mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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34710 lines
1.4 MiB
34710 lines
1.4 MiB
=== release 0.10.25 ===
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2009-10-05 Jan Schmidt <jan.schmidt@sun.com>
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* configure.ac:
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releasing 0.10.25, "Standard disclaimers apply"
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2009-10-05 13:49:10 +0100 Jan Schmidt <thaytan@noraisin.net>
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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/en_GB.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/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/ru.po:
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* po/sk.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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Update .po files
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2009-10-01 17:17:55 +0100 Jan Schmidt <thaytan@noraisin.net>
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* ChangeLog:
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* configure.ac:
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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/en_GB.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/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/ru.po:
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* po/sk.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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0.10.24.4 pre-release
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2009-10-01 10:37:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* ext/pango/gsttextoverlay.c:
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* ext/pango/gsttextrender.c:
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pango: Unpremultiply Cairo's ARGB to match GStreamers ARGB
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2009-09-28 22:06:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/playback/gstplaysink.c:
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playsink: make the lock recursive for now
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Fixes #583255
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2009-09-28 21:54:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst/playback/gstplaysink.c:
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playsink: fix the vis property getter
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2009-09-30 18:06:56 +0100 Christian F.K. Schaller <christian.schaller@collabora.co.uk>
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* gst-plugins-base.spec.in:
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Add missing file to spec file
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2009-09-17 16:57:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* gst-libs/gst/cdda/gstcddabasesrc.c:
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* tests/check/libs/cddabasesrc.c:
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cddabasesrc: Fix string leaks in the unit test and a leak in cddabasesrc
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2009-09-17 23:42:52 +1000 Jonathan Matthew <jonathan@d14n.org>
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* gst-libs/gst/cdda/gstcddabasesrc.c:
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* tests/check/libs/cddabasesrc.c:
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cddabasesrc: ignore URI fragments that look like device paths
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Rhythmbox uses cdda:// URIs of the form cdda://track#device, which
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worked before the fix for bug #321532.
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Also adds a check for negative track numbers and some unit tests for URI
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parsing.
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Fixes bug #595454.
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2009-09-17 01:20:45 +0100 Jan Schmidt <thaytan@noraisin.net>
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* configure.ac:
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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/en_GB.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/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/ru.po:
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* po/sk.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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0.10.24.3 pre-release
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2009-09-15 15:23:49 -0700 Michael Smith <msmith@songbirdnest.com>
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* gst-libs/gst/tag/gstvorbistag.c:
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vorbistag: don't ever return NULL in list of strings.
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2009-09-14 12:18:33 +0200 Edward Hervey <bilboed@bilboed.com>
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* gst/playback/gstplaysink.c:
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playsink: Expose mute,volume,vis-plugin and font-desc properties
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https://bugzilla.gnome.org/show_bug.cgi?id=594623
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2009-09-09 12:42:04 +0200 Edward Hervey <bilboed@bilboed.com>
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* gst/playback/gstplaysink.c:
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GstPlaySink: Expose 'reconfigure' as an action signal.
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2009-09-09 11:17:28 +0200 Edward Hervey <bilboed@bilboed.com>
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* gst/playback/gstplaysink.c:
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GstPlaySink: Expose flags as a gobject property.
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2009-09-08 11:35:20 +0200 Edward Hervey <bilboed@bilboed.com>
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* gst/playback/gstplayback.c:
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* gst/playback/gstplaysink.c:
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* gst/playback/gstplaysink.h:
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playback: Register playsink as an element.
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This allows using playsink from outside the playback plugin.
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Add code to be able to request the sink pads using standard GStreamer API.
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TODO : expose GObject properties/signals.
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2009-09-12 14:55:06 +0300 Stefan Kost <ensonic@users.sf.net>
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* docs/libs/gst-plugins-base-libs.types:
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docs: add new gst_stream_volume_get_type to types file
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This is needs to get Gobject features to show up in the docs.
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2009-09-12 15:48:11 -0700 David Schleef <ds@schleef.org>
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* ext/ogg/gstoggdemux.c:
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oggdemux: Fix duration calculation for truncated files
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If the last page of a stream has a granulepos of -1, that is,
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it doesn't complete a packet, we need to continue to search
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for the last granulepos.
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2009-09-12 14:01:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* Makefile.am:
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* gst-libs/gst/app/Makefile.am:
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* gst-libs/gst/audio/Makefile.am:
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* gst-libs/gst/cdda/Makefile.am:
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* gst-libs/gst/fft/Makefile.am:
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* gst-libs/gst/interfaces/Makefile.am:
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* gst-libs/gst/netbuffer/Makefile.am:
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* gst-libs/gst/pbutils/Makefile.am:
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* gst-libs/gst/riff/Makefile.am:
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* gst-libs/gst/rtp/Makefile.am:
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* gst-libs/gst/rtsp/Makefile.am:
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* gst-libs/gst/sdp/Makefile.am:
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* gst-libs/gst/tag/Makefile.am:
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* gst-libs/gst/video/Makefile.am:
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introspection: Build pkgconfig before all libraries and set PKG_CONFIG_PATH
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This way g-ir-scanner can find the gstreamer-*-0.10 pkg-config files.
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2009-09-12 02:23:07 +0100 Jan Schmidt <thaytan@noraisin.net>
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* ext/theora/theoraenc.c:
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theoraenc: Fix a string leak in _getcaps()
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2009-09-11 23:49:11 +0100 Jan Schmidt <thaytan@noraisin.net>
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* ChangeLog:
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* configure.ac:
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* po/LINGUAS:
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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/en_GB.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/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/ru.po:
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* po/sk.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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0.10.24.2 pre-release
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2009-09-11 21:44:18 +0100 Jan Schmidt <thaytan@noraisin.net>
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* tests/check/elements/audioresample.c:
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check: Improve audioresample test
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Make the audioresample test work with CK_FORK=no, and
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turn a g_print into a GST_INFO.
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2009-09-11 22:09:06 +0200 Benjamin Otte <otte@gnome.org>
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* gst/videotestsrc/videotestsrc.c:
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videotestsrc: Fix crashes with even widths
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The fix for green lines introduced by commit
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35fdfcc6258c66ba462a4330a35deffb0f2b501d caused invalid memory accesses
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for even widths. This patch fixes it.
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2009-09-11 15:11:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* gst/playback/gstplaybin2.c:
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playbin2: Implement GstStreamVolume interface
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2009-09-11 15:04:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* gst/volume/gstvolume.c:
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* gst/volume/gstvolume.h:
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* tests/check/Makefile.am:
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* tests/check/elements/volume.c:
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volume: Implement GstStreamVolume interface
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2009-09-11 14:54:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* docs/libs/gst-plugins-base-libs-docs.sgml:
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* docs/libs/gst-plugins-base-libs-sections.txt:
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* gst-libs/gst/interfaces/Makefile.am:
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* gst-libs/gst/interfaces/streamvolume.c:
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* gst-libs/gst/interfaces/streamvolume.h:
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* gst/playback/Makefile.am:
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* win32/common/libgstinterfaces.def:
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interfaces: API: Add GstStreamVolume interface
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Fixes bug #567660.
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2009-09-11 12:20:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst-libs/gst/rtsp/gstrtspconnection.c:
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rtsp: properly fix the HTTP manual mode
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When we're not parsing HTTP, return EPARSE when we get an HTTP
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message.
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2009-09-11 10:16:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
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* gst-libs/gst/interfaces/mixertrack.h:
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mixertrack: add READONLY and WRITEONLY flags
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Should really have been READABLE and WRITABLE, but those are hard to
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add whilst maintaining backwards compatibility. See #343615.
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API: GST_MIXER_TRACK_READONLY
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API: GST_MIXER_TRACK_WRITEONLY
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2009-09-11 10:02:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
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* gst-libs/gst/audio/gstringbuffer.c:
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ringbuffer: fix build against core that has debugging disabled
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The macro is called GST_DISABLE_GST_DEBUG, not GST_DISABLE_DEBUG.
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2009-09-11 07:38:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* gst/videorate/gstvideorate.c:
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videorate: Add Since marker for the new skip-to-first property
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2009-09-11 07:36:10 +0200 Olivier Crête <olivier.crete@collabora.co.uk>
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* gst/videorate/gstvideorate.c:
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* gst/videorate/gstvideorate.h:
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videorate: Make videorate work with a live source
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Add a property that makes videorate skip to the first buffer it
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receives instead of padding the stream from segment start to the
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first real buffer.
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Fixes bug #567928.
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2009-09-11 07:20:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* gst-libs/gst/fft/gstfft.h:
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* gst-libs/gst/fft/gstfftf32.h:
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* gst-libs/gst/fft/gstfftf64.h:
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* gst-libs/gst/fft/gstffts16.h:
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* gst-libs/gst/fft/gstffts32.h:
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fft: Mark one function as const and add notes that the structs should be private in 0.11
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2009-09-10 22:28:19 +0300 Stefan Kost <ensonic@users.sf.net>
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* gst-libs/gst/audio/gstringbuffer.c:
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ringbuffer: add human readable format names when logging
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Add string array with human readable names for format and type to be used in log
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statements.
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2009-09-10 18:19:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst-libs/gst/rtp/gstbasertppayload.c:
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basertppay: don't print RTP timestamps as clocktime
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Don't try to print the RTP timestamp as a GstClockTime, it's just a guint32.
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Fixes #594757
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2009-09-10 16:55:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* gst/playback/gstplaybin.c:
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* gst/playback/gstplaybin2.c:
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playbin(2): Document that the volume property uses a linear scale
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Fixes bug #571610.
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2009-09-10 14:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst-libs/gst/rtsp/gstrtspconnection.c:
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rtsp: don't return EPARSE
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Don't blindly return EPARSE when http mode is disabled.
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Restore old http mode after temporarily setting it to TRUE.
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2009-09-10 12:38:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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baseaudiosink: add ugly backward compat hack
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Check for pulsesink < 0.10.17 because it includes code that is now included in
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baseaudiosink. Disable that code in baseaudiosink to be compatible with the
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older version.
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2009-09-10 10:56:29 +0200 Benjamin Otte <otte@gnome.org>
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* gst/ffmpegcolorspace/imgconvert.c:
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ffmpegcolorspace: Handle YVU9/YUV9 conversion with odd widths
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A green border could be visible when converting to Y444 or RGB, because
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the last chroma samples weren't copied correctly
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2009-09-10 10:43:37 +0200 Benjamin Otte <otte@gnome.org>
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* gst/videotestsrc/videotestsrc.c:
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videotestsrc: Fix YVU9 and YUV9
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- Buffer sizes were computed different from ffmpegcolorspace
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- Green bar on right size for widths not divisable by 4
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2009-09-10 10:08:28 +0200 Benjamin Otte <otte@gnome.org>
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* gst/videotestsrc/videotestsrc.c:
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videotestsrc: Fix image for odd widths in some formats
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videotestsrc rounds chroma down. This causes it to omit the last chroma
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value completely for odd widths when the chroma is downsampled.
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This patch special cases the last pixel to not be rounded down.
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2009-09-10 10:02:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* ext/ogg/gstoggdemux.c:
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oggdemux: Handle kate and cmml as sparse streams too
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2009-09-10 10:00:16 +0200 Tim-Philipp Müller <tim.muller@collabora.co.uk>
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* ext/ogg/gstoggdemux.c:
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* ext/ogg/gstoggdemux.h:
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oggdemux: Better handling of sparse streams by sending segment updates
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Fixes bug #397419.
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2009-09-10 09:43:28 +0300 Stefan Kost <ensonic@users.sf.net>
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* gst/playback/gsturidecodebin.c:
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docs: tell a biit more about uri-decodebin and buffering
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2009-09-09 18:24:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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baseaudiosink: take clock time in setcaps
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Take the time of the clock so that the last_time field is set. This is important
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for sinks that restart their internal ringbuffer after a caps change and need to
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know the last know position.
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2009-09-09 18:24:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
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* gst-libs/gst/audio/gstaudioclock.c:
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audioclock: add some more debug
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2009-09-09 16:44:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
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* ext/theora/theoraenc.c:
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theoraenc: Print a debug message with supported formats
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2009-09-07 17:29:38 +0200 Benjamin Otte <otte@gnome.org>
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* ext/theora/theoraenc.c:
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theora: Check supported input formats in getcaps function
|
|
We want to fail early when an older libtheora release is used that does
|
|
not support Y444 or Y42B formats, so use a getcaps function that does
|
|
this.
|
|
|
|
2009-09-04 21:37:04 +0200 Benjamin Otte <otte@gnome.org>
|
|
|
|
* ext/theora/theoraenc.c:
|
|
theora: Implement support in theoraenc for Y444 and Y42B
|
|
Fixes bug #594165.
|
|
|
|
2009-09-04 20:23:52 +0200 Benjamin Otte <otte@gnome.org>
|
|
|
|
* ext/theora/theoraenc.c:
|
|
theora: Refactor the buffer copy code
|
|
|
|
2009-09-04 16:59:49 +0200 Benjamin Otte <otte@gnome.org>
|
|
|
|
* ext/theora/theoraenc.c:
|
|
theora: Split yuv_buffer creation into its own function
|
|
|
|
2009-09-04 16:49:08 +0200 Benjamin Otte <otte@gnome.org>
|
|
|
|
* ext/theora/theoraenc.c:
|
|
theora: Split out buffer resize in its own function
|
|
|
|
2009-09-04 14:06:09 +0200 Benjamin Otte <otte@gnome.org>
|
|
|
|
* ext/theora/theoraenc.c:
|
|
theora: Add assertions that functions don't fail
|
|
Some functions in libtheora can return an error, but that error cannot
|
|
ever happen inside theoraenc. In those cases assert that it doesn't.
|
|
|
|
2009-09-09 16:21:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/examples/seek/seek.c:
|
|
seek: make stop state configurable
|
|
Make it easy to experiment with different stop states (NULL and READY)
|
|
|
|
2009-09-09 16:19:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
baseaudiosink: correct for clock reset
|
|
When going to NULL, we reset the ringbuffer so that it starts beck from 0. We
|
|
also make sure that the clock is updated with the elapsed time so that it
|
|
alsways increments even when the ringbuffer goes back to 0. When this happened
|
|
we need to adjust the sample position for the reset ringbuffer.
|
|
Fixes #594136
|
|
|
|
2009-09-09 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosink.h:
|
|
baseaudiosink: whitespace fixes
|
|
|
|
2009-09-09 16:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstringbuffer.c:
|
|
ringbuffer: add more debug
|
|
|
|
2009-09-09 10:25:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/interfaces/colorbalance.h:
|
|
* gst-libs/gst/interfaces/mixer.h:
|
|
whitespace fixes
|
|
|
|
2009-09-08 17:59:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/video/gstvideosink.c:
|
|
* gst-libs/gst/video/gstvideosink.h:
|
|
videosink: add "show-preroll-frame" property
|
|
Add a property to disable rendering of video frames during preroll. This
|
|
will only work for videosinks that use the new ::show_frame() vfunc instead
|
|
of overriding basesink's preroll and render vfuncs directly.
|
|
API: GstVideoSink:show-preroll-frame
|
|
|
|
2009-09-08 17:43:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* sys/ximage/ximagesink.c:
|
|
* sys/xvimage/xvimagesink.c:
|
|
ximagesink, xvimagesink: use new GstVideoSink::show_frame() vfunc
|
|
|
|
2009-09-08 18:19:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/video/gstvideosink.c:
|
|
* gst-libs/gst/video/gstvideosink.h:
|
|
video: add GstVideoSinkClass::show_frame()
|
|
Add ::show_frame() vfunc which maps to basesink's ::preroll and ::render
|
|
vfuncs and add some gtk-doc chunks.
|
|
API: GstVideoSinkClass::show_frame()
|
|
|
|
2009-09-08 16:00:47 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/interfaces/navigation.c:
|
|
navigation: don't do stuff inside g_return_val_if_fail() statements
|
|
Or it will all fall apart if someone compiles with -DG_DISABLE_ASSERT.
|
|
|
|
2009-08-31 20:24:22 +0200 Havard Graff <havard.graff@tandberg.com>
|
|
|
|
* gst-libs/gst/interfaces/navigation.c:
|
|
navigation: Fix compiler warning with MSVC
|
|
Fixes bug #594275.
|
|
|
|
2009-08-31 20:31:56 +0200 Havard Graff <havard.graff@tandberg.com>
|
|
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
basertpdepayload: fix event forwarding
|
|
|
|
2009-08-31 20:36:37 +0200 Havard Graff <havard.graff@tandberg.com>
|
|
|
|
* gst-libs/gst/rtp/gstrtcpbuffer.c:
|
|
rtcpbuffer: add missing break in handling of GST_RTCP_TYPE_PSFB
|
|
Fixes #594258
|
|
|
|
2009-09-08 13:02:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
* gst/playback/gstplaysink.c:
|
|
* gst/playback/gstplaysink.h:
|
|
fix whitespace
|
|
|
|
2009-09-08 12:59:20 +0200 Håvard Graff <havard.graff@tandberg.com>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
baseaudiosrc: improve slave skew resync
|
|
The old one did the mistake of not actually advancing the ringbuffer, it just
|
|
adjusted the segbase, introducing the whole lenght of the ringbuffer as an
|
|
extra delay in the pipeline.
|
|
Also make sure that the resync can never go back in time, producing the same
|
|
timestamps that has already been produced, as this can cause severe problems
|
|
for sinks and other synching mechanisms.
|
|
Fixes #594256
|
|
|
|
2009-09-07 17:13:12 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefinding: disable typefinder for headerless flac
|
|
Disable headerless flac typefinder as long as it happily typefinds anything
|
|
including /dev/urandom as flac and as long as it's not particularly useful
|
|
given that such streams don't really exist in the wild.
|
|
Also fix up some comments so that gtk-doc doesn't complain about them.
|
|
|
|
2009-09-06 15:21:43 +0300 René Stadler <mail@renestadler.de>
|
|
|
|
* sys/ximage/ximagesink.c:
|
|
ximagesink: fix small memory leak when setting window title
|
|
|
|
2009-09-06 01:42:42 +0300 René Stadler <mail@renestadler.de>
|
|
|
|
* sys/xvimage/xvimagesink.c:
|
|
xvimagesink: fix small memory leak when setting window title
|
|
|
|
2009-09-05 13:55:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* .gitignore:
|
|
introspection: Add *.gir and *.typelib to .gitignore
|
|
|
|
2009-09-05 13:46:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/app/Makefile.am:
|
|
* gst-libs/gst/audio/Makefile.am:
|
|
* gst-libs/gst/interfaces/Makefile.am:
|
|
* gst-libs/gst/pbutils/Makefile.am:
|
|
* gst-libs/gst/rtsp/Makefile.am:
|
|
* gst-libs/gst/video/Makefile.am:
|
|
introduction: Fix out-of-tree build
|
|
|
|
2009-09-05 13:13:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/Makefile.am:
|
|
rtsp: Fix introspection build by ordering sources/headers in dependency order
|
|
|
|
2009-09-05 13:09:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/Makefile.am:
|
|
audio: Remove debug echo
|
|
|
|
2009-09-05 13:08:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/Makefile.am:
|
|
audio: Fix build of introspection data by using dependency order for the headers/sources
|
|
|
|
2009-09-05 12:31:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/app/Makefile.am:
|
|
* gst-libs/gst/audio/Makefile.am:
|
|
* gst-libs/gst/cdda/Makefile.am:
|
|
* gst-libs/gst/fft/Makefile.am:
|
|
* gst-libs/gst/interfaces/Makefile.am:
|
|
* gst-libs/gst/netbuffer/Makefile.am:
|
|
* gst-libs/gst/pbutils/Makefile.am:
|
|
* gst-libs/gst/riff/Makefile.am:
|
|
* gst-libs/gst/rtp/Makefile.am:
|
|
* gst-libs/gst/rtsp/Makefile.am:
|
|
* gst-libs/gst/sdp/Makefile.am:
|
|
* gst-libs/gst/tag/Makefile.am:
|
|
* gst-libs/gst/video/Makefile.am:
|
|
introspection: Strip Gst prefix from all types/functions
|
|
|
|
2009-09-05 11:49:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/Makefile.am:
|
|
* gst-libs/gst/app/Makefile.am:
|
|
* gst-libs/gst/audio/Makefile.am:
|
|
* gst-libs/gst/fft/Makefile.am:
|
|
* gst-libs/gst/interfaces/Makefile.am:
|
|
* gst-libs/gst/netbuffer/Makefile.am:
|
|
* gst-libs/gst/pbutils/Makefile.am:
|
|
* gst-libs/gst/riff/Makefile.am:
|
|
* gst-libs/gst/rtp/Makefile.am:
|
|
* gst-libs/gst/rtsp/Makefile.am:
|
|
* gst-libs/gst/sdp/Makefile.am:
|
|
* gst-libs/gst/tag/Makefile.am:
|
|
* gst-libs/gst/video/Makefile.am:
|
|
introspection: Fix build if gir-repository is not installed
|
|
|
|
2009-09-05 11:37:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/video/Makefile.am:
|
|
video: Add gobject-introspection support
|
|
|
|
2009-09-05 11:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/tag/Makefile.am:
|
|
tag: Add gobject-introspection support
|
|
|
|
2009-09-05 11:34:11 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/sdp/Makefile.am:
|
|
sdp: Add gobject-introspection support
|
|
|
|
2009-09-05 11:31:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/app/Makefile.am:
|
|
* gst-libs/gst/audio/Makefile.am:
|
|
* gst-libs/gst/interfaces/Makefile.am:
|
|
* gst-libs/gst/pbutils/Makefile.am:
|
|
libs: Add nodist headers and sources to the introspection files
|
|
|
|
2009-09-05 11:28:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/Makefile.am:
|
|
rtsp: Add gobject-introspection support
|
|
|
|
2009-09-05 11:25:42 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/Makefile.am:
|
|
rtp: Add gobject-introspection support
|
|
|
|
2009-09-05 11:23:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/riff/Makefile.am:
|
|
riff: Add gobject-introspection support
|
|
|
|
2009-09-05 11:20:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/pbutils/Makefile.am:
|
|
pbutils: Add gobject-introspection support
|
|
|
|
2009-09-05 11:17:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/netbuffer/Makefile.am:
|
|
netbuffer: Add gobject-introspection support
|
|
|
|
2009-09-05 11:15:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/interfaces/Makefile.am:
|
|
interfaces: Add gobject-introspection support
|
|
|
|
2009-09-05 11:04:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/fft/Makefile.am:
|
|
fft: Add gobject-introspection support
|
|
|
|
2009-09-05 11:01:44 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/cdda/Makefile.am:
|
|
cdda: Add gobject-introspection support
|
|
This is disabled for now until gobject-introspection is fixed
|
|
|
|
2009-09-05 10:50:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/Makefile.am:
|
|
audio: Add gobject-introspection support
|
|
|
|
2009-09-05 10:40:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* configure.ac:
|
|
* gst-libs/gst/app/Makefile.am:
|
|
app: Add gobject-introspection support
|
|
|
|
2009-09-05 10:20:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* common:
|
|
Automatic update of common submodule
|
|
From 00a859e to 19fa4f3
|
|
|
|
2009-09-04 15:48:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefind: fix midi typefinding
|
|
We already have a audio/midi typefinder so don't override it with the midi in
|
|
RIFF typefinder or else we fail to detect plain midi files.
|
|
|
|
2009-09-04 11:29:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gsturidecodebin.c:
|
|
uridecodebin: do buffering for more uris
|
|
Add ssh://, ftp://, sftp://, myth:// to the list of uris that require
|
|
buffering.
|
|
Fixes #594020
|
|
|
|
2009-09-04 07:36:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefindfunctions: Add typefinder for Midi inside RIFF
|
|
This is a standard Midi file format that should be supported by
|
|
all Midi decoders and also has the mimetype audio/mid according to
|
|
the Midi specification homepage.
|
|
Fixes bug #594094.
|
|
|
|
2009-09-03 18:53:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
audiortppay: add some debugging
|
|
|
|
2009-09-03 17:53:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
audiortppay: handle gaps
|
|
Add various conversion functions between time<->bytes<->rtptime that will be
|
|
used later on.
|
|
Refactor the min/max packet length code so that it can be used for both
|
|
sample/frame based payloaders. Cache the returned values.
|
|
code cleanups.
|
|
When we discover a DISCONT buffer, make the outgoing RTP timestamps have the
|
|
same gap as the GStreamer timestamps gap.
|
|
|
|
2009-09-03 14:13:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
audiortppay: fix frame duration calculations
|
|
Fix the calculation of the frame duration and rtp timestamps.
|
|
Add some debugging
|
|
|
|
2009-09-03 14:13:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstbasertppayload.c:
|
|
rtppay: add some debugging
|
|
|
|
2009-09-02 19:49:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
audiortppay: use offsets for RTP timestamps
|
|
Have a custom sample/frame function to generate an offset that the base class
|
|
will use for generating RTP timestamps. This results in perfect RTP timestamps
|
|
on the output buffers.
|
|
Refactor setting metadata on output buffers.
|
|
Add some more functionality to _flush().
|
|
Handle DISCONT on the input buffers and set the marker bit and DISCONT flag on
|
|
the next outgoing buffer.
|
|
Flush the pending data on EOS.
|
|
|
|
2009-09-02 13:13:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
audiortppay: move function around
|
|
|
|
2009-09-02 13:12:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
audiortppay: fix sample duration calculation
|
|
|
|
2009-09-02 12:24:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
audiortppay: more refactoring
|
|
Unify the sample/frame buffer handling code by making the functions plugable.
|
|
|
|
2009-09-02 12:03:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
|
|
audiortppayload: refactor some more
|
|
Refactor getting the packet min/max size and alignment code.
|
|
Refactor converting bytes to time.
|
|
change some variable to something shorter.
|
|
|
|
2009-09-02 10:46:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
|
|
* win32/common/libgstrtp.def:
|
|
audiortppayload: refactor and cleanup
|
|
Always use the adapter when we need to fragment the incomming buffer. Use more
|
|
modern adapter functions to avoid malloc and memcpy. The overall result is that
|
|
the code looks cleaner while it should be equally fast and in some case avoid a
|
|
memcpy and malloc.
|
|
Use the adapter timestamping functions for more precise timestamps in case of
|
|
weird disconts.
|
|
Cache some values instead of recalculating them.
|
|
Add gst_base_rtp_audio_payload_flush() to flush a certain amount of bytes from
|
|
the internal adapter.
|
|
API: GstBaseRTPAudioPayload::gst_base_rtp_audio_payload_flush()
|
|
|
|
2009-09-03 16:56:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* common:
|
|
Update common
|
|
|
|
2009-09-03 11:29:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstbasertppayload.c:
|
|
basertppay: add property to disable perfect RTP time
|
|
Add a property to disable the generation of perfect RTP timestamps. By default
|
|
it is active.
|
|
API: GstBaseRTPPayload::perfect-rtptime
|
|
|
|
2009-09-02 19:47:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstbasertppayload.c:
|
|
basertppay: allow subclasses to influence RTP time
|
|
Allow subclasses to use the OFFSET field on RTP buffers to influence the way in
|
|
which RTP timestamps are generated. Usually timestamps are created from the
|
|
GStreamer timestamps on the buffer, which could result in imperfect RTP
|
|
timestamps.
|
|
|
|
2009-09-02 19:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstbasertppayload.h:
|
|
basertppay: add macro to cast
|
|
|
|
2009-09-01 18:26:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
audiopayload: code cleanups
|
|
|
|
2009-09-01 18:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
audiortppayload: don't check adapter
|
|
the adapter is never NULL so we don't need to check it.
|
|
Use _scale functions to avoid overflows.
|
|
|
|
2009-09-03 00:14:02 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* configure.ac:
|
|
* gst/typefind/Makefile.am:
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefinding: move gio-based xdg mime typefinder from -bad to -base
|
|
Its purposes is mainly to avoid false positives (e.g. mp3 typefinder
|
|
reporting a 20% probability and somesuch). Won't be registered if
|
|
the gio plugin has been disabled via ./configure --disable-gio.
|
|
|
|
2009-09-01 15:06:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/subparse/gstsubparse.c:
|
|
subparse: GstAdapter is not a GstObject and should be freed with g_object_unref
|
|
|
|
2009-09-01 15:02:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* sys/v4l/v4lsrc_calls.c:
|
|
v4lsrc: fix timestamping for when we do not have a clock yet
|
|
Should fix #559049.
|
|
|
|
2009-09-01 14:30:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* sys/v4l/v4lsrc_calls.c:
|
|
v4lsrc: don't log not-yet-initialised integer value
|
|
|
|
2009-09-01 14:28:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* sys/v4l/v4lsrc_calls.c:
|
|
v4lsrc: avoid unnecessary run-time type checks in custom buffer finalize
|
|
And reflow code to be more indent friendly.
|
|
|
|
2009-09-01 10:39:52 +0200 Jonas Holmberg <jonas.holmberg@axis.com>
|
|
|
|
* gst-libs/gst/rtp/gstbasertppayload.c:
|
|
* gst-libs/gst/rtp/gstbasertppayload.h:
|
|
basertppayload: Make instance init faster by not reading /dev/urandom 3 times
|
|
... which is the default seed when creating a new GRand. Because
|
|
GLib in older versions used buffered IO this would take a lot of time.
|
|
Instead use the global GRand for getting random numbers and keep the
|
|
three instance GRand for backward compatibility with a simple seed.
|
|
Fixes bug #593284.
|
|
|
|
2009-08-31 22:48:01 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/adder/gstadder.c:
|
|
adder: improve caps filter functionality. Fixes #590146.
|
|
Also use the capsfilter if there is no src-peer as the caps constrain what
|
|
we can do. Don't create any_caps as a default, as we check for NULL to skip the
|
|
filtering. This is a (small) performance regression as we always intersect
|
|
otherwise.
|
|
|
|
2009-08-31 11:10:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
decodebin2: Post missing plugin messages before any error messages
|
|
|
|
2009-08-28 19:06:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/cdda/gstcddabasesrc.c:
|
|
cddabasesrc: safely handle the indexes
|
|
|
|
2009-08-28 19:06:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* win32/common/libgstrtsp.def:
|
|
def: add new rtsp symbols
|
|
|
|
2009-08-28 14:08:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstbasertppayload.h:
|
|
basertppayload: whitespace fixes.
|
|
|
|
2009-08-27 18:59:49 +0200 Marc-André Lureau <mlureau@flumotion.com>
|
|
|
|
* gst/gdp/gstgdppay.c:
|
|
Bug 593035 - set IN_CAPS for streamheader buffer
|
|
|
|
2009-08-26 16:56:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/playback/gstinputselector.c:
|
|
* gst/playback/gststreamselector.c:
|
|
playbin: The internally linked pad of the selector might be NULL in some cases
|
|
|
|
2009-08-26 16:45:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/playback/gstinputselector.c:
|
|
* gst/playback/gststreamselector.c:
|
|
playbin: Fix iterate internal linked pads functions for the stream selectors
|
|
This now used the new gst_iterator_new_single() function and as a side effect
|
|
fixes bug #592864.
|
|
|
|
2009-08-26 09:08:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/riff/riff-ids.h:
|
|
* gst-libs/gst/riff/riff-read.c:
|
|
riff: Add support for AVF files
|
|
AVF is valid RIFF but has AVF0 has first fourcc instead of RIFF.
|
|
Fixes bug #593117.
|
|
|
|
2009-08-26 09:08:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefindfunctions: Detect AVF files as RIFF files too
|
|
AVF is valid RIFF but has AVF0 as first fourcc instead of RIFF.
|
|
Partially fixes bug #593117.
|
|
|
|
2009-08-21 11:51:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* tests/check/elements/audioresample.c:
|
|
audioresample: Add unit test for checking for timestamp drifts
|
|
This also checks for perfect timestamping and offsetting.
|
|
|
|
2009-08-21 10:11:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/audioresample/gstaudioresample.c:
|
|
audioresample: Fix drain processing
|
|
In case we have to convert internally don't process output length input samples
|
|
but history length input samples.
|
|
|
|
2009-08-21 10:02:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* tests/check/elements/audioresample.c:
|
|
audioresample: Improve debugging a bit in the unit test
|
|
|
|
2009-08-21 10:00:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/audioresample/gstaudioresample.c:
|
|
audioresample: On the first buffer we need discont handling
|
|
Otherwise we won't get upstream timestamps and everything and all
|
|
output buffers would have -1 timestamps.
|
|
|
|
2009-08-21 08:23:39 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
|
|
|
|
* configure.ac:
|
|
* gst/subparse/gstsubparse.c:
|
|
subparse: Remove dependency on regex.h as it's not used anyway
|
|
Fixes bug #592544.
|
|
|
|
2009-08-21 06:58:31 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
|
|
|
|
* gst/audioresample/gstaudioresample.c:
|
|
audioresample: Fix buffer overflow when pushing the drain
|
|
|
|
2009-08-21 06:57:58 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
|
|
|
|
* gst/audioresample/gstaudioresample.c:
|
|
* gst/audioresample/gstaudioresample.h:
|
|
audioresample: Fix timestamp drift
|
|
Fixes bug #591934.
|
|
|
|
2009-08-24 11:34:35 -0700 David Schleef <ds@schleef.org>
|
|
|
|
* ext/gnomevfs/gstgnomevfssrc.c:
|
|
* ext/ogg/gstogmparse.c:
|
|
* ext/pango/gsttextrender.c:
|
|
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
|
|
* gst/playback/gstinputselector.c:
|
|
* gst/playback/gststreamselector.c:
|
|
* gst/subparse/gstsubparse.c:
|
|
* sys/v4l/gstv4lmjpegsink.c:
|
|
* sys/v4l/gstv4lmjpegsrc.c:
|
|
* sys/v4l/gstv4lsrc.c:
|
|
Remove Ronald Bultje from Authors field
|
|
Replaced with "GStreamer maintainers
|
|
<gstreamer-devel@lists.sourceforge.net>" or just removed,
|
|
depending on the number of other authors.
|
|
|
|
2009-08-24 15:06:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
playbin2: fix refcounting of _get_sink()
|
|
g_value_set_object() increases the refcount of the sink, which is not needed
|
|
because the object should already be refcounted. Make sure this is always the
|
|
case and use g_value_take_object().
|
|
Fixes: #592884
|
|
|
|
2009-08-24 14:39:16 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspdefs.c:
|
|
rtsp: Mark Transport as supporting multiple values.
|
|
|
|
2009-08-24 13:58:17 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
* gst-libs/gst/rtsp/gstrtspdefs.h:
|
|
* gst-libs/gst/rtsp/gstrtspmessage.h:
|
|
rtsp: Added missing Since tags.
|
|
|
|
2009-08-24 13:27:55 +0200 Eero Nurkkala <ext-eero.nurkkala at nokia.com>
|
|
|
|
* gst-libs/gst/audio/gstringbuffer.c:
|
|
ringbuffer: Improve audiosink startup performance
|
|
When we start the ringbuffer, immediatly continue processing samples if the
|
|
writer prepared some for us.
|
|
Fixes #545807
|
|
|
|
2009-08-17 11:53:43 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
rtsp: Added new API for sending using GstRTSPWatch.
|
|
The new API to send messages using GstRTSPWatch will first try to send the
|
|
message immediately. Then, if that failed (or the message was not sent
|
|
fully), it will queue the remaining message for later delivery. This avoids
|
|
unnecessary context switches, and makes it possible to keep track of
|
|
whether the connection is blocked (the unblocking of the connection is
|
|
indicated by the reception of the message_sent signal).
|
|
This also deprecates the old API (gst_rtsp_watch_queue_data() and
|
|
gst_rtsp_watch_queue_message().)
|
|
API: gst_rtsp_watch_write_data()
|
|
API: gst_rtsp_watch_send_message()
|
|
|
|
2009-08-17 11:46:32 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Made gst_rtsp_watch_queue_data() thread safe.
|
|
|
|
2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
rtsp: Added gst_rtsp_connection_set_http_mode().
|
|
With gst_rtsp_connection_set_http_mode() it is possible to tell the
|
|
connection whether to allow HTTP messages to be supported. By enabling HTTP
|
|
support the automatic HTTP tunnel support will also be disabled.
|
|
API: gst_rtsp_connection_set_http_mode()
|
|
|
|
2009-06-16 19:35:23 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Allow gst_rtsp_connection_do_tunnel() to just setup decoding context.
|
|
If the second connection passed to gst_rtsp_connection_do_tunnel() is NULL
|
|
then just setup the base64 decoding context for the first connection.
|
|
|
|
2009-06-16 19:04:54 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Write as much as possible in gst_rtsp_source_dispatch().
|
|
Try to write as much as possible if there are multiple messages queued.
|
|
|
|
2009-06-16 18:38:02 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
rtsp: Add error_full callback to GstRTSPWatchFuncs.
|
|
The error_full callback is similar to the error callback, but allows for
|
|
better error handling. For read errors a partial message is provided to
|
|
help an RTSP server generate a more correct error response, and for write
|
|
errors the write queue id of the failed message is returned.
|
|
|
|
2009-08-17 18:29:17 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Made read_line() support LWS.
|
|
Rewrote read_line() to support LWS (Line White Space), the method used by
|
|
RTSP (and HTTP) to break long lines. Also added support for \r and \n as
|
|
line endings (in addition to the official \r\n).
|
|
|
|
2009-08-20 14:12:50 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspdefs.c:
|
|
* gst-libs/gst/rtsp/gstrtspdefs.h:
|
|
rtsp: Do not split headers which should not be split.
|
|
From RFC 2068 section 4.2: "Multiple message-header fields with the same
|
|
field-name may be present in a message if and only if the entire
|
|
field-value for that header field is defined as a comma-separated list
|
|
[i.e., #(values)]." This means that we should not split other headers which
|
|
may contain a comma, e.g., Range and Date.
|
|
|
|
2009-08-20 14:12:09 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Parse WWW-Authenticate headers correctly.
|
|
Due to the odd syntax for WWW-Authenticate (and Proxy-Authenticate) which
|
|
allows commas both to separate between multiple challenges, and within the
|
|
challenges themself, we need to take some extra care to split these headers
|
|
correctly.
|
|
|
|
2009-06-17 21:46:27 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Improve parse_line().
|
|
Make parse_line() handle keys with multiple values on one line correctly.
|
|
|
|
2009-06-17 23:15:23 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Rewrote setup_tunneling().
|
|
Rewrote setup_tunneling() to use normal GstRTSPMessages instead of hard
|
|
coded strings and duplicates of the message parsing code.
|
|
|
|
2009-08-24 10:20:16 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspdefs.c:
|
|
* gst-libs/gst/rtsp/gstrtspdefs.h:
|
|
rtsp: Rewrote gen_tunnel_reply().
|
|
Rewrote gen_tunnel_reply() to generate a normal GstRTSPMessage rather
|
|
than a hard coded string.
|
|
|
|
2009-08-24 10:19:35 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Ignore the Content-Length for POST requests.
|
|
The Content-Length for POST requests with an x-sessioncookie header should
|
|
be ignored as the length is bogus and only there to fool proxies.
|
|
|
|
2009-06-17 20:52:48 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Normalize lines (remove extra whitespace) before parsing.
|
|
|
|
2009-06-10 13:11:31 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Made parse_string() return a result.
|
|
This will catch parsing errors when a too long string is received.
|
|
|
|
2009-06-10 11:43:31 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Improved parsing of messages.
|
|
Do not abort message parsing as soon as there is an error. Instead parse
|
|
as much as possible to allow a server to return as meaningful an error as
|
|
possible.
|
|
|
|
2009-06-09 17:54:20 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspdefs.c:
|
|
* gst-libs/gst/rtsp/gstrtspdefs.h:
|
|
* gst-libs/gst/rtsp/gstrtspmessage.c:
|
|
* gst-libs/gst/rtsp/gstrtspmessage.h:
|
|
rtsp: Added support for HTTP messages
|
|
|
|
2009-06-09 16:22:17 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
rtsp: Added gst_rtsp_connection_create_from_fd().
|
|
API: gst_rtsp_connection_create_from_fd()
|
|
|
|
2009-06-09 15:27:17 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Add initial buffer support.
|
|
The initial buffer contains data for a connection which should be used
|
|
before starting to actually read anything from the socket.
|
|
|
|
2009-08-24 13:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/app/gstappsink.c:
|
|
appsink: don't block in paused
|
|
When we are asked to unlock we should either leave the render function or call
|
|
the wait_preroll method to release the stream lock.
|
|
Fixes #592657
|
|
|
|
2009-08-24 13:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
docs: fix includes for appsrc/appsink
|
|
|
|
2009-08-24 11:24:27 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspdefs.c:
|
|
* gst-libs/gst/rtsp/gstrtspdefs.h:
|
|
rtsp: Add support for the Authentication-Info header.
|
|
The Authentication-Info header is defined in RFC 2617 (Digest Access
|
|
Authentication).
|
|
|
|
2009-08-20 13:11:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* ext/ogg/gstoggmux.c:
|
|
* tests/check/pipelines/oggmux.c:
|
|
oggmux: don't drop the streamheader field from the output caps
|
|
Revert previous 'fix' for bug #588717 and fix it properly, whilst
|
|
maintaining the streamheader field on the output caps. Also make
|
|
sure we don't leak header buffers we couldn't push when downstream
|
|
is unlinked. Add unit test for the presence of the streamheader
|
|
field on the output caps and for the issue from bug #588717.
|
|
|
|
2009-08-18 21:45:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/playback/gstinputselector.c:
|
|
* gst/playback/gststreamselector.c:
|
|
streamselector/inputselector: Use iterate internal links instead of deprecated get internal links
|
|
|
|
2009-08-19 09:31:51 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Avoid duplicated headers.
|
|
Remove any existing Session and Date headers before adding new ones
|
|
when sending a request. This may happen if the user of this code reuses
|
|
a request (rtspsrc does this when resending after authorization fails).
|
|
|
|
2009-08-18 16:49:58 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Corrected the HTTP digest authorization computation.
|
|
Do not use sizeof() on an array passed as an argument to a function and
|
|
expect to get anything but the size of a pointer. As a result only the
|
|
first 4 (or 8) bytes of the response buffer were initialized to 0 in
|
|
auth_digest_compute_response() which caused it to return a string which
|
|
was not NUL-terminated...
|
|
|
|
2009-08-18 11:15:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playsink: Also send SEEK events directly to a subpicture sink
|
|
|
|
2009-08-18 08:39:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playsink: If a custom text sink is used, send events to it too
|
|
Before, SEEK events would be sent to the video sink, which wouldn't
|
|
be linked in any way to the subtitle part of the pipeline and
|
|
subparse would never see the SEEK event. This would then seek
|
|
the audio/video but the subtitles would continue from the old
|
|
position instead.
|
|
Fixes bug #591664.
|
|
|
|
2009-08-18 08:20:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/playback/gsturidecodebin.c:
|
|
uridecodebin: Make missing plugins emit a warning message, not an error message
|
|
The problem with an error message is, that it will stop playback completely
|
|
while it could be that only a audio decoder plugin is missing and the video
|
|
could be played with the available plugins.
|
|
See bug #591677.
|
|
|
|
2009-08-13 17:42:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/playback/gsturidecodebin.c:
|
|
uridecodebin: Post a correct error message for unknown types
|
|
Before we had STREAM/WRONG_TYPE but it's really CORE/MISSING_PLUGIN
|
|
because a plugin is missing and nothing else is wrong.
|
|
Also make it an error instead of a warning.
|
|
Really fixes bug #591677.
|
|
|
|
2009-08-13 15:48:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/playback/gsturidecodebin.c:
|
|
uridecodebin: Post a missing plugin message additional to the error message on unknown types
|
|
Fixes bug #591677.
|
|
|
|
2009-08-13 10:59:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/bg.po:
|
|
* po/ca.po:
|
|
* po/cs.po:
|
|
* po/da.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/es.po:
|
|
* po/fi.po:
|
|
* po/fr.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/ru.po:
|
|
* po/sk.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/tr.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
* po/zh_CN.po:
|
|
playbin2: fix error message string
|
|
Fixes #591577.
|
|
|
|
2009-08-05 15:38:32 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
|
|
|
|
* gst-libs/gst/riff/riff-read.c:
|
|
riff: align API doc of gst_riff_parse_chunk with reality
|
|
|
|
2009-08-05 15:36:30 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
decodebin2: avoid assertion failure on empty/NULL caps
|
|
|
|
2009-08-12 12:09:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefindfunctions: Also detect SVG by the <svg> starting tag
|
|
Not all SVG images have the DOCTYPE specified.
|
|
|
|
2009-08-10 20:18:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtspconnection: don't use GLib-2.18 function
|
|
g_checksum_reset() was added only in GLib 2.18, but we still require
|
|
only 2.16, so work around that if we only have 2.16. Fixes #591357.
|
|
|
|
2009-08-10 15:40:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* tests/check/pipelines/streamheader.c:
|
|
streamheader: Fix caps leak in the vorbisenc unit test
|
|
|
|
2009-08-10 14:14:30 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* tests/check/pipelines/streamheader.c:
|
|
checks: fix stream header unit test hanging in gst_task_cleanup_all()
|
|
Set pipelines to NULL state and unref when done.
|
|
|
|
2009-08-10 10:17:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/Makefile.am:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/md5.c:
|
|
* gst-libs/gst/rtsp/md5.h:
|
|
rtsp: Use GLib's GChecksum instead of our own MD5 implementation
|
|
|
|
2009-08-10 03:46:39 +0300 Mart Raudsepp <leio@gentoo.org>
|
|
|
|
* gst-libs/gst/interfaces/navigation.c:
|
|
navigation: Fix doc blurb typo for gst_navigation_send_key_event
|
|
|
|
2009-08-09 12:13:16 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/subparse/gstsubparse.c:
|
|
subparse: Allow . instead of , as millisecond delimiter in srt subtitles
|
|
Fixes bug #591207.
|
|
|
|
2009-08-08 17:51:10 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstaudiosrc.c:
|
|
* gst/playback/gstinputselector.c:
|
|
* gst/playback/gststreamselector.c:
|
|
Revert inlines that cause compiler warnings and are not needed anyway
|
|
|
|
2009-08-08 15:54:57 +0200 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst-libs/gst/audio/gstaudioclock.c:
|
|
* gst-libs/gst/audio/gstaudiosink.c:
|
|
* gst-libs/gst/audio/gstaudiosrc.c:
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
* gst-libs/gst/audio/gstringbuffer.c:
|
|
* gst-libs/gst/interfaces/propertyprobe.c:
|
|
* gst-libs/gst/riff/riff-media.c:
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
* gst-libs/gst/video/gstvideofilter.c:
|
|
* gst-libs/gst/video/gstvideosink.c:
|
|
gst-libs: Remove dead assignments and resulting unused variables.
|
|
|
|
2009-08-08 15:54:41 +0200 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* ext/alsa/gstalsadeviceprobe.c:
|
|
* ext/alsa/gstalsasink.c:
|
|
* ext/alsa/gstalsasrc.c:
|
|
* ext/gnomevfs/gstgnomevfssrc.c:
|
|
* ext/ogg/gstoggaviparse.c:
|
|
* ext/ogg/gstoggdemux.c:
|
|
* ext/ogg/gstoggmux.c:
|
|
* ext/pango/gsttextrender.c:
|
|
* ext/vorbis/vorbisenc.c:
|
|
ext: Remove dead assignments and resulting unused variables.
|
|
|
|
2009-08-08 15:54:02 +0200 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst/adder/gstadder.c:
|
|
* gst/audioconvert/gstaudioconvert.c:
|
|
* gst/audioresample/gstaudioresample.c:
|
|
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
|
|
* gst/ffmpegcolorspace/imgconvert.c:
|
|
* gst/playback/gstdecodebin.c:
|
|
* gst/playback/gstdecodebin2.c:
|
|
* gst/playback/gstfactorylists.c:
|
|
* gst/playback/gstinputselector.c:
|
|
* gst/playback/gstplaysink.c:
|
|
* gst/playback/gststreamselector.c:
|
|
* gst/tcp/gsttcpclientsink.c:
|
|
* gst/videoscale/gstvideoscale.c:
|
|
* gst/videoscale/vs_image.c:
|
|
* gst/videotestsrc/gstvideotestsrc.c:
|
|
gst: Remove dead assignments and resulting unused variables
|
|
|
|
2009-08-07 13:05:42 +0200 Josep Torra <n770galaxy@gmail.com>
|
|
|
|
* docs/design/draft-va.txt:
|
|
docs: add draft for generic introduction of video acceleration APIs idea
|
|
|
|
2009-08-07 08:53:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* ext/theora/gsttheoradec.h:
|
|
* ext/theora/theoradec.c:
|
|
Revert "theora: Convert theoradec to libtheora 1.0 API"
|
|
This reverts commit f1e142ac9dcfb754d85357b9077d5aee48559dd9.
|
|
Temporarily revert until we have a workaround for debian/ubuntu
|
|
packaging failure (see http://bugs.debian.org/528710).
|
|
|
|
2009-08-07 09:32:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefindfunctions: Add typefinders for many game sound console formats supported by gme
|
|
These are AY, GBS, GYM, KSS, SAP and VGM. SPC and NSF already had typefinders.
|
|
|
|
2009-07-16 11:29:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* ext/ogg/gstoggmux.c:
|
|
oggmux: fix warning when we're not linked downstream and error out properly
|
|
Fix caps warning when there's no element linked downstream, and pass
|
|
not-linked flow return value correctly up the chain, so we error out
|
|
correctly. Fixes #588717.
|
|
|
|
2009-07-31 14:59:03 -0700 David Schleef <ds@schleef.org>
|
|
|
|
* ext/theora/gsttheoradec.h:
|
|
* ext/theora/theoradec.c:
|
|
theora: Convert theoradec to libtheora 1.0 API
|
|
|
|
2009-08-06 20:47:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/pango/gsttextrender.c:
|
|
textrender: Fix blitting of text over the output buffer and cairo painting
|
|
|
|
2009-08-06 09:13:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/pango/gsttextrender.c:
|
|
textrender: Fix endianness problems (i.e. make it work again on big endian architectures)
|
|
|
|
2009-07-31 14:27:28 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* tests/icles/test-colorkey.c:
|
|
colorkey-test: fix xsync error
|
|
|
|
2009-07-06 23:06:50 +0300 Siarhei Siamashka <siarhei.siamashka@nokia.com>
|
|
|
|
* gst/ffmpegcolorspace/imgconvert.c:
|
|
* gst/ffmpegcolorspace/imgconvert_template.h:
|
|
ffmpegcolorspace: support for direct conversion from uyvy422 to rgb formats
|
|
|
|
2009-07-14 12:33:29 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: smarter sink selection. Fixes #588523
|
|
Don't do fallbacks if application specified a sink element. When doing the
|
|
fallback use configured default elements instead of hardcoded linux only
|
|
elements. Improve error messages accordingly.
|
|
|
|
2009-08-06 12:18:36 +0200 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
|
|
|
|
* gst/playback/gstqueue2.c:
|
|
queue2: post error message when pausing task if so appropriate
|
|
If a downstream element returns an error while upstream has already
|
|
put all data into queue2 (including EOS), upstream will no longer
|
|
chain into queue2, so it is up to queue2 to perform some
|
|
EOS handling / message posting in such cases. See #589991.
|
|
|
|
2009-08-06 12:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
baseaudiosrc: change default slave method
|
|
Set the default slave method to the much better skew slaving algortihm.
|
|
|
|
2009-08-06 12:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* ext/pango/gsttextoverlay.c:
|
|
textoverlay: make buffer writable
|
|
Make the input buffer writable before changing its contents.
|
|
|
|
2009-08-06 09:55:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefinding: fix postscript typefinder probability
|
|
Two bytes for a rare format hardly warrants MAXIMUM typefinding
|
|
probability, POSSIBLE seems more appropriate.
|
|
|
|
2009-08-04 14:55:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/pango/gsttextoverlay.c:
|
|
pango: Send queries from the srcpad directly to the video sinkpad
|
|
|
|
2009-08-04 14:32:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/subparse/gstsubparse.c:
|
|
subparse: Implement POSITION query
|
|
|
|
2009-08-04 14:29:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/subparse/gstsubparse.c:
|
|
* gst/subparse/samiparse.c:
|
|
subparse: Implement SEEKING query
|
|
|
|
2009-08-04 14:14:53 +0200 John Millikin <jmillikin@gmail.com>
|
|
|
|
* configure.ac:
|
|
* gst-libs/gst/tag/gstid3tag.c:
|
|
* gst-libs/gst/tag/gstvorbistag.c:
|
|
tag: Add support for ALBUM_ARTIST tag in vorbiscomments and ID3v2 tags
|
|
Require latest core for this.
|
|
Fixes bug #590430.
|
|
|
|
2009-08-04 12:46:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/pango/gsttextoverlay.c:
|
|
* ext/pango/gsttextoverlay.h:
|
|
pango: Add support for xRGB and BGRx formats
|
|
|
|
2009-08-04 12:22:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/pango/gsttextoverlay.c:
|
|
pango: Fix endianness issues from the pangocairo switch
|
|
cairo's ARGB is in native endianness, i.e. ARGB on big endian architectures
|
|
and BGRA on little endian architectures.
|
|
|
|
2009-08-04 12:11:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/pango/gsttextoverlay.c:
|
|
pango: Re-add shading support which was dropped by a previous patch
|
|
|
|
2009-08-04 11:58:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* configure.ac:
|
|
* ext/pango/gsttextoverlay.c:
|
|
pango: Check if pangocairo supports vertical rendering and fix properties
|
|
|
|
2009-08-04 11:45:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/pango/gsttextrender.c:
|
|
textrender: Use PROP_X instead of ARG_X consistently
|
|
|
|
2009-08-04 11:42:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/pango/gstclockoverlay.c:
|
|
* ext/pango/gsttextoverlay.c:
|
|
* ext/pango/gsttextrender.c:
|
|
* ext/pango/gsttimeoverlay.c:
|
|
pango: Some minor cleanup
|
|
|
|
2009-08-04 11:36:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* configure.ac:
|
|
pango: Check for pangocairo instead of pangoft2
|
|
|
|
2009-08-04 11:35:10 +0200 Young-Ho Cha <ganadist@chollian.net>
|
|
|
|
* ext/pango/gsttextoverlay.c:
|
|
* ext/pango/gsttextoverlay.h:
|
|
* ext/pango/gsttextrender.c:
|
|
* ext/pango/gsttextrender.h:
|
|
pango: Use pango-cairo instead of pango-ft2
|
|
pango-cairo will always use the native font rendering backend
|
|
of the platform and provides better results.
|
|
Fixes bug #340887.
|
|
|
|
2009-08-04 10:35:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefindfunctions: Add SVG typefinder
|
|
|
|
2009-08-04 10:29:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefindfunctions: Add postscript typefinder
|
|
|
|
2009-07-30 15:08:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefindfunctions: Use static caps again for MPEG4 typefinding
|
|
|
|
2009-07-30 15:05:28 +0200 Arnout Vandecappelle <arnout@mind.be>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefindfunctions: Implement better & more flexible MPEG4 typefinding
|
|
This detects more MPEG4 streams as MPEG4.
|
|
Fixes bug #556537.
|
|
|
|
2009-07-30 14:04:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/cdda/gstcddabasesrc.c:
|
|
cddabasesrc: Allow to specify the device name in the URI
|
|
The allowed URI scheme is now:
|
|
cdda://(device#)?track
|
|
Also allow every combination of uppercase and lowercase
|
|
characters for the protocol part.
|
|
Fixes bug #321532.
|
|
|
|
2009-07-30 12:37:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/gstvideoscale.c:
|
|
videoscale: Restrict width/height to 2^15 - 1
|
|
Otherwise integer overflows will happen, resulting in segmentation faults.
|
|
Fixes bug #590243.
|
|
|
|
2009-07-29 14:55:04 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/ffmpegcolorspace/imgconvert_template.h:
|
|
ffmpegcolorspace: Fix indention of template header
|
|
|
|
2009-07-29 14:10:35 +0200 Philip Jägenstedt <philipj@opera.com>
|
|
|
|
* gst-libs/gst/app/gstappsrc.c:
|
|
appsrc: Clarify documentation about caps and linkage
|
|
Fixes bug #589095.
|
|
|
|
2009-07-29 07:42:05 +0200 Benjamin Gaignard <benjamin@gaignard.net>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefindfunctions: Fix typefinding of SDP files
|
|
Fixes bug #589574.
|
|
|
|
2009-07-28 20:50:06 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
|
|
|
|
* gst/audioresample/gstaudioresample.c:
|
|
audioresample: Take the output offsets from the input if possible
|
|
Fixes bug #588915.
|
|
|
|
2009-07-28 15:54:14 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/gstvideoscale.c:
|
|
videoscale: Make sure to allocate enough memory for the temporary buffer
|
|
and fix scaling of odd-height interlaced video.
|
|
|
|
2009-07-28 15:18:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/gstvideoscale.c:
|
|
videoscale: Fix interlaced scaling for I420
|
|
...and some other minor mistakes in the previous change.
|
|
|
|
2009-07-28 14:12:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/ffmpegcolorspace/avcodec.h:
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.h:
|
|
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
|
|
* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
|
|
* gst/ffmpegcolorspace/imgconvert.c:
|
|
ffmpegcolorspace: Include interlacing information in the AVPicture
|
|
This later allows to handle interlaced AVPicture different than
|
|
progressive ones which is needed for horizontally subsampled YUV
|
|
formats, see bug #589242.
|
|
|
|
2009-07-28 13:55:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/gstvideoscale.c:
|
|
* gst/videoscale/gstvideoscale.h:
|
|
videoscale: Add support for interlaced content
|
|
videoscale is not mixing content of two seperate fields anymore
|
|
and does scaling on every field separately.
|
|
Fixes bug #588761.
|
|
|
|
2009-08-06 01:44:24 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* configure.ac:
|
|
back to development -> 0.10.24.1
|
|
|
|
2009-08-05 02:03:44 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* gst-plugins-base.doap:
|
|
Add 0.10.24 release to the doap file
|
|
|
|
=== release 0.10.24 ===
|
|
|
|
2009-08-05 00:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* ChangeLog:
|
|
* NEWS:
|
|
* RELEASE:
|
|
* configure.ac:
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
|
* docs/plugins/gst-plugins-base-plugins.interfaces:
|
|
* docs/plugins/gst-plugins-base-plugins.prerequisites:
|
|
* docs/plugins/gst-plugins-base-plugins.signals:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.xml:
|
|
* docs/plugins/inspect/plugin-app.xml:
|
|
* docs/plugins/inspect/plugin-audioconvert.xml:
|
|
* docs/plugins/inspect/plugin-audiorate.xml:
|
|
* docs/plugins/inspect/plugin-audioresample.xml:
|
|
* docs/plugins/inspect/plugin-audiotestsrc.xml:
|
|
* docs/plugins/inspect/plugin-cdparanoia.xml:
|
|
* docs/plugins/inspect/plugin-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gdp.xml:
|
|
* docs/plugins/inspect/plugin-gio.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.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-queue2.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-uridecodebin.xml:
|
|
* docs/plugins/inspect/plugin-video4linux.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:
|
|
Release 0.10.24
|
|
|
|
2009-08-05 00:38:40 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/bg.po:
|
|
* po/ca.po:
|
|
* po/cs.po:
|
|
* po/da.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/es.po:
|
|
* po/fi.po:
|
|
* po/fr.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/ru.po:
|
|
* po/sk.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
|
|
|
|
2009-08-01 17:26:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
* tests/check/gst/typefindfunctions.c:
|
|
typefinding: fix detection of fLaC id packet in broken flac-in-ogg
|
|
There are flac-in-ogg files without the usual flac packet framing
|
|
and these files just have a 4-byte fLaC ID packet as first packet.
|
|
We need to recognise the type just from these four bytes if we
|
|
want oggdemux to recognise these streams correctly.
|
|
|
|
2009-07-30 14:40:50 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* ChangeLog:
|
|
* configure.ac:
|
|
* po/LINGUAS:
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/bg.po:
|
|
* po/ca.po:
|
|
* po/cs.po:
|
|
* po/da.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/es.po:
|
|
* po/fi.po:
|
|
* po/fr.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/ru.po:
|
|
* po/sk.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/tr.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
* po/zh_CN.po:
|
|
0.10.24.5 pre-release
|
|
|
|
2009-07-29 14:15:53 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstaudiofilter.c:
|
|
audiofilter: Don't assert on slightly different caps
|
|
Plugins should not assert on incompatible caps, caps negotiation will
|
|
fail anyway.
|
|
|
|
2009-07-30 13:42:21 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/adder/gstadder.c:
|
|
adder: reset pending flush-stop flag in state_changed. (mostly) Fixes #590146.
|
|
|
|
2009-07-30 09:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* configure.ac:
|
|
configure: bump Gtk+ requirement of GUI examples from 2.12 to 2.14
|
|
The gio mount example needs GtkMountOperation, which is new in 2.14.
|
|
|
|
2009-07-27 10:29:27 +0100 Balachandran C <balachandran_c@rediffmail.com>
|
|
|
|
* ext/alsa/gstalsasrc.c:
|
|
alsasrc: set alsasrc->handle back to NULL when closing device
|
|
Fixes crashes in gst_alsa_find_device_name() when probing or
|
|
reading the device-name property (e.g. when doing a dot-file
|
|
dump). Fixes #589797.
|
|
|
|
2009-07-24 19:26:40 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/playback/gststreamselector.c:
|
|
playbin: rename GType of stream selector pad to avoid clash with input-selector from -bad
|
|
Rename the GType of the pads of playbin's internal stream selector
|
|
element so they don't use the same type name as input-selector's
|
|
pads. Fixes #589622.
|
|
|
|
2009-07-24 13:39:55 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* ChangeLog:
|
|
* configure.ac:
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/bg.po:
|
|
* po/ca.po:
|
|
* po/cs.po:
|
|
* po/da.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/es.po:
|
|
* po/fi.po:
|
|
* po/fr.po:
|
|
* po/hu.po:
|
|
* po/id.po:
|
|
* po/it.po:
|
|
* po/ja.po:
|
|
* po/lt.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/pl.po:
|
|
* po/pt_BR.po:
|
|
* po/ru.po:
|
|
* po/sk.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/tr.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
* po/zh_CN.po:
|
|
0.10.23.4 pre-release
|
|
|
|
2009-07-24 13:46:15 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* tests/examples/v4l/.gitignore:
|
|
ignores: Ignore v4l probing example binary
|
|
|
|
2009-07-24 09:35:38 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefind: recognise Kate spu subtitles as well
|
|
Recognise spu-subtitles, SUB and K-SPU as valid categories for
|
|
Kate subtitles as well.
|
|
|
|
2009-07-24 00:42:16 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* common:
|
|
Automatic update of common submodule
|
|
From fedaaee to 94f95e3
|
|
|
|
2009-07-22 14:21:43 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
|
|
|
|
* gst-plugins-base.spec.in:
|
|
Update spec file with latest changes
|
|
|
|
2009-07-20 17:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* configure.ac:
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/bg.po:
|
|
* po/ca.po:
|
|
* po/cs.po:
|
|
* po/da.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/es.po:
|
|
* po/fi.po:
|
|
* po/fr.po:
|
|
* po/hu.po:
|
|
* po/id.po:
|
|
* po/it.po:
|
|
* po/ja.po:
|
|
* po/lt.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/pl.po:
|
|
* po/pt_BR.po:
|
|
* po/ru.po:
|
|
* po/sk.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/tr.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
* po/zh_CN.po:
|
|
* win32/common/_stdint.h:
|
|
* win32/common/audio-enumtypes.c:
|
|
* win32/common/config.h:
|
|
* win32/common/gstrtsp-enumtypes.c:
|
|
* win32/common/interfaces-enumtypes.c:
|
|
* win32/common/video-enumtypes.c:
|
|
0.10.23.3 pre-release
|
|
|
|
2009-07-20 12:51:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/audiotestsrc/gstaudiotestsrc.c:
|
|
audiotestsrc: call send_event directly
|
|
We can't call gst_element_send_event() from a streaming thread as it gets the
|
|
state lock. Instead call the send_event method directly until we have a nice API
|
|
for this in basesrc.
|
|
Fixes #588746
|
|
|
|
2009-07-03 04:42:24 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstaudiosink.c:
|
|
audiosink: Add stream-status messages
|
|
Fixes #587695
|
|
|
|
2009-07-03 04:41:05 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstaudiosrc.c:
|
|
audiosrc: Add stream-status messages
|
|
See #587695
|
|
|
|
2009-07-20 10:53:11 +0200 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst/adder/gstadder.c:
|
|
gstadder: Don't forget to free pending events on flush/dispose.
|
|
Fixes #588747
|
|
|
|
2009-07-12 10:08:12 +0200 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* tests/check/elements/adder.c:
|
|
tests/adder: Add stream consistency checking. Fixes #588748
|
|
|
|
2009-07-12 10:07:34 +0200 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst/audiotestsrc/gstaudiotestsrc.c:
|
|
audiotestsrc: Make sure tags are properly serialized. Fixes #588746
|
|
We do this by letting the basesrc base class handle the tags.
|
|
|
|
2009-07-13 09:28:54 +0200 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst/adder/gstadder.c:
|
|
* gst/adder/gstadder.h:
|
|
adder: Collect incoming tag events and send them after newsegment. Fixes #588747
|
|
|
|
2009-07-16 09:32:46 +0200 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* ext/vorbis/vorbisdec.c:
|
|
vorbisdec: Check for empty tag strings. Fixes #588724
|
|
|
|
2009-07-14 17:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstqueue2.c:
|
|
queue2: fix leak and improve buffering
|
|
Keep track of the max requested position and compare this to the write position
|
|
in the temp file to get the current amount of buffered data.
|
|
Fix memleak of all incomming buffers.
|
|
Fixes #588551
|
|
|
|
2009-07-15 17:40:14 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/playback/Makefile.am:
|
|
* gst/playback/gstinputselector.c:
|
|
* gst/playback/gstinputselector.h:
|
|
* gst/playback/gstplay-marshal.list:
|
|
* gst/playback/gstplaybin2.c:
|
|
playbin2: use private copy of input-selector
|
|
We shouldn't really depend on elements from -bad for stream
|
|
selection in playbin2, so use a private copy of input-selector
|
|
until the selector plugin is ready to be moved to -base or -good.
|
|
Fixes #586356.
|
|
|
|
2009-07-15 17:26:32 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/playback/gstinputselector.c:
|
|
* gst/playback/gstinputselector.h:
|
|
playback: add private copy of the input-selector from gst-plugins-bad
|
|
Not hooked up yet though. See #586356.
|
|
|
|
2009-07-14 19:00:36 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
|
|
|
|
* tests/examples/v4l/Makefile.am:
|
|
examples: fix v4l probe example build
|
|
Fixes bug #588550.
|
|
|
|
2009-07-14 19:00:10 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* ChangeLog:
|
|
* configure.ac:
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/bg.po:
|
|
* po/ca.po:
|
|
* po/cs.po:
|
|
* po/da.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/es.po:
|
|
* po/fi.po:
|
|
* po/fr.po:
|
|
* po/hu.po:
|
|
* po/id.po:
|
|
* po/it.po:
|
|
* po/ja.po:
|
|
* po/lt.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/pl.po:
|
|
* po/pt_BR.po:
|
|
* po/ru.po:
|
|
* po/sk.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/tr.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
* po/zh_CN.po:
|
|
0.10.23.2 pre-release
|
|
|
|
2009-07-14 16:24:10 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* po/LINGUAS:
|
|
* po/tr.po:
|
|
Add Turkish translations
|
|
|
|
2009-07-14 15:31:13 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* tests/check/elements/adder.c:
|
|
adder: One more attempt to fix the adder test
|
|
Give up and discard and recreate the alsasrc after checking it can
|
|
be opened, due to some strange crash inside alsa when we don't.
|
|
|
|
2009-07-14 15:06:41 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* tests/check/elements/adder.c:
|
|
adder: Perform get_state() in the unit test
|
|
Wait for the alsasrc to return to NULL after setting it to PAUSED for
|
|
testing, otherwise it leads to segfaults later on.
|
|
|
|
2009-07-14 14:39:32 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* tests/check/elements/adder.c:
|
|
adder: Don't fail when alsasrc is unavailable
|
|
Make the liveadder test succeed silently when it can't be completed
|
|
either because alsasrc is unavailable, or because the device is
|
|
inaccessible.
|
|
|
|
2009-07-13 22:51:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/pbutils/descriptions.c:
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefinding: use subtitle/x-kate for Kate subtitle streams and application/x-kate for the rest
|
|
Differentiate subtitle streams and lyrics/cracktastic/complex streams via
|
|
the category string in the headers. This seems like a useful distinction
|
|
to make, and also seems more future-proof. See #525743.
|
|
|
|
2009-02-21 13:18:10 +0000 Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
|
|
|
|
* ext/ogg/gstoggmux.c:
|
|
oggmux: add Kate caps to the list of accepted types
|
|
See #525743.
|
|
|
|
2009-07-13 21:56:46 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/playback/gsturidecodebin.c:
|
|
uridecodebin: treat uri-schemas incasesensitive
|
|
Treat uri-schemas incasesensitive. This is mandated in rfc2396 section 3.1.
|
|
Fixes not showing buffering messages e.g. for HTTP://...
|
|
|
|
2009-07-13 21:54:47 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst-libs/gst/interfaces/navigation.c:
|
|
navigation: simplify docs
|
|
Make short-desc short - its used in the toc. Strip uneeded markup.
|
|
|
|
2009-07-13 18:31:15 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* win32/common/libgstnetbuffer.def:
|
|
* win32/common/libgstvideo.def:
|
|
win32: Fix exports
|
|
Remove methods from video base classes that have moved to -bad.
|
|
Add gst_netaddress_to_string
|
|
|
|
2009-07-13 17:56:58 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* tests/examples/gio/.gitignore:
|
|
ignores: ignore the giosrc-mounting example binary
|
|
|
|
2009-07-13 17:54:40 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* gst-libs/gst/interfaces/navigation.c:
|
|
navigation: Add some partial documentation
|
|
Add a general documentation blurb for the GstNavigation functionality.
|
|
Still lacks some example code and detail on how to implement it.
|
|
|
|
2009-07-13 17:52:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/pbutils/descriptions.c:
|
|
pbutils: add description for Siren codec and make two descriptions non-translatable
|
|
|
|
2009-07-13 12:23:20 -0400 Olivier Crête <olivier.crete@collabora.co.uk>
|
|
|
|
* common:
|
|
Automatic update of common submodule
|
|
From 5845b63 to fedaaee
|
|
|
|
2009-07-13 18:21:49 +0200 Elliott Sales de Andrade <quantum.analyst at gmail.com>
|
|
|
|
* gst-libs/gst/riff/riff-ids.h:
|
|
* gst-libs/gst/riff/riff-media.c:
|
|
riff: add siren to the RIFF parser
|
|
Add siren7 caps to the RIFF parser.
|
|
|
|
2009-07-13 14:55:59 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
|
|
|
|
* configure.ac:
|
|
* tests/examples/Makefile.am:
|
|
* tests/examples/v4l/Makefile.am:
|
|
* tests/examples/v4l/probe.c:
|
|
v4lsrc: add a simple test case for device probing
|
|
|
|
2009-07-03 11:38:01 +0200 Filippo Argiolas <filippo.argiolas@gmail.com>
|
|
|
|
* configure.ac:
|
|
* sys/v4l/Makefile.am:
|
|
* sys/v4l/gstv4lelement.c:
|
|
v4lsrc: optional support for device probing with gudev
|
|
Enumerate v4l devices using gudev if available.
|
|
Fixes bug #583640.
|
|
|
|
2009-07-10 23:24:36 +0100 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/adder/gstadder.c:
|
|
adder: add since tags to docs
|
|
|
|
2009-07-10 21:29:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/examples/seek/seek.c:
|
|
seek: don't automatically start pipeline in DB
|
|
Keep the pipeline paused when we detect download buffering. The user has to
|
|
manually start the pipeline for now because we can't estimate when the buffering
|
|
will finish or when we have underrun.
|
|
|
|
2009-07-10 21:01:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstqueue2.c:
|
|
queue2: flush differently, avoiding deadlocks
|
|
Don't flush the file by closing and opening it but instead use g_freopen. This
|
|
avoids a deadlock in shutdown because we emit the temp-location property change
|
|
with the wrong lock held.
|
|
|
|
2009-07-10 20:25:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/examples/seek/seek.c:
|
|
seek: add a checkbox for progressive download
|
|
|
|
2009-07-10 20:24:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gsturidecodebin.c:
|
|
uridecodebin: Fix template construction
|
|
Fix the construction of the temporary filename construction as the application
|
|
name can be NULL and we don't want a separator between the prgname and the
|
|
template.
|
|
|
|
2009-07-10 20:04:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplay-enum.c:
|
|
* gst/playback/gstplay-enum.h:
|
|
* gst/playback/gstplaybin2.c:
|
|
playbin2: add support for progressive download
|
|
Add a new playbin2 flag (initially disabled) to enable progressive download
|
|
buffering in uridecodebin.
|
|
|
|
2009-07-10 19:59:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gsturidecodebin.c:
|
|
uridecodebin: add download property
|
|
Add a download property that will attempt to configure queue2 into progressive
|
|
download buffering.
|
|
Make sure we only enable download buffering for quicktime and flv formats.
|
|
|
|
2009-07-10 19:49:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstqueue2.c:
|
|
queue2: add temp-template property
|
|
Add a new temp-template property so that queue2 can securely allocate a
|
|
temporary filename. Deprecate the temp-location property for setting the
|
|
location but still use it to notify the allocated temp file.
|
|
|
|
2009-07-10 20:06:28 +0100 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/adder/gstadder.c:
|
|
* gst/adder/gstadder.h:
|
|
adder: add a caps-property to avoid to need to plug a capsfilter afterwards
|
|
Adder can only handle one common format accross the pads. Thus one needed to add
|
|
a capsfilter afterwards and manage the caps. Now one can simply set the caps on
|
|
the property.
|
|
|
|
2009-07-10 18:59:05 +0100 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* tests/check/elements/adder.c:
|
|
adder: skip live-seek text if we have no audiosrc, add new test
|
|
The seek-test needs a real audiosrc. Also add a test that checks that adder is
|
|
reusable. Finaly handle warnings as warnings to fix a assertion.
|
|
|
|
2009-07-10 19:16:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/gio/gstgiosink.c:
|
|
gio: Also post a "not-mounted" message from giosink
|
|
|
|
2009-07-10 17:15:48 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* tests/examples/gio/giosrc-mounting.c:
|
|
gio: Remove workaround for playbin2 bug in the sample application
|
|
The playbin2 bug was #588078.
|
|
|
|
2009-07-10 17:08:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
playbin2: Make it possible for READY->PAUSED to succeed after it failed the first time
|
|
If READY->PAUSED failed in the source element we would've swapped
|
|
the current and next group already. To allow READY->PAUSED to succeed
|
|
after the first failure we have to swap the current and next group
|
|
back again. This also ensure that we're again in the same state
|
|
as before the failed state change and not at the next group.
|
|
This was especially a problem for playbin2 pipelines that use the
|
|
new mounting support in giosrc as the source would fail for READY->PAUSED
|
|
the first time, the application mounts the location and then tries
|
|
to go READY->PAUSED again (and this time it would succeed).
|
|
Fixes bug #588078.
|
|
|
|
2009-07-10 11:42:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* configure.ac:
|
|
* tests/examples/Makefile.am:
|
|
* tests/examples/gio/Makefile.am:
|
|
* tests/examples/gio/giosrc-mounting.c:
|
|
gio: Add example application that shows how to handle the "not-mounted" message
|
|
|
|
2009-07-10 11:24:57 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* configure.ac:
|
|
gio: Remove the experimental status from the GIO plugin
|
|
Fixes bug #510417.
|
|
|
|
2009-07-10 11:24:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/gio/gstgiosink.c:
|
|
* ext/gio/gstgiosrc.c:
|
|
gio: Add documentation for the new "not-mounted" and "file-exists" messages
|
|
|
|
2009-07-09 13:45:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/gio/gstgiobasesrc.c:
|
|
gio: Make sure that we have the correct stream position when starting
|
|
|
|
2009-07-08 17:24:19 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/gio/gstgiobasesink.c:
|
|
gio: Make sure to flush the output stream if it shouldn't be closed
|
|
Otherwise there might still be unwritten data after the element
|
|
has stopped.
|
|
|
|
2009-07-08 17:19:29 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/gio/gstgiobasesink.c:
|
|
* ext/gio/gstgiobasesink.h:
|
|
* ext/gio/gstgiobasesrc.c:
|
|
* ext/gio/gstgiobasesrc.h:
|
|
* ext/gio/gstgiosink.c:
|
|
* ext/gio/gstgiosrc.c:
|
|
gio: Don't close the GIO streams for the giostream{src,sink} elements
|
|
This makes it possible to do something useful with the streams
|
|
after the element has stopped. Fixes bug #587896.
|
|
|
|
2009-07-08 17:19:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* tests/check/pipelines/gio.c:
|
|
gio: Try to reuse the pipeline with the same stream objects
|
|
|
|
2009-07-08 17:02:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/gio/gstgiobasesink.c:
|
|
* ext/gio/gstgiobasesrc.c:
|
|
gio: Improve the error message if a stream is already closed before usage
|
|
|
|
2009-07-08 16:55:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/gio/gstgiosink.c:
|
|
gio: Post a custom file-exists message on the bus if the file already exists
|
|
An application can handle this message, remove the file in question
|
|
and restart the pipeline again without showing an error.
|
|
This fixes bug #529300.
|
|
|
|
2009-07-08 16:54:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/gio/gstgiosrc.c:
|
|
gio: Use OPEN_READ instead of NOT_FOUND if a location is not mounted
|
|
|
|
2009-07-08 16:50:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/gio/gstgiosink.c:
|
|
gio: Use OPEN_WRITE instead of OPEN_READ as error category in giosink
|
|
|
|
2009-07-08 15:52:35 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/gio/gstgiosrc.c:
|
|
gio: Post a custom "not-mounted" message on the bus
|
|
This allows applications to mount the GFile if possible and restart
|
|
the pipeline instead of simply giving an error.
|
|
|
|
2009-07-08 15:08:32 +0200 Philip Jägenstedt <philipj@opera.com>
|
|
|
|
* gst/audioconvert/gstchannelmix.c:
|
|
audioconvert: Fix compilation when debugging is disabled
|
|
Fixes bug #587980.
|
|
|
|
2009-07-07 20:23:23 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/gio/gstgiobasesink.c:
|
|
* ext/gio/gstgiobasesink.h:
|
|
* ext/gio/gstgiobasesrc.h:
|
|
* ext/gio/gstgiosink.c:
|
|
* ext/gio/gstgiosink.h:
|
|
* ext/gio/gstgiostreamsink.c:
|
|
* ext/gio/gstgiostreamsink.h:
|
|
gio: Add vfunc for requesting the stream for the sinks too
|
|
|
|
2009-07-07 20:21:36 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/gio/gstgiobasesink.c:
|
|
* ext/gio/gstgiobasesink.h:
|
|
* ext/gio/gstgiobasesrc.c:
|
|
* ext/gio/gstgiosink.c:
|
|
* ext/gio/gstgiosrc.c:
|
|
* ext/gio/gstgiostreamsink.c:
|
|
* ext/gio/gstgiostreamsrc.c:
|
|
gio: Some more random cleanup
|
|
|
|
2009-07-07 20:20:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/gio/gstgio.c:
|
|
* ext/gio/gstgiobasesink.c:
|
|
* ext/gio/gstgiobasesrc.c:
|
|
* ext/gio/gstgiobasesrc.h:
|
|
* ext/gio/gstgiosink.c:
|
|
* ext/gio/gstgiosrc.c:
|
|
* ext/gio/gstgiosrc.h:
|
|
* ext/gio/gstgiostreamsink.c:
|
|
* ext/gio/gstgiostreamsrc.c:
|
|
* ext/gio/gstgiostreamsrc.h:
|
|
gio: Update my mail address and copyright
|
|
|
|
2009-07-07 20:18:00 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/gio/gstgiobasesrc.c:
|
|
* ext/gio/gstgiobasesrc.h:
|
|
* ext/gio/gstgiosrc.c:
|
|
* ext/gio/gstgiostreamsrc.c:
|
|
* ext/gio/gstgiostreamsrc.h:
|
|
gio: General clean up and simplification
|
|
The GInputStreams are now requested by a vfunc from
|
|
the subclasses instead of relying that the subclass
|
|
sets it until it's needed.
|
|
This might also fix bug #587896.
|
|
|
|
2009-07-06 22:31:12 +0100 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/adder/gstadder.c:
|
|
adder: keep sending newsegments after seeking
|
|
Adder sends with timestamps from 0 upwards. After seeking we need to send
|
|
new-segments to get correct positions-queries.
|
|
|
|
2009-07-06 20:44:00 +0100 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* tests/check/elements/adder.c:
|
|
adder: make test more robust
|
|
Add audioconverts to the live-seeking test to make it negotiate.
|
|
|
|
2009-06-30 17:19:50 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* sys/xvimage/xvimagesink.c:
|
|
xvimagesink: use core performance log category
|
|
|
|
2009-07-05 21:29:40 +0200 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst/adder/gstadder.c:
|
|
adder: Call set_flushing(TRUE) for flushing seeks *when* the streaming is stopped.
|
|
This ensures that collectpads' cookie is properly updated so that when the streaming
|
|
threads will restart and be checking for the flushing status of all pads there will
|
|
be no inconsistent state.
|
|
|
|
2009-07-05 18:01:38 +0200 Hans-Peter Nilsson <hp@gcc.gnu.org>
|
|
|
|
* ext/pango/gstclockoverlay.c:
|
|
pango: Call tzset() before localtime_r()
|
|
POSIX and your local friendly ctime(3) manual entry says that localtime_r isn't
|
|
required to set the state variables that define the current timezone. Indeed,
|
|
glibc (at least 2.9) doesn't do this for subsequent calls. The effect is that
|
|
if the system timezone is changed for a running program between two calls to
|
|
gst_clock_overlay_render_time, it won't be noticed. For glibc, changing the
|
|
timezone equals /etc/localtime being modified.
|
|
Fixes bug #587676.
|
|
|
|
2009-07-01 17:33:14 -0700 David Schleef <ds@schleef.org>
|
|
|
|
* ext/Makefile.am:
|
|
build: remove spurious schroedinger reference
|
|
|
|
2009-07-01 10:25:43 -0700 David Schleef <ds@schleef.org>
|
|
|
|
* configure.ac:
|
|
* ext/Makefile.am:
|
|
* ext/schroedinger/Makefile.am:
|
|
* ext/schroedinger/gstschro.c:
|
|
* ext/schroedinger/gstschrodec.c:
|
|
* ext/schroedinger/gstschroenc.c:
|
|
* ext/schroedinger/gstschroparse.c:
|
|
* ext/schroedinger/gstschroutils.c:
|
|
* ext/schroedinger/gstschroutils.h:
|
|
* gst-libs/gst/video/Makefile.am:
|
|
* gst-libs/gst/video/gstbasevideocodec.c:
|
|
* gst-libs/gst/video/gstbasevideocodec.h:
|
|
* gst-libs/gst/video/gstbasevideodecoder.c:
|
|
* gst-libs/gst/video/gstbasevideodecoder.h:
|
|
* gst-libs/gst/video/gstbasevideoencoder.c:
|
|
* gst-libs/gst/video/gstbasevideoencoder.h:
|
|
* gst-libs/gst/video/gstbasevideoparse.c:
|
|
* gst-libs/gst/video/gstbasevideoparse.h:
|
|
* gst-libs/gst/video/gstbasevideoutils.c:
|
|
* gst-libs/gst/video/gstbasevideoutils.h:
|
|
basevideo: send basevideo back to remedial school
|
|
Move basevideo classes and schroedinger plugin to -bad.
|
|
|
|
2009-07-01 12:54:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/netbuffer/gstnetbuffer.h:
|
|
netaddress: add constant for max len
|
|
|
|
2009-07-01 12:48:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/netbuffer/gstnetbuffer.c:
|
|
* gst-libs/gst/netbuffer/gstnetbuffer.h:
|
|
netbuffer: add gst_netaddress_to_string
|
|
Add function to serialize a net address to a string.
|
|
API: GstNetAddress::gst_netaddress_to_string()
|
|
|
|
2009-06-30 18:44:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gsturidecodebin.c:
|
|
uridecodebin: make fd:// uri use buffering too
|
|
fd:// usually operate in push mode only and are thus suitable for buffering.
|
|
|
|
2009-06-30 14:46:38 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
* gst/volume/gstvolume.c:
|
|
volume: include "1.0=100%" in property description
|
|
|
|
2009-06-30 14:45:51 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playsink: remove unused property defs
|
|
|
|
2009-06-29 17:11:50 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst-libs/gst/audio/multichannel.c:
|
|
multichannel: rewrite the new doc comment a bit
|
|
Its part of the audio lib.
|
|
|
|
2009-06-29 14:34:02 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playsink: Avoid a segfault when the video sink fails to start
|
|
Don't attempt to display the subpictures and segfault when the
|
|
video sink failed to start (and hence the videochain is NULL).
|
|
|
|
2009-06-29 15:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstringbuffer.c:
|
|
* gst-libs/gst/audio/gstringbuffer.h:
|
|
ringbuffer: add vmethod to clear the ringbuffer
|
|
Add a vmethod so that subclasses can be notified when they should clear the data
|
|
in the ringbuffer.
|
|
|
|
2009-06-29 14:00:14 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* gst-libs/gst/riff/riff-media.c:
|
|
riff-media: Fix the fourcc caps property for VC-1/WMVA
|
|
The caps property for carrying fourccs is 'format', not 'fourcc'
|
|
|
|
2009-06-29 12:20:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: include in.h for FreeBSD compat
|
|
Fixes #586920
|
|
|
|
2009-06-29 12:20:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* win32/common/libgstapp.def:
|
|
defs: add defs for new appsink buffer-list method
|
|
|
|
2009-06-29 12:14:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/app/gstappsink.c:
|
|
* gst-libs/gst/app/gstappsink.h:
|
|
appsink: add docs and signals
|
|
Add docs for the new callback.
|
|
Add signals for the new buffer-list support.
|
|
|
|
2009-06-29 10:24:36 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
|
|
|
|
* tests/check/elements/appsink.c:
|
|
Added unit tests for buffer list support in appsink.
|
|
|
|
2009-06-17 11:12:08 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
|
|
|
|
* gst-libs/gst/app/gstappsink.c:
|
|
Added buffer list support.
|
|
|
|
2009-06-17 09:23:11 +0200 Branko Subasic <branko@lnxbranko2.se.axis.com>
|
|
|
|
* gst-libs/gst/app/gstappsink.h:
|
|
Added buffer list support.
|
|
|
|
2009-06-29 09:36:27 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/sdp/gstsdpmessage.c:
|
|
sdp: Include winsock2.h after defining WINVER.
|
|
Similar to bug #587080.
|
|
|
|
2009-06-29 09:31:40 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Moved a comment.
|
|
|
|
2009-06-27 23:23:02 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst-libs/gst/audio/audio.c:
|
|
* gst-libs/gst/audio/multichannel.c:
|
|
docs: add basic section docs for multichannel and relocate the ones for audio
|
|
Add section docs for multichannel, so that it has a short desc in the toc too.
|
|
Move the section docs in adio up, so that the follow the copyright like
|
|
elsewhere.
|
|
|
|
2009-06-26 21:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* sys/v4l/gstv4lelement.c:
|
|
* sys/v4l/gstv4lsrc.c:
|
|
v4l: open/close device in ready.
|
|
Simillar change like in v4l2src. This allows probing feature in paused, where
|
|
streaming is noit yet started.
|
|
|
|
2009-06-10 17:05:22 +0300 René Stadler <rene.stadler@nokia.com>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: fix initial volume handling also when reusing the element
|
|
This is a follow-up to commit 452988, making it work correctly when the audio
|
|
chain is reused.
|
|
|
|
2009-06-26 21:48:58 +0400 Руслан Ижбулатов <lrn1986@gmail.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
Define WINVER before including any win headers
|
|
Fixes bug #587080.
|
|
|
|
2009-06-27 00:50:54 +0300 René Stadler <mail@renestadler.de>
|
|
|
|
* gst-libs/gst/riff/riff-read.c:
|
|
riff: prevent crash if rounded up tag size exceeds data size
|
|
When rounding up `tsize' exceeds the remaining buffer size, `size' underflows
|
|
and an invalid read past the buffer data follows.
|
|
|
|
2009-06-26 15:17:21 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/video/gstbasevideocodec.c:
|
|
basevideocodec: By default don't allow caps changes on the srcpad
|
|
This fixed playback of Dirac files with schrodec when upstream wants
|
|
a different width/height, basevideocodec accepts this and then
|
|
pushes buffers with new caps but content of the old caps.
|
|
In the best case this will just result in wrong unit size and a
|
|
failure in basestransform elements.
|
|
|
|
2009-06-26 14:11:21 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* autogen.sh:
|
|
autogen.sh: Use printf instead of 'echo -n'. Check for automake-1.1[01]
|
|
Check for more automake command variants. Use printf instead of 'echo -n'
|
|
for portability
|
|
|
|
2009-06-26 13:41:38 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* common:
|
|
Automatic update of common submodule
|
|
From f810030 to 5845b63
|
|
|
|
2009-06-26 13:14:02 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/playback/gstscreenshot.c:
|
|
screenshot: don't leak message
|
|
|
|
2009-06-25 12:04:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefinding: lower the h264 typefinder's probability
|
|
A NEARLY_CERTAIN is absolutely not warranted given the kind
|
|
of things it checks for. Even a LIKELY is probably not entirely
|
|
appropriate.
|
|
|
|
2009-06-24 15:13:56 +0100 Jan Schmidt <jan.schmidt@sun.com>
|
|
|
|
* common:
|
|
Automatic update of common submodule
|
|
From f3bb51b to f810030
|
|
|
|
2009-06-24 09:48:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/pbutils/descriptions.c:
|
|
pbutils: add description for multipart
|
|
So we get slightly nicer error messages when multipartdemux is missing.
|
|
|
|
2009-06-23 18:07:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/adder/gstadder.c:
|
|
adder: only unflush when we flushed before
|
|
Ass suggested by Stefan Kost:
|
|
Keep track of when the sinkpad was set to flushing and unflush the pad when an
|
|
upstream flushing seek failed.
|
|
|
|
2009-06-23 15:10:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/playback/gsturidecodebin.c:
|
|
uridecodebin: fix leak when the source fails to change state
|
|
|
|
2009-06-23 12:40:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/subparse/gstssaparse.c:
|
|
ssaparse: avoid leaking all buffers
|
|
|
|
2009-06-22 22:18:03 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* tests/check/elements/adder.c:
|
|
adder: test seek handling in adder
|
|
This tests seeking on an adder that has a normal and a live source connected.
|
|
Wheter the current behavior is the desired one needs to be discussed still
|
|
(see #586033)
|
|
|
|
2009-06-22 16:17:10 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* sys/ximage/ximagesink.c:
|
|
* sys/xvimage/xvimagesink.c:
|
|
x(v)imagesink: pass the xwindow along to not look at the yet unset var.
|
|
When we call this from xwindow_new, x(v)imagesink->xwindow is not yet set.
|
|
|
|
2009-06-22 11:40:33 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* sys/ximage/ximagesink.c:
|
|
* sys/ximage/ximagesink.h:
|
|
* sys/xvimage/xvimagesink.c:
|
|
* sys/xvimage/xvimagesink.h:
|
|
x(v)imagesink: catch tags and show title in own window
|
|
Refactor the code that sets the window title. Catch tag-events and use title
|
|
metadata for the window title.
|
|
|
|
2009-06-21 19:42:15 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/audiotestsrc/gstaudiotestsrc.c:
|
|
audiotestsrc: Name gaussian noise "gaussian-noise" instead of just "gaussian"
|
|
Also make all the function arrays constant.
|
|
|
|
2009-06-21 12:27:37 +0200 Kipp Cannon <kcannon@ligo.caltech.edu>
|
|
|
|
* gst/audiotestsrc/gstaudiotestsrc.c:
|
|
* gst/audiotestsrc/gstaudiotestsrc.h:
|
|
audiotestsrc: Add support for generating gaussian white noise
|
|
This patch adds support for stationary white Gaussian noise.
|
|
The Box-Muller algorithm is used to generate pairs of independent
|
|
normally-distributed random numbers.
|
|
Fixes bug #586519.
|
|
|
|
2009-06-20 23:46:28 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* gst/ffmpegcolorspace/imgconvert.c:
|
|
* gst/ffmpegcolorspace/imgconvert_template.h:
|
|
ffmpegcolorspace: Fix NV12 and NV21 transformations
|
|
Fix some stride problems, fix the nv12 to nv21 direct transformation,
|
|
and implement a direct conversion to yuv444 to save CPU.
|
|
|
|
2009-06-20 22:36:21 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* gst/videotestsrc/videotestsrc.c:
|
|
videotestsrc: Fix NV12 painting for odd strides/heights
|
|
|
|
2009-06-19 22:16:43 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* ext/cdparanoia/gstcdparanoiasrc.c:
|
|
cdparanoia: run-time license is LGPL now that we require cdparanoia 0.10.2
|
|
cdparanoia has an LGPL v2.1 license since 0.10.1 and we now require 0.10.2.
|
|
Finally fixes #531035.
|
|
|
|
2009-06-19 21:25:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* ext/cdparanoia/gstcdparanoiasrc.c:
|
|
cdparanoia: try to guess a good cache size if it's set to -1
|
|
Try to guess from the paranoia-mode setting whether playback or
|
|
ripping is wanted, and use a smaller cache size if we're likely
|
|
to be doing playback, to avoid a long startup delay. Since this
|
|
was the value used in older cdparanoia versions, it should be
|
|
fine in any case. See #586331.
|
|
|
|
2009-06-19 11:27:40 +1000 Jonathan Matthew <jonathan@d14n.org>
|
|
|
|
* configure.ac:
|
|
* ext/cdparanoia/gstcdparanoiasrc.c:
|
|
* ext/cdparanoia/gstcdparanoiasrc.h:
|
|
cdparanoia: expose cache size setting
|
|
This setting was added in cdparanoia 10.2. The default value is good
|
|
for audio extraction, but lower values (previous versions of cdparanoia
|
|
used 150) are better for realtime playback.
|
|
Fixes #586331.
|
|
|
|
2009-06-19 17:43:03 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
|
|
|
|
* gst-plugins-base.spec.in:
|
|
Make build of schro plugin conditional
|
|
|
|
2009-06-19 15:52:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/rtp/gstbasertppayload.c:
|
|
* gst-libs/gst/rtp/gstbasertppayload.h:
|
|
* win32/common/libgstrtp.def:
|
|
basertppayload: add support for bufferlists
|
|
Based on patch from Ognyan Tonchev.
|
|
See #585559
|
|
|
|
2009-06-19 15:33:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstrtpbuffer.c:
|
|
rtpbuffer: use new convenience functions
|
|
New core convenience functions makes the list getters and setters trivial.
|
|
Maybe even too trivial...
|
|
|
|
2009-06-18 19:07:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* win32/common/libgstrtp.def:
|
|
defs: add new symbol to win32 defs file
|
|
Based on patches by Ognyan Tonchev.
|
|
See #585559
|
|
|
|
2009-06-18 19:04:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/rtp/gstrtpbuffer.c:
|
|
rtp: cleanups, add _list_get_seq() too
|
|
Clean up the docs a little.
|
|
Add missing _list_get_seq method.
|
|
Add new symbols to the docs
|
|
|
|
2009-06-18 18:47:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstrtpbuffer.c:
|
|
* win32/common/libgstrtp.def:
|
|
rtp: cleanups
|
|
Add Since tags to docs
|
|
Move some code around
|
|
Add win32 symbols
|
|
|
|
2009-06-18 17:46:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstrtpbuffer.c:
|
|
* gst-libs/gst/rtp/gstrtpbuffer.h:
|
|
* tests/check/libs/rtp.c:
|
|
rtp: add bufferlist support
|
|
|
|
2009-06-18 18:03:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstrtpbuffer.c:
|
|
rtp: pass data to macros instead of GstBuffer
|
|
|
|
2009-06-18 17:42:10 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* win32/common/libgstrtsp.def:
|
|
win32: Add gst_rtsp_watch_queue_data() to the exports
|
|
Fix the tests by exporting the new symbol from the win32 dlls
|
|
|
|
2009-06-18 18:13:22 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* sys/xvimage/xvimagesink.c:
|
|
xvimagesink: appname might be NULL
|
|
Don't set title if appname is unknown.
|
|
|
|
2009-06-18 17:58:06 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* sys/xvimage/xvimagesink.c:
|
|
xvimagesink: set window title from application name
|
|
|
|
2009-06-09 19:14:00 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspurl.c:
|
|
rtsp: Made the parsing of the RTSP URL scheme more generic.
|
|
|
|
2009-06-15 13:58:26 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
rtsp: Added gst_rtsp_watch_queue_data().
|
|
gst_rtsp_watch_queue_data() is similar to gst_rtsp_watch_queue_message()
|
|
but allows for queuing any data block for writing (much like
|
|
gst_rtsp_connection_write() vs. gst_rtsp_connection_send().)
|
|
API: gst_rtsp_watch_queue_data()
|
|
|
|
2009-06-09 16:37:09 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Only extract the session ID from RTSP responses.
|
|
|
|
2009-06-09 19:06:57 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspurl.c:
|
|
rtsp: Added support for parsing IPv6 addresses in RTSP URLs.
|
|
|
|
2009-06-09 14:31:18 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Use getaddrinfo() to support both IPv4 and IPv6.
|
|
|
|
2009-06-17 15:37:53 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Improved base64 decoding in fill_bytes().
|
|
The base64 decoding in fill_bytes() expected the size of the read data to
|
|
be evenly divisible by four (which is true for the base64 encoded data
|
|
itself). This did not, however, take whitespace (especially line breaks)
|
|
into account and would fail the decoding if any whitespace was present.
|
|
|
|
2009-06-17 14:00:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
audiosrc: fix get_offset
|
|
When we need to jump to the most recently captured sample, jump to where the
|
|
next sample will be written instead of to some old data.
|
|
Fixes #581460
|
|
|
|
2009-06-17 13:18:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
audiosink: free the ringbuffer when going to NULL
|
|
Unparent and free the ringbuffer when going to NULL, like we do with the
|
|
audiosrc element. We can do this now because we correctly manage the time
|
|
jumping back to 0.
|
|
|
|
2009-06-17 13:17:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstaudiosink.c:
|
|
* gst-libs/gst/audio/gstaudiosrc.c:
|
|
audio: correctly handle short read/writes
|
|
|
|
2009-05-05 15:37:54 +0300 René Stadler <rene.stadler@nokia.com>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
baseaudiosrc: add some extra logging for buffer timestamps
|
|
|
|
2009-06-17 11:22:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/adder/gstadder.c:
|
|
adder: more seeking fixes.
|
|
When a seek failed upstream, make sure the adder sinkpad is set unflushing again
|
|
so that streaming can continue.
|
|
We only have a pending segment when we flushed.
|
|
Set the flush_stop_pending flag inside the appropriate locks and before we
|
|
attempt to perform the upstream seek.
|
|
Add some more comments.
|
|
Use the right lock to protect the flags in flush_stop.
|
|
See #585708
|
|
|
|
2009-06-17 07:24:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
decodebin2: Free iterator after removing all groups
|
|
|
|
2009-06-16 19:38:17 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/video/gstvideofilter.c:
|
|
videofilter: Add a default get_unit_size function
|
|
This returns the correct values for all formats that are handled by
|
|
GstVideoFormat and makes all the custom get_unit_size functions in
|
|
many elements unnecessary.
|
|
|
|
2009-06-16 18:57:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspdefs.c:
|
|
* gst-libs/gst/rtsp/gstrtspdefs.h:
|
|
rtsp: add Timestamp header field
|
|
fixes #585994
|
|
|
|
2009-06-16 18:15:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
playbin2: set smarter target state on uridecodebin
|
|
Set the target state of the newly added uridecodebins to somthing else that
|
|
PAUSED so that we keep their state in sync with the playsink state.
|
|
Fixes #585268
|
|
|
|
2009-06-16 18:13:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playsink: set the sink flag on the element
|
|
|
|
2009-06-16 18:09:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gsturidecodebin.c:
|
|
uridecodebin: add debug message
|
|
|
|
2009-06-16 14:05:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstaudiosink.c:
|
|
* gst-libs/gst/audio/gstaudiosrc.c:
|
|
audiosink, audiosrc: do the class_ref()s in the right class_init functions
|
|
Spotted by Philip Jägenstedt. Hopefully fixes #585970 for real.
|
|
|
|
2009-06-15 15:39:09 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstaudiosink.c:
|
|
* gst-libs/gst/audio/gstaudiosrc.c:
|
|
audiosink,audiosrc: ref the audio ring buffer class and type in class_init
|
|
Hack around thread-safety issues in GObject and our racy _get_type()
|
|
functions (we could easily fix the _get_type() functions, but we still
|
|
need to hack around the GObject class races until we require a newer
|
|
GLib version, I think).
|
|
|
|
2009-06-15 12:57:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
audiosrc: return FALSE when receiving a SEEK event
|
|
When receiving a seek event, return FALSE as we don't implement seeking.
|
|
|
|
2009-06-15 11:06:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* tests/examples/seek/seek.c:
|
|
Don't use deprecated GTK API
|
|
Fixes bug #585758.
|
|
|
|
2009-06-15 11:40:00 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/adder/gstadder.c:
|
|
adder: send flush_stop when seeking failed
|
|
At least do the fix to sent the flush_stop when seeking failed to ensure we
|
|
keep no pads flushing. before it was send when the seeking worked which is just
|
|
plain wrong and was not the intention.
|
|
|
|
2009-06-12 15:17:14 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Use a more consistent naming of GstRTSPRec variables.
|
|
|
|
2009-06-12 15:11:05 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
rtsp: Call message_sent() callback for all sent messages.
|
|
Previously the messages_sent() callback was only called for messages
|
|
which had a CSeq, which excluded all data messages. Instead of using the
|
|
CSeq as ID, use a simple index counter.
|
|
|
|
2009-06-14 22:13:41 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* ext/ogg/gstoggdemux.c:
|
|
* ext/theora/theoradec.c:
|
|
* ext/vorbis/vorbisdec.c:
|
|
oggdemux: post/send tags with the container-format tag
|
|
For this to work properly, theoradec and vorbisdec need to put
|
|
tag events received from upstream into the pending_events list
|
|
so they get pushed out after any newsegment event, not before.
|
|
|
|
2009-06-14 20:30:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* tests/examples/seek/scrubby.c:
|
|
* tests/examples/seek/seek.c:
|
|
* tests/old/examples/seek/cdplayer.c:
|
|
Don't use deprecated GTK API
|
|
Fixes bug #585758.
|
|
|
|
2009-06-12 16:31:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/adder/gstadder.c:
|
|
adder: send flush-stop earlier
|
|
When no flush-stop has been sent by upstream, we have to send one ourselves to
|
|
continue playback. Do this as soon as the collect function is called instead of
|
|
after we possibly pushed segment events (that got then flushed out)
|
|
|
|
2009-06-12 13:55:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/examples/seek/seek.c:
|
|
seek: add shuttle controls
|
|
|
|
2009-06-12 13:55:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/examples/seek/stepping2.c:
|
|
example: fix compile
|
|
|
|
2009-06-12 13:52:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/examples/seek/Makefile.am:
|
|
examples: build the stepping2 example
|
|
|
|
2009-06-12 13:52:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playsink: update for new step API
|
|
|
|
2009-06-12 13:22:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* ext/ogg/gstoggdemux.c:
|
|
oggdemux: do reverse seeks more accurate
|
|
For reverse seeking with the accurate flag set, try to be more precise by
|
|
seeking a little bit after the requested position.
|
|
|
|
2009-06-11 22:32:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* ext/ogg/gstogmparse.c:
|
|
* gst/subparse/gstssaparse.c:
|
|
* gst/subparse/gstssaparse.h:
|
|
* gst/subparse/gstsubparse.c:
|
|
* gst/subparse/gstsubparse.h:
|
|
subparse, ogmparse: post tags with GST_TAG_SUBTITLE_CODEC
|
|
Make subtitle parsers post a taglist with codec tags, so the application
|
|
knows what kind of subtitle a subtitle stream is. Fixes #576552.
|
|
|
|
2009-06-11 19:12:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstringbuffer.c:
|
|
ringbuffer: handle border cases in resampler
|
|
|
|
2009-06-11 13:28:20 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* common:
|
|
* docs/libs/Makefile.am:
|
|
* docs/plugins/Makefile.am:
|
|
docs: Update common. Use upload-doc.mak instead of upload.mak
|
|
|
|
2009-06-11 12:39:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstbasertppayload.c:
|
|
docs: fix typo
|
|
|
|
2009-06-11 12:17:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
baseaudiosink: reset accum when dropping samples
|
|
When we are resampling and we drop samples because we paused, reset the accum
|
|
counter because it's now invalid.
|
|
|
|
2009-06-11 11:16:15 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/interfaces/mixer.h:
|
|
* gst-libs/gst/video/gstbasevideodecoder.h:
|
|
docs: Fix a couple of warnings from the docs build.
|
|
|
|
2009-06-10 21:36:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/testchannels.c:
|
|
Don't include config.h multiple times when build audio testchannel app.
|
|
Fixes build problem on win32 (#585075).
|
|
|
|
2009-06-10 16:56:51 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
* gst/playback/gsturidecodebin.c:
|
|
playbin2/uridecodebin: Fix connection-speed propagation
|
|
uridecodebin expects the passed connection-speed value in kbps, so we
|
|
need to divide the value stored in bps by 1000. Also, lower the upper
|
|
limit on the properties to the value that we can actually store in our
|
|
internal guint (which is plenty high enough)
|
|
|
|
2009-06-10 14:37:36 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/subparse/gstsubparse.c:
|
|
* tests/check/elements/subparse.c:
|
|
subparse: recognise more subrip timestamp variants
|
|
Be even less restrictive in what we accept for .srt timestamps when
|
|
typefinding and parsing subrip subtitles and add a unit test for
|
|
the 'new' format. Fixes #585197.
|
|
|
|
2009-06-09 22:00:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtsptransport.h:
|
|
rtsp: add some more docs
|
|
|
|
2009-06-09 18:24:55 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspmessage.c:
|
|
rtsp: Avoid a compiler warning.
|
|
|
|
2009-06-09 18:23:28 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspdefs.h:
|
|
rtsp: Updated documentation for GstRTSPResult.
|
|
Moved GST_RTSP_ELAST to be last in the documentation to match the actual
|
|
enum values.
|
|
|
|
2009-05-20 17:30:23 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* autogen.sh:
|
|
autogen: remove -Wno-portability from here
|
|
as it is in configure.ac now.
|
|
|
|
2009-06-09 16:28:20 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Plug a memory leak.
|
|
Free memory related to any partially read and/or written RTSP messages.
|
|
|
|
2009-06-09 12:09:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
baseaudiosink: no need to cause discont when clipping
|
|
Remove the discont-when-clipping hack now that basesink provides us with
|
|
correctly clipped samples when stepping.
|
|
|
|
2009-06-08 17:26:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
audiosink: don't align when we clip
|
|
Don't align samples when they were clipped. Not entirely correct but better than
|
|
nothing for now.
|
|
|
|
2009-06-08 16:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/examples/seek/.gitignore:
|
|
* tests/examples/seek/stepping2.c:
|
|
examples: add stepping example in PLAYING
|
|
Add stepping example in PLAYING, audio is a bit distorted because basesink does
|
|
not provide good clipping info yet.
|
|
|
|
2009-06-08 10:25:00 +0200 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst-libs/gst/pbutils/descriptions.c:
|
|
pbutils: Add description for hdv/aux-* formats.
|
|
|
|
2009-06-07 22:20:33 +0400 LRN <lrn1986@gmail.com>
|
|
|
|
* ext/schroedinger/Makefile.am:
|
|
Added libgstbase to schro's LIBADD
|
|
Fixes #585079
|
|
|
|
2009-06-06 02:15:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/tag/gstid3tag.c:
|
|
libgsttag: don't extract genres from empty ID3v1 tags
|
|
If we don't have any other info, don't try to interpret the
|
|
genre field. In particular we don't want to interpret a genre
|
|
of 0 as 'Blues' if no other fields are set and the entire tag
|
|
is just empty.
|
|
|
|
2009-06-05 18:13:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
decodebin2: make sure varargs are of right type
|
|
Explicitly cast the variables to g_object_set to their right types.
|
|
|
|
2009-06-05 16:49:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
decodebin2: increase stream probing queues
|
|
When we are probing for streams, we want to set the queue size in such a way
|
|
that we can scan a maximum amount of data without consuming too much memory.
|
|
Therefore, remove the time limit on the queue and only stop scanning after 2MB
|
|
of data.
|
|
See #584104.
|
|
|
|
2009-06-05 14:06:17 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Fixed a typo.
|
|
|
|
2009-06-05 14:05:54 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Remove an unused variable.
|
|
|
|
2009-06-05 13:59:14 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Removed duplicate initialization of conn->writefd.
|
|
|
|
2009-06-05 13:55:08 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Use #defined status codes.
|
|
|
|
2009-06-05 13:53:29 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Correct gen_tunnel_reply().
|
|
Prevent gen_tunnel_reply() from generating an incomplete response
|
|
in case an error response code is given.
|
|
|
|
2009-06-05 10:57:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* configure.ac:
|
|
* win32/common/_stdint.h:
|
|
* win32/common/config.h:
|
|
* win32/common/video-enumtypes.c:
|
|
configure: remove AC_C_INLINE which is not needed and causes problems with MSVC
|
|
See #584835. Also update win32 files while we're at it.
|
|
|
|
2009-06-04 08:57:24 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
playbin2: API: Add {audio,video,text}-tags-changed signals
|
|
Fixes bug #584686.
|
|
|
|
2009-06-03 20:42:39 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* ext/vorbis/vorbisdec.c:
|
|
vorbisdec: don't put invalid bitrate values into the taglist
|
|
Bitrates are stored as 32-bit signed integers in the vorbis
|
|
identification headers, but seem to be read incorrectly,
|
|
namely as unsigned 32-bit integers, into the vorbis structure
|
|
members which are of type long, which makes our check for
|
|
values <= 0 fail with files that put -1 in there for unset
|
|
values.
|
|
|
|
2009-06-03 15:52:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/examples/seek/.gitignore:
|
|
ignore: add new stepping app to ignore
|
|
|
|
2009-06-03 15:31:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/examples/seek/Makefile.am:
|
|
* tests/examples/seek/stepping.c:
|
|
examples: add stepping example.
|
|
Add an example of using playbin2 and frame stepping to simulate variable rate
|
|
playback based on a sine wave.
|
|
|
|
2009-06-03 12:45:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
* gst/playback/gstplaysink.h:
|
|
playbin2: also set custom text and subp sinks
|
|
Set the custom subpicture and text sinks along with the custom audio and video
|
|
sinks when needed.
|
|
Fix a little docs blurb too.
|
|
|
|
2009-06-02 12:10:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
rtsp: add G_LIKELY because we can
|
|
|
|
2009-06-02 09:53:05 +0200 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefindfunctions: Fix caps for ogg typefinder.
|
|
|
|
2009-05-29 11:10:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
docs: remove some cruft from -sections.txt file
|
|
|
|
2009-06-01 11:31:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
* tests/examples/seek/seek.c:
|
|
add framestepping to playbin2 and seek
|
|
|
|
2009-06-01 09:59:22 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Avoid compiler warnings with -Wextra.
|
|
|
|
2009-06-01 09:58:27 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
rtsp: Include gst/gstconfig.h to make sure GST_PADDING is defined.
|
|
|
|
2009-06-01 09:43:04 +0200 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/sdp/gstsdpmessage.c:
|
|
sdp: Remove an unused variable.
|
|
|
|
2009-05-30 14:17:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/ffmpegcolorspace/imgconvert.c:
|
|
* gst/ffmpegcolorspace/imgconvert_template.h:
|
|
ffmpegcolorspace: Add a lot more conversions from/to 16 bit grayscale
|
|
|
|
2009-05-29 00:09:15 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
playbin2: Have playbin recognise PGS subpicture streams
|
|
Recognise PGS subpicture streams and connect them to the SPU pad
|
|
in playsink. Unfortunately this fails badly with negotiation errors
|
|
if the SPU is not recent enough to support the stream. I'm not sure
|
|
how to add format negotiation in yet.
|
|
|
|
2009-05-21 23:11:29 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
* gst/playback/gsturidecodebin.c:
|
|
decodebin/uridecodebin: Recognise subpicture/x-pgs pads and output them.
|
|
|
|
2009-05-28 20:37:59 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: fix volume handling for audio sinks without "volume" property
|
|
When using an audio sink without a "volume" property, volume control
|
|
would only work for the first song. For the next song, we'd try to
|
|
re-use the existing audio chain, but inadvertently set chain->volume
|
|
to NULL instead of to the existing volume element.
|
|
|
|
2009-05-28 17:05:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: cosmetic change to avoid unnecessary line breaks
|
|
Looks nicer and works around gst-indent silliness.
|
|
|
|
2009-05-28 17:21:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: don't lose the ref to the volume element
|
|
Only release the ref to the volume element when it is controled by a sink. For
|
|
software volume we never have to fear that it will change.
|
|
|
|
2009-05-28 15:21:42 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: actually use configured audio/video sinks
|
|
playbin2 inadvertently used autoaudiosink and autovideosink up to now,
|
|
since it would overwrite the sinks configured via the "audio-sink"
|
|
and "video-sink" properties with the stream-specific group sinks when
|
|
configuring the outputs. Those are usually NULL however, so that would
|
|
overwrite the configured sinks with NULL which makes playbin2 then
|
|
default to the auto sinks. Fix this by keeping a reference to each
|
|
configured sink in playbin2 and setting up the right sinks depending
|
|
on whether there is a stream-specific sink or not.
|
|
Fixes #584020.
|
|
|
|
2009-05-27 17:37:38 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* tests/examples/seek/seek.c:
|
|
seek: add volume label and sync with sink volume
|
|
Look at the volume and have the pulsemixer open at same time. Unfortunately
|
|
playbin2 does not emit notify on volume right, so this polls for now.
|
|
|
|
2009-05-27 18:12:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
decodebin2: remove leftover elements
|
|
Remove all of the elements inside decodebin2 when goint to READY and NULL.
|
|
Makes decodebin2 reusable.
|
|
Fixes #583750
|
|
|
|
2009-05-27 15:36:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2; release refs to volume/mute properties
|
|
Release the refs to the volume and mute property elemens before setting the
|
|
child elements to READY or NULL.
|
|
Fixes #583318
|
|
|
|
2009-05-27 12:10:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/gdp/gstgdppay.c:
|
|
gdppay: set caps on outgoing buffers
|
|
Set caps on outgoing buffers because NULL caps confuse basetransform.
|
|
Fixes #583867
|
|
|
|
2009-05-27 11:08:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/netbuffer/gstnetbuffer.c:
|
|
netbuffer: also note the order of IP4 addresses
|
|
IP4 addresses are also stored in network byte order. Make a note of this in the
|
|
docs.
|
|
|
|
2009-05-26 22:43:34 +0200 Alessandro Decina <alessandro.d@gmail.com>
|
|
|
|
* ext/theora/theoraparse.c:
|
|
theoraparse: fix assertions in make_granulepos when using the new theora granulepos mapping. Fixes #583903.
|
|
|
|
2009-05-26 11:13:35 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
Revert "rtspconnection: don't use GLib-2.16 API, we require only 2.14"
|
|
This reverts commit 418760cf740332c12c3fd9cf3244af134fa9534b.
|
|
We now require GLib 2.16.
|
|
|
|
2009-05-26 15:18:09 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* common:
|
|
Update common
|
|
|
|
2009-05-26 15:37:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/netbuffer/gstnetbuffer.c:
|
|
netbuffer: document that the port is network order
|
|
Document the fact that we store the port number in network order in
|
|
GstNetAddress and that the caller should byteswap appropriately.
|
|
|
|
2009-05-26 15:23:45 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/gstvideoscale.c:
|
|
* gst/videoscale/vs_4tap.c:
|
|
* gst/videoscale/vs_4tap.h:
|
|
* gst/videoscale/vs_image.c:
|
|
* gst/videoscale/vs_image.h:
|
|
* gst/videoscale/vs_scanline.c:
|
|
* gst/videoscale/vs_scanline.h:
|
|
videoscale: Add support for 16 bit grayscale in native endianness
|
|
|
|
2009-05-26 14:58:28 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/ffmpegcolorspace/avcodec.h:
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
|
|
* gst/ffmpegcolorspace/imgconvert.c:
|
|
ffmpegcolorspace: Add support for 16 bit grayscale in little/big endian
|
|
|
|
2009-05-26 14:38:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videotestsrc/videotestsrc.c:
|
|
* gst/videotestsrc/videotestsrc.h:
|
|
videotestsrc: Add support for 16 bit grayscale in native endianness
|
|
|
|
2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
|
|
|
|
add can-activate-pull property to baseaudiosink
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
|
|
to baseaudiosink.
|
|
|
|
2009-05-26 13:14:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* ext/ogg/gstoggdemux.c:
|
|
oggdemux: fix boundary case for seeking.
|
|
When we have exactly 0 bytes left to search, make sure we stop instead of going
|
|
into an infinite loop.
|
|
|
|
2009-05-26 11:11:03 +0200 Bastien Nocera <hadess at hadess.net>
|
|
|
|
* gst-libs/gst/cdda/Makefile.am:
|
|
* gst-libs/gst/cdda/gstcddabasesrc.c:
|
|
* gst-libs/gst/cdda/sha1.c:
|
|
* gst-libs/gst/cdda/sha1.h:
|
|
cddabasesrc: Remove copy of sha1 digest
|
|
Remove our copy of sha1 digest now that we depend on glib 2.16.
|
|
Fixes #536313
|
|
|
|
2009-05-25 17:54:01 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
|
|
|
|
* gst-plugins-base.spec.in:
|
|
Update spec file
|
|
|
|
2009-05-23 00:33:04 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/video/gstbasevideodecoder.c:
|
|
* gst-libs/gst/video/gstbasevideoparse.c:
|
|
* gst-libs/gst/video/gstbasevideoutils.c:
|
|
* gst-libs/gst/video/gstbasevideoutils.h:
|
|
* win32/common/libgstvideo.def:
|
|
video: don't expose internal gst_adapter_get_buffer() helper function
|
|
If it's really needed it should go into GstAdapter in core.
|
|
|
|
2009-05-22 21:29:51 -0700 David Schleef <ds@schleef.org>
|
|
|
|
* gst-libs/gst/video/gstbasevideodecoder.c:
|
|
basevideo: Fix memleak
|
|
|
|
2009-05-22 21:27:58 -0700 David Schleef <ds@schleef.org>
|
|
|
|
* ext/schroedinger/gstschrodec.c:
|
|
* ext/schroedinger/gstschroparse.c:
|
|
schro: Fix usage of adapter_masked_scan_uint32
|
|
Because *somebody* changed the API without telling me.
|
|
|
|
2009-05-22 21:25:06 -0700 David Schleef <ds@schleef.org>
|
|
|
|
* ext/schroedinger/gstschro.c:
|
|
schro: Change package name to GST_PACKAGE_NAME
|
|
|
|
2009-05-22 17:34:10 -0700 David Schleef <ds@schleef.org>
|
|
|
|
* gst-libs/gst/video/gstbasevideoencoder.c:
|
|
basevideo: Add preset interface to encoder
|
|
|
|
2009-05-22 17:31:14 -0700 David Schleef <ds@schleef.org>
|
|
|
|
* gst/audioresample/gstaudioresample.c:
|
|
Run liboil benchmark multiple times
|
|
The statistics function requires multiple runs, otherwise
|
|
it causes a divide by zero error.
|
|
|
|
2009-05-22 19:36:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* m4/gst-fionread.m4:
|
|
m4: fix 'suspicious cache value' warning for gst-fionread.m4
|
|
.. here as well (should really be moved to common, but I'm too lazy).
|
|
|
|
2009-05-22 17:41:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* ext/vorbis/vorbisdec.c:
|
|
vorbisdec: detect and report errors better
|
|
Check the return values of a couple more libvorbis functions and post an error
|
|
when something is wrong instead of continuing and crashing.
|
|
|
|
2009-05-22 15:49:14 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: fix initial volume and mute handling
|
|
Use two flags to remember volume/mute changes at times when we don't have the
|
|
audiochain yet (e.g. construction). Only set values when they were actualy
|
|
changed. This makes pulseaudio's stream restore functional.
|
|
|
|
2009-05-22 10:19:51 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* common:
|
|
Automatic update of common submodule
|
|
From d3a8fab to 888e0a2
|
|
|
|
2009-05-22 09:03:22 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* win32/common/libgstvideo.def:
|
|
win32: Remove gst_adapter_masked_scan_uint32 from the exports
|
|
|
|
2009-05-21 10:48:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
audiosink: improve debug message
|
|
|
|
2009-05-19 18:10:55 -0700 Michael Smith <msmith@songbirdnest.com>
|
|
|
|
* gst-libs/gst/tag/gstid3tag.c:
|
|
gstid3tag: Don't extract a track number unless present.
|
|
In ID3v1, a track number is present only if byte 125 is null AND
|
|
byte 126 is non-null. If the track number is not present, don't add
|
|
a track number tag with value 0.
|
|
|
|
2009-05-20 00:48:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/video/gstbasevideoutils.c:
|
|
* gst-libs/gst/video/gstbasevideoutils.h:
|
|
videoutils: remove adapter methods
|
|
Remove adapter methods now that they are in core.
|
|
|
|
2009-05-20 00:42:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* win32/common/libgstvideo.def:
|
|
defs: add new symbols
|
|
|
|
2009-05-19 17:47:34 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* configure.ac:
|
|
autogen: pass -Wno-portability to automake to suppress warnings
|
|
GNU make is needed.
|
|
|
|
2009-05-19 02:28:20 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* docs/libs/.gitignore:
|
|
gitignore: remove bogus *.sgml wildcard - these files are tracked in git
|
|
|
|
2009-05-19 18:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/tcp/gsttcpclientsrc.c:
|
|
tcpclientsrc: this is not a live source
|
|
Don't mark us as a live source because we are not.
|
|
|
|
2009-05-19 18:41:02 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/adder/gstadder.c:
|
|
adder: only send flush_stop when seek failed
|
|
This is still not the ultimate fix. Added some comment to explain the troubles.
|
|
|
|
2009-05-19 17:17:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
audiosink: return the return value of wait_preroll
|
|
Return the value that _wait_preroll() returned instead of always WRONG_STATE.
|
|
|
|
2009-05-19 16:45:56 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/adder/gstadder.c:
|
|
* gst/adder/gstadder.h:
|
|
adder: send flush_stop to match flush_start
|
|
Adder was relying that something else sends a flush stop. When using adder with
|
|
a livesource it was not getting a flush_stop and thus all pads downstream where
|
|
keept flushing. Mark a pending flush_stop and send it when we are working on
|
|
the new segment back in the streaming thread.
|
|
|
|
2009-05-19 16:02:44 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* tests/examples/seek/seek.c:
|
|
seek: ui improvements
|
|
Repaint the window black on expose, as this looks nicer when resizing or using
|
|
the expander. Also show time after slider, as this saves a whole line (nice on
|
|
small displays).
|
|
|
|
2009-04-29 18:36:17 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/playback/gstdecodebin.c:
|
|
decodebin: use iterators instead of list
|
|
The list api is deprecated. Use threadsafe iterators instead.
|
|
|
|
2009-05-19 15:35:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gsturidecodebin.c:
|
|
uridecodebin: configure caps on decodebin2
|
|
Implement the caps property by setting the configured caps on new decodebin2
|
|
objects.
|
|
Fixes #582749
|
|
|
|
2009-05-19 15:34:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
decodebin2: avoid some _caps_ref in some cases
|
|
Only mess with the caps refcount when we configure different caps.
|
|
|
|
2009-05-19 15:27:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gsturidecodebin.c:
|
|
uridecodebin: fix potential caps leak
|
|
Free the user-configured caps in finalize.
|
|
|
|
2009-05-19 15:20:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gsturidecodebin.c:
|
|
uridecodebin: add queue after cdda://
|
|
Add a queue2 after the raw output pads of certain sources such as those for uris
|
|
like cdda://
|
|
No tuning of the queue is done yet as the defaults seem to work fine for me.
|
|
Fixes #582528
|
|
|
|
2009-05-19 12:45:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* ext/ogg/gstoggdemux.c:
|
|
oggdemux: don't loop when at EOS
|
|
When we try to read the last page, don't try to read past the upper boundary, as
|
|
this might cause endless loops.
|
|
See #582942
|
|
|
|
2009-05-19 11:20:19 +0200 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst/audioresample/gstaudioresample.c:
|
|
audioresample: Don't drain remaining buffers after a flush.
|
|
If we were resetted (due to a flush), we can not drain the remaining
|
|
buffers since they would be pushed before a valid new newsegment event.
|
|
|
|
2009-05-18 22:29:07 -0700 Michael Smith <msmith@syncword.(none)>
|
|
|
|
* ext/theora/theoradec.c:
|
|
theoradec: for 4:2:2, use Y42B (planar) rather than a packed format.
|
|
|
|
2009-05-19 01:13:34 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/adder/gstadder.c:
|
|
adder: add more logging and return value checking
|
|
|
|
2009-05-19 01:11:45 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/adder/gstadder.c:
|
|
adder: handle the return value from iterator_fold
|
|
|
|
2009-05-19 01:03:44 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/adder/gstadder.c:
|
|
adder: use the pad in logging as objects
|
|
Helps to differenciate between source and sinks pads.
|
|
|
|
2009-04-21 22:54:19 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* tests/examples/seek/seek.c:
|
|
seek: use parser for mp3 and rename variable
|
|
|
|
2009-05-18 11:08:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/examples/seek/seek.c:
|
|
seek: add playbin2 options in expander
|
|
Add the playbin2 stream selection options inside an expander to preserve some
|
|
space on screen.
|
|
|
|
2009-02-10 15:29:10 -0800 David Schleef <ds@schleef.org>
|
|
|
|
* gst/videotestsrc/videotestsrc.c:
|
|
videotestsrc: Add support for v210 and v216 formats
|
|
|
|
2009-05-15 16:21:15 -0700 David Schleef <ds@schleef.org>
|
|
|
|
* gst-libs/gst/video/gstbasevideocodec.c:
|
|
* gst-libs/gst/video/gstbasevideodecoder.c:
|
|
* gst-libs/gst/video/gstbasevideoencoder.c:
|
|
* gst-libs/gst/video/gstbasevideoparse.c:
|
|
video: remove // comments
|
|
|
|
2009-05-15 16:18:18 -0700 David Schleef <ds@schleef.org>
|
|
|
|
* gst-libs/gst/video/video.c:
|
|
* gst-libs/gst/video/video.h:
|
|
video: Add Y444, v210, v216 formats
|
|
|
|
2009-05-15 16:12:37 -0700 David Schleef <ds@schleef.org>
|
|
|
|
* configure.ac:
|
|
* ext/Makefile.am:
|
|
* ext/schroedinger/Makefile.am:
|
|
* ext/schroedinger/gstschro.c:
|
|
* ext/schroedinger/gstschrodec.c:
|
|
* ext/schroedinger/gstschroenc.c:
|
|
* ext/schroedinger/gstschroparse.c:
|
|
* ext/schroedinger/gstschroutils.c:
|
|
* ext/schroedinger/gstschroutils.h:
|
|
schro: Move schro plugin from Schroedinger
|
|
Previous history is in Schroedinger. Depends on, and is an example
|
|
of using, GstBaseVideo* base classes.
|
|
Code was reindented, and an #ifdef HAVE_ENCODER removed.
|
|
|
|
2009-05-15 10:23:08 -0700 David Schleef <ds@schleef.org>
|
|
|
|
* gst-libs/gst/video/Makefile.am:
|
|
* gst-libs/gst/video/gstbasevideocodec.c:
|
|
* gst-libs/gst/video/gstbasevideocodec.h:
|
|
* gst-libs/gst/video/gstbasevideodecoder.c:
|
|
* gst-libs/gst/video/gstbasevideodecoder.h:
|
|
* gst-libs/gst/video/gstbasevideoencoder.c:
|
|
* gst-libs/gst/video/gstbasevideoencoder.h:
|
|
* gst-libs/gst/video/gstbasevideoparse.c:
|
|
* gst-libs/gst/video/gstbasevideoparse.h:
|
|
* gst-libs/gst/video/gstbasevideoutils.c:
|
|
* gst-libs/gst/video/gstbasevideoutils.h:
|
|
video: Copy BaseVideo classes from Schroedinger
|
|
|
|
2009-05-15 23:05:45 +0200 Arnout Vandecappelle <arnout@mind.be>
|
|
|
|
* gst/tcp/gstmultifdsink.c:
|
|
multifdsink: add num-fds property
|
|
multifdsink::num-fds
|
|
|
|
2009-05-15 20:36:29 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/pbutils/descriptions.c:
|
|
pbutils: add descriptions for 3GP, JPEG 2000 and Motion JPEG 2000
|
|
|
|
2009-05-14 11:44:27 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/vorbis/vorbisenc.c:
|
|
vorbisenc: Implement Preset interface
|
|
|
|
2009-05-14 11:43:07 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/theora/theoraenc.c:
|
|
theoraenc: Implement Preset interface
|
|
|
|
2009-05-14 11:41:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/ogg/gstoggmux.c:
|
|
oggmux: Implement Preset interface
|
|
|
|
2009-05-14 21:37:22 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: Fix cdda:// playback
|
|
Don't send async-start when the playsink has already been configured
|
|
before changing state.
|
|
|
|
2009-05-14 01:31:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* configure.ac:
|
|
configure: require core CVS for gst_adapter_prev_timestamp()
|
|
which is used in the libvisual plugin.
|
|
|
|
2009-04-22 18:34:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* AUTHORS:
|
|
AUTHORS: fix my email
|
|
|
|
2009-04-22 18:35:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstaudioclock.c:
|
|
audioclock: make our internal time monotonic
|
|
Make the internal time increase monotonically.
|
|
|
|
2009-05-13 19:27:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* ext/libvisual/visual.c:
|
|
visual: remove next_ts variable
|
|
We can remove the next_ts variable as we don't use it anymore.
|
|
|
|
2009-05-13 19:24:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* ext/libvisual/visual.c:
|
|
visual: use new adapter timestamp code
|
|
Use the new adapter timestamp tracking code to make things easier and produce
|
|
vastly better output timestamps.
|
|
|
|
2009-05-13 01:35:07 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* po/Makevars:
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/bg.po:
|
|
* po/ca.po:
|
|
* po/cs.po:
|
|
* po/da.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/es.po:
|
|
* po/fi.po:
|
|
* po/fr.po:
|
|
* po/hu.po:
|
|
* po/id.po:
|
|
* po/it.po:
|
|
* po/ja.po:
|
|
* po/lt.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/pl.po:
|
|
* po/pt_BR.po:
|
|
* po/ru.po:
|
|
* po/sk.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
* po/zh_CN.po:
|
|
po: avoid conflicts of local *.po files with files in git
|
|
Make it so that filenames and line numbers are only stored in the *.pot file
|
|
(which is not in git), but not in the individual *.po files. This information
|
|
is hardly useful for translators in our case, and it should avoid the constant
|
|
conflicts of local *.po files with the ones in git which are caused by the
|
|
source files changing and the line numbers being updated. This commit might
|
|
cause one last merge conflict for you, which you can work around with
|
|
"git checkout po/*.po" before merging or pulling. After that there should
|
|
(hopefully) not be any more local modifications of these files (unless
|
|
someone committed additions or changes to translated strings and the
|
|
*.po files haven't been updated yet, that is).
|
|
|
|
2009-05-12 23:51:08 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* tests/check/elements/.gitignore:
|
|
* tests/check/elements/audioresample.c:
|
|
tests: fix audioresample unit test on big endian architectures
|
|
Don't hardcode endianness=1234 in the filtercaps, it will cause
|
|
pad link failures which will result in the test timing out.
|
|
|
|
2009-05-12 17:18:37 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/audiotestsrc/gstaudiotestsrc.c:
|
|
audiotestsrc: fix broken enum nick - it should have a hyphen
|
|
The enum nick should be 'sine-table', not 'sine table'. Technically this is
|
|
an API/ABI change I guess, but anyone who was using this and didn't report
|
|
it deserves this.
|
|
|
|
2009-05-01 01:04:48 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/audiotestsrc/gstaudiotestsrc.c:
|
|
audiotestsrc: seek to the requested byte offset, not the expected byte offset
|
|
|
|
2009-05-01 01:03:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/audiotestsrc/gstaudiotestsrc.c:
|
|
* gst/audiotestsrc/gstaudiotestsrc.h:
|
|
audiotestsrc: support more than just one channel
|
|
|
|
2009-05-12 15:52:41 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/interfaces/propertyprobe.h:
|
|
propertyprobe: Fix typo in the docs
|
|
|
|
2009-05-12 12:17:55 +0100 Christian Schaller <christian.schaller@collabora.co.uk>
|
|
|
|
* ext/ogg/gstoggmux.c:
|
|
* ext/theora/theora.c:
|
|
* ext/vorbis/vorbis.c:
|
|
Add ranks to the Oggmuxer, Vorbis encoder and Theora encoder
|
|
|
|
2009-04-30 16:37:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/videorate/gstvideorate.c:
|
|
* gst/videorate/gstvideorate.h:
|
|
videorate: handle invalid timestamps better
|
|
Handle buffers with -1 timestamps better by keeping track of the en time of the
|
|
previous buffer and assuming the -1 timestamp buffer goes right after the
|
|
previous one.
|
|
when we have two buffers that are equally good, output the oldest buffer once to
|
|
minimize latency.
|
|
don't try to calculate latency when the input framerate is unknown.
|
|
|
|
2009-04-28 11:37:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* ext/ogg/gstoggmux.c:
|
|
oggmux: small debug statement in DISCONT
|
|
|
|
2009-04-28 11:24:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* ext/ogg/gstoggdemux.c:
|
|
* ext/ogg/gstoggdemux.h:
|
|
oggdemux: fix abuse of ogg API, handle broken oggs
|
|
When we feed the ogg sync layer, we need to feed it contiguous data even if the
|
|
sync layer did not consume all of it yet. This makes sure that it always finds
|
|
the next page even for more corrupted files. Use a different read_offset for
|
|
this purpose. since we now keep track of the sync layer, we don't have to reset
|
|
after finding a start of a page.
|
|
Add some more debug info for the error paths.
|
|
Only reset the sync layer when we perform a seek operation.
|
|
Avoid failure when the next chain has no bos pages but instead simply ignore it.
|
|
when we receive unknown page serial numbers mid stream, don't fail but post a
|
|
warning and hope that we get back on track later.
|
|
Fixes #579642
|
|
|
|
2009-04-30 16:41:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
decodebin2: make subpictures a raw output format
|
|
Subpictures are a raw format, we want those pads exposed so that playbin2 can do
|
|
the subpicture mixing.
|
|
|
|
2009-04-27 10:15:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstbasertppayload.c:
|
|
* gst-libs/gst/rtp/gstbasertppayload.h:
|
|
rtpdepay: add some more comments
|
|
|
|
2009-04-17 10:54:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstaudioclock.c:
|
|
audioclock: make sure values are ever increasing
|
|
|
|
2009-05-05 17:17:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: make fallback identity silent
|
|
Set the signal-handoffs to FALSE and silent to TRUE for the fallback identity
|
|
element so that it consumes less CPU.
|
|
|
|
2009-04-17 10:57:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: handle custom audiosinks differently
|
|
Keep track of the autoplugged custom sinks and configure them in the playsink
|
|
element when we have collected all streams.
|
|
Also make sure that we only select one custom sink.
|
|
When unreffing the internal sink, we don't need to change the state to NULL.
|
|
|
|
2009-05-12 10:36:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
* gst/playback/gstplaysink.c:
|
|
* gst/playback/gstplaysink.h:
|
|
playbin2: unify custom sink get/set functions
|
|
Use one function to set/get all of the different sink types.
|
|
cleanup up the subpicture chain too.
|
|
Allow setting a custom subpicture sink.
|
|
|
|
2009-05-11 18:29:34 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/interfaces/tunernorm.h:
|
|
interfaces: Seperate some more struct definitions from typedefs
|
|
|
|
2009-05-11 15:48:56 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/interfaces/navigation.h:
|
|
* gst-libs/gst/interfaces/videoorientation.h:
|
|
* gst-libs/gst/interfaces/xoverlay.h:
|
|
interfaces: Seperate some more struct definitions from typedefs
|
|
|
|
2009-05-10 17:28:53 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* win32/common/libgstinterfaces.def:
|
|
Add new functions to win32 exports
|
|
|
|
2009-05-10 17:28:05 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
Add new functions to the docs
|
|
|
|
2009-05-10 17:25:58 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/interfaces/mixer.c:
|
|
* gst-libs/gst/interfaces/mixer.h:
|
|
interfaces: API: Add gst_mixer_get_mixer_type()
|
|
This is a convenience function that returns the mixer_type
|
|
of the interface struct.
|
|
|
|
2009-05-10 17:25:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/interfaces/colorbalance.c:
|
|
interfaces: Add docs for gst_color_balance_get_balance_type()
|
|
|
|
2009-05-10 11:17:19 +0200 Marc-Andre Lureau <marcandre.lureau@gmail.com>
|
|
|
|
* autogen.sh:
|
|
Run libtoolize before aclocal
|
|
This unbreaks the build in some cases. Fixes bug #582021
|
|
|
|
2009-05-07 17:38:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/pango/gsttextrender.c:
|
|
textrender: Correctly initialize the background for ARGB too
|
|
|
|
2009-05-07 16:59:32 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/pango/gsttextrender.c:
|
|
* ext/pango/gsttextrender.h:
|
|
textrender: Use libgstvideo functions to create caps
|
|
Also check if downstream wants ARGB always when we get
|
|
new caps.
|
|
|
|
2009-05-07 16:52:02 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/pango/gsttextrender.c:
|
|
textrender: Don't always use ARGB if downstream supports it but take it's preference
|
|
|
|
2009-05-07 16:48:08 +0200 Kapil Agrawal <kapil@mediamagictechnologies.com>
|
|
|
|
* ext/pango/gsttextrender.c:
|
|
* ext/pango/gsttextrender.h:
|
|
textrender: Add support for ARGB and alignment properties
|
|
Fixes bug #581571.
|
|
|
|
2009-05-07 16:42:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/pango/gsttextrender.c:
|
|
textrender: Add ; after GST_BOILERPLATE to fix indention
|
|
|
|
2009-05-07 15:10:30 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/tag/gstvorbistag.c:
|
|
vorbistag: Use text/uri-list as mimetype instead of ---> for URI lists
|
|
|
|
2009-05-07 14:59:36 +0200 Arnout Vandecappelle <arnout@mind.be>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefindfunctions: made mp3_type_find less aggressive
|
|
mp3_type_find could suggest already when only a single valid header
|
|
was found, if it ran out of data before the end of the next frame.
|
|
Therefore, ignore the last found frame if it was incomplete.
|
|
Fixes bug #579692.
|
|
|
|
2009-05-07 14:48:29 +0200 John Millikin <jmillikin@gmail.com>
|
|
|
|
* gst-libs/gst/tag/gstvorbistag.c:
|
|
vorbistag: Store cover art in vorbiscomments
|
|
Fixes bug #513373.
|
|
|
|
2009-05-07 06:14:18 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/interfaces/colorbalance.c:
|
|
* gst-libs/gst/interfaces/colorbalance.h:
|
|
interfaces: API: Add gst_color_balance_get_balance_type()
|
|
This is a convenience function that returns the balance_type
|
|
of the interface struct.
|
|
|
|
2009-05-06 17:59:13 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/interfaces/colorbalance.h:
|
|
* gst-libs/gst/interfaces/colorbalancechannel.h:
|
|
* gst-libs/gst/interfaces/tuner.h:
|
|
* gst-libs/gst/interfaces/tunerchannel.h:
|
|
interfaces: Separate struct definitions from typedefs
|
|
|
|
2009-05-06 14:03:01 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* pkgconfig/gstreamer-app-uninstalled.pc.in:
|
|
Fix libdir for uninstalled gstreamer-app library
|
|
|
|
2009-05-12 01:59:01 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/pbutils/descriptions.c:
|
|
pbutils: add description for APE tag caps
|
|
|
|
2009-05-12 01:35:27 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* configure.ac:
|
|
configure: bump core requirement to last release
|
|
as that's more likely to be true than that we need
|
|
only 0.21.1.
|
|
|
|
2009-05-12 01:21:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* common:
|
|
* configure.ac:
|
|
configure: rename CVS -> git in a couple of places
|
|
|
|
2009-05-12 01:17:53 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* configure.ac:
|
|
configure: bump GLib requirement to GLib >= 2.16
|
|
as per the New Regime (see wiki).
|
|
|
|
2009-05-01 00:09:15 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/tag/gsttagdemux.c:
|
|
tagdemux: cache events from upstream and re-send them once we have a source pad
|
|
Makes sure tags don't get dropped when we have multiple tag demuxers in a row.
|
|
Fixes #580318.
|
|
|
|
2009-05-07 14:07:44 -0700 Michael Smith <msmith@songbirdnest.com>
|
|
|
|
* gst-libs/gst/riff/riff-media.c:
|
|
riff: support UYVY raw 4:2:2 in riff.
|
|
|
|
2009-05-11 21:20:07 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* configure.ac:
|
|
Back to development -> 0.10.23.1
|
|
|
|
2009-04-27 22:42:55 -0700 Michael Smith <msmith@syncword.(none)>
|
|
|
|
* ext/theora/theoradec.c:
|
|
theoradec: fix buffer overrun on 422 decode.
|
|
|
|
2009-04-27 21:39:01 -0700 Michael Smith <msmith@syncword.(none)>
|
|
|
|
* ext/theora/theoradec.c:
|
|
theoradec: 444 support.
|
|
|
|
2009-04-27 21:30:04 -0700 Michael Smith <msmith@syncword.(none)>
|
|
|
|
* ext/theora/theoradec.c:
|
|
theoradec: handle 422 images (as YUY2).
|
|
|
|
2009-04-27 21:01:51 -0700 Michael Smith <msmith@syncword.(none)>
|
|
|
|
* ext/theora/gsttheoradec.h:
|
|
* ext/theora/theoradec.c:
|
|
theoradec: rearrange code in preparation for 422 and 444 support.
|
|
|
|
=== release 0.10.23 ===
|
|
|
|
2009-05-10 23:57:01 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* ChangeLog:
|
|
* NEWS:
|
|
* RELEASE:
|
|
* configure.ac:
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
|
* docs/plugins/gst-plugins-base-plugins.interfaces:
|
|
* docs/plugins/gst-plugins-base-plugins.prerequisites:
|
|
* docs/plugins/gst-plugins-base-plugins.signals:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.xml:
|
|
* docs/plugins/inspect/plugin-app.xml:
|
|
* docs/plugins/inspect/plugin-audioconvert.xml:
|
|
* docs/plugins/inspect/plugin-audiorate.xml:
|
|
* docs/plugins/inspect/plugin-audioresample.xml:
|
|
* docs/plugins/inspect/plugin-audiotestsrc.xml:
|
|
* docs/plugins/inspect/plugin-cdparanoia.xml:
|
|
* docs/plugins/inspect/plugin-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gdp.xml:
|
|
* docs/plugins/inspect/plugin-gio.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.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-queue2.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-uridecodebin.xml:
|
|
* docs/plugins/inspect/plugin-video4linux.xml:
|
|
* docs/plugins/inspect/plugin-videorate.xml:
|
|
* docs/plugins/inspect/plugin-videoscale.xml:
|
|
* docs/plugins/inspect/plugin-videotestsrc.xml:
|
|
* docs/plugins/inspect/plugin-volume.xml:
|
|
* docs/plugins/inspect/plugin-vorbis.xml:
|
|
* docs/plugins/inspect/plugin-ximagesink.xml:
|
|
* docs/plugins/inspect/plugin-xvimagesink.xml:
|
|
* gst-plugins-base.doap:
|
|
* win32/common/_stdint.h:
|
|
* win32/common/config.h:
|
|
Release 0.10.23
|
|
|
|
2009-05-10 23:56:05 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/bg.po:
|
|
* po/ca.po:
|
|
* po/cs.po:
|
|
* po/da.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/es.po:
|
|
* po/fi.po:
|
|
* po/fr.po:
|
|
* po/hu.po:
|
|
* po/id.po:
|
|
* po/it.po:
|
|
* po/ja.po:
|
|
* po/lt.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/pl.po:
|
|
* po/pt_BR.po:
|
|
* po/ru.po:
|
|
* po/sk.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
* po/zh_CN.po:
|
|
Update .po files
|
|
|
|
2009-05-08 20:32:20 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* configure.ac:
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/bg.po:
|
|
* po/ca.po:
|
|
* po/cs.po:
|
|
* po/da.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/es.po:
|
|
* po/fi.po:
|
|
* po/fr.po:
|
|
* po/hu.po:
|
|
* po/id.po:
|
|
* po/it.po:
|
|
* po/ja.po:
|
|
* po/lt.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/pl.po:
|
|
* po/pt_BR.po:
|
|
* po/ru.po:
|
|
* po/sk.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
* po/zh_CN.po:
|
|
* win32/common/_stdint.h:
|
|
* win32/common/config.h:
|
|
0.10.22.6 pre-release
|
|
|
|
2009-05-08 13:09:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: fix resume after pause
|
|
Don't ignore the state change of the children, they might be doing an ASYNC
|
|
state change.
|
|
|
|
2009-05-08 11:05:41 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* ChangeLog:
|
|
* configure.ac:
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/bg.po:
|
|
* po/ca.po:
|
|
* po/cs.po:
|
|
* po/da.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/es.po:
|
|
* po/fi.po:
|
|
* po/fr.po:
|
|
* po/hu.po:
|
|
* po/id.po:
|
|
* po/it.po:
|
|
* po/ja.po:
|
|
* po/lt.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/pl.po:
|
|
* po/pt_BR.po:
|
|
* po/ru.po:
|
|
* po/sk.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
* po/zh_CN.po:
|
|
0.10.22.5 pre-release
|
|
|
|
2009-05-07 22:01:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/tcp/gstmultifdsink.c:
|
|
* gst/tcp/gsttcp-marshal.list:
|
|
multifdsink: fix signature of the add-full signal
|
|
The second parameter is a GstSyncMethod enum, not a boolean.
|
|
|
|
2009-05-07 15:19:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playsink: initialize variable too
|
|
|
|
2009-05-07 14:28:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: make playsink go ASYNC to PAUSED
|
|
Make playsink go async to the PAUSED state instead of relying on uridecodebin
|
|
for async behaviour in playbin. This solves some problems (mainly with DVD)
|
|
where the pipeline would go to PLAYING before preroll completed, failing to
|
|
select the audiosink clock.
|
|
Fixes #581727
|
|
|
|
2009-05-06 16:09:52 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* configure.ac:
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/bg.po:
|
|
* po/ca.po:
|
|
* po/cs.po:
|
|
* po/da.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/es.po:
|
|
* po/fi.po:
|
|
* po/fr.po:
|
|
* po/hu.po:
|
|
* po/id.po:
|
|
* po/it.po:
|
|
* po/ja.po:
|
|
* po/lt.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/pl.po:
|
|
* po/pt_BR.po:
|
|
* po/ru.po:
|
|
* po/sk.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
* po/zh_CN.po:
|
|
* win32/common/_stdint.h:
|
|
* win32/common/config.h:
|
|
0.10.22.4 pre-release
|
|
|
|
2009-05-06 13:19:34 +0100 Zaheer Merali <zaheerabbas@merali.org>
|
|
|
|
* ext/theora/theoraenc.c:
|
|
* ext/vorbis/vorbisenc.c:
|
|
vorbisenc, theoraenc: Ensure gp is computed consistently + clip to segment
|
|
With vorbisenc, compute the granulepos with running time and clip incoming
|
|
buffers to segment.
|
|
With theoraenc, drop out of segment buffers.
|
|
|
|
2009-05-01 16:47:53 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* gst/audioresample/gstaudioresample.c:
|
|
audioresample: Fix buffer size transformations
|
|
When calculating the input/output buffer sizes in the transform_size function,
|
|
take the number of channels into account, so we don't end up calculating
|
|
a buffer size that only contains a partial number of audio frames.
|
|
Also, when going from output size to input size, round down rather than
|
|
up, so as to calculate the minimum number of samples that *might* yield
|
|
a buffer of the intended destination size.
|
|
Fixes: #580470 and #580952
|
|
|
|
2009-04-29 16:45:27 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* ext/vorbis/gstvorbisenc.h:
|
|
* ext/vorbis/vorbisenc.c:
|
|
vorbisenc: Ensure output buffers fall within the segment
|
|
Add the start position of the first segment to the running time
|
|
used to generate buffer timestamps in vorbisenc. This avoids generating
|
|
buffers which fall outside the initial segment. The element segment
|
|
handling requires more extensive fixing, but this at least prevents
|
|
regressions. Fixes: #580020
|
|
|
|
2009-04-29 11:18:42 +0200 Andy Wingo <wingo@oblong.net>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
Revert "add can-activate-pull property to baseaudiosink"
|
|
This reverts commit c4074a2ee4f1e6cac734a145bf675bbb16fac985.
|
|
|
|
2009-04-29 11:18:33 +0200 Andy Wingo <wingo@oblong.net>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
Revert "[baseaudiosink] add docs for can-activate-pull"
|
|
This reverts commit 416ce16f26b39c76ab35e1ef6a75dc41ec69f75b.
|
|
|
|
2009-04-28 18:48:33 +0200 Andy Wingo <wingo@oblong.net>
|
|
|
|
[baseaudiosink] add docs for can-activate-pull
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c: Add documentation for
|
|
can-activate-pull.
|
|
|
|
2009-01-21 12:33:59 +0100 Andy Wingo <wingo@oblong.net>
|
|
|
|
add can-activate-pull property to baseaudiosink
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c: Add can-activate-pull property
|
|
to baseaudiosink.
|
|
|
|
2009-04-28 11:32:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/videorate/gstvideorate.c:
|
|
* gst/videorate/gstvideorate.h:
|
|
videorate: clear discont on duplicated buffers
|
|
When videorate duplicates a buffer with a DISCONT flag, it copies the discont on
|
|
the first pushed buffer but fails to clear it for subsequent buffers. This
|
|
causes theoraenc!oggmux and possibly other elements to consider this a discont
|
|
stream.
|
|
Fix videorate to produce discont as the first buffer and after a flushing seek.
|
|
Fixes #580271.
|
|
|
|
2009-04-24 18:13:00 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* tests/check/Makefile.am:
|
|
check: Disable the playbin2 for this release, as it is a bit racy.
|
|
Disable the test, as per the discussion in #580120. Needs re-enabling
|
|
after the release, when playbin2 is fixed.
|
|
|
|
2009-04-23 08:41:19 +0200 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
decodebin2: Don't reduce max-size-time of exposed groups. Fixes #579912
|
|
The 2s limit is way too small for a lot of files (which have an interleave
|
|
in time of between 3 and 5s). Instead, leave it to the initial 5s value
|
|
and reduce the other limits (allowing us to stay memory-efficient).
|
|
|
|
2009-04-21 21:06:59 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* configure.ac:
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/bg.po:
|
|
* po/ca.po:
|
|
* po/cs.po:
|
|
* po/da.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/es.po:
|
|
* po/fi.po:
|
|
* po/fr.po:
|
|
* po/hu.po:
|
|
* po/id.po:
|
|
* po/it.po:
|
|
* po/ja.po:
|
|
* po/lt.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/pl.po:
|
|
* po/pt_BR.po:
|
|
* po/ru.po:
|
|
* po/sk.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
* po/zh_CN.po:
|
|
* win32/common/_stdint.h:
|
|
* win32/common/config.h:
|
|
0.10.22.3 pre-release
|
|
|
|
2009-04-21 20:41:23 +0100 René Stadler <mail@renestadler.de>
|
|
|
|
* gst/audioresample/gstaudioresample.c:
|
|
audioresample: Fix unused variable in compilation with --disable-gst-debug
|
|
Fixes: #579668
|
|
|
|
2009-04-21 22:12:28 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* common:
|
|
Automatic update of common submodule
|
|
From b3941ea to 6ab11d1
|
|
|
|
2009-04-21 20:57:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybasebin.c:
|
|
playbin: only use raw_decoding_mode when it's true
|
|
First check the pad caps if they are raw before setting the raw_decoding_mode to
|
|
TRUE. Fixes playback of transport streams and other streams that require large
|
|
queues.
|
|
Fixes #579734
|
|
|
|
2009-04-19 18:15:28 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/cdda/gstcddabasesrc.c:
|
|
* tests/check/libs/cddabasesrc.c:
|
|
cddabasesrc: fix posting of discid tags after MERGE_MODE_REPLACE_ALL changes in core
|
|
Don't use REPLACE_ALL merge mode when that's not really what we want,
|
|
as now that REPLACE_ALL actually does what it's supposed to do in
|
|
core, we drop tags we wanted to keep, such as the various disc id
|
|
tags. Add unit test for this as well. Fixes #579463.
|
|
|
|
2009-04-17 10:34:54 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtspconnection: don't use GLib-2.16 API, we require only 2.14
|
|
Fixes #579267.
|
|
|
|
2009-04-17 10:55:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
baseaudiosink: don't unparent the ringbuffer
|
|
when going to NULL, don't unparent the ringbuffer because we don't support going
|
|
back to 0 very well yet.
|
|
Fixes #579203
|
|
|
|
2009-04-17 10:53:10 +0200 Olivier Crete <tester at tester.ca>
|
|
|
|
* gst-libs/gst/rtp/gstrtcpbuffer.c:
|
|
RTCP: don't fail when retrieving invalid PT
|
|
We can't meaningfully assert on valid packet types so just return the type as it
|
|
is. Update the comments to reflect this.
|
|
Fixes #579192.
|
|
|
|
2009-04-16 12:12:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/app/gstappsink.h:
|
|
* gst-libs/gst/app/gstappsrc.h:
|
|
app: add trivial cast macros
|
|
Add trivial cast macros for appsrc and appsink. Mark them as being since 0.10.23
|
|
and add the macros to the standard macros in the docs.
|
|
Fixes #579130
|
|
|
|
2009-04-16 12:09:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
|
|
pkgconfig: add the app/ directory to Libs
|
|
Add the appsrc/appsink directory to the Libs in the uninstalled
|
|
pkgconfig file so that one can build against it.
|
|
Fixes #579129
|
|
|
|
2009-04-15 22:59:31 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* configure.ac:
|
|
0.10.22.2 pre-release
|
|
|
|
2009-04-15 22:56:15 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* ChangeLog:
|
|
ChangeLog: regenerate changelog with the gen-changelog script
|
|
|
|
2009-04-16 00:41:13 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/bg.po:
|
|
* po/ca.po:
|
|
* po/cs.po:
|
|
* po/da.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/es.po:
|
|
* po/fi.po:
|
|
* po/fr.po:
|
|
* po/hu.po:
|
|
* po/id.po:
|
|
* po/it.po:
|
|
* po/ja.po:
|
|
* po/lt.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/pl.po:
|
|
* po/pt_BR.po:
|
|
* po/ru.po:
|
|
* po/sk.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
* po/zh_CN.po:
|
|
po: Update po files from TP
|
|
|
|
2009-04-16 00:40:59 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* win32/common/_stdint.h:
|
|
* win32/common/config.h:
|
|
* win32/common/gstrtsp-enumtypes.c:
|
|
* win32/common/interfaces-enumtypes.c:
|
|
* win32/common/interfaces-enumtypes.h:
|
|
* win32/common/video-enumtypes.c:
|
|
win32: Update win32 build files
|
|
|
|
2009-04-16 00:31:55 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* tests/check/libs/video.c:
|
|
check: Add GST_VIDEO_FORMAT_YVYU to the test so it passes.
|
|
|
|
2009-04-16 00:31:00 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* tests/check/elements/playbin2.c:
|
|
check: Fix the input uri in playbin2 test.
|
|
Don't try and use a random file in wim's home directory as a test input
|
|
|
|
2009-04-15 15:35:59 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/video/video.h:
|
|
video: Fix typo in the docs
|
|
|
|
2009-04-15 14:53:47 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/video/video.c:
|
|
* gst-libs/gst/video/video.h:
|
|
video: Add support for YVYU YUV colorspace
|
|
|
|
2009-04-15 00:17:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-docs.sgml:
|
|
* gst-libs/gst/fft/gstfft.c:
|
|
docs: fix hyperlink and move fft attribution to the right place
|
|
|
|
2009-04-15 00:02:39 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
log: use G_GUINT64_FORMAT instead of llu
|
|
|
|
2009-04-14 18:31:52 +0200 Josep Torra <n770galaxy at gmail.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspdefs.c:
|
|
* gst-libs/gst/rtsp/gstrtspdefs.h:
|
|
RTSP: add missing headers for WMS RTSP
|
|
Add missing headers related to Windows Media RTSP extension.
|
|
Fixes #578942
|
|
|
|
2009-04-14 18:16:37 +0200 Olivier Crete <tester at tester.ca>
|
|
|
|
* docs/design/draft-keyframe-force.txt:
|
|
* ext/theora/gsttheoraenc.h:
|
|
* ext/theora/theoraenc.c:
|
|
theoraenc: implement upstream keyframe force
|
|
Implement handling of upstream keyframe forcing.
|
|
Update the design documents too.
|
|
Fixes #578656
|
|
|
|
2009-04-14 17:31:31 +0200 Olivier Crete <tester at tester.ca>
|
|
|
|
* ext/theora/theoraenc.c:
|
|
theoraenc: factor out keyframe forcing
|
|
See #578656
|
|
|
|
2009-04-14 17:01:51 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* AUTHORS:
|
|
* gst-libs/gst/fft/gstfft.c:
|
|
Give credit to Mark Borgerding (kissfft author)
|
|
and add myself to AUTHORS as well. Fixes #575638.
|
|
|
|
2009-04-14 17:04:06 +0200 Jan Urbanski <j.urbanski at students.mimuw.edu.pl>
|
|
|
|
* gst/tcp/gstmultifdsink.c:
|
|
* gst/tcp/gstmultifdsink.h:
|
|
multifdsink: add property to resend streamheaders
|
|
Adds a new property in multifdsink, resend-streamheader.
|
|
If this property is false, the multifdsink will not send the streamheader if
|
|
there's already one set for a particular client.
|
|
There are some formats in which every stream needs to start with a certain
|
|
blob, but you can't inject this blob at leisure. If the producer wants to
|
|
change the blob in question and sets in as the streamheader on the outgoing
|
|
buffers' caps, new clients of multifdsink will get the new streamheader, but
|
|
old clients will break, because they'll see the blob in the middle of the
|
|
stream.
|
|
The property is true by default, so existing code will not see any difference.
|
|
Fixes #578118.
|
|
|
|
2009-04-14 16:53:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/tcp/gstmultifdsink.c:
|
|
* gst/tcp/gstmultifdsink.h:
|
|
multifdsink: add property to handle client write
|
|
Add a property to disable listening to client writes. This property is usefull
|
|
when other code will deal with reading from the client socket.
|
|
API: GstMultiFdSink::handle-read property
|
|
|
|
2009-04-14 16:45:20 +0200 Johann Prieur <johann.prieur at gmail.com>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/rtp/gstrtcpbuffer.c:
|
|
* gst-libs/gst/rtp/gstrtcpbuffer.h:
|
|
* win32/common/libgstrtp.def:
|
|
RTCP: add beginnings of Feedback messages
|
|
Add the beginnings of parsing and constructing Feedback messages.
|
|
Fixes #577610.
|
|
|
|
2009-04-14 13:51:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: clear the target
|
|
Clear the target of our ghostpads before we remove the pad from the element.
|
|
This to make sure that the internal pad is not left linked to whatever pad we
|
|
were ghosted to. This should only be a problem when we leak the ghostpads.
|
|
Also release our subpicture pads.
|
|
Fixes #577288.
|
|
|
|
2009-04-14 12:10:30 +0100 Hannes Bistry <hannesb@gmx.net>
|
|
|
|
* sys/ximage/ximagesink.c:
|
|
ximagesink: fix mouse pointer offsets in navigation event if window is smaller than the image
|
|
Fixes #570768.
|
|
|
|
2009-04-14 13:16:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
baseaudiosrc: adjust the internal timestamp
|
|
Adjust the internal timestamp before comparing it against the adjusted clock
|
|
time.
|
|
Fixes #578506
|
|
|
|
2009-04-14 13:12:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
baseaudiosink: use new clock time methods
|
|
Use the unadjusted internal clock times to calculate the internal/external
|
|
offset when calibrating the clock.
|
|
When going to NULL, unparent and free the ringbuffer, like we do in the source
|
|
element.
|
|
See #578506
|
|
|
|
2009-04-14 13:08:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstaudioclock.c:
|
|
* gst-libs/gst/audio/gstaudioclock.h:
|
|
* win32/common/libgstaudio.def:
|
|
audioclock: add methods for the internal offset
|
|
Add two methods for getting the unadjusted time of the clock and one for
|
|
adjusting an internal time. We will need these methods for correctly handling
|
|
the time after a gst_audio_clock_reset().
|
|
Add a debug category and some debug lines to the audio clock.
|
|
API: gst_audio_clock_get_time()
|
|
API: gst_audio_clock_adjust()
|
|
API: GST_AUDIO_CLOCK_CAST()
|
|
|
|
2009-04-14 11:34:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
decodebin2: fix up the debugs and warnings
|
|
Use _OBJECT variants because we can. Go over some log statements and put them in
|
|
the right category.
|
|
Fixes #567740.
|
|
|
|
2009-04-12 22:26:33 +0200 Luca Ognibene <luca.ognibene at gmail.com>
|
|
|
|
* gst/tcp/gstmultifdsink.c:
|
|
multifdsink: fix error in sync-method
|
|
Multifdsink did not handle sync-method=latest-keyframe correctly when the
|
|
soft-limit is set to -1 (unlimited).
|
|
Fixes #578583.
|
|
|
|
2009-04-10 21:49:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
baseaudiosink: use the internal clock time
|
|
We can't assume that the internal clock time is the same as the function we
|
|
installed on our provided clock because somebody might have changed it.
|
|
|
|
2009-04-10 14:12:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/examples/seek/seek.c:
|
|
seek: handle clock-lost messages
|
|
When we receive a clock-lost message we need to pause and play to select a new
|
|
clock.
|
|
|
|
2009-04-10 13:44:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/check/Makefile.am:
|
|
* tests/check/elements/playbin2.c:
|
|
check: add a unit test for playbin2
|
|
Add unit test for playbin2 and include the refcount test in #577794.
|
|
|
|
2009-04-10 13:42:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: fix refcounting of visualisations
|
|
See #577794.
|
|
|
|
2009-04-10 13:27:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playsink: fix refcounting of custom elements
|
|
Sink the custom sinks, let other elements we create be sunken by the bin we add
|
|
them to.
|
|
Fixes #577794.
|
|
|
|
2009-04-10 12:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/check/elements/appsink.c:
|
|
check: fix appsink test
|
|
Fix the appsink test now that the method signature changed.
|
|
|
|
2009-04-10 12:26:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
playbin2: handle missing input-selector
|
|
Gracefully degrade and disable stream selection when input-selector is
|
|
missing.
|
|
|
|
2009-04-09 23:46:17 +0200 Martin Samuelsson <martin.samuelsson at axis.com>
|
|
|
|
* gst-libs/gst/app/gstappsink.c:
|
|
* gst-libs/gst/app/gstappsink.h:
|
|
appsink: make callbacks return GstFlowReturn
|
|
Make the new_buffer and new_preroll callbacks return a GstFlowReturn so that
|
|
errors can be reported properly.
|
|
Fixes #577827.
|
|
|
|
2009-04-09 18:04:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstringbuffer.c:
|
|
* gst-libs/gst/audio/gstringbuffer.h:
|
|
ringbuffer: allow for custom commit functions
|
|
Allow subclasses to override the commit method.
|
|
|
|
2009-04-08 18:04:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
baseaudiosink: fix a small glitch after pause
|
|
After we pause the stream and interrupt the writeout to the ringbuffer, also adjust
|
|
the amount of output samples we consumed. We can't do this reliably with the
|
|
current API when we are doing trick modes but we can do the right thing for
|
|
normal playback.
|
|
|
|
2009-04-08 16:43:27 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: better error message on sink failure
|
|
If we could create the sinks, but the don't work, don't send the missing plugin
|
|
message and report that the state-changed failed.
|
|
|
|
2009-04-07 22:38:29 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst-libs/gst/audio/gstaudiofilter.c:
|
|
audiofilter: don't leak pad-template
|
|
gst_element_class_add_pad_template() does not take ownership.
|
|
|
|
2009-04-04 21:18:38 +0300 Felipe Contreras <felipe.contreras@gmail.com>
|
|
|
|
* common:
|
|
Automatic update of common submodule
|
|
From d0ea89e to b3941ea
|
|
|
|
2009-04-04 16:28:14 +0200 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst-libs/gst/interfaces/navigation.c:
|
|
* sys/v4l/v4lsrc_calls.c:
|
|
navigation/v4l: Don't use g_return_val_if_fail for computed/used values.
|
|
|
|
2009-03-22 09:46:37 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* ext/theora/theoradec.c:
|
|
theoradec: return GST_CLOCK_TIME_NONE for negative framecounts.
|
|
This fixes most seeking issues when used with gnonlin.
|
|
Fixes #543591
|
|
|
|
2009-04-04 14:53:42 +0200 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* common:
|
|
Automatic update of common submodule
|
|
From f8b3d91 to d0ea89e
|
|
|
|
2009-04-03 10:51:42 -0700 Michael Smith <msmith@songbirdnest.com>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
playbin2: don't leak selector when getting current stream numbers.
|
|
|
|
2009-04-02 22:28:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: use fully qualified urls when using a proxy
|
|
Use a fully qualified url when specifying the url for tunneled requests through
|
|
a proxy.
|
|
See #573173
|
|
|
|
2009-03-31 00:54:30 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/interfaces/navigation.c:
|
|
* gst-libs/gst/interfaces/navigation.h:
|
|
* tests/check/Makefile.am:
|
|
* tests/check/libs/.gitignore:
|
|
* tests/check/libs/navigation.c:
|
|
* win32/common/libgstinterfaces.def:
|
|
navigation: Extend the navigation interface
|
|
Add support for a set of standard commands that can be queried and executed to
|
|
support applications like DVD. Add query construction and parsing functions.
|
|
Add new messages that can be sent on the bus to provide notifications related
|
|
to commands, multiangle changes, and button highlight activity.
|
|
Add some helper functions to parse the existing GstNavigation events that
|
|
elements might receive.
|
|
Document it all and add unit tests.
|
|
|
|
2009-02-04 17:03:07 +0000 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* gst/playback/gstplaybasebin.c:
|
|
* gst/playback/gstplaybasebin.h:
|
|
playbin: Add simple 'raw decoding mode'.
|
|
Raw decoding mode removes almost all buffering in video and audio queues
|
|
when a source providing already decoded video/audio is detected, on the
|
|
possibly bogus assumption that such a source should provide sufficient
|
|
internal queueing. Fixes playback on some DVDs, and improves it
|
|
on all.
|
|
|
|
2009-04-02 09:27:07 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* tests/check/elements/.gitignore:
|
|
ignores: Ignore the videoscale check binary
|
|
|
|
2009-04-02 12:13:57 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* win32/common/libgstrtsp.def:
|
|
win32: Add gst_rtsp_connection_set_proxy to the win32 exports
|
|
|
|
2009-04-02 10:42:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* ext/alsa/gstalsamixer.c:
|
|
alsamixer: don't forget to release locks in a few places
|
|
Might fix #576585.
|
|
|
|
2009-04-02 11:10:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/vs_4tap.c:
|
|
videoscale: Don't read over line ends when taking the last Cr or Cb
|
|
|
|
2009-04-02 10:52:06 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/vs_4tap.c:
|
|
videoscale: Don't write to few pixels and don't mix Cr and Cb
|
|
Fixes bug #577054.
|
|
|
|
2009-04-01 15:15:57 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/audioresample/gstaudioresample.c:
|
|
* tests/check/elements/audioresample.c:
|
|
audioresample: fix negotiation so that upstream can actually fixate to downstream's rate
|
|
If one side has a preference for a particular sample rate or set of sample rates, we
|
|
should honour this in the caps we advertise and transform to and from, so that elements
|
|
actually know about the other side's sample rate preference and can negotiate to it
|
|
if supported. Also add unit test for this.
|
|
|
|
2009-03-26 19:34:23 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
docs: add a blurb about redirect messages to playbin2 docs
|
|
|
|
2009-04-01 09:03:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: fix little typo in the comments
|
|
|
|
2009-03-31 17:52:44 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtspconnection: make gst_rtsp_watch_queue_message() thread-safe
|
|
People might queue messages from a thread other than the thread in which
|
|
the main context which this watch is attached is iterated from, so use
|
|
a GAsyncQueue instead of a GList, so g_list_append() doesn't trample
|
|
over list nodes just freed in the other thread. This just fixes issues
|
|
I've had with gst-rtsp-server. We might need more locking in various
|
|
places here.
|
|
|
|
2009-03-31 18:13:19 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspmessage.c:
|
|
rtsp: clear the entire builder structure
|
|
And use structure instead of variable with sizeof when
|
|
clearing the rtsp message structure, for clarity.
|
|
|
|
2009-03-31 17:56:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspmessage.c:
|
|
docs: fix typo in gst_rtsp_message_unset() API docs
|
|
|
|
2009-03-31 19:00:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
rtsp: add support for proxies
|
|
Add suport for proxy servers. Currently only used for tunneled HTTP
|
|
connections without authentication.
|
|
|
|
2009-03-31 18:57:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspmessage.c:
|
|
Revert "rtsp: reset whole message (was sizeof pointer instead of sizeof type)"
|
|
This reverts commit 79de0b8d67df6fbbe79455adc2e06858295f5c03.
|
|
|
|
2009-03-26 18:54:56 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* sys/xvimage/xvimagesink.c:
|
|
xvimagesink: use xcontext->depth instead of bits in attr.max_value for colorkey
|
|
According to the drivers in http://cgit.freedesktop.org/xorg/driver/ we should
|
|
format the colorkey depending on xcontext->depth. This is what they will use to
|
|
interprete the value. The max_value in turn is usualy a constant regardless of
|
|
the depth.
|
|
|
|
2009-03-31 12:22:14 +0300 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspmessage.c:
|
|
rtsp: reset whole message (was sizeof pointer instead of sizeof type)
|
|
|
|
2009-03-31 00:56:18 +0100 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* gst-libs/gst/interfaces/mixer.c:
|
|
doc: Fix a typo in the GstMixer docs
|
|
|
|
2009-03-29 12:01:33 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/vs_scanline.c:
|
|
videoscale: Fix linear scaling for one byte components
|
|
Fixes bug #577054.
|
|
|
|
2009-03-29 11:53:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/vs_4tap.c:
|
|
videoscale: Fix 4tap scaling of YUYV and friends
|
|
|
|
2009-03-28 16:08:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/vs_image.c:
|
|
* gst/videoscale/vs_scanline.c:
|
|
* gst/videoscale/vs_scanline.h:
|
|
videoscale: Rewrite YUYV (and friends) scaling and don't read/write over line ends
|
|
Partially fixes bug #577054, there's just one issue left now.
|
|
|
|
2009-03-28 12:48:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* tests/check/elements/videoscale.c:
|
|
videoscale: Add some more unit tests
|
|
|
|
2009-03-28 11:51:01 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/gstvideoscale.c:
|
|
videoscale: Use bilinear instead of 4tap scaling for heights < 4
|
|
Partially fixes bug #577054.
|
|
|
|
2009-03-28 11:45:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/vs_scanline.c:
|
|
videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY/RGB/RGBA
|
|
This case is for upscaling a frame with width=1
|
|
Partially fixes bug #577054.
|
|
|
|
2009-03-28 11:27:56 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/vs_scanline.c:
|
|
videoscale: Don't read after the end of a line when lineary scaling YUYV/UYVY
|
|
Partially fixes bug #577054.
|
|
|
|
2009-03-28 10:40:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videotestsrc/gstvideotestsrc.c:
|
|
videotestsrc: Initialize buffer memory with zeroes
|
|
This prevents valgrind warnings when accessing the "x" parts
|
|
of xRGB and friends in other elements that handle (and can handle)
|
|
xRGB like ARGB (for example videoscale).
|
|
|
|
2009-03-28 10:25:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* tests/check/Makefile.am:
|
|
* tests/check/elements/videoscale.c:
|
|
videoscale: Add a lot of unit tests
|
|
|
|
2009-03-28 10:06:24 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/gstvideoscale.c:
|
|
videocale: Add support for video/x-raw-gray with bpp=depth=8
|
|
|
|
2009-03-28 10:01:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videotestsrc/videotestsrc.c:
|
|
videotestsrc: Add support for generating video/x-raw-gray with bpp=depth=8
|
|
|
|
2009-03-28 09:43:23 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
|
|
ffmpegcolorspace: video/x-raw-gray is the same as the YUV Y800 format
|
|
|
|
2009-03-27 19:12:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/vs_4tap.c:
|
|
videoscale: Take the next luma value instead of every second next when scaling UYVY and friends
|
|
|
|
2009-03-27 19:09:47 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/gstvideoscale.c:
|
|
videoscale: Add support for v308 YUV colorspace
|
|
|
|
2009-03-27 13:15:11 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/vs_4tap.c:
|
|
videoscale: Add my copyright to the 4tap scalers
|
|
|
|
2009-03-27 13:14:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/gstvideoscale.c:
|
|
videoscale: Enable 4-tap scaling for all supported formats
|
|
|
|
2009-03-27 13:14:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/vs_4tap.c:
|
|
* gst/videoscale/vs_4tap.h:
|
|
videoscale: Implement 4-tap scaling for RGB565 and RGB555
|
|
|
|
2009-03-27 10:47:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/vs_4tap.c:
|
|
* gst/videoscale/vs_4tap.h:
|
|
videoscale: Implement 4-tap scaling for UYVY
|
|
|
|
2009-03-27 09:33:58 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/vs_4tap.c:
|
|
* gst/videoscale/vs_4tap.h:
|
|
videoscale: Implement 4-tap scaling for YUY2 and YVYU
|
|
|
|
2009-03-26 22:14:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/vs_4tap.c:
|
|
* gst/videoscale/vs_4tap.h:
|
|
videoscale: Implement 4-tap scaling for RGB and BGR
|
|
|
|
2009-03-26 22:08:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videoscale/vs_4tap.c:
|
|
* gst/videoscale/vs_4tap.h:
|
|
videoscale: Implement 4-tap scaling for RGBA and other 4 byte formats
|
|
|
|
2009-03-26 11:02:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/pango/gsttextoverlay.c:
|
|
textoverlay: Fix drawing of UYVY text borders
|
|
|
|
2009-03-26 10:36:27 +0100 Zeeshan Ali <zeeshan.ali@nokia.com>
|
|
|
|
* ext/pango/gsttextoverlay.c:
|
|
* ext/pango/gsttextoverlay.h:
|
|
textoverlay: Add support for UYVY colorspace
|
|
Fixes bug #378094.
|
|
|
|
2009-03-25 19:01:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
decodebin2: do some more cleanup
|
|
Free the groups when we go to READY.
|
|
Allow for NO_PREROLL elements.
|
|
|
|
2009-03-25 16:37:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: start CSeq counting from 1 instead of 0
|
|
Start counting from 1 instead of 0 as this is what most other clients
|
|
seem to do.
|
|
|
|
2009-03-25 16:35:22 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspdefs.c:
|
|
* gst-libs/gst/rtsp/gstrtspdefs.h:
|
|
rtsp: add ETag and If-Match headers
|
|
Add new headers, we need them for RealMedia support.
|
|
|
|
2009-03-25 14:16:25 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* sys/xvimage/xvimagesink.c:
|
|
xvimagesink: scale the colorkey components in case of 16bit visuals
|
|
Use a default that won't be scales to 0,0,0
|
|
|
|
2009-03-25 11:27:44 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
audiosrc: improve 'Dropped n samples' warning message
|
|
|
|
2009-03-24 19:41:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/examples/app/appsrc-ra.c:
|
|
* tests/examples/app/appsrc-seekable.c:
|
|
examples: use new method to set flags
|
|
Use the new core method for setting object enum properties by name.
|
|
|
|
2009-03-24 18:29:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
* gst/playback/gstplaysink.h:
|
|
playbin2: add more support for subpictures
|
|
|
|
2009-03-24 17:12:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
* gst/playback/gstplaysink.c:
|
|
* gst/playback/gstplaysink.h:
|
|
playbin2: first support for subpictures
|
|
Add beginnings of subpicture support.
|
|
|
|
2009-03-24 15:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/examples/seek/seek.c:
|
|
seek: print tags from the different tracks
|
|
|
|
2009-03-24 12:22:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
playbin2: blacklist subpictures for now
|
|
Blacklist the subpictures until we add support for them.
|
|
Add some small debug info.
|
|
See #576408.
|
|
|
|
2009-03-24 12:19:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gsturidecodebin.c:
|
|
uridecodebin: expose more media types
|
|
Expose more media types from a raw source, such as the subpicture and various
|
|
text pads.
|
|
Small cleanups and add some more debugging.
|
|
See #576408.
|
|
|
|
2009-03-24 10:42:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: rescan audio sinks for volume/mute
|
|
Rescan the audio sinks for the mute and volume properties.
|
|
fixes #576180.
|
|
|
|
2009-03-23 19:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: fix reuse of the video chains
|
|
When reusing playbin with visualisations, reset the async property on the video
|
|
sink because some sinks might dynamically recreate their sinks.
|
|
Fixes #576188
|
|
|
|
2009-03-23 17:37:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: allow dynamic swtiching of subtitles
|
|
When we have the textpad configured, enable and disable the subtitles by setting
|
|
the silent flag on the overlay element instead of trying to remove elements.
|
|
See #576187
|
|
|
|
2009-03-23 16:59:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/icles/playbin-text.c:
|
|
tests: print some more info in the text example
|
|
Print both the position and the running_time when the subtitle becomes available
|
|
in the application.
|
|
|
|
2009-03-23 16:04:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: fix dynamic switching of visualisations
|
|
Fix the switching of visualisations by requesting and releasing the tee request
|
|
pads on demand.
|
|
See #576187.
|
|
|
|
2009-03-23 16:19:11 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/tcp/README:
|
|
* gst/tcp/gsttcpclientsink.c:
|
|
* gst/tcp/gsttcpclientsrc.c:
|
|
* gst/tcp/gsttcpserversink.c:
|
|
* gst/tcp/gsttcpserversrc.c:
|
|
docs: add examples for tcp elements, also use correct section name. Fixes #564139
|
|
Updated the examples in the README to actually work. Add them to api docs. Tests
|
|
the api-docs and fix the section names to make the docs actualy show up.
|
|
The example for "tcpserversrc" needs review (might be an element bug).
|
|
|
|
2009-03-17 09:14:02 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/videoscale/gstvideoscale.c:
|
|
indent: fix damange that gst-indent did some time ago
|
|
|
|
2009-03-23 15:27:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: fix linking order
|
|
Link after doing the state change and unlink before shutting down. Makes the
|
|
window for causing races in toggling the visualisations smaller.
|
|
See #576187.
|
|
|
|
2009-03-23 12:26:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gsturidecodebin.c:
|
|
uridecodebin: reset counter
|
|
reset the number of pending dynamic operations back to 0 when we reuse
|
|
uridecodebin.
|
|
Fixes #576190
|
|
|
|
2009-03-23 11:38:53 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* ext/theora/theoradec.c:
|
|
theoradec: Use GST_CLOCK_TIME_NONE for invalid positions. Fixes #543591
|
|
The problem was that previously we didn't check whether _theora_granule_frame
|
|
returned a negative framecount or not, resulting in bogus timestamps.
|
|
|
|
2009-03-21 09:46:28 +0100 René Stadler <mail@renestadler.de>
|
|
|
|
* ext/vorbis/vorbisenc.c:
|
|
vorbisenc: Set caps on non-header ouput buffers.
|
|
Fixes #576142.
|
|
|
|
2009-03-20 16:13:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/examples/seek/seek.c:
|
|
seek: Add some more debug
|
|
Add some more info about the selected streams.
|
|
|
|
2009-03-20 15:47:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
decodebin2: a pad starts out being not drained.
|
|
Mark a new pad as not drained until we get EOS on it.
|
|
|
|
2009-03-20 14:17:19 +0100 LRN <lrn1986 at gmail dot com>
|
|
|
|
* gst/playback/gstqueue2.c:
|
|
win32: fix seeking in large files
|
|
Fix Seeking in large files by using the 64-bit seek functions.
|
|
Fixes #576019
|
|
|
|
2009-03-19 20:31:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
decodebin2: recover from failing to add a pad
|
|
When we cannot add a pad to the decodebin2 for some reason, print a warning but
|
|
continue adding the remaining pads.
|
|
|
|
2009-03-19 19:35:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
decodebin2: more cleanups and docs.
|
|
Add some more comments and use g_list_prepend().
|
|
|
|
2009-03-19 19:19:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
decodebin2: refactoring and race fixes
|
|
Refactor some code so that we can take the right locks and in the right order.
|
|
Fixes quite a bit of races already.
|
|
|
|
2009-03-19 19:03:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
playbin2: remove the group cond + cleanups
|
|
Remove the group GCond that we used for waiting for groups to finish because we
|
|
use pad blocking on the selectors and counters instead for waiting for the
|
|
groups to complete.
|
|
remove the obsolete about_to_finish variable set while emiting the
|
|
about-to-finish signal and fix some old comments.
|
|
We don't need to take the playbin lock when querying the uridecodebin.
|
|
|
|
2009-03-18 10:45:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/icles/playbin-text.c:
|
|
icles: print better error and warning messages
|
|
--
|
|
|
|
2009-03-17 22:53:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspbase64.c:
|
|
* gst-libs/gst/rtsp/gstrtspbase64.h:
|
|
rtsp: Use GLib base64 functions and deprecate gst_rtsp_base64_encode
|
|
This also fixes another instance of CVE-2008-4316.
|
|
|
|
2009-03-17 19:53:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* ext/ogg/gstoggdemux.c:
|
|
oggdemux: report -1 for duration in push mode
|
|
In push mode we must return TRUE from the duration query with a value of -1
|
|
meaning that we know that we don't know the duration.
|
|
|
|
2009-03-17 19:09:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
decodebin2: add extra dynamic ref for demuxers
|
|
When we make a group connected to a demuxer, keep an extra dynamic refcount for
|
|
the group which is only decremented when no_more_pads or a multiqueue overrun is
|
|
detected. This way we avoid a race between exposing the group while more dynamic
|
|
refs are added from new pads.
|
|
Fixes #575588.
|
|
|
|
2009-03-17 15:39:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: sync state of the sink correctly
|
|
Sync the state of the newly added chains to the state of the parent sink element
|
|
to avoid lost async-start messages. Fixes cdda:// async-done message storm.
|
|
|
|
2009-03-17 11:54:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
playbin2: return NOT_LINKED for unselected streams
|
|
When streams are not selected in the selector, return NOT_LINKED so that
|
|
upstream elements can skip decoding. Only do this for audio and video pads
|
|
because for text streams the overhead is smaller and they could come from
|
|
external files.
|
|
|
|
2009-03-17 11:51:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin: set custom text sink properties
|
|
Set the custom sink async=FALSE to not make it participate in preroll because we
|
|
are dealing with sparse streams.
|
|
Try to set sync=TRUE on the custom text sink.
|
|
|
|
2009-03-17 11:30:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/icles/playbin-text.c:
|
|
example: use appsink instead of fakesink
|
|
Use appsink instead of fakesink to get the subtitles.
|
|
Make things more pretty.
|
|
|
|
2009-03-17 11:24:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/icles/.gitignore:
|
|
* tests/icles/Makefile.am:
|
|
* tests/icles/playbin-text.c:
|
|
examples: add example of intercepting subtitles
|
|
Add an example of how to install a custom sink for receiving subtitles in
|
|
playbin2.
|
|
|
|
2009-03-17 11:03:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/check/elements/appsink.c:
|
|
tests: fix include in the appsink test
|
|
Fix dist by doing the right include.
|
|
|
|
2009-03-16 16:42:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
playbin2: don't try to set invalid stream numbers
|
|
Fix a problem with setting the stream numbers because we check for the wrong
|
|
range.
|
|
See #575239.
|
|
|
|
2009-03-16 16:16:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
playbin2: release the shutdown lock
|
|
Release the shutdown lock when we wait for other groups to complete or else we
|
|
have a deadlock when the other group completes and tries to grab the shutdown
|
|
lock.
|
|
Fixes #575550.
|
|
|
|
2009-03-16 15:31:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/examples/app/appsrc-ra.c:
|
|
* tests/examples/app/appsrc-seekable.c:
|
|
* tests/examples/app/appsrc-stream.c:
|
|
* tests/examples/app/appsrc-stream2.c:
|
|
examples: fix g_object_set() value type.
|
|
Make sure we cast the length value as a gint64 to the vararg g_object_set() just
|
|
incase sizeof(gsize) != sizeof(gint64).
|
|
|
|
2009-03-15 19:57:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefinding: make flac typefinder return lower probability for frame headers
|
|
The flac frame header typefinder overstates the likelihood of a match, leading
|
|
to false positives with e.g. aac streams and PDF files. Reduce probabilty
|
|
returned from LIKELY to POSSIBLE for the frame header matchin code.
|
|
Fixes #574939.
|
|
|
|
2009-03-11 12:59:05 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefinding: improve image/bmp typefinder
|
|
Detect more variations and also bail out in more cases where the values
|
|
don't make sense. Furthermore, add width/height and bpp to the caps,
|
|
because we can.
|
|
|
|
2009-03-13 15:22:42 +0000 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* tests/check/Makefile.am:
|
|
check: Ignore alsamixer in the states test too
|
|
|
|
2009-03-13 15:22:11 +0000 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* sys/v4l/v4l_calls.c:
|
|
v4lsrc: Fix some valgrind warnings about leaked memory and uninitialised data.
|
|
|
|
2009-03-13 16:19:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: fix resolving of hostnames
|
|
We were returning a pointer to a stack variable with the resolved hostname,
|
|
which doesn't work.
|
|
return a copy of the resolved ip address instead.
|
|
Fixes #575256.
|
|
|
|
2009-03-13 15:29:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* ext/vorbis/vorbisparse.c:
|
|
vorbisparse: be smarter when queueing headers
|
|
Look at the first buffer byte to see if a buffer is a header instead of counting
|
|
packets.
|
|
|
|
2009-03-13 15:27:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* ext/theora/gsttheoraparse.h:
|
|
* ext/theora/theoraparse.c:
|
|
theoraparse: be smarter when queuing headers
|
|
Look at the first byte of the buffer data (if we can) to decide if the packet is
|
|
a header packet or not instead of counting packets.
|
|
|
|
2009-03-13 15:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* ext/ogg/gstoggdemux.c:
|
|
oggdemux: add some debug info
|
|
Add some debug info to log when the seek worked.
|
|
|
|
2009-03-13 15:14:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/app/gstappsrc.c:
|
|
appsrc: release lock in _eos flushing case
|
|
Release the mutex when we are flushing in gst_app_src_end_of_stream()
|
|
Fixes #574964.
|
|
|
|
2009-03-13 11:49:10 +0000 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* ext/vorbis/vorbisdec.c:
|
|
vorbisdec: Avoid an unnecessary memory allocation in vorbiscomment handling.
|
|
|
|
2009-03-13 11:48:28 +0000 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* ext/theora/theoradec.c:
|
|
theoradec: Avoid an unnecessary memory allocation in vorbiscomment handling.
|
|
|
|
2009-03-12 18:27:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gsturidecodebin.c:
|
|
playbin2: fix raw elements like cdda://
|
|
Fix a fixme with a one liner and make cd playback work again.
|
|
|
|
2009-03-12 17:47:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
* gst/playback/gstplaysink.c:
|
|
* gst/playback/gstplaysink.h:
|
|
playbin2: improve subtitle handling
|
|
Add property to playbin2 to configure a custom sink that receives the raw
|
|
subtitle buffers instead of using a textoverlay.
|
|
Improve the property finding code to make it more usable.
|
|
Use property find code to find async properties in custom sinks that are bins.
|
|
Improve text overlay code to gracefully handle missing elements.
|
|
|
|
2009-02-24 15:58:42 +0000 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* gst-libs/gst/tag/gstvorbistag.c:
|
|
vorbistag: Protect memory allocation calculation from overflow.
|
|
Patch by: Tomas Hoger <thoger@redhat.com> Fixes CVE-2009-0586
|
|
|
|
2009-03-12 11:34:20 +0000 Jan Urbanski <jurbanski@flumotion.com>
|
|
|
|
* gst-plugins-base.spec.in:
|
|
Spec: fix up deps
|
|
|
|
2009-03-11 18:45:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: fix parsing of the timeout parameter
|
|
--
|
|
|
|
2009-03-11 16:20:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspmessage.c:
|
|
rtsp: fix g_return condition
|
|
when parsing a data message, we require a data message.
|
|
|
|
2009-03-11 13:33:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefinding: flac typefinder fixes
|
|
Use scan context for initial peek as well. Peek 6 bytes in the initial
|
|
peek rather than 5 bytes, to match the length of the memcmp we're doing
|
|
on that data later. Return immediately when we found caps from looking
|
|
at the beginning of the data - no point in continuing to scan the next
|
|
64kB for something matching a frame header.
|
|
|
|
2009-03-11 14:08:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspmessage.c:
|
|
rtsp: free the right string.
|
|
Free the key value before we remove the header item from the array. The item we
|
|
retrieved from the array is only valid until we remove it from the array.
|
|
|
|
2009-03-11 14:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: keep track of amount of decoded bytes
|
|
Keep track of the actual amount of decoded bytes, which can be less than 3 when
|
|
we decode the last bits of a base64 message.
|
|
|
|
2009-03-10 21:00:26 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/adder/gstadder.c:
|
|
adder: log details in getcaps like in setcaps
|
|
|
|
2009-03-10 13:11:09 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* win32/MANIFEST:
|
|
win32: update MANIFEST, fixing 'make dist'
|
|
|
|
2009-03-09 23:12:00 +0000 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* common:
|
|
Automatic update of common submodule
|
|
From 7032163 to f8b3d91
|
|
|
|
2009-03-09 16:19:40 +0100 Jonathan Matthew <notverysmart at gmail dot com>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefind: add photoshop typefind functions
|
|
Add photoshop typefind functions.
|
|
Fixes #574516.
|
|
|
|
2009-03-09 15:46:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
decodebin2: only remove pads that were added
|
|
Flag pads that were added so that we can see if we need to remove them later or
|
|
not.
|
|
|
|
2009-03-09 13:53:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtsptransport.c:
|
|
rtsp: only add ports when not using TCP
|
|
Only add the port numbers in the transport string when we are using udp or
|
|
multicast.
|
|
|
|
2009-03-09 13:53:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspmessage.c:
|
|
rtsp: use gstreamer dump mem
|
|
--
|
|
|
|
2009-03-09 13:51:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: use glib base64 encoder
|
|
--
|
|
|
|
2009-03-06 19:28:37 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
Unblock blocked ghostpads when shutting down. Fixes #574293.
|
|
|
|
2009-03-09 10:03:13 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst-libs/gst/riff/riff-media.c:
|
|
Riff: Add mapping for Fraps video codec.
|
|
Found through insanity testrun. Confirmed mapping in libavformat.
|
|
|
|
2009-03-09 09:07:13 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst-libs/gst/riff/riff-media.c:
|
|
riff: Add the 'DVR ' mapping for mpeg2video.
|
|
Found this in 3 files from the insanity suite and mapping is also present
|
|
in libavformat.
|
|
|
|
2009-03-09 09:06:40 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefind: Use the proper data pointer instead of poking random memory.
|
|
|
|
2009-03-08 18:17:48 +0100 LRN <lrn1986@gmail.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: fix compilation on windows.
|
|
Remove unused variable when building for windows.
|
|
Fixes #574443.
|
|
|
|
2009-03-08 12:03:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* common:
|
|
Automatic update of common submodule
|
|
From ffa738d to 7032163
|
|
|
|
2009-03-08 11:19:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* common:
|
|
Automatic update of common submodule
|
|
From 3f13e4e to ffa738d
|
|
|
|
2009-03-07 11:44:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* common:
|
|
Automatic update of common submodule
|
|
From 3c7456b to 3f13e4e
|
|
|
|
2009-03-07 10:44:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* common:
|
|
Automatic update of common submodule
|
|
From 57c83f2 to 3c7456b
|
|
|
|
2009-03-06 19:02:58 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* ext/theora/theoradec.c:
|
|
theoradec: parse and use codec_data in the caps
|
|
Parse the codec_data in the caps and use this as the headers.
|
|
Fixes #574169.
|
|
|
|
2009-03-06 18:53:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/riff/riff-media.c:
|
|
riff: add theora mapping
|
|
Add theora mappings. See #574169.
|
|
|
|
2009-03-06 16:31:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
* win32/common/libgstrtsp.def:
|
|
rtsp: Add methods for getting the read/write fds
|
|
API:gst_rtsp_connection_get_readfd()
|
|
API:gst_rtsp_connection_get_writefd()
|
|
|
|
2009-03-06 10:35:01 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* Makefile.am:
|
|
* win32/common/audio-enumtypes.c:
|
|
win32: indent copied *-enumtypes.c files in make win32-update
|
|
|
|
2009-03-06 10:35:56 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* win32/MANIFEST:
|
|
win32: update MANIFEST
|
|
|
|
2009-03-06 10:30:28 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* configure.ac:
|
|
* win32/common/config.h:
|
|
win32: fix configure logic for GST_INSTALL_PLUGINS_HELPER define
|
|
|
|
2009-03-06 10:05:11 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* win32/common/_stdint.h:
|
|
* win32/common/config.h:
|
|
* win32/common/gstrtsp-enumtypes.c:
|
|
* win32/common/interfaces-enumtypes.c:
|
|
* win32/common/multichannel-enumtypes.c:
|
|
* win32/common/pbutils-enumtypes.c:
|
|
* win32/common/video-enumtypes.c:
|
|
* win32/common/video-enumtypes.h:
|
|
win32: update windows files via make win32-update
|
|
Updates win32 files using the new system/hook, and defines HAVE_PROCESS_H,
|
|
which fixes the build of pbutils on windows (#574319).
|
|
|
|
2009-03-06 10:03:31 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* .gitignore:
|
|
gitignore: ignore more
|
|
|
|
2009-03-06 10:37:38 +0100 Julien Moutte <julien@fluendo.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
Fix build on Mac OS X
|
|
|
|
2009-03-05 15:42:23 -0800 Michael Smith <msmith@songbirdnest.com>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
decodebin2: don't stay connected to notify::caps after negotiation
|
|
Disconnect the notify::caps signal in our callback (it'll be re-added
|
|
if we're not, in fact, finished getting complete caps). Ensures that
|
|
caps changes mid-stream (e.g. from an mp3 that changes from
|
|
stereo->mono mid-file) don't cause us to try to add a new pad.
|
|
|
|
2009-03-05 13:48:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtsprange.c:
|
|
rtsp: fix parsing of 'now-' ranges.
|
|
--
|
|
|
|
2009-03-05 12:43:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/examples/dynamic/.gitignore:
|
|
* tests/examples/dynamic/Makefile.am:
|
|
* tests/examples/dynamic/sprinkle.c:
|
|
* tests/examples/dynamic/sprinkle2.c:
|
|
* tests/examples/dynamic/sprinkle3.c:
|
|
examples: add some more sprinkle examples
|
|
Add some more sprinle examples and add some more comments.
|
|
See #574160.
|
|
|
|
2009-03-05 11:57:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
|
docs: add appsrc symbols to standard section
|
|
--
|
|
|
|
2009-03-05 12:27:16 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/adder/gstadder.c:
|
|
adder: add variants for unsigned to fix warnings for unneeded check
|
|
For unsigned int out+in can't be < 0.
|
|
|
|
2009-03-05 10:58:12 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/subparse/gstsubparse.c:
|
|
subparse: use the right variable in debug log, encoding is not yet initialized
|
|
|
|
2009-03-05 10:51:25 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* sys/v4l/v4l_calls.c:
|
|
v4l: add a fixme for broken code, that someone who has a v4l tuner device should fix
|
|
|
|
2009-03-05 10:39:33 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/audioresample/gstaudioresample.c:
|
|
audioresample: add missing break in event handling, remove dead code
|
|
|
|
2009-03-04 16:24:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: do some more cleanup in _close
|
|
Do som more cleanup in gst_rtsp_connection_close() so that it's back into the
|
|
unconnected state as it was allocated.
|
|
|
|
2009-03-04 16:11:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
rtsp: fix the memory management of the url
|
|
Constify the url parameter in _create.
|
|
Make a copy of the url stored in the connection.
|
|
Free the url when the connection is freed.
|
|
|
|
2009-03-04 12:21:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
* win32/common/libgstrtsp.def:
|
|
RTSP: Add support for server tunneling
|
|
Save the tunnelid in the connection. Add a method to retrieve the tunnelid so
|
|
that a server can store and match the id against other tunnel requests.
|
|
Fix the URI in the tunnel requests so that they contain the absolute uri and the
|
|
query string if any instead of just the hostname.
|
|
Transparently base64 decode the input stream when tunneling.
|
|
Add method to set the connection ip address so that it can be included in the
|
|
tunnel response.
|
|
Add method to connect the two tunnel requests.
|
|
Add two callbacks for the async mode to notify a tunnel start and tunnel
|
|
complete event.
|
|
Add method to reset the watch after the connection has been tunneled.
|
|
Various little refactoring to make more stuff reusable.
|
|
API: RTSP::gst_rtsp_connection_set_ip()
|
|
API: RTSP::gst_rtsp_connection_get_tunnelid()
|
|
API: RTSP::gst_rtsp_connection_do_tunnel()
|
|
API: RTSP::gst_rtsp_watch_reset()
|
|
|
|
2009-03-04 12:18:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspdefs.c:
|
|
* gst-libs/gst/rtsp/gstrtspdefs.h:
|
|
rtsp: add new defines for tunneling
|
|
Add two more result codes for tunneling support.
|
|
|
|
2009-03-04 12:12:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspmessage.h:
|
|
rtsp: remove , from last enum member
|
|
Remove , from last enum member to improve compatibility with other compilers.
|
|
|
|
2009-02-28 15:23:20 -0800 LRN <lrn1986@gmail.com>
|
|
|
|
* gst/subparse/gstsubparse.c:
|
|
subparse: Convert regex code to GRegex code
|
|
Fixes: #572993. Patch author prefers to use an alias, contact
|
|
ds if you actually need a real name.
|
|
Signed-off-by: David Schleef <ds@schleef.org>
|
|
|
|
2009-03-02 16:13:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: remove debugging g_message
|
|
--
|
|
|
|
2009-03-02 16:03:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
* win32/common/libgstrtsp.def:
|
|
RTSP: add support for Quicktime tunneled RTSP
|
|
Add support for tunneling RTSP over HTTP.
|
|
Fix documentation some more.
|
|
See also #573173.
|
|
API: RTSP:gst_rtsp_connection_is_tunneled()
|
|
API: RTSP:gst_rtsp_connection_set_tunneled()
|
|
|
|
2009-03-02 15:48:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtsptransport.h:
|
|
* gst-libs/gst/rtsp/gstrtspurl.c:
|
|
RTSP: parse rtsph uris as RTSP tunneled over HTTP
|
|
Add transport define for RTSP tunneled over HTTP.
|
|
Parse rtsph:// uris as tunneled HTTP over TCP.
|
|
API: GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP
|
|
See also #573173.
|
|
|
|
2009-03-02 12:48:18 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* win32/common/libgstrtsp.def:
|
|
win32: Add gst_rtsp_connection_get_url definition
|
|
No, I'm not wim's buildslave, seriously.
|
|
|
|
2009-03-02 10:58:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
rtsp: add _get_url method and separate sockets
|
|
Add gst_rtsp_connection_get_url() method.
|
|
Reserve space for 2 sockets, one for reading and one for writing. Use socket
|
|
pointers to select the read and write sockets. This should allow us to implement
|
|
tunneling over HTTP soon.
|
|
API: RTSP::gst_rtsp_connection_get_url()
|
|
|
|
2009-03-01 18:31:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/app/gstapp-marshal.list:
|
|
app: force automatic rebuild of gstapp-marshal.[ch] after previous change
|
|
The previous change to appsrc/appsink requires people to 'make clean'
|
|
to get the marshallers rebuilt (causing a build failure otherwise).
|
|
Change some lines in the .list file around to force a rebuild of
|
|
these files automatically.
|
|
|
|
2009-02-28 11:07:04 -0800 David Schleef <ds@schleef.org>
|
|
|
|
* configure.ac:
|
|
Bump glib requirement to 2.14
|
|
|
|
2009-02-28 19:37:53 +0100 LRN <lrn1986@gmail.com>
|
|
|
|
* ext/gio/gstgiobasesink.c:
|
|
gio: Use correct format modifier for size_t
|
|
Fixes bug #573528.
|
|
|
|
2009-02-28 19:35:33 +0100 LRN <lrn1986@gmail.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtspconnection: Use correct types for some functions on Win32
|
|
Fixes bug #573529.
|
|
|
|
2009-02-28 13:11:59 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtspconnection: Fix warning about using unitialized value.
|
|
|
|
2009-02-28 12:41:28 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst-libs/gst/riff/riff-ids.h:
|
|
* gst-libs/gst/riff/riff-media.c:
|
|
riff: Add more codec mappings.
|
|
This comes mostly from a review of ffmpeg/libavformat/riff.c
|
|
|
|
2009-02-27 11:14:25 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* ext/alsa/gstalsa.c:
|
|
alsa: release pcminfo after the strdup
|
|
|
|
2009-02-26 17:38:47 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst-libs/gst/rtsp/gstrtsprange.c:
|
|
rtsprange: don't leak the range in case of parsing error.
|
|
Free the gstRTSPTimeRange if we don't return it. Also simplify
|
|
gst_rtsp_range_free() as it is valid to pass NULL to g_free().
|
|
|
|
2009-02-26 16:47:39 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* ext/alsa/gstalsa.c:
|
|
alsa: cleanup name lookup.
|
|
We can break, once we have a name to make sure, we won't read it ever twice.
|
|
|
|
2009-02-26 16:09:03 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/subparse/gstsubparse.c:
|
|
subparse: don't leak line, if flushing
|
|
|
|
2009-02-26 16:03:39 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* ext/gio/gstgiosink.c:
|
|
giosink: reflow error handling to not leak uri
|
|
|
|
2009-02-26 15:53:10 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
|
|
* gst/ffmpegcolorspace/imgconvert.c:
|
|
ffmpegcolorspace: remove unused code/variables
|
|
|
|
2009-02-26 12:10:47 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* sys/ximage/ximagesink.c:
|
|
ximagesink: use GST_FLOW_NOT_NEGOTIATED for partial caps
|
|
|
|
2009-02-26 16:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/app/gstappsink.c:
|
|
* gst-libs/gst/app/gstappsrc.c:
|
|
* gst-libs/gst/app/gstappsrc.h:
|
|
* win32/common/libgstapp.def:
|
|
app: add callbacks to appsrc, cleanups
|
|
Add a uri handler to appsink.
|
|
don't emit signals when we have installed callbacks on appsink.
|
|
Add callbacks to appsrc to replace the signals.
|
|
Add property to disable callbacks in appsrc, default to TRUE for backwards
|
|
compatibility but disable when callbacks are installed.
|
|
API: GstAppSrc::emit-signals
|
|
API: GstAppSrc::gst_app_src_set_emit_signals()
|
|
API: GstAppSrc::gst_app_src_get_emit_signals()
|
|
API: GstAppSrc::gst_app_src_set_callbacks()
|
|
|
|
2009-02-26 11:42:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/app/gstappsink.h:
|
|
* tests/check/elements/appsink.c:
|
|
Appsink: add padding for callbacks + docs
|
|
Add some padding to the callbacks structure just to be safe.
|
|
Remove the now invisible marshaller methods from the docs.
|
|
Fix a comment in the unit test.
|
|
|
|
2009-02-26 09:52:59 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* win32/common/libgstapp.def:
|
|
win32: Add new libgstapp symbol
|
|
|
|
2009-02-26 10:07:21 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
|
docs: clean section.txt file.
|
|
Add appsrc/sink symbols to private, as they are covered in the libs docs.
|
|
|
|
2009-02-26 10:06:23 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/playback/gstplaybasebin.c:
|
|
docs: fix random text after since: tag. Also fix class name to make the docs actual appear.
|
|
|
|
2009-02-26 09:56:16 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
|
* docs/plugins/gst-plugins-base-plugins.interfaces:
|
|
* docs/plugins/gst-plugins-base-plugins.prerequisites:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.xml:
|
|
* docs/plugins/inspect/plugin-app.xml:
|
|
* docs/plugins/inspect/plugin-audioconvert.xml:
|
|
* docs/plugins/inspect/plugin-audiorate.xml:
|
|
* docs/plugins/inspect/plugin-audioresample.xml:
|
|
* docs/plugins/inspect/plugin-audiotestsrc.xml:
|
|
* docs/plugins/inspect/plugin-cdparanoia.xml:
|
|
* docs/plugins/inspect/plugin-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gdp.xml:
|
|
* docs/plugins/inspect/plugin-gio.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.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-queue2.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-uridecodebin.xml:
|
|
* docs/plugins/inspect/plugin-video4linux.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/playback/gstplaybin2.c:
|
|
docs: playbin2 has no stream-info
|
|
|
|
2009-02-26 09:53:03 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst-libs/gst/video/video.h:
|
|
docs: fix newly added interlace constants and plug holes in video format docs
|
|
|
|
2009-02-26 09:35:43 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst-libs/gst/app/gstappsink.c:
|
|
* gst-libs/gst/app/gstappsrc.c:
|
|
* gst-libs/gst/audio/gstaudiofilter.c:
|
|
* gst-libs/gst/audio/gstringbuffer.c:
|
|
* gst-libs/gst/rtp/gstrtcpbuffer.c:
|
|
docs: don't put random stuff in tags.
|
|
Tags like Since: or Returns: can only be followed by more tags. gtk-doc has no
|
|
tag to append text again to the documentation body.
|
|
|
|
2009-02-06 11:10:15 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* sys/ximage/ximagesink.c:
|
|
ximagsink: do not access uninitialized height variable.
|
|
Exit like in xvimagesink, if we have partial caps.
|
|
|
|
2009-02-25 20:26:05 -0800 David Schleef <ds@schleef.org>
|
|
|
|
* Makefile.am:
|
|
* configure.ac:
|
|
* win32/common/config.h.in:
|
|
Change how win32/common/config.h is updated
|
|
Generate win32/common/config.h-new directly from config.h.in,
|
|
using shell variables in configure and some hard-coded information.
|
|
Change top-level makefile so that 'make win32-update' copies the
|
|
generated file to win32/common/config.h, which we keep in source
|
|
control. It's kept in source control so that the git tree is
|
|
buildable from VS.
|
|
This change is similar to the one recently applied to GStreamer,
|
|
except that it adds a few -base specific defines.
|
|
|
|
2009-02-25 19:40:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/app/Makefile.am:
|
|
* gst-libs/gst/app/gstappsink.c:
|
|
* gst-libs/gst/app/gstappsrc.c:
|
|
* win32/common/libgstapp.def:
|
|
app: add win32 .def file and only export functions we want exported
|
|
Add a .def file for win32 builds (and make check-exports).
|
|
Fix LDFLAGS in Makefile.am, so the usual export regexps are used (fixes #573165).
|
|
Make sure private marshaller functions aren't exported by prefixing them with __gst;
|
|
also rename gst_app_marshal_OBJECT__VOID to _BUFFER__VOID, make it static and add
|
|
a comment why we're not using glib-genmarshal for this one.
|
|
|
|
2009-02-25 17:08:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/examples/dynamic/.gitignore:
|
|
* tests/examples/dynamic/Makefile.am:
|
|
* tests/examples/dynamic/sprinkle.c:
|
|
sprinkle: Add another example app
|
|
Add an example app that dynamically adds and removes audiotestsrc elements from
|
|
adder.
|
|
|
|
2009-02-25 16:25:33 +0100 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
Fixed a typo.
|
|
|
|
2009-02-25 11:31:02 +0100 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst/tcp/gstmultifdsink.c:
|
|
rtsp, multifdsink: Unify the use of union gst_sockaddr.
|
|
|
|
2009-02-25 14:22:35 +0000 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* common:
|
|
* configure.ac:
|
|
build: Update shave init statement for changes in common. Bump common.
|
|
|
|
2009-02-25 13:16:32 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* sys/xvimage/xvimagesink.c:
|
|
* sys/xvimage/xvimagesink.h:
|
|
xvimageink: protect buffer_alloc from shutdown
|
|
Use the pool_lock in the buffer_alloc function to detect shutdown. Avoids
|
|
crashes when the sink is shutdown.
|
|
|
|
2009-02-25 12:43:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
playbin: use flushing pads instead of fakesink
|
|
Use the flushing pads on playsink to terminate on shutdown instead of plugging
|
|
fakesinks. this should be a little cheaper.
|
|
|
|
2009-02-25 12:42:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
* gst/playback/gstplaysink.h:
|
|
playsink: Add FLUSHING pad type
|
|
Make it possible to request a flushing pad from the playsink. We can eventually
|
|
use these flushing pads to quickly terminate the dataflow when we are shutting
|
|
down.
|
|
|
|
2009-02-25 11:31:52 +0000 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* common:
|
|
Automatic update of common submodule
|
|
From 9cf8c9b to a6ce5c6
|
|
|
|
2009-02-25 09:52:38 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/riff/riff-media.c:
|
|
riff: add fourcc for mpeg2-in-avi (as produced by mencoder)
|
|
Fixes: #565777
|
|
|
|
2009-02-25 12:07:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/icles/stress-playbin.c:
|
|
stress-playbin: print the current uri
|
|
Print the current uri so that we can more easily see what uri caused a crash or
|
|
error.
|
|
|
|
2009-02-25 11:07:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* tests/icles/stress-playbin.c:
|
|
Print the errors more clearly
|
|
Print some more verbose messages when dealing with errors.
|
|
|
|
2009-02-25 10:08:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
Release the group lock when setting states
|
|
Release the group lock while we perform the state changes on the uridecodebins
|
|
because that might trigger callbacks that we need to handle with the group lock
|
|
taken. Avoids a possible deadly embrace in some id3/flac files.
|
|
Fixes #567396.
|
|
|
|
2009-02-25 10:05:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
Combine finding and creating groups
|
|
Combine the search for the current group and optionally creating one into one
|
|
function so that we can avoid taking the lock multiple times.
|
|
|
|
2009-02-25 08:22:00 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
Playbin2: Don't leave unused parameters in debug statements.
|
|
Fixes build on macosx
|
|
|
|
2009-02-24 10:33:05 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst-libs/gst/riff/riff-media.c:
|
|
Riff: Add fourcc for mpeg1-in-avi (as produced by mencoder)
|
|
|
|
2009-02-24 18:43:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
Add some G_UNLIKELY because we can
|
|
Add a G_UNLIKELY when checking the shutdown variable.
|
|
|
|
2009-02-24 17:23:58 +0000 Garret D'Amore <garrett.damore@sun.com>
|
|
|
|
* gst-libs/gst/interfaces/mixer.h:
|
|
* gst-libs/gst/interfaces/mixertrack.h:
|
|
mixer interface: Add flags to enhance mixer interfaces
|
|
This patch adds a few flags to the mixer and mixerctrl interface to
|
|
better support OSSv4 (and potentially other backends).
|
|
Patch By: Garret D'Amore <garrett.damore@sun.com>
|
|
Signed-Off-By: Jan Schmidt <jan.schmidt@sun.com>
|
|
API: GST_MIXER_FLAG_HAS_WHITELIST, GST_MIXER_FLAG_GROUPING,
|
|
API: GST_MIXER_TRACK_NO_RECORD, GST_MIXER_TRACK_NO_MUTE,
|
|
API: GST_MIXER_TRACK_WHITELIST
|
|
|
|
2009-02-24 17:03:08 +0000 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* gst/tcp/gstmultifdsink.c:
|
|
multifdsink: Fix strict aliasing error using a union
|
|
|
|
2009-02-24 16:49:40 +0000 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
rtsp: Fix a strict aliasing warning
|
|
Fix strict aliasing warnings from casting a sockaddr_storage and
|
|
using it as a sockaddr_in6. Use a union instead.
|
|
|
|
2009-02-24 16:08:49 +0000 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* docs/libs/.gitignore:
|
|
* docs/libs/tmpl/.gitignore:
|
|
* docs/plugins/.gitignore:
|
|
* docs/plugins/tmpl/.gitignore:
|
|
Remove .gitignore files from the docs tmpl dirs, that are killed by make clean.
|
|
|
|
2009-02-24 14:36:39 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* docs/plugins/Makefile.am:
|
|
* ext/vorbis/Makefile.am:
|
|
* ext/vorbis/gstvorbisdec.h:
|
|
* ext/vorbis/gstvorbisenc.h:
|
|
* ext/vorbis/gstvorbisparse.h:
|
|
* ext/vorbis/gstvorbistag.h:
|
|
* ext/vorbis/vorbis.c:
|
|
* ext/vorbis/vorbisdec.c:
|
|
* ext/vorbis/vorbisdec.h:
|
|
* ext/vorbis/vorbisenc.c:
|
|
* ext/vorbis/vorbisenc.h:
|
|
* ext/vorbis/vorbisparse.c:
|
|
* ext/vorbis/vorbisparse.h:
|
|
* ext/vorbis/vorbistag.c:
|
|
* ext/vorbis/vorbistag.h:
|
|
vorbis: Rename vorbis*.h to gstvorbis*.h to prevent name conflicts
|
|
|
|
2009-02-24 14:06:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/ffmpegcolorspace/avcodec.h:
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
|
|
* gst/ffmpegcolorspace/imgconvert.c:
|
|
ffmpegcolorspace: Add conversion from/to YVYU colorspace
|
|
Fixes bug #572872.
|
|
|
|
2009-02-24 13:42:01 +0100 Jonas Danielsson <jonas.danielsson@axis.com>
|
|
|
|
* gst/ffmpegcolorspace/imgconvert.c:
|
|
ffmpegcolorspace: Add direct UYVY->GRAY8 conversion
|
|
The conversion from UYVY to RGB24 and then to GRAY8
|
|
is quite slow. Fixes bug #569655.
|
|
|
|
2009-02-19 17:16:51 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
playbin2: fix deadlock when shutting down. Fixes #572577.
|
|
|
|
2009-02-19 17:15:18 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
|
|
|
|
* tests/icles/stress-playbin.c:
|
|
stress-playbin: make more flexible, e.g. also useful for playbin2
|
|
|
|
2009-02-24 12:11:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
Match WSAStartup and WSACleanup correctly
|
|
Don't randomly call WSAStartup and WSACleanup but instead call the startup when
|
|
we create a connection and cleanup when we free it again. Because the internal
|
|
datastructure is refcounted, this should not cause any refcounting leaks when
|
|
the connection is managed correctly.
|
|
Fixes #562794.
|
|
|
|
2009-02-18 11:59:58 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2/playsink: Set audiotee to PAUSED state in all cases. Fixes #565105.
|
|
|
|
2009-02-23 10:57:42 -0800 David Flynn <davidf@rd.bbc.co.uk>
|
|
|
|
* pkgconfig/gstreamer-app-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-audio-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-cdda-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-fft-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-riff-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-rtp-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-sdp-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-tag-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-video-uninstalled.pc.in:
|
|
Add srcdir to includes for out-of-source builds
|
|
When you use gstreamer uninstalled and build outside
|
|
the source tree, the includes need to be specified for
|
|
both the source tree and the build tree.
|
|
Signed-off-by: David Schleef <ds@schleef.org>
|
|
|
|
2009-02-22 17:23:52 +0000 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* configure.ac:
|
|
* docs/libs/Makefile.am:
|
|
* docs/plugins/Makefile.am:
|
|
Use shave for the build output
|
|
|
|
2009-02-23 12:17:07 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* win32/common/libgstrtsp.def:
|
|
win32: Add new symbol to libgstrtsp.def
|
|
|
|
2009-02-23 10:57:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspextension.c:
|
|
* gst-libs/gst/rtsp/gstrtspextension.h:
|
|
Add method for handling server requests
|
|
Add a receive_request so that extensions can react to server requests.
|
|
|
|
2009-02-22 19:20:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* tests/check/libs/netbuffer.c:
|
|
Correctly cast to GstBuffer * before passing to gst_buffer_(copy|unref)
|
|
|
|
2009-02-22 19:19:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* ext/theora/theoraparse.c:
|
|
theoraparse: Use the correct unref functions
|
|
|
|
2009-02-22 19:18:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* sys/ximage/ximagesink.c:
|
|
* sys/xvimage/xvimagesink.c:
|
|
x(v)imagesink: Correctly cast to GstBuffer * before passing to gst_buffer_unref()
|
|
|
|
2009-02-22 19:12:00 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst-libs/gst/tag/gsttagdemux.c:
|
|
tagdemux: Unref the actual buffer instead of the memory address of the buffer
|
|
|
|
2009-02-22 15:47:53 +0000 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* common:
|
|
Automatic update of common submodule
|
|
From 5d7c9cc to 9cf8c9b
|
|
|
|
2009-02-22 14:49:29 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* win32/common/libgstrtsp.def:
|
|
* win32/common/libgstvideo.def:
|
|
win32/common: Update .def files for recent API addition
|
|
|
|
2009-02-22 13:43:35 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* tests/check/libs/rtp.c:
|
|
tests: Fix indentation
|
|
|
|
2009-02-22 13:42:33 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst-libs/gst/video/video.c:
|
|
libs/video: Fix gst_video_format_new_caps* functions.
|
|
Only add a 'interlaced=True' property to caps *IF* it is interlaced, else
|
|
don't add anything.
|
|
|
|
2009-02-21 11:13:36 -0800 David Schleef <ds@schleef.org>
|
|
|
|
* common:
|
|
Automatic update of common submodule
|
|
From 80c627d to 5d7c9cc
|
|
|
|
2009-02-20 17:26:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspmessage.c:
|
|
Improve key/value parsing
|
|
Improve header field parsing by keeping a ref to the key/value instead of
|
|
copying it into a local variable.
|
|
|
|
2009-02-20 12:35:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
Add trailing \0 to message length
|
|
We always put a trailing 0 at the end of the message body. Reflect this fact in
|
|
the length of the message.
|
|
|
|
2009-02-20 09:50:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
Don't parse headers for data messages
|
|
Don't try to parse the headers on a data message because they don't have
|
|
headers.
|
|
|
|
2009-02-19 12:18:29 -0800 Benjamin M. Schwartz <bens@alum.mit.edu>
|
|
|
|
* ext/theora/gsttheoraenc.h:
|
|
* ext/theora/theoraenc.c:
|
|
theoraenc: Add property for speed level control
|
|
Add property "speed-level" to control the amount of motion searching
|
|
the encoder does. This is only available in libtheora >= 1.0 and
|
|
will silently fail with earlier libraries. Fixes: #572275.
|
|
Signed-off-by: David Schleef <ds@schleef.org>
|
|
|
|
2009-02-19 17:40:45 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst-libs/gst/video/video.c:
|
|
* gst-libs/gst/video/video.h:
|
|
video: Fix 'Since' tags
|
|
|
|
2009-01-26 10:30:53 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/video/video.c:
|
|
* gst-libs/gst/video/video.h:
|
|
video: Add flags for interlaced video along with convenience methods for interlaced caps.
|
|
These three flags allow all know combinations of interlaced formats. They should
|
|
only be used when the caps contain 'interlaced=True'.
|
|
Fixes #163577 (yes, it's a 4 year old bug).
|
|
|
|
2009-02-19 15:51:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
Make RTSPConnection opaque and rename RTSPChannel
|
|
Make the RTSPConnection object opaque so that we can extend it in the future.
|
|
Rename GstRTSPChannel to GstRTSPWatch to avoid confusing with the RTSP channels.
|
|
|
|
2009-01-26 10:31:14 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* gst-libs/gst/riff/riff-media.c:
|
|
Add some more mappings for h264 in riff
|
|
|
|
2009-02-19 10:49:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* win32/common/libgstrtsp.def:
|
|
Add new RTSP symbols to def files
|
|
Add the new RTSP symbols to the windows def file.
|
|
|
|
2009-02-19 10:44:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/app/gstappsink.c:
|
|
* gst-libs/gst/app/gstappsink.h:
|
|
* tests/check/Makefile.am:
|
|
* tests/check/elements/.gitignore:
|
|
* tests/check/elements/appsink.c:
|
|
Add method to install callbacks on appsink
|
|
Based on pacth by Martin Samuelsson <martin dot samuelsson at axis dot com>
|
|
Fixes #571299.
|
|
Add gst_app_sink_set_callbacks() to install a set of callbacks. This is a more
|
|
performant alternative to connecting to the signals.
|
|
Add a unit test for appsink.
|
|
Clean up some of the appsink docs.
|
|
API: GstAppSink::gst_app_sink_set_callbacks()
|
|
|
|
2009-02-18 18:46:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
Add RTSP accept method
|
|
Add a method to accept a connection on a socket and create a GstRTSPConnection
|
|
for it.
|
|
API: gst_rtsp_connection_accept()
|
|
|
|
2009-02-18 17:42:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
Add RTSP channel object for async io
|
|
Add a GstRTSPChannel object that wraps a GSource around the RTSP connection so
|
|
that the connection can be monitored from a maincontext. This allows us to
|
|
operate in ASYNC mode, which is handy when building a server.
|
|
Rework the old code to use the async code under the hood.
|
|
API: gst_rtsp_channel_new()
|
|
API: gst_rtsp_channel_unref()
|
|
API: gst_rtsp_channel_attach()
|
|
API: gst_rtsp_channel_queue_message()
|
|
|
|
2009-02-15 07:30:17 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/audioresample/gstaudioresample.c:
|
|
audioresample: Add locking to protect the resampling context
|
|
When setting the quality/filter-length while PLAYING the
|
|
resampling context will be destroyed and created again in
|
|
some cases, which will cause crashes in the transform function
|
|
if it's called at that time.
|
|
|
|
2009-02-13 10:10:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
|
|
* gst/videotestsrc/videotestsrc.c:
|
|
ffmpegcolorspace/videotestsrc: Use v308 instead of V308
|
|
|
|
2009-02-12 19:02:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/ffmpegcolorspace/avcodec.h:
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
|
|
* gst/ffmpegcolorspace/imgconvert.c:
|
|
* gst/ffmpegcolorspace/imgconvert_template.h:
|
|
ffmpegcolorspace: Add support for packed 4:4:4 YUV (format=V308)
|
|
Only conversions from/to are implemented, which
|
|
gives (indirect) support for all possible conversions.
|
|
Partially fixes bug #571147.
|
|
|
|
2009-02-12 18:17:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videotestsrc/videotestsrc.c:
|
|
videotestsrc: Add support for packed 4:4:4 YUV (format=V308)
|
|
Partially fixes bug #571147.
|
|
|
|
2009-02-12 09:18:20 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/tag/gsttagdemux.c:
|
|
tagdemux: don't abort when downstream pulls a buffer of size 0
|
|
Pulling a 0-sized buffer is allowed, and we should handle this correctly instead of
|
|
aborting. Fixes #571009 (wma file with ID3v2 tag).
|
|
|
|
2009-02-11 16:39:55 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/riff/riff-read.c:
|
|
riff: error out on nonsensical chunk sizes instead of aborting
|
|
When encountering a nonsensical chunk size such as (guint)-1, error out cleanly instead of
|
|
continuing and trying to g_memdup() 4GB of data that doesn't exist, which will either abort
|
|
in g_malloc() or crash.
|
|
Fixes #553295, crash with fuzzed AVI file.
|
|
|
|
2009-02-11 16:39:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* .gitignore:
|
|
Make git ignore backup files.
|
|
|
|
2009-02-10 20:38:58 -0800 Michael Smith <msmith@syncword.(none)>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
Revert "Remove pad-removed handlers after setting the decodebins to NULL."
|
|
This reverts commit b36d8f3e119f9edc5993c08025614ee32642972e.
|
|
This brought back some deadlocks. A small leak is better, for now. Need to
|
|
figure out a way to fix the leak properly.
|
|
|
|
2009-02-10 17:16:07 -0800 Michael Smith <msmith@songbirdnest.com>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
playbin2: Fix segfault on notify after group change.
|
|
If our group has been switched, then we get a selector active-pad
|
|
notification, we don't need to notify.
|
|
|
|
2009-02-10 17:10:33 -0800 Michael Smith <msmith@songbirdnest.com>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: Look for volume/mute properties recursively in audio element.
|
|
Rather than only checking for volume property on the audio sink
|
|
directly, recursively look for it on sinks within it (if it's a bin).
|
|
Allows use of sink-as-volume-control where the application has supplied
|
|
an audio-sink bin that includes a real audio sink internally.
|
|
|
|
2009-02-10 18:29:22 +0000 Christian Schaller <cschalle@crazyhorse.localdomain>
|
|
|
|
* gst-plugins-base.spec.in:
|
|
Update spec file with latest additions and changes, most noteably the move of appsrc appsink into -base
|
|
|
|
2009-02-10 17:39:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* gst/videotestsrc/videotestsrc.c:
|
|
videotestsrc: Add support for Y444 (planar 4:4:4 YUV)
|
|
Partially fixes bug #571147.
|
|
|
|
2009-02-10 17:37:06 +0100 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspmessage.c:
|
|
gstrtspmessage: Minor documentation correction.
|
|
Corrected documentation about what needs to be freed after calling
|
|
gst_rtsp_message_new(), gst_rtsp_message_new_request(),
|
|
gst_rtsp_message_new_response() and gst_rtsp_message_new_data().
|
|
|
|
2009-02-10 11:00:12 +0100 Antoine Tremblay <hexa00@gmail.com>
|
|
|
|
* ext/alsa/gstalsamixer.c:
|
|
alsamixer: Fix race condition that made alsamixer not working properly
|
|
This is due to race conditions between functions that
|
|
modified the mixer like set_volume and
|
|
snd_mixer_handle_events since the handle_events
|
|
can now be called at any time.
|
|
Fixed by adding locking around any snd_mixer call
|
|
since even read functions can modify the mixer stucture, since
|
|
alsa likes to clear it's values before reading new ones.
|
|
The favorite race condition seemed to be that set_volume
|
|
called read_elem (in alsalib) that reset the volumes to
|
|
0 and then read them with read_x_volume. This read looped
|
|
on each channel and as the race condition occured the
|
|
channels value could be anything , most of the time
|
|
it was 0. Thus no value was read or only the value of
|
|
one channel was and the volume was reset to 0.
|
|
Fixes bug #478512.
|
|
|
|
2009-02-09 12:02:21 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* common:
|
|
Bump revision to use for common submodule.
|
|
|
|
2009-02-05 15:47:00 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* sys/xvimage/xvimagesink.c:
|
|
xvimagesink: do not call _xwindow_clear on ready->paused.
|
|
Calling clear at that transition does things like stopping xvideo (which is not
|
|
running at that time) and also clearing anything what the application might have drawn.
|
|
This breaks handle-expose and autopaint-colorkey features.
|
|
|
|
2009-02-04 17:03:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/rtsp/gstrtsprange.c:
|
|
* gst-libs/gst/rtsp/gstrtsprange.h:
|
|
RTSPRange: Add method to serialize ranges
|
|
Add gst_rtsp_range_to_string() to serialize a GstRTSPRange to a string that can
|
|
be used by a server.
|
|
API: GstRTSPRange::gst_rtsp_range_to_string()
|
|
|
|
2009-02-04 13:16:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspurl.c:
|
|
* gst-libs/gst/rtsp/gstrtspurl.h:
|
|
GstRTSPUrl: Add some const to methods
|
|
Add const to the methods that do not modify the object.
|
|
|
|
2009-02-04 13:53:30 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
playbin2: implement GST_PLAY_FLAG_NATIVE_{AUDIO,VIDEO}
|
|
The flags where present but actually not been taken into account.
|
|
|
|
2009-02-04 12:06:38 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/audioresample/gstaudioresample.c:
|
|
audioresample: Add a proper deprecation comment and also drop G_PARAM_CONSTRUCT.
|
|
The comment will ensure that is is marked properly in the docs and the
|
|
GParamSpecflag was causing a duplicated initialisation of the same value.
|
|
|
|
2009-02-04 11:18:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
Add more g_return_if_fail() calls
|
|
Check that we have a valid file descriptor before entering certain functions in
|
|
order to avoid undesirable situations.
|
|
Add some more debugging in the connect method.
|
|
|
|
2009-02-04 10:31:21 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* configure.ac:
|
|
* gst/audioresample/Makefile.am:
|
|
* gst/audioresample/gstaudioresample.c:
|
|
audioresample: Only pull in liboil if its actualy used.
|
|
Liboil still has quite significant startup overhead especialy on embedded
|
|
platforms. In audioresample it was only used for the profiling timer.
|
|
|
|
2009-02-03 15:26:08 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
typefind: Make the flac check more tight to not mistace some aac files for flac. Fixes #570356.
|
|
Add comments about the flac format. Tighten the check to not allow values that
|
|
refer to headers.
|
|
|
|
2009-02-03 10:52:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* win32/common/libgstrtsp.def:
|
|
Add new methods
|
|
Add new methods to the windows def file.
|
|
|
|
2009-02-02 17:25:21 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
* gst-libs/gst/pbutils/install-plugins.c:
|
|
* tests/check/libs/pbutils.c:
|
|
pbutils: remove duplicate detail strings when calling the external codec installer
|
|
It doesn't make sense to ask installers for the same codec or element twice, so filter out duplicate requests before calling the external helper script and make the unit test check this works right. Fixes #567636.
|
|
|
|
2009-02-02 18:05:42 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst-libs/gst/audio/gstaudiosink.c:
|
|
* gst-libs/gst/audio/gstaudiosink.h:
|
|
Add a FIXME 0.11. Make the log message a bit more detailed and add comments.
|
|
|
|
2009-02-02 15:43:03 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* configure.ac:
|
|
* gst/audioresample/gstaudioresample.c:
|
|
Allow to configure the resampler function for integer to skip the benchmarking. Fix releasing the intger resampler in benchmark.
|
|
|
|
2009-02-02 13:30:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* sys/ximage/ximagesink.c:
|
|
Fix buffer_alloc in ximagesink
|
|
Remove some useless debug info that reported wrong image sizes.
|
|
When upstream does not accept out suggested size, fall back to allocating an
|
|
image of the requested width/height instead of the currently configured size.
|
|
The problem is that an image is reused from the pool because the width/height
|
|
match but the caps on the new buffer are the requested caps with possibly
|
|
different height/width resulting in errors.
|
|
|
|
2009-02-02 12:54:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gstdecodebin2.c:
|
|
* gst/playback/gsturidecodebin.c:
|
|
Fix documentation for autoplug-select
|
|
fix the documentation strings for the autoplug-select signal.
|
|
Fixes #570142.
|
|
|
|
2009-02-02 10:09:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspmessage.c:
|
|
Fix string leak in rtspmessage
|
|
when we remove a header field from a message we must free the value associated
|
|
with the key to avoid a memory leak.
|
|
|
|
2009-01-31 18:45:47 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* docs/libs/gst-plugins-base-libs-docs.sgml:
|
|
Its "Base Library" and not just "Library".
|
|
|
|
2009-01-31 18:44:32 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* gst-libs/gst/audio/gstaudiofilter.c:
|
|
Link to the class, as we can't link to the members yet.
|
|
|
|
2009-01-30 17:48:23 -0800 Michael Smith <msmith@songbirdnest.com>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
Remove pad-removed handlers after setting the decodebins to NULL.
|
|
They do needed cleanup; without this we leak selector requestpads.
|
|
|
|
2009-01-30 17:47:07 -0800 Michael Smith <msmith@songbirdnest.com>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
Unref selector request pad even if we no longer have a selector.
|
|
During destruction, we won't have a selector any more, but we still need
|
|
to unref the pad to avoid leaking it.
|
|
|
|
2009-01-30 15:23:23 -0800 Michael Smith <msmith@songbirdnest.com>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
Unref source in playbin2's finalize method
|
|
|
|
2009-01-30 12:04:01 -0800 Michael Smith <msmith@songbirdnest.com>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
Fix more leaks of pads and elements in gstplaysink.
|
|
Don't keep extra references to volume and mute elements; we don't need
|
|
to do so.
|
|
Ensure we unref pads that we have references to, and release request
|
|
pads.
|
|
|
|
2009-01-30 11:04:37 -0800 Michael Smith <msmith@songbirdnest.com>
|
|
|
|
* gst/playback/gstplaysink.c:
|
|
Avoid leaking all playsinks. Fix some internal leaks.
|
|
Playsink was holding references to itself. Don't do that, it's not cool.
|
|
Also, free all chains in dispose.
|
|
|
|
2009-01-30 10:54:12 -0800 Michael Smith <msmith@songbirdnest.com>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
Unref peer request pad after releasing it, since we hold a reference.
|
|
|
|
2009-01-30 10:52:52 -0800 Michael Smith <msmith@songbirdnest.com>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
Fix caps leak in playbin2.
|
|
|
|
2009-01-30 10:51:11 -0800 Michael Smith <msmith@songbirdnest.com>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
Unref active pad from selector when finding active stream.
|
|
|
|
2009-01-30 10:49:55 -0800 Michael Smith <msmith@songbirdnest.com>
|
|
|
|
* gst/playback/gstplaybin2.c:
|
|
Free uris when finalizing playbin2 instance.
|
|
|
|
2009-01-30 10:38:17 -0800 Michael Smith <msmith@songbirdnest.com>
|
|
|
|
* gst/playback/gsturidecodebin.c:
|
|
Unref pads when iterating over them in analyse_source.
|
|
Fixes leak of source's srcpad when using uridecodebin.
|
|
|
|
2009-01-30 22:22:07 +0200 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
|
|
Add releaseinfo with online url.
|
|
|
|
2009-01-30 17:58:15 +0000 Jan Schmidt <jan.schmidt@sun.com>
|
|
|
|
* gst/playback/gstplaybasebin.c:
|
|
Fix compilation warning on Forte
|
|
|
|
2009-01-30 17:16:39 +0000 Jan Schmidt <jan.schmidt@sun.com>
|
|
|
|
* gst/adder/gstadder.c:
|
|
Don't do void pointer arithmetic.
|
|
|
|
2009-01-30 17:25:51 +0000 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* common:
|
|
Bump common
|
|
|
|
2009-01-30 08:50:53 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* autogen.sh:
|
|
* common:
|
|
Use a symbolic link for the pre-commit client-side hook
|
|
|
|
2009-01-30 08:12:42 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* .gitignore:
|
|
Add more files/directories to ignore
|
|
|
|
2009-01-29 14:00:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspdefs.c:
|
|
fix some typos
|
|
Fix some typos in the doc string of the new
|
|
gst_rtsp_options_as_string() method.
|
|
|
|
2009-01-29 11:55:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
* gst-libs/gst/rtsp/gstrtspmessage.c:
|
|
* gst-libs/gst/rtsp/gstrtspmessage.h:
|
|
Add new RTSP message method to set header
|
|
Add gst_rtsp_message_take_header() that takes ownership of the passed header
|
|
value. This allows us to avoid an allocations and memory copy in some
|
|
situations.
|
|
API: GstRTSPMessage::gst_rtsp_message_take_header()
|
|
|
|
2009-01-29 11:51:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
Add new method to docs
|
|
Add the new gst_rtsp_options_as_text() method to the docs.
|
|
|
|
2009-01-28 11:48:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtsp/gstrtspdefs.c:
|
|
* gst-libs/gst/rtsp/gstrtspdefs.h:
|
|
Add method to serialize RTSP options
|
|
Add gst_rtsp_options_as_text() method to serialize a set of RTSP options to a
|
|
string.
|
|
API: GstRTSP::gst_rtsp_options_as_text()
|
|
|
|
2009-01-26 17:59:37 -0800 Michael Smith <msmith@songbirdnest.com>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
Ensure we have sufficient data when using data scan contexts.
|
|
Fixes crashes typefinding things that look like they might contain AAC
|
|
data (but probably aren't actually AAC).
|
|
|
|
2009-01-26 23:32:09 +0000 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* ext/gio/Makefile.am:
|
|
Fix include order for gio plugin
|
|
|
|
2009-01-23 23:59:48 +0000 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* win32/common/config.h:
|
|
Update win32 config.h for 0.10.22.1 dev cycle
|
|
|
|
2009-01-23 23:16:11 +0000 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* .gitignore:
|
|
* docs/libs/.gitignore:
|
|
* gst-libs/gst/audio/.gitignore:
|
|
* gst-libs/gst/video/.gitignore:
|
|
* po/.gitignore:
|
|
* tests/examples/dynamic/.gitignore:
|
|
Extend and clean up git ignores
|
|
|
|
2009-01-23 12:31:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
|
|
* configure.ac:
|
|
* docs/plugins/Makefile.am:
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
|
* docs/plugins/gst-plugins-base-plugins.interfaces:
|
|
* docs/plugins/gst-plugins-base-plugins.prerequisites:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.xml:
|
|
* docs/plugins/inspect/plugin-app.xml:
|
|
* docs/plugins/inspect/plugin-audioconvert.xml:
|
|
* docs/plugins/inspect/plugin-audiorate.xml:
|
|
* docs/plugins/inspect/plugin-audioresample.xml:
|
|
* docs/plugins/inspect/plugin-audiotestsrc.xml:
|
|
* docs/plugins/inspect/plugin-cdparanoia.xml:
|
|
* docs/plugins/inspect/plugin-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gdp.xml:
|
|
* docs/plugins/inspect/plugin-gio.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.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-queue2.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-uridecodebin.xml:
|
|
* docs/plugins/inspect/plugin-video4linux.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/audioresample/Makefile.am:
|
|
* gst/audioresample/README:
|
|
* gst/audioresample/arch.h:
|
|
* gst/audioresample/buffer.c:
|
|
* gst/audioresample/buffer.h:
|
|
* gst/audioresample/debug.c:
|
|
* gst/audioresample/debug.h:
|
|
* gst/audioresample/fixed_arm4.h:
|
|
* gst/audioresample/fixed_arm5e.h:
|
|
* gst/audioresample/fixed_bfin.h:
|
|
* gst/audioresample/fixed_debug.h:
|
|
* gst/audioresample/fixed_generic.h:
|
|
* gst/audioresample/functable.c:
|
|
* gst/audioresample/functable.h:
|
|
* gst/audioresample/gstaudioresample.c:
|
|
* gst/audioresample/gstaudioresample.h:
|
|
* gst/audioresample/resample.c:
|
|
* gst/audioresample/resample.h:
|
|
* gst/audioresample/resample_chunk.c:
|
|
* gst/audioresample/resample_functable.c:
|
|
* gst/audioresample/resample_ref.c:
|
|
* gst/audioresample/resample_sse.h:
|
|
* gst/audioresample/speex_resampler.h:
|
|
* gst/audioresample/speex_resampler_double.c:
|
|
* gst/audioresample/speex_resampler_float.c:
|
|
* gst/audioresample/speex_resampler_int.c:
|
|
* gst/audioresample/speex_resampler_wrapper.h:
|
|
* gst/speexresample/Makefile.am:
|
|
* gst/speexresample/README:
|
|
* gst/speexresample/arch.h:
|
|
* gst/speexresample/fixed_arm4.h:
|
|
* gst/speexresample/fixed_arm5e.h:
|
|
* gst/speexresample/fixed_bfin.h:
|
|
* gst/speexresample/fixed_debug.h:
|
|
* gst/speexresample/fixed_generic.h:
|
|
* gst/speexresample/gstspeexresample.c:
|
|
* gst/speexresample/gstspeexresample.h:
|
|
* gst/speexresample/resample.c:
|
|
* gst/speexresample/resample_sse.h:
|
|
* gst/speexresample/speex_resampler.h:
|
|
* gst/speexresample/speex_resampler_double.c:
|
|
* gst/speexresample/speex_resampler_float.c:
|
|
* gst/speexresample/speex_resampler_int.c:
|
|
* gst/speexresample/speex_resampler_wrapper.h:
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
* tests/check/Makefile.am:
|
|
* tests/check/elements/audioresample.c:
|
|
* tests/check/elements/speexresample.c:
|
|
Rename files and types from speexresample to audioresample
|
|
Rename files and types from speexresample to audioresample
|
|
to finish the move and to prevent any confusion.
|
|
|
|
2009-01-23 11:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* sys/xvimage/xvimagesink.c:
|
|
Add some more debugging to the Xv strides
|
|
Add some more debugging to the strides as they are received from the server and
|
|
the expected strides.
|
|
|
|
2009-01-23 11:40:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
Add typefind function for gsm
|
|
Because core now supports typefindfactories without a typefind function we can
|
|
register a factory fo GSM that will --if all else fails-- assume the file is a
|
|
GSM file based on the registered extension.
|
|
Fixes #566661.
|
|
|
|
2009-01-23 11:37:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst/playback/gsturidecodebin.c:
|
|
Use more performant link function
|
|
We can use gst_element_link_pads() instead of the more generic
|
|
gst_element_link() function because we know the pads. This saves some cycles
|
|
because the more generic function needs to search for possible compatible caps
|
|
etc.
|
|
|
|
2009-01-23 11:33:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/riff/riff-ids.h:
|
|
* gst-libs/gst/riff/riff-media.c:
|
|
Add more codec ids for RIFF formats
|
|
Handle codec ID for various other AAC formats.
|
|
Sync the list of possible codec ids with that of ffmpeg.
|
|
Fixes #567255
|
|
|
|
2009-01-23 11:27:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* ext/theora/theoradec.c:
|
|
Use rounded values for image strides and sizes
|
|
Round up the height before calculating the expected size and
|
|
strides of the output image.
|
|
|
|
2009-01-23 11:23:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* ext/alsa/gstalsasink.c:
|
|
Improve debug message
|
|
Improve the debug message when alsa returns an error.
|
|
|
|
2009-01-23 11:07:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/app/gstappsrc.c:
|
|
Reset queued_bytes counter when flushing
|
|
Set the amount of queued bytes in the internal queue back to 0 when we clear the
|
|
queue.
|
|
Fixes #567982
|
|
|
|
2009-01-23 10:19:27 +0100 Benjamin Gaignard <benjamin@gaignard.net>
|
|
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
Add typefinder for Mobile XMF. Fixes bug #568707.
|
|
|
|
2009-01-23 10:00:11 +0100 Brian Cameron <brian.cameron@sun.com>
|
|
|
|
* configure.ac:
|
|
Fix linking on Solaris. Fixes bug #568482.
|
|
Check for nsl and socket libraries and add them to
|
|
LIBS if they're found. They're needed for socket()
|
|
and gethostbyname() on Solaris.
|
|
|
|
2009-01-22 22:09:47 +0000 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
* gst/playback/gstplaybasebin.c:
|
|
Fix use-after-unref problem noticed by Josep Torra Valles, and run
|
|
gst-indent
|
|
|
|
2009-01-22 17:46:59 +0200 Stefan Kost <ensonic@users.sf.net>
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* common:
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|
Update common snapshot.
|
|
|
|
2009-01-22 13:47:24 +0100 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
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* common:
|
|
Fix pre-commit hook
|
|
|
|
2009-01-22 13:12:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-base
|
|
|
|
2009-01-22 10:14:28 +0100 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
* gst-libs/gst/fft/gstfftf32.c:
|
|
* gst-libs/gst/fft/gstfftf64.c:
|
|
* gst-libs/gst/fft/gstffts16.c:
|
|
* gst-libs/gst/fft/gstffts32.c:
|
|
Reduce the number of allocations for creating FFT contexts
|
|
Reduce the number of allocations from 2 to 1 for every FFT
|
|
context by allocating enough memory for the FFT context
|
|
and passing parts of it to the kissfft allocation functions.
|
|
|
|
2009-01-22 11:32:56 +0000 Jan Schmidt <thaytan@noraisin.net>
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|
|
|
* configure.ac:
|
|
Back to devel -> 0.10.22.1
|
|
|
|
2009-01-22 05:57:53 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* autogen.sh:
|
|
* common:
|
|
Install and use pre-commit indentation hook from common
|
|
|
|
2009-01-21 13:09:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
|
|
|
|
* gst-libs/gst/rtp/gstrtpbuffer.c:
|
|
* tests/check/libs/rtp.c:
|
|
Avoid overflows in the padding checks by doing the check slightly
|
|
differently.
|
|
Add a unit test to check for correct behaviour.
|
|
|
|
2009-01-21 04:31:32 +0100 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
* autogen.sh:
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|
autogen.sh : Use git submodule
|
|
|
|
=== release 0.10.22 ===
|
|
|
|
2009-01-19 23:10:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
* ChangeLog:
|
|
* NEWS:
|
|
* RELEASE:
|
|
* configure.ac:
|
|
* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
|
* docs/plugins/gst-plugins-base-plugins.interfaces:
|
|
* docs/plugins/gst-plugins-base-plugins.prerequisites:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.xml:
|
|
* docs/plugins/inspect/plugin-app.xml:
|
|
* docs/plugins/inspect/plugin-audioconvert.xml:
|
|
* docs/plugins/inspect/plugin-audiorate.xml:
|
|
* docs/plugins/inspect/plugin-audioresample.xml:
|
|
* docs/plugins/inspect/plugin-audiotestsrc.xml:
|
|
* docs/plugins/inspect/plugin-cdparanoia.xml:
|
|
* docs/plugins/inspect/plugin-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gdp.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.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-queue2.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-uridecodebin.xml:
|
|
* docs/plugins/inspect/plugin-video4linux.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:
|
|
* po/LINGUAS:
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/bg.po:
|
|
* po/ca.po:
|
|
* po/cs.po:
|
|
* po/da.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/es.po:
|
|
* po/fi.po:
|
|
* po/fr.po:
|
|
* po/hu.po:
|
|
* po/id.po:
|
|
* po/it.po:
|
|
* po/ja.po:
|
|
* po/lt.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/pl.po:
|
|
* po/pt_BR.po:
|
|
* po/ru.po:
|
|
* po/sk.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
* po/zh_CN.po:
|
|
* win32/common/config.h:
|
|
Release 0.10.22
|
|
Original commit message from CVS:
|
|
Release 0.10.22
|
|
|
|
2009-01-19 22:01:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/bg.po:
|
|
* po/ca.po:
|
|
* po/cs.po:
|
|
* po/da.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/es.po:
|
|
* po/fi.po:
|
|
* po/fr.po:
|
|
* po/hu.po:
|
|
* po/id.po:
|
|
* po/it.po:
|
|
* po/ja.po:
|
|
* po/lt.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/pl.po:
|
|
* po/pt_BR.po:
|
|
* po/ru.po:
|
|
* po/sk.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
* po/zh_CN.po:
|
|
Update .po files
|
|
Original commit message from CVS:
|
|
Update .po files
|
|
|
|
2009-01-16 11:44:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst-libs/gst/fft/: Use correct struct alignment everywhere to prevent unaligned memory accesses, resulting in SIGBUS ...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/fft/_kiss_fft_guts_f32.h:
|
|
* gst-libs/gst/fft/_kiss_fft_guts_f64.h:
|
|
* gst-libs/gst/fft/_kiss_fft_guts_s16.h:
|
|
* gst-libs/gst/fft/_kiss_fft_guts_s32.h:
|
|
* gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc):
|
|
* gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc):
|
|
* gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc):
|
|
* gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc):
|
|
Use correct struct alignment everywhere to prevent unaligned
|
|
memory accesses, resulting in SIGBUS on sparc and probably others.
|
|
Fixes bug #500833.
|
|
|
|
2009-01-16 11:40:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst-libs/gst/tag/gsttagdemux.c: Forward unknown events upstream to allow latency configuration.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
|
|
Forward unknown events upstream to allow latency configuration.
|
|
Fixes bug #567960.
|
|
|
|
2009-01-13 14:47:19 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybin2.c: Provide the right arguments to a debug line.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin2.c: (groups_set_locked_state):
|
|
Provide the right arguments to a debug line.
|
|
|
|
2009-01-13 06:51:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
sys/xvimage/xvimagesink.c: Don't reset the colorkey when element is reused. Fixes #567511.
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c:
|
|
Don't reset the colorkey when element is reused. Fixes #567511.
|
|
|
|
2009-01-09 23:42:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
configure.ac: 0.10.21.3 pre-release
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
0.10.21.3 pre-release
|
|
|
|
2009-01-09 23:13:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/app/gstappsink.c: Store the returned signal id in the right slot when registering the pull-buffer signal.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/app/gstappsink.c:
|
|
Store the returned signal id in the right slot when
|
|
registering the pull-buffer signal.
|
|
Fixes #567168
|
|
Spotted by: Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
2009-01-09 17:17:50 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/interfaces/mixer.c: Small docs addition to clarify that one really mustn't free the constant GList retur...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/mixer.c:
|
|
Small docs addition to clarify that one really mustn't free
|
|
the constant GList returned (#566812).
|
|
|
|
2009-01-08 17:18:24 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
Add GType for GstRTSPUrl and expose a copy function because we can.
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/rtsp/gstrtspurl.c: (register_rtsp_url_type),
|
|
(gst_rtsp_url_get_type), (gst_rtsp_url_copy):
|
|
* gst-libs/gst/rtsp/gstrtspurl.h:
|
|
* win32/common/libgstrtsp.def:
|
|
Add GType for GstRTSPUrl and expose a copy function because we can.
|
|
API: gst_rtsp_url_copy()
|
|
Fixes #567027.
|
|
|
|
2009-01-07 18:36:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
Add plugin dependency for the GIO and GVfs modules.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* ext/gio/gstgio.c: (plugin_init):
|
|
Add plugin dependency for the GIO and GVfs modules.
|
|
Fixes bug #566876.
|
|
|
|
2009-01-07 18:32:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
Add plugin dependency for the gnomevfs modules.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
|
|
Add plugin dependency for the gnomevfs modules.
|
|
Fixes bug #566875.
|
|
|
|
2009-01-07 18:30:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
win32/common/libgstcdda.def: Add new symbol to the list of exported symbols.
|
|
Original commit message from CVS:
|
|
* win32/common/libgstcdda.def:
|
|
Add new symbol to the list of exported symbols.
|
|
|
|
2009-01-07 13:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybin2.c: Fix some comments and docs.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
|
|
(gst_play_bin_set_uri), (gst_play_bin_set_suburi),
|
|
(no_more_pads_cb), (drained_cb), (group_set_locked_state_unlocked),
|
|
(activate_group), (deactivate_group), (groups_set_locked_state),
|
|
(gst_play_bin_change_state):
|
|
Fix some comments and docs.
|
|
Post an error message when we fail to link the selector to the sink.
|
|
Remove pushing of EOS, this seems unneeded.
|
|
Lock the state of deactivated groups so that they don't accidentally
|
|
reactivate when the playbin2 state changes.
|
|
Reuse uridecodebins.
|
|
Unlock and relock state of groups when playbin goes to NULL.
|
|
Fixes #566654.
|
|
Fixes #566341.
|
|
* gst/playback/gsturidecodebin.c: (pad_removed_cb), (type_found):
|
|
Only do something in the pad removed callback when we are dealing with
|
|
our sourcepads because the sinkpads don't have a ghostpad.
|
|
|
|
2009-01-07 10:50:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst-libs/gst/cdda/gstcddabasesrc.*: Make the GType of GstCDDABaseSrcMode public for bindings.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/cdda/gstcddabasesrc.c:
|
|
* gst-libs/gst/cdda/gstcddabasesrc.h:
|
|
Make the GType of GstCDDABaseSrcMode public for bindings.
|
|
Fixes bug #566837.
|
|
|
|
2009-01-06 18:03:51 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
Use new core API to make registry re-scan the plugin whenever visualisations are added or removed (see #350477).
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* ext/libvisual/visual.c: (plugin_init):
|
|
Use new core API to make registry re-scan the plugin
|
|
whenever visualisations are added or removed (see #350477).
|
|
|
|
2009-01-06 17:30:31 +0000 José Alburquerque <jaalburqu@svn.gnome.org>
|
|
|
|
gst-libs/gst/audio/gstaudioclock.*: Make gst_audio_clock_new use const gchar* to ease the wrapping of
|
|
Original commit message from CVS:
|
|
Patch by: José Alburquerque <jaalburqu svn gnome org>
|
|
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_new):
|
|
* gst-libs/gst/audio/gstaudioclock.h:
|
|
Make gst_audio_clock_new use const gchar* to ease the wrapping of
|
|
C++ bindings. Fixes #566723.
|
|
|
|
2009-01-06 12:16:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
Add pkg-config files for libgstapp. Fixes bug #566761.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* pkgconfig/Makefile.am:
|
|
* pkgconfig/gstreamer-app-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-app.pc.in:
|
|
Add pkg-config files for libgstapp. Fixes bug #566761.
|
|
|
|
2009-01-06 11:10:29 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/app/: Make debug categories static. Use _element_class_set_details_simple().
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/app/gstappsink.c:
|
|
* gst-libs/gst/app/gstappsink.h:
|
|
* gst-libs/gst/app/gstappsrc.c:
|
|
* gst-libs/gst/app/gstappsrc.h:
|
|
Make debug categories static. Use _element_class_set_details_simple().
|
|
|
|
2009-01-06 10:56:45 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/app/: Move private data into a private instance struct. Add padding to instance and class structures exp...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/app/gstappsink.c: (_GstAppSinkPrivate),
|
|
(gst_app_sink_class_init), (gst_app_sink_init),
|
|
(gst_app_sink_dispose), (gst_app_sink_finalize),
|
|
(gst_app_sink_unlock_start), (gst_app_sink_unlock_stop),
|
|
(gst_app_sink_flush_unlocked), (gst_app_sink_start),
|
|
(gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
|
|
(gst_app_sink_render), (gst_app_sink_getcaps),
|
|
(gst_app_sink_set_caps), (gst_app_sink_get_caps),
|
|
(gst_app_sink_is_eos), (gst_app_sink_set_emit_signals),
|
|
(gst_app_sink_get_emit_signals), (gst_app_sink_set_max_buffers),
|
|
(gst_app_sink_get_max_buffers), (gst_app_sink_set_drop),
|
|
(gst_app_sink_get_drop), (gst_app_sink_pull_preroll),
|
|
(gst_app_sink_pull_buffer)::
|
|
* gst-libs/gst/app/gstappsink.h: (GstAppSinkPrivate), (_GstAppSink)::
|
|
* gst-libs/gst/app/gstappsrc.c: (_GstAppSrcPrivate),
|
|
(gst_app_src_class_init), (gst_app_src_init),
|
|
(gst_app_src_flush_queued), (gst_app_src_dispose),
|
|
(gst_app_src_finalize), (gst_app_src_set_property),
|
|
(gst_app_src_get_property), (gst_app_src_unlock),
|
|
(gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
|
|
(gst_app_src_is_seekable), (gst_app_src_check_get_range),
|
|
(gst_app_src_query), (gst_app_src_do_seek), (gst_app_src_create),
|
|
(gst_app_src_set_caps), (gst_app_src_get_caps),
|
|
(gst_app_src_set_size), (gst_app_src_get_size),
|
|
(gst_app_src_set_stream_type), (gst_app_src_get_stream_type),
|
|
(gst_app_src_set_max_bytes), (gst_app_src_get_max_bytes),
|
|
(gst_app_src_set_latencies), (gst_app_src_set_latency),
|
|
(gst_app_src_get_latency), (gst_app_src_push_buffer_full),
|
|
(gst_app_src_push_buffer_action), (gst_app_src_end_of_stream)::
|
|
* gst-libs/gst/app/gstappsrc.h: (GstAppSrcPrivate)::
|
|
Move private data into a private instance struct. Add padding to
|
|
instance and class structures exposed in public headers. Add
|
|
Since markers to the gtk-doc blurbs (#566750).
|
|
|
|
2009-01-06 10:50:37 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
tests/examples/app/appsrc_ex.c: Some comments.
|
|
Original commit message from CVS:
|
|
* tests/examples/app/appsrc_ex.c: (main):
|
|
Some comments.
|
|
When pulling a buffer we can get NULL when the element is EOS, don't try
|
|
to unref this NULL buffer.
|
|
|
|
2009-01-06 10:16:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/video/: Fix up build flags and include statement for the new generated enumtypes files, to fix dist.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/video/Makefile.am:
|
|
* gst-libs/gst/video/video.h:
|
|
Fix up build flags and include statement for the new generated
|
|
enumtypes files, to fix dist.
|
|
|
|
2009-01-05 23:04:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* docs/libs/Makefile.am:
|
|
* docs/libs/gst-plugins-base-libs-docs.sgml:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-base-plugins-docs.sgml:
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* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
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* docs/plugins/gst-plugins-base-plugins.args:
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* docs/plugins/gst-plugins-base-plugins.hierarchy:
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* docs/plugins/gst-plugins-base-plugins.interfaces:
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* docs/plugins/gst-plugins-base-plugins.prerequisites:
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* docs/plugins/gst-plugins-base-plugins.signals:
|
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* docs/plugins/inspect/plugin-app.xml:
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* gst-libs/gst/Makefile.am:
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* gst-libs/gst/app/gstappsink.c:
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* gst-libs/gst/app/gstappsrc.c:
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* tests/examples/Makefile.am:
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* tests/examples/app/Makefile.am:
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Move AppSrc/AppSink from gst-plugins-bad. Fixes #564421
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2009-01-05 17:13:13 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstbaseaudiosink.c: Avoid holding the OBJECT_LOCK when calling ringbuffer functions that take the ...
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|
Original commit message from CVS:
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_change_state):
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Avoid holding the OBJECT_LOCK when calling ringbuffer functions that
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take the ringbuffer lock because rinbuffer lock > OBJECT_LOCK. We can do
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this because the async_play method is deprecated and usually not called
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anymore.
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2009-01-05 12:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstplaybin2.c: Disconnect signal handlers before destroying a previous decodebin so that we don't end up...
|
|
Original commit message from CVS:
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* gst/playback/gstplaybin2.c: (notify_source_cb), (activate_group):
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Disconnect signal handlers before destroying a previous decodebin so
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that we don't end up causing deadlocks. Fixes #566586.
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2009-01-05 10:59:35 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/audiotestsrc/gstaudiotestsrc.*: Add property to control pull/push based scheduling.
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Original commit message from CVS:
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* gst/audiotestsrc/gstaudiotestsrc.c:
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(gst_audio_test_src_class_init), (gst_audio_test_src_init),
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(gst_audio_test_src_check_get_range),
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(gst_audio_test_src_set_property),
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(gst_audio_test_src_get_property):
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* gst/audiotestsrc/gstaudiotestsrc.h:
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Add property to control pull/push based scheduling.
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2009-01-02 15:04:13 +0000 Alessandro Decina <alessandro.d@gmail.com>
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Make the seek and colorkey examples depend on gtk+-x11 as they use
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Original commit message from CVS:
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* configure.ac:
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* tests/examples/seek/Makefile.am:
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* tests/icles/Makefile.am:
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Make the seek and colorkey examples depend on gtk+-x11 as they use
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GDK_WINDOW_XID.
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Fixes the build with gtk+-quartz.
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2008-12-31 16:04:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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win32/common/: Add new exports to win32 files.
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Original commit message from CVS:
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* win32/common/libgstaudio.def:
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* win32/common/libgsttag.def:
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* win32/common/libgstvideo.def:
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Add new exports to win32 files.
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2008-12-31 13:31:55 +0000 Edward Hervey <bilboed@bilboed.com>
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gst-libs/gst/tag/gsttagdemux.*: Add GType for GstTagDemuxResult enum.
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Original commit message from CVS:
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* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_result_get_type):
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* gst-libs/gst/tag/gsttagdemux.h:
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Add GType for GstTagDemuxResult enum.
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2008-12-31 13:01:30 +0000 Edward Hervey <bilboed@bilboed.com>
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gst-libs/gst/video/: Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
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Original commit message from CVS:
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* gst-libs/gst/video/Makefile.am:
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* gst-libs/gst/video/video.h:
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Add glib-mkenum for GstVideoFormat enum GTYPE auto-generation.
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This will help bindings to use it.
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2008-12-31 11:20:26 +0000 Edward Hervey <bilboed@bilboed.com>
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Switch glib-mkenum for gst-libs/gst/audio from multichannel- to audio- in order to wrap all enums declarations of tha...
|
|
Original commit message from CVS:
|
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* gst-libs/gst/audio/Makefile.am:
|
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* gst-libs/gst/audio/audio.c:
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* gst-libs/gst/audio/multichannel.h:
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* gst-libs/gst/audio/testchannels.c:
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* win32/MANIFEST:
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* win32/common/audio-enumtypes.c:
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(gst_audio_channel_position_get_type),
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(gst_ring_buffer_state_get_type),
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(gst_ring_buffer_seg_state_get_type),
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(gst_buffer_format_type_get_type), (gst_buffer_format_get_type):
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* win32/common/audio-enumtypes.h:
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* win32/common/multichannel-enumtypes.c:
|
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* win32/common/multichannel-enumtypes.h:
|
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* win32/vs6/grammar.dsp:
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* win32/vs6/libgstaudio.dsp:
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* win32/vs7/libgstaudio.vcproj:
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* win32/vs8/libgstaudio.vcproj:
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Switch glib-mkenum for gst-libs/gst/audio from multichannel- to
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audio- in order to wrap all enums declarations of that library.
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This modification should not matter since that header file is not a
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public header (it will be included by public headers).
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Modify win32 crap^Wfiles accordingly.
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2008-12-30 17:55:07 +0000 Edward Hervey <bilboed@bilboed.com>
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gst-libs/gst/audio/: Complete Sebastien's commit from the 13th by exporting the _slave_method_get_type() methods.
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|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.h:
|
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* gst-libs/gst/audio/gstbaseaudiosink.h:
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Complete Sebastien's commit from the 13th by exporting the
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_slave_method_get_type() methods.
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|
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2008-12-29 16:45:20 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/app/gstappsrc.*: Add properties and methods to configure and retrieve the min and max latencies.
|
|
Original commit message from CVS:
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|
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
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|
(gst_app_src_init), (gst_app_src_set_property),
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(gst_app_src_get_property), (gst_app_src_query),
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(gst_app_src_set_latencies), (gst_app_src_set_latency),
|
|
(gst_app_src_get_latency), (gst_app_src_push_buffer_full):
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* gst-libs/gst/app/gstappsrc.h:
|
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Add properties and methods to configure and retrieve the min and max
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|
latencies.
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|
|
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2008-12-20 17:38:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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ext/: Implement URI query. Fixes bug #562949.
|
|
Original commit message from CVS:
|
|
* ext/gio/gstgiobasesink.c: (gst_gio_base_sink_query):
|
|
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_class_init),
|
|
(gst_gio_base_src_query):
|
|
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_query):
|
|
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
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(gst_gnome_vfs_src_query):
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Implement URI query. Fixes bug #562949.
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|
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2008-12-20 12:48:43 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
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gst/playback/gstplaybin2.c: Add some debug info.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin2.c: (no_more_pads_cb):
|
|
Add some debug info.
|
|
* gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
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(gst_play_sink_reconfigure), (gst_play_sink_request_pad),
|
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(gst_play_sink_release_pad):
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Add some more debug info.
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Reconfigure the audio chain when we switch between raw and encoded audio
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in gapless playback.
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2008-12-20 12:45:03 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
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gst-libs/gst/audio/gstbaseaudiosink.c: Pause the write thread before deactivating and releasing the ringbuffer to avo...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_setcaps):
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|
Pause the write thread before deactivating and releasing the ringbuffer
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to avoid a deadlock when we do gapless playback with different sample
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rates in playbin2. Fixes #564929.
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|
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2008-12-19 13:03:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
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gst-libs/gst/audio/gstbaseaudiosrc.c: Make GstAudioSrcSlaveMethod get_type() function non-static as it's public now.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
Make GstAudioSrcSlaveMethod get_type() function non-static
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|
as it's public now.
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* win32/common/libgstaudio.def:
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* win32/common/libgstnetbuffer.def:
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Add some missing functions to the list of exported symbols.
|
|
|
|
2008-12-18 12:37:33 +0000 Andrew Feren <acferen@yahoo.com>
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|
|
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gst-libs/gst/netbuffer/gstnetbuffer.*: Make gst_netaddress_get_ip4_address fail for v6 addresses.
|
|
Original commit message from CVS:
|
|
Patch by: Andrew Feren <acferen at yahoo dot com>
|
|
* gst-libs/gst/netbuffer/gstnetbuffer.c:
|
|
(gst_netaddress_get_ip4_address), (gst_netaddress_get_ip6_address),
|
|
(gst_netaddress_get_address_bytes),
|
|
(gst_netaddress_set_address_bytes):
|
|
* gst-libs/gst/netbuffer/gstnetbuffer.h:
|
|
Make gst_netaddress_get_ip4_address fail for v6 addresses.
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Make gst_netaddress_get_ip6_address either fail or return the v4
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|
address as a transitional v6 address.
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Add two convenience functions:
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API: gst_netaddress_get_address_bytes()
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API: gst_netaddress_set_address_bytes()
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Fixes #564896.
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2008-12-17 13:51:46 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
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Add appsrc and appsink documentation.
|
|
Original commit message from CVS:
|
|
* docs/plugins/Makefile.am:
|
|
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
|
|
* gst-libs/gst/app/gstappsink.c:
|
|
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init):
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Add appsrc and appsink documentation.
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|
|
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2008-12-17 08:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/adder/: Cleanup variable names to make the adder-loop easier to understand.
|
|
Original commit message from CVS:
|
|
* gst/adder/Makefile.am:
|
|
* gst/adder/gstadder.c:
|
|
Cleanup variable names to make the adder-loop easier to understand.
|
|
Also try to use liboil to spee it up, but ifdef it out as it does not
|
|
make any change for me (Intel pentim M (sse,sse2) please try on other
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|
systems).
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|
|
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2008-12-16 20:16:17 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
Add minimal docs to make the remaining tcp elements show up.
|
|
Original commit message from CVS:
|
|
* docs/plugins/Makefile.am:
|
|
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
|
* gst/tcp/gsttcpclientsink.c:
|
|
* gst/tcp/gsttcpclientsrc.c:
|
|
* gst/tcp/gsttcpserversrc.c:
|
|
Add minimal docs to make the remaining tcp elements show up.
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|
Fixes #564139.
|
|
|
|
2008-12-15 12:02:26 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
examples/app/: Fix example to unref after emiting the push-buffer action.
|
|
Original commit message from CVS:
|
|
* examples/app/appsrc-ra.c: (feed_data):
|
|
* examples/app/appsrc-seekable.c: (feed_data):
|
|
* examples/app/appsrc-stream.c: (read_data):
|
|
* examples/app/appsrc-stream2.c: (feed_data):
|
|
Fix example to unref after emiting the push-buffer action.
|
|
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
|
|
(gst_app_src_push_buffer_full), (gst_app_src_push_buffer),
|
|
(gst_app_src_push_buffer_action):
|
|
Don't take the ref on the buffer in push-buffer action because it's too
|
|
awkward for bindings. Fixes #564482.
|
|
|
|
2008-12-13 19:32:13 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
win32/common/config.h: Update to CVS version.
|
|
Original commit message from CVS:
|
|
* win32/common/config.h:
|
|
Update to CVS version.
|
|
* win32/common/config.h.in:
|
|
Hardcode path to plugin install helper exe, just like we hardcode
|
|
the paths in core. Removes another source of VCS conflicts for
|
|
people hacking gst-plugins-base on systems with autotools.
|
|
|
|
2008-12-13 16:21:12 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
m4/Makefile.am: And a couple more .m4 that don't exist anymore with gettext 0.17
|
|
Original commit message from CVS:
|
|
* m4/Makefile.am:
|
|
And a couple more .m4 that don't exist anymore with gettext 0.17
|
|
|
|
2008-12-13 12:41:56 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
m4/Makefile.am: inttypes.m4 hasn't been available since gettext-0.15, and since we now require gettext >= 0.17 ... we...
|
|
Original commit message from CVS:
|
|
* m4/Makefile.am:
|
|
inttypes.m4 hasn't been available since gettext-0.15, and since we now
|
|
require gettext >= 0.17 ... we can remove it from the list of files to
|
|
dist.
|
|
|
|
2008-12-13 06:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst-libs/gst/audio/: API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API. This is needed for the C...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_slave_method_get_type),
|
|
(gst_base_audio_sink_class_init):
|
|
* gst-libs/gst/audio/gstbaseaudiosink.h:
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
(gst_base_audio_src_slave_method_get_type),
|
|
(gst_base_audio_src_class_init):
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.h:
|
|
API: Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the
|
|
public API. This is needed for the C++ bindings to be able
|
|
to use this base classes. Fixes bug #564200, #564206.
|
|
|
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2008-12-12 19:41:28 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
gst-libs/gst/cdda/gstcddabasesrc.c: Remove erroneous gst_buffer_ref().
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/cdda/gstcddabasesrc.c:
|
|
(gst_cdda_base_src_handle_event):
|
|
Remove erroneous gst_buffer_ref().
|
|
* tests/check/libs/rtp.c: (GST_START_TEST):
|
|
Don't forget to unref the buffer once you're done with it.
|
|
|
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2008-12-12 13:06:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/playback/: XRef to GstXOverlay.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin.c:
|
|
* gst/playback/gstplaybin2.c:
|
|
XRef to GstXOverlay.
|
|
|
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2008-12-12 10:54:45 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
gst/playback/gsturidecodebin.c: Free the factory array when finalizing.
|
|
Original commit message from CVS:
|
|
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_finalize):
|
|
Free the factory array when finalizing.
|
|
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_base_init):
|
|
Use a GstStaticPadTemplate since the src pad caps are fixed.
|
|
|
|
2008-12-12 07:17:21 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
ext/vorbis/vorbisenc.c: Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with pad templates.
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_base_init),
|
|
(gst_vorbis_enc_init):
|
|
Make vorbisenc's pad template behave like vorbisdec's. Fixes a leak with
|
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pad templates.
|
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|
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2008-12-12 07:15:22 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
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gst-libs/gst/riff/riff-media.c: Add mapping for VP6 in avi/riff.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
|
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(gst_riff_create_video_template_caps):
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Add mapping for VP6 in avi/riff.
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|
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2008-12-11 15:49:12 +0000 Edward Hervey <bilboed@bilboed.com>
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|
|
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gst/subparse/samiparse.c: Some versions of libxml seem to be very picky as to strict formatting of the input and neve...
|
|
Original commit message from CVS:
|
|
* gst/subparse/samiparse.c: (sami_context_push_state),
|
|
(sami_context_pop_state), (start_sami_element), (end_sami_element):
|
|
Some versions of libxml seem to be very picky as to strict formatting
|
|
of the input and never 'close' the final </body> tag.
|
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In order to fix that bad behaviour, we trigger the flushing of
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remaining data on both </body> and </sami>.
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Fixes #557365
|
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2008-12-11 12:32:03 +0000 Guillaume Emont <guillaume@fluendo.com>
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|
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gst/typefind/gsttypefindfunctions.c: Add typefinders for MS Word files and OS X .DS_Store files to prevent them to be...
|
|
Original commit message from CVS:
|
|
Patch by: Guillaume Emont <guillaume at fluendo dot com>
|
|
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
|
|
Add typefinders for MS Word files and OS X .DS_Store files to
|
|
prevent them to be recognized as MPEG files. Fixes bug #564098.
|
|
|
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2008-12-11 11:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst/playback/gstplaysink.c: Add some more debug info.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaysink.c: (gen_audio_chain),
|
|
(gst_play_sink_reconfigure):
|
|
Add some more debug info.
|
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Fix linking of just an encoded sink.
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Handle failure to create a sink chain more gracefully than crashing.
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|
|
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2008-12-11 10:33:48 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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tests/check/pipelines/theoraenc.c: Pushing 10 buffers is enough to run the test.
|
|
Original commit message from CVS:
|
|
* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
|
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Pushing 10 buffers is enough to run the test.
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|
|
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2008-12-11 10:28:43 +0000 Wim Taymans <wim.taymans@gmail.com>
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tests/examples/seek/seek.c: Hook up the SKIP seek flag.
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Original commit message from CVS:
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* tests/examples/seek/seek.c: (do_seek), (stop_cb),
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(skip_toggle_cb), (rate_spinbutton_changed_cb), (msg_segment_done),
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(main):
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Hook up the SKIP seek flag.
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2008-12-10 18:43:32 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstplaybin2.c: Error out with a missing-plugin error when the input-selector was not found.
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Original commit message from CVS:
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* gst/playback/gstplaybin2.c: (pad_added_cb):
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Error out with a missing-plugin error when the input-selector was not
|
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found.
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* gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
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Indentation.
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2008-12-10 17:39:32 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstplaysink.c: Use G_DEFINE_TYPE.
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Original commit message from CVS:
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* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
|
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(gst_play_sink_dispose), (gst_play_sink_finalize), (try_element),
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(gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
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(gst_play_sink_send_event), (gst_play_sink_change_state):
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Use G_DEFINE_TYPE.
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Try to set the selected sink to READY before using it. This will allow
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for detection of incompatible formats sooner.
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Don't cause a fatal error when conversion elements are missing but post
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a missing-element message and a warning instead because things might
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still link and run fine.
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Simplyfy the construction of audio and video sink chains.
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2008-12-10 14:55:10 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/ogg/gstoggdemux.c: Use G_DEFINE_TYPE for the OggPad to get some threadsafe type init from glib.
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Original commit message from CVS:
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* ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init),
|
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(gst_ogg_pad_dispose), (gst_ogg_pad_finalize):
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Use G_DEFINE_TYPE for the OggPad to get some threadsafe type
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init from glib.
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2008-12-10 08:19:13 +0000 Luis Menina <liberforce@freeside.fr>
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gst/: Include glib.h instead of a specific GLib header. Including single
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Original commit message from CVS:
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Patch by: Luis Menina <liberforce at freeside dot fr>
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* gst-libs/gst/floatcast/floatcast.h:
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* gst/typefind/gsttypefindfunctions.c:
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Include glib.h instead of a specific GLib header. Including single
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GLib headers is deprecated. Fixes bug #563904.
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2008-12-09 18:30:10 +0000 Julien Moutte <julien@moutte.net>
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gst-libs/gst/riff/riff-media.c: Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
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Original commit message from CVS:
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2008-12-09 Julien Moutte <julien@fluendo.com>
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* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
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Support higher max audio rates for some formats (WAV, Vorbis, LPCM).
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2008-12-09 17:21:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst-libs/gst/riff/riff-read.c: Fix handling of odd chunks in riff metadata.
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Original commit message from CVS:
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* gst-libs/gst/riff/riff-read.c:
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Fix handling of odd chunks in riff metadata.
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2008-12-08 18:44:22 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/volume/gstvolume.c: Use new basetransform vmethod to reconfigure the dynamic properties and any pending volume/mu...
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Original commit message from CVS:
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* gst/volume/gstvolume.c: (gst_volume_class_init),
|
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(volume_before_transform), (volume_transform_ip):
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Use new basetransform vmethod to reconfigure the dynamic properties and
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any pending volume/mute changes. Fixes #563508.
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2008-12-08 18:12:18 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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configure.ac: First check for "theoraenc theoradec" and if that failed check for "theora >= 1.0alpha5". The former ap...
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Original commit message from CVS:
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* configure.ac:
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First check for "theoraenc theoradec" and if that failed check
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for "theora >= 1.0alpha5". The former appeared in 1.0beta3 and
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deprecate the latter. Also linking on Windows fails with just "theora"
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and the version check would fail for the release candidates.
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Fixes bug #563718.
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2008-12-08 15:25:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst/playback/: Add basic docs to decodebin and link to decodebin from decodebin2.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c:
|
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* gst/playback/gstdecodebin2.c:
|
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Add basic docs to decodebin and link to decodebin from decodebin2.
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2008-12-08 12:08:32 +0000 Olivier Crete <tester@tester.ca>
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gst-libs/gst/rtp/gstrtcpbuffer.*: Implement gst_rtcp_packet_remove(). Fixes #563174.
|
|
Original commit message from CVS:
|
|
Patch by: Olivier Crete <tester at tester ca>
|
|
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_remove):
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* gst-libs/gst/rtp/gstrtcpbuffer.h:
|
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Implement gst_rtcp_packet_remove(). Fixes #563174.
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* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
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Add unit test for some RTCP functions.
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2008-12-04 20:09:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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configure.ac: Apparently AC_CONFIG_MACRO_DIR breaks when using more than one macro directory, reverting last change.
|
|
Original commit message from CVS:
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* configure.ac:
|
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Apparently AC_CONFIG_MACRO_DIR breaks when using more
|
|
than one macro directory, reverting last change.
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2008-12-04 19:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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configure.ac: Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to our M4 macros.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Set AC_CONFIG_MACRO_DIR to common/m4 to point autoconf to
|
|
our M4 macros.
|
|
|
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2008-12-03 17:47:44 +0000 Edward Hervey <bilboed@bilboed.com>
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|
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sys/: Clear all flags on buffers returned from the image pool.
|
|
Original commit message from CVS:
|
|
* sys/ximage/ximagesink.c: (gst_ximagesink_buffer_alloc):
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
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Clear all flags on buffers returned from the image pool.
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Fixes #563143
|
|
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2008-12-01 19:36:35 +0000 이문형 <iwings@gmail.com>
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|
|
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gst-libs/gst/app/gstappsrc.c: Don't forget to release the lock again if we bail out because some pad is flushing or w...
|
|
Original commit message from CVS:
|
|
Patch by: 이문형 <iwings at gmail dot com>
|
|
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_push_buffer):
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Don't forget to release the lock again if we bail out because some
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pad is flushing or we've reached EOS, otherwise things will lock up
|
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next time _push_buffer() is called (#562802).
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|
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2008-11-29 13:31:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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Require gettext 0.17 because older versions don't mix with libtool 2.2. At build time an older gettext version will s...
|
|
Original commit message from CVS:
|
|
Patch by: Cygwin Ports maintainer
|
|
<yselkowitz at users dot sourceforge dot net>
|
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* autogen.sh:
|
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* configure.ac:
|
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Require gettext 0.17 because older versions don't mix with libtool
|
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2.2. At build time an older gettext version will still work.
|
|
Fixes bug #556091.
|
|
|
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2008-11-28 13:30:36 +0000 Christian Schaller <uraeus@gnome.org>
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|
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* ChangeLog:
|
|
* gst/speexresample/Makefile.am:
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|
fix build
|
|
Original commit message from CVS:
|
|
fix build
|
|
|
|
2008-11-28 09:44:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
Update documentation of speexresample for the new element name.
|
|
Original commit message from CVS:
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
|
* docs/plugins/gst-plugins-base-plugins.interfaces:
|
|
* docs/plugins/gst-plugins-base-plugins.prerequisites:
|
|
* docs/plugins/inspect/plugin-videorate.xml:
|
|
* gst/speexresample/gstspeexresample.c:
|
|
Update documentation of speexresample for the new element name.
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|
|
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2008-11-28 09:04:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/speexresample/README: Update README with the latest diff between the Speex resampler and our copy.
|
|
Original commit message from CVS:
|
|
* gst/speexresample/README:
|
|
Update README with the latest diff between the Speex resampler
|
|
and our copy.
|
|
|
|
2008-11-28 08:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
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gst/speexresample/gstspeexresample.c: Update the debug category from speex_resample to audioresample.
|
|
Original commit message from CVS:
|
|
* gst/speexresample/gstspeexresample.c: (plugin_init):
|
|
Update the debug category from speex_resample to audioresample.
|
|
|
|
2008-11-27 19:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
Remove audioresample files.
|
|
Original commit message from CVS:
|
|
* gst/audioresample/Makefile.am:
|
|
* gst/audioresample/buffer.c:
|
|
* gst/audioresample/buffer.h:
|
|
* gst/audioresample/debug.c:
|
|
* gst/audioresample/debug.h:
|
|
* gst/audioresample/functable.c:
|
|
* gst/audioresample/functable.h:
|
|
* gst/audioresample/gstaudioresample.c:
|
|
* gst/audioresample/gstaudioresample.h:
|
|
* gst/audioresample/resample.c:
|
|
* gst/audioresample/resample.h:
|
|
* gst/audioresample/resample_chunk.c:
|
|
* gst/audioresample/resample_functable.c:
|
|
* gst/audioresample/resample_ref.c:
|
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* tests/check/elements/audioresample.c:
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|
Remove audioresample files.
|
|
|
|
2008-11-27 17:04:07 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
docs/plugins/inspect/plugin-audioresample.xml: Regenerated for library filename change.
|
|
Original commit message from CVS:
|
|
* docs/plugins/inspect/plugin-audioresample.xml:
|
|
Regenerated for library filename change.
|
|
|
|
2008-11-27 16:57:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
Rename the moved speexresample to audioresample, integrate into the build system and remove the old audioresample fro...
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* docs/plugins/Makefile.am:
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
|
* docs/plugins/gst-plugins-base-plugins.interfaces:
|
|
* docs/plugins/gst-plugins-base-plugins.prerequisites:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.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-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gdp.xml:
|
|
* docs/plugins/inspect/plugin-gio.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.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-queue2.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-uridecodebin.xml:
|
|
* docs/plugins/inspect/plugin-video4linux.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/speexresample/gstspeexresample.c: (plugin_init):
|
|
* gst/speexresample/Makefile.am:
|
|
* tests/check/Makefile.am:
|
|
* tests/check/elements/speexresample.c: (setup_speexresample),
|
|
(GST_START_TEST), (test_pipeline):
|
|
Rename the moved speexresample to audioresample, integrate into the
|
|
build system and remove the old audioresample from the build system.
|
|
Fixes bug #558124, #385061, #346218, #116051.
|
|
|
|
2008-11-27 16:47:41 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosrc.c: Avoid nasty int overflows after about 12 hours and 25 minutes when these code p...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
(gst_base_audio_src_get_offset), (gst_base_audio_src_create):
|
|
Avoid nasty int overflows after about 12 hours and 25 minutes when these
|
|
code paths are triggered.
|
|
A free beer to Håvard Graff for finding this!
|
|
|
|
2008-11-27 11:16:44 +0000 이문형 <iwings@gmail.com>
|
|
|
|
gst-libs/gst/rtsp/gstrtspconnection.c: A successful gst_poll_wait() doesn't always mean successful connect() on
|
|
Original commit message from CVS:
|
|
Patch by: 이문형 <iwings at gmail dot com>
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
(gst_rtsp_connection_connect):
|
|
A successful gst_poll_wait() doesn't always mean successful connect() on
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|
Windows. We should check errors by calling gst_poll_fd_has_error().
|
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See #561924.
|
|
|
|
2008-11-25 16:37:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
tests/check/elements/speexresample.c: Make unit test again faster to prevent timeouts with valgrind.
|
|
Original commit message from CVS:
|
|
* tests/check/elements/speexresample.c: (test_pipeline):
|
|
Make unit test again faster to prevent timeouts with valgrind.
|
|
|
|
2008-11-25 15:33:30 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstrtcpbuffer.c: Fix typo in the docs.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstrtcpbuffer.c:
|
|
Fix typo in the docs.
|
|
|
|
2008-11-25 15:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/ogg/gstoggdemux.c: If no stream was found before receiving EOS, post an error message.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_event):
|
|
If no stream was found before receiving EOS, post an error message.
|
|
Fixes #561924.
|
|
|
|
2008-11-25 15:14:30 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/theora/: Parse segment events.
|
|
Original commit message from CVS:
|
|
* ext/theora/gsttheoraenc.h:
|
|
* ext/theora/theoraenc.c: (gst_theora_enc_init),
|
|
(theora_buffer_from_packet), (theora_push_packet),
|
|
(theora_enc_sink_event), (theora_enc_is_discontinuous),
|
|
(theora_enc_chain):
|
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Parse segment events.
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Pass incomming buffer timestamps to outgoing buffers.
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Use the running_time to construct the granulepos.
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Fixes #562163.
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|
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2008-11-25 11:00:55 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybin2.c: Fix buffer-duration property.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin2.c: (activate_group):
|
|
Fix buffer-duration property.
|
|
|
|
2008-11-25 10:32:49 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.c: Really fix audiosink drain handling by keeping track of the running_time of th...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
|
|
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
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(gst_base_audio_sink_change_state):
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Really fix audiosink drain handling by keeping track of the running_time
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of the last sample.
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2008-11-24 20:25:24 +0000 Michael Smith <msmith@xiph.org>
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|
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gst/playback/gstplaybin2.c: Add notification of current stream. Add ability to configure buffer sizes.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin2.c:
|
|
Add notification of current stream. Add ability to configure buffer
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|
sizes.
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|
* gst/playback/gsturidecodebin.c:
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Add ability to configure buffer sizes for streaming mode.
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Bug #561734.
|
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2008-11-24 20:11:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
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gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove base_time handling. Fixes audiosinks n...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
Time is already in running_time. Remove base_time handling. Fixes
|
|
audiosinks not draining and thus chopping some audio in the end.
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|
|
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2008-11-24 19:18:59 +0000 David Schleef <ds@schleef.org>
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|
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ext/ogg/gstoggmux.*: If we're muxing a dirac stream, flush the page after every picture.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggmux.c:
|
|
* ext/ogg/gstoggmux.h:
|
|
If we're muxing a dirac stream, flush the page after every picture.
|
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|
|
2008-11-24 12:56:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
|
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gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for audio_drained. Sync one log message with the ...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
Add one log message to check for audio_drained. Sync one log message
|
|
with the condition. Send EOS after draining audio in pull mode.
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|
|
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2008-11-24 12:07:10 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
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ext/: Use gst_buffer_try_new_and_alloc() and fail properly if the allocation failed. This prevents abort() if downstr...
|
|
Original commit message from CVS:
|
|
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
|
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* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
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|
Use gst_buffer_try_new_and_alloc() and fail properly if the
|
|
allocation failed. This prevents abort() if downstream elements
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request an insane amount of memory.
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|
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2008-11-24 12:03:11 +0000 Jon Trowbridge <trow@ximian.com>
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|
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gst/volume/gstvolume.*: Cleanup volume, define and use default values.
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|
Original commit message from CVS:
|
|
* gst/volume/gstvolume.c: (volume_choose_func),
|
|
(volume_update_volume), (gst_volume_set_volume),
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|
(gst_volume_get_volume), (gst_volume_set_mute),
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|
(gst_volume_class_init), (gst_volume_init),
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|
(volume_process_double), (volume_process_float),
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|
(volume_process_int32), (volume_process_int32_clamp),
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|
(volume_process_int24), (volume_process_int24_clamp),
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|
(volume_process_int16), (volume_process_int16_clamp),
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|
(volume_process_int8), (volume_process_int8_clamp), (volume_setup),
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(volume_transform_ip), (volume_set_property),
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(volume_get_property):
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* gst/volume/gstvolume.h:
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Cleanup volume, define and use default values.
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Recalculate new volume and mute setup before processing. Fixes #561789.
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* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
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Add controller unit test. Patch by: Jonathan Matthew
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Fix bogus test that messed with basetransform's internal state.
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|
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2008-11-22 15:02:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
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tests/check/elements/speexresample.c: Make the unit test a bit faster to prevent timeouts, especially with valgrind.
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|
Original commit message from CVS:
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|
* tests/check/elements/speexresample.c: (GST_START_TEST):
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Make the unit test a bit faster to prevent timeouts, especially
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|
with valgrind.
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2008-11-22 14:44:26 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/videorate/gstvideorate.c: Add jpeg and png image media types to the caps. Fixes #561436.
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|
Original commit message from CVS:
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|
* gst/videorate/gstvideorate.c:
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Add jpeg and png image media types to the caps. Fixes #561436.
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|
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2008-11-22 14:31:43 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst/playback/gstplaysink.c: Don't post an error when we can't configure the volume but post a warning instead. Fixes ...
|
|
Original commit message from CVS:
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|
* gst/playback/gstplaysink.c: (gen_audio_chain):
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|
Don't post an error when we can't configure the volume but post a
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|
warning instead. Fixes #561780.
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|
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2008-11-21 20:32:56 +0000 Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
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|
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gst/videotestsrc/: Add a zone plate pattern generator based on BBC R&D Report 1978/23 (yeah *that* 1978). Try 'video...
|
|
Original commit message from CVS:
|
|
Patch by: Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk>
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* gst/videotestsrc/gstvideotestsrc.c:
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* gst/videotestsrc/gstvideotestsrc.h:
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* gst/videotestsrc/videotestsrc.c:
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* gst/videotestsrc/videotestsrc.h:
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Add a zone plate pattern generator based on BBC R&D Report
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1978/23 (yeah *that* 1978). Try 'videotestsrc pattern=zone-plate
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kx2=20 ky2=20 kt=1'.
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2008-11-21 15:45:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
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gst/speexresample/gstspeexresample.c: Add a "filter-length" property that maps to the quality values for compatibilty...
|
|
Original commit message from CVS:
|
|
* gst/speexresample/gstspeexresample.c:
|
|
(gst_speex_resample_class_init), (gst_speex_resample_set_property),
|
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(gst_speex_resample_get_property):
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Add a "filter-length" property that maps to the quality values
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for compatibilty with audioresample.
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2008-11-21 00:04:48 +0000 Michael Smith <msmith@xiph.org>
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gst/playback/gstdecodebin2.c: Fix random fat-fingering making this not compile.
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|
Original commit message from CVS:
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|
* gst/playback/gstdecodebin2.c:
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Fix random fat-fingering making this not compile.
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|
|
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2008-11-20 22:11:38 +0000 Michael Smith <msmith@xiph.org>
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|
|
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gst/playback/gstdecodebin2.c: If the top-level type of the stream is plain text, don't try to decode it, matching beh...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin2.c:
|
|
If the top-level type of the stream is plain text, don't try to decode
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|
it, matching behaviour of decodebin.
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* gst/playback/gstplaysink.c:
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If we fail to generate a text chain (e.g. due to missing optional
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plugins), don't crash.
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|
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2008-11-20 22:06:05 +0000 Michael Smith <msmith@xiph.org>
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|
|
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gst-libs/gst/rtsp/gstrtspdefs.c: Fix win32 build. Oops.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtsp/gstrtspdefs.c:
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|
Fix win32 build. Oops.
|
|
|
|
2008-11-20 21:40:49 +0000 Michael Smith <msmith@xiph.org>
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|
|
|
gst-libs/gst/rtsp/gstrtspdefs.c: Use WSAGetLastError() rather than errno/h_errno on win32.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtsp/gstrtspdefs.c:
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|
Use WSAGetLastError() rather than errno/h_errno on win32.
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|
|
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2008-11-20 21:20:27 +0000 Michael Smith <msmith@xiph.org>
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|
|
|
gst-libs/gst/riff/riff-media.c: Support WMA Lossless properly.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c:
|
|
Support WMA Lossless properly.
|
|
|
|
2008-11-19 00:24:44 +0000 David Schleef <ds@schleef.org>
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|
|
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gst/videotestsrc/: Add "colorspec" property, specifying whether to generate BT.601 or BT.709 video. This only affect...
|
|
Original commit message from CVS:
|
|
* gst/videotestsrc/gstvideotestsrc.c:
|
|
* gst/videotestsrc/gstvideotestsrc.h:
|
|
* gst/videotestsrc/videotestsrc.c:
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* gst/videotestsrc/videotestsrc.h:
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Add "colorspec" property, specifying whether to generate BT.601
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or BT.709 video. This only affects YCbCr values, not RGB, since
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if you're generating a 709 test pattern, presumably you want
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709 RGB primaries, not 601. Also add "smpte75" pattern, which
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uses 75% colors instead of 100%, since this is often more useful
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for testing (and also follows the SMPTE EG-1 guideline).
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|
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2008-11-18 18:08:42 +0000 Alessandro Decina <alessandro.d@gmail.com>
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|
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gst/playback/gstdecodebin.c: Add a "sink-caps" property to decodebin like it's done for decodebin2.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c:
|
|
Add a "sink-caps" property to decodebin like it's done for decodebin2.
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Fixes #560380.
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|
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2008-11-14 21:44:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
|
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gst/audioresample/gstaudioresample.c: Guard against a NULL dereference I somehow encountered - with a FLUSH_STOP arri...
|
|
Original commit message from CVS:
|
|
* gst/audioresample/gstaudioresample.c:
|
|
Guard against a NULL dereference I somehow encountered -
|
|
with a FLUSH_STOP arriving either before basetransform _start(),
|
|
or after _stop().
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* gst/typefind/gsttypefindfunctions.c:
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|
Make sure we never jump backwards when typefinding corrupt mov files.
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|
|
|
2008-11-14 21:39:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/interfaces/propertyprobe.c: Fix random type causing a docs warning.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/propertyprobe.c:
|
|
Fix random type causing a docs warning.
|
|
|
|
2008-11-14 15:40:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
sys/v4l/gstv4l.c: Give it a minimal rank for autovideosrc.
|
|
Original commit message from CVS:
|
|
* sys/v4l/gstv4l.c:
|
|
Give it a minimal rank for autovideosrc.
|
|
|
|
2008-11-13 21:11:13 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Improve typefinding of ISO JPEG2000 mime types.
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (jp2_type_find),
|
|
(plugin_init):
|
|
Improve typefinding of ISO JPEG2000 mime types.
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|
|
|
2008-11-13 18:18:32 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
sys/xvimage/xvimagesink.*: Avoid typechecking when we do trivial casts.
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
|
|
(gst_xvimagesink_xvimage_put), (gst_xvimagesink_setcaps),
|
|
(gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
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|
* sys/xvimage/xvimagesink.h:
|
|
Avoid typechecking when we do trivial casts.
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|
Move error handling out of the main program flow.
|
|
Sneak in the display-region caps property, not completely correct yet.
|
|
Cache the width/height in buffer_alloc instead of parsing it from the
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caps all the time.
|
|
|
|
2008-11-13 17:27:37 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybin2.c: don't try to unlink the selector sinkpad when we don't have it yet. This can happen if an...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin2.c: (deactivate_group):
|
|
don't try to unlink the selector sinkpad when we don't have it yet. This
|
|
can happen if an error occured before the group was complete.
|
|
|
|
2008-11-13 15:37:40 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstrtpbuffer.c: Avoid expensive type checks we already did as part of the _validate() function that ...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
|
|
(gst_rtp_buffer_set_packet_len), (gst_rtp_buffer_get_packet_len),
|
|
(gst_rtp_buffer_get_header_len), (gst_rtp_buffer_get_version),
|
|
(gst_rtp_buffer_set_version), (gst_rtp_buffer_get_padding),
|
|
(gst_rtp_buffer_set_padding), (gst_rtp_buffer_pad_to),
|
|
(gst_rtp_buffer_get_extension), (gst_rtp_buffer_set_extension),
|
|
(gst_rtp_buffer_get_extension_data),
|
|
(gst_rtp_buffer_set_extension_data), (gst_rtp_buffer_get_ssrc),
|
|
(gst_rtp_buffer_set_ssrc), (gst_rtp_buffer_get_csrc_count),
|
|
(gst_rtp_buffer_get_csrc), (gst_rtp_buffer_set_csrc),
|
|
(gst_rtp_buffer_get_marker), (gst_rtp_buffer_set_marker),
|
|
(gst_rtp_buffer_get_payload_type),
|
|
(gst_rtp_buffer_set_payload_type), (gst_rtp_buffer_get_seq),
|
|
(gst_rtp_buffer_set_seq), (gst_rtp_buffer_get_timestamp),
|
|
(gst_rtp_buffer_set_timestamp),
|
|
(gst_rtp_buffer_get_payload_subbuffer),
|
|
(gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload):
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|
Avoid expensive type checks we already did as part of the
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|
_validate() function that should be called first.
|
|
|
|
2008-11-11 16:40:50 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some cases where a newsegment event was not sent.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c: (create_segment_event),
|
|
(gst_base_rtp_depayload_push_full),
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|
(gst_base_rtp_depayload_set_gst_timestamp):
|
|
Fix some cases where a newsegment event was not sent.
|
|
|
|
2008-11-11 15:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybin2.c: Catch state change errors and stop from the uridecodebin elements instead of trying to co...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin2.c: (activate_group):
|
|
Catch state change errors and stop from the uridecodebin elements
|
|
instead of trying to continue in vain.
|
|
|
|
2008-11-10 14:53:45 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
gst/: Wim, you're a bad boy. You don't want people to contact you or what?
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/app/gstappsink.c:
|
|
* gst-libs/gst/app/gstappsrc.c:
|
|
* gst/h264parse/gsth264parse.c:
|
|
Wim, you're a bad boy. You don't want people to contact you or what?
|
|
|
|
2008-11-10 14:22:09 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting for the ...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
|
|
(gst_base_audio_sink_callback):
|
|
Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting
|
|
for the latency to expire, fixes #559567.
|
|
|
|
2008-11-10 13:55:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
gst/adder/gstadder.c: Change author string after seeing output of gst-inspector.
|
|
Original commit message from CVS:
|
|
* gst/adder/gstadder.c:
|
|
Change author string after seeing output of gst-inspector.
|
|
|
|
2008-11-10 10:33:26 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaysink.c: Don't try to do crazy things when we only have a text pad without a video pad. Fixes #559...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaysink.c: (gst_play_sink_reconfigure):
|
|
Don't try to do crazy things when we only have a text pad without a
|
|
video pad. Fixes #559478.
|
|
|
|
2008-11-07 17:35:46 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/app/gstappsrc.*: Add is-live property.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
|
|
(gst_app_src_init), (gst_app_src_set_property),
|
|
(gst_app_src_get_property), (gst_app_src_push_buffer):
|
|
* gst-libs/gst/app/gstappsrc.h:
|
|
Add is-live property.
|
|
Add some more docs.
|
|
|
|
2008-11-06 12:14:51 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/riff/riff-media.c: Fix case where we don't have a range for the rates or channels as is the case with tr...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
|
|
Fix case where we don't have a range for the rates or channels as is the
|
|
case with truespeech.
|
|
|
|
2008-11-05 19:18:25 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/volume/gstvolume.*: Keep negotiated state in a separate variable.
|
|
Original commit message from CVS:
|
|
* gst/volume/gstvolume.c: (volume_update_real_volume),
|
|
(gst_volume_set_volume), (gst_volume_get_volume),
|
|
(gst_volume_set_mute), (gst_volume_init), (volume_setup),
|
|
(volume_transform_ip), (volume_update_mute),
|
|
(volume_update_volume), (volume_get_property):
|
|
* gst/volume/gstvolume.h:
|
|
Keep negotiated state in a separate variable.
|
|
Protect the volume and mute properties with the object lock.
|
|
Protect modifying the transform with the transform lock.
|
|
|
|
2008-11-05 12:20:21 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/ffmpegcolorspace/gstffmpegcodecmap.c: Only convert caps to string when debug is enabled.
|
|
Original commit message from CVS:
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
|
|
(gst_ffmpeg_pixfmt_to_caps):
|
|
Only convert caps to string when debug is enabled.
|
|
|
|
2008-11-04 18:17:24 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/theora/: Copy seqnum.
|
|
Original commit message from CVS:
|
|
* ext/theora/gsttheoradec.h:
|
|
* ext/theora/theoradec.c: (gst_theora_dec_init),
|
|
(gst_theora_dec_reset), (theora_dec_src_event),
|
|
(theora_dec_sink_event), (theora_handle_type_packet):
|
|
Copy seqnum.
|
|
Keep events in a pending list, like vorbisdec, instead of trying
|
|
to construct a segment event ourselves.
|
|
* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
|
|
(vorbis_dec_src_event), (vorbis_dec_sink_event):
|
|
* ext/vorbis/vorbisdec.h:
|
|
Copy seqnum.
|
|
|
|
2008-11-04 17:24:35 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/ogg/gstoggdemux.*: Copy seqnums around to track playback segments and messages.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
|
|
(gst_ogg_demux_deactivate_current_chain),
|
|
(gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page),
|
|
(gst_ogg_demux_loop):
|
|
* ext/ogg/gstoggdemux.h:
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|
Copy seqnums around to track playback segments and messages.
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|
|
|
2008-11-04 12:42:18 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
Don't install static libs for plugins. Fixes #550851 for -bad.
|
|
Original commit message from CVS:
|
|
* ext/alsaspdif/Makefile.am:
|
|
* ext/amrwb/Makefile.am:
|
|
* ext/apexsink/Makefile.am:
|
|
* ext/arts/Makefile.am:
|
|
* ext/artsd/Makefile.am:
|
|
* ext/audiofile/Makefile.am:
|
|
* ext/audioresample/Makefile.am:
|
|
* ext/bz2/Makefile.am:
|
|
* ext/cdaudio/Makefile.am:
|
|
* ext/celt/Makefile.am:
|
|
* ext/dc1394/Makefile.am:
|
|
* ext/dirac/Makefile.am:
|
|
* ext/directfb/Makefile.am:
|
|
* ext/divx/Makefile.am:
|
|
* ext/dts/Makefile.am:
|
|
* ext/faac/Makefile.am:
|
|
* ext/faad/Makefile.am:
|
|
* ext/gsm/Makefile.am:
|
|
* ext/hermes/Makefile.am:
|
|
* ext/ivorbis/Makefile.am:
|
|
* ext/jack/Makefile.am:
|
|
* ext/jp2k/Makefile.am:
|
|
* ext/ladspa/Makefile.am:
|
|
* ext/lcs/Makefile.am:
|
|
* ext/libfame/Makefile.am:
|
|
* ext/libmms/Makefile.am:
|
|
* ext/metadata/Makefile.am:
|
|
* ext/mpeg2enc/Makefile.am:
|
|
* ext/mplex/Makefile.am:
|
|
* ext/musepack/Makefile.am:
|
|
* ext/musicbrainz/Makefile.am:
|
|
* ext/mythtv/Makefile.am:
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|
* ext/nas/Makefile.am:
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|
* ext/neon/Makefile.am:
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|
* ext/ofa/Makefile.am:
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|
* ext/polyp/Makefile.am:
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|
* ext/resindvd/Makefile.am:
|
|
* ext/sdl/Makefile.am:
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|
* ext/shout/Makefile.am:
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|
* ext/snapshot/Makefile.am:
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|
* ext/sndfile/Makefile.am:
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|
* ext/soundtouch/Makefile.am:
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|
* ext/spc/Makefile.am:
|
|
* ext/swfdec/Makefile.am:
|
|
* ext/tarkin/Makefile.am:
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|
* ext/theora/Makefile.am:
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|
* ext/timidity/Makefile.am:
|
|
* ext/twolame/Makefile.am:
|
|
* ext/x264/Makefile.am:
|
|
* ext/xine/Makefile.am:
|
|
* ext/xvid/Makefile.am:
|
|
* gst-libs/gst/app/Makefile.am:
|
|
* gst-libs/gst/dshow/Makefile.am:
|
|
* gst/aiffparse/Makefile.am:
|
|
* gst/app/Makefile.am:
|
|
* gst/audiobuffer/Makefile.am:
|
|
* gst/bayer/Makefile.am:
|
|
* gst/cdxaparse/Makefile.am:
|
|
* gst/chart/Makefile.am:
|
|
* gst/colorspace/Makefile.am:
|
|
* gst/dccp/Makefile.am:
|
|
* gst/deinterlace/Makefile.am:
|
|
* gst/deinterlace2/Makefile.am:
|
|
* gst/dvdspu/Makefile.am:
|
|
* gst/festival/Makefile.am:
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|
* gst/filter/Makefile.am:
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|
* gst/flacparse/Makefile.am:
|
|
* gst/flv/Makefile.am:
|
|
* gst/games/Makefile.am:
|
|
* gst/h264parse/Makefile.am:
|
|
* gst/librfb/Makefile.am:
|
|
* gst/mixmatrix/Makefile.am:
|
|
* gst/modplug/Makefile.am:
|
|
* gst/mpeg1sys/Makefile.am:
|
|
* gst/mpeg4videoparse/Makefile.am:
|
|
* gst/mpegdemux/Makefile.am:
|
|
* gst/mpegtsmux/Makefile.am:
|
|
* gst/mpegvideoparse/Makefile.am:
|
|
* gst/mve/Makefile.am:
|
|
* gst/nsf/Makefile.am:
|
|
* gst/nuvdemux/Makefile.am:
|
|
* gst/overlay/Makefile.am:
|
|
* gst/passthrough/Makefile.am:
|
|
* gst/pcapparse/Makefile.am:
|
|
* gst/playondemand/Makefile.am:
|
|
* gst/rawparse/Makefile.am:
|
|
* gst/real/Makefile.am:
|
|
* gst/rtjpeg/Makefile.am:
|
|
* gst/rtpmanager/Makefile.am:
|
|
* gst/scaletempo/Makefile.am:
|
|
* gst/sdp/Makefile.am:
|
|
* gst/selector/Makefile.am:
|
|
* gst/smooth/Makefile.am:
|
|
* gst/smoothwave/Makefile.am:
|
|
* gst/speed/Makefile.am:
|
|
* gst/speexresample/Makefile.am:
|
|
* gst/stereo/Makefile.am:
|
|
* gst/subenc/Makefile.am:
|
|
* gst/tta/Makefile.am:
|
|
* gst/vbidec/Makefile.am:
|
|
* gst/videodrop/Makefile.am:
|
|
* gst/videosignal/Makefile.am:
|
|
* gst/virtualdub/Makefile.am:
|
|
* gst/vmnc/Makefile.am:
|
|
* gst/y4m/Makefile.am:
|
|
* sys/acmenc/Makefile.am:
|
|
* sys/cdrom/Makefile.am:
|
|
* sys/dshowdecwrapper/Makefile.am:
|
|
* sys/dshowsrcwrapper/Makefile.am:
|
|
* sys/dvb/Makefile.am:
|
|
* sys/dxr3/Makefile.am:
|
|
* sys/fbdev/Makefile.am:
|
|
* sys/oss4/Makefile.am:
|
|
* sys/qcam/Makefile.am:
|
|
* sys/qtwrapper/Makefile.am:
|
|
* sys/vcd/Makefile.am:
|
|
* sys/wininet/Makefile.am:
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|
* win32/common/config.h:
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|
Don't install static libs for plugins. Fixes #550851 for -bad.
|
|
|
|
2008-11-03 15:30:14 +0000 Matthias Kretz <kretz@kde.org>
|
|
|
|
ext/alsa/gstalsasink.c: Make all access non-blocking so that we can better handle unplugging of usb devices. Fixes #5...
|
|
Original commit message from CVS:
|
|
Based on patch by: Matthias Kretz <kretz at kde dot org>
|
|
* ext/alsa/gstalsasink.c: (gst_alsasink_open),
|
|
(gst_alsasink_prepare), (gst_alsasink_unprepare),
|
|
(gst_alsasink_write):
|
|
Make all access non-blocking so that we can better handle unplugging
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|
of usb devices. Fixes #559111
|
|
|
|
2008-11-03 10:49:24 +0000 Damien Lespiau <damien.lespiau@gmail.com>
|
|
|
|
gst-libs/gst/rtsp/gstrtspconnection.c: Make the next call to poll not depend on previous calls to poll with or withou...
|
|
Original commit message from CVS:
|
|
Patch by: Damien Lespiau <damien.lespiau gmail com>
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
(gst_rtsp_connection_write):
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|
Make the next call to poll not depend on previous calls to poll with or
|
|
without reading from the active descriptor. Fixes #544293.
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|
|
|
2008-11-03 08:55:49 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
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|
gst/speexresample/gstspeexresample.c: Add TODO at the top of the file for enabling SSE/ARM specific optimizations and...
|
|
Original commit message from CVS:
|
|
* gst/speexresample/gstspeexresample.c:
|
|
(gst_speex_resample_convert_buffer):
|
|
Add TODO at the top of the file for enabling SSE/ARM specific
|
|
optimizations and choosing the fastest implementation at runtime.
|
|
Add g_assert_not_reached() at two places that should really never
|
|
be reached.
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|
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|
2008-11-02 09:19:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/speexresample/gstspeexresample.c: Fix format string and arguments.
|
|
Original commit message from CVS:
|
|
* gst/speexresample/gstspeexresample.c:
|
|
(gst_speex_resample_check_discont):
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|
Fix format string and arguments.
|
|
* gst/speexresample/resample_sse.h:
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|
Add missing file.
|
|
|
|
2008-11-01 19:38:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/speexresample/: Add missing headers to Makefile.am.
|
|
Original commit message from CVS:
|
|
* gst/speexresample/Makefile.am:
|
|
* gst/speexresample/gstspeexresample.c:
|
|
(gst_speex_resample_base_init), (gst_speex_resample_get_funcs),
|
|
(gst_speex_resample_convert_buffer), (_benchmark_int_float),
|
|
(_benchmark_int_int), (_benchmark_integer_resampling),
|
|
(plugin_init):
|
|
* gst/speexresample/gstspeexresample.h:
|
|
* gst/speexresample/resample.c:
|
|
* gst/speexresample/speex_resampler_double.c:
|
|
* gst/speexresample/speex_resampler_float.c:
|
|
* gst/speexresample/speex_resampler_int.c:
|
|
* gst/speexresample/speex_resampler_wrapper.h:
|
|
Add missing headers to Makefile.am.
|
|
Update copyright, years and my mail address.
|
|
Benchmark the integer resampling implementation against the
|
|
float implementation and use the faster one for 8/16 bit integer
|
|
input. On most recent systems the floating point version is faster.
|
|
|
|
2008-10-31 09:49:57 +0000 Nick Haddad <nick@haddads.net>
|
|
|
|
gst-libs/gst/riff/: Add support for other fourcc codes that are commonly used for 'uncompressed RGB', including 'RGB ...
|
|
Original commit message from CVS:
|
|
Patch by: Nick Haddad <nick at haddads dot net>
|
|
* gst-libs/gst/riff/riff-ids.h:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
|
|
Add support for other fourcc codes that are commonly used for
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|
'uncompressed RGB', including 'RGB ', 'RAW ', and 0.
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|
Fixes #558553.
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|
|
|
2008-10-30 14:55:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/speexresample/gstspeexresample.c: The length for the buffer conversion function is the number of audio frames, i....
|
|
Original commit message from CVS:
|
|
* gst/speexresample/gstspeexresample.c:
|
|
(gst_speex_resample_convert_buffer):
|
|
The length for the buffer conversion function is the number of
|
|
audio frames, i.e. we need to multiply it by the number of channels
|
|
to get the number of values. Also spotted by the unit test after
|
|
running in valgrind.
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|
|
|
2008-10-30 14:46:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
tests/check/elements/speexresample.c: Add pipeline unit tests for testing all supported formats with up/downsampling ...
|
|
Original commit message from CVS:
|
|
* tests/check/elements/speexresample.c: (element_message_cb),
|
|
(eos_message_cb), (test_pipeline), (GST_START_TEST),
|
|
(speexresample_suite):
|
|
Add pipeline unit tests for testing all supported formats with
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|
up/downsampling and different in/outrates.
|
|
* gst/speexresample/gstspeexresample.c:
|
|
(gst_speex_resample_push_drain), (gst_speex_resample_process):
|
|
* gst/speexresample/speex_resampler_wrapper.h:
|
|
Fix bugs identified by the testsuite.
|
|
|
|
2008-10-30 13:44:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/speexresample/: Add support for int8, int24 and int32 input by converting internally to/from int16 or double.
|
|
Original commit message from CVS:
|
|
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
|
|
(gst_speex_resample_get_funcs),
|
|
(gst_speex_resample_transform_size),
|
|
(gst_speex_resample_convert_buffer),
|
|
(gst_speex_resample_push_drain), (gst_speex_resample_process):
|
|
* gst/speexresample/gstspeexresample.h:
|
|
* gst/speexresample/speex_resampler_wrapper.h:
|
|
Add support for int8, int24 and int32 input by converting internally
|
|
to/from int16 or double.
|
|
|
|
2008-10-30 12:43:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
Add support for double samples as input and refactor the usage of the different compilation flavors of the speex resa...
|
|
Original commit message from CVS:
|
|
* gst/speexresample/Makefile.am:
|
|
* gst/speexresample/arch.h:
|
|
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_stop),
|
|
(gst_speex_resample_get_unit_size), (gst_speex_resample_get_funcs),
|
|
(gst_speex_resample_init_state), (gst_speex_resample_update_state),
|
|
(gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
|
|
(_gcd), (gst_speex_resample_transform_size),
|
|
(gst_speex_resample_set_caps), (gst_speex_resample_push_drain),
|
|
(gst_speex_resample_process), (gst_speex_resample_transform),
|
|
(gst_speex_resample_query), (gst_speex_resample_set_property):
|
|
* gst/speexresample/gstspeexresample.h:
|
|
* gst/speexresample/resample.c:
|
|
* gst/speexresample/speex_resampler.h:
|
|
* gst/speexresample/speex_resampler_double.c:
|
|
* gst/speexresample/speex_resampler_wrapper.h:
|
|
* tests/check/elements/speexresample.c: (setup_speexresample),
|
|
(test_perfect_stream_instance), (GST_START_TEST),
|
|
(test_discont_stream_instance):
|
|
Add support for double samples as input and refactor the usage
|
|
of the different compilation flavors of the speex resampler.
|
|
|
|
2008-10-30 11:43:12 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/audioresample/gstaudioresample.c: Return the result of parent_class->event().
|
|
Original commit message from CVS:
|
|
* gst/audioresample/gstaudioresample.c:
|
|
Return the result of parent_class->event().
|
|
|
|
2008-10-29 17:02:55 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/app/gstappsink.c: Fix the docs.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
|
|
Fix the docs.
|
|
|
|
2008-10-29 12:11:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/speexresample/gstspeexresample.*: Rewrite timestamp tracking to make it more robust and guarantee a continous str...
|
|
Original commit message from CVS:
|
|
* gst/speexresample/gstspeexresample.c: (gst_speex_resample_start),
|
|
(gst_speex_resample_get_unit_size),
|
|
(gst_speex_resample_push_drain), (gst_speex_resample_event),
|
|
(gst_speex_resample_check_discont), (gst_speex_resample_process),
|
|
(gst_speex_resample_transform):
|
|
* gst/speexresample/gstspeexresample.h:
|
|
Rewrite timestamp tracking to make it more robust and guarantee
|
|
a continous stream.
|
|
* tests/check/Makefile.am:
|
|
* tests/check/elements/speexresample.c: (setup_speexresample),
|
|
(cleanup_speexresample), (fail_unless_perfect_stream),
|
|
(test_perfect_stream_instance), (GST_START_TEST),
|
|
(test_discont_stream_instance), (live_switch_alloc_only_48000),
|
|
(live_switch_get_sink_caps), (live_switch_push),
|
|
(speexresample_suite):
|
|
Add unit tests for speexresample based on the audioresample unit tests.
|
|
|
|
2008-10-28 19:30:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/speexresample/gstspeexresample.*: Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT instead of ...
|
|
Original commit message from CVS:
|
|
* gst/speexresample/gstspeexresample.c:
|
|
(gst_speex_resample_get_unit_size),
|
|
(gst_speex_resample_fixate_caps), (gst_speex_resample_init_state),
|
|
(gst_speex_resample_update_state), (gst_speex_resample_parse_caps),
|
|
(gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
|
|
(gst_speex_resample_push_drain), (gst_speex_resample_event),
|
|
(gst_speex_resample_check_discont), (gst_speex_fix_output_buffer),
|
|
(gst_speex_resample_process), (gst_speex_resample_transform),
|
|
(gst_speex_resample_query), (gst_speex_resample_set_property):
|
|
* gst/speexresample/gstspeexresample.h:
|
|
Some random cleanup, add G_LIKELY and friends, use GST_DEBUG_OBJECT
|
|
instead of GST_DEBUG, ...
|
|
|
|
2008-10-28 16:28:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/speexresample/gstspeexresample.c: Fixate to the nearest supported rate instead of the first one.
|
|
Original commit message from CVS:
|
|
* gst/speexresample/gstspeexresample.c:
|
|
(gst_speex_resample_class_init), (gst_speex_resample_fixate_caps),
|
|
(gst_speex_resample_process):
|
|
Fixate to the nearest supported rate instead of the first one.
|
|
|
|
2008-10-28 16:25:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/audioresample/gstaudioresample.c: Fixate the rate to the nearest supported rate instead of the first one. Fixes b...
|
|
Original commit message from CVS:
|
|
* gst/audioresample/gstaudioresample.c:
|
|
(gst_audioresample_class_init), (audioresample_fixate_caps):
|
|
Fixate the rate to the nearest supported rate instead of
|
|
the first one. Fixes bug #549510.
|
|
|
|
2008-10-28 11:46:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/speexresample/: Update Speex resampler with latest version from Speex GIT.
|
|
Original commit message from CVS:
|
|
* gst/speexresample/README:
|
|
* gst/speexresample/arch.h:
|
|
* gst/speexresample/fixed_arm4.h:
|
|
* gst/speexresample/fixed_arm5e.h:
|
|
* gst/speexresample/fixed_bfin.h:
|
|
* gst/speexresample/fixed_debug.h:
|
|
* gst/speexresample/fixed_generic.h:
|
|
* gst/speexresample/resample.c: (compute_func), (main), (sinc),
|
|
(cubic_coef), (resampler_basic_direct_single),
|
|
(resampler_basic_direct_double),
|
|
(resampler_basic_interpolate_single),
|
|
(resampler_basic_interpolate_double), (update_filter),
|
|
(speex_resampler_init_frac), (speex_resampler_process_native),
|
|
(speex_resampler_magic), (speex_resampler_process_float),
|
|
(speex_resampler_process_int),
|
|
(speex_resampler_process_interleaved_float),
|
|
(speex_resampler_process_interleaved_int),
|
|
(speex_resampler_set_rate_frac), (speex_resampler_skip_zeros),
|
|
(speex_resampler_reset_mem):
|
|
* gst/speexresample/speex_resampler.h:
|
|
Update Speex resampler with latest version from Speex GIT.
|
|
|
|
2008-10-27 14:57:34 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
win32/common/libgstaudio.def: Add new symbols.
|
|
Original commit message from CVS:
|
|
* win32/common/libgstaudio.def:
|
|
Add new symbols.
|
|
|
|
2008-10-23 09:57:06 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/vorbis/vorbisdec.c: Attempt to make obfuscated code clearer.
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
|
|
Attempt to make obfuscated code clearer.
|
|
|
|
2008-10-23 07:11:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
Move float endianness conversion macros to core. Second part of bug ##555196.
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/floatcast/floatcast.h:
|
|
Move float endianness conversion macros to core. Second part of
|
|
bug ##555196.
|
|
|
|
2008-10-22 12:29:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
sys/: Don't mark as gtk-doc docs as they aren't public.
|
|
Original commit message from CVS:
|
|
* sys/ximage/ximagesink.h:
|
|
* sys/xvimage/xvimagesink.h:
|
|
Don't mark as gtk-doc docs as they aren't public.
|
|
|
|
2008-10-22 12:25:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
Allow setting colorkey if possible. Implement property probe interface for optional X features (autopaint-colorkey, d...
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c:
|
|
* sys/xvimage/xvimagesink.h:
|
|
* tests/icles/Makefile.am:
|
|
* tests/icles/test-colorkey.c:
|
|
Allow setting colorkey if possible. Implement property probe interface
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|
for optional X features (autopaint-colorkey, double-buffer and
|
|
colorkey). Fixes #554533
|
|
|
|
2008-10-22 12:01:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
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gst-libs/gst/tag/tags.c: Remove useless buffer size assignment. It already has this value.
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Original commit message from CVS:
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* gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
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Remove useless buffer size assignment. It already has this value.
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2008-10-20 15:35:37 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstaudiosink.c: Implement a separate activate functions to start monitoring the segments or, in pu...
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Original commit message from CVS:
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* gst-libs/gst/audio/gstaudiosink.c:
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(gst_audioringbuffer_class_init), (gst_audioringbuffer_acquire),
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(gst_audioringbuffer_activate), (gst_audioringbuffer_release),
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(gst_audioringbuffer_stop):
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Implement a separate activate functions to start monitoring the segments
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or, in pull mode, pulling in data.
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
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(gst_base_audio_sink_query_pad), (gst_base_audio_sink_query),
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(gst_base_audio_sink_setcaps), (gst_base_audio_sink_callback),
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(gst_base_audio_sink_activate_pull),
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(gst_base_audio_sink_async_play),
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(gst_base_audio_sink_change_state):
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Implement pad and element convert query function.
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Activate the ringbuffer.
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Use the segment last_stop value as the offset to pull.
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Use new basesink _do_preroll() method to preroll in the pulling thread.
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Take appropriate locking in the pulling thread.
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* gst-libs/gst/audio/gstringbuffer.h:
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Update some docs.
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2008-10-20 14:08:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/typefind/gsttypefindfunctions.c: Improve MXF typefinding a bit by searching for a header partition pack instead o...
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Original commit message from CVS:
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* gst/typefind/gsttypefindfunctions.c: (mxf_type_find):
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Improve MXF typefinding a bit by searching for a header partition
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pack instead of just a general partition pack and checking more
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bytes for valid values.
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2008-10-20 13:45:55 +0000 Wim Taymans <wim.taymans@gmail.com>
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tests/icles/.cvsignore: update ignore file.
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Original commit message from CVS:
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* tests/icles/.cvsignore:
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update ignore file.
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* tests/icles/Makefile.am:
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* tests/icles/test-box.c: (make_pipeline), (main):
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Add another interactive command line experimentation suite for
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dynamically boxing/cropping/saling an input video.
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2008-10-17 13:19:05 +0000 Wim Taymans <wim.taymans@gmail.com>
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Add methods to more accuratly control the pulling thread of a ringbuffer.
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Original commit message from CVS:
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* docs/libs/gst-plugins-base-libs-sections.txt:
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* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_convert),
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(gst_ring_buffer_activate), (gst_ring_buffer_is_active):
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* gst-libs/gst/audio/gstringbuffer.h:
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Add methods to more accuratly control the pulling thread of a
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ringbuffer.
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Add format conversion helper code to the ringbuffer.
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API: GstRingBuffer:gst_ring_buffer_activate()
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API: GstRingBuffer:gst_ring_buffer_is_active()
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API: GstRingBuffer:gst_ring_buffer_convert()
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2008-10-16 15:44:37 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstaudiosink.c: Signal thread startup earlier so that we can immediatly go into pull mode when we ...
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Original commit message from CVS:
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* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
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(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
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(gst_audioringbuffer_stop):
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Signal thread startup earlier so that we can immediatly go into pull
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mode when we have to and block on preroll.
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2008-10-16 15:38:50 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstringbuffer.c: In pull mode we want the callback to prepull a buffer we can preroll on even when...
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Original commit message from CVS:
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* gst-libs/gst/audio/gstringbuffer.c:
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(gst_ring_buffer_prepare_read):
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In pull mode we want the callback to prepull a buffer we can preroll on
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even when we are not yet playing.
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2008-10-16 15:07:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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Don't install static libs for plugins. Fixes #550851 for base.
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Original commit message from CVS:
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* ext/alsa/Makefile.am:
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* ext/cdparanoia/Makefile.am:
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* ext/gio/Makefile.am:
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* ext/gnomevfs/Makefile.am:
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* ext/libvisual/Makefile.am:
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* ext/ogg/Makefile.am:
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* ext/pango/Makefile.am:
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* ext/theora/Makefile.am:
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* ext/vorbis/Makefile.am:
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* gst/adder/Makefile.am:
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* gst/audioconvert/Makefile.am:
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* gst/audiorate/Makefile.am:
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* gst/audioresample/Makefile.am:
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* gst/audiotestsrc/Makefile.am:
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* gst/ffmpegcolorspace/Makefile.am:
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* gst/gdp/Makefile.am:
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* gst/playback/Makefile.am:
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* gst/subparse/Makefile.am:
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* gst/tcp/Makefile.am:
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* gst/typefind/Makefile.am:
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* gst/videorate/Makefile.am:
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* gst/videoscale/Makefile.am:
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* gst/videotestsrc/Makefile.am:
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* gst/volume/Makefile.am:
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* sys/v4l/Makefile.am:
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* sys/ximage/Makefile.am:
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* sys/xvimage/Makefile.am:
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Don't install static libs for plugins. Fixes #550851 for base.
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2008-10-16 13:50:00 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/audiotestsrc/gstaudiotestsrc.c: Set the default blocksize to -1 because we will then use the configured samplespe...
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Original commit message from CVS:
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* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init):
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Set the default blocksize to -1 because we will then use the configured
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samplesperbuffer to create our output buffer.
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2008-10-15 15:28:41 +0000 Edward Hervey <bilboed@bilboed.com>
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gst-libs/gst/riff/riff-media.c: Add mappping for the KMVC (Karl Morton's Video) Codec.
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Original commit message from CVS:
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* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
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(gst_riff_create_video_template_caps):
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Add mappping for the KMVC (Karl Morton's Video) Codec.
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2008-10-15 14:25:50 +0000 Edward Hervey <bilboed@bilboed.com>
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gst/typefind/gsttypefindfunctions.c: Don't forget to advance the offset of what we're matching against, else we end u...
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Original commit message from CVS:
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* gst/typefind/gsttypefindfunctions.c: (flac_type_find):
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Don't forget to advance the offset of what we're matching against, else
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we end up in a forever loop.
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2008-10-15 11:25:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/subparse/gstsubparse.c: Improve typefinding a bit. If we don't have a Unicode charset try GST_SUBTITLE_ENCODING a...
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Original commit message from CVS:
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* gst/subparse/gstsubparse.c: (gst_subparse_type_find):
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Improve typefinding a bit. If we don't have a Unicode charset
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try GST_SUBTITLE_ENCODING and otherwise try ISO-8859-15.
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2008-10-14 11:13:59 +0000 Edward Hervey <bilboed@bilboed.com>
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ext/theora/theoradec.c: Fix build on macosx.
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Original commit message from CVS:
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* ext/theora/theoradec.c: (theora_dec_decode_buffer):
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Fix build on macosx.
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2008-10-13 11:36:13 +0000 Robin Stocker <robin@nibor.org>
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ext/theora/: Parse input caps and make the PAR override the encoded PAR when specified by a container. Fixes #555699.
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Original commit message from CVS:
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Based on patch by: Robin Stocker <robin at nibor dot org>
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* ext/theora/gsttheoradec.h:
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* ext/theora/theoradec.c: (gst_theora_dec_init),
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(theora_dec_setcaps), (theora_handle_type_packet),
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(theora_dec_decode_buffer), (theora_dec_change_state):
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Parse input caps and make the PAR override the encoded PAR when
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specified by a container. Fixes #555699.
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2008-10-13 09:16:59 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/rtp/gstbasertpdepayload.*: Add some more G_LIKELY
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Original commit message from CVS:
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* gst-libs/gst/rtp/gstbasertpdepayload.c:
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(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
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(gst_base_rtp_depayload_set_gst_timestamp),
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(gst_base_rtp_depayload_change_state):
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* gst-libs/gst/rtp/gstbasertpdepayload.h:
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Add some more G_LIKELY
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Fail when the setcaps function was not called.
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* gst-libs/gst/rtp/gstbasertppayload.c:
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(gst_basertppayload_set_outcaps):
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Propagate return value of setcaps.
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2008-10-13 08:58:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/subparse/: Add support for UTF16/UTF32 subtitles as long as the first bytes of the first buffer contain the BOM. ...
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Original commit message from CVS:
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* gst/subparse/Makefile.am:
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* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
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(gst_sub_parse_class_init), (gst_sub_parse_init),
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(gst_convert_to_utf8), (detect_encoding), (convert_encoding),
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(get_next_line), (gst_sub_parse_data_format_autodetect),
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(feed_textbuf), (handle_buffer), (gst_sub_parse_change_state),
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(gst_subparse_type_find):
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* gst/subparse/gstsubparse.h:
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Add support for UTF16/UTF32 subtitles as long as the first bytes of
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the first buffer contain the BOM. This also adds support for other
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encodings that allow NUL bytes via the encoding property.
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Fixes bugs #552237 and #456788.
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2008-10-13 08:15:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst-libs/gst/tag/tags.c: Don't drop the last byte of image tags if they're not an URI list.
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Original commit message from CVS:
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* gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
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Don't drop the last byte of image tags if they're not an URI list.
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Fixes bug #556066.
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2008-10-13 08:00:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/typefind/gsttypefindfunctions.c: For looking at the 4th byte we have to get 4 bytes of course and not 3.
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Original commit message from CVS:
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* gst/typefind/gsttypefindfunctions.c: (flac_type_find):
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For looking at the 4th byte we have to get 4 bytes of course
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and not 3.
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2008-10-13 07:52:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/typefind/gsttypefindfunctions.c: Improve FLAC-without-headers typefinding by looking at most of the frame header ...
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Original commit message from CVS:
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* gst/typefind/gsttypefindfunctions.c: (flac_type_find):
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Improve FLAC-without-headers typefinding by looking at most of the
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frame header and checking if invalid values are used. Should prevent
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quite some false positives compared to the old version which only
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check if the first 14 bits are set.
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2008-10-11 16:27:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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sys/xvimage/xvimagesink.c: Don't assert on caps==NULL.
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Original commit message from CVS:
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* sys/xvimage/xvimagesink.c:
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Don't assert on caps==NULL.
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2008-10-10 17:13:40 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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Add support for subtitle files with UTF-8 BOM at the beginning by simple stripping it from the first line before pass...
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Original commit message from CVS:
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* gst/subparse/gstsubparse.c:
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(gst_sub_parse_data_format_autodetect), (handle_buffer),
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(gst_sub_parse_change_state):
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* gst/subparse/gstsubparse.h:
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* tests/check/elements/subparse.c: (GST_START_TEST):
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Add support for subtitle files with UTF-8 BOM at the beginning
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by simple stripping it from the first line before passing it
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to any parsing code. Fixes bug #555257 and playback of files
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created by Gnome Subtitles.
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2008-10-10 15:45:15 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/audiotestsrc/gstaudiotestsrc.*: Define the default property values in the usual place.
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Original commit message from CVS:
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* gst/audiotestsrc/gstaudiotestsrc.c:
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(gst_audio_test_src_class_init), (gst_audio_test_src_init),
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(gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
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(gst_audio_test_src_start), (gst_audio_test_src_stop),
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(gst_audio_test_src_do_seek), (gst_audio_test_src_check_get_range),
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(gst_audio_test_src_create):
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* gst/audiotestsrc/gstaudiotestsrc.h:
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Define the default property values in the usual place.
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Implement start/stop to reset values correctly.
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Calculate the sample size only once when we negotiate.
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Rename some values to make more sense.
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Keep track of our byte range.
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Add support for pull based scheduling. Disabled for now until we have
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the whole stack working.
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Set the BUFFER_OFFSET correctly.
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2008-10-10 15:32:10 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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Make the detection of the used subtitle a bit less strict for srt subtitles. Fixes bug #555607.
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Original commit message from CVS:
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Based on a patch by: xavierb at gmail dot com
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* gst/subparse/gstsubparse.c:
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(gst_sub_parse_data_format_autodetect):
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* tests/check/elements/subparse.c: (GST_START_TEST):
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Make the detection of the used subtitle a bit less strict
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for srt subtitles. Fixes bug #555607.
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2008-10-10 15:21:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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ext/vorbis/vorbisenc.c: Fix discontinuity detection which was broken by last commit.
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Original commit message from CVS:
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* ext/vorbis/vorbisenc.c:
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(gst_vorbis_enc_buffer_check_discontinuous):
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Fix discontinuity detection which was broken by last commit.
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2008-10-09 11:18:09 +0000 Tim-Philipp Müller <tim@centricular.net>
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configure.ac: Require core CVS for ghostpad API additions used by decodebin2.
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Original commit message from CVS:
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* configure.ac::
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Require core CVS for ghostpad API additions used by decodebin2.
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2008-10-08 15:30:33 +0000 Edward Hervey <bilboed@bilboed.com>
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gst-libs/gst/audio/gstbaseaudiosrc.c: Fix debug statements (space between '%' and actual format).
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Original commit message from CVS:
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* gst-libs/gst/audio/gstbaseaudiosrc.c:
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(gst_base_audio_src_create):
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Fix debug statements (space between '%' and actual format).
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2008-10-08 14:44:04 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstdecodebin2.c: Remove bogus assert, the decodepad could have been created inside an already existing g...
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Original commit message from CVS:
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* gst/playback/gstdecodebin2.c: (gst_decode_pad_activate):
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Remove bogus assert, the decodepad could have been created inside an
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already existing group.
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2008-10-08 14:01:42 +0000 Andy Wingo <wingo@pobox.com>
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* ChangeLog:
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changelog
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Original commit message from CVS:
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changelog
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2008-10-08 14:00:07 +0000 Andy Wingo <wingo@pobox.com>
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gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset target instead of setting it.
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Original commit message from CVS:
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2008-10-08 Andy Wingo <wingo@pobox.com>
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* gst/playback/gstdecodebin2.c (expose_pad): Fix typo: unset
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target instead of setting it.
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(gst_decode_pad_activate, gst_decode_pad_unblock): This is now the
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API for a decode pad. The bugfix is that we set the group in
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activate(), not when the pad was created because it might be NULL
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then.
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(gst_decode_group_control_source_pad, gst_decode_group_expose):
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Update to use the API.
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2008-10-08 12:49:40 +0000 Andy Wingo <wingo@pobox.com>
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gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to be a subclass of GstGhostPad.
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Original commit message from CVS:
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2008-10-08 Andy Wingo <wingo@pobox.com>
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* gst/playback/gstdecodebin2.c (struct _GstDecodePad): Change to
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be a subclass of GstGhostPad.
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(analyze_new_pad): So, when emitting the signals that determine
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how we do autoplugging, already create the ghost pad and use it as
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the pad in the signal arguments. This allows applications to make
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a connection between the pad passed in e.g. autoplug-continue, and
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the pad passed in new-decoded-pad.
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(connect_pad, expose_pad): Update to receive the ghosted decode
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pad in the args, retargetting it as necessary if we have to plug
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the target pad through a multiqueue.
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(gst_decode_group_control_source_pad): Adapt to receive an
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already-ghosted pad that just needs activation, blocking, and
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drain notification.
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(sort_end_pads): Adapt for decode pads actually being pads.
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(gst_decode_group_expose): Adapt for decode pads actually being
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pads. Rewrite the decode pad names so they appear in order. Adds a
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new error case if we couldn't set the name.
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(gst_decode_group_free, gst_decode_group_hide): Adapt cleanup
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logic.
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(gst_decode_pad_set_blocked, gst_decode_pad_add_drained_check):
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New API for the decode pad, needed because we shouldn't do these
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things inside gst_decode_pad_new(), but after.
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(gst_decode_pad_new): Change to actually make the real pad, and
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delay the blocking/drainage bits.
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2008-10-08 12:12:01 +0000 Daniel Drake <dsd@laptop.org>
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ext/ogg/gstoggmux.c: Unref all buffers when clearing collectpads. Fixes bug #546955.
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Original commit message from CVS:
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Patch by: Daniel Drake <dsd at laptop dot org>
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* ext/ogg/gstoggmux.c: (gst_ogg_mux_clear_collectpads):
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Unref all buffers when clearing collectpads. Fixes bug #546955.
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2008-10-08 12:08:01 +0000 Klaas <klaas@rivercrew.net>
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ext/vorbis/vorbisenc.*: Keep track of the upstream segments and use the running time on that segment instead of the b...
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Original commit message from CVS:
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Based on a patch by: Klaas <klaas at rivercrew dot net>
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* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_sink_event),
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(gst_vorbis_enc_buffer_check_discontinuous),
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(gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
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* ext/vorbis/vorbisenc.h:
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Keep track of the upstream segments and use the running time on that
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segment instead of the buffer timestamp everywhere. Fixes bug #525807.
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|
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2008-10-08 11:50:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/audioconvert/audioconvert.c: Prevent overflows with big buffer when calculating the size of the intermediate buff...
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|
Original commit message from CVS:
|
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* gst/audioconvert/audioconvert.c: (audio_convert_convert):
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Prevent overflows with big buffer when calculating the size of
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the intermediate buffer by using gst_util_uint64_scale() instead of
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plain arithmetics. Fixes bug #552801.
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2008-10-08 10:49:15 +0000 Pavel Zeldin <pzeldin@gmail.com>
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ext/pango/gstclockoverlay.*: API: Add ability to specify format for date/time display by adding a "time-format" prope...
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Original commit message from CVS:
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|
Patch by: Pavel Zeldin <pzeldin at gmail dot com>
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* ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time),
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(gst_clock_overlay_class_init), (gst_clock_overlay_finalize),
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(gst_clock_overlay_init), (gst_clock_overlay_set_property),
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(gst_clock_overlay_get_property):
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* ext/pango/gstclockoverlay.h:
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API: Add ability to specify format for date/time display by
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adding a "time-format" property.
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Fixes bug #554879.
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2008-10-08 09:22:26 +0000 Jan Gerber <j@oil21.org>
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gst-libs/gst/riff/riff-media.c: Add FFV1 fourcc to support playback of FFMPEG lossless video in AVI. Fixes bug #555319.
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Original commit message from CVS:
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Patch by: Jan Gerber <j at oil21 dot org>
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* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
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(gst_riff_create_video_template_caps):
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Add FFV1 fourcc to support playback of FFMPEG lossless video
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in AVI. Fixes bug #555319.
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2008-10-08 09:12:36 +0000 Håvard Graff <havard.graff@tandberg.com>
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gst-libs/gst/audio/gstbaseaudiosrc.c: Implement skew clock slaving. Fixes #552559.
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Original commit message from CVS:
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Patch by: Håvard Graff <havard dot graff at tandberg dot com>
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* gst-libs/gst/audio/gstbaseaudiosrc.c:
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(gst_base_audio_src_create):
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Implement skew clock slaving. Fixes #552559.
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2008-10-08 09:10:23 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/: Fix include of config.h
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Original commit message from CVS:
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* gst-libs/gst/audio/multichannel.c:
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* gst-libs/gst/audio/testchannels.c:
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Fix include of config.h
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2008-10-06 16:36:20 +0000 Tero Saarni <tero.saarni@gmail.com>
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gst-libs/gst/sdp/gstsdpmessage.c: Fix parsing of the c= field containing multicast addresses.
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Original commit message from CVS:
|
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Based on Patch by: Tero Saarni <tero dot saarni at gmail dot com>
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* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_parse_line),
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(print_media), (gst_sdp_message_dump):
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Fix parsing of the c= field containing multicast addresses.
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Fixes #552199.
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Add the connection info to the session or streams.
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Fix parsing of the bandwidth.
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Add debugging for the connections and bandwidths for a media.
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Add debugging for the bandwidth of the session.
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2008-10-06 16:31:27 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/rtp/gstbasertppayload.c: Configure the next seqnum and timestamp in the state change so that they can be...
|
|
Original commit message from CVS:
|
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* gst-libs/gst/rtp/gstbasertppayload.c:
|
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(gst_basertppayload_change_state):
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Configure the next seqnum and timestamp in the state change so that they
|
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can be queried soon after.
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2008-10-06 16:29:33 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/rtp/gstbasertpdepayload.c: Improve debugging of the rtptime.
|
|
Original commit message from CVS:
|
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* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
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(gst_base_rtp_depayload_chain):
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Improve debugging of the rtptime.
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2008-10-05 11:33:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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configure.ac: Back to development -> 0.10.21.1
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Original commit message from CVS:
|
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* configure.ac:
|
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Back to development -> 0.10.21.1
|
|
|
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2008-10-05 08:18:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
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* ChangeLog:
|
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ChangeLog surgery
|
|
Original commit message from CVS:
|
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ChangeLog surgery
|
|
|
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2008-10-05 08:11:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
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gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
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Original commit message from CVS:
|
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* gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
|
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(plugin_init):
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Add typefinder for MXF.
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2008-10-05 08:10:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/typefind/gsttypefindfunctions.c: Add typefinder for MXF.
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|
Original commit message from CVS:
|
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* gst/typefind/gsttypefindfunctions.c: (mxf_type_find),
|
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(plugin_init):
|
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Add typefinder for MXF.
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2008-10-03 15:19:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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tests/icles/Makefile.am: Only build test-colorkey if GTK+ is available.
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Original commit message from CVS:
|
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* tests/icles/Makefile.am:
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Only build test-colorkey if GTK+ is available.
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|
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=== release 0.10.21 ===
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2008-10-03 00:03:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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* ChangeLog:
|
|
* NEWS:
|
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* RELEASE:
|
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* configure.ac:
|
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* docs/plugins/gst-plugins-base-plugins.args:
|
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* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
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* docs/plugins/gst-plugins-base-plugins.interfaces:
|
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* docs/plugins/gst-plugins-base-plugins.prerequisites:
|
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* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.xml:
|
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* docs/plugins/inspect/plugin-audioconvert.xml:
|
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* docs/plugins/inspect/plugin-audiorate.xml:
|
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* docs/plugins/inspect/plugin-audioresample.xml:
|
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* docs/plugins/inspect/plugin-audiotestsrc.xml:
|
|
* docs/plugins/inspect/plugin-cdparanoia.xml:
|
|
* docs/plugins/inspect/plugin-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gdp.xml:
|
|
* docs/plugins/inspect/plugin-gio.xml:
|
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* docs/plugins/inspect/plugin-gnomevfs.xml:
|
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* docs/plugins/inspect/plugin-libvisual.xml:
|
|
* docs/plugins/inspect/plugin-ogg.xml:
|
|
* docs/plugins/inspect/plugin-pango.xml:
|
|
* docs/plugins/inspect/plugin-playback.xml:
|
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* docs/plugins/inspect/plugin-queue2.xml:
|
|
* docs/plugins/inspect/plugin-subparse.xml:
|
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* docs/plugins/inspect/plugin-tcp.xml:
|
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* docs/plugins/inspect/plugin-theora.xml:
|
|
* docs/plugins/inspect/plugin-typefindfunctions.xml:
|
|
* docs/plugins/inspect/plugin-uridecodebin.xml:
|
|
* docs/plugins/inspect/plugin-video4linux.xml:
|
|
* docs/plugins/inspect/plugin-videorate.xml:
|
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* docs/plugins/inspect/plugin-videoscale.xml:
|
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* docs/plugins/inspect/plugin-videotestsrc.xml:
|
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* docs/plugins/inspect/plugin-volume.xml:
|
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* docs/plugins/inspect/plugin-vorbis.xml:
|
|
* docs/plugins/inspect/plugin-ximagesink.xml:
|
|
* docs/plugins/inspect/plugin-xvimagesink.xml:
|
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* gst-plugins-base.doap:
|
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* win32/common/config.h:
|
|
Release 0.10.21
|
|
Original commit message from CVS:
|
|
Release 0.10.21
|
|
|
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2008-10-02 23:44:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
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* po/af.po:
|
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* po/az.po:
|
|
* po/bg.po:
|
|
* po/ca.po:
|
|
* po/cs.po:
|
|
* po/da.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/es.po:
|
|
* po/fi.po:
|
|
* po/fr.po:
|
|
* po/hu.po:
|
|
* po/id.po:
|
|
* po/it.po:
|
|
* po/lt.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/pl.po:
|
|
* po/pt_BR.po:
|
|
* po/ru.po:
|
|
* po/sk.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
* po/zh_CN.po:
|
|
Update .po files
|
|
Original commit message from CVS:
|
|
Update .po files
|
|
|
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2008-09-28 22:58:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
configure.ac: 0.10.20.4 pre-release
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
0.10.20.4 pre-release
|
|
|
|
2008-09-25 10:46:00 +0000 ogg.k.ogg.k <ogg.k.ogg.k@googlemail.com>
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|
|
|
ext/theora/theoraparse.c: Set the BOS flag on the BOS packet. Fixes #553244.
|
|
Original commit message from CVS:
|
|
Patch by: ogg.k.ogg.k <ogg dot k dot ogg dot k at googlemail dot com>
|
|
* ext/theora/theoraparse.c: (theora_parse_set_streamheader):
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Set the BOS flag on the BOS packet. Fixes #553244.
|
|
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2008-09-23 17:48:14 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtsp/gstrtspmessage.c: Fix the g_return_val_if_fail() statements.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtsp/gstrtspmessage.c:
|
|
(gst_rtsp_message_parse_request),
|
|
(gst_rtsp_message_parse_response):
|
|
Fix the g_return_val_if_fail() statements.
|
|
|
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2008-09-22 17:44:14 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst-libs/gst/tag/gsttagdemux.c: Fail to activate if there's insufficient data in the file to be usable, preventing an...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/gsttagdemux.c:
|
|
Fail to activate if there's insufficient data in the file to be usable,
|
|
preventing an assertion fail later. Fixes #552960
|
|
|
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2008-09-16 15:36:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
Commit stuff that should have gone in last week when I made the pre-releases:
|
|
Original commit message from CVS:
|
|
Commit stuff that should have gone in last week when I made the pre-releases:
|
|
2008-09-10 Jan Schmidt <jan.schmidt@sun.com>
|
|
* configure.ac:
|
|
0.10.20.2 pre-release
|
|
* po/LINGUAS:
|
|
* po/id.po:
|
|
* po/pt_BR.po:
|
|
New translations.
|
|
|
|
2008-09-15 15:11:18 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/: Recognise Kate subtitle streams (#550582).
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/pbutils/descriptions.c:
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
Recognise Kate subtitle streams (#550582).
|
|
|
|
2008-09-13 11:04:02 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/audio/audio.h: Remove trailing comma from enum list, which causes problems with -pendantic (#550729).
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/audio.h: (GST_AUDIO_FIELD_SIGNED):
|
|
Remove trailing comma from enum list, which causes problems
|
|
with -pendantic (#550729).
|
|
|
|
2008-09-05 19:04:47 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/interfaces/propertyprobe.c: More sanity checks for our second-favourite interface.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/propertyprobe.c:
|
|
(gst_property_probe_get_properties),
|
|
(gst_property_probe_get_property),
|
|
(gst_property_probe_probe_property),
|
|
(gst_property_probe_probe_property_name),
|
|
(gst_property_probe_needs_probe),
|
|
(gst_property_probe_needs_probe_name),
|
|
(gst_property_probe_get_values),
|
|
(gst_property_probe_get_values_name),
|
|
(gst_property_probe_probe_and_get_values),
|
|
(gst_property_probe_probe_and_get_values_name):
|
|
More sanity checks for our second-favourite interface.
|
|
|
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2008-09-05 14:12:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
|
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gst-libs/gst/interfaces/propertyprobe.c: Check for NULL pointer, in the hope that this fixes #532864.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/propertyprobe.c:
|
|
Check for NULL pointer, in the hope that this fixes #532864.
|
|
|
|
2008-09-05 10:24:05 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
sys/xvimage/xvimagesink.c: No really, the next release is 0.10.21 (fix Since: tags in docs).
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
|
|
No really, the next release is 0.10.21 (fix Since: tags in docs).
|
|
|
|
2008-09-04 16:25:06 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstaudiosrc.c: Disable a code path that is now called but causes a deadlock for some reason and is...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_stop):
|
|
Disable a code path that is now called but causes a deadlock for some
|
|
reason and is unneeded.
|
|
|
|
2008-09-04 13:46:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
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sys/xvimage/xvimagesink.*: Add a "draw-border" property that can be set to false to disable drawing borders.
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c:
|
|
* sys/xvimage/xvimagesink.h:
|
|
Add a "draw-border" property that can be set to false to disable
|
|
drawing borders.
|
|
* tests/icles/test-colorkey.c:
|
|
* tests/icles/Makefile.am:
|
|
Add new test application for the colorkey handling.
|
|
|
|
2008-09-03 14:00:06 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
gst-libs/gst/riff/riff-media.c: Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
|
|
Use a decent caps for TrueSpeech instead of a ffmpeg-specific one.
|
|
This will also be fixed for upcoming gst-ffmpeg release so that once
|
|
this release of -base is out, it will work with the latest gst-ffmpeg
|
|
release.
|
|
|
|
2008-09-03 13:27:20 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
gst-libs/gst/riff/riff-media.c: Add Truespeech mapping for RIFF formats (AVI/WAV).
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
|
|
(gst_riff_create_audio_template_caps):
|
|
Add Truespeech mapping for RIFF formats (AVI/WAV).
|
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Fixes #550656
|
|
|
|
2008-09-03 12:23:44 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
|
|
Typefind video/mj2 and image/jp2 ISO JPEG2000 mime types.
|
|
Fixes #550638.
|
|
|
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2008-09-03 10:12:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
Rework last change, so that we build subparse, but just disable the sami parse functionality, if we're configured to ...
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* gst/subparse/Makefile.am:
|
|
* gst/subparse/gstsubparse.c:
|
|
* gst/subparse/samiparse.c:
|
|
* tests/check/elements/subparse.c:
|
|
Rework last change, so that we build subparse, but just disable the
|
|
sami parse functionality, if we're configured to not use xml. In the
|
|
tests only the sami test is disabled now.
|
|
|
|
2008-09-02 15:07:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
configure.ac: Disable subparse when xml is disabled. It woundn't work anyway. Fixes test runs.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Disable subparse when xml is disabled. It woundn't work anyway. Fixes
|
|
test runs.
|
|
|
|
2008-09-02 09:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
po/POTFILES.in: Add some more files with strings for translation.
|
|
Original commit message from CVS:
|
|
* po/POTFILES.in:
|
|
Add some more files with strings for translation.
|
|
|
|
2008-09-02 06:37:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
Use new geo location tags from core. Fixes #481169
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/gstvorbistag.c:
|
|
* tests/check/libs/tag.c:
|
|
Use new geo location tags from core. Fixes #481169
|
|
|
|
2008-09-01 16:05:45 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
tests/check/elements/audioresample.c: Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
|
|
Original commit message from CVS:
|
|
* tests/check/elements/audioresample.c: (setup_audioresample),
|
|
(fail_unless_perfect_stream), (test_perfect_stream_instance),
|
|
(test_discont_stream_instance):
|
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Now that GstBaseTransform is 'fixed' ... remove cruft from tests.
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Add debugging for coherence.
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|
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2008-08-30 15:55:06 +0000 Jonathan Matthew <notverysmart@gmail.com>
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|
|
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gst/typefind/gsttypefindfunctions.c: Add typefinder for PDF documents (which is nice to have, since it's a common for...
|
|
Original commit message from CVS:
|
|
Patch by: Jonathan Matthew <notverysmart gmail com>
|
|
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
|
|
Add typefinder for PDF documents (which is nice to have, since it's a
|
|
common format, but also helps prevent false positives). Fixes #549814.
|
|
|
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2008-08-27 15:30:16 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst/playback/gstplaybin2.c: Fix nasty race where multiple decodebins could start pushing data before we manage to con...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin2.c: (selector_blocked), (pad_added_cb),
|
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(no_more_pads_cb):
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Fix nasty race where multiple decodebins could start pushing data before
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we manage to configure the sinks, resulting in not-linked errors in
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typical RTSP streaming cases.
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2008-08-26 17:24:31 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstaudiosink.c: Since we now call stop, we trigger this code path that causes a deadlock is appare...
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Original commit message from CVS:
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* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
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Since we now call stop, we trigger this code path that causes a deadlock
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is apparently not needed.
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2008-08-26 15:45:36 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstringbuffer.c: Also allow the case where the ringbuffer was paused when we try to stop it so tha...
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Original commit message from CVS:
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* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
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(gst_ring_buffer_stop):
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Also allow the case where the ringbuffer was paused when we try to stop
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it so that the basesrc stop function is still called.
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2008-08-23 15:25:44 +0000 Mike Ruprecht <cmaiku@gmail.com>
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sys/v4l/gstv4lelement.c: Reprobe devices again instead of taking a cached list as new devices could've been plugged i...
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Original commit message from CVS:
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Patch by: Mike Ruprecht <cmaiku at gmail dot com>
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* sys/v4l/gstv4lelement.c: (gst_v4l_class_probe_devices):
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Reprobe devices again instead of taking a cached list as new
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devices could've been plugged in. Fixes bug #549062.
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2008-08-23 15:19:59 +0000 Alessandro Dessina <alessandro@nnva.org>
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ext/ogg/gstoggdemux.c: Don't add pads and activate them for skeleton streams. These are already handled inside oggdem...
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Original commit message from CVS:
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Patch by: Alessandro Dessina <alessandro nnva org>
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* ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
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(gst_ogg_demux_activate_chain):
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Don't add pads and activate them for skeleton streams. These are already
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handled inside oggdemux. Fixes bug #537599.
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2008-08-22 15:54:15 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/vorbis/vorbisdec.c: Reset variable so that query and convert fail after going back to
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Original commit message from CVS:
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* ext/vorbis/vorbisdec.c: (vorbis_dec_change_state):
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Reset variable so that query and convert fail after going back to
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READY. Fixes #548898.
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2008-08-22 07:24:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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ext/vorbis/vorbisenc.c: If a buffer arrives with a timestamp before the timestamp+duration of the previous buffer cli...
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Original commit message from CVS:
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* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
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If a buffer arrives with a timestamp before the timestamp+duration
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of the previous buffer clip it instead of dropping it completely.
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Slight improvement for the unfixable bug #548913.
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2008-08-21 14:19:21 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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ext/vorbis/vorbisdec.c: Take the current timestamp instead of timestamp+duration for the offset.
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Original commit message from CVS:
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* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
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Take the current timestamp instead of timestamp+duration for the offset.
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This offset will later be used for calculating the timestamp and
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otherwise vorbisdec will interpolate timestamps wrong if upstream
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only sends timestamps and no granulepos.
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2008-08-21 11:20:36 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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tests/examples/seek/seek.c: Don't crash when having no visualisations.
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Original commit message from CVS:
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* tests/examples/seek/seek.c:
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Don't crash when having no visualisations.
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2008-08-16 20:57:27 +0000 David Schleef <ds@schleef.org>
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gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
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Original commit message from CVS:
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* gst/typefind/gsttypefindfunctions.c: DV typefinding. Remove
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check for a bit that is 0 in IEC 61384, but not SMPTE 314M.
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Fixes #548065.
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2008-08-15 07:24:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst-libs/gst/pbutils/missing-plugins.c: When cleaning up the caps fields also remove "depth" for the same reason we r...
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Original commit message from CVS:
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* gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
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When cleaning up the caps fields also remove "depth" for the same
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reason we remove "width".
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2008-08-14 17:14:53 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/pbutils/descriptions.c: Add Lead H.264 here as well.
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Original commit message from CVS:
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* gst-libs/gst/pbutils/descriptions.c: (format_info_get_desc):
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Add Lead H.264 here as well.
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2008-08-14 15:17:31 +0000 Julien Moutte <julien@moutte.net>
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gst-libs/gst/riff/riff-media.c: Add Lead H.264 variant.
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Original commit message from CVS:
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2008-08-14 Julien Moutte <julien@fluendo.com>
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* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
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(gst_riff_create_video_template_caps): Add Lead H.264 variant.
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2008-08-13 09:17:38 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstbaseaudiosrc.c: When not slaved to another clock also subtract the base_time from our internal ...
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|
Original commit message from CVS:
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* gst-libs/gst/audio/gstbaseaudiosrc.c:
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(gst_base_audio_src_create):
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When not slaved to another clock also subtract the base_time from our
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internal clock time to get the running time.
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2008-08-13 00:59:07 +0000 David Schleef <ds@schleef.org>
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ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate, since it has no basis in libtheora.
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Original commit message from CVS:
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* ext/theora/theoraenc.c: Remove the 2000 kbit limit to bitrate,
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since it has no basis in libtheora.
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2008-08-12 06:31:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst-libs/gst/interfaces/propertyprobe.h: Remove double "interface" from doc-string.
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Original commit message from CVS:
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* gst-libs/gst/interfaces/propertyprobe.h:
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Remove double "interface" from doc-string.
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* gst-libs/gst/interfaces/xoverlay.h:
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Document interface.
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* gst-libs/gst/riff/riff.c:
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Add basic doc blobs.
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2008-08-11 15:05:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst-libs/gst/audio/Makefile.am: Don't try to build that example anymore.
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Original commit message from CVS:
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* gst-libs/gst/audio/Makefile.am:
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Don't try to build that example anymore.
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2008-08-11 14:51:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst-libs/gst/audio/: Move audiofiltertemplate to gst-template.
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Original commit message from CVS:
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* gst-libs/gst/audio/.cvsignore:
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* gst-libs/gst/audio/Makefile.am:
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* gst-libs/gst/audio/gstaudiofiltertemplate.c:
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* gst-libs/gst/audio/make_filter:
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Move audiofiltertemplate to gst-template.
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2008-08-11 09:20:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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More docs and shuffling. What can we do with the hundreds of #defines.
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Original commit message from CVS:
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* docs/libs/gst-plugins-base-libs-sections.txt:
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* gst-libs/gst/audio/gstaudiosrc.h:
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More docs and shuffling. What can we do with the hundreds of #defines.
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2008-08-11 08:34:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst-libs/gst/: Reducing number of dundocumented symbols.
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Original commit message from CVS:
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* gst-libs/gst/audio/audio.h:
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* gst-libs/gst/audio/gstaudiofilter.h:
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* gst-libs/gst/audio/gstringbuffer.h:
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* gst-libs/gst/interfaces/propertyprobe.h:
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* gst-libs/gst/tag/gsttagdemux.h:
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Reducing number of dundocumented symbols.
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2008-08-11 07:16:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst-libs/gst/audio/audio.c: Fix doc comment syntax.
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Original commit message from CVS:
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* gst-libs/gst/audio/audio.c:
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Fix doc comment syntax.
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* gst-libs/gst/interfaces/propertyprobe.c:
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Add more doc-comments and a FIXME: for the signal.
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2008-08-07 16:11:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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ext/ogg/gstoggmux.*: Don't pretend to support NEWSEGMENT events, instead override the
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|
Original commit message from CVS:
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* ext/ogg/gstoggmux.c: (gst_ogg_mux_sink_event),
|
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(gst_ogg_mux_request_new_pad):
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* ext/ogg/gstoggmux.h:
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Don't pretend to support NEWSEGMENT events, instead override the
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GstCollectPads event function to return FALSE on NEWSEGMENT events
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and do the normal work for other events.
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This prevents elements like flacenc to seek to the start and rewrite
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some data which then results in a broken Ogg packet.
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2008-08-07 15:58:58 +0000 Frederic Crozat <fcrozat@mandriva.org>
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Make sure gettext returns translations in UTF-8 encoding rather than in the current locale encoding (#546822).
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Original commit message from CVS:
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Patch by: Frederic Crozat <fcrozat@mandriva.org>
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* ext/alsa/gstalsaplugin.c: (plugin_init):
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* ext/cdparanoia/gstcdparanoiasrc.c: (plugin_init):
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* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
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* ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
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* gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init):
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* gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
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* gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal):
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* gst/playback/gstdecodebin.c: (plugin_init):
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* gst/playback/gstdecodebin2.c: (gst_decode_bin_plugin_init):
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* gst/playback/gstplayback.c: (plugin_init):
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* gst/playback/gstqueue2.c: (plugin_init):
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* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_plugin_init):
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* sys/v4l/gstv4l.c: (plugin_init):
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Make sure gettext returns translations in UTF-8 encoding rather
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than in the current locale encoding (#546822).
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2008-08-06 13:12:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst-libs/gst/pbutils/descriptions.c: Add audio/x-qdm for qtdemux.
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|
Original commit message from CVS:
|
|
* gst-libs/gst/pbutils/descriptions.c:
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Add audio/x-qdm for qtdemux.
|
|
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2008-08-05 15:38:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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ext/vorbis/vorbisdec.c: Do not leak old taglist.
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|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisdec.c:
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Do not leak old taglist.
|
|
|
|
2008-08-04 12:35:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
|
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tests/icles/test-scale.c: Include <stdlib.h> for atoi().
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Original commit message from CVS:
|
|
* tests/icles/test-scale.c:
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Include <stdlib.h> for atoi().
|
|
|
|
2008-08-04 09:11:08 +0000 Andy Wingo <wingo@pobox.com>
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gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important documentation fix.
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|
Original commit message from CVS:
|
|
2008-08-04 Andy Wingo <wingo@pobox.com>
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* gst/audiotestsrc/gstaudiotestsrc.c: Very crucial and important
|
|
documentation fix.
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2008-08-01 13:06:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst/adder/gstadder.c: Cleanup lots of empty lines that came from gst-indent going havoc before I added the INDENT_ON/...
|
|
Original commit message from CVS:
|
|
* gst/adder/gstadder.c:
|
|
Cleanup lots of empty lines that came from gst-indent going havoc
|
|
before I added the INDENT_ON/OFF marker some time agao.
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2008-08-01 11:55:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
|
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Bump requirement to latest core and use new tag for riff formats.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
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* gst-libs/gst/riff/riff-read.c:
|
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Bump requirement to latest core and use new tag for riff formats.
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Needed for #520694.
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|
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2008-08-01 11:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
tests/examples/dynamic/: Add example app that dynamically switches between 3 'encoders'.
|
|
Original commit message from CVS:
|
|
* tests/examples/dynamic/Makefile.am:
|
|
* tests/examples/dynamic/codec-select.c: (make_encoder),
|
|
(make_pipeline), (do_switch), (my_bus_callback), (main):
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Add example app that dynamically switches between 3 'encoders'.
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|
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2008-07-31 13:06:13 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaysink.c: Add some more comments.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaysink.c: (gst_play_sink_set_vis_plugin):
|
|
Add some more comments.
|
|
|
|
2008-07-31 12:58:44 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst/videotestsrc/gstvideotestsrc.c: Discard buffers of the wrong size after renegotiation, this is perfectly possible...
|
|
Original commit message from CVS:
|
|
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_getcaps),
|
|
(gst_video_test_src_create):
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Discard buffers of the wrong size after renegotiation, this is perfectly
|
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possible with things like capsfilter that could suggest caps changes
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upstream without knowing the size of the buffer.
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2008-07-31 11:39:44 +0000 Wim Taymans <wim.taymans@gmail.com>
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tests/icles/: Add dynamic rescaling tests for the new basetransform.
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|
Original commit message from CVS:
|
|
* tests/icles/.cvsignore:
|
|
* tests/icles/Makefile.am:
|
|
* tests/icles/test-scale.c: (make_pipeline), (main):
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Add dynamic rescaling tests for the new basetransform.
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2008-07-30 19:51:36 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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gst/audioconvert/Makefile.am: Dist recently-added gstfastrandom.h.
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|
Original commit message from CVS:
|
|
* gst/audioconvert/Makefile.am:
|
|
Dist recently-added gstfastrandom.h.
|
|
|
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2008-07-30 15:29:44 +0000 Edward Hervey <bilboed@bilboed.com>
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sys/xvimage/xvimagesink.c: Fix a "may be used uninitialized in this function" which weirdly only appears on macosx (?).
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
|
|
Fix a "may be used uninitialized in this function" which weirdly only
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appears on macosx (?).
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|
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2008-07-30 09:02:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst-libs/gst/riff/riff-ids.h: Adding acid chunk for tempo and loop information.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-ids.h:
|
|
Adding acid chunk for tempo and loop information.
|
|
|
|
2008-07-29 13:01:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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sys/xvimage/Makefile.am: floor() needs linking to $(LIBM).
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|
Original commit message from CVS:
|
|
* sys/xvimage/Makefile.am:
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floor() needs linking to $(LIBM).
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|
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2008-07-29 12:35:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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ext/gnomevfs/gstgnomevfssrc.c: Aggregate short reads and add some comments and debug logging.
|
|
Original commit message from CVS:
|
|
* ext/gnomevfs/gstgnomevfssrc.c:
|
|
Aggregate short reads and add some comments and debug logging.
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Fixes #537380
|
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2008-07-29 10:26:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
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gst/playback/gstplaybasebin.c: Fix property doc markup (its not a signal).
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c:
|
|
Fix property doc markup (its not a signal).
|
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* sys/xvimage/xvimagesink.c:
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Add since tag for new proeprties (also add sice tags fro the last two
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other additions).
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2008-07-29 08:59:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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sys/xvimage/xvimagesink.*: Add autofill/colorkey properties. Fixes #538656.
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Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c:
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* sys/xvimage/xvimagesink.h:
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Add autofill/colorkey properties. Fixes #538656.
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2008-07-29 01:58:05 +0000 David Schleef <ds@schleef.org>
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sys/xvimage/xvimagesink.c: Fix rounding errors when converting colorbalance values between hardware and object proper...
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c:
|
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Fix rounding errors when converting colorbalance values
|
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between hardware and object property ranges. Partial
|
|
fix for #537889, however, there still seems to be a small
|
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drift problem that could be totem's fault.
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2008-07-28 15:34:13 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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ext/ogg/gstoggdemux.c: Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
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|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
|
|
(gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
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Don't use GST_CLOCK_TIME_NONE as start of NEWSEGMENT events.
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This fixes a critical warning.
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2008-07-28 13:12:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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ext/ogg/gstoggmux.c: Allow muxing of CELT into Ogg streams.
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Original commit message from CVS:
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* ext/ogg/gstoggmux.c:
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Allow muxing of CELT into Ogg streams.
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|
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2008-07-28 12:47:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/typefind/gsttypefindfunctions.c: Add simple typefinder for the CELT codec (www.celt-codec.org).
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|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (celt_type_find),
|
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(plugin_init):
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Add simple typefinder for the CELT codec (www.celt-codec.org).
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2008-07-27 11:12:41 +0000 Jan Gerber <j@oil21.org>
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ext/ogg/gstoggdemux.c: Fix calculation of the start time from skeleton streams.
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Original commit message from CVS:
|
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Patch by: Jan Gerber <j at oil21 dot org>
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* ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone):
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Fix calculation of the start time from skeleton streams.
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Fixes bug #530068.
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2008-07-24 13:19:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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tests/examples/seek/seek.c: Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
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Original commit message from CVS:
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* tests/examples/seek/seek.c:
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Use 64 bit constant GST_CLOCK_TIME_NONE instead of plain -1.
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2008-07-23 18:34:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/audioconvert/: Implement a linear congruential generator as pseudo random number generator for the dither noise. ...
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|
Original commit message from CVS:
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* gst/audioconvert/audioconvert.h:
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* gst/audioconvert/gstaudioquantize.c:
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(gst_audio_quantize_setup_dither),
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(gst_audio_quantize_free_dither):
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* gst/audioconvert/gstfastrandom.h:
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Implement a linear congruential generator as pseudo random number
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generator for the dither noise. This is about 2 times faster than
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using GLib's mersenne twister. Also this uses only integer math for
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generating integers while GLib internally uses floating point math.
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2008-07-23 18:27:15 +0000 Michael Smith <msmith@xiph.org>
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configure.ac: Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
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Original commit message from CVS:
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* configure.ac:
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Remove AC_ISC_POSIX; it breaks on some systems and is not needed.
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2008-07-23 13:17:31 +0000 Damien Lespiau <damien.lespiau@gmail.com>
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gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL to avoid crashes with libcs that don't like NULL strings in printf...
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Original commit message from CVS:
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Patch by: Damien Lespiau <damien.lespiau gmail com>
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* gst-libs/gst/sdp/gstsdpmessage.c: (print_media):
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Use GST_STR_NULL to avoid crashes with libcs that don't
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like NULL strings in printf args (such as the win32 one).
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Fixes #544306.
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2008-07-17 14:21:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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sys/xvimage/xvimagesink.c: Oops - set the size of the image used for probing back to 1x1, for consistency with ximage...
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Original commit message from CVS:
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
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Oops - set the size of the image used for probing back to 1x1, for
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consistency with ximagesink
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2008-07-17 13:57:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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sys/: it's not legal to ask the
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Original commit message from CVS:
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* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
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(gst_ximagesink_ximage_new):
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
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(gst_xvimagesink_xvimage_new):
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Apparently on Solaris and OS/X (at least), it's not legal to ask the
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X server to attach to a shared memory segment after we've deleted it,
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with the result that MIT-SHM is disabled. Instead, remove it only after
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X succeeds in attaching too.
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2008-07-17 02:30:24 +0000 David Schleef <ds@schleef.org>
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gst/audiotestsrc/gstaudiotestsrc.*: Add 'ticks', a 1/30 second sine wave pulse every second.
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Original commit message from CVS:
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* gst/audiotestsrc/gstaudiotestsrc.c:
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* gst/audiotestsrc/gstaudiotestsrc.h:
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Add 'ticks', a 1/30 second sine wave pulse every second.
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2008-07-15 22:43:16 +0000 David Schleef <ds@schleef.org>
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gst-libs/gst/video/video.c: Revert ABI change.
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Original commit message from CVS:
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* gst-libs/gst/video/video.c: Revert ABI change.
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2008-07-15 13:05:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst-libs/gst/riff/riff-media.c: Make it impossible to have NULL caps at the point where we set framerate and other th...
|
|
Original commit message from CVS:
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* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
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Make it impossible to have NULL caps at the point where we set
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framerate and other things. Also don't return immediately for "3ivd"
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video and let framerate, etc be set. Might fix bug #542508.
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2008-07-14 17:06:26 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
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gst-libs/gst/video/video.c: Video format can also be conveniently determined from (many) non-fixed caps.
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Original commit message from CVS:
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* gst-libs/gst/video/video.c: (gst_video_format_parse_caps):
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Video format can also be conveniently determined from (many)
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non-fixed caps.
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2008-07-14 08:18:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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gst/playback/: First stab at integrating DVD subpicture overlay into playbin. Successfully plugs and plays, but the q...
|
|
Original commit message from CVS:
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* gst/playback/gstplaybasebin.c:
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* gst/playback/gstplaybasebin.h:
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* gst/playback/gstplaybin.c:
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* gst/playback/gststreamselector.c:
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First stab at integrating DVD subpicture overlay into
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playbin. Successfully plugs and plays, but the queues need
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shrinking - 3 seconds of video is too much buffering.
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2008-07-11 18:06:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst/audioconvert/gstaudioconvert.c: Remove now obsolete note in the docs.
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|
Original commit message from CVS:
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* gst/audioconvert/gstaudioconvert.c:
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Remove now obsolete note in the docs.
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|
|
|
2008-07-11 06:10:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
|
|
Original commit message from CVS:
|
|
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/gst-plugins-base-plugins.hierarchy:
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|
* docs/plugins/gst-plugins-base-plugins.interfaces:
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|
* docs/plugins/gst-plugins-base-plugins.prerequisites:
|
|
* docs/plugins/gst-plugins-base-plugins.signals:
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* docs/plugins/inspect/plugin-adder.xml:
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* docs/plugins/inspect/plugin-alsa.xml:
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* docs/plugins/inspect/plugin-audioconvert.xml:
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* docs/plugins/inspect/plugin-audiorate.xml:
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* docs/plugins/inspect/plugin-audioresample.xml:
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* docs/plugins/inspect/plugin-audiotestsrc.xml:
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* docs/plugins/inspect/plugin-cdparanoia.xml:
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|
* docs/plugins/inspect/plugin-decodebin.xml:
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* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
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|
* docs/plugins/inspect/plugin-gdp.xml:
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|
* docs/plugins/inspect/plugin-gnomevfs.xml:
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|
* docs/plugins/inspect/plugin-libvisual.xml:
|
|
* docs/plugins/inspect/plugin-ogg.xml:
|
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* docs/plugins/inspect/plugin-pango.xml:
|
|
* docs/plugins/inspect/plugin-playback.xml:
|
|
* docs/plugins/inspect/plugin-queue2.xml:
|
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* docs/plugins/inspect/plugin-subparse.xml:
|
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* docs/plugins/inspect/plugin-tcp.xml:
|
|
* docs/plugins/inspect/plugin-theora.xml:
|
|
* docs/plugins/inspect/plugin-typefindfunctions.xml:
|
|
* docs/plugins/inspect/plugin-uridecodebin.xml:
|
|
* docs/plugins/inspect/plugin-video4linux.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:
|
|
* ext/alsa/gstalsamixer.c:
|
|
* ext/alsa/gstalsasink.c:
|
|
* ext/alsa/gstalsasrc.c:
|
|
* ext/gio/gstgiosink.c:
|
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* ext/gio/gstgiosrc.c:
|
|
* ext/gio/gstgiostreamsink.c:
|
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* ext/gio/gstgiostreamsrc.c:
|
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* ext/gnomevfs/gstgnomevfssink.c:
|
|
* ext/gnomevfs/gstgnomevfssrc.c:
|
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* ext/ogg/gstoggdemux.c:
|
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* ext/ogg/gstoggmux.c:
|
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* ext/pango/gstclockoverlay.c:
|
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* ext/pango/gsttextoverlay.c:
|
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* ext/pango/gsttextrender.c:
|
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* ext/pango/gsttimeoverlay.c:
|
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* ext/theora/theoradec.c:
|
|
* ext/theora/theoraenc.c:
|
|
* ext/theora/theoraparse.c:
|
|
* ext/vorbis/vorbisdec.c:
|
|
* ext/vorbis/vorbisenc.c:
|
|
* ext/vorbis/vorbisparse.c:
|
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* ext/vorbis/vorbistag.c:
|
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* gst/adder/gstadder.c:
|
|
* gst/audioconvert/gstaudioconvert.c:
|
|
* gst/audioresample/gstaudioresample.c:
|
|
* gst/audiotestsrc/gstaudiotestsrc.c:
|
|
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
|
|
* gst/gdp/gstgdpdepay.c:
|
|
* gst/gdp/gstgdppay.c:
|
|
* gst/playback/gstdecodebin2.c:
|
|
* gst/playback/gstplaybin.c:
|
|
* gst/playback/gstplaybin2.c:
|
|
* gst/playback/gstqueue2.c:
|
|
* gst/playback/gsturidecodebin.c:
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* gst/tcp/gstmultifdsink.c:
|
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* gst/tcp/gsttcpserversink.c:
|
|
* gst/videorate/gstvideorate.c:
|
|
* gst/videoscale/gstvideoscale.c:
|
|
* gst/videotestsrc/gstvideotestsrc.c:
|
|
* gst/volume/gstvolume.c:
|
|
* sys/ximage/ximagesink.c:
|
|
* sys/xvimage/xvimagesink.c:
|
|
Cleanup Plugin docs. Link to signals and properties. Fix sub-section
|
|
titles. Drop mentining that all our example pipelines are "simple"
|
|
pipelines.
|
|
|
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2008-07-10 21:06:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe...
|
|
Original commit message from CVS:
|
|
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
|
* docs/plugins/gst-plugins-base-plugins.interfaces:
|
|
* docs/plugins/gst-plugins-base-plugins.prerequisites:
|
|
* docs/plugins/gst-plugins-base-plugins.signals:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.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-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gdp.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.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-queue2.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-uridecodebin.xml:
|
|
* docs/plugins/inspect/plugin-video4linux.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:
|
|
* ext/alsa/gstalsamixer.c:
|
|
* ext/alsa/gstalsasink.c:
|
|
* ext/alsa/gstalsasrc.c:
|
|
* ext/gio/gstgiosink.c:
|
|
* ext/gio/gstgiosrc.c:
|
|
* ext/gio/gstgiostreamsink.c:
|
|
* ext/gio/gstgiostreamsrc.c:
|
|
* ext/gnomevfs/gstgnomevfssink.c:
|
|
* ext/gnomevfs/gstgnomevfssrc.c:
|
|
* ext/ogg/gstoggdemux.c:
|
|
* ext/ogg/gstoggmux.c:
|
|
* ext/pango/gstclockoverlay.c:
|
|
* ext/pango/gsttextoverlay.c:
|
|
* ext/pango/gsttextrender.c:
|
|
* ext/pango/gsttimeoverlay.c:
|
|
* ext/theora/theoradec.c:
|
|
* ext/theora/theoraenc.c:
|
|
* ext/theora/theoraparse.c:
|
|
* ext/vorbis/vorbisdec.c:
|
|
* ext/vorbis/vorbisenc.c:
|
|
* ext/vorbis/vorbisparse.c:
|
|
* ext/vorbis/vorbistag.c:
|
|
* gst/adder/gstadder.c:
|
|
* gst/audioconvert/gstaudioconvert.c:
|
|
* gst/audioresample/gstaudioresample.c:
|
|
* gst/audiotestsrc/gstaudiotestsrc.c:
|
|
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
|
|
* gst/gdp/gstgdpdepay.c:
|
|
* gst/gdp/gstgdppay.c:
|
|
* gst/playback/gstdecodebin2.c:
|
|
* gst/playback/gstplaybin.c:
|
|
* gst/playback/gstplaybin2.c:
|
|
* gst/playback/gstqueue2.c:
|
|
* gst/playback/gsturidecodebin.c:
|
|
* gst/tcp/gstmultifdsink.c:
|
|
* gst/tcp/gsttcpserversink.c:
|
|
* gst/videorate/gstvideorate.c:
|
|
* gst/videoscale/gstvideoscale.c:
|
|
* gst/videotestsrc/gstvideotestsrc.c:
|
|
* gst/volume/gstvolume.c:
|
|
* sys/ximage/ximagesink.c:
|
|
* sys/xvimage/xvimagesink.c:
|
|
Cleanup Plugin docs. Link to signals and properties. Fix sub-section
|
|
titles. Drop mentining that all our example pipelines are "simple"
|
|
pipelines.
|
|
|
|
2008-07-07 17:25:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
tests/examples/seek/Makefile.am: Fix out of tree build by adding all required CFLAGS.
|
|
Original commit message from CVS:
|
|
* tests/examples/seek/Makefile.am:
|
|
Fix out of tree build by adding all required CFLAGS.
|
|
|
|
2008-07-07 09:55:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/playback/gstdecodebin.c: And ref the pad before returning it again when linking to the queue failed. Otherwise we...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (add_raw_queue):
|
|
And ref the pad before returning it again when linking to the queue
|
|
failed. Otherwise we will unref the pad twice later and things break.
|
|
|
|
2008-07-07 09:48:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/playback/gstdecodebin.c: If linking the raw pad with a queue fails, try it without a queue instead of failing com...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (add_raw_queue):
|
|
If linking the raw pad with a queue fails, try it without a queue
|
|
instead of failing completely. This should never happen.
|
|
|
|
2008-07-06 23:22:12 +0000 Evgeniy Stepanov <eugeni.stepanov@gmail.com>
|
|
|
|
gst/playback/gstdecodebin.c: Add a queue after a demuxer if the demuxer outputs raw data. This was done before only f...
|
|
Original commit message from CVS:
|
|
Patch by: Evgeniy Stepanov <eugeni dot stepanov at gmail dot com>
|
|
* gst/playback/gstdecodebin.c: (add_raw_queue), (close_pad_link):
|
|
Add a queue after a demuxer if the demuxer outputs raw data. This was
|
|
done before only for non-raw data but is required in this case too.
|
|
Fixes bug #540215.
|
|
decodebin2 doesn't have this issue because all streams of a group
|
|
go through multiqueue.
|
|
|
|
2008-07-03 09:12:49 +0000 Damien Lespiau <damien.lespiau@gmail.com>
|
|
|
|
gst-libs/gst/sdp/gstsdpmessage.c: Makes libgstsdp compile with mingw32 by defining the right WINVER so that getaddrin...
|
|
Original commit message from CVS:
|
|
Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
|
|
* gst-libs/gst/sdp/gstsdpmessage.c:
|
|
Makes libgstsdp compile with mingw32 by defining the right WINVER so
|
|
that getaddrinfo() can be used. Fixes #541358.
|
|
|
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2008-07-01 13:22:49 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst/videotestsrc/gstvideotestsrc.*: Cleanups, use default property values as defines.
|
|
Original commit message from CVS:
|
|
* gst/videotestsrc/gstvideotestsrc.c:
|
|
(gst_video_test_src_class_init), (gst_video_test_src_init),
|
|
(gst_video_test_src_set_property),
|
|
(gst_video_test_src_get_property), (gst_video_test_src_create):
|
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* gst/videotestsrc/gstvideotestsrc.h:
|
|
Cleanups, use default property values as defines.
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Add property to enable/disable peer buffer allocation.
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|
|
2008-06-30 09:46:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
tests/check/: Enable unit tests on PPC again as the bugs are now fixed.
|
|
Original commit message from CVS:
|
|
* tests/check/elements/gdpdepay.c: (gdpdepay_suite):
|
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* tests/check/pipelines/streamheader.c: (streamheader_suite):
|
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Enable unit tests on PPC again as the bugs are now fixed.
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2008-06-30 09:20:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
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gst-libs/gst/riff/: Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-ids.h:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
|
|
(gst_riff_create_audio_template_caps):
|
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Add support for ADPCM IMA DK3 and DK4 variant in RIFF containers.
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Fixes bug #540351.
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|
|
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2008-06-30 08:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
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gst/ffmpegcolorspace/: Only set/get on the PAL8 format, ffmpegcolorspace doesn't support it on other formats. Also ad...
|
|
Original commit message from CVS:
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
|
|
(gst_ffmpeg_pixfmt_to_caps):
|
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* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
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(gst_ffmpegcsp_get_unit_size):
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Only set/get on the PAL8 format, ffmpegcolorspace doesn't support
|
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it on other formats. Also adjust the unit size only for that format
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to not include the palette. Fixes bug #540497.
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|
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2008-06-29 13:45:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst/adder/gstadder.c: Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
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Original commit message from CVS:
|
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* gst/adder/gstadder.c:
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Use GST_DEBUG_FUNCPTR and remove some extra vlnak lines.
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2008-06-27 07:55:40 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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ChangeLog: ChangeLog surgery.
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Original commit message from CVS:
|
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* ChangeLog:
|
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ChangeLog surgery.
|
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* tests/examples/seek/seek.c:
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Move variable into ifdef too.
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2008-06-27 07:42:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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tests/examples/seek/seek.c: Include config.h and check if we have X. Fixes: #540334.
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Original commit message from CVS:
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* tests/examples/seek/seek.c:
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Include config.h and check if we have X. Fixes: #540334.
|
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2008-06-26 06:03:38 +0000 Sam Morris <sam@robots.org.to.uk>
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gst-libs/gst/interfaces/mixertrack.c: API: Add "index" property to GstMixerTrack to differantiate between multiple mi...
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Original commit message from CVS:
|
|
Patch by: Sam Morris <sam at robots dot org to uk>
|
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* gst-libs/gst/interfaces/mixertrack.c:
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(gst_mixer_track_class_init), (gst_mixer_track_get_property),
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(gst_mixer_track_set_property):
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API: Add "index" property to GstMixerTrack to differantiate between
|
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multiple mixer tracks with the same label.
|
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* ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
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* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
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Set the "index" property of GstMixerTrack to the index given by ALSA.
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Fixes bug #528299.
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2008-06-25 13:15:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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tests/examples/seek/: Remove libgstvideo usage. Use gtk_get_option_group instead of gtk_init().
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Original commit message from CVS:
|
|
* tests/examples/seek/Makefile.am:
|
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* tests/examples/seek/seek.c:
|
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Remove libgstvideo usage. Use gtk_get_option_group instead of
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gtk_init().
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2008-06-24 16:27:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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tests/check/Makefile.am: Name the test registry format neutral.
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Original commit message from CVS:
|
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* tests/check/Makefile.am:
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Name the test registry format neutral.
|
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|
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2008-06-24 16:22:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
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gst/playback/gstqueue2.c: Do not double notify. Remove the unsued return value.
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|
Original commit message from CVS:
|
|
* gst/playback/gstqueue2.c:
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Do not double notify. Remove the unsued return value.
|
|
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2008-06-24 16:15:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
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ext/alsa/gstalsamixer.c: Also consider "speaker" as a name for master volume. If that doesn't help look for the first...
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsamixer.c:
|
|
Also consider "speaker" as a name for master volume. If that doesn't
|
|
help look for the first non-mono volume control that also has a
|
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playback switch.
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|
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2008-06-24 16:10:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
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ChangeLog: Forgot to save the ChangeLog :/
|
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Original commit message from CVS:
|
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* ChangeLog:
|
|
Forgot to save the ChangeLog :/
|
|
|
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2008-06-24 16:05:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
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|
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tests/examples/seek/: Embedd the xwindow.
|
|
Original commit message from CVS:
|
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* tests/examples/seek/Makefile.am:
|
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* tests/examples/seek/seek.c:
|
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Embedd the xwindow.
|
|
|
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2008-06-24 01:14:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
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sys/ximage/ximagesink.h: When the caps change, make sure to re-draw borders in force-aspect-ratio=true mode.
|
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Original commit message from CVS:
|
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* sys/ximage/ximagesink.c (gst_ximagesink_ximage_put),
|
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(gst_ximagesink_setcaps):
|
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* sys/ximage/ximagesink.h:
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When the caps change, make sure to re-draw borders in
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force-aspect-ratio=true mode.
|
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* sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_put):
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Don't clear the border_draw flag until we actually draw the border.
|
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* tests/check/Makefile.am:
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Ignore alsasink/src during the states test too, so it doesn't fail
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when running without access to the sound device.
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|
|
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2008-06-22 18:35:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
|
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tests/examples/seek/seek.c: Fix crasher when playing a parse-launch line the 2nd time.
|
|
Original commit message from CVS:
|
|
* tests/examples/seek/seek.c:
|
|
Fix crasher when playing a parse-launch line the 2nd time.
|
|
|
|
2008-06-21 18:56:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
tests/check/pipelines/oggmux.c: Properly ifdef tests to fix compilation.
|
|
Original commit message from CVS:
|
|
* tests/check/pipelines/oggmux.c:
|
|
Properly ifdef tests to fix compilation.
|
|
|
|
2008-06-21 10:25:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* ChangeLog:
|
|
break long lines
|
|
Original commit message from CVS:
|
|
break long lines
|
|
|
|
2008-06-20 18:24:24 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/playback/: Add get-video-pad, get-audio-pad, get-text-pad action signals to playbin2. This allows the user to get...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplay-marshal.list:
|
|
* gst/playback/gstplaybin2.c:
|
|
Add get-video-pad, get-audio-pad, get-text-pad action signals to
|
|
playbin2. This allows the user to get to the selector's sinkpads, and
|
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thus inspect a range of things - caps, tags, etc.
|
|
|
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2008-06-20 17:27:03 +0000 Michael Smith <msmith@xiph.org>
|
|
|
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gst/playback/gstplaybin2.c: Use a different constant for the convert-frame signal id.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin2.c:
|
|
Use a different constant for the convert-frame signal id.
|
|
Fixes #537009.
|
|
|
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2008-06-20 17:18:55 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/playback/: Fix a whole bunch of typos in comments and log statements.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin2.c:
|
|
* gst/playback/gstplaysink.c:
|
|
Fix a whole bunch of typos in comments and log statements.
|
|
|
|
2008-06-20 17:02:48 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
sys/xvimage/xvimagesink.c: Don't set colour balance values on the Xv port if the user hasn't changed them (via proper...
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c:
|
|
Don't set colour balance values on the Xv port if the user hasn't
|
|
changed them (via properties or the interface). Avoids accumulating
|
|
rounding errors for the common case.
|
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Partial fix for bug #537889.
|
|
|
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2008-06-20 16:56:18 +0000 Michael Smith <msmith@xiph.org>
|
|
|
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gst/playback/gstdecodebin2.c: Ensure decodebin2 emits 'drained' signal once, and only once, when all pads are drained.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin2.c:
|
|
Ensure decodebin2 emits 'drained' signal once, and only once, when all
|
|
pads are drained.
|
|
|
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2008-06-20 16:12:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
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* gst/tcp/README:
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apparently it's an error to specify nc -l -p 3000 - though the short usage does not make it very clear that you can d...
|
|
Original commit message from CVS:
|
|
apparently it's an error to specify nc -l -p 3000 - though the short usage
|
|
does not make it very clear that you can drop the host arg with -l
|
|
|
|
2008-06-20 09:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/vorbis/vorbisenc.c: Report the encoder latency. Fixes #538232.
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_get_latency),
|
|
(gst_vorbis_enc_src_query), (gst_vorbis_enc_chain):
|
|
Report the encoder latency. Fixes #538232.
|
|
|
|
2008-06-20 09:19:59 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybin2.c: Implement the source property, emit notify when it changes in the underlying uridecodebin.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin2.c: (gst_play_bin_get_property),
|
|
(notify_source), (activate_group):
|
|
Implement the source property, emit notify when it changes in the
|
|
underlying uridecodebin.
|
|
|
|
2008-06-20 09:14:26 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
tests/examples/seek/seek.c: Free and clear the seek element list so that we don't use invalid references when seeking...
|
|
Original commit message from CVS:
|
|
* tests/examples/seek/seek.c: (stop_cb):
|
|
Free and clear the seek element list so that we don't use invalid
|
|
references when seeking after recreating a gst-launch line.
|
|
|
|
2008-06-20 09:09:37 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.c: Report latency even if we are not live instead of hiding it.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_query), (gst_base_audio_sink_skew_slaving),
|
|
(gst_base_audio_sink_render):
|
|
Report latency even if we are not live instead of hiding it.
|
|
Take ts-offset and render-delay of the basesink into account when
|
|
scheduling samples.
|
|
Rework the clipping code so that we can take the various offsets into
|
|
account and still do correct clipping.
|
|
|
|
2008-06-20 08:52:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
configure.ac: Bump verion back to devel -> 0.10.20.1
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Bump verion back to devel -> 0.10.20.1
|
|
|
|
2008-06-20 08:47:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst-libs/gst/tag/tags.c: Don't increase the size of non-string image buffers by one as this might in theory confuse d...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/tags.c: (gst_tag_image_data_to_image_buffer):
|
|
Don't increase the size of non-string image buffers by one as this
|
|
might in theory confuse decoders. Still increase it by one for string
|
|
image buffers to append '\0'.
|
|
|
|
2008-06-20 08:45:13 +0000 Antoine Tremblay <hexa00@gmail.com>
|
|
|
|
gst/gdp/gstgdppay.c: Fix a buffer memleak and remove a confusing and wrong debug output.
|
|
Original commit message from CVS:
|
|
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
|
|
* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset):
|
|
Fix a buffer memleak and remove a confusing and wrong debug output.
|
|
Fixes bug #538663.
|
|
|
|
2008-06-19 11:25:37 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
examples/app/appsink-src.c: Don't use a buffer after unreffing it.
|
|
Original commit message from CVS:
|
|
* examples/app/appsink-src.c: (on_new_buffer_from_source):
|
|
Don't use a buffer after unreffing it.
|
|
|
|
=== release 0.10.20 ===
|
|
|
|
2008-06-18 14:36:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
* ChangeLog:
|
|
* NEWS:
|
|
* RELEASE:
|
|
* configure.ac:
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
|
* docs/plugins/gst-plugins-base-plugins.interfaces:
|
|
* docs/plugins/gst-plugins-base-plugins.prerequisites:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.xml:
|
|
* docs/plugins/inspect/plugin-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-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gdp.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.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-queue2.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-uridecodebin.xml:
|
|
* docs/plugins/inspect/plugin-video4linux.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:
|
|
* po/LINGUAS:
|
|
* win32/common/config.h:
|
|
Release 0.10.20
|
|
Original commit message from CVS:
|
|
Release 0.10.20
|
|
|
|
2008-06-18 14:32:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/bg.po:
|
|
* po/ca.po:
|
|
* po/cs.po:
|
|
* po/da.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/es.po:
|
|
* po/fi.po:
|
|
* po/fr.po:
|
|
* po/hu.po:
|
|
* po/it.po:
|
|
* po/lt.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/pl.po:
|
|
* po/ru.po:
|
|
* po/sk.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
* po/zh_CN.po:
|
|
Update .po files
|
|
Original commit message from CVS:
|
|
Update .po files
|
|
|
|
2008-06-18 06:31:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
Fix gtk-doc warnings. Also don't misuse api-doc comments for normal comments.
|
|
Original commit message from CVS:
|
|
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
|
|
* examples/app/appsrc-ra.c:
|
|
* examples/app/appsrc-seekable.c:
|
|
* examples/app/appsrc-stream.c:
|
|
* examples/app/appsrc-stream2.c:
|
|
* ext/directfb/dfbvideosink.h:
|
|
* ext/metadata/gstbasemetadata.c:
|
|
* ext/metadata/gstbasemetadata.h:
|
|
* ext/metadata/metadata.c:
|
|
* ext/metadata/metadataexif.c:
|
|
* ext/theora/theoradec.h:
|
|
* gst/deinterlace2/gstdeinterlace2.h:
|
|
* gst/deinterlace2/tvtime/speedy.c:
|
|
* gst/deinterlace2/tvtime/speedy.h:
|
|
* gst/deinterlace2/tvtime/vfir.c:
|
|
Fix gtk-doc warnings. Also don't misuse api-doc comments for normal
|
|
comments.
|
|
|
|
2008-06-16 14:11:36 +0000 Andy Wingo <wingo@pobox.com>
|
|
|
|
* gst-libs/gst/app/gstappsrc.c:
|
|
gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
|
|
Original commit message from CVS:
|
|
2008-06-16 Andy Wingo <wingo@pobox.com>
|
|
* gst-libs/gst/app/gstappsrc.c (gst_app_src_set_max_bytes)
|
|
(gst_app_src_get_max_bytes, gst_app_src_push_buffer): Use
|
|
G_GUINT64_FORMAT. Avoid overflow in get_max_bytes().
|
|
|
|
2008-06-16 07:30:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
Final round of doc updates.
|
|
Original commit message from CVS:
|
|
* gst/rtpmanager/gstrtpjitterbuffer.c:
|
|
* gst/speed/gstspeed.c:
|
|
* gst/speexresample/gstspeexresample.c:
|
|
* gst/videosignal/gstvideoanalyse.c:
|
|
* gst/videosignal/gstvideodetect.c:
|
|
* gst/videosignal/gstvideomark.c:
|
|
* sys/dvb/gstdvbsrc.c:
|
|
* sys/oss4/oss4-mixer.c:
|
|
* sys/oss4/oss4-sink.c:
|
|
* sys/oss4/oss4-source.c:
|
|
* sys/wininet/gstwininetsrc.c:
|
|
Final round of doc updates.
|
|
|
|
2008-06-13 11:59:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml
|
|
Original commit message from CVS:
|
|
* docs/plugins/Makefile.am:
|
|
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
|
|
* docs/plugins/gst-plugins-bad-plugins.args:
|
|
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
|
|
* docs/plugins/gst-plugins-bad-plugins.interfaces:
|
|
* docs/plugins/gst-plugins-bad-plugins.prerequisites:
|
|
* docs/plugins/gst-plugins-bad-plugins.signals:
|
|
* docs/plugins/inspect/plugin-alsaspdif.xml:
|
|
* docs/plugins/inspect/plugin-amrwb.xml:
|
|
* docs/plugins/inspect/plugin-app.xml:
|
|
* docs/plugins/inspect/plugin-bayer.xml:
|
|
* docs/plugins/inspect/plugin-bz2.xml:
|
|
* docs/plugins/inspect/plugin-cdaudio.xml:
|
|
* docs/plugins/inspect/plugin-cdxaparse.xml:
|
|
* docs/plugins/inspect/plugin-dtsdec.xml:
|
|
* docs/plugins/inspect/plugin-dvb.xml:
|
|
* docs/plugins/inspect/plugin-dvdspu.xml:
|
|
* docs/plugins/inspect/plugin-faac.xml:
|
|
* docs/plugins/inspect/plugin-faad.xml:
|
|
* docs/plugins/inspect/plugin-fbdevsink.xml:
|
|
* docs/plugins/inspect/plugin-festival.xml:
|
|
* docs/plugins/inspect/plugin-filter.xml:
|
|
* docs/plugins/inspect/plugin-flvdemux.xml:
|
|
* docs/plugins/inspect/plugin-freeze.xml:
|
|
* docs/plugins/inspect/plugin-gsm.xml:
|
|
* docs/plugins/inspect/plugin-gstinterlace.xml:
|
|
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
|
|
* docs/plugins/inspect/plugin-h264parse.xml:
|
|
* docs/plugins/inspect/plugin-interleave.xml:
|
|
* docs/plugins/inspect/plugin-jack.xml:
|
|
* docs/plugins/inspect/plugin-ladspa.xml:
|
|
* docs/plugins/inspect/plugin-metadata.xml:
|
|
* docs/plugins/inspect/plugin-mms.xml:
|
|
* docs/plugins/inspect/plugin-modplug.xml:
|
|
* docs/plugins/inspect/plugin-mpeg2enc.xml:
|
|
* docs/plugins/inspect/plugin-mpeg4videoparse.xml:
|
|
* docs/plugins/inspect/plugin-mpegtsparse.xml:
|
|
* docs/plugins/inspect/plugin-mpegvideoparse.xml:
|
|
* docs/plugins/inspect/plugin-musepack.xml:
|
|
* docs/plugins/inspect/plugin-musicbrainz.xml:
|
|
* docs/plugins/inspect/plugin-mve.xml:
|
|
* docs/plugins/inspect/plugin-mythtv.xml
|
|
* docs/plugins/inspect/plugin-nas.xml:
|
|
* docs/plugins/inspect/plugin-neon.xml:
|
|
* docs/plugins/inspect/plugin-nsfdec.xml:
|
|
* docs/plugins/inspect/plugin-nuvdemux.xml:
|
|
* docs/plugins/inspect/plugin-oss4.xml
|
|
* docs/plugins/inspect/plugin-rawparse.xml:
|
|
* docs/plugins/inspect/plugin-real.xml:
|
|
* docs/plugins/inspect/plugin-replaygain.xml:
|
|
* docs/plugins/inspect/plugin-rfbsrc.xml:
|
|
* docs/plugins/inspect/plugin-sdl.xml:
|
|
* docs/plugins/inspect/plugin-sdp.xml:
|
|
* docs/plugins/inspect/plugin-selector.xml:
|
|
* docs/plugins/inspect/plugin-sndfile.xml:
|
|
* docs/plugins/inspect/plugin-soundtouch.xml:
|
|
* docs/plugins/inspect/plugin-spcdec.xml:
|
|
* docs/plugins/inspect/plugin-speed.xml:
|
|
* docs/plugins/inspect/plugin-speexresample.xml:
|
|
* docs/plugins/inspect/plugin-stereo.xml:
|
|
* docs/plugins/inspect/plugin-subenc.xml
|
|
* docs/plugins/inspect/plugin-timidity.xml:
|
|
* docs/plugins/inspect/plugin-tta.xml:
|
|
* docs/plugins/inspect/plugin-vcdsrc.xml:
|
|
* docs/plugins/inspect/plugin-videosignal.xml:
|
|
* docs/plugins/inspect/plugin-vmnc.xml:
|
|
* docs/plugins/inspect/plugin-wildmidi.xml:
|
|
* docs/plugins/inspect/plugin-x264.xml:
|
|
* docs/plugins/inspect/plugin-xvid.xml:
|
|
* docs/plugins/inspect/plugin-y4menc.xml:
|
|
* ext/amrwb/gstamrwbdec.c:
|
|
* ext/amrwb/gstamrwbenc.c:
|
|
* ext/amrwb/gstamrwbparse.c:
|
|
* ext/dc1394/gstdc1394.c:
|
|
* ext/directfb/dfbvideosink.c:
|
|
* ext/ivorbis/vorbisdec.c:
|
|
* ext/jack/gstjackaudiosink.c:
|
|
* ext/mpeg2enc/gstmpeg2enc.cc:
|
|
* ext/mplex/gstmplex.cc:
|
|
* ext/musicbrainz/gsttrm.c:
|
|
* ext/mythtv/gstmythtvsrc.c:
|
|
* ext/theora/theoradec.c:
|
|
* ext/timidity/gsttimidity.c:
|
|
* ext/timidity/gstwildmidi.c:
|
|
* gst-libs/gst/app/gstappsink.c:
|
|
* gst/deinterlace/gstdeinterlace.c:
|
|
* gst/dvdspu/gstdvdspu.c:
|
|
* gst/festival/gstfestival.c:
|
|
* gst/freeze/gstfreeze.c:
|
|
* gst/interleave/deinterleave.c:
|
|
* gst/interleave/interleave.c:
|
|
* gst/modplug/gstmodplug.cc:
|
|
* gst/nuvdemux/gstnuvdemux.c:
|
|
Add missing elements to docs. Fix doc-markup: use convinience syntax
|
|
for examples (produces valid docbook), add several refsec2 when we
|
|
have several titles. Fix some types.
|
|
|
|
2008-06-12 15:47:03 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
examples/app/: Add beefed up example app from bug #413418. It now also uses appsink instead of fakesink for more ulti...
|
|
Original commit message from CVS:
|
|
* examples/app/.cvsignore:
|
|
* examples/app/Makefile.am:
|
|
* examples/app/appsink-src.c: (on_new_buffer_from_source),
|
|
(on_source_message), (on_sink_message), (main):
|
|
Add beefed up example app from bug #413418. It now also uses appsink
|
|
instead of fakesink for more ultimate coolness.
|
|
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
|
|
(gst_app_src_init), (gst_app_src_set_property),
|
|
(gst_app_src_get_property), (gst_app_src_unlock),
|
|
(gst_app_src_unlock_stop), (gst_app_src_create),
|
|
(gst_app_src_set_max_bytes), (gst_app_src_push_buffer),
|
|
(gst_app_src_end_of_stream):
|
|
* gst-libs/gst/app/gstappsrc.h:
|
|
Add block property to allow push based implementation to block when we
|
|
fill up the appsrc queues.
|
|
Emit the enough-data signal while releasing our lock.
|
|
|
|
2008-06-12 14:50:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
examples/app/.cvsignore: Ignore more.
|
|
Original commit message from CVS:
|
|
* examples/app/.cvsignore:
|
|
Ignore more.
|
|
|
|
2008-06-12 14:49:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
Do not use short_description in section docs for elements. We extract them from element details and there will be war...
|
|
Original commit message from CVS:
|
|
* ext/dc1394/gstdc1394.c:
|
|
* ext/ivorbis/vorbisdec.c:
|
|
* ext/jack/gstjackaudiosink.c:
|
|
* ext/metadata/gstmetadatademux.c:
|
|
* ext/mythtv/gstmythtvsrc.c:
|
|
* ext/theora/theoradec.c:
|
|
* gst-libs/gst/app/gstappsink.c:
|
|
* gst/bayer/gstbayer2rgb.c:
|
|
* gst/deinterlace/gstdeinterlace.c:
|
|
* gst/rawparse/gstaudioparse.c:
|
|
* gst/rawparse/gstvideoparse.c:
|
|
* gst/rtpmanager/gstrtpbin.c:
|
|
* gst/rtpmanager/gstrtpclient.c:
|
|
* gst/rtpmanager/gstrtpjitterbuffer.c:
|
|
* gst/rtpmanager/gstrtpptdemux.c:
|
|
* gst/rtpmanager/gstrtpsession.c:
|
|
* gst/rtpmanager/gstrtpssrcdemux.c:
|
|
* gst/selector/gstinputselector.c:
|
|
* gst/selector/gstoutputselector.c:
|
|
* gst/videosignal/gstvideoanalyse.c:
|
|
* gst/videosignal/gstvideodetect.c:
|
|
* gst/videosignal/gstvideomark.c:
|
|
* sys/oss4/oss4-mixer.c:
|
|
* sys/oss4/oss4-sink.c:
|
|
* sys/oss4/oss4-source.c:
|
|
Do not use short_description in section docs for elements. We extract
|
|
them from element details and there will be warnings if they differ.
|
|
Also fixing up the ChangeLog order.
|
|
|
|
2008-06-11 21:17:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
configure.ac: 0.10.19.3 pre-release
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
0.10.19.3 pre-release
|
|
|
|
2008-06-11 20:13:00 +0000 David Schleef <ds@schleef.org>
|
|
|
|
gst-libs/gst/rtsp/gstrtspconnection.c: Fix build on win32.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
Fix build on win32.
|
|
Patch By: David Schleef <ds@schleef.org>
|
|
Fixes: #536874
|
|
|
|
2008-06-11 09:35:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
ext/gio/gstgiobasesrc.*: Try to read the requested number of bytes, even if the first read returns less than requeste...
|
|
Original commit message from CVS:
|
|
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_finalize),
|
|
(gst_gio_base_src_create):
|
|
* ext/gio/gstgiobasesrc.h:
|
|
Try to read the requested number of bytes, even if the first
|
|
read returns less than requested, until nothing is read anymore
|
|
or we have the requested amount of bytes. This fixes playback of
|
|
files via Samba as Samba only allows to read 64k at once.
|
|
Implement a caching algorithm that makes sure that we read at
|
|
least 4k of data every time. Some elements will try to read a few
|
|
bytes, then seek, read again a few bytes and so on and this is
|
|
painfully slow as every operation has to go over DBus if GVfs is
|
|
used as backend.
|
|
Fixes bug #536849 and #536848.
|
|
* ext/gio/gstgiosrc.c: (gst_gio_src_class_init),
|
|
(gst_gio_src_check_get_range):
|
|
Override check_get_range() to blacklist http/https URIs
|
|
and whitelist file URIs. More to be added on demand.
|
|
|
|
2008-06-06 16:50:51 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
examples/app/: Added 3 more example application for using appsrc in random-access mode, pull-mode streaming and pull ...
|
|
Original commit message from CVS:
|
|
* examples/app/Makefile.am:
|
|
* examples/app/appsrc-ra.c: (feed_data), (seek_data),
|
|
(found_source), (bus_message), (main):
|
|
* examples/app/appsrc-seekable.c: (feed_data), (seek_data),
|
|
(found_source), (bus_message), (main):
|
|
* examples/app/appsrc-stream2.c: (feed_data), (found_source),
|
|
(bus_message), (main):
|
|
Added 3 more example application for using appsrc in random-access mode,
|
|
pull-mode streaming and pull mode seekable.
|
|
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
|
|
(gst_app_src_start), (gst_app_src_do_get_size),
|
|
(gst_app_src_create):
|
|
* gst-libs/gst/app/gstappsrc.h:
|
|
Make stream-type property writable.
|
|
Unset flushing when starting so that we reuse appsrc.
|
|
Inform basesrc about the configured size.
|
|
Emit seek-data signal when we are going to a different offset in
|
|
random-access mode.
|
|
|
|
2008-06-06 14:19:54 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
examples/app/appsrc-stream.c: Use deep-notify until we can depend on a playbin2 with support for the source property.
|
|
Original commit message from CVS:
|
|
* examples/app/appsrc-stream.c: (found_source), (main):
|
|
Use deep-notify until we can depend on a playbin2 with support for the
|
|
source property.
|
|
|
|
2008-06-05 16:38:50 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
examples/app/: Added an example on how to use appsrc in playbin in streaming mode from an mmapped file.
|
|
Original commit message from CVS:
|
|
* examples/app/.cvsignore:
|
|
* examples/app/Makefile.am:
|
|
* examples/app/appsrc-stream.c: (read_data), (start_feed),
|
|
(stop_feed), (found_source), (bus_message), (main):
|
|
Added an example on how to use appsrc in playbin in streaming mode from
|
|
an mmapped file.
|
|
* examples/app/appsrc_ex.c: (main):
|
|
Set pipeline to NULL to free queued buffers.
|
|
* gst-libs/gst/app/gstapp-marshal.list:
|
|
* gst-libs/gst/app/gstappsrc.c: (stream_type_get_type), (_do_init),
|
|
(gst_app_src_class_init), (gst_app_src_init),
|
|
(gst_app_src_flush_queued), (gst_app_src_dispose),
|
|
(gst_app_src_set_property), (gst_app_src_get_property),
|
|
(gst_app_src_unlock), (gst_app_src_unlock_stop),
|
|
(gst_app_src_start), (gst_app_src_stop), (gst_app_src_is_seekable),
|
|
(gst_app_src_check_get_range), (gst_app_src_do_seek),
|
|
(gst_app_src_create), (gst_app_src_set_stream_type),
|
|
(gst_app_src_get_stream_type), (gst_app_src_set_max_bytes),
|
|
(gst_app_src_get_max_bytes), (gst_app_src_push_buffer),
|
|
(gst_app_src_end_of_stream), (gst_app_src_uri_get_type),
|
|
(gst_app_src_uri_get_protocols), (gst_app_src_uri_get_uri),
|
|
(gst_app_src_uri_set_uri), (gst_app_src_uri_handler_init):
|
|
* gst-libs/gst/app/gstappsrc.h:
|
|
Measure max queue size in bytes instead.
|
|
Add support for 3 modes of operation, streaming, seekable and
|
|
random-access, making basesrc handle the scheduling modes for each.
|
|
Add appsrc:// uri handler so that automatic plugging can be done from
|
|
playbin2 or uridecodebin, for example.
|
|
Added support for custom segment formats.
|
|
Add support for push and pull based operations from the application.
|
|
Expand the methods so that errors can be detected.
|
|
Flush the queued buffers on seeks and when shutting down.
|
|
Add signals to inform the app that a seek must happen.
|
|
|
|
2008-06-05 09:47:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
configure.ac: 0.10.19.2 pre-release
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
0.10.19.2 pre-release
|
|
|
|
2008-06-04 21:48:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
win32/common/: Add new API functions to the dll exports
|
|
Original commit message from CVS:
|
|
* win32/common/libgstrtsp.def:
|
|
* win32/common/libgsttag.def:
|
|
Add new API functions to the dll exports
|
|
|
|
2008-06-04 17:42:38 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/playback/gstplaybasebin.c: Disconnect signals from decodebins we created before we remove it from playbin, to avo...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c:
|
|
Disconnect signals from decodebins we created before we remove it from
|
|
playbin, to avoid crashes if the decodebin is eventually disposed after
|
|
the playbin itself (possible if the app takes a reference on the
|
|
decodebin).
|
|
Fixes #536521.
|
|
|
|
2008-06-04 17:12:40 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't copy caps fo...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (aac_type_find),
|
|
(mp3_type_find), (musepack_type_find), (MULTIPART_MAX_HEADER_SIZE),
|
|
(mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
|
|
(h264_video_type_find), (mpeg_video_stream_type_find),
|
|
(dv_type_find), (mmsh_type_find):
|
|
Bunch of small clean-ups: use gst_type_find_suggest_simple(); don't
|
|
copy caps for no good reason (this may be desirable to make it easier
|
|
to detect leaks, but then it should probably be done for all caps
|
|
in the typefinder somewhere).
|
|
|
|
2008-06-04 16:06:49 +0000 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
tests/check/Makefile.am: Do not try to run the check tests for subparse unless it has been built.
|
|
Original commit message from CVS:
|
|
* tests/check/Makefile.am:
|
|
Do not try to run the check tests for subparse unless it has been
|
|
built.
|
|
|
|
2008-06-04 16:00:26 +0000 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
tests/check/pipelines/streamheader.c: Do not try to run a test which requires vorbisenc unless we have actually built...
|
|
Original commit message from CVS:
|
|
* tests/check/pipelines/streamheader.c: (buffer_probe_cb),
|
|
(test_multifdsink_gdp_vorbisenc), (streamheader_suite):
|
|
Do not try to run a test which requires vorbisenc unless we have
|
|
actually built it.
|
|
|
|
2008-06-04 11:53:53 +0000 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
gst-libs/gst/rtsp/gstrtspconnection.*: Add a couple of missing argument guards.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
(gst_rtsp_connection_set_auth), (gst_rtsp_connection_set_auth_param),
|
|
(gst_rtsp_connection_clear_auth_params),
|
|
(gst_rtsp_connection_set_qos_dscp), (gst_rtsp_connection_get_ip):
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
Add a couple of missing argument guards.
|
|
Add a way of setting the DSCP for an RTSP connection.
|
|
Add an accessor method for the ip member of GstRTSPConnection as all
|
|
members are supposed to be private.
|
|
|
|
2008-06-04 11:33:23 +0000 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
gst/tcp/gstmultifdsink.c: Fixed accidental use of IPv4 options for all IPv6 addresses.
|
|
Original commit message from CVS:
|
|
* gst/tcp/gstmultifdsink.c: (setup_dscp_client):
|
|
Fixed accidental use of IPv4 options for all IPv6 addresses.
|
|
|
|
2008-06-04 10:18:42 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/interfaces/mixertrack.h: Document mixer track flags.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/mixertrack.h:
|
|
Document mixer track flags.
|
|
|
|
2008-06-04 05:58:38 +0000 Antoine Tremblay <hexa00@gmail.com>
|
|
|
|
gst/gdp/gstgdppay.c: Don't set caps on the buffers that contain a copy of the buffer including the caps of them resul...
|
|
Original commit message from CVS:
|
|
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
|
|
* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader):
|
|
Don't set caps on the buffers that contain a copy of the buffer
|
|
including the caps of them resulting in an always increasing refcount
|
|
of the caps and insanely large caps. Instead include a buffer without
|
|
caps in the new caps. Fixes bug #536475.
|
|
|
|
2008-06-04 05:44:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/videoscale/gstvideoscale.c: Transform a given PAR to a range on the struct with the generic height/width instead ...
|
|
Original commit message from CVS:
|
|
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
|
|
Transform a given PAR to a range on the struct with the generic
|
|
height/width instead of the struct with the possibly restricted
|
|
height/width.
|
|
|
|
2008-06-04 04:24:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/videoscale/gstvideoscale.c: Prefer the given format if it contains something stricter than [1,MAX] for height or ...
|
|
Original commit message from CVS:
|
|
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
|
|
Prefer the given format if it contains something stricter than [1,MAX]
|
|
for height or width and only put a structure that requires rescaling
|
|
as second. This makes it possible to use videoscale in pipelines where
|
|
the source can actually produce the wanted height/width but usually
|
|
selects a different one from the requested.
|
|
|
|
2008-06-03 20:01:58 +0000 John Millikin <jmillikin@gmail.com>
|
|
|
|
gst-libs/gst/tag/gstvorbistag.c: Retrieve COVERART tags from vorbis comments (#512333)
|
|
Original commit message from CVS:
|
|
Based on patch by: John Millikin <jmillikin gmail com>
|
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* gst-libs/gst/tag/gstvorbistag.c: (tag_matches), (gst_vorbis_tag_add),
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(gst_vorbis_tag_add_coverart):
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Retrieve COVERART tags from vorbis comments (#512333)
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2008-06-03 19:44:48 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/tag/: Don't forget to add new enum value here too (should probably use glib-mkenums here...).
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Original commit message from CVS:
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* gst-libs/gst/tag/tag.h:
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* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
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Don't forget to add new enum value here too (should probably use
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glib-mkenums here...).
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2008-06-03 19:29:06 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/tag/: API: add gst_tag_image_data_to_image_buffer()
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Original commit message from CVS:
|
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* gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_add_id3_image):
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* gst-libs/gst/tag/tag.h: (GST_TAG_IMAGE_TYPE_NONE),
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* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
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(gst_tag_image_type_get_type), (gst_tag_image_type_is_valid),
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(gst_tag_image_data_to_image_buffer):
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Add two utility functions to avoid code duplication (#512333):
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API: add gst_tag_image_data_to_image_buffer()
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API: add gst_tag_list_add_id3_image()
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2008-06-03 08:54:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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win32/common/libgstaudio.def: Add gst_audio_check_channel_positions() to the exported symbols.
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Original commit message from CVS:
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* win32/common/libgstaudio.def:
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Add gst_audio_check_channel_positions() to the exported symbols.
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2008-06-03 08:48:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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API: Make gst_audio_check_channel_positions() public.
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Original commit message from CVS:
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* docs/libs/gst-plugins-base-libs-sections.txt:
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* gst-libs/gst/audio/multichannel.c:
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(gst_audio_check_channel_positions):
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* gst-libs/gst/audio/multichannel.h:
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API: Make gst_audio_check_channel_positions() public.
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* tests/check/libs/audio.c: (GST_START_TEST):
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Add some simple checks for gst_audio_check_channel_positions().
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2008-06-02 20:09:14 +0000 Tim-Philipp Müller <tim@centricular.net>
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sys/v4l/v4l_calls.c: minrange and maxrange are scaled according to the frequency multiplier.
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Original commit message from CVS:
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* sys/v4l/v4l_calls.c: (gst_v4l_get_chan_names):
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minrange and maxrange are scaled according to the frequency
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multiplier.
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2008-06-02 18:37:02 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/pango/: Use gstvideo functions to calculate strides and plane offsets. Fixes rendering issue ('ghost' images of t...
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|
Original commit message from CVS:
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* ext/pango/Makefile.am:
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* ext/pango/gsttextoverlay.c: (gst_text_overlay_shade_y),
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(gst_text_overlay_blit_yuv420), (gst_text_overlay_push_frame):
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Use gstvideo functions to calculate strides and plane offsets. Fixes
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rendering issue ('ghost' images of the text on the chroma planes)
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with widths or heights that are not multiples of 8 (#506659 and
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probably also #485729).
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* tests/icles/test-textoverlay.c: (show_text), (test_textoverlay),
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(main):
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Test with odd height/width too.
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2008-06-02 12:20:35 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/adder/gstadder.c: When using gst_element_iterate_pads() one has to unref every pad after usage.
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Original commit message from CVS:
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* gst/adder/gstadder.c: (gst_adder_query_duration),
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(gst_adder_query_latency):
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When using gst_element_iterate_pads() one has to unref every pad
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after usage.
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2008-05-31 19:57:57 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
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gst-libs/gst/audio/gstbaseaudiosrc.c: Add a gtk-doc chunk for the new properties to have a Since: indication.
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Original commit message from CVS:
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* gst-libs/gst/audio/gstbaseaudiosrc.c:
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(gst_base_audio_src_class_init):
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Add a gtk-doc chunk for the new properties to have a Since: indication.
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2008-05-31 19:50:59 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
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* ChangeLog:
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ChangeLog surgery, mark API change
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Original commit message from CVS:
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ChangeLog surgery, mark API change
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2008-05-31 18:10:47 +0000 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
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gst-libs/gst/audio/gstbaseaudiosrc.c: Provide readable actual-buffer-time and actual-latency-time properties that ref...
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|
Original commit message from CVS:
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* gst-libs/gst/audio/gstbaseaudiosrc.c:
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(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
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(gst_base_audio_src_get_property), (gst_base_audio_src_setcaps),
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(gst_base_audio_src_change_state):
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Provide readable actual-buffer-time and actual-latency-time properties
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that reflect the configured ringbuffer values. Fixes #524724.
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2008-05-30 15:29:20 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/rtp/gstbasertppayload.c: Simply converting the running time into an RTP timestamp by scaling it based on...
|
|
Original commit message from CVS:
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|
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push),
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(gst_basertppayload_change_state):
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Simply converting the running time into an RTP timestamp by scaling it
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based on the clock-rate is good enough for making an RTP timestamp. This
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has the added benefit that we can later on expose a property with the
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RTP timestamp of running time 0, as is needed for RTSP servers to
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generate the response of the PLAY request.
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2008-05-30 08:42:17 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/audioconvert/gstaudioconvert.c: Allow up to 11 positioned channels now that audioconvert can handle this but add ...
|
|
Original commit message from CVS:
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|
* gst/audioconvert/gstaudioconvert.c:
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(structure_has_fixed_channel_positions),
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(gst_audio_convert_transform_caps):
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Allow up to 11 positioned channels now that audioconvert can handle
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this but add no default positions for > 8 channels.
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* tests/check/elements/audioconvert.c: (GST_START_TEST):
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Add some unit tests for the above change: Test conversion of
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11 positioned channels to stereo and the other way around, test
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conversion of 15 unpositioned channels in different ways.
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2008-05-29 19:45:40 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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win32/common/libgstaudio.def: Add gst_audio_clock_reset to the list of exported symbols.
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|
Original commit message from CVS:
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|
* win32/common/libgstaudio.def:
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Add gst_audio_clock_reset to the list of exported symbols.
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|
|
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2008-05-29 19:37:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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tests/check/elements/vorbisdec.c: Remove wrong_channels_identification_header unit test as we now support 7 (and more...
|
|
Original commit message from CVS:
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* tests/check/elements/vorbisdec.c: (vorbisdec_suite):
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Remove wrong_channels_identification_header unit test as we now
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support 7 (and more channels).
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2008-05-29 12:17:16 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/audioconvert/gstchannelmix.c: If mixing left or right to center (or the other way around) only take the complete ...
|
|
Original commit message from CVS:
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|
* gst/audioconvert/gstchannelmix.c:
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(gst_channel_mix_fill_one_other):
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If mixing left or right to center (or the other way around) only take
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the complete value if we don't already have the original position in
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the source.
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2008-05-29 11:34:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst-libs/gst/audio/multichannel.c: Allow rear center together with rear left/right and other previously conflicting c...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/multichannel.c:
|
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(gst_audio_check_channel_positions),
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|
(gst_audio_set_structure_channel_positions_list),
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(gst_audio_fixate_channel_positions):
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Allow rear center together with rear left/right and other previously
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conflicting channel positions. The reason why they weren't allowed
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was the channel mixing implementation in audioconvert.
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Also take this into account when fixing channel layouts.
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Allow setting channel positions for 1/2 channels when using
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gst_audio_set_structure_channel_position().
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* gst/audioconvert/gstchannelmix.c:
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(gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
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|
(gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
|
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(gst_channel_mix_fill_special), (gst_channel_mix_fill_matrix):
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Major rewrite of the channel mixing.
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We now allow previously conflicting channel positions to appear
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together (rear center and rear left/right for example).
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Fixes bug #533817.
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Rework the way channels are mixed together to take more possible
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channel positions into account, properly mix from/to side channels
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and don't assume that either center, left&right or nothing of a
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|
specific position is available anymore.
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* tests/check/elements/audioconvert.c: (GST_START_TEST):
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Adjust unit tests with non-standard 1/2 channel layouts to the more
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correct new behaviour.
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Add a unit test for 5.1->Stereo downmixing.
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2008-05-29 07:02:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
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ext/vorbis/: Add sane defaults for the 7 and 8 channel layouts as those are undefined in the Vorbis spec. Use NONE ch...
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
|
|
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps):
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|
Add sane defaults for the 7 and 8 channel layouts as those are
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|
undefined in the Vorbis spec. Use NONE channel layouts when decoding
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|
more than 8 channels instead of erroring out. Fixes bug #535356.
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2008-05-28 16:10:20 +0000 Wim Taymans <wim.taymans@gmail.com>
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Add theoraparse to the docs and fix some docs.
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|
Original commit message from CVS:
|
|
* docs/plugins/Makefile.am:
|
|
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
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* ext/theora/theoraparse.c:
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Add theoraparse to the docs and fix some docs.
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|
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2008-05-28 15:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst-libs/gst/cdda/gstcddabasesrc.c: Fix EOS condition and track addition check, the track.end sector is included in t...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/cdda/gstcddabasesrc.c:
|
|
(gst_cdda_base_src_add_track), (gst_cdda_base_src_create):
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Fix EOS condition and track addition check, the track.end sector is
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|
included in the track. Fixes #533265.
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2008-05-28 14:49:24 +0000 Mark Nauwelaerts <manauw@skynet.be>
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gst/videorate/gstvideorate.*: React (more) to NEWSEGMENT
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|
Original commit message from CVS:
|
|
Patch by: Mark Nauwelaerts <manauw at skynet be>
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|
* gst/videorate/gstvideorate.c: (gst_video_rate_reset),
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|
(gst_video_rate_flush_prev), (gst_video_rate_event),
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|
(gst_video_rate_chain):
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* gst/videorate/gstvideorate.h:
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React (more) to NEWSEGMENT
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Small adjustment in timestamp calculation to prevent mismatches
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Fixes #435633.
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2008-05-28 11:31:44 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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|
tests/examples/seek/seek.c: Initialise error to NULL as we should.
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|
Original commit message from CVS:
|
|
* tests/examples/seek/seek.c: (make_parselaunch_pipeline):
|
|
Initialise error to NULL as we should.
|
|
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|
2008-05-28 08:14:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/adder/gstadder.c: Implement latency query.
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|
Original commit message from CVS:
|
|
* gst/adder/gstadder.c: (gst_adder_query_duration),
|
|
(gst_adder_query_latency), (gst_adder_query):
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|
Implement latency query.
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|
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2008-05-27 18:10:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
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gst/adder/gstadder.c: Correctly resync the iterator if gst_iterator_next() returns
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|
Original commit message from CVS:
|
|
* gst/adder/gstadder.c: (gst_adder_query_duration):
|
|
Correctly resync the iterator if gst_iterator_next() returns
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GST_ITERATOR_RESYNC.
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2008-05-27 17:14:07 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
win32/vs6/libgstpbutils.dsp: Add pbutils-enumtypes.c to sources (#518037).
|
|
Original commit message from CVS:
|
|
* win32/vs6/libgstpbutils.dsp:
|
|
Add pbutils-enumtypes.c to sources (#518037).
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|
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2008-05-27 16:20:17 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
|
|
(gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
|
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* gst-libs/gst/audio/gstaudioclock.h:
|
|
Add method to inform the clock that the time starts from 0 again. We use
|
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this info to calculate a clock offset so that the time we report in
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|
internal_time is monotonically increasing, as required by the clock base
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class. Fixes #521761.
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API: GstAudioClock::gst_audio_clock_reset()
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* gst-libs/gst/audio/gstbaseaudiosink.c:
|
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(gst_base_audio_sink_skew_slaving),
|
|
(gst_base_audio_sink_change_state):
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
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(gst_base_audio_src_create), (gst_base_audio_src_change_state):
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Reset reported time when we (re)create the ringbuffer.
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2008-05-27 16:11:32 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
|
ext/alsa/gstalsamixertrack.c: Make sure playback volumes aren't accidentally overwritten by capture volumes if an als...
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsamixertrack.c:
|
|
(gst_alsa_mixer_track_update_alsa_capabilities):
|
|
Make sure playback volumes aren't accidentally overwritten by
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|
capture volumes if an alsa mixer track has both playback and
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capture capabilities: we create two GstMixerTracks in that
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case, so make sure we query only the alsa capabilities that
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|
refer to the type of GstMixerTrack we created from the dual
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|
capability alsa element. Should fix issues with Audigy2 sound
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cards (#518082).
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2008-05-27 10:57:56 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
|
tests/check/pipelines/oggmux.c: Don't use deprecated function.
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|
Original commit message from CVS:
|
|
* tests/check/pipelines/oggmux.c: (test_pipeline):
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|
Don't use deprecated function.
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|
|
|
2008-05-27 10:35:55 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
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gst/playback/gstdecodebin2.c: Check for NULL cases and log them, creating ghostpads can, for example, fail when the p...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin2.c:
|
|
(gst_decode_group_control_source_pad), (gst_decode_group_expose):
|
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Check for NULL cases and log them, creating ghostpads can, for example,
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fail when the pad returns wrong caps.
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* gst/playback/gstplaybin2.c: (perform_eos):
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When pushing out the EOS event, collect the return value and warn when
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something failed.
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2008-05-26 17:18:52 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
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gst-libs/gst/riff/riff-media.c: Add support for DVCPRO.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
|
|
(gst_riff_create_video_template_caps):
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Add support for DVCPRO.
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2008-05-26 10:29:20 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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gst/videoscale/gstvideoscale.c: Change default scaling method from nearest-neighbour to bilinear.
|
|
Original commit message from CVS:
|
|
* gst/videoscale/gstvideoscale.c: (DEFAULT_PROP_METHOD):
|
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Change default scaling method from nearest-neighbour to bilinear.
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|
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2008-05-26 10:26:00 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
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|
tests/check/libs/video.c: More checks.
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|
Original commit message from CVS:
|
|
* tests/check/libs/video.c:
|
|
More checks.
|
|
|
|
2008-05-25 20:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
|
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|
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Limit duration to a maximum of five seconds for tmplayer format where we can guess the duration only from the timesta...
|
|
Original commit message from CVS:
|
|
* gst/subparse/gstsubparse.c: (parser_state_init),
|
|
(gst_sub_parse_format_autodetect), (handle_buffer):
|
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* gst/subparse/gstsubparse.h:
|
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* tests/check/elements/subparse.c: (test_tmplayer_style3b):
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Limit duration to a maximum of five seconds for tmplayer format where
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we can guess the duration only from the timestamp of the next line of
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text. We don't want to show a text for eternities just because nothing
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else is being said for a while.
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2008-05-23 14:14:28 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst-libs/gst/rtp/gstbasertpdepayload.c: Check sequence numbers, mark input buffers with a discont flag for the subcla...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_chain),
|
|
(gst_base_rtp_depayload_handle_sink_event),
|
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(gst_base_rtp_depayload_push_full),
|
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(gst_base_rtp_depayload_change_state):
|
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Check sequence numbers, mark input buffers with a discont flag for the
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subclass when we detected a gap, drop duplicate buffers. We do this
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because one can use the element without a jitterbuffer in front and we
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don't want to feed the subclasses invalid or reordered data.
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Do an error when the subclass did not provide a process function instead
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of crashing.
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Some other small cleanups.
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2008-05-22 22:35:40 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/videotestsrc/videotestsrc.c: May just as well use the precalculated uvstride here.
|
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Original commit message from CVS:
|
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* gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
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May just as well use the precalculated uvstride here.
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2008-05-22 22:09:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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Add some documentation comments, and some new headers to be scanned.
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|
Original commit message from CVS:
|
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* docs/plugins/Makefile.am:
|
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* docs/plugins/gst-plugins-base-plugins-overrides.txt:
|
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* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
|
* docs/plugins/gst-plugins-base-plugins.interfaces:
|
|
* docs/plugins/gst-plugins-base-plugins.prerequisites:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.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-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gdp.xml:
|
|
* docs/plugins/inspect/plugin-gio.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.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-queue2.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-uridecodebin.xml:
|
|
* docs/plugins/inspect/plugin-video4linux.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:
|
|
* ext/cdparanoia/gstcdparanoiasrc.c:
|
|
* ext/ogg/gstoggdemux.c:
|
|
* ext/ogg/gstoggdemux.h:
|
|
* ext/ogg/gstoggmux.c:
|
|
* ext/ogg/gstoggmux.h:
|
|
* gst/audioconvert/audioconvert.c:
|
|
* gst/audioconvert/audioconvert.h:
|
|
* gst/audioconvert/gstaudioconvert.h:
|
|
* gst/gdp/gstgdpdepay.h:
|
|
* gst/gdp/gstgdppay.h:
|
|
* gst/playback/gstdecodebin.c:
|
|
* gst/playback/gstdecodebin2.c:
|
|
* gst/playback/gstplaybin.c:
|
|
* gst/playback/gstplaybin2.c:
|
|
* gst/playback/gsturidecodebin.c:
|
|
* gst/tcp/gstmultifdsink.c:
|
|
* gst/tcp/gstmultifdsink.h:
|
|
* gst/tcp/gsttcp.h:
|
|
Add some documentation comments, and some new headers to be scanned.
|
|
Rename some internal enum declarations (audioconvert's DitherType and
|
|
NoiseShapingType, GstUnitType from the TCP elements) to match the
|
|
documented GObject type names so that the docs pick them up.
|
|
Name the playbin2 docs markups properly so they get picked up. They'll
|
|
need renaming back when/if playbin2 becomes playbin.
|
|
100% symbol coverage for the plugin docs, booya.
|
|
|
|
2008-05-22 18:30:15 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
|
|
|
|
gst/videotestsrc/videotestsrc.c: Fix generation of NV12/NV21 frames. Fixes bug #532454.
|
|
Original commit message from CVS:
|
|
Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
|
|
* gst/videotestsrc/videotestsrc.c: (paint_hline_NV12_NV21):
|
|
Fix generation of NV12/NV21 frames. Fixes bug #532454.
|
|
|
|
2008-05-22 11:59:33 +0000 Sjoerd Simons <sjoerd@luon.net>
|
|
|
|
gst/playback/gstdecodebin.c: Lock the fakesink before setting the state to NULL and removing it from the bin so that ...
|
|
Original commit message from CVS:
|
|
Patch by: Sjoerd Simons <sjoerd at luon dot net>
|
|
* gst/playback/gstdecodebin.c: (remove_fakesink):
|
|
Lock the fakesink before setting the state to NULL and removing it from
|
|
the bin so that a concurrent state change cannot interfere.
|
|
Fixes #534331.
|
|
|
|
2008-05-21 17:09:42 +0000 Felipe Contreras <felipe.contreras@nokia.com>
|
|
|
|
docs/Makefile.am: Fix installing plugin documentation when gtk-doc is disabled.
|
|
Original commit message from CVS:
|
|
* docs/Makefile.am:
|
|
Fix installing plugin documentation when gtk-doc is disabled.
|
|
|
|
2008-05-21 17:01:16 +0000 Felipe Contreras <felipe.contreras@nokia.com>
|
|
|
|
gst-libs/gst/rtsp/Makefile.am: Distribute, don't install md5.h
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtsp/Makefile.am:
|
|
Distribute, don't install md5.h
|
|
|
|
2008-05-21 16:47:58 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
gst/tcp/gstmultifdsink.c: Use IPPROTO_IP instead of SOL_IP, works on more platforms.
|
|
Original commit message from CVS:
|
|
2008-05-21 Julien Moutte <julien@fluendo.com>
|
|
* gst/tcp/gstmultifdsink.c: (setup_dscp_client): Use IPPROTO_IP
|
|
instead of SOL_IP, works on more platforms.
|
|
* gst/typefind/gsttypefindfunctions.c: (aac_type_find): Fix printf
|
|
arguments.
|
|
|
|
2008-05-21 16:44:15 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
Some debug and comment fixes.
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisdec.c:
|
|
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_show_frame):
|
|
Some debug and comment fixes.
|
|
* tests/examples/dynamic/addstream.c: (main):
|
|
Fix , to ;
|
|
|
|
2008-05-21 16:36:50 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
Don't use bad gst_element_get_pad().
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
|
|
* gst/playback/decodetest.c: (new_decoded_pad_cb):
|
|
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
|
|
(try_to_link_1), (elem_is_dynamic), (close_link), (type_found),
|
|
(cleanup_decodebin):
|
|
* gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
|
|
(connect_element), (gst_decode_group_control_demuxer_pad):
|
|
* gst/playback/gstplaybasebin.c: (queue_remove_probe),
|
|
(queue_out_of_data), (gen_preroll_element), (preroll_unlinked),
|
|
(mute_group_type):
|
|
* gst/playback/gstplaybin.c: (gst_play_bin_vis_blocked),
|
|
(gst_play_bin_set_property), (handoff), (gen_video_element),
|
|
(gen_text_element), (gen_audio_element), (gen_vis_element),
|
|
(remove_sinks), (add_sink), (setup_sinks):
|
|
* gst/playback/gstplaybin2.c: (pad_added_cb), (no_more_pads_cb):
|
|
* gst/playback/gstplaysink.c: (gst_play_sink_get_video_sink),
|
|
(gst_play_sink_get_audio_sink), (gst_play_sink_vis_unblocked),
|
|
(gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
|
|
(gst_play_sink_get_vis_plugin), (gst_play_sink_set_mute),
|
|
(gen_video_chain), (gen_text_chain), (gen_audio_chain),
|
|
(gen_vis_chain), (gst_play_sink_reconfigure),
|
|
(gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
|
|
(gst_play_sink_request_pad):
|
|
* gst/playback/gsturidecodebin.c: (type_found), (setup_source):
|
|
* gst/playback/test.c: (gen_video_element), (gen_audio_element),
|
|
(cb_newpad):
|
|
* gst/playback/test6.c: (new_decoded_pad_cb):
|
|
* tests/check/elements/audioconvert.c: (GST_START_TEST):
|
|
* tests/check/elements/audiorate.c: (test_injector_chain),
|
|
(do_perfect_stream_test):
|
|
* tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
|
|
* tests/check/elements/gdpdepay.c: (GST_START_TEST):
|
|
* tests/check/elements/gnomevfssink.c:
|
|
* tests/check/elements/textoverlay.c:
|
|
(notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2):
|
|
* tests/check/elements/videotestsrc.c: (GST_START_TEST):
|
|
* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
|
|
* tests/check/pipelines/oggmux.c: (test_pipeline):
|
|
* tests/check/pipelines/streamheader.c: (GST_START_TEST):
|
|
* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
|
|
* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
|
|
* tests/examples/seek/scrubby.c: (make_wav_pipeline):
|
|
* tests/examples/seek/seek.c: (make_mod_pipeline),
|
|
(make_dv_pipeline), (make_wav_pipeline), (make_flac_pipeline),
|
|
(make_sid_pipeline), (make_parse_pipeline), (make_vorbis_pipeline),
|
|
(make_theora_pipeline), (make_vorbis_theora_pipeline),
|
|
(make_avi_msmpeg4v3_mp3_pipeline), (make_mp3_pipeline),
|
|
(make_avi_pipeline), (make_mpeg_pipeline), (make_mpegnt_pipeline),
|
|
(update_fill), (msg_buffering):
|
|
Don't use bad gst_element_get_pad().
|
|
|
|
2008-05-21 14:35:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst-libs/gst/riff/riff-media.c: Fix wrong method name in docs. Fix calculation of strf fields for broken mulaw/alaw.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c:
|
|
Fix wrong method name in docs. Fix calculation of strf fields for
|
|
broken mulaw/alaw.
|
|
* gst-libs/gst/riff/riff-read.c:
|
|
Whitespace fix and removing double ';'.
|
|
|
|
2008-05-21 11:52:30 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
docs/design/part-playbin2.txt: Add some leftover doc.
|
|
Original commit message from CVS:
|
|
* docs/design/part-playbin2.txt:
|
|
Add some leftover doc.
|
|
|
|
2008-05-21 11:36:37 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/audioconvert/gstchannelmix.c: Fix copy & paste error in last commit.
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
|
|
Fix copy & paste error in last commit.
|
|
|
|
2008-05-21 11:30:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/audioconvert/gstchannelmix.c: Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to other channel posi...
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
|
|
Add support for mixing GST_AUDIO_CHANNEL_POSITION_SIDE_* from/to
|
|
other channel positions when source has SIDE channels and dest doesn't
|
|
or the other way around.
|
|
|
|
2008-05-21 11:29:25 +0000 Henrik Eriksson <henriken@axis.com>
|
|
|
|
gst/tcp/gstmultifdsink.*: Add support for DSCP QOS. Fixes #469933.
|
|
Original commit message from CVS:
|
|
Patch by: Henrik Eriksson <henriken at axis dot com>
|
|
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
|
|
(gst_multi_fd_sink_init), (setup_dscp_client), (setup_dscp),
|
|
(gst_multi_fd_sink_add_full), (gst_multi_fd_sink_set_property),
|
|
(gst_multi_fd_sink_get_property):
|
|
* gst/tcp/gstmultifdsink.h:
|
|
Add support for DSCP QOS. Fixes #469933.
|
|
|
|
2008-05-21 07:46:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
tests/check/elements/audioconvert.c: Add another test that checks if conversion between standard 1 and 2 channel layo...
|
|
Original commit message from CVS:
|
|
* tests/check/elements/audioconvert.c: (GST_START_TEST):
|
|
Add another test that checks if conversion between standard 1 and 2
|
|
channel layouts with and without positions set is working.
|
|
|
|
2008-05-21 07:39:56 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst-libs/gst/audio/multichannel.c: Allow non-standard 2 channel layouts.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/multichannel.c:
|
|
(gst_audio_check_channel_positions):
|
|
Allow non-standard 2 channel layouts.
|
|
* tests/check/elements/audioconvert.c: (GST_START_TEST):
|
|
Add some tests for converting and remapping non-standard 1 and 2
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|
channel layouts.
|
|
|
|
2008-05-21 07:28:04 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/audioconvert/gstchannelmix.c: Prevent division by zero if the channel mix matrix contains only zeroes.
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/gstchannelmix.c:
|
|
(gst_channel_mix_fill_normalize):
|
|
Prevent division by zero if the channel mix matrix contains only
|
|
zeroes.
|
|
|
|
2008-05-21 06:45:22 +0000 Antoine Tremblay <hexa00@gmail.com>
|
|
|
|
gst/gdp/gstgdppay.c: Close a buffer memory leak. Fixes bug #534071.
|
|
Original commit message from CVS:
|
|
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
|
|
* gst/gdp/gstgdppay.c: (gst_gdp_pay_chain):
|
|
Close a buffer memory leak. Fixes bug #534071.
|
|
|
|
2008-05-21 06:39:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst-libs/gst/rtsp/gstrtsptransport.h: Make the GstRTSPTransport struct members public as there are no setters/getters...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtsp/gstrtsptransport.h:
|
|
Make the GstRTSPTransport struct members public as there are no
|
|
setters/getters and it's supposed to be changed directly.
|
|
Fixes bug #533087.
|
|
|
|
2008-05-21 05:48:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/adder/gstadder.c: Adder also doesn't support audio/x-raw-int with width!=depth so don't claim this on the pad tem...
|
|
Original commit message from CVS:
|
|
* gst/adder/gstadder.c:
|
|
Adder also doesn't support audio/x-raw-int with width!=depth so don't
|
|
claim this on the pad template caps.
|
|
|
|
2008-05-20 16:26:53 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.c: We can only use our optimal calibration if we prerolled before the latency exp...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_sync_latency):
|
|
We can only use our optimal calibration if we prerolled before the
|
|
latency expired.
|
|
|
|
2008-05-20 14:35:42 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
configure.ac: Require core CVS for GstBaseSrc buffer caps setting magic.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Require core CVS for GstBaseSrc buffer caps setting magic.
|
|
|
|
2008-05-20 12:26:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/audioconvert/gstaudioconvert.c: Fix logic in last commit.
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/gstaudioconvert.c:
|
|
(gst_audio_convert_fixate_channels):
|
|
Fix logic in last commit.
|
|
|
|
2008-05-20 12:15:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/audioconvert/gstaudioconvert.c: Passthrough the channel positions if the number of output channels is the same as...
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/gstaudioconvert.c:
|
|
(gst_audio_convert_fixate_channels):
|
|
Passthrough the channel positions if the number of output channels is
|
|
the same as the number of input channels, the input had a channel
|
|
layout and downstream requests no special one. We did this already for
|
|
> 2 channels but now it's also done for 1 channel. Fixes bug #533617.
|
|
|
|
2008-05-20 11:13:27 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/gnomevfs/gstgnomevfssrc.*: Set the ICY caps on the srcpad from where they get picked up by the base class now and...
|
|
Original commit message from CVS:
|
|
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
|
|
(gst_gnome_vfs_src_finalize),
|
|
(gst_gnome_vfs_src_received_headers_callback),
|
|
(gst_gnome_vfs_src_create), (gst_gnome_vfs_src_stop):
|
|
* ext/gnomevfs/gstgnomevfssrc.h:
|
|
Set the ICY caps on the srcpad from where they get picked up by the base
|
|
class now and set on the outgoing buffers.
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
(gst_base_audio_src_create):
|
|
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
|
|
BaseSrc now sets the caps on outgoing buffers automatically.
|
|
|
|
2008-05-20 11:09:06 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.c: Change the way in which the ringbuffer is started when dealing with a slaved c...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_resample_slaving),
|
|
(gst_base_audio_sink_skew_slaving),
|
|
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
|
|
(gst_base_audio_sink_async_play),
|
|
(gst_base_audio_sink_change_state):
|
|
Change the way in which the ringbuffer is started when dealing with a
|
|
slaved clock and latency. We now sync to the clock until we reach
|
|
upstream latency before starting the ringbuffer. This has the effect
|
|
that we can accurately align the master and slave clocks and let the
|
|
rate correction code take care of the initial drift or rounding errors
|
|
instead of leaving them uncorrected with the old approach.
|
|
|
|
2008-05-20 08:12:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/audioconvert/gstaudioconvert.c: Correctly set the default channel positions when converting to 8 channels.
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/gstaudioconvert.c:
|
|
(gst_audio_convert_fixate_channels):
|
|
Correctly set the default channel positions when converting to 8
|
|
channels.
|
|
|
|
2008-05-19 16:13:25 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
configure.ac: Error out if we don't have the required version of core.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Error out if we don't have the required version of core.
|
|
|
|
2008-05-19 15:59:40 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Use data scan helper in aac typefinder and stop scanning for headers when we've ...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (aac_type_find):
|
|
Use data scan helper in aac typefinder and stop scanning
|
|
for headers when we've found a type. Also fix potential invalid
|
|
memory access when calculating the frame length.
|
|
|
|
2008-05-19 14:09:08 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/typefind/gsttypefindfunctions.c: Don't modify scan context when we return FALSE in ensure_data, so it's possible ...
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|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (data_scan_ctx_ensure_data),
|
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(mpeg_sys_is_valid_pack):
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Don't modify scan context when we return FALSE in ensure_data, so
|
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it's possible to continue scanning, and we don't end up with a NULL
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data pointer and a positive size, which might bite us the next time
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we're called. Small constification.
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2008-05-16 21:12:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/adder/gstadder.c: Adder doesn't support 24 bit samples so don't claim it supports them in the pad template caps.
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Original commit message from CVS:
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* gst/adder/gstadder.c:
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Adder doesn't support 24 bit samples so don't claim it supports them
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in the pad template caps.
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2008-05-14 20:28:02 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/rtp/gstbasertpdepayload.c: Validate the RTP packet before further processing it. It's just too dangerous...
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|
Original commit message from CVS:
|
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* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
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(gst_base_rtp_depayload_chain):
|
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Validate the RTP packet before further processing it. It's just too
|
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dangerous to accept random packets and people are not forced to use a
|
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jitterbuffer or session manager to filter out the bad packets.
|
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* gst-libs/gst/rtp/gstrtpbuffer.c:
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(gst_rtp_buffer_set_extension_data),
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(gst_rtp_buffer_get_payload_subbuffer):
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Small cleanups.
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When setting extension data in a buffer that is too small, we fail and
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we should not set the extension bit.
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Change GST_WARNINGS into g_warning because they really are
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programming errors.
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* tests/check/libs/rtp.c: (GST_START_TEST):
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Catch the g_warnings now in the unit tests and that fact that failing to
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set extension data left the extension bit untouched.
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2008-05-14 13:57:41 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/audioresample/gstaudioresample.c: Revert previous change which made basetransform handle buffer_alloc and which b...
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|
Original commit message from CVS:
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|
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
|
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Revert previous change which made basetransform handle buffer_alloc
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and which breaks things badly in the non-passthrough case since it
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returned buffers with a different (ie. sometimes smaller) size than
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the size requested.
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2008-05-14 13:43:12 +0000 Bernard B <b-gnome@largestprime.net>
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gst-libs/gst/rtp/gstrtpbuffer.c: Fix seqnum compare function for bordercase values and fix the docs again. Fixes #533...
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|
Original commit message from CVS:
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Patch by: Bernard B <b-gnome at largestprime dot net>
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* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum):
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Fix seqnum compare function for bordercase values and fix the docs
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again. Fixes #533075.
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* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
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Add a testcase for seqnum compare function.
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2008-05-14 10:58:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/adder/gstadder.c: Correctly declare the supported endianness on the pad templates and check for correct endiannes...
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|
Original commit message from CVS:
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* gst/adder/gstadder.c: (gst_adder_setcaps),
|
|
(gst_adder_class_init):
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Correctly declare the supported endianness on the pad templates
|
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and check for correct endianness in the set caps function. Adder
|
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only supports native endianness.
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Also use gst_element_class_set_details_simple().
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2008-05-14 09:12:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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sys/xvimage/xvimagesink.c: Better debug logging in port value handling. Merging separate port value loops into one.
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|
Original commit message from CVS:
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* sys/xvimage/xvimagesink.c:
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Better debug logging in port value handling. Merging separate port
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value loops into one.
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2008-05-13 16:02:19 +0000 Hannes Bistry <hannesb@gmx.de>
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gst/tcp/: Fix regression in clientsrc because we did not add the fd to the poll set anymore. Fixes #532364.
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Original commit message from CVS:
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|
Patch by: Hannes Bistry <hannesb at gmx dot de>
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* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
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* gst/tcp/gsttcpserversink.c:
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(gst_tcp_server_sink_handle_server_read),
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(gst_tcp_server_sink_handle_wait), (gst_tcp_server_sink_init_send):
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Fix regression in clientsrc because we did not add the fd to the poll
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set anymore. Fixes #532364.
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Do some cleanups here and there.
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2008-05-13 13:04:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/playback/: Use correct marshallers. GstCaps are a boxed type and no GObject subclass.
|
|
Original commit message from CVS:
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|
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
|
|
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
|
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* gst/playback/gstplay-marshal.list:
|
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* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
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Use correct marshallers. GstCaps are a boxed type and no GObject
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subclass.
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|
|
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2008-05-13 11:37:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
|
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win32/common/libgstrtsp.def: Add gst_rtsp_connection_(set|clear)_auth_param() to the exported symbols.
|
|
Original commit message from CVS:
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|
* win32/common/libgstrtsp.def:
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Add gst_rtsp_connection_(set|clear)_auth_param() to the exported
|
|
symbols.
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|
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2008-05-13 10:59:49 +0000 Sjoerd Simons <sjoerd@luon.net>
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tests/check/elements/audioresample.c: Add unit test for the latest basetransform negotiation changes.
|
|
Original commit message from CVS:
|
|
Patch by: Sjoerd Simons <sjoerd at luon dot net>
|
|
* tests/check/elements/audioresample.c:
|
|
(live_switch_alloc_only_48000), (live_switch_get_sink_caps),
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|
(live_switch_push), (GST_START_TEST):
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Add unit test for the latest basetransform negotiation changes.
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See bug #526768.
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|
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|
2008-05-13 09:14:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
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gst/ffmpegcolorspace/imgconvert.c: Fix nv12<->nv21 conversion if stride is larger than width.
|
|
Original commit message from CVS:
|
|
* gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
|
|
Fix nv12<->nv21 conversion if stride is larger than width.
|
|
|
|
2008-05-13 07:28:21 +0000 j^ <j@oil21.org>
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|
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ext/ogg/gstoggdemux.*: Parse presentation time from skeleton streams and use it as offset for the timestamps. Fixes b...
|
|
Original commit message from CVS:
|
|
Patch by: j^ <j at oil21 dot org>
|
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* ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
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(gst_ogg_pad_parse_skeleton_fisbone):
|
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* ext/ogg/gstoggdemux.h:
|
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Parse presentation time from skeleton streams and use it as offset
|
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for the timestamps. Fixes bug #530068.
|
|
|
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2008-05-12 08:45:11 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.c: Revert previous patch that attempted to more accurately calculate the initial ...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
|
|
Revert previous patch that attempted to more accurately calculate the
|
|
initial offset between master and slave clock. The best thing we can do
|
|
in general is take the time of both clocks as the diff since we don't
|
|
know when the actual preroll happened.
|
|
|
|
2008-05-11 19:52:59 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/pbutils/install-plugins.c: Fix docs: type and missing word.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/pbutils/install-plugins.c:
|
|
Fix docs: type and missing word.
|
|
|
|
2008-05-10 20:16:21 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Don't do lots of 4-byte peeks, but use the 'new' data scan helper for this inste...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
|
|
Don't do lots of 4-byte peeks, but use the 'new' data scan helper
|
|
for this instead; don't check if we've found enough markers after
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each and every step, it's enough to do that only if we've actually
|
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found a new marker.
|
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Embed a G_UNLIKELY into the IS_MPEG_HEADER macro.
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|
|
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2008-05-10 18:19:17 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Move scan helper thingy to the beginning of the file so we can use it in other t...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
(DATA_SCAN_CTX_CHUNK_SIZE), (DataScanCtx), (data_scan_ctx_advance),
|
|
(data_scan_ctx_ensure_data), (GST_MPEGVID_TYPEFIND_TRY_SYNC),
|
|
(mpeg_video_stream_type_find):
|
|
Move scan helper thingy to the beginning of the file so we can use
|
|
it in other typefind functions. Rename it to something more
|
|
generic. Also improve handling of things towards the end of the
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typefind data: peek as much as we can if we know the size of the
|
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data, rather than just min_size.
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|
|
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2008-05-09 21:42:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
Document the GstTuner and GstColorBalance interfaces, and some other random API functions that needed it. 70% symbol ...
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/interfaces/colorbalance.c:
|
|
* gst-libs/gst/interfaces/colorbalance.h:
|
|
* gst-libs/gst/interfaces/colorbalancechannel.c:
|
|
* gst-libs/gst/interfaces/colorbalancechannel.h:
|
|
* gst-libs/gst/interfaces/tuner.c:
|
|
* gst-libs/gst/interfaces/tunerchannel.c:
|
|
* gst-libs/gst/interfaces/tunerchannel.h:
|
|
* gst-libs/gst/interfaces/tunernorm.c:
|
|
* gst-libs/gst/interfaces/tunernorm.h:
|
|
* gst-libs/gst/video/video.c:
|
|
* gst-libs/gst/video/video.h:
|
|
Document the GstTuner and GstColorBalance interfaces, and some
|
|
other random API functions that needed it. 70% symbol coverage, woo.
|
|
|
|
2008-05-09 16:38:10 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstaudiosink.c: Choose to allocate one less segment but require one additional segment as latency.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
|
|
Choose to allocate one less segment but require one additional segment
|
|
as latency.
|
|
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_acquire):
|
|
No need to increment the number of segments in the source.
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_get_time), (clock_convert_external),
|
|
(gst_base_audio_sink_resample_slaving),
|
|
(gst_base_audio_sink_skew_slaving),
|
|
(gst_base_audio_sink_none_slaving), (gst_base_audio_sink_render),
|
|
(gst_base_audio_sink_async_play):
|
|
Remove adding latency when returning the internal time while subtracting
|
|
it again when we use the value a little later.
|
|
When calculating the end timestamp, we are making a rounding error
|
|
with the current algorithm. Ensure that we don't accumulate these
|
|
rounding errors when aligning samples by not resampling at all if we
|
|
don't need to. Fixes #419351.
|
|
Make the initial calibration of the clock slaving a little more
|
|
predictable and accurate. Also handle the case where we don't do
|
|
clock slaving.
|
|
|
|
2008-05-09 08:34:52 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/ffmpegcolorspace/: Add conversions from/to NV12 and NV21 and conversions between those two formats. Fixes bug #53...
|
|
Original commit message from CVS:
|
|
Based on a patch by:
|
|
Björn Benderius <bjoern dot benderius at axis dot com>
|
|
* gst/ffmpegcolorspace/avcodec.h:
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
|
|
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
|
|
(gst_ffmpegcsp_avpicture_fill):
|
|
* gst/ffmpegcolorspace/imgconvert.c: (nv12_to_nv21):
|
|
* gst/ffmpegcolorspace/imgconvert_template.h:
|
|
Add conversions from/to NV12 and NV21 and conversions between those
|
|
two formats. Fixes bug #532166.
|
|
|
|
2008-05-08 17:35:44 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Abort the h264 typefinding as soon as _peek() doesn't return anything, which hap...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find):
|
|
Abort the h264 typefinding as soon as _peek() doesn't return anything,
|
|
which happens for example with files smaller than 128kb.
|
|
|
|
2008-05-08 14:46:27 +0000 Wouter Cloetens <zombie@e2big.org>
|
|
|
|
gst-libs/gst/rtsp/: Add Digest authorization support for RTSP connections. See #532065.
|
|
Original commit message from CVS:
|
|
Patch by: Wouter Cloetens <zombie at e2big dot org>
|
|
* gst-libs/gst/rtsp/Makefile.am:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
(gst_rtsp_connection_create), (md5_digest_to_hex_string),
|
|
(auth_digest_compute_hex_urp), (auth_digest_compute_response),
|
|
(add_auth_header), (gst_rtsp_connection_free),
|
|
(gst_rtsp_connection_set_auth), (str_case_hash), (str_case_equal),
|
|
(gst_rtsp_connection_set_auth_param),
|
|
(gst_rtsp_connection_clear_auth_params):
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
Add Digest authorization support for RTSP connections. See #532065.
|
|
* gst-libs/gst/rtsp/md5.c:
|
|
* gst-libs/gst/rtsp/md5.h:
|
|
Yeap, another md5 implementation until we can depend on a glib that has
|
|
support for it.
|
|
|
|
2008-05-08 06:20:42 +0000 Sjoerd Simons <sjoerd@luon.net>
|
|
|
|
gst/audioresample/gstaudioresample.c: Let audioresample use the buffer allocation of basetransform instead of it's ow...
|
|
Original commit message from CVS:
|
|
Patch by: Sjoerd Simons <sjoerd at luon dot net>
|
|
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
|
|
Let audioresample use the buffer allocation of basetransform instead
|
|
of it's own stuff.
|
|
* tests/check/elements/audioresample.c: (alloc_only_48000),
|
|
(GST_START_TEST), (audioresample_suite):
|
|
Add unit test for the recent basetransform bugfix, where upstream
|
|
changes caps to something that can't be passed through anymore.
|
|
|
|
2008-05-07 19:50:27 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
|
|
|
|
win32/common/config.h.in: Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather use the real thing than h...
|
|
Original commit message from CVS:
|
|
* win32/common/config.h.in:
|
|
Don't define GST_FUNCTION, if GLib supports MSVC we'd much rather
|
|
use the real thing than having "???" unconditionally.
|
|
|
|
2008-05-07 15:47:03 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.c: Report the latency with the new seglatency parameter.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_query):
|
|
Report the latency with the new seglatency parameter.
|
|
* gst-libs/gst/audio/gstringbuffer.c:
|
|
(gst_ring_buffer_debug_spec_buff), (gst_ring_buffer_parse_caps),
|
|
(gst_ring_buffer_acquire):
|
|
* gst-libs/gst/audio/gstringbuffer.h:
|
|
Add new field to the ringbufferspec to specify the expected latency
|
|
between the underlying device read/write pointer, this is needed
|
|
when writing sinks that sit a little closer to the hardware.
|
|
Add some more docs for other fields.
|
|
|
|
2008-05-07 10:38:23 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/app/: Add marshal.list, make it compile and add to cvsignore.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/app/.cvsignore:
|
|
* gst-libs/gst/app/Makefile.am:
|
|
* gst-libs/gst/app/gstapp-marshal.list:
|
|
Add marshal.list, make it compile and add to cvsignore.
|
|
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose),
|
|
(gst_app_sink_stop):
|
|
Small cleanups.
|
|
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init),
|
|
(gst_app_src_init), (gst_app_src_set_property),
|
|
(gst_app_src_get_property), (gst_app_src_unlock),
|
|
(gst_app_src_unlock_stop), (gst_app_src_start), (gst_app_src_stop),
|
|
(gst_app_src_create), (gst_app_src_set_caps),
|
|
(gst_app_src_get_caps), (gst_app_src_set_size),
|
|
(gst_app_src_get_size), (gst_app_src_set_seekable),
|
|
(gst_app_src_get_seekable), (gst_app_src_set_max_buffers),
|
|
(gst_app_src_get_max_buffers), (gst_app_src_push_buffer),
|
|
(gst_app_src_end_of_stream):
|
|
* gst-libs/gst/app/gstappsrc.h:
|
|
Beat appsrc in shape, add signals and actions.
|
|
Add some docs.
|
|
Add properties for caps, size, seekability and max-buffers.
|
|
Fix unlock/stop code.
|
|
|
|
2008-05-06 12:35:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/volume/gstvolume.c: Return NOT_NEGOTIATED if we didn't set a process function yet for some reason instead of cras...
|
|
Original commit message from CVS:
|
|
* gst/volume/gstvolume.c: (volume_transform_ip):
|
|
Return NOT_NEGOTIATED if we didn't set a process function yet for some
|
|
reason instead of crashing later. Might fix bug #509125.
|
|
|
|
2008-05-06 12:12:16 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
|
|
|
|
gst/audioconvert/: Add support for more than 8 channels and NONE channel layouts. For more than 8 channels no channel...
|
|
Original commit message from CVS:
|
|
Based on a patch by: Tim-Philipp Müller <tim.muller at collabora co uk>
|
|
* gst/audioconvert/audioconvert.c: (audio_convert_prepare_context):
|
|
* gst/audioconvert/audioconvert.h:
|
|
* gst/audioconvert/gstaudioconvert.c:
|
|
(gst_audio_convert_parse_caps),
|
|
(structure_has_fixed_channel_positions),
|
|
(gst_audio_convert_transform_caps):
|
|
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_matrix):
|
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Add support for more than 8 channels and NONE channel layouts. For
|
|
more than 8 channels no channel conversion is supported yet, only
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|
format conversions are supported. Fixes bug #398033.
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* tests/check/elements/audioconvert.c: (verify_convert),
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(GST_START_TEST), (audioconvert_suite):
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Add some unit tests by Tim for checking the NONE channel layouts
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and more than 8 channels and add some more unit tests for channel
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conversions.
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2008-05-06 10:16:49 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstdecodebin2.c: When autoplugging fails, set the element back to NULL before unreffing it.
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|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin2.c: (connect_pad):
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|
When autoplugging fails, set the element back to NULL before
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|
unreffing it.
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2008-05-06 09:59:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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win32/common/libgstaudio.def: Add gst_base_audio_src_[sg]et_slave_method() to the exported symbols.
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|
Original commit message from CVS:
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|
* win32/common/libgstaudio.def:
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Add gst_base_audio_src_[sg]et_slave_method() to the exported
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symbols.
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2008-05-05 12:33:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/subparse/samiparse.c: Remove trailing, leading and double whitespaces.
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|
Original commit message from CVS:
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* gst/subparse/samiparse.c: (handle_start_sync),
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(end_sami_element), (characters_sami):
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Remove trailing, leading and double whitespaces.
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Correctly timestamp buffers and output the last buffer too.
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* tests/check/elements/subparse.c: (GST_START_TEST),
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(subparse_suite):
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Add a simple unit test for SAMI parsing.
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2008-05-05 11:14:48 +0000 Young-Ho Cha <ganadist@chollian.net>
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gst/subparse/samiparse.c: Only output characters inside the "sync" elements. There could be other elements like "styl...
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Original commit message from CVS:
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Patch by: Young-Ho Cha <ganadist at chollian dot net>
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* gst/subparse/samiparse.c: (handle_start_sync),
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(start_sami_element), (end_sami_element), (characters_sami),
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(sami_context_reset):
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Only output characters inside the "sync" elements. There could be
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other elements like "style" that have some content but should
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not be printed. Fixes bug #467911.
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2008-05-05 10:27:45 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/app/gstappsink.*: Start some docs.
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Original commit message from CVS:
|
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* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
|
|
(gst_app_sink_init), (gst_app_sink_set_property),
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(gst_app_sink_get_property), (gst_app_sink_unlock_start),
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(gst_app_sink_unlock_stop), (gst_app_sink_flush_unlocked),
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(gst_app_sink_start), (gst_app_sink_stop), (gst_app_sink_event),
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(gst_app_sink_preroll), (gst_app_sink_render),
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(gst_app_sink_set_caps), (gst_app_sink_set_drop),
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(gst_app_sink_get_drop):
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* gst-libs/gst/app/gstappsink.h:
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Start some docs.
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Add property to drop buffers when the queue is filled
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Fix unlocking and flushing when the queues are filled.
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2008-05-05 10:03:51 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/playback/: Allow setting -1 as current-audio to mute the current audio stream, similar to what is done for subtit...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (set_audio_mute),
|
|
(set_active_source):
|
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* gst/playback/gstplaybasebin.h:
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* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
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(playbin_set_audio_mute):
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Allow setting -1 as current-audio to mute the current audio stream,
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similar to what is done for subtitles. Fixes bug #342294.
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2008-05-05 07:41:03 +0000 Edward Hervey <bilboed@bilboed.com>
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gst-libs/gst/pbutils/descriptions.c: It's SorensOn and not SorensEn.
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|
Original commit message from CVS:
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* gst-libs/gst/pbutils/descriptions.c: (formats):
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It's SorensOn and not SorensEn.
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2008-05-04 15:23:36 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/pbutils/descriptions.c: Fix description of video/x-flash-video.
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|
Original commit message from CVS:
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|
* gst-libs/gst/pbutils/descriptions.c: (formats):
|
|
Fix description of video/x-flash-video.
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|
|
|
2008-05-04 15:02:20 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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Remove some unused code.
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|
Original commit message from CVS:
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|
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
|
|
* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
|
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* gst/tcp/gsttcp.c: (gst_tcp_socket_write):
|
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* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_get_fps_list):
|
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Remove some unused code.
|
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* gst/audioconvert/gstaudioquantize.c:
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(gst_audio_quantize_free_noise_shaping):
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Don't return before freeing the noise shaping history.
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2008-05-03 16:00:04 +0000 Tim-Philipp Müller <tim@centricular.net>
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tests/check/elements/subparse.c: Add unit test for the tmplayer variant from bug #530962.
|
|
Original commit message from CVS:
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|
* tests/check/elements/subparse.c: (do_test),
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|
(test_tmplayer_style3b), (subparse_suite):
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Add unit test for the tmplayer variant from bug #530962.
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2008-05-03 15:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/subparse/: Fix parsing of tmplayer subtitle variant where every single line contains text and there isn't an empt...
|
|
Original commit message from CVS:
|
|
* gst/subparse/gstsubparse.c: (handle_buffer),
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|
(gst_sub_parse_sink_event):
|
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* gst/subparse/tmplayerparse.c: (tmplayer_process_buffer),
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(tmplayer_parse_line):
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Fix parsing of tmplayer subtitle variant where every single line contains
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text and there isn't an empty line after each line to determine the
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duration (#530962). Improve EOS handling for tmplayer subtitles a bit by
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making sure that we push out the last line of text without a duration if
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there's still text left in the buffer at the end.
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2008-05-03 15:39:04 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/subparse/gstsubparse.c: Fix detection of discontinuities based on the buffer offset (doesn't work so well if no b...
|
|
Original commit message from CVS:
|
|
* gst/subparse/gstsubparse.c: (feed_textbuf):
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|
Fix detection of discontinuities based on the buffer offset (doesn't work
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so well if no buffer offset is set) and also check for the DISCONT buffer
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flag. This keeps the parser state from being reset after each buffer in
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the unit test.
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2008-05-03 12:09:16 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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gst/typefind/gsttypefindfunctions.c: Further fine-tuning: don't absolutely require sequence or GOP headers but adjust...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (mpeg_video_stream_type_find):
|
|
Further fine-tuning: don't absolutely require sequence or GOP headers
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(as introduced in the previous commit), but adjust the typefind
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probabilities returned accordingly if we don't see them. Also make sure
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picture header and first slice are somewhat close to each other (which
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is not perfect but still better than requiring a fixed offset or having
|
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no limit at all).
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2008-05-02 12:13:08 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
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gst-libs/gst/rtp/gstbasertppayload.c: Rename the setcaps/getcaps function internally to make it clear that they are c...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
|
|
(gst_basertppayload_sink_setcaps),
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(gst_basertppayload_sink_getcaps):
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Rename the setcaps/getcaps function internally to make it clear that
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they are called for the sink pad.
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|
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2008-05-02 12:11:07 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst-libs/gst/rtp/gstbasertpdepayload.*: Catch packet-lost events from the jitterbuffer and convert them into a vmetho...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_class_init),
|
|
(gst_base_rtp_depayload_handle_sink_event), (create_segment_event),
|
|
(gst_base_rtp_depayload_packet_lost),
|
|
(gst_base_rtp_depayload_set_gst_timestamp):
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|
* gst-libs/gst/rtp/gstbasertpdepayload.h:
|
|
Catch packet-lost events from the jitterbuffer and convert them into a
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|
vmethod call (lost-packet) so that depayloaders can do something smart.
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Also add a default packet-lost function that sends out a segment update
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|
to the decoders.
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|
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2008-05-02 11:13:05 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/playback/: Also include config.h when relying on defines from it. Fixes the build. Its been a please to serve :)
|
|
Original commit message from CVS:
|
|
* gst/playback/test4.c:
|
|
* gst/playback/test5.c:
|
|
* gst/playback/test6.c:
|
|
* gst/playback/test7.c:
|
|
Also include config.h when relying on defines from it. Fixes the
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|
build. Its been a please to serve :)
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|
|
|
2008-05-02 10:54:51 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
|
|
|
|
* ChangeLog:
|
|
* gst/videotestsrc/videotestsrc.c:
|
|
Add support for NV12 and NV21 in videotestsrc
|
|
Original commit message from CVS:
|
|
* gst/videotestsrc/videotestsrc.c (paint_setup_NV12),
|
|
(paint_setup_NV21), (paint_hline_NV12_NV21):
|
|
Add support for NV12 and NV21 in videotestsrc
|
|
|
|
2008-05-02 10:02:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/videoscale/: Support 1x1 images as input and output as for example the BBC HQ new streams have 1x1 GIFs in the pl...
|
|
Original commit message from CVS:
|
|
* gst/videoscale/gstvideoscale.c:
|
|
* gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
|
|
* gst/videoscale/vs_image.c: (vs_image_scale_nearest_RGBA),
|
|
(vs_image_scale_linear_RGBA), (vs_image_scale_nearest_RGB),
|
|
(vs_image_scale_linear_RGB), (vs_image_scale_nearest_YUYV),
|
|
(vs_image_scale_linear_YUYV), (vs_image_scale_nearest_UYVY),
|
|
(vs_image_scale_linear_UYVY), (vs_image_scale_nearest_Y),
|
|
(vs_image_scale_linear_Y), (vs_image_scale_nearest_RGB565),
|
|
(vs_image_scale_linear_RGB565), (vs_image_scale_nearest_RGB555),
|
|
(vs_image_scale_linear_RGB555):
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|
Support 1x1 images as input and output as for example the BBC HQ new
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streams have 1x1 GIFs in the playlists for some reason.
|
|
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2008-05-01 19:11:42 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/playback/gstdecodebin.c: If we can't activate one of the decoders we plugged in (such as, say, musepackdec) for s...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (free_pad_probe_for_element),
|
|
(try_to_link_1):
|
|
If we can't activate one of the decoders we plugged in (such as,
|
|
say, musepackdec) for some reason (it might not support push mode,
|
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for example), remove any pad probes that close_pad_link() might
|
|
have set up. This makes sure we later don't try to remove a probe
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|
for a pad that doesn't exist any longer, and avoids nast warnings
|
|
and probably other things too.
|
|
|
|
2008-04-30 20:54:56 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Rework mpeg video stream typefinding a bit more: make sure sequence,
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
(mpeg_video_stream_ctx_ensure_data), (mpeg_video_stream_type_find),
|
|
(plugin_init):
|
|
Rework mpeg video stream typefinding a bit more: make sure sequence,
|
|
GOP, picture and slice headers appear in the order they should and
|
|
that we've in fact at least had one of each; fix picture header
|
|
detection; decouple picture and slice header check - don't assume
|
|
they're at a fixed offset, there may be extra data in between. Also,
|
|
announce varying degrees of probability depending on what we found
|
|
exactly (multiple pictures, at least one picture, just sequence and
|
|
GOP headers). Finally, in _ensure_data(), take into account that we
|
|
might be typefinding smaller amounts of data, such as the first
|
|
buffer of a stream, so fall back to the minimum size needed as long
|
|
as that's available, instead of erroring out if there's less than
|
|
2kB of data. Fixes #526173. Conveniently also doesn't recognise the
|
|
fuzzed file from #399342 as valid.
|
|
|
|
2008-04-30 17:06:45 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
ext/theora/theoradec.c: Cool kids don't divide by zero.
|
|
Original commit message from CVS:
|
|
* ext/theora/theoradec.c:
|
|
Cool kids don't divide by zero.
|
|
Treat PAR of x:0 as 1:1.
|
|
Fixes #530719.
|
|
|
|
2008-04-30 14:37:52 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Refactor a bit: use context structure to track parsing offset and size of availa...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (MpegVideoStreamCtx),
|
|
(mpeg_video_stream_ctx_advance), (mpeg_video_stream_ctx_ensure_data),
|
|
(mpeg_video_stream_type_find):
|
|
Refactor a bit: use context structure to track parsing offset and size of
|
|
available data and make the code a bit clearer. Fixes bad memory access
|
|
in #356937.
|
|
|
|
2008-04-28 22:18:49 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/: Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro is defined.
|
|
Original commit message from CVS:
|
|
* gst/playback/test4.c:
|
|
* gst/playback/test5.c:
|
|
* gst/playback/test6.c:
|
|
* gst/tcp/gstmultifdsink.c:
|
|
Include stdlib.h and unistd.h only if the appropriate HAVE_*_H macro
|
|
is defined.
|
|
|
|
2008-04-28 08:51:38 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.h: Clarify some docs.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.h:
|
|
Clarify some docs.
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c: (slave_method_get_type),
|
|
(gst_base_audio_src_class_init), (gst_base_audio_src_init),
|
|
(gst_base_audio_src_set_slave_method),
|
|
(gst_base_audio_src_get_slave_method),
|
|
(gst_base_audio_src_set_property),
|
|
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.h:
|
|
Add property and methods for selecting the clock slave method in the
|
|
source, like in the sink.
|
|
We only implement "none" and "re-timestamp" for now.
|
|
API: gst_base_audio_src_set_slave_method()
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API: gst_base_audio_src_get_slave_method()
|
|
|
|
2008-04-25 18:18:47 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/app/gstappsink.*: Add more docs.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init),
|
|
(gst_app_sink_init), (gst_app_sink_set_property),
|
|
(gst_app_sink_get_property), (gst_app_sink_event),
|
|
(gst_app_sink_preroll), (gst_app_sink_render),
|
|
(gst_app_sink_set_emit_signals), (gst_app_sink_get_emit_signals),
|
|
(gst_app_sink_set_max_buffers), (gst_app_sink_get_max_buffers),
|
|
(gst_app_sink_pull_buffer):
|
|
* gst-libs/gst/app/gstappsink.h:
|
|
Add more docs.
|
|
Add signals for when preroll and render buffers are available.
|
|
Add property to control signal emission.
|
|
Add property to control the max queue size.
|
|
|
|
2008-04-25 07:37:09 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstrtpbuffer.c: Fix the docs about the seqnum compare function, it returns a difference.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstrtpbuffer.c:
|
|
Fix the docs about the seqnum compare function, it returns a difference.
|
|
|
|
2008-04-24 09:27:35 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
ext/alsa/gstalsadeviceprobe.c: Don't return before freeing up the allocated structures.
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsadeviceprobe.c:
|
|
(gst_alsa_get_device_list): Don't return before freeing up
|
|
the allocated structures.
|
|
|
|
2008-04-24 08:19:35 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/playback/gstplaybin.c: Remove obsolete streaminfo code and fix a leak. Fixes #529546
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin.c:
|
|
Remove obsolete streaminfo code and fix a leak. Fixes #529546
|
|
|
|
2008-04-23 13:50:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
ext/ogg/gstoggdemux.c: Revert the event part, that should not go in.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c:
|
|
Revert the event part, that should not go in.
|
|
|
|
2008-04-23 13:45:29 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
ext/ogg/gstoggdemux.c: Don't leak GstPluginFeatures when filtering.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c:
|
|
Don't leak GstPluginFeatures when filtering.
|
|
|
|
2008-04-23 08:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
sys/xvimage/xvimagesink.c: Add some logging for cases when grabbing the xv failed.
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c:
|
|
Add some logging for cases when grabbing the xv failed.
|
|
|
|
2008-04-22 06:18:04 +0000 David Schleef <ds@schleef.org>
|
|
|
|
ext/ogg/gstoggmux.c: Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos packet. Should conform to what we cu...
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggmux.c:
|
|
Update Ogg/Dirac muxing. Removes the weird "KW-DIRAC" bos
|
|
packet. Should conform to what we currently think is the
|
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final Ogg/Dirac muxing spec.
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|
|
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2008-04-22 06:13:43 +0000 David Schleef <ds@schleef.org>
|
|
|
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sys/xvimage/xvimagesink.c: Fix typo that causes the overlay keying color to bright green on a 16-bit display. Dark g...
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c:
|
|
Fix typo that causes the overlay keying color to bright green
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|
on a 16-bit display. Dark grey good. Bright green bad.
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|
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2008-04-21 13:47:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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ext/gnomevfs/gstgnomevfsuri.c: Add FIXME comment about using uri-list for source and sink.
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Original commit message from CVS:
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|
* ext/gnomevfs/gstgnomevfsuri.c:
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Add FIXME comment about using uri-list for source and sink.
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2008-04-20 11:42:37 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
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ext/ogg/gstogmparse.c: GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to vaargs functions to gin...
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|
Original commit message from CVS:
|
|
* ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
|
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GST_TYPE_FRACTION contains gints so correctly cast gint64 arguments to
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vaargs functions to gint. Otherwise the fractions will get 0 set
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instead of the correct value on big endian systems. Fixes bug #529018.
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2008-04-20 10:17:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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ext/gnomevfs/: Get the list of supported URI schemes in a threadsafe way and use the same list for the source and sink.
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|
Original commit message from CVS:
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* ext/gnomevfs/gstgnomevfssink.c:
|
|
(gst_gnome_vfs_sink_uri_get_protocols):
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* ext/gnomevfs/gstgnomevfssrc.c:
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(gst_gnome_vfs_src_uri_get_protocols):
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* ext/gnomevfs/gstgnomevfsuri.c: (_internal_get_supported_uris),
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(gst_gnomevfs_get_supported_uris):
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Get the list of supported URI schemes in a threadsafe way and use the
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same list for the source and sink.
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2008-04-20 10:11:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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ext/gio/gstgio.c: Don't generate a new supported protocols list on each call but cache it. It's supposed to be static...
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|
Original commit message from CVS:
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* ext/gio/gstgio.c: (_internal_get_supported_protocols),
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|
(gst_gio_get_supported_protocols):
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Don't generate a new supported protocols list on each call but cache
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it. It's supposed to be static anyway, this way we only leak it once
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per process.
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* ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
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(gst_gio_sink_class_init), (gst_gio_sink_finalize),
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(gst_gio_sink_set_property), (gst_gio_sink_get_property),
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(gst_gio_sink_start):
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* ext/gio/gstgiosink.h:
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* ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
|
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(gst_gio_src_class_init), (gst_gio_src_finalize),
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(gst_gio_src_set_property), (gst_gio_src_get_property),
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(gst_gio_src_start):
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* ext/gio/gstgiosrc.h:
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API: Add "file" properties where one can set a GFile as source/destination.
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Add locking to the properties and use gst_element_class_set_details_simple()
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instead of a static GstElementDetails struct.
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2008-04-19 20:06:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/typefind/gsttypefindfunctions.c: Add "mpp" and "mp+" as possible extensions for MusePack files.
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|
Original commit message from CVS:
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|
* gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
|
|
(plugin_init):
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Add "mpp" and "mp+" as possible extensions for MusePack files.
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Add typefinding for MusePack StreamVersion 8 files and include the
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stream version in the caps.
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2008-04-19 16:33:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst-libs/gst/rtp/gstrtppayloads.c: Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
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|
Original commit message from CVS:
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* gst-libs/gst/rtp/gstrtppayloads.c:
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(gst_rtp_payload_info_for_name):
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Use g_ascii_strcasecmp() instead of the deprecated g_strcasecmp().
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2008-04-18 17:10:43 +0000 Tim-Philipp Müller <tim@centricular.net>
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configure.ac: Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level (NB: this only affects compilation of some...
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Original commit message from CVS:
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* configure.ac:
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Bump Gtk+ requirement to 2.12.0 for gtk_range_set_fill_level
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(NB: this only affects compilation of some of the examples).
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Remove some configure.ac cruft that's not needed any longer.
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2008-04-18 14:54:01 +0000 Edward Hervey <bilboed@bilboed.com>
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gst/gdp/gstgdpdepay.c: Don't validate the payload if there isn't any.
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|
Original commit message from CVS:
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* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
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Don't validate the payload if there isn't any.
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Fixes #525915
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2008-04-17 07:33:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst-libs/gst/audio/gstringbuffer.c: Use g_atomic_int_set() instead of gst_atomic_int_set().
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|
Original commit message from CVS:
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* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_may_start):
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Use g_atomic_int_set() instead of gst_atomic_int_set().
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2008-04-17 07:29:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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ext/gio/gstgio.c: Return NULL instead of a gchar * array with one NULL element if we don't get any supported URI sche...
|
|
Original commit message from CVS:
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* ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
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Return NULL instead of a gchar * array with one NULL element if we
|
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don't get any supported URI schemes from GIO.
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2008-04-15 19:06:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst/audiotestsrc/gstaudiotestsrc.c: Remove cpp style commented old code.
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|
Original commit message from CVS:
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* gst/audiotestsrc/gstaudiotestsrc.c:
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Remove cpp style commented old code.
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|
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2008-04-15 19:02:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst/playback/gstdecodebin2.c: Fix signal docs.
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|
Original commit message from CVS:
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* gst/playback/gstdecodebin2.c:
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Fix signal docs.
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|
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2008-04-14 17:58:19 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/pango/gsttextoverlay.c: Fix textoverlay unit test again by making the supposed default value for the wait-text pr...
|
|
Original commit message from CVS:
|
|
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
|
|
(gst_text_overlay_init):
|
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Fix textoverlay unit test again by making the supposed default
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value for the wait-text property the actual default value.
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Also fix Since: tag for new property.
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2008-04-11 17:13:52 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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gst-libs/gst/video/video.c: Add guards to these functions to ensure sane input values.
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|
Original commit message from CVS:
|
|
* gst-libs/gst/video/video.c: (gst_video_format_new_caps),
|
|
(gst_video_format_to_fourcc), (gst_video_format_get_row_stride),
|
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(gst_video_format_get_pixel_stride),
|
|
(gst_video_format_get_component_width),
|
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(gst_video_format_get_component_height),
|
|
(gst_video_format_get_component_offset), (gst_video_format_get_size),
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(gst_video_format_convert):
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Add guards to these functions to ensure sane input values.
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* tests/check/libs/video.c:
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Fix unit test not to create caps with width=0 and height=0.
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2008-04-11 01:25:01 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
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docs/design/draft-keyframe-force.txt: Fix typo.
|
|
Original commit message from CVS:
|
|
* docs/design/draft-keyframe-force.txt:
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Fix typo.
|
|
* gst/playback/gstqueue2.c: (update_buffering),
|
|
(gst_queue_handle_src_query):
|
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Set buffering mode in the messages.
|
|
Set buffering percent in the query.
|
|
* tests/examples/seek/seek.c: (update_fill), (msg_state_changed),
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(do_stream_buffering), (do_download_buffering), (msg_buffering):
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Do some more fancy things based on the buffering method in use.
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|
|
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2008-04-09 21:42:24 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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tests/examples/seek/seek.c: Add basic download reports to seek using the new buffering API.
|
|
Original commit message from CVS:
|
|
* tests/examples/seek/seek.c: (update_fill), (set_update_fill),
|
|
(play_cb), (pause_cb), (stop_cb), (msg_state_changed),
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(msg_buffering), (main):
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Add basic download reports to seek using the new buffering API.
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|
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2008-04-09 21:40:17 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst/playback/gstqueue2.c: Include extra buffering stats in the buffering message.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstqueue2.c: (update_buffering),
|
|
(gst_queue_close_temp_location_file), (gst_queue_handle_src_query),
|
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(gst_queue_src_checkgetrange_function):
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Include extra buffering stats in the buffering message.
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Implement BUFFERING query.
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* gst/playback/gsturidecodebin.c: (do_async_start),
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(do_async_done), (type_found), (setup_streaming), (setup_source),
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(gst_uri_decode_bin_change_state):
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Only add decodebin2 when the type is found in streaming mode.
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Make uridecodebin async to PAUSED even when we don't have decodebin2
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added yet.
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2008-04-09 08:38:19 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
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ext/gio/gstgio.c: Filter cdda from the supported URI schemes. We can't support musicbrainz tags and everything else o...
|
|
Original commit message from CVS:
|
|
* ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
|
|
Filter cdda from the supported URI schemes. We can't support
|
|
musicbrainz tags and everything else one expects from a cdda source
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|
with GIO. Fixes bug #526794.
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2008-04-07 22:37:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
|
|
* sys/xvimage/xvimagesink.c:
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|
Fix calculation of 'expected size' for YV12 buffers.
|
|
Original commit message from CVS:
|
|
2008-04-07 Jan Schmidt <jan.schmidt@sun.com>
|
|
* sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_new),
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|
(gst_xvimagesink_buffer_alloc):
|
|
Fix calculation of 'expected size' for YV12 buffers.
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Be a little more verbose in the debug output for buffer-alloc'ed
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buffers which turn out to have the wrong size.
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|
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2008-04-07 22:26:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
|
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* ChangeLog:
|
|
Fix calculation of 'expected size' for YV12 buffers.
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c (gst_xvimagesink_xvimage_new),
|
|
(gst_xvimagesink_buffer_alloc):
|
|
Fix calculation of 'expected size' for YV12 buffers.
|
|
Be a little more verbose in the debug output for buffer-alloc'ed
|
|
buffers which turn out to have the wrong size.
|
|
|
|
2008-04-07 10:50:11 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
Merge other changes from 0.10.19 release branch.
|
|
Original commit message from CVS:
|
|
* NEWS:
|
|
* RELEASE:
|
|
* gst-plugins-base.doap:
|
|
Merge other changes from 0.10.19 release branch.
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|
|
|
2008-04-06 20:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/: Work around missing bits of thread-safety on older GLibs some more to avoid assertions when starting up multipl...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_class_init):
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
(gst_base_audio_src_class_init):
|
|
* gst/playback/gstplayback.c: (plugin_init):
|
|
* gst/volume/gstvolume.c: (plugin_init):
|
|
Work around missing bits of thread-safety on older GLibs some
|
|
more to avoid assertions when starting up multiple playbin
|
|
objects concurrently (see #512382).
|
|
|
|
2008-04-06 17:19:39 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/pbutils/missing-plugins.c: Remove some more fields.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
|
|
Remove some more fields.
|
|
|
|
2008-04-06 08:56:07 +0000 Damien Lespiau <damien.lespiau@gmail.com>
|
|
|
|
configure.ac: Actually build dlls when cross-compiling with mingw32.
|
|
Original commit message from CVS:
|
|
Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
|
|
* configure.ac:
|
|
Actually build dlls when cross-compiling with mingw32.
|
|
Fixes bug #526247.
|
|
|
|
2008-04-03 23:01:11 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
configure.ac: Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Bump version to 0.10.19.1 after the unplanned 0.10.19 release.
|
|
|
|
2008-04-03 16:10:53 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
tests/examples/seek/seek.c: Add statusbar.
|
|
Original commit message from CVS:
|
|
* tests/examples/seek/seek.c: (play_cb), (pause_cb), (stop_cb),
|
|
(msg_buffering), (connect_bus_signals), (main):
|
|
Add statusbar.
|
|
Add buffering support with feedback in the statusbar.
|
|
|
|
2008-04-03 15:58:37 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/ogg/gstoggmux.c: Fix sample pipeline description.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggmux.c:
|
|
Fix sample pipeline description.
|
|
|
|
2008-04-03 14:58:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
docs/plugins/: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
|
|
Original commit message from CVS:
|
|
* docs/plugins/Makefile.am:
|
|
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-base-plugins-overrides.txt:
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
|
Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
|
* docs/plugins/gst-plugins-base-plugins.interfaces:
|
|
* docs/plugins/gst-plugins-base-plugins.prerequisites:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.xml:
|
|
* docs/plugins/inspect/plugin-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-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gdp.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.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-queue2.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-uridecodebin.xml:
|
|
* docs/plugins/inspect/plugin-video4linux.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:
|
|
Update introspection data.
|
|
* ext/ogg/gstoggmux.c:
|
|
Document oggmux.
|
|
* gst/playback/gstdecodebin2.c:
|
|
Don't use gtk-doc style comment start for private stuff, but make it
|
|
formatted like this for consistency.
|
|
|
|
2008-04-03 12:16:04 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstdecodebin2.c: Remove fakesink hack, we can now implement this more elegantly.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
|
|
(gst_decode_bin_init), (gst_decode_bin_dispose),
|
|
(gst_decode_bin_set_sink_caps), (gst_decode_bin_get_sink_caps),
|
|
(gst_decode_bin_set_property), (gst_decode_bin_get_property),
|
|
(analyze_new_pad), (connect_pad), (expose_pad),
|
|
(gst_decode_group_new), (gst_decode_group_control_demuxer_pad),
|
|
(gst_decode_group_expose), (gst_decode_group_free),
|
|
(do_async_start), (do_async_done), (gst_decode_bin_change_state):
|
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Remove fakesink hack, we can now implement this more elegantly.
|
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Added property to bypass typefinding.
|
|
Removed underrun callback and demuxer pad probe, we now use the srcpad
|
|
probe to expose groups.
|
|
API::sink-caps property
|
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* gst/playback/gstplaybin2.c: (no_more_pads_cb):
|
|
Guard against multiple emissions of the no_more_pads signal, which
|
|
happens when we are dealing with chained oggs.
|
|
* gst/playback/gsturidecodebin.c: (remove_decoders),
|
|
(make_decoder), (type_found), (setup_streaming), (source_new_pad),
|
|
(setup_source):
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For streams, use our own typefind element and plug our queue after it.
|
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We will need this to determine the type of buffering to use for the
|
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queue soon.
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|
|
|
2008-04-03 10:37:03 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst-libs/gst/audio/gstbaseaudiosink.c: Guard against over and underflows because of clock slaving.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render):
|
|
Guard against over and underflows because of clock slaving.
|
|
When we are using our own clock, still compensate for any calibrations
|
|
that we might have done to our clock.
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|
|
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2008-04-03 10:22:33 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/theora/theoradec.c: Don't try to do anything fancy with the return code from pushing an event, it does not have e...
|
|
Original commit message from CVS:
|
|
* ext/theora/theoradec.c: (theora_handle_type_packet),
|
|
(theora_dec_chain):
|
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Don't try to do anything fancy with the return code from pushing an
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event, it does not have enough information to turn it into a
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GST_FLOW_ERROR.
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2008-04-03 10:19:43 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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ext/ogg/gstoggdemux.c: Add small debug line.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_reset),
|
|
(gst_ogg_demux_chain_elem_pad):
|
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Add small debug line.
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Pass return code from the internal decoder instead of the too generic
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GST_FLOW_ERROR.
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|
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2008-04-03 06:39:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
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gst-libs/gst/cdda/: Use GLib's base64 implementation instead of our own.
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Original commit message from CVS:
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|
* gst-libs/gst/cdda/Makefile.am:
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* gst-libs/gst/cdda/base64.c:
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* gst-libs/gst/cdda/base64.h:
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|
* gst-libs/gst/cdda/gstcddabasesrc.c:
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(gst_cddabasesrc_calculate_musicbrainz_discid):
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Use GLib's base64 implementation instead of our own.
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|
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2008-04-02 15:41:50 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/ogg/gstoggdemux.c: Refix oggdemux, we only have a problem if we failed to find a chain and we are not EOF.
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Original commit message from CVS:
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* ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
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(gst_ogg_demux_read_chain):
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Refix oggdemux, we only have a problem if we failed to find a chain and
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we are not EOF.
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2008-04-02 15:07:01 +0000 Victor STINNER <victor.stinner@haypocalc.com>
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ext/ogg/gstoggdemux.c: When we fail to find a BOS page and we and up with no chain, error out properly instead of seg...
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Original commit message from CVS:
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Patch by: Victor STINNER <victor dot stinner at haypocalc dot com>
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* ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
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(gst_ogg_demux_read_chain):
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When we fail to find a BOS page and we and up with no chain, error out
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properly instead of segfaulting. Fixes #525665.
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2008-04-02 14:58:05 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/ogg/gstoggdemux.c: The new-pad-group sequence is add-pads, no-more-pads, add-pads, no-more-pads...
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Original commit message from CVS:
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* ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain),
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(gst_ogg_demux_read_chain), (gst_ogg_demux_handle_page):
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The new-pad-group sequence is add-pads, no-more-pads, add-pads,
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no-more-pads...
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2008-04-02 11:08:05 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstqueue2.c: Update the estimated input data when we push out a buffer.
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Original commit message from CVS:
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* gst/playback/gstqueue2.c: (update_out_rates),
|
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(gst_queue_open_temp_location_file),
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(gst_queue_close_temp_location_file), (gst_queue_handle_src_event),
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(gst_queue_handle_src_query), (gst_queue_set_property):
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Update the estimated input data when we push out a buffer.
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Add some debug info about the temp file.
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Only forward src events when we are not using a temp file.
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Don't block the duration query, we need to find something better.
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Don't leak the temp filename.
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2008-04-01 14:01:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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configure.ac: Require GLib 2.12 and liboil 0.3.14.
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Original commit message from CVS:
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* configure.ac:
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Require GLib 2.12 and liboil 0.3.14.
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* gst/volume/gstvolume.c: (volume_process_double):
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Unconditionally use liboil 0.3.14 function.
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2008-03-31 16:08:45 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/riff/riff-media.c: ms-gsm can have arbitrarty sample rates. See #481354.
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Original commit message from CVS:
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* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
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ms-gsm can have arbitrarty sample rates. See #481354.
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2008-03-28 16:22:35 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/riff/riff-media.c: MP4S is generic MPEG-4, not a microsoft variant.
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Original commit message from CVS:
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* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
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MP4S is generic MPEG-4, not a microsoft variant.
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2008-03-27 15:26:38 +0000 Michael Smith <msmith@xiph.org>
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gst/gdp/gstgdpdepay.c: Check the body CRC (if set) when depayloading.
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Original commit message from CVS:
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* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
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Check the body CRC (if set) when depayloading.
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Fixes #522401.
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2008-03-24 17:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/pango/gsttextoverlay.c: Fix Since: version for new property.
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Original commit message from CVS:
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* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
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Fix Since: version for new property.
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|
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2008-03-24 16:40:08 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst-libs/gst/rtsp/gstrtspconnection.c: Don't error when poll_wait returns EAGAIN.
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|
Original commit message from CVS:
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|
* gst-libs/gst/rtsp/gstrtspconnection.c:
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(gst_rtsp_connection_connect), (gst_rtsp_connection_write),
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(gst_rtsp_connection_read_internal), (gst_rtsp_connection_poll):
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Don't error when poll_wait returns EAGAIN.
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2008-03-24 14:08:22 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst/playback/gstqueue2.c: The queue is never filled when there are no buffers in the queue at all.
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|
Original commit message from CVS:
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|
* gst/playback/gstqueue2.c: (gst_queue_is_filled):
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The queue is never filled when there are no buffers in the queue at all.
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Fixes #523993.
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2008-03-24 12:26:30 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstplaybin2.c: Update some docs.
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|
Original commit message from CVS:
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* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
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(init_group), (free_group), (gst_play_bin_init),
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(gst_play_bin_finalize), (gst_play_bin_set_uri),
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|
(gst_play_bin_set_suburi), (gst_play_bin_get_video_tags),
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(gst_play_bin_get_audio_tags), (gst_play_bin_get_text_tags),
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(gst_play_bin_set_current_video_stream),
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(gst_play_bin_set_current_audio_stream),
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(gst_play_bin_set_current_text_stream),
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(gst_play_bin_set_encoding), (gst_play_bin_set_property),
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(gst_play_bin_get_property), (pad_added_cb), (pad_removed_cb),
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(no_more_pads_cb), (perform_eos), (autoplug_select_cb),
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(activate_group), (deactivate_group), (setup_next_source),
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(save_current_group), (gst_play_bin_change_state):
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Update some docs.
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Add new locks and conds to protect pipeline creation and group
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switching.
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Implement the sub-uri property.
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Keep track of pending uridecodebin creation and configure the output
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pipeline after all streams are configured.
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Propagate subtitle encoding to the uridecodebins.
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Implement getting the video/audio/visualisation elements.
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Use input-selector for stream switching.
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If we are asked to do visualisation, prefer to autoplug raw sinks
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instead of sinks that accept encoded data.
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2008-03-24 12:15:26 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst/playback/gstplaysink.*: Add methods to get audio/video/vis elements.
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|
Original commit message from CVS:
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|
* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
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|
(gst_play_sink_init), (gst_play_sink_dispose),
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(gst_play_sink_set_video_sink), (gst_play_sink_get_video_sink),
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(gst_play_sink_set_audio_sink), (gst_play_sink_get_audio_sink),
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(gst_play_sink_vis_unblocked), (gst_play_sink_vis_blocked),
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(gst_play_sink_set_vis_plugin), (gst_play_sink_get_vis_plugin),
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(gst_play_sink_set_volume), (gst_play_sink_get_volume),
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(gst_play_sink_set_mute), (gen_video_chain), (gen_text_chain),
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(gen_audio_chain), (gen_vis_chain), (gst_play_sink_reconfigure),
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(gst_play_sink_set_font_desc), (gst_play_sink_get_font_desc),
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(gst_play_sink_send_event_to_sink), (gst_play_sink_change_state):
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* gst/playback/gstplaysink.h:
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Add methods to get audio/video/vis elements.
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Add methods to set the font description for the overlay.
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Remove properties, we're using this element with its methods only.
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Add support for subtitles.
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Rearrange the locking a bit to not use the object lock for protecting
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the pipeline construction.
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Try to use the volume and mute property on the sink when its available.
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Implement the mute option with volume when the sink does not have a mute
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property.
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Only add volume element when the sink has no volume property.
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Only do visualisations with raw audio pads.
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2008-03-24 12:03:02 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
|
ext/pango/gsttextoverlay.*: Add property to configure waiting for text on the textpad or not, with the default behavi...
|
|
Original commit message from CVS:
|
|
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
|
|
(gst_text_overlay_init), (gst_text_overlay_set_property),
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|
(gst_text_overlay_get_property), (gst_text_overlay_src_event),
|
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(gst_text_overlay_text_event), (gst_text_overlay_video_event),
|
|
(gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
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(gst_text_overlay_change_state):
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* ext/pango/gsttextoverlay.h:
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Add property to configure waiting for text on the textpad or not, with
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the default behaviour being the old one (always wait for text before
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rendering the video). This default behaviour is usually not the best one
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because the text stream can very sparse and could require queueing a lot
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of video.
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Fix the flushing and EOS handing so that we don't mix up their meaning.
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2008-03-24 11:54:02 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
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gst/playback/gsturidecodebin.c: Add a readonly source property and notify.
|
|
Original commit message from CVS:
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|
* gst/playback/gsturidecodebin.c:
|
|
(gst_uri_decode_bin_autoplug_factories),
|
|
(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
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|
(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_encoding),
|
|
(gst_uri_decode_bin_set_property),
|
|
(gst_uri_decode_bin_get_property), (no_more_pads_full),
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|
(new_decoded_pad_cb), (gen_source_element), (remove_decoders),
|
|
(proxy_autoplug_factories_signal), (make_decoder),
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(source_new_pad), (setup_source):
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Add a readonly source property and notify.
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Add new lock for protecting the construction of the pipeline.
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Keep track of the decodebins we plugged.
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Correctly proxy the autoplug signal so that it actually continues.
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Proxy subtitle-encoding to the decodebins.
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2008-03-24 11:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
tests/examples/seek/seek.c: Rearrange some buttons in playbin2 and make some other boxes insensitive when needed.
|
|
Original commit message from CVS:
|
|
* tests/examples/seek/seek.c: (audio_toggle_cb), (video_toggle_cb),
|
|
(text_toggle_cb), (update_streams), (main):
|
|
Rearrange some buttons in playbin2 and make some other boxes insensitive
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when needed.
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Add language codes to subtitle selection boxes when we gind the right
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tags for the streams.
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|
|
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2008-03-24 11:36:08 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst/playback/gstdecodebin2.c: Protect caps property with the object lock.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
|
|
(gst_decode_bin_set_caps), (gst_decode_bin_get_caps),
|
|
(gst_decode_bin_set_subs_encoding),
|
|
(gst_decode_bin_get_subs_encoding),
|
|
(gst_decode_bin_autoplug_factories), (connect_pad), (are_raw_caps),
|
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(deactivate_free_recursive):
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|
Protect caps property with the object lock.
|
|
Protect encoding property with the object lock.
|
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Keep list of elements we added that have the subtitle-encoding property.
|
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Distribute the subtitle-encoding to all of the elements when it
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|
changes.
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|
|
|
2008-03-24 11:24:22 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstaudiosink.c: Small debug improvement.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_release):
|
|
Small debug improvement.
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_render):
|
|
Fix bug in determining the sample start/stop position, we want to base
|
|
this decision on the fact that we are going forwards or backwards, not
|
|
slower or faster. This fixes some ugly resync warnings when playing at
|
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very slow speeds.
|
|
|
|
2008-03-23 13:41:28 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
ext/gio/gstgio.c: Correctly set the supported URI schemes and don't leave some schemes in the middle or at the start ...
|
|
Original commit message from CVS:
|
|
* ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
|
|
Correctly set the supported URI schemes and don't leave
|
|
some schemes in the middle or at the start at NULL.
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|
|
|
2008-03-23 13:12:41 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/elements/gdpdepay.c: Make test compile without unused function/variable warnings on PPC.
|
|
Original commit message from CVS:
|
|
* tests/check/elements/gdpdepay.c:
|
|
Make test compile without unused function/variable warnings on PPC.
|
|
|
|
2008-03-22 15:00:53 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u...
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* ext/alsa/gstalsamixerelement.c:
|
|
(gst_alsa_mixer_element_class_init):
|
|
* ext/alsa/gstalsasink.c: (gst_alsasink_class_init):
|
|
* ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
|
|
* ext/cdparanoia/gstcdparanoiasrc.c:
|
|
(gst_cd_paranoia_src_class_init):
|
|
* ext/gio/gstgiosink.c: (gst_gio_sink_class_init):
|
|
* ext/gio/gstgiosrc.c: (gst_gio_src_class_init):
|
|
* ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init):
|
|
* ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init):
|
|
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
|
|
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init):
|
|
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
|
|
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
|
|
* ext/pango/gsttextrender.c: (gst_text_render_class_init):
|
|
* ext/theora/theoradec.c: (gst_theora_dec_class_init):
|
|
* ext/theora/theoraenc.c: (gst_theora_enc_class_init):
|
|
* ext/theora/theoraparse.c: (gst_theora_parse_class_init):
|
|
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init):
|
|
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
|
|
(gst_audio_filter_template_class_init):
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_class_init):
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
(gst_base_audio_src_class_init):
|
|
* gst-libs/gst/cdda/gstcddabasesrc.c:
|
|
(gst_cdda_base_src_class_init):
|
|
* gst-libs/gst/interfaces/mixertrack.c:
|
|
(gst_mixer_track_class_init):
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_class_init):
|
|
* gst-libs/gst/rtp/gstbasertppayload.c:
|
|
(gst_basertppayload_class_init):
|
|
* gst/audioconvert/gstaudioconvert.c:
|
|
(gst_audio_convert_class_init):
|
|
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init):
|
|
* gst/audioresample/gstaudioresample.c:
|
|
(gst_audioresample_class_init):
|
|
* gst/audiotestsrc/gstaudiotestsrc.c:
|
|
(gst_audio_test_src_class_init):
|
|
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init):
|
|
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
|
|
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
|
|
(preroll_unlinked):
|
|
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
|
|
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init):
|
|
* gst/playback/gstplaysink.c: (gst_play_sink_class_init):
|
|
* gst/playback/gstqueue2.c: (gst_queue_class_init):
|
|
* gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
|
|
* gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
|
|
(gst_stream_selector_class_init):
|
|
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init):
|
|
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
|
|
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
|
|
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
|
|
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
|
|
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
|
|
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
|
|
* gst/videorate/gstvideorate.c: (gst_video_rate_class_init):
|
|
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
|
|
* gst/videotestsrc/gstvideotestsrc.c:
|
|
(gst_video_test_src_class_init):
|
|
* gst/volume/gstvolume.c: (gst_volume_class_init):
|
|
* sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init):
|
|
* sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
|
|
* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
|
|
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init):
|
|
* sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
|
|
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use
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|
static strings (i.e. all). This gives us less memory usage,
|
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fewer allocations and thus less memory defragmentation. Depend
|
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on core CVS for this. Fixes bug #523806.
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|
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2008-03-22 14:13:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
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ext/gio/gstgio.c: Filter http and https protocols. GIO/GVfs handles them but it's impossible to implement iradio/icec...
|
|
Original commit message from CVS:
|
|
* ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
|
|
Filter http and https protocols. GIO/GVfs handles them but it's
|
|
impossible to implement iradio/icecast with it. Better use
|
|
souphttpsrc or something else for this.
|
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* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
|
|
If getting the file informations by a query fails try it with the
|
|
seek-to-end trick too.
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|
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2008-03-21 16:46:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
|
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gst/volume/gstvolume.c: memset buffers to zero if we get a GAP buffer. We usually see a buffer as one unit so let's h...
|
|
Original commit message from CVS:
|
|
* gst/volume/gstvolume.c: (gst_volume_interface_supported),
|
|
(gst_volume_base_init), (gst_volume_class_init),
|
|
(volume_process_double), (volume_process_float),
|
|
(volume_transform_ip), (plugin_init):
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|
memset buffers to zero if we get a GAP buffer. We usually see a
|
|
buffer as one unit so let's handle it as one and don't care about
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|
volume changes while processing one buffer.
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Also clean up some stuff a bit.
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|
|
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2008-03-21 15:58:44 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
|
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gst/audioconvert/gstaudioconvert.c: Make audioconvert GAP-aware by outputting silence buffers when the input has the ...
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
|
|
(gst_audio_convert_create_silence_buffer),
|
|
(gst_audio_convert_transform):
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|
Make audioconvert GAP-aware by outputting silence buffers when the
|
|
input has the GAP flag set. This is up to 8x faster.
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Based on a patch by Stefan Kost. Fixes bug #517813.
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|
|
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2008-03-21 15:54:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
|
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gst/volume/gstvolume.c: Use oil_scalarmultiply_f64_ns() for double processing when it's available at compile time.
|
|
Original commit message from CVS:
|
|
* gst/volume/gstvolume.c: (volume_process_double):
|
|
Use oil_scalarmultiply_f64_ns() for double processing when it's
|
|
available at compile time.
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|
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2008-03-21 13:27:47 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
|
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configure.ac: Fix lrint/lrintf checks to actually work. These functions are in libm on Linux at least so try to link ...
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
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Fix lrint/lrintf checks to actually work. These functions are
|
|
in libm on Linux at least so try to link to it.
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|
|
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2008-03-21 00:36:20 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
|
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configure.ac: Back to development - 0.10.18.1
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|
Original commit message from CVS:
|
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* configure.ac:
|
|
Back to development - 0.10.18.1
|
|
|
|
=== release 0.10.18 ===
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|
|
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2008-03-21 00:26:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
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* ChangeLog:
|
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* NEWS:
|
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* RELEASE:
|
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* configure.ac:
|
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* docs/plugins/gst-plugins-base-plugins.args:
|
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* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
|
* docs/plugins/gst-plugins-base-plugins.interfaces:
|
|
* docs/plugins/gst-plugins-base-plugins.prerequisites:
|
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* docs/plugins/gst-plugins-base-plugins.signals:
|
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* docs/plugins/inspect/plugin-adder.xml:
|
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* docs/plugins/inspect/plugin-alsa.xml:
|
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* docs/plugins/inspect/plugin-audioconvert.xml:
|
|
* docs/plugins/inspect/plugin-audiorate.xml:
|
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* docs/plugins/inspect/plugin-audioresample.xml:
|
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* docs/plugins/inspect/plugin-audiotestsrc.xml:
|
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* docs/plugins/inspect/plugin-cdparanoia.xml:
|
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* docs/plugins/inspect/plugin-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
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* docs/plugins/inspect/plugin-gdp.xml:
|
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* docs/plugins/inspect/plugin-gnomevfs.xml:
|
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* docs/plugins/inspect/plugin-libvisual.xml:
|
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* docs/plugins/inspect/plugin-ogg.xml:
|
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* docs/plugins/inspect/plugin-pango.xml:
|
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* docs/plugins/inspect/plugin-playback.xml:
|
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* docs/plugins/inspect/plugin-queue2.xml:
|
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* docs/plugins/inspect/plugin-subparse.xml:
|
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* docs/plugins/inspect/plugin-tcp.xml:
|
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* docs/plugins/inspect/plugin-theora.xml:
|
|
* docs/plugins/inspect/plugin-typefindfunctions.xml:
|
|
* docs/plugins/inspect/plugin-uridecodebin.xml:
|
|
* docs/plugins/inspect/plugin-video4linux.xml:
|
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* 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:
|
|
* po/LINGUAS:
|
|
* win32/common/config.h:
|
|
Release 0.10.18
|
|
Original commit message from CVS:
|
|
Release 0.10.18
|
|
|
|
2008-03-21 00:16:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
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* po/af.po:
|
|
* po/az.po:
|
|
* po/bg.po:
|
|
* po/ca.po:
|
|
* po/cs.po:
|
|
* po/da.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/es.po:
|
|
* po/fi.po:
|
|
* po/hu.po:
|
|
* po/it.po:
|
|
* po/lt.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/pl.po:
|
|
* po/sk.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
* po/zh_CN.po:
|
|
Update .po files
|
|
Original commit message from CVS:
|
|
Update .po files
|
|
|
|
2008-03-18 12:19:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
0.10.17.4 pre-release
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* win32/common/config.h:
|
|
0.10.17.4 pre-release
|
|
|
|
2008-03-18 11:20:05 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/sdp/gstsdpmessage.c: Use GST_STR_NULL when trying to print strings that could be NULL because this might...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_dump):
|
|
Use GST_STR_NULL when trying to print strings that could be NULL because
|
|
this might crash on some platforms. See #520808.
|
|
|
|
2008-03-18 11:10:12 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
|
|
|
|
gst-libs/gst/rtsp/gstrtspconnection.c: Generic Windows fixes that makes libgstrtsp work on Windows when coupled with ...
|
|
Original commit message from CVS:
|
|
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
(gst_rtsp_connection_connect), (gst_rtsp_connection_write),
|
|
(read_line), (gst_rtsp_connection_read_internal):
|
|
Generic Windows fixes that makes libgstrtsp work on Windows when
|
|
coupled with the new GstPoll API. See #520808.
|
|
|
|
2008-03-17 22:06:56 +0000 Milosz Derezynski <internalerror@gmail.com>
|
|
|
|
ext/gio/gstgiobasesrc.c: If seeking to a new position succeeds don't simply return from create() without creating a b...
|
|
Original commit message from CVS:
|
|
Patch by: Milosz Derezynski <internalerror at gmail dot com>
|
|
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create):
|
|
If seeking to a new position succeeds don't simply return from
|
|
create() without creating a buffer. Do this only in the case
|
|
seeking to the new position fails. Fixes bug #523054.
|
|
|
|
2008-03-17 10:32:28 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/video/video.c: Fix gst_video_format_parse_caps() for RGB caps with alpha channel (#522635).
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/video/video.c: (gst_video_format_parse_caps),
|
|
(gst_video_format_from_rgba32_masks):
|
|
Fix gst_video_format_parse_caps() for RGB caps with alpha channel
|
|
(#522635).
|
|
* tests/check/libs/video.c: (test_parse_caps_rgb), (video_suite):
|
|
Add unit test for the RGB caps parsing and creation, checking for
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|
internal consistency of the new API and consistency of the API with
|
|
the old GST_VIDEO_CAPS_* defines.
|
|
|
|
2008-03-14 18:42:35 +0000 David Schleef <ds@schleef.org>
|
|
|
|
gst/videotestsrc/videotestsrc.c: Oops, revert last change because -base is in freeze.
|
|
Original commit message from CVS:
|
|
* gst/videotestsrc/videotestsrc.c: Oops, revert last change
|
|
because -base is in freeze.
|
|
|
|
2008-03-14 17:33:09 +0000 William M. Brack <wbrack@mmm.hk>
|
|
|
|
gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
|
|
Original commit message from CVS:
|
|
Patch by: William M. Brack
|
|
* gst/videotestsrc/videotestsrc.c: Fix Bayer pattern generation.
|
|
|
|
2008-03-14 09:54:44 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gststreamselector.*: Revert change that caused regression until a real fix is found.
|
|
Original commit message from CVS:
|
|
* gst/playback/gststreamselector.c: (gst_selector_pad_event),
|
|
(gst_selector_pad_chain):
|
|
* gst/playback/gststreamselector.h:
|
|
Revert change that caused regression until a real fix is found.
|
|
Fixes #522203.
|
|
|
|
2008-03-12 12:39:13 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst-libs/gst/audio/gstringbuffer.*: Rename recently added buffer types to make more sense.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
|
|
* gst-libs/gst/audio/gstringbuffer.h:
|
|
Rename recently added buffer types to make more sense.
|
|
* ext/alsa/gstalsasink.c: (alsasink_parse_spec),
|
|
(gst_alsasink_write):
|
|
Adapt for above API changes.
|
|
Fixes bug #520523.
|
|
|
|
2008-03-11 13:23:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
win32/common/libgstnetbuffer.def: Add new symbol gst_netaddress_equal. Fixes bug #521743.
|
|
Original commit message from CVS:
|
|
* win32/common/libgstnetbuffer.def:
|
|
Add new symbol gst_netaddress_equal. Fixes bug #521743.
|
|
|
|
2008-03-11 00:25:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
0.10.17.3 pre-release
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* win32/common/config.h:
|
|
0.10.17.3 pre-release
|
|
|
|
2008-03-10 17:19:56 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosrc.c: Fix duration when no clock was provided. Fixes #520300.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
(gst_base_audio_src_create):
|
|
Fix duration when no clock was provided. Fixes #520300.
|
|
|
|
2008-03-07 18:17:44 +0000 Olivier Crete <tester@tester.ca>
|
|
|
|
Add trivial function to compare GstNetAddress. See #520626.
|
|
Original commit message from CVS:
|
|
Patch by: Olivier Crete <tester at tester ca>
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netaddress_equal):
|
|
* gst-libs/gst/netbuffer/gstnetbuffer.h:
|
|
Add trivial function to compare GstNetAddress. See #520626.
|
|
API: GstNetBuffer::gst_netaddress_equal
|
|
|
|
2008-03-07 16:10:51 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/tcp/gstmultifdsink.c: Update mode property docs, it's deprecated now.
|
|
Original commit message from CVS:
|
|
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
|
|
Update mode property docs, it's deprecated now.
|
|
|
|
2008-03-07 15:48:51 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/: Remove GstPollMode from gstpoll constructor.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
(gst_rtsp_connection_create):
|
|
* gst/tcp/gstmultifdsink.c: (gst_fdset_mode_get_type),
|
|
(gst_multi_fd_sink_class_init), (gst_multi_fd_sink_start):
|
|
* gst/tcp/gstmultifdsink.h:
|
|
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_start):
|
|
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_start):
|
|
Remove GstPollMode from gstpoll constructor.
|
|
|
|
2008-03-04 00:26:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
0.10.17.2 pre-release
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* win32/common/config.h:
|
|
0.10.17.2 pre-release
|
|
|
|
2008-03-03 23:59:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst/Makefile.am: GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean them twice
|
|
Original commit message from CVS:
|
|
* gst/Makefile.am:
|
|
GST_PLUGINS_ALL correctly lists subparse and tcp now, don't distclean
|
|
them twice
|
|
* win32/common/libgstinterfaces.def:
|
|
* win32/common/libgstrtp.def:
|
|
Add new API to the defs
|
|
|
|
2008-03-03 16:11:50 +0000 Mersad Jelacic <mersad@axis.com>
|
|
|
|
gst-libs/gst/rtp/gstbasertpaudiopayload.*: API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it po...
|
|
Original commit message from CVS:
|
|
Patch by: Mersad Jelacic <mersad at axis dot com>
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
|
|
API: add gst_base_rtp_audio_payload_set_samplebits_options() to make it
|
|
possible to specify the sample size in bits. (#509637)
|
|
|
|
2008-03-03 13:59:19 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/libs/mixer.c: Add a few simple checks for the new message types.
|
|
Original commit message from CVS:
|
|
* tests/check/libs/mixer.c:
|
|
Add a few simple checks for the new message types.
|
|
|
|
2008-03-03 13:56:38 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed() and gst_mixer_message_parse_options_list_changed...
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/interfaces/mixer.c: (gst_mixer_option_changed),
|
|
(gst_mixer_options_list_changed), (gst_mixer_mixer_changed),
|
|
(gst_mixer_message_get_type),
|
|
(gst_mixer_message_parse_option_changed),
|
|
(gst_mixer_message_parse_options_list_changed):
|
|
* gst-libs/gst/interfaces/mixer.h: (GstMixerType),
|
|
(GST_MIXER_MESSAGE_OPTION_CHANGED),
|
|
(GST_MIXER_MESSAGE_OPTIONS_LIST_CHANGED),
|
|
(GST_MIXER_MESSAGE_MIXER_CHANGED):
|
|
API: add gst_mixer_options_list_changed(), gst_mixer_mixer_changed()
|
|
and gst_mixer_message_parse_options_list_changed(). Fixes #519916.
|
|
|
|
2008-03-03 13:50:18 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/interfaces/mixeroptions.*: API: add GstMixerOptions::get_values vfunc (#519906)
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/mixeroptions.c: (gst_mixer_options_init),
|
|
(gst_mixer_options_get_values):
|
|
* gst-libs/gst/interfaces/mixeroptions.h:
|
|
(GST_MIXER_OPTIONS_GET_CLASS), (GstMixerOptionsClass),
|
|
(_GstMixerOptions), (_GstMixerOptionsClass):
|
|
API: add GstMixerOptions::get_values vfunc (#519906)
|
|
|
|
2008-03-03 12:01:15 +0000 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
configure.ac: Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which plug-ins are included/excluded. (#4...
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Use AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to simplify which
|
|
plug-ins are included/excluded. (#498222)
|
|
|
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2008-03-03 06:22:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Add typefinder for IMelody files, using audio/x-imelody.
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
|
|
Add typefinder for IMelody files, using audio/x-imelody.
|
|
See bug #519516.
|
|
|
|
2008-03-03 06:04:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static...
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_get_type):
|
|
* ext/alsa/gstalsasink.c: (set_hwparams):
|
|
* ext/alsa/gstalsasrc.c: (set_hwparams):
|
|
* ext/gio/gstgio.c: (gst_gio_uri_handler_get_uri):
|
|
* ext/ogg/gstoggmux.h:
|
|
* ext/ogg/gstogmparse.c:
|
|
* gst-libs/gst/audio/audio.c:
|
|
* gst-libs/gst/fft/kiss_fft_f64.c: (kiss_fft_f64_alloc):
|
|
* gst-libs/gst/pbutils/missing-plugins.c:
|
|
(gst_missing_uri_sink_message_new),
|
|
(gst_missing_element_message_new),
|
|
(gst_missing_decoder_message_new),
|
|
(gst_missing_encoder_message_new):
|
|
* gst-libs/gst/rtp/gstbasertppayload.c:
|
|
* gst-libs/gst/rtp/gstrtcpbuffer.c:
|
|
(gst_rtcp_packet_bye_get_reason):
|
|
* gst/audioconvert/gstaudioconvert.c:
|
|
* gst/audioresample/gstaudioresample.c:
|
|
* gst/ffmpegcolorspace/imgconvert.c:
|
|
* gst/playback/test.c: (gen_video_element), (gen_audio_element):
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
* gst/videoscale/vs_4tap.c:
|
|
* gst/videoscale/vs_4tap.h:
|
|
* sys/v4l/gstv4lelement.c:
|
|
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_get_any_caps):
|
|
* sys/v4l/v4l_calls.c:
|
|
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_capture_init),
|
|
(gst_v4lsrc_try_capture):
|
|
* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
|
|
(gst_ximagesink_ximage_new):
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls),
|
|
(gst_xvimagesink_xvimage_new):
|
|
* tests/check/elements/audioconvert.c:
|
|
* tests/check/elements/audioresample.c:
|
|
(fail_unless_perfect_stream):
|
|
* tests/check/elements/audiotestsrc.c: (setup_audiotestsrc):
|
|
* tests/check/elements/decodebin.c:
|
|
* tests/check/elements/gdpdepay.c: (setup_gdpdepay),
|
|
(setup_gdpdepay_streamheader):
|
|
* tests/check/elements/gdppay.c: (setup_gdppay), (GST_START_TEST),
|
|
(setup_gdppay_streamheader):
|
|
* tests/check/elements/gnomevfssink.c: (setup_gnomevfssink):
|
|
* tests/check/elements/multifdsink.c: (setup_multifdsink):
|
|
* tests/check/elements/textoverlay.c:
|
|
* tests/check/elements/videorate.c: (setup_videorate):
|
|
* tests/check/elements/videotestsrc.c: (setup_videotestsrc):
|
|
* tests/check/elements/volume.c: (setup_volume):
|
|
* tests/check/elements/vorbisdec.c: (setup_vorbisdec):
|
|
* tests/check/elements/vorbistag.c:
|
|
* tests/check/generic/clock-selection.c:
|
|
* tests/check/generic/states.c: (setup), (teardown):
|
|
* tests/check/libs/cddabasesrc.c:
|
|
* tests/check/libs/video.c:
|
|
* tests/check/pipelines/gio.c:
|
|
* tests/check/pipelines/oggmux.c:
|
|
* tests/check/pipelines/simple-launch-lines.c:
|
|
(simple_launch_lines_suite):
|
|
* tests/check/pipelines/streamheader.c:
|
|
* tests/check/pipelines/theoraenc.c:
|
|
* tests/check/pipelines/vorbisdec.c:
|
|
* tests/check/pipelines/vorbisenc.c:
|
|
* tests/examples/seek/scrubby.c:
|
|
* tests/examples/seek/seek.c: (query_positions_elems),
|
|
(query_positions_pads):
|
|
* tests/icles/stress-xoverlay.c: (myclock):
|
|
Correct all relevant warnings found by the sparse semantic code
|
|
analyzer. This include marking several symbols static, using
|
|
NULL instead of 0 for pointers and using "foo (void)" instead
|
|
of "foo ()" for declarations.
|
|
* win32/common/libgstrtp.def:
|
|
Add gst_rtp_buffer_set_extension_data to the symbol definition file.
|
|
|
|
2008-03-02 18:43:15 +0000 José Alburquerque <jaalburqu@svn.gnome.org>
|
|
|
|
gst/playback/gstplaybin2.c: Make the function signature of the _get_*_tags() functions match the signature of the vfu...
|
|
Original commit message from CVS:
|
|
Patch by: José Alburquerque <jaalburqu svn gnome org>
|
|
* gst/playback/gstplaybin2.c:
|
|
Make the function signature of the _get_*_tags() functions match
|
|
the signature of the vfuncs they implement, ie. return a
|
|
GstTagList rather than a GstStructure, which is more correct,
|
|
even if one is typedef'ed to the other (#518940).
|
|
|
|
2008-03-02 18:32:36 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/rtsp/gstrtspconnection.c: Don't include unix headers unconditionally (fixes #518037).
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
Don't include unix headers unconditionally (fixes #518037).
|
|
|
|
2008-03-02 18:24:37 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/libs/video.c: Add unit test that makes sure that the strides, offsets and sizes returned for the various ...
|
|
Original commit message from CVS:
|
|
* tests/check/libs/video.c: (paintinfo), (paintinfo_struct),
|
|
(fourcc_list_struct), (fourcc_list), (fourcc_get_size),
|
|
(paint_setup_I420), (paint_setup_YV12), (paint_setup_AYUV),
|
|
(paint_setup_YUY2), (paint_setup_UYVY), (paint_setup_YVYU),
|
|
(paint_setup_IYU2), (paint_setup_Y41B), (paint_setup_Y42B),
|
|
(paint_setup_Y800), (paint_setup_YVU9), (paint_setup_YUV9),
|
|
(gst_video_format_is_packed), (video_format_is_packed):
|
|
Add unit test that makes sure that the strides, offsets and
|
|
sizes returned for the various YUV formats by the new video API
|
|
match the old reference implementation in videotestsrc.
|
|
|
|
2008-03-02 18:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/video/video.*: API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio),
|
|
(gst_video_format_from_fourcc), (gst_video_format_to_fourcc),
|
|
(gst_video_format_is_rgb), (gst_video_format_is_yuv),
|
|
(gst_video_format_has_alpha), (gst_video_format_get_row_stride),
|
|
(gst_video_format_get_pixel_stride),
|
|
(gst_video_format_get_component_width),
|
|
(gst_video_format_get_component_height),
|
|
(gst_video_format_get_component_offset), (gst_video_format_get_size):
|
|
* gst-libs/gst/video/video.h: (GST_VIDEO_FORMAT_Y41B),
|
|
(GST_VIDEO_FORMAT_Y42B):
|
|
API: add GST_VIDEO_FORMAT_Y41B and GST_VIDEO_FORMAT_Y42B.
|
|
|
|
2008-03-02 18:07:10 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/video/video.c: YV12 is I420 with swapped components 1 and 2, so the offset of component 1 for I420 shoul...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/video/video.c: (gst_video_format_get_component_offset):
|
|
YV12 is I420 with swapped components 1 and 2, so the offset of
|
|
component 1 for I420 should be the offset for component 2 for YV12
|
|
and vice versa.
|
|
|
|
2008-02-29 21:48:00 +0000 Rene Stadler <mail@renestadler.de>
|
|
|
|
sys/v4l/gstv4lelement.c: Add missing semicolon to fix indentation.
|
|
Original commit message from CVS:
|
|
* sys/v4l/gstv4lelement.c:
|
|
Add missing semicolon to fix indentation.
|
|
|
|
2008-02-29 18:44:36 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
ext/alsa/gstalsa.c: Probe for IEC958 pcm to detect if we can do SPDIF output.
|
|
Original commit message from CVS:
|
|
2008-02-29 Julien Moutte <julien@fluendo.com>
|
|
* ext/alsa/gstalsa.c: (gst_alsa_open_iec958_pcm),
|
|
(gst_alsa_probe_supported_formats): Probe for IEC958 pcm to
|
|
detect
|
|
if we can do SPDIF output.
|
|
* ext/alsa/gstalsa.h:
|
|
* ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec),
|
|
(gst_alsasink_prepare), (gst_alsasink_close),
|
|
(gst_alsasink_write):
|
|
* ext/alsa/gstalsasink.h: Initial support for SPDIF.
|
|
* gst-libs/gst/audio/gstringbuffer.c:
|
|
(gst_ring_buffer_parse_caps):
|
|
* gst-libs/gst/audio/gstringbuffer.h: Add non linear buffer
|
|
types
|
|
to support AC3, EC3 and IEC958 buffers.
|
|
|
|
2008-02-29 17:59:16 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/interfaces/mixer.c: De-cruft and fix message type assertions (NULL is not a really valid mixer message t...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/mixer.c: (GST_MIXER_MESSAGE_HAS_TYPE),
|
|
(gst_mixer_message_parse_mute_toggled),
|
|
(gst_mixer_message_parse_record_toggled),
|
|
(gst_mixer_message_parse_volume_changed),
|
|
(gst_mixer_message_parse_option_changed):
|
|
De-cruft and fix message type assertions (NULL is not a really
|
|
valid mixer message type string).
|
|
|
|
2008-02-29 14:52:02 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/libvisual/visual.c: When negotiating, actually start from a format that we can support instead of from the too ge...
|
|
Original commit message from CVS:
|
|
* ext/libvisual/visual.c: (gst_vis_src_negotiate):
|
|
When negotiating, actually start from a format that we can support
|
|
instead of from the too generic template.
|
|
|
|
2008-02-29 12:26:48 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybin2.c: Enable vis setting.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin2.c: (gst_play_bin_set_property):
|
|
Enable vis setting.
|
|
* gst/playback/gstplaysink.c: (gst_play_sink_init),
|
|
(gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
|
|
(gst_play_sink_vis_blocked), (gst_play_sink_set_vis_plugin),
|
|
(gen_vis_chain):
|
|
Implement vis switching while playing.
|
|
|
|
2008-02-29 00:04:57 +0000 David Schleef <ds@schleef.org>
|
|
|
|
gst-libs/gst/riff/riff-media.c: Add Dirac mapping
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: Add Dirac mapping
|
|
|
|
2008-02-28 10:54:14 +0000 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
gst/tcp/: Removed fdset and stress test, they are now known as GstPoll in core.
|
|
Original commit message from CVS:
|
|
Patch by: Peter Kjellerstedt <pkj at axis com>
|
|
* gst/tcp/Makefile.am:
|
|
* gst/tcp/fdsetstress.c:
|
|
* gst/tcp/gstfdset.c:
|
|
* gst/tcp/gstfdset.h:
|
|
Removed fdset and stress test, they are now known as GstPoll in
|
|
core.
|
|
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
|
|
(gst_multi_fd_sink_add_full), (gst_multi_fd_sink_remove),
|
|
(gst_multi_fd_sink_clear), (gst_multi_fd_sink_remove_client_link),
|
|
(gst_multi_fd_sink_handle_client_write),
|
|
(gst_multi_fd_sink_queue_buffer),
|
|
(gst_multi_fd_sink_handle_clients), (gst_multi_fd_sink_start),
|
|
(gst_multi_fd_sink_stop):
|
|
* gst/tcp/gstmultifdsink.h:
|
|
* gst/tcp/gsttcp.c: (gst_tcp_socket_read), (gst_tcp_socket_close),
|
|
(gst_tcp_read_buffer), (gst_tcp_gdp_read_buffer),
|
|
(gst_tcp_gdp_read_caps):
|
|
* gst/tcp/gsttcp.h:
|
|
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_init),
|
|
(gst_tcp_client_sink_setcaps), (gst_tcp_client_sink_render),
|
|
(gst_tcp_client_sink_start), (gst_tcp_client_sink_stop):
|
|
* gst/tcp/gsttcpclientsink.h:
|
|
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_init),
|
|
(gst_tcp_client_src_create), (gst_tcp_client_src_start),
|
|
(gst_tcp_client_src_stop), (gst_tcp_client_src_unlock):
|
|
* gst/tcp/gsttcpclientsrc.h:
|
|
* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_handle_wait),
|
|
(gst_tcp_server_sink_init_send), (gst_tcp_server_sink_close):
|
|
* gst/tcp/gsttcpserversink.h:
|
|
* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_init),
|
|
(gst_tcp_server_src_create), (gst_tcp_server_src_start),
|
|
(gst_tcp_server_src_stop), (gst_tcp_server_src_unlock):
|
|
* gst/tcp/gsttcpserversrc.h:
|
|
Port to GstPoll. See #505417.
|
|
|
|
2008-02-28 09:54:14 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
* ChangeLog:
|
|
Patch Changelog a bit to give credit and refer to the relevant bug.
|
|
Original commit message from CVS:
|
|
Patch Changelog a bit to give credit and refer to the
|
|
relevant bug.
|
|
|
|
2008-02-28 09:50:52 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtsp/gstrtspconnection.*: Use GstPoll for the rtsp connection.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
(gst_rtsp_connection_create), (gst_rtsp_connection_connect),
|
|
(gst_rtsp_connection_write), (gst_rtsp_connection_read_internal),
|
|
(gst_rtsp_connection_receive), (gst_rtsp_connection_close),
|
|
(gst_rtsp_connection_free), (gst_rtsp_connection_poll),
|
|
(gst_rtsp_connection_flush):
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
Use GstPoll for the rtsp connection.
|
|
|
|
2008-02-27 12:19:31 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
tests/examples/seek/seek.c: Add combo box for visualisations, populate it with a factory list of all visualisation pl...
|
|
Original commit message from CVS:
|
|
* tests/examples/seek/seek.c: (vis_toggle_cb), (filter_features),
|
|
(init_visualization_features), (vis_combo_cb), (shot_cb), (main):
|
|
Add combo box for visualisations, populate it with a factory list
|
|
of all visualisation plugins, configure vis plugin instance in
|
|
playbin2.
|
|
|
|
2008-02-27 10:55:03 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
tests/check/libs/rtp.c: Add check for RTP buffer defaults, padding and marker bit API.
|
|
Original commit message from CVS:
|
|
* tests/check/libs/rtp.c: (GST_START_TEST):
|
|
Add check for RTP buffer defaults, padding and marker bit API.
|
|
|
|
2008-02-27 10:42:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst-libs/gst/cdda/sha1.c: Use memcpy() instead of upcasting a byte array to long *. This fixes an unaligned memory ac...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/cdda/sha1.c: (sha_transform):
|
|
Use memcpy() instead of upcasting a byte array to long *. This
|
|
fixes an unaligned memory access, resulting in SIGBUS on IA64.
|
|
This should be ported to GCheckSum once we can use GLib 2.16.
|
|
Partially fixes bug #500833.
|
|
|
|
2008-02-27 10:23:27 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/tag/gsttagdemux.c: Push tag event after the newsegment event. Log the pointer of the buffer we're actual...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_chain):
|
|
Push tag event after the newsegment event. Log the pointer of
|
|
the buffer we're actually going to push rather than the buffer
|
|
we're feeding to _make_metadata_writable().
|
|
|
|
2008-02-25 07:21:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Comment smoke typefinder for now. The smokedec plugin needs one frame per buffer...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
|
|
Comment smoke typefinder for now. The smokedec plugin needs one
|
|
frame per buffer but we have no parser yet, thus it simply crashes
|
|
in most situations.
|
|
|
|
2008-02-25 06:48:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Add typefinder for the smoke video codec. Copied from the jpeg plugin.
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
|
|
Add typefinder for the smoke video codec. Copied from the jpeg plugin.
|
|
|
|
2008-02-25 06:29:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Add midi typefinder, copied from the timidity plugin.
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (mid_type_find),
|
|
(plugin_init):
|
|
Add midi typefinder, copied from the timidity plugin.
|
|
|
|
2008-02-23 09:51:26 +0000 Tomasz Sałaciński <tsalacinski@gmail.com>
|
|
|
|
Forward slashes at the beginning and end of a line also signify italics (Fixes: #518162).
|
|
Original commit message from CVS:
|
|
Based on patch by: Tomasz Sałaciński <tsalacinski gmail com>
|
|
* gst/subparse/gstsubparse.c: (parse_mdvdsub):
|
|
* tests/check/elements/subparse.c: (test_microdvd_with_italics),
|
|
(subparse_suite):
|
|
Forward slashes at the beginning and end of a line also signify
|
|
italics (Fixes: #518162).
|
|
|
|
2008-02-22 06:38:08 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
tests/check/gst-plugins-base.supp: Add a suppression for a cached value in GIO that wasn't moved while moving gio fro...
|
|
Original commit message from CVS:
|
|
* tests/check/gst-plugins-base.supp:
|
|
Add a suppression for a cached value in GIO that wasn't moved
|
|
while moving gio from -bad to -base.
|
|
|
|
2008-02-22 05:27:24 +0000 Brian Cameron <brian.cameron@sun.com>
|
|
|
|
configure.ac: Don't hardcode -Wall and -Werror for configure checks, this fails with non-GCC compilers. Fixes bug #51...
|
|
Original commit message from CVS:
|
|
Patch by: Brian Cameron <brian dot cameron at sun dot com>
|
|
* configure.ac:
|
|
Don't hardcode -Wall and -Werror for configure checks, this fails
|
|
with non-GCC compilers. Fixes bug #517991.
|
|
|
|
2008-02-21 08:05:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/audiotestsrc/gstaudiotestsrc.c: Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
|
|
Original commit message from CVS:
|
|
* gst/audiotestsrc/gstaudiotestsrc.c:
|
|
Mark buffers as GAP,if volume is 0.0 and fix the previous logic.
|
|
|
|
2008-02-20 15:37:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
ext/gnomevfs/gstgnomevfssink.c: Return FALSE when seeking for a new segment fails instead of silently ignoring the fa...
|
|
Original commit message from CVS:
|
|
* ext/gnomevfs/gstgnomevfssink.c:
|
|
(gst_gnome_vfs_sink_handle_event):
|
|
Return FALSE when seeking for a new segment fails instead
|
|
of silently ignoring the failure and appending every buffer
|
|
that comes for the new segment.
|
|
|
|
2008-02-20 11:52:28 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaysink.c: Recursively search the sink element for a last-frame property so that we can also find th...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaysink.c: (find_property),
|
|
(gst_play_sink_find_property), (gen_video_chain),
|
|
(gst_play_sink_reconfigure), (gst_play_sink_get_last_frame):
|
|
Recursively search the sink element for a last-frame property so that we
|
|
can also find the property in autovideosink and friends that don't
|
|
always proxy the internal sink properties.
|
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2008-02-19 20:42:09 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/audio/multichannel.c: Fix confusing terminology in docs and code: structure fields are 'fields' and not ...
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Original commit message from CVS:
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* gst-libs/gst/audio/multichannel.c:
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(GST_AUDIO_CHANNEL_POSITIONS_FIELD_NAME),
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(gst_audio_get_channel_positions), (gst_audio_set_channel_positions),
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(gst_audio_set_structure_channel_positions_list),
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(add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
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(gst_audio_fixate_channel_positions):
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Fix confusing terminology in docs and code: structure fields are
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'fields' and not 'properties'.
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2008-02-19 20:36:58 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/audio/multichannel.c: Give more useful warning messages if one of the channel layout enums passed to us ...
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Original commit message from CVS:
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* gst-libs/gst/audio/multichannel.c:
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(gst_audio_check_channel_positions), (add_list_to_struct):
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Give more useful warning messages if one of the channel
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layout enums passed to us is invalid and if the "channels"
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field in the caps has a GType we don't expect.
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2008-02-19 20:22:09 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/audio/multichannel.c: Fix typo in docs blurb.
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Original commit message from CVS:
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* gst-libs/gst/audio/multichannel.c:
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Fix typo in docs blurb.
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|
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2008-02-19 16:16:55 +0000 Josep Torra Valles <josep@fluendo.com>
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gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS typefind lookup to fix typefinding on HD clips.
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Original commit message from CVS:
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2008-02-19 Julien Moutte <julien@fluendo.com>
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Patch by: Josep Torra Valles <josep@fluendo.com>
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* gst/typefind/gsttypefindfunctions.c: Increase the MPEG PS
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typefind lookup to fix typefinding on HD clips.
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2008-02-19 15:50:37 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gstscreenshot.*: Fix up copyright (I rewrote the GStreamer-0.10 code for this from scratch back in the d...
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Original commit message from CVS:
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* gst/playback/gstscreenshot.c:
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* gst/playback/gstscreenshot.h:
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Fix up copyright (I rewrote the GStreamer-0.10 code for
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this from scratch back in the days).
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2008-02-19 15:02:33 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/: Add screenshot conversion code from totem.
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Original commit message from CVS:
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* gst/playback/Makefile.am:
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* gst/playback/gstscreenshot.c: (feed_fakesrc), (save_result),
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(create_element), (gst_play_frame_conv_convert):
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* gst/playback/gstscreenshot.h:
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Add screenshot conversion code from totem.
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* gst/playback/gstplay-marshal.list:
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* gst/playback/gstplaybin2.c: (gst_play_marshal_BUFFER__BOXED),
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(gst_play_bin_class_init), (gst_play_bin_convert_frame),
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(gst_play_bin_get_property), (no_more_pads_cb), (activate_group):
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Implement frame property to get a color-unconverted snapshot.
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Implement convert-frame action signal to get a converted snapshot image.
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Configure connection speed in uridecodebin.
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Document some more properties.
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* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
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(gen_video_chain), (gen_audio_chain), (gst_play_sink_reconfigure),
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(gst_play_sink_get_last_frame):
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* gst/playback/gstplaysink.h:
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Use last-buffer property of the video sink to get a video snapshot.
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* tests/examples/seek/seek.c: (shot_cb), (main):
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Add snapshot button for playbin2 and use the frame property to save the
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frame as a png in the current directory.
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2008-02-19 11:45:56 +0000 Josep Torra Valles <josep@fluendo.com>
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gst/typefind/gsttypefindfunctions.c: Add typefinding support for h264 elementary streams.
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Original commit message from CVS:
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Patch by: Josep Torra Valles <josep at fluendo dot com>
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* gst/typefind/gsttypefindfunctions.c: (h264_video_type_find),
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(plugin_init):
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Add typefinding support for h264 elementary streams.
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Fixes bug #517420.
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2008-02-18 13:51:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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configure.ac: Require CVS of core for new API in collectpads.
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Original commit message from CVS:
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* configure.ac:
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Require CVS of core for new API in collectpads.
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* gst/adder/gstadder.c:
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Use new API to make adder sparse stream aware.
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2008-02-18 11:54:15 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstplaybin2.c: Get the object data correct so that we can remove our channels correctly.
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Original commit message from CVS:
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* gst/playback/gstplaybin2.c: (pad_added_cb), (pad_removed_cb),
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(no_more_pads_cb):
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Get the object data correct so that we can remove our channels
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correctly.
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* gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
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(gen_vis_chain), (gst_play_sink_reconfigure),
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(gst_play_sink_request_pad):
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Add option to disable async behaviour in the sinks when possible. This
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makes it possible to avoid an audio queue when dealing with
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visualisations.
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Add option to add a queue for the audio path.
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* tests/examples/seek/seek.c: (clear_streams), (update_streams),
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(main):
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Disable the vis checkbox to match the defaults of playbin2.
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Only get the stream info when we need to.
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2008-02-17 05:15:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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ext/gio/: Don't use async operations as they require a running main loop.
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Original commit message from CVS:
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* ext/gio/gstgiobasesink.c: (gst_gio_base_sink_stop),
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(gst_gio_base_sink_set_stream):
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* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_stop),
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(gst_gio_base_src_set_stream):
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* ext/gio/gstgiosink.c: (gst_gio_sink_start):
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* ext/gio/gstgiosrc.c: (gst_gio_src_start):
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Don't use async operations as they require a running main loop.
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This makes us block again when closing streams and unable
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to mount the enclosing volume of an URI if it isn't yet.
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2008-02-15 18:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstplaysink.c: Move tee in front of the audio and vis pipelines.
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Original commit message from CVS:
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* gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
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|
(gst_play_sink_get_mute), (gen_video_chain), (gen_audio_chain),
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(gen_vis_chain), (gst_play_sink_reconfigure),
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(gst_play_sink_request_pad):
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Move tee in front of the audio and vis pipelines.
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Add queue for audio for now.
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Add visualisation support.
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* tests/examples/seek/seek.c: (main):
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Visualisation is by default disabled.
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2008-02-15 11:58:06 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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ext/gio/: Improve debugging a bit.
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|
Original commit message from CVS:
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|
* ext/gio/gstgiobasesink.c: (close_stream_cb):
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* ext/gio/gstgiobasesrc.c: (close_stream_cb):
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Improve debugging a bit.
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* ext/gio/gstgiosink.c: (mount_cb), (gst_gio_sink_start):
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* ext/gio/gstgiosink.h:
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* ext/gio/gstgiosrc.c: (mount_cb), (gst_gio_src_start):
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* ext/gio/gstgiosrc.h:
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Try to mount the enclosing volume of a GFile if it isn't mounted
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yet. This requires us to wait for an async operation to finish, done
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with an nested GMainLoop. Authentication is not supported yet, will
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come later.
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2008-02-14 18:24:42 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/: Add mute property.
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|
Original commit message from CVS:
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* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
|
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(gst_play_bin_set_property), (gst_play_bin_get_property),
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(pad_added_cb), (pad_removed_cb), (no_more_pads_cb):
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* gst/playback/gstplaysink.c: (gst_play_sink_set_mute),
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(gst_play_sink_get_mute), (gen_audio_chain):
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* gst/playback/gstplaysink.h:
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Add mute property.
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* gst/playback/gststreamselector.c: (gst_selector_pad_event),
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(gst_selector_pad_chain):
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* gst/playback/gststreamselector.h:
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Make sure we forward the event only once.
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* tests/examples/seek/seek.c: (stop_cb), (mute_toggle_cb), (main):
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Add and implement the mute button for playbin2.
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2008-02-13 14:34:55 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
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ext/alsa/gstalsasink.c: Add some more debug info.
|
|
Original commit message from CVS:
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|
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
|
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* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_delay):
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Add some more debug info.
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Make sure we never return a negative delay. Fixes #516246.
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|
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2008-02-12 20:09:07 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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ext/alsa/gstalsasink.c: Revert patch that makes the sink hold the object lock when calling snd_pcm_delay(), since it ...
|
|
Original commit message from CVS:
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|
* ext/alsa/gstalsasink.c: (gst_alsasink_delay):
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Revert patch that makes the sink hold the object lock when
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calling snd_pcm_delay(), since it breaks playback for me.
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2008-02-12 19:50:36 +0000 Julien Moutte <julien@moutte.net>
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tests/examples/seek/seek.c: Add some seek flags when changing rate.
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Original commit message from CVS:
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2008-02-12 Julien Moutte <julien@fluendo.com>
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* tests/examples/seek/seek.c: (rate_spinbutton_changed_cb): Add
|
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some seek flags when changing rate.
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|
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2008-02-12 14:51:26 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fix potential leaks.
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|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
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(gst_base_rtp_audio_payload_handle_frame_based_buffer),
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(gst_base_rtp_audio_payload_handle_sample_based_buffer):
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Fix potential leaks.
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* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_chain):
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Fix leak when there is no function configured.
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|
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2008-02-12 11:36:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
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sys/v4l/v4lsrc_calls.c: Correctly chain up the finalize method.
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|
Original commit message from CVS:
|
|
* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_class_init),
|
|
(gst_v4lsrc_buffer_finalize):
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Correctly chain up the finalize method.
|
|
|
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2008-02-12 09:24:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
ext/gio/: Add documentation and example code for giostreamsink/giostreamsrc.
|
|
Original commit message from CVS:
|
|
* ext/gio/gstgiostreamsink.c:
|
|
* ext/gio/gstgiostreamsrc.c:
|
|
Add documentation and example code for giostreamsink/giostreamsrc.
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|
* tests/check/pipelines/gio.c: (GST_START_TEST):
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Ask the GMemoryOutputStream for the data instead of assuming that
|
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the pointer to the data stayed the same. It could've been realloc'ed.
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|
|
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2008-02-12 08:55:57 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
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ext/gio/: Make the documentation of giosink/giosrc complete, large parts are based on the gnomevfssink/gnomevfssrc docs.
|
|
Original commit message from CVS:
|
|
* ext/gio/gstgiosink.c:
|
|
* ext/gio/gstgiosrc.c:
|
|
Make the documentation of giosink/giosrc complete, large parts
|
|
are based on the gnomevfssink/gnomevfssrc docs.
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|
|
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2008-02-12 08:13:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
docs/plugins/: Add the GIO documentation again and while at that run make update.
|
|
Original commit message from CVS:
|
|
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
|
* docs/plugins/gst-plugins-base-plugins.interfaces:
|
|
* docs/plugins/gst-plugins-base-plugins.prerequisites:
|
|
* docs/plugins/gst-plugins-base-plugins.signals:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-audioconvert.xml:
|
|
* docs/plugins/inspect/plugin-audiorate.xml:
|
|
* docs/plugins/inspect/plugin-audioresample.xml:
|
|
* docs/plugins/inspect/plugin-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gdp.xml:
|
|
* docs/plugins/inspect/plugin-gio.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.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-queue2.xml:
|
|
* docs/plugins/inspect/plugin-subparse.xml:
|
|
* docs/plugins/inspect/plugin-theora.xml:
|
|
* docs/plugins/inspect/plugin-uridecodebin.xml:
|
|
* docs/plugins/inspect/plugin-videorate.xml:
|
|
* docs/plugins/inspect/plugin-videoscale.xml:
|
|
* docs/plugins/inspect/plugin-volume.xml:
|
|
* docs/plugins/inspect/plugin-vorbis.xml:
|
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Add the GIO documentation again and while at that run make update.
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|
|
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2008-02-11 20:23:44 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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ext/alsa/: Don't use snd_pcm_sw_params_set_xfer_align() if we're compiling against libasound >= 1.0.16, since it's be...
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsa.h: (GST_CHECK_ALSA_VERSION):
|
|
* ext/alsa/gstalsasink.c: (set_swparams):
|
|
* ext/alsa/gstalsasrc.c: (set_swparams), (gst_alsasrc_open):
|
|
Don't use snd_pcm_sw_params_set_xfer_align() if we're compiling
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against libasound >= 1.0.16, since it's been deprecated in
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|
0.10.16, and alignment is always 1 then, apparently. (#512899)
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|
|
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2008-02-11 18:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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gst/playback/: Handle case where we can't create the volume element a bit better (#514307).
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin.c: (gen_audio_element):
|
|
* gst/playback/gstplaysink.c: (gen_audio_chain):
|
|
Handle case where we can't create the volume element a bit
|
|
better (#514307).
|
|
|
|
2008-02-11 18:02:13 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/gnomevfs/: Add support for https protocol. Fixes #510229.
|
|
Original commit message from CVS:
|
|
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_check_get_range):
|
|
* ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
|
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Add support for https protocol. Fixes #510229.
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|
|
|
2008-02-11 17:03:18 +0000 Alan Peevers <peeves@pacbell.net>
|
|
|
|
ext/alsa/gstalsasink.c: Take appropriate lock when calling alsa methods.
|
|
Original commit message from CVS:
|
|
2008-02-11 Julien Moutte <julien@fluendo.com>
|
|
Patch by: Alan Peevers <peeves@pacbell.net>
|
|
* ext/alsa/gstalsasink.c: (gst_alsasink_delay): Take appropriate
|
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lock when calling alsa methods.
|
|
|
|
2008-02-11 13:03:13 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Bump rank of jpeg and png typefinders, which will return maximum probability in ...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
Bump rank of jpeg and png typefinders, which will return maximum
|
|
probability in the most common cases (thus short-circuiting more
|
|
expensive typefinders like the mp3 one for these two quite common
|
|
image types).
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|
|
|
2008-02-11 09:48:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
ext/theora/theoraparse.c: Fix long description of the theora parser to be more verbose than just the type name.
|
|
Original commit message from CVS:
|
|
* ext/theora/theoraparse.c:
|
|
Fix long description of the theora parser to be more verbose than just
|
|
the type name.
|
|
|
|
2008-02-11 06:47:50 +0000 Branko Čibej <brane@xbc.nu>
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|
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sys/xvimage/xvimagesink.c: Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
|
|
Original commit message from CVS:
|
|
Patch by: Branko Čibej <brane at xbc dot nu>
|
|
* sys/xvimage/xvimagesink.c:
|
|
Fix build of xvimagesink if we don't have XShm, e.g. on Mac OS X.
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Fixes bug #515654.
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|
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2008-02-09 10:41:36 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
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|
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gst/playback/gstplaybasebin.c: Set is_dynamic as True if there are elements with both request and sometimes src pad t...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c:
|
|
Set is_dynamic as True if there are elements with both request
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and sometimes src pad templates instead of breaking out when it
|
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finds the first pad template that is a src.
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2008-02-08 18:17:51 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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tests/examples/seek/seek.c: Add some stream switching and volume gui for playbin2.
|
|
Original commit message from CVS:
|
|
* tests/examples/seek/seek.c: (stop_cb), (clear_streams),
|
|
(update_streams), (video_combo_cb), (audio_combo_cb),
|
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(text_combo_cb), (volume_spinbutton_changed_cb), (main):
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Add some stream switching and volume gui for playbin2.
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|
|
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2008-02-08 17:47:37 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst/playback/gstplay-marshal.list: Added marshal for streamselector Tags.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplay-marshal.list:
|
|
Added marshal for streamselector Tags.
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* gst/playback/gstplaybasebin.c: (set_active_source):
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Streamselector now selects pads based on the pad object instead of its
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|
name.
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* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
|
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(init_group), (gst_play_bin_init), (get_group), (get_tags),
|
|
(gst_play_bin_get_video_tags), (gst_play_bin_get_audio_tags),
|
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(gst_play_bin_get_text_tags),
|
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(gst_play_bin_set_current_video_stream),
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(gst_play_bin_set_current_audio_stream),
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(gst_play_bin_set_current_text_stream),
|
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(gst_play_bin_set_property), (gst_play_bin_get_property),
|
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(pad_added_cb), (pad_removed_cb), (autoplug_select_cb):
|
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Remove option to mute streams with the current-a/v/t property, we have
|
|
this functionality in the flags.
|
|
Add signals to notify when the number of A/V/T channels changed.
|
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Add action signals to get tags for the A/V/T streams.
|
|
Implement setting the current A/V/T stream.
|
|
Rearrange some things to simplify stream selection.
|
|
Implement volume.
|
|
* gst/playback/gstplaysink.c: (gst_play_sink_set_volume),
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|
(gst_play_sink_get_volume), (gst_play_sink_set_property),
|
|
(gst_play_sink_get_property), (gen_video_chain), (gen_audio_chain),
|
|
(activate_vis), (gst_play_sink_reconfigure):
|
|
* gst/playback/gstplaysink.h:
|
|
Add and implement volume setting methods.
|
|
* gst/playback/gststreamselector.c: (gst_selector_pad_class_init),
|
|
(gst_selector_pad_finalize), (gst_selector_pad_get_property),
|
|
(gst_selector_pad_event), (gst_stream_selector_class_init),
|
|
(gst_stream_selector_init), (gst_stream_selector_finalize),
|
|
(gst_stream_selector_set_property),
|
|
(gst_stream_selector_get_property),
|
|
(gst_stream_selector_get_linked_pad),
|
|
(gst_stream_selector_request_new_pad):
|
|
* gst/playback/gststreamselector.h:
|
|
Add pad properties for tags and status of pads.
|
|
Keep tags on pads.
|
|
Make active pad selection based on pad object instead of name.
|
|
|
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2008-02-08 16:10:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
|
|
configure.ac: Revert last change as we now check in gtk-doc.m4 for sed.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Revert last change as we now check in gtk-doc.m4 for sed.
|
|
|
|
2008-02-08 14:54:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
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configure.ac: Find and subst SED when building the docs.
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|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Find and subst SED when building the docs.
|
|
|
|
2008-02-08 14:34:41 +0000 Julien Moutte <julien@moutte.net>
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|
|
|
tests/examples/seek/seek.c: Make sure bus signals are reconnected when pressing STOP and then PLAY again for a parse ...
|
|
Original commit message from CVS:
|
|
2008-02-08 Julien Moutte <julien@fluendo.com>
|
|
* tests/examples/seek/seek.c: (stop_cb), (connect_bus_signals),
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(main): Make sure bus signals are reconnected when pressing STOP
|
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and then PLAY again for a parse launch pipeline. Fix a ref leak
|
|
on the bus.
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* win32/common/config.h: Updated.
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|
|
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2008-02-08 00:57:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
|
|
configure.ac: Make DISABLE_DEPRECATED defined *only* during CVS, not during pre-releases or releases.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Make DISABLE_DEPRECATED defined *only* during CVS, not during
|
|
pre-releases or releases.
|
|
|
|
2008-02-08 00:45:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is reporting
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* ext/gio/Makefile.am:
|
|
Subst GIO_LDFLAGS to avoid undefined Makefile var error Zaheer is
|
|
reporting
|
|
|
|
2008-02-07 23:40:30 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
docs/plugins/Makefile.am: Add the headers which need scanning for the GIO plugin. The rest of the docs still need mig...
|
|
Original commit message from CVS:
|
|
* docs/plugins/Makefile.am:
|
|
Add the headers which need scanning for the GIO plugin. The rest of
|
|
the docs still need migrating.
|
|
|
|
2008-02-07 23:22:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
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Add gio in a few more places.
|
|
Original commit message from CVS:
|
|
* ext/Makefile.am:
|
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* tests/check/Makefile.am:
|
|
* tests/check/pipelines/.cvsignore:
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|
Add gio in a few more places.
|
|
|
|
2008-02-07 23:18:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
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Move gio plugin from -bad and mark as experimental.
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|
Original commit message from CVS:
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* configure.ac:
|
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* ext/Makefile.am:
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* tests/check/Makefile.am:
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Move gio plugin from -bad and mark as experimental.
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|
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2008-02-07 22:39:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
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gst-libs/gst/interfaces/: Comment out a couple of other things which break the build when
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/mixeroptions.c:
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|
* gst-libs/gst/interfaces/mixertrack.c:
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Comment out a couple of other things which break the build when
|
|
GST_DISABLE_DEPRECATED isn't on but -Werror is.
|
|
|
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2008-02-07 18:28:29 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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docs/libs/gst-plugins-base-libs-sections.txt: Fix pbutils header.
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
Fix pbutils header.
|
|
|
|
2008-02-07 18:07:41 +0000 Christian Schaller <uraeus@gnome.org>
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|
|
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* gst-plugins-base.spec.in:
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|
commit spec file update which includes all the split .pc files
|
|
Original commit message from CVS:
|
|
commit spec file update which includes all the split .pc files
|
|
|
|
2008-02-07 12:17:49 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst-libs/gst/rtsp/gstrtspmessage.c: Fix compiler warning.
|
|
Original commit message from CVS:
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|
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
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Fix compiler warning.
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|
|
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2008-02-07 11:00:45 +0000 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
gst-libs/gst/sdp/gstsdpmessage.c: Clear the addrinfo struct using memset. Fixes #514937.
|
|
Original commit message from CVS:
|
|
Patch by: Peter Kjellerstedt <pkj at axis com>
|
|
* gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
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Clear the addrinfo struct using memset. Fixes #514937.
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|
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2008-02-06 15:07:30 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/tcp/gstfdset.h: Remove unused field to same some memory.
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|
Original commit message from CVS:
|
|
* gst/tcp/gstfdset.h:
|
|
Remove unused field to same some memory.
|
|
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
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Mark action signals as such.
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|
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2008-02-06 13:35:58 +0000 Michael Smith <msmith@xiph.org>
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|
|
|
ext/theora/theoradec.c: Increment granulepos for new-bitstream versions appropriately.
|
|
Original commit message from CVS:
|
|
* ext/theora/theoradec.c: (_theora_granule_frame),
|
|
(_inc_granulepos):
|
|
Increment granulepos for new-bitstream versions appropriately.
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Fixes #514623.
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|
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2008-02-04 11:51:31 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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tests/examples/seek/seek.c: Remove obsolete stream_time reset after flushing seek, core does that automatically now.
|
|
Original commit message from CVS:
|
|
* tests/examples/seek/seek.c: (do_seek),
|
|
(rate_spinbutton_changed_cb), (update_streams), (main):
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|
Remove obsolete stream_time reset after flushing seek, core does that
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|
automatically now.
|
|
Improve accuracy of speed spinbutton.
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Only do playbin2 stuff when we actually use it.
|
|
|
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2008-02-02 17:29:32 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/Makefile.am: Revert previous change of the test environment's GST_PLUGIN_PATH.
|
|
Original commit message from CVS:
|
|
* tests/check/Makefile.am:
|
|
Revert previous change of the test environment's GST_PLUGIN_PATH.
|
|
The problem is not with the plugins, but with element factories
|
|
and only occurs if elements are split out from existing plugins
|
|
or if plugins change name (see #512740).
|
|
|
|
2008-02-02 15:32:23 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/Makefile.am: Fix the tests environment's GST_PLUGIN_PATH: we want the directory with the core's plugins f...
|
|
Original commit message from CVS:
|
|
* tests/check/Makefile.am:
|
|
Fix the tests environment's GST_PLUGIN_PATH: we want the directory
|
|
with the core's plugins first and our local build directories last,
|
|
since we might be building against an installed core, and that
|
|
core's plugin directory may contain older or other versions of
|
|
our own -base plugins, but we really do want to test our local
|
|
ones (if there are multiple plugins or element factories with the
|
|
same name, those inspected last will trump those read in earlier).
|
|
Fixes #512740 for the most part.
|
|
|
|
2008-02-02 07:13:15 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
Use gmtime_r if available as gmtime is not MT-safe.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
|
|
Use gmtime_r if available as gmtime is not MT-safe.
|
|
Fixes bug #511810.
|
|
|
|
2008-02-02 06:52:41 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
|
|
Cast glong to time_t as time_t might have a different type on
|
|
other platforms, like FreeBSD, and we get a compiler warning
|
|
otherwise. Fixes bug #511825.
|
|
|
|
2008-02-01 16:44:21 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybin2.c: Remove stream-info, we going for something easier.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
|
|
(get_group), (get_n_pads), (gst_play_bin_get_property),
|
|
(pad_added_cb), (no_more_pads_cb), (perform_eos),
|
|
(autoplug_select_cb), (deactivate_group):
|
|
Remove stream-info, we going for something easier.
|
|
Refactor getting the current group.
|
|
Implement getting the number of audio/video/text streams.
|
|
* gst/playback/gststreamselector.c:
|
|
(gst_stream_selector_class_init), (gst_stream_selector_init),
|
|
(gst_stream_selector_get_property),
|
|
(gst_stream_selector_request_new_pad),
|
|
(gst_stream_selector_release_pad):
|
|
* gst/playback/gststreamselector.h:
|
|
Add property for number of pads.
|
|
* tests/examples/seek/seek.c: (set_scale), (update_flag),
|
|
(vis_toggle_cb), (audio_toggle_cb), (video_toggle_cb),
|
|
(text_toggle_cb), (update_streams), (msg_async_done),
|
|
(msg_state_changed), (main):
|
|
Block slider callback when updating the slider position.
|
|
Add gui elements for controlling playbin2.
|
|
Add callback for async_done that updates position/duration.
|
|
|
|
2008-02-01 12:56:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
docs/plugins/: First round of plugin docs cleansups.
|
|
Original commit message from CVS:
|
|
* docs/plugins/Makefile.am:
|
|
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
|
* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
|
* docs/plugins/gst-plugins-base-plugins.interfaces:
|
|
* docs/plugins/gst-plugins-base-plugins.prerequisites:
|
|
First round of plugin docs cleansups.
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.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-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gdp.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.xml:
|
|
* docs/plugins/inspect/plugin-libvisual.xml:
|
|
* docs/plugins/inspect/plugin-ogg.xml:
|
|
* docs/plugins/inspect/plugin-pango.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-video4linux.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:
|
|
Regenerate.
|
|
* ext/ogg/Makefile.am:
|
|
* ext/ogg/gstoggmux.c:
|
|
* ext/ogg/gstoggmux.h:
|
|
Add header for oggmux. the c-file needs a doc blob still.
|
|
|
|
2008-02-01 11:09:16 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
|
|
|
|
Add gst_rtp_buffer_set_extension_data()
|
|
Original commit message from CVS:
|
|
Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
|
|
* gst-libs/gst/rtp/gstrtpbuffer.c:
|
|
(gst_rtp_buffer_set_extension_data):
|
|
* gst-libs/gst/rtp/gstrtpbuffer.h:
|
|
* tests/check/libs/rtp.c: (GST_START_TEST), (rtp_suite):
|
|
Add gst_rtp_buffer_set_extension_data()
|
|
Add a unit test for this addition. Fixes #511478.
|
|
API: GstRTPBuffer:gst_rtp_buffer_set_extension_data()
|
|
|
|
2008-01-31 17:18:46 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/app/gstappsink.c: Really clean up the queue instead of just unreffing all buffers in it.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_dispose):
|
|
Really clean up the queue instead of just unreffing all buffers
|
|
in it.
|
|
* gst-libs/gst/app/gstappsrc.c: (gst_app_src_base_init),
|
|
(gst_app_src_class_init), (gst_app_src_init),
|
|
(gst_app_src_dispose), (gst_app_src_finalize):
|
|
Fix dispose/finalize.
|
|
|
|
2008-01-30 15:34:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
ext/gio/: Use async variants of the close stream functions to prevent blocking for a long time there and add some mor...
|
|
Original commit message from CVS:
|
|
* ext/gio/gstgiobasesink.c: (close_stream_cb),
|
|
(gst_gio_base_sink_stop), (gst_gio_base_sink_event),
|
|
(gst_gio_base_sink_render), (gst_gio_base_sink_set_stream):
|
|
* ext/gio/gstgiobasesrc.c: (close_stream_cb),
|
|
(gst_gio_base_src_stop), (gst_gio_base_src_create),
|
|
(gst_gio_base_src_set_stream):
|
|
Use async variants of the close stream functions to prevent blocking
|
|
for a long time there and add some more sanity checks for a correct
|
|
stream.
|
|
|
|
2008-01-30 14:42:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
configure.ac: Back to CVS
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Back to CVS
|
|
|
|
=== release 0.10.17 ===
|
|
|
|
2008-01-30 14:19:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
* ChangeLog:
|
|
* NEWS:
|
|
* RELEASE:
|
|
* configure.ac:
|
|
* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.xml:
|
|
* docs/plugins/inspect/plugin-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-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gdp.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.xml:
|
|
* docs/plugins/inspect/plugin-libvisual.xml:
|
|
* docs/plugins/inspect/plugin-ogg.xml:
|
|
* docs/plugins/inspect/plugin-pango.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-video4linux.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/config.h:
|
|
Release 0.10.17
|
|
Original commit message from CVS:
|
|
Release 0.10.17
|
|
|
|
2008-01-30 13:45:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/interfaces/: Also remove the conditional registration of the signals that disappeared with the ABI chang...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/mixeroptions.c:
|
|
* gst-libs/gst/interfaces/mixertrack.c:
|
|
Also remove the conditional registration of the signals
|
|
that disappeared with the ABI change in 0.10.14
|
|
|
|
2008-01-30 12:28:59 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/rtsp/gstrtspconnection.c: Revert patch to gstrtspconnection.c for brown paper bag release of -base. Re-o...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
Revert patch to gstrtspconnection.c for brown paper bag
|
|
release of -base. Re-opens: #511825
|
|
|
|
2008-01-30 12:20:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/mixeroptions.h:
|
|
* gst-libs/gst/interfaces/mixertrack.h:
|
|
Change the way these deprecated function pointers are removed
|
|
so that the compiled ABI is unconditionally smaller. This
|
|
sets in stone an ABI break that actually occurred when the
|
|
things were deprecated in 0.10.14, which seems to be the best
|
|
fix as the only known users are oss-mixer and sunaudio-mixer in
|
|
gst-plugins-good.
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Fixes: #513018
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2008-01-30 12:19:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
|
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gst-libs/gst/interfaces/: Change the way these deprecated function pointers are removed so that the compiled ABI is u...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/mixeroptions.h:
|
|
* gst-libs/gst/interfaces/mixertrack.h:
|
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Change the way these deprecated function pointers are removed
|
|
so that the compiled ABI is unconditionally smaller. This
|
|
sets in stone an ABI break that actually occurred when the
|
|
things were deprecated in 0.10.14, which seems to be the best
|
|
fix as the only known users are oss-mixer and sunaudio-mixer in
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gst-plugins-good.
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2008-01-30 11:43:53 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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win32/common/libgstpbutils.def: Export the two new _get_type() functions which are needed by the python bindings.
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|
Original commit message from CVS:
|
|
* win32/common/libgstpbutils.def:
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Export the two new _get_type() functions which are needed
|
|
by the python bindings.
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|
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2008-01-29 09:59:03 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
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gst-libs/gst/rtsp/gstrtspconnection.c: Cast glong to time_t as time_t might have a different type on other platforms,...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c: (add_date_header):
|
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Cast glong to time_t as time_t might have a different type on
|
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other platforms, like FreeBSD, and we get a compiler warning
|
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otherwise. Fixes bug #511825.
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2008-01-29 09:47:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst-libs/gst/audio/gstaudiofilter.c: Initialize the GstRingerBuffer class to get it's debug category initialized. gst...
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|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstaudiofilter.c:
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(gst_audio_filter_class_init):
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Initialize the GstRingerBuffer class to get it's debug category
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initialized. gst_ring_buffer_parse_caps() uses the ringbuffer debug
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category and otherwise we get some g_critical(). Fixes bug #512334.
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2008-01-28 23:35:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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configure.ac: Back to CVS
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Original commit message from CVS:
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* configure.ac:
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Back to CVS
|
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=== release 0.10.16 ===
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2008-01-28 23:31:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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* ChangeLog:
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* NEWS:
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* RELEASE:
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* configure.ac:
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* docs/plugins/gst-plugins-base-plugins.args:
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* docs/plugins/gst-plugins-base-plugins.hierarchy:
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* docs/plugins/gst-plugins-base-plugins.interfaces:
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* docs/plugins/gst-plugins-base-plugins.prerequisites:
|
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* docs/plugins/gst-plugins-base-plugins.signals:
|
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* docs/plugins/inspect/plugin-adder.xml:
|
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* docs/plugins/inspect/plugin-alsa.xml:
|
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* docs/plugins/inspect/plugin-audioconvert.xml:
|
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* docs/plugins/inspect/plugin-audiorate.xml:
|
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* docs/plugins/inspect/plugin-audioresample.xml:
|
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* docs/plugins/inspect/plugin-audiotestsrc.xml:
|
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* docs/plugins/inspect/plugin-cdparanoia.xml:
|
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* docs/plugins/inspect/plugin-decodebin.xml:
|
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* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
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* docs/plugins/inspect/plugin-gdp.xml:
|
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* docs/plugins/inspect/plugin-gnomevfs.xml:
|
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* docs/plugins/inspect/plugin-libvisual.xml:
|
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* docs/plugins/inspect/plugin-ogg.xml:
|
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* docs/plugins/inspect/plugin-pango.xml:
|
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* docs/plugins/inspect/plugin-subparse.xml:
|
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* docs/plugins/inspect/plugin-tcp.xml:
|
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* docs/plugins/inspect/plugin-theora.xml:
|
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* docs/plugins/inspect/plugin-typefindfunctions.xml:
|
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* docs/plugins/inspect/plugin-video4linux.xml:
|
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* docs/plugins/inspect/plugin-videorate.xml:
|
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* docs/plugins/inspect/plugin-videoscale.xml:
|
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* docs/plugins/inspect/plugin-videotestsrc.xml:
|
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* docs/plugins/inspect/plugin-volume.xml:
|
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* docs/plugins/inspect/plugin-vorbis.xml:
|
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* docs/plugins/inspect/plugin-ximagesink.xml:
|
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* docs/plugins/inspect/plugin-xvimagesink.xml:
|
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* gst-plugins-base.doap:
|
|
* win32/common/config.h:
|
|
Release 0.10.16
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Original commit message from CVS:
|
|
Release 0.10.16
|
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|
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2008-01-28 22:15:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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* common:
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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:
|
|
* po/en_GB.po:
|
|
* po/es.po:
|
|
* po/fi.po:
|
|
* po/hu.po:
|
|
* po/it.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/pl.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
* po/zh_CN.po:
|
|
Update .po files
|
|
Original commit message from CVS:
|
|
Update .po files
|
|
|
|
2008-01-22 15:37:49 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstrtpbuffer.c: Fix typos and wrong extension check. Fixes #511274.
|
|
Original commit message from CVS:
|
|
Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
|
|
* gst-libs/gst/rtp/gstrtpbuffer.c:
|
|
(gst_rtp_buffer_get_extension_data):
|
|
Fix typos and wrong extension check. Fixes #511274.
|
|
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|
2008-01-18 00:03:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
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po/sk.po: Oops - add new sk.po mentioned in the LINGUAS I just committed
|
|
Original commit message from CVS:
|
|
* po/sk.po:
|
|
Oops - add new sk.po mentioned in the LINGUAS I just committed
|
|
|
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2008-01-17 22:31:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
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po/LINGUAS: Add ca translation to the disted list.
|
|
Original commit message from CVS:
|
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* po/LINGUAS:
|
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Add ca translation to the disted list.
|
|
* win32/vs6/libgstsdp.dsp:
|
|
Convert line endings to CRLF
|
|
|
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2008-01-17 21:58:53 +0000 Sébastien Moutte <sebastien@moutte.net>
|
|
|
|
win32/MANIFEST: Add win32/vs6/libgstrtsp.dsp to MANIFEST
|
|
Original commit message from CVS:
|
|
* win32/MANIFEST:
|
|
Add win32/vs6/libgstrtsp.dsp to MANIFEST
|
|
|
|
2008-01-16 05:40:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
Update for API changes in GIO and require GIO 2.15.2 for this.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* tests/check/pipelines/gio.c: (GST_START_TEST):
|
|
Update for API changes in GIO and require GIO 2.15.2 for this.
|
|
|
|
2008-01-14 22:20:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
win32/common/: Add new API declarations
|
|
Original commit message from CVS:
|
|
* win32/common/libgstsdp.def:
|
|
* win32/common/libgstvideo.def:
|
|
Add new API declarations
|
|
|
|
2008-01-14 17:00:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
ext/theora/: Take a 2nd stab at handling libtheora granulepos changes in the decoder and parser by inspecting the bit...
|
|
Original commit message from CVS:
|
|
* ext/theora/gsttheoradec.h:
|
|
* ext/theora/gsttheoraparse.h:
|
|
* ext/theora/theoradec.c:
|
|
* ext/theora/theoraparse.c:
|
|
Take a 2nd stab at handling libtheora granulepos changes in the decoder
|
|
and parser by inspecting the bitstream version of the incoming data.
|
|
|
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2008-01-14 13:11:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
Provide one pkg-config file for every gst-plugins-base library.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* pkgconfig/Makefile.am:
|
|
* pkgconfig/gstreamer-audio-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-audio.pc.in:
|
|
* pkgconfig/gstreamer-cdda-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-cdda.pc.in:
|
|
* pkgconfig/gstreamer-fft-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-fft.pc.in:
|
|
* pkgconfig/gstreamer-floatcast-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-floatcast.pc.in:
|
|
* pkgconfig/gstreamer-interfaces-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-interfaces.pc.in:
|
|
* pkgconfig/gstreamer-netbuffer-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-netbuffer.pc.in:
|
|
* pkgconfig/gstreamer-pbutils-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-pbutils.pc.in:
|
|
* pkgconfig/gstreamer-riff-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-riff.pc.in:
|
|
* pkgconfig/gstreamer-rtp-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-rtp.pc.in:
|
|
* pkgconfig/gstreamer-rtsp-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-rtsp.pc.in:
|
|
* pkgconfig/gstreamer-sdp-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-sdp.pc.in:
|
|
* pkgconfig/gstreamer-tag-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-tag.pc.in:
|
|
* pkgconfig/gstreamer-video-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-video.pc.in:
|
|
Provide one pkg-config file for every gst-plugins-base library.
|
|
This makes linking to those libraries much more intuitive and
|
|
provides standard pkg-config behaviour for them. Fixes bug #499697.
|
|
|
|
2008-01-14 01:19:34 +0000 David Schleef <ds@schleef.org>
|
|
|
|
gst/videoscale/vs_4tap.c: Fix valgrind error on 4tap scaling method.
|
|
Original commit message from CVS:
|
|
* gst/videoscale/vs_4tap.c:
|
|
Fix valgrind error on 4tap scaling method.
|
|
|
|
2008-01-13 21:40:45 +0000 Sébastien Moutte <sebastien@moutte.net>
|
|
|
|
gst-libs/gst/sdp/gstsdpmessage.c: Include Winsock2.h for VS6 and use a different way initialize hints structure so it...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address):
|
|
Include Winsock2.h for VS6 and use a different way initialize
|
|
hints structure so it can build with VS6.
|
|
* win32/MANIFEST:
|
|
* win32/vs6/libgstsdp.dsp:
|
|
* win32/common/libgstsdp.def:
|
|
Add new files for libgstsdp.
|
|
* win32/vs6/grammar.dsp:
|
|
Copy pbutils-enumtypes* from win32/common to pbutils sources folder.
|
|
* win32/vs6/gst_plugins_base.dsw:
|
|
* win32/vs6/libgstdecodebin.dsp:
|
|
* win32/vs6/libgstdecodebin2.dsp:
|
|
* win32/vs6/libgstplaybin.dsp:
|
|
* win32/vs6/libgstvolume.dsp:
|
|
Add new dependencies to the link list.
|
|
|
|
2008-01-13 17:24:49 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
win32/common/: Update/Add generated files in the win32 build directory.
|
|
Original commit message from CVS:
|
|
2008-01-13 Julien Moutte <julien@fluendo.com>
|
|
* win32/common/config.h:
|
|
* win32/common/gstrtsp-enumtypes.c: (gst_rtsp_result_get_type),
|
|
(gst_rtsp_event_get_type), (gst_rtsp_family_get_type),
|
|
(gst_rtsp_state_get_type), (gst_rtsp_version_get_type),
|
|
(gst_rtsp_method_get_type), (gst_rtsp_auth_method_get_type),
|
|
(gst_rtsp_header_field_get_type),
|
|
(gst_rtsp_status_code_get_type):
|
|
* win32/common/interfaces-enumtypes.c:
|
|
(gst_color_balance_type_get_type), (gst_mixer_type_get_type),
|
|
(gst_mixer_message_type_get_type), (gst_mixer_flags_get_type),
|
|
(gst_mixer_track_flags_get_type),
|
|
(gst_tuner_channel_flags_get_type):
|
|
* win32/common/multichannel-enumtypes.c:
|
|
(gst_audio_channel_position_get_type):
|
|
* win32/common/pbutils-enumtypes.c:
|
|
(gst_install_plugins_return_get_type):
|
|
* win32/common/pbutils-enumtypes.h: Update/Add generated files
|
|
in the win32 build directory.
|
|
|
|
2008-01-12 23:24:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
tests/check/Makefile.am: Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
|
|
Original commit message from CVS:
|
|
* tests/check/Makefile.am:
|
|
Fix CFLAGS to also pull in the gstcheck cflags from AM_CFLAGS.
|
|
* tests/check/elements/audiorate.c: (do_perfect_stream_test):
|
|
* tests/check/elements/playbin.c:
|
|
* tests/check/libs/mixer.c: (test_element_interface_supported),
|
|
(gst_implements_interface_init):
|
|
* tests/check/libs/rtp.c: (GST_START_TEST):
|
|
Fix various assignment type mismatches.
|
|
|
|
2008-01-12 23:08:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
Add test to see if hstrerror is available or if we need libresolv (Solaris) for it, then use it in libgstrtsp.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* gst-libs/gst/rtsp/Makefile.am:
|
|
Add test to see if hstrerror is available or if we need libresolv
|
|
(Solaris) for it, then use it in libgstrtsp.
|
|
|
|
2008-01-12 14:54:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/tag/Makefile.am: Fix include path order
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/Makefile.am:
|
|
Fix include path order
|
|
|
|
2008-01-11 17:15:23 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
* gst-libs/gst/pbutils/.gitignore:
|
|
Ignore more and make buildbot happy
|
|
Original commit message from CVS:
|
|
Ignore more and make buildbot happy
|
|
|
|
2008-01-11 16:18:10 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
gst-libs/gst/pbutils/install-plugins.*: Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping for bi...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/pbutils/install-plugins.c:
|
|
(gst_install_plugins_context_copy),
|
|
(gst_install_plugins_context_get_type):
|
|
* gst-libs/gst/pbutils/install-plugins.h:
|
|
Add GBoxed GType for GstInstallPluginsContext, this eases the wrapping
|
|
for bindings.
|
|
|
|
2008-01-11 15:48:11 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
ext/theora/theoradec.c: Adapt for post-alpha meaning of granulepos, when we have a newer version of libtheora.
|
|
Original commit message from CVS:
|
|
* ext/theora/theoradec.c: (gst_theora_dec_class_init),
|
|
(_theora_granule_frame), (_theora_granule_start_time),
|
|
(theora_dec_sink_convert), (theora_dec_decode_buffer):
|
|
Adapt for post-alpha meaning of granulepos, when we
|
|
have a newer version of libtheora.
|
|
* ext/theora/theoraenc.c: (gst_theora_enc_class_init),
|
|
(theora_enc_get_ogg_packet_end_time), (theora_enc_sink_event),
|
|
(theora_enc_is_discontinuous), (theora_enc_chain):
|
|
Likewise.
|
|
* tests/check/Makefile.am:
|
|
Link libtheora into theoraenc test so we can check which version of
|
|
libtheora we're testing against.
|
|
* tests/check/pipelines/theoraenc.c: (check_libtheora),
|
|
(check_buffer_granulepos),
|
|
(check_buffer_granulepos_from_starttime), (GST_START_TEST),
|
|
(theoraenc_suite):
|
|
Adapt tests to check the values that are now defined for theora; make
|
|
the tests backwards-adapt the passed values if we're running against an
|
|
old libtheora.
|
|
Fixes #497964
|
|
|
|
2008-01-10 17:55:53 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/audio/: Ref audio clock class from a thread-safe context to make sure however unlikely that may be in pr...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_class_init):
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
(gst_base_audio_src_class_init):
|
|
Ref audio clock class from a thread-safe context to make sure
|
|
we're not bit by GObjects lack of thread-safety here (#349410),
|
|
however unlikely that may be in practice.
|
|
|
|
2008-01-10 12:22:46 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
autogen.sh: Add -Wno-portability to the automake parameters to stop warnings about GNU make extensions being used. We...
|
|
Original commit message from CVS:
|
|
* autogen.sh:
|
|
Add -Wno-portability to the automake parameters to stop warnings
|
|
about GNU make extensions being used. We require GNU make in almost
|
|
every Makefile anyway.
|
|
* configure.ac:
|
|
Use AM_PROG_CC_C_O as a compiler that accepts both -c and -o
|
|
at the same time is required for per target flags.
|
|
|
|
2008-01-08 21:10:02 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/tag/gsttagdemux.c: Post an error message if we can't pull as many bytes as we need for the tag. This mak...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/gsttagdemux.c: (gst_tag_demux_pull_start_tag):
|
|
Post an error message if we can't pull as many bytes as we need
|
|
for the tag. This makes sure the user gets to see a proper error
|
|
message if a file with a partial ID3 tag is fed to decodebin, and
|
|
not a 'no ID3 tag demuxer' error, which would be confusing
|
|
(see #508138).
|
|
|
|
2008-01-08 20:59:20 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/pbutils/descriptions.c: Add description strings for ID3, APE, and ICY tags.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/pbutils/descriptions.c: (formats):
|
|
Add description strings for ID3, APE, and ICY tags.
|
|
|
|
2008-01-08 20:48:00 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/playback/gstdecodebin.c: Make sure we error out correctly if we can't activate one of the elements we've added. ...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (try_to_link_1):
|
|
Make sure we error out correctly if we can't activate one of
|
|
the elements we've added. Fixes #508138.
|
|
|
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2008-01-07 13:59:43 +0000 Bastien Nocera <hadess@hadess.net>
|
|
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ext/alsa/gstalsamixer.c: Use snd_mixer_selem_set_{playback|capture}_volume_all() if the volume is the same for all ch...
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Original commit message from CVS:
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Patch by: Bastien Nocera <hadess at hadess net>
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* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume),
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(check_if_volumes_are_the_same), (gst_alsa_mixer_set_volume):
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Use snd_mixer_selem_set_{playback|capture}_volume_all() if
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the volume is the same for all channels. This works around
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some problem in alsa that leaves us with inconsistent state
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for some reason (#486840).
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2008-01-07 13:19:50 +0000 Jerone Young <jerone@gmail.com>
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ext/alsa/gstalsamixer.c: If there's no mixer track by the name of 'Master' or 'Front', check if there's one called 'P...
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Original commit message from CVS:
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Patch by: Jerone Young <jerone at gmail com>
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* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer):
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If there's no mixer track by the name of 'Master' or 'Front',
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check if there's one called 'PCM' before trying the generic
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fallback logic (fixes #506928, where we pick 'Mic' as master
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track for the AD1984 card in a Thinkpad T61/X61 laptop).
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2008-01-07 11:40:04 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstplay-enum.*: Add enums for configuration flags.
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Original commit message from CVS:
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* gst/playback/gstplay-enum.c:
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(register_gst_autoplug_select_result),
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(gst_autoplug_select_result_get_type), (register_gst_play_flags),
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(gst_play_flags_get_type):
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* gst/playback/gstplay-enum.h:
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Add enums for configuration flags.
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* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
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(init_group), (gst_play_bin_init), (gst_play_bin_set_property),
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(gst_play_bin_get_property), (no_more_pads_cb),
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(autoplug_select_cb), (gst_play_bin_change_state):
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Merge mode with flags.
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Add more property getters/setters, defaults and docs.
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Add properties to get number of audio/video/text streams.
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Create sink object in _init so that we can always rely on it being
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there.
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* gst/playback/gstplaysink.c: (gst_play_sink_init),
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(gen_video_chain), (gen_audio_chain), (gen_vis_chain),
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(activate_vis), (gst_play_sink_reconfigure),
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(gst_play_sink_set_flags), (gst_play_sink_get_flags),
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(gst_play_sink_change_state):
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* gst/playback/gstplaysink.h:
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Use flags to configure the sink pipelines.
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Add tee before audio pipeline so that we can use it for visualisations.
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Start working on integrating visualisations.
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Remove mode, we can do everything with the flags now.
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Add method to configue the sink pipeline.
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2008-01-06 16:36:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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Update to GMemoryInputStream API changes in GLib SVN and require gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
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Original commit message from CVS:
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* configure.ac:
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* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
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* tests/check/pipelines/gio.c: (free_input), (GST_START_TEST):
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Update to GMemoryInputStream API changes in GLib SVN and require
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gio-2.0 >= 2.15.1 for this. Fixes bug #507584.
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We can also report the duration for every GSeekable, not only
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GFileInputStream and GMemoryInputStream.
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2008-01-06 14:39:19 +0000 Tim-Philipp Müller <tim@centricular.net>
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tests/check/pipelines/theoraenc.c: Turn these functions into macros so we can see right away where the failure occured.
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Original commit message from CVS:
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* tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
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(check_buffer_timestamp), (check_buffer_duration):
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Turn these functions into macros so we can see right away
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where the failure occured.
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2008-01-05 22:25:05 +0000 Julien Moutte <julien@moutte.net>
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sys/xvimage/xvimagesink.c: Add debugging information to understand how X calculates the stride for XvImages.
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Original commit message from CVS:
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2008-01-05 Julien Moutte <julien@fluendo.com>
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new): Add
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debugging information to understand how X calculates the stride
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for XvImages.
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2008-01-03 20:33:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/volume/: Use GstAudioFilter as base class for the volume element instead of plain GstBaseTransform.
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Original commit message from CVS:
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* gst/volume/Makefile.am:
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* gst/volume/gstvolume.c: (volume_choose_func),
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(gst_volume_base_init), (gst_volume_class_init), (gst_volume_init),
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(volume_setup):
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* gst/volume/gstvolume.h:
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Use GstAudioFilter as base class for the volume element instead of
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plain GstBaseTransform.
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2008-01-03 07:17:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst-libs/gst/audio/gstaudiofilter.c: Don't set element details for the abstract GstAudioFilter class.
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Original commit message from CVS:
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* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type):
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Don't set element details for the abstract GstAudioFilter class.
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2008-01-02 12:09:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst-libs/gst/audio/gstaudiofilter.c: Implement get_unit_size() vmethod of GstBaseTransform.
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Original commit message from CVS:
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* gst-libs/gst/audio/gstaudiofilter.c:
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(gst_audio_filter_class_init), (gst_audio_filter_get_unit_size):
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Implement get_unit_size() vmethod of GstBaseTransform.
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2008-01-01 12:53:48 +0000 Edward Hervey <bilboed@bilboed.com>
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gst-libs/gst/pbutils/: Use glib-enum generator to have a proper enum GType for
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Original commit message from CVS:
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* gst-libs/gst/pbutils/Makefile.am:
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* gst-libs/gst/pbutils/pbutils.h:
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Use glib-enum generator to have a proper enum GType for
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GST_TYPE_INSTALL_PLUGINS_RETURN so we can easily wrap it in bindings.
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2008-01-01 01:21:47 +0000 David Schleef <ds@schleef.org>
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tests/check/: Reenable theoraenc test, which fails on the buildbot but not locally.
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Original commit message from CVS:
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* tests/check/Makefile.am:
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* tests/check/pipelines/theoraenc.c:
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Reenable theoraenc test, which fails on the buildbot but
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not locally.
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2007-12-31 21:31:01 +0000 David Schleef <ds@schleef.org>
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docs/: Add *-undeclared.txt to fix buildbot.
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Original commit message from CVS:
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* docs/libs/.cvsignore:
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* docs/plugins/.cvsignore:
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Add *-undeclared.txt to fix buildbot.
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2007-12-31 20:45:28 +0000 David Schleef <ds@schleef.org>
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tests/check/Makefile.am: Second attempt at disabling theoraenc test long enough to get buildbot to compile -base.
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Original commit message from CVS:
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* tests/check/Makefile.am:
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Second attempt at disabling theoraenc test long enough to
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get buildbot to compile -base.
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2007-12-31 20:21:20 +0000 David Schleef <ds@schleef.org>
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tests/check/pipelines/theoraenc.c: Disable theoraenc test long enough to get the buildbot to compile a recent -base.
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Original commit message from CVS:
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* tests/check/pipelines/theoraenc.c:
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Disable theoraenc test long enough to get the buildbot to
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compile a recent -base.
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2007-12-31 13:17:29 +0000 Wim Taymans <wim.taymans@gmail.com>
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tests/examples/seek/seek.c: Make sure we reset the slider value to 0.0 without racing against a possible g_idle that ...
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|
Original commit message from CVS:
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* tests/examples/seek/seek.c: (stop_cb):
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Make sure we reset the slider value to 0.0 without racing against a
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possible g_idle that sets it to something else.
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2007-12-31 00:32:53 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
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sys/ximage/ximagesink.c: fix typo
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Original commit message from CVS:
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* sys/ximage/ximagesink.c:
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fix typo
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2007-12-30 19:21:16 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/rtsp/gstrtspdefs.*: Add Location header so that we can start implementing redirects.
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|
Original commit message from CVS:
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* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
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* gst-libs/gst/rtsp/gstrtspdefs.h:
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Add Location header so that we can start implementing redirects.
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See #506025.
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2007-12-29 20:55:39 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
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gst/subparse/gstssaparse.c: combine if's
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|
Original commit message from CVS:
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|
* gst/subparse/gstssaparse.c:
|
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combine if's
|
|
|
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2007-12-29 19:23:59 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
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gst/subparse/gstssaparse.c: remove duplicate log message
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|
Original commit message from CVS:
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* gst/subparse/gstssaparse.c:
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remove duplicate log message
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2007-12-29 17:29:17 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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Update to latest API changes in GLib/GIO and require at least gio-2.0 2.15.0 for this.
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|
Original commit message from CVS:
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* configure.ac:
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* ext/gio/gstgio.c:
|
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* ext/gio/gstgio.h:
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* ext/gio/gstgiobasesink.h:
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* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_get_size):
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* ext/gio/gstgiobasesrc.h:
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* ext/gio/gstgiosink.c: (gst_gio_sink_start):
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* ext/gio/gstgiosink.h:
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* ext/gio/gstgiosrc.h:
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* ext/gio/gstgiostreamsink.h:
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* ext/gio/gstgiostreamsrc.h:
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* tests/check/pipelines/gio.c:
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Update to latest API changes in GLib/GIO and require at least
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gio-2.0 2.15.0 for this.
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* ext/gio/Makefile.am:
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Add GST_PLUGIN_LDFLAGS to LDFLAGS.
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2007-12-29 16:23:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
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ext/libvisual/visual.c: Fix 'xyz may be used uninitialized' compiler warnings caused by broken g_assert_not_reached()...
|
|
Original commit message from CVS:
|
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* ext/libvisual/visual.c: (gst_visual_chain):
|
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Fix 'xyz may be used uninitialized' compiler warnings caused
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by broken g_assert_not_reached() macro in GLib-2.15.x and don't
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abort() in any case but properly report the error.
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2007-12-28 09:00:27 +0000 Wim Taymans <wim.taymans@gmail.com>
|
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|
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gst/playback/gstplaybin2.c: Code cleanups.
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|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
|
|
(gst_play_bin_finalize), (gst_play_bin_set_uri),
|
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(gst_play_bin_set_suburi), (gst_play_bin_set_property),
|
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(gst_play_bin_get_property), (pad_removed_cb), (drained_cb),
|
|
(autoplug_select_cb), (activate_group), (deactivate_group),
|
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(setup_next_source), (save_current_group),
|
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(gst_play_bin_change_state):
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Code cleanups.
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Remove next-uri, we can use the uri property just fine.
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Fix some crasher.
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Unref uridecodebin when switching.
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Fix going to READY.
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* gst/playback/gstplaysink.c: (gst_play_sink_class_init),
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(gst_play_sink_init), (gst_play_sink_dispose),
|
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(gst_play_sink_finalize), (gst_play_sink_vis_unblocked),
|
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(gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
|
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(gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
|
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(gst_play_sink_set_property), (gst_play_sink_get_property),
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(gen_video_chain), (gen_text_element), (gen_audio_chain),
|
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(gen_vis_element), (gst_play_sink_get_mode),
|
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(gst_play_sink_set_mode), (gst_play_sink_set_flags),
|
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(gst_play_sink_get_flags), (gst_play_sink_request_pad),
|
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(gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
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(gst_play_sink_change_state):
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* gst/playback/gstplaysink.h:
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Add some locking to make things threadsafe.
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* gst/playback/test7.c: (about_to_finish_cb):
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Fix test.
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2007-12-22 12:06:47 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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gst/videoscale/gstvideoscale.c: Don't claim to be able to handle/transform caps that can't really be handled by the c...
|
|
Original commit message from CVS:
|
|
* gst/videoscale/gstvideoscale.c: (gst_video_scale_set_property),
|
|
(gst_video_scale_get_property), (gst_video_scale_transform_caps),
|
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(gst_video_scale_transform):
|
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Don't claim to be able to handle/transform caps that can't really
|
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be handled by the currently selected scaling method (here: RGB or
|
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packed YUV with 4-tap method). Also add locking to method property.
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* tests/check/pipelines/simple-launch-lines.c: (setup_pipeline),
|
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(test_basetransform_based):
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Some test pipelines for the above (not entirely valgrind clean yet
|
|
apparently).
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|
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2007-12-22 05:19:00 +0000 David Schleef <ds@schleef.org>
|
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gst-libs/gst/video/video.*: Add additional RGBA and RGB-24 video formats.
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|
Original commit message from CVS:
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* gst-libs/gst/video/video.c:
|
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* gst-libs/gst/video/video.h:
|
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Add additional RGBA and RGB-24 video formats.
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2007-12-21 22:46:56 +0000 Tim-Philipp Müller <tim@centricular.net>
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tests/check/: Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be deprecated in the future (see #498924).
|
|
Original commit message from CVS:
|
|
* tests/check/elements/playbin.c: (test_sink_usage_video_only_stream),
|
|
(test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
|
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(test_suburi_error_wrongproto), (test_missing_primary_decoder):
|
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* tests/check/libs/cddabasesrc.c: (GST_START_TEST), (GST_START_TEST),
|
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(cddabasesrc_suite):
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Don't use GST_PLUGIN_DEFINE_STATIC, it's not portable and will be
|
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deprecated in the future (see #498924).
|
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2007-12-21 22:26:47 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gststreamselector.c: Don't leak event.
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|
Original commit message from CVS:
|
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* gst/playback/gststreamselector.c: (gst_selector_pad_event):
|
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Don't leak event.
|
|
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2007-12-20 19:43:25 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
|
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gst-libs/gst/riff/riff-read.c: Use GST_ROUND_UP_2 macro
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|
Original commit message from CVS:
|
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* gst-libs/gst/riff/riff-read.c:
|
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Use GST_ROUND_UP_2 macro
|
|
|
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2007-12-20 17:13:37 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/.cvsignore: Ignore more.
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|
Original commit message from CVS:
|
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* gst/playback/.cvsignore:
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Ignore more.
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|
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2007-12-20 10:41:29 +0000 Tim-Philipp Müller <tim@centricular.net>
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Make switching off of subtitles work. To avoid all kind of problems with unlinking of the subtitle input, we just kee...
|
|
Original commit message from CVS:
|
|
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init):
|
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* gst/playback/gstplaybasebin.c: (set_subtitles_visible),
|
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(set_active_source):
|
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* gst/playback/gstplaybasebin.h:
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* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
|
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(setup_sinks), (playbin_set_subtitles_visible):
|
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Make switching off of subtitles work. To avoid all kind of
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problems with unlinking of the subtitle input, we just keep
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the subtitle inputs linked as they are and tell textoverlay
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not to render them. Fixes #373011.
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Other subtitle switching issues (esp. when there are both
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external and in-stream subtitles) remain. They'll be solved
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in playbin2.
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2007-12-18 16:21:35 +0000 Wim Taymans <wim.taymans@gmail.com>
|
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gst/playback/gststreamselector.c: Init the pad segment too.
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|
Original commit message from CVS:
|
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* gst/playback/gststreamselector.c: (gst_selector_pad_init):
|
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Init the pad segment too.
|
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2007-12-18 15:56:51 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
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gst-libs/gst/audio/gstaudiosink.c: Improve debug output.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
|
|
(gst_audioringbuffer_open_device),
|
|
(gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
|
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(gst_audioringbuffer_release), (gst_audioringbuffer_start),
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(gst_audioringbuffer_pause), (gst_audioringbuffer_stop),
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(gst_audio_sink_create_ringbuffer):
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Improve debug output.
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* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_start),
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(gst_ring_buffer_pause), (gst_ring_buffer_delay):
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Prevent some functions from doing things and failing when the
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ringbuffer is not yet acquired.
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2007-12-18 15:32:49 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst-libs/gst/interfaces/interfaces.h: Also remove interfaces.h from CVS as it is not needed anymore.
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Original commit message from CVS:
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* gst-libs/gst/interfaces/interfaces.h:
|
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Also remove interfaces.h from CVS as it is not needed anymore.
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2007-12-18 15:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst-libs/gst/interfaces/Makefile.am: interfaces.h is not used anymore so remove it from the build process.
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|
Original commit message from CVS:
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* gst-libs/gst/interfaces/Makefile.am:
|
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interfaces.h is not used anymore so remove it from the build
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process.
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2007-12-18 01:01:23 +0000 David Schleef <ds@schleef.org>
|
|
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gst/videotestsrc/gstvideotestsrc.*: Add a "blink" pattern. Turn on the pain. Apologies. It's useful for testing ve...
|
|
Original commit message from CVS:
|
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* gst/videotestsrc/gstvideotestsrc.c:
|
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* gst/videotestsrc/gstvideotestsrc.h:
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Add a "blink" pattern. Turn on the pain. Apologies. It's useful
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for testing vertical refresh synchronization.
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2007-12-18 00:13:26 +0000 David Schleef <ds@schleef.org>
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Add new GstVideFormat enum and write a bunch of helper functions based around it.
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Original commit message from CVS:
|
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* docs/libs/gst-plugins-base-libs-sections.txt:
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* gst-libs/gst/video/video.c:
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* gst-libs/gst/video/video.h:
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Add new GstVideFormat enum and write a bunch of helper functions
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based around it.
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2007-12-17 23:41:14 +0000 Tim-Philipp Müller <tim@centricular.net>
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Makefile.am: Use new common/win32.mak.
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Original commit message from CVS:
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* Makefile.am:
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Use new common/win32.mak.
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|
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2007-12-17 16:44:51 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstbaseaudiosrc.c: Add debug info.
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Original commit message from CVS:
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* gst-libs/gst/audio/gstbaseaudiosrc.c:
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|
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
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Add debug info.
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When going from PLAYING to PAUSED, pause the ringbuffer before calling
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the parent state change function, just like the audiosink, because the
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parent waits for the element to finish its processing before completing
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the state change. This makes going to PAUSED a lot snappier.
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When going from READY to PAUSED, don't allow the ringbuffer to start
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yet.
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2007-12-17 00:01:00 +0000 Edward Hervey <bilboed@bilboed.com>
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gst-libs/gst/riff/riff-media.c: Yet another fix for broken software that produce files with an empty blockalign field...
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Original commit message from CVS:
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* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
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Yet another fix for broken software that produce files with an empty
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blockalign field. Instead of completely failing, make a second attempt
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at guessing the width/depth by looking at strf->size.
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2007-12-16 23:52:58 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/: Turn a few g_assert_not_reached() into g_return_val_if_reached() to avoid compiler warnings (#503930).
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Original commit message from CVS:
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* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_do_seek),
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(gst_cdda_base_src_handle_track_seek), (gst_cdda_base_src_create):
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* gst-libs/gst/pbutils/install-plugins.c:
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(gst_install_plugins_spawn_child), (gst_install_plugins_supported):
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* gst-libs/gst/pbutils/missing-plugins.c:
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(gst_missing_plugin_message_get_installer_detail),
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(gst_missing_encoder_installer_detail_new):
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* gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_send):
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* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
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Turn a few g_assert_not_reached() into g_return_val_if_reached() to
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avoid compiler warnings (#503930).
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2007-12-16 23:46:16 +0000 Edward Hervey <bilboed@bilboed.com>
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gst-libs/gst/riff/riff-media.c: Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC for jpeg video...
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|
Original commit message from CVS:
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* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
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Be apologetic of software that use the 'jpeg' instead of 'JPEG' FOURCC
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for jpeg video streams.
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Add the 'avc1'/'AVC1' fourcc mapping for h264, same software-comment as
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for the above modification.
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2007-12-15 17:27:48 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/interfaces/xoverlay.c: More guards (we don't want klass to end up being NULL).
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Original commit message from CVS:
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* gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_expose),
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(gst_x_overlay_handle_events):
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More guards (we don't want klass to end up being NULL).
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2007-12-15 03:40:34 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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Use new gst_base_transform_set_gap_aware() function as volume correctly handles GST_BUFFER_FLAG_GAP. Require core 0.1...
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|
Original commit message from CVS:
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* configure.ac:
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* gst/volume/gstvolume.c: (gst_volume_init):
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Use new gst_base_transform_set_gap_aware() function as volume
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correctly handles GST_BUFFER_FLAG_GAP. Require core 0.10.15.1
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for this.
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2007-12-14 19:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
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tests/examples/seek/seek.c: Don't go to READY on EOS as this avoids testing of seeking and restarting after EOS, use ...
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|
Original commit message from CVS:
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* tests/examples/seek/seek.c: (msg_segment_done), (main):
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Don't go to READY on EOS as this avoids testing of seeking and
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restarting after EOS, use the stop button when you want to READY.
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Don't try to do a flushing seek in segment-done, it does not make
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sense to use this for gapless playback and is not needed.
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2007-12-14 18:46:12 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstqueue2.c: Use separate timers for input and output rates.
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|
Original commit message from CVS:
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* gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_finalize),
|
|
(reset_rate_timer), (update_in_rates), (update_out_rates),
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(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
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(gst_queue_chain), (gst_queue_loop):
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Use separate timers for input and output rates.
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Pause measuring the output rate when we block for more data.
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See #503262.
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|
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2007-12-14 16:23:06 +0000 Christian Schaller <uraeus@gnome.org>
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* gst/speexresample/Makefile.am:
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update spec file and add two missing files for disting
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|
Original commit message from CVS:
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|
update spec file and add two missing files for disting
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|
|
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2007-12-14 09:24:55 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst/playback/gstqueue2.c: Pause the timer to measure the input rate when we block because the queue is filled. See #5...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstqueue2.c: (gst_queue_chain):
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Pause the timer to measure the input rate when we block because the
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|
queue is filled. See #503262.
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|
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2007-12-13 15:54:00 +0000 Peter Kjellerstedt <pkj@axis.com>
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|
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gst-libs/gst/rtsp/gstrtspconnection.c: Close control sockets. Fixes #503440.
|
|
Original commit message from CVS:
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|
Patch by: Peter Kjellerstedt <pkj at axis com>
|
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* gst-libs/gst/rtsp/gstrtspconnection.c:
|
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(gst_rtsp_connection_free):
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Close control sockets. Fixes #503440.
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|
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2007-12-13 12:31:38 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst/playback/gstdecodebin2.c: Expose the right pad in the right place with the right element.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin2.c: (analyze_new_pad), (connect_pad):
|
|
Expose the right pad in the right place with the right element.
|
|
|
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2007-12-13 11:40:10 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/pbutils/descriptions.c: Add description for 'private' dts caps (who come up with that name?).
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/pbutils/descriptions.c: (formats):
|
|
Add description for 'private' dts caps (who come up with that name?).
|
|
|
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2007-12-13 10:10:35 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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Makefile.am: Add check-exports target and run it with 'make check'.
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|
Original commit message from CVS:
|
|
* Makefile.am:
|
|
Add check-exports target and run it with 'make check'.
|
|
* configure.ac:
|
|
Be stricter about what we export in our libraries: change regexp so that
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we only export _gst_foo(), but not __gst_foo().
|
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* gst-libs/gst/cdda/base64.h: (rfc822_binary):
|
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* gst-libs/gst/cdda/sha1.h: (sha_init), (sha_update), (sha_final):
|
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Change internal functions to __gst_foo so they dont' get exported.
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* win32/common/libgstaudio.def:
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Add missing symbols.
|
|
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2007-12-11 21:18:57 +0000 David Schleef <ds@schleef.org>
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|
|
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* ChangeLog:
|
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ChangeLog: remove conflict markers
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|
Original commit message from CVS:
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|
ChangeLog: remove conflict markers
|
|
|
|
2007-12-11 17:14:13 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/gnomevfs/: Use gst_tag_freeform_string_to_utf8() here, which also takes into account any character sets specified...
|
|
Original commit message from CVS:
|
|
* ext/gnomevfs/Makefile.am:
|
|
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_unicodify):
|
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Use gst_tag_freeform_string_to_utf8() here, which also takes
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into account any character sets specified by the user via
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environment variables.
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2007-12-10 15:21:41 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst/audioconvert/Makefile.am: Also link to libm.
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/Makefile.am:
|
|
Also link to libm.
|
|
|
|
2007-12-10 15:13:55 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/riff/riff-media.c: No need for floating point operations here. avoids having to link against the math li...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
|
|
No need for floating point operations here. avoids having to link
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against the math library too.
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|
|
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2007-12-10 11:16:25 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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Add one or two missing formats. Generate ADPCM description dynamically depending on layout/format.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/pbutils/descriptions.c: (formats),
|
|
(format_info_get_desc):
|
|
* tests/check/libs/pbutils.c: (GST_START_TEST), (caps_strings),
|
|
(GST_START_TEST):
|
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Add one or two missing formats. Generate ADPCM description
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dynamically depending on layout/format.
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|
|
|
2007-12-09 04:28:38 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
configure.ac: Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Don't define GST_DISABLE_DEPRECATED for releases. Fixes #498181.
|
|
|
|
2007-12-08 18:38:39 +0000 Robin Stocker <robin.stocker@gmx.ch>
|
|
|
|
gst/subparse/gstsubparse.c: Some .srt files start with chunk number 0 and not chunk number 1, recognise and accept th...
|
|
Original commit message from CVS:
|
|
Patch by: Robin Stocker <robin dot stocker at gmx dot ch>
|
|
* gst/subparse/gstsubparse.c: (gst_sub_parse_data_format_autodetect):
|
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Some .srt files start with chunk number 0 and not chunk number 1,
|
|
recognise and accept those as well (fixes #502497).
|
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* tests/check/elements/subparse.c: (srt_input), (srt_input0),
|
|
(test_src):
|
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Add unit test for the above.
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|
|
|
2007-12-06 12:08:21 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplay-enum.*: Add missing files.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplay-enum.c:
|
|
(register_gst_autoplug_select_result),
|
|
(gst_autoplug_select_result_get_type):
|
|
* gst/playback/gstplay-enum.h:
|
|
Add missing files.
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|
|
|
2007-12-05 17:11:48 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/Makefile.am: Group decodebin2 and uridecodebin into the same plugin so that they can share the GEnumType.
|
|
Original commit message from CVS:
|
|
* gst/playback/Makefile.am:
|
|
Group decodebin2 and uridecodebin into the same plugin so that they
|
|
can share the GEnumType.
|
|
* gst/playback/gstdecodebin2.c: (_gst_array_accumulator),
|
|
(_gst_select_accumulator), (gst_decode_bin_class_init),
|
|
(gst_decode_bin_init), (gst_decode_bin_autoplug_sort),
|
|
(gst_decode_bin_autoplug_select), (gst_decode_bin_autoplug_add),
|
|
(analyze_new_pad), (connect_pad), (gst_decode_bin_plugin_init):
|
|
Add signal to sort factories instead of the more awkward autoplug-select
|
|
signal.
|
|
Modify autoplug_select so that we can try, skip or expose the
|
|
autopluggin of an element on a pad.
|
|
* gst/playback/gstfactorylists.c: (compare_ranks),
|
|
(decoders_filter), (sinks_filter), (gst_factory_list_is_type),
|
|
(element_filter), (gst_factory_list_get_elements),
|
|
(gst_factory_list_debug), (gst_factory_list_filter):
|
|
* gst/playback/gstfactorylists.h:
|
|
Simplify the API, allow getting elements based on mask.
|
|
* gst/playback/gstplay-marshal.list:
|
|
Add some more marshallers.
|
|
* gst/playback/gstplaybin2.c: (init_group), (gst_play_bin_init),
|
|
(gst_play_bin_finalize), (pad_removed_cb), (autoplug_factories_cb),
|
|
(autoplug_select_cb), (activate_group):
|
|
Add support for managing non-raw sinks by providing a custom element and
|
|
sink list to decodebin2.
|
|
Try to plug non-raw sinks when decodebin2 using autoplug-select of
|
|
decodebin2.
|
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* gst/playback/gstplaysink.c: (gen_video_chain), (gen_audio_chain),
|
|
(gst_play_sink_set_mode), (gst_play_sink_request_pad):
|
|
* gst/playback/gstplaysink.h:
|
|
Add support for raw and non-raw sinks.
|
|
Add support to force sinks selected by playbin2.
|
|
Don't plug raw converters for non-raw sinks.
|
|
* gst/playback/gsturidecodebin.c: (_gst_array_accumulator),
|
|
(_gst_select_accumulator), (gst_uri_decode_bin_class_init),
|
|
(proxy_autoplug_select_signal), (gst_uri_decode_bin_plugin_init),
|
|
(plugin_init):
|
|
Use right accumulators.
|
|
Proxy new signal.
|
|
|
|
2007-12-03 13:47:00 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstbasertppayload.c: Use runnning time as the base time instead of the timestamp.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
|
|
Use runnning time as the base time instead of the timestamp.
|
|
Spotted by Saur on IRC.
|
|
|
|
2007-12-03 11:32:30 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
gst-libs/gst/riff/riff-media.c: Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
|
|
Add 'WVC1' codec mapping for Windows Media VC-1 video codec.
|
|
|
|
2007-12-03 10:58:14 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/ogg/gstoggdemux.c: If we find a new serial number but it does not contain a BOS page, make sure we initialize the...
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_bisect_forward_serialno),
|
|
(gst_ogg_demux_read_chain):
|
|
If we find a new serial number but it does not contain a BOS page, make
|
|
sure we initialize the chain to NULL because else we will try to scan it
|
|
and crash. Fixes #500763
|
|
|
|
2007-11-30 17:47:15 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/: Refactor some common code to filter factories and check caps compat.
|
|
Original commit message from CVS:
|
|
* gst/playback/Makefile.am:
|
|
* gst/playback/gstfactorylists.c: (compare_ranks), (print_feature),
|
|
(get_feature_array), (decoders_filter), (sinks_filter),
|
|
(gst_factory_list_get_decoders), (gst_factory_list_get_sinks),
|
|
(gst_factory_list_filter):
|
|
* gst/playback/gstfactorylists.h:
|
|
Refactor some common code to filter factories and check caps compat.
|
|
* gst/playback/gstdecodebin.c:
|
|
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
|
|
(gst_decode_bin_init), (gst_decode_bin_dispose),
|
|
(gst_decode_bin_autoplug_continue),
|
|
(gst_decode_bin_autoplug_factories),
|
|
(gst_decode_bin_autoplug_select), (analyze_new_pad),
|
|
(find_compatibles):
|
|
* gst/playback/gstplaybin.c:
|
|
* gst/playback/gstplaybin2.c: (gst_play_bin_class_init),
|
|
(gst_play_bin_init), (gst_play_bin_finalize),
|
|
(autoplug_factories_cb), (activate_group):
|
|
* gst/playback/gstqueue2.c:
|
|
* gst/playback/gsturidecodebin.c: (proxy_unknown_type_signal),
|
|
(proxy_autoplug_continue_signal),
|
|
(proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
|
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(proxy_drained_signal):
|
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Add some more debug info and use factor filtering code.
|
|
|
|
2007-11-26 13:19:46 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
configure.ac: Add QuickTime Wrapper plug-in.
|
|
Original commit message from CVS:
|
|
2007-11-26 Julien Moutte <julien@fluendo.com>
|
|
* configure.ac: Add QuickTime Wrapper plug-in.
|
|
* gst/speexresample/gstspeexresample.c:
|
|
(gst_speex_resample_push_drain), (gst_speex_resample_process): Fix
|
|
build on Mac OS X Leopard. Incorrect printf format arguments.
|
|
* sys/Makefile.am:
|
|
* sys/qtwrapper/Makefile.am:
|
|
* sys/qtwrapper/audiodecoders.c:
|
|
(qtwrapper_audio_decoder_base_init),
|
|
(qtwrapper_audio_decoder_class_init),
|
|
(qtwrapper_audio_decoder_init),
|
|
(clear_AudioStreamBasicDescription), (fill_indesc_mp3),
|
|
(fill_indesc_aac), (fill_indesc_samr), (fill_indesc_generic),
|
|
(make_samr_magic_cookie), (open_decoder),
|
|
(qtwrapper_audio_decoder_sink_setcaps), (process_buffer_cb),
|
|
(qtwrapper_audio_decoder_chain),
|
|
(qtwrapper_audio_decoder_sink_event),
|
|
(qtwrapper_audio_decoders_register):
|
|
* sys/qtwrapper/codecmapping.c: (audio_caps_from_string),
|
|
(fourcc_to_caps):
|
|
* sys/qtwrapper/codecmapping.h:
|
|
* sys/qtwrapper/imagedescription.c: (image_description_for_avc1),
|
|
(image_description_for_mp4v), (image_description_from_stsd_buffer),
|
|
(image_description_from_codec_data):
|
|
* sys/qtwrapper/imagedescription.h:
|
|
* sys/qtwrapper/qtutils.c: (get_name_info_from_component),
|
|
(get_output_info_from_component), (dump_avcc_atom),
|
|
(dump_image_description), (dump_codec_decompress_params),
|
|
(addSInt32ToDictionary), (dump_cvpixel_buffer),
|
|
(DestroyAudioBufferList), (AllocateAudioBufferList):
|
|
* sys/qtwrapper/qtutils.h:
|
|
* sys/qtwrapper/qtwrapper.c: (plugin_init):
|
|
* sys/qtwrapper/qtwrapper.h:
|
|
* sys/qtwrapper/videodecoders.c:
|
|
(qtwrapper_video_decoder_base_init),
|
|
(qtwrapper_video_decoder_class_init),
|
|
(qtwrapper_video_decoder_init), (qtwrapper_video_decoder_finalize),
|
|
(fill_image_description), (new_image_description), (close_decoder),
|
|
(open_decoder), (qtwrapper_video_decoder_sink_setcaps),
|
|
(decompressCb), (qtwrapper_video_decoder_chain),
|
|
(qtwrapper_video_decoder_sink_event),
|
|
(qtwrapper_video_decoders_register): Initial import of QuickTime
|
|
wrapper jointly developped by Songbird authors (Pioneers of the
|
|
Inevitable) and Fluendo.
|
|
|
|
2007-11-26 12:25:55 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
|
|
gst/: Add GAP-flag support.
|
|
Original commit message from CVS:
|
|
* gst/audiotestsrc/gstaudiotestsrc.c:
|
|
* gst/volume/gstvolume.c:
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|
* gst/volume/gstvolume.h:
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Add GAP-flag support.
|
|
|
|
2007-11-26 08:43:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/speexresample/: Update speex resampler to latest SVN. We're now down to only the changes noted in README again.
|
|
Original commit message from CVS:
|
|
* gst/speexresample/README:
|
|
* gst/speexresample/arch.h:
|
|
* gst/speexresample/resample.c: (resampler_basic_direct_single),
|
|
(resampler_basic_direct_double),
|
|
(resampler_basic_interpolate_single),
|
|
(resampler_basic_interpolate_double),
|
|
(speex_resampler_process_native), (speex_resampler_process_float),
|
|
(speex_resampler_process_int),
|
|
(speex_resampler_process_interleaved_float),
|
|
(speex_resampler_process_interleaved_int),
|
|
(speex_resampler_get_input_latency),
|
|
(speex_resampler_get_output_latency):
|
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* gst/speexresample/speex_resampler.h:
|
|
Update speex resampler to latest SVN. We're now down to only the
|
|
changes noted in README again.
|
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* gst/speexresample/speex_resampler_wrapper.h:
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|
* gst/speexresample/gstspeexresample.c:
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(gst_speex_resample_push_drain), (gst_speex_resample_query):
|
|
Adjust to API changes.
|
|
|
|
2007-11-24 15:02:01 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
tests/examples/seek/seek.c: Increase the range of the rate selector as I would like to test QOS behavior at higher fo...
|
|
Original commit message from CVS:
|
|
2007-11-24 Julien MOUTTE <julien@moutte.net>
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|
* tests/examples/seek/seek.c: (main): Increase the range of the
|
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rate selector as I would like to test QOS behavior at higher
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forward and reverse playback speed like say 64x.
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|
|
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2007-11-23 10:21:31 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
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gst/speexresample/gstspeexresample.c: Only post the latency message if we have a resampler state already.
|
|
Original commit message from CVS:
|
|
* gst/speexresample/gstspeexresample.c:
|
|
(gst_speex_resample_update_state):
|
|
Only post the latency message if we have a resampler state already.
|
|
|
|
2007-11-23 10:21:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/audioresample/gstaudioresample.c: Implement latency query.
|
|
Original commit message from CVS:
|
|
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
|
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(audioresample_query), (audioresample_query_type),
|
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(gst_audioresample_set_property):
|
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Implement latency query.
|
|
|
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2007-11-23 10:01:33 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
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gst/speexresample/gstspeexresample.c: Also post GST_MESSAGE_LATENCY if the latency changes.
|
|
Original commit message from CVS:
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|
* gst/speexresample/gstspeexresample.c:
|
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(gst_speex_resample_update_state):
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Also post GST_MESSAGE_LATENCY if the latency changes.
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|
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2007-11-23 08:48:50 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
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gst/speexresample/: Add functions to push the remaining samples and to get the latency of the resampler. These will g...
|
|
Original commit message from CVS:
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|
* gst/speexresample/resample.c: (speex_resampler_get_latency),
|
|
(speex_resampler_drain_float), (speex_resampler_drain_int),
|
|
(speex_resampler_drain_interleaved_float),
|
|
(speex_resampler_drain_interleaved_int):
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* gst/speexresample/speex_resampler.h:
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* gst/speexresample/speex_resampler_wrapper.h:
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Add functions to push the remaining samples and to get the latency
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of the resampler. These will get added to Speex SVN in this or a
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slightly changed form at some point too and should get merged then
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|
again.
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* gst/speexresample/gstspeexresample.c: (gst_speex_resample_init),
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(gst_speex_resample_init_state),
|
|
(gst_speex_resample_transform_size),
|
|
(gst_speex_resample_push_drain), (gst_speex_resample_event),
|
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(gst_speex_fix_output_buffer), (gst_speex_resample_process),
|
|
(gst_speex_resample_query), (gst_speex_resample_query_type):
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Drop the prepending zeroes and output the remaining samples on EOS.
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Also properly implement the latency query for this. speexresample
|
|
should be completely ready for production use now.
|
|
|
|
2007-11-21 18:02:21 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
|
gst-libs/gst/audio/gstbaseaudiosink.c: Our EOS time contains the base_time, _wait_eos() expects a running_time so we ...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_drain):
|
|
Our EOS time contains the base_time, _wait_eos() expects a running_time
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so we have to subtract the base_time again before calling the function.
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This fixes an EOS regression where the base_time was added twice and EOS
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took longer and longer in certain situations.
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Fixes #498767.
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|
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2007-11-21 13:04:17 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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Expose methods for some object properties so that subclasses can more easily configure them.
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
|
|
(gst_base_audio_sink_set_provide_clock),
|
|
(gst_base_audio_sink_get_provide_clock),
|
|
(gst_base_audio_sink_set_slave_method),
|
|
(gst_base_audio_sink_get_slave_method),
|
|
(gst_base_audio_sink_set_property),
|
|
(gst_base_audio_sink_get_property), (gst_base_audio_sink_drain),
|
|
(gst_base_audio_sink_none_slaving),
|
|
(gst_base_audio_sink_handle_slaving):
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|
* gst-libs/gst/audio/gstbaseaudiosink.h:
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|
Expose methods for some object properties so that subclasses can more
|
|
easily configure them.
|
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Added slave method none, that completely disables slaving to the
|
|
internal clock.
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API: gst_base_audio_sink_set_provide_clock()
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|
API: gst_base_audio_sink_get_provide_clock()
|
|
API: gst_base_audio_sink_set_slave_method()
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|
API: gst_base_audio_sink_get_slave_method()
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* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
(gst_base_audio_src_set_provide_clock),
|
|
(gst_base_audio_src_get_provide_clock),
|
|
(gst_base_audio_src_set_property),
|
|
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.h:
|
|
Expose methods for some object properties so that subclasses can more
|
|
easily configure them.
|
|
API: gst_base_audio_src_set_provide_clock()
|
|
API: gst_base_audio_src_get_provide_clock()
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|
|
|
2007-11-21 10:18:56 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/speexresample/README: Add README explaining where the resampling code was taken from and which changes were done.
|
|
Original commit message from CVS:
|
|
* gst/speexresample/README:
|
|
Add README explaining where the resampling code was taken from
|
|
and which changes were done.
|
|
* gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
|
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(speex_free):
|
|
Use g_malloc() and friends instead of malloc() to achieve higher
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|
portability and define the functions inline.
|
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* gst/speexresample/speex_resampler.h:
|
|
Add back some useless preprocessor stuff to keep the diff between
|
|
our version and the one from the Speex SVN repository lower.
|
|
|
|
2007-11-20 20:23:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/speexresample/gstspeexresample.c: Some small cleanup and addition of a TODO item.
|
|
Original commit message from CVS:
|
|
* gst/speexresample/gstspeexresample.c:
|
|
(gst_speex_fix_output_buffer), (gst_speex_resample_transform):
|
|
Some small cleanup and addition of a TODO item.
|
|
|
|
2007-11-20 12:56:00 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/speexresample/Makefile.am: Add missing file.
|
|
Original commit message from CVS:
|
|
* gst/speexresample/Makefile.am:
|
|
Add missing file.
|
|
|
|
2007-11-20 07:53:56 +0000 Joe Peterson <lavajoe@gentoo.org>
|
|
|
|
gst-libs/gst/sdp/gstsdpmessage.c: Fix compilation on FreeBSD (Gentoo). Fixes #498228.
|
|
Original commit message from CVS:
|
|
Patch by: Joe Peterson <lavajoe at gentoo dot org>
|
|
* gst-libs/gst/sdp/gstsdpmessage.c:
|
|
Fix compilation on FreeBSD (Gentoo). Fixes #498228.
|
|
|
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2007-11-20 07:47:27 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
Add speexresample to the docs and while at that do a make update.
|
|
Original commit message from CVS:
|
|
* docs/plugins/Makefile.am:
|
|
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
|
|
* docs/plugins/gst-plugins-bad-plugins.args:
|
|
* docs/plugins/gst-plugins-bad-plugins.signals:
|
|
* docs/plugins/inspect/plugin-bz2.xml:
|
|
* docs/plugins/inspect/plugin-cdxaparse.xml:
|
|
* docs/plugins/inspect/plugin-dtsdec.xml:
|
|
* docs/plugins/inspect/plugin-equalizer.xml:
|
|
* docs/plugins/inspect/plugin-faac.xml:
|
|
* docs/plugins/inspect/plugin-faad.xml:
|
|
* docs/plugins/inspect/plugin-filter.xml:
|
|
* docs/plugins/inspect/plugin-freeze.xml:
|
|
* docs/plugins/inspect/plugin-gio.xml:
|
|
* docs/plugins/inspect/plugin-gsm.xml:
|
|
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
|
|
* docs/plugins/inspect/plugin-h264parse.xml:
|
|
* docs/plugins/inspect/plugin-modplug.xml:
|
|
* docs/plugins/inspect/plugin-mpeg2enc.xml:
|
|
* docs/plugins/inspect/plugin-musepack.xml:
|
|
* docs/plugins/inspect/plugin-musicbrainz.xml:
|
|
* docs/plugins/inspect/plugin-nsfdec.xml:
|
|
* docs/plugins/inspect/plugin-replaygain.xml:
|
|
* docs/plugins/inspect/plugin-soundtouch.xml:
|
|
* docs/plugins/inspect/plugin-spcdec.xml:
|
|
* docs/plugins/inspect/plugin-spectrum.xml:
|
|
* docs/plugins/inspect/plugin-speed.xml:
|
|
* docs/plugins/inspect/plugin-tta.xml:
|
|
* docs/plugins/inspect/plugin-videosignal.xml:
|
|
* docs/plugins/inspect/plugin-xingheader.xml:
|
|
* docs/plugins/inspect/plugin-xvid.xml:
|
|
* gst/speexresample/gstspeexresample.h:
|
|
Add speexresample to the docs and while at that do a make update.
|
|
|
|
2007-11-20 07:30:30 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/speexresample/gstspeexresample.c: If the resampler gives less output samples than expected adjust the output buff...
|
|
Original commit message from CVS:
|
|
* gst/speexresample/gstspeexresample.c:
|
|
(gst_speex_fix_output_buffer), (gst_speex_resample_process):
|
|
If the resampler gives less output samples than expected
|
|
adjust the output buffer and print a warning.
|
|
|
|
2007-11-20 07:02:45 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
Add resample element based on the Speex resampling algorithm.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* gst/speexresample/arch.h:
|
|
* gst/speexresample/fixed_generic.h:
|
|
* gst/speexresample/gstspeexresample.c:
|
|
(gst_speex_resample_base_init), (gst_speex_resample_class_init),
|
|
(gst_speex_resample_init), (gst_speex_resample_start),
|
|
(gst_speex_resample_stop), (gst_speex_resample_get_unit_size),
|
|
(gst_speex_resample_transform_caps),
|
|
(gst_speex_resample_init_state), (gst_speex_resample_update_state),
|
|
(gst_speex_resample_reset_state), (gst_speex_resample_parse_caps),
|
|
(gst_speex_resample_transform_size), (gst_speex_resample_set_caps),
|
|
(gst_speex_resample_event), (gst_speex_resample_check_discont),
|
|
(gst_speex_resample_process), (gst_speex_resample_transform),
|
|
(gst_speex_resample_set_property),
|
|
(gst_speex_resample_get_property), (plugin_init):
|
|
* gst/speexresample/gstspeexresample.h:
|
|
* gst/speexresample/resample.c: (speex_alloc), (speex_realloc),
|
|
(speex_free), (compute_func), (main), (sinc), (cubic_coef),
|
|
(resampler_basic_direct_single), (resampler_basic_direct_double),
|
|
(resampler_basic_interpolate_single),
|
|
(resampler_basic_interpolate_double), (update_filter),
|
|
(speex_resampler_init), (speex_resampler_init_frac),
|
|
(speex_resampler_destroy), (speex_resampler_process_native),
|
|
(speex_resampler_process_float), (speex_resampler_process_int),
|
|
(speex_resampler_process_interleaved_float),
|
|
(speex_resampler_process_interleaved_int),
|
|
(speex_resampler_set_rate), (speex_resampler_get_rate),
|
|
(speex_resampler_set_rate_frac), (speex_resampler_get_ratio),
|
|
(speex_resampler_set_quality), (speex_resampler_get_quality),
|
|
(speex_resampler_set_input_stride),
|
|
(speex_resampler_get_input_stride),
|
|
(speex_resampler_set_output_stride),
|
|
(speex_resampler_get_output_stride), (speex_resampler_skip_zeros),
|
|
(speex_resampler_reset_mem), (speex_resampler_strerror):
|
|
* gst/speexresample/speex_resampler.h:
|
|
* gst/speexresample/speex_resampler_float.c:
|
|
* gst/speexresample/speex_resampler_int.c:
|
|
* gst/speexresample/speex_resampler_wrapper.h:
|
|
Add resample element based on the Speex resampling algorithm.
|
|
|
|
2007-11-19 12:30:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
tests/check/libs/fft.c: Fix scaling to really have dB instead of something else.
|
|
Original commit message from CVS:
|
|
* tests/check/libs/fft.c: (GST_START_TEST):
|
|
Fix scaling to really have dB instead of something else.
|
|
|
|
2007-11-19 12:08:16 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
tests/examples/seek/seek.c: There's a nice macro to check
|
|
Original commit message from CVS:
|
|
2007-11-19 Julien MOUTTE <julien@moutte.net>
|
|
* tests/examples/seek/seek.c: (main): There's a nice macro to
|
|
check
|
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GTK version, use it.
|
|
|
|
2007-11-19 11:59:20 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
tests/examples/seek/seek.c: Try to support stable version of GTK.
|
|
Original commit message from CVS:
|
|
2007-11-19 Julien MOUTTE <julien@moutte.net>
|
|
* tests/examples/seek/seek.c: (main): Try to support stable version
|
|
of GTK.
|
|
|
|
2007-11-17 15:25:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/playback/: Fix the build + little README update.
|
|
Original commit message from CVS:
|
|
* gst/playback/README:
|
|
* gst/playback/test7.c:
|
|
Fix the build + little README update.
|
|
|
|
2007-11-16 16:02:45 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
tests/examples/seek/seek.c: Add playbin2 seek pipeline.
|
|
Original commit message from CVS:
|
|
* tests/examples/seek/seek.c: (make_playerbin2_pipeline), (main):
|
|
Add playbin2 seek pipeline.
|
|
|
|
2007-11-16 15:44:48 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/: Add playbin2.
|
|
Original commit message from CVS:
|
|
* gst/playback/Makefile.am:
|
|
* gst/playback/gstplayback.c: (plugin_init):
|
|
* gst/playback/test7.c: (update_scale), (warning_cb), (error_cb),
|
|
(eos_cb), (about_to_finish_cb), (main):
|
|
Add playbin2.
|
|
Added gapless playback example.
|
|
* gst/playback/gstplaybasebin.c:
|
|
* gst/playback/gstplaybasebin.h:
|
|
* gst/playback/gstplaybin.c: (gst_play_bin_plugin_init):
|
|
* gst/playback/gstqueue2.c:
|
|
* gst/playback/test.c:
|
|
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
|
|
(pad_removed_cb):
|
|
* gst/playback/gststreaminfo.h:
|
|
Change email.
|
|
* gst/playback/gstplaybin2.c: (gst_play_bin_get_type),
|
|
(gst_play_bin_class_init), (init_group), (gst_play_bin_init),
|
|
(gst_play_bin_dispose), (gst_play_bin_set_uri),
|
|
(gst_play_bin_set_suburi), (gst_play_bin_set_property),
|
|
(gst_play_bin_get_property), (gst_play_bin_handle_message),
|
|
(pad_added_cb), (pad_removed_cb), (no_more_pads_cb), (perform_eos),
|
|
(drained_cb), (unlink_group), (activate_group),
|
|
(setup_next_source), (gst_play_bin_change_state),
|
|
(gst_play_bin2_plugin_init):
|
|
Added raw first version of playbin2. Does chained oggs and gapless
|
|
playback fine. No support for raw sinks yet. No visualisations or
|
|
subtitles yet.
|
|
* gst/playback/gstplaysink.c: (gst_play_sink_get_type),
|
|
(gst_play_sink_class_init), (gst_play_sink_init),
|
|
(gst_play_sink_dispose), (gst_play_sink_vis_unblocked),
|
|
(gst_play_sink_vis_blocked), (gst_play_sink_set_video_sink),
|
|
(gst_play_sink_set_audio_sink), (gst_play_sink_set_vis_plugin),
|
|
(gst_play_sink_set_property), (gst_play_sink_get_property),
|
|
(post_missing_element_message), (free_chain), (add_chain),
|
|
(activate_chain), (gen_video_chain), (gen_text_element),
|
|
(gen_audio_chain), (gen_vis_element), (gst_play_sink_get_mode),
|
|
(gst_play_sink_set_mode), (gst_play_sink_request_pad),
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(gst_play_sink_release_pad), (gst_play_sink_send_event_to_sink),
|
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(gst_play_sink_send_event), (gst_play_sink_change_state):
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* gst/playback/gstplaysink.h:
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|
Added Element that abstracts the sinks and their pipelines for playbin2.
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|
|
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2007-11-16 15:05:07 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst/playback/gststreamselector.*: Improve streamselector, make it select and unselect the current pad more intelligen...
|
|
Original commit message from CVS:
|
|
* gst/playback/gststreamselector.c: (gst_selector_pad_get_type),
|
|
(gst_selector_pad_class_init), (gst_selector_pad_init),
|
|
(gst_selector_pad_finalize), (gst_selector_pad_reset),
|
|
(gst_selector_pad_get_linked_pads), (gst_selector_pad_event),
|
|
(gst_selector_pad_getcaps), (gst_selector_pad_bufferalloc),
|
|
(gst_selector_pad_chain), (gst_stream_selector_get_type),
|
|
(gst_stream_selector_base_init), (gst_stream_selector_class_init),
|
|
(gst_stream_selector_init), (gst_stream_selector_set_property),
|
|
(gst_stream_selector_get_linked_pad),
|
|
(gst_stream_selector_getcaps),
|
|
(gst_stream_selector_is_active_sinkpad),
|
|
(gst_stream_selector_activate_sinkpad),
|
|
(gst_stream_selector_get_linked_pads),
|
|
(gst_stream_selector_request_new_pad),
|
|
(gst_stream_selector_release_pad):
|
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* gst/playback/gststreamselector.h:
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|
Improve streamselector, make it select and unselect the current pad more
|
|
intelligently.
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Subclass GstPad for the sinkpads of the selector.
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Handle segments more correctly.
|
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Fix caps negotiation.
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Implement release_pad.
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2007-11-16 12:51:44 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst/playback/gstdecodebin2.c: Add drained signal fired when decodebin finishes decoding the data.
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Original commit message from CVS:
|
|
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
|
|
(gst_decode_group_check_if_drained), (source_pad_event_probe),
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(remove_fakesink):
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Add drained signal fired when decodebin finishes decoding the data.
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|
Remove deprecated STATE_DIRTY message.
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* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
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(unknown_type_cb), (new_decoded_pad_cb), (pad_removed_cb),
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|
(analyse_source), (proxy_drained_signal), (make_decoder),
|
|
(source_new_pad), (value_list_append_structure_list),
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(handle_redirect_message), (handle_message):
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Proxy the new drained signal.
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Handle pad removed from decodebin.
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Handle redirect messages by sorting multiple redirections based on the
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connection speed.
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|
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2007-11-16 11:22:09 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
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|
|
|
gst-libs/gst/rtsp/gstrtspmessage.c: Fix leaking headers. Fixes #496761.
|
|
Original commit message from CVS:
|
|
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
|
|
* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_unset):
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Fix leaking headers. Fixes #496761.
|
|
|
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2007-11-16 11:16:58 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
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|
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sys/: Don't leak the PAR on errors. Fixes #496731.
|
|
Original commit message from CVS:
|
|
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
|
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* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get),
|
|
(gst_ximagesink_change_state):
|
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_get):
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Don't leak the PAR on errors. Fixes #496731.
|
|
|
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2007-11-16 10:14:34 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/tag/gstid3tag.c: Add mapping for audio cd discid tags, so we can extract them from tags as well (see #34...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/gstid3tag.c: (user_tag_matches),
|
|
(gst_tag_from_id3_user_tag):
|
|
Add mapping for audio cd discid tags, so we can extract
|
|
them from tags as well (see #347848). Also compare identifiers
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in ID3v2 TXXX frames in a case-insensitive way to increase
|
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compatibility when reading tags (discid vs. DiscID vs. DiscId).
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|
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2007-11-16 01:21:40 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
|
|
gst-plugins-base.doap: Oops, fix the release name.
|
|
Original commit message from CVS:
|
|
* gst-plugins-base.doap:
|
|
Oops, fix the release name.
|
|
|
|
2007-11-16 00:44:58 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-plugins-base.doap: Add 0.10.15 release
|
|
Original commit message from CVS:
|
|
* gst-plugins-base.doap:
|
|
Add 0.10.15 release
|
|
|
|
2007-11-16 00:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
|
|
configure.ac: Back to CVS
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Back to CVS
|
|
|
|
=== release 0.10.15 ===
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|
|
|
2007-11-16 00:14:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
|
|
configure.ac: releasing 0.10.15, "No need to argue"
|
|
Original commit message from CVS:
|
|
=== release 0.10.15 ===
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|
2007-11-15 Jan Schmidt <jan.schmidt@sun.com>
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|
* configure.ac:
|
|
releasing 0.10.15, "No need to argue"
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|
|
|
2007-11-16 00:04:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
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* po/af.po:
|
|
* po/az.po:
|
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* po/bg.po:
|
|
* po/ca.po:
|
|
* po/cs.po:
|
|
* po/da.po:
|
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* po/de.po:
|
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* po/en_GB.po:
|
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* po/es.po:
|
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* po/fi.po:
|
|
* po/hu.po:
|
|
* po/it.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/pl.po:
|
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* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
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* po/zh_CN.po:
|
|
Update .po files
|
|
Original commit message from CVS:
|
|
Update .po files
|
|
|
|
2007-11-15 21:40:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
win32/vs6/libgstfft.dsp: Convert line endings to DOS.
|
|
Original commit message from CVS:
|
|
* win32/vs6/libgstfft.dsp:
|
|
Convert line endings to DOS.
|
|
|
|
2007-11-15 21:14:04 +0000 Sébastien Moutte <sebastien@moutte.net>
|
|
|
|
win32/: Add a project file for fft plugin and remove socket based plugin which don't build from the workspace.* win32...
|
|
Original commit message from CVS:
|
|
* win32/vs6/gst_plugins_base.dsw:
|
|
* win32/vs6/libgstfft.dsp:
|
|
* win32/MANIFEST:
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|
Add a project file for fft plugin and remove socket
|
|
based plugin which don't build from the workspace.* win32/vs6/libgstaudio.dsp:
|
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* win32/vs6/libgstrtp.dsp:
|
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* win32/vs6/libgsttag.dsp:
|
|
Convert line endings back to DOS.
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Fixes #496724
|
|
|
|
2007-11-14 12:27:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
win32/vs6/: Convert line endings back to DOS
|
|
Original commit message from CVS:
|
|
* win32/vs6/libgstinterfaces.dsp:
|
|
* win32/vs6/libgstrtsp.dsp:
|
|
Convert line endings back to DOS
|
|
|
|
2007-11-14 11:08:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/fft/: Don't include malloc.h which doesn't exist on Mac OSX.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/fft/kiss_fft_f32.h:
|
|
* gst-libs/gst/fft/kiss_fft_f64.h:
|
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* gst-libs/gst/fft/kiss_fft_s16.h:
|
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* gst-libs/gst/fft/kiss_fft_s32.h:
|
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Don't include malloc.h which doesn't exist on Mac OSX.
|
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Instead, pull in glib.h and use g_malloc/g_free for
|
|
consistency. Fixes: #496548
|
|
|
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2007-11-09 15:54:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
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gst/playback/gstdecodebin2.c: Dont leak ghostpad. Fixes #475451.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin2.c:
|
|
Dont leak ghostpad. Fixes #475451.
|
|
|
|
2007-11-09 12:21:52 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
Update some more docs and comments.
|
|
Original commit message from CVS:
|
|
* docs/design/design-decodebin.txt:
|
|
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
|
|
Update some more docs and comments.
|
|
|
|
2007-11-07 16:47:32 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
Require GIO >= 0.1.2 and adjust unit test for an API change.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* tests/check/pipelines/gio.c: (GST_START_TEST):
|
|
Require GIO >= 0.1.2 and adjust unit test for an API change.
|
|
|
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2007-11-07 15:18:54 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
ext/gio/gstgio.h: Add macro to check if a stream supports seeking.
|
|
Original commit message from CVS:
|
|
* ext/gio/gstgio.h:
|
|
Add macro to check if a stream supports seeking.
|
|
* ext/gio/Makefile.am:
|
|
* ext/gio/gstgiobasesink.c: (gst_gio_base_sink_base_init),
|
|
(gst_gio_base_sink_class_init), (gst_gio_base_sink_init),
|
|
(gst_gio_base_sink_finalize), (gst_gio_base_sink_start),
|
|
(gst_gio_base_sink_stop), (gst_gio_base_sink_unlock),
|
|
(gst_gio_base_sink_unlock_stop), (gst_gio_base_sink_event),
|
|
(gst_gio_base_sink_render), (gst_gio_base_sink_query),
|
|
(gst_gio_base_sink_set_stream):
|
|
* ext/gio/gstgiobasesink.h:
|
|
* ext/gio/gstgiobasesrc.c: (gst_gio_base_src_base_init),
|
|
(gst_gio_base_src_class_init), (gst_gio_base_src_init),
|
|
(gst_gio_base_src_finalize), (gst_gio_base_src_start),
|
|
(gst_gio_base_src_stop), (gst_gio_base_src_get_size),
|
|
(gst_gio_base_src_is_seekable), (gst_gio_base_src_unlock),
|
|
(gst_gio_base_src_unlock_stop), (gst_gio_base_src_check_get_range),
|
|
(gst_gio_base_src_create), (gst_gio_base_src_set_stream):
|
|
* ext/gio/gstgiobasesrc.h:
|
|
Refactor common GIO functions to GstGioBaseSink and GstGioBaseSrc
|
|
base classes that only require a GInputStream or GOutputStream to
|
|
work.
|
|
* ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
|
|
(gst_gio_sink_class_init), (gst_gio_sink_init),
|
|
(gst_gio_sink_finalize), (gst_gio_sink_start):
|
|
* ext/gio/gstgiosink.h:
|
|
* ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
|
|
(gst_gio_src_class_init), (gst_gio_src_init),
|
|
(gst_gio_src_finalize), (gst_gio_src_start):
|
|
* ext/gio/gstgiosrc.h:
|
|
Use the newly created base classes here.
|
|
* ext/gio/gstgio.c: (plugin_init):
|
|
* ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_base_init),
|
|
(gst_gio_stream_sink_class_init), (gst_gio_stream_sink_init),
|
|
(gst_gio_stream_sink_finalize), (gst_gio_stream_sink_set_property),
|
|
(gst_gio_stream_sink_get_property):
|
|
* ext/gio/gstgiostreamsink.h:
|
|
* ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_base_init),
|
|
(gst_gio_stream_src_class_init), (gst_gio_stream_src_init),
|
|
(gst_gio_stream_src_finalize), (gst_gio_stream_src_set_property),
|
|
(gst_gio_stream_src_get_property):
|
|
* ext/gio/gstgiostreamsrc.h:
|
|
Implement GstGioStreamSink and GstGioStreamSrc that have a property
|
|
to set the GInputStream/GOutputStream that should be used.
|
|
* tests/check/Makefile.am:
|
|
* tests/check/pipelines/.cvsignore:
|
|
* tests/check/pipelines/gio.c: (message_handler), (GST_START_TEST),
|
|
(gio_testsuite), (main):
|
|
Add unit test for giostreamsrc and giostreamsink.
|
|
|
|
2007-11-07 11:48:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
ext/gio/gstgio.c: Remove nowadays unnecessary workaround for a crash.
|
|
Original commit message from CVS:
|
|
* ext/gio/gstgio.c: (plugin_init):
|
|
Remove nowadays unnecessary workaround for a crash.
|
|
* ext/gio/gstgiosink.c: (gst_gio_sink_finalize),
|
|
(gst_gio_sink_start), (gst_gio_sink_stop),
|
|
(gst_gio_sink_unlock_stop):
|
|
* ext/gio/gstgiosink.h:
|
|
* ext/gio/gstgiosrc.c: (gst_gio_src_finalize), (gst_gio_src_start),
|
|
(gst_gio_src_stop), (gst_gio_src_unlock_stop):
|
|
* ext/gio/gstgiosrc.h:
|
|
Make the finalize function safer, clean up everything that could stay
|
|
around.
|
|
Reset the cancellable instead of creating a new one after cancelling
|
|
some operation.
|
|
Don't store the GFile in the element, it's only necessary for creating
|
|
the streams.
|
|
|
|
2007-11-06 23:35:39 +0000 Sebastien Moutte <sebastien@moutte.net>
|
|
|
|
gst-libs/gst/rtp/: Fix some C99-isms and and a missing function that some versions of
|
|
Original commit message from CVS:
|
|
Patch by: Sebastien Moutte <sebastien moutte net>
|
|
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
|
|
(gst_rtcp_unix_to_ntp):
|
|
* gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_name):
|
|
Fix some C99-isms and and a missing function that some versions of
|
|
MSVC don't like too much (#494346).
|
|
* win32/vs6/gst_plugins_base.dsw:
|
|
* win32/vs6/libgstaudio.dsp:
|
|
* win32/vs6/libgstrtp.dsp:
|
|
* win32/vs6/libgsttag.dsp:
|
|
Update vs6 projects files (#494346).
|
|
|
|
2007-11-06 16:38:49 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
|
|
|
|
win32/common/: More missing symbols to export (fixes #493986).
|
|
Original commit message from CVS:
|
|
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
|
|
* win32/common/libgstaudio.def:
|
|
* win32/common/libgstcdda.def:
|
|
* win32/common/libgstinterfaces.def:
|
|
* win32/common/libgstnetbuffer.def:
|
|
* win32/common/libgstpbutils.def:
|
|
* win32/common/libgstrtp.def:
|
|
* win32/common/libgstrtsp.def:
|
|
* win32/common/libgsttag.def:
|
|
* win32/common/libgstvideo.def:
|
|
More missing symbols to export (fixes #493986).
|
|
|
|
2007-11-06 11:58:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
Remove the magnitude and phase calculation functions as these have very special use cases and can't even be used for ...
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/fft/gstfftf32.c:
|
|
* gst-libs/gst/fft/gstfftf32.h:
|
|
* gst-libs/gst/fft/gstfftf64.c:
|
|
* gst-libs/gst/fft/gstfftf64.h:
|
|
* gst-libs/gst/fft/gstffts16.c:
|
|
* gst-libs/gst/fft/gstffts16.h:
|
|
* gst-libs/gst/fft/gstffts32.c:
|
|
* gst-libs/gst/fft/gstffts32.h:
|
|
* tests/check/libs/fft.c: (GST_START_TEST):
|
|
Remove the magnitude and phase calculation functions as these have
|
|
very special use cases and can't even be used for the spectrum
|
|
element. Also adjust the docs to mention some properties of the used
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|
FFT implemention, i.e. how the values are scaled. Fixes #492098.
|
|
|
|
2007-11-06 11:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/playback/gstplaybasebin.c: Avoid crash when there are external subtitles (fixes #491722).
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (queue_threshold_reached),
|
|
(finish_source):
|
|
Avoid crash when there are external subtitles (fixes #491722).
|
|
|
|
2007-11-03 10:39:21 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/alsa/: 'Could not open resource for writing' is not an acceptable even less so when we're trying to open it to re...
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsasink.c: (gst_alsasink_open):
|
|
* ext/alsa/gstalsasrc.c: (gst_alsasrc_open):
|
|
'Could not open resource for writing' is not an acceptable
|
|
error message when we can't open the audio device (see #492334),
|
|
even less so when we're trying to open it to record something.
|
|
|
|
2007-11-02 21:03:01 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
|
|
|
|
win32/common/libgstrtp.def: Add some more missing symbols (#492813).
|
|
Original commit message from CVS:
|
|
Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
|
|
* win32/common/libgstrtp.def:
|
|
Add some more missing symbols (#492813).
|
|
|
|
2007-11-02 14:59:06 +0000 Thijs Vermeir <thijsvermeir@gmail.com>
|
|
|
|
tests/check/elements/audioconvert.c: Add check to make sure that the out caps have a channel layout set on them where...
|
|
Original commit message from CVS:
|
|
Patch by: Thijs Vermeir <thijsvermeir@gmail.com>
|
|
* tests/check/elements/audioconvert.c: (verify_convert):
|
|
Add check to make sure that the out caps have a channel layout
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|
set on them where they should have one.
|
|
|
|
2007-11-01 13:28:59 +0000 Vincent Torri <vtorri@univ-evry.fr>
|
|
|
|
gst-libs/gst/fft/: Include our own _stdint.h instead of sys/types.h, makes MingW happy (#492306).
|
|
Original commit message from CVS:
|
|
Patch by: Vincent Torri <vtorri at univ-evry dot fr>
|
|
* gst-libs/gst/fft/kiss_fft_s16.h: (KISS_FFT_S16_MALLOC):
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* gst-libs/gst/fft/kiss_fft_s32.h: (KISS_FFT_S32_MALLOC):
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Include our own _stdint.h instead of sys/types.h, makes MingW happy
|
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(#492306).
|
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* gst-libs/gst/rtsp/gstrtspconnection.c: (gst_rtsp_connection_create):
|
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Use _pipe directly, GLib doesn't have a pipe() macro any longer
|
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(it disappeared in GLib 2.14.0) (#492306).
|
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* gst-libs/gst/sdp/Makefile.am:
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* gst-libs/gst/sdp/gstsdpmessage.c:
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Fix includes and LIBS for win32/Mingw (#492306).
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* tests/examples/dynamic/addstream.c (pause_play_stream):
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Use more portable g_usleep() instead of sleep() (#492306).
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2007-11-01 12:51:57 +0000 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
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gst-libs/gst/audio/gstringbuffer.c: Return NULL instead of an enum that happens to be 0, fixes warning on MSVC (#4921...
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Original commit message from CVS:
|
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Patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
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* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
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(gst_ring_buffer_parse_caps):
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Return NULL instead of an enum that happens to be 0, fixes warning
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on MSVC (#492114).
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* gst-libs/gst/audio/gstringbuffer.h:
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No trailing commas in enum list (for gcc-2.9x).
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* gst/videotestsrc/videotestsrc.c: (random_char):
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Make information loss explicit instead of implicitly truncating to
|
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eight bits via the return value. Fixes runtime error on MSVC when
|
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using the debug CRT (#492114).
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* win32/common/config.h.in:
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Fix a bunch of '#undef FOO bar', which MSVC doesn't like (#492114).
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* win32/common/libgstinterfaces.def:
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* win32/common/libgstrtp.def:
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Export a few more symbols (#492114).
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2007-11-01 08:06:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst-libs/gst/audio/audio.*: Readd the deprecation guards, but preserve compilability.
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Original commit message from CVS:
|
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* gst-libs/gst/audio/audio.c:
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* gst-libs/gst/audio/audio.h:
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Readd the deprecation guards, but preserve compilability.
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2007-10-31 17:54:48 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/audioconvert/gstaudioconvert.c: Preserve channel layout when fixating the number of channels in the output caps, ...
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Original commit message from CVS:
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* gst/audioconvert/gstaudioconvert.c: (find_suitable_channel_layout),
|
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(gst_audio_convert_fixate_channels), (gst_audio_convert_fixate_caps):
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Preserve channel layout when fixating the number of channels in the
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output caps, or make sure there's a suitable channel position layout
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set on the caps if required. Fixes #430677.
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2007-10-31 17:32:22 +0000 Tim-Philipp Müller <tim@centricular.net>
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tests/check/elements/decodebin.c: Make sure the pipeline really operates in push mode as it should in this case.
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Original commit message from CVS:
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* tests/check/elements/decodebin.c: (test_text_plain_streams):
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Make sure the pipeline really operates in push mode as it should
|
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in this case.
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2007-10-31 15:30:15 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/audio/audio.h: Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or compilation with DISABLE_...
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Original commit message from CVS:
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* gst-libs/gst/audio/audio.h:
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Don't guard deprecated enum with #ifndef DISABLE_DEPRECATED, or
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compilation with DISABLE_DEPRECATED and without REMOVE_DEPRECATED
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(ie. normal cvs builds) will fail.
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2007-10-31 12:47:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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tell gtk-doc about the deprecation guard. Apply more doc fixes.
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Original commit message from CVS:
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* docs/libs/Makefile.am:
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* gst-libs/gst/audio/audio.c:
|
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* gst-libs/gst/audio/audio.h:
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* gst-libs/gst/interfaces/mixer.c:
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tell gtk-doc about the deprecation guard. Apply more doc fixes.
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2007-10-31 12:30:28 +0000 Tim-Philipp Müller <tim@centricular.net>
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tests/check/libs/audio.c: Add simple unit test to make sure GstValue intersection of channel layouts works the way I ...
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|
Original commit message from CVS:
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* tests/check/libs/audio.c: (init_value_to_channel_layout),
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|
(test_channel_layout_value_intersect), (audio_suite):
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Add simple unit test to make sure GstValue intersection
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of channel layouts works the way I think it does.
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2007-10-30 20:32:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
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Fix the docs according to what gtk-doc complained about.
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|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
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* gst-libs/gst/audio/gstaudiofilter.h:
|
|
* gst-libs/gst/interfaces/mixer.h:
|
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* gst-libs/gst/rtp/gstbasertpdepayload.c:
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* gst-libs/gst/rtp/gstbasertpdepayload.h:
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* gst-libs/gst/sdp/gstsdpmessage.c:
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Fix the docs according to what gtk-doc complained about.
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|
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2007-10-30 19:46:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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tests/icles/stress-playbin.c: Fix the build.
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|
Original commit message from CVS:
|
|
* tests/icles/stress-playbin.c:
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Fix the build.
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|
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2007-10-30 15:54:46 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/: Post nice/more useful error message if we don't have a decoder for the primary type.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (close_pad_link), (type_found):
|
|
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
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Post nice/more useful error message if we don't have a decoder for
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the primary type.
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2007-10-30 15:07:58 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstdecodebin2.c: Be a bit more useful, unblock the pads after we fired the no-more-pads signal so that w...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin2.c: (gst_decode_group_expose):
|
|
Be a bit more useful, unblock the pads after we fired the no-more-pads
|
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signal so that we can use the signal to inspect and connect all pads
|
|
without having to keep extra state outside of decodebin.
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|
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2007-10-30 15:00:06 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst/playback/gsturidecodebin.c: Implement default signal handler so that we return TRUE when nothing is connected.
|
|
Original commit message from CVS:
|
|
* gst/playback/gsturidecodebin.c:
|
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(gst_uri_decode_bin_autoplug_continue),
|
|
(gst_uri_decode_bin_class_init), (no_more_pads_full):
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Implement default signal handler so that we return TRUE when nothing is
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connected.
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|
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2007-10-28 11:53:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
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gst-libs/gst/riff/riff-media.c: Use the ALSA channel layout as default for wav files without channel layout informati...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c:
|
|
(gst_riff_wavext_add_channel_layout),
|
|
(gst_riff_wave_add_default_channel_layout),
|
|
(gst_riff_wavext_get_default_channel_mask),
|
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(gst_riff_create_audio_caps):
|
|
Use the ALSA channel layout as default for wav files without channel
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|
layout information. This fixes playback of chan-id.wav on 5.1 systems
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|
for example. Also refactor the channel layout setting a bit and add
|
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more default channel orders. Fixes #489010.
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|
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2007-10-28 11:46:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
|
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* ChangeLog:
|
|
Use the ALSA channel layout as default for wav files without channel layout information. This fixes playback of chan-...
|
|
Original commit message from CVS:
|
|
(gst_riff_wavext_add_channel_layout),
|
|
(gst_riff_wave_add_default_channel_layout),
|
|
(gst_riff_wavext_get_default_channel_mask),
|
|
(gst_riff_create_audio_caps):
|
|
Use the ALSA channel layout as default for wav files without channel
|
|
layout information. This fixes playback of chan-id.wav on 5.1 systems
|
|
for example. Also refactor the channel layout setting a bit and add
|
|
more default channel orders. Fixes #489010.
|
|
|
|
2007-10-26 18:57:33 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/libs/tag.c: GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
|
|
Original commit message from CVS:
|
|
* tests/check/libs/tag.c: (test_musicbrainz_tag_registration):
|
|
GST_TAG_MUSICBRAINZ_SORTNAME is deprecated and we compile with
|
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-DGST_DISABLE_DEPRECATED, so use new GST_TAG_ARTIST_SORTNAME
|
|
instead.
|
|
|
|
2007-10-26 12:07:14 +0000 Christian Schaller <uraeus@gnome.org>
|
|
|
|
* gst-plugins-base.spec.in:
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|
update spec file
|
|
Original commit message from CVS:
|
|
update spec file
|
|
|
|
2007-10-25 17:36:49 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstdecodebin2.c: Move subtitle encoding property to decodebin2 so that it can set the property value on ...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
|
|
(gst_decode_bin_dispose), (gst_decode_bin_set_caps),
|
|
(gst_decode_bin_set_subs_encoding),
|
|
(gst_decode_bin_get_subs_encoding), (gst_decode_bin_set_property),
|
|
(gst_decode_bin_get_property), (analyze_new_pad):
|
|
Move subtitle encoding property to decodebin2 so that it can set the
|
|
property value on all elements that it autoplugs and that require it.
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|
Make caps refcounting more consistent in get/set.
|
|
* gst/playback/gsturidecodebin.c: (_gst_boolean_accumulator),
|
|
(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
|
|
(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
|
|
(gst_uri_decode_bin_get_property), (proxy_unknown_type_signal),
|
|
(proxy_autoplug_continue_signal),
|
|
(proxy_autoplug_factories_signal), (proxy_autoplug_select_signal),
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(make_decoder):
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|
Proxy properties and relevant signals from the internal decodebin.
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|
Make properties MT safe.
|
|
|
|
2007-10-25 15:10:59 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/tag/: Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/tag.h: (GST_TAG_MUSICBRAINZ_SORTNAME):
|
|
* gst-libs/gst/tag/tags.c:
|
|
Deprecate GST_TAG_MUSICBRAINZ_SORTNAME, replaced by the newly-added
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|
GST_TAG_ARTIST_SORTNAME (in an API and ABI compatible way).
|
|
* gst-libs/gst/tag/gstid3tag.c: (tag_matches):
|
|
Map ID3v2 TSOP, TSOA and TSOT frames to new SORTNAME tags (#414539).
|
|
* gst-libs/gst/tag/gstvorbistag.c: (tag_matches),
|
|
(gst_tag_to_vorbis_comments):
|
|
Map new SORTNAME tags (these tags aren't even semi-official, so I'm
|
|
just mapping everything I found in the wild) (#414539).
|
|
|
|
2007-10-24 11:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/: Remove the autoplug-sort signal and replace it with a binding friendly autoplug-select signal.
|
|
Original commit message from CVS:
|
|
Inspired by patch of: René Stadler <mail at renestadler dot de>
|
|
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init),
|
|
(gst_decode_bin_autoplug_continue),
|
|
(gst_decode_bin_autoplug_factories),
|
|
(gst_decode_bin_autoplug_select), (analyze_new_pad), (connect_pad),
|
|
(find_compatibles):
|
|
* gst/playback/gstplay-marshal.list:
|
|
Remove the autoplug-sort signal and replace it with a binding friendly
|
|
autoplug-select signal.
|
|
Add an autoplug-factories signal that can be used to generate a list of
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factories to try to autoplug.
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|
Add the GstPad to the autoplugging signal args as it might be needed to
|
|
make a good factory selection.
|
|
Fix up the marshallers for this. Fixes #407282.
|
|
|
|
2007-10-23 14:23:14 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/tag/gsttagdemux.c: Don't abort with an assertion if we receive a seek event with a start type of NONE (s...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/gsttagdemux.c:
|
|
Don't abort with an assertion if we receive a seek event with
|
|
a start type of NONE (see launchpad bug #155878).
|
|
|
|
2007-10-22 10:21:46 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
sys/: Make sure that before we clean up the X resources, we shutdown and join the event thread.
|
|
Original commit message from CVS:
|
|
* sys/ximage/ximagesink.c: (gst_ximagesink_event_thread),
|
|
(gst_ximagesink_xcontext_get), (gst_ximagesink_xcontext_clear),
|
|
(gst_ximagesink_change_state), (gst_ximagesink_reset):
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_event_thread),
|
|
(gst_xvimagesink_xcontext_get), (gst_xvimagesink_xcontext_clear),
|
|
(gst_xvimagesink_change_state), (gst_xvimagesink_reset):
|
|
Make sure that before we clean up the X resources, we shutdown and join
|
|
the event thread.
|
|
Also make sure the event thread does not shut down immediatly after
|
|
startup because the running variable is not yet correctly set.
|
|
Fixes #378770.
|
|
|
|
2007-10-16 16:48:38 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstdecodebin.c: Make the window for a race in typefind and shutting down smaller until we figure out the...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (new_pad), (type_found):
|
|
Make the window for a race in typefind and shutting down smaller until
|
|
we figure out the right locking here. Avoids #485753 usually.
|
|
* gst/playback/gstdecodebin2.c: (type_found), (pad_added_group_cb):
|
|
Remove unneeded lock causing a race in typefind and shutting down.
|
|
Fixes #485753.
|
|
* gst/playback/gstplaybin.c: (gst_play_bin_change_state):
|
|
Also remove sinks when going to NULL because we might not complete the
|
|
state change to PAUSED, causing the PAUSED->READY state change not to
|
|
happen.
|
|
|
|
2007-10-16 15:33:31 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.c: Also explicitly release the ringbuffer when going to NULL because it is requir...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_drain), (gst_base_audio_sink_change_state):
|
|
Also explicitly release the ringbuffer when going to NULL because it
|
|
is required in the setcaps function, before the state change to PAUSED
|
|
completes.
|
|
|
|
2007-10-16 14:58:53 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/icles/: Does what it says on the tin.
|
|
Original commit message from CVS:
|
|
* tests/icles/.cvsignore:
|
|
* tests/icles/Makefile.am:
|
|
* tests/icles/stress-playbin.c:
|
|
Does what it says on the tin.
|
|
|
|
2007-10-15 11:38:39 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstqueue2.c: Fix queue negotiation. See #486758.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstqueue2.c: (gst_queue_init), (gst_queue_push_one):
|
|
Fix queue negotiation. See #486758.
|
|
|
|
2007-10-12 10:52:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
Actual code change to go along with:
|
|
Original commit message from CVS:
|
|
Actual code change to go along with:
|
|
2007-10-12 Jan Schmidt <Jan.Schmidt@sun.com>
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_decorate),
|
|
(gst_xvimagesink_xwindow_new),
|
|
(gst_xvimagesink_update_colorbalance),
|
|
(gst_xvimagesink_handle_xevents), (gst_xvimagesink_xcontext_get):
|
|
Fix handling of some of the X atoms. If the last parameter is True,
|
|
XInternAtom won't create the atom if it doesn't exist, and therefore
|
|
might return None. This causes X errors on Xv implementations that
|
|
don't provide the colour balance attributes.
|
|
|
|
2007-10-12 10:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
* ChangeLog:
|
|
Remove stray character from the changelog.
|
|
Original commit message from CVS:
|
|
Remove stray character from the changelog.
|
|
|
|
2007-10-12 10:33:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
* ChangeLog:
|
|
I'm too lazy to comment this
|
|
Original commit message from CVS:
|
|
*** empty log message ***
|
|
|
|
2007-10-11 18:24:09 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
Extract vorbis comment LICENSE tags correctly.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/gstvorbistag.c:
|
|
* tests/check/libs/tag.c:
|
|
Extract vorbis comment LICENSE tags correctly.
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|
|
|
2007-10-11 16:12:21 +0000 Jason Kivlighn <jkivlighn@gmail.com>
|
|
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|
Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
|
|
Original commit message from CVS:
|
|
Patch by: Jason Kivlighn <jkivlighn gmail com>
|
|
* gst-libs/gst/tag/gstid3tag.c:
|
|
* tests/check/libs/tag.c:
|
|
Map ID3v2 WCOP frame to GST_TAG_COPYRIGHT_URI (#447000).
|
|
|
|
2007-10-10 17:01:51 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/tag/gsttagdemux.c: Don't error out when a buggy downstream element doesn't handle the newsegment event w...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/gsttagdemux.c:
|
|
Don't error out when a buggy downstream element doesn't
|
|
handle the newsegment event we send properly (especially
|
|
not without posting a meaningful error message on the
|
|
bus). See bug #471370 and launchpad bug #136264.
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|
|
|
2007-10-10 15:36:56 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst-libs/gst/audio/gstbaseaudiosink.c: Use new basesink method to make our EOS drain interruptable.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_drain):
|
|
Use new basesink method to make our EOS drain interruptable.
|
|
|
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2007-10-10 09:37:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/rtp/gstrtppayloads.c: Fix silly search-replace oversight.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstrtppayloads.c:
|
|
Fix silly search-replace oversight.
|
|
|
|
2007-10-09 09:57:17 +0000 Laurent Glayal <spglegle@yahoo.fr>
|
|
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gst-libs/gst/rtp/gstbasertppayload.c: Fix caps memleak. Fixes #484989.
|
|
Original commit message from CVS:
|
|
Patch by: Laurent Glayal <spglegle at yahoo dot fr>
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* gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
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(gst_basertppayload_set_outcaps):
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Fix caps memleak. Fixes #484989.
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2007-10-08 18:04:34 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst-libs/gst/rtp/gstbasertpdepayload.c: Fix debug output.
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|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain):
|
|
Fix debug output.
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|
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2007-10-08 18:02:53 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst-libs/gst/audio/gstbaseaudiosrc.c: Also handle the case where there is no clock set on the audio source, like in t...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
(gst_base_audio_src_create):
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Also handle the case where there is no clock set on the audio source,
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like in the unit tests.
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|
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2007-10-08 17:40:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
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gst-libs/gst/rtp/gstrtppayloads.c: Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8 to avoid compiler war...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstrtppayloads.c:
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Use unsigned G_MAXUINT8 instead of -1 to initialise a guint8
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to avoid compiler warnings
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2007-10-08 17:12:32 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst/playback/: Don't disconnect the have_type signal because we never reconnect it later on. Instead keep a variable ...
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|
Original commit message from CVS:
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|
* gst/playback/gstdecodebin.c: (type_found),
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|
(gst_decode_bin_change_state):
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* gst/playback/gstdecodebin2.c: (type_found),
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(gst_decode_bin_change_state):
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Don't disconnect the have_type signal because we never reconnect it
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later on. Instead keep a variable to see if we already detected a type.
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2007-10-08 10:47:26 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst/playback/: Unlink the signal handler when we found the type, we're not going to do anything sensible with more ty...
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|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (add_fakesink), (type_found):
|
|
* gst/playback/gstdecodebin2.c: (gst_decode_bin_init),
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(type_found):
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Unlink the signal handler when we found the type, we're not going to do
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anything sensible with more type_found signals anyway.
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2007-10-08 06:07:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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ext/gio/gstgio.c: Use GIO function to get a list of supported URI schemes instead of hard coding something.
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|
Original commit message from CVS:
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* ext/gio/gstgio.c: (gst_gio_get_supported_protocols):
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Use GIO function to get a list of supported URI schemes instead of
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hard coding something.
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2007-10-06 16:49:55 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/tag/gsttagdemux.c: Don't leak caps.
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|
Original commit message from CVS:
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|
* gst-libs/gst/tag/gsttagdemux.c:
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Don't leak caps.
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|
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2007-10-06 15:04:53 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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gst-libs/gst/tag/: API: add GstTagDemux base class for simple tag demuxers.
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|
Original commit message from CVS:
|
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* gst-libs/gst/tag/Makefile.am:
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* gst-libs/gst/tag/gsttagdemux.c:
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* gst-libs/gst/tag/gsttagdemux.h:
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API: add GstTagDemux base class for simple tag demuxers.
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* docs/libs/gst-plugins-base-libs-docs.sgml:
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* docs/libs/gst-plugins-base-libs-sections.txt:
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Add GstTagDemux to docs.
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2007-10-05 07:49:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst-libs/gst/rtp/gstrtpbuffer.c: Fix bug introduced with last commit which inverted the logic and caused all buffers ...
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|
Original commit message from CVS:
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* gst-libs/gst/rtp/gstrtpbuffer.c:
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(gst_rtp_buffer_get_payload_subbuffer):
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Fix bug introduced with last commit which inverted the logic and
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caused all buffers to be dropped. Fixes #483620.
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Thanks to Laurent Glayal <spglegle at yahoo dot fr> for noticing.
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2007-10-04 06:50:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst-libs/gst/rtp/gstrtpbuffer.c: with regular return and warning.
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|
Original commit message from CVS:
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* gst-libs/gst/rtp/gstrtpbuffer.c:
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Replace g_return_if_val (as it could be disabled), with regular return
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and warning.
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2007-10-03 14:51:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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tests/check/pipelines/simple-launch-lines.c: Print message name and not just number.
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|
Original commit message from CVS:
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|
* tests/check/pipelines/simple-launch-lines.c:
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Print message name and not just number.
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|
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2007-10-02 11:11:13 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst-libs/gst/audio/gstbaseaudiosink.c: When slaved to the clock, don't try to align a sample with the previous one wh...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
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(gst_base_audio_sink_async_play):
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When slaved to the clock, don't try to align a sample with the previous
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one when going to PLAYING again.
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2007-10-02 09:04:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
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tests/examples/snapshot/snapshot.c: Fix the build.
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|
Original commit message from CVS:
|
|
* tests/examples/snapshot/snapshot.c:
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Fix the build.
|
|
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2007-10-02 07:43:57 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
|
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ext/gio/gstgiosink.c: Update to API changes in GIO.
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|
Original commit message from CVS:
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|
* ext/gio/gstgiosink.c: (gst_gio_sink_start):
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Update to API changes in GIO.
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|
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2007-10-01 16:33:00 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst-libs/gst/sdp/gstsdpmessage.h: Add RFC 3556 bandwidth modifiers.
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|
Original commit message from CVS:
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* gst-libs/gst/sdp/gstsdpmessage.h:
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Add RFC 3556 bandwidth modifiers.
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|
|
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2007-10-01 13:37:31 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
|
Update documentation.
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-docs.sgml:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
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* gst-libs/gst/rtp/gstrtppayloads.c:
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Update documentation.
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|
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2007-10-01 13:22:14 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
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gst-libs/gst/rtp/: Added new file and header to deal with payload info.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/Makefile.am:
|
|
* gst-libs/gst/rtp/gstrtppayloads.c: (gst_rtp_payload_info_for_pt),
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|
(gst_rtp_payload_info_for_name):
|
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* gst-libs/gst/rtp/gstrtppayloads.h:
|
|
Added new file and header to deal with payload info.
|
|
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
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|
(gst_rtp_buffer_default_clock_rate):
|
|
* gst-libs/gst/rtp/gstrtpbuffer.h:
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|
Payload specific stuff is move to new headers.
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Implement _default_clock rate using the new payload function.
|
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* gst-libs/gst/sdp/gstsdpmessage.c: (is_multicast_address),
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(gst_sdp_parse_line):
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* gst-libs/gst/sdp/gstsdpmessage.h:
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Add some more comments.
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|
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2007-10-01 10:22:46 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
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gst/typefind/gsttypefindfunctions.c: Add typefind function for application/sdp.
|
|
Original commit message from CVS:
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|
* gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
|
|
(sdp_check_header), (sdp_type_find), (plugin_init):
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|
Add typefind function for application/sdp.
|
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Remove some old dirac typefind code that was ifdeffed out.
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|
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2007-09-29 12:04:02 +0000 Sébastien Moutte <sebastien@moutte.net>
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|
|
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win32/common/libgstaudio.def: Add new exported functions.
|
|
Original commit message from CVS:
|
|
* win32/common/libgstaudio.def:
|
|
Add new exported functions.
|
|
* win32/vs6/grammar.dsp:
|
|
Add autogeneration and copy of some autegenerated files from win32/common
|
|
for rtsp library.
|
|
* win32/vs6/libgstaudioconvert.dsp:
|
|
Add gstaudioquantize.c to the build.
|
|
* win32/vs6/libgstinterfaces.dsp:
|
|
Add videoorientation.c to the build.
|
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* win32/vs6/libgstriff.dsp:
|
|
Add libgsttag to the link libraries list.
|
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* win32/vs6/libgstvolume.dsp:
|
|
Add liboil to the link.
|
|
* win32/vs6/gst_plugins_base.dsw:
|
|
* win32/vs6/libgstrtsp.dsp:
|
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* win32/common/libgstrtsp.def:
|
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Add files to build libgstrtsp library.
|
|
|
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2007-09-29 07:01:55 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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ext/gio/: Some minor cleanup and allow setting the location only when the element is not playing or paused.
|
|
Original commit message from CVS:
|
|
* ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
|
|
(gst_gio_sink_set_property), (gst_gio_sink_render):
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* ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
|
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(gst_gio_src_set_property):
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Some minor cleanup and allow setting the location only when the
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element is not playing or paused.
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|
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2007-09-26 15:14:37 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
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tests/examples/snapshot/snapshot.c: Print error when pipeline failed to construct.
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|
Original commit message from CVS:
|
|
* tests/examples/snapshot/snapshot.c: (main):
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Print error when pipeline failed to construct.
|
|
|
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2007-09-25 19:06:47 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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Add mappings for the new GST_TAG_COMPOSER for vorbis comments and ID3v2 tags.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* gst-libs/gst/tag/gstid3tag.c:
|
|
* gst-libs/gst/tag/gstvorbistag.c:
|
|
Add mappings for the new GST_TAG_COMPOSER for vorbis comments
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and ID3v2 tags.
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|
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2007-09-25 11:54:09 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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gst-libs/gst/floatcast/floatcast.h: Don't include config.h in an installed public header, this might break compilatio...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/floatcast/floatcast.h:
|
|
Don't include config.h in an installed public header, this
|
|
might break compilation of applications that don't have such
|
|
a header and doesn't necessarily do what it's supposed to do
|
|
anyway (ie. check for the lrint/lrintf defines) (#442065).
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Add docs for the various macros and document how this header
|
|
has to be used (link against libm, etc.); add a few FIXMEs;
|
|
include math.h for non-c99 code path. Based on patch by
|
|
Jan Schmidt.
|
|
|
|
2007-09-25 07:50:59 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
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configure.ac: Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead of duplicating these macros in confi...
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Use AG_GST_ARG_WITH_PLUGINS and AG_GST_ARG_ENABLE_EXTERNAL instead
|
|
of duplicating these macros in configure.ac.
|
|
|
|
2007-09-22 17:58:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
po/: Updated translations to 0.10.14
|
|
Original commit message from CVS:
|
|
* po/hu.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
Updated translations to 0.10.14
|
|
|
|
2007-09-22 17:57:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* po/LINGUAS:
|
|
add languages
|
|
Original commit message from CVS:
|
|
add languages
|
|
|
|
2007-09-22 17:56:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
po/pl.po: Added Polish translation.
|
|
Original commit message from CVS:
|
|
translated by: Jakub Bogusz <qboosh@pld-linux.org>
|
|
* po/pl.po:
|
|
Added Polish translation.
|
|
|
|
2007-09-22 17:55:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
po/fi.po: Added Finnish translation.
|
|
Original commit message from CVS:
|
|
translated by: Ilkka Tuohela <hile@iki.fi>
|
|
* po/fi.po:
|
|
Added Finnish translation.
|
|
|
|
2007-09-22 17:54:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
po/es.po: Added Spanish translation.
|
|
Original commit message from CVS:
|
|
translated by: Jorge González González <aloriel@gmail.com>
|
|
* po/es.po:
|
|
Added Spanish translation.
|
|
|
|
2007-09-22 17:53:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
po/da.po: Added Danish translation.
|
|
Original commit message from CVS:
|
|
translated by: Mogens Jaeger <mogens@jaeger.tf>
|
|
* po/da.po:
|
|
Added Danish translation.
|
|
|
|
2007-09-22 17:52:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
po/zh_CN.po: Added Chinese (simplified) translation.
|
|
Original commit message from CVS:
|
|
translated by: Funda Wang <fundawang@linux.net.cn>
|
|
* po/zh_CN.po:
|
|
Added Chinese (simplified) translation.
|
|
|
|
2007-09-22 17:51:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
po/bg.po: Added Bulgarian translation.
|
|
Original commit message from CVS:
|
|
translated by: Alexander Shopov <ash@contact.bg>
|
|
* po/bg.po:
|
|
Added Bulgarian translation.
|
|
|
|
2007-09-21 18:00:24 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
docs/plugins/gst-plugins-bad-plugins.hierarchy: Update hierarchy.
|
|
Original commit message from CVS:
|
|
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
|
|
Update hierarchy.
|
|
* ext/gio/gstgiosink.h:
|
|
* ext/gio/gstgiosrc.h:
|
|
Mark private fields of the instance structs private.
|
|
|
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2007-09-21 17:31:05 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
docs/plugins/: Add the GIO plugin to the docs and do a make update while doing that.
|
|
Original commit message from CVS:
|
|
* docs/plugins/Makefile.am:
|
|
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
|
|
* docs/plugins/gst-plugins-bad-plugins.args:
|
|
* docs/plugins/gst-plugins-bad-plugins.signals:
|
|
* docs/plugins/inspect/plugin-bz2.xml:
|
|
* docs/plugins/inspect/plugin-cdxaparse.xml:
|
|
* docs/plugins/inspect/plugin-dfbvideosink.xml:
|
|
* docs/plugins/inspect/plugin-dtsdec.xml:
|
|
* docs/plugins/inspect/plugin-equalizer.xml:
|
|
* docs/plugins/inspect/plugin-faac.xml:
|
|
* docs/plugins/inspect/plugin-faad.xml:
|
|
* docs/plugins/inspect/plugin-filter.xml:
|
|
* docs/plugins/inspect/plugin-freeze.xml:
|
|
* docs/plugins/inspect/plugin-gio.xml:
|
|
* docs/plugins/inspect/plugin-gsm.xml:
|
|
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
|
|
* docs/plugins/inspect/plugin-h264parse.xml:
|
|
* docs/plugins/inspect/plugin-modplug.xml:
|
|
* docs/plugins/inspect/plugin-mpeg2enc.xml:
|
|
* docs/plugins/inspect/plugin-musepack.xml:
|
|
* docs/plugins/inspect/plugin-musicbrainz.xml:
|
|
* docs/plugins/inspect/plugin-nsfdec.xml:
|
|
* docs/plugins/inspect/plugin-replaygain.xml:
|
|
* docs/plugins/inspect/plugin-soundtouch.xml:
|
|
* docs/plugins/inspect/plugin-spcdec.xml:
|
|
* docs/plugins/inspect/plugin-spectrum.xml:
|
|
* docs/plugins/inspect/plugin-speed.xml:
|
|
* docs/plugins/inspect/plugin-tta.xml:
|
|
* docs/plugins/inspect/plugin-videosignal.xml:
|
|
* docs/plugins/inspect/plugin-xingheader.xml:
|
|
* docs/plugins/inspect/plugin-xvid.xml:
|
|
Add the GIO plugin to the docs and do a make update
|
|
while doing that.
|
|
* ext/gio/gstgiosrc.c: (gst_gio_src_start):
|
|
Fix a small memleak.
|
|
|
|
2007-09-21 17:07:56 +0000 René Stadler <mail@renestadler.de>
|
|
|
|
Add a GIO/GVFS plugin with source and sink elements. This will only be enabled when --enable-experimental is given to...
|
|
Original commit message from CVS:
|
|
Patch by: René Stadler <mail at renestadler dot de>
|
|
* configure.ac:
|
|
* ext/Makefile.am:
|
|
* ext/gio/Makefile.am:
|
|
* ext/gio/gstgio.c: (gst_gio_error), (gst_gio_seek),
|
|
(gst_gio_get_supported_protocols),
|
|
(gst_gio_uri_handler_get_type_sink),
|
|
(gst_gio_uri_handler_get_type_src),
|
|
(gst_gio_uri_handler_get_protocols), (gst_gio_uri_handler_get_uri),
|
|
(gst_gio_uri_handler_set_uri), (gst_gio_uri_handler_init),
|
|
(gst_gio_uri_handler_do_init), (plugin_init):
|
|
* ext/gio/gstgio.h:
|
|
* ext/gio/gstgiosink.c: (gst_gio_sink_base_init),
|
|
(gst_gio_sink_class_init), (gst_gio_sink_init),
|
|
(gst_gio_sink_finalize), (gst_gio_sink_set_property),
|
|
(gst_gio_sink_get_property), (gst_gio_sink_start),
|
|
(gst_gio_sink_stop), (gst_gio_sink_unlock),
|
|
(gst_gio_sink_unlock_stop), (gst_gio_sink_event),
|
|
(gst_gio_sink_render), (gst_gio_sink_query):
|
|
* ext/gio/gstgiosink.h:
|
|
* ext/gio/gstgiosrc.c: (gst_gio_src_base_init),
|
|
(gst_gio_src_class_init), (gst_gio_src_init),
|
|
(gst_gio_src_finalize), (gst_gio_src_set_property),
|
|
(gst_gio_src_get_property), (gst_gio_src_start),
|
|
(gst_gio_src_stop), (gst_gio_src_get_size),
|
|
(gst_gio_src_is_seekable), (gst_gio_src_unlock),
|
|
(gst_gio_src_unlock_stop), (gst_gio_src_check_get_range),
|
|
(gst_gio_src_create):
|
|
* ext/gio/gstgiosrc.h:
|
|
Add a GIO/GVFS plugin with source and sink elements. This will
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only be enabled when --enable-experimental is given to configure
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for now as the GIO API is not stable yet. Fixes #476916.
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2007-09-21 14:37:26 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstqueue2.c: Fix compilation wrt printf arguments.
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Original commit message from CVS:
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* gst/playback/gstqueue2.c: (gst_queue_push_one):
|
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Fix compilation wrt printf arguments.
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2007-09-20 17:38:10 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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examples/app/appsrc_ex.c: Fix compilation after changing the name of a method.
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Original commit message from CVS:
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* examples/app/appsrc_ex.c: (main):
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Fix compilation after changing the name of a method.
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|
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2007-09-20 14:09:24 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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Add simple snapshot example program using appsink.
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Original commit message from CVS:
|
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* configure.ac:
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* tests/examples/Makefile.am:
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* tests/examples/snapshot/.cvsignore:
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* tests/examples/snapshot/Makefile.am:
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* tests/examples/snapshot/snapshot.c: (main):
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Add simple snapshot example program using appsink.
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2007-09-20 13:59:50 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/app/gstappsink.*: Add properties, signals and actions to access the element even without linking to the ...
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|
Original commit message from CVS:
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* gst-libs/gst/app/gstappsink.c: (gst_app_marshal_OBJECT__VOID),
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(gst_app_sink_class_init), (gst_app_sink_init),
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(gst_app_sink_dispose), (gst_app_sink_finalize),
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(gst_app_sink_set_property), (gst_app_sink_get_property),
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(gst_app_sink_flush_unlocked), (gst_app_sink_start),
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(gst_app_sink_event), (gst_app_sink_getcaps),
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(gst_app_sink_set_caps), (gst_app_sink_get_caps),
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(gst_app_sink_is_eos), (gst_app_sink_pull_preroll),
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(gst_app_sink_pull_buffer):
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* gst-libs/gst/app/gstappsink.h:
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Add properties, signals and actions to access the element even without
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linking to the library.
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Fix some method names and signatures.
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2007-09-20 10:37:02 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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tests/check/generic/states.c: Improved state change unit test.
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Original commit message from CVS:
|
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* tests/check/generic/states.c:
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Improved state change unit test.
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2007-09-19 18:16:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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Ignore registries in any format.
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|
Original commit message from CVS:
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* docs/plugins/.cvsignore:
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* tests/check/.cvsignore:
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Ignore registries in any format.
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2007-09-19 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/rtp/gstbasertpdepayload.c: Only copy timestamp on outgoing packets if the depayloader did not set one.
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Original commit message from CVS:
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* gst-libs/gst/rtp/gstbasertpdepayload.c:
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(gst_base_rtp_depayload_chain),
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(gst_base_rtp_depayload_set_gst_timestamp):
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Only copy timestamp on outgoing packets if the depayloader did not set
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one.
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Also copy duration on outgoing packets.
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2007-09-19 15:55:08 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/rtp/gstbasertppayload.c: Fix compilation because of missing %d in printf.
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Original commit message from CVS:
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* gst-libs/gst/rtp/gstbasertppayload.c: (copy_fixed),
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(gst_basertppayload_set_outcaps):
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Fix compilation because of missing %d in printf.
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When fixating caps, fixate what we can and throw away all remaining
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unfixed caps, subclasses should do something smart if they need to.
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2007-09-19 12:04:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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ext/gnomevfs/gstgnomevfssrc.c: Improve debug logs a bit and be more verbose if things go wrong.
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Original commit message from CVS:
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* ext/gnomevfs/gstgnomevfssrc.c:
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Improve debug logs a bit and be more verbose if things go wrong.
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2007-09-17 17:24:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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Fix a bunch of compile warnings shown with Forte.
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Original commit message from CVS:
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* ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
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(gst_text_overlay_set_property):
|
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* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_render):
|
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* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_ntp_to_unix),
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(gst_rtcp_unix_to_ntp):
|
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* gst-libs/gst/rtsp/gstrtspmessage.c: (gst_rtsp_message_get_type):
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* gst/playback/gstqueue2.c:
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* tests/examples/seek/seek.c: (set_scale):
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Fix a bunch of compile warnings shown with Forte.
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* gst/audiorate/gstaudiorate.c:
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Always pull in config.h before including any system headers.
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2007-09-17 16:22:17 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstqueue2.c: Also fix #476514 for queue2.
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|
Original commit message from CVS:
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* gst/playback/gstqueue2.c: (update_buffering),
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(gst_queue_locked_flush), (gst_queue_locked_enqueue),
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(gst_queue_handle_sink_event), (gst_queue_chain),
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(gst_queue_push_one), (gst_queue_sink_activate_push),
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(gst_queue_src_activate_push), (gst_queue_src_activate_pull):
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Also fix #476514 for queue2.
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2007-09-16 19:31:06 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/rtp/gstbasertpdepayload.c: Remove code to deal with RTP to GST time conversion, we now just copy the GST...
|
|
Original commit message from CVS:
|
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* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_init), (gst_base_rtp_depayload_setcaps),
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(gst_base_rtp_depayload_chain),
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(gst_base_rtp_depayload_handle_sink_event),
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(gst_base_rtp_depayload_push_full),
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(gst_base_rtp_depayload_set_gst_timestamp),
|
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(gst_base_rtp_depayload_change_state):
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Remove code to deal with RTP to GST time conversion, we now just copy
|
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the GST timestamp we receive to the outgoing buffers.
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Handle segment and flushes correctly.
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* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
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When we have no valid input timestamp, use the previous rtp timestamp on
|
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the outgoing RTP packet instead of the RTP base time.
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2007-09-16 01:56:21 +0000 David Schleef <ds@schleef.org>
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ext/alsa/: Change alsa alloca's to malloc to fix warnings on gcc-4.2.
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsa.c:
|
|
* ext/alsa/gstalsadeviceprobe.c:
|
|
* ext/alsa/gstalsamixer.c:
|
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* ext/alsa/gstalsasink.c:
|
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* ext/alsa/gstalsasrc.c:
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Change alsa alloca's to malloc to fix warnings on gcc-4.2.
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2007-09-15 18:41:27 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/rtp/gstbasertppayload.c: Add some debug info when negotiating caps.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertppayload.c:
|
|
(gst_basertppayload_set_outcaps), (gst_basertppayload_push):
|
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Add some debug info when negotiating caps.
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|
|
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2007-09-15 00:29:11 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst-libs/gst/rtp/gstrtpbuffer.c: A buffer with an empty payload is also a valid buffer.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
|
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A buffer with an empty payload is also a valid buffer.
|
|
|
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2007-09-14 20:52:00 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst-libs/gst/rtp/gstbasertppayload.c: Make sure we start our RTP timestamp from the random base RTP timestamp even if...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_event),
|
|
(gst_basertppayload_set_outcaps), (gst_basertppayload_push),
|
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(gst_basertppayload_change_state):
|
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Make sure we start our RTP timestamp from the random base RTP
|
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timestamp even if the buffer timestamp starts from some random value.
|
|
|
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2007-09-14 16:56:16 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
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Add simple exmple app to demonstrate starting and pausing live and non-live bins in a PLAYING pipeline.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* tests/examples/Makefile.am:
|
|
* tests/examples/dynamic/.cvsignore:
|
|
* tests/examples/dynamic/Makefile.am:
|
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* tests/examples/dynamic/addstream.c: (create_stream),
|
|
(pause_play_stream), (message_received), (eos_message_received),
|
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(perform_step), (main):
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Add simple exmple app to demonstrate starting and pausing live and
|
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non-live bins in a PLAYING pipeline.
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|
|
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2007-09-14 10:42:00 +0000 Julien Moutte <julien@moutte.net>
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|
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gst/typefind/gsttypefindfunctions.c: Add some typefind for QCP files (RFC #3625)
|
|
Original commit message from CVS:
|
|
2007-09-14 Julien MOUTTE <julien@moutte.net>
|
|
* gst/typefind/gsttypefindfunctions.c: (plugin_init): Add some
|
|
typefind for QCP files (RFC #3625)
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|
|
|
2007-09-13 22:52:09 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.c: Disable pull mode scheduling, we're not ready for it yet and it subtly breaks ...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_init):
|
|
Disable pull mode scheduling, we're not ready for it yet and it subtly
|
|
breaks a lot of things.
|
|
|
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2007-09-12 17:35:52 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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tests/check/elements/libvisual.c: Test all libvisual plugins, not just the first one; this reproduces bug #450336 qui...
|
|
Original commit message from CVS:
|
|
* tests/check/elements/libvisual.c:
|
|
Test all libvisual plugins, not just the first one; this reproduces
|
|
bug #450336 quite easily. Looks like a problem with the 'jess'
|
|
visualisation.
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|
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|
2007-09-12 17:15:12 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/: Add basic libvisual test case in an attempt to reproduce bug #450336.
|
|
Original commit message from CVS:
|
|
* tests/check/Makefile.am:
|
|
* tests/check/elements/.cvsignore:
|
|
* tests/check/elements/libvisual.c:
|
|
Add basic libvisual test case in an attempt to reproduce bug #450336.
|
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Doesn't reproduce that bug, but some other crasher instead (invalid
|
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free), at least with make elements/libvisual.forever and the bumscope
|
|
plugin on x86-64/gutsy. Leaving test disabled for now.
|
|
|
|
2007-09-12 08:38:21 +0000 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
gst/: Printf format fixes (#476128).
|
|
Original commit message from CVS:
|
|
Patch by: Peter Kjellerstedt <pkj at axis com>
|
|
* gst-libs/gst/app/gstappsink.c:
|
|
* gst/flv/gstflvdemux.c:
|
|
* gst/flv/gstflvparse.c:
|
|
* gst/interleave/deinterleave.c:
|
|
* gst/switch/gstswitch.c:
|
|
Printf format fixes (#476128).
|
|
|
|
2007-09-11 19:07:57 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
|
|
|
|
gst-libs/gst/rtsp/gstrtspconnection.c: Make sure we can not cancel in the middle of receiving a message.
|
|
Original commit message from CVS:
|
|
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
(gst_rtsp_connection_read_internal), (gst_rtsp_connection_read),
|
|
(read_body), (gst_rtsp_connection_receive):
|
|
Make sure we can not cancel in the middle of receiving a message.
|
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Fixes #475731.
|
|
|
|
2007-09-11 11:29:12 +0000 Josep Torra Valles <josep@fluendo.com>
|
|
|
|
gst/playback/gstplaybasebin.c: Increase upper limit for audio queue a bit; fixes preroll problem with playbin and dec...
|
|
Original commit message from CVS:
|
|
Patch by: Josep Torra Valles <josep@fluendo.com>
|
|
* gst/playback/gstplaybasebin.c:
|
|
Increase upper limit for audio queue a bit; fixes preroll problem
|
|
with playbin and decodebin2 when playing a quicktime trailer with
|
|
multichannel audio via http (#464666).
|
|
|
|
2007-09-10 22:10:54 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosrc.*: Allow othe clocks than the internal clock to be used for the pipeline.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
(gst_base_audio_src_class_init), (gst_base_audio_src_init),
|
|
(gst_base_audio_src_provide_clock),
|
|
(gst_base_audio_src_set_property),
|
|
(gst_base_audio_src_get_property), (gst_base_audio_src_create):
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.h:
|
|
Allow othe clocks than the internal clock to be used for the pipeline.
|
|
Add property to disable clock provide.
|
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API: GstBaseAudioSrc::provide-clock
|
|
|
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2007-09-10 12:05:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/playback/gstdecodebin2.c: Don't leak request pads. Fixes #475395.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin2.c:
|
|
Don't leak request pads. Fixes #475395.
|
|
|
|
2007-09-09 10:25:43 +0000 René Stadler <mail@renestadler.de>
|
|
|
|
sys/: Correctly chain up finalize with the parent class to prevent memory leaks. Fixes #474880.
|
|
Original commit message from CVS:
|
|
Patch by: René Stadler <mail at renestadler dot de>
|
|
* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
|
|
(gst_ximage_buffer_class_init):
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
|
|
(gst_xvimage_buffer_class_init):
|
|
Correctly chain up finalize with the parent class to prevent
|
|
memory leaks. Fixes #474880.
|
|
|
|
2007-09-09 04:08:48 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
Revert the latest change: floating point samples are allowed to have any value, not only values in the range [-1,1]. ...
|
|
Original commit message from CVS:
|
|
* gst/volume/gstvolume.c: (volume_choose_func):
|
|
* tests/check/elements/volume.c: (GST_START_TEST):
|
|
Revert the latest change: floating point samples are allowed to
|
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have any value, not only values in the range [-1,1]. Thanks to Andy
|
|
Wingo for noticing.
|
|
Also fix processing of int32 samples with volumes > 4 by making the
|
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unity value smaller which prevents overflows.
|
|
|
|
2007-09-07 17:37:03 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstrtpbuffer.c:
|
|
* tests/check/libs/rtp.c:
|
|
Fix gst_rtp_buffer_set_csrc() and enable csrc-related unit test checks.
|
|
|
|
2007-09-07 16:46:05 +0000 Haakon Sporsheim <haakon.sporsheim@tandberg.com>
|
|
|
|
gst-libs/gst/rtp/gstrtpbuffer.c: Fix up GstRTPHeader helper struct so that compilers will not under any circumstances...
|
|
Original commit message from CVS:
|
|
Based on patch by: Haakon Sporsheim <haakon.sporsheim at tandberg com>
|
|
* gst-libs/gst/rtp/gstrtpbuffer.c:
|
|
Fix up GstRTPHeader helper struct so that compilers will not under
|
|
any circumstances add padding in between our fields, as currently
|
|
happens with MSVC on win32, because that would lead to us sending
|
|
out RTP payloads with broken RTP headers (#471194).
|
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Fix assertion guards for gst_rtp_buffer_get_csrc() and _set_csrc().
|
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* tests/check/Makefile.am:
|
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* tests/check/libs/.cvsignore:
|
|
* tests/check/libs/rtp.c:
|
|
Add some simple unit tests for GstRTPBuffer. Some are disabled
|
|
because the code tested still needs fixing (set_csrc() does not work).
|
|
|
|
2007-09-07 15:05:24 +0000 Christian Schaller <uraeus@gnome.org>
|
|
|
|
* gst-plugins-base.spec.in:
|
|
update spec file to include latest RTSP libraries and headers and more
|
|
Original commit message from CVS:
|
|
update spec file to include latest RTSP libraries and headers and more
|
|
|
|
2007-09-07 12:41:01 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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win32/: Add rtsp enumtypes (#474384) and update others.
|
|
Original commit message from CVS:
|
|
* win32/MANIFEST:
|
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* win32/common/gstrtsp-enumtypes.c:
|
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* win32/common/gstrtsp-enumtypes.h:
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* win32/common/interfaces-enumtypes.c:
|
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* win32/common/interfaces-enumtypes.h:
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* win32/common/multichannel-enumtypes.c:
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Add rtsp enumtypes (#474384) and update others.
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|
|
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2007-09-06 20:31:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
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configure.ac: Fix configure check for HAVE_LIBXML_HTML.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Fix configure check for HAVE_LIBXML_HTML.
|
|
|
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2007-09-06 12:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
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tests/check/libs/.cvsignore: Ignore more, in case the build bots work again one day.
|
|
Original commit message from CVS:
|
|
* tests/check/libs/.cvsignore:
|
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Ignore more, in case the build bots work again one day.
|
|
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2007-09-06 07:00:36 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
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Add libgstfft, a FFT library based on Kiss FFT which is
|
|
Original commit message from CVS:
|
|
Reviewed by: Stefan Kost <ensonic@users.sf.net>
|
|
* configure.ac:
|
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* gst-libs/gst/Makefile.am:
|
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* gst-libs/gst/fft/Makefile.am:
|
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* gst-libs/gst/fft/_kiss_fft_guts_f32.h:
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* gst-libs/gst/fft/_kiss_fft_guts_f64.h:
|
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* gst-libs/gst/fft/_kiss_fft_guts_s16.h:
|
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* gst-libs/gst/fft/_kiss_fft_guts_s32.h:
|
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* gst-libs/gst/fft/gstfft.c: (gst_fft_next_fast_length):
|
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* gst-libs/gst/fft/gstfft.h:
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* gst-libs/gst/fft/gstfftf32.c: (gst_fft_f32_new),
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(gst_fft_f32_fft), (gst_fft_f32_inverse_fft), (gst_fft_f32_free),
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(gst_fft_f32_window), (gst_fft_f32_magnitude), (gst_fft_f32_phase):
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* gst-libs/gst/fft/gstfftf32.h:
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|
* gst-libs/gst/fft/gstfftf64.c: (gst_fft_f64_new),
|
|
(gst_fft_f64_fft), (gst_fft_f64_inverse_fft), (gst_fft_f64_free),
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|
(gst_fft_f64_window), (gst_fft_f64_magnitude), (gst_fft_f64_phase):
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* gst-libs/gst/fft/gstfftf64.h:
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* gst-libs/gst/fft/gstffts16.c: (gst_fft_s16_new),
|
|
(gst_fft_s16_fft), (gst_fft_s16_inverse_fft), (gst_fft_s16_free),
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(gst_fft_s16_window), (gst_fft_s16_magnitude), (gst_fft_s16_phase):
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* gst-libs/gst/fft/gstffts16.h:
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|
* gst-libs/gst/fft/gstffts32.c: (gst_fft_s32_new),
|
|
(gst_fft_s32_fft), (gst_fft_s32_inverse_fft), (gst_fft_s32_free),
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|
(gst_fft_s32_window), (gst_fft_s32_magnitude), (gst_fft_s32_phase):
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* gst-libs/gst/fft/gstffts32.h:
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|
* gst-libs/gst/fft/kiss_fft_f32.c: (kf_bfly2), (kf_bfly4),
|
|
(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
|
|
(kiss_fft_f32_alloc), (kiss_fft_f32_stride), (kiss_fft_f32),
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|
(kiss_fft_f32_cleanup), (kiss_fft_f32_next_fast_size):
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* gst-libs/gst/fft/kiss_fft_f32.h:
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|
* gst-libs/gst/fft/kiss_fft_f64.c: (kf_bfly2), (kf_bfly4),
|
|
(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
|
|
(kiss_fft_f64_alloc), (kiss_fft_f64_stride), (kiss_fft_f64),
|
|
(kiss_fft_f64_cleanup), (kiss_fft_f64_next_fast_size):
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* gst-libs/gst/fft/kiss_fft_f64.h:
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|
* gst-libs/gst/fft/kiss_fft_s16.c: (kf_bfly2), (kf_bfly4),
|
|
(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
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|
(kiss_fft_s16_alloc), (kiss_fft_s16_stride), (kiss_fft_s16),
|
|
(kiss_fft_s16_cleanup), (kiss_fft_s16_next_fast_size):
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* gst-libs/gst/fft/kiss_fft_s16.h:
|
|
* gst-libs/gst/fft/kiss_fft_s32.c: (kf_bfly2), (kf_bfly4),
|
|
(kf_bfly3), (kf_bfly5), (kf_bfly_generic), (kf_work), (kf_factor),
|
|
(kiss_fft_s32_alloc), (kiss_fft_s32_stride), (kiss_fft_s32),
|
|
(kiss_fft_s32_cleanup), (kiss_fft_s32_next_fast_size):
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|
* gst-libs/gst/fft/kiss_fft_s32.h:
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|
* gst-libs/gst/fft/kiss_fftr_f32.c: (kiss_fftr_f32_alloc),
|
|
(kiss_fftr_f32), (kiss_fftri_f32):
|
|
* gst-libs/gst/fft/kiss_fftr_f32.h:
|
|
* gst-libs/gst/fft/kiss_fftr_f64.c: (kiss_fftr_f64_alloc),
|
|
(kiss_fftr_f64), (kiss_fftri_f64):
|
|
* gst-libs/gst/fft/kiss_fftr_f64.h:
|
|
* gst-libs/gst/fft/kiss_fftr_s16.c: (kiss_fftr_s16_alloc),
|
|
(kiss_fftr_s16), (kiss_fftri_s16):
|
|
* gst-libs/gst/fft/kiss_fftr_s16.h:
|
|
* gst-libs/gst/fft/kiss_fftr_s32.c: (kiss_fftr_s32_alloc),
|
|
(kiss_fftr_s32), (kiss_fftri_s32):
|
|
* gst-libs/gst/fft/kiss_fftr_s32.h:
|
|
* gst-libs/gst/fft/kiss_version:
|
|
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
|
|
* pkgconfig/gstreamer-plugins-base.pc.in:
|
|
Add libgstfft, a FFT library based on Kiss FFT which is
|
|
BSD licensed. Supported sample formats are int16, int32,
|
|
float and double. For those formats a real FFT and IFFT
|
|
can be done, different windowing functions can be applied
|
|
and functions for extracting the magnitude and phase exist.
|
|
Fixes #468619.
|
|
* docs/libs/Makefile.am:
|
|
* docs/libs/gst-plugins-base-libs-docs.sgml:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
Integrate libgstfft into the docs.
|
|
* tests/check/Makefile.am:
|
|
* tests/check/libs/fft.c: (GST_START_TEST), (fft_suite), (main):
|
|
Add unit tests for libgstfft, currently only testing the FFT.
|
|
Unit tests for IFFT will follow soon.
|
|
|
|
2007-09-05 23:07:40 +0000 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
gst-libs/gst/sdp/gstsdpmessage.*: Separate INIT_ARRAY() and related macros into two versions, one for structures and ...
|
|
Original commit message from CVS:
|
|
Patch by: Peter Kjellerstedt <pkj at axis com>
|
|
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_time_init),
|
|
(gst_sdp_message_init), (gst_sdp_message_uninit),
|
|
(is_multicast_address), (gst_sdp_message_as_text),
|
|
(gst_sdp_message_get_origin), (gst_sdp_message_set_connection),
|
|
(gst_sdp_message_get_connection), (gst_sdp_message_add_bandwidth),
|
|
(gst_sdp_message_add_time), (gst_sdp_message_add_zone),
|
|
(gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
|
|
(gst_sdp_message_get_attribute_val), (gst_sdp_message_add_media),
|
|
(gst_sdp_media_init), (gst_sdp_media_uninit),
|
|
(gst_sdp_media_as_text), (gst_sdp_media_set_port_info),
|
|
(gst_sdp_media_connections_len), (gst_sdp_media_add_connection),
|
|
(gst_sdp_media_bandwidths_len), (gst_sdp_media_get_bandwidth),
|
|
(gst_sdp_media_add_bandwidth), (gst_sdp_media_attributes_len),
|
|
(gst_sdp_parse_line), (print_media), (gst_sdp_message_dump):
|
|
* gst-libs/gst/sdp/gstsdpmessage.h:
|
|
Separate INIT_ARRAY() and related macros into two versions, one for
|
|
structures and one for pointers (e.g., INIT_ARRAY() and
|
|
INIT_PTR_ARRAY()). This fixes a segmentation error on freeing the
|
|
lists of emails and phone numbers.
|
|
Add missing const as appropriate.
|
|
Change all gint to guint since they all actually represent unsigned
|
|
values.
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|
Do not use time as a variable name as it shadows the global time().
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|
Add gst_sdp_message_as_text() and gst_sdp_media_as_text().
|
|
Actually implement gst_sdp_message_add_time().
|
|
Make gst_sdp_message_add_time() take repeat times as an argument.
|
|
Store repeat times in GstSDPTime as a GArray rather than as gchar**.
|
|
Corrected the definition of gst_sdp_media_get_bandwidth() (was
|
|
misspelled as badwidth).
|
|
gst-indented and a little clean up. Fixes #471067.
|
|
|
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2007-09-05 21:20:12 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/volume/gstvolume.c: Correctly clamp float/double samples in the [-1.0,1.0] range to prevent weird effects.
|
|
Original commit message from CVS:
|
|
* gst/volume/gstvolume.c: (volume_choose_func),
|
|
(volume_process_double), (volume_process_double_clamp),
|
|
(volume_process_float_clamp):
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|
Correctly clamp float/double samples in the [-1.0,1.0] range to
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|
prevent weird effects.
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* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
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Add unit tests for all samples types that had none before.
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|
|
|
2007-09-05 14:09:15 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
|
gst-libs/gst/rtp/gstrtpbuffer.c: Need to include stdlib.h for abs() here too.
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|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstrtpbuffer.c:
|
|
Need to include stdlib.h for abs() here too.
|
|
|
|
2007-09-05 14:01:25 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/playback/gststreaminfo.c: Fix build.
|
|
Original commit message from CVS:
|
|
* gst/playback/gststreaminfo.c:
|
|
Fix build.
|
|
|
|
2007-09-05 10:32:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/playback/gststreaminfo.c: Clean up some half-disabled code and comment.
|
|
Original commit message from CVS:
|
|
* gst/playback/gststreaminfo.c:
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|
Clean up some half-disabled code and comment.
|
|
|
|
2007-09-04 16:18:48 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstbasertpaudiopayload.c: Return FALSE from the event handler to let the parent class handle the event.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
(gst_base_rtp_payload_audio_handle_event):
|
|
Return FALSE from the event handler to let the parent class handle the
|
|
event.
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_chain), (gst_base_rtp_depayload_push_full):
|
|
Mark outgoing buffers as DISCONT if the incomming buffer was DISCONT.
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|
* gst-libs/gst/rtp/gstbasertppayload.c:
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|
Bump the MTU to 1400.
|
|
|
|
2007-09-04 01:50:55 +0000 Johan Dahlin <johan@gnome.org>
|
|
|
|
gst/typefind/gsttypefindfunctions.c (plugin_init): Add an audio/x-nsf typefind function for the nsfdec element.
|
|
Original commit message from CVS:
|
|
2007-09-03 Johan Dahlin <jdahlin@async.com.br>
|
|
* gst/typefind/gsttypefindfunctions.c (plugin_init):
|
|
Add an audio/x-nsf typefind function for the nsfdec element.
|
|
|
|
2007-09-03 20:46:38 +0000 Renato Filho <renato.filho@indt.org.br>
|
|
|
|
gst/playback/gstplaybasebin.c: Included "myth://" on stream_uris list for enable buffering to mythtv files
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c:
|
|
Included "myth://" on stream_uris list for enable buffering to mythtv files
|
|
|
|
2007-09-03 19:31:11 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
Fix parsing of RB blocks.
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
|
|
(gst_rtcp_packet_sdes_copy_entry), (gst_rtcp_ntp_to_unix),
|
|
(gst_rtcp_unix_to_ntp):
|
|
* gst-libs/gst/rtp/gstrtcpbuffer.h:
|
|
Fix parsing of RB blocks.
|
|
Fix docs.
|
|
Added helper functions to convert to/from UNIX and NTP time.
|
|
API: gst_rtcp_ntp_to_unix()
|
|
API: gst_rtcp_unix_to_ntp()
|
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* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data),
|
|
(gst_rtp_buffer_get_header_len),
|
|
(gst_rtp_buffer_get_extension_data),
|
|
(gst_rtp_buffer_get_payload_subbuffer),
|
|
(gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload),
|
|
(gst_rtp_buffer_ext_timestamp):
|
|
* gst-libs/gst/rtp/gstrtpbuffer.h:
|
|
Fix some more docs.
|
|
Implement handling of packets with extensions.
|
|
Fix padding check in _validate().
|
|
Added function to get extension data.
|
|
API: gst_rtp_buffer_get_header_len()
|
|
API: gst_rtp_buffer_get_extension_data()
|
|
|
|
2007-09-03 19:19:35 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstbasertpdepayload.c: Add some more docs for the queue-delay property and fix a typo in a comment.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_class_init),
|
|
(gst_base_rtp_depayload_set_gst_timestamp):
|
|
Add some more docs for the queue-delay property and fix a typo in a
|
|
comment.
|
|
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
|
|
Fix typo.
|
|
|
|
2007-09-03 19:17:33 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.c: When skew slaving, try to hover around the middle of a segment so that we at m...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_skew_slaving), (gst_base_audio_sink_render),
|
|
(gst_base_audio_sink_change_state):
|
|
When skew slaving, try to hover around the middle of a segment so that
|
|
we at most drift by half a segment.
|
|
If we are aligning in the oposite direction of the clock skew, we don't
|
|
have to resync.
|
|
|
|
2007-08-31 21:07:20 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstbasertpdepayload.c: Be less silly with the segment start, just apply the clock-base to the timest...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_setcaps),
|
|
(gst_base_rtp_depayload_set_gst_timestamp):
|
|
Be less silly with the segment start, just apply the clock-base to the
|
|
timestamp.
|
|
|
|
2007-08-31 15:58:30 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstbasertpdepayload.*: Deprecate the queue handling thread thing and remove the code.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_class_init),
|
|
(gst_base_rtp_depayload_finalize),
|
|
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
|
|
(gst_base_rtp_depayload_handle_sink_event),
|
|
(gst_base_rtp_depayload_set_gst_timestamp),
|
|
(gst_base_rtp_depayload_change_state):
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.h:
|
|
Deprecate the queue handling thread thing and remove the code.
|
|
Use new method to calculate the extended timestamp.
|
|
|
|
2007-08-31 15:21:13 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstrtcpbuffer.c: Use g_strndup which does exactly what we want.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstrtcpbuffer.c:
|
|
(gst_rtcp_packet_sdes_copy_entry):
|
|
Use g_strndup which does exactly what we want.
|
|
* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_compare_seqnum),
|
|
(gst_rtp_buffer_ext_timestamp):
|
|
* gst-libs/gst/rtp/gstrtpbuffer.h:
|
|
Add helper function to compare seqnums.
|
|
Add helper function to calculate extended timestamps.
|
|
API: gst_rtp_buffer_compare_seqnum()
|
|
API: gst_rtp_buffer_ext_timestamp()
|
|
|
|
2007-08-30 21:59:23 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstrtcpbuffer.*: Fix and document SDES item data function.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstrtcpbuffer.c:
|
|
(gst_rtcp_packet_sdes_get_entry),
|
|
(gst_rtcp_packet_sdes_copy_entry):
|
|
* gst-libs/gst/rtp/gstrtcpbuffer.h:
|
|
Fix and document SDES item data function.
|
|
Add new function that makes a proper copy of SDES item data.
|
|
API: gst_rtcp_packet_sdes_copy_entry()
|
|
|
|
2007-08-30 07:29:55 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
The tcp and subparse plugins are under gst, but not totaly free of dependencies. Handle selection inconfigure.ac, so ...
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* gst/Makefile.am:
|
|
The tcp and subparse plugins are under gst, but not totaly free of
|
|
dependencies. Handle selection inconfigure.ac, so that they show up
|
|
on the final list of what is build and what is not. Maybe they should
|
|
better be moved to ext.
|
|
|
|
2007-08-30 06:58:46 +0000 Daniel Díaz <yosoy@danieldiaz.org>
|
|
|
|
Check if libxml provides HTML parser which subparse needs.
|
|
Original commit message from CVS:
|
|
Patch by: Daniel Díaz <yosoy@danieldiaz.org>
|
|
* configure.ac:
|
|
* gst/Makefile.am:
|
|
Check if libxml provides HTML parser which subparse needs.
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|
Fixes #451970.
|
|
|
|
2007-08-29 14:22:04 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/alsa/gstalsa.c: Fix typo and compilation on big endian systems.
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsa.c:
|
|
Fix typo and compilation on big endian systems.
|
|
|
|
2007-08-29 12:16:46 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/subparse/gstssaparse.c: Convert SSA newline codes into actual newline characters (#470766).
|
|
Original commit message from CVS:
|
|
* gst/subparse/gstssaparse.c:
|
|
Convert SSA newline codes into actual newline characters (#470766).
|
|
|
|
2007-08-28 14:58:17 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
API: also add gst_install_plugins_supported() while we're at it (see #470456).
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/pbutils/install-plugins.c:
|
|
* gst-libs/gst/pbutils/install-plugins.h:
|
|
* tests/check/libs/pbutils.c:
|
|
API: also add gst_install_plugins_supported() while we're at it
|
|
(see #470456).
|
|
|
|
2007-08-28 14:23:55 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
API: add gst_missing_*_installer_detail_new() convenience API so that applications that know exactly what they're mis...
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/pbutils/missing-plugins.c:
|
|
* gst-libs/gst/pbutils/missing-plugins.h:
|
|
* tests/check/libs/pbutils.c:
|
|
API: add gst_missing_*_installer_detail_new() convenience API so
|
|
that applications that know exactly what they're missing can request
|
|
installer detail strings for those items directly instead of having
|
|
to first create a dummy missing-plugin message and then get the
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installer detail string from that. Fixes #470456.
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|
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2007-08-27 11:59:56 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
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gst/playback/gstdecodebin.c: We need to set up delayed-linking whenever the caps are non-fixed, not just when there a...
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Original commit message from CVS:
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* gst/playback/gstdecodebin.c: (close_pad_link):
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We need to set up delayed-linking whenever the caps are non-fixed,
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not just when there are multiple types - use gst_pad_is_fixed()
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to test.
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2007-08-26 14:14:33 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/pbutils/missing-plugins.c: Add missing separator in PID fallback case.
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Original commit message from CVS:
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* gst-libs/gst/pbutils/missing-plugins.c:
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(gst_missing_plugin_message_get_installer_detail):
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Add missing separator in PID fallback case.
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2007-08-24 15:28:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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ext/alsa/Makefile.am: There is no GST_PLUGINS_BASE_LIBS defined.
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Original commit message from CVS:
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* ext/alsa/Makefile.am:
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There is no GST_PLUGINS_BASE_LIBS defined.
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* ext/alsa/gstalsa.c:
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* ext/alsa/gstalsasink.c: (gst_alsasink_delay):
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* ext/alsa/gstalsasrc.c: (gst_alsasrc_delay):
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Add support for ALSA 24-bit formats.
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snd_pcm_delay can return an error code, especially
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during XRUNS. In that case, the best we can do is assume
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delay = 0.
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* gst/audioconvert/Makefile.am:
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Add flags from -base before any more-remote dependencies.
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2007-08-23 20:45:45 +0000 Davyd <davyd@madeley.id.au>
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gst/volume/gstvolume.*: Add support for int32, int24 and int8 to the volume element.
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Original commit message from CVS:
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Based on a patch by: Davyd <davyd at madeley dot id dot au>
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* gst/volume/gstvolume.c: (volume_choose_func),
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(volume_update_real_volume), (gst_volume_set_volume),
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(gst_volume_init), (volume_process_int32),
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(volume_process_int32_clamp), (volume_process_int24),
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(volume_process_int24_clamp), (volume_process_int16),
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(volume_process_int16_clamp), (volume_process_int8),
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(volume_process_int8_clamp), (volume_update_volume), (plugin_init):
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* gst/volume/gstvolume.h:
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Add support for int32, int24 and int8 to the volume element.
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Fixes #445529.
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2007-08-23 12:37:42 +0000 Tim-Philipp Müller <tim@centricular.net>
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tests/examples/Makefile.am: Fix even more.
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Original commit message from CVS:
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* tests/examples/Makefile.am:
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Fix even more.
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2007-08-23 10:58:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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Revert unwanted commit. many thanks to moap. I want a fix for https://thomas.apestaart.org/moap/trac/ticket/239
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Original commit message from CVS:
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* configure.ac:
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* docs/libs/Makefile.am:
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* docs/libs/gst-plugins-base-libs-docs.sgml:
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* docs/libs/gst-plugins-base-libs-sections.txt:
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* ext/gnomevfs/gstgnomevfssrc.c:
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* ext/gnomevfs/gstgnomevfssrc.h:
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* gst-libs/gst/Makefile.am:
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* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
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* pkgconfig/gstreamer-plugins-base.pc.in:
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* sys/v4l/v4lsrc_calls.c:
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* tests/examples/Makefile.am:
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* win32/common/config.h:
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Revert unwanted commit. many thanks to moap. I want a fix for
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https://thomas.apestaart.org/moap/trac/ticket/239
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2007-08-23 08:33:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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Original commit message from CVS:
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reviewed by: <delete if not using a buddy>
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patch by: <delete if not someone else's patch>
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* configure.ac:
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* docs/libs/Makefile.am:
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* docs/libs/gst-plugins-base-libs-docs.sgml:
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* docs/libs/gst-plugins-base-libs-sections.txt:
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* ext/gnomevfs/gstgnomevfssrc.c:
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* ext/gnomevfs/gstgnomevfssrc.h:
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* gst-libs/gst/Makefile.am:
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* gst-libs/gst/audio/gstaudiofilter.h:
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* gst/typefind/gsttypefindfunctions.c:
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* gst/volume/gstvolume.c:
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* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
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* pkgconfig/gstreamer-plugins-base.pc.in:
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* sys/v4l/v4lsrc_calls.c:
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* tests/examples/Makefile.am:
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* win32/common/config.h:
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2007-08-22 15:29:04 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/audio.c: Clarify the docs a little.
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Original commit message from CVS:
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* gst-libs/gst/audio/audio.c:
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Clarify the docs a little.
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2007-08-22 11:20:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst/volume/gstvolume.c: Enable liboil for float and add more details about problems with int16.
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Original commit message from CVS:
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* gst/volume/gstvolume.c:
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Enable liboil for float and add more details about problems with
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int16.
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2007-08-21 15:43:24 +0000 Wim Taymans <wim.taymans@gmail.com>
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sys/v4l/gstv4lsrc.c: Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
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Original commit message from CVS:
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* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_palette_to_caps):
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Fix RGB24 masks as spotted by _ke (Daniel G. Siegel) on IRC.
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2007-08-21 12:08:43 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/vorbis/vorbisdec.c: When calculating the first timestamp of the buffers, don't go below 0 and clip the samples be...
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|
Original commit message from CVS:
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* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
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When calculating the first timestamp of the buffers, don't go below 0
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and clip the samples because the offset was on the eos page.
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Fixes #466717.
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2007-08-21 11:42:39 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/ogg/gstoggdemux.c: Also submit the eos page when trying to find the first timestamp.
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|
Original commit message from CVS:
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* ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain),
|
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(gst_ogg_demux_collect_chain_info):
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Also submit the eos page when trying to find the first timestamp.
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See #466717.
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2007-08-17 15:24:43 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst-libs/gst/audio/audio.h: Use gst_util_uint64_scale() instead of doing the math with double for GST_FRAMES_TO_CLOCK...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/audio.h:
|
|
Use gst_util_uint64_scale() instead of doing the math
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|
with double for GST_FRAMES_TO_CLOCK_TIME() and
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GST_CLOCK_TIME_TO_FRAMES(). For large timestamps this
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prevents rounding errors. Fixes #467667.
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2007-08-17 13:42:49 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/rtsp/gstrtspconnection.*: Small cleanups.
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|
Original commit message from CVS:
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|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
(gst_rtsp_connection_connect), (gst_rtsp_connection_write),
|
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(gst_rtsp_connection_read), (gst_rtsp_connection_poll):
|
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* gst-libs/gst/rtsp/gstrtspconnection.h:
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Small cleanups.
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On shutdown, don't read the control socket yet.
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Set timeout value correctly in all cases.
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Add function to check if the server accepts reads or writes.
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API: gst_rtsp_connection_poll()
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* gst-libs/gst/rtsp/gstrtspdefs.h:
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Fix compilation with -pedantic.
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Add enum for _poll.
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2007-08-16 17:11:48 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/app/gstappsink.c: Override the preroll vmethod instead of overriding the render method twice.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_class_init):
|
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Override the preroll vmethod instead of overriding the render method
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twice.
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2007-08-16 16:06:21 +0000 Olivier Crete <tester@tester.ca>
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gst-libs/gst/rtp/gstbasertppayload.*: Add getcaps vfunc to basertppayload. See #465146.
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|
Original commit message from CVS:
|
|
Patch by: Olivier Crete <tester at tester ca>
|
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* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
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(gst_basertppayload_getcaps):
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* gst-libs/gst/rtp/gstbasertppayload.h:
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Add getcaps vfunc to basertppayload. See #465146.
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2007-08-16 11:20:56 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstplaybasebin.c: Only post buffering messages when we are a stream.
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|
Original commit message from CVS:
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* gst/playback/gstplaybasebin.c: (queue_threshold_reached):
|
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Only post buffering messages when we are a stream.
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|
|
|
2007-08-15 17:05:45 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
|
gst-libs/gst/pbutils/: Small docs fix and addition.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/pbutils/install-plugins.c:
|
|
* gst-libs/gst/pbutils/missing-plugins.c:
|
|
Small docs fix and addition.
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2007-08-14 17:47:34 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/app/gstappsink.c: Don't use new API.
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|
Original commit message from CVS:
|
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* gst-libs/gst/app/gstappsink.c: (gst_app_sink_flush_unlocked):
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Don't use new API.
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|
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2007-08-14 17:38:05 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst-libs/gst/app/gstappsink.*: Make love to appsink.
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|
Original commit message from CVS:
|
|
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_base_init),
|
|
(gst_app_sink_class_init), (gst_app_sink_dispose),
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(gst_app_sink_flush_unlocked), (gst_app_sink_start),
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(gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
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(gst_app_sink_render), (gst_app_sink_get_caps),
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|
(gst_app_sink_set_caps), (gst_app_sink_end_of_stream),
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|
(gst_app_sink_pull_preroll), (gst_app_sink_pull_buffer):
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* gst-libs/gst/app/gstappsink.h:
|
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Make love to appsink.
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|
Make it support pulling of the preroll buffer.
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Add docs and debug statements.
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Fix some races wrt to EOS handling and stopping.
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Implement getcaps.
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Implement FLUSHING.
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API: gst_app_sink_pull_preroll()
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2007-08-13 15:37:29 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/icles/: Add a dumb little test for textoverlay alignments.
|
|
Original commit message from CVS:
|
|
* tests/icles/.cvsignore:
|
|
* tests/icles/Makefile.am:
|
|
* tests/icles/test-textoverlay.c:
|
|
Add a dumb little test for textoverlay alignments.
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|
|
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2007-08-13 15:26:54 +0000 Dan Williams <dcbw@redhat.com>
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ext/pango/gsttextoverlay.*: API: add "line-alignment" property (#459334). Add gtk-doc blurb for "silent" property so ...
|
|
Original commit message from CVS:
|
|
Patch by: Dan Williams <dcbw redhat com>
|
|
* ext/pango/gsttextoverlay.c:
|
|
* ext/pango/gsttextoverlay.h:
|
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API: add "line-alignment" property (#459334). Add gtk-doc blurb for
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"silent" property so there's a Since tag in the API reference.
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2007-08-13 11:21:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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|
|
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* ChangeLog:
|
|
fix ... by: lines
|
|
Original commit message from CVS:
|
|
fix ... by: lines
|
|
|
|
2007-08-12 16:30:36 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstbasertppayload.*: Improve caps negotiation so that downstream elements can confiure certain RTP p...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertppayload.c:
|
|
(gst_basertppayload_set_outcaps):
|
|
* gst-libs/gst/rtp/gstbasertppayload.h:
|
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Improve caps negotiation so that downstream elements can confiure
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certain RTP properties by fixing them on the caps. See #465146.
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Add docs.
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|
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2007-08-11 12:39:51 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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Mark as deprecated some macros which were presumably meant to be private API and accidentally exposed in the public h...
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.h:
|
|
Mark as deprecated some macros which were presumably meant to be
|
|
private API and accidentally exposed in the public header file.
|
|
Also actually _init() lock (only works at the moment because the
|
|
struct is zeroed out when created and the initial values in the
|
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mutex struct are zeroes too). (#459585)
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|
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2007-08-10 17:35:52 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
docs/libs/Makefile.am: Remove cruft and do some cleanups.
|
|
Original commit message from CVS:
|
|
* docs/libs/Makefile.am:
|
|
Remove cruft and do some cleanups.
|
|
* docs/libs/gst-plugins-base-libs-docs.sgml:
|
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Prepare for comming gtkdoc features (rebase against online docs).
|
|
|
|
2007-08-10 13:55:44 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/audiorate/gstaudiorate.c: Debug output fixes.
|
|
Original commit message from CVS:
|
|
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
|
|
Debug output fixes.
|
|
* tests/check/elements/audiorate.c: (do_perfect_stream_test),
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(GST_START_TEST):
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Change the number of buffers used; 500 is too many and leads to
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timeouts.
|
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|
|
2007-08-10 10:08:05 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/: Printf format fixes (#465028).
|
|
Original commit message from CVS:
|
|
* gst/playback/gstqueue2.c:
|
|
* gst/videorate/gstvideorate.c:
|
|
Printf format fixes (#465028).
|
|
|
|
2007-08-09 15:44:02 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/audiorate/gstaudiorate.c: If we have a large (> 1 second) discontinuity, push a series of smaller buffers rather ...
|
|
Original commit message from CVS:
|
|
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
|
|
If we have a large (> 1 second) discontinuity, push a series of
|
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smaller buffers rather than a single very large buffer. Avoids
|
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unreasonably large single buffer allocations when encountering a
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large gap.
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* tests/check/elements/audiorate.c: (GST_START_TEST),
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(audiorate_suite):
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Add a test for this.
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|
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2007-08-09 12:06:43 +0000 Josep Torra Valles <josep@fluendo.com>
|
|
|
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gst/playback/gstplaybasebin.c: Fixes: #465015
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (group_commit),
|
|
(queue_remove_probe), (queue_threshold_reached):
|
|
Patch by: Josep Torra Valles <josep@fluendo.com>
|
|
Fixes: #465015
|
|
Make sure we remove the check_queues buffer probe from the
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correct queue to avoid racily going back to "buffering 99%" when
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buffering is actually complete.
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Also, fix the spelling of Josep's surname in the ChangeLog.
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2007-08-09 11:37:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
ext/ogg/gstoggmux.c: Do not leak oggmux instance.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggmux.c:
|
|
Do not leak oggmux instance.
|
|
* ext/vorbis/vorbisenc.c:
|
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Also log values.
|
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|
|
2007-08-09 10:51:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
po/: Updated translations.
|
|
Original commit message from CVS:
|
|
* po/hu.po:
|
|
* po/it.po:
|
|
* po/nl.po:
|
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* po/uk.po:
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* po/vi.po:
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Updated translations.
|
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2007-08-08 16:07:21 +0000 Yang Hong <hongyang@redflag-linux.com>
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ext/pango/gsttextoverlay.*: Add 'silent' property to GstTimeOverlay. Fixes #462979
|
|
Original commit message from CVS:
|
|
patch by: Yang Hong <hongyang@redflag-linux.com>
|
|
* ext/pango/gsttextoverlay.c:
|
|
* ext/pango/gsttextoverlay.h:
|
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Add 'silent' property to GstTimeOverlay. Fixes #462979
|
|
|
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2007-08-08 15:05:22 +0000 Josep Torre Valles <josep@fluendo.com>
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|
|
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Add connection-speed property. Fixes #464690.
|
|
Original commit message from CVS:
|
|
Patch by: Josep Torre Valles <josep@fluendo.com>
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init),
|
|
(gst_uri_decode_bin_init), (gst_uri_decode_bin_set_property),
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(gst_uri_decode_bin_get_property), (gen_source_element):
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Add connection-speed property. Fixes #464690.
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2007-08-07 15:13:46 +0000 Damien Lespiau <damien.lespiau@gmail.com>
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Fix compilation on windows. Fixes #464320.
|
|
Original commit message from CVS:
|
|
Patch by: Damien Lespiau <damien dot lespiau at gmail dot com>
|
|
* configure.ac:
|
|
* gst-libs/gst/rtsp/Makefile.am:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
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(gst_rtsp_connection_connect):
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Fix compilation on windows. Fixes #464320.
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2007-08-07 14:14:54 +0000 Josep Torre Valles <josep@fluendo.com>
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gst/playback/: Move connection-speed property from playbin to playbasebin so that we can also configure it in source ...
|
|
Original commit message from CVS:
|
|
Patch by: Josep Torre Valles <josep@fluendo.com>
|
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* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
|
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(gst_play_base_bin_init), (queue_threshold_reached),
|
|
(gen_source_element), (setup_substreams),
|
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(gst_play_base_bin_set_property), (gst_play_base_bin_get_property),
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(gst_play_base_bin_get_streaminfo_value_array):
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* gst/playback/gstplaybasebin.h:
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* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
|
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(gst_play_bin_set_property), (gst_play_bin_get_property),
|
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(gst_play_bin_handle_redirect_message):
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|
Move connection-speed property from playbin to playbasebin so that we
|
|
can also configure it in source elements that have the connection-speed
|
|
property. Fixes #464028.
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|
Add some debug info here and there.
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|
|
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2007-08-06 16:42:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
|
|
gst/audiotestsrc/gstaudiotestsrc.c: Properly respond to conversion queries. Fixes #464079.
|
|
Original commit message from CVS:
|
|
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
|
|
Properly respond to conversion queries. Fixes #464079.
|
|
|
|
2007-08-03 19:53:11 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/audiotestsrc/gstaudiotestsrc.*: Add float/double and int32 support to audiotestsrc. Fixes #460422.
|
|
Original commit message from CVS:
|
|
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_init),
|
|
(gst_audio_test_src_src_fixate), (gst_audio_test_src_setcaps),
|
|
(gst_audio_test_src_init_sine_table),
|
|
(gst_audio_test_src_change_wave), (gst_audio_test_src_create):
|
|
* gst/audiotestsrc/gstaudiotestsrc.h:
|
|
Add float/double and int32 support to audiotestsrc. Fixes #460422.
|
|
Also set the default volume to the default value specified in the
|
|
GParamSpec.
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|
|
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2007-08-03 19:40:14 +0000 Jens Granseuer <jensgr@gmx.net>
|
|
|
|
gst/audioconvert/gstaudioquantize.c: Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
|
|
Original commit message from CVS:
|
|
Patch by: Jens Granseuer <jensgr at gmx dot net>
|
|
* gst/audioconvert/gstaudioquantize.c:
|
|
Fix C89 incompatibilities and spelling of explanations. Fixes #463215.
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|
|
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2007-08-03 15:44:01 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst-libs/gst/rtsp/gstrtsptransport.c: Add rdt manager for rdt transport.
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|
Original commit message from CVS:
|
|
* gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_parse):
|
|
Add rdt manager for rdt transport.
|
|
Fix parsing of RDT transport.
|
|
|
|
2007-08-03 14:43:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
configure.ac: Back to CVS
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Back to CVS
|
|
|
|
=== release 0.10.14 ===
|
|
|
|
2007-08-03 14:41:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
|
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* ChangeLog:
|
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* NEWS:
|
|
* RELEASE:
|
|
* configure.ac:
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.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-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-decodebin2.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gdp.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.xml:
|
|
* docs/plugins/inspect/plugin-libvisual.xml:
|
|
* docs/plugins/inspect/plugin-ogg.xml:
|
|
* docs/plugins/inspect/plugin-pango.xml:
|
|
* docs/plugins/inspect/plugin-playbin.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-video4linux.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/config.h:
|
|
Release 0.10.14
|
|
Original commit message from CVS:
|
|
Release 0.10.14
|
|
|
|
2007-08-03 14:24:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/cs.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/hu.po:
|
|
* po/it.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
Update .po files
|
|
Original commit message from CVS:
|
|
Update .po files
|
|
|
|
2007-07-27 17:37:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
tests/check/libs/audio.c: Fix the test to reflect the behaviour of gst_audio_clip_buffer.
|
|
Original commit message from CVS:
|
|
* tests/check/libs/audio.c: (GST_START_TEST):
|
|
Fix the test to reflect the behaviour of gst_audio_clip_buffer.
|
|
|
|
2007-07-27 17:10:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/audio/audio.c: When clipping a buffer with no timestamp, assume it is within the segment without warnings.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/audio.c:
|
|
When clipping a buffer with no timestamp, assume it is
|
|
within the segment without warnings.
|
|
Fixes: #460978
|
|
|
|
2007-07-27 11:16:23 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtsp/gstrtspextension.c: Fire the signal on the object, not the interface.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtsp/gstrtspextension.c: (gst_rtsp_extension_send):
|
|
Fire the signal on the object, not the interface.
|
|
|
|
2007-07-27 09:17:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/rtsp/.cvsignore: Ber. Don't include the full path, idiot.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtsp/.cvsignore:
|
|
Ber. Don't include the full path, idiot.
|
|
|
|
2007-07-27 08:29:29 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/rtsp/.cvsignore: Ignore generated files.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtsp/.cvsignore:
|
|
Ignore generated files.
|
|
|
|
2007-07-26 19:57:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/: Move the rtspextension.h interface into gstrtspextension.h as part of libgstrtsp instead of libgstinte...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/Makefile.am:
|
|
* gst-libs/gst/interfaces/interfaces-marshal.list:
|
|
* gst-libs/gst/interfaces/rtspextension.c:
|
|
* gst-libs/gst/interfaces/rtspextension.h:
|
|
* gst-libs/gst/rtsp/Makefile.am:
|
|
* gst-libs/gst/rtsp/gstrtsp.h:
|
|
* gst-libs/gst/rtsp/gstrtspextension.c:
|
|
(gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
|
|
(gst_rtsp_extension_detect_server),
|
|
(gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
|
|
(gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
|
|
(gst_rtsp_extension_configure_stream),
|
|
(gst_rtsp_extension_get_transports),
|
|
(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
|
|
* gst-libs/gst/rtsp/gstrtspextension.h:
|
|
* gst-libs/gst/rtsp/rtsp-marshal.list:
|
|
Move the rtspextension.h interface into gstrtspextension.h
|
|
as part of libgstrtsp instead of libgstinterfaces, because it's
|
|
only for use within plugins, not applications.
|
|
Add stuff to do the enum & marshal generation needed in libgstrtsp now.
|
|
Use the GST_TYPE_RTSP_RESULT enum type for the return value of the
|
|
signal that the GstRTSPExtension interface emits, since G_TYPE_ENUM
|
|
is abstract.
|
|
|
|
2007-07-26 15:48:01 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/interfaces/: Fix marshaller for the send signal.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/Makefile.am:
|
|
* gst-libs/gst/interfaces/interfaces-marshal.list:
|
|
* gst-libs/gst/interfaces/rtspextension.c:
|
|
(gst_rtsp_extension_iface_init),
|
|
(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
|
|
* gst-libs/gst/interfaces/rtspextension.h:
|
|
Fix marshaller for the send signal.
|
|
Add URL to stream selection interface method.
|
|
|
|
2007-07-26 15:35:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/riff/Makefile.am: Pull in our dependencies from -base before those from outside.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/Makefile.am:
|
|
Pull in our dependencies from -base before those from outside.
|
|
|
|
2007-07-26 14:33:01 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
API: gst_rtsp_base64_decode_ip()
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_decode_ip):
|
|
* gst-libs/gst/rtsp/gstrtspbase64.h:
|
|
API: gst_rtsp_base64_decode_ip()
|
|
Added function to decode Base64 in-place.
|
|
|
|
2007-07-26 14:08:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
tests/check/libs/.cvsignore: Ignore the mixer test binary.
|
|
Original commit message from CVS:
|
|
* tests/check/libs/.cvsignore:
|
|
Ignore the mixer test binary.
|
|
|
|
2007-07-26 10:00:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
ext/vorbis/vorbisdec.c: Gratuitous comment change to trigger a rebuild on the buildbots.
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward):
|
|
Gratuitous comment change to trigger a rebuild on the buildbots.
|
|
|
|
2007-07-25 18:20:36 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/sdp/gstsdpmessage.*: Constify args where we can.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_media_get_media),
|
|
(gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
|
|
(gst_sdp_media_get_proto), (gst_sdp_media_formats_len),
|
|
(gst_sdp_media_get_format), (gst_sdp_media_get_information),
|
|
(gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
|
|
(gst_sdp_media_bandwidths_len), (gst_sdp_media_get_badwidth),
|
|
(gst_sdp_media_get_key), (gst_sdp_media_attributes_len),
|
|
(gst_sdp_media_get_attribute), (gst_sdp_media_get_attribute_val_n),
|
|
(gst_sdp_media_get_attribute_val):
|
|
* gst-libs/gst/sdp/gstsdpmessage.h:
|
|
Constify args where we can.
|
|
|
|
2007-07-25 18:18:49 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/interfaces/: Move interface for RTSP extensions from -good to here.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/Makefile.am:
|
|
* gst-libs/gst/interfaces/rtspextension.c:
|
|
(gst_rtsp_extension_get_type), (gst_rtsp_extension_iface_init),
|
|
(gst_rtsp_extension_detect_server),
|
|
(gst_rtsp_extension_before_send), (gst_rtsp_extension_after_send),
|
|
(gst_rtsp_extension_parse_sdp), (gst_rtsp_extension_setup_media),
|
|
(gst_rtsp_extension_configure_stream),
|
|
(gst_rtsp_extension_get_transports),
|
|
(gst_rtsp_extension_stream_select), (gst_rtsp_extension_send):
|
|
* gst-libs/gst/interfaces/rtspextension.h:
|
|
Move interface for RTSP extensions from -good to here.
|
|
Added helper methods to invoke interface methods.
|
|
|
|
2007-07-25 11:22:30 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
Fix some more RTSP docs.
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/rtsp/gstrtspdefs.h:
|
|
* gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
|
|
(gst_rtsp_message_get_type), (gst_rtsp_message_parse_request),
|
|
(gst_rtsp_message_init_response),
|
|
(gst_rtsp_message_parse_response), (gst_rtsp_message_new_data),
|
|
(gst_rtsp_message_parse_data), (gst_rtsp_message_add_header),
|
|
(gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
|
|
(gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
|
|
(gst_rtsp_message_get_body), (dump_key_value):
|
|
* gst-libs/gst/rtsp/gstrtspmessage.h:
|
|
* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
|
|
(parse_npt_range), (parse_clock_range), (parse_smpte_range),
|
|
(gst_rtsp_range_parse):
|
|
* gst-libs/gst/rtsp/gstrtsprange.h:
|
|
* gst-libs/gst/rtsp/gstrtsptransport.c:
|
|
* gst-libs/gst/rtsp/gstrtspurl.c:
|
|
Fix some more RTSP docs.
|
|
Add some missing methods for dealing with messages.
|
|
|
|
2007-07-24 19:19:33 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
Added beginnings of RTSP documentation.
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-docs.sgml:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
|
|
* gst-libs/gst/rtsp/gstrtspbase64.h:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c:
|
|
(gst_rtsp_connection_connect), (add_auth_header),
|
|
(gst_rtsp_connection_write), (gst_rtsp_connection_send),
|
|
(read_body), (gst_rtsp_connection_receive),
|
|
(gst_rtsp_connection_next_timeout),
|
|
(gst_rtsp_connection_reset_timeout),
|
|
(gst_rtsp_connection_set_auth):
|
|
* gst-libs/gst/rtsp/gstrtspconnection.h:
|
|
* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status):
|
|
* gst-libs/gst/rtsp/gstrtspdefs.h:
|
|
* gst-libs/gst/rtsp/gstrtspmessage.h:
|
|
* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
|
|
(parse_npt_range), (parse_clock_range), (parse_smpte_range),
|
|
(gst_rtsp_range_parse):
|
|
* gst-libs/gst/rtsp/gstrtspurl.h:
|
|
Added beginnings of RTSP documentation.
|
|
|
|
2007-07-24 17:37:03 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
Document the SDP library.
|
|
Original commit message from CVS:
|
|
* docs/libs/Makefile.am:
|
|
* docs/libs/gst-plugins-base-libs-docs.sgml:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/sdp/gstsdp.h:
|
|
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_message_set_origin),
|
|
(gst_sdp_message_set_connection), (gst_sdp_message_add_bandwidth),
|
|
(gst_sdp_message_add_time), (gst_sdp_message_add_zone),
|
|
(gst_sdp_message_set_key), (gst_sdp_message_get_attribute_val_n),
|
|
(gst_sdp_message_get_attribute_val),
|
|
(gst_sdp_message_add_attribute), (gst_sdp_media_new),
|
|
(gst_sdp_media_init), (gst_sdp_media_uninit), (gst_sdp_media_free),
|
|
(gst_sdp_media_get_media), (gst_sdp_media_set_media),
|
|
(gst_sdp_media_get_port), (gst_sdp_media_get_num_ports),
|
|
(gst_sdp_media_set_port_info), (gst_sdp_media_get_proto),
|
|
(gst_sdp_media_set_proto), (gst_sdp_media_formats_len),
|
|
(gst_sdp_media_get_format), (gst_sdp_media_add_format),
|
|
(gst_sdp_media_get_information), (gst_sdp_media_set_information),
|
|
(gst_sdp_media_connections_len), (gst_sdp_media_get_connection),
|
|
(gst_sdp_media_add_connection), (gst_sdp_media_bandwidths_len),
|
|
(gst_sdp_media_get_badwidth), (gst_sdp_media_add_bandwidth),
|
|
(gst_sdp_media_set_key), (gst_sdp_media_get_key),
|
|
(gst_sdp_media_attributes_len), (gst_sdp_media_add_attribute),
|
|
(gst_sdp_media_get_attribute_val_n),
|
|
(gst_sdp_media_get_attribute_val), (gst_sdp_message_parse_buffer),
|
|
(print_media), (gst_sdp_message_dump):
|
|
* gst-libs/gst/sdp/gstsdpmessage.h:
|
|
Document the SDP library.
|
|
Add some of the missing SDPMedia methods.
|
|
|
|
2007-07-24 11:52:56 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
Move SDP and RTSP from helper objects in -good to a reusable library.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* gst-libs/gst/Makefile.am:
|
|
* gst-libs/gst/rtsp/Makefile.am:
|
|
* gst-libs/gst/rtsp/gstrtspbase64.c: (gst_rtsp_base64_encode):
|
|
* gst-libs/gst/rtsp/gstrtspbase64.h:
|
|
* gst-libs/gst/rtsp/gstrtspconnection.c: (inet_aton),
|
|
(gst_rtsp_connection_create), (gst_rtsp_connection_connect),
|
|
(add_auth_header), (add_date_header), (gst_rtsp_connection_write),
|
|
(gst_rtsp_connection_send), (read_line), (read_string), (read_key),
|
|
(parse_response_status), (parse_request_line), (parse_line),
|
|
(gst_rtsp_connection_read), (read_body),
|
|
(gst_rtsp_connection_receive), (gst_rtsp_connection_close),
|
|
(gst_rtsp_connection_free), (gst_rtsp_connection_next_timeout),
|
|
(gst_rtsp_connection_reset_timeout), (gst_rtsp_connection_flush),
|
|
(gst_rtsp_connection_set_auth):
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* gst-libs/gst/rtsp/gstrtspconnection.h:
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* gst-libs/gst/rtsp/gstrtspdefs.c: (rtsp_init_status),
|
|
(gst_rtsp_strresult), (gst_rtsp_method_as_text),
|
|
(gst_rtsp_version_as_text), (gst_rtsp_header_as_text),
|
|
(gst_rtsp_status_as_text), (gst_rtsp_find_header_field),
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|
(gst_rtsp_find_method):
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* gst-libs/gst/rtsp/gstrtspdefs.h:
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|
* gst-libs/gst/rtsp/gstrtspmessage.c: (key_value_foreach),
|
|
(gst_rtsp_message_new), (gst_rtsp_message_init),
|
|
(gst_rtsp_message_new_request), (gst_rtsp_message_init_request),
|
|
(gst_rtsp_message_new_response), (gst_rtsp_message_init_response),
|
|
(gst_rtsp_message_init_data), (gst_rtsp_message_unset),
|
|
(gst_rtsp_message_free), (gst_rtsp_message_add_header),
|
|
(gst_rtsp_message_remove_header), (gst_rtsp_message_get_header),
|
|
(gst_rtsp_message_append_headers), (gst_rtsp_message_set_body),
|
|
(gst_rtsp_message_take_body), (gst_rtsp_message_get_body),
|
|
(gst_rtsp_message_steal_body), (dump_mem), (dump_key_value),
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|
(gst_rtsp_message_dump):
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|
* gst-libs/gst/rtsp/gstrtspmessage.h:
|
|
* gst-libs/gst/rtsp/gstrtsprange.c: (parse_npt_time),
|
|
(parse_npt_range), (parse_clock_range), (parse_smpte_range),
|
|
(gst_rtsp_range_parse), (gst_rtsp_range_free):
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* gst-libs/gst/rtsp/gstrtsprange.h:
|
|
* gst-libs/gst/rtsp/gstrtsptransport.c: (gst_rtsp_transport_new),
|
|
(gst_rtsp_transport_init), (gst_rtsp_transport_get_mime),
|
|
(gst_rtsp_transport_get_manager), (parse_mode), (parse_range),
|
|
(range_as_text), (rtsp_transport_mode_as_text),
|
|
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
|
|
(gst_rtsp_transport_parse), (gst_rtsp_transport_as_text),
|
|
(gst_rtsp_transport_free):
|
|
* gst-libs/gst/rtsp/gstrtsptransport.h:
|
|
* gst-libs/gst/rtsp/gstrtspurl.c: (gst_rtsp_url_parse),
|
|
(gst_rtsp_url_free), (gst_rtsp_url_set_port),
|
|
(gst_rtsp_url_get_port), (gst_rtsp_url_get_request_uri):
|
|
* gst-libs/gst/rtsp/gstrtspurl.h:
|
|
* gst-libs/gst/sdp/Makefile.am:
|
|
* gst-libs/gst/sdp/gstsdp.h:
|
|
* gst-libs/gst/sdp/gstsdpmessage.c: (gst_sdp_origin_init),
|
|
(gst_sdp_connection_init), (gst_sdp_bandwidth_init),
|
|
(gst_sdp_time_init), (gst_sdp_zone_init), (gst_sdp_key_init),
|
|
(gst_sdp_attribute_init), (gst_sdp_message_new),
|
|
(gst_sdp_message_init), (gst_sdp_message_uninit),
|
|
(gst_sdp_message_free), (gst_sdp_media_new), (gst_sdp_media_init),
|
|
(gst_sdp_media_uninit), (gst_sdp_media_free),
|
|
(gst_sdp_message_set_origin), (gst_sdp_message_get_origin),
|
|
(gst_sdp_message_set_connection), (gst_sdp_message_get_connection),
|
|
(gst_sdp_message_add_bandwidth), (gst_sdp_message_add_time),
|
|
(gst_sdp_message_add_zone), (gst_sdp_message_set_key),
|
|
(gst_sdp_message_get_key), (gst_sdp_message_get_attribute_val_n),
|
|
(gst_sdp_message_get_attribute_val),
|
|
(gst_sdp_message_add_attribute), (gst_sdp_message_add_media),
|
|
(gst_sdp_media_add_attribute), (gst_sdp_media_add_bandwidth),
|
|
(gst_sdp_media_add_format), (gst_sdp_media_get_attribute),
|
|
(gst_sdp_media_get_attribute_val_n),
|
|
(gst_sdp_media_get_attribute_val), (gst_sdp_media_get_format),
|
|
(read_string), (read_string_del), (gst_sdp_parse_line),
|
|
(gst_sdp_message_parse_buffer), (print_media),
|
|
(gst_sdp_message_dump):
|
|
* gst-libs/gst/sdp/gstsdpmessage.h:
|
|
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
|
|
Move SDP and RTSP from helper objects in -good to a reusable library.
|
|
Use a proper gst_ namespace.
|
|
|
|
2007-07-23 18:42:22 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
ext/vorbis/vorbisdec.c: Use the new buffer clipping function from gstaudio here.
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
|
|
(vorbis_dec_flush_decode):
|
|
Use the new buffer clipping function from gstaudio here.
|
|
|
|
2007-07-23 18:26:09 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
API: Add buffer clipping function for raw audio buffers. Fixes #456656.
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/audio/audio.c: (gst_audio_buffer_clip):
|
|
* gst-libs/gst/audio/audio.h:
|
|
* tests/check/libs/audio.c: (GST_START_TEST), (audio_suite):
|
|
API: Add buffer clipping function for raw audio buffers. Fixes #456656.
|
|
Also add deprecation guards for gst_audio_structure_set_int() to the
|
|
header.
|
|
|
|
2007-07-23 14:45:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
docs/libs/gst-plugins-base-libs-sections.txt: Cleanup the docs.
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
Cleanup the docs.
|
|
|
|
2007-07-23 11:18:35 +0000 Dan Williams <dcbw@redhat.com>
|
|
|
|
gst/playback/gstplaybasebin.c: Don't return NULL when querying the stream info value array but instead return an empt...
|
|
Original commit message from CVS:
|
|
Patch by: Dan Williams <dcbw at redhat dot com>
|
|
* gst/playback/gstplaybasebin.c:
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|
(gst_play_base_bin_get_streaminfo_value_array):
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|
Don't return NULL when querying the stream info value array but instead
|
|
return an empty array. Fixes #459204.
|
|
|
|
2007-07-23 10:41:18 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/playback/gsturidecodebin.c: Init debug category before using it.
|
|
Original commit message from CVS:
|
|
* gst/playback/gsturidecodebin.c:
|
|
Init debug category before using it.
|
|
|
|
2007-07-21 09:56:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/interfaces/mixer.h: Add padding vars in place of the signal pointers when building with DISABLE_DEPRECAT...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/mixer.h:
|
|
Add padding vars in place of the signal pointers
|
|
when building with DISABLE_DEPRECATED so that the
|
|
interface structure doesn't change size.
|
|
|
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2007-07-21 09:21:12 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
|
|
|
|
Fixes: #152864
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* ext/alsa/gstalsamixer.c:
|
|
* ext/alsa/gstalsamixer.h:
|
|
* ext/alsa/gstalsamixerelement.c:
|
|
* ext/alsa/gstalsamixertrack.c:
|
|
* gst-libs/gst/interfaces/mixer.c:
|
|
* gst-libs/gst/interfaces/mixer.h:
|
|
* gst-libs/gst/interfaces/mixeroptions.c:
|
|
* gst-libs/gst/interfaces/mixeroptions.h:
|
|
* gst-libs/gst/interfaces/mixertrack.c:
|
|
* gst-libs/gst/interfaces/mixertrack.h:
|
|
* tests/check/Makefile.am:
|
|
* tests/check/libs/mixer.c:
|
|
Patch By: Marc-Andre Lureau <marcandre.lureau@gmail.com>
|
|
Fixes: #152864
|
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Add support for notifying mixer changes on the message bus, and
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implement it in alsamixer.
|
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API: gst_mixer_get_mixer_flags
|
|
API: gst_mixer_message_parse_mute_toggled
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|
API: gst_mixer_message_parse_record_toggled
|
|
API: gst_mixer_message_parse_volume_changed
|
|
API: gst_mixer_message_parse_option_changed
|
|
API: GstMixerMessageType
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|
API: GstMixerFlags
|
|
|
|
2007-07-20 16:09:03 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
sys/xvimage/xvimagesink.c: xcontext->im_format is only for testing XShm support (as the header file comments document...
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
|
|
(gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps):
|
|
xcontext->im_format is only for testing XShm support (as the header
|
|
file comments document). Use xvimage->im_format for everything else.
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|
Avoids spurious warnings on buffer allocation before setcaps.
|
|
|
|
2007-07-20 07:22:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
tests/: We should use $(LIBM).
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|
Original commit message from CVS:
|
|
* tests/examples/volume/Makefile.am:
|
|
* tests/icles/Makefile.am:
|
|
We should use $(LIBM).
|
|
|
|
2007-07-20 06:13:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
tests/icles/Makefile.am: This needs -lm.
|
|
Original commit message from CVS:
|
|
* tests/icles/Makefile.am:
|
|
This needs -lm.
|
|
|
|
2007-07-18 07:35:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
Add stdlib include (free, atoi, exit).
|
|
Original commit message from CVS:
|
|
* examples/app/appsrc_ex.c:
|
|
* examples/switch/switcher.c:
|
|
* ext/neon/gstneonhttpsrc.c:
|
|
* ext/timidity/gstwildmidi.c:
|
|
* ext/x264/gstx264enc.c:
|
|
* gst/mve/mveaudioenc.c: (mve_compress_audio):
|
|
* gst/rtpmanager/gstrtpclient.c:
|
|
* gst/rtpmanager/gstrtpjitterbuffer.c:
|
|
* gst/spectrum/demo-audiotest.c:
|
|
* gst/spectrum/demo-osssrc.c:
|
|
* sys/dvb/gstdvbsrc.c:
|
|
Add stdlib include (free, atoi, exit).
|
|
|
|
2007-07-16 10:10:28 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstbasertppayload.c: Don't break ABI, restore previous ranges. Keep the default random selection of ...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertppayload.c:
|
|
(gst_basertppayload_class_init), (gst_basertppayload_init),
|
|
(gst_basertppayload_set_property),
|
|
(gst_basertppayload_get_property):
|
|
Don't break ABI, restore previous ranges. Keep the default random
|
|
selection of timestamp and seqnum offset but as soon as the app sets a
|
|
specific value, use that one.
|
|
|
|
2007-07-14 18:33:15 +0000 Bastien Nocera <hadess@hadess.net>
|
|
|
|
sys/xvimage/xvimagesink.*: Add option to turn off double-buffering for debugging purposes.
|
|
Original commit message from CVS:
|
|
Patch by: Bastien Nocera <hadess at hadess dot net>
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
|
|
(gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
|
|
(gst_xvimagesink_init), (gst_xvimagesink_class_init):
|
|
* sys/xvimage/xvimagesink.h:
|
|
Add option to turn off double-buffering for debugging purposes.
|
|
Fixes #437169.
|
|
|
|
2007-07-14 18:20:41 +0000 Jorn Baayen <jorn@openedhand.com>
|
|
|
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sys/: add 'handle-expose' property. Useful for video widgets which may want to be in control of Expose behaviour. Fix...
|
|
Original commit message from CVS:
|
|
Patch by: Jorn Baayen <jorn at openedhand dot com>
|
|
* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents),
|
|
(gst_ximagesink_set_property), (gst_ximagesink_get_property),
|
|
(gst_ximagesink_init), (gst_ximagesink_class_init):
|
|
* sys/ximage/ximagesink.h:
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
|
|
(gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
|
|
(gst_xvimagesink_init), (gst_xvimagesink_class_init):
|
|
* sys/xvimage/xvimagesink.h:
|
|
add 'handle-expose' property. Useful for video widgets which may want to
|
|
be in control of Expose behaviour. Fixes #380625
|
|
|
|
2007-07-14 17:23:42 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstbasertppayload.*: Fix ranges of rtp payloader properties so that the full range can be used in ad...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertppayload.c:
|
|
(gst_basertppayload_class_init), (gst_basertppayload_init),
|
|
(gst_basertppayload_event), (gst_basertppayload_push),
|
|
(gst_basertppayload_set_property),
|
|
(gst_basertppayload_get_property),
|
|
(gst_basertppayload_change_state):
|
|
* gst-libs/gst/rtp/gstbasertppayload.h:
|
|
Fix ranges of rtp payloader properties so that the full range can be
|
|
used in addition to -1 (random).
|
|
Fix wrong seqnum reporting in caps.
|
|
Fixes #420326.
|
|
|
|
2007-07-13 18:12:19 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/videorate/gstvideorate.c: Use boilerplate.
|
|
Original commit message from CVS:
|
|
* gst/videorate/gstvideorate.c: (gst_video_rate_init),
|
|
(gst_video_rate_query):
|
|
Use boilerplate.
|
|
Add latency query, might not be perfect yet but already works a lot
|
|
better. Fixes #442557.
|
|
|
|
2007-07-13 16:05:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
sys/xvimage/xvimagesink.*: After a caps change, redraw our borders to avoid garbage left there when the image format ...
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
|
|
(gst_xvimagesink_setcaps):
|
|
* sys/xvimage/xvimagesink.h:
|
|
After a caps change, redraw our borders to avoid garbage left there
|
|
when the image format changes to a smaller size, like 16:9 -> 4:3
|
|
Also, hold the flow_lock a bit longer in the set_caps while we're
|
|
fiddling with the xcontext.
|
|
|
|
2007-07-13 16:02:23 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
Remove bogus check for libcheck, since we check for gstreamer-check and it pulls in the required info from there, and...
|
|
Original commit message from CVS:
|
|
* Makefile.am:
|
|
* configure.ac:
|
|
* tests/Makefile.am:
|
|
Remove bogus check for libcheck, since we check for
|
|
gstreamer-check and it pulls in the required info from there, and we
|
|
weren't actually _using_ the information for libcheck ourselves
|
|
anyway.
|
|
|
|
2007-07-13 15:52:02 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix the r_mask test for RGBA32 on little-endian.
|
|
Original commit message from CVS:
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
|
|
(gst_ffmpeg_caps_to_pixfmt):
|
|
Fix the r_mask test for RGBA32 on little-endian.
|
|
Fix a stupid typo that would have obviously broken
|
|
compilation on big-endian, if anyone was testing.
|
|
|
|
2007-07-12 15:02:43 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/videotestsrc/videotestsrc.*: Add alpha to the color struct.
|
|
Original commit message from CVS:
|
|
* gst/videotestsrc/videotestsrc.c: (paint_hline_AYUV),
|
|
(paint_hline_str4):
|
|
* gst/videotestsrc/videotestsrc.h:
|
|
Add alpha to the color struct.
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|
Use a default alpha value of 255 instead of 128.
|
|
|
|
2007-07-12 12:01:20 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybasebin.c: Clear the dynamic pads counter when starting a new uri. This makes reusing playbin wor...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (no_more_pads_full),
|
|
(setup_source):
|
|
Clear the dynamic pads counter when starting a new uri. This makes
|
|
reusing playbin work again.
|
|
Fixes #454264.
|
|
|
|
2007-07-12 11:13:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
configure.ac: Use pkg-config to locate check.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Use pkg-config to locate check.
|
|
|
|
2007-07-11 23:12:12 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
Fix 'make check' build against core CVS.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* tests/check/elements/volume.c: (GST_START_TEST):
|
|
Fix 'make check' build against core CVS.
|
|
|
|
2007-07-10 20:46:41 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst-libs/gst/: Make gtk-doc happy.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/propertyprobe.c:
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
* gst-libs/gst/tag/gstvorbistag.c:
|
|
Make gtk-doc happy.
|
|
|
|
2007-07-08 13:07:38 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.c: Quick hack to make audiosinks stop at EOS when operating in pull-mode; needs t...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_callback):
|
|
Quick hack to make audiosinks stop at EOS when operating in
|
|
pull-mode; needs to be fixed properly some day.
|
|
|
|
2007-07-06 18:19:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
docs/libs/gst-plugins-base-libs-sections.txt: Fix location of includes in the docs.
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
Fix location of includes in the docs.
|
|
|
|
2007-07-06 11:40:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst/ffmpegcolorspace/: Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections of the existing BGRA32 and ...
|
|
Original commit message from CVS:
|
|
* gst/ffmpegcolorspace/avcodec.h:
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
|
|
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
|
|
(gst_ffmpegcsp_avpicture_fill):
|
|
* gst/ffmpegcolorspace/imgconvert.c: (img_convert),
|
|
(img_get_alpha_info):
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Add 2 new pixel formats - ABGR32 and ARGB32, which are reflections
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of the existing BGRA32 and RGBA32 formats with the alpha at the other
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end of the word. Partially fixes #451908
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2007-07-05 08:43:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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docs/: Simplify --extra-dir as gtkdoc scans recursively.
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Original commit message from CVS:
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* docs/libs/Makefile.am:
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* docs/plugins/Makefile.am:
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Simplify --extra-dir as gtkdoc scans recursively.
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2007-07-03 11:52:47 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/adder/gstadder.c: Make getcaps more robust by not using the proxycaps function. This makes sure that we don't end...
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Original commit message from CVS:
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* gst/adder/gstadder.c: (gst_adder_sink_getcaps),
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(gst_adder_request_new_pad):
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Make getcaps more robust by not using the proxycaps function. This makes
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sure that we don't end up recursively calling getcaps upstream.
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See #316248.
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2007-06-29 17:21:18 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/audioconvert/audioconvert.c: Include math.h to fix compilation.
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Original commit message from CVS:
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* gst/audioconvert/audioconvert.c:
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Include math.h to fix compilation.
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2007-06-29 14:47:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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gst/ffmpegcolorspace/gstffmpegcodecmap.c: Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel format, ...
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|
Original commit message from CVS:
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* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
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(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
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Add a mapping for YUV format "IYU1", which is a 4:1:1 packed pixel
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format, as produced by some dc1394 cameras like the iSight.
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See http://www.fourcc.org/yuv.php#IYU1
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2007-06-28 20:37:58 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/audioconvert/: Implement dithering and noise shaping in audioconvert. By default now
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Original commit message from CVS:
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* gst/audioconvert/Makefile.am:
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* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
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(check_default), (audio_convert_prepare_context),
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(audio_convert_clean_context), (audio_convert_convert):
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* gst/audioconvert/audioconvert.h:
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* gst/audioconvert/gstaudioconvert.c:
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(gst_audio_convert_dithering_get_type),
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(gst_audio_convert_ns_get_type), (gst_audio_convert_class_init),
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(gst_audio_convert_init), (gst_audio_convert_set_caps),
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(gst_audio_convert_set_property), (gst_audio_convert_get_property):
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* gst/audioconvert/gstaudioconvert.h:
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* gst/audioconvert/gstaudioquantize.c:
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(gst_audio_quantize_setup_noise_shaping),
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(gst_audio_quantize_free_noise_shaping),
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(gst_audio_quantize_setup_dither),
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(gst_audio_quantize_free_dither),
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(gst_audio_quantize_setup_quantize_func),
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(gst_audio_quantize_setup), (gst_audio_quantize_free):
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* gst/audioconvert/gstaudioquantize.h:
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Implement dithering and noise shaping in audioconvert. By default now
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TPDF dithering (and no noise shaping) will be used when converting
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from a higher bit depth to 20 bit depth or smaller, otherwise
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everything will be as it is now.
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For the last audioconvert in a pipeline it would make sense to
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use some kind of noise shaping, enabling it by default for all
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conversions would give undesired results though. Fixes #360246.
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* tests/check/elements/audioconvert.c: (setup_audioconvert),
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(GST_START_TEST):
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Adjust unit test for the new audioconvert.
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2007-06-28 11:06:56 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstqueue2.c: Use other metrics as well when estimating the buffer level.
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Original commit message from CVS:
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* gst/playback/gstqueue2.c: (apply_segment), (update_buffering):
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Use other metrics as well when estimating the buffer level.
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2007-06-28 10:21:19 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstplaybasebin.c: Small debug improvement.
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Original commit message from CVS:
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* gst/playback/gstplaybasebin.c: (make_decoder), (setup_source):
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Small debug improvement.
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* gst/playback/gstqueue2.c: (apply_segment), (update_buffering),
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(plugin_init):
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Tweak the rate estimation period.
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When calculating the buffer filledness in rate estimation mode, don't
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mix it with other metrics.
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2007-06-28 09:46:11 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstdecodebin2.c: When creating the groups, allow for a 5 second, unlimited buffers preroll phase after w...
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|
Original commit message from CVS:
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* gst/playback/gstdecodebin2.c: (gst_decode_group_new),
|
|
(gst_decode_group_expose), (gst_decode_group_free), (add_fakesink):
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When creating the groups, allow for a 5 second, unlimited buffers
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preroll phase after which we expose the group.
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When the group is exposed, use a small number of buffers up to a 2
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second limit. Also disconnect the overrun signal from multiqueue when we
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exposed the group because it is not needed anymore.
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2007-06-27 22:30:19 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/tag/tags.c: Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags to utf8-validate; fixes...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
|
|
Don't pass trailing zeroes in fixed-size string arrays in ID3v1 tags
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to utf8-validate; fixes recognition of ID3v1 tags in UTF-8 encoding
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(#451707); also, output some debugging info when dealing with
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freeform strings.
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* tests/check/libs/tag.c: (GST_START_TEST), (tag_suite):
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Add unit test for the above.
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2007-06-27 12:55:20 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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gst-libs/gst/pbutils/descriptions.c: Add description for Windows Media RTP caps.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/pbutils/descriptions.c: (caps_are_rtp_caps):
|
|
Add description for Windows Media RTP caps.
|
|
* gst-libs/gst/pbutils/missing-plugins.c: (copy_and_clean_caps):
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Remove RTP fields that don't define the format from caps.
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|
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2007-06-27 10:14:03 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/vorbis/vorbisdec.c: Skip empty buffers, but not empty header buffers. That way the original vorbisdec unit test s...
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
|
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Skip empty buffers, but not empty header buffers. That way the original
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|
vorbisdec unit test still passes (#451145); also, take into account
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that those empty packets might carry a granulepos.
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* tests/check/Makefile.am:
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* tests/check/elements/vorbisdec.c:
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(_create_codebook_header_buffer), (_create_audio_buffer),
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(GST_START_TEST), (vorbisdec_suite):
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Add unit test that sends an empty packet.
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2007-06-27 09:49:51 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/vorbis/vorbisdec.c: Don't error out on 0-sized packets, just emit a warning because this is not a fatal error. Fi...
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisdec.c: (vorbis_dec_decode_buffer):
|
|
Don't error out on 0-sized packets, just emit a warning because this is
|
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not a fatal error. Fixes #451145.
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2007-06-25 12:43:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
|
|
docs/plugins/: Update docs with caps info.
|
|
Original commit message from CVS:
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/gst-plugins-base-plugins.signals:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.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-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-decodebin2.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gdp.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.xml:
|
|
* docs/plugins/inspect/plugin-libvisual.xml:
|
|
* docs/plugins/inspect/plugin-ogg.xml:
|
|
* docs/plugins/inspect/plugin-pango.xml:
|
|
* docs/plugins/inspect/plugin-playbin.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-video4linux.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:
|
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* docs/plugins/inspect/plugin-ximagesink.xml:
|
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* docs/plugins/inspect/plugin-xvimagesink.xml:
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Update docs with caps info.
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|
|
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2007-06-25 12:04:15 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
|
po/POTFILES.in: Add more files with translatable strings (#450875).
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|
Original commit message from CVS:
|
|
* po/POTFILES.in:
|
|
Add more files with translatable strings (#450875).
|
|
|
|
2007-06-23 14:44:07 +0000 Edward Hervey <bilboed@bilboed.com>
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|
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ext/ogg/gstoggdemux.c: The chain should be freed if we error out here, else it will leak.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_find_chains):
|
|
The chain should be freed if we error out here, else it will leak.
|
|
* gst/playback/gstdecodebin.c: (disconnect_unlinked_signals),
|
|
(cleanup_decodebin):
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|
Don't forget to *properly* remove the signals, else it will leak.
|
|
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2007-06-22 14:25:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
|
|
MAINTAINERS: Updating all the maintainers files
|
|
Original commit message from CVS:
|
|
* MAINTAINERS:
|
|
Updating all the maintainers files
|
|
|
|
2007-06-21 08:34:46 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
|
|
tests/examples/seek/seek.c: Destroy and recreate parse-launch based pipeline after stop to be able to play again. Reo...
|
|
Original commit message from CVS:
|
|
* tests/examples/seek/seek.c: (update_scale), (play_cb), (stop_cb),
|
|
(main):
|
|
Destroy and recreate parse-launch based pipeline after stop to be able
|
|
to play again. Reorder some code and add more comments.
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|
|
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2007-06-20 11:09:03 +0000 Wim Taymans <wim@fluendo.com>
|
|
|
|
gst/playback/gstdecodebin2.c: When handling a delayed-caps notification case, mark the group as dynamic so that the n...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
|
|
When handling a delayed-caps notification case, mark
|
|
the group as dynamic so that the nbdynamic count is
|
|
incremented and decremented correctly. Fixes: #449156
|
|
Patch by: Wim Taymans <wim@fluendo.com>
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|
|
|
2007-06-19 19:13:04 +0000 Andy Wingo <wingo@pobox.com>
|
|
|
|
* ChangeLog:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
* win32/common/config.h:
|
|
gst-libs/gst/audio/gstbaseaudiosink.c
|
|
Original commit message from CVS:
|
|
2007-06-19 Andy Wingo <wingo@pobox.com>
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c
|
|
(gst_base_audio_sink_init): Enable pull-mode operation.
|
|
|
|
2007-06-19 09:34:35 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst-libs/gst/riff/riff-media.c: Change minimum rate back to 1000 to allow low-sample-rate wav files to play back.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
|
|
Change minimum rate back to 1000 to allow low-sample-rate wav files
|
|
to play back.
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|
|
|
2007-06-17 17:27:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
po/vi.po: Update translations.
|
|
Original commit message from CVS:
|
|
* po/vi.po:
|
|
Update translations.
|
|
|
|
2007-06-16 03:42:14 +0000 David Schleef <ds@schleef.org>
|
|
|
|
gst/playback/gstqueue2.c: Fix compile error from ignored return value.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstqueue2.c:
|
|
Fix compile error from ignored return value.
|
|
|
|
2007-06-15 15:23:36 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/videoscale/vs_4tap.c: Update tmpbuf for all neccesary rows, not just one, as is required when downscaling.
|
|
Original commit message from CVS:
|
|
* gst/videoscale/vs_4tap.c: (vs_image_scale_4tap_Y):
|
|
Update tmpbuf for all neccesary rows, not just one, as is required
|
|
when downscaling.
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Fixes #402076.
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|
|
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2007-06-15 11:15:28 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
tests/check/pipelines/oggmux.c: Add a test that ensures we set DELTA_UNIT on all non-header, non-video buffers, if we...
|
|
Original commit message from CVS:
|
|
* tests/check/pipelines/oggmux.c: (validate_ogg_page), (is_video),
|
|
(eos_buffer_probe):
|
|
Add a test that ensures we set DELTA_UNIT on all non-header,
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|
non-video buffers, if we have a video stream.
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|
* ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
|
|
(gst_ogg_mux_process_best_pad):
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|
Move setting delta_pad to earlier, where we inspect all pads, so
|
|
that leading audio pages don't get DELTA_UNIT unset if they come
|
|
before the first DELTA_UNIT from video pages. Fixes the newly-added
|
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test. Fixes #385527.
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2007-06-14 19:53:27 +0000 Tim-Philipp Müller <tim@centricular.net>
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tests/check/pipelines/streamheader.c: Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it fails on the p5-ppc6...
|
|
Original commit message from CVS:
|
|
* tests/check/pipelines/streamheader.c: (streamheader_suite):
|
|
Disable test_multifdsink_gdp_vorbisenc() on ppc64 since it
|
|
fails on the p5-ppc64 build bot and the failure looks like it is due
|
|
to the same issue as #348114, ie. a compiler bug.
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|
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2007-06-13 18:20:57 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
gst/playback/gstqueue2.c: Fix build on MacOSX.
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|
Original commit message from CVS:
|
|
* gst/playback/gstqueue2.c: (gst_queue_create_read):
|
|
Fix build on MacOSX.
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|
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|
2007-06-13 09:01:32 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
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|
ext/ogg/gstoggdemux.c: Fix compilation on mingw. Fixes #446972.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
|
|
(gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain):
|
|
Fix compilation on mingw. Fixes #446972.
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|
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2007-06-12 08:38:06 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst/playback/gstqueue2.c: Fix a division by zero when the max percent is <= 0. Fixes #446572. also update the bufferi...
|
|
Original commit message from CVS:
|
|
Patches by: Thiago Sousa Santos <thiagossantos at gmail dot com>
|
|
* gst/playback/gstqueue2.c: (update_buffering),
|
|
(gst_queue_locked_enqueue):
|
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Fix a division by zero when the max percent is <= 0. Fixes #446572.
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also update the buffering status when receiving events. Fixes #446551.
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|
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2007-06-11 11:32:26 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
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|
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gst/playback/gstqueue2.c: Wait for preroll before attempting to forward a duration query upstream.
|
|
Original commit message from CVS:
|
|
Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
|
|
* gst/playback/gstqueue2.c: (gst_queue_peer_query),
|
|
(gst_queue_handle_src_query):
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|
Wait for preroll before attempting to forward a duration query upstream.
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Fixes #445505.
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|
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|
2007-06-07 21:08:38 +0000 Sébastien Moutte <sebastien@moutte.net>
|
|
|
|
gst-libs/gst/rtp/gstbasertpdepayload.c: Use G_GINT64_CONSTANT macro for int64 constant.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_set_gst_timestamp):
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|
Use G_GINT64_CONSTANT macro for int64 constant.
|
|
* win32/common/libgstinterfaces.def:
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* win32/common/libgsttag.def:
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Add new exported functions.
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|
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|
2007-06-07 14:25:32 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/ogg/gstoggmux.c: The BOS page of the first Dirac video stream needs to come before the BOS page of any Vorbis str...
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers):
|
|
The BOS page of the first Dirac video stream needs to come before
|
|
the BOS page of any Vorbis streams or other audio streams, just like
|
|
it is with Theora.
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2007-06-07 09:11:27 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst/playback/gstqueue2.c: Fix compilation.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstqueue2.c: (gst_queue_get_range):
|
|
Fix compilation.
|
|
|
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2007-06-06 13:36:26 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
|
|
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gst/playback/gstqueue2.c: Add pull based scheduling and fix some deadlocks. Fixes #444523.
|
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Original commit message from CVS:
|
|
Patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
|
|
* gst/playback/gstqueue2.c: (gst_queue_init),
|
|
(gst_queue_handle_sink_event), (gst_queue_chain),
|
|
(gst_queue_get_range), (gst_queue_src_checkgetrange_function),
|
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(gst_queue_sink_activate_push), (gst_queue_src_activate_push),
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(gst_queue_src_activate_pull):
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Add pull based scheduling and fix some deadlocks. Fixes #444523.
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Does not yet completely work because duration queries upstream won't
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block yet.
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2007-06-06 09:08:50 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
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Some more fseeko checks.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
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* gst/playback/gstqueue2.c: (gst_queue_create_read):
|
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Some more fseeko checks.
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2007-06-06 08:01:42 +0000 Wim Taymans <wim.taymans@gmail.com>
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configure.ac: check for large file support.
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Original commit message from CVS:
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* configure.ac:
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check for large file support.
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2007-06-05 21:36:11 +0000 Sven Arvidsson <sa@whiz.se>
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gst/subparse/gstsubparse.*: Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
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Original commit message from CVS:
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|
Based on a patch by Sven Arvidsson <sa at whiz dot se>:
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* gst/subparse/gstsubparse.c: (parse_subrip),
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(subviewer_unescape_newlines), (parse_subviewer),
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(gst_sub_parse_data_format_autodetect),
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(gst_sub_parse_format_autodetect), (gst_subparse_type_find):
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* gst/subparse/gstsubparse.h:
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Add support for SubViewer version 1 and 2 subtitles. Fixes #394061.
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* tests/check/elements/subparse.c: (GST_START_TEST),
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(subparse_suite):
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Add a unit test for both SubViewer formats.
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2007-06-05 17:08:04 +0000 Michael Smith <msmith@xiph.org>
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gst/audiotestsrc/gstaudiotestsrc.c: Don't overflow intermediate values when seeking to large time values in audiotest...
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Original commit message from CVS:
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* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
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Don't overflow intermediate values when seeking to large time values
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in audiotestsrc.
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2007-06-05 17:02:13 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstqueue2.c: Include stdio to define fseeko.
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Original commit message from CVS:
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* gst/playback/gstqueue2.c: (gst_queue_have_data),
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(gst_queue_create_read), (gst_queue_read_item_from_file),
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(gst_queue_open_temp_location_file), (gst_queue_locked_enqueue):
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Include stdio to define fseeko.
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2007-06-05 16:37:09 +0000 Edward Hervey <edward@fluendo.com>
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sys/v4l/gstv4lsrc.c: Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
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Original commit message from CVS:
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|
Patch by: Edward Hervey <edward@fluendo.com>
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* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_init), (gst_v4lsrc_fixate),
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(gst_v4lsrc_query):
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Make v4lsrc output segments in GST_FORMAT_TIME. Fixes #442553.
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2007-06-05 16:20:44 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/riff/: Use gst_tag_utf8_from_freeform_string() from libgsttag instead of our own implementation.
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Original commit message from CVS:
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* gst-libs/gst/riff/Makefile.am:
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* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_info):
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Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
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our own implementation.
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2007-06-05 16:19:30 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/rtp/gstbasertpdepayload.c: Handle timestamp wraparound.
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Original commit message from CVS:
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* gst-libs/gst/rtp/gstbasertpdepayload.c:
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(gst_base_rtp_depayload_setcaps),
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(gst_base_rtp_depayload_set_gst_timestamp),
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(gst_base_rtp_depayload_change_state):
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Handle timestamp wraparound.
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2007-06-05 16:17:30 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gsturidecodebin.c: Make sure we name srcpads uniquely even when using different internal decodebins.
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Original commit message from CVS:
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* gst/playback/gsturidecodebin.c: (no_more_pads_full),
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(new_decoded_pad), (remove_pads), (make_decoder), (setup_source),
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(gst_uri_decode_bin_change_state):
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Make sure we name srcpads uniquely even when using different internal
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decodebins.
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Signal no-more-pads when no more dynamic elements exist.
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Remove pads on cleanup.
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2007-06-05 16:14:23 +0000 Thiago Sousa Santos <thiagossantos@gmail.com>
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gst/playback/gstqueue2.c: Add support for filebased buffering. Fixes #441264.
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Original commit message from CVS:
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|
Based on patch by: Thiago Sousa Santos <thiagossantos at gmail dot com>
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* gst/playback/gstqueue2.c: (gst_queue_class_init),
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(gst_queue_init), (gst_queue_finalize),
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(gst_queue_write_buffer_to_file), (gst_queue_have_data),
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(gst_queue_create_read), (gst_queue_read_item_from_file),
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(gst_queue_open_temp_location_file),
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(gst_queue_close_temp_location_file), (gst_queue_locked_flush),
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(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
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(gst_queue_is_empty), (gst_queue_is_filled),
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(gst_queue_change_state), (gst_queue_set_temp_location),
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(gst_queue_set_property):
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Add support for filebased buffering. Fixes #441264.
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2007-06-05 16:05:19 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstdecodebin2.c: Add support for delayed caps fixation when autoplugging.
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|
Original commit message from CVS:
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* gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter),
|
|
(analyze_new_pad), (connect_pad), (expose_pad), (caps_notify_cb),
|
|
(caps_notify_group_cb), (gst_decode_group_new),
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(gst_decode_group_free):
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Add support for delayed caps fixation when autoplugging.
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Optimize cases where a multiqueue is not needed/wanted, like right after
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anything that is not a demuxer.
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2007-06-05 16:02:57 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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ext/ogg/gstoggdemux.c: consideratly speedup ogg chain detection by not trying to find a base timestamp for skeleton s...
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fisbone),
|
|
(gst_ogg_pad_submit_packet), (gst_ogg_demux_perform_seek),
|
|
(gst_ogg_demux_read_chain), (gst_ogg_demux_collect_chain_info):
|
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consideratly speedup ogg chain detection by not trying to find a base
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timestamp for skeleton streams.
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|
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2007-06-05 16:00:33 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
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gst/tcp/gstmultifdsink.*: Add support for remuve_flush.
|
|
Original commit message from CVS:
|
|
* gst/tcp/gstmultifdsink.c: (gst_client_status_get_type),
|
|
(gst_multi_fd_sink_class_init), (gst_multi_fd_sink_add_full),
|
|
(gst_multi_fd_sink_remove_flush),
|
|
(gst_multi_fd_sink_remove_client_link),
|
|
(gst_multi_fd_sink_handle_client_write),
|
|
(gst_multi_fd_sink_handle_clients):
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|
* gst/tcp/gstmultifdsink.h:
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|
Add support for remuve_flush.
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|
|
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2007-06-05 15:59:00 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
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Add draft design for forcing keyframes in encoders and implement in theoraenc.
|
|
Original commit message from CVS:
|
|
* docs/design/draft-keyframe-force.txt:
|
|
* ext/theora/theoraenc.c: (theora_enc_sink_event),
|
|
(theora_enc_chain):
|
|
Add draft design for forcing keyframes in encoders and implement in
|
|
theoraenc.
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|
|
|
2007-06-05 13:22:18 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
configure.ac: Back to CVS
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Back to CVS
|
|
|
|
=== release 0.10.13 ===
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|
|
|
2007-06-05 12:50:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
|
|
* ChangeLog:
|
|
* NEWS:
|
|
* RELEASE:
|
|
* configure.ac:
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.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-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-decodebin2.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gdp.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.xml:
|
|
* docs/plugins/inspect/plugin-libvisual.xml:
|
|
* docs/plugins/inspect/plugin-ogg.xml:
|
|
* docs/plugins/inspect/plugin-pango.xml:
|
|
* docs/plugins/inspect/plugin-playbin.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-video4linux.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/config.h:
|
|
* win32/vs6/grammar.dsp:
|
|
* win32/vs6/gst_plugins_base.dsw:
|
|
* win32/vs6/libgstadder.dsp:
|
|
* win32/vs6/libgstaudio.dsp:
|
|
* win32/vs6/libgstaudioconvert.dsp:
|
|
* win32/vs6/libgstaudiorate.dsp:
|
|
* win32/vs6/libgstaudioresample.dsp:
|
|
* win32/vs6/libgstaudioscale.dsp:
|
|
* win32/vs6/libgstaudiotestsrc.dsp:
|
|
* win32/vs6/libgstcdda.dsp:
|
|
* win32/vs6/libgstdecodebin.dsp:
|
|
* win32/vs6/libgstdecodebin2.dsp:
|
|
* win32/vs6/libgstdirectsound.dsp:
|
|
* win32/vs6/libgstffmpegcolorspace.dsp:
|
|
* win32/vs6/libgstgdp.dsp:
|
|
* win32/vs6/libgstinterfaces.dsp:
|
|
* win32/vs6/libgstnetbuffer.dsp:
|
|
* win32/vs6/libgstogg.dsp:
|
|
* win32/vs6/libgstpbutils.dsp:
|
|
* win32/vs6/libgstplaybin.dsp:
|
|
* win32/vs6/libgstriff.dsp:
|
|
* win32/vs6/libgstrtp.dsp:
|
|
* win32/vs6/libgstsinesrc.dsp:
|
|
* win32/vs6/libgstsubparse.dsp:
|
|
* win32/vs6/libgsttag.dsp:
|
|
* win32/vs6/libgsttheora.dsp:
|
|
* win32/vs6/libgsttypefindfunctions.dsp:
|
|
* win32/vs6/libgstutils.dsp:
|
|
* win32/vs6/libgstvideo.dsp:
|
|
* win32/vs6/libgstvideorate.dsp:
|
|
* win32/vs6/libgstvideoscale.dsp:
|
|
* win32/vs6/libgstvideotestsrc.dsp:
|
|
* win32/vs6/libgstvolume.dsp:
|
|
* win32/vs6/libgstvorbis.dsp:
|
|
Release 0.10.13 "What's going on?"
|
|
Original commit message from CVS:
|
|
Release 0.10.13 "What's going on?"
|
|
|
|
2007-06-05 12:32:03 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/cs.po:
|
|
* po/de.po:
|
|
* po/en_GB.po:
|
|
* po/hu.po:
|
|
* po/it.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
Update .po files
|
|
Original commit message from CVS:
|
|
Update .po files
|
|
|
|
2007-05-31 17:08:58 +0000 Wim Taymans <wim@fluendo.com>
|
|
|
|
gst-libs/gst/riff/riff-media.c: In riff, the depth is stored in the size field but it just means that the least signi...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
|
|
In riff, the depth is stored in the size field but it just means that
|
|
the least significant bits are cleared. We can therefore just play
|
|
the sample as if it had a depth == width. Fixes: #440997
|
|
Patch by: Wim Taymans <wim@fluendo.com>
|
|
Patch by: Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
2007-05-31 16:36:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/floatcast/floatcast.h: Define inline when needed on win32 builds. Fixes: #441295
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/floatcast/floatcast.h:
|
|
Define inline when needed on win32 builds. Fixes: #441295
|
|
|
|
2007-05-29 13:38:35 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybasebin.c: Stop buffering when the group is commited because the queues filled up.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (queue_overrun),
|
|
(no_more_pads_full):
|
|
Stop buffering when the group is commited because the queues filled up.
|
|
Fixes #442024.
|
|
|
|
2007-05-25 10:07:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
Revert commits towards #152864 made so far. We'll pick it up again after the 0.10.13 release.
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
|
|
(gst_alsa_mixer_free), (gst_alsa_mixer_update),
|
|
(gst_alsa_mixer_get_volume), (gst_alsa_mixer_set_volume),
|
|
(gst_alsa_mixer_set_mute), (gst_alsa_mixer_set_record),
|
|
(gst_alsa_mixer_set_option), (gst_alsa_mixer_get_option):
|
|
* ext/alsa/gstalsamixer.h:
|
|
* ext/alsa/gstalsamixerelement.c:
|
|
(gst_alsa_mixer_element_interface_supported),
|
|
(gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
|
|
(gst_alsa_mixer_element_set_property),
|
|
(gst_alsa_mixer_element_get_property),
|
|
(gst_alsa_mixer_element_change_state):
|
|
* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_update):
|
|
* gst-libs/gst/interfaces/mixer.c: (gst_mixer_volume_changed),
|
|
(gst_mixer_option_changed):
|
|
* gst-libs/gst/interfaces/mixer.h:
|
|
Revert commits towards #152864 made so far. We'll pick it up again
|
|
after the 0.10.13 release.
|
|
|
|
2007-05-24 16:22:23 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.c: After an interrupt (PAUSED/flush) assume that the next sample should not be al...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_render):
|
|
After an interrupt (PAUSED/flush) assume that the next sample should not
|
|
be aligned to the previous sample. Fixes #417992.
|
|
|
|
2007-05-24 15:16:59 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/riff/riff-media.c: Don't add channels and rate fields to the template caps for audio/x-dts, as wavparse ...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
|
|
Don't add channels and rate fields to the template caps for
|
|
audio/x-dts, as wavparse might not always be able to set them,
|
|
which would then lead to 'caps are not a real subset of the
|
|
template caps' warnings.
|
|
|
|
2007-05-24 11:15:32 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst/playback/gstplaybasebin.c: Handle unknown or invalid pads without crashing, as might occur if a media file like a...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
|
|
Handle unknown or invalid pads without crashing, as might occur if
|
|
a media file like an mp3 is specified as a subtitle file.
|
|
Fixes: #410039
|
|
|
|
2007-05-24 10:19:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst/playback/gstplaybin.c: Block the subtitle bin output queue before ghosting it and linking, then unblock after. Th...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin.c: (add_sink), (dummy_blocked_cb),
|
|
(setup_sinks):
|
|
Block the subtitle bin output queue before ghosting it and linking,
|
|
then unblock after. This avoids spurious not-linked errors caused
|
|
by the queue starting up (because it gets linked when it is ghosted).
|
|
Fixes: #350299
|
|
|
|
2007-05-23 15:54:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
tests/check/elements/playbin.c: Use /dev/zero instead of /dev/urandom to produce an invalid subtitle file. Avoids flu...
|
|
Original commit message from CVS:
|
|
* tests/check/elements/playbin.c: (test_suburi_error_unknowntype):
|
|
Use /dev/zero instead of /dev/urandom to produce an invalid subtitle
|
|
file. Avoids flukes where the input gets typefound to some valid but
|
|
useless type.
|
|
|
|
2007-05-22 15:45:19 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/: Add unit test for gnomevfssink seeking and position reporting for file:// URIs.
|
|
Original commit message from CVS:
|
|
* tests/check/Makefile.am:
|
|
* tests/check/elements/.cvsignore:
|
|
* tests/check/elements/gnomevfssink.c: (setup_gnomevfssink),
|
|
(cleanup_gnomevfssink), (GST_START_TEST), (gnomevfssink_suite):
|
|
Add unit test for gnomevfssink seeking and position reporting for
|
|
file:// URIs.
|
|
|
|
2007-05-22 15:30:26 +0000 Mark Nauwelaerts <manauw@skynet.be>
|
|
|
|
ext/gnomevfs/gstgnomevfssink.*: see #412648.
|
|
Original commit message from CVS:
|
|
Patch by: Mark Nauwelaerts <manauw at skynet be>
|
|
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_init),
|
|
(gst_gnome_vfs_sink_open_file), (gst_gnome_vfs_sink_handle_event),
|
|
(gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render):
|
|
* ext/gnomevfs/gstgnomevfssink.h:
|
|
Fix position reporting, especially after a seek (from upstream),
|
|
see #412648.
|
|
|
|
2007-05-22 15:04:41 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/cdparanoia/gstcdparanoiasrc.c: Repair umlaut.
|
|
Original commit message from CVS:
|
|
* ext/cdparanoia/gstcdparanoiasrc.c:
|
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Repair umlaut.
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2007-05-22 11:40:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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gst-libs/gst/riff/riff-media.c: Specify the full valid range for MP3 samplerates. Fixes a regression caused by extra ...
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Original commit message from CVS:
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* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
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Specify the full valid range for MP3 samplerates. Fixes a regression
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caused by extra header checks since the last release.
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2007-05-21 15:32:42 +0000 Mike Smith <msmith@xiph.org>
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sys/: Fix a locking-order bug I introduced with my changes the other day.
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Original commit message from CVS:
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* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
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Fix a locking-order bug I introduced with my changes the other day.
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Patch by Mike Smith.
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2007-05-21 15:24:21 +0000 Michael Smith <msmith@xiph.org>
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ext/theora/theoradec.c: Don't look inside 0-length packets (which indicate duplicated frames)
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Original commit message from CVS:
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* ext/theora/theoradec.c: (theora_handle_data_packet):
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Don't look inside 0-length packets (which indicate duplicated
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frames)
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2007-05-21 10:25:44 +0000 Wim Taymans <wim.taymans@gmail.com>
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Small cleanups.
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Original commit message from CVS:
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* ext/cdparanoia/gstcdparanoiasrc.c:
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(gst_cd_paranoia_src_read_sector):
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* gst-libs/gst/audio/gstbaseaudiosrc.c:
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(gst_base_audio_src_create):
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Small cleanups.
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* ext/theora/theoradec.c: (theora_dec_sink_event):
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Fix typo.
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* gst-libs/gst/rtp/gstbasertpdepayload.c:
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(gst_base_rtp_depayload_set_gst_timestamp):
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Add some FIXME
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* gst/playback/gstdecodebin.c: (queue_underrun_cb):
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And some debug info when a FIXME path is hit.
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2007-05-21 09:45:28 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/rtp/gstbasertpaudiopayload.c: Some cleanups, remove minptime property as it is now in the parent class.
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Original commit message from CVS:
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* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
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(gst_base_rtp_audio_payload_class_init),
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(gst_base_rtp_audio_payload_init),
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(gst_base_rtp_audio_payload_finalize),
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(gst_base_rtp_audio_payload_handle_frame_based_buffer),
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(gst_base_rtp_audio_payload_handle_sample_based_buffer),
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(gst_base_rtp_payload_audio_handle_event):
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Some cleanups, remove minptime property as it is now in the parent
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class.
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Override parent class event function.
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* gst-libs/gst/rtp/gstbasertppayload.c:
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(gst_basertppayload_class_init), (gst_basertppayload_init),
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(gst_basertppayload_event), (gst_basertppayload_set_property),
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(gst_basertppayload_get_property):
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* gst-libs/gst/rtp/gstbasertppayload.h:
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Add min-ptime property.
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Add handle-event vmethod. Fixes #415001.
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2007-05-18 17:10:03 +0000 Christian Schaller <uraeus@gnome.org>
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* gst-plugins-base.spec.in:
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update spec
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Original commit message from CVS:
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update spec
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2007-05-18 15:23:43 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst-libs/gst/audio/gstbaseaudiosink.c
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Original commit message from CVS:
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* gst-libs/gst/audio/gstbaseaudiosink.c
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(gst_base_audio_sink_change_state):
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Fix typo in comment.
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* gst/playback/gstdecodebin.c (gst_decode_bin_class_init,
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free_dynamics, pad_probe, close_pad_link, try_to_link_1,
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get_our_ghost_pad, remove_element_chain, queue_underrun_cb,
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close_link):
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* gst/playback/gstplaybin.c (gst_play_bin_set_property,
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gen_audio_element, remove_sinks, gst_play_bin_send_event_to_sink):
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Remove trailing whitespaces in comments.
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* gst/volume/Makefile.am:
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Fix tabs.
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2007-05-18 15:10:08 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
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* ChangeLog:
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* gst-libs/gst/interfaces/mixer.h:
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gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed, set_option, get_option, _gst_reserved):
|
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Original commit message from CVS:
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patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
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* gst-libs/gst/interfaces/mixer.h (mixer_type, option_changed,
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set_option, get_option, _gst_reserved):
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Revert reordering functions (keep ABI).
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2007-05-17 17:35:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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sys/: When we create our own window, indicate that we handle the
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Original commit message from CVS:
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* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put),
|
|
(gst_ximagesink_xwindow_new), (gst_ximagesink_handle_xevents),
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(gst_ximagesink_show_frame):
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
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(gst_xvimagesink_xwindow_new), (gst_xvimagesink_handle_xevents),
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(gst_xvimagesink_show_frame):
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When we create our own window, indicate that we handle the
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WM_DELETE client message from the window manager, so that it won't
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kill our window (and our app) along with it. Handle ClientMessage,
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post an error on the bus, and close the window. Further buffers
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arriving will result in a FlowError because the window has been
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destroyed.
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Fixes: #393975
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Clean up the X event handling loop and make them the same for
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both xvimagesink and ximagesink while I'm at it.
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2007-05-17 16:27:32 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstdecodebin2.c: Make decodebin2 autoplug depayloaders too.
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|
Original commit message from CVS:
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* gst/playback/gstdecodebin2.c: (gst_decode_bin_factory_filter):
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Make decodebin2 autoplug depayloaders too.
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* gst/playback/gsturidecodebin.c: (source_new_pad):
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Set the newly created decoder in a usable state when autoplugging a
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dynamic source such as RTSP.
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2007-05-17 16:11:03 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gststreaminfo.c: Ignore video-codec tag for audio streams and ignore audio-codec tags for video streams....
|
|
Original commit message from CVS:
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|
* gst/playback/gststreaminfo.c: (cb_probe):
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Ignore video-codec tag for audio streams and ignore audio-codec tags
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for video streams. Should make codec name collection a bit more
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robust against sloppy demuxers that send tag events containing both
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tags down each pad.
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2007-05-17 15:22:44 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstqueue2.c: Tweak the buffering thresholds a little.
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|
Original commit message from CVS:
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* gst/playback/gstqueue2.c: (update_rates):
|
|
Tweak the buffering thresholds a little.
|
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Update the buffer size with the previously calculate rate instead of
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only when we calculate a new rate so that we get smoother buffering
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updates.
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* gst/playback/Makefile.am:
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* gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_base_init),
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(gst_uri_decode_bin_class_init), (gst_uri_decode_bin_init),
|
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(gst_uri_decode_bin_finalize), (gst_uri_decode_bin_set_property),
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(gst_uri_decode_bin_get_property), (unknown_type),
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(add_element_stream), (no_more_pads_full), (no_more_pads),
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(source_no_more_pads), (new_decoded_pad), (array_has_value),
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(gen_source_element), (has_all_raw_caps), (analyse_source),
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(remove_decoders), (make_decoder), (remove_source),
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(source_new_pad), (setup_source), (decoder_query_init),
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(decoder_query_duration_fold), (decoder_query_duration_done),
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(decoder_query_position_fold), (decoder_query_position_done),
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(decoder_query_latency_fold), (decoder_query_latency_done),
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(decoder_query_seeking_fold), (decoder_query_seeking_done),
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(decoder_query_generic_fold), (gst_uri_decode_bin_query),
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(gst_uri_decode_bin_change_state), (plugin_init):
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New element that intergrates a source, optional buffering element and
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decodebin.
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2007-05-17 14:17:17 +0000 Tim-Philipp Müller <tim@centricular.net>
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configure.ac: Bump libtheora requirement to 1.0alpha5 for the pixformat check (also has a .pc file, so we don't need ...
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Bump libtheora requirement to 1.0alpha5 for the pixformat check
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(also has a .pc file, so we don't need the fallback check any
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longer). Fixes #438840.
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2007-05-17 13:36:11 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstqueue2.c: fix build.
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|
Original commit message from CVS:
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|
* gst/playback/gstqueue2.c: (gst_queue_get_type),
|
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(gst_queue_class_init), (gst_queue_finalize), (update_time_level),
|
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(apply_segment), (apply_buffer), (update_buffering),
|
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(reset_rate_timer), (update_rates), (gst_queue_locked_flush),
|
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(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
|
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(gst_queue_handle_sink_event), (gst_queue_is_filled),
|
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(gst_queue_chain), (gst_queue_push_one), (gst_queue_loop),
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(plugin_init):
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fix build.
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2007-05-17 11:57:44 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst/playback/: On our way to playbin2 this is the new network queue that does buffering all by itself using high and ...
|
|
Original commit message from CVS:
|
|
* gst/playback/Makefile.am:
|
|
* gst/playback/gstqueue2.c: (gst_queue_get_type),
|
|
(gst_queue_class_init), (gst_queue_init), (gst_queue_finalize),
|
|
(gst_queue_getcaps), (gst_queue_bufferalloc),
|
|
(gst_queue_acceptcaps), (update_time_level), (apply_segment),
|
|
(apply_buffer), (update_buffering), (reset_rate_timer),
|
|
(update_rates), (gst_queue_locked_flush),
|
|
(gst_queue_locked_enqueue), (gst_queue_locked_dequeue),
|
|
(gst_queue_handle_sink_event), (gst_queue_is_empty),
|
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(gst_queue_is_filled), (gst_queue_chain), (gst_queue_push_one),
|
|
(gst_queue_loop), (gst_queue_handle_src_event),
|
|
(gst_queue_handle_src_query), (gst_queue_sink_activate_push),
|
|
(gst_queue_src_activate_push), (gst_queue_change_state),
|
|
(gst_queue_set_property), (gst_queue_get_property), (plugin_init):
|
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On our way to playbin2 this is the new network queue that does buffering
|
|
all by itself using high and low watermarks. It can also measure up and
|
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downstream bandwidth to optimally size the queue.
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|
|
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2007-05-17 11:16:14 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/: Use the segment->last_stop value to calculate the next timestamp to generate after a seek; not the segment->sta...
|
|
Original commit message from CVS:
|
|
* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_do_seek):
|
|
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_do_seek):
|
|
Use the segment->last_stop value to calculate the next timestamp to
|
|
generate after a seek; not the segment->start value.
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|
|
|
2007-05-15 20:14:06 +0000 David Schleef <ds@schleef.org>
|
|
|
|
docs/Makefile.am: Install docs even when --disable-gtk-doc is disabled. This matches the behavior of gtk+. Fixes #3...
|
|
Original commit message from CVS:
|
|
* docs/Makefile.am: Install docs even when --disable-gtk-doc
|
|
is disabled. This matches the behavior of gtk+. Fixes #349099.
|
|
|
|
2007-05-15 17:11:09 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/ogg/gstoggdemux.c: Some more chained streaming ogg timestamp fixes.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
|
|
(gst_ogg_demux_perform_seek), (gst_ogg_demux_handle_page):
|
|
Some more chained streaming ogg timestamp fixes.
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|
|
|
2007-05-15 16:46:10 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/ogg/gstoggdemux.c: Add some FIXMEs.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
|
|
(gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
|
|
(gst_ogg_demux_handle_page):
|
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Add some FIXMEs.
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Fix chain start/stop segment handling based on patch by
|
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<ahalda at cs dot mcgill dot ca> see #320984.
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2007-05-15 15:33:54 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
configure.ac: We don't require a C++ compiler. So don't require one.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
We don't require a C++ compiler. So don't require one.
|
|
|
|
2007-05-15 15:29:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
* ChangeLog:
|
|
* ext/alsa/gstalsamixer.c:
|
|
ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds, gst_alsa_mixer_check, gst_alsa_mixer_dispatch, gst_alsa_mixer_...
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsamixer.c (source, n_poll_fds, poll_fds,
|
|
gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
|
|
gst_alsa_mixer_finalize, gst_alsa_mixer_handle_source_callback,
|
|
gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
|
|
gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_update_option,
|
|
gst_alsa_mixer_update_track):
|
|
Apply some of the cleanup Tim suggested in #152864 afterwards.
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|
|
|
2007-05-15 14:01:26 +0000 Marc-Andre Lureau <marcandre.lureau@gmail.com>
|
|
|
|
ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch, _GstAlsaMixerWatch, source, n_poll_fds, poll_fds, gst_alsa_...
|
|
Original commit message from CVS:
|
|
patch by: Marc-Andre Lureau <marcandre.lureau@gmail.com>
|
|
* ext/alsa/gstalsamixer.c (main_context, GstAlsaMixerWatch,
|
|
_GstAlsaMixerWatch, source, n_poll_fds, poll_fds,
|
|
gst_alsa_mixer_watch_funcs, gst_alsa_mixer_prepare,
|
|
gst_alsa_mixer_check, gst_alsa_mixer_dispatch,
|
|
gst_alsa_mixer_finalize, gst_alsa_mixer_find_master_mixer,
|
|
gst_alsa_mixer_handle_source_callback,
|
|
gst_alsa_mixer_handle_callback, gst_alsa_mixer_elem_handle_callback,
|
|
gst_alsa_mixer_ensure_track_list, gst_alsa_mixer_free,
|
|
gst_alsa_mixer_get_volume, gst_alsa_mixer_set_volume,
|
|
gst_alsa_mixer_set_mute, gst_alsa_mixer_set_record,
|
|
gst_alsa_mixer_get_option, gst_alsa_mixer_update_option,
|
|
gst_alsa_mixer_update_track, _gst_alsa_mixer_set_interface):
|
|
* ext/alsa/gstalsamixer.h (handle_source, interface, dir):
|
|
* ext/alsa/gstalsamixerelement.c (gst_alsa_mixer_element_details,
|
|
gst_alsa_mixer_element_interface_supported,
|
|
gst_alsa_mixer_element_finalize, gst_alsa_mixer_element_init,
|
|
gst_alsa_mixer_element_set_property,
|
|
gst_alsa_mixer_element_get_property,
|
|
gst_alsa_mixer_element_change_state):
|
|
* ext/alsa/gstalsamixertrack.c (gst_alsa_mixer_track_update):
|
|
* gst-libs/gst/interfaces/mixer.c (gst_mixer_volume_changed,
|
|
gst_mixer_option_changed):
|
|
* gst-libs/gst/interfaces/mixer.h (set_option, get_option,
|
|
volume_changed, option_changed, _gst_reserved):
|
|
Implement notification for alsamixer. Fixes #152864
|
|
|
|
2007-05-15 03:53:11 +0000 David Schleef <ds@schleef.org>
|
|
|
|
gst/videotestsrc/videotestsrc.*: Add support for video/x-raw-bayer.
|
|
Original commit message from CVS:
|
|
* gst/videotestsrc/videotestsrc.c:
|
|
* gst/videotestsrc/videotestsrc.h:
|
|
Add support for video/x-raw-bayer.
|
|
|
|
2007-05-13 01:06:19 +0000 David Schleef <ds@schleef.org>
|
|
|
|
sys/xvimage/xvimagesink.c: Add some sanity checking for the XVImage size returned by X.
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c:
|
|
Add some sanity checking for the XVImage size returned by X.
|
|
Related to #377400.
|
|
|
|
2007-05-12 16:18:39 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstbasertpdepayload.c: Parse and use additional caps fields as described in updated application/x-rt...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_setcaps),
|
|
(gst_base_rtp_depayload_set_gst_timestamp):
|
|
Parse and use additional caps fields as described in updated
|
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application/x-rtp caps spec.
|
|
|
|
2007-05-12 16:16:22 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/ogg/gstoggdemux.c: If there is a stream in a chain without any data packets, ignore the stream in the total lengt...
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
|
|
(gst_ogg_demux_collect_chain_info):
|
|
If there is a stream in a chain without any data packets, ignore the
|
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stream in the total length calculations. Might be related to #436820.
|
|
|
|
2007-05-11 17:33:43 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Consolidate and re-work our mpeg system stream detection to probe more packets a...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (mpeg_sys_is_valid_pack),
|
|
(mpeg_sys_is_valid_pes), (mpeg_sys_is_valid_sys),
|
|
(mpeg_sys_type_find), (mpeg_ts_type_find), (mpeg4_video_type_find),
|
|
(mpeg_video_type_find), (mpeg_video_stream_type_find),
|
|
(plugin_init):
|
|
Consolidate and re-work our mpeg system stream detection to probe
|
|
more packets and produce a higher confidence result. Fixes a
|
|
regression caused by lowering the typefind probability last year
|
|
- related to bug #397810. Remove the redundant MPEG-1 specific
|
|
typefind function, as the new one detects both MPEG-1 & MPEG-2
|
|
happily.
|
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Also cleanup the MPEG elementary and MPEG-TS detection functions a
|
|
little.
|
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Tested against my media test directory, with some improvements and
|
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no regressions.
|
|
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2007-05-10 15:28:13 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst/playback/gstplaybasebin.c: Connect to the new queue "pushing" signal instead of the broken "running" one.
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Original commit message from CVS:
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* gst/playback/gstplaybasebin.c: (fill_buffer), (check_queue),
|
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(queue_out_of_data):
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Connect to the new queue "pushing" signal instead of the broken
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"running" one.
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2007-05-09 21:17:40 +0000 Sébastien Moutte <sebastien@moutte.net>
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gst-libs/gst/rtp/gstbasertpaudiopayload.c: Move variable declaration before the first instruction.
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Original commit message from CVS:
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* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
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(gst_base_rtp_audio_payload_handle_frame_based_buffer):
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Move variable declaration before the first instruction.
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* gst/videotestsrc/videotestsrc.c:
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Define M_PI if it's not defined yet.
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* win32/common/libgstrtp.def:
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Add new exported functions.
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2007-05-09 11:54:32 +0000 Michael Smith <msmith@xiph.org>
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ext/theora/theoradec.c: gst_pad_push_event() does not return a GstFlowReturn!
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Original commit message from CVS:
|
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* ext/theora/theoradec.c: (theora_handle_type_packet):
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gst_pad_push_event() does not return a GstFlowReturn!
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2007-05-09 11:25:34 +0000 Wim Taymans <wim.taymans@gmail.com>
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tests/examples/seek/: Some small cosmetic changes.
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Original commit message from CVS:
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* tests/examples/seek/scrubby.c: (stop_cb), (main):
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* tests/examples/seek/seek.c: (do_seek):
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Some small cosmetic changes.
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2007-05-08 19:24:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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* ChangeLog:
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* gst/adder/gstadder.c:
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* gst/adder/gstadder.h:
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gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected, gst_adder_change_state): gst/adder/gstadder.h (bps, o...
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Original commit message from CVS:
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* gst/adder/gstadder.c (gst_adder_src_event, gst_adder_collected,
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gst_adder_change_state):
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* gst/adder/gstadder.h (bps, offset, collect_event, segment,
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segment_pending, segment_position, segment_rate):
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Handle playback-rate on adder.
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2007-05-07 11:43:31 +0000 Michael Smith <msmith@xiph.org>
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ext/theora/: Don't push events (newsegment, tags) before initialising the decoder.
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Original commit message from CVS:
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* ext/theora/gsttheoradec.h:
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* ext/theora/theoradec.c: (gst_theora_dec_reset),
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(theora_dec_sink_event), (theora_handle_comment_packet),
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(theora_handle_type_packet), (theora_dec_change_state):
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Don't push events (newsegment, tags) before initialising the
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decoder.
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This is neccesary for seeking to work correctly in gnonlin.
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2007-05-04 13:10:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst/: gst/audiotestsrc/gstaudiotestsrc.c
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Original commit message from CVS:
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* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
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* gst/adder/gstadder.c:
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* gst/audiotestsrc/gstaudiotestsrc.c
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(gst_audio_test_src_create_white_noise):
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* gst/videotestsrc/gstvideotestsrc.c:
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* gst/volume/gstvolume.c (VOLUME_UNITY_INT16,
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VOLUME_UNITY_INT16_BIT_SHIFT, VOLUME_MAX_DOUBLE,
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volume_sink_template, volume_src_template, gst_volume_init,
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volume_process_double, volume_process_int16,
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volume_process_int16_clamp):
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Doc fixes and formatting.
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2007-05-04 12:41:21 +0000 Tim-Philipp Müller <tim@centricular.net>
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tests/check/: Minimal check for volume's GstController usability; also another test for #422295.
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Original commit message from CVS:
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* tests/check/Makefile.am:
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* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
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Minimal check for volume's GstController usability; also another
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test for #422295.
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2007-05-04 09:06:38 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/cdda/gstcddabasesrc.c: Fix it so that it (a) makes sense and (b) doesn't break everything cdda-related i...
|
|
Original commit message from CVS:
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* gst-libs/gst/cdda/gstcddabasesrc.c:
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(gst_cdda_base_src_add_track):
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Fix it so that it (a) makes sense and (b) doesn't break
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everything cdda-related including the unit test.
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2007-05-04 08:46:59 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst-libs/gst/cdda/gstcddabasesrc.c: Fix build when disabling asserts.
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Original commit message from CVS:
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* gst-libs/gst/cdda/gstcddabasesrc.c:
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(gst_cdda_base_src_add_track):
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Fix build when disabling asserts.
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2007-05-03 16:29:10 +0000 Tim-Philipp Müller <tim@centricular.net>
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sys/ximage/ximagesink.c: When XShm is not available, we might get row strides that are not rounded up to multiples of...
|
|
Original commit message from CVS:
|
|
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new):
|
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When XShm is not available, we might get row strides that are not
|
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rounded up to multiples of four; this is bad, because virtually
|
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every RGB-processing element in GStreamer assumes rowstrides are
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rounded up to multiples of four, so let's allocate at least enough
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memory to avoid crashes in this case. The image will still be
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displayed distorted though if this happens, so that still needs
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fixing (maybe by allocating a bigger image with an 'even' width
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and then clipping it appropriately when rendering - something for
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Xlib aficionados in any case).
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2007-05-03 13:16:21 +0000 Michael Smith <msmith@xiph.org>
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|
|
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gst/audiorate/gstaudiorate.c: If a buffer doesn't have a timestamp, assume it's contiguous with the previous buffer, ...
|
|
Original commit message from CVS:
|
|
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
|
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If a buffer doesn't have a timestamp, assume it's contiguous with
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the previous buffer, and synthesise timestamps appropriately.
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2007-05-03 11:24:00 +0000 Edward Hervey <bilboed@bilboed.com>
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tests/check/elements/videorate.c: Set buffer timestamp to a valid value in order to test the buffer really does stay ...
|
|
Original commit message from CVS:
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|
* tests/check/elements/videorate.c: (GST_START_TEST):
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Set buffer timestamp to a valid value in order to test the buffer
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really does stay in videorate.
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2007-05-03 10:47:22 +0000 Edward Hervey <bilboed@bilboed.com>
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gst/videorate/gstvideorate.c: There is no sensible way to handle incoming buffers which don't have a valid timestamp....
|
|
Original commit message from CVS:
|
|
* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
|
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There is no sensible way to handle incoming buffers which don't have a
|
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valid timestamp. We therefore discard them and wait for the next one.
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2007-05-01 18:45:36 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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gst/playback/: Better error message for text files.
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|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (type_found), (plugin_init):
|
|
* gst/playback/gstdecodebin2.c: (plugin_init):
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Better error message for text files.
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|
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2007-04-29 14:38:05 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstrtcpbuffer.c: Fix offset bug in generation RR packets.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_add_rb):
|
|
Fix offset bug in generation RR packets.
|
|
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2007-04-27 15:33:46 +0000 Julien Moutte <julien@moutte.net>
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|
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ext/theora/theoradec.c: Calculate buffer duration correctly to generate a perfect stream (#433888).
|
|
Original commit message from CVS:
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|
2007-04-27 Julien MOUTTE <julien@moutte.net>
|
|
* ext/theora/theoradec.c: (_theora_granule_time),
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(theora_dec_push_forward), (theora_handle_data_packet),
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(theora_dec_decode_buffer): Calculate buffer duration correctly
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|
to generate a perfect stream (#433888).
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* gst/audioresample/gstaudioresample.c:
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(audioresample_check_discont): Glib provides ABS.
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|
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2007-04-27 15:01:40 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstrtcpbuffer.*: Fix RB block parsing and writing.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_packet_get_rb),
|
|
(gst_rtcp_packet_add_rb), (gst_rtcp_packet_sdes_add_item),
|
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(gst_rtcp_packet_sdes_add_entry), (gst_rtcp_packet_bye_add_ssrc),
|
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(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
|
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(gst_rtcp_packet_bye_set_reason):
|
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* gst-libs/gst/rtp/gstrtcpbuffer.h:
|
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Fix RB block parsing and writing.
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Add support for constructing BYE packets.
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|
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2007-04-25 08:54:34 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
When posting a warning message because samples were dropped, post something more intelligible than he default error m...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c: (_do_init),
|
|
(gst_base_audio_src_create):
|
|
* po/POTFILES.in:
|
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When posting a warning message because samples were dropped, post
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something more intelligible than he default error message for clock
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errors which is just confusing in this context (#432984).
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|
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2007-04-25 08:10:26 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstrtcpbuffer.*: Implement code to write SR, RR and SDES packets.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstrtcpbuffer.c: (gst_rtcp_buffer_new),
|
|
(gst_rtcp_buffer_end), (gst_rtcp_buffer_get_packet_count),
|
|
(read_packet_header), (gst_rtcp_packet_move_to_next),
|
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(gst_rtcp_buffer_add_packet), (gst_rtcp_packet_sr_set_sender_info),
|
|
(gst_rtcp_packet_rr_set_ssrc), (gst_rtcp_packet_add_rb),
|
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(gst_rtcp_packet_sdes_get_item_count),
|
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(gst_rtcp_packet_sdes_first_item),
|
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(gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_ssrc),
|
|
(gst_rtcp_packet_sdes_first_entry),
|
|
(gst_rtcp_packet_sdes_next_entry),
|
|
(gst_rtcp_packet_sdes_get_entry), (gst_rtcp_packet_sdes_add_item),
|
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(gst_rtcp_packet_sdes_add_entry):
|
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* gst-libs/gst/rtp/gstrtcpbuffer.h:
|
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Implement code to write SR, RR and SDES packets.
|
|
|
|
2007-04-24 20:45:24 +0000 Christian Kirbach <Christian.Kirbach@googlemail.com>
|
|
|
|
sys/ximage/ximagesink.c: Fix build if XShm is not available (#432362).
|
|
Original commit message from CVS:
|
|
Patch by: Christian Kirbach <Christian dot Kirbach at googlemail com>
|
|
* sys/ximage/ximagesink.c:
|
|
Fix build if XShm is not available (#432362).
|
|
|
|
2007-04-24 18:58:25 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst/audioconvert/gstaudioconvert.c: Initalize the AudioConvertCtx with zeroes, otherwise it will contain pointers to ...
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init):
|
|
Initalize the AudioConvertCtx with zeroes, otherwise it will contain
|
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pointers to random memory which are passed to g_free() when
|
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audio_convert_prepare_context() is called the first time.
|
|
|
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2007-04-24 15:00:07 +0000 Dan Williams <dcbw@redhat.com>
|
|
|
|
gst/videorate/gstvideorate.c: Don't leak incoming buffer if gst_pad_push() returns a non-OK flow. Fixes #432755.
|
|
Original commit message from CVS:
|
|
Patch by: Dan Williams <dcbw redhat com>
|
|
* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
|
|
Don't leak incoming buffer if gst_pad_push() returns a
|
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non-OK flow. Fixes #432755.
|
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* tests/check/elements/videorate.c: (GST_START_TEST),
|
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(videorate_suite):
|
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Unit test for the above by Yours Truly.
|
|
|
|
2007-04-23 20:04:28 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/adder/gstadder.c: Fix non-flushing segmented seeks, Fixes #340060 for me
|
|
Original commit message from CVS:
|
|
* gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
|
|
(gst_adder_sink_event), (gst_adder_collected):
|
|
Fix non-flushing segmented seeks, Fixes #340060 for me
|
|
|
|
2007-04-21 15:29:27 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
* ChangeLog:
|
|
ChangeLog surgery: add API keyword
|
|
Original commit message from CVS:
|
|
ChangeLog surgery: add API keyword
|
|
|
|
2007-04-21 15:25:22 +0000 Olivier Crete <tester@tester.ca>
|
|
|
|
gst-libs/gst/rtp/gstbasertpaudiopayload.c: Chain up to parent class in dispose function; get rid of unnecessary 'dipo...
|
|
Original commit message from CVS:
|
|
Patch by: Olivier Crete <tester at tester ca>
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
(gst_base_rtp_audio_payload_class_init),
|
|
(gst_base_rtp_audio_payload_init),
|
|
(gst_base_rtp_audio_payload_dispose):
|
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Chain up to parent class in dispose function; get rid of
|
|
unnecessary 'diposed' flag in private structure (#415001).
|
|
|
|
2007-04-21 15:10:25 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
Some minor docs fixes and additions; also add missing 'Since' bits.
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs.types:
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
(gst_base_rtp_audio_payload_class_init):
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
* gst-libs/gst/rtp/gstbasertppayload.c:
|
|
Some minor docs fixes and additions; also add missing 'Since' bits.
|
|
|
|
2007-04-21 14:40:45 +0000 Zeeshan Ali <zeenix@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstbasertpaudiopayload.*: The recently-added gst_base_rtp_audio_payload_push() should take an object...
|
|
Original commit message from CVS:
|
|
Patch by: Zeeshan Ali <zeenix gmail com>
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
|
|
(gst_base_rtp_audio_payload_handle_sample_based_buffer),
|
|
(gst_base_rtp_audio_payload_push):
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
|
|
The recently-added gst_base_rtp_audio_payload_push() should take an
|
|
object of type GstBaseRTPAudioPayload as first argument (#431672).
|
|
|
|
2007-04-21 14:14:24 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/audioresample/gstaudioresample.c: Make more functions static, just because we can.
|
|
Original commit message from CVS:
|
|
* gst/audioresample/gstaudioresample.c:
|
|
Make more functions static, just because we can.
|
|
|
|
2007-04-21 13:54:39 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/elements/audioresample.c: Add unit test for audioresample shutdown crasher (#420106).
|
|
Original commit message from CVS:
|
|
* tests/check/elements/audioresample.c:
|
|
Add unit test for audioresample shutdown crasher (#420106).
|
|
|
|
2007-04-20 10:42:24 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/subparse/: Use GST_DISABLE_XML here
|
|
Original commit message from CVS:
|
|
* gst/subparse/gstsubparse.c:
|
|
* gst/subparse/samiparse.c:
|
|
Use GST_DISABLE_XML here
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put),
|
|
(gst_xvimagesink_xwindow_new), (gst_xvimagesink_get_xv_support),
|
|
(gst_xvimagesink_buffer_alloc),
|
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(gst_xvimagesink_navigation_send_event):
|
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* sys/xvimage/xvimagesink.h:
|
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Include stdlib.h when using atoi.
|
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* tests/check/elements/playbin.c: (playbin_suite):
|
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Use GST_DISABLE_REGISTRY here
|
|
|
|
2007-04-19 16:58:53 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
ext/theora/: Track initialisation state; don't try to use encoder state if we're not initialised (it'll segfault).
|
|
Original commit message from CVS:
|
|
* ext/theora/gsttheoraenc.h:
|
|
* ext/theora/theoraenc.c: (theora_enc_sink_setcaps),
|
|
(theora_enc_sink_event), (theora_enc_change_state):
|
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Track initialisation state; don't try to use encoder state if we're
|
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not initialised (it'll segfault).
|
|
|
|
2007-04-18 11:06:42 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
tests/check/pipelines/.cvsignore: Fix build.
|
|
Original commit message from CVS:
|
|
* tests/check/pipelines/.cvsignore:
|
|
Fix build.
|
|
|
|
2007-04-17 10:56:37 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/app/Makefile.am: Fix CFLAGS and hopefully #430594.
|
|
Original commit message from CVS:
|
|
* gst/app/Makefile.am:
|
|
Fix CFLAGS and hopefully #430594.
|
|
|
|
2007-04-17 02:53:16 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
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gst-libs/gst/riff/riff-media.c: Allow random depths between 1 and 32 instead of only multiplies of 8.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
|
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Allow random depths between 1 and 32 instead of only multiplies of 8.
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|
|
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2007-04-17 02:04:21 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
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|
gst-libs/gst/riff/riff-media.c: Set the maximum number of channels for PCM and float in the correct place to have it ...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
|
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Set the maximum number of channels for PCM and float in the correct
|
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place to have it also used when creating the template caps.
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|
|
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2007-04-17 01:56:07 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
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gst-libs/gst/riff/riff-media.c: Correctly support 4, 6 and 8 channels with normal PCM and float wav files.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
|
|
Correctly support 4, 6 and 8 channels with normal PCM and float
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wav files.
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Fix the depth and signedness calculation in extensible wav files and
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also handle 1, 2, 4, 6, 8 channels here when a file without channel
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mask is found.
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Add support for float, alaw and mulaw in extensible wav files.
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This allows correct playback of all but 5 files from
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http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/Samples.html
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(gst_riff_create_audio_template_caps):
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Add voxware and float formats to the template caps.
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|
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2007-04-16 22:20:03 +0000 Vincent Torri <vtorri@univ-evry.fr>
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ext/pango/gstclockoverlay.c: Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
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Original commit message from CVS:
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Patch by: Vincent Torri <vtorri at univ-evry dot fr>
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* ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
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Fix unused variable warning if HAVE_LOCALTIME_R is undefinied
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* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
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* gst/audioresample/gstaudioresample.c: (audioresample_do_output):
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Use the correct format strings for integer formats.
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2007-04-16 21:44:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* gst-plugins-base.doap:
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fix release date
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Original commit message from CVS:
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fix release date
|
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2007-04-16 21:42:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* gst-plugins-base.doap:
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fix release date
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Original commit message from CVS:
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fix release date
|
|
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2007-04-15 14:35:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain): Don't use pad_alloc_buffer_and_set_caps to crea...
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Original commit message from CVS:
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* ext/theora/theoraenc.c (theora_buffer_from_packet, theora_enc_chain):
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Don't use pad_alloc_buffer_and_set_caps to create a small header
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packet, or, worse, to create a big temporary video buffer using the
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src pad.
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2007-04-14 12:34:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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gst/gdp/gstgdppay.c (gst_gdp_pay_chain): tests/check/pipelines/streamheader.c (tag_event_probe_cb,
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Original commit message from CVS:
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* gst/gdp/gstgdppay.c (gst_gdp_pay_chain):
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* tests/check/pipelines/streamheader.c (tag_event_probe_cb,
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GST_START_TEST, buffer_probe_cb, GST_START_TEST):
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Fix a bug where serialized IN_CAPS buffers needed to be set IN_CAPS.
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2007-04-13 22:10:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* gst/tcp/gstmultifdsink.c:
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add debug
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Original commit message from CVS:
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add debug
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2007-04-13 21:55:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* ChangeLog:
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* tests/check/pipelines/streamheader.c:
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tests/check/pipelines/streamheader.c (tag_event_probe_cb,
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Original commit message from CVS:
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* tests/check/pipelines/streamheader.c (tag_event_probe_cb,
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GST_START_TEST, n_in_caps, buffer_probe_cb, GST_START_TEST,
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streamheader_suite):
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Add another test set up for failure
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2007-04-13 21:09:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* ext/ogg/gstoggmux.c:
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* gst/gdp/gstgdpdepay.c:
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debug changes
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Original commit message from CVS:
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debug changes
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2007-04-13 21:08:11 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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tests/check/Makefile.am: tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
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Original commit message from CVS:
|
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* tests/check/Makefile.am:
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* tests/check/pipelines/streamheader.c (n_tags, tag_event_probe_cb,
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GST_START_TEST, streamheader_suite, main):
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Add a test for the streamheader bug Wim fixed.
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2007-04-13 11:42:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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ext/theora/theoradec.c: Fix misleading comment.
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Original commit message from CVS:
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* ext/theora/theoradec.c: (theora_dec_sink_event):
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Fix misleading comment.
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2007-04-13 06:17:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst-libs/gst/riff/riff-media.c: More sanity checks for the header fields.
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Original commit message from CVS:
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* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
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More sanity checks for the header fields.
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2007-04-12 16:36:36 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/tag/tags.c: Try encodings from all environment variables, not just those in the first environment variab...
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|
Original commit message from CVS:
|
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* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
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Try encodings from all environment variables, not just those in the
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first environment variable that is set.
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2007-04-12 15:00:03 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/videorate/gstvideorate.c: Add some debug.
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|
Original commit message from CVS:
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* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
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(gst_video_rate_chain):
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Add some debug.
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* tests/check/elements/videorate.c: (GST_START_TEST),
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(videorate_suite):
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Added check for videorate changing caps handling. Closes #421834.
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2007-04-12 12:57:33 +0000 Michael Smith <msmith@xiph.org>
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ext/vorbis/vorbisdec.c: Use scale functions to avoid overflow when calculating duration of vorbis buffers.
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|
Original commit message from CVS:
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* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
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Use scale functions to avoid overflow when calculating duration of
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vorbis buffers.
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2007-04-12 12:19:20 +0000 Tim-Philipp Müller <tim@centricular.net>
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API: add gst_tag_freeform_string_to_utf8() (#405072).
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|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
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* gst-libs/gst/tag/tag.h:
|
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* gst-libs/gst/tag/tags.c: (gst_tag_freeform_string_to_utf8):
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API: add gst_tag_freeform_string_to_utf8() (#405072).
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* gst-libs/gst/tag/gstid3tag.c: (gst_tag_extract_id3v1_string):
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Use gst_tag_freeform_string_to_utf8() here.
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2007-04-12 10:38:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* gst/tcp/gstmultifdsink.c:
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log tweaking
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|
Original commit message from CVS:
|
|
log tweaking
|
|
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2007-04-12 10:03:22 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/gdp/gstgdppay.c: Make sure we set the IN_CAPS flag correctly.
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|
Original commit message from CVS:
|
|
* gst/gdp/gstgdppay.c: (gst_gdp_pay_chain),
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(gst_gdp_pay_sink_event):
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Make sure we set the IN_CAPS flag correctly.
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* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
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Get the IN_CAPS flag before we call functions that mess with the flags.
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2007-04-10 20:37:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* ChangeLog:
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* gst/gdp/gstgdppay.c:
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gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader, gst_gdp_pay_chain, gst_gdp_pay_sink_event):
|
|
Original commit message from CVS:
|
|
* gst/gdp/gstgdppay.c (gst_gdp_pay_reset_streamheader,
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gst_gdp_pay_chain, gst_gdp_pay_sink_event):
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Only stamp buffers with offset/offset_end right before they get
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pushed. This ensures offset continuity, which was not the case
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before as shown by
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gst-launch -v -m audiotestsrc num-buffers=10 ! audioconvert ! vorbisenc ! gdppay ! identity check-imperfect-offset=TRUE ! fakesink silent=TRUE
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2007-04-10 20:25:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* gst/gdp/gstgdpdepay.c:
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* gst/gdp/gstgdppay.c:
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adding debugging
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|
Original commit message from CVS:
|
|
adding debugging
|
|
|
|
2007-04-10 11:23:18 +0000 Christian Schaller <uraeus@gnome.org>
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* common:
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* gst-plugins-base.spec.in:
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update spec file for RTP changes
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|
Original commit message from CVS:
|
|
update spec file for RTP changes
|
|
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2007-04-06 12:58:06 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstplaybin.c: Activate sync in playbin, we are ready to handle it for live streams.
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|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin.c: (add_sink),
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(gst_play_bin_change_state):
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Activate sync in playbin, we are ready to handle it for live streams.
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2007-04-06 09:56:18 +0000 Tim-Philipp Müller <tim@centricular.net>
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tests/check/elements/playbin.c: Add small test for stream-info-value-array code paths.
|
|
Original commit message from CVS:
|
|
* tests/check/elements/playbin.c:
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|
(test_sink_usage_video_only_stream), (playbin_suite):
|
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Add small test for stream-info-value-array code paths.
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2007-04-05 15:44:40 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to create invalid calibration parameters by making the internal time...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_skew_slaving):
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Don't try to create invalid calibration parameters by making the
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internal time go backwards, instead make external time go forward.
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2007-04-05 10:27:06 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
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gst/playback/gstplaybasebin.c: Fix leak in add_stream(), when g_value_set_object() increases the refcount of streamin...
|
|
Original commit message from CVS:
|
|
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
|
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* gst/playback/gstplaybasebin.c: (add_stream):
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Fix leak in add_stream(), when g_value_set_object() increases the
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refcount of streaminfo object. Fixes #426250.
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2007-04-04 02:45:03 +0000 David Schleef <ds@schleef.org>
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|
|
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gst/videotestsrc/: Add a test pattern called "circular", which has concentric rings with varying radial frequency. T...
|
|
Original commit message from CVS:
|
|
* gst/videotestsrc/gstvideotestsrc.c:
|
|
* gst/videotestsrc/gstvideotestsrc.h:
|
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* gst/videotestsrc/videotestsrc.c:
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* gst/videotestsrc/videotestsrc.h:
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Add a test pattern called "circular", which has concentric
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rings with varying radial frequency. The main purpose of this
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pattern is to test fidelity loss in a filter or scaler element.
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Notably, this pattern is scale invariant, and is optimally viewed
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with a width (and height) of 400.
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2007-04-03 11:10:52 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
|
|
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|
gst/playback/gstdecodebin2.c: Decodebin2 doesn't unref pads it obtains in some occasions:
|
|
Original commit message from CVS:
|
|
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
|
|
* gst/playback/gstdecodebin2.c: (connect_pad), (expose_pad),
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(deactivate_free_recursive):
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Decodebin2 doesn't unref pads it obtains in some occasions:
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- multiqueue src pads, when either connecting further or exposing
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- sink pads of new autoplugged elements
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- peer pads when recursively freeing elements
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Fixes #425455.
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2007-03-30 17:05:23 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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|
|
|
gst-libs/gst/riff/riff-media.c: Add audio/x-raw-float support, now that audioconvert support non-native endianness fl...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
|
|
Add audio/x-raw-float support, now that audioconvert support
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non-native endianness floats.
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|
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2007-03-30 15:00:49 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
docs/libs/gst-plugins-base-libs-docs.sgml: gstreamer-plugins-base.pc doesn't exist, it's gstreamer-plugins-base-0.10.pc.
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-docs.sgml:
|
|
gstreamer-plugins-base.pc doesn't exist, it's
|
|
gstreamer-plugins-base-0.10.pc.
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|
|
|
2007-03-29 18:42:34 +0000 René Stadler <mail@renestadler.de>
|
|
|
|
with some minor changes
|
|
Original commit message from CVS:
|
|
Patch by: René Stadler <mail at renestadler dot de>
|
|
with some minor changes
|
|
* gst-libs/gst/floatcast/floatcast.h:
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Use more efficient float endianness conversion functions that don't
|
|
involve 2 function calls per value.
|
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* gst/audioconvert/audioconvert.c: (audio_convert_get_func_index),
|
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(check_default), (audio_convert_prepare_context):
|
|
* gst/audioconvert/gstaudioconvert.c:
|
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(gst_audio_convert_parse_caps), (make_lossless_changes):
|
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Support non-native endianness floats as input and output.
|
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Fixes #339838.
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* tests/check/elements/audioconvert.c: (verify_convert),
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(GST_START_TEST):
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Add unit tests for the non-native endianness float conversions.
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|
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2007-03-29 16:23:53 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstbasertpdepayload.*: Add Private structure.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_base_init),
|
|
(gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
|
|
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
|
|
(gst_base_rtp_depayload_set_gst_timestamp),
|
|
(gst_base_rtp_depayload_change_state),
|
|
(gst_base_rtp_depayload_set_property),
|
|
(gst_base_rtp_depayload_get_property):
|
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* gst-libs/gst/rtp/gstbasertpdepayload.h:
|
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Add Private structure.
|
|
Bring element code to 2007.
|
|
Parse clock-base caps param and use it when generating the
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newsegment.
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Reset variables before going to PAUSED.
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Fix some docs.
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|
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2007-03-29 16:20:31 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
Add RTCP docs.
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-docs.sgml:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
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(gst_base_rtp_audio_payload_get_adapter):
|
|
Add RTCP docs.
|
|
Fix some more docs.
|
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* gst-libs/gst/rtp/Makefile.am:
|
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* gst-libs/gst/rtp/gstrtcpbuffer.c:
|
|
(gst_rtcp_buffer_new_take_data), (gst_rtcp_buffer_new_copy_data),
|
|
(gst_rtcp_buffer_validate_data), (gst_rtcp_buffer_validate),
|
|
(gst_rtcp_buffer_get_packet_count), (read_packet_header),
|
|
(gst_rtcp_buffer_get_first_packet), (gst_rtcp_packet_move_to_next),
|
|
(gst_rtcp_buffer_add_packet), (gst_rtcp_packet_remove),
|
|
(gst_rtcp_packet_get_padding), (gst_rtcp_packet_get_type),
|
|
(gst_rtcp_packet_get_count), (gst_rtcp_packet_get_length),
|
|
(gst_rtcp_packet_sr_get_sender_info),
|
|
(gst_rtcp_packet_sr_set_sender_info),
|
|
(gst_rtcp_packet_rr_get_ssrc), (gst_rtcp_packet_rr_set_ssrc),
|
|
(gst_rtcp_packet_get_rb_count), (gst_rtcp_packet_get_rb),
|
|
(gst_rtcp_packet_add_rb), (gst_rtcp_packet_set_rb),
|
|
(gst_rtcp_packet_sdes_get_chunk_count),
|
|
(gst_rtcp_packet_sdes_first_chunk),
|
|
(gst_rtcp_packet_sdes_next_chunk), (gst_rtcp_packet_sdes_get_ssrc),
|
|
(gst_rtcp_packet_sdes_first_item),
|
|
(gst_rtcp_packet_sdes_next_item), (gst_rtcp_packet_sdes_get_item),
|
|
(gst_rtcp_packet_bye_get_ssrc_count),
|
|
(gst_rtcp_packet_bye_get_nth_ssrc), (gst_rtcp_packet_bye_add_ssrc),
|
|
(gst_rtcp_packet_bye_add_ssrcs), (get_reason_offset),
|
|
(gst_rtcp_packet_bye_get_reason_len),
|
|
(gst_rtcp_packet_bye_get_reason), (gst_rtcp_packet_bye_set_reason):
|
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* gst-libs/gst/rtp/gstrtcpbuffer.h:
|
|
Add new helper object for parsing and creating RTCP messages.
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|
|
|
2007-03-29 12:07:02 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
gst-libs/gst/riff/riff-media.c: PCM samples with width=8 must be always unsigned, no matter what depth they have.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
|
|
PCM samples with width=8 must be always unsigned, no matter what
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|
depth they have.
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|
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2007-03-29 11:24:47 +0000 Andy Wingo <wingo@pobox.com>
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|
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gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make perfect offsets also, not just timestamps.
|
|
Original commit message from CVS:
|
|
2007-03-29 Andy Wingo <wingo@pobox.com>
|
|
* gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Make
|
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perfect offsets also, not just timestamps.
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* tests/check/elements/videorate.c (test_more): Test that given
|
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any incoming offsets, that videorate produces perfect offsets.
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|
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2007-03-29 10:19:45 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/riff/riff-ids.h: Add some more RIFF formats.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-ids.h:
|
|
Add some more RIFF formats.
|
|
|
|
2007-03-29 10:17:52 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstrtpbuffer.*: Fix fixed payload names and docs.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstrtpbuffer.c:
|
|
(gst_rtp_buffer_default_clock_rate):
|
|
* gst-libs/gst/rtp/gstrtpbuffer.h:
|
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Fix fixed payload names and docs.
|
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Added method to get the default clock rates of fixed payload types.
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API: GstRTPBuffer::gst_rtp_buffer_default_clock_rate()
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2007-03-28 15:24:40 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
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|
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tests/check/pipelines/.cvsignore: Add new vorbisdec test to cvsignore.
|
|
Original commit message from CVS:
|
|
* tests/check/pipelines/.cvsignore:
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Add new vorbisdec test to cvsignore.
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2007-03-28 14:50:47 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstbaseaudiosink.*: Store private stuff in GstBaseAudioSinkPrivate.
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Original commit message from CVS:
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* gst-libs/gst/audio/gstbaseaudiosink.c: (slave_method_get_type),
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(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
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(gst_base_audio_sink_query), (gst_base_audio_sink_get_time),
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(gst_base_audio_sink_set_property),
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(gst_base_audio_sink_get_property), (gst_base_audio_sink_event),
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(clock_convert_external), (gst_base_audio_sink_resample_slaving),
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(gst_base_audio_sink_skew_slaving),
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(gst_base_audio_sink_handle_slaving), (gst_base_audio_sink_render),
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(gst_base_audio_sink_async_play):
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* gst-libs/gst/audio/gstbaseaudiosink.h:
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Store private stuff in GstBaseAudioSinkPrivate.
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Add configurable clock slaving modes property.
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API:: GstBaseAudioSink::slave-method property
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Some more latency reporting tweaks.
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Added skew based clock slaving correction and make it the default until
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the resampling method is more robust.
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2007-03-27 12:44:14 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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gst/audioconvert/audioconvert.c: Add docs to the integer pack functions and implement proper rounding. Before we had ...
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Original commit message from CVS:
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* gst/audioconvert/audioconvert.c:
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Add docs to the integer pack functions and implement proper
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rounding. Before we had rounding towards negative infinity, i.e.
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always the smaller number was taken. Now we use natural rounding,
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i.e. rounding to the nearest integer and to the one with the largest
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absolute value for X.5. The old rounding introduced some minor
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distortions. Fixes #420079
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* tests/check/elements/audioconvert.c: (GST_START_TEST):
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Fix one unit test that assumed the old rounding and added unit tests
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for checking signed/unsigned int16 <-> signed/unsigned int16 with
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depth 8, one for signed int16 <-> unsigned int16 and one for the new
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rounding from signed int32 to signed/unsigned int16.
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2007-03-27 11:31:17 +0000 Michael Smith <msmith@xiph.org>
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gst/audioconvert/gstaudioconvert.c: Fix typo in debug line introduced recently, as pointed out on irc.
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Original commit message from CVS:
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* gst/audioconvert/gstaudioconvert.c: (strip_width_64),
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(gst_audio_convert_transform_caps):
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Fix typo in debug line introduced recently, as pointed out on irc.
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2007-03-27 10:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
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Make sure we parse floating-point numbers in vorbis comments correctly with either '.' or ',' as separator, no matter...
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Original commit message from CVS:
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* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
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* tests/check/libs/tag.c: (GST_START_TEST):
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Make sure we parse floating-point numbers in vorbis comments
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correctly with either '.' or ',' as separator, no matter what
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the current locale is. Add unit test for this too.
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2007-03-27 09:37:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* tests/check/pipelines/vorbisdec.c:
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commit new file
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Original commit message from CVS:
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commit new file
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2007-03-26 22:38:19 +0000 René Stadler <mail@renestadler.de>
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gst-libs/gst/tag/gstvorbistag.c: When writing out floating-point numbers to vorbis comment tags, always use the same ...
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Original commit message from CVS:
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Patch by: René Stadler <mail at renestadler de>
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* gst-libs/gst/tag/gstvorbistag.c: (gst_tag_to_vorbis_comments):
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When writing out floating-point numbers to vorbis comment tags, always
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use the same character as separator no matter what the current locale is
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(fixes #423051).
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* tests/check/libs/tag.c: (GST_START_TEST):
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Add unit tests for replaygain tags in vorbis comments (closes #423055).
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2007-03-26 20:56:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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ext/vorbis/vorbisdec.c (vorbis_dec_push_forward, vorbis_handle_data_packet):
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Original commit message from CVS:
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* ext/vorbis/vorbisdec.c (vorbis_dec_push_forward,
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vorbis_handle_data_packet):
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Correctly set DURATION to generate a timestamp-continuous stream.
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One bug left at the end; see
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ihttp://bugzilla.gnome.org/show_bug.cgi?id=423086
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* tests/check/Makefile.am:
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* tests/check/pipelines/vorbisenc.c (GST_START_TEST):
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Add a test to check this. Without the above patch this test fails.
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2007-03-26 11:44:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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gst-libs/gst/rtp/Makefile.am: The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
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Original commit message from CVS:
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* gst-libs/gst/rtp/Makefile.am:
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The base audio payloader uses GstAdapter - we need GST_BASE_LIBS.
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2007-03-23 15:43:24 +0000 Christian Schaller <uraeus@gnome.org>
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* gst-plugins-base.spec.in:
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update spec file
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Original commit message from CVS:
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update spec file
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2007-03-23 12:32:33 +0000 Michael Smith <msmith@xiph.org>
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gst/videorate/gstvideorate.c: If videorate changes caps, we can no longer use the old buffer (which may have a differ...
|
|
Original commit message from CVS:
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* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps),
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(gst_video_rate_reset), (gst_video_rate_chain):
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If videorate changes caps, we can no longer use the old buffer
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(which may have a different size, incompatible with our caps).
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So don't do that; just duplicate the new frame more times.
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2007-03-22 17:43:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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gst/playback/gstplaybin.c: Remove playbin's override of the set_clock vmethod. It's irrelevant after Wim's commit on ...
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|
Original commit message from CVS:
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* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
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Remove playbin's override of the set_clock vmethod. It's irrelevant
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after Wim's commit on the 19th.
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2007-03-22 14:37:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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gst-libs/gst/app/Makefile.am: Use GST_ALL_LDFLAGS, which actually exists, but maybe David can confirm that was what h...
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|
Original commit message from CVS:
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* gst-libs/gst/app/Makefile.am:
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Use GST_ALL_LDFLAGS, which actually exists, but maybe David
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can confirm that was what he wanted.
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2007-03-22 09:26:02 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/gnomevfs/gstgnomevfssrc.*: Don't cache file sizes. Fixes #341078.
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Original commit message from CVS:
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* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_size),
|
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(gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
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* ext/gnomevfs/gstgnomevfssrc.h:
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Don't cache file sizes. Fixes #341078.
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2007-03-21 11:03:23 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gstplaybin.c: Use GST_PTR_FORMAT to log caps.
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Original commit message from CVS:
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* gst/playback/gstplaybin.c: (add_sink):
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Use GST_PTR_FORMAT to log caps.
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2007-03-21 10:23:11 +0000 Young-Ho Cha <ganadist@chollian.net>
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gst/subparse/samiparse.c: Special-case some more colour names that pango doesn't handle by default. Fixes #420578.
|
|
Original commit message from CVS:
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|
Patch by: Young-Ho Cha <ganadist at chollian net>
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* gst/subparse/samiparse.c: (handle_start_font):
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Special-case some more colour names that pango doesn't handle by
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default. Fixes #420578.
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2007-03-20 11:49:55 +0000 Michael Smith <msmith@xiph.org>
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ext/vorbis/vorbisenc.c: If we get a zero-sized input buffer, don't pass it to libvorbis, as that marks EOS internally...
|
|
Original commit message from CVS:
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* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
|
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If we get a zero-sized input buffer, don't pass it to libvorbis, as
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that marks EOS internally. After that, libvorbis will buffer all
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input data, and encode none of it, eventually leading to memory
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exhaustion.
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2007-03-19 10:52:50 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstdecodebin.c: Don't post STATE_DIRTY anymore.
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Original commit message from CVS:
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* gst/playback/gstdecodebin.c: (remove_fakesink):
|
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Don't post STATE_DIRTY anymore.
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* gst/playback/gstplaybin.c: (add_sink), (gst_play_bin_send_event),
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(gst_play_bin_change_state):
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Remove stream_time reset in seek handling, core does that now.
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Disable clocking for live pipelines by forcing a NULL clock to the
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complete pipeline, core is too smart now for our previous hack.
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We can always autoplug in PAUSED now.
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2007-03-18 03:14:01 +0000 David Schleef <ds@schleef.org>
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REQUIREMENTS: Update this file, change the formatting to make it more consistent, plus more machine readable.
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|
Original commit message from CVS:
|
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* REQUIREMENTS: Update this file, change the formatting to make
|
|
it more consistent, plus more machine readable.
|
|
|
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2007-03-16 17:29:09 +0000 Michael Smith <msmith@xiph.org>
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|
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gst/audioconvert/gstaudioconvert.c: Previous fix was too simplistic, and broke the tests. Use a better approach; only...
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
|
|
(strip_width_64), (append_with_other_format):
|
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Previous fix was too simplistic, and broke the tests. Use a better
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approach; only strip 64 from widths for integer audio.
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2007-03-16 16:42:23 +0000 Michael Smith <msmith@xiph.org>
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|
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gst/audioconvert/gstaudioconvert.c: We don't support 64 bit integer audio, so don't try to claim we can.
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
|
|
(gst_audio_convert_transform_caps):
|
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We don't support 64 bit integer audio, so don't try to claim we can.
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Stops us producing caps don't match our template caps.
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Update comments.
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2007-03-15 10:52:21 +0000 Michael Smith <msmith@xiph.org>
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|
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gst/audioresample/gstaudioresample.c: Don't trigger discontinuities for very small imperfections; a filter flush will...
|
|
Original commit message from CVS:
|
|
* gst/audioresample/gstaudioresample.c:
|
|
(audioresample_check_discont), (audioresample_transform):
|
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Don't trigger discontinuities for very small imperfections; a filter
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flush will sound bad, and many plugins have rounding errors leading
|
|
to these.
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2007-03-14 21:11:18 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
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gst-libs/gst/rtp/gstbasertpaudiopayload.*: olivier.crete@collabora.co.uk.
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|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
|
|
Add min-ptime property to RTP base audio payloader. Patch by
|
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olivier.crete@collabora.co.uk.
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Fixes #415001
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Indentation/whitespace/documentation fixes.
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2007-03-14 17:16:30 +0000 Julien Moutte <julien@moutte.net>
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gst/audioresample/gstaudioresample.c: Handle discontinuous streams.
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|
Original commit message from CVS:
|
|
2007-03-14 Julien MOUTTE <julien@moutte.net>
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* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
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(audioresample_transform_size), (audioresample_do_output),
|
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(audioresample_transform), (audioresample_pushthrough): Handle
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discontinuous streams.
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* gst/audioresample/gstaudioresample.h:
|
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* tests/check/elements/audioresample.c:
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(test_discont_stream_instance), (GST_START_TEST),
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(audioresample_suite): Add a test for discontinuous streams.
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* win32/common/config.h: Updated.
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2007-03-14 15:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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|
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po/: Update translations from translation project.
|
|
Original commit message from CVS:
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/cs.po:
|
|
* po/en_GB.po:
|
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* po/it.po:
|
|
* po/nb.po:
|
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* po/nl.po:
|
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* po/or.po:
|
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* po/sq.po:
|
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* po/sr.po:
|
|
* po/sv.po:
|
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* po/uk.po:
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* po/vi.po:
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Update translations from translation project.
|
|
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2007-03-14 15:05:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
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* gst/gdp/gstgdpdepay.c:
|
|
add buffer logging
|
|
Original commit message from CVS:
|
|
add buffer logging
|
|
|
|
2007-03-14 14:48:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
gst/audioresample/: Since I really am not interested in a debug line for each sample being processed, move the librar...
|
|
Original commit message from CVS:
|
|
* gst/audioresample/debug.h:
|
|
* gst/audioresample/resample.c: (resample_init):
|
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Since I really am not interested in a debug line for each sample
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being processed, move the library's debugging to its own category,
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libaudioresample
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2007-03-14 14:09:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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|
|
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* gst/audioresample/gstaudioresample.c:
|
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add debugging and reformat docs
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|
Original commit message from CVS:
|
|
add debugging and reformat docs
|
|
|
|
2007-03-12 23:29:07 +0000 Michael Smith <msmith@xiph.org>
|
|
|
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ext/theora/theoradec.c: Since the plugin doesn't support anything other than 4:2:0 right now, post an error and fail ...
|
|
Original commit message from CVS:
|
|
* ext/theora/theoradec.c: (theora_handle_type_packet):
|
|
Since the plugin doesn't support anything other than 4:2:0 right
|
|
now, post an error and fail if we get something else. Won't matter
|
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until libtheora supports the other pixel formats, but hopefully
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that'll be soon...
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2007-03-12 15:50:35 +0000 Alex Lancaster <alexlan@fedoraproject.org>
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|
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* ChangeLog:
|
|
I'm too lazy to comment this
|
|
Original commit message from CVS:
|
|
Mention Patch by: Alex Lancaster in a recent commit.
|
|
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2007-03-12 11:47:42 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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examples/app/.cvsignore: The buildbot demands .cvsignore files, and I comply.
|
|
Original commit message from CVS:
|
|
* examples/app/.cvsignore:
|
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The buildbot demands .cvsignore files, and I comply.
|
|
|
|
2007-03-11 00:48:26 +0000 David Schleef <ds@schleef.org>
|
|
|
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Add appsrc/appsink example.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* examples/Makefile.am:
|
|
* examples/app/Makefile.am:
|
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* examples/app/appsrc_ex.c:
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Add appsrc/appsink example.
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* gst-libs/gst/app/Makefile.am:
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* gst-libs/gst/app/gstapp.c:
|
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* gst-libs/gst/app/gstappsink.c:
|
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* gst-libs/gst/app/gstappsink.h:
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* gst/app/gstapp.c:
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Add appsink.
|
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2007-03-10 15:59:33 +0000 Sébastien Moutte <sebastien@moutte.net>
|
|
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gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_guint64_to_gdouble for conversion.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:(gst_base_audio_sink_render):
|
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Use gst_guint64_to_gdouble for conversion.
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* win32/MANIFEST:
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Add new files to the win32 MANIFEST.
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* win32/common/libgstaudio.def:
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* win32/common/libgstpbutils.def:
|
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Add new exported functions.
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* win32/vs6/gst_plugins_base.dsw:
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* win32/vs6/libgstdecodebin.dsp:
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* win32/vs6/libgstplaybin.dsp:
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Change the link to libgstpbutils.lib.
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* win32/vs6/libgstdecodebin2.dsp:
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Add a new project for decodebin2.
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* win32/vs6/libgstpbutils.dsp:
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Add a new project for pbutils.
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2007-03-10 12:18:58 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/tag/gstvorbistag.c: Also accept partial dates with only year and month, like 1999-12-00 (fixes #410396 e...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
|
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Also accept partial dates with only year and month,
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like 1999-12-00 (fixes #410396 even more).
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* tests/check/libs/tag.c: (GST_START_TEST):
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Add unit test for the above.
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2007-03-10 11:21:08 +0000 Tim-Philipp Müller <tim@centricular.net>
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tests/check/elements/subparse.c: Add unit test for MPL2 subtitle format (#413799).
|
|
Original commit message from CVS:
|
|
* tests/check/elements/subparse.c: (GST_START_TEST),
|
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(subparse_suite):
|
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Add unit test for MPL2 subtitle format (#413799).
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2007-03-10 11:17:52 +0000 Kamil Pawlowski <kamilpe@gmail.com>
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gst/subparse/: Add support for MPL2 subtitle format (#413799).
|
|
Original commit message from CVS:
|
|
Patch by: Kamil Pawlowski <kamilpe gmail com>
|
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* gst/subparse/Makefile.am:
|
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* gst/subparse/gstsubparse.c:
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(gst_sub_parse_data_format_autodetect),
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(gst_sub_parse_format_autodetect), (gst_sub_parse_sink_event),
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(gst_subparse_type_find):
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* gst/subparse/gstsubparse.h:
|
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* gst/subparse/mpl2parse.c: (mpl2_parse_line), (parse_mpl2):
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* gst/subparse/mpl2parse.h:
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Add support for MPL2 subtitle format (#413799).
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2007-03-09 17:33:17 +0000 Tim-Philipp Müller <tim@centricular.net>
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configure.ac: We require core CVS for the new buffer metadata copy functions.
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Original commit message from CVS:
|
|
* configure.ac:
|
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We require core CVS for the new buffer metadata copy functions.
|
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2007-03-09 16:51:13 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst-libs/gst/tag/gstid3tag.c: Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
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Original commit message from CVS:
|
|
* gst-libs/gst/tag/gstid3tag.c:
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Add read support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
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Fixes #414496.
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2007-03-09 16:46:35 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/libvisual/visual.c: Improve adapter usage and comments.
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Original commit message from CVS:
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* ext/libvisual/visual.c: (gst_visual_sink_setcaps),
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(gst_vis_src_negotiate), (get_buffer), (gst_visual_chain):
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Improve adapter usage and comments.
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2007-03-09 16:38:06 +0000 Wim Taymans <wim.taymans@gmail.com>
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Use new metadata copy function.
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Original commit message from CVS:
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* ext/pango/gsttextrender.c: (gst_text_render_chain):
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* ext/vorbis/vorbistag.c: (gst_vorbis_tag_parse_packet):
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* gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_copy):
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Use new metadata copy function.
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* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
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(gst_ffmpegcsp_transform):
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* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform):
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Basetransform copied the metadata for us.
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2007-03-09 16:28:04 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/pango/gsttextoverlay.c: Some more logging. Only accept newsegment events in TIME format and send a WARNING messag...
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Original commit message from CVS:
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* ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
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(gst_text_overlay_video_event):
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Some more logging. Only accept newsegment events in TIME format and
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send a WARNING message if they are not in TIME format.
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* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
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(gst_sub_parse_init), (gst_sub_parse_src_event), (handle_buffer),
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(gst_sub_parse_chain), (gst_sub_parse_sink_event):
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* gst/subparse/gstsubparse.h:
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No need to allocate GstSegment structure dynamically, just put it
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into the instance structure; ignore newsegment events in BYTE
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format and in particular don't let it overwrite our saved TIME
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segment from the last seek.
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2007-03-09 13:05:04 +0000 Michael Smith <msmith@xiph.org>
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gst/typefind/gsttypefindfunctions.c: Replace AC3 typefinder with one that isn't terrible, and actually works usefully.
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Original commit message from CVS:
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* gst/typefind/gsttypefindfunctions.c: (ac3_type_find):
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Replace AC3 typefinder with one that isn't terrible, and actually
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works usefully.
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2007-03-09 12:22:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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gst/audioconvert/gstaudioconvert.c: fix error category and translatable string
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Original commit message from CVS:
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* gst/audioconvert/gstaudioconvert.c:
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(gst_audio_convert_transform):
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fix error category and translatable string
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2007-03-09 11:23:32 +0000 Tim-Philipp Müller <tim@centricular.net>
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pkgconfig/: Fix up utils => pbutils here too.
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Original commit message from CVS:
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* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
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* pkgconfig/gstreamer-plugins-base.pc.in:
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Fix up utils => pbutils here too.
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2007-03-09 10:49:53 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/subparse/gstsubparse.c: Break out of loop in chain function as soon as possible if we get a non-OK flow return.
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Original commit message from CVS:
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* gst/subparse/gstsubparse.c: (handle_buffer):
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Break out of loop in chain function as soon as possible if we get
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a non-OK flow return.
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2007-03-08 18:26:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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tests/check/elements/alsa.c: Unref the mixer if the state change fails too (if the alsa devices are inaccessible, for...
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Original commit message from CVS:
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* tests/check/elements/alsa.c: (GST_START_TEST):
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Unref the mixer if the state change fails too (if the
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alsa devices are inaccessible, for example)
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2007-03-08 17:49:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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tests/check/Makefile.am: Don't test libvisual elements in the states check, because libvisual seems to leak internally.
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Original commit message from CVS:
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* tests/check/Makefile.am:
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Don't test libvisual elements in the states check, because libvisual
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seems to leak internally.
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Re-enable the alsa and states tests now that there's new suppressions
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in gst.supp.
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* tests/check/elements/alsa.c: (GST_START_TEST):
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Don't leak the alsamixer we instantiated.
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2007-03-08 15:22:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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sys/: Move some cleanup stuff from the state change handler into a _reset() function that can be called from _finaliz...
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Original commit message from CVS:
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* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_clear),
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(gst_ximagesink_change_state), (gst_ximagesink_reset),
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(gst_ximagesink_finalize):
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state),
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(gst_xvimagesink_reset), (gst_xvimagesink_finalize):
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Move some cleanup stuff from the state change handler into a _reset()
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function that can be called from _finalize(). This ensures that things
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get freed even if (for some reason) the NULL->READY state transition
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fails in the parent class.
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Even if a parent state change fails, process our downward state change
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logic instead of bailing out early.
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Free the correct xcontext pointer in ximagesink's xcontext_clear.
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2007-03-08 12:53:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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ext/alsa/gstalsasink.c: Extra log line.
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Original commit message from CVS:
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* ext/alsa/gstalsasink.c: (gst_alsasink_open):
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Extra log line.
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* ext/pango/gstclockoverlay.c: (gst_clock_overlay_init):
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* ext/pango/gsttimeoverlay.c: (gst_time_overlay_init):
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Use pango_font_description_set_family_static instead of
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pango_font_description_set_family to save a string copy (it was
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leaking due to the strdup anyway)
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* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_finalize):
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* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_finalize):
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* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_finalize):
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* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_finalize):
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Chain up in finalize.
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2007-03-07 18:50:10 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/interfaces/mixertrack.c: API: add "untranslated-label" property which should be set by implementations a...
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Original commit message from CVS:
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* gst-libs/gst/interfaces/mixertrack.c:
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(gst_mixer_track_class_init), (gst_mixer_track_get_property),
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(gst_mixer_track_set_property):
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API: add "untranslated-label" property which should be set by
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implementations at construct time (#414645).
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* ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_new):
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* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
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Set "untranslated-label" when constructing mixer track objects.
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* tests/check/elements/alsa.c: (GST_START_TEST), (alsa_suite):
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Unit test to check the above.
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2007-03-07 17:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/ogg/gstoggdemux.c: Fix confusing debug message.
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Original commit message from CVS:
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* ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
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Fix confusing debug message.
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2007-03-07 17:12:54 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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gst-plugins-base.doap: update doap file with new version
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Original commit message from CVS:
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* gst-plugins-base.doap:
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update doap file with new version
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2007-03-07 17:05:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* gst/tcp/gstmultifdsink.c:
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update docs
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Original commit message from CVS:
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update docs
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2007-03-07 16:56:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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configure.ac: Back to CVS
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Original commit message from CVS:
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* configure.ac:
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Back to CVS
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=== release 0.10.12 ===
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2007-03-07 16:46:51 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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* ChangeLog:
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* NEWS:
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* RELEASE:
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* configure.ac:
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* docs/plugins/gst-plugins-base-plugins.args:
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* docs/plugins/inspect/plugin-adder.xml:
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* docs/plugins/inspect/plugin-alsa.xml:
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* docs/plugins/inspect/plugin-audioconvert.xml:
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* docs/plugins/inspect/plugin-audiorate.xml:
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* docs/plugins/inspect/plugin-audioresample.xml:
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* docs/plugins/inspect/plugin-audiotestsrc.xml:
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* docs/plugins/inspect/plugin-cdparanoia.xml:
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* docs/plugins/inspect/plugin-decodebin.xml:
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* docs/plugins/inspect/plugin-decodebin2.xml:
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* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
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* docs/plugins/inspect/plugin-gdp.xml:
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* docs/plugins/inspect/plugin-gnomevfs.xml:
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* docs/plugins/inspect/plugin-libvisual.xml:
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* docs/plugins/inspect/plugin-ogg.xml:
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* docs/plugins/inspect/plugin-pango.xml:
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* docs/plugins/inspect/plugin-playbin.xml:
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* docs/plugins/inspect/plugin-subparse.xml:
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* docs/plugins/inspect/plugin-tcp.xml:
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* docs/plugins/inspect/plugin-theora.xml:
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* docs/plugins/inspect/plugin-typefindfunctions.xml:
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* docs/plugins/inspect/plugin-video4linux.xml:
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* docs/plugins/inspect/plugin-videorate.xml:
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* docs/plugins/inspect/plugin-videoscale.xml:
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* docs/plugins/inspect/plugin-videotestsrc.xml:
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* docs/plugins/inspect/plugin-volume.xml:
|
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* docs/plugins/inspect/plugin-vorbis.xml:
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* docs/plugins/inspect/plugin-ximagesink.xml:
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* docs/plugins/inspect/plugin-xvimagesink.xml:
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* win32/common/config.h:
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Release 0.10.12
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|
Original commit message from CVS:
|
|
Release 0.10.12
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|
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2007-03-07 15:35:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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* common:
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* po/af.po:
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* po/az.po:
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* po/cs.po:
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* po/de.po:
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* po/en_GB.po:
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* po/hu.po:
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* po/it.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/sq.po:
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* po/sr.po:
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* po/sv.po:
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* po/uk.po:
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* po/vi.po:
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Update .po files
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|
Original commit message from CVS:
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Update .po files
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2007-03-06 12:31:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
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configure.ac: Bump version to 0.10.11.4 pre-release
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Original commit message from CVS:
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* configure.ac:
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Bump version to 0.10.11.4 pre-release
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2007-03-06 12:10:08 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstbaseaudiosink.c: Fix regression that made GStreamer skip the first samples of audio.
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|
Original commit message from CVS:
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|
* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_async_play):
|
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Fix regression that made GStreamer skip the first samples of audio.
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Fixes #414684.
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2007-03-05 11:21:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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configure.ac: Bump version to 0.10.11.3 pre-release
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|
Original commit message from CVS:
|
|
* configure.ac:
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Bump version to 0.10.11.3 pre-release
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2007-03-05 09:35:29 +0000 Sebastian Dröge <slomo@circular-chaos.org>
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po/POTFILES.in: Update paths for the rename from utils to pbutils to fix the build.
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|
Original commit message from CVS:
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* po/POTFILES.in:
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Update paths for the rename from utils to pbutils to fix the build.
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2007-03-05 09:27:55 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/pbutils/Makefile.am: Change directory to install headers in from gst/utils to gst/pbutils as well.
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|
Original commit message from CVS:
|
|
* gst-libs/gst/pbutils/Makefile.am:
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Change directory to install headers in from gst/utils to gst/pbutils
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as well.
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2007-03-04 23:41:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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|
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* tests/check/libs/.gitignore:
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|
moap ignore
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|
Original commit message from CVS:
|
|
moap ignore
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|
|
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2007-03-04 23:41:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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|
|
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* win32/common/config.h:
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* win32/common/libgstutils.def:
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update defs
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|
Original commit message from CVS:
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|
update defs
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2007-03-04 23:39:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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|
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rename utils to pbutils
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|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* docs/libs/gst-plugins-base-libs-docs.sgml:
|
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* docs/libs/gst-plugins-base-libs-sections.txt:
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* gst-libs/gst/Makefile.am:
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* gst-libs/gst/interfaces/mixer.c:
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|
* gst-libs/gst/pbutils/Makefile.am:
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* gst-libs/gst/pbutils/descriptions.c:
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(gst_pb_utils_get_source_description),
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(gst_pb_utils_get_sink_description),
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(gst_pb_utils_get_decoder_description),
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(gst_pb_utils_get_encoder_description),
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(gst_pb_utils_get_element_description),
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(gst_pb_utils_add_codec_description_to_tag_list),
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(gst_pb_utils_get_codec_description), (gst_pb_utils_list_all):
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* gst-libs/gst/pbutils/descriptions.h:
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* gst-libs/gst/pbutils/install-plugins.c:
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|
* gst-libs/gst/pbutils/install-plugins.h:
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|
* gst-libs/gst/pbutils/missing-plugins.c:
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(gst_missing_uri_source_message_new),
|
|
(gst_missing_uri_sink_message_new),
|
|
(gst_missing_element_message_new),
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(gst_missing_decoder_message_new),
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(gst_missing_encoder_message_new),
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(gst_missing_plugin_message_get_description):
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* gst-libs/gst/pbutils/missing-plugins.h:
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|
* gst-libs/gst/pbutils/pbutils.c: (gst_pb_utils_init):
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|
* gst-libs/gst/pbutils/pbutils.h:
|
|
* gst-libs/gst/utils/Makefile.am:
|
|
* gst-libs/gst/utils/base-utils.c:
|
|
* gst-libs/gst/utils/base-utils.h:
|
|
* gst-libs/gst/utils/descriptions.c:
|
|
* gst-libs/gst/utils/descriptions.h:
|
|
* gst-libs/gst/utils/install-plugins.c:
|
|
* gst-libs/gst/utils/install-plugins.h:
|
|
* gst-libs/gst/utils/missing-plugins.c:
|
|
* gst-libs/gst/utils/missing-plugins.h:
|
|
* gst-plugins-base.spec.in:
|
|
* gst/playback/Makefile.am:
|
|
* gst/playback/gstdecodebin.c:
|
|
* gst/playback/gstdecodebin2.c:
|
|
* gst/playback/gstplaybasebin.c: (setup_subtitle),
|
|
(gen_source_element):
|
|
* gst/playback/gstplaybin.c: (plugin_init):
|
|
* tests/check/Makefile.am:
|
|
* tests/check/libs/pbutils.c: (GST_START_TEST),
|
|
(test_pb_utils_install_plugins_do_callout), (libgstpbutils_suite):
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* tests/check/libs/utils.c:
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rename utils to pbutils
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2007-03-03 10:23:03 +0000 David Schleef <ds@schleef.org>
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gst-libs/gst/app/Makefile.am: Install the headers.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/app/Makefile.am:
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Install the headers.
|
|
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|
2007-03-03 10:10:30 +0000 David Schleef <ds@schleef.org>
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|
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gst-libs/gst/app/: Add GstAppBuffer that includes a callback and closure for proper handling of data chunks.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/app/Makefile.am:
|
|
* gst-libs/gst/app/gstappbuffer.c:
|
|
* gst-libs/gst/app/gstappbuffer.h:
|
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* gst-libs/gst/app/gstappsrc.c:
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Add GstAppBuffer that includes a callback and closure for
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proper handling of data chunks.
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2007-03-03 09:06:06 +0000 David Schleef <ds@schleef.org>
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gst-libs/gst/app/gstappsrc.*: Hacking to address issues in 413418.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/app/gstappsrc.c:
|
|
* gst-libs/gst/app/gstappsrc.h:
|
|
Hacking to address issues in 413418.
|
|
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2007-03-03 08:16:57 +0000 David Schleef <ds@schleef.org>
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|
|
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Move the app library to gst-libs/gst/app (duh!)
|
|
Original commit message from CVS:
|
|
* Makefile.am:
|
|
* configure.ac:
|
|
* ext/Makefile.am:
|
|
* gst-libs/gst/Makefile.am:
|
|
* gst-libs/gst/app/Makefile.am:
|
|
* gst-libs/gst/app/gstapp.c:
|
|
* gst-libs/gst/app/gstappsrc.c:
|
|
* gst-libs/gst/app/gstappsrc.h:
|
|
* gst/app/Makefile.am:
|
|
* gst/app/gstapp.c:
|
|
* gst/app/gstappsrc.c:
|
|
* gst/app/gstappsrc.h:
|
|
Move the app library to gst-libs/gst/app (duh!)
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|
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2007-03-02 12:59:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
Add documentation for decodebin2 that indicates that the API is still unstable.
|
|
Original commit message from CVS:
|
|
* docs/plugins/Makefile.am:
|
|
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
|
* docs/plugins/inspect/plugin-decodebin2.xml:
|
|
* gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init):
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Add documentation for decodebin2 that indicates that the API
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|
is still unstable.
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2007-03-01 18:50:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
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configure.ac: Update to 0.10.11.2 (0.10.12 pre-release)
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Update to 0.10.11.2 (0.10.12 pre-release)
|
|
|
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2007-03-01 17:29:55 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstbaseaudiosink.c: base time is irrelevant here.
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Original commit message from CVS:
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_async_play):
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base time is irrelevant here.
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2007-03-01 17:01:43 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/: Improve debugging.
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Original commit message from CVS:
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* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
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* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
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Improve debugging.
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_query), (gst_base_audio_sink_event),
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(gst_base_audio_sink_render), (gst_base_audio_sink_async_play):
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Improve latency and clock slaving calculations.
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Improve slave clock calibration.
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* gst-libs/gst/audio/gstringbuffer.c:
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(gst_ring_buffer_commit_full):
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When we are asked to render N sample to 0 bytes, return N.
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2007-03-01 16:48:45 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/alsa/gstalsasink.*: Remove unused dispose function.
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Original commit message from CVS:
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* ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
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(gst_alsasink_write), (gst_alsasink_reset):
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* ext/alsa/gstalsasink.h:
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Remove unused dispose function.
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Rename lock to not interfere with alsasrc lock.
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* ext/alsa/gstalsasrc.c: (gst_alsasrc_finalize),
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(gst_alsasrc_class_init), (gst_alsasrc_init), (set_swparams),
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(gst_alsasrc_read), (gst_alsasrc_reset):
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* ext/alsa/gstalsasrc.h:
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Implement finalize function.
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Use lock to protect alsa access.
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Implement _reset.
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Fine tune sw params.
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2007-03-01 10:20:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* common:
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* configure.ac:
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typo
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Original commit message from CVS:
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typo
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2007-02-28 19:27:28 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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configure.ac: Convert to new AG_GST style.
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Original commit message from CVS:
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* configure.ac:
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Convert to new AG_GST style.
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2007-02-28 15:17:20 +0000 Ed Catmur <ed@catmur.co.uk>
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gst/playback/gstplaybin.c: Fix race condition when rapidly switching visualisations in playbin.
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Original commit message from CVS:
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Patch by: Ed Catmur <ed at catmur dot co dot uk>
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* gst/playback/gstplaybin.c: (gst_play_bin_vis_unblocked),
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(gst_play_bin_vis_blocked), (gst_play_bin_set_property):
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Fix race condition when rapidly switching visualisations in playbin.
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Fixes #401029.
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2007-02-28 15:11:59 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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tests/check/Makefile.am: Include local stuff before system installed things in LDFLAGS and
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Original commit message from CVS:
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* tests/check/Makefile.am:
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Include local stuff before system installed things in LDFLAGS and
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CFLAGS.
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2007-02-28 15:10:06 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/ogg/gstoggdemux.c: Improve debugging.
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Original commit message from CVS:
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* ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_activate):
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Improve debugging.
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2007-02-28 15:05:03 +0000 Wim Taymans <wim.taymans@gmail.com>
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sys/v4l/: Fix duration and timestamping, taking latency into account.
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Original commit message from CVS:
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* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init), (gst_v4lsrc_init),
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(gst_v4lsrc_fixate), (gst_v4lsrc_query):
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* sys/v4l/v4lsrc_calls.c: (gst_v4lsrc_buffer_new):
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Fix duration and timestamping, taking latency into account.
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Implement latency query.
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2007-02-28 15:02:25 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstaudioclock.c: Fix clock name.
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Original commit message from CVS:
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* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
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(gst_audio_clock_new):
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Fix clock name.
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_init), (gst_base_audio_sink_query):
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* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
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(gst_base_audio_src_query), (gst_base_audio_src_get_offset),
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(gst_base_audio_src_create):
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Improve latency query code.
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Use proper clock names.
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2007-02-28 12:57:46 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* tests/check/generic/states.c:
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plug test leak
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Original commit message from CVS:
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plug test leak
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2007-02-28 12:44:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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tests/check/generic/states.c: Copy the states.c test from core again
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Original commit message from CVS:
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* tests/check/generic/states.c: (GST_START_TEST):
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Copy the states.c test from core again
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* tests/check/Makefile.am:
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ignore cdio and cdparanoiasrc
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2007-02-28 12:08:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst/audioconvert/audioconvert.c: Also make valgrind happy and avoid copying data in some cases.
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Original commit message from CVS:
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* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
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(double_hq), (audio_convert_get_func_index), (check_default),
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(audio_convert_prepare_context), (audio_convert_convert):
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Also make valgrind happy and avoid copying data in some cases.
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2007-02-28 11:58:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* tests/check/generic/states.c:
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use a macro
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Original commit message from CVS:
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use a macro
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2007-02-28 11:47:45 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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Don't run inplace if that overwrites source data as we go. Add more tests. Fixes #339837 even more.
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Original commit message from CVS:
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* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
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(double_hq), (audio_convert_get_func_index),
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(audio_convert_prepare_context), (audio_convert_convert):
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* gst/audioconvert/gstaudioconvert.c:
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(gst_audio_convert_class_init), (gst_audio_convert_get_unit_size),
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(gst_audio_convert_transform_caps):
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* tests/check/elements/audioconvert.c: (GST_START_TEST),
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(audioconvert_suite):
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Don't run inplace if that overwrites source data as we go. Add more
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tests. Fixes #339837 even more.
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2007-02-27 18:45:37 +0000 Julien Moutte <julien@moutte.net>
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tests/examples/seek/seek.c: Fix various seeking bugs (Slider was not updating when doing a non flushing seek, Reverse...
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Original commit message from CVS:
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2007-02-27 Julien MOUTTE <julien@moutte.net>
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* tests/examples/seek/seek.c: (do_seek), (set_update_scale),
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(msg_segment_done): Fix various seeking bugs (Slider was not
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updating when doing a non flushing seek, Reverse playback
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on segment seek was wrong).
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2007-02-26 21:01:03 +0000 David Schleef <ds@schleef.org>
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Add a new plugin/library to make it easy for apps to shove data into a pipeline.
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Original commit message from CVS:
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* configure.ac:
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* gst/app/Makefile.am:
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* gst/app/gstapp.c:
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* gst/app/gstappsrc.c:
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* gst/app/gstappsrc.h:
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Add a new plugin/library to make it easy for apps to shove
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data into a pipeline.
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2007-02-26 11:48:49 +0000 Wim Taymans <wim.taymans@gmail.com>
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tests/examples/seek/seek.c: When we stop scrubbing, don't leave the pipeline PLAYING when we requested a PAUSED state.
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Original commit message from CVS:
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* tests/examples/seek/seek.c: (stop_seek):
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When we stop scrubbing, don't leave the pipeline PLAYING when we
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requested a PAUSED state.
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2007-02-25 23:51:03 +0000 René Stadler <mail@renestadler.de>
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gst-libs/gst/tag/gstvorbistag.c: Parse date strings in vorbis comments that have an invalid (zero) month or day (#410...
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Original commit message from CVS:
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Patch by: René Stadler <mail at renestadler de>
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* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
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Parse date strings in vorbis comments that have an invalid (zero)
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month or day (#410396).
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* tests/check/libs/tag.c: (GST_START_TEST):
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Test case for the above.
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2007-02-24 20:12:49 +0000 Loïc Minier <lool+gnome@via.ecp.fr>
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Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
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Original commit message from CVS:
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Patch by: Loïc Minier <lool+gnome at via ecp fr>
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* configure.ac:
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* ext/alsa/Makefile.am:
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* gst/audiotestsrc/Makefile.am:
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Fix compilation with LDFLAGS='-Wl,-z,defs' (#410963).
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2007-02-23 18:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gstplaybin.c: Improve docs: point out that the application needs to assist playbin with buffering.
|
|
Original commit message from CVS:
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* gst/playback/gstplaybin.c:
|
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Improve docs: point out that the application needs to assist playbin
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|
with buffering.
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2007-02-23 13:10:50 +0000 Tim-Philipp Müller <tim@centricular.net>
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Change GStreamer marker prefix in detail string from 'gstreamer.net' to just 'gstreamer'. Document the caps string co...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/utils/install-plugins.c:
|
|
* gst-libs/gst/utils/missing-plugins.c:
|
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* tests/check/libs/utils.c: (missing_msg_check_getters):
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Change GStreamer marker prefix in detail string from 'gstreamer.net'
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to just 'gstreamer'. Document the caps string component of the
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decoder/encoder detail a bit better, since not everyone will be
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familiar with the GStreamer media type/caps system (but they better
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enjoy nested itemized lists).
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2007-02-22 12:57:47 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/netbuffer/gstnetbuffer.c: Fix copying of GstNetBuffer (would crash before, or at least lead to invalid m...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/netbuffer/gstnetbuffer.c:
|
|
(notgst_buffer_copy_fields_in_place), (gst_netbuffer_copy):
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Fix copying of GstNetBuffer (would crash before, or at least lead to
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invalid memory access, #410772), for now by copying the GstBuffer copy
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code from the core over here so we can copy the GstBuffer fields on a
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provided buffer instance (of type GstNetBuffer in this case). Would be
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better to fix this with some support by the core though (and in the long
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run change the broken GstBuffer/GstMiniObject copy semantics, #393099).
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* tests/check/Makefile.am:
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Enable unit test for GstNetBuffer.
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2007-02-22 11:04:10 +0000 Andy Wingo <wingo@pobox.com>
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* ChangeLog:
|
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* gst-libs/gst/audio/gstbaseaudiosink.c:
|
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gst-libs/gst/audio/gstbaseaudiosink.c
|
|
Original commit message from CVS:
|
|
2007-02-22 Andy Wingo <wingo@pobox.com>
|
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* gst-libs/gst/audio/gstbaseaudiosink.c
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(gst_base_audio_sink_init): Disable pull-mode activation until we
|
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figure out how to make audio sinks go to PLAYING.
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2007-02-22 09:04:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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Add float as an intermediate format, as well as float mixing. Enable test that was failing before. Fixes #339837
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/audioconvert.c: (float), (double), (float_hq),
|
|
(double_hq), (audio_convert_get_func_index),
|
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(audio_convert_prepare_context), (audio_convert_convert):
|
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* gst/audioconvert/audioconvert.h:
|
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* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_setup_matrix),
|
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(gst_channel_mix_mix_int), (gst_channel_mix_mix_float):
|
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* gst/audioconvert/gstchannelmix.h:
|
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* tests/check/elements/audioconvert.c: (GST_START_TEST):
|
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Add float as an intermediate format, as well as float mixing. Enable
|
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test that was failing before. Fixes #339837
|
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2007-02-21 16:12:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
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tests/examples/seek/seek.c: Undo the previous commit: -1 as a stop time implies that the stop time is the end of file...
|
|
Original commit message from CVS:
|
|
* tests/examples/seek/seek.c: (do_seek):
|
|
Undo the previous commit: -1 as a stop time implies that the stop
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|
time is the end of file, clearing any previously configured segment.
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|
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2007-02-21 15:36:26 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
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tests/examples/seek/seek.c: Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
|
|
Original commit message from CVS:
|
|
* tests/examples/seek/seek.c: (do_seek):
|
|
Don't SEEK_SET with a stop time of -1, use SEEK_NONE instead.
|
|
|
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2007-02-21 13:55:54 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/volume/gstvolume.c: Unbreak volume, value remains gint.
|
|
Original commit message from CVS:
|
|
* gst/volume/gstvolume.c: (volume_process_int16),
|
|
(volume_process_int16_clamp), (volume_set_caps):
|
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Unbreak volume, value remains gint.
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2007-02-21 13:08:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
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gst/volume/gstvolume.*: Extend float audio support (double) and some int->uint cleanups.
|
|
Original commit message from CVS:
|
|
* gst/volume/gstvolume.c: (volume_choose_func),
|
|
(volume_update_real_volume), (gst_volume_set_volume),
|
|
(gst_volume_init), (volume_process_double), (volume_process_float),
|
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(volume_process_int16), (volume_process_int16_clamp),
|
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(volume_set_caps), (volume_transform_ip), (volume_update_volume):
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* gst/volume/gstvolume.h:
|
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Extend float audio support (double) and some int->uint cleanups.
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2007-02-20 15:44:32 +0000 Edward Hervey <bilboed@bilboed.com>
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gst/playback/gstdecodebin2.c: Don't free groups from the streaming threads. Just put them aside and free them in disp...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose),
|
|
(multi_queue_underrun_cb), (gst_decode_group_check_if_drained),
|
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(sort_end_pads), (gst_decode_group_expose),
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(gst_decode_group_hide):
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Don't free groups from the streaming threads. Just put them aside and
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free them in dispose.
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2007-02-20 11:20:52 +0000 Edward Hervey <bilboed@bilboed.com>
|
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gst/playback/gstdecodebin2.c: Handle dynamic pads within groups.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin2.c: (connect_element),
|
|
(pad_added_group_cb), (gst_decode_group_check_if_blocked),
|
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(sort_end_pads), (gst_decode_group_expose):
|
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Handle dynamic pads within groups.
|
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Sort pads before exposing them in order to make playbin happy.
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There still is a race with the multiqueue filling up. This should be
|
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solved separately.
|
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Fixes #398721
|
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2007-02-18 21:02:36 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/utils/: Some more docs (and descriptions for two subtitle formats).
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/utils/base-utils.c:
|
|
* gst-libs/gst/utils/descriptions.c:
|
|
* gst-libs/gst/utils/install-plugins.c:
|
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* gst-libs/gst/utils/missing-plugins.c:
|
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Some more docs (and descriptions for two subtitle formats).
|
|
|
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2007-02-16 10:19:45 +0000 Tim-Philipp Müller <tim@centricular.net>
|
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|
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gst-libs/gst/audio/audio.c: Fix documentation.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/audio.c:
|
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Fix documentation.
|
|
|
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2007-02-16 10:15:46 +0000 Yves Lefebvre <ivanohe@abacom.com>
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gst/videorate/gstvideorate.c: Don't leak caps. Fixes #408278.
|
|
Original commit message from CVS:
|
|
Patch by: Yves Lefebvre <ivanohe abacom com>
|
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* gst/videorate/gstvideorate.c: (gst_video_rate_setcaps):
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Don't leak caps. Fixes #408278.
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2007-02-15 15:17:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
|
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More docs coverage and some ChangeLog surgery (add missing names)
|
|
Original commit message from CVS:
|
|
* ext/cdparanoia/gstcdparanoiasrc.h:
|
|
* ext/ogg/gstoggdemux.h:
|
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* gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
|
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(gst_audio_frame_length), (gst_audio_duration_from_pad_buffer),
|
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(gst_audio_is_buffer_framed), (gst_audio_structure_set_int):
|
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* gst-libs/gst/audio/audio.h:
|
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* gst-libs/gst/audio/gstaudiofilter.h:
|
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* gst-libs/gst/interfaces/videoorientation.h:
|
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* gst/adder/gstadder.h:
|
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More docs coverage and some ChangeLog surgery (add missing names)
|
|
|
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2007-02-15 12:07:57 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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sys/: Small constifications.
|
|
Original commit message from CVS:
|
|
* sys/ximage/ximagesink.c:
|
|
(gst_ximagesink_calculate_pixel_aspect_ratio):
|
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* sys/xvimage/xvimagesink.c:
|
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(gst_xvimagesink_calculate_pixel_aspect_ratio):
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Small constifications.
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2007-02-15 12:06:25 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst-libs/gst/audio/gstbaseaudiosink.c: Answer latency query.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_class_init), (gst_base_audio_sink_query),
|
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(gst_base_audio_sink_render), (gst_base_audio_sink_callback),
|
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(gst_base_audio_sink_async_play),
|
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(gst_base_audio_sink_change_state):
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Answer latency query.
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Use configured latency when syncing.
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Fix clock slaving.
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* gst-libs/gst/audio/gstbaseaudiosrc.c:
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(gst_base_audio_src_class_init), (gst_base_audio_src_dispose),
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(gst_base_audio_src_query), (gst_base_audio_src_change_state):
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Fix possible memleak.
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Implement latency query.
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Small cleanups.
|
|
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2007-02-15 11:59:41 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/alsa/gstalsasink.c: Ignore errors in reset, these are not fatal. They also grab the element lock which is already...
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsasink.c: (gst_alsasink_reset):
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Ignore errors in reset, these are not fatal. They also grab the element
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lock which is already taking when this function is called. Fixes
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#405451.
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2007-02-13 13:50:56 +0000 Christian Schaller <uraeus@gnome.org>
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* gst-plugins-base.spec.in:
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add header file for easy codec install
|
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Original commit message from CVS:
|
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add header file for easy codec install
|
|
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2007-02-13 10:24:13 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
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configure.ac: Remove 'tests/examples/xerror/Makefile' from output files again.
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Original commit message from CVS:
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* configure.ac:
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Remove 'tests/examples/xerror/Makefile' from output files again.
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2007-02-13 09:12:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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Also crossref against gst-plugins-base-libs.
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Original commit message from CVS:
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* configure.ac:
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* docs/plugins/Makefile.am:
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Also crossref against gst-plugins-base-libs.
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|
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2007-02-12 20:42:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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Add crossreferences to glib/gobject/gstream docs.
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Original commit message from CVS:
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* configure.ac:
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* docs/libs/Makefile.am:
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* docs/plugins/Makefile.am:
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Add crossreferences to glib/gobject/gstream docs.
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* gst-libs/gst/audio/audio.h:
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Source formatting.
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* gst/audiotestsrc/gstaudiotestsrc.c: (plugin_init):
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Add own debug category.
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2007-02-12 11:01:04 +0000 René Stadler <mail@renestadler.de>
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gst-libs/gst/tag/gstvorbistag.c: Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL (#403597).
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Original commit message from CVS:
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Patch by: René Stadler <mail at renestadler de>
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* gst-libs/gst/tag/gstvorbistag.c:
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Add vorbis/FLAC-tag mapping for new GST_TAG_REFERENCE_LEVEL
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(#403597).
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2007-02-12 10:33:40 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gstplaybasebin.c: When we have external subtitles and wait for the subtitle decodebin to get up and runn...
|
|
Original commit message from CVS:
|
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* gst/playback/gstplaybasebin.c: (setup_source):
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When we have external subtitles and wait for the subtitle decodebin
|
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to get up and running, we set up a (sync) bus handler for the
|
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subtitle decodebin, so we can stop waiting when it posts an error
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message. However, we should do that before we set the subtitle
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decodebin's state to playing, otherwise things are racy and we might
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miss error messages posted before we had a chance to set up the bus.
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This should finally fix totem hanging on .txt pseudo-subtitle files.
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2007-02-10 19:27:48 +0000 Sébastien Moutte <sebastien@moutte.net>
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gst-libs/gst/rtp/gstbasertpaudiopayload.c: Use gst_gdouble_to_guint64 for conversions.
|
|
Original commit message from CVS:
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* gst-libs/gst/rtp/gstbasertpaudiopayload.c:(gst_base_rtp_audio_payload_handle_frame_based_buffer):
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Use gst_gdouble_to_guint64 for conversions.
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* win32/common/config.h.in:
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Add a define for GST_INSTALL_PLUGINS_HELPER
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* win32/common/libgstaudio.def:
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* win32/common/libgstcdda.def:
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* win32/common/libgstnetbuffer.def:
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* win32/common/libgstrtp.def:
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* win32/common/libgutils.def:
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Add new exported functions.
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* win32/vs6/gst_plugins_base.dsw:
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* win32/vs6/libgstdecodebin.dsp:
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* win32/vs6/libgstnetbuffer.dsp:
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* win32/vs6/libgstplaybin.dsp:
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* win32/vs6/libgstrtp.dsp:
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* win32/vs6/libgstvorbis.dsp:
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* win32/vs6/libgstcdda.dsp:
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* win32/vs6/libgstgdp.dsp:
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* win32/vs6/libgstutils.dsp:
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Update and add new project files.
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2007-02-10 18:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/subparse/gstsubparse.c: For SubRip (.srt) subtitles, ignore all markup tags we don't handle (like font tags, for ...
|
|
Original commit message from CVS:
|
|
* gst/subparse/gstsubparse.c: (subrip_remove_unhandled_tag),
|
|
(subrip_remove_unhandled_tags), (parse_subrip):
|
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For SubRip (.srt) subtitles, ignore all markup tags we don't
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handle (like font tags, for example).
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* tests/check/elements/subparse.c:
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Add test for this.
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2007-02-09 13:28:01 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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* ChangeLog:
|
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ChangeLog surgery
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|
Original commit message from CVS:
|
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ChangeLog surgery
|
|
|
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2007-02-09 13:16:27 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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gst/playback/: Don't error out if there is no fakesink in the READY to NULL state change, since when decodebin is re-...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (add_fakesink),
|
|
(gst_decode_bin_change_state):
|
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* gst/playback/gstdecodebin2.c: (add_fakesink),
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(gst_decode_bin_change_state):
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Don't error out if there is no fakesink in the READY to NULL state
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change, since when decodebin is re-used, we're only adding the
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fakesink element in READY to PAUSED.
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* tests/check/elements/decodebin.c:
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(new_decoded_pad_plug_fakesink_cb), (GST_START_TEST),
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(decodebin_suite):
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Minimal unit test to make sure we can use the same decodebin
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instance twice (at least with audiotestsrc input).
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2007-02-09 09:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/alsa/gstalsa.c: Try to get devic-name from device string first, and from handle only as fallback (seems to yield ...
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsa.c: (gst_alsa_find_device_name):
|
|
Try to get devic-name from device string first, and from handle only
|
|
as fallback (seems to yield better results and is more robust
|
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against buggy probing code on the application side).
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2007-02-08 15:43:26 +0000 Julien Puydt <julien.puydt@laposte.net>
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ext/alsa/: Improve device-name detection a bit, especially in the case where the device is not actually open (#405020...
|
|
Original commit message from CVS:
|
|
Based on patch by: Julien Puydt <julien.puydt at laposte net>
|
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* ext/alsa/gstalsa.c: (gst_alsa_find_device_name_no_handle),
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(gst_alsa_find_device_name):
|
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* ext/alsa/gstalsa.h:
|
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* ext/alsa/gstalsasink.c: (gst_alsasink_get_property):
|
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* ext/alsa/gstalsasrc.c: (gst_alsasrc_get_property):
|
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Improve device-name detection a bit, especially in the case where
|
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the device is not actually open (#405020, #405024). Move common code
|
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into gstalsa.c instead of duplicating it.
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2007-02-07 13:05:01 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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gst/audioconvert/gstaudioconvert.c: Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/gstaudioconvert.c:
|
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Fix up docs chunk so that gtk-doc doesn't complain, and fix typo.
|
|
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2007-02-06 17:47:32 +0000 Julien Moutte <julien@moutte.net>
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sys/xvimage/xvimagesink.*: Implement PropertyProbe Interface for XVAdaptors so that one can choose the adaptor to use...
|
|
Original commit message from CVS:
|
|
2007-02-06 Julien MOUTTE <julien@moutte.net>
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents),
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(gst_xvimagesink_get_xv_support),
|
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(gst_xvimagesink_xcontext_clear),
|
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(gst_xvimagesink_interface_supported),
|
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(gst_xvimagesink_probe_get_properties),
|
|
(gst_xvimagesink_probe_probe_property),
|
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(gst_xvimagesink_probe_needs_probe),
|
|
(gst_xvimagesink_probe_get_values),
|
|
(gst_xvimagesink_property_probe_interface_init),
|
|
(gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
|
|
(gst_xvimagesink_init), (gst_xvimagesink_class_init),
|
|
(gst_xvimagesink_get_type):
|
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* sys/xvimage/xvimagesink.h: Implement PropertyProbe Interface
|
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for XVAdaptors so that one can choose the adaptor to use with
|
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gstreamer-properties.
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|
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2007-02-06 14:00:31 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
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gst/audioconvert/gstaudioconvert.c: Also mention that a conversion from double to float is suboptimal still.
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/gstaudioconvert.c:
|
|
Also mention that a conversion from double to float is suboptimal still.
|
|
|
|
2007-02-06 09:42:05 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/audio/gstaudiofilter.c: Clear our formats structure and free the caps contained in it when shutting down.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstaudiofilter.c:
|
|
(gst_audio_filter_class_init), (gst_audio_filter_change_state):
|
|
Clear our formats structure and free the caps contained in it when
|
|
shutting down.
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|
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2007-02-05 18:39:51 +0000 Andy Wingo <wingo@pobox.com>
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|
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* ChangeLog:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
gst-libs/gst/audio/gstbaseaudiosink.c
|
|
Original commit message from CVS:
|
|
2007-02-05 Andy Wingo <wingo@pobox.com>
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c
|
|
(gst_base_audio_sink_callback): Update basesink->offset so that we
|
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pull monotonically increasing offsets instead of, um, seeking back
|
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to 0 each time. Fixes alsasrc ! alsasink!
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|
|
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2007-02-05 11:44:52 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/videoscale/gstvideoscale.c: A width and height of 1 makes us crash, so increase minimum size to 2x2 pixels until ...
|
|
Original commit message from CVS:
|
|
* gst/videoscale/gstvideoscale.c:
|
|
A width and height of 1 makes us crash, so increase minimum size to
|
|
2x2 pixels until someone feels like fixing this (#404512).
|
|
|
|
2007-02-04 16:23:37 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/pipelines/oggmux.c: Add small test to make sure request pads are cleaned up properly even if oggmux never...
|
|
Original commit message from CVS:
|
|
* tests/check/pipelines/oggmux.c: (GST_START_TEST), (oggmux_suite):
|
|
Add small test to make sure request pads are cleaned up properly
|
|
even if oggmux never changes state out of NULL.
|
|
|
|
2007-02-04 14:11:51 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/libs/utils.c: Fix unit test. Turns out things work much better when you
|
|
Original commit message from CVS:
|
|
* tests/check/libs/utils.c: (GST_START_TEST):
|
|
Fix unit test. Turns out things work much better when you
|
|
NULL-terminate string arrays. Should make p5 build bot happy again.
|
|
|
|
2007-02-03 23:28:45 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/audio/: Oops, forgot to commit fixed-up example.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/Makefile.am:
|
|
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
|
|
(gst_audio_filter_template_base_init),
|
|
(gst_audio_filter_template_class_init),
|
|
(gst_audio_filter_template_init),
|
|
(gst_audio_filter_template_set_property),
|
|
(gst_audio_filter_template_get_property),
|
|
(gst_audio_filter_template_setup),
|
|
(gst_audio_filter_template_filter),
|
|
(gst_audio_filter_template_filter_inplace), (plugin_init):
|
|
Oops, forgot to commit fixed-up example.
|
|
|
|
2007-02-03 20:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
Port GstAudioFilter to 0.10. This change technically breaks but seems justifiable on the grounds that the base class ...
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_get_type),
|
|
(gst_audio_filter_class_init), (gst_audio_filter_init),
|
|
(gst_audio_filter_set_caps),
|
|
(gst_audio_filter_class_add_pad_templates):
|
|
* gst-libs/gst/audio/gstaudiofilter.h:
|
|
Port GstAudioFilter to 0.10. This change technically breaks
|
|
API and ABI (and thus also every library developer's heart),
|
|
but seems justifiable on the grounds that the base class was
|
|
completely unusable before (ie. would crash immediately when
|
|
actually used). Fixes #403963 (and eventually also #403572).
|
|
Also document all of this a bit.
|
|
|
|
2007-02-03 14:26:54 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
Lowering log level to see why things fail on the p5 build bot; fix some typos in unit test messages.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/utils/install-plugins.c:
|
|
(gst_install_plugins_spawn_child):
|
|
* tests/check/libs/utils.c:
|
|
(test_base_utils_install_plugins_do_callout):
|
|
Lowering log level to see why things fail on the p5 build bot;
|
|
fix some typos in unit test messages.
|
|
|
|
2007-02-03 13:59:27 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/libs/utils.c: Don't hard-code temp directory for test helper; use GLib functions to write out file and do...
|
|
Original commit message from CVS:
|
|
* tests/check/libs/utils.c:
|
|
(test_base_utils_install_plugins_do_callout):
|
|
Don't hard-code temp directory for test helper; use GLib functions
|
|
to write out file and do error checking etc.
|
|
|
|
2007-02-02 20:42:08 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/utils/: API: add API for applications to initiate installation of missing plugins, ie. gst_install_plugi...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/utils/Makefile.am:
|
|
* gst-libs/gst/utils/base-utils.h:
|
|
* gst-libs/gst/utils/install-plugins.c:
|
|
(gst_install_plugins_context_set_xid),
|
|
(gst_install_plugins_context_new),
|
|
(gst_install_plugins_context_free),
|
|
(gst_install_plugins_get_helper),
|
|
(gst_install_plugins_spawn_child),
|
|
(gst_install_plugins_return_from_status),
|
|
(gst_install_plugins_installer_exited),
|
|
(gst_install_plugins_async), (gst_install_plugins_sync),
|
|
(gst_install_plugins_return_get_name),
|
|
(gst_install_plugins_installation_in_progress):
|
|
* gst-libs/gst/utils/install-plugins.h:
|
|
API: add API for applications to initiate installation of missing
|
|
plugins, ie. gst_install_plugins_async() primarily.
|
|
Based on libgimme-codec by Ryan Lortie.
|
|
* configure.ac:
|
|
Add --with-install-plugins-helper configure option so distros can specify
|
|
the path of the helper script or program to call when plugin installation
|
|
is requested (distros: please do any argument munging in this helper
|
|
script instead of patching GStreamer to pass arguments differently
|
|
to another program directly).
|
|
* docs/libs/gst-plugins-base-libs-docs.sgml:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
Build and document new API.
|
|
* tests/check/libs/utils.c: (result_cb),
|
|
(test_base_utils_install_plugins_do_callout), (GST_START_TEST),
|
|
(libgstbaseutils_suite):
|
|
Some simple checks for the new API.
|
|
|
|
2007-02-02 14:44:29 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/elements/audioconvert.c: Add small test for 32bit float <=> 64bit float conversion (works only one way so...
|
|
Original commit message from CVS:
|
|
* tests/check/elements/audioconvert.c: (test_float_conversion):
|
|
Add small test for 32bit float <=> 64bit float conversion (works
|
|
only one way so far, 32=>64 produces structured noise).
|
|
|
|
2007-02-02 11:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/audioconvert/gstaudioconvert.c: We don't support floats with a width of 40, 48 or 56 bits.
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/gstaudioconvert.c:
|
|
(set_structure_widths_32_and_64), (make_lossless_changes):
|
|
We don't support floats with a width of 40, 48 or 56 bits.
|
|
|
|
2007-02-02 09:48:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/audioconvert/: Support for 64-bit float audio in audioconvert (#339837)
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/audioconvert.c: (float), (double),
|
|
(audio_convert_get_func_index):
|
|
* gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
|
|
(make_lossless_changes):
|
|
Support for 64-bit float audio in audioconvert (#339837)
|
|
|
|
2007-02-01 18:50:08 +0000 Holger Wansing <linux@wansing-online.de>
|
|
|
|
po/: Add German translation (#352069).
|
|
Original commit message from CVS:
|
|
Patch by: Holger Wansing <linux wansing-online de>
|
|
* po/LINGUAS:
|
|
* po/de.po:
|
|
Add German translation (#352069).
|
|
|
|
2007-02-01 17:52:39 +0000 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
ext/ogg/gstoggmux.c: Use newly added GstCollectPads API to free the allocated resources in the GstOggPad structures (...
|
|
Original commit message from CVS:
|
|
reviewed by: Wim Taymans <wim@fluendo.com>
|
|
* ext/ogg/gstoggmux.c: (gst_ogg_mux_ogg_pad_destroy_notify),
|
|
(gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad):
|
|
Use newly added GstCollectPads API to free the allocated resources in
|
|
the GstOggPad structures (#402393).
|
|
|
|
2007-01-31 15:58:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst/playback/gstplaybin.c: Add audioresample+audioconvert in front of the visualisation element, so that elements lik...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin.c: (gen_vis_element):
|
|
Add audioresample+audioconvert in front of the visualisation
|
|
element, so that elements like libvisual 0.4 that don't support all
|
|
samplerates can work.
|
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Fixes: #402505
|
|
|
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2007-01-30 19:19:37 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/playback/gstplaybasebin.c: Take some locks and make a copy of the streaminfo value array we maintain while holdin...
|
|
Original commit message from CVS:
|
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* gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property),
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(gst_play_base_bin_get_streaminfo_value_array):
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Take some locks and make a copy of the streaminfo value array we
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maintain while holding the lock, so that the application can
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retrieve the stream-info as a value array in a thread-safe way.
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2007-01-30 11:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/audioconvert/gstaudioconvert.c: Don't fail on 0 sized buffers. Fixes #396835.
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Original commit message from CVS:
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* gst/audioconvert/gstaudioconvert.c:
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Don't fail on 0 sized buffers. Fixes #396835.
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2007-01-29 21:13:07 +0000 David Schleef <ds@schleef.org>
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gst/typefind/gsttypefindfunctions.c: Detect BBCD as video/x-dirac, so we can play raw dirac streams.
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Original commit message from CVS:
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* gst/typefind/gsttypefindfunctions.c:
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Detect BBCD as video/x-dirac, so we can play raw dirac
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streams.
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2007-01-29 18:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/theora/theoraenc.c: Check return value of theora_encode_header(), or we might try to allocate a random number of ...
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|
Original commit message from CVS:
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* ext/theora/theoraenc.c: (theora_enc_chain):
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Check return value of theora_encode_header(), or we might try to
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allocate a random number of bytes. theora_encode_header() can fail
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if libtheora has been compiled with encoding support disabled.
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Fixes #398110.
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2007-01-29 10:53:06 +0000 Wim Taymans <wim.taymans@gmail.com>
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tests/check/gst/.cvsignore: Do as buildbot says.
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Original commit message from CVS:
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* tests/check/gst/.cvsignore:
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Do as buildbot says.
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2007-01-29 10:25:11 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/libvisual/visual.c: Fix strides in libvisual. Gst uses X strides.
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Original commit message from CVS:
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* ext/libvisual/visual.c: (gst_visual_src_setcaps):
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Fix strides in libvisual. Gst uses X strides.
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Inspired by: <ed at catmur dot co dot uk> and
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<tim at centricular dot net>
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Fixes #401118.
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2007-01-27 13:32:24 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/ogg/gstoggdemux.*: Properly propagate streaming errors when we are scanning the file for chains so that we don't ...
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Original commit message from CVS:
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* ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
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(gst_ogg_demux_get_data), (gst_ogg_demux_get_next_page),
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(gst_ogg_demux_get_prev_page), (gst_ogg_demux_do_seek),
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(gst_ogg_demux_perform_seek),
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(gst_ogg_demux_bisect_forward_serialno),
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(gst_ogg_demux_read_chain), (gst_ogg_demux_read_end_chain),
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(gst_ogg_demux_find_chains), (gst_ogg_demux_handle_page),
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(gst_ogg_demux_chain), (gst_ogg_demux_combine_flows),
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(gst_ogg_demux_loop_reverse), (gst_ogg_demux_loop):
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* ext/ogg/gstoggdemux.h:
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Properly propagate streaming errors when we are scanning the file for
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chains so that we don't crash when shut down. Might fix some crashers
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when quickly switching oggs in RB such as #332503 and #378436.
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2007-01-26 12:44:46 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/gnomevfs/gstgnomevfssrc.c: Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND error code as well.
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Original commit message from CVS:
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* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
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Map a gnome-vfs HOST_NOT_FOUND error into a GStreamer NOT_FOUND
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error code as well.
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2007-01-25 16:02:41 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstplaybasebin.c: Don't try to disconnect a signal from a finalized object.
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Original commit message from CVS:
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* gst/playback/gstplaybasebin.c: (remove_source):
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Don't try to disconnect a signal from a finalized object.
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2007-01-25 14:29:21 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gstdecodebin2.c: Cast lock macro parameters to make sure we're actually accessing the lock member at the...
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Original commit message from CVS:
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* gst/playback/gstdecodebin2.c: (gst_decode_bin_dispose):
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Cast lock macro parameters to make sure we're actually accessing the
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lock member at the right class level. Free list itself in _dispose()
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as well and NULL it in case dispose gets called multiple times.
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2007-01-25 14:02:37 +0000 Edward Hervey <bilboed@bilboed.com>
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gst/playback/gstdecodebin2.c: Free GstDecodeGroups no longer used.
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Original commit message from CVS:
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* gst/playback/gstdecodebin2.c:
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(gst_decode_bin_dispose),(gst_decode_bin_finalize):
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Free GstDecodeGroups no longer used.
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(gst_decode_group_expose):
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Don't unlock too many times !
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(deactivate_free_recursive):
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Free iterator once we're done with it.
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Fix for recursively deactivating elements (stop at ghostpads).
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2007-01-25 12:24:18 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gstplaybin.c: Fix up caps on the frame buffer before we save it and potentially make it accessible to ot...
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|
Original commit message from CVS:
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* gst/playback/gstplaybin.c: (handoff):
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Fix up caps on the frame buffer before we save it and potentially
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make it accessible to other threads via g_object_get; also use
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gst_buffer_replace() instead of gst_mini_object_replace().
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2007-01-25 12:06:59 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gstplaybin.c: Make getting the current frame thread-safe.
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Original commit message from CVS:
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* gst/playback/gstplaybin.c: (gst_play_bin_get_property):
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Make getting the current frame thread-safe.
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2007-01-25 11:48:10 +0000 Edward Hervey <bilboed@bilboed.com>
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gst/playback/gstdecodebin2.c: Set queues to bigger sizes to cope with HD contents.
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|
Original commit message from CVS:
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* gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize),
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(gst_decode_group_new), (gst_decode_group_free):
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Set queues to bigger sizes to cope with HD contents.
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Fix some mutex freeing and add comment about MT safe methods.
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2007-01-24 12:51:20 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/pango/gsttextoverlay.c: Don't unnecessarily ref (and then leak) upstream events if the text pad is not linked. Fi...
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|
Original commit message from CVS:
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* ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
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(gst_text_overlay_text_event):
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Don't unnecessarily ref (and then leak) upstream events if the text
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pad is not linked. Fixes #399948.
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* tests/check/gst-plugins-base.supp:
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Add suppression for pango on edgy/x86 for textoverlay test.
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2007-01-24 12:10:56 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/rtp/gstrtpbuffer.h: Add some more fixed payloads.
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Original commit message from CVS:
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* gst-libs/gst/rtp/gstrtpbuffer.h:
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Add some more fixed payloads.
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|
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2007-01-23 18:39:45 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/ogg/gstoggdemux.c: Error out properly if we get an error from libogg while reading the
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|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_read_chain):
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Error out properly if we get an error from libogg while reading the
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BOS page(s). Fixes crash parsing 'fuzzed' ogg file (#399340).
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2007-01-23 17:49:29 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gstdecodebin2.c: Don't leak mutex.
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|
Original commit message from CVS:
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* gst/playback/gstdecodebin2.c: (gst_decode_bin_finalize):
|
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Don't leak mutex.
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* tests/check/elements/playbin.c:
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(test_sink_usage_video_only_stream),
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(test_suburi_error_unknowntype), (test_suburi_error_invalidfile),
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(test_suburi_error_wrongproto), (test_missing_urisource_handler),
|
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(test_missing_suburisource_handler),
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(test_missing_primary_decoder), (playbin_suite):
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Run all tests once with decodebin and once with decodebin2.
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One test does not pass yet with decodebin2.
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2007-01-23 14:30:28 +0000 Edward Hervey <bilboed@bilboed.com>
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ext/ogg/gstoggmux.c: Fix the cases where oggmux doesn't properly figure out that all sinkpads have gone EOS, and ther...
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggmux.c: (all_pads_eos), (gst_ogg_mux_collected):
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|
Fix the cases where oggmux doesn't properly figure out that all
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sinkpads have gone EOS, and therefore doesn't push out the remaining
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buffers and the final EOS event.
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Fixes #363379
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2007-01-23 13:19:19 +0000 Julien Moutte <julien@moutte.net>
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sys/: Don't lock on navigation event push, just on keysym to string.
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|
Original commit message from CVS:
|
|
2007-01-23 Julien MOUTTE <julien@moutte.net>
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|
* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
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Don't lock on navigation event push, just on keysym to string.
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Fixes #397673 again.
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|
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2007-01-22 17:37:38 +0000 Edward Hervey <bilboed@bilboed.com>
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|
|
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gst/playback/gstdecodebin2.c: Cleanups.
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|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin2.c: (gst_decode_group_new),
|
|
(get_current_group), (group_demuxer_event_probe),
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(gst_decode_group_expose), (deactivate_free_recursive),
|
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(gst_decode_group_free):
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Cleanups.
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Don't forget to emit 'no-more-pads' once a group is exposed.
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Cleanup elements from a DecodeGroup once we remove it.
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Protect call to gst_decode_group_expose() with the decodebin lock.
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2007-01-22 13:16:42 +0000 Julien Moutte <julien@moutte.net>
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sys/: Looking at Xorg code i can't figure out if that XKeysymToString function is thread sensible or not. Lock it jus...
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|
Original commit message from CVS:
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|
2007-01-22 Julien MOUTTE <julien@moutte.net>
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|
* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
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Looking at Xorg code i can't figure out if that XKeysymToString
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function is thread sensible or not. Lock it just in case as
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recommended by Radek Doulik <rodo at ximian dot com>.
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2007-01-22 13:10:13 +0000 Julien Moutte <julien@moutte.net>
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sys/: Lock that X Call as well. Fixes #397673.
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Original commit message from CVS:
|
|
2007-01-22 Julien MOUTTE <julien@moutte.net>
|
|
* sys/ximage/ximagesink.c: (gst_ximagesink_handle_xevents):
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_handle_xevents):
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Lock that X Call as well. Fixes #397673.
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|
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2007-01-22 12:03:27 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Don't go into an endless loop if the file starts with 00 00 01 2X, like quicktim...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
|
|
Don't go into an endless loop if the file starts with 00 00 01 2X,
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like quicktime redirect files might. Fixes #396042.
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* tests/check/Makefile.am:
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* tests/check/gst/.cvsignore:
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* tests/check/gst/typefindfunctions.c: (GST_START_TEST),
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(typefindfunctions_suite):
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Add unit test for the above.
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|
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2007-01-22 10:27:26 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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gst-libs/gst/riff/riff-media.c: On second thought, use "depth" field rather than "bpp" field.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
|
|
On second thought, use "depth" field rather than "bpp" field.
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|
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2007-01-22 09:23:01 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/riff/riff-media.c: Camtasia caps apparently need a bpp field (#398875).
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
|
|
Camtasia caps apparently need a bpp field (#398875).
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|
|
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2007-01-19 19:09:05 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/playback/gstplaybasebin.c: Attempt at a better error message in case we don't have the required
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (setup_subtitle),
|
|
(gen_source_element), (gst_play_base_bin_change_state):
|
|
Attempt at a better error message in case we don't have the required
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URI handler installed; post missing-plugin message also when we're
|
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missing an URI handler for the subtitle URI; clean up properly also
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when an error occurs and we never made it to PAUSED state.
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* tests/check/elements/playbin.c: (GST_START_TEST),
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(playbin_suite):
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Check that we're also getting a missing-plugin messsage for a
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missing subtitle URI handler (and clean up properly).
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|
|
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2007-01-19 18:47:30 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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gst/playback/gstplaybasebin.c: Plug a few reference leaks.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (analyse_source), (setup_source):
|
|
Plug a few reference leaks.
|
|
|
|
2007-01-19 12:23:06 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Lower probability a bit if the marker isn't right at the start, to decrease the ...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
|
|
Lower probability a bit if the marker isn't right at the start,
|
|
to decrease the chance of false positives.
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|
|
|
2007-01-19 11:31:50 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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gst/typefind/gsttypefindfunctions.c: Small mpeg2 system stream typefinding improvement: make typefinder probe a bit i...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
|
|
Small mpeg2 system stream typefinding improvement: make typefinder
|
|
probe a bit into the stream instead of just looking for a marker
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at the beginning. Fixes #397810.
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|
|
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2007-01-18 16:23:35 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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gst/audioconvert/gstchannelmix.c: Remove compatibility cruft for prehistoric GLib versions.
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/gstchannelmix.c:
|
|
Remove compatibility cruft for prehistoric GLib versions.
|
|
|
|
2007-01-17 16:11:14 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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gst/playback/: Let decodebin be the element to post missing-plugin messages for missing decoders (rather than playbin...
|
|
Original commit message from CVS:
|
|
* gst/playback/Makefile.am:
|
|
* gst/playback/gstdecodebin.c: (close_pad_link):
|
|
* gst/playback/gstdecodebin2.c: (analyze_new_pad):
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* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
|
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(gst_play_base_bin_handle_message_func), (unknown_type):
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Let decodebin be the element to post missing-plugin messages for
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missing decoders (rather than playbin); make playbin implement
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GstBin::handle_message so we can suppress missing-plugin messages
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for types we're not handling on purpose (don't want to bring up an
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installer in those cases).
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2007-01-16 19:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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gst/: Fix potentially unaligned access (#397207).
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
|
|
* gst-libs/gst/tag/gstvorbistag.c:
|
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(gst_tag_list_to_vorbiscomment_buffer):
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* gst/typefind/gsttypefindfunctions.c: (vorbis_type_find):
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Fix potentially unaligned access (#397207).
|
|
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2007-01-16 12:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
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tests/examples/seek/seek.c: Allow to toggle looping while it plays. Fix callback prototype. Clean up code a bit more....
|
|
Original commit message from CVS:
|
|
* tests/examples/seek/seek.c: (set_scale), (update_scale),
|
|
(do_seek), (stop_seek), (pause_cb), (stop_cb), (loop_toggle_cb),
|
|
(rate_spinbutton_changed_cb), (msg_eos), (msg_segment_done),
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(main):
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Allow to toggle looping while it plays. Fix callback prototype. Clean
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up code a bit more. Add copyright header.
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2007-01-16 11:41:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
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sys/xvimage/xvimagesink.c: Red and blue mask was swapped (spotted by Dan Williams).
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
|
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Red and blue mask was swapped (spotted by Dan Williams).
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2007-01-15 13:58:58 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
|
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gst-libs/gst/tag/: Use new beats-per-minute tag from core.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/gstid3tag.c:
|
|
* gst-libs/gst/tag/gstvorbistag.c:
|
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Use new beats-per-minute tag from core.
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2007-01-15 11:30:53 +0000 Tim-Philipp Müller <tim@centricular.net>
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po/POTFILES.in: Add new files with translatable strings, so they actually make it into the template file one day.
|
|
Original commit message from CVS:
|
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* po/POTFILES.in:
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Add new files with translatable strings, so they actually make it
|
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into the template file one day.
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2007-01-12 21:19:35 +0000 Andy Wingo <wingo@pobox.com>
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* ChangeLog:
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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* gst-libs/gst/audio/gstbaseaudiosrc.c:
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gst-libs/gst/audio/gstbaseaudiosink.c
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Original commit message from CVS:
|
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2007-01-12 Andy Wingo <wingo@pobox.com>
|
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* gst-libs/gst/audio/gstbaseaudiosink.c
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(gst_base_audio_sink_fixate): Implement, stolen from baseaudiosrc.
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(gst_base_audio_sink_activate_pull): Remove the handwavey nego
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stuff, as the base class handles this now. Actually tell the ring
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buffer to start.
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(gst_base_audio_sink_callback): Cast the ring buffer correctly.
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How did this work before? Maybe I'm not as awesome a programmer as
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I think.
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* gst-libs/gst/audio/gstbaseaudiosrc.c
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(gst_base_audio_src_fixate): Rework as a basesrc vmethod instead
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of a pad function.
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2007-01-12 18:08:23 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/utils/missing-plugins.c: Remove more fields so that the application can better blacklist formats that ha...
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Original commit message from CVS:
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* gst-libs/gst/utils/missing-plugins.c: (copy_and_clean_caps):
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Remove more fields so that the application can better blacklist
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formats that have been tried before.
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2007-01-12 17:43:40 +0000 Christian Schaller <uraeus@gnome.org>
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* gst-plugins-base.spec.in:
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add latest files
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Original commit message from CVS:
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add latest files
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|
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2007-01-12 12:47:29 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/audio/mixerutils.h: Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be used when compiling...
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Original commit message from CVS:
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* gst-libs/gst/audio/mixerutils.h:
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Add G_BEGIN_DECLS and G_END_DECLS guards so these helpers can be
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used when compiling with c++ compilers as well.
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2007-01-12 09:45:23 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/typefind/gsttypefindfunctions.c: Fix comment.
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Original commit message from CVS:
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* gst/typefind/gsttypefindfunctions.c:
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Fix comment.
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2007-01-11 13:12:17 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gstplaybin.c: Post missing-plugin messages also when we error out because converters, textoverlay or aut...
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|
Original commit message from CVS:
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* gst/playback/gstplaybin.c: (post_missing_element_message),
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(gen_video_element), (gen_text_element), (gen_audio_element),
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(gen_vis_element):
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Post missing-plugin messages also when we error out because
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converters, textoverlay or auto*sinks are missing (#161922).
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2007-01-10 16:08:18 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/: Fix the case where we try to ref a NULL element when we delay a link because of unfixed caps.
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Original commit message from CVS:
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* gst/playback/gstdecodebin.c: (dynamic_add), (close_pad_link),
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(is_demuxer_element), (new_caps):
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* gst/playback/gstplaybasebin.c: (source_new_pad):
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Fix the case where we try to ref a NULL element when we delay a link
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because of unfixed caps.
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Set the state of autoplugged decodebins to PAUSED.
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RTSP now works in playbin, we can remove it from the blacklist.
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2007-01-09 14:33:24 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/: Post missing-plugin messages on the bus for missing sources and missing decoders/demuxers/depayloaders...
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Original commit message from CVS:
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* gst/playback/Makefile.am:
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* gst/playback/gstplaybasebin.c: (string_arr_has_str),
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(unknown_type), (setup_subtitle), (gen_source_element):
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* gst/playback/gstplaybin.c: (plugin_init):
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Post missing-plugin messages on the bus for missing sources and
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missing decoders/demuxers/depayloaders; fix error code used when
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we're missing an URI handler source; for media types that we are not
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handling on purpose at the moment, don't print "don't know how to
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handle xyz" messages to the terminal or post missing-plugin
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messages on the bus.
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* tests/check/elements/playbin.c: (create_playbin),
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(GST_START_TEST), (gst_codec_src_uri_get_type),
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(gst_codec_src_uri_get_protocols), (gst_codec_src_uri_get_uri),
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(gst_codec_src_uri_set_uri), (gst_codec_src_uri_handler_init),
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(gst_codec_src_init_type), (gst_codec_src_base_init),
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(gst_codec_src_create), (gst_codec_src_class_init),
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(gst_codec_src_init), (plugin_init), (playbin_suite):
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Add some tests for the missing-plugin stuff.
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2007-01-09 14:20:08 +0000 Tim-Philipp Müller <tim@centricular.net>
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API: add new libgstbaseutils library with functions
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Original commit message from CVS:
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* configure.ac:
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* gst-libs/gst/Makefile.am:
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* gst-libs/gst/utils/Makefile.am:
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* gst-libs/gst/utils/base-utils.c: (gst_base_utils_init):
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* gst-libs/gst/utils/base-utils.h:
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* gst-libs/gst/utils/descriptions.c: (format_info_get_desc),
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(find_format_info), (caps_are_rtp_caps),
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(gst_base_utils_get_source_description),
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(gst_base_utils_get_sink_description),
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(gst_base_utils_get_decoder_description),
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(gst_base_utils_get_encoder_description),
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(gst_base_utils_get_element_description),
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(gst_base_utils_add_codec_description_to_tag_list),
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(gst_base_utils_get_codec_description), (gst_base_utils_list_all):
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* gst-libs/gst/utils/descriptions.h:
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* gst-libs/gst/utils/missing-plugins.c:
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(missing_structure_get_type), (copy_and_clean_caps),
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(gst_missing_uri_source_message_new),
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(gst_missing_uri_sink_message_new),
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(gst_missing_element_message_new),
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(gst_missing_decoder_message_new),
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(gst_missing_encoder_message_new),
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(missing_structure_get_string_detail),
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(missing_structure_get_caps_detail),
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(gst_missing_plugin_message_get_installer_detail),
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(gst_missing_plugin_message_get_description),
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(gst_is_missing_plugin_message):
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* gst-libs/gst/utils/missing-plugins.h:
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API: add new libgstbaseutils library with functions
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- to create and parse missing-plugins messages
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- that provide (translated) descriptions for caps/decoders/sources/etc.
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Closes #392393.
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* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
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* pkgconfig/gstreamer-plugins-base.pc.in:
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Add new lib.
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* docs/libs/gst-plugins-base-libs-docs.sgml:
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* docs/libs/gst-plugins-base-libs-sections.txt:
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Generate docs for new lib and API.
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* tests/check/Makefile.am:
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* tests/check/libs/.cvsignore:
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* tests/check/libs/utils.c: (missing_msg_check_getters),
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(GST_START_TEST), (libgstbaseutils_suite):
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Add some basic unit tests.
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2007-01-09 13:35:08 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/ogg/Makefile.am: Dist gstoggdemux.h to fix 'make distcheck'.
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Original commit message from CVS:
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* ext/ogg/Makefile.am:
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Dist gstoggdemux.h to fix 'make distcheck'.
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* sys/v4l/Makefile.am:
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Fix 'make distcheck' even more.
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2007-01-09 12:30:46 +0000 Wim Taymans <wim.taymans@gmail.com>
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Added docs.
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Original commit message from CVS:
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-base-plugins-docs.sgml:
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* docs/plugins/gst-plugins-base-plugins-sections.txt:
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* ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
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(gst_ogg_pad_query_types), (gst_ogg_pad_submit_page),
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(gst_ogg_chain_reset), (gst_ogg_chain_new_stream),
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(gst_ogg_demux_perform_seek):
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* ext/ogg/gstoggdemux.h:
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Added docs.
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Add some more comments.
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Small cleanups.
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2007-01-09 11:15:57 +0000 Wim Taymans <wim.taymans@gmail.com>
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Small documentation updates/fixes
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|
Original commit message from CVS:
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* ext/theora/theoradec.c:
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* ext/vorbis/vorbisdec.c:
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* gst-libs/gst/audio/gstringbuffer.c:
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(gst_ring_buffer_commit_full):
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* gst-libs/gst/audio/gstringbuffer.h:
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* gst-libs/gst/rtp/gstrtpbuffer.c:
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* gst-libs/gst/tag/gstvorbistag.c:
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Small documentation updates/fixes
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2007-01-09 10:37:01 +0000 Tim-Philipp Müller <tim@centricular.net>
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configure.ac: Require core CVS HEAD for Andy's basesrc/sink API additions.
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Original commit message from CVS:
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* configure.ac:
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Require core CVS HEAD for Andy's basesrc/sink API additions.
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2007-01-08 14:01:23 +0000 Günter Thelen <daedalus.inc@gmx.net>
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gst/typefind/gsttypefindfunctions.c: Add typefinder for flac-in-ogg in conformance with the ogg-mapping on flac.sf.ne...
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Original commit message from CVS:
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Patch by: Günter Thelen <daedalus dot inc at gmx net>
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* gst/typefind/gsttypefindfunctions.c: (flac_type_find),
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(plugin_init):
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Add typefinder for flac-in-ogg in conformance with the ogg-mapping
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on flac.sf.net (there appear to be other versions of the first
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ogg page in the wild) (#391365).
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2007-01-08 13:32:32 +0000 Tim-Philipp Müller <tim@centricular.net>
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configure.ac: Check if localtime_r() is available.
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Original commit message from CVS:
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* configure.ac:
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Check if localtime_r() is available.
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* ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time):
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If localtime_r() is not available, fall back to localtime(). Should
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fix build on MingW (#393310).
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2007-01-08 12:30:03 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/subparse/gstsubparse.*: Remove spurious 1000 subtrahend when calculating the timestamp from the frame number and ...
|
|
Original commit message from CVS:
|
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* gst/subparse/gstsubparse.c: (parse_mdvdsub):
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* gst/subparse/gstsubparse.h:
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Remove spurious 1000 subtrahend when calculating the timestamp from
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the frame number and the frame rate . Also, use the frames/second
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value specified in the first line of the file, if one is specified
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there. Should fix #357503.
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* tests/check/elements/subparse.c: (do_test),
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(test_tmplayer_do_test), (test_microdvd_do_test), (GST_START_TEST),
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(subparse_suite):
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Add some basic unit tests for the microdvd subtitle format.
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2007-01-07 21:53:38 +0000 Young-Ho Cha <ganadist@chollian.net>
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sys/xvimage/xvimagesink.c: Fixes : #390076.
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Original commit message from CVS:
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|
2007-01-07 Julien MOUTTE <julien@moutte.net>
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* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
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(gst_xvimage_buffer_finalize), (gst_xvimagesink_xvimage_new),
|
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(gst_xvimagesink_xvimage_put),
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(gst_lookup_xv_port_from_adaptor),
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(gst_xvimagesink_get_xv_support), (gst_xvimagesink_setcaps),
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(gst_xvimagesink_set_xwindow_id),
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(gst_xvimagesink_set_event_handling),
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(gst_xvimagesink_set_property), (gst_xvimagesink_get_property),
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(gst_xvimagesink_init), (gst_xvimagesink_class_init):
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Patch by : Young-Ho Cha <ganadist at chollian dot net>
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Fixes : #390076.
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Add an adaptor property to select a specific XV adaptor.
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* sys/xvimage/xvimagesink.h:
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2007-01-07 18:50:13 +0000 Julien Moutte <julien@moutte.net>
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sys/: Use flow_lock much more to protect every access to xwindow.
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|
Original commit message from CVS:
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|
2007-01-07 Julien MOUTTE <julien@moutte.net>
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* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
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(gst_ximagesink_handle_xerror), (gst_ximagesink_ximage_new),
|
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(gst_ximagesink_ximage_destroy), (gst_ximagesink_ximage_put),
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(gst_ximagesink_handle_xevents), (gst_ximagesink_setcaps),
|
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(gst_ximagesink_change_state), (gst_ximagesink_set_xwindow_id),
|
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(gst_ximagesink_expose), (gst_ximagesink_set_event_handling):
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* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
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(gst_xvimage_buffer_finalize), (gst_xvimagesink_handle_xerror),
|
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(gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
|
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(gst_xvimagesink_handle_xevents), (gst_xvimagesink_setcaps),
|
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(gst_xvimagesink_change_state),
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(gst_xvimagesink_set_xwindow_id),
|
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(gst_xvimagesink_expose), (gst_xvimagesink_set_event_handling):
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Use flow_lock much more to protect every access to xwindow.
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Try to catch erros while creating images in case some drivers
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are
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just generating an XError when the requested image is too big.
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Should fix : #354698, #384008, #384060.
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* tests/icles/stress-xoverlay.c: (cycle_window),
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(create_window):
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Implement some stress testing of setting window xid.
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|
|
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2007-01-07 10:33:55 +0000 Sébastien Moutte <sebastien@moutte.net>
|
|
|
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win32/common/libgsaudio.def: Add new exported function.
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|
Original commit message from CVS:
|
|
* win32/common/libgsaudio.def:
|
|
Add new exported function.
|
|
* win32/common/libgstogg.dsp:
|
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Add gstoggaviparse.c to the build.
|
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* win32/common/libgstvideoscale.dsp:
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Add vs_4tap.c to the build.
|
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* win32/common/libgstvorbis.dsp:
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Add vorbistag.c to the build.
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2007-01-06 17:28:40 +0000 Andy Wingo <wingo@pobox.com>
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|
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* ChangeLog:
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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gst-libs/gst/audio/gstbaseaudiosink.c (gst_base_audio_sink_class_init)
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|
Original commit message from CVS:
|
|
2007-01-06 Andy Wingo <wingo@pobox.com>
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c
|
|
(gst_base_audio_sink_class_init)
|
|
(gst_base_audio_sink_init):
|
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(gst_base_audio_sink_activate_pull): Add an activate_pull function
|
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to baseaudiosink, and tell basesink that we can work in pull mode.
|
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This way the ring buffer thread drives the pipeline directly, if
|
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pull mode is possible. There is some lingering nastiness regarding
|
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capsnego, however.
|
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(gst_base_audio_sink_callback): Implement the callback to pull
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data. This interface is a bit light, though -- it should get a
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GstFlowReturn return value at least.
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|
|
|
2007-01-05 19:43:55 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
Printf format and missing argument fixes.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_stream_out):
|
|
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
|
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* gst/playback/gstdecodebin2.c:
|
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(gst_decode_group_check_if_blocked):
|
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Printf format and missing argument fixes.
|
|
|
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2007-01-05 18:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
ext/ogg/gstogmparse.c: Activate pads before adding them to the element.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header),
|
|
(gst_ogm_parse_change_state):
|
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Activate pads before adding them to the element.
|
|
|
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2007-01-05 16:02:50 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/examples/seek/: Call g_thread_init() first thing in main() (see #391278).
|
|
Original commit message from CVS:
|
|
* tests/examples/seek/scrubby.c: (main):
|
|
* tests/examples/seek/seek.c: (main):
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Call g_thread_init() first thing in main() (see #391278).
|
|
|
|
2007-01-05 12:19:34 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/: Add test for GstNetBuffer + gst_buffer_copy(). Disabled for the time being, since it's broken, see #393...
|
|
Original commit message from CVS:
|
|
* tests/check/Makefile.am:
|
|
* tests/check/libs/.cvsignore:
|
|
* tests/check/libs/netbuffer.c: (GST_START_TEST),
|
|
(netbuffer_suite):
|
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Add test for GstNetBuffer + gst_buffer_copy(). Disabled
|
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for the time being, since it's broken, see #393099.
|
|
|
|
2007-01-05 12:13:24 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/Makefile.am: Update to use GST_PLUGINS_BASE_CFLAGS as well.
|
|
Original commit message from CVS:
|
|
* tests/check/Makefile.am:
|
|
Update to use GST_PLUGINS_BASE_CFLAGS as well.
|
|
|
|
2007-01-04 12:49:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
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configure.ac: split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS so that GST_BASE_CFLAGS can go inbetwe...
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
split out GST_CFLAGS into GST_PLUGINS_BASE_CFLAGS and GST_CFLAGS
|
|
so that GST_BASE_CFLAGS can go inbetween them, making sure
|
|
we use uninstalled gst-libs headers
|
|
* docs/libs/Makefile.am:
|
|
* ext/alsa/Makefile.am:
|
|
* ext/cdparanoia/Makefile.am:
|
|
* ext/gnomevfs/Makefile.am:
|
|
* ext/libvisual/Makefile.am:
|
|
* ext/ogg/Makefile.am:
|
|
* ext/theora/Makefile.am:
|
|
* ext/vorbis/Makefile.am:
|
|
* gst-libs/gst/audio/Makefile.am:
|
|
* gst-libs/gst/cdda/Makefile.am:
|
|
* gst-libs/gst/interfaces/Makefile.am:
|
|
* gst-libs/gst/riff/Makefile.am:
|
|
* gst-libs/gst/rtp/Makefile.am:
|
|
* gst-libs/gst/tag/Makefile.am:
|
|
* gst/adder/Makefile.am:
|
|
* gst/audioconvert/Makefile.am:
|
|
* gst/audiorate/Makefile.am:
|
|
* gst/audioresample/Makefile.am:
|
|
* gst/playback/Makefile.am:
|
|
* gst/tcp/Makefile.am:
|
|
* gst/videoscale/Makefile.am:
|
|
* gst/volume/Makefile.am:
|
|
* sys/ximage/Makefile.am:
|
|
* sys/xvimage/Makefile.am:
|
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* tests/icles/Makefile.am:
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adapt
|
|
|
|
2007-01-04 11:30:53 +0000 Julien Moutte <julien@moutte.net>
|
|
|
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Add a method to the XOverlay interface to allow disabling of event handling in x[v]imagesink elements. This will let ...
|
|
Original commit message from CVS:
|
|
2007-01-04 Julien MOUTTE <julien@moutte.net>
|
|
* gst-libs/gst/interfaces/xoverlay.c:
|
|
(gst_x_overlay_handle_events):
|
|
* gst-libs/gst/interfaces/xoverlay.h:
|
|
* sys/ximage/ximagesink.c: (gst_ximagesink_xwindow_new),
|
|
(gst_ximagesink_set_xwindow_id),
|
|
(gst_ximagesink_set_event_handling),
|
|
(gst_ximagesink_xoverlay_init), (gst_ximagesink_set_property),
|
|
(gst_ximagesink_get_property), (gst_ximagesink_init),
|
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(gst_ximagesink_class_init):
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* sys/ximage/ximagesink.h:
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|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xwindow_new),
|
|
(gst_xvimagesink_set_xwindow_id),
|
|
(gst_xvimagesink_set_event_handling),
|
|
(gst_xvimagesink_xoverlay_init), (gst_xvimagesink_set_property),
|
|
(gst_xvimagesink_get_property), (gst_xvimagesink_init),
|
|
(gst_xvimagesink_class_init):
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|
* sys/xvimage/xvimagesink.h:
|
|
* tests/icles/stress-xoverlay.c: (toggle_events),
|
|
(create_window):
|
|
Add a method to the XOverlay interface to allow disabling of
|
|
event handling in x[v]imagesink elements. This will let X events
|
|
propagate to parent windows which can be usefull in some cases.
|
|
Be carefull that the application is then responsible of pushing
|
|
navigation events and expose events to the video sink.
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|
Fixes: #387138.
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|
|
|
2007-01-03 15:45:06 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
|
Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION (fixes #392070).
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/gstvorbistag.c:
|
|
* tests/check/libs/tag.c: (GST_START_TEST):
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|
Add vorbistag <=> GStreamer tag mapping for GST_TAG_LOCATION
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(fixes #392070).
|
|
|
|
2007-01-01 18:59:49 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
Dist design docs.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* docs/Makefile.am:
|
|
* docs/design/Makefile.am:
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|
Dist design docs.
|
|
|
|
2006-12-27 17:15:35 +0000 Julien Moutte <julien@moutte.net>
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|
|
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docs/libs/gst-plugins-base-libs-sections.txt: Fix a documentation typo. Fixes: #390063.
|
|
Original commit message from CVS:
|
|
2006-12-27 Julien MOUTTE <julien@moutte.net>
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|
* docs/libs/gst-plugins-base-libs-sections.txt: Fix a
|
|
documentation
|
|
typo. Fixes: #390063.
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|
|
|
2006-12-27 12:08:13 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
sys/: Plug a caps leak.
|
|
Original commit message from CVS:
|
|
2006-12-27 Julien MOUTTE <julien@moutte.net>
|
|
* sys/ximage/ximagesink.c: (gst_ximagesink_setcaps):
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps): Plug a
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|
caps leak.
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* win32/common/config.h: Updated.
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|
|
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2006-12-22 12:10:18 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
|
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tests/check/elements/: Fix the dp tests, but activating the pads for the streamheader tests too and cleaning up condi...
|
|
Original commit message from CVS:
|
|
* tests/check/elements/gdpdepay.c: (cleanup_gdpdepay),
|
|
(setup_gdpdepay_streamheader):
|
|
* tests/check/elements/gdppay.c: (cleanup_gdppay),
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|
(setup_gdppay_streamheader):
|
|
Fix the dp tests, but activating the pads for the streamheader tests
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|
too and cleaning up conditionaly
|
|
|
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2006-12-22 11:09:34 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
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gst/ffmpegcolorspace/: Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the other end of the wo...
|
|
Original commit message from CVS:
|
|
* gst/ffmpegcolorspace/avcodec.h:
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
|
|
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
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(gst_ffmpegcsp_avpicture_fill):
|
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* gst/ffmpegcolorspace/imgconvert.c: (img_convert),
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|
(img_get_alpha_info):
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Add 2 new caps arrangements, for 24-bit RGB and BGR in 32-bits, but at the
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other end of the word. Fixes: #387073.
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Add some inconsequential branch hints in a couple of places.
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|
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2006-12-21 12:30:11 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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gst/ffmpegcolorspace/gstffmpegcodecmap.c: The "signed" field in raw audio caps is of boolean type, trying to extract ...
|
|
Original commit message from CVS:
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
|
|
(gst_ffmpeg_caps_to_smpfmt):
|
|
The "signed" field in raw audio caps is of boolean type, trying to
|
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extract the value with _get_int() will fail (fix to keep in sync with
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the copy in gst-ffmpeg)
|
|
|
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2006-12-21 08:12:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
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tests/check/elements/: consistent pad (de)activation
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|
Original commit message from CVS:
|
|
* tests/check/elements/audioresample.c: (cleanup_audioresample):
|
|
* tests/check/elements/audiotestsrc.c: (cleanup_audiotestsrc):
|
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* tests/check/elements/gdpdepay.c: (setup_gdpdepay),
|
|
(cleanup_gdpdepay):
|
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* tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay):
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* tests/check/elements/subparse.c: (teardown_subparse):
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* tests/check/elements/textoverlay.c: (cleanup_textoverlay):
|
|
* tests/check/elements/videorate.c: (cleanup_videorate):
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* tests/check/elements/videotestsrc.c: (cleanup_videotestsrc):
|
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* tests/check/elements/volume.c: (cleanup_volume):
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|
* tests/check/elements/vorbisdec.c: (setup_vorbisdec),
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|
(cleanup_vorbisdec):
|
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* tests/check/elements/vorbistag.c: (setup_vorbistag),
|
|
(cleanup_vorbistag):
|
|
consistent pad (de)activation
|
|
|
|
2006-12-20 10:29:58 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Forgot to register the extensions.
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
|
|
Forgot to register the extensions.
|
|
|
|
2006-12-20 09:25:55 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Add typefinder for VIVO files (my christmas present to the 90s).
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (vivo_type_find),
|
|
(plugin_init):
|
|
Add typefinder for VIVO files (my christmas present to the 90s).
|
|
|
|
2006-12-16 13:59:09 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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gst/playback/gstdecodebin.c: Special-case the text/plain media type: we only want to recognise it as a 'raw' decoded ...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (type_found):
|
|
Special-case the text/plain media type: we only want to recognise it
|
|
as a 'raw' decoded media type if it comes from a demuxer or subtitle
|
|
parser, but not if the entire stream is of text/plain type. If the
|
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entire stream is text/plain, we should just error out.
|
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This fixes playback of audio files with lyrics in totem. Totem can't
|
|
distinguish between text files and subtitle files and passes any
|
|
.txt file with the same basename as the main file to playbin as
|
|
suburi, and playbin will then throw a 'subtitle found, but no video
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|
stream' error, which isn't entirely helpful. See #380342.
|
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Also, with this change we'll show a slightly more correct error
|
|
message in case totem passes a playlist file to us (although a
|
|
custom error message wording instead of the default text would
|
|
probably not be a bad idea either).
|
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Same problem also needs to be fixed for playbin+decodebin2.
|
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* tests/check/Makefile.am:
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* tests/check/elements/decodebin.c: (src_handoff_cb),
|
|
(decodebin_new_decoded_pad_cb), (GST_START_TEST),
|
|
(decodebin_suite):
|
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Add simple unit test for decodebin for the above.
|
|
|
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2006-12-16 12:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/playback/: Refuse to change state to READY when we failed to create any of the required elements in our instance ...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (gst_decode_bin_change_state):
|
|
* gst/playback/gstdecodebin2.c: (gst_decode_bin_change_state):
|
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Refuse to change state to READY when we failed to create any of the
|
|
required elements in our instance init function.
|
|
|
|
2006-12-15 10:52:23 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
docs/libs/gst-plugins-base-libs-sections.txt: Small docs fixes/updates.
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
Small docs fixes/updates.
|
|
* gst-libs/gst/video/gstvideosink.h:
|
|
Remove nonfunctional GST_VIDEO_SINK_CLOCK macro which is a leftover
|
|
from the 0.9 days (GST_BASE_SINK_CLOCK, which it points to, was
|
|
removed from the base sink API between 0.9.6 and 0.9.7).
|
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API: add GST_VIDEO_SINK_CAST and use it for the height/width
|
|
accessor macros, so we don't do a runtime GObject type check every
|
|
time we use them.
|
|
|
|
2006-12-15 00:20:37 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
add doap file
|
|
Original commit message from CVS:
|
|
* Makefile.am:
|
|
* gst-plugins-base.doap:
|
|
* gst-plugins-base.spec.in:
|
|
add doap file
|
|
|
|
2006-12-09 15:12:38 +0000 Jens Granseuer <jensgr@gmx.net>
|
|
|
|
Declare variables at the beginning of a block. Fixes #383195.
|
|
Original commit message from CVS:
|
|
Patch by: Jens Granseuer <jensgr at gmx net>
|
|
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
(gst_base_rtp_audio_payload_handle_frame_based_buffer),
|
|
(gst_base_rtp_audio_payload_handle_sample_based_buffer):
|
|
* sys/v4l/gstv4lsrc.c: (gst_v4lsrc_fixate):
|
|
Declare variables at the beginning of a block. Fixes #383195.
|
|
|
|
2006-12-07 02:38:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
configure.ac: Bump version nano - back to CVS.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Bump version nano - back to CVS.
|
|
|
|
=== release 0.10.11 ===
|
|
|
|
2006-12-07 02:30:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
configure.ac: releasing 0.10.11, "Dumb things"
|
|
Original commit message from CVS:
|
|
=== release 0.10.11 ===
|
|
2006-12-06 Jan Schmidt <thaytan@mad.scientist.com>
|
|
* configure.ac:
|
|
releasing 0.10.11, "Dumb things"
|
|
|
|
2006-12-05 12:44:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst/playback/gstdecodebin.c: Handle the case where an element has multiple pads with unfixed caps as well as still po...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (find_dynamic), (dynamic_add),
|
|
(close_pad_link), (elem_is_dynamic), (unlinked), (close_link):
|
|
Handle the case where an element has multiple pads with
|
|
unfixed caps as well as still possibly producing more dynamic
|
|
pads by storing each case as a distinct entry in the dynamic list.
|
|
Fixes #38223 again.
|
|
|
|
2006-12-04 13:02:37 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstdecodebin.c: Fix #382223, add more dynamic caps handling.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (close_pad_link):
|
|
Fix #382223, add more dynamic caps handling.
|
|
|
|
2006-12-01 11:35:57 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
|
|
|
|
* po/.gitignore:
|
|
Ignore all pot files
|
|
Original commit message from CVS:
|
|
Ignore all pot files
|
|
|
|
2006-12-01 10:36:50 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/audiorate/gstaudiorate.c: Delete bad debug code.
|
|
Original commit message from CVS:
|
|
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
|
|
Delete bad debug code.
|
|
Fixes #381219
|
|
|
|
2006-12-01 10:27:54 +0000 Sergey Scobich <sergey.scobich@gmail.com>
|
|
|
|
Fix compilation on win32 under VS8
|
|
Original commit message from CVS:
|
|
* gst/videoscale/vs_4tap.c:
|
|
* win32/MANIFEST:
|
|
* win32/common/config.h:
|
|
* win32/vs8/libgstvideoscale.vcproj:
|
|
Fix compilation on win32 under VS8
|
|
Patch by: Sergey Scobich <sergey dot scobich at gmail dot com>
|
|
Partially fixes #381175
|
|
|
|
2006-11-30 23:46:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/cs.po:
|
|
* po/en_GB.po:
|
|
* po/hu.po:
|
|
* po/it.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
Update .po files
|
|
Original commit message from CVS:
|
|
Update .po files
|
|
|
|
2006-11-30 12:50:42 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
tests/check/pipelines/theoraenc.c: It would be very bad if, after a discont buffer, we thought every single following...
|
|
Original commit message from CVS:
|
|
* tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
|
|
(GST_START_TEST):
|
|
It would be very bad if, after a discont buffer, we thought every
|
|
single following buffer was also discont. So, add to the test to
|
|
ensure that this isn't the case.
|
|
* ext/theora/theoraenc.c: (theora_enc_is_discontinuous):
|
|
... it was the case. So fix it.
|
|
|
|
2006-11-28 16:43:18 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybasebin.c: Improve debug.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (check_queue_event):
|
|
Improve debug.
|
|
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps):
|
|
Fix width and height range from 16 - 4096 to 1 - MAXINT, just like the
|
|
padtemplate caps. Refixes #357577.
|
|
|
|
2006-11-28 16:21:27 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybasebin.c: Add event probe to see when EOS is in a queue and we can disable the underrun signals....
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (check_queue_event),
|
|
(queue_threshold_reached), (queue_out_of_data),
|
|
(gen_preroll_element):
|
|
Add event probe to see when EOS is in a queue and we can disable the
|
|
underrun signals. Fixes #357577.
|
|
|
|
2006-11-28 14:40:39 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
gst/playback/: New decodebin2 element.
|
|
Original commit message from CVS:
|
|
* gst/playback/Makefile.am:
|
|
* gst/playback/gstdecodebin2.c: (gst_decode_bin_get_type),
|
|
(_gst_boolean_accumulator), (gst_decode_bin_class_init),
|
|
(gst_decode_bin_factory_filter), (compare_ranks), (print_feature),
|
|
(gst_decode_bin_init), (gst_decode_bin_dispose),
|
|
(gst_decode_bin_finalize), (gst_decode_bin_set_property),
|
|
(gst_decode_bin_get_property), (gst_decode_bin_set_caps),
|
|
(gst_decode_bin_get_caps), (gst_decode_bin_autoplug_continue),
|
|
(gst_decode_bin_autoplug_sort), (analyze_new_pad), (connect_pad),
|
|
(connect_element), (expose_pad), (type_found),
|
|
(pad_added_group_cb), (pad_removed_group_cb),
|
|
(no_more_pads_group_cb), (pad_added_cb), (pad_removed_cb),
|
|
(no_more_pads_cb), (find_compatibles), (is_demuxer_element),
|
|
(are_raw_caps), (multi_queue_overrun_cb),
|
|
(multi_queue_underrun_cb), (gst_decode_group_new),
|
|
(get_current_group), (group_demuxer_event_probe),
|
|
(gst_decode_group_control_demuxer_pad),
|
|
(gst_decode_group_control_source_pad),
|
|
(gst_decode_group_check_if_blocked),
|
|
(gst_decode_group_check_if_drained), (gst_decode_group_expose),
|
|
(gst_decode_group_hide), (gst_decode_group_free),
|
|
(gst_decode_group_set_complete), (source_pad_blocked_cb),
|
|
(source_pad_event_probe), (gst_decode_pad_new), (add_fakesink),
|
|
(remove_fakesink), (find_sink_pad), (gst_decode_bin_change_state),
|
|
(plugin_init):
|
|
New decodebin2 element.
|
|
Closes #370092
|
|
* gst/playback/gstplay-marshal.list:
|
|
Added marshallers for new signals in decodebin2
|
|
* gst/playback/gstplaybasebin.c: (setup_subtitle), (make_decoder):
|
|
Use decodebin2 if *and only if* the USE_DECODEBIN2 environment variable
|
|
is set.
|
|
|
|
2006-11-28 10:45:40 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybasebin.c: Disable rtsp:// uris for the release, it's not good enough yet.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (setup_source),
|
|
(gst_play_base_bin_change_state):
|
|
Disable rtsp:// uris for the release, it's not good enough yet.
|
|
Remove unused var.
|
|
|
|
2006-11-26 16:39:41 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/theora/theoradec.c: Implement reverse playback.
|
|
Original commit message from CVS:
|
|
* ext/theora/theoradec.c: (gst_theora_dec_reset),
|
|
(theora_dec_push_forward), (theora_dec_push_reverse),
|
|
(theora_handle_data_packet), (theora_dec_decode_buffer),
|
|
(theora_dec_flush_decode), (theora_dec_chain_reverse),
|
|
(theora_dec_chain_forward), (theora_dec_chain):
|
|
Implement reverse playback.
|
|
* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
|
|
(vorbis_dec_decode_buffer), (vorbis_dec_flush_decode),
|
|
(vorbis_dec_chain_forward):
|
|
Clear buffers used for reverse playback in _reset.
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|
No need to set the eos flag, we clip samples using the segment.
|
|
|
|
2006-11-24 15:40:58 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/ogg/gstoggdemux.c: Some cleanups.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_page_copy), (gst_ogg_page_free),
|
|
(gst_ogg_pad_init), (gst_ogg_pad_dispose), (gst_ogg_pad_reset),
|
|
(gst_ogg_pad_stream_out), (gst_ogg_pad_submit_page),
|
|
(gst_ogg_chain_reset), (gst_ogg_demux_perform_seek):
|
|
Some cleanups.
|
|
Handle continued pages in reverse mode.
|
|
|
|
2006-11-24 15:39:03 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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ext/vorbis/vorbisdec.c: Small cleanups.
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisdec.c: (vorbis_dec_push_forward),
|
|
(vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
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(vorbis_dec_flush_decode):
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Small cleanups.
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Don't try to add invalid timestamps.
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Clipping will unref the buffer.
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|
|
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2006-11-24 08:56:10 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
|
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gst/: remove obsolete _factory_init protos
|
|
Original commit message from CVS:
|
|
* gst/adder/gstadder.h:
|
|
* gst/audiotestsrc/gstaudiotestsrc.h:
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remove obsolete _factory_init protos
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|
|
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2006-11-24 08:35:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
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|
sys/xvimage/xvimagesink.c: Fix spacing in debug message.
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
|
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Fix spacing in debug message.
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|
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2006-11-23 11:07:23 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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ext/ogg/gstoggdemux.c: Don't just ignore return values from _pad_push().
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|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
|
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(gst_ogg_demux_chain):
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Don't just ignore return values from _pad_push().
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Small debug improvements.
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2006-11-23 11:02:11 +0000 Michael Smith <msmith@xiph.org>
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ext/ogg/gstoggmux.c: If our incoming buffer is marked as DISCONT, then increment the page number (so that the discont...
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggmux.c: (gst_ogg_mux_process_best_pad):
|
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If our incoming buffer is marked as DISCONT, then increment the page
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number (so that the discontinuity is marked in the final ogg
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bitstream) and flush the previous page.
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2006-11-22 14:34:03 +0000 Michael Smith <msmith@xiph.org>
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ext/theora/: Mark discontinuities of > 3/4 of a frame, reinit encoder.
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|
Original commit message from CVS:
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* ext/theora/gsttheoraenc.h:
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* ext/theora/theoraenc.c: (gst_theora_enc_init),
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|
(theora_enc_reset), (theora_enc_clear), (theora_enc_sink_setcaps),
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(theora_buffer_from_packet), (theora_enc_is_discontinuous),
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(theora_enc_chain), (theora_enc_change_state):
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Mark discontinuities of > 3/4 of a frame, reinit encoder.
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* tests/check/pipelines/theoraenc.c: (check_buffer_granulepos),
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(GST_START_TEST), (theoraenc_suite):
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Enable discontinuity test, fix it.
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2006-11-21 18:39:34 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/pango/gsttextoverlay.*: Some textoverlay fixes: for one, in the video chain function, actually wait for a text bu...
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|
Original commit message from CVS:
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|
* ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
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|
(gst_text_overlay_text_pad_unlink), (gst_text_overlay_text_event),
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(gst_text_overlay_video_event), (gst_text_overlay_pop_text),
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(gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
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(gst_text_overlay_change_state):
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* ext/pango/gsttextoverlay.h:
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Some textoverlay fixes: for one, in the video chain function,
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actually wait for a text buffer to come in if there is none at the
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moment and there should be one; also, deal more gracefully with
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incoming buffers that do not have a timestamp or duration; discard
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text buffer when not needed any longer. Fixes #341681.
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* tests/check/Makefile.am:
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* tests/check/elements/.cvsignore:
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* tests/check/elements/textoverlay.c:
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(notgst_check_setup_src_pad2), (notgst_check_teardown_src_pad2),
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(setup_textoverlay), (buffer_is_all_black), (create_black_buffer),
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(create_text_buffer), (cleanup_textoverlay), (GST_START_TEST),
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(test_video_waits_for_text_send_text_newsegment_thread),
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(test_video_waits_for_text_shutdown_element),
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(test_render_continuity_push_video_buffers_thread),
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(textoverlay_suite):
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Add some unit tests for textoverlay.
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2006-11-21 09:29:56 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/typefind/gsttypefindfunctions.c: Avoid integer underflow when the found probability for mp3 is smaller than the '...
|
|
Original commit message from CVS:
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|
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
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Avoid integer underflow when the found probability for mp3 is
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smaller than the 'penalty' we subtract if there's not a clean
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mp3 header sync at offset 0.
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2006-11-21 08:17:16 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
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docs/libs/gst-plugins-base-libs-sections.txt: Add some new symbols to the docs
|
|
Original commit message from CVS:
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|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
Add some new symbols to the docs
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|
|
|
2006-11-20 16:44:28 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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tests/check/: Enable ffmpegcolorspace test now that the RGBA32 issue is fixed (for now not for valgrinding though, si...
|
|
Original commit message from CVS:
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|
* tests/check/Makefile.am:
|
|
* tests/check/elements/ffmpegcolorspace.c:
|
|
(ffmpegcolorspace_suite):
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Enable ffmpegcolorspace test now that the RGBA32 issue is fixed
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(for now not for valgrinding though, since it takes too long).
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|
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2006-11-20 15:01:09 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst/ffmpegcolorspace/gstffmpegcodecmap.c: Fix RGBA32 caps. Fixes #357038.
|
|
Original commit message from CVS:
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
|
|
(gst_ffmpeg_pixfmt_to_caps):
|
|
Fix RGBA32 caps. Fixes #357038.
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|
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|
2006-11-20 12:20:39 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
|
gst-libs/gst/interfaces/mixertrack.h: Add FIXME so we can add some padding here in 0.11
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|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/mixertrack.h:
|
|
Add FIXME so we can add some padding here in 0.11
|
|
|
|
2006-11-19 17:07:34 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
|
gst-libs/gst/rtp/gstbasertpaudiopayload.h: Fix GstBaseRTPAudioPayload structure so the whole GObject inheritance busi...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
|
|
Fix GstBaseRTPAudioPayload structure so the whole GObject
|
|
inheritance business actually works (parent class instance structure
|
|
must always come first in the derived class instance structure).
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|
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2006-11-16 14:35:30 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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Make sure our checks and the videotestsrc plugin link against the local uninstalled gst libs and not any installed gs...
|
|
Original commit message from CVS:
|
|
* gst/videotestsrc/Makefile.am:
|
|
* tests/check/Makefile.am:
|
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Make sure our checks and the videotestsrc plugin link against the
|
|
local uninstalled gst libs and not any installed gst libs that
|
|
might happen to exist as well.
|
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* tests/check/elements/adder.c: (message_received),
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(test_event_message_received), (test_play_twice_message_received):
|
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* tests/check/elements/ffmpegcolorspace.c: (GST_START_TEST):
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Fix compiler warnings when compiling against core with disabled
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|
debugging system.
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|
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2006-11-16 12:55:08 +0000 Michael Smith <msmith@xiph.org>
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|
|
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gst/audiorate/gstaudiorate.c: Fix audiorate, so that it accurately sets offsets and timestamps.
|
|
Original commit message from CVS:
|
|
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
|
|
(gst_audio_rate_sink_event), (gst_audio_rate_chain):
|
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Fix audiorate, so that it accurately sets offsets and timestamps.
|
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Doesn't change the fundamental algorithmic decisions; so should be
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|
safe.
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* tests/check/Makefile.am:
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Enable audiorate test now that it passes.
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|
|
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2006-11-15 10:05:33 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
|
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sys/xvimage/xvimagesink.c: clear xv when going to NULL, remove // commented non-existant proto
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
|
|
clear xv when going to NULL, remove // commented non-existant proto
|
|
* tests/examples/seek/seek.c: (main):
|
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add missing tooltip description for scrub and play_scrub
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|
|
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2006-11-14 23:34:19 +0000 David Schleef <ds@schleef.org>
|
|
|
|
configure.ac: Bump liboil requirement to 0.3.8.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Bump liboil requirement to 0.3.8.
|
|
* gst-libs/gst/riff/riff-media.c:
|
|
Add Dirac fourcc.
|
|
* gst/videoscale/vs_image.h:
|
|
* gst/videoscale/vs_scanline.h:
|
|
Use liboil's stdint.h.
|
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* gst/videotestsrc/videotestsrc.c:
|
|
Remove liboil related ifdef's, since they aren't needed now, and
|
|
won't work with future versions.
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|
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2006-11-14 23:08:38 +0000 David Schleef <ds@schleef.org>
|
|
|
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gst/videoscale/: Add a 4-tap image scaler. Theoretically looks much prettier.
|
|
Original commit message from CVS:
|
|
* gst/videoscale/Makefile.am:
|
|
* gst/videoscale/gstvideoscale.c:
|
|
* gst/videoscale/gstvideoscale.h:
|
|
* gst/videoscale/vs_4tap.c:
|
|
* gst/videoscale/vs_4tap.h:
|
|
* gst/videoscale/vs_image.c:
|
|
* gst/videoscale/vs_image.h:
|
|
* gst/videoscale/vs_scanline.c:
|
|
* gst/videoscale/vs_scanline.h:
|
|
Add a 4-tap image scaler. Theoretically looks much prettier.
|
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The tap calculation could use some improvement.
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|
|
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2006-11-14 11:54:14 +0000 Jan David Mol <j.j.d.mol@tudelft.nl>
|
|
|
|
Various gsize and gssize printf fixes. Fixes #372507.
|
|
Original commit message from CVS:
|
|
Patch by: Jan David Mol <j dot j dot d dot mol at tudelft dot nl>
|
|
* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_auds),
|
|
(gst_riff_parse_strf_iavs):
|
|
* gst/subparse/gstsubparse.c: (convert_encoding):
|
|
* gst/tcp/gstmultifdsink.c:
|
|
(gst_multi_fd_sink_handle_client_write):
|
|
* gst/tcp/gsttcp.c: (gst_tcp_socket_write), (gst_tcp_socket_read),
|
|
(gst_tcp_read_buffer), (gst_tcp_gdp_read_caps),
|
|
(gst_tcp_gdp_write_buffer), (gst_tcp_gdp_write_caps):
|
|
* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_render):
|
|
* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
|
|
(gst_ximagesink_ximage_new):
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
|
|
Various gsize and gssize printf fixes. Fixes #372507.
|
|
|
|
2006-11-13 18:14:48 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/vorbis/vorbisdec.*: First stab at vorbis reverse playback.
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
|
|
(vorbis_dec_push_forward), (vorbis_dec_push_reverse),
|
|
(vorbis_handle_data_packet), (vorbis_dec_decode_buffer),
|
|
(vorbis_dec_flush_decode), (vorbis_dec_chain_reverse),
|
|
(vorbis_dec_chain_forward), (vorbis_dec_chain):
|
|
* ext/vorbis/vorbisdec.h:
|
|
First stab at vorbis reverse playback.
|
|
|
|
2006-11-13 17:30:17 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.*: Make the clock sync code more accurate wrt resampling and playback at differen...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
|
|
* gst-libs/gst/audio/gstbaseaudiosink.h:
|
|
Make the clock sync code more accurate wrt resampling and playback
|
|
at different rates.
|
|
* gst-libs/gst/audio/gstringbuffer.c:
|
|
(gst_ring_buffer_commit_full), (gst_ring_buffer_commit):
|
|
* gst-libs/gst/audio/gstringbuffer.h:
|
|
Use better algorithm to interpolate sample rates.
|
|
|
|
2006-11-13 15:31:01 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
ext/ogg/gstoggdemux.c: Improve a debug line slightly.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page):
|
|
Improve a debug line slightly.
|
|
* ext/ogg/gstogmparse.c: (gst_ogm_parse_plugin_init):
|
|
Call gst_riff_init() in plugin_init, to avoid getting errors from
|
|
the debug system (unrelated changes to another plugin made this turn
|
|
up; not sure why).
|
|
|
|
2006-11-10 19:20:21 +0000 Sergey Scobich <sergery.scobich@gmail.com>
|
|
|
|
win32/common/libgsttag.def: Add missing symbol (#366492).
|
|
Original commit message from CVS:
|
|
Patch by: Sergey Scobich <sergery.scobich at gmail com>
|
|
* win32/common/libgsttag.def:
|
|
Add missing symbol (#366492).
|
|
|
|
2006-11-10 00:52:55 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/playback/gststreamselector.c: Don't unref a NULL pad.
|
|
Original commit message from CVS:
|
|
* gst/playback/gststreamselector.c: (gst_stream_selector_dispose):
|
|
Don't unref a NULL pad.
|
|
|
|
2006-11-09 00:50:00 +0000 Christian Schaller <uraeus@gnome.org>
|
|
|
|
ext/ogg/gstoggdemux.c: Implement first stab at reverse playback.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_page),
|
|
(gst_ogg_demux_get_prev_page), (gst_ogg_demux_perform_seek),
|
|
(gst_ogg_demux_handle_page), (gst_ogg_demux_chain),
|
|
(gst_ogg_demux_loop_forward), (gst_ogg_demux_loop_reverse),
|
|
(gst_ogg_demux_loop):
|
|
Implement first stab at reverse playback.
|
|
|
|
2006-11-07 07:22:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst-libs/gst/riff/riff-media.c: add h263/h264 variants to the caps, Fixes #363118
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
|
|
(gst_riff_create_video_template_caps):
|
|
add h263/h264 variants to the caps, Fixes #363118
|
|
|
|
2006-11-06 18:24:59 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/audio/: Use g_strerror instead of strerror so we get UTF-8.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func):
|
|
* gst-libs/gst/audio/gstaudiosrc.c: (audioringbuffer_thread_func):
|
|
Use g_strerror instead of strerror so we get UTF-8.
|
|
|
|
2006-11-04 07:25:58 +0000 David Schleef <ds@schleef.org>
|
|
|
|
ext/ogg/: Add/remove KW-DIRAC header here, since it is ogg-specific.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c:
|
|
* ext/ogg/gstoggmux.c:
|
|
Add/remove KW-DIRAC header here, since it is ogg-specific.
|
|
|
|
2006-11-03 15:44:31 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Recognise more mpeg4 elementary video streams.
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find):
|
|
Recognise more mpeg4 elementary video streams.
|
|
|
|
2006-11-02 17:26:03 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Lower the probability of mp3 typefinding functions if we don't find a valid mp3 ...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
|
|
Lower the probability of mp3 typefinding functions if we don't find a
|
|
valid mp3 header at the start of the file.
|
|
Closes #369482
|
|
|
|
2006-11-02 15:06:36 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/theora/: Document and partially implement an algorithm for doing reverse playback of theora video.
|
|
Original commit message from CVS:
|
|
* ext/theora/gsttheoradec.h:
|
|
* ext/theora/theoradec.c: (gst_theora_dec_init),
|
|
(theora_dec_sink_event), (theora_dec_chain_forward),
|
|
(theora_dec_flush_decode), (theora_dec_chain_reverse),
|
|
(theora_dec_chain):
|
|
Document and partially implement an algorithm for doing reverse playback
|
|
of theora video.
|
|
|
|
2006-11-02 14:18:45 +0000 Sergey Scobich <sergey.scobich@gmail.com>
|
|
|
|
win32/: Misc. VS8 build fixes: fix syntax in config.h, add missing entries to libgsttag.def; add missing dependencies...
|
|
Original commit message from CVS:
|
|
Patch by: Sergey Scobich <sergey.scobich at gmail com>
|
|
* win32/common/config.h:
|
|
* win32/common/interfaces-enumtypes.c:
|
|
* win32/common/libgsttag.def:
|
|
* win32/vs8/gst-plugins-base.sln:
|
|
* win32/vs8/libgstaudioresample.vcproj:
|
|
* win32/vs8/libgstinterfaces.vcproj:
|
|
* win32/vs8/libgstogg.vcproj:
|
|
* win32/vs8/libgstriff.vcproj:
|
|
* win32/vs8/libgsttag.vcproj:
|
|
* win32/vs8/libgsttheora.vcproj:
|
|
* win32/vs8/libgstvideoscale.vcproj:
|
|
* win32/vs8/libgstvorbis.vcproj:
|
|
Misc. VS8 build fixes: fix syntax in config.h, add missing entries
|
|
to libgsttag.def; add missing dependencies for some vs8 projects;
|
|
re-arrange placement of .def files in vs8 projects (#366334).
|
|
|
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2006-11-01 14:08:31 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/ogg/gstogg.c: Remove unused variable.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstogg.c:
|
|
Remove unused variable.
|
|
* ext/ogg/gstoggdemux.c:
|
|
Fix Wim's surname in plugin description.
|
|
|
|
2006-10-31 15:05:33 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-plugins-base.spec.in: spec new .h file. Fixes #368310.
|
|
Original commit message from CVS:
|
|
* gst-plugins-base.spec.in:
|
|
spec new .h file. Fixes #368310.
|
|
|
|
2006-10-31 14:19:07 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/tcp/gstmultifdsink.*: Make using the remove or clear signals threadsafe.
|
|
Original commit message from CVS:
|
|
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
|
|
(gst_multi_fd_sink_remove), (gst_multi_fd_sink_clear),
|
|
(gst_multi_fd_sink_get_stats),
|
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(gst_multi_fd_sink_remove_client_link),
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(gst_multi_fd_sink_queue_buffer),
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(gst_multi_fd_sink_handle_clients):
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* gst/tcp/gstmultifdsink.h:
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Make using the remove or clear signals threadsafe.
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Make calling get-stats with an invalid fd not segfault.
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Fixes 368273.
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2006-10-31 10:49:19 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/rtp/: Fix and activate base audio payloader.
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Original commit message from CVS:
|
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* gst-libs/gst/rtp/Makefile.am:
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* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
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(gst_base_rtp_audio_payload_init):
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Fix and activate base audio payloader.
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2006-10-28 17:22:57 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/typefind/gsttypefindfunctions.c: Add typefinder for QuickTime Image Files (see #366156).
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Original commit message from CVS:
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* gst/typefind/gsttypefindfunctions.c: (qtif_type_find),
|
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(plugin_init):
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Add typefinder for QuickTime Image Files (see #366156).
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2006-10-28 16:00:51 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/audioresample/gstaudioresample.c: Another typo fix (#366212).
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Original commit message from CVS:
|
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* gst/audioresample/gstaudioresample.c: (gst_audioresample_init):
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Another typo fix (#366212).
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2006-10-27 17:13:35 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/volume/gstvolume.c: Use stream time to synchronize volume property instead of rather random timestamps. This is n...
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|
Original commit message from CVS:
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* gst/volume/gstvolume.c: (volume_transform_ip):
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Use stream time to synchronize volume property instead of rather random
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timestamps. This is needed when gnonlin does its time shifting.
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2006-10-27 16:46:15 +0000 Wim Taymans <wim.taymans@gmail.com>
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* ChangeLog:
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I'm too lazy to comment this
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Original commit message from CVS:
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*** empty log message ***
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2006-10-27 16:45:30 +0000 Mark Nauwelaerts <manauw@skynet.be>
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ext/ogg/gstoggmux.c: Remove the pad from the element in release_pad.
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Original commit message from CVS:
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Patch by: Mark Nauwelaerts <manauw at skynet dot be>
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* ext/ogg/gstoggmux.c: (gst_ogg_mux_release_pad):
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Remove the pad from the element in release_pad.
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2006-10-27 11:57:18 +0000 Tim-Philipp Müller <tim@centricular.net>
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sys/: Explicitly create our custom buffer classes at a thread-safe location as well, since g_type_class_ref() doesn't...
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Original commit message from CVS:
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* sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
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Explicitly create our custom buffer classes at a thread-safe
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location as well, since g_type_class_ref() doesn't seem to be
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entirely thread-safe either (#365501; also see #349410).
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2006-10-26 10:49:00 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/riff/riff-read.c: If strings in INFO chunk are not UTF-8, do something similar to what we do for ID3v1 t...
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Original commit message from CVS:
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* gst-libs/gst/riff/riff-read.c: (freeform_string_to_utf8),
|
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(gst_riff_parse_info):
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If strings in INFO chunk are not UTF-8, do something similar to
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what we do for ID3v1 tags: check a number of environment variables
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(GST_AVI_TAG_ENCODING, GST_RIFF_TAG_ENCODING, GST_TAG_ENCODING) for
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character sets to try, otherwise try the current locale and/or fall
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back on ISO-8859-1. Fixes #360552.
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2006-10-23 12:46:41 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/videotestsrc/: Add a bunch of exciting new checkers patterns.
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Original commit message from CVS:
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* gst/videotestsrc/gstvideotestsrc.c:
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(gst_video_test_src_pattern_get_type),
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(gst_video_test_src_set_pattern):
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* gst/videotestsrc/gstvideotestsrc.h:
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* gst/videotestsrc/videotestsrc.c: (gst_video_test_src_checkers1),
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(gst_video_test_src_checkers2), (gst_video_test_src_checkers4),
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(gst_video_test_src_checkers8):
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* gst/videotestsrc/videotestsrc.h:
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Add a bunch of exciting new checkers patterns.
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2006-10-23 12:06:44 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/subparse/: Add support for TMPlayer-type subtitles (#362845).
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Original commit message from CVS:
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* gst/subparse/Makefile.am:
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* gst/subparse/gstsubparse.c:
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(gst_sub_parse_data_format_autodetect),
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(gst_sub_parse_format_autodetect), (handle_buffer),
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(gst_sub_parse_chain), (gst_subparse_type_find), (plugin_init):
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* gst/subparse/gstsubparse.h:
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* gst/subparse/tmplayerparse.c: (tmplayer_parse_line),
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(parse_tmplayer):
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* gst/subparse/tmplayerparse.h:
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Add support for TMPlayer-type subtitles (#362845).
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* tests/check/elements/subparse.c: (test_tmplayer_do_test),
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(GST_START_TEST), (subparse_suite):
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Add some basic unit tests for the above.
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2006-10-23 11:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
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tests/check/elements/audiorate.c: More tests for audiorate: inject buffers to check behaviour when buffers overlap.
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|
Original commit message from CVS:
|
|
* tests/check/elements/audiorate.c: (test_injector_base_init),
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(test_injector_class_init), (test_injector_chain),
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(test_injector_init), (probe_cb), (do_perfect_stream_test),
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(GST_START_TEST), (audiorate_suite):
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More tests for audiorate: inject buffers to check behaviour when
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buffers overlap.
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2006-10-21 16:39:54 +0000 Tim-Philipp Müller <tim@centricular.net>
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tests/check/: Add some basic unit tests for audiorate. Disabled at the moment since it doesn't pass yet (see bug #363...
|
|
Original commit message from CVS:
|
|
* tests/check/Makefile.am:
|
|
* tests/check/elements/.cvsignore:
|
|
* tests/check/elements/audiorate.c: (probe_cb), (got_buf),
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(do_perfect_stream_test), (GST_START_TEST), (audiorate_suite):
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Add some basic unit tests for audiorate. Disabled at the moment
|
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since it doesn't pass yet (see bug #363119).
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|
|
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2006-10-20 17:02:19 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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gst/subparse/gstsubparse.c: Add missing closing tags for markup and fix broken markup, otherwise pango won't render a...
|
|
Original commit message from CVS:
|
|
* gst/subparse/gstsubparse.c: (subrip_fix_up_markup),
|
|
(parse_subrip), (handle_buffer):
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Add missing closing tags for markup and fix broken markup,
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otherwise pango won't render anything (fixes #357531). Also,
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|
make sure the text we send out is always NUL-terminated
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(better safe than sorry etc.).
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* tests/check/elements/subparse.c: (test_srt_do_test),
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(test_srt):
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Some more tests for .srt incl. tests for the above stuff.
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2006-10-20 13:56:55 +0000 Stefan Kost <ensonic@users.sf.net>
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sys/: Try to redraw borders only when needed. Apparently this consumes resources on small devices... :-O (#363607)
|
|
Original commit message from CVS:
|
|
2006-10-20 Julien MOUTTE <julien@moutte.net>
|
|
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
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Patch by: Stefan Kost <ensonic@users.sf.net>
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Try to redraw borders only when needed. Apparently this consumes
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resources on small devices... :-O (#363607)
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|
|
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2006-10-20 13:54:19 +0000 Michael Smith <msmith@xiph.org>
|
|
|
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gst/tcp/gstmultifdsink.c: If caps change, then update the client's idea of the caps so that we don't end up re-sendin...
|
|
Original commit message from CVS:
|
|
* gst/tcp/gstmultifdsink.c:
|
|
(gst_multi_fd_sink_client_queue_buffer):
|
|
If caps change, then update the client's idea of the caps so that we
|
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don't end up re-sending streamheaders for every single buffer after
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the caps change.
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2006-10-20 12:31:02 +0000 Michael Smith <msmith@xiph.org>
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|
|
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ext/ogg/gstoggparse.c: Set caps on pushed buffers; fix up refcounting of caps objects.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggparse.c: (gst_ogg_parse_dispose),
|
|
(gst_ogg_parse_append_header), (gst_ogg_parse_chain):
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Set caps on pushed buffers; fix up refcounting of caps objects.
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|
|
|
2006-10-19 14:09:30 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Typefind mmsh header data packet to application/x-mmsh (#362625).
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (mmsh_type_find),
|
|
(plugin_init):
|
|
Typefind mmsh header data packet to application/x-mmsh (#362625).
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|
|
|
2006-10-19 09:17:48 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/: Add very simple unit test for subparse.
|
|
Original commit message from CVS:
|
|
* tests/check/Makefile.am:
|
|
* tests/check/elements/.cvsignore:
|
|
* tests/check/elements/subparse.c: (buffer_from_static_string),
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|
(setup_subparse), (teardown_subparse), (test_srt_do_test),
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(GST_START_TEST), (subparse_suite):
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|
Add very simple unit test for subparse.
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|
|
|
2006-10-19 09:00:21 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/subparse/gstsubparse.c: Strip trailing newlines from subtitle text output.
|
|
Original commit message from CVS:
|
|
* gst/subparse/gstsubparse.c: (strip_trailing_newlines),
|
|
(parse_subrip):
|
|
Strip trailing newlines from subtitle text output.
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|
|
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2006-10-18 18:40:12 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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gst/subparse/gstsubparse.c: Fix memleak; clear subparse->textbuf n state change function.
|
|
Original commit message from CVS:
|
|
* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
|
|
(gst_sub_parse_change_state):
|
|
Fix memleak; clear subparse->textbuf n state change function.
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|
|
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2006-10-18 15:13:59 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/subparse/gstsubparse.c: Don't require subrip (.srt) files to start with a chunk number of 1.
|
|
Original commit message from CVS:
|
|
* gst/subparse/gstsubparse.c:
|
|
(gst_sub_parse_data_format_autodetect):
|
|
Don't require subrip (.srt) files to start with a chunk number of 1.
|
|
|
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2006-10-18 13:42:49 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.*: Extract rate from the NEWSEGMENT event.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
|
|
* gst-libs/gst/audio/gstbaseaudiosink.h:
|
|
Extract rate from the NEWSEGMENT event.
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|
Use commit_full to also take rate adjustment into account when writing
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|
samples to the ringbuffer.
|
|
* gst-libs/gst/audio/gstringbuffer.c:
|
|
(gst_ring_buffer_commit_full), (gst_ring_buffer_commit),
|
|
(gst_ring_buffer_read):
|
|
* gst-libs/gst/audio/gstringbuffer.h:
|
|
Added _commit_full() to also take rate into account.
|
|
Use simple interpolation algorithm to resample audio.
|
|
API: gst_ring_buffer_commit_full()
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|
* tests/examples/seek/scrubby.c: (speed_cb), (do_seek):
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|
* tests/examples/seek/seek.c: (segment_done):
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|
Don't try to seek with 0.0 rate, just pause instead.
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Remove bogus debug line.
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|
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|
2006-10-18 12:57:54 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/playback/gstplaybasebin.c: Catch async errors when starting up the subtitle bin, so we can stop waiting and conti...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (subbin_startup_sync_msg),
|
|
(setup_source):
|
|
Catch async errors when starting up the subtitle bin, so we can
|
|
stop waiting and continue with the main film instead of hanging
|
|
forever. Fixes #339366.
|
|
* tests/check/elements/playbin.c: (playbin_suite):
|
|
Enable unit test for the above.
|
|
|
|
2006-10-18 09:53:03 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/: Some small and basic unit tests for playbin; not very useful yet, but at least a start.
|
|
Original commit message from CVS:
|
|
* tests/check/Makefile.am:
|
|
* tests/check/elements/.cvsignore:
|
|
* tests/check/elements/playbin.c: (GST_START_TEST),
|
|
(gst_red_video_src_uri_get_type),
|
|
(gst_red_video_src_uri_get_protocols),
|
|
(gst_red_video_src_uri_get_uri), (gst_red_video_src_uri_set_uri),
|
|
(gst_red_video_src_uri_handler_init),
|
|
(gst_red_video_src_init_type), (gst_red_video_src_base_init),
|
|
(gst_red_video_src_create), (gst_red_video_src_class_init),
|
|
(gst_red_video_src_init), (plugin_init), (playbin_suite):
|
|
Some small and basic unit tests for playbin; not very useful yet,
|
|
but at least a start.
|
|
|
|
2006-10-18 09:46:35 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/playback/gstplaybin.c: The old pad activation spiel.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin.c: (setup_sinks):
|
|
The old pad activation spiel.
|
|
|
|
2006-10-18 09:31:49 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/playback/gstplaybasebin.c: Don't hang forever if the subbin already fails to start up in the state change to PAUS...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (setup_source):
|
|
Don't hang forever if the subbin already fails to start up in
|
|
the state change to PAUSED (#339366).
|
|
|
|
2006-10-17 17:17:16 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/interfaces/tuner.c: Fix some function guards, add some more function guards.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/tuner.c: (gst_tuner_list_channels),
|
|
(gst_tuner_set_channel), (gst_tuner_get_channel),
|
|
(gst_tuner_list_norms), (gst_tuner_set_norm), (gst_tuner_get_norm),
|
|
(gst_tuner_set_frequency), (gst_tuner_get_frequency),
|
|
(gst_tuner_signal_strength), (gst_tuner_find_norm_by_name),
|
|
(gst_tuner_find_channel_by_name):
|
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Fix some function guards, add some more function guards.
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|
|
|
2006-10-17 11:34:32 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst/playback/gstdecodebin.c: Don't return a pad from get_our_ghost_pad unless it is actually the one we want.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (get_our_ghost_pad),
|
|
(remove_element_chain):
|
|
Don't return a pad from get_our_ghost_pad unless it is actually the
|
|
one we want.
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Change a cast in remove_element_chain slightly.
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|
|
|
2006-10-13 15:20:29 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
tests/examples/seek/seek.c: Segment seeking needs to use the rate and set stop to -1.
|
|
Original commit message from CVS:
|
|
2006-10-13 Julien MOUTTE <julien@moutte.net>
|
|
* tests/examples/seek/seek.c: (do_seek), (start_seek),
|
|
(rate_spinbutton_changed_cb), (segment_done),
|
|
(msg_state_changed):
|
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Segment seeking needs to use the rate and set stop to -1.
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|
|
|
2006-10-13 14:15:42 +0000 Ville Syrjala <ville.syrjala@movial.fi>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.c: Don't crash when ringbuffer is not yet created.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_setcaps):
|
|
Don't crash when ringbuffer is not yet created.
|
|
Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
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|
Fixes #361634.
|
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* gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
|
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* gst/playback/gststreamselector.c:
|
|
(gst_stream_selector_request_new_pad):
|
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Activate pads befre adding them to running elements.
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|
|
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2006-10-13 11:25:10 +0000 Julien Moutte <julien@moutte.net>
|
|
|
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tests/examples/seek/seek.c: Stop the scale updater when we start grabing the slider. Don't wait for the pipeline to b...
|
|
Original commit message from CVS:
|
|
2006-10-13 Julien MOUTTE <julien@moutte.net>
|
|
* tests/examples/seek/seek.c: (do_seek), (start_seek),
|
|
(rate_spinbutton_changed_cb), (msg_state_changed): Stop the
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scale
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updater when we start grabing the slider. Don't wait for the
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pipeline to be PAUSED.
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|
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2006-10-13 08:57:52 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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gst-libs/gst/interfaces/mixer.c: Guard mixer interface functions against bogus arguments.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/mixer.c: (gst_mixer_list_tracks),
|
|
(gst_mixer_set_volume), (gst_mixer_get_volume),
|
|
(gst_mixer_set_mute), (gst_mixer_set_option),
|
|
(gst_mixer_get_option), (gst_mixer_mute_toggled),
|
|
(gst_mixer_record_toggled), (gst_mixer_volume_changed),
|
|
(gst_mixer_option_changed):
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Guard mixer interface functions against bogus arguments.
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|
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2006-10-12 19:39:07 +0000 Julien Moutte <julien@moutte.net>
|
|
|
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tests/examples/seek/seek.c: Use state-changed messages to trigger start/stop of scale update timer. Indeed the scale ...
|
|
Original commit message from CVS:
|
|
2006-10-12 Julien MOUTTE <julien@moutte.net>
|
|
* tests/examples/seek/seek.c: (do_seek), (start_seek),
|
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(stop_seek),
|
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(play_cb), (pause_cb), (stop_cb),
|
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(rate_spinbutton_changed_cb),
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(msg_state_changed), (main): Use state-changed messages to
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trigger
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start/stop of scale update timer. Indeed the scale slider was
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jumping here and there because the update timer was activated
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before seek completed. This fixes instant applying of rate
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changes
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by pressing the spinbutton like a crazy man !
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2006-10-12 19:09:06 +0000 Sebastien Cote <sebas642@yahoo.ca>
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|
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gst-libs/gst/rtp/gstbasertppayload.c: Fix two small memory leaks (#361456).
|
|
Original commit message from CVS:
|
|
Patch by: Sebastien Cote <sebas642 at yahoo.ca>
|
|
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_init),
|
|
(gst_basertppayload_finalize):
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|
Fix two small memory leaks (#361456).
|
|
|
|
2006-10-10 18:56:01 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
tests/examples/seek/seek.c: When changing spinbutton we try to change the rate on the fly.
|
|
Original commit message from CVS:
|
|
2006-10-10 Julien MOUTTE <julien@moutte.net>
|
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* tests/examples/seek/seek.c: (do_seek),
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|
(rate_spinbutton_changed_cb): When changing spinbutton we try
|
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to change the rate on the fly.
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|
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2006-10-10 16:50:06 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
|
gst-libs/gst/riff/: Add WMS caps.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-ids.h:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
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|
(gst_riff_create_audio_template_caps):
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Add WMS caps.
|
|
|
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2006-10-10 12:49:03 +0000 Josep Torre Valles <josep@fluendo.com>
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|
|
|
ext/gnomevfs/: Fix URI interface implementation return type.
|
|
Original commit message from CVS:
|
|
2006-10-10 Zaheer Abbas Merali <zaheerabbas at merali dot org>
|
|
Patch by: Josep Torre Valles <josep@fluendo.com>
|
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* ext/gnomevfs/gstgnomevfssink.c:
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|
* ext/gnomevfs/gstgnomevfssrc.c:
|
|
Fix URI interface implementation return type.
|
|
* ext/pango/gsttextoverlay.c: (gst_text_overlay_set_property):
|
|
Fix what looks like a copy/paste issue when assigning values.
|
|
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
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|
(gst_audio_filter_template_get_type):
|
|
Cast to prevent Forte warnings.
|
|
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
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|
Fix URI interface implementation return type.
|
|
gst_pad_query_position requires a signed integer pointer as
|
|
3rd parameter, GstClockTime is unsigned.
|
|
* gst/audioconvert/audioconvert.c:
|
|
Fix integer overflow when treated as signed.
|
|
* gst/audioresample/resample.c: (resample_add_input_data):
|
|
Cast to prevent warnings on Forte.
|
|
* gst/ffmpegcolorspace/imgconvert.c: (build_rgb_palette):
|
|
Fix integer overflow when treated as signed.
|
|
* gst/ffmpegcolorspace/imgconvert_template.h:
|
|
Fix integer overflow when treated as signed. RGBA_OUT shifts bits.
|
|
* gst/playback/gstdecodebin.c: (queue_filled_cb),
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|
(cleanup_decodebin):
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|
Who initialises a guint to -1!
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|
Cast function pointers to prevent warnings on Forte.
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* gst/playback/gstplaybasebin.c: (queue_deadlock_check),
|
|
(queue_threshold_reached):
|
|
Cast function pointers correctly to prevent warnings on Forte.
|
|
* gst/playback/gststreaminfo.c: (gst_stream_info_dispose):
|
|
Cast function pointers correctly to prevent warnings on Forte.
|
|
* gst/subparse/gstssaparse.c: (gst_ssa_parse_setcaps):
|
|
Obvious change to unsigned, 0xEF > max signed char.
|
|
* gst/tcp/gstmultifdsink.c: (get_buffers_max), (count_burst_unit):
|
|
GstClockTime is unsigned, initialise correctly.
|
|
* gst/tcp/gsttcp.c: (gst_tcp_socket_write):
|
|
Cast so pointer arithemetic doesn't cause warnings on Forte.
|
|
* gst/videorate/gstvideorate.c:
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|
Use correct return value.
|
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* tests/examples/seek/scrubby.c:
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GstClockTime is unsigned, initialise correctly.
|
|
|
|
2006-10-10 11:20:03 +0000 Ferenc Gerlits <fgerlits@gmail.com>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Recognise XML files and XML-like files shorter than 256 bytes as well (fixes #35...
|
|
Original commit message from CVS:
|
|
Patch by: Ferenc Gerlits <fgerlits at gmail com>
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
Recognise XML files and XML-like files shorter than 256 bytes as
|
|
well (fixes #359237).
|
|
|
|
2006-10-09 15:01:30 +0000 Edgard Lima <edgard.lima@indt.org.br>
|
|
|
|
* ChangeLog:
|
|
* common:
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
Added typefind functions to video/x-nuv media.
|
|
Original commit message from CVS:
|
|
Added typefind functions to video/x-nuv media.
|
|
|
|
2006-10-08 16:59:31 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/interfaces/xoverlay.c: Some more guards against invalid input.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/xoverlay.c:
|
|
(gst_x_overlay_set_xwindow_id), (gst_x_overlay_expose):
|
|
Some more guards against invalid input.
|
|
|
|
2006-10-07 18:35:39 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
ext/pango/gsttextoverlay.c: Useless goto.
|
|
Original commit message from CVS:
|
|
2006-10-07 Julien MOUTTE <julien@moutte.net>
|
|
* ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event):
|
|
Useless goto.
|
|
* tests/examples/seek/seek.c: (do_seek),
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|
(rate_spinbutton_changed_cb), (main): Add a rate spinbutton in
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|
seek example to experiment with rates != 1.0 (reverse playback
|
|
!)
|
|
|
|
2006-10-06 19:20:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst-libs/gst/interfaces/xoverlay.c: Unref message in doc-example (spotted by Robert McQueen)
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/xoverlay.c:
|
|
Unref message in doc-example (spotted by Robert McQueen)
|
|
|
|
2006-10-06 17:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: printf fix.
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
|
|
(mpeg1_parse_header), (mpeg1_sys_type_find):
|
|
printf fix.
|
|
|
|
2006-10-06 14:37:43 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/: Activate dynamic pads before adding them to the element.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
|
|
(close_pad_link):
|
|
* gst/playback/gstplaybasebin.c: (new_decoded_pad_full):
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Activate dynamic pads before adding them to the element.
|
|
|
|
2006-10-06 14:04:53 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst-libs/gst/floatcast/floatcast.h: Fix obviously-bogus macros; use the correct types.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/floatcast/floatcast.h:
|
|
Fix obviously-bogus macros; use the correct types.
|
|
|
|
2006-10-06 13:34:46 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstbasertpdepayload.c: Also call parent state change function to activate pads.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_change_state):
|
|
Also call parent state change function to activate pads.
|
|
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
|
|
(mpeg1_parse_header), (mpeg1_sys_type_find):
|
|
Add some more debug info in mpeg typefinding.
|
|
|
|
2006-10-06 12:57:10 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
ext/theora/theoradec.c: Zero byte theora packets are valid and well-defined; don't warn on them.
|
|
Original commit message from CVS:
|
|
* ext/theora/theoradec.c: (theora_dec_chain):
|
|
Zero byte theora packets are valid and well-defined; don't warn on
|
|
them.
|
|
|
|
2006-10-06 10:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
gst/tcp/gstmultifdsink.c: API: add dropped_buffers to the get-stats GValueArray
|
|
Original commit message from CVS:
|
|
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
|
|
(gst_multi_fd_sink_get_stats), (find_limits),
|
|
(gst_multi_fd_sink_queue_buffer):
|
|
API: add dropped_buffers to the get-stats GValueArray
|
|
|
|
2006-10-05 15:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
Printf format fixes.
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsadeviceprobe.c:
|
|
(gst_alsa_device_property_probe_get_values):
|
|
* ext/alsa/gstalsasink.c: (set_hwparams):
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad),
|
|
(gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain):
|
|
* ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers),
|
|
(gst_ogg_mux_process_best_pad):
|
|
* ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream),
|
|
(gst_ogg_parse_chain):
|
|
* ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header):
|
|
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
|
|
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup),
|
|
(gst_vorbis_enc_buffer_check_discontinuous):
|
|
* ext/vorbis/vorbisparse.c: (vorbis_parse_src_query):
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_render):
|
|
* gst-libs/gst/cdda/gstcddabasesrc.c:
|
|
(gst_cdda_base_src_handle_track_seek):
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_push_full):
|
|
* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
|
|
* gst/audioresample/resample.c: (resample_input_pushthrough):
|
|
* gst/playback/gstplaybasebin.c: (queue_out_of_data):
|
|
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
|
|
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
|
|
(wavpack_type_find):
|
|
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
|
|
(gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
|
|
* tests/check/elements/volume.c: (GST_START_TEST):
|
|
Printf format fixes.
|
|
|
|
2006-10-04 13:18:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
gst/tcp/gsttcp.c: Fix a simple mistake (see the docs)
|
|
Original commit message from CVS:
|
|
* gst/tcp/gsttcp.c: (gst_tcp_gdp_read_caps):
|
|
Fix a simple mistake (see the docs)
|
|
Fixes #359580
|
|
|
|
2006-10-04 13:15:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* win32/common/config.h:
|
|
bump version
|
|
Original commit message from CVS:
|
|
bump version
|
|
|
|
2006-10-03 12:11:45 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
docs/plugins/: Add vorbistag element to docs; update version numbers to 0.10.10.1.
|
|
Original commit message from CVS:
|
|
* docs/plugins/Makefile.am:
|
|
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.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-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gdp.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.xml:
|
|
* docs/plugins/inspect/plugin-libvisual.xml:
|
|
* docs/plugins/inspect/plugin-ogg.xml:
|
|
* docs/plugins/inspect/plugin-pango.xml:
|
|
* docs/plugins/inspect/plugin-playbin.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-video4linux.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:
|
|
Add vorbistag element to docs; update version numbers to 0.10.10.1.
|
|
|
|
2006-10-03 11:51:48 +0000 James Doc Livingston <doclivingston@gmail.com>
|
|
|
|
ext/vorbis/: Add new vorbistag element which derives from vorbisparse and is essentially the same as well, only that ...
|
|
Original commit message from CVS:
|
|
Patch by: James "Doc" Livingston <doclivingston at gmail com>
|
|
* ext/vorbis/Makefile.am:
|
|
* ext/vorbis/vorbis.c: (plugin_init):
|
|
* ext/vorbis/vorbisparse.c: (gst_vorbis_parse_class_init),
|
|
(vorbis_parse_parse_packet), (vorbis_parse_chain):
|
|
* ext/vorbis/vorbisparse.h:
|
|
* ext/vorbis/vorbistag.c: (gst_vorbis_tag_base_init),
|
|
(gst_vorbis_tag_class_init), (gst_vorbis_tag_init),
|
|
(gst_vorbis_tag_parse_packet):
|
|
* ext/vorbis/vorbistag.h:
|
|
Add new vorbistag element which derives from vorbisparse
|
|
and is essentially the same as well, only that it implements
|
|
the GstTagSetter interface and can modify the stream's
|
|
vorbiscomment on the fly (#335635).
|
|
* tests/check/Makefile.am:
|
|
* tests/check/elements/.cvsignore:
|
|
* tests/check/elements/vorbistag.c: (setup_vorbistag),
|
|
(cleanup_vorbistag), (buffer_probe), (start_pipeline),
|
|
(get_buffer), (stop_pipeline), (_create_codebook_header_buffer),
|
|
(_create_audio_buffer), (GST_START_TEST), (vorbistag_suite):
|
|
Add unit test for new vorbistag element.
|
|
|
|
2006-10-03 10:36:38 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/vorbis/vorbisparse.c: Set BOS flag in packet structure to fix 'jump depends on unitialized value' errors in valgr...
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisparse.c: (gst_vorbis_parse_init),
|
|
(vorbis_parse_push_headers), (vorbis_parse_chain):
|
|
Set BOS flag in packet structure to fix 'jump depends
|
|
on unitialized value' errors in valgrind; various minor
|
|
clean-ups.
|
|
|
|
2006-09-30 15:30:07 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst/playback/gstdecodebin.c: Fix typo in a debug statement.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (close_pad_link):
|
|
Fix typo in a debug statement.
|
|
* gst/playback/gstplaybasebin.c: (probe_triggered),
|
|
(new_decoded_pad_full), (new_decoded_pad), (subs_new_decoded_pad),
|
|
(gen_source_element), (source_new_pad), (analyse_source),
|
|
(setup_source):
|
|
When handling no_more_pads in new_decoded_pad, make sure to treat
|
|
subtitle pads correctly. Fixes playback with subtitle files.
|
|
Move a recurring message to LOG level.
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
|
|
The maximum value for the Xv colorkey on this Radeon is 0xFFFFFFFF,
|
|
which ends up as -1 when cast to an int. Make the logic handle the
|
|
max value as an unsigned mask and only change the colorkey when it's
|
|
a value we recognise.
|
|
|
|
2006-09-30 00:14:20 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
|
|
|
|
gst-libs/gst/rtp/gstbasertpaudiopayload.c: Removed empty * between paragraphs
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
Removed empty * between paragraphs
|
|
|
|
2006-09-29 23:50:53 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
|
|
|
|
gst-libs/gst/rtp/: Moved some documentation into .c file
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
* gst-libs/gst/rtp/README:
|
|
Moved some documentation into .c file
|
|
|
|
2006-09-29 17:35:01 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstdecodebin.c: Fix compilation.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (no_more_pads):
|
|
Fix compilation.
|
|
|
|
2006-09-29 16:04:05 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstdecodebin.c: Remove g_print
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (new_caps):
|
|
Remove g_print
|
|
* gst/playback/gstplaybin.c:
|
|
Add some docs.
|
|
|
|
2006-09-29 15:16:32 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/Makefile.am: Re-enable cddabasesrc test to see if it works again now.
|
|
Original commit message from CVS:
|
|
* tests/check/Makefile.am:
|
|
Re-enable cddabasesrc test to see if it works again
|
|
now.
|
|
|
|
2006-09-29 13:46:45 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybasebin.c: Handle invalid URIs a bit more gracefully.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (setup_subtitle),
|
|
(gen_source_element):
|
|
Handle invalid URIs a bit more gracefully.
|
|
|
|
2006-09-29 12:54:28 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/pipelines/oggmux.c: Remove obsolete comment.
|
|
Original commit message from CVS:
|
|
* tests/check/pipelines/oggmux.c:
|
|
Remove obsolete comment.
|
|
|
|
2006-09-29 10:43:05 +0000 James Doc Livingston <doclivingston@gmail.com>
|
|
|
|
ext/ogg/gstoggmux.c: Commit patch from James "Doc" Livingston, adds proper EOS handling in oggmux. GStreamer can, for...
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggmux.c: (gst_ogg_mux_request_new_pad),
|
|
(gst_ogg_mux_release_pad), (gst_ogg_mux_push_buffer),
|
|
(gst_ogg_mux_compare_pads), (gst_ogg_mux_queue_pads),
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(gst_ogg_mux_send_headers), (gst_ogg_mux_process_best_pad),
|
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(gst_ogg_mux_collected):
|
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Commit patch from James "Doc" Livingston, adds proper EOS handling
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in oggmux. GStreamer can, for the first time ever, create a valid
|
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Ogg file! Yay!
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* tests/check/pipelines/oggmux.c: (check_chain_final_state),
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(oggmux_suite):
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Reenable tests now that they pass.
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|
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2006-09-29 08:20:03 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/tcp/gstmultifdsink.c: Stop reading commands when EOF (we read 0) as well.
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|
Original commit message from CVS:
|
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* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients):
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Stop reading commands when EOF (we read 0) as well.
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2006-09-28 15:29:17 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst/playback/gstdecodebin.c: Implement delayed caps linking needed for element with a lot of different caps on the sr...
|
|
Original commit message from CVS:
|
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* gst/playback/gstdecodebin.c: (dynamic_create), (dynamic_free),
|
|
(close_pad_link), (dynamic_remove), (no_more_pads), (new_caps),
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|
(find_dynamic), (unlinked), (close_link):
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Implement delayed caps linking needed for element with a lot of
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different caps on the src pads that get fixed at runtime.
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Improve management of dynamic elements.
|
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* gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
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(group_destroy), (group_commit), (check_queue), (queue_overrun),
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(gen_preroll_element), (remove_groups), (unknown_type),
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(add_element_stream), (no_more_pads_full), (no_more_pads),
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(sub_no_more_pads), (source_no_more_pads), (preroll_unlinked),
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(new_decoded_pad), (setup_subtitle), (array_has_value),
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(gen_source_element), (source_new_pad), (has_all_raw_caps),
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(analyse_source), (remove_decoders), (make_decoder),
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(remove_source), (setup_source), (finish_source), (prepare_output),
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(gst_play_base_bin_change_state):
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* gst/playback/gstplaybasebin.h:
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Use more _CAST instead of full type checking casts.
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Small cleanups, plug some leaks.
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Handle dynamic sources.
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Add some helper functions to create lists of strings used for
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blacklisting and other stuff.
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Refactor some code dealing with analysing the source.
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Re-enable sources without pads (like cd:// or other selfcontained
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elements).
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2006-09-28 15:08:15 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
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gst-libs/gst/audio/gstbaseaudiosink.c: When we have a timestamp, we can still perform clipping.
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|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_render):
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When we have a timestamp, we can still perform clipping.
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When we have no clock, we must play the sample ASAP.
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|
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2006-09-28 11:46:26 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
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gst/audiorate/gstaudiorate.c: Set caps on outgoing buffers.
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|
Original commit message from CVS:
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|
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
|
|
Set caps on outgoing buffers.
|
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* gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
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(gst_video_rate_event), (gst_video_rate_chain):
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* gst/videorate/gstvideorate.h:
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Fix videorate some more. Fixes #357977
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2006-09-28 11:34:05 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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tests/check/elements/adder.c: Don't set timeout to 6 seconds when we're running in valgrind ... (and how is 6 seconds...
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|
Original commit message from CVS:
|
|
* tests/check/elements/adder.c: (adder_suite):
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Don't set timeout to 6 seconds when we're running
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|
in valgrind ... (and how is 6 seconds longer than
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the default anyway?)
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2006-09-28 10:49:56 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/audiorate/gstaudiorate.c: Keep sink and src segment to keep track of time and support more input formats.
|
|
Original commit message from CVS:
|
|
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
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|
(gst_audio_rate_sink_event), (gst_audio_rate_convert),
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(gst_audio_rate_convert_segments), (gst_audio_rate_chain):
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Keep sink and src segment to keep track of time and support more
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|
input formats.
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Fix bogus next_offset and run_time calculation, don't understand how
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this could have worked before. Fixes #357976.
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Remove some unneeded vars.
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2006-09-28 09:41:20 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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gst/playback/gstplaybin.c: Only remove visualisation from visbin if there is a visbin (or: don't throw warnings when ...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin.c: (remove_sinks):
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Only remove visualisation from visbin if there is a visbin (or:
|
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don't throw warnings when closing totem without playing a file).
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|
|
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2006-09-27 13:52:14 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst-libs/gst/audio/gstbaseaudiosink.c: Add some more info in a WARNING.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_render):
|
|
Add some more info in a WARNING.
|
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* gst-libs/gst/audio/gstbaseaudiosrc.c:
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(gst_base_audio_src_create):
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Handle PAUSE in create function, use new -core addition to
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|
wait for playing. Fixes pausing and resuming capture from an
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|
audiosrc.
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* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
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(gst_ring_buffer_read):
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Constify some more.
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|
Caller supports interrupted reads now.
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|
|
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2006-09-27 13:29:49 +0000 Christian Schaller <uraeus@gnome.org>
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|
|
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* gst-plugins-base.spec.in:
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add new header file to spec
|
|
Original commit message from CVS:
|
|
add new header file to spec
|
|
|
|
2006-09-27 12:55:45 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/Makefile.am: Another attempt to make the gen64 buildbot happy.
|
|
Original commit message from CVS:
|
|
* tests/check/Makefile.am:
|
|
Another attempt to make the gen64 buildbot happy.
|
|
|
|
2006-09-27 11:58:17 +0000 Jonathan Matthew <jonathan@kaolin.wh9.net>
|
|
|
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ext/libvisual/visual.c: Libvisual plugin was not passing audio data to libvisual 0.4.0 correctly. Fixes #357800
|
|
Original commit message from CVS:
|
|
Patch by: Jonathan Matthew <jonathan@kaolin.wh9.net>
|
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* ext/libvisual/visual.c: (gst_visual_clear_actors),
|
|
(gst_visual_chain), (gst_visual_change_state):
|
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Libvisual plugin was not passing audio data to libvisual 0.4.0
|
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correctly. Fixes #357800
|
|
|
|
2006-09-27 11:31:43 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/pipelines/simple-launch-lines.c: Add timeout to _get_state() so we see which pipeline it is that causes t...
|
|
Original commit message from CVS:
|
|
* tests/check/pipelines/simple-launch-lines.c: (run_pipeline):
|
|
Add timeout to _get_state() so we see which pipeline it is
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that causes trouble on the gen64 build bot.
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|
|
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2006-09-27 11:06:54 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstbasertpdepayload.c: the source pad always uses fixed caps.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_init), (gst_base_rtp_depayload_push_full),
|
|
(gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_process),
|
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(gst_base_rtp_depayload_set_gst_timestamp):
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the source pad always uses fixed caps.
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|
|
|
2006-09-27 11:05:08 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
Added docs for the audio libs.
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-docs.sgml:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/audio/gstaudioclock.c:
|
|
* gst-libs/gst/audio/gstaudioclock.h:
|
|
* gst-libs/gst/audio/gstaudiosink.c:
|
|
* gst-libs/gst/audio/gstaudiosink.h:
|
|
* gst-libs/gst/audio/gstaudiosrc.c:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_render):
|
|
* gst-libs/gst/audio/gstbaseaudiosink.h:
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init):
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.h:
|
|
* gst-libs/gst/audio/gstringbuffer.h:
|
|
Added docs for the audio libs.
|
|
|
|
2006-09-27 10:59:24 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/Makefile.am: Temporarily disable test that fails on the bots for unknown reasons.
|
|
Original commit message from CVS:
|
|
* tests/check/Makefile.am:
|
|
Temporarily disable test that fails on the bots for unknown reasons.
|
|
|
|
2006-09-27 00:13:29 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
|
|
|
|
gst-libs/gst/rtp/gstbasertpaudiopayload.*: Moved AudioCodecType into priv
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
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|
Moved AudioCodecType into priv
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|
Renamed all gst_basertpaudiopayload to gst_base_rtp_audio_payload prefixes
|
|
|
|
2006-09-25 15:47:25 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstdecodebin.c: Cleanups and small leak fixes.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
|
|
(add_fakesink), (remove_fakesink), (pad_probe), (close_pad_link),
|
|
(is_demuxer_element), (try_to_link_1), (get_our_ghost_pad),
|
|
(new_pad):
|
|
Cleanups and small leak fixes.
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|
Added Depayloaders to valid list of autopluggable elements.
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|
|
|
2006-09-25 13:24:59 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybin.c: Detect NO_PREROLL state change returns and disable clock distribution to the sinks so that...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
|
|
(gst_play_bin_vis_blocked), (gst_play_bin_set_property),
|
|
(gen_video_element), (gen_text_element), (gen_audio_element),
|
|
(gen_vis_element), (remove_sinks), (add_sink), (setup_sinks),
|
|
(gst_play_bin_set_clock_func), (gst_play_bin_change_state):
|
|
Detect NO_PREROLL state change returns and disable clock distribution to
|
|
the sinks so that sync is disabled.
|
|
Avoid some type checking and do simple casts instead.
|
|
Small cleanups, fix some FIXMEs.
|
|
Be more robust when linking user specified elements, catch an report
|
|
errors. Fixes #357404.
|
|
Fix some leaks in the error paths.
|
|
|
|
2006-09-25 12:55:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
* ChangeLog:
|
|
ChangeLog surgery for missing bug-number
|
|
Original commit message from CVS:
|
|
ChangeLog surgery for missing bug-number
|
|
|
|
2006-09-25 11:28:15 +0000 Peter Kjellerstedt <pkj@axis.com>
|
|
|
|
gst/playback/test.c: Fix compilation with uClibc and -Werror (#357591).
|
|
Original commit message from CVS:
|
|
Patch by: Peter Kjellerstedt <pkj at axis com>
|
|
* gst/playback/test.c:
|
|
Fix compilation with uClibc and -Werror (#357591).
|
|
|
|
2006-09-25 10:21:31 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/tag/gstvorbistag.c: Parse dates that are followed by a time as well (#357532).
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
|
|
Parse dates that are followed by a time as well (#357532).
|
|
* tests/check/libs/tag.c: (test_vorbis_tags):
|
|
Add unit test for this.
|
|
|
|
2006-09-23 15:24:55 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/: A few array const-ifications.
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
|
|
(gst_audio_convert_transform_caps):
|
|
* gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor):
|
|
* gst/videotestsrc/videotestsrc.h:
|
|
A few array const-ifications.
|
|
|
|
2006-09-23 15:02:51 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/Makefile.am: See if this makes the build bots happy.
|
|
Original commit message from CVS:
|
|
* tests/check/Makefile.am:
|
|
See if this makes the build bots happy.
|
|
* tests/check/libs/cddabasesrc.c:
|
|
UTF8-ise my name.
|
|
|
|
2006-09-23 14:30:53 +0000 Young-Ho Cha <ganadist@chollian.net>
|
|
|
|
gst/subparse/samiparse.c: More case-insensitivity for certain tags; recognise entities with decimal codes as special ...
|
|
Original commit message from CVS:
|
|
Patch by: Young-Ho Cha <ganadist at chollian dot net>
|
|
* gst/subparse/samiparse.c: (handle_start_font),
|
|
(fix_invalid_entities):
|
|
More case-insensitivity for certain tags; recognise entities with
|
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decimal codes as special entities as well (#357330).
|
|
|
|
2006-09-23 13:32:07 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/Makefile.am: Need to build tag directory before cdda.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/Makefile.am:
|
|
Need to build tag directory before cdda.
|
|
|
|
2006-09-23 13:21:07 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc depend on libgsttag. This is required so we can ex...
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/cdda/Makefile.am:
|
|
* gst-libs/gst/cdda/gstcddabasesrc.c:
|
|
(gst_cdda_base_src_base_init):
|
|
* gst-libs/gst/cdda/gstcddabasesrc.h:
|
|
* gst-libs/gst/tag/tag.h:
|
|
* gst-libs/gst/tag/tags.c: (gst_tag_register_tags_internal),
|
|
(gst_tag_register_musicbrainz_tags):
|
|
Move GST_TAG_CDDA_* tags into libgsttag and make libgstcddabasesrc
|
|
depend on libgsttag. This is required so we can extract/read tags like
|
|
DISCID without depending on libgstcddabasesrc (which used to register
|
|
them).
|
|
* gst-libs/gst/tag/gstvorbistag.c:
|
|
Add vorbiscomment mapping for CDDB_DISCID and MUSICBRAINZ_DISCID
|
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tags (also see #347848).
|
|
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1):
|
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Log vorbis comments we are actually writing. Const-ify array.
|
|
|
|
2006-09-23 08:53:30 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybasebin.c: Improve buffering a bit by avoiding a deadlock because we cannot assume the underrun i...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (gen_preroll_element):
|
|
Improve buffering a bit by avoiding a deadlock because we cannot assume
|
|
the underrun is always called.
|
|
|
|
2006-09-23 08:51:14 +0000 Young-Ho Cha <ganadist@chollian.net>
|
|
|
|
gst-libs/gst/riff/: Added MPEG-4 AAC and id and caps. Fixes #357289
|
|
Original commit message from CVS:
|
|
Patch by: Young-Ho Cha <ganadist at chollian dot net>
|
|
* gst-libs/gst/riff/riff-ids.h:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
|
|
(gst_riff_create_audio_template_caps):
|
|
Added MPEG-4 AAC and id and caps. Fixes #357289
|
|
Added WMA9 Lossless id.
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|
|
|
2006-09-22 14:50:01 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/gnomevfs/gstgnomevfssrc.c: Fix misleading docs addition.
|
|
Original commit message from CVS:
|
|
* ext/gnomevfs/gstgnomevfssrc.c:
|
|
Fix misleading docs addition.
|
|
* tests/check/elements/videotestsrc.c: (check_rgb_buf):
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Get rid of compiler warning the right way.
|
|
|
|
2006-09-22 14:13:34 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/rtp/gstbasertpdepayload.*: Small cleanups.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_finalize),
|
|
(gst_base_rtp_depayload_setcaps), (gst_base_rtp_depayload_chain),
|
|
(gst_base_rtp_depayload_push_full),
|
|
(gst_base_rtp_depayload_push_ts), (gst_base_rtp_depayload_push),
|
|
(gst_base_rtp_depayload_process),
|
|
(gst_base_rtp_depayload_set_gst_timestamp),
|
|
(gst_base_rtp_depayload_queue_release):
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.h:
|
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Small cleanups.
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|
Fix some leaks.
|
|
Refactored the process method and added methods to push from the process
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|
vmethod.
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Use _scale functions.
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API: gst_base_rtp_depayload_push_ts
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API: gst_base_rtp_depayload_push
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* gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push):
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timestamps are uint.
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|
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2006-09-22 11:59:00 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst-libs/gst/interfaces/xoverlay.c: Remove unused statement from doc example.
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|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/xoverlay.c:
|
|
Remove unused statement from doc example.
|
|
|
|
2006-09-22 09:52:21 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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|
|
|
* gst/videorate/gstvideorate.c:
|
|
update docs
|
|
Original commit message from CVS:
|
|
update docs
|
|
|
|
2006-09-21 13:49:47 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
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gst-libs/gst/interfaces/videoorientation.c: Add since tags to new API docs, ChangeLog surgery (forgot API keyword in ...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/videoorientation.c:
|
|
(gst_video_orientation_iface_init),
|
|
(gst_video_orientation_get_hflip),
|
|
(gst_video_orientation_get_vflip),
|
|
(gst_video_orientation_get_hcenter),
|
|
(gst_video_orientation_get_vcenter),
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|
(gst_video_orientation_set_hflip),
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|
(gst_video_orientation_set_vflip),
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|
(gst_video_orientation_set_hcenter),
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|
(gst_video_orientation_set_vcenter):
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|
Add since tags to new API docs, ChangeLog surgery (forgot API keyword
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|
in ChangeLog)
|
|
|
|
2006-09-21 09:27:47 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/: but disable for now since it doesn't pass (something wrong with
|
|
Original commit message from CVS:
|
|
* tests/check/Makefile.am:
|
|
* tests/check/elements/.cvsignore:
|
|
* tests/check/elements/ffmpegcolorspace.c: (rgb_format_to_caps),
|
|
(create_rgb_conversions), (rgb_conversion_free),
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|
(right_shift_colour), (fix_expected_colour), (check_rgb_buf),
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|
(got_buf_cb), (GST_START_TEST), (ffmpegcolorspace_suite):
|
|
Add unit test for ffmpegcolorspace (RGB <=> RGB only so far),
|
|
but disable for now since it doesn't pass (something wrong with
|
|
RGBA somewhere).
|
|
|
|
2006-09-21 07:01:48 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybasebin.c: Refactor handling of overrun detection.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (group_commit),
|
|
(queue_deadlock_check), (queue_overrun), (queue_threshold_reached),
|
|
(queue_out_of_data), (gen_preroll_element),
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|
(preroll_remove_overrun), (probe_triggered):
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|
Refactor handling of overrun detection.
|
|
Separate handling of group completion and deadlock detection when doing
|
|
network buffering. This should fix some deadlocks that were not detected
|
|
because the group was completed.
|
|
Add more comments, improve debugging.
|
|
|
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2006-09-21 05:31:00 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
tests/check/: Some more compilation fixes.
|
|
Original commit message from CVS:
|
|
* tests/check/elements/gdpdepay.c: (GST_START_TEST):
|
|
* tests/check/libs/audio.c:
|
|
Some more compilation fixes.
|
|
|
|
2006-09-21 05:12:18 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstringbuffer.c: Early morning compilation fix.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstringbuffer.c:
|
|
(gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
|
|
(gst_ring_buffer_read):
|
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Early morning compilation fix.
|
|
|
|
2006-09-20 18:09:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
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* configure.ac:
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|
bump nano
|
|
Original commit message from CVS:
|
|
bump nano
|
|
|
|
2006-09-20 17:04:57 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
tests/check/: Fix some warnings.
|
|
Original commit message from CVS:
|
|
* tests/check/elements/gdpdepay.c: (GST_START_TEST):
|
|
* tests/check/elements/multifdsink.c: (GST_START_TEST):
|
|
* tests/check/elements/videorate.c: (GST_START_TEST):
|
|
* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
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* tests/check/pipelines/oggmux.c: (eos_buffer_probe):
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Fix some warnings.
|
|
|
|
2006-09-20 10:59:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
sys/xvimage/xvimagesink.c: change colorkey behaviour back according to #354773 comment 6/7
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
|
|
(gst_xvimagesink_get_times):
|
|
change colorkey behaviour back according to #354773 comment 6/7
|
|
|
|
2006-09-20 10:42:34 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
* ChangeLog:
|
|
ChangeLog surgery: remove junk
|
|
Original commit message from CVS:
|
|
ChangeLog surgery: remove junk
|
|
|
|
2006-09-19 11:31:06 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/tcp/gstmultifdsink.*: Implement stubbed out properties unit-type, units-soft-max, units-max, to allow specifying ...
|
|
Original commit message from CVS:
|
|
* gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
|
|
(gst_multi_fd_sink_class_init), (get_buffers_max), (find_limits),
|
|
(gst_multi_fd_sink_recover_client),
|
|
(gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_set_property),
|
|
(gst_multi_fd_sink_get_property):
|
|
* gst/tcp/gstmultifdsink.h:
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|
Implement stubbed out properties unit-type, units-soft-max,
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|
units-max, to allow specifying maximum sizes in units other than
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|
buffers.
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Fixes #355935
|
|
|
|
2006-09-19 10:23:03 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/riff/riff-media.c: Reorder the audio formats a bit for clarity.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
|
|
(gst_riff_create_audio_template_caps):
|
|
Reorder the audio formats a bit for clarity.
|
|
Detect and create caps for MSGSM and MSN (WAV49).
|
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Fixes #356596.
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|
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
|
|
(gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
|
|
(gst_xvimagesink_get_xv_support), (gst_xvimagesink_show_frame):
|
|
Small cleanups, move error handling out of normal flow for clarity.
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|
|
|
2006-09-18 15:59:39 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
Add new interface to control video orientation (fixes #354908)
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-docs.sgml:
|
|
* docs/libs/gst-plugins-base-libs.types:
|
|
* gst-libs/gst/interfaces/Makefile.am:
|
|
* gst-libs/gst/interfaces/videoorientation.c:
|
|
(gst_video_orientation_get_type),
|
|
(gst_video_orientation_iface_init),
|
|
(gst_video_orientation_get_hflip),
|
|
(gst_video_orientation_get_vflip),
|
|
(gst_video_orientation_get_hcenter),
|
|
(gst_video_orientation_get_vcenter),
|
|
(gst_video_orientation_set_hflip),
|
|
(gst_video_orientation_set_vflip),
|
|
(gst_video_orientation_set_hcenter),
|
|
(gst_video_orientation_set_vcenter):
|
|
* gst-libs/gst/interfaces/videoorientation.h:
|
|
Add new interface to control video orientation (fixes #354908)
|
|
|
|
2006-09-18 15:48:01 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/videotestsrc/gstvideotestsrc.c: Use G_UNLIKELY in _create and log one more detail.
|
|
Original commit message from CVS:
|
|
* gst/videotestsrc/gstvideotestsrc.c:
|
|
Use G_UNLIKELY in _create and log one more detail.
|
|
(gst_video_test_src_get_times), (gst_video_test_src_create):
|
|
* sys/ximage/ximagesink.c: (gst_ximagesink_get_times):
|
|
Use gst_util_uint64_scale_int in _get_times().
|
|
|
|
2006-09-18 15:00:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support)
|
|
Give better warning message (add object and detail).
|
|
|
|
2006-09-18 14:42:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
sys/xvimage/xvimagesink.c: xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes #354773), use gst_util...
|
|
Original commit message from CVS:
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support),
|
|
(gst_xvimagesink_get_times):
|
|
xvimage assumed that XV_COLORKEY can be set in RGB888 format (fixes
|
|
#354773), use gst_util_uint64_scale_int in _get_times()
|
|
|
|
2006-09-18 14:21:45 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
ext/ogg/gstoggmux.c: Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was always true, leading to dro...
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggmux.c: (gst_ogg_mux_push_buffer):
|
|
Timestamps are unsigned; comparision against GST_CLOCK_TIME_NONE was
|
|
always true, leading to dropping all timestamps.
|
|
|
|
2006-09-18 11:40:14 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
ext/libvisual/visual.c: update to work also with libvisual 0.4 API
|
|
Original commit message from CVS:
|
|
* ext/libvisual/visual.c: (gst_vis_src_negotiate),
|
|
(gst_visual_chain), (gst_visual_change_state):
|
|
update to work also with libvisual 0.4 API
|
|
* tools/gst-launch-ext.1.in:
|
|
* tools/gst-visualise.1.in:
|
|
remove references to old man-pages
|
|
* tests/examples/seek/seek.c: (main):
|
|
add real meadi-buttons, add tool-tips for the seek-options, arrange
|
|
seek options in a table
|
|
|
|
2006-09-18 10:57:28 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
ext/ogg/gstoggmux.c: Don't generate out-of-order timestamps from oggmux, instead clamp output timestamps to be >= the...
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggmux.c: (gst_ogg_mux_clear),
|
|
(gst_ogg_mux_push_buffer):
|
|
Don't generate out-of-order timestamps from oggmux, instead clamp
|
|
output timestamps to be >= the previously output ts.
|
|
Fixes #355595
|
|
|
|
2006-09-18 10:18:22 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/tcp/gstmultifdsink.c: Updates, fixes, and typo corrections for multifdsink. No functional changes.
|
|
Original commit message from CVS:
|
|
* gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
|
|
(gst_multi_fd_sink_class_init):
|
|
Updates, fixes, and typo corrections for multifdsink. No functional
|
|
changes.
|
|
|
|
2006-09-17 21:58:06 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Don't crash on truncated files - check that we got an 8 byte buffer before tryin...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (ogganx_type_find):
|
|
Don't crash on truncated files - check that we got an 8 byte buffer
|
|
before trying to memcmp it.
|
|
|
|
2006-09-17 20:32:09 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/playback/gstplaybasebin.c: Make stream-switching appear instant to the application (ie. make sure that a g_object...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (get_active_source):
|
|
Make stream-switching appear instant to the application
|
|
(ie. make sure that a g_object_get on 'current-foo' returns
|
|
the stream previously set with g_object_set(). Totem needs
|
|
this to update stream-related meta-info (like audio-codec)
|
|
correctly when switching streams.
|
|
|
|
2006-09-17 20:14:43 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/alsa/gstalsamixer.c: Try harder to guess which mixer track is the master mixer track (instead of just taking the ...
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_find_master_mixer),
|
|
(gst_alsa_mixer_ensure_track_list):
|
|
Try harder to guess which mixer track is the master mixer
|
|
track (instead of just taking the first one that has a pvolume).
|
|
Fixes #342228.
|
|
|
|
2006-09-17 11:24:21 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/audioconvert/gstaudioconvert.c: Get structure-name just once.
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/gstaudioconvert.c: (set_structure_widths),
|
|
(gst_audio_convert_transform_caps):
|
|
Get structure-name just once.
|
|
|
|
2006-09-16 22:30:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
tests/check/: Fix big batch of compiler warnings.
|
|
Original commit message from CVS:
|
|
* tests/check/elements/audioresample.c: (GST_START_TEST):
|
|
* tests/check/elements/videotestsrc.c: (check_rgb_buf):
|
|
* tests/check/elements/volume.c: (GST_START_TEST):
|
|
* tests/check/elements/vorbisdec.c: (GST_START_TEST):
|
|
* tests/check/pipelines/oggmux.c: (validate_ogg_page), (eos_watch),
|
|
(test_pipeline), (GST_START_TEST):
|
|
* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
|
|
* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
|
|
Fix big batch of compiler warnings.
|
|
|
|
2006-09-16 21:54:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
ext/gnomevfs/gstgnomevfssrc.c: Add docs about icydemux usage in connection with gnomevfssrc
|
|
Original commit message from CVS:
|
|
* ext/gnomevfs/gstgnomevfssrc.c:
|
|
Add docs about icydemux usage in connection with gnomevfssrc
|
|
* ext/libvisual/visual.c:
|
|
* ext/ogg/gstoggaviparse.c:
|
|
* ext/ogg/gstoggdemux.c:
|
|
* ext/ogg/gstoggmux.c:
|
|
* ext/ogg/gstoggparse.c:
|
|
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
|
|
* gst-libs/gst/audio/gstaudiosink.c:
|
|
* gst-libs/gst/audio/gstaudiosrc.c:
|
|
* gst/audiorate/gstaudiorate.c:
|
|
More G_OBJECT macro fixing.
|
|
* gst/audiotestsrc/gstaudiotestsrc.h:
|
|
Fix wrong info in header due to copy & paste
|
|
|
|
2006-09-15 14:53:44 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/: Do the delay calculation in the source/sink base classes as this is specific for the capture/pla...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_get_time), (gst_base_audio_sink_callback):
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
(gst_base_audio_src_get_time), (gst_base_audio_src_fixate),
|
|
(gst_base_audio_src_get_times), (gst_base_audio_src_get_offset),
|
|
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
|
|
Do the delay calculation in the source/sink base classes as this is
|
|
specific for the capture/playback mode.
|
|
Try to fixate a bit better, like round depth up to a multiple of 8
|
|
bigger than width.
|
|
Handle underruns correctly by marking DISCONT on buffers and adjusting
|
|
timestamps to handle the gap.
|
|
Set offset/offset_end correctly on buffers.
|
|
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_pause),
|
|
(gst_ring_buffer_samples_done), (gst_ring_buffer_commit),
|
|
(gst_ring_buffer_read):
|
|
Remove resync and underrun recovery from the ringbuffer.
|
|
Fix ringbuffer read code on under/overrun.
|
|
|
|
2006-09-15 11:17:02 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybasebin.*: Don't use a 0 low watermark when buffering, it is catching starvation way too late. In...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
|
|
(gst_play_base_bin_init), (fill_buffer), (check_queue),
|
|
(queue_threshold_reached), (gst_play_base_bin_set_property),
|
|
(gst_play_base_bin_get_property):
|
|
* gst/playback/gstplaybasebin.h:
|
|
Don't use a 0 low watermark when buffering, it is catching starvation
|
|
way too late. Instead, use a 3 second queue with 30 and 95
|
|
percent low/high watermarks.
|
|
Added queue-min-threshold property to configure low watermark.
|
|
Use new _buffering message API.
|
|
Make queue_threshold variable big enough to store a uint64 time value.
|
|
API: playbin::queue-min-threshold property.
|
|
|
|
2006-09-15 09:13:50 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
configure.ac: We require 0.10.10.1 now because of _wait_preroll().
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
We require 0.10.10.1 now because of _wait_preroll().
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_render):
|
|
Use gst_base_sink_wait_preroll().
|
|
|
|
2006-09-15 09:09:00 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/alsa/: Use DEBUG_OBJECT more.
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsasink.c: (xrun_recovery), (gst_alsasink_write):
|
|
* ext/alsa/gstalsasrc.c: (xrun_recovery), (gst_alsasrc_read):
|
|
Use DEBUG_OBJECT more.
|
|
|
|
=== release 0.10.10 ===
|
|
|
|
2006-09-14 20:09:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* ChangeLog:
|
|
* NEWS:
|
|
* RELEASE:
|
|
* common:
|
|
* configure.ac:
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.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-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gdp.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.xml:
|
|
* docs/plugins/inspect/plugin-libvisual.xml:
|
|
* docs/plugins/inspect/plugin-ogg.xml:
|
|
* docs/plugins/inspect/plugin-pango.xml:
|
|
* docs/plugins/inspect/plugin-playbin.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-video4linux.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:
|
|
* ext/theora/theoraparse.c:
|
|
* gst-libs/gst/rtp/gstrtpbuffer.c:
|
|
* gst/playback/gstplaybin.c:
|
|
* tests/check/Makefile.am:
|
|
* win32/common/config.h:
|
|
releasing 0.10.10
|
|
Original commit message from CVS:
|
|
releasing 0.10.10
|
|
|
|
2006-09-09 16:08:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* configure.ac:
|
|
* win32/common/config.h:
|
|
second prerelease
|
|
Original commit message from CVS:
|
|
second prerelease
|
|
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2006-09-07 19:01:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* ChangeLog:
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update bug in changelog
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Original commit message from CVS:
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update bug in changelog
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2006-09-07 19:00:33 +0000 Michael Smith <msmith@fluendo.com>
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Fix implementation of sync-method 'next-keyframe'
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Original commit message from CVS:
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patch by: Michael Smith <msmith at fluendo dot com>
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* gst/tcp/gstmultifdsink.c: (is_sync_frame),
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(gst_multi_fd_sink_client_queue_buffer),
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(gst_multi_fd_sink_new_client):
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* tests/check/elements/multifdsink.c: (GST_START_TEST),
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(multifdsink_suite):
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Fix implementation of sync-method 'next-keyframe'
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2006-09-07 15:00:08 +0000 Wim Taymans <wim@fluendo.com>
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ext/gnomevfs/gstgnomevfssrc.c: This patch removes the RANDOM flag that was incorrectly introduced with revision 1.91....
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Original commit message from CVS:
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patch by: Wim Taymans <wim at fluendo dot com>
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* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
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This patch removes the RANDOM flag that was incorrectly introduced with
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revision 1.91. Fixes #354590
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2006-09-07 14:56:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* configure.ac:
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* win32/common/config.h:
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first prerelease
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Original commit message from CVS:
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first prerelease
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2006-09-07 14:56:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* po/af.po:
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* po/az.po:
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* po/cs.po:
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* po/en_GB.po:
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* po/hu.po:
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* po/it.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/sq.po:
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* po/sr.po:
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* po/sv.po:
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* po/uk.po:
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* po/vi.po:
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update po files
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Original commit message from CVS:
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update po files
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2006-09-05 09:12:25 +0000 Tim-Philipp Müller <tim@centricular.net>
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tests/check/Makefile.am: Random variation in Makefile line to see if it makes the gen64-base-full bot any happier.
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Original commit message from CVS:
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* tests/check/Makefile.am:
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Random variation in Makefile line to see if it makes the
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gen64-base-full bot any happier.
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2006-09-04 19:04:35 +0000 Tim-Philipp Müller <tim@centricular.net>
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tests/check/pipelines/oggmux.c: Disable test that fails at the moment (killed after timeout).
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Original commit message from CVS:
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* tests/check/pipelines/oggmux.c: (oggmux_suite):
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Disable test that fails at the moment (killed after timeout).
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2006-09-04 18:19:06 +0000 James Livingston <doclivingston@gmail.com>
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tests/check/: Add simple unit test for oggmux from #337026 with checking for the
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Original commit message from CVS:
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Patch by: James Livingston <doclivingston at gmail.com>
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* tests/check/Makefile.am:
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* tests/check/pipelines/.cvsignore:
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* tests/check/pipelines/oggmux.c: (get_page_codec),
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(check_chain_final_state), (fail_if_audio), (validate_ogg_page),
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(eos_buffer_probe), (start_pipeline), (stop_pipeline), (eos_watch),
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(test_pipeline), (test_vorbis), (test_theora), (test_vorbis_theora),
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(test_theora_vorbis), (oggmux_suite):
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Add simple unit test for oggmux from #337026 with checking for the
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EOS flags disabled for the time being.
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2006-09-04 09:13:01 +0000 Alessandro Dessina <alessandro@nnva.org>
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ext/ogg/gstoggmux.c: Add cmml caps to oggmux. Fixes #353912
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Original commit message from CVS:
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patch by: Alessandro Dessina <alessandro nnva org>
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* ext/ogg/gstoggmux.c:
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Add cmml caps to oggmux. Fixes #353912
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2006-09-02 13:20:59 +0000 Tim-Philipp Müller <tim@centricular.net>
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tests/check/elements/videotestsrc.c: Returning a return value often helps. In this case, we don't need the return val...
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Original commit message from CVS:
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* tests/check/elements/videotestsrc.c: (check_rgb_buf):
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Returning a return value often helps. In this case, we
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don't need the return value anyway, so just get rid of it.
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Should make build bots much happier.
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2006-09-02 12:59:48 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/videotestsrc/videotestsrc.*: Add support for AYUV and the various RGBA formats. Initialise fields of paintinfo st...
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|
Original commit message from CVS:
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* gst/videotestsrc/videotestsrc.c: (paintinfo_find_by_structure),
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(paint_get_structure), (gst_video_test_src_get_size),
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(gst_video_test_src_smpte), (gst_video_test_src_snow),
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(gst_video_test_src_unicolor), (paint_setup_AYUV),
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(paint_hline_AYUV), (paint_setup_ARGB8888), (paint_setup_ABGR8888),
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(paint_setup_RGBA8888), (paint_setup_BGRA8888), (paint_hline_str4):
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* gst/videotestsrc/videotestsrc.h:
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Add support for AYUV and the various RGBA formats. Initialise
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fields of paintinfo structs allocated on the stack.
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* tests/check/elements/videotestsrc.c: (right_shift_colour),
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(fix_expected_colour), (check_rgb_buf), (got_buf_cb),
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(GST_START_TEST), (videotestsrc_suite):
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Add unit tests for videotestsrc's RGB output.
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2006-09-01 16:12:35 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/videotestsrc/: Add more uni-colour patterns ("white", "red", "green", and "blue").
|
|
Original commit message from CVS:
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|
* gst/videotestsrc/gstvideotestsrc.c:
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(gst_video_test_src_pattern_get_type),
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(gst_video_test_src_set_pattern):
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* gst/videotestsrc/gstvideotestsrc.h:
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* gst/videotestsrc/videotestsrc.c: (gst_video_test_src_unicolor),
|
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(gst_video_test_src_black), (gst_video_test_src_white),
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(gst_video_test_src_red), (gst_video_test_src_green),
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(gst_video_test_src_blue):
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* gst/videotestsrc/videotestsrc.h:
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Add more uni-colour patterns ("white", "red", "green", and "blue").
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2006-09-01 10:07:05 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/videotestsrc/videotestsrc.c: Fix stride for YVYU, should be word-aligned (#353658).
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|
Original commit message from CVS:
|
|
* gst/videotestsrc/videotestsrc.c: (paint_setup_YVYU):
|
|
Fix stride for YVYU, should be word-aligned (#353658).
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2006-08-31 14:37:33 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
gst/adder/gstadder.c: Fix build.
|
|
Original commit message from CVS:
|
|
* gst/adder/gstadder.c: (gst_adder_src_event):
|
|
Fix build.
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|
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2006-08-31 12:39:17 +0000 Edward Hervey <bilboed@bilboed.com>
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gst/adder/gstadder.*: Remember the start position asked in the incoming seeks, so we can output GST_EVENT_NEW_SEGMENT...
|
|
Original commit message from CVS:
|
|
* gst/adder/gstadder.c: (forward_event_func),
|
|
(gst_adder_src_event), (gst_adder_collected),
|
|
(gst_adder_change_state):
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* gst/adder/gstadder.h:
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Remember the start position asked in the incoming seeks, so we can
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output GST_EVENT_NEW_SEGMENT with a correct position value (instead
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of assuming it will always be 0).
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2006-08-31 12:31:00 +0000 Edward Hervey <bilboed@bilboed.com>
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ext/ogg/gstoggdemux.c: Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_init),
|
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(gst_ogg_demux_finalize), (gst_ogg_demux_perform_seek),
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(gst_ogg_demux_loop):
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Send the GST_EVENT_NEW_SEGMENT from the streaming thread.
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2006-08-30 17:22:27 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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gst/ffmpegcolorspace/gstffmpegcolorspace.c: Return FALSE instead of returning a random false unit size when the forma...
|
|
Original commit message from CVS:
|
|
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
|
|
(gst_ffmpegcsp_get_unit_size):
|
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Return FALSE instead of returning a random false unit
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size when the format isn't known/supported (even if
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this shouldn't happen under normal circumstances).
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2006-08-29 15:23:46 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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ext/gnomevfs/gstgnomevfssrc.c: Try harder to get the size from a uri by using _info_uri() when _info_from_handle() do...
|
|
Original commit message from CVS:
|
|
Patch by: Tim-Philipp Müller <tim at centricular dot net>
|
|
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create),
|
|
(gst_gnome_vfs_src_start):
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|
Try harder to get the size from a uri by using _info_uri() when
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_info_from_handle() does not give us enough info.
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Also follow symlinks when getting the size.
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Partially Fixes #332864.
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2006-08-29 11:50:51 +0000 Viktor Peters <viktor.peters@gmail.com>
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|
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ext/alsa/: Improve and fix mixer track handling, in particular better handling of alsa's pvolume/pswitch/cvolume/cswi...
|
|
Original commit message from CVS:
|
|
Patch by: Viktor Peters <viktor dot peters at gmail dot com>
|
|
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_ensure_track_list),
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|
(gst_alsa_mixer_update), (gst_alsa_mixer_get_volume),
|
|
(gst_alsa_mixer_set_volume), (gst_alsa_mixer_set_mute),
|
|
(gst_alsa_mixer_set_record):
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|
* ext/alsa/gstalsamixertrack.c:
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|
(gst_alsa_mixer_track_update_alsa_capabilities),
|
|
(alsa_track_has_cap), (gst_alsa_mixer_track_new),
|
|
(gst_alsa_mixer_track_update):
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|
* ext/alsa/gstalsamixertrack.h:
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|
Improve and fix mixer track handling, in particular better handling
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of alsa's pvolume/pswitch/cvolume/cswitch capabilities; create separate
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track objects for tracks that have both capture and playback volume
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(and label them differently as well so they're not mistakenly
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assumed to be duplicates); classify mixer tracks that only affect
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the audible volume of something (rather than the capture volume)
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|
as playback tracks. Redefine/fix meaning of RECORD and MUTE flags
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for capture tracks to correspond to alsa-pswitch alsa-cswitch
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|
(following the meaning documented in the mixer interface header
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file); add support for alsa's exclusive cswitch groups; update/sync
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state/flags better if mixer settings are changed by another
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application. Fixes #336075.
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2006-08-29 10:58:43 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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gst/playback/gstplaybin.c: Improve docs: add section about BUFFERING messages sent by playbin.
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|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin.c:
|
|
Improve docs: add section about BUFFERING messages sent by playbin.
|
|
|
|
2006-08-29 10:51:12 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
ext/vorbis/vorbisenc.c: Ignore explicit DISCONT marked on buffers (which is often spurious, particularly when using m...
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_push_buffer),
|
|
(gst_vorbis_enc_buffer_check_discontinuous),
|
|
(gst_vorbis_enc_chain):
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Ignore explicit DISCONT marked on buffers (which is often spurious,
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particularly when using multiple segments), in favour of solely
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|
using the timestamps/durations.
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|
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2006-08-29 10:32:34 +0000 Edward Hervey <bilboed@bilboed.com>
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|
|
|
gst/audiorate/gstaudiorate.c: Don't rely on incoming buffers offset anymore, since it is completely broken when using...
|
|
Original commit message from CVS:
|
|
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
|
|
Don't rely on incoming buffers offset anymore, since it is completely
|
|
broken when using multiple segments.
|
|
Instead convert the incoming buffers timestamp to running time, and
|
|
then convert that value to the offsets.
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Also inform GstSegment of the last outputted stop position, which is
|
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needed if we received several segments with an unknown stop value.
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|
|
|
2006-08-29 08:03:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
ext/vorbis/vorbisenc.c: fix buffer unreffing on a header push failure
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):
|
|
fix buffer unreffing on a header push failure
|
|
|
|
2006-08-28 16:17:13 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/audiorate/gstaudiorate.c: Make the metadata of the buffer writable before changing its flags.
|
|
Original commit message from CVS:
|
|
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_sink_event),
|
|
(gst_audio_rate_chain):
|
|
Make the metadata of the buffer writable before changing its
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|
flags.
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|
|
|
2006-08-28 16:09:57 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
* ChangeLog:
|
|
Fix changelog with bugzilla bug it fixed.
|
|
Original commit message from CVS:
|
|
Fix changelog with bugzilla bug it fixed.
|
|
|
|
2006-08-28 16:08:18 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/audiorate/gstaudiorate.c: Fix audiorate some more.
|
|
Original commit message from CVS:
|
|
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
|
|
(gst_audio_rate_setcaps), (gst_audio_rate_init),
|
|
(gst_audio_rate_sink_event), (gst_audio_rate_src_event),
|
|
(gst_audio_rate_chain), (gst_audio_rate_change_state):
|
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Fix audiorate some more.
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Reset and resync counters on flush and READY.
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Handle the DISCONT flag correctly.
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Use GstSegment to track position.
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Fail when not negotiated.
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|
|
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2006-08-25 16:48:28 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/tcp/gstmultifdsink.c: Fix spelling.
|
|
Original commit message from CVS:
|
|
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
|
|
Fix spelling.
|
|
Remove accidently included debug line.
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|
|
|
2006-08-25 16:39:38 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/tcp/gstmultifdsink.c: Small cleanups.
|
|
Original commit message from CVS:
|
|
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
|
|
Small cleanups.
|
|
If a buffer is received with no caps, make the buffer metadata
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writable and set the caps, making sure that we don't screw up the
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refcounts.
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2006-08-25 16:19:55 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/gdp/gstgdppay.c: Fix memory leaks and misleading debug messages, add a couple of comments.
|
|
Original commit message from CVS:
|
|
* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset),
|
|
(gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain):
|
|
Fix memory leaks and misleading debug messages, add a couple of
|
|
comments.
|
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* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats),
|
|
(gst_multi_fd_sink_render):
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|
Do not use gst_buffer_make_writable() in a basesink render method,
|
|
as it may incorrectly unref the buffer. Instead, use convoluted
|
|
dance to avoid copying the buffer except when we need to.
|
|
|
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2006-08-25 09:54:56 +0000 Michael Smith <msmith@xiph.org>
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|
|
|
ext/vorbis/vorbisenc.c: Allow very small discontinuities in the timestamps. These we can't do anything useful with an...
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisenc.c:
|
|
(gst_vorbis_enc_buffer_check_discontinuous):
|
|
Allow very small discontinuities in the timestamps. These we can't
|
|
do anything useful with anyway (because vorbis's timestamps have
|
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only sample granularity), and are commonly produced by elements with
|
|
minor bugs. Allow up to 1/2 a sample out.
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Fixes #351742.
|
|
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|
2006-08-24 11:18:56 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
tests/examples/seek/seek.c: Add a checkbox to enable play scrubbing. Makes it possible to disable normal scrubbing.
|
|
Original commit message from CVS:
|
|
* tests/examples/seek/seek.c: (seek_cb), (start_seek), (stop_seek),
|
|
(play_scrub_toggle_cb), (main):
|
|
Add a checkbox to enable play scrubbing. Makes it possible to disable
|
|
normal scrubbing.
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|
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2006-08-23 19:37:50 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
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tests/check/elements/.cvsignore: make buildbot happy
|
|
Original commit message from CVS:
|
|
* tests/check/elements/.cvsignore:
|
|
make buildbot happy
|
|
|
|
2006-08-23 16:43:03 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/ogg/gstogmparse.c: Refactor ogm parse, do better input checking, misc. clean-ups.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
|
|
(gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init),
|
|
(gst_ogm_parse_class_init), (gst_ogm_parse_dispose),
|
|
(gst_ogm_parse_init), (gst_ogm_audio_parse_init),
|
|
(gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
|
|
(gst_ogm_parse_stream_header), (gst_ogm_parse_comment_packet),
|
|
(gst_ogm_text_parse_strip_trailing_zeroes),
|
|
(gst_ogm_parse_data_packet), (gst_ogm_parse_chain),
|
|
(gst_ogm_parse_sink_event), (gst_ogm_parse_change_state):
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Refactor ogm parse, do better input checking, misc. clean-ups.
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Cache incoming events and push them once the source pad has
|
|
been created. Don't pass unterminated strings to sscanf().
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Strip trailing zeroes from subtitle text output, since they
|
|
are not valid UTF-8. Don't push vorbiscomment packets on
|
|
the subtitle text pad. Output perfect streams if possible.
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2006-08-23 15:27:38 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
tests/check/libs/cddabasesrc.c: Waits for tasks to settle down so that we clean up correctly for valgrind.
|
|
Original commit message from CVS:
|
|
* tests/check/libs/cddabasesrc.c: (GST_START_TEST):
|
|
Waits for tasks to settle down so that we clean up correctly for
|
|
valgrind.
|
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|
|
2006-08-23 15:11:56 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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tests/check/libs/tag.c: Unit test fixes: \377 is more likely to fit into 8 bits than \777; actually return return val...
|
|
Original commit message from CVS:
|
|
* tests/check/libs/tag.c: (GST_START_TEST), (taglists_are_equal):
|
|
Unit test fixes: \377 is more likely to fit into 8 bits than \777;
|
|
actually return return value in taglists_are_equal.
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|
|
|
2006-08-23 12:14:20 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
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ext/ogg/gstogmparse.c: Fix crash due to broken bitstream parsing on x86-64: can't make any assumptions about sizeof(s...
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Original commit message from CVS:
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* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
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Fix crash due to broken bitstream parsing on x86-64: can't make
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any assumptions about sizeof(struct) due to alignment/packing
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differences on different architectures. Fixes #351790.
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2006-08-22 16:31:47 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/riff/riff-read.c: Protect public functions against bad input.
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Original commit message from CVS:
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* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
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(gst_riff_parse_chunk), (gst_riff_parse_file_header),
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(gst_riff_parse_strh), (gst_riff_parse_strf_vids),
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(gst_riff_parse_strf_auds), (gst_riff_parse_strf_iavs),
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(gst_riff_parse_info):
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Protect public functions against bad input.
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Do some cleanups.
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Fix documentation.
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2006-08-22 15:50:36 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/riff/: Add voxware audio IDs (even if we can't play it) (#351795).
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Original commit message from CVS:
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* gst-libs/gst/riff/riff-ids.h:
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* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
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Add voxware audio IDs (even if we can't play it) (#351795).
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2006-08-22 15:11:52 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/riff/riff-media.c: Const-ify some arrays and use G_N_ELEMENTS instead of wasting oodles of RAM on termin...
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Original commit message from CVS:
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* gst-libs/gst/riff/riff-media.c:
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(gst_riff_create_video_template_caps),
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(gst_riff_create_audio_template_caps),
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(gst_riff_create_iavs_template_caps):
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Const-ify some arrays and use G_N_ELEMENTS instead
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of wasting oodles of RAM on terminator bits.
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2006-08-22 08:27:07 +0000 Tim-Philipp Müller <tim@centricular.net>
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And the same for _to_vorbiscomment_buffer(): allow id_data_len == 0 for speex.
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Original commit message from CVS:
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* gst-libs/gst/tag/gstvorbistag.c:
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(gst_tag_list_to_vorbiscomment_buffer):
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* tests/check/libs/tag.c: (GST_START_TEST):
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And the same for _to_vorbiscomment_buffer(): allow
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id_data_len == 0 for speex.
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2006-08-21 19:04:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* gst/gdp/README:
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adding a README
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Original commit message from CVS:
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adding a README
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2006-08-21 19:01:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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Move GDP plugin to -base from -bad. Closes #347783.
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Original commit message from CVS:
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* configure.ac:
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-base-plugins-docs.sgml:
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* docs/plugins/gst-plugins-base-plugins-sections.txt:
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* docs/plugins/inspect/plugin-gdp.xml:
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* gst/gdp/Makefile.am:
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* tests/check/Makefile.am:
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Move GDP plugin to -base from -bad. Closes #347783.
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2006-08-21 18:34:46 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/tag/gstvorbistag.c: Allow id_data_len == 0 (needed for vorbis comments in Speex files).
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Original commit message from CVS:
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* gst-libs/gst/tag/gstvorbistag.c:
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(gst_tag_list_from_vorbiscomment_buffer):
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Allow id_data_len == 0 (needed for vorbis comments in Speex files).
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Also add some checks to make sure we don't memcmp() beyond the end of
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vorbiscomment buffer if the ID to check for is larger than the buffer.
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* tests/check/libs/tag.c: (GST_START_TEST):
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Some more tests for gst_tag_list_from_vorbiscomment_buffer().
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2006-08-21 16:39:25 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/vorbis/vorbisenc.c: Use vorbis comment utility functions from libgsttag instead of re-inventing the wheel (partia...
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Original commit message from CVS:
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* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_metadata_set1),
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(gst_vorbis_enc_set_metadata):
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Use vorbis comment utility functions from libgsttag
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instead of re-inventing the wheel (partially fixes #347091).
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2006-08-21 11:42:12 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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tests/check/elements/audioconvert.c: Fix leaks. Wait for state transitions that might happen ASYNC, as well as some t...
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Original commit message from CVS:
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* tests/check/elements/audioconvert.c: (GST_START_TEST):
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Fix leaks. Wait for state transitions that might happen ASYNC, as well
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as some that won't.
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2006-08-21 10:32:51 +0000 Wim Taymans <wim.taymans@gmail.com>
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docs/libs/: Don't try to GObject scan the netbuffer as it's not a GObject.
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Original commit message from CVS:
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* docs/libs/Makefile.am:
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* docs/libs/gst-plugins-base-libs-sections.txt:
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* docs/libs/gst-plugins-base-libs.types:
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Don't try to GObject scan the netbuffer as it's not a GObject.
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Fixes #351308.
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* gst-libs/gst/netbuffer/gstnetbuffer.c:
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* gst-libs/gst/netbuffer/gstnetbuffer.h:
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Document GstNetBuffer.
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2006-08-21 08:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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tests/check/elements/audioconvert.c: Add testcase for caps-size-explosion
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Original commit message from CVS:
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* tests/check/elements/audioconvert.c: (GST_START_TEST),
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(audioconvert_suite):
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Add testcase for caps-size-explosion
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2006-08-20 13:05:44 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst/audioconvert/gstaudioconvert.c: Lower debug, use g_assert in _get_unit_size
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Original commit message from CVS:
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* gst/audioconvert/gstaudioconvert.c:
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(gst_audio_convert_get_unit_size), (set_structure_widths):
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Lower debug, use g_assert in _get_unit_size
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* gst/audioresample/gstaudioresample.c:
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(audioresample_get_unit_size):
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* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
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(gst_ffmpegcsp_get_unit_size):
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* gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size):
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use g_assert in _get_unit_size
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2006-08-18 21:21:48 +0000 Tim-Philipp Müller <tim@centricular.net>
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* ChangeLog:
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ChangeLog surgery: fix bug number
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Original commit message from CVS:
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ChangeLog surgery: fix bug number
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2006-08-18 16:43:26 +0000 Wim Taymans <wim.taymans@gmail.com>
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Document GstRTPBuffer.
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Original commit message from CVS:
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* docs/libs/gst-plugins-base-libs-sections.txt:
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* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_set_packet_len),
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(gst_rtp_buffer_pad_to), (gst_rtp_buffer_get_payload_subbuffer),
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(gst_rtp_buffer_get_payload_buffer):
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* gst-libs/gst/rtp/gstrtpbuffer.h:
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Document GstRTPBuffer.
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Added function to efficiently strip payload headers.
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API: gst_rtp_buffer_get_payload_subbuffer()
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2006-08-17 16:52:06 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/tag/gstvorbistag.c: Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT tags and deserialise...
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|
Original commit message from CVS:
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|
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add),
|
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(gst_tag_to_vorbis_comments):
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Serialise unknown vorbis comments into GST_TAG_EXTENDED_COMMENT
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tags and deserialise them properly as well (#351768).
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Add some more gtk-doc blurbs and also some g_return_if_fail().
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* tests/check/libs/tag.c: (GST_START_TEST),
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(back_to_vorbis_comments), (taglists_are_equal), (tag_suite):
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More tests.
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2006-08-17 15:43:40 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/ogg/: Added ogg-in-avi parser element. Fixes #140139.
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|
Original commit message from CVS:
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|
* ext/ogg/Makefile.am:
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* ext/ogg/gstogg.c: (plugin_init):
|
|
* ext/ogg/gstoggaviparse.c: (gst_ogg_avi_parse_get_type),
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(gst_ogg_avi_parse_base_init), (gst_ogg_avi_parse_class_init),
|
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(gst_ogg_avi_parse_init), (gst_ogg_avi_parse_finalize),
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(gst_ogg_avi_parse_setcaps), (gst_ogg_avi_parse_event),
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(gst_ogg_avi_parse_push_packet), (gst_ogg_avi_parse_chain),
|
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(gst_ogg_avi_parse_change_state), (gst_ogg_avi_parse_plugin_init):
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Added ogg-in-avi parser element. Fixes #140139.
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* ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
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Fixed a bug in oggdemux debug code.
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* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
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(gst_riff_create_audio_template_caps):
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Recognise Ogg in the AVI extensible wave format.
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2006-08-17 10:00:00 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/cdda/gstcddabasesrc.c: Make buffer durations add up (duration should be next_ts-ts for perfect streams)....
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|
Original commit message from CVS:
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|
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
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Make buffer durations add up (duration should be next_ts-ts for
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perfect streams). Fixes CD ripping to Ogg/Vorbis with vorbisenc
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from CVS.
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* tests/check/libs/cddabasesrc.c: (gst_cd_foo_src_close),
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(test_buffer_timestamps), (cddabasesrc_suite):
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Add unit test for the above.
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* tests/check/Makefile.am:
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Don't know why cddabasesrc test was in VALGRIND_TO_FIX, remove
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to see what happens.
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2006-08-16 11:38:52 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/alsa/: Avoid setting and using a NULL device name.
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Original commit message from CVS:
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|
* ext/alsa/gstalsasink.c: (gst_alsasink_set_property),
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|
(gst_alsasink_open):
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* ext/alsa/gstalsasrc.c: (gst_alsasrc_set_property),
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(gst_alsasrc_open):
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Avoid setting and using a NULL device name.
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Print more info when we fail to open a device.
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2006-08-16 11:28:57 +0000 Tim-Philipp Müller <tim@centricular.net>
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API: add gst_tag_parse_extended_comment() (#351426).
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|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/tag/tag.h:
|
|
* gst-libs/gst/tag/tags.c: (gst_tag_parse_extended_comment):
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API: add gst_tag_parse_extended_comment() (#351426).
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* tests/check/Makefile.am:
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* tests/check/libs/.cvsignore:
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* tests/check/libs/tag.c: (GST_START_TEST), (tag_suite), (main):
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Add unit test for gst_tag_parse_extended_comment().
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2006-08-15 19:20:16 +0000 Tim-Philipp Müller <tim@centricular.net>
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sys/: Fix leak (#351502).
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|
Original commit message from CVS:
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|
* sys/ximage/ximagesink.c: (gst_ximagesink_get_property):
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_property):
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Fix leak (#351502).
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2006-08-15 17:21:33 +0000 Tim-Philipp Müller <tim@centricular.net>
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Document playbin.
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|
Original commit message from CVS:
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|
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
|
* docs/plugins/gst-plugins-base-plugins.args:
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* gst/playback/gstplaybin.c:
|
|
Document playbin.
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|
* docs/plugins/inspect/plugin-adder.xml:
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|
* docs/plugins/inspect/plugin-alsa.xml:
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|
* docs/plugins/inspect/plugin-audioconvert.xml:
|
|
* docs/plugins/inspect/plugin-audiorate.xml:
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|
* docs/plugins/inspect/plugin-audioresample.xml:
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|
* docs/plugins/inspect/plugin-audiotestsrc.xml:
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|
* docs/plugins/inspect/plugin-cdparanoia.xml:
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|
* docs/plugins/inspect/plugin-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.xml:
|
|
* docs/plugins/inspect/plugin-ogg.xml:
|
|
* docs/plugins/inspect/plugin-pango.xml:
|
|
* docs/plugins/inspect/plugin-playbin.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-video4linux.xml:
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|
* 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:
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* docs/plugins/inspect/plugin-ximagesink.xml:
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* docs/plugins/inspect/plugin-xvimagesink.xml:
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Update to CVS version.
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2006-08-14 17:54:01 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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gst/playback/gstplaybin.c: Add "connection-speed" property; re-order redirect messages with multiple redirect locatio...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin.c: (gst_play_bin_class_init),
|
|
(gst_play_bin_set_property), (gst_play_bin_get_property),
|
|
(value_list_append_structure_list),
|
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(gst_play_bin_handle_redirect_message),
|
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(gst_play_bin_handle_message):
|
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Add "connection-speed" property; re-order redirect messages with
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multiple redirect locations depending on the minimum bitrate if
|
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that information is available and a connection speed is set
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(#350399).
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2006-08-14 11:41:04 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gstplaybin.c: Update max volume to the same value that the volume element uses.
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|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin.c:
|
|
Update max volume to the same value that the volume element uses.
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|
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2006-08-14 10:50:15 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
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|
ext/alsa/gstalsamixer.c: Less uglyness..
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|
Original commit message from CVS:
|
|
* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
|
|
Less uglyness..
|
|
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|
2006-08-14 10:49:10 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/ogg/gstoggdemux.c: Add some more debug info.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_submit_buffer),
|
|
(gst_ogg_demux_get_next_page), (gst_ogg_demux_perform_seek),
|
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(gst_ogg_demux_read_chain), (gst_ogg_demux_loop):
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Add some more debug info.
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Don't crash when a seek failed.
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Actually return the result of the seek instead of TRUE.
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Ignore multiple BOS pages with the same serial so that we don't create
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the same stream multiple times.
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Post an error when we fail to do the initial seek.
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2006-08-13 14:34:48 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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ext/alsa/gstalsa.c: Small code cleanup.
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|
Original commit message from CVS:
|
|
* ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
|
|
(gst_alsa_detect_channels), (gst_alsa_probe_supported_formats):
|
|
Small code cleanup.
|
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* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open),
|
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(gst_alsa_mixer_new):
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Remove hack that always set the device to hw:0*.
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Properly find the card name for whatever device was configured.
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Do some better debugging.
|
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Fixes #350784.
|
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* ext/alsa/gstalsamixerelement.c:
|
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(gst_alsa_mixer_element_set_property),
|
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(gst_alsa_mixer_element_change_state):
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Cleanups.
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Handle setting of a NULL device name better.
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2006-08-11 15:53:43 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
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gst/adder/gstadder.c: Don't clip float values. Fixes #350900.
|
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Original commit message from CVS:
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|
* gst/adder/gstadder.c:
|
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Don't clip float values. Fixes #350900.
|
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|
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2006-08-11 15:33:17 +0000 Andy Wingo <wingo@pobox.com>
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gst/tcp/gsttcp.c: Really fix the build?
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|
Original commit message from CVS:
|
|
2006-08-11 Andy Wingo <wingo@pobox.com>
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* gst/tcp/gsttcp.c: Really fix the build?
|
|
|
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2006-08-11 15:29:56 +0000 Andy Wingo <wingo@pobox.com>
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gst/tcp/gsttcp.h: For now, always disable deprecation here -- fixes the build.
|
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Original commit message from CVS:
|
|
2006-08-11 Andy Wingo <wingo@pobox.com>
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* gst/tcp/gsttcp.h: For now, always disable deprecation here --
|
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fixes the build.
|
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2006-08-10 13:01:31 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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gst/audioconvert/gstaudioconvert.c: Float caps shouldn't have a "signed" field.
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Original commit message from CVS:
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* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes):
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Float caps shouldn't have a "signed" field.
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2006-08-10 08:56:22 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/ogg/gstoggdemux.c: Implement SEEKING query in its most basic form, so that we can at least check if we're seekabl...
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query):
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Implement SEEKING query in its most basic form, so that we can
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at least check if we're seekable or not (#350655).
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|
|
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2006-08-09 14:42:58 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
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gst/typefind/gsttypefindfunctions.c: The checks here are not even close to anything that would justify MAXIMUM probab...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find):
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The checks here are not even close to anything that would
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|
justify MAXIMUM probability, lowering to POSSIBLE until someone
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fixes the checks (case at hand: quicktime redirection files
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might start with 00 00 01 XX and pass the checks here just
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fine, see #350399).
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2006-08-08 13:57:29 +0000 Edward Hervey <bilboed@bilboed.com>
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tests/check/elements/gdpdepay.c: I forgot to include the file containing the #define :)
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Original commit message from CVS:
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* tests/check/elements/gdpdepay.c: (gdpdepay_suite):
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I forgot to include the file containing the #define :)
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Now includes "config.h"
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2006-08-08 13:45:44 +0000 Edward Hervey <bilboed@bilboed.com>
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tests/check/elements/gdpdepay.c: Ignore test known to fail on PPC64. See #348114.
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Original commit message from CVS:
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* tests/check/elements/gdpdepay.c: (gdpdepay_suite):
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Ignore test known to fail on PPC64. See #348114.
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2006-08-08 08:41:13 +0000 Sjoerd Simons <sjoerd@luon.net>
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gst/typefind/gsttypefindfunctions.c: Better detection for multipart/x-mixed-replace: accept leading whitespaces befor...
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Original commit message from CVS:
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Patch by: Sjoerd Simons <sjoerd at luon net>
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* gst/typefind/gsttypefindfunctions.c: (multipart_type_find):
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Better detection for multipart/x-mixed-replace: accept leading
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whitespaces before the boundary marker as well (as our very own
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multipartmux used to produce) (#349068).
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2006-08-07 08:26:03 +0000 Young-Ho Cha <ganadist@chollian.net>
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gst-libs/gst/riff/: Detect DTS audio streams (#350157).
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Original commit message from CVS:
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Patch by: Young-Ho Cha <ganadist at chollian net>
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* gst-libs/gst/riff/riff-ids.h:
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* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
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(gst_riff_create_audio_template_caps):
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Detect DTS audio streams (#350157).
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2006-08-05 17:08:05 +0000 Andy Wingo <wingo@pobox.com>
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ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (gst_theora_parse_class_init) (theora_parse_dispose, theora_par...
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Original commit message from CVS:
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2006-08-05 Andy Wingo <wingo@pobox.com>
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* ext/theora/gsttheoraparse.h:
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* ext/theora/theoraparse.c (gst_theora_parse_class_init)
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(theora_parse_dispose, theora_parse_set_property)
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(theora_parse_get_property, theora_parse_munge_granulepos)
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(theora_parse_push_buffer, theora_parse_change_state): Add a
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property 'synchronization-points' to fix badly synchronized oggs.
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2006-08-04 13:20:23 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
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gst/gdp/gstgdpdepay.c: Fix event parsing by gdpdepay. Fixes #349916.
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Original commit message from CVS:
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2006-08-04 Zaheer Abbas Merali <zaheerabbas at merali dot org>
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* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
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Fix event parsing by gdpdepay. Fixes #349916.
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2006-08-03 15:04:42 +0000 Tim-Philipp Müller <tim@centricular.net>
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tests/check/: Add a few tests for the channel position stuff in libgstaudio.
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Original commit message from CVS:
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* tests/check/Makefile.am:
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* tests/check/libs/.cvsignore:
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* tests/check/libs/audio.c: (structure_contains_channel_positions),
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(fixed_caps_have_channel_positions), (GST_START_TEST),
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(audio_suite), (main):
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Add a few tests for the channel position stuff in libgstaudio.
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2006-08-03 14:16:06 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/alsa/: Add support for cards that (only) do more than 8 channels, like the Delta 44 (#345188).
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Original commit message from CVS:
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* ext/alsa/gstalsa.c: (caps_add_channel_configuration),
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(gst_alsa_detect_channels):
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* ext/alsa/gstalsasink.c:
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Add support for cards that (only) do more than 8 channels,
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like the Delta 44 (#345188).
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* gst-libs/gst/audio/multichannel.c:
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(gst_audio_check_channel_positions):
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* gst-libs/gst/audio/multichannel.h:
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API: add GST_AUDIO_CHANNEL_POSITION_NONE, which stands for an
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unspecified channel position and cannot be combined with any
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of the other audio channel positions; adjust position layout
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checks accordingly (#345188).
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2006-08-03 11:15:53 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/typefind/gsttypefindfunctions.c: Recognise ancient RealAudio files (see #349779).
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Original commit message from CVS:
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* gst/typefind/gsttypefindfunctions.c: (plugin_init):
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Recognise ancient RealAudio files (see #349779).
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2006-08-03 09:01:25 +0000 Jens Granseuer <jensgr@gmx.net>
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gst/typefind/gsttypefindfunctions.c: Add typefinder for Interplay's MVE format (#348973).
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Original commit message from CVS:
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Patch by: Jens Granseuer <jensgr at gmx net>
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* gst/typefind/gsttypefindfunctions.c: (plugin_init):
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Add typefinder for Interplay's MVE format (#348973).
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2006-08-02 17:03:29 +0000 Marcel Moreaux <marcelm@luon.net>
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gst-libs/gst/rtp/gstbasertpdepayload.*: Handle RTP sequence number rollover.
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Original commit message from CVS:
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Patch by: Marcel Moreaux <marcelm at luon dot net>
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* gst-libs/gst/rtp/gstbasertpdepayload.c:
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(gst_base_rtp_depayload_add_to_queue):
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* gst-libs/gst/rtp/gstbasertpdepayload.h:
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Handle RTP sequence number rollover.
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Disable jitterbuffer by default.
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2006-08-02 16:56:19 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/gdp/gstgdpdepay.c: Disable seeking.
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Original commit message from CVS:
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* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
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(gst_gdp_depay_finalize), (gst_gdp_depay_sink_event),
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(gst_gdp_depay_src_event), (gst_gdp_depay_chain),
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(gst_gdp_depay_change_state):
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Disable seeking.
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Small cleanups.
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Clear adapter on disconts.
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Clear caps when going to READY instead of NULL
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* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
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(gst_gdp_pay_init), (gst_gdp_pay_finalize), (gst_gdp_pay_reset),
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(gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
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(gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
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(gst_gdp_queue_buffer), (gst_gdp_pay_chain),
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(gst_gdp_pay_sink_event), (gst_gdp_pay_src_event),
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(gst_gdp_pay_change_state):
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* gst/gdp/gstgdppay.h:
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Reset payloader when going to READY.
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Fix leaked buffers in ->queue on push errors.
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Disable seeking.
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Code cleanups.
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Create packetizer in _init, free in _finalize.
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2006-07-31 08:48:36 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/gdp/gstgdpdepay.c: Consume all events except EOS because we generate events from the gdp payload instead. Fixes #...
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|
Original commit message from CVS:
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* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init),
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(gst_gdp_depay_sink_event), (gst_gdp_depay_chain):
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Consume all events except EOS because we generate events from
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the gdp payload instead. Fixes #349204
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2006-07-28 17:17:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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gst/audioresample/gstaudioresample.c: Don't leak references to the incoming caps. Clean them up when stopping.
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Original commit message from CVS:
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* gst/audioresample/gstaudioresample.c: (audioresample_stop),
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(audioresample_set_caps):
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Don't leak references to the incoming caps. Clean them up when
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stopping.
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* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
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(gst_video_scale_finalize):
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Don't leak our temporary pixel buffer.
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* tests/check/Makefile.am:
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* tests/check/pipelines/simple-launch-lines.c: (run_pipeline),
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(GST_START_TEST), (simple_launch_lines_suite):
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Fix leaks and re-enable the test for valgrind checking.
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2006-07-28 16:39:31 +0000 Sjoerd Simons <sjoerd@luon.net>
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gst/typefind/gsttypefindfunctions.c: Add typefind function for multipart/x-mixed-replace (#348916).
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Original commit message from CVS:
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|
Patch by: Sjoerd Simons <sjoerd at luon net>
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* gst/typefind/gsttypefindfunctions.c: (multipart_type_find),
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(plugin_init):
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Add typefind function for multipart/x-mixed-replace (#348916).
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2006-07-28 14:14:58 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/adder/gstadder.c: Fix leak in duration query.
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Original commit message from CVS:
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* gst/adder/gstadder.c: (gst_adder_setcaps),
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(gst_adder_query_duration):
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Fix leak in duration query.
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Reflow some docs and notes.
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2006-07-28 13:42:48 +0000 Michael Smith <msmith@xiph.org>
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tests/check/pipelines/vorbisenc.c: Enable Andy's extra vorbisenc test, now that it passes. Also fix one aspect of it.
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|
Original commit message from CVS:
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* tests/check/pipelines/vorbisenc.c: (GST_START_TEST),
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(vorbisenc_suite):
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Enable Andy's extra vorbisenc test, now that it passes. Also fix one
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aspect of it.
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2006-07-28 12:48:21 +0000 Michael Smith <msmith@xiph.org>
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ext/vorbis/vorbisenc.*: Handle discontinuities in the input vorbis stream correctly, so that the output is properly t...
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|
Original commit message from CVS:
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* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_generate_sink_caps),
|
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(gst_vorbis_enc_sink_getcaps), (gst_vorbis_enc_buffer_from_packet),
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(gst_vorbis_enc_push_buffer),
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(gst_vorbis_enc_buffer_check_discontinuous),
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(gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
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* ext/vorbis/vorbisenc.h:
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Handle discontinuities in the input vorbis stream correctly,
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so that the output is properly timestamped (and has good granulepos
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values). Needs some oggmux fixes too.
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2006-07-27 10:52:52 +0000 Kai Vehmanen <kv2004@eca.cx>
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gst-libs/gst/rtp/gstbasertpdepayload.c: Don't send multiple newsegments with different formats.
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|
Original commit message from CVS:
|
|
patch by: Kai Vehmanen <kv2004 eca cx>
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* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
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(gst_base_rtp_depayload_chain),
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(gst_base_rtp_depayload_handle_sink_event),
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(gst_base_rtp_depayload_change_state):
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Don't send multiple newsegments with different formats.
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Fixes #348677.
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2006-07-26 15:20:56 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/ogg/gstoggdemux.c: Make seeking in ogg more accurate again by doing the more correct granuletime to stream time c...
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
|
|
(gst_ogg_demux_do_seek), (gst_ogg_demux_read_chain):
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Make seeking in ogg more accurate again by doing the more correct
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granuletime to stream time conversion.
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2006-07-26 10:59:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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gst/tcp/gstmultifdsink.c: debug a little more understandably do not use goto as a substitute for break, especially if...
|
|
Original commit message from CVS:
|
|
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_add_full),
|
|
(gst_multi_fd_sink_new_client):
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|
debug a little more understandably
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do not use goto as a substitute for break, especially if
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break is also being used
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|
2006-07-26 10:55:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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|
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* gst/tcp/gsttcp.c:
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|
move a recurring normal event to LOG, where it should be
|
|
Original commit message from CVS:
|
|
move a recurring normal event to LOG, where it should be
|
|
|
|
2006-07-26 10:54:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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|
|
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* ext/vorbis/vorbisdec.c:
|
|
tweak debug output
|
|
Original commit message from CVS:
|
|
tweak debug output
|
|
|
|
2006-07-26 10:52:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
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gst/gdp/gstgdpdepay.c: proxying get/set caps is the wrong thing to do, since we really do change caps quite fundament...
|
|
Original commit message from CVS:
|
|
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_init):
|
|
proxying get/set caps is the wrong thing to do, since we really
|
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do change caps quite fundamentally
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* tests/check/elements/gdpdepay.c:
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* tests/check/elements/gdppay.c:
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remove declaration of buffers, it's already done in gstcheck.h
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2006-07-26 10:31:06 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/playback/: Remove GLib-2.6 compatibility cruft.
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|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
|
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* gst/playback/gstplaybin.c: (gst_play_bin_get_property):
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Remove GLib-2.6 compatibility cruft.
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2006-07-24 16:47:10 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to align a sample to an unknown value.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
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(gst_base_audio_sink_render):
|
|
Don't try to align a sample to an unknown value.
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2006-07-24 15:14:17 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst-libs/gst/audio/gstbaseaudiosink.c: When the audio clock is slaved to another clock, never try to align samples bu...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_provide_clock), (gst_base_audio_sink_render):
|
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When the audio clock is slaved to another clock, never try to align
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samples but trust the rate interpolation algorithm.
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2006-07-24 14:34:42 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/alsa/gstalsasink.c: Don't try to calculate silence samples, base class does this much better now.
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
|
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Don't try to calculate silence samples, base class does this much
|
|
better now.
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* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
|
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(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps),
|
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(gst_ring_buffer_acquire):
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Calculate silence samples correctly.
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* gst-libs/gst/audio/gstringbuffer.h:
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Add _CAST macro.
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2006-07-22 17:01:12 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/typefind/gsttypefindfunctions.c: Limit search for the first markup tag to the first few kB of the file. If we don...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (xml_check_first_element):
|
|
Limit search for the first markup tag to the first few kB of
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|
the file. If we don't find one there, it's highly unlikely that
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this is an XML(-ish) file.
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2006-07-21 17:04:06 +0000 Andy Wingo <wingo@pobox.com>
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|
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tests/check/pipelines/theoraenc.c (test_discontinuity): Similar test to the one in vorbisenc. Also commented out.
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|
Original commit message from CVS:
|
|
2006-07-21 Andy Wingo <wingo@pobox.com>
|
|
* tests/check/pipelines/theoraenc.c (test_discontinuity): Similar
|
|
test to the one in vorbisenc. Also commented out.
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2006-07-21 16:54:19 +0000 Andy Wingo <wingo@pobox.com>
|
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tests/check/pipelines/vorbisenc.c: New test, commented out until Mike lands some elite vorbisenc patches.
|
|
Original commit message from CVS:
|
|
2006-07-21 Andy Wingo <wingo@pobox.com>
|
|
* tests/check/pipelines/vorbisenc.c:
|
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(test_discontinuity): New test, commented out until Mike lands
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some elite vorbisenc patches.
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2006-07-21 15:59:24 +0000 Andy Wingo <wingo@pobox.com>
|
|
|
|
tests/check/pipelines/: Port to bufferstraw.
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|
Original commit message from CVS:
|
|
2006-07-21 Andy Wingo <wingo@pobox.com>
|
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* tests/check/pipelines/vorbisenc.c:
|
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* tests/check/pipelines/theoraenc.c: Port to bufferstraw.
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Bufferstraw was actually factored out of these tests. Now we share
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code yay.
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2006-07-21 11:03:28 +0000 Wim Taymans <wim.taymans@gmail.com>
|
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ext/theora/theoradec.c: Better clipping.
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|
Original commit message from CVS:
|
|
* ext/theora/theoradec.c: (clip_buffer):
|
|
Better clipping.
|
|
|
|
2006-07-21 10:43:54 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst-libs/gst/audio/gstaudiosink.c: Fix leak.
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|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstaudiosink.c: (audioringbuffer_thread_func),
|
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(gst_audioringbuffer_finalize), (gst_audioringbuffer_acquire),
|
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(gst_audioringbuffer_release), (gst_audioringbuffer_stop):
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Fix leak.
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Avoid type casting when we can.
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* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_dispose):
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Fix mem leak.
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2006-07-20 16:57:29 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/alsa/gstalsamixerelement.c: Make state change fail if the specified device can't be opened for some reason.
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|
Original commit message from CVS:
|
|
* ext/alsa/gstalsamixerelement.c:
|
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(gst_alsa_mixer_element_change_state):
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Make state change fail if the specified device can't be opened
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for some reason.
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2006-07-20 10:42:21 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/test.c: Example of a small audio/video player using decodebin.
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|
Original commit message from CVS:
|
|
* gst/playback/test.c: (gen_video_element), (gen_audio_element),
|
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(cb_newpad), (main):
|
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Example of a small audio/video player using decodebin.
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2006-07-20 05:56:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst-libs/gst/riff/riff-ids.h: Add 'fact' chunk id
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|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-ids.h:
|
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Add 'fact' chunk id
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|
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2006-07-19 18:20:43 +0000 Wim Taymans <wim.taymans@gmail.com>
|
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gst-libs/gst/rtp/gstbasertpdepayload.c: Don't assert when not negotiated but post a meaningfull error message. Fixes ...
|
|
Original commit message from CVS:
|
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* gst-libs/gst/rtp/gstbasertpdepayload.c:
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(gst_base_rtp_depayload_chain),
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(gst_base_rtp_depayload_change_state):
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Don't assert when not negotiated but post a meaningfull
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error message. Fixes #347918.
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* gst-libs/gst/rtp/gstbasertppayload.c:
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Add comment about better default MTU size.
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* gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data):
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Small cleanups, start docs.
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2006-07-19 14:46:36 +0000 Martin Szulecki <compiz@sukimashita.com>
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sys/v4l/gstv4lelement.c: If "device-name" is requested and the device is not open, try to temporarily open it to obta...
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Original commit message from CVS:
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Patch by: Martin Szulecki
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* sys/v4l/gstv4lelement.c: (gst_v4lelement_get_property):
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If "device-name" is requested and the device is not
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open, try to temporarily open it to obtain this
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information (#342494).
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2006-07-19 12:25:00 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/tag/gstid3tag.c: Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
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Original commit message from CVS:
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* gst-libs/gst/tag/gstid3tag.c:
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Add TSSE <=> GST_TAG_ENCODER mapping (see #347898).
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* gst-libs/gst/tag/gsttageditingprivate.h:
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* gst-libs/gst/tag/gstvorbistag.c:
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Some more random const-ifications.
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2006-07-18 19:48:48 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst-libs/gst/riff/: add comment what those 16 bytes in struct _gst_riff_strh according to one avi-dumper are
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Original commit message from CVS:
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* gst-libs/gst/riff/riff-ids.h:
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* gst-libs/gst/riff/riff-media.c:
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(gst_riff_create_video_template_caps):
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Add more FOURCCs (sort list to make stuff easier to find),
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add comment what those 16 bytes in struct _gst_riff_strh according to
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one avi-dumper are
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2006-07-17 14:17:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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gst/gdp/: remove parent_class setting, BOILERPLATE does this fix typo in comment
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Original commit message from CVS:
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2006-07-17 Thomas Vander Stichele <thomas at apestaart dot org>
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* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_class_init):
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* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
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remove parent_class setting, BOILERPLATE does this
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(gst_gdp_pay_reset_streamheader):
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fix typo in comment
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2006-07-17 13:48:10 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/audio/multichannel.c: Const-ify two arrays.
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Original commit message from CVS:
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* gst-libs/gst/audio/multichannel.c:
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(gst_audio_check_channel_positions),
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(gst_audio_fixate_channel_positions):
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Const-ify two arrays.
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2006-07-17 12:33:42 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/alsa/gstalsa.c: Fix typo, so that alsasink also advertises 8 channels if that's supported (tags: can, worms, open...
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Original commit message from CVS:
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* ext/alsa/gstalsa.c: (caps_add_channel_configuration):
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Fix typo, so that alsasink also advertises 8 channels
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if that's supported (tags: can, worms, open, alsa, ph34r).
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2006-07-17 12:01:04 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/ogg/gstoggdemux.c: *sigh*, when is the compiler going to warn when the comments are out-of-sync with the code.. R...
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Original commit message from CVS:
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* ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
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(gst_ogg_pad_submit_packet), (gst_ogg_demux_read_chain):
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*sigh*, when is the compiler going to warn when the comments
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are out-of-sync with the code.. Refix case of busted theora
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headers with 0 granule pos.
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2006-07-14 17:56:59 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/rtp/gstbasertpdepayload.c: Fix 99% cpu load by waiting for absolute times on the clock. Fixes #347300.
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Original commit message from CVS:
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* gst-libs/gst/rtp/gstbasertpdepayload.c:
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(gst_base_rtp_depayload_wait),
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(gst_base_rtp_depayload_change_state),
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(gst_base_rtp_depayload_set_property),
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(gst_base_rtp_depayload_get_property):
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Fix 99% cpu load by waiting for absolute times on the
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clock. Fixes #347300.
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2006-07-14 17:07:08 +0000 Andy Wingo <wingo@pobox.com>
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ext/theora/gsttheoraparse.h: ext/theora/theoraparse.c (theora_parse_drain_event_queue) (theora_parse_push_headers, th...
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Original commit message from CVS:
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2006-07-14 Andy Wingo <wingo@pobox.com>
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* ext/theora/gsttheoraparse.h:
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* ext/theora/theoraparse.c (theora_parse_drain_event_queue)
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(theora_parse_push_headers, theora_parse_clear_queue)
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(theora_parse_drain_queue_prematurely, )
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(theora_parse_sink_event, theora_parse_change_state): Queue events
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until we initialized our state, like in vorbisparse.
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2006-07-14 16:45:17 +0000 Iain * <iaingnome@gmail.com>
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ext/vorbis/vorbisparse.h: ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue) (vorbis_parse_push_headers, vorbi...
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Original commit message from CVS:
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2006-07-14 Andy Wingo <wingo@pobox.com>
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* ext/vorbis/vorbisparse.h:
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* ext/vorbis/vorbisparse.c (vorbis_parse_drain_event_queue)
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(vorbis_parse_push_headers, vorbis_parse_clear_queue)
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(vorbis_parse_drain_queue_prematurely, )
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(vorbis_parse_sink_event, vorbis_parse_change_state): Queue events
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until we have initialized our state. Fixes seeking after an
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initial pad block.
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2006-07-14 Andy Wingo <wingo@pobox.com>
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Patch by: Iain * <iaingnome@gmail.com>
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* ext/ogg/gstoggdemux.c (gst_ogg_demux_finalize): Fix memleak.
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2006-07-14 15:52:39 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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configure.ac: Bump nano back to CVS
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Original commit message from CVS:
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* configure.ac:
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Bump nano back to CVS
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=== release 0.10.9 ===
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2006-07-14 15:51:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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configure.ac: releasing 0.10.9, "I walk the line"
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Original commit message from CVS:
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2006-07-13 Jan Schmidt <thaytan@mad.scientist.com>
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* configure.ac:
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releasing 0.10.9, "I walk the line"
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2006-07-14 14:12:40 +0000 Michael Smith <msmith@xiph.org>
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tests/check/pipelines/vorbisenc.c: Move a g_cond_signal to earlier to avoid sometimes deadlocking (commonly happens w...
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Original commit message from CVS:
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* tests/check/pipelines/vorbisenc.c: (stop_pipeline):
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Move a g_cond_signal to earlier to avoid sometimes deadlocking
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(commonly happens when running this test under valgrind) when trying
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to remove the buffer probe.
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2006-07-14 10:34:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* gst/gdp/Makefile.am:
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build as a plugin, not a lib
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Original commit message from CVS:
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build as a plugin, not a lib
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2006-07-13 16:43:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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sys/xvimage/xvimagesink.c: Fix missing g_unlock from the previous commit
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Original commit message from CVS:
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new):
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Fix missing g_unlock from the previous commit
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2006-07-13 16:34:04 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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sys/: Implement a locking order to ensure we always take the object lock before the x_lock and never vice-versa.
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Original commit message from CVS:
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* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
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(gst_ximagesink_change_state):
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
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(gst_xvimagesink_change_state):
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Implement a locking order to ensure we always take the object lock
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before the x_lock and never vice-versa.
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2006-07-13 15:25:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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docs/plugins/: add more plugins and elements to docs
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Original commit message from CVS:
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
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* docs/plugins/gst-plugins-bad-plugins-sections.txt:
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add more plugins and elements to docs
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* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
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fix segfaults due to wrong g_free
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add example
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* gst/gdp/gstgdppay.c:
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add example
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2006-07-13 14:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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gst/playback/gstdecodebin.c: Fix a caps leak when linking (#347304)
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Original commit message from CVS:
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* gst/playback/gstdecodebin.c: (find_compatibles):
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Fix a caps leak when linking (#347304)
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* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
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(gst_ximagesink_ximage_destroy), (gst_ximagesink_xcontext_clear),
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(gst_ximagesink_change_state):
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* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
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(gst_xvimage_buffer_finalize), (gst_xvimagesink_check_xshm_calls),
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(gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
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(gst_xvimagesink_xcontext_clear), (gst_xvimagesink_change_state):
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Don't leak shared memory resources. Use the object lock to protect
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against the xcontext disappearing while returning a buffer from the
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pipeline. (#347304)
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2006-07-12 14:20:43 +0000 Edward Hervey <bilboed@bilboed.com>
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ext/vorbis/vorbisdec.c: gst_tag_list_merge() returns a new object. Take that into account when using it. This avoids ...
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Original commit message from CVS:
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* ext/vorbis/vorbisdec.c: (vorbis_dec_finalize),
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(vorbis_handle_comment_packet):
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gst_tag_list_merge() returns a new object. Take that into account when
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using it. This avoids memleak.
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Revert previous commit which is not needed.
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2006-07-12 13:30:20 +0000 Edward Hervey <bilboed@bilboed.com>
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ext/vorbis/vorbisdec.c: Reset the decoder in finalize so that all fields get cleared.
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Original commit message from CVS:
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* ext/vorbis/vorbisdec.c: (vorbis_dec_finalize):
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Reset the decoder in finalize so that all fields get cleared.
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2006-07-12 13:24:19 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstbaseaudiosrc.c: Don't try to post an error message when setting the clock fails as this can hap...
|
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Original commit message from CVS:
|
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* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
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(gst_base_audio_src_set_clock),
|
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(gst_base_audio_src_check_get_range), (gst_base_audio_src_create):
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Don't try to post an error message when setting the clock fails
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as this can happen when adding an element to a bin which will then
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deadlock. Fixes #347296.
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2006-07-12 13:04:15 +0000 Edward Hervey <bilboed@bilboed.com>
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ext/vorbis/vorbisdec.c: Post tag messages on the bus even if we're not initialized.
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Original commit message from CVS:
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* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
|
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(vorbis_dec_sink_event), (vorbis_handle_comment_packet),
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(vorbis_handle_type_packet):
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Post tag messages on the bus even if we're not initialized.
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If we're not initialized, we still postpone the event pushing of tags.
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2006-07-12 11:28:37 +0000 Wim Taymans <wim.taymans@gmail.com>
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Revert last two changes that broke the freeze.
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Original commit message from CVS:
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|
* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
|
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* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
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(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
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Revert last two changes that broke the freeze.
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2006-07-12 10:59:55 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/alsa/gstalsasink.c: basesink calculates silence sample correctly for us.
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Original commit message from CVS:
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* ext/alsa/gstalsasink.c: (set_hwparams), (gst_alsasink_prepare):
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basesink calculates silence sample correctly for us.
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2006-07-12 10:58:42 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstringbuffer.c: Calculate correct silence samples so we don't fill our ringbuffer with noise.
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Original commit message from CVS:
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* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format),
|
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(gst_ring_buffer_debug_spec_caps), (gst_ring_buffer_parse_caps):
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Calculate correct silence samples so we don't fill our ringbuffer
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with noise.
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2006-07-12 10:37:18 +0000 Edward Hervey <bilboed@bilboed.com>
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ext/vorbis/vorbisdec.*: Delay sending events (newsegment, tags) until the decoder is properly initialized.
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Original commit message from CVS:
|
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* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_init),
|
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(gst_vorbis_dec_reset), (vorbis_dec_sink_event),
|
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(vorbis_handle_comment_packet), (vorbis_handle_type_packet):
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* ext/vorbis/vorbisdec.h:
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Delay sending events (newsegment, tags) until the decoder is properly
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initialized.
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Fixes #347295
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2006-07-11 22:40:13 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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* po/af.po:
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* po/az.po:
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* po/cs.po:
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* po/en_GB.po:
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* po/hu.po:
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* po/it.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/sq.po:
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* po/sr.po:
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* po/sv.po:
|
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* po/uk.po:
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* po/vi.po:
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Update .po files
|
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Original commit message from CVS:
|
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Update .po files
|
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2006-07-11 21:04:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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tests/check/elements/audioconvert.c: Patch from #347221 adding a test for audioconvert channel remappings.
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|
Original commit message from CVS:
|
|
* tests/check/elements/audioconvert.c: (get_float_mc_caps),
|
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(get_int_mc_caps), (GST_START_TEST), (audioconvert_suite):
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Patch from #347221 adding a test for audioconvert
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channel remappings.
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2006-07-11 12:03:25 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/subparse/gstssaparse.c: Don't include the terminating NUL in the buffer size, it's only there for extra paranoia ...
|
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Original commit message from CVS:
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* gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
|
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(gst_ssa_parse_parse_line):
|
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Don't include the terminating NUL in the buffer size,
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it's only there for extra paranoia (would add random
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'*' characters at the end of each subtitle since the
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terminator itself is not valid UTF-8 technically).
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Also fix indenting after boilerplate macro.
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2006-07-10 14:59:03 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gstdecodebin.c: Also emit 'unknown-type' signal (which should really be called unhandled-type) if we fou...
|
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Original commit message from CVS:
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* gst/playback/gstdecodebin.c: (close_pad_link):
|
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Also emit 'unknown-type' signal (which should really be
|
|
called unhandled-type) if we found potential decoders/demuxers
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in the registry but none of them worked in the end (as in the
|
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case where the plugins don't exist any longer but are still
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listed in the registry). Fixes #329798.
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2006-07-08 13:48:58 +0000 Andy Wingo <wingo@pobox.com>
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* ChangeLog:
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* ext/theora/theoraparse.c:
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theoraparse.c (theora_parse_push_buffer)
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Original commit message from CVS:
|
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2006-07-08 Andy Wingo <wingo@pobox.com>
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* theoraparse.c (theora_parse_push_buffer)
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(theora_parse_drain_queue_prematurely, theora_parse_drain_queue):
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Add some more debugging. Fix granulepos reconstruction in the face
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of discontinuities.
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2006-07-06 15:54:50 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstbaseaudiosink.c: Use gobject_class instead of G_OBJECT_CLASS (klass)
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Original commit message from CVS:
|
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* gst-libs/gst/audio/gstbaseaudiosink.c:
|
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(gst_base_audio_sink_class_init),
|
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(gst_base_audio_sink_provide_clock):
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Use gobject_class instead of G_OBJECT_CLASS (klass)
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* gst-libs/gst/audio/gstbaseaudiosrc.c:
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(gst_base_audio_src_class_init), (gst_base_audio_src_init),
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(gst_base_audio_src_set_clock), (gst_base_audio_src_provide_clock),
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(gst_base_audio_src_get_time),
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(gst_base_audio_src_check_get_range), (gst_base_audio_src_create),
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(gst_base_audio_src_create_ringbuffer):
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Fix latency and buffer-time constants and properties ala basesink.
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Implement pull based scheduling. Fixes #346527.
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Set default blocksize in GstBaseSrc to 0, we default to pushing out
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one segment.
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Refuse slaving to another clock instead of silently not working.
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Only provide a clock when we are actually able to do so.
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Various small cleanups and compiler hints.
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2006-07-06 13:23:07 +0000 Lutz Mueller <lutz@topfrose.de>
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gst/typefind/gsttypefindfunctions.c: Add typefinding for text/html (#346581).
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Original commit message from CVS:
|
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Patch by: Lutz Mueller <lutz at topfrose de>
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* gst/typefind/gsttypefindfunctions.c: (html_type_find),
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(plugin_init):
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Add typefinding for text/html (#346581).
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2006-07-06 13:12:02 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/typefind/gsttypefindfunctions.c: Fix SMIL typefinding, make xml_check_first_element() more useful.
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Original commit message from CVS:
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* gst/typefind/gsttypefindfunctions.c: (utf8_type_find),
|
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(xml_check_first_element), (xml_type_find), (smil_type_find):
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Fix SMIL typefinding, make xml_check_first_element() more
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useful.
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2006-07-06 13:04:24 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gstplaybasebin.*: Protect list of elements with a subtitle-encoding property and the subtitle encoding m...
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Original commit message from CVS:
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* gst/playback/gstplaybasebin.c: (gst_play_base_bin_init),
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(gst_play_base_bin_finalize), (decodebin_element_added_cb),
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(decodebin_element_removed_cb), (gst_play_base_bin_set_property):
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* gst/playback/gstplaybasebin.h:
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Protect list of elements with a subtitle-encoding property and
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the subtitle encoding member itself with a lock of their own
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instead of using the object lock. This prevents a dead-lock in
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the element-remove callback in some circumstances when shutting
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down playbin.
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2006-07-05 20:11:13 +0000 Sébastien Moutte <sebastien@moutte.net>
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win32/common/libgsttag.def: Export some new functions.
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Original commit message from CVS:
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* win32/common/libgsttag.def:
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Export some new functions.
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* win32/vs6/libgstogg.dsp:
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Add a link to libgsttag-0.10.lib.
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2006-07-04 16:50:21 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/alsa/gstalsamixertrack.c: Some const-ification.
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Original commit message from CVS:
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* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_new):
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Some const-ification.
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2006-07-04 14:06:03 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/playback/gstplaybasebin.c: Improve checking if we are dealing with a stream. Added some more uris that need buffe...
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Original commit message from CVS:
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* gst/playback/gstplaybasebin.c: (is_stream), (gen_source_element):
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Improve checking if we are dealing with a stream. Added some
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more uris that need buffering.
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2006-07-03 10:43:31 +0000 Edward Hervey <bilboed@bilboed.com>
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ext/vorbis/vorbisdec.c: Remove unused variable.
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Original commit message from CVS:
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* ext/vorbis/vorbisdec.c: (vorbis_do_clip):
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Remove unused variable.
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2006-07-02 21:48:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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Makefile.am: include lcov.mak
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Original commit message from CVS:
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* Makefile.am:
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include lcov.mak
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* configure.ac:
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add GCOV_LIBS to GST_LIBS
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2006-07-02 11:08:58 +0000 Michael Sheldon <webmaster@mikeasoft.com>
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ext/alsa/gstalsasrc.c: Add 32 bps to template caps and increase channels range from [1,2] to [1,MAX]. See #346326.
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Original commit message from CVS:
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Patch by: Michael Sheldon <webmaster at mikeasoft com>
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* ext/alsa/gstalsasrc.c:
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Add 32 bps to template caps and increase channels range
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from [1,2] to [1,MAX]. See #346326.
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2006-06-30 12:04:51 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/riff/riff-media.c: Recognise 'WMVA' video codec fourcc (#345879).
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Original commit message from CVS:
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* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
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Recognise 'WMVA' video codec fourcc (#345879).
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2006-06-29 12:21:06 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
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gst-libs/gst/rtp/gstbasertpaudiopayload.c: Fixed nasty memory leak
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Original commit message from CVS:
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* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
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Fixed nasty memory leak
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2006-06-26 13:19:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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gst/tcp/gsttcp.c: fix logging
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Original commit message from CVS:
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* gst/tcp/gsttcp.c: (gst_tcp_read_buffer),
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(gst_tcp_gdp_read_buffer), (gst_tcp_gdp_read_caps):
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fix logging
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2006-06-23 16:45:50 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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gst/playback/gstdecodebin.c: Protect remove_fakesink using a mutex, so that we don't try and remove the fakesink simu...
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|
Original commit message from CVS:
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* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init),
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(gst_decode_bin_init), (gst_decode_bin_finalize), (add_fakesink),
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(remove_fakesink), (pad_probe), (gst_decode_bin_change_state):
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Protect remove_fakesink using a mutex, so that we don't try and
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remove the fakesink simultaneously from multiple threads.
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When going from READY to PAUSED, restore the fakesink, so that
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it is there when decodebin gets reused.
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2006-06-23 09:53:09 +0000 Tim-Philipp Müller <tim@centricular.net>
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Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
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Original commit message from CVS:
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* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
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* gst-libs/gst/rtp/gstbasertpdepayload.c:
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* gst-libs/gst/rtp/gstbasertppayload.c:
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* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
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* gst/tcp/gstmultifdsink.c:
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* gst/tcp/gsttcpclientsink.c:
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* gst/tcp/gsttcpclientsrc.c:
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* gst/tcp/gsttcpserversink.c:
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* gst/tcp/gsttcpserversrc.c:
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* gst/videorate/gstvideorate.c:
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* gst/videotestsrc/gstvideotestsrc.c:
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* sys/v4l/gstv4ljpegsrc.c:
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* sys/v4l/gstv4lmjpegsink.c:
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* sys/v4l/gstv4lsrc.c:
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* tests/examples/seek/scrubby.c:
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* tests/examples/seek/seek.c:
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Use GST_DEBUG_CATEGORY_STATIC where possible (#342503).
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2006-06-23 09:09:44 +0000 Tim-Philipp Müller <tim@centricular.net>
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Use GST_DEBUG_CATEGORY_STATIC where possible (#342503) and fix one GObject boilerplate macro.
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Original commit message from CVS:
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* ext/directfb/dfbvideosink.c:
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* ext/gsm/gstgsmdec.c:
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* ext/gsm/gstgsmenc.c:
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* ext/libmms/gstmms.c:
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* ext/neon/gstneonhttpsrc.c:
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* ext/theora/theoradec.c:
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* gst/freeze/gstfreeze.c:
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* gst/gdp/gstgdpdepay.c:
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* gst/gdp/gstgdppay.c:
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* sys/glsink/glimagesink.c:
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Use GST_DEBUG_CATEGORY_STATIC where possible (#342503)
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and fix one GObject boilerplate macro.
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2006-06-22 12:13:31 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/tag/tags.c: Second field in GEnumValue shouldn't be a description, but a stringified version of the enum...
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|
Original commit message from CVS:
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* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum):
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Second field in GEnumValue shouldn't be a description,
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but a stringified version of the enum value.
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2006-06-22 12:03:14 +0000 Wim Taymans <wim.taymans@gmail.com>
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sys/ximage/ximagesink.c: Avoid type checking in buffer casts.
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Original commit message from CVS:
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* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
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(gst_ximage_buffer_free), (gst_ximagesink_ximage_put),
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(gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
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Avoid type checking in buffer casts.
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Avoid caps copy in buffer_alloc when we can.
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Use pad_peer_accept.
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2006-06-22 11:01:41 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/tag/tag.h: Oops, make that 'Since: 0.10.9'.
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Original commit message from CVS:
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* gst-libs/gst/tag/tag.h:
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Oops, make that 'Since: 0.10.9'.
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|
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2006-06-22 10:55:05 +0000 Tim-Philipp Müller <tim@centricular.net>
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API: add GstTagImageType enum to describe images contained in image tags (#345641).
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Original commit message from CVS:
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|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
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* gst-libs/gst/tag/tag.h:
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* gst-libs/gst/tag/tags.c: (register_tag_image_type_enum),
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(gst_tag_image_type_get_type):
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API: add GstTagImageType enum to describe images contained
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in image tags (#345641).
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|
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2006-06-22 10:31:22 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/tcp/gstmultifdsink.c: Fix warnings with gst-inspect: "buffers-min" property should be of G_TYPE_INT and not G_TYP...
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|
Original commit message from CVS:
|
|
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
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Fix warnings with gst-inspect: "buffers-min" property
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should be of G_TYPE_INT and not G_TYPE_INT64. Also fix
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typo in property description.
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2006-06-22 10:10:51 +0000 Cody Russell <bratsche@gnome.org>
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gst/: Avoid unnecessary class cast check in class_init functions (#337747).
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|
Original commit message from CVS:
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|
Patch by: Cody Russell <bratsche at gnome org>
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* gst/audioresample/gstaudioresample.c:
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|
(gst_audioresample_class_init):
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* gst/playback/gststreamselector.c:
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(gst_stream_selector_class_init):
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* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
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* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
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* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
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* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init):
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* gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init):
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* gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init):
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* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init):
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* gst/videotestsrc/gstvideotestsrc.c:
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(gst_video_test_src_class_init):
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* gst/volume/gstvolume.c: (gst_volume_class_init):
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Avoid unnecessary class cast check in class_init
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functions (#337747).
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2006-06-21 18:39:07 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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ext/pango/gsttextoverlay.c: g_markup_escape_text() REALLY doesn't like non-UTF8 input and doesn't validate its input ...
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|
Original commit message from CVS:
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|
* ext/pango/gsttextoverlay.c: (gst_text_overlay_make_utf8),
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|
(gst_text_overlay_video_chain):
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g_markup_escape_text() REALLY doesn't like non-UTF8 input
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and doesn't validate its input either (and neither did
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textoverlay it seems). Let's do that then and fix #345206.
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2006-06-19 17:12:57 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst/tcp/gstmultifdsink.*: Added shiny new burst-on-connect methods.
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|
Original commit message from CVS:
|
|
* gst/tcp/gstmultifdsink.c: (gst_sync_method_get_type),
|
|
(gst_unit_type_get_type), (gst_multi_fd_sink_class_init),
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|
(gst_multi_fd_sink_init), (gst_multi_fd_sink_add_full),
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|
(gst_multi_fd_sink_add), (gst_multi_fd_sink_handle_client_read),
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|
(find_syncframe), (find_limits), (assign_value),
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(count_burst_unit), (gst_multi_fd_sink_new_client),
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|
(gst_multi_fd_sink_handle_client_write),
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(gst_multi_fd_sink_queue_buffer), (gst_multi_fd_sink_render),
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(gst_multi_fd_sink_set_property), (gst_multi_fd_sink_get_property),
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(gst_multi_fd_sink_change_state):
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* gst/tcp/gstmultifdsink.h:
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Added shiny new burst-on-connect methods.
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Add properties to control the minimal amount of data queued.
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Small cleanups.
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API: bytes-min property
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API: time-min property
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API: buffers-min property
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API: burst-unit property
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API: burst-value property
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API: add-full signal
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* gst/tcp/gsttcp-marshal.list:
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Added new marshaller code for the new signal.
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* tests/check/elements/multifdsink.c: (GST_START_TEST),
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(multifdsink_suite):
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Added testcases for new burst methods.
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|
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2006-06-19 11:35:47 +0000 Christian Schaller <uraeus@gnome.org>
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|
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* gst-plugins-base.spec.in:
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|
update for latest changes
|
|
Original commit message from CVS:
|
|
update for latest changes
|
|
|
|
2006-06-19 09:57:50 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
ext/theora/theoradec.c: Implement clipping for accurate seeking.
|
|
Original commit message from CVS:
|
|
* ext/theora/theoradec.c: (clip_buffer), (theora_dec_push):
|
|
Implement clipping for accurate seeking.
|
|
Closes #345225
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|
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2006-06-19 09:08:05 +0000 Philip Jaegenstedt <philip@lysator.liu.se>
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|
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gst/videoscale/gstvideoscale.c: Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
|
|
Original commit message from CVS:
|
|
Patch by: Philip Jaegenstedt <philip at lysator dot liu dot se>
|
|
* gst/videoscale/gstvideoscale.c: (gst_video_scale_prepare_size),
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(gst_video_scale_transform):
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Make videoscale support RGBA, ARGB, BGRA and ABGR. Fixes #345131
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|
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2006-06-17 14:18:41 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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* ChangeLog:
|
|
ChangeLog surgery
|
|
Original commit message from CVS:
|
|
ChangeLog surgery
|
|
|
|
2006-06-17 14:13:03 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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configure.ac: Fix --disable-extern (can't set conditionals conditionally, #343602).
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|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Fix --disable-extern (can't set conditionals conditionally,
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|
#343602).
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|
|
|
2006-06-16 15:43:23 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
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|
tests/check/elements/audioresample.c: Add test case for bug #342789 fixed below.
|
|
Original commit message from CVS:
|
|
* tests/check/elements/audioresample.c: (test_reuse),
|
|
(audioresample_suite):
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Add test case for bug #342789 fixed below.
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|
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2006-06-16 15:17:44 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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gst/audioresample/gstaudioresample.c: Implement GstBaseTransform::start and ::stop so that audioresample can clear it...
|
|
Original commit message from CVS:
|
|
* gst/audioresample/gstaudioresample.c:
|
|
(gst_audioresample_class_init), (gst_audioresample_init),
|
|
(audioresample_start), (audioresample_stop),
|
|
(gst_audioresample_set_property), (gst_audioresample_get_property):
|
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Implement GstBaseTransform::start and ::stop so that audioresample
|
|
can clear its internal state properly and be reused insted of
|
|
causing non-negotiated errors with playbin under some circumstances
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(#342789).
|
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* tests/check/elements/audioresample.c: (setup_audioresample),
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(cleanup_audioresample):
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Need to set element state here so that ::start and ::stop are
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called.
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|
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2006-06-16 13:59:29 +0000 Young-Ho Cha <ganadist@chollian.net>
|
|
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gst-libs/gst/riff/riff-read.c: Parse extra data better, apparently it's right behind the normal strf header size. Fix...
|
|
Original commit message from CVS:
|
|
Patch by: Young-Ho Cha <ganadist at chollian dot net>
|
|
* gst-libs/gst/riff/riff-read.c: (gst_riff_parse_strf_vids):
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Parse extra data better, apparently it's right behind
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the normal strf header size. Fixes #343500.
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2006-06-16 11:04:21 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
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ext/alsa/gstalsasink.c: If we fail to set the buffer_time and period_time alsa parameters, post a warning and leave a...
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsasink.c: (set_hwparams):
|
|
If we fail to set the buffer_time and period_time alsa
|
|
parameters, post a warning and leave alsa select a
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default instead of failing. Fixes #342085
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2006-06-16 10:30:25 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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* ChangeLog:
|
|
ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
|
|
Original commit message from CVS:
|
|
ChangeLog surgery (it wouldn't have crashed, just shown bogus values)
|
|
|
|
2006-06-16 10:20:10 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
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Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed out in the header file and shouldn't be listed in the docs.
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/cdda/gstcddabasesrc.h:
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Remove GST_CDDA_TAG_TRACK_TAGS again, it is #ifdef 0'ed
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out in the header file and shouldn't be listed in the docs.
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* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
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Fix it so that it doesn't crash in the debug statement.
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2006-06-16 10:02:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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docs/libs/: add remaining symbols into correct setions
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|
Original commit message from CVS:
|
|
* docs/libs/Makefile.am:
|
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* docs/libs/gst-plugins-base-libs-docs.sgml:
|
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* docs/libs/gst-plugins-base-libs-sections.txt:
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* docs/libs/gst-plugins-base-libs.types:
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add remaining symbols into correct setions
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* gst-libs/gst/audio/gstringbuffer.c:
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fix incomplete docs
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* gst-libs/gst/audio/gstringbuffer.h:
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comment out not yet implemented function
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* gst-libs/gst/floatcast/floatcast.h:
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* gst-libs/gst/netbuffer/gstnetbuffer.c:
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add short descriptions
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* gst-libs/gst/interfaces/propertyprobe.c:
|
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fix return value docs
|
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* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
|
|
simplify debug logging
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* gst-libs/gst/riff/riff-read.h:
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sync function prototype and docs
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* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
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remove left over symbol
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2006-06-16 09:56:41 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
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Use GST_PLUGIN_DOCS macro in configure.ac, add
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|
Original commit message from CVS:
|
|
* autogen.sh:
|
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* configure.ac:
|
|
* docs/Makefile.am:
|
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Use GST_PLUGIN_DOCS macro in configure.ac, add
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--enable-plugin-docs default to autogen.sh and use
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ENABLE_PLUGIN_DOCS conditional in Makefile.am (#344039).
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2006-06-15 15:27:49 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/ogg/gstoggdemux.c: Combine GstFlowReturn from the source pads to give a meaningfull result to the upstream peer o...
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Original commit message from CVS:
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* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
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(gst_ogg_demux_activate_chain), (gst_ogg_demux_combine_flows),
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(gst_ogg_demux_loop):
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Combine GstFlowReturn from the source pads to give a
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meaningfull result to the upstream peer or to stop the
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processing task in case of errors.
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2006-06-14 14:49:33 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gststreaminfo.c: Try GST_TAG_CODEC as fallback when extracting the codec name; more debug info.
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Original commit message from CVS:
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* gst/playback/gststreaminfo.c: (cb_probe):
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Try GST_TAG_CODEC as fallback when extracting the
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codec name; more debug info.
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2006-06-14 14:34:28 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/ogg/: Extract language tags from ogm subtitle streams, so that the subtitle menu choices are labelled correctly in
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Original commit message from CVS:
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* ext/ogg/Makefile.am:
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* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
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Extract language tags from ogm subtitle streams, so that
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the subtitle menu choices are labelled correctly in
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Totem (fixes #344708).
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2006-06-14 09:13:54 +0000 Alessandro Decina <alessandro@nnva.org>
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ext/ogg/gstoggmux.c: Fix various leaks. Fixes #343699.
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Original commit message from CVS:
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Patch by: Alessandro Decina <alessandro at nnva dot org>
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* ext/ogg/gstoggmux.c: (gst_ogg_mux_get_type), (gst_ogg_mux_clear),
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(gst_ogg_mux_release_pad), (gst_ogg_mux_get_headers),
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(gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_init_collectpads),
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(gst_ogg_mux_clear_collectpads), (gst_ogg_mux_change_state):
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Fix various leaks. Fixes #343699.
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Add x-smoke mime type.
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2006-06-14 08:17:45 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/riff/riff-ids.h: Add IDs for 'bext' chunks (see #343837).
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Original commit message from CVS:
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* gst-libs/gst/riff/riff-ids.h:
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Add IDs for 'bext' chunks (see #343837).
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2006-06-12 12:44:38 +0000 Young-Ho Cha <ganadist@chollian.net>
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gst/subparse/samiparse.c: Honour font face tags in SAMI subtitles (#344503).
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Original commit message from CVS:
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Patch by: Young-Ho Cha <ganadist at chollian net>
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* gst/subparse/samiparse.c: (sami_context_pop_state),
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(handle_start_font), (end_sami_element):
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Honour font face tags in SAMI subtitles (#344503).
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2006-06-11 20:41:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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po/POTFILES.in: add missing files containing translatable strings
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Original commit message from CVS:
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* po/POTFILES.in:
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add missing files containing translatable strings
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2006-06-11 19:55:32 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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docs/libs/tmpl/.cvsignore: we don't want those *.sgml files in CVS either
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Original commit message from CVS:
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* docs/libs/tmpl/.cvsignore:
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we don't want those *.sgml files in CVS either
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2006-06-11 19:44:49 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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ignore more
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Original commit message from CVS:
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* docs/libs/.cvsignore:
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* tests/check/elements/.cvsignore:
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* tests/check/libs/.cvsignore:
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ignore more
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2006-06-11 18:33:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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docs/libs/Makefile.am: also commiting the changed Makefile.am (added more libs to the doc-build)
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Original commit message from CVS:
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* docs/libs/Makefile.am:
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also commiting the changed Makefile.am (added more libs to the
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doc-build)
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2006-06-11 17:08:26 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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docs/libs/: first batch of reordering things, add index & hierarchy
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Original commit message from CVS:
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* docs/libs/gst-plugins-base-libs-docs.sgml:
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* docs/libs/gst-plugins-base-libs-sections.txt:
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* docs/libs/gst-plugins-base-libs.types:
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first batch of reordering things, add index & hierarchy
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2006-06-11 14:08:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* common:
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* ext/alsa/Makefile.am:
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* ext/cdparanoia/Makefile.am:
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* ext/gnomevfs/Makefile.am:
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* ext/libvisual/Makefile.am:
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* ext/ogg/Makefile.am:
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* ext/pango/Makefile.am:
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* ext/theora/Makefile.am:
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* ext/vorbis/Makefile.am:
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* sys/v4l/Makefile.am:
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* sys/ximage/Makefile.am:
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* sys/xvimage/Makefile.am:
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further clean up build
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|
Original commit message from CVS:
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further clean up build
|
|
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2006-06-11 12:14:18 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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configure.ac: use GST_PKG_CHECK_MODULES, cleans up output
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Original commit message from CVS:
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* configure.ac:
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use GST_PKG_CHECK_MODULES, cleans up output
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|
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2006-06-11 12:10:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* common:
|
|
* win32/common/config.h:
|
|
update to cvs
|
|
Original commit message from CVS:
|
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update to cvs
|
|
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2006-06-10 18:52:03 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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ext/gnomevfs/gstgnomevfsuri.c: Add support for burn:// URIs (#343385); const-ify things a bit, use G_N_ELEMENTS inste...
|
|
Original commit message from CVS:
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|
* ext/gnomevfs/gstgnomevfsuri.c: (gst_gnomevfs_get_supported_uris):
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Add support for burn:// URIs (#343385); const-ify things a bit,
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use G_N_ELEMENTS instead of hard-coded array size.
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|
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2006-06-10 18:25:07 +0000 Young-Ho Cha <ganadist@chollian.net>
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gst/subparse/samiparse.c: Fix up broken entities before passing them to libxml *sigh*. (#343303).
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Original commit message from CVS:
|
|
Patch by: Young-Ho Cha <ganadist at chollian net>
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* gst/subparse/samiparse.c: (fix_invalid_entities), (parse_sami):
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Fix up broken entities before passing them to libxml *sigh*.
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(#343303).
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2006-06-09 18:52:35 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
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* ChangeLog:
|
|
* configure.ac:
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|
back to trunk
|
|
Original commit message from CVS:
|
|
back to trunk
|
|
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|
=== release 0.10.8 ===
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|
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2006-06-09 18:49:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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|
|
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* ChangeLog:
|
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* NEWS:
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* RELEASE:
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* configure.ac:
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* docs/plugins/gst-plugins-base-plugins.args:
|
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* docs/plugins/inspect/plugin-adder.xml:
|
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* docs/plugins/inspect/plugin-alsa.xml:
|
|
* docs/plugins/inspect/plugin-audioconvert.xml:
|
|
* docs/plugins/inspect/plugin-audiorate.xml:
|
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* docs/plugins/inspect/plugin-audioresample.xml:
|
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* docs/plugins/inspect/plugin-audiotestsrc.xml:
|
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* docs/plugins/inspect/plugin-cdparanoia.xml:
|
|
* docs/plugins/inspect/plugin-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.xml:
|
|
* docs/plugins/inspect/plugin-libvisual.xml:
|
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* docs/plugins/inspect/plugin-ogg.xml:
|
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* docs/plugins/inspect/plugin-pango.xml:
|
|
* docs/plugins/inspect/plugin-playbin.xml:
|
|
* docs/plugins/inspect/plugin-subparse.xml:
|
|
* docs/plugins/inspect/plugin-tcp.xml:
|
|
* docs/plugins/inspect/plugin-theora.xml:
|
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* docs/plugins/inspect/plugin-typefindfunctions.xml:
|
|
* docs/plugins/inspect/plugin-video4linux.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:
|
|
* win32/common/config.h:
|
|
releasing 0.10.8
|
|
Original commit message from CVS:
|
|
releasing 0.10.8
|
|
|
|
2006-06-07 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
0.10.7.2 prerelease
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/cs.po:
|
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* po/en_GB.po:
|
|
* po/hu.po:
|
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* po/it.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/sq.po:
|
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* po/sr.po:
|
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* po/sv.po:
|
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* po/uk.po:
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* po/vi.po:
|
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* win32/common/config.h:
|
|
0.10.7.2 prerelease
|
|
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2006-06-07 11:03:03 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
move last template doc snippets to source code and delete them
|
|
Original commit message from CVS:
|
|
* docs/libs/tmpl/gstaudio.sgml:
|
|
* docs/libs/tmpl/gstcolorbalance.sgml:
|
|
* docs/libs/tmpl/gstmixer.sgml:
|
|
* docs/libs/tmpl/gstringbuffer.sgml:
|
|
* docs/libs/tmpl/gsttuner.sgml:
|
|
* docs/libs/tmpl/gstxoverlay.sgml:
|
|
* gst-libs/gst/audio/audio.c:
|
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* gst-libs/gst/audio/gstringbuffer.c:
|
|
* gst-libs/gst/interfaces/colorbalance.c:
|
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* gst-libs/gst/interfaces/mixer.c:
|
|
* gst-libs/gst/interfaces/tuner.c:
|
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* gst-libs/gst/interfaces/xoverlay.c:
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move last template doc snippets to source code and delete them
|
|
|
|
2006-06-06 16:26:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* gst/gdp/gstgdppay.c:
|
|
adapt to new api
|
|
Original commit message from CVS:
|
|
adapt to new api
|
|
|
|
2006-06-06 14:39:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
configure.ac: enable building of GDP elements
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
enable building of GDP elements
|
|
* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain):
|
|
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
|
|
(gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
|
|
(gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
|
|
(gst_gdp_pay_reset_streamheader), (gst_gdp_pay_sink_event),
|
|
(gst_gdp_pay_set_property), (gst_gdp_pay_get_property),
|
|
(gst_gdp_pay_change_state):
|
|
* gst/gdp/gstgdppay.h:
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|
add version 1.0
|
|
|
|
2006-06-06 11:13:18 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
ext/theora/theoraparse.c: Mark DELTA_UNIT on non-keyframes.
|
|
Original commit message from CVS:
|
|
* ext/theora/theoraparse.c: (theora_parse_drain_queue_prematurely),
|
|
(theora_parse_drain_queue):
|
|
Mark DELTA_UNIT on non-keyframes.
|
|
|
|
2006-06-03 21:06:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/audio/: Document better the fact that latency_time and buffer_time are values stored in microseconds, an...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_class_init), (gst_base_audio_sink_setcaps):
|
|
* gst-libs/gst/audio/gstbaseaudiosink.h:
|
|
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps),
|
|
(gst_ring_buffer_samples_done):
|
|
* gst-libs/gst/audio/gstringbuffer.h:
|
|
Document better the fact that latency_time and buffer_time are values
|
|
stored in microseconds, and not the usual GStreamer nanoseconds.
|
|
Change the variables (compatibly) that store them from GstClockTime
|
|
to guint64 to make it more clear that they're not storing clock times.
|
|
Also, remove the bogus property description that says the user can
|
|
specify -1 to get the default value, since that's never been the case.
|
|
When computing the default segment size for the ring buffer, make it
|
|
an integer number of samples.
|
|
When the sub-class indicates a delay greater than the number of
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|
samples we've written return 0 from the audio sink get_time method.
|
|
|
|
2006-06-02 17:01:02 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
tests/check/: Don't busy-wait in tests; this was causing test timeouts very frequently when running under valgrind.
|
|
Original commit message from CVS:
|
|
* tests/check/elements/audioconvert.c: (set_channel_positions),
|
|
(get_float_mc_caps), (get_int_mc_caps):
|
|
* tests/check/elements/audioresample.c:
|
|
* tests/check/elements/audiotestsrc.c: (GST_START_TEST):
|
|
* tests/check/elements/videorate.c:
|
|
* tests/check/elements/videotestsrc.c: (GST_START_TEST):
|
|
* tests/check/elements/volume.c:
|
|
* tests/check/elements/vorbisdec.c:
|
|
* tests/check/pipelines/vorbisenc.c: (GST_START_TEST):
|
|
Don't busy-wait in tests; this was causing test timeouts very
|
|
frequently when running under valgrind.
|
|
|
|
2006-06-02 16:45:59 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* gst/gdp/gstgdpdepay.c:
|
|
* gst/gdp/gstgdppay.h:
|
|
small fixes
|
|
Original commit message from CVS:
|
|
small fixes
|
|
|
|
2006-06-02 16:35:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* tests/check/elements/multifdsink.c:
|
|
fail_if_can_read is racy
|
|
Original commit message from CVS:
|
|
fail_if_can_read is racy
|
|
|
|
2006-06-02 16:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
gst/tcp/: make multifdsink properly deal with streamheader:
|
|
Original commit message from CVS:
|
|
* gst/tcp/README:
|
|
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_init),
|
|
(gst_multi_fd_sink_remove_client_link),
|
|
(gst_multi_fd_sink_client_queue_caps),
|
|
(gst_multi_fd_sink_client_queue_buffer),
|
|
(gst_multi_fd_sink_handle_client_write),
|
|
(gst_multi_fd_sink_render):
|
|
* gst/tcp/gstmultifdsink.h:
|
|
make multifdsink properly deal with streamheader:
|
|
- streamheader is taken from caps
|
|
- buffers marked with IN_CAPS are not sent
|
|
- streamheaders are sent, on connection, from the caps of the
|
|
buffer where the client gets positioned to
|
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- further streamheader changes are done every time the client
|
|
will receive a buffer with different caps
|
|
* tests/check/elements/multifdsink.c: (GST_START_TEST),
|
|
(gst_multifdsink_create_streamheader):
|
|
add tests for this
|
|
|
|
2006-06-02 15:06:59 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
ext/vorbis/vorbisdec.c: Reinstate limit on channel count. Vorbis does not define the meaning of > 6 channels, so they...
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
|
|
Reinstate limit on channel count. Vorbis does not define the meaning
|
|
of > 6 channels, so they're just independent channels. Gstreamer
|
|
currently has no mechanism to represent N independent channels.
|
|
|
|
2006-06-02 14:23:34 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
ext/vorbis/vorbisdec.c: Don't arbitrarily restrict channel counts and rate in vorbis.
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisdec.c: (vorbis_handle_identification_packet):
|
|
Don't arbitrarily restrict channel counts and rate in vorbis.
|
|
In terms of effects likely on real-world files, this fixes 96kHz
|
|
playback of vorbis.
|
|
|
|
2006-06-02 14:19:18 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/audioconvert/audioconvert.c: More correct float->int conversion.
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/audioconvert.c: (float):
|
|
More correct float->int conversion.
|
|
|
|
2006-06-02 14:07:42 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
ext/ogg/gstoggdemux.c: Don't accidently send GST_CLOCK_TIME_NONE as a new segment start value. Fixes g-critical on tr...
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_perform_seek):
|
|
Don't accidently send GST_CLOCK_TIME_NONE as a new segment start
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value. Fixes g-critical on trying to play back ogg containing
|
|
unknown codec.
|
|
|
|
2006-06-02 10:34:12 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybasebin.*: Make the subtitle detection work from any thread so we don't deadlock. Fixes #343397.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (group_create), (group_commit),
|
|
(setup_source):
|
|
* gst/playback/gstplaybasebin.h:
|
|
Make the subtitle detection work from any thread so we don't
|
|
deadlock. Fixes #343397.
|
|
|
|
2006-06-02 10:28:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
gst/gdp/gstgdppay.c: add crc-header and crc-payload properties don't error out on some things that are recoverable
|
|
Original commit message from CVS:
|
|
* gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init),
|
|
(gst_gdp_pay_init), (gst_gdp_buffer_from_caps),
|
|
(gst_gdp_pay_buffer_from_buffer), (gst_gdp_buffer_from_event),
|
|
(gst_gdp_pay_reset_streamheader), (gst_gdp_pay_chain),
|
|
(gst_gdp_pay_sink_event), (gst_gdp_pay_set_property),
|
|
(gst_gdp_pay_get_property):
|
|
add crc-header and crc-payload properties
|
|
don't error out on some things that are recoverable
|
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* tests/check/elements/gdppay.c: (GST_START_TEST), (gdppay_suite):
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add test for crc
|
|
|
|
2006-06-02 09:17:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* gst/tcp/gsttcp.c:
|
|
show type number when packet is of the wrong type
|
|
Original commit message from CVS:
|
|
show type number when packet is of the wrong type
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2006-06-01 23:04:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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gst/volume/Makefile.am: Seriously, it's not *that* hard to get compilation right. Even a drunk can do it ! Add LIBOI...
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Original commit message from CVS:
|
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* gst/volume/Makefile.am:
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Seriously, it's not *that* hard to get compilation right. Even
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a drunk can do it ! Add LIBOIL CFLAGS and LIBS
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2006-06-01 22:00:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
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|
Original commit message from CVS:
|
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* ext/alsaspdif/alsaspdifsink.h:
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* ext/amrwb/gstamrwbdec.h:
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|
* ext/amrwb/gstamrwbenc.h:
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|
* ext/amrwb/gstamrwbparse.h:
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|
* ext/arts/gst_arts.h:
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* ext/artsd/gstartsdsink.h:
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* ext/audiofile/gstafparse.h:
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|
* ext/audiofile/gstafsink.h:
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* ext/audiofile/gstafsrc.h:
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* ext/audioresample/gstaudioresample.h:
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* ext/bz2/gstbz2dec.h:
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* ext/bz2/gstbz2enc.h:
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* ext/dirac/gstdiracdec.h:
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* ext/directfb/dfbvideosink.h:
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* ext/divx/gstdivxdec.h:
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* ext/divx/gstdivxenc.h:
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* ext/dts/gstdtsdec.h:
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* ext/faac/gstfaac.h:
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* ext/gsm/gstgsmdec.h:
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* ext/gsm/gstgsmenc.h:
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* ext/ivorbis/vorbisenc.h:
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* ext/libfame/gstlibfame.h:
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* ext/nas/nassink.h:
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* ext/neon/gstneonhttpsrc.h:
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* ext/polyp/polypsink.h:
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* ext/sdl/sdlaudiosink.h:
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* ext/sdl/sdlvideosink.h:
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* ext/shout/gstshout.h:
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* ext/snapshot/gstsnapshot.h:
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* ext/sndfile/gstsf.h:
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* ext/swfdec/gstswfdec.h:
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* ext/tarkin/gsttarkindec.h:
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* ext/tarkin/gsttarkinenc.h:
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* ext/theora/theoradec.h:
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* ext/wavpack/gstwavpackdec.h:
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* ext/wavpack/gstwavpackparse.h:
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* ext/xine/gstxine.h:
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* ext/xvid/gstxviddec.h:
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* ext/xvid/gstxvidenc.h:
|
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* gst/cdxaparse/gstcdxaparse.h:
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* gst/cdxaparse/gstcdxastrip.h:
|
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* gst/colorspace/gstcolorspace.h:
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* gst/festival/gstfestival.h:
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* gst/freeze/gstfreeze.h:
|
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* gst/gdp/gstgdpdepay.h:
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* gst/gdp/gstgdppay.h:
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* gst/modplug/gstmodplug.h:
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* gst/mpeg1sys/gstmpeg1systemencode.h:
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* gst/mpeg1videoparse/gstmp1videoparse.h:
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* gst/mpeg2sub/gstmpeg2subt.h:
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* gst/mpegaudioparse/gstmpegaudioparse.h:
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* gst/multifilesink/gstmultifilesink.h:
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* gst/overlay/gstoverlay.h:
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* gst/playondemand/gstplayondemand.h:
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* gst/qtdemux/qtdemux.h:
|
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* gst/rtjpeg/gstrtjpegdec.h:
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* gst/rtjpeg/gstrtjpegenc.h:
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* gst/smooth/gstsmooth.h:
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* gst/smoothwave/gstsmoothwave.h:
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* gst/spectrum/gstspectrum.h:
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* gst/speed/gstspeed.h:
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* gst/stereo/gststereo.h:
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* gst/switch/gstswitch.h:
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* gst/tta/gstttadec.h:
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* gst/tta/gstttaparse.h:
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* gst/videodrop/gstvideodrop.h:
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* gst/xingheader/gstxingmux.h:
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* sys/directdraw/gstdirectdrawsink.h:
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* sys/directsound/gstdirectsoundsink.h:
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* sys/dxr3/dxr3audiosink.h:
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* sys/dxr3/dxr3spusink.h:
|
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* sys/dxr3/dxr3videosink.h:
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* sys/qcam/gstqcamsrc.h:
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* sys/vcd/vcdsrc.h:
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Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
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2006-06-01 20:39:30 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst/volume/gstvolume.*: rewrite the passthrough check, split _int16 and _int16_clamp, fix another property desc., rem...
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|
Original commit message from CVS:
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* gst/volume/gstvolume.c: (volume_choose_func),
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|
(volume_update_real_volume), (gst_volume_class_init),
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(gst_volume_init), (volume_process_float), (volume_process_int16),
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(volume_process_int16_clamp), (volume_set_caps),
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(volume_transform_ip), (plugin_init):
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* gst/volume/gstvolume.h:
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rewrite the passthrough check, split _int16 and _int16_clamp, fix
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another property desc., remove unused param from process function
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* tests/check/elements/volume.c: (volume_suite):
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reactivate the passthrough test
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2006-06-01 19:19:51 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
|
|
Original commit message from CVS:
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|
* ext/alsa/gstalsamixerelement.h:
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* ext/alsa/gstalsamixeroptions.h:
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|
* ext/alsa/gstalsamixertrack.h:
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* ext/gnomevfs/gstgnomevfssink.h:
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* ext/gnomevfs/gstgnomevfssrc.h:
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* ext/theora/gsttheoradec.h:
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* ext/theora/gsttheoraenc.h:
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* ext/theora/gsttheoraparse.h:
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* ext/vorbis/vorbisparse.h:
|
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* gst-libs/gst/audio/gstaudioclock.h:
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* gst-libs/gst/audio/gstaudiofilter.h:
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* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
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* gst/audioconvert/gstaudioconvert.h:
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* gst/audioresample/gstaudioresample.h:
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* gst/audiotestsrc/gstaudiotestsrc.h:
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* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
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* gst/playback/gststreamselector.h:
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* gst/tcp/gstmultifdsink.h:
|
|
* gst/tcp/gsttcpclientsink.h:
|
|
* gst/tcp/gsttcpclientsrc.h:
|
|
* gst/tcp/gsttcpserversink.h:
|
|
* gst/tcp/gsttcpserversrc.h:
|
|
* gst/videorate/gstvideorate.h:
|
|
* gst/videoscale/gstvideoscale.h:
|
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* gst/videotestsrc/gstvideotestsrc.h:
|
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* gst/volume/gstvolume.h:
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* sys/v4l/gstv4ljpegsrc.h:
|
|
* sys/v4l/gstv4lmjpegsink.h:
|
|
* sys/v4l/gstv4lmjpegsrc.h:
|
|
* sys/v4l/gstv4lsrc.h:
|
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* sys/ximage/ximagesink.h:
|
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* sys/xvimage/xvimagesink.h:
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* tests/old/testsuite/alsa/sinesrc.h:
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Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
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2006-05-31 16:56:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* tests/check/elements/multifdsink.c:
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remove wrong commit
|
|
Original commit message from CVS:
|
|
remove wrong commit
|
|
|
|
2006-05-31 16:21:48 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/libvisual/visual.c: Handle DISCONT.
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|
Original commit message from CVS:
|
|
* ext/libvisual/visual.c: (gst_visual_reset),
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|
(gst_visual_sink_setcaps), (gst_visual_sink_event),
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(gst_visual_src_event), (get_buffer), (gst_visual_chain):
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Handle DISCONT.
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Use running time before doing QoS.
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Handle mono too.
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2006-05-31 14:17:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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docs/libs/Makefile.am: set a magic variable to indicate we know the docs are incomplete
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|
Original commit message from CVS:
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* docs/libs/Makefile.am:
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set a magic variable to indicate we know the docs are incomplete
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2006-05-30 20:33:59 +0000 Sébastien Moutte <sebastien@moutte.net>
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win32/common/libgstvideo.def: export gst_video_calculate_display_ratio
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|
Original commit message from CVS:
|
|
* win32/common/libgstvideo.def:
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export gst_video_calculate_display_ratio
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* win32/vs6/libgstvideoscale.dsp:
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add link to libgstvideo-0.10.lib
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2006-05-30 19:00:39 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gstplaybasebin.c: Throw a more comprehensible error for rtsp:// URIs (rather than erroring out with a ne...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (gen_source_element):
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Throw a more comprehensible error for rtsp:// URIs (rather
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than erroring out with a negotiation error later on) until
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we fix playbin to handle rtspsrc etc.
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2006-05-30 16:09:36 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/pango/gsttextoverlay.c: Added some FIXMEs.
|
|
Original commit message from CVS:
|
|
* ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event),
|
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(gst_text_overlay_text_event):
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Added some FIXMEs.
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2006-05-30 16:07:50 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/adder/gstadder.*: Implement release_request_pad.
|
|
Original commit message from CVS:
|
|
* gst/adder/gstadder.c: (gst_adder_class_init), (gst_adder_init),
|
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(gst_adder_request_new_pad), (gst_adder_release_pad):
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* gst/adder/gstadder.h:
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Implement release_request_pad.
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Make padcounter atomic.
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* tests/check/elements/adder.c: (GST_START_TEST), (adder_suite):
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Added check for release_pad in adder.
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2006-05-30 16:04:14 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/ogg/gstoggdemux.c: Fix build again.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_chain_new_stream):
|
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Fix build again.
|
|
|
|
2006-05-30 14:59:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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|
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ext/ogg/gstoggdemux.c: add more debugging clean up printf formats for granulepos and serialno
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind),
|
|
(gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data),
|
|
(gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
|
|
(gst_ogg_pad_submit_page), (gst_ogg_chain_new_stream),
|
|
(gst_ogg_demux_seek), (gst_ogg_demux_get_data),
|
|
(gst_ogg_demux_get_next_page), (gst_ogg_demux_do_seek),
|
|
(gst_ogg_demux_bisect_forward_serialno),
|
|
(gst_ogg_demux_read_chain), (gst_ogg_demux_find_chains),
|
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(gst_ogg_demux_chain), (gst_ogg_demux_loop), (gst_ogg_print):
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add more debugging
|
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clean up printf formats for granulepos and serialno
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2006-05-30 14:31:43 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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|
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* tests/check/elements/multifdsink.c:
|
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* tests/check/generic/states.c:
|
|
properly fail if we can't make an element
|
|
Original commit message from CVS:
|
|
properly fail if we can't make an element
|
|
|
|
2006-05-30 13:22:58 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
ext/vorbis/vorbisenc.*: Multi-channel caps negotiation, so we can do proper multichannel vorbis encoding, negotiated ...
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisenc.c: (raw_caps_factory),
|
|
(gst_vorbis_enc_class_init), (gst_vorbis_enc_dispose),
|
|
(gst_vorbis_enc_generate_sink_caps), (gst_vorbis_enc_sink_getcaps),
|
|
(gst_vorbis_enc_init), (gst_vorbis_enc_buffer_from_header_packet),
|
|
(gst_vorbis_enc_chain), (gst_vorbis_enc_change_state):
|
|
* ext/vorbis/vorbisenc.h:
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Multi-channel caps negotiation, so we can do proper multichannel
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vorbis encoding, negotiated through audioconvert.
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2006-05-30 11:45:52 +0000 Wim Taymans <wim.taymans@gmail.com>
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tests/check/elements/adder.c: Added check to show that #339935 is fixed with ongoing adder and collectpads fixes.
|
|
Original commit message from CVS:
|
|
* tests/check/elements/adder.c: (test_event_message_received),
|
|
(test_play_twice_message_received), (GST_START_TEST),
|
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(adder_suite):
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Added check to show that #339935 is fixed with ongoing
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adder and collectpads fixes.
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2006-05-29 17:19:48 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
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gst/adder/gstadder.c: Don't leak pad name.
|
|
Original commit message from CVS:
|
|
* gst/adder/gstadder.c: (gst_adder_request_new_pad):
|
|
Don't leak pad name.
|
|
|
|
2006-05-29 15:49:53 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/adder/gstadder.c: Fix adder seeking.
|
|
Original commit message from CVS:
|
|
* gst/adder/gstadder.c: (gst_adder_query_duration),
|
|
(forward_event_func), (forward_event), (gst_adder_src_event):
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Fix adder seeking.
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Make query/seeking code threadsafe.
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* tests/check/Makefile.am:
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* tests/check/elements/adder.c: (test_event_message_received),
|
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(GST_START_TEST), (test_play_twice_message_received):
|
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Fix adder test case.
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2006-05-29 13:21:00 +0000 Young-Ho Cha <ganadist@chollian.net>
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gst/playback/gstplaybasebin.*: Add 'subtitle-encoding' property to playbin, so applications can force a subtitle enco...
|
|
Original commit message from CVS:
|
|
Patch by: Young-Ho Cha <ganadist at chollian net>
|
|
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
|
|
(gst_play_base_bin_init), (gst_play_base_bin_dispose),
|
|
(set_encoding_element), (decodebin_element_added_cb),
|
|
(decodebin_element_removed_cb), (setup_subtitle), (setup_source),
|
|
(gst_play_base_bin_set_property), (gst_play_base_bin_get_property):
|
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* gst/playback/gstplaybasebin.h:
|
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Add 'subtitle-encoding' property to playbin, so applications can
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force a subtitle encoding for non-UTF8 subtitles (#342268).
|
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* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init),
|
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(gst_sub_parse_set_property):
|
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Rename recently-added 'encoding' property to 'subtitle-encoding'
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(so it can be proxied by playbin/decodebin in a generic way
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with less danger of false positives).
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2006-05-29 11:04:48 +0000 Michael Smith <msmith@xiph.org>
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|
|
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gst/audioconvert/gstaudioconvert.c: Patch from #341562: give more specific audio caps in get_caps, so that basetransf...
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/gstaudioconvert.c: (make_lossless_changes),
|
|
(append_with_other_format), (set_structure_widths),
|
|
(gst_audio_convert_transform_caps):
|
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Patch from #341562: give more specific audio caps in get_caps, so
|
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that basetransform can make better decisions on what caps to
|
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negotiate.
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2006-05-28 20:04:12 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
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tests/check/elements/volume.c: make it compile again
|
|
Original commit message from CVS:
|
|
* tests/check/elements/volume.c:
|
|
make it compile again
|
|
|
|
2006-05-28 19:56:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
tests/check/elements/volume.c: disable test until #343196 gets resolved
|
|
Original commit message from CVS:
|
|
* tests/check/elements/volume.c: (volume_suite):
|
|
disable test until #343196 gets resolved
|
|
|
|
2006-05-28 19:42:27 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/adder/gstadder.c: Make it easier to copy&paste
|
|
Original commit message from CVS:
|
|
* gst/adder/gstadder.c: (gst_adder_get_type):
|
|
Make it easier to copy&paste
|
|
* gst/volume/Makefile.am:
|
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* gst/volume/gstvolume.c: (volume_update_real_volume),
|
|
(gst_volume_set_volume), (gst_volume_set_mute),
|
|
(gst_volume_class_init), (volume_process_int16), (volume_set_caps),
|
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(volume_transform_ip), (volume_update_mute),
|
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(volume_update_volume):
|
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* gst/volume/gstvolume.h:
|
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Add own debug category, move duplicate code to helper function, fix
|
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property texts, add more comments and prepare ffor liboil-goodness
|
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* tests/check/Makefile.am:
|
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* tests/check/elements/volume.c: (GST_START_TEST), (volume_suite):
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add test for mute and passtrough case, be a bit more verbose to track
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failure
|
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* tests/check/generic/states.c: (GST_START_TEST):
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catch elements that fail to instantiate
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2006-05-28 09:37:18 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
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tests/check/pipelines/: Comment out tests using parse_launch() if core was built without parsing capabilities.
|
|
Original commit message from CVS:
|
|
* tests/check/pipelines/simple-launch-lines.c:
|
|
* tests/check/pipelines/theoraenc.c:
|
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* tests/check/pipelines/vorbisenc.c:
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Comment out tests using parse_launch() if core was built without
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parsing capabilities.
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2006-05-27 13:34:03 +0000 Edward Hervey <bilboed@bilboed.com>
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tests/check/Makefile.am: Extra bonus points for whoever explains to ensonic that you are meant to test unit tests tho...
|
|
Original commit message from CVS:
|
|
* tests/check/Makefile.am:
|
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Extra bonus points for whoever explains to ensonic that you are meant
|
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to test unit tests thoroughly before commiting them, especially if
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you know it's going to break.
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De-activated element/adder tests.
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2006-05-27 13:09:16 +0000 Edward Hervey <bilboed@bilboed.com>
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gst/ffmpegcolorspace/gstffmpegcodecmap.c: Marking caps conversion issues as GST_WARNING is way too verbose,
|
|
Original commit message from CVS:
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
|
|
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_smpfmt_to_caps):
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Marking caps conversion issues as GST_WARNING is way too verbose,
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Moving them to GST_LOG.
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2006-05-27 11:26:18 +0000 Tim-Philipp Müller <tim@centricular.net>
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README: Replace current README (containing the release notes from some 0.9.x version) with a proper README taken from...
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Original commit message from CVS:
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* README:
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Replace current README (containing the release notes from
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some 0.9.x version) with a proper README taken from the core.
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2006-05-26 15:52:23 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/vorbis/vorbisdec.c: Small cleanups.
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Original commit message from CVS:
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* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
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(vorbis_dec_src_event), (vorbis_dec_sink_event), (vorbis_do_clip),
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(vorbis_dec_push), (vorbis_handle_data_packet), (vorbis_dec_chain),
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(vorbis_dec_change_state):
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Small cleanups.
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Add some FIXMEs
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Clip output samples to segment boundaries.
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2006-05-26 11:17:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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sys/ximage/ximagesink.c: Improve the errors produced on bad output, including some human readable description strings.
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Original commit message from CVS:
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* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
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(gst_ximagesink_xcontext_get), (gst_ximagesink_show_frame):
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Improve the errors produced on bad output, including some human
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readable description strings.
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Handle the (theoretical for ximagesink) case where the XServer
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has a different idea about the size required for a particular
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frame and gives us too small a memory allocation.
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2006-05-26 10:18:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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* ChangeLog:
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Mention bugs fixed by previous commit
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Original commit message from CVS:
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Mention bugs fixed by previous commit
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2006-05-26 09:40:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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sys/xvimage/xvimagesink.c: Improve the errors produced on bad output, including some human readable description strings.
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Original commit message from CVS:
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
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(gst_xvimagesink_get_xv_support), (gst_xvimagesink_xcontext_get),
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(gst_xvimagesink_get_format_from_caps), (gst_xvimagesink_setcaps),
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(gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
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Improve the errors produced on bad output, including some human
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readable description strings.
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Handle RGB Xv formats properly by transforming them into our
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big-endian caps description.
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Use gst_caps_truncate to ensure that we never try and choose a
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non-fixed caps in buffer_alloc.
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Handle the case where the XServer has a different idea about the size
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required for a particular frame and gives us too small a memory
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allocation.
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Use -1 to indicate 'no image format', because 0 is a valid XServer
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image format number.
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Put RGB Xv formats at the end of the caps, so that we always prefer
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YUV format frames.
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Iterate the available Xv Encodings to determine the maximum width and
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height, and then return that in our caps.
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2006-05-25 16:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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gst/playback/gstdecodebin.c: When there is only one unfinished pad and it receives an event that doesn't match our re...
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Original commit message from CVS:
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* gst/playback/gstdecodebin.c: (remove_fakesink), (pad_probe):
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When there is only one unfinished pad and it receives an event that
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doesn't match our requirements, we need to set alldone=FALSE so that
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the fakesink is not removed yet.
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2006-05-25 09:32:31 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/ogg/gstoggdemux.c: Use gst_type_find_helper_for_buffer() to find the type of stream from the first packet.
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Original commit message from CVS:
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* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
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Use gst_type_find_helper_for_buffer() to find the type
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of stream from the first packet.
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* configure.ac:
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Bump requirements to core CVS (needed for vorbis
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typefinding to work).
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2006-05-24 08:34:53 +0000 Edward Hervey <bilboed@bilboed.com>
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gst/typefind/gsttypefindfunctions.c: Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
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Original commit message from CVS:
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* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
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Added the 'prfl' atom type which MQV (no, it's not a typo) files contain.
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Else they play perfectly fine with qtdemux.
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2006-05-23 20:38:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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make more debug catagories static
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Original commit message from CVS:
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* ext/theora/theoradec.c:
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* ext/theora/theoraenc.c:
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* ext/theora/theoraparse.c: (gst_theora_parse_class_init):
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* gst/audiorate/gstaudiorate.c:
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make more debug catagories static
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* tests/check/Makefile.am:
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* tests/check/elements/adder.c: (message_received),
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(test_event_message_received), (GST_START_TEST),
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(test_play_twice_message_received), (adder_suite):
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added test case for using element twice, extra bonus points for anyone
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who can make these test run reliably
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2006-05-23 15:18:40 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/theora/theoradec.c: Make work with time-stamped input buffers that do not have a granulepos in BUFFER_OFFSET_END ...
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Original commit message from CVS:
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* ext/theora/theoradec.c: (theora_dec_chain):
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Make work with time-stamped input buffers that do not
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have a granulepos in BUFFER_OFFSET_END (like theora
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buffers coming from matroskademux). Fixes #342448.
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2006-05-22 15:53:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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gst/gdp/: Handle error cases when calling functions do downwards state change after parent's change_state
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Original commit message from CVS:
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* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_chain),
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(gst_gdp_depay_change_state):
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* gst/gdp/gstgdpdepay.h:
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* gst/gdp/gstgdppay.c: (gst_gdp_pay_reset_streamheader),
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(gst_gdp_pay_chain), (gst_gdp_pay_sink_event),
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(gst_gdp_pay_change_state):
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* gst/gdp/gstgdppay.h:
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Handle error cases when calling functions
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do downwards state change after parent's change_state
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* tests/check/elements/gdpdepay.c: (GST_START_TEST):
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* tests/check/elements/gdppay.c: (GST_START_TEST):
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clean up more
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2006-05-22 13:25:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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adding GDP payloader and depayloader. Build integration will follow later when the GDP issues for core are sorted out.
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Original commit message from CVS:
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* gst/gdp/Makefile.am:
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* gst/gdp/gstgdp.c: (plugin_init):
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* gst/gdp/gstgdpdepay.c: (gst_gdp_depay_base_init),
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(gst_gdp_depay_class_init), (gst_gdp_depay_init),
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(gst_gdp_depay_finalize), (gst_gdp_depay_chain),
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(gst_gdp_depay_change_state), (gst_gdp_depay_plugin_init):
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* gst/gdp/gstgdpdepay.h:
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* gst/gdp/gstgdppay.c: (gst_gdp_pay_base_init),
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(gst_gdp_pay_class_init), (gst_gdp_pay_init),
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(gst_gdp_pay_dispose), (gst_gdp_stamp_buffer),
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(gst_gdp_buffer_from_caps), (gst_gdp_pay_buffer_from_buffer),
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(gst_gdp_buffer_from_event), (gst_gdp_pay_reset_streamheader),
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(gst_gdp_queue_buffer), (gst_gdp_pay_chain),
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(gst_gdp_pay_sink_event), (gst_gdp_pay_change_state),
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(gst_gdp_pay_plugin_init):
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* gst/gdp/gstgdppay.h:
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* tests/check/Makefile.am:
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* tests/check/elements/gdpdepay.c: (setup_gdpdepay),
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(cleanup_gdpdepay), (gdpdepay_push_per_byte), (GST_START_TEST),
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(setup_gdpdepay_streamheader), (gdpdepay_suite), (main):
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* tests/check/elements/gdppay.c: (setup_gdppay), (cleanup_gdppay),
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(GST_START_TEST), (setup_gdppay_streamheader), (gdppay_suite),
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(main):
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adding GDP payloader and depayloader. Build integration will
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follow later when the GDP issues for core are sorted out.
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2006-05-22 11:42:03 +0000 Peter Kjellerstedt <pkj@axis.com>
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gst/tcp/Makefile.am: fdstresstest doesn't need Gtk+, fix compilation if gtk is not available (#342566).
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Original commit message from CVS:
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Patch by: Peter Kjellerstedt <pkj at axis com>
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* gst/tcp/Makefile.am:
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fdstresstest doesn't need Gtk+, fix compilation if
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gtk is not available (#342566).
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2006-05-19 17:57:56 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
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gst-libs/gst/rtp/gstbasertpaudiopayload.c: 80 line columns
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Original commit message from CVS:
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* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
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80 line columns
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Removed redundant floor()
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2006-05-19 15:00:43 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/riff/riff-read.c: On second thought, just skip JUNK chunks automatically, so the caller doesn't have to ...
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Original commit message from CVS:
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* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
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On second thought, just skip JUNK chunks automatically, so
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the caller doesn't have to handle this. Fixes #342345.
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Also, return GST_FLOW_UNEXPECTED if we get a short read,
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not GST_FLOW_ERROR.
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2006-05-19 13:37:55 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/riff/riff-read.c: Don't bail out on JUNK chunks with a size of 0 (would try to pull_range 0 bytes before...
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Original commit message from CVS:
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* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk):
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Don't bail out on JUNK chunks with a size of 0 (would try to
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pull_range 0 bytes before, which sources don't like too much).
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See #342345.
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2006-05-19 13:02:46 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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Use the gstutil scaling function to preserve 64 bits while calculating output width and height from the display-aspec...
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Original commit message from CVS:
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* gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
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Use the gstutil scaling function to preserve 64 bits while calculating
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output width and height from the display-aspect-ratio. (A continuation
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of #341542)
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2006-05-19 11:50:17 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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sys/xvimage/xvimagesink.*: When performing buffer allocations, remember the caps and image format we return so that i...
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Original commit message from CVS:
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xcontext_clear),
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(gst_xvimagesink_buffer_alloc):
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* sys/xvimage/xvimagesink.h:
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When performing buffer allocations, remember the caps and image format
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we return so that if the same caps are asked for next time we can
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return them immediately without doing any caps intersections.
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2006-05-18 23:00:02 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
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gst-libs/gst/rtp/README: Some new documentation
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Original commit message from CVS:
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2006-05-18 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
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* gst-libs/gst/rtp/README:
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Some new documentation
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* gst-libs/gst/rtp/gstrtpbuffer.h:
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Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
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* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
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* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
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New RTP audio base payloader class. Supports frame or sample based codecs.
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Not enabled in Makefile.am until approved.
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2006-05-18 20:30:26 +0000 Tim-Philipp Müller <tim@centricular.net>
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tests/check/elements/alsa.c: Fix test case: don't try to free NULL GValueArray when there are no devices.
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Original commit message from CVS:
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* tests/check/elements/alsa.c: (test_device_property_probe):
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Fix test case: don't try to free NULL GValueArray when there
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are no devices.
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2006-05-18 19:21:53 +0000 Tim-Philipp Müller <tim@centricular.net>
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tests/check/: Add simple test that runs a device property probe on alsasrc, alsasink and alsamixer. Disable valgrind ...
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|
Original commit message from CVS:
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* tests/check/Makefile.am:
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* tests/check/elements/alsa.c: (test_device_property_probe),
|
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(alsa_suite), (main):
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Add simple test that runs a device property probe on alsasrc,
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alsasink and alsamixer. Disable valgrind check for now (too
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many leaks in libasound, and valgrind ignored my suppressions
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additions).
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2006-05-18 17:19:39 +0000 Martin Szulecki <gnomebugzilla@sukimashita.com>
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ext/alsa/: Clean up and simplify alsa device probing. Make it actually work for multiple classes. Don't cache results...
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|
Original commit message from CVS:
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* ext/alsa/gstalsadeviceprobe.c: (gst_alsa_get_device_list),
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(gst_alsa_device_property_probe_probe_property),
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(gst_alsa_device_property_probe_needs_probe),
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(gst_alsa_device_property_probe_get_values),
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(gst_alsa_type_add_device_property_probe_interface):
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* ext/alsa/gstalsadeviceprobe.h:
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* ext/alsa/gstalsamixerelement.c:
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(gst_alsa_mixer_element_init_interfaces):
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* ext/alsa/gstalsamixerelement.h:
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Clean up and simplify alsa device probing. Make it actually work
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for multiple classes. Don't cache results any longer.
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* ext/alsa/gstalsasink.c: (gst_alsasink_init_interfaces),
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(gst_alsasink_init):
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* ext/alsa/gstalsasrc.c: (gst_alsasrc_dispose),
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(gst_alsasrc_interface_supported), (gst_implements_interface_init),
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(gst_alsasrc_init_interfaces), (gst_alsasrc_set_property):
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Make alsasink and alsasrc implement the GstPropertyProbe interface
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for device probing (#342181).
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Patch by: Martin Szulecki <gnomebugzilla at sukimashita com>
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2006-05-18 10:05:23 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/subparse/samiparse.c: Don't ignore return value of strtol (++compiler_happiness).
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Original commit message from CVS:
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|
* gst/subparse/samiparse.c: (handle_start_font):
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Don't ignore return value of strtol (++compiler_happiness).
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2006-05-17 17:49:10 +0000 Young-Ho Cha <ganadist@chollian.net>
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gst/subparse/gstsubparse.*: Add 'encoding' property (#341681).
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Original commit message from CVS:
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Patch by: Young-Ho Cha <ganadist chollian net>
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* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
|
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(gst_sub_parse_class_init), (gst_sub_parse_init),
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(gst_sub_parse_set_property), (gst_sub_parse_get_property),
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(convert_encoding):
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* gst/subparse/gstsubparse.h:
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Add 'encoding' property (#341681).
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* gst/subparse/samiparse.c: (characters_sami):
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Output is pango markup, so we need to escape text
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between tags (#342143).
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2006-05-16 17:34:14 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/audio/multichannel.c: It's okay to have caps with channels=1 and a channel position different from GST_A...
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|
Original commit message from CVS:
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|
* gst-libs/gst/audio/multichannel.c:
|
|
(gst_audio_check_channel_positions):
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It's okay to have caps with channels=1 and a channel position
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different from GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
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(deinterleavers might want to keep the position in the caps,
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so that they can be re-interleaved again properly later).
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Leave check for unexpected 2-channel layouts intact for now.
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2006-05-16 16:28:10 +0000 Zaheer Abbas Merali <zaheerabbas@merali.org>
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gst/tcp/gsttcp.c: Return GST_FLOW_UNEXPECTED when we have an eos on the socket so basesrc can do its job correctly.
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Original commit message from CVS:
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2006-05-16 Zaheer Abbas Merali <zaheerabbas at merali dot org>
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* gst/tcp/gsttcp.c: (gst_tcp_socket_read):
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Return GST_FLOW_UNEXPECTED when we have an eos on the socket so
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basesrc can do its job correctly.
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2006-05-16 15:52:17 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/alsa/: Refactor and improve caps probing code: probe signedness when we probe the supported formats/widths; set e...
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|
Original commit message from CVS:
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|
* ext/alsa/Makefile.am:
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* ext/alsa/gstalsa.c: (gst_alsa_detect_rates),
|
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(gst_alsa_detect_formats), (get_channel_free_structure),
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(caps_add_channel_configuration), (gst_alsa_detect_channels),
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(gst_alsa_probe_supported_formats):
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* ext/alsa/gstalsa.h:
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* ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
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Refactor and improve caps probing code: probe signedness
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when we probe the supported formats/widths; set endianness
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to the one we actually probed for (ie. cpu endianness).
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* ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (gst_alsasrc_getcaps),
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(gst_alsasrc_close):
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* ext/alsa/gstalsasrc.h:
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Implement caps probing for alsasrc.
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2006-05-15 17:42:19 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/theora/theoradec.c: Cleanups, add some G_LIKELY.
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Original commit message from CVS:
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|
* ext/theora/theoradec.c: (gst_theora_dec_reset),
|
|
(theora_dec_src_query), (theora_dec_src_event),
|
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(theora_dec_sink_event), (theora_handle_comment_packet),
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(theora_handle_data_packet), (theora_dec_change_state):
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Cleanups, add some G_LIKELY.
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Use segment helpers instead of our own wrong code.
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Clear queued buffers on seek and READY.
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* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_reset),
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(vorbis_dec_convert), (vorbis_dec_src_query),
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(vorbis_dec_src_event), (vorbis_dec_sink_event),
|
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(vorbis_handle_comment_packet), (vorbis_dec_push),
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(vorbis_handle_data_packet), (vorbis_dec_chain),
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(vorbis_dec_change_state):
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* ext/vorbis/vorbisdec.h:
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Remove old useless packetno variable.
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Do position query properly.
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Add some G_LIKELY.
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Do cleanup of queued buffers in new helper function
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and use it.
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2006-05-15 17:17:22 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/alsa/gstalsasink.c: Query supported sample rates. Fixes #341732.
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Original commit message from CVS:
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* ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
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Query supported sample rates. Fixes #341732.
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2006-05-15 17:01:02 +0000 Julien Moutte <julien@moutte.net>
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gst/playback/gstdecodebin.c: Make decodebin reusable when going from PAUSE_TO_READY and then back to PAUSED.
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Original commit message from CVS:
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2006-05-15 Julien MOUTTE <julien@moutte.net>
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* gst/playback/gstdecodebin.c: (cleanup_decodebin),
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(gst_decode_bin_change_state): Make decodebin reusable
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when going from PAUSE_TO_READY and then back to PAUSED.
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Fixes #331678.
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2006-05-15 16:49:31 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/vorbis/vorbisdec.c: Cleanups. Use refcounting and DEBUG_OBJECT.
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Original commit message from CVS:
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* ext/vorbis/vorbisdec.c: (vorbis_get_query_types),
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(vorbis_dec_convert), (vorbis_dec_src_query),
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(vorbis_dec_sink_query), (vorbis_dec_src_event),
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(vorbis_dec_sink_event), (vorbis_handle_identification_packet),
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(vorbis_dec_clean_queued), (vorbis_dec_push),
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(vorbis_handle_data_packet), (vorbis_dec_change_state):
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Cleanups. Use refcounting and DEBUG_OBJECT.
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Reset segment on flush, use code methods instead of our
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own wrong version.
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Fix potential memleak.
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2006-05-15 16:46:44 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/alsa/gstalsasink.*: Don't leak allocated snd_output_t structure if there's more than one alsasink instance at a t...
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Original commit message from CVS:
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* ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
|
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(gst_alsasink_init):
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* ext/alsa/gstalsasink.h:
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Don't leak allocated snd_output_t structure if there's
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more than one alsasink instance at a time (#341873).
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Also fix GObject macros in header file.
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2006-05-15 15:31:30 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/subparse/gstsubparse.c: Don't use libxml functions in the typefinding code.
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Original commit message from CVS:
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* gst/subparse/gstsubparse.c:
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(gst_sub_parse_data_format_autodetect):
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Don't use libxml functions in the typefinding code.
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2006-05-15 15:01:08 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/ogg/gstoggdemux.c: Fix seeking performance in the case where a non-header packet has a 0 granulepos (busted theor...
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Original commit message from CVS:
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* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet):
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Fix seeking performance in the case where a non-header
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packet has a 0 granulepos (busted theora case).
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Fixes #341719
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2006-05-15 14:19:35 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/subparse/gstsubparse.c: Improve SAMI typefinding: handle case where there are whitespaces or newlines in front of...
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Original commit message from CVS:
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* gst/subparse/gstsubparse.c:
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(gst_sub_parse_data_format_autodetect):
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Improve SAMI typefinding: handle case where there are
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whitespaces or newlines in front of the first <SAMI>
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tag (#169936).
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2006-05-15 12:18:13 +0000 Tim-Philipp Müller <tim@centricular.net>
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configure.ac: Build video4linux plugin even if there's no XVIDEO, just without implementing the GstXOverlay interface...
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Original commit message from CVS:
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* configure.ac:
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Build video4linux plugin even if there's no XVIDEO, just
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without implementing the GstXOverlay interface (#334002).
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2006-05-15 10:17:04 +0000 Tim-Philipp Müller <tim@centricular.net>
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Add tentative support for libvisual-0.4 (#336881).
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Original commit message from CVS:
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* configure.ac:
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* ext/libvisual/visual.c: (gst_visual_actor_plugin_is_gl),
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(plugin_init):
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Add tentative support for libvisual-0.4 (#336881).
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2006-05-15 09:41:03 +0000 Young-Ho Cha <ganadist@chollian.net>
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gst/subparse/samiparse.c: Need to map "silver" colour explicitly (#169936).
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Original commit message from CVS:
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Patch by: Young-Ho Cha <ganadist at chollian net>
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* gst/subparse/samiparse.c: (handle_start_font):
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Need to map "silver" colour explicitly (#169936).
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2006-05-15 09:14:35 +0000 Young-Ho Cha <ganadist@chollian.net>
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gst/subparse/: Add support for SAMI subtitles (#169936).
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Original commit message from CVS:
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Patch by: Young-Ho Cha <ganadist at chollian net>
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* gst/subparse/Makefile.am:
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* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
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(parser_state_dispose), (gst_sub_parse_data_format_autodetect),
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(gst_sub_parse_format_autodetect), (feed_textbuf),
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(gst_subparse_type_find), (plugin_init):
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* gst/subparse/gstsubparse.h:
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* gst/subparse/samiparse.c:
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* gst/subparse/samiparse.h:
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Add support for SAMI subtitles (#169936).
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2006-05-14 21:18:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* win32/common/config.h:
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update config.h
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Original commit message from CVS:
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update config.h
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2006-05-14 21:18:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* ext/ogg/README:
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fix mistakes in README
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Original commit message from CVS:
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fix mistakes in README
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2006-05-14 18:15:17 +0000 Michael Smith <msmith@xiph.org>
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gst/audioconvert/gstchannelmix.c: Fix #341696: crash when mixing L+R+C to mono or stereo.
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Original commit message from CVS:
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* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_fill_others):
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Fix #341696: crash when mixing L+R+C to mono or stereo.
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* tests/check/Makefile.am:
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* tests/check/elements/audioconvert.c: (set_channel_positions),
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(get_float_mc_caps), (get_int_mc_caps), (GST_START_TEST),
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(audioconvert_suite):
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Add test for the above, including some generic framework bits for
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testing multichannel things.
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2006-05-14 16:05:47 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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* configure.ac:
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Back to CVS
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Original commit message from CVS:
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Back to CVS
|
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=== release 0.10.7 ===
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2006-05-14 16:00:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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configure.ac: releasing 0.10.7, "Leave the gun"
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Original commit message from CVS:
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2006-05-14 Jan Schmidt <thaytan@mad.scientist.com>
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* configure.ac:
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releasing 0.10.7, "Leave the gun"
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2006-05-14 15:55:16 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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* common:
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* po/af.po:
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* po/az.po:
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* po/cs.po:
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* po/en_GB.po:
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* po/hu.po:
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* po/it.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/sq.po:
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* po/sr.po:
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* po/sv.po:
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* po/uk.po:
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* po/vi.po:
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Update .po files
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|
Original commit message from CVS:
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Update .po files
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|
|
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2006-05-12 22:22:37 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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Fix the build.
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|
Original commit message from CVS:
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* gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
|
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
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Fix the build.
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2006-05-12 21:30:00 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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Fix integer overflow problem with pixel-aspect-ratio calculations in videoscale and xvimagesink (#341542)
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Original commit message from CVS:
|
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* docs/libs/gst-plugins-base-libs-docs.sgml:
|
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* docs/libs/gst-plugins-base-libs-sections.txt:
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* gst-libs/gst/video/video.c: (gst_video_calculate_display_ratio):
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* gst-libs/gst/video/video.h:
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* gst/videoscale/Makefile.am:
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* gst/videoscale/gstvideoscale.c: (gst_video_scale_fixate_caps):
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps):
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* tests/check/Makefile.am:
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* tests/check/libs/video.c: (GST_START_TEST), (video_suite),
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(main):
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Fix integer overflow problem with pixel-aspect-ratio calculations
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in videoscale and xvimagesink (#341542)
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2006-05-12 16:56:52 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/tag/gstid3tag.c: Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
|
|
Original commit message from CVS:
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* gst-libs/gst/tag/gstid3tag.c:
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Map GST_IMAGE_TAG to and from ID2v2 APIC frames (#341557).
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2006-05-12 10:39:08 +0000 Sébastien Moutte <sebastien@moutte.net>
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win32/MANIFEST: update win32 files listing
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|
Original commit message from CVS:
|
|
* win32/MANIFEST:
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update win32 files listing
|
|
|
|
2006-05-11 21:47:01 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* tests/check/elements/multifdsink.c:
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disable failing check on gentoo64
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|
Original commit message from CVS:
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|
disable failing check on gentoo64
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|
|
|
2006-05-11 21:35:44 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* tests/check/elements/multifdsink.c:
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disable failing check on gentoo64
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|
Original commit message from CVS:
|
|
disable failing check on gentoo64
|
|
|
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2006-05-11 21:20:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* tests/check/elements/multifdsink.c:
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macros show the correct line
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|
Original commit message from CVS:
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|
macros show the correct line
|
|
|
|
2006-05-11 21:04:08 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
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* tests/check/elements/multifdsink.c:
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macros show the correct line
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|
Original commit message from CVS:
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|
macros show the correct line
|
|
|
|
2006-05-11 21:01:05 +0000 Sjoerd Simons <sjoerd@luon.net>
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gst/playback/gstplaybasebin.*: API: GstPlayBaseBin::stream-info-value-array property use a more bindings-friendly way...
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|
Original commit message from CVS:
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|
2006-05-11 Thomas Vander Stichele <thomas at apestaart dot org>
|
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patch by: Sjoerd Simons (sjoerd@luon.net)
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* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
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(group_create), (group_destroy), (add_stream),
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(gst_play_base_bin_get_property),
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(gst_play_base_bin_get_streaminfo_value_array):
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* gst/playback/gstplaybasebin.h:
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API: GstPlayBaseBin::stream-info-value-array property
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use a more bindings-friendly way of exposing streaminfo
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using a GValueArray. Tested in ipython.
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Closes #341114
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2006-05-11 19:44:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
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|
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* tests/check/elements/multifdsink.c:
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fix some type warnings
|
|
Original commit message from CVS:
|
|
fix some type warnings
|
|
|
|
2006-05-11 19:38:22 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst/playback/gstdecodebin.c: Also catch queue underruns but don't do anything yet.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (try_to_link_1), (queue_enlarge),
|
|
(queue_underrun_cb), (queue_filled_cb):
|
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Also catch queue underruns but don't do anything yet.
|
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Refactor and comment queue enlarging code a bit.
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* gst/playback/gstplaybasebin.c: (queue_overrun),
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(queue_threshold_reached), (queue_out_of_data),
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(gen_preroll_element):
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If a queue over/underruns check that we don't create nasty
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deadlocks when the min-threshold is not reached but the
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max-bytes is. In those cases disable max-bytes when we
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know that the queue is fed timed data.
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Add more comments.
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2006-05-11 18:06:18 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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gst/playback/gstplaybin.c: Make playbin automatically plug an 'audioresample' element before the audio sink as well. ...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin.c: (gen_audio_element):
|
|
Make playbin automatically plug an 'audioresample'
|
|
element before the audio sink as well. This solves
|
|
problems with sinks that only accept a very specific
|
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sample rate, like esdsink (e.g. #340379).
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|
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2006-05-11 16:04:28 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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gst/playback/gstplaybasebin.c: Make http sources send special headers so that we receive icecast metadata if the http...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (gen_source_element):
|
|
Make http sources send special headers so that we receive
|
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icecast metadata if the http stream is an icecast stream
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(otherwise the server will just ignore them). This also
|
|
means that from now on users will need the 'icydemux'
|
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element from gst-plugins-good installed if they want to
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listen to icecast radio streams. (#341432, #333657).
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2006-05-11 12:34:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
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* gst/tcp/gstmultifdsink.c:
|
|
more commenting
|
|
Original commit message from CVS:
|
|
more commenting
|
|
|
|
2006-05-11 11:40:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
gst/tcp/gstmultifdsink.c: remove stupid example from docs - it should come with a simple
|
|
Original commit message from CVS:
|
|
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_remove_client_link),
|
|
(gst_multi_fd_sink_new_client), (gst_multi_fd_sink_stop):
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remove stupid example from docs - it should come with a simple
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C program instead.
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Clean up/fix docs
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* tests/check/elements/multifdsink.c: (wait_bytes_served),
|
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(fail_if_can_read), (GST_START_TEST),
|
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(gst_multifdsink_create_streamheader), (multifdsink_suite):
|
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add a test for changing streamheader which exposes a bug in
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|
multifdsink
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|
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2006-05-11 10:33:46 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
ext/gnomevfs/gstgnomevfssrc.*: Don't set icy-caps unless we have a sane interval value. Move interval to a local vari...
|
|
Original commit message from CVS:
|
|
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_init),
|
|
(gst_gnome_vfs_src_received_headers_callback):
|
|
* ext/gnomevfs/gstgnomevfssrc.h:
|
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Don't set icy-caps unless we have a sane interval value. Move
|
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interval to a local variable; we never use it outside this function.
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|
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2006-05-11 10:14:20 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
sys/: Register special buffer types along with the objects so that they are not registered at runtime from N differen...
|
|
Original commit message from CVS:
|
|
* sys/ximage/ximagesink.c: (gst_ximagesink_get_type):
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_type):
|
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Register special buffer types along with the objects so
|
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that they are not registered at runtime from N different
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streaming threads since they are not threadsafe.
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|
2006-05-10 18:31:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* tests/check/elements/multifdsink.c:
|
|
set caps and plug leaks
|
|
Original commit message from CVS:
|
|
set caps and plug leaks
|
|
|
|
2006-05-10 18:16:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
tests/check/elements/multifdsink.c: add two more tests, one doing streamheader
|
|
Original commit message from CVS:
|
|
* tests/check/elements/multifdsink.c: (wait_bytes_served),
|
|
(GST_START_TEST), (fail_unless_read), (multifdsink_suite):
|
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add two more tests, one doing streamheader
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|
|
2006-05-10 16:34:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
gst/tcp/gstmultifdsink.c: clean up the bufqueue when shutting down
|
|
Original commit message from CVS:
|
|
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_stop):
|
|
clean up the bufqueue when shutting down
|
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* tests/check/Makefile.am:
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* tests/check/elements/multifdsink.c: (setup_multifdsink),
|
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(cleanup_multifdsink), (GST_START_TEST), (multifdsink_suite),
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(main):
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add a test for the leak that was just fixed
|
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|
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2006-05-10 15:16:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* gst/tcp/gstmultifdsink.c:
|
|
DEBUG_FUNCPTR'ing
|
|
Original commit message from CVS:
|
|
DEBUG_FUNCPTR'ing
|
|
|
|
2006-05-10 15:14:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* gst/tcp/gstmultifdsink.c:
|
|
* gst/tcp/gstmultifdsink.h:
|
|
whitespace fixes
|
|
Original commit message from CVS:
|
|
whitespace fixes
|
|
|
|
2006-05-10 11:54:36 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/adder/gstadder.*: Updated some docs. Added comments and FIXMEs all over the place.
|
|
Original commit message from CVS:
|
|
* gst/adder/gstadder.c: (gst_adder_setcaps),
|
|
(gst_adder_query_duration), (gst_adder_query), (forward_event),
|
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(gst_adder_src_event), (gst_adder_sink_event),
|
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(gst_adder_class_init), (gst_adder_finalize),
|
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(gst_adder_request_new_pad), (gst_adder_collected):
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* gst/adder/gstadder.h:
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Updated some docs. Added comments and FIXMEs all over the place.
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Improve debugging info.
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Fix leak on finalize by not calling the parent.
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Implement duration query.
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Make event forwarding threadsafe.
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Correctly send NEWSEGMENT at start and after flush.
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Handle EOS correctly.
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Post error when not negotiated.
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* tests/check/elements/adder.c: (GST_START_TEST):
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Added FIXME in the test.
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2006-05-09 19:24:46 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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Const-ify GEnumValue and GFlagsValue arrays. Use
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|
Original commit message from CVS:
|
|
* ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
|
|
(gst_text_overlay_halign_get_type),
|
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(gst_text_overlay_wrap_mode_get_type):
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* ext/theora/theoradec.c: (theora_handle_type_packet),
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(theora_handle_data_packet):
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* ext/theora/theoraenc.c: (gst_border_mode_get_type),
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(theora_enc_sink_setcaps), (theora_enc_chain):
|
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* gst-libs/gst/cdda/gstcddabasesrc.c:
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(gst_cdda_base_src_mode_get_type):
|
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* gst/audiotestsrc/gstaudiotestsrc.c:
|
|
(gst_audiostestsrc_wave_get_type):
|
|
* gst/playback/gststreaminfo.c: (gst_stream_type_get_type):
|
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* gst/tcp/gstfdset.c: (gst_fdset_mode_get_type):
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* gst/tcp/gstmultifdsink.c: (gst_recover_policy_get_type),
|
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(gst_sync_method_get_type), (gst_unit_type_get_type),
|
|
(gst_client_status_get_type):
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* gst/videoscale/gstvideoscale.c:
|
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(gst_video_scale_method_get_type):
|
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* gst/videotestsrc/gstvideotestsrc.c:
|
|
(gst_video_test_src_pattern_get_type):
|
|
* gst/videotestsrc/videotestsrc.c: (paint_setup_I420),
|
|
(paint_setup_YV12), (paint_setup_YUY2), (paint_setup_UYVY),
|
|
(paint_setup_YVYU), (paint_setup_IYU2), (paint_setup_Y41B),
|
|
(paint_setup_Y42B), (paint_setup_Y800), (paint_setup_YVU9),
|
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(paint_setup_YUV9), (paint_setup_RGB888), (paint_setup_BGR888),
|
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(paint_setup_RGB565), (paint_setup_xRGB1555):
|
|
Const-ify GEnumValue and GFlagsValue arrays. Use
|
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GST_ROUND_UP_* macros instead of home-made ones.
|
|
|
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2006-05-09 17:40:41 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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configure.ac: Require core CVS for the new newsegment stuff.
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|
Original commit message from CVS:
|
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* configure.ac:
|
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Require core CVS for the new newsegment stuff.
|
|
|
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2006-05-09 17:30:48 +0000 Sjoerd Simons <sjoerd@luon.net>
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|
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gst/tcp/gstmultifdsink.c: Register nick for enum value (#341160).
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|
Original commit message from CVS:
|
|
Patch by: Sjoerd Simons <sjoerd at luon net>
|
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* gst/tcp/gstmultifdsink.c: (gst_client_status_get_type):
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Register nick for enum value (#341160).
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|
|
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2006-05-09 16:46:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
|
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gst/typefind/gsttypefindfunctions.c: backout typefind patch #340375
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (m4a_type_find),
|
|
(plugin_init):
|
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backout typefind patch #340375
|
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* tests/check/elements/adder.c: (message_received),
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|
(GST_START_TEST), (adder_suite):
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redo, signal-handling of test
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|
|
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2006-05-09 16:14:49 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/adder/gstadder.*: Remove bogus segment merging and forwarding, we don't care about timestamps anyway and we just ...
|
|
Original commit message from CVS:
|
|
* gst/adder/gstadder.c: (gst_adder_request_new_pad),
|
|
(gst_adder_collected):
|
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* gst/adder/gstadder.h:
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Remove bogus segment merging and forwarding, we don't
|
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care about timestamps anyway and we just produce a
|
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continuous stream.
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Also create a nice NEWSEGMENT event when we start.
|
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Use _scale_int some more.
|
|
|
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2006-05-09 11:59:13 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
tests/icles/stress-xoverlay.c: Fix if core was built without parsing support.
|
|
Original commit message from CVS:
|
|
* tests/icles/stress-xoverlay.c:
|
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Fix if core was built without parsing support.
|
|
|
|
2006-05-09 11:37:22 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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gst-libs/gst/riff/riff-media.c: Add SEDG (Samsung MPEG-4) fourcc.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
|
|
Add SEDG (Samsung MPEG-4) fourcc.
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|
|
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2006-05-09 11:31:47 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
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tests/examples/volume/volume.c: Fox if core was built without parsing support.
|
|
Original commit message from CVS:
|
|
* tests/examples/volume/volume.c:
|
|
Fox if core was built without parsing support.
|
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* tests/examples/seek/seek.c:
|
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Disable the parse_launch example if core was built without parsing
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support.
|
|
|
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2006-05-09 11:21:24 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
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tests/examples/seek/seek.c: Disable the parse_launch example if core was built without parsing support.
|
|
Original commit message from CVS:
|
|
* tests/examples/seek/seek.c:
|
|
Disable the parse_launch example if core was built without parsing
|
|
support.
|
|
|
|
2006-05-08 15:51:15 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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|
|
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* docs/libs/tmpl/gstcolorbalance.sgml:
|
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* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
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* gst/tcp/gstmultifdsink.c:
|
|
* gst/videoscale/gstvideoscale.c:
|
|
doc reparagraphing and DEBUG_FUNCPTRing
|
|
Original commit message from CVS:
|
|
doc reparagraphing and DEBUG_FUNCPTRing
|
|
|
|
2006-05-08 11:51:43 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
autogen.sh: libtoolize on Darwin/MacOSX is called glibtoolize
|
|
Original commit message from CVS:
|
|
* autogen.sh: (CONFIGURE_DEF_OPT):
|
|
libtoolize on Darwin/MacOSX is called glibtoolize
|
|
|
|
2006-05-07 17:39:04 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
tests/check/: Disable the adder test, until the build-slaves posses the kindness to either like it or to give valid r...
|
|
Original commit message from CVS:
|
|
* tests/check/Makefile.am:
|
|
* tests/check/elements/adder.c: (event_loop), (GST_START_TEST):
|
|
Disable the adder test, until the build-slaves posses the kindness to
|
|
either like it or to give valid reason for not doing so
|
|
|
|
2006-05-07 17:25:56 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
tests/check/elements/adder.c: Shuffle NULL state change around and raise timeout more
|
|
Original commit message from CVS:
|
|
* tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
|
|
(adder_suite):
|
|
Shuffle NULL state change around and raise timeout more
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|
|
|
2006-05-07 17:07:03 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Add typefind to distinguish between "audio/x-m4a" and new type "video/mp4". Fixe...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (mp4_find_box),
|
|
(mp4_type_find), (plugin_init):
|
|
Add typefind to distinguish between "audio/x-m4a" and new type
|
|
"video/mp4". Fixes #340375
|
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* tests/check/elements/adder.c: (adder_suite):
|
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Raise timeout to make buildbot happy
|
|
|
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2006-05-07 16:39:36 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
Add sink-event handling to adder. It tries to merge incomming newsegment-events. Added test to check if segment_done ...
|
|
Original commit message from CVS:
|
|
* gst/adder/gstadder.c: (gst_adder_sink_event),
|
|
(gst_adder_request_new_pad), (gst_adder_change_state):
|
|
* gst/adder/gstadder.h:
|
|
* tests/check/Makefile.am:
|
|
* tests/check/elements/adder.c: (event_loop), (GST_START_TEST),
|
|
(adder_suite), (main):
|
|
Add sink-event handling to adder. It tries to merge incomming
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|
newsegment-events. Added test to check if segment_done is comming
|
|
through.
|
|
|
|
2006-05-05 16:34:15 +0000 Andy Wingo <wingo@pobox.com>
|
|
|
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* ChangeLog:
|
|
* ext/theora/theoraparse.c:
|
|
* ext/vorbis/vorbisparse.c:
|
|
ext/theora/theoraparse.c (gst_theora_parse_init) ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
|
|
Original commit message from CVS:
|
|
2006-05-05 Andy Wingo <wingo@pobox.com>
|
|
* ext/theora/theoraparse.c (gst_theora_parse_init)
|
|
(theora_parse_src_convert, theora_parse_src_query):
|
|
* ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
|
|
(vorbis_parse_convert, vorbis_parse_src_query): Add convert and
|
|
query functions on the source pads of the theora and vorbis parse
|
|
elements. Fixes position querying when doing a remux.
|
|
|
|
2006-05-05 13:46:37 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
ext/theora/theoraparse.c: Fix flushing.
|
|
Original commit message from CVS:
|
|
* ext/theora/theoraparse.c: (parse_granulepos),
|
|
(theora_parse_drain_queue_prematurely),
|
|
(theora_parse_queue_buffer), (theora_parse_sink_event):
|
|
Fix flushing.
|
|
Fix invalid granulepos outputs when starting with a non-keyframe.
|
|
|
|
2006-05-05 12:37:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Rearrange MPEG system stream detection, fixing some memleaks in the process.
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (mpeg2_sys_type_find),
|
|
(mpeg1_sys_type_find), (ogganx_type_find), (sw_data_destroy):
|
|
Rearrange MPEG system stream detection, fixing some memleaks in the
|
|
process.
|
|
Constify the data for STARTS_WITH and RIFF helper handlers. Make sure
|
|
they clean up their data correctly.
|
|
Remove unused ogganx caps and move the 'is_annodex' check to inside
|
|
the 'is_ogg' if statement.
|
|
|
|
2006-05-05 11:33:37 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstdecodebin.c: Properly remove ghostpads. Fixes #340392
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (cleanup_decodebin):
|
|
Properly remove ghostpads. Fixes #340392
|
|
|
|
2006-05-04 18:43:58 +0000 David Schleef <ds@schleef.org>
|
|
|
|
gst/typefind/gsttypefindfunctions.c:
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
|
|
2006-05-03 16:32:19 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: When typefinding an MP3 in push-based mode, don't penalise the probability down ...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
|
|
(mpeg_ts_probe_headers), (mpeg_ts_type_find):
|
|
When typefinding an MP3 in push-based mode, don't penalise the
|
|
probability down to 74% when we found 5 valid frames just because we
|
|
can't peek the end of the file.
|
|
Make the probability for detecting MPEG Transport Streams based on the
|
|
number of sequential headers we successfully detected.
|
|
|
|
2006-05-03 15:52:46 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/vorbis/vorbisdec.c: Still produce an error when we receive an empty packet.
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
|
|
(vorbis_dec_push), (vorbis_dec_chain):
|
|
Still produce an error when we receive an empty packet.
|
|
|
|
2006-05-03 15:34:48 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/ogg/gstoggdemux.c: Mark buffers with DISCONT after seek and after activating new chains.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_peer),
|
|
(gst_ogg_chain_mark_discont), (gst_ogg_chain_new_stream),
|
|
(gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek):
|
|
Mark buffers with DISCONT after seek and after activating new
|
|
chains.
|
|
* ext/theora/gsttheoradec.h:
|
|
* ext/theora/theoradec.c: (gst_theora_dec_reset),
|
|
(theora_get_query_types), (theora_dec_sink_event),
|
|
(theora_dec_push), (theora_handle_data_packet), (theora_dec_chain),
|
|
(theora_dec_change_state):
|
|
Fix frame counter.
|
|
Detect and mark DISCONT buffers.
|
|
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
|
|
(vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_dec_chain),
|
|
(vorbis_dec_change_state):
|
|
* ext/vorbis/vorbisdec.h:
|
|
Use GstSegment.
|
|
Detect and mark DISCONT buffers.
|
|
Don't crash on 0 sized buffers.
|
|
|
|
2006-05-03 08:58:13 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/volume/gstvolume.c: Increase "volume" property to 10.0. Fixes #340369.
|
|
Original commit message from CVS:
|
|
* gst/volume/gstvolume.c: (volume_funcfind), (volume_set_caps),
|
|
(volume_transform_ip):
|
|
Increase "volume" property to 10.0. Fixes #340369.
|
|
Set the process function to NULL when capsnego fails so that
|
|
we properly error out.
|
|
|
|
2006-05-02 18:15:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/playback/: free cpas using gst_caps_unref, don't leak caps-strings
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin.c: (add_sink):
|
|
* gst/playback/test.c: (main):
|
|
* gst/playback/test5.c: (dump_element_stats):
|
|
* gst/playback/test6.c: (main):
|
|
free cpas using gst_caps_unref, don't leak caps-strings
|
|
|
|
2006-05-02 06:33:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* gst-libs/gst/rtp/gstbasertppayload.c:
|
|
some RTP debug
|
|
Original commit message from CVS:
|
|
some RTP debug
|
|
|
|
2006-05-01 19:08:40 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Refine musepack typefinding a bit. Return MAXIMUM probability when we detect str...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (musepack_type_find),
|
|
(plugin_init):
|
|
Refine musepack typefinding a bit. Return MAXIMUM
|
|
probability when we detect stream version 7 to make
|
|
sure the mpeg audio typefinder doesn't trump us.
|
|
|
|
2006-04-29 16:25:58 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/riff/riff-media.c: Protect against unexpected NULL strf_data buffer.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
|
|
Protect against unexpected NULL strf_data buffer.
|
|
|
|
2006-04-29 13:09:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
tests/check/elements/audioconvert.c: interpret the out[] buffer in the order the bytes are actually put in, which is ...
|
|
Original commit message from CVS:
|
|
* tests/check/elements/audioconvert.c: (verify_convert),
|
|
(GST_START_TEST):
|
|
interpret the out[] buffer in the order the bytes are actually
|
|
put in, which is LITTLE_ENDIAN, not BYTE_ORDER.
|
|
Other tests should use BYTE_ORDER since the array is filled in
|
|
with actual values
|
|
|
|
2006-04-29 12:10:52 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* tests/check/elements/audioconvert.c:
|
|
dump expected data when audioconvert test fails
|
|
Original commit message from CVS:
|
|
dump expected data when audioconvert test fails
|
|
|
|
2006-04-29 11:55:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
tests/check/elements/audioconvert.c: when a test fails, give an indication of which it is
|
|
Original commit message from CVS:
|
|
* tests/check/elements/audioconvert.c: (verify_convert),
|
|
(GST_START_TEST):
|
|
when a test fails, give an indication of which it is
|
|
|
|
2006-04-29 09:48:16 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* ext/ogg/gstoggmux.c:
|
|
* ext/theora/theoraenc.c:
|
|
add another include
|
|
Original commit message from CVS:
|
|
add another include
|
|
|
|
2006-04-29 01:24:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* gst/subparse/gstssaparse.c:
|
|
atoi() needs stdlib.h
|
|
Original commit message from CVS:
|
|
atoi() needs stdlib.h
|
|
|
|
2006-04-29 01:18:05 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* gst/playback/test4.c:
|
|
* gst/playback/test5.c:
|
|
* gst/playback/test6.c:
|
|
exit needs stdlib.h
|
|
Original commit message from CVS:
|
|
exit needs stdlib.h
|
|
|
|
2006-04-29 01:10:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
gst-libs/gst/cdda/gstcddabasesrc.c: compile fix; strtol() needs <stdlib.h>
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/cdda/gstcddabasesrc.c:
|
|
compile fix; strtol() needs <stdlib.h>
|
|
|
|
2006-04-29 01:04:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* ChangeLog:
|
|
* common:
|
|
* docs/Makefile.am:
|
|
* docs/libs/Makefile.am:
|
|
* docs/libs/tmpl/gstcolorbalance.sgml:
|
|
* docs/plugins/Makefile.am:
|
|
* docs/upload.mak:
|
|
use common upload.mak
|
|
Original commit message from CVS:
|
|
use common upload.mak
|
|
|
|
2006-04-28 19:46:37 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
make GstElementDetails const
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsamixerelement.c:
|
|
* ext/alsa/gstalsasrc.c:
|
|
* ext/cdparanoia/gstcdparanoiasrc.c:
|
|
* ext/gnomevfs/gstgnomevfssink.c:
|
|
* ext/gnomevfs/gstgnomevfssrc.c:
|
|
* ext/ogg/gstoggdemux.c:
|
|
* ext/ogg/gstoggmux.c:
|
|
* ext/ogg/gstoggparse.c:
|
|
* ext/ogg/gstogmparse.c:
|
|
* ext/pango/gstclockoverlay.c:
|
|
* ext/pango/gsttextoverlay.c:
|
|
* ext/pango/gsttextrender.c:
|
|
* ext/pango/gsttimeoverlay.c:
|
|
* ext/theora/theoradec.c:
|
|
* ext/theora/theoraenc.c:
|
|
* ext/vorbis/vorbisdec.c:
|
|
* ext/vorbis/vorbisenc.c:
|
|
* gst-libs/gst/audio/gstaudiofilter.c:
|
|
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
|
|
* gst/audioconvert/gstaudioconvert.c:
|
|
* gst/audiorate/gstaudiorate.c:
|
|
* gst/audioresample/gstaudioresample.c:
|
|
* gst/audiotestsrc/gstaudiotestsrc.c:
|
|
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
|
|
* gst/playback/gstdecodebin.c:
|
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* gst/playback/gstplaybin.c:
|
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* gst/playback/gststreamselector.c:
|
|
* gst/subparse/gstsubparse.c:
|
|
* gst/tcp/gstmultifdsink.c:
|
|
* gst/tcp/gsttcpclientsink.c:
|
|
* gst/tcp/gsttcpclientsrc.c:
|
|
* gst/tcp/gsttcpserversink.c:
|
|
* gst/tcp/gsttcpserversrc.c:
|
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* gst/typefind/gsttypefindfunctions.c: (plugin_init):
|
|
* gst/videorate/gstvideorate.c:
|
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* gst/videoscale/gstvideoscale.c:
|
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* gst/videotestsrc/gstvideotestsrc.c:
|
|
* gst/volume/gstvolume.c:
|
|
* sys/v4l/gstv4ljpegsrc.c:
|
|
* sys/v4l/gstv4lmjpegsink.c:
|
|
* sys/v4l/gstv4lmjpegsrc.c:
|
|
* sys/v4l/gstv4lsrc.c:
|
|
* sys/ximage/ximagesink.c:
|
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* sys/xvimage/xvimagesink.c:
|
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* tests/check/libs/cddabasesrc.c:
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make GstElementDetails const
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|
|
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2006-04-28 19:17:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
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gst/adder/gstadder.c: send events from src-pad to all sink-pads fixes #338657
|
|
Original commit message from CVS:
|
|
* gst/adder/gstadder.c: (gst_adder_setcaps), (gst_adder_src_event),
|
|
(gst_adder_init):
|
|
send events from src-pad to all sink-pads fixes #338657
|
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|
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2006-04-28 19:08:34 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
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ext/alsa/gstalsasink.c: query witdh capabilities from alsa, fixes #338919
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsasink.c: (gst_alsasink_getcaps),
|
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(alsasink_parse_spec):
|
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query witdh capabilities from alsa, fixes #338919
|
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|
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2006-04-28 15:31:28 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst/tcp/gstmultifdsink.*: Fix race condition in multifdsink that can lead to spurious duplicate clients. this patch a...
|
|
Original commit message from CVS:
|
|
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init),
|
|
(gst_multi_fd_sink_remove_client_link):
|
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* gst/tcp/gstmultifdsink.h:
|
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Fix race condition in multifdsink that can lead to spurious
|
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duplicate clients. this patch adds a new signal that is fired when
|
|
multifdsink has removed all references to the fd.
|
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Fixes #339574.
|
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Updated documentation.
|
|
API: client-fd-removed signal added
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|
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2006-04-28 15:24:00 +0000 Michael Smith <msmith@xiph.org>
|
|
|
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gst/tcp/gstmultifdsink.c: When asking g_value_array_new to prealloc elements, we may as well ask for the right number...
|
|
Original commit message from CVS:
|
|
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_get_stats):
|
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When asking g_value_array_new to prealloc elements, we may as well
|
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ask for the right number of elements.
|
|
|
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2006-04-28 15:08:09 +0000 Wim Taymans <wim.taymans@gmail.com>
|
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|
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gst-libs/gst/audio/gstbaseaudiosink.c: patch to make timestamp checking more tollerant to rounding errors given that ...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
|
|
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
|
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patch to make timestamp checking more tollerant to rounding
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errors given that real discontinuities are to be marked on
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buffers. Fixes some asf files and #338778.
|
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Also avoid some crashers when we receive an event in the
|
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NULL state.
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2006-04-28 15:01:58 +0000 Michael Smith <msmith@xiph.org>
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|
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ext/gnomevfs/gstgnomevfssrc.*: Remove ICY handling (mostly) from gnomevfssrc, in favour of proper shared support with...
|
|
Original commit message from CVS:
|
|
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
|
|
(gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
|
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(gst_gnome_vfs_src_get_property),
|
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(gst_gnome_vfs_src_send_additional_headers_callback),
|
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(gst_gnome_vfs_src_received_headers_callback),
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(gst_gnome_vfs_src_create), (gst_gnome_vfs_src_start),
|
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(gst_gnome_vfs_src_stop):
|
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* ext/gnomevfs/gstgnomevfssrc.h:
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Remove ICY handling (mostly) from gnomevfssrc, in favour of
|
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proper shared support within icydemux.
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|
|
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2006-04-28 14:49:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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|
|
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gst/videorate/gstvideorate.c: fix up docs fix a leak when no caps negotiated fix counting of input frames
|
|
Original commit message from CVS:
|
|
* gst/videorate/gstvideorate.c: (gst_video_rate_reset),
|
|
(gst_video_rate_swap_prev), (gst_video_rate_chain):
|
|
fix up docs
|
|
fix a leak when no caps negotiated
|
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fix counting of input frames
|
|
* tests/check/elements/.cvsignore:
|
|
* tests/check/elements/videorate.c: (assert_videorate_stats),
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|
(GST_START_TEST), (videorate_suite):
|
|
add tests for these
|
|
|
|
2006-04-28 14:48:11 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstringbuffer.c: Check arguments passed to public functions instead of crashing.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstringbuffer.c:
|
|
(gst_ring_buffer_set_callback), (gst_ring_buffer_acquire),
|
|
(gst_ring_buffer_release), (gst_ring_buffer_is_acquired),
|
|
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
|
|
(gst_ring_buffer_pause), (gst_ring_buffer_stop),
|
|
(gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
|
|
(gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
|
|
(gst_ring_buffer_commit), (gst_ring_buffer_read),
|
|
(gst_ring_buffer_prepare_read), (gst_ring_buffer_advance),
|
|
(gst_ring_buffer_clear), (gst_ring_buffer_may_start):
|
|
Check arguments passed to public functions instead of
|
|
crashing.
|
|
|
|
2006-04-28 14:37:46 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosrc.c: GstBaseAudioSrc must be live or it does not work.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
|
|
(gst_base_audio_src_get_time), (gst_base_audio_src_create):
|
|
GstBaseAudioSrc must be live or it does not work.
|
|
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audio_src_init):
|
|
Don't set live to TRUE as this is the default in the parentclass.
|
|
|
|
2006-04-28 14:37:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* win32/common/config.h:
|
|
update config.h
|
|
Original commit message from CVS:
|
|
update config.h
|
|
|
|
2006-04-28 14:33:45 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/videoscale/gstvideoscale.c: Videoscale doesn't pass on pixel-aspect ratio. Handle all fixation cases better. Fixe...
|
|
Original commit message from CVS:
|
|
* gst/videoscale/gstvideoscale.c: (gst_video_scale_transform_caps),
|
|
(gst_video_scale_fixate_caps), (gst_video_scale_src_event):
|
|
Videoscale doesn't pass on pixel-aspect ratio. Handle all
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|
fixation cases better. Fixes #338991
|
|
|
|
2006-04-28 14:24:38 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/videotestsrc/gstvideotestsrc.c: Handle 0/1 framerate correctly Fixes #331901.
|
|
Original commit message from CVS:
|
|
* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create):
|
|
Handle 0/1 framerate correctly Fixes #331901.
|
|
|
|
2006-04-28 14:22:16 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
tests/check/elements/audioconvert.c: Added check for correct clipping when doing float samples in audioconvert.
|
|
Original commit message from CVS:
|
|
* tests/check/elements/audioconvert.c: (get_float_caps),
|
|
(GST_START_TEST), (audioconvert_suite):
|
|
Added check for correct clipping when doing float samples
|
|
in audioconvert.
|
|
|
|
2006-04-28 14:19:49 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/videorate/gstvideorate.c: Print more debugging info.
|
|
Original commit message from CVS:
|
|
* gst/videorate/gstvideorate.c: (gst_video_rate_event),
|
|
(gst_video_rate_chain):
|
|
Print more debugging info.
|
|
|
|
2006-04-28 14:17:00 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/audioresample/gstaudioresample.c: Add support for other formats audioresample can handle such as 32 bits in and f...
|
|
Original commit message from CVS:
|
|
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
|
|
(resample_set_state_from_caps):
|
|
Add support for other formats audioresample can handle such as
|
|
32 bits in and float and 64 bits float. Fixes #301759
|
|
|
|
2006-04-28 14:12:28 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/audioconvert/audioconvert.c: correctly clip float samples > 1.0. Fixes #338718
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/audioconvert.c: (float):
|
|
correctly clip float samples > 1.0. Fixes #338718
|
|
|
|
2006-04-28 13:35:34 +0000 Young-Ho Cha <ganadist@chollian.net>
|
|
|
|
ext/pango/gsttextoverlay.c: Don't strip newlines from the text. Also, center lines within multi-line paragraphs (#339...
|
|
Original commit message from CVS:
|
|
Patch by: Young-Ho Cha <ganadist at chollian net>
|
|
* ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
|
|
(gst_text_overlay_render_text):
|
|
Don't strip newlines from the text. Also, center lines
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|
within multi-line paragraphs (#339405).
|
|
|
|
2006-04-28 12:15:33 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Fix wavpack typefinding to work in more cases (don't peek for chunks of multiple...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (wavpack_type_find):
|
|
Fix wavpack typefinding to work in more cases (don't peek
|
|
for chunks of multiple hundred kBs at once, but process
|
|
things step-by-step in smaller units). Fixes #339786.
|
|
|
|
2006-04-28 10:58:41 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* ChangeLog:
|
|
* configure.ac:
|
|
back to HEAD
|
|
Original commit message from CVS:
|
|
back to HEAD
|
|
|
|
=== release 0.10.6 ===
|
|
|
|
2006-04-28 10:53:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* ChangeLog:
|
|
* NEWS:
|
|
* RELEASE:
|
|
* configure.ac:
|
|
* docs/plugins/gst-plugins-base-plugins.signals:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.xml:
|
|
* docs/plugins/inspect/plugin-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-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.xml:
|
|
* docs/plugins/inspect/plugin-libvisual.xml:
|
|
* docs/plugins/inspect/plugin-ogg.xml:
|
|
* docs/plugins/inspect/plugin-pango.xml:
|
|
* docs/plugins/inspect/plugin-playbin.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-video4linux.xml:
|
|
* docs/plugins/inspect/plugin-videorate.xml:
|
|
* docs/plugins/inspect/plugin-videoscale.xml:
|
|
* docs/plugins/inspect/plugin-videotestsrc.xml:
|
|
* docs/plugins/inspect/plugin-volume.xml:
|
|
* docs/plugins/inspect/plugin-vorbis.xml:
|
|
* docs/plugins/inspect/plugin-ximagesink.xml:
|
|
* docs/plugins/inspect/plugin-xvimagesink.xml:
|
|
* docs/upload.mak:
|
|
releasing 0.10.6
|
|
Original commit message from CVS:
|
|
releasing 0.10.6
|
|
|
|
2006-04-28 10:42:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* win32/MANIFEST:
|
|
* win32/common/config.h:
|
|
dist more win32 files
|
|
Original commit message from CVS:
|
|
dist more win32 files
|
|
|
|
2006-04-28 10:41:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/cs.po:
|
|
* po/en_GB.po:
|
|
* po/hu.po:
|
|
* po/it.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
Update .po files
|
|
Original commit message from CVS:
|
|
Update .po files
|
|
|
|
2006-04-27 00:19:29 +0000 David Schleef <ds@schleef.org>
|
|
|
|
gst/videoscale/gstvideoscale.c: Add call to oil_init().
|
|
Original commit message from CVS:
|
|
* gst/videoscale/gstvideoscale.c: Add call to oil_init().
|
|
Fixes #338897.
|
|
|
|
2006-04-26 17:20:31 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* ChangeLog:
|
|
* configure.ac:
|
|
* win32/common/config.h:
|
|
new prerelease
|
|
Original commit message from CVS:
|
|
new prerelease
|
|
|
|
2006-04-26 17:17:39 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/ogg/gstoggdemux.c: make sure correct newsegments are sent, so that the decoder and the demuxer agree on timestamp...
|
|
Original commit message from CVS:
|
|
2006-04-26 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
patch by: Wim Taymans
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
|
|
(gst_ogg_demux_perform_seek):
|
|
make sure correct newsegments are sent, so that the decoder
|
|
and the demuxer agree on timestamps. Fixes playback of a lot
|
|
of Ogg files that do not start from 0. Fixes #339833.
|
|
|
|
2006-04-26 16:44:20 +0000 Edward Hervey <edward@fluendo.com>
|
|
|
|
Fix an infinite loop if frames are passed in with wrongly ordered timestamps. Fixes #339013.
|
|
Original commit message from CVS:
|
|
Patch by: Edward Hervey <edward@fluendo.com>
|
|
* gst/videorate/gstvideorate.c: (gst_video_rate_chain):
|
|
* tests/check/Makefile.am:
|
|
* tests/check/elements/videorate.c: (assert_videorate_stats),
|
|
(setup_videorate), (cleanup_videorate), (GST_START_TEST),
|
|
(videorate_suite), (main):
|
|
Fix an infinite loop if frames are passed in with wrongly ordered
|
|
timestamps. Fixes #339013.
|
|
|
|
2006-04-26 13:55:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* configure.ac:
|
|
* win32/common/config.h:
|
|
prerelease
|
|
Original commit message from CVS:
|
|
prerelease
|
|
|
|
2006-04-22 21:25:41 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: fix typefinding on some ISO files. Fixes #339212.
|
|
Original commit message from CVS:
|
|
Patch by: Tim-Philipp Müller <tim at centricular dot net>
|
|
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
|
|
fix typefinding on some ISO files. Fixes #339212.
|
|
|
|
2006-04-22 21:19:06 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/riff/riff-media.c: add another H264 fourcc. Fixes #339047.
|
|
Original commit message from CVS:
|
|
Patch by: Tim-Philipp Müller <tim at centricular dot net>
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
|
|
add another H264 fourcc. Fixes #339047.
|
|
|
|
2006-04-22 21:12:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst/playback/gststreamselector.c: Restore old StreamSelector behaviour.
|
|
Original commit message from CVS:
|
|
Patch by: Jan Schmidt
|
|
* gst/playback/gststreamselector.c:
|
|
(gst_stream_selector_bufferalloc):
|
|
Restore old StreamSelector behaviour.
|
|
Fixes #338419.
|
|
|
|
2006-04-13 09:26:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* ChangeLog:
|
|
* gst-libs/gst/rtp/Makefile.am:
|
|
* gst-libs/gst/rtp/gstrtpbuffer.h:
|
|
reverting rtp patches to fix freeze break on -base as explained on the list
|
|
Original commit message from CVS:
|
|
reverting rtp patches to fix freeze break on -base as explained on the list
|
|
|
|
2006-04-13 03:55:12 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
|
|
|
|
gst-libs/gst/rtp/gstrtpbuffer.h: Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
|
|
Original commit message from CVS:
|
|
2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
|
|
* gst-libs/gst/rtp/gstrtpbuffer.h:
|
|
Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
|
|
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
|
|
New RTP audio base payloader class. Supports frame or sample based codecs
|
|
|
|
2006-04-12 11:04:53 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* ChangeLog:
|
|
* configure.ac:
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/cs.po:
|
|
* po/en_GB.po:
|
|
* po/hu.po:
|
|
* po/it.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
update libtool versioning
|
|
Original commit message from CVS:
|
|
update libtool versioning
|
|
|
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2006-04-12 10:58:00 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* ChangeLog:
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* configure.ac:
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* win32/common/config.h:
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prerelease
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Original commit message from CVS:
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prerelease
|
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2006-04-11 17:31:29 +0000 Antoine Tremblay <hexa00@gmail.com>
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gst-libs/gst/rtp/gstbasertpdepayload.c: Fix some memory leaks: on finalize, free buffers left in the queue before des...
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Original commit message from CVS:
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Patch by: Antoine Tremblay <hexa00 at gmail dot com>
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* gst-libs/gst/rtp/gstbasertpdepayload.c:
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(gst_base_rtp_depayload_finalize), (gst_base_rtp_depayload_push):
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Fix some memory leaks: on finalize, free buffers left in the queue
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before destroying the queue; in _push(), unref rtp_buf even if
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the process vfunc returned a NULL buffer as output buffer (#337548);
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demote some recuring debug messages to LOG level.
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2006-04-11 15:01:51 +0000 Christian Schaller <uraeus@gnome.org>
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* gst-plugins-base.spec.in:
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fix version number macro
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Original commit message from CVS:
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fix version number macro
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2006-04-11 14:42:33 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/ogg/gstoggdemux.c: More cleanups.
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Original commit message from CVS:
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* ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
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(gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
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(gst_ogg_chain_free), (gst_ogg_demux_sink_event),
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(gst_ogg_demux_loop):
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More cleanups.
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Respect segment stop when emiting EOS or SEGMENT_DONE.
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Fixes (#337945).
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2006-04-11 10:45:32 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gststreamselector.c: Don't leak pad name.
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Original commit message from CVS:
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* gst/playback/gststreamselector.c:
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(gst_stream_selector_get_property):
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Don't leak pad name.
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2006-04-11 09:42:52 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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* ChangeLog:
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Mention bug #336617 closed by recent commit
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Original commit message from CVS:
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Mention bug #336617 closed by recent commit
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2006-04-10 20:32:46 +0000 Michael Smith <msmith@xiph.org>
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tests/check/: so that FC4 buildslaves can pass.
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Original commit message from CVS:
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* tests/check/Makefile.am:
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* tests/check/gst-plugins-base.supp:
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Suppress an old libtheora bug (fixed in more recent versions), so
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that FC4 buildslaves can pass.
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2006-04-10 19:13:30 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/ogg/gstoggdemux.c: Don't leak events.
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Original commit message from CVS:
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* ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
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(gst_ogg_demux_receive_event), (gst_ogg_pad_event),
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(gst_ogg_demux_init), (gst_ogg_demux_finalize),
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(gst_ogg_demux_sink_event), (gst_ogg_demux_get_data),
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(gst_ogg_demux_loop):
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Don't leak events.
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Remember what error we got when finding chains, if we
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were shutdown, that would not be an error.
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2006-04-10 17:05:46 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstbaseaudiosink.c: Starting the ringbuffer when we did not acquire it can cause a deadlock, is po...
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Original commit message from CVS:
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_event):
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Starting the ringbuffer when we did not acquire it can cause
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a deadlock, is pointless and causes nasty things for
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subclasses.
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Fixes gst-launch audiotestsrc num-buffers=0 ! alsasink.
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2006-04-10 15:17:24 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/ogg/gstoggdemux.c: Add some more debugging.
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Original commit message from CVS:
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* ext/ogg/gstoggdemux.c: (gst_ogg_pad_src_query),
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(gst_ogg_demux_receive_event), (gst_ogg_pad_event),
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(gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
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(gst_ogg_demux_submit_buffer), (gst_ogg_demux_get_data),
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(gst_ogg_demux_deactivate_current_chain),
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(gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek),
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(gst_ogg_demux_bisect_forward_serialno),
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(gst_ogg_demux_find_chains), (gst_ogg_demux_chain):
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Add some more debugging.
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2006-04-10 14:52:10 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* common:
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* ext/theora/theoraenc.c:
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fix width of docs
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Original commit message from CVS:
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fix width of docs
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2006-04-10 10:29:21 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/theora/theoradec.c: Some more debug info.
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|
Original commit message from CVS:
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* ext/theora/theoradec.c: (theora_dec_src_event),
|
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(theora_handle_data_packet):
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Some more debug info.
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* tests/examples/seek/seek.c: (start_seek), (main):
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Print element messages too.
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2006-04-09 17:14:22 +0000 Sébastien Moutte <sebastien@moutte.net>
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gst/audioresample/debug.h: replace debug macros with variable number of parameters by a simple alias to gstreamer sta...
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|
Original commit message from CVS:
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* gst/audioresample/debug.h:
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replace debug macros with variable number of parameters
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by a simple alias to gstreamer standard debug macros
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(#define RESAMPLE_ERROR GST_ERROR, __VA_ARGS__ is not
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supported by MSVC 6.0 and 7.1)
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* gst/audioresample/resample.h:
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define M_PI and rint for WIN32
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* win32/common/libgstaudio.def:
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* win32/common/libgstriff.def:
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* win32/common/libgsttag.def:
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* win32/common/libgstvideo.def:
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add new exported functions
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* win32/vs6:
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update project files
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2006-04-08 21:02:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
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|
Original commit message from CVS:
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* ext/alsa/gstalsamixeroptions.c:
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(gst_alsa_mixer_options_class_init):
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* ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init):
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* ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init):
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* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init):
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* ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init):
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* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init):
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* gst-libs/gst/audio/gstaudiofilter.c:
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(gst_audio_filter_class_init):
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* gst-libs/gst/audio/gstaudiosink.c:
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(gst_audioringbuffer_class_init):
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* gst-libs/gst/audio/gstaudiosrc.c:
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(gst_audioringbuffer_class_init):
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* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init):
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* gst-libs/gst/interfaces/colorbalancechannel.c:
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(gst_color_balance_channel_class_init):
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* gst-libs/gst/interfaces/mixeroptions.c:
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(gst_mixer_options_class_init):
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* gst-libs/gst/interfaces/mixertrack.c:
|
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(gst_mixer_track_class_init):
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* gst-libs/gst/interfaces/tunerchannel.c:
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(gst_tuner_channel_class_init):
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* gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init):
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* gst-libs/gst/netbuffer/gstnetbuffer.c:
|
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(gst_netbuffer_class_init):
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* gst-libs/gst/rtp/gstbasertppayload.c:
|
|
(gst_basertppayload_class_init):
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|
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
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* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init):
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* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
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* gst/playback/gststreaminfo.c: (gst_stream_info_class_init):
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* gst/playback/gststreamselector.c:
|
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(gst_stream_selector_class_init):
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|
* gst/subparse/gstsubparse.c: (gst_sub_parse_class_init):
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* gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init):
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* sys/v4l/gstv4lcolorbalance.c:
|
|
(gst_v4l_color_balance_channel_class_init):
|
|
* sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init):
|
|
* sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init):
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* sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init):
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* sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init),
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(gst_v4l_tuner_norm_class_init):
|
|
* sys/ximage/ximagesink.c: (gst_ximagesink_class_init):
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init):
|
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* tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init):
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|
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
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|
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2006-04-08 18:09:17 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
|
|
Fix broken GObject macros
|
|
Original commit message from CVS:
|
|
* ext/pango/gsttextrender.h:
|
|
* gst-libs/gst/audio/gstaudiosink.h:
|
|
* gst-libs/gst/audio/gstaudiosrc.h:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.h:
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.h:
|
|
* gst-libs/gst/audio/gstringbuffer.h:
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.h:
|
|
* gst-libs/gst/rtp/gstbasertppayload.h:
|
|
* gst-libs/gst/video/gstvideofilter.h:
|
|
* gst-libs/gst/video/gstvideosink.h:
|
|
* gst/playback/gstplaybasebin.h:
|
|
* gst/tcp/gstmultifdsink.h:
|
|
* sys/v4l/gstv4lelement.h:
|
|
Fix broken GObject macros
|
|
|
|
2006-04-08 16:21:15 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
ext/alsa/gstalsasink.c: More debug to trace why my USB headset is not working with gst
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsasink.c: (set_hwparams), (alsasink_parse_spec):
|
|
More debug to trace why my USB headset is not working with gst
|
|
|
|
2006-04-07 17:18:11 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst/playback/gstplaybasebin.c: Clean up our group elements properly in the case where it never got committed - it sti...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (group_destroy):
|
|
Clean up our group elements properly in the case where it never
|
|
got committed - it still got added unconditionally to the bin.
|
|
|
|
2006-04-07 15:14:32 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/theora/theoradec.c: Unref unhandled events.
|
|
Original commit message from CVS:
|
|
* ext/theora/theoradec.c: (theora_dec_sink_event),
|
|
(theora_handle_data_packet), (theora_dec_chain):
|
|
Unref unhandled events.
|
|
Protect against empty buffers.
|
|
Perform QoS on running time.
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|
|
|
2006-04-07 13:24:54 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
ext/vorbis/vorbisenc.c: Remove leaks from vorbisenc.
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_set_header_on_caps),
|
|
(gst_vorbis_enc_chain):
|
|
Remove leaks from vorbisenc.
|
|
Mostly minor changes, the only significant one is that now the
|
|
buffers we set as 'streamheader' on the caps are copies of the
|
|
original buffers, to avoid circular refcounting problems.
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|
|
|
2006-04-07 09:51:35 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst/playback/gstplaybasebin.c: Don't remove our mute-probe if someone else already did so.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (mute_stream), (setup_substreams):
|
|
Don't remove our mute-probe if someone else already did so.
|
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Don't set a 2nd one if there is already one pending on the pad.
|
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* gst/playback/gstplaybin.c: (gst_play_bin_send_event_to_sink),
|
|
(do_playbin_seek):
|
|
When a seek fails, ensure that playbin is still set back to playing.
|
|
* gst/typefind/gsttypefindfunctions.c: (mpeg_ts_probe_headers),
|
|
(mpeg_ts_type_find), (plugin_init):
|
|
Add a typefind function for mpeg-ts streams.
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|
|
|
2006-04-06 11:40:45 +0000 Andy Wingo <wingo@pobox.com>
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|
|
|
* ChangeLog:
|
|
* gst/audiotestsrc/gstaudiotestsrc.c:
|
|
* gst/videorate/gstvideorate.c:
|
|
gst/videorate/gstvideorate.c (gst_video_rate_reset)
|
|
Original commit message from CVS:
|
|
2006-04-06 Andy Wingo <wingo@pobox.com>
|
|
* gst/videorate/gstvideorate.c (gst_video_rate_reset)
|
|
(gst_video_rate_init): Caps-related parameters should not be reset
|
|
by a flush -- move their inits to the instance init function.
|
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(gst_video_rate_flush_prev): Don't complain if gst_pad_push
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|
is not OK, just return the result.
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* gst/audiotestsrc/gstaudiotestsrc.c
|
|
(gst_audio_test_src_class_init)
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|
(gst_audio_test_src_get_times): Re-enable is-live=true, as was
|
|
broken by Stefan's commit on 24 March.
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|
|
|
2006-04-06 10:50:14 +0000 Andy Wingo <wingo@pobox.com>
|
|
|
|
ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on buffers being pushed out. Fixes oggmux ! multifdsink.
|
|
Original commit message from CVS:
|
|
2006-04-06 Andy Wingo <wingo@pobox.com>
|
|
* ext/ogg/gstoggmux.c (gst_ogg_mux_push_buffer): Set caps on
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buffers being pushed out. Fixes oggmux ! multifdsink.
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|
|
|
2006-04-05 13:05:25 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/vorbis/: Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make vorbisenc adhere to the official nomenclature; u...
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisdec.c: (gst_vorbis_dec_class_init),
|
|
(gst_vorbis_dec_init), (vorbis_dec_finalize):
|
|
* ext/vorbis/vorbisdec.h:
|
|
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_add_interfaces),
|
|
(gst_vorbis_enc_base_init), (gst_vorbis_enc_class_init),
|
|
(gst_vorbis_enc_sink_setcaps), (gst_vorbis_enc_convert_src),
|
|
(gst_vorbis_enc_convert_sink), (gst_vorbis_enc_get_query_types),
|
|
(gst_vorbis_enc_src_query), (gst_vorbis_enc_sink_query),
|
|
(gst_vorbis_enc_init), (gst_vorbis_enc_get_tag_value),
|
|
(gst_vorbis_enc_metadata_set1), (gst_vorbis_enc_set_metadata),
|
|
(gst_vorbis_enc_setup), (gst_vorbis_enc_clear),
|
|
(gst_vorbis_enc_buffer_from_packet),
|
|
(gst_vorbis_enc_buffer_from_header_packet),
|
|
(gst_vorbis_enc_push_buffer), (gst_vorbis_enc_push_packet),
|
|
(gst_vorbis_enc_set_header_on_caps), (gst_vorbis_enc_sink_event),
|
|
(gst_vorbis_enc_chain), (gst_vorbis_enc_output_buffers),
|
|
(gst_vorbis_enc_get_property), (gst_vorbis_enc_set_property),
|
|
(gst_vorbis_enc_change_state):
|
|
* ext/vorbis/vorbisenc.h:
|
|
Remove left-over 0.8 cruft; use GST_DEBUG_FUNCPTR; make
|
|
vorbisenc adhere to the official nomenclature; use boilerplate
|
|
macro.
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|
|
|
2006-04-04 11:20:24 +0000 Andy Wingo <wingo@pobox.com>
|
|
|
|
gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Whoops, fix bug introduced. Bad hacker!
|
|
Original commit message from CVS:
|
|
2006-04-04 Andy Wingo <wingo@pobox.com>
|
|
* gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
|
|
Whoops, fix bug introduced. Bad hacker!
|
|
|
|
2006-04-04 11:15:00 +0000 Andy Wingo <wingo@pobox.com>
|
|
|
|
gst/videorate/gstvideorate.c (gst_video_rate_flush_prev): Properly handle the case where you get EOS before any buffe...
|
|
Original commit message from CVS:
|
|
2006-04-04 Andy Wingo <wingo@pobox.com>
|
|
* gst/videorate/gstvideorate.c (gst_video_rate_flush_prev):
|
|
Properly handle the case where you get EOS before any buffers are
|
|
received. Use gst_buffer_make_metadata_writable where appropriate.
|
|
|
|
2006-04-04 10:16:46 +0000 Andy Wingo <wingo@pobox.com>
|
|
|
|
ext/theora/theoradec.c (theora_handle_data_packet): This value is often negative -- make it signed so as not to wrap ...
|
|
Original commit message from CVS:
|
|
2006-04-04 Andy Wingo <wingo@pobox.com>
|
|
* ext/theora/theoradec.c (theora_handle_data_packet): This value
|
|
is often negative -- make it signed so as not to wrap around.
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Fixes segfaults introduced on 9 March.
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|
|
|
2006-04-03 16:43:10 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/theora/: Don't try to store a gdouble in a gboolean.
|
|
Original commit message from CVS:
|
|
* ext/theora/gsttheoradec.h:
|
|
* ext/theora/theoradec.c: (theora_dec_src_event):
|
|
Don't try to store a gdouble in a gboolean.
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|
Small cleanups.
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|
|
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2006-04-03 12:55:18 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
ext/ogg/gstoggmux.c: Oggmux sucks.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads):
|
|
Oggmux sucks.
|
|
Make it suck slightly less by writing out the final page.
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|
Still can't encode a vorbis-in-ogg file correctly, though.
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|
|
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2006-04-03 08:49:06 +0000 Andy Wingo <wingo@pobox.com>
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|
|
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ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove a g_print.
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|
Original commit message from CVS:
|
|
2006-04-03 Andy Wingo <wingo@pobox.com>
|
|
* ext/theora/theoraparse.c (theora_parse_drain_queue): Um, remove
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a g_print.
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2006-04-03 08:32:21 +0000 Andy Wingo <wingo@pobox.com>
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ext/theora/theora.c (plugin_init): Register theoraparse.
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Original commit message from CVS:
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2006-04-03 Andy Wingo <wingo@pobox.com>
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* ext/theora/theora.c (plugin_init): Register theoraparse.
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* ext/theora/gsttheoraparse.h:
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* ext/theora/theoraparse.c: New files implementing a theora
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parser. Now we can properly remux ogg/theora+vorbis, yay.
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2006-04-03 08:28:58 +0000 Andy Wingo <wingo@pobox.com>
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ext/vorbis/vorbisparse.c: Add some docs and a copyright.
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Original commit message from CVS:
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2006-04-03 Andy Wingo <wingo@pobox.com>
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* ext/vorbis/vorbisparse.c: Add some docs and a copyright.
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2006-04-01 15:34:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* common:
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* configure.ac:
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don't use AS_LIBTOOL_TAGS, it doesn't work
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Original commit message from CVS:
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don't use AS_LIBTOOL_TAGS, it doesn't work
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2006-04-01 11:41:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* common:
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* ext/pango/gsttextoverlay.c:
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* sys/v4l/gstv4lsrc.c:
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remove BT8x8 from description, works for more devices
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Original commit message from CVS:
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remove BT8x8 from description, works for more devices
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2006-04-01 11:21:30 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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gst/audiotestsrc/gstaudiotestsrc.c: Fixed the sample pipeline (see #323798)
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Original commit message from CVS:
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* gst/audiotestsrc/gstaudiotestsrc.c:
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Fixed the sample pipeline (see #323798)
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2006-04-01 09:50:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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use AS_VERSION and AS_NANO more cleanups
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Original commit message from CVS:
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* configure.ac:
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* win32/common/config.h:
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* win32/common/config.h.in:
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use AS_VERSION and AS_NANO
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more cleanups
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2006-03-31 17:08:41 +0000 Andy Wingo <wingo@pobox.com>
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ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix uninitialized variable return that would happen.
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Original commit message from CVS:
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2006-03-31 Andy Wingo <wingo@pobox.com>
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* ext/vorbis/vorbisparse.c (vorbis_parse_sink_event): Fix
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uninitialized variable return that would happen.
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2006-03-31 16:57:47 +0000 Andy Wingo <wingo@pobox.com>
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ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix uninitialized variable return that would never happen.
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Original commit message from CVS:
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2006-03-31 Andy Wingo <wingo@pobox.com>
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* ext/vorbis/vorbisparse.c (vorbis_parse_drain_queue): Fix
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uninitialized variable return that would never happen.
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2006-03-31 16:43:43 +0000 Andy Wingo <wingo@pobox.com>
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ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
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Original commit message from CVS:
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2006-03-31 Andy Wingo <wingo@pobox.com>
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* ext/vorbis/vorbisparse.c (gst_vorbis_parse_init)
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(vorbis_parse_sink_event): Add an event function to flush our
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state on a seek, and to drain buffers on a premature EOS.
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(vorbis_parse_push_headers, vorbis_parse_clear_queue)
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(vorbis_parse_push_buffer, vorbis_parse_drain_queue_prematurely)
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(vorbis_parse_chain, vorbis_parse_queue_buffer)
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(vorbis_parse_drain_queue): Queue up buffers until we can set
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their timestamps and granulepos values.
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* ext/vorbis/vorbisparse.h: Include the vorbis decoder headers,
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and keep track of data needed for deriving granulepos and
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timestamps for buffers.
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2006-03-30 11:05:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
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* pkgconfig/gstreamer-plugins-base.pc.in:
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expose pluginsdir so gonlin can use it for tests
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Original commit message from CVS:
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expose pluginsdir so gonlin can use it for tests
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2006-03-30 10:03:56 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
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* pkgconfig/gstreamer-plugins-base.pc.in:
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add ccda to libraries
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Original commit message from CVS:
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add ccda to libraries
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2006-03-29 14:00:08 +0000 j^ <j@bootlab.org>
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better/unified long descriptions
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Original commit message from CVS:
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Patch by: j^ <j at bootlab dot org>
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* ext/alsa/gstalsamixerelement.c:
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(gst_alsa_mixer_element_class_init):
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* ext/alsa/gstalsasink.c:
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* ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init):
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* ext/ogg/gstoggdemux.c:
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* ext/ogg/gstoggmux.c:
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* ext/ogg/gstoggparse.c:
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* ext/pango/gstclockoverlay.c:
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* ext/pango/gsttextoverlay.c:
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* ext/pango/gsttextrender.c:
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* ext/pango/gsttimeoverlay.c:
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* ext/theora/theoradec.c:
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* ext/theora/theoraenc.c:
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* ext/vorbis/vorbisdec.c:
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* ext/vorbis/vorbisenc.c:
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* gst/audioconvert/gstaudioconvert.c:
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* gst/subparse/gstsubparse.c:
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* gst/tcp/gstmultifdsink.c:
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* gst/tcp/gsttcpclientsink.c:
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* gst/tcp/gsttcpclientsrc.c:
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* gst/tcp/gsttcpserversink.c:
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* gst/tcp/gsttcpserversrc.c:
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better/unified long descriptions
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Fixes #336477
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2006-03-29 13:54:24 +0000 Wim Taymans <wim.taymans@gmail.com>
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tests/examples/seek/seek.c: Don't let double and tripple clicks mess up our state.
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Original commit message from CVS:
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* tests/examples/seek/seek.c: (end_scrub), (seek_cb), (start_seek),
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(stop_seek):
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Don't let double and tripple clicks mess up our state.
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2006-03-28 13:13:43 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gstplaybin.c: Error out gracefully when we can't create any of the usual conversion elements for some re...
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Original commit message from CVS:
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* gst/playback/gstplaybin.c: (gen_video_element),
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(gen_text_element), (gen_audio_element), (gen_vis_element):
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Error out gracefully when we can't create any of the usual
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conversion elements for some reason. Also, don't try to
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create an audioscale (sic) element that's not used anyway.
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2006-03-28 10:21:52 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gstplaybasebin.c: Don't post RESOURCE_NOT_FOUND error when we can't find a source element for a particul...
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Original commit message from CVS:
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* gst/playback/gstplaybasebin.c: (setup_source):
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Don't post RESOURCE_NOT_FOUND error when we can't find a source
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element for a particular protocol, that's confusing for users.
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Instead, post a RESOURCE_FAILED error, so that our own error
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message is actually shown in totem etc. (#336303).
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2006-03-27 16:36:46 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
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ext/gnomevfs/gstgnomevfssrc.c: Fix some minor memory leaks (#336194).
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Original commit message from CVS:
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Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
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* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_finalize),
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(gst_gnome_vfs_src_get_icy_metadata):
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Fix some minor memory leaks (#336194).
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2006-03-27 16:15:00 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/gnomevfs/: Make gnomevfssink accept filenames as well as URIs for the "location" property, just like gnomevfssrc ...
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Original commit message from CVS:
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* ext/gnomevfs/gstgnomevfs.c:
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(gst_gnome_vfs_location_to_uri_string):
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* ext/gnomevfs/gstgnomevfs.h:
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* ext/gnomevfs/gstgnomevfssink.c:
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(gst_gnome_vfs_sink_set_property):
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* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_set_property):
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Make gnomevfssink accept filenames as well as URIs for the
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"location" property, just like gnomevfssrc does (and
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filesrc/filesink do) (#336190).
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2006-03-24 20:35:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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tests/check/generic/clock-selection.c: set to NULL before unreffing, fixes a valgrind leak.
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Original commit message from CVS:
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* tests/check/generic/clock-selection.c: (GST_START_TEST):
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set to NULL before unreffing, fixes a valgrind leak.
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Why was this not triggering the error that an object needs to
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be NULL before unreffing ?
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* win32/common/config.h:
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update
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2006-03-24 17:57:39 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/subparse/gstsubparse.*: Text subtitle files may or may not be UTF-8. If it's not, we don't really want to see '?'...
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Original commit message from CVS:
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* gst/subparse/gstsubparse.c: (convert_encoding),
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(gst_sub_parse_change_state):
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* gst/subparse/gstsubparse.h:
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Text subtitle files may or may not be UTF-8. If it's not, we
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don't really want to see '?' characters in place of non-ASCII
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characters like accented characters. So let's assume the input
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is UTF-8 until we come across text that is clearly not. If it's
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not UTF-8, we don't really know what it is, so try the following:
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(a) see whether the GST_SUBTITLE_ENCODING environment variable
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is set; if not, check (b) if the current locale encoding is
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non-UTF-8 and use that if it is, or (c) assume ISO-8859-15 if
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the current locale encoding is UTF-8 and the environment variable
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was not set to any particular encoding. Not perfect, but better
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than nothing (and better than before, I think) (fixes #172848).
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2006-03-24 17:39:45 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* docs/plugins/tmpl/.gitignore:
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* tests/check/libs/.gitignore:
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* tests/check/pipelines/.gitignore:
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* tests/examples/volume/.gitignore:
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ignore more
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|
Original commit message from CVS:
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ignore more
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|
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2006-03-24 17:26:54 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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configure.ac: update core requirement to 0.10.4.1 because of async_playback vmethod on GstBaseSink
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Original commit message from CVS:
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|
2006-03-24 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac:
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update core requirement to 0.10.4.1 because of async_playback
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vmethod on GstBaseSink
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2006-03-24 17:11:53 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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|
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use DEBUG_FUNCPTR for collectpads
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|
Original commit message from CVS:
|
|
* ext/ogg/gstoggmux.c: (gst_ogg_mux_init):
|
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* gst/adder/gstadder.c: (gst_adder_init):
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use DEBUG_FUNCPTR for collectpads
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2006-03-24 14:11:20 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* Makefile.am:
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don't go through check-torture if no check installed
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|
Original commit message from CVS:
|
|
don't go through check-torture if no check installed
|
|
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2006-03-24 10:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
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|
Original commit message from CVS:
|
|
* docs/plugins/Makefile.am:
|
|
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
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* ext/cdparanoia/gstcdparanoiasrc.c:
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* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init),
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(gst_gnome_vfs_sink_class_init):
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* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init):
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* ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init):
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* ext/ogg/gstoggmux.c:
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* ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init):
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* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init),
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(gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init):
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* ext/pango/gsttextoverlay.c:
|
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* ext/pango/gsttextrender.c:
|
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* ext/theora/theoradec.c:
|
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* ext/theora/theoraenc.c:
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* ext/vorbis/vorbisdec.c:
|
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* ext/vorbis/vorbisenc.c:
|
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* gst-libs/gst/audio/gstaudiofilter.c:
|
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(gst_audio_filter_base_init):
|
|
* gst-libs/gst/audio/gstaudiofiltertemplate.c:
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(gst_audio_filter_template_base_init):
|
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* gst/adder/gstadder.c: (gst_adder_get_type):
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* gst/adder/gstadder.h:
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* gst/audioconvert/gstaudioconvert.c:
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* gst/audiotestsrc/gstaudiotestsrc.c:
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(gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
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(gst_audio_test_src_create):
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* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
|
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* gst/playback/gstdecodebin.c:
|
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* gst/playback/gstplaybin.c:
|
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* gst/playback/gststreamselector.c:
|
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(gst_stream_selector_base_init):
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* gst/subparse/gstsubparse.c: (gst_sub_parse_base_init):
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* gst/volume/gstvolume.c:
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* sys/v4l/gstv4lmjpegsink.c:
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* sys/v4l/gstv4lmjpegsrc.c:
|
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* tests/check/libs/cddabasesrc.c:
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* tests/old/examples/gob/gst-identity2.gob:
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Add docs for adder, use GST_ELEMENT_DETAILS macro,
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define GstElementDetails at the top
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2006-03-23 21:48:18 +0000 Sébastien Moutte <sebastien@moutte.net>
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win32/common/libgstinterfaces.def: Add a lot of export functions for gst-python
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|
Original commit message from CVS:
|
|
* win32/common/libgstinterfaces.def:
|
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Add a lot of export functions for gst-python
|
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* win32/common/libgstinterfaces.dsp:
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Add a missing include folder in the project configuration
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2006-03-23 16:58:03 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
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gst-libs/gst/audio/gstbaseaudiosrc.c: Fix audio sources, forgot to make the ringbuffer startable...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
(gst_base_audio_src_get_time), (gst_base_audio_src_create),
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(gst_base_audio_src_change_state):
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Fix audio sources, forgot to make the ringbuffer
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startable...
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2006-03-23 16:29:58 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
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gst-libs/gst/audio/gstbaseaudiosrc.c: unparent instead of unref the ringbuffer.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
(gst_base_audio_src_get_time), (gst_base_audio_src_create),
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(gst_base_audio_src_change_state):
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unparent instead of unref the ringbuffer.
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2006-03-23 16:24:23 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
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gst-libs/gst/audio/gstbaseaudiosink.c: Implement new async_play vmethod to start slaving and allow playback start in ...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_class_init), (gst_base_audio_sink_async_play),
|
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(gst_base_audio_sink_do_play), (gst_base_audio_sink_change_state):
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Implement new async_play vmethod to start slaving and allow
|
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playback start in case of async PLAY state changes.
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* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
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Enable QoS with new method in base class.
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2006-03-23 11:18:19 +0000 Julien MOUTTE <julien@moutte.net>
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gst/videotestsrc/gstvideotestsrc.c: Partially handle 0 framerate, only EOS after the first frame is missing.
|
|
Original commit message from CVS:
|
|
Patch by: Julien MOUTTE <julien at moutte dot net>
|
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* gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_query),
|
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(gst_video_test_src_do_seek), (gst_video_test_src_create):
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Partially handle 0 framerate, only EOS after the first frame
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is missing.
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2006-03-23 09:38:59 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
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gst/: Patch for support of YVU9 AVI files (#334822)
|
|
Original commit message from CVS:
|
|
Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
|
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(gst_riff_create_video_template_caps):
|
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* gst/ffmpegcolorspace/avcodec.h:
|
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* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
|
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(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
|
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(gst_ffmpegcsp_avpicture_fill):
|
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* gst/ffmpegcolorspace/imgconvert.c:
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Patch for support of YVU9 AVI files (#334822)
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2006-03-22 15:29:25 +0000 Edward Hervey <bilboed@bilboed.com>
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|
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docs/design/design-decodebin.txt: Added design document for new decodebin text/x-pango-markup is also a default targe...
|
|
Original commit message from CVS:
|
|
* docs/design/design-decodebin.txt:
|
|
Added design document for new decodebin
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(Target Caps): text/x-pango-markup is also a default target caps.
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2006-03-22 15:11:47 +0000 Edward Hervey <bilboed@bilboed.com>
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docs/design/design-decodebin.txt: Added design document for new decodebin
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|
Original commit message from CVS:
|
|
* docs/design/design-decodebin.txt:
|
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Added design document for new decodebin
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|
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2006-03-22 12:33:09 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
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gst-libs/gst/audio/gstbaseaudiosink.c: Since we _parent the ringbuffer, we also need to _unparent instead of a plain ...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_dispose):
|
|
Since we _parent the ringbuffer, we also need to
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_unparent instead of a plain _unref.
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|
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2006-03-22 12:28:36 +0000 Wim Taymans <wim.taymans@gmail.com>
|
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|
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tests/examples/seek/seek.c: Add scrub checkbox.
|
|
Original commit message from CVS:
|
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* tests/examples/seek/seek.c: (end_scrub), (do_seek), (seek_cb),
|
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(stop_seek), (scrub_toggle_cb), (main):
|
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Add scrub checkbox.
|
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2006-03-21 17:47:04 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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ext/ogg/gstoggparse.c: Fix very inefficient usage of linked lists (#335365).
|
|
Original commit message from CVS:
|
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* ext/ogg/gstoggparse.c: (gst_ogg_parse_find_stream),
|
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(gst_ogg_parse_chain):
|
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Fix very inefficient usage of linked lists (#335365).
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|
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2006-03-21 14:26:01 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
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gcc 4.1 unreferenced pointer fixes.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
|
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* gst/playback/gstplaybin.c: (handoff):
|
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* gst/playback/gststreamselector.c:
|
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(gst_stream_selector_set_property):
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gcc 4.1 unreferenced pointer fixes.
|
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* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_put):
|
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* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_put):
|
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gst_buffer_ref() now takes a GstBuffer*.
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2006-03-20 18:09:41 +0000 Julien Moutte <julien@moutte.net>
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sys/xvimage/xvimagesink.c: Fix a memleak reported by Jan Schmidt.
|
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Original commit message from CVS:
|
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2006-03-20 Julien MOUTTE <julien@moutte.net>
|
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* sys/xvimage/xvimagesink.c:
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(gst_xvimagesink_get_format_from_caps): Fix a memleak reported
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by Jan Schmidt.
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2006-03-19 11:37:46 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/typefind/gsttypefindfunctions.c: Can't do tag preferences via probability, as tags would then lose against types ...
|
|
Original commit message from CVS:
|
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* gst/typefind/gsttypefindfunctions.c: (id3v2_type_find),
|
|
(id3v1_type_find), (apetag_type_find), (plugin_init):
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Can't do tag preferences via probability, as tags would then
|
|
lose against types that are recognised with MAXIMUM probability
|
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(like .wav); so let all tag typefinders return MAXIMUM themselves
|
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and order them via the rank. Split ID3v1 and ID3v2 typefinders so
|
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that we can prefer APE to ID3v1 (fixes #335028).
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2006-03-17 17:48:33 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst-libs/gst/audio/: Only start playback if we are playing. should fix #330748.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_change_state):
|
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* gst-libs/gst/audio/gstringbuffer.c: (wait_segment),
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(gst_ring_buffer_may_start):
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* gst-libs/gst/audio/gstringbuffer.h:
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Only start playback if we are playing.
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should fix #330748.
|
|
|
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2006-03-17 13:11:45 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
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Revert accidental commits to these files.
|
|
Original commit message from CVS:
|
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* ext/alsa/gstalsasink.c: (gst_alsasink_getcaps):
|
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* win32/common/config.h:
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Revert accidental commits to these files.
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2006-03-16 20:01:03 +0000 Michal Benes <michal.benes@xeris.cz>
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tests/Makefile.am: Don't try to build tests in tests/icles if we don't have X (#323852)
|
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Original commit message from CVS:
|
|
Patch by: Michal Benes <michal dot benes at xeris dot cz>
|
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* tests/Makefile.am:
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Don't try to build tests in tests/icles if we
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don't have X (#323852)
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|
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2006-03-16 13:08:01 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/tag/gstid3tag.c: Add TXXX frame identifiers for replaygain stuff as used by some taggers (see #323721).
|
|
Original commit message from CVS:
|
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* gst-libs/gst/tag/gstid3tag.c:
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Add TXXX frame identifiers for replaygain stuff as used
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by some taggers (see #323721).
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|
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2006-03-16 10:22:27 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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|
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gst/playback/gststreamselector.c: Preserve the existing buggy streamselector behaviour by performing a fallback buffe...
|
|
Original commit message from CVS:
|
|
* gst/playback/gststreamselector.c:
|
|
(gst_stream_selector_set_property),
|
|
(gst_stream_selector_bufferalloc):
|
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Preserve the existing buggy streamselector behaviour by performing
|
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a fallback buffer allocation when downstream isn't linked yet.
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This should really be fixed in playbin by blocking pads until it's
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linked them.
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Also, use gst_pad_alloc_buffer instead of
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gst_pad_alloc_buffer_and_set.
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|
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2006-03-15 22:40:08 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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gst-libs/gst/tag/gstid3tag.c: Don't crash on unknown ID3v2 TXXX frames.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/gstid3tag.c:
|
|
Don't crash on unknown ID3v2 TXXX frames.
|
|
|
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2006-03-15 17:59:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
ext/alsa/gstalsasink.c: Chain up to the parent finalize method.
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsasink.c: (gst_alsasink_finalise):
|
|
Chain up to the parent finalize method.
|
|
Add 32-bit sample size to the template caps.
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
|
|
(gst_riff_create_video_template_caps):
|
|
Add the fourcc that the VMWare codec uses.
|
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* gst/playback/gststreamselector.c:
|
|
(gst_stream_selector_set_property),
|
|
(gst_stream_selector_bufferalloc),
|
|
(gst_stream_selector_request_new_pad):
|
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For the active pad, forward buffer-alloc requests, otherwise
|
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return GST_FLOW_NOT_LINKED. This also prevents xvimagesink
|
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having to memcpy every frame when used by playbin.
|
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* gst/tcp/gstmultifdsink.c:
|
|
(gst_multi_fd_sink_handle_client_write):
|
|
Get negotiated caps from the sink pad, rather than the sink
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|
pad's peer.
|
|
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2006-03-15 17:11:34 +0000 Tommi Myöhänen <ext-tommi.myohanen@nokia.com>
|
|
|
|
ext/gnomevfs/gstgnomevfssrc.c: Don't forget to set src->callbacks_pushed to FALSE again when popping them, otherwise ...
|
|
Original commit message from CVS:
|
|
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
|
|
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_pop_callbacks):
|
|
Don't forget to set src->callbacks_pushed to FALSE again when
|
|
popping them, otherwise re-activation in a different mode won't
|
|
work (#334620).
|
|
|
|
2006-03-15 11:30:29 +0000 Sebastien Moutte <sebastien@moutte.net>
|
|
|
|
gst/ffmpegcolorspace/gstffmpegcodecmap.c: Replace __VA_ARGS__ caps creation macros with varargs functions. looks nice...
|
|
Original commit message from CVS:
|
|
Patch by: Sebastien Moutte <sebastien moutte net>
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.c: (gst_ff_vid_caps_new),
|
|
(gst_ff_aud_caps_new), (gst_ffmpeg_pixfmt_to_caps),
|
|
(gst_ffmpeg_smpfmt_to_caps):
|
|
Replace __VA_ARGS__ caps creation macros with varargs functions.
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|
Makes things compile on MSVC (#320765), looks nicer, and we can
|
|
tell the compiler to check for the NULL terminator.
|
|
|
|
2006-03-14 15:13:04 +0000 Fabrizio Gennari <fabrizio.ge@tiscali.it>
|
|
|
|
gst-libs/gst/riff/riff-media.c: Make sure the buffer we copy into is really always big enough, this time for real (#3...
|
|
Original commit message from CVS:
|
|
Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
|
|
Make sure the buffer we copy into is really always big
|
|
enough, this time for real (#333488).
|
|
|
|
2006-03-14 13:16:49 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/riff/riff-media.c: Add support for 24bpp DIB (#305279).
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
|
|
Add support for 24bpp DIB (#305279).
|
|
|
|
2006-03-14 11:11:59 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/: Re-enable QoS after the release.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_init):
|
|
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
|
|
* gst/ffmpegcolorspace/gstffmpegcolorspace.c: (gst_ffmpegcsp_init):
|
|
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
|
|
(gst_video_scale_init), (gst_video_scale_src_event):
|
|
Re-enable QoS after the release.
|
|
Rework videoscale to use the base class src_event handler.
|
|
|
|
2006-03-14 09:51:01 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
configure.ac: back to CVS.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
back to CVS.
|
|
|
|
=== release 0.10.5 ===
|
|
|
|
2006-03-13 19:50:04 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* ChangeLog:
|
|
* NEWS:
|
|
* RELEASE:
|
|
* configure.ac:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.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-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.xml:
|
|
* docs/plugins/inspect/plugin-libvisual.xml:
|
|
* docs/plugins/inspect/plugin-ogg.xml:
|
|
* docs/plugins/inspect/plugin-pango.xml:
|
|
* docs/plugins/inspect/plugin-playbin.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-video4linux.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:
|
|
* win32/common/config.h:
|
|
releasing 0.10.5
|
|
Original commit message from CVS:
|
|
releasing 0.10.5
|
|
|
|
2006-03-13 17:28:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/cs.po:
|
|
* po/en_GB.po:
|
|
* po/hu.po:
|
|
* po/it.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
Update .po files
|
|
Original commit message from CVS:
|
|
Update .po files
|
|
|
|
2006-03-13 11:17:19 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
docs/plugins/Makefile.am: Part of previous cdparanoiasrc docs fixes, forgot to commit.
|
|
Original commit message from CVS:
|
|
* docs/plugins/Makefile.am:
|
|
Part of previous cdparanoiasrc docs fixes, forgot to commit.
|
|
|
|
2006-03-12 14:56:31 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
docs/plugins/: Add cdparanoiasrc to docs.
|
|
Original commit message from CVS:
|
|
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
|
* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
|
Add cdparanoiasrc to docs.
|
|
* gst-libs/gst/cdda/gstcddabasesrc.c:
|
|
More GstCddaBaseSrc docs.
|
|
|
|
2006-03-12 13:47:22 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
Add new API to libgsttag: gst_tag_from_id3_user_tag().
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_user_tag):
|
|
* gst-libs/gst/tag/tag.h:
|
|
Add new API to libgsttag: gst_tag_from_id3_user_tag().
|
|
|
|
2006-03-11 19:47:16 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: NULL-terminate array of mpeg4 video file extensions.
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
|
|
NULL-terminate array of mpeg4 video file extensions.
|
|
Fixes crash on PPC (#334226).
|
|
|
|
2006-03-11 16:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/gnomevfs/gstgnomevfssrc.c: gnome_vfs_uri_is_local() alone is not a good indicator whether we can operate in pull-...
|
|
Original commit message from CVS:
|
|
* ext/gnomevfs/gstgnomevfssrc.c:
|
|
(gst_gnome_vfs_src_check_get_range):
|
|
gnome_vfs_uri_is_local() alone is not a good indicator
|
|
whether we can operate in pull-mode with a specific URI,
|
|
as it returns FALSE for file:// URIs that point to an
|
|
NFS-mounted path. Be more conservative here: whitelist
|
|
local files, blacklist http URIs and use the old
|
|
mechanism for anything else (fixes #334216).
|
|
|
|
2006-03-10 19:15:34 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
configure.ac: back to trunk
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
back to trunk
|
|
|
|
=== release 0.10.4 ===
|
|
|
|
2006-03-10 19:05:13 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* ChangeLog:
|
|
* NEWS:
|
|
* RELEASE:
|
|
* configure.ac:
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.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-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.xml:
|
|
* docs/plugins/inspect/plugin-libvisual.xml:
|
|
* docs/plugins/inspect/plugin-ogg.xml:
|
|
* docs/plugins/inspect/plugin-pango.xml:
|
|
* docs/plugins/inspect/plugin-playbin.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-video4linux.xml:
|
|
* docs/plugins/inspect/plugin-videorate.xml:
|
|
* docs/plugins/inspect/plugin-videoscale.xml:
|
|
* docs/plugins/inspect/plugin-videotestsrc.xml:
|
|
* docs/plugins/inspect/plugin-volume.xml:
|
|
* docs/plugins/inspect/plugin-vorbis.xml:
|
|
* docs/plugins/inspect/plugin-ximagesink.xml:
|
|
* docs/plugins/inspect/plugin-xvimagesink.xml:
|
|
* docs/upload.mak:
|
|
* win32/common/config.h:
|
|
releasing 0.10.4
|
|
Original commit message from CVS:
|
|
releasing 0.10.4
|
|
|
|
2006-03-10 12:37:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/video/gstvideosink.c: Disable max-lateness by setting it to -1 for now, so that we can bed QoS stuff in ...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
|
|
Disable max-lateness by setting it to -1 for now, so that
|
|
we can bed QoS stuff in thoroughly between now and the next
|
|
release.
|
|
|
|
2006-03-10 11:09:23 +0000 Fabrizio <fabrizio.ge@tiscali.it>
|
|
|
|
gst-libs/gst/riff/riff-media.c: Make sure we don't read beyond the palette buffer in case of
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
|
|
Make sure we don't read beyond the palette buffer in case of
|
|
broken or manipulated files (#333488, patch by: Fabrizio
|
|
Gennari)
|
|
|
|
2006-03-10 10:44:02 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Fix for variable not initialized.
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
|
|
Fix for variable not initialized.
|
|
|
|
2006-03-09 19:02:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* configure.ac:
|
|
* docs/libs/tmpl/gstringbuffer.sgml:
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/cs.po:
|
|
* po/en_GB.po:
|
|
* po/hu.po:
|
|
* po/it.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
* win32/common/config.h:
|
|
prereleasing
|
|
Original commit message from CVS:
|
|
prereleasing
|
|
|
|
2006-03-09 17:58:00 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/libvisual/visual.c: Small cleanups.
|
|
Original commit message from CVS:
|
|
* ext/libvisual/visual.c: (gst_visual_get_type),
|
|
(gst_visual_src_setcaps), (gst_vis_src_negotiate),
|
|
(gst_visual_chain):
|
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Small cleanups.
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* ext/theora/gsttheoradec.h:
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* ext/theora/theoradec.c: (gst_theora_dec_init),
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(gst_theora_dec_reset), (_theora_granule_time),
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(theora_dec_src_convert), (theora_dec_sink_convert),
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(theora_dec_src_query), (theora_dec_src_event),
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(theora_dec_sink_event), (theora_handle_comment_packet),
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(theora_handle_header_packet), (theora_dec_push),
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(theora_handle_data_packet), (theora_dec_chain),
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(theora_dec_change_state):
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Add simple QoS.
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2006-03-09 17:50:59 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
|
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ext/gnomevfs/gstgnomevfssrc.c: Some cleanups.
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Original commit message from CVS:
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|
* ext/gnomevfs/gstgnomevfssrc.c: (audiocast_init),
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(audiocast_register_listener), (gst_gnome_vfs_src_start):
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Some cleanups.
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2006-03-09 17:45:39 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/ogg/gstoggdemux.c: Don't try to activate NULL chains.
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Original commit message from CVS:
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* ext/ogg/gstoggdemux.c: (gst_ogg_demux_activate_chain):
|
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Don't try to activate NULL chains.
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2006-03-09 16:30:41 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/typefind/gsttypefindfunctions.c: Fix invalid memory access to region before peek'd data (#332964).
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Original commit message from CVS:
|
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* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset):
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Fix invalid memory access to region before peek'd data (#332964).
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2006-03-09 15:05:03 +0000 Christophe Fergeau <teuf@gnome.org>
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closes #333510.
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Original commit message from CVS:
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* ext/pango/gsttextoverlay.c: (gst_text_overlay_init):
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* ext/pango/gsttextrender.c: (gst_text_render_init):
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* gst/adder/gstadder.c: (gst_adder_init):
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Don't leak padtemplates, patch by Christophe Fergeau,
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closes #333510.
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2006-03-09 12:56:35 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/subparse/gstsubparse.c: Fix invalid memory access: make sure string passed to regexec() is NUL-termianted.
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Original commit message from CVS:
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* gst/subparse/gstsubparse.c: (gst_subparse_type_find):
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Fix invalid memory access: make sure string passed to
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regexec() is NUL-termianted.
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2006-03-09 12:37:59 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/typefind/gsttypefindfunctions.c: Refactor mpeg/audio typefinding to make it more maintainable and easier to fine-...
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Original commit message from CVS:
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* gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset),
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(mp3_type_find):
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Refactor mpeg/audio typefinding to make it more maintainable
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and easier to fine-tune. Make probing into middle of the file
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work properly (fixes #333900, also see #152688).
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2006-03-09 11:10:03 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/typefind/gsttypefindfunctions.c: Remove part from previous commit that was bogus: g_utf8_validate() does in fact ...
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Original commit message from CVS:
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* gst/typefind/gsttypefindfunctions.c:
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(utf8_type_find_have_valid_utf8_at_offset):
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Remove part from previous commit that was bogus:
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g_utf8_validate() does in fact not accept embedded
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zeroes, so we don't need to check for those (thanks
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to Mike for the hint).
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2006-03-08 17:11:29 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/typefind/gsttypefindfunctions.c: Make plain/text typefinder more conservative: firstly, check for embedded zeroes...
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|
Original commit message from CVS:
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* gst/typefind/gsttypefindfunctions.c:
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(utf8_type_find_count_embedded_zeroes),
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(utf8_type_find_have_valid_utf8_at_offset), (utf8_type_find):
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Make plain/text typefinder more conservative: firstly, check
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for embedded zeroes, which are perfectly valid UTF-8 characters,
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but also a fairly good sign that something is not a plain text
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file; secondly, probe into the middle of the file if possible.
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If we can't probe into the middle, limit the probability value
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to be returned to TYPE_FIND_POSSIBLE (see #333900).
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2006-03-08 11:34:45 +0000 Michael Smith <msmith@xiph.org>
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gst/typefind/gsttypefindfunctions.c: Make typefind function name for mpeg4 video unique.
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|
Original commit message from CVS:
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* gst/typefind/gsttypefindfunctions.c: (plugin_init):
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Make typefind function name for mpeg4 video unique.
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2006-03-08 09:53:31 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/libvisual/visual.c: Cleanups, post nice errors.
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|
Original commit message from CVS:
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* ext/libvisual/visual.c: (gst_visual_init),
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|
(gst_visual_clear_actors), (gst_visual_dispose),
|
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(gst_visual_reset), (gst_visual_src_setcaps),
|
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(gst_visual_sink_setcaps), (gst_vis_src_negotiate),
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(gst_visual_sink_event), (gst_visual_src_event), (get_buffer),
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(gst_visual_chain), (gst_visual_change_state):
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Cleanups, post nice errors.
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Handle sink and src events.
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Implement simple QoS.
|
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* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
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Use new basesink methods to configure max-lateness.
|
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Small doc update.
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* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
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(gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps):
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Debug statement cleanups.
|
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* gst/volume/gstvolume.c: (gst_volume_class_init):
|
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Simple cleanup.
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2006-03-08 09:50:23 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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ext/pango/gsttextoverlay.c: Revert API/ABI break from March 1. Keep 'halign' and 'valign' as string type properties, ...
|
|
Original commit message from CVS:
|
|
* ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init),
|
|
(gst_text_overlay_init), (gst_text_overlay_set_property),
|
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(gst_text_overlay_get_property):
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Revert API/ABI break from March 1. Keep 'halign' and 'valign'
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as string type properties, but mark them deprecated. Add
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'halignment' and 'valignment' properties that use enums
|
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instead of strings.
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|
|
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2006-03-08 09:37:12 +0000 Fabrizio <fabrizio.ge@tiscali.it>
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|
|
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gst-libs/gst/riff/riff-media.c: Allow palettes with less than 256 colours in AVI files
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
|
|
Allow palettes with less than 256 colours in AVI files
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|
(#333488, patch by: Fabrizio Gennari).
|
|
|
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2006-03-07 21:56:09 +0000 Julien Moutte <julien@moutte.net>
|
|
|
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ext/pango/gsttextoverlay.c: Fix wrong EOS handling on text pad. We were releasing the queued text buffer when we shou...
|
|
Original commit message from CVS:
|
|
2006-03-07 Julien MOUTTE <julien@moutte.net>
|
|
* ext/pango/gsttextoverlay.c: (gst_text_overlay_text_event),
|
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(gst_text_overlay_video_event): Fix wrong EOS handling on text
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pad. We were releasing the queued text buffer when we should keep
|
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it until video pad gets EOS or discard the text buffer because it's
|
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too old. That was eating the last subtitle buffer. Add some more
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debug.
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|
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2006-03-07 17:28:36 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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ext/pango/gsttextoverlay.c: Fix invalid memory access (we can't access a buffer after it's been pushed downstream wit...
|
|
Original commit message from CVS:
|
|
* ext/pango/gsttextoverlay.c: (gst_text_overlay_render_text),
|
|
(gst_text_overlay_video_chain):
|
|
Fix invalid memory access (we can't access a buffer after it's been
|
|
pushed downstream without taking a reference); fix memory leak (if
|
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there's no text to render, bail out before allocating stuff).
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|
|
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2006-03-07 15:08:15 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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ext/pango/gsttextoverlay.*: If input is plain text, escape it before passing it to pango_layout_set_markup().
|
|
Original commit message from CVS:
|
|
* ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
|
|
(gst_text_overlay_setcaps_txt), (gst_text_overlay_video_chain):
|
|
* ext/pango/gsttextoverlay.h:
|
|
If input is plain text, escape it before passing it to
|
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pango_layout_set_markup().
|
|
|
|
2006-03-07 13:01:21 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/audio/gstaudiofilter.c: Don't ignore flow return from gst_pad_push().
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_chain):
|
|
Don't ignore flow return from gst_pad_push().
|
|
|
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2006-03-07 12:49:03 +0000 Christophe Fergeau <teuf@gnome.org>
|
|
|
|
Don't leak references returned by gst_pad_get_parent()
|
|
Original commit message from CVS:
|
|
* ext/libvisual/visual.c: (gst_visual_getcaps),
|
|
(gst_visual_src_setcaps), (gst_visual_sink_setcaps):
|
|
* ext/ogg/gstoggmux.c: (gst_ogg_mux_sinkconnect):
|
|
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
|
|
(gst_vorbisenc_convert_sink):
|
|
* gst-libs/gst/audio/audio.c: (gst_audio_frame_byte_size),
|
|
(gst_audio_duration_from_pad_buffer):
|
|
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_link),
|
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(gst_audio_filter_chain):
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_setcaps):
|
|
* gst-libs/gst/video/video.c: (gst_video_frame_rate),
|
|
(gst_video_get_size):
|
|
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
|
|
Don't leak references returned by gst_pad_get_parent()
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|
(#333663, based on patch by: Christophe Fergeau).
|
|
|
|
2006-03-06 20:52:25 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
ext/gnomevfs/gstgnomevfssink.c: change location param details
|
|
Original commit message from CVS:
|
|
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
|
|
change location param details
|
|
* gst/volume/gstvolume.c: (plugin_init):
|
|
correct plugin description
|
|
|
|
2006-03-06 20:07:55 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/gnomevfs/gstgnomevfssrc.c: Override GstBaseSrc::check_get_range() in order to avoid opening the resource just to ...
|
|
Original commit message from CVS:
|
|
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init),
|
|
(gst_gnome_vfs_src_check_get_range):
|
|
Override GstBaseSrc::check_get_range() in order to avoid opening
|
|
the resource just to check whether we can operate in pull-mode or
|
|
not - we can predict that pretty well from the URI alone. Should
|
|
fix problems with last.fm (#331690). (Requires latest core CVS).
|
|
|
|
2006-03-06 16:18:51 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/video/gstvideosink.c: Throw away frames that are later than 20 ms.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init),
|
|
(gst_video_sink_class_init):
|
|
Throw away frames that are later than 20 ms.
|
|
|
|
2006-03-06 14:14:47 +0000 Fabrizio <fabrizio.ge@tiscali.it>
|
|
|
|
gst-libs/gst/riff/riff-media.c:
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
|
|
Set depth on WMA caps (#333545, patch by: Fabrizio Gennari).
|
|
|
|
2006-03-05 23:39:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
ext/ogg/gstoggmux.c: put Theora BOS pages before others. This hardcodes the Ogg/Theora I profile, but hey.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
|
|
(gst_ogg_mux_send_headers), (gst_ogg_mux_collected):
|
|
put Theora BOS pages before others. This hardcodes
|
|
the Ogg/Theora I profile, but hey.
|
|
|
|
2006-03-05 23:06:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* ext/ogg/gstoggmux.c:
|
|
changed more than 5 lines
|
|
Original commit message from CVS:
|
|
changed more than 5 lines
|
|
|
|
2006-03-05 22:57:58 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
ogg muxing of vorbis and theora now has pages ordered correctly again, even with delays.
|
|
Original commit message from CVS:
|
|
ogg muxing of vorbis and theora now has pages ordered correctly again,
|
|
even with delays.
|
|
* ext/ogg/README:
|
|
updated with some examples
|
|
* ext/theora/theoraenc.c: (granulepos_to_timestamp),
|
|
(granulepos_add), (theora_buffer_from_packet):
|
|
* ext/vorbis/vorbisenc.c: (granulepos_to_timestamp_offset),
|
|
(granulepos_to_timestamp), (gst_vorbisenc_buffer_from_packet),
|
|
(gst_vorbisenc_chain):
|
|
implement strategy from ext/ogg/README
|
|
* ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
|
|
(gst_ogg_mux_push_buffer), (gst_ogg_mux_dequeue_page),
|
|
(gst_ogg_mux_pad_queue_page), (gst_ogg_mux_compare_pads),
|
|
(gst_ogg_mux_queue_pads), (gst_ogg_mux_collected):
|
|
Fix muxer so that oggz-validate is happy with all streams;
|
|
except for no eos mark, and the BOS page ordering
|
|
* tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
|
|
(check_buffer_granulepos):
|
|
* tests/check/pipelines/vorbisenc.c: (check_buffer_granulepos):
|
|
update tests to check for OFFSET being set as requested
|
|
fixed type of granulepos, it's not a ClockTime
|
|
|
|
2006-03-05 21:34:23 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
sys/xvimage/xvimagesink.c: Check that the xvimage we are creating has a correct size before returning it. (#3...
|
|
Original commit message from CVS:
|
|
2006-03-05 Julien MOUTTE <julien@moutte.net>
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
|
|
(gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
|
|
Check that the xvimage we are creating has a correct size before returning it. (#314897)
|
|
|
|
2006-03-05 13:44:05 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Give id3 and ape tag typefinders a rank slightly higher than PRIMARY to ensure t...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
|
|
Give id3 and ape tag typefinders a rank slightly higher
|
|
than PRIMARY to ensure they're always run before any of
|
|
the other typefinders (in particular wav and mp3) (#324186).
|
|
|
|
2006-03-05 13:08:37 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/riff/riff-media.c: Add support for '3IVD' fourcc (#333403).
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
|
|
Add support for '3IVD' fourcc (#333403).
|
|
|
|
2006-03-04 14:35:10 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
configure.ac: Bump requirements to GStreamer CVS for the new error enum.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Bump requirements to GStreamer CVS for the new error enum.
|
|
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_render):
|
|
Emit new GST_RESOURCE_ERROR_NO_SPACE_LEFT when there's no
|
|
space left on the device (fixes #333352).
|
|
|
|
2006-03-03 23:53:50 +0000 Sébastien Moutte <sebastien@moutte.net>
|
|
|
|
win32/vs6: add a project file for libgstvolume update the workspace
|
|
Original commit message from CVS:
|
|
* win32/vs6:
|
|
add a project file for libgstvolume
|
|
update the workspace
|
|
|
|
2006-03-03 15:26:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* ext/ogg/README:
|
|
* ext/ogg/gstoggmux.c:
|
|
debug updates
|
|
Original commit message from CVS:
|
|
debug updates
|
|
|
|
2006-03-03 15:22:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
|
|
Original commit message from CVS:
|
|
2006-03-03 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
* ext/theora/theoraenc.c: (theora_set_header_on_caps):
|
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* tests/check/pipelines/theoraenc.c: (check_buffer_is_header),
|
|
(GST_START_TEST):
|
|
Fix for http://bugzilla.gnome.org/show_bug.cgi?id=333254
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|
Set IN_CAPS on header buffers
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|
|
|
2006-03-02 18:23:55 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
docs/plugins/: Add audioresample to docs.
|
|
Original commit message from CVS:
|
|
* docs/plugins/Makefile.am:
|
|
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
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Add audioresample to docs.
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|
* gst/audioconvert/gstaudioconvert.c:
|
|
Add revision date.
|
|
* gst/audioresample/gstaudioresample.c:
|
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(gst_audioresample_base_init), (gst_audioresample_class_init),
|
|
(gst_audioresample_init), (gst_audioresample_dispose),
|
|
(audioresample_get_unit_size), (audioresample_transform_caps),
|
|
(resample_set_state_from_caps), (audioresample_transform_size),
|
|
(audioresample_set_caps), (audioresample_event),
|
|
(audioresample_do_output), (audioresample_transform),
|
|
(audioresample_pushthrough), (gst_audioresample_set_property),
|
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(gst_audioresample_get_property), (plugin_init):
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* gst/audioresample/gstaudioresample.h:
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Added docs.
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Small code cleanups.
|
|
|
|
2006-03-02 18:12:33 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
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* gst/videorate/Makefile.am:
|
|
fix wim's commit
|
|
Original commit message from CVS:
|
|
fix wim's commit
|
|
|
|
2006-03-02 17:48:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* ext/ogg/gstoggmux.c:
|
|
debug using the actual GstPad, that allows us to see the serialno in the padname
|
|
Original commit message from CVS:
|
|
debug using the actual GstPad, that allows us to see the serialno in the padname
|
|
|
|
2006-03-02 17:46:36 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
docs/plugins/: Added videoscale to docs.
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Original commit message from CVS:
|
|
* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-base-plugins-docs.sgml:
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* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
|
Added videoscale to docs.
|
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* gst/videorate/gstvideorate.c: (gst_video_rate_flush_prev),
|
|
(gst_video_rate_swap_prev), (gst_video_rate_event),
|
|
(gst_video_rate_chain):
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Fix typo in docs.
|
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* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
|
|
(gst_video_scale_init), (gst_video_scale_prepare_size),
|
|
(gst_video_scale_set_caps), (gst_video_scale_get_unit_size),
|
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(gst_video_scale_fixate_caps), (gst_video_scale_transform):
|
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* gst/videoscale/gstvideoscale.h:
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|
Added docs, examples.
|
|
Some code cleanups.
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|
Post errors instead of g_warning.
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2006-03-02 17:30:57 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* ext/ogg/gstoggmux.c:
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clean up debug messages
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|
Original commit message from CVS:
|
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clean up debug messages
|
|
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2006-03-02 17:15:38 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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|
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* ext/ogg/gstoggmux.c:
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extra debugging from older version, makes it easier to compare
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|
Original commit message from CVS:
|
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extra debugging from older version, makes it easier to compare
|
|
|
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2006-03-02 17:04:55 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* ext/ogg/gstoggmux.c:
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some space cleanup and debug fixes
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|
Original commit message from CVS:
|
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some space cleanup and debug fixes
|
|
|
|
2006-03-02 16:47:34 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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docs/: Added some more docs to libs and plugins.
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|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-docs.sgml:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* docs/libs/gst-plugins-base-libs.types:
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-base-plugins-docs.sgml:
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* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
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Added some more docs to libs and plugins.
|
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* gst-libs/gst/audio/gstringbuffer.c:
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(gst_ring_buffer_prepare_read), (gst_ring_buffer_clear):
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* gst-libs/gst/audio/gstringbuffer.h:
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Document ringbuffer some more.
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* gst/videorate/gstvideorate.c: (gst_video_rate_class_init),
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(gst_video_rate_setcaps), (gst_video_rate_reset),
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(gst_video_rate_init), (gst_video_rate_flush_prev),
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(gst_video_rate_swap_prev), (gst_video_rate_event),
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(gst_video_rate_chain), (gst_video_rate_change_state):
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* gst/videorate/gstvideorate.h:
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Fix videorate to use segments.
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Make it work with 0/1 framerates (closes #331903)
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Handle EOS correctly.
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Added docs.
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2006-03-02 13:13:00 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/ogg/gstogmparse.c: In state change function, first chain up to parent class, then handle downwards state change s...
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|
Original commit message from CVS:
|
|
* ext/ogg/gstogmparse.c: (gst_ogm_parse_class_init),
|
|
(gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
|
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(gst_ogm_text_parse_init), (gst_ogm_parse_change_state):
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In state change function, first chain up to parent class,
|
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then handle downwards state change stuff. Remove some
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commented out cruft from 0.8 code.
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2006-03-02 12:35:59 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/ogg/gstogmparse.c: Don't remove/re-add source pad if the new caps are the same as the old caps anyway (#333042). ...
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_init),
|
|
(gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
|
|
(gst_ogm_parse_sink_convert), (gst_ogm_parse_sink_query),
|
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(gst_ogm_parse_chain):
|
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Don't remove/re-add source pad if the new caps are the same as
|
|
the old caps anyway (#333042). When removing source pad, don't
|
|
unref it afterwards - we didn't ref it when adding. Sprinkle some
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GST_DEBUG_FUNCPTR goodness here and there. Don't leak references
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after using gst_pad_get_parent(). Return downstream flow return
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value in chain function.
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2006-03-02 11:28:23 +0000 Wim Taymans <wim.taymans@gmail.com>
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docs/plugins/: Fix hierarchy, added some more elements to the docs.
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|
Original commit message from CVS:
|
|
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/gst-plugins-base-plugins.hierarchy:
|
|
* docs/plugins/gst-plugins-base-plugins.interfaces:
|
|
* docs/plugins/gst-plugins-base-plugins.signals:
|
|
Fix hierarchy, added some more elements to the docs.
|
|
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
|
|
(gst_ffmpegcsp_get_type):
|
|
* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
|
|
Fix docs for ffmpegcolorspace.
|
|
|
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2006-03-01 19:24:44 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
|
gst/typefind/gsttypefindfunctions.c: Some typefinding fine-tuning:
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (id3_type_find),
|
|
(apetag_type_find), (ape_type_find), (plugin_init):
|
|
Some typefinding fine-tuning:
|
|
- rank ID3/APE tags in order of preference via probabilities, so that
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ID3v2 > APEv2 > APEv1 > ID3v1.
|
|
- three or four bytes don't really justify MAXIMUM probability,
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change those to 'very likely' (musepack and monkeysaudio).
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|
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2006-03-01 18:25:18 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
Added alsa docs.
|
|
Original commit message from CVS:
|
|
* docs/plugins/Makefile.am:
|
|
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
|
* ext/alsa/gstalsamixer.c:
|
|
* ext/alsa/gstalsamixer.h:
|
|
* ext/alsa/gstalsamixerelement.c:
|
|
(gst_alsa_mixer_element_class_init), (gst_alsa_mixer_element_init):
|
|
* ext/alsa/gstalsamixerelement.h:
|
|
* ext/alsa/gstalsasink.c:
|
|
* ext/alsa/gstalsasink.h:
|
|
* ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init),
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|
(gst_alsasrc_init):
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|
* ext/alsa/gstalsasrc.h:
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|
Added alsa docs.
|
|
Small code cleanups.
|
|
|
|
2006-03-01 17:52:45 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/theora/Makefile.am: Dist new header too,
|
|
Original commit message from CVS:
|
|
* ext/theora/Makefile.am:
|
|
Dist new header too,
|
|
|
|
2006-03-01 17:39:28 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
Fix some more docs.
|
|
Original commit message from CVS:
|
|
* docs/plugins/Makefile.am:
|
|
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
|
* ext/gnomevfs/gstgnomevfssink.h:
|
|
* ext/gnomevfs/gstgnomevfssrc.h:
|
|
* ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet):
|
|
* ext/vorbis/vorbisdec.h:
|
|
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_sink):
|
|
* ext/vorbis/vorbisenc.h:
|
|
* ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps),
|
|
(vorbis_parse_chain), (vorbis_parse_change_state):
|
|
* ext/vorbis/vorbisparse.h:
|
|
* gst/audioconvert/gstaudioconvert.h:
|
|
* gst/tcp/gsttcpserversink.h:
|
|
* gst/videotestsrc/gstvideotestsrc.c:
|
|
* gst/videotestsrc/gstvideotestsrc.h:
|
|
* gst/volume/gstvolume.c:
|
|
* gst/volume/gstvolume.h:
|
|
Fix some more docs.
|
|
Added docs for vorbisdec and vorbisparse.
|
|
Fix vorbisparse.
|
|
|
|
2006-03-01 16:24:37 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
Updated/added documentation.
|
|
Original commit message from CVS:
|
|
* docs/plugins/Makefile.am:
|
|
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
|
|
* docs/plugins/gst-plugins-base-plugins-sections.txt:
|
|
* ext/pango/gstclockoverlay.h:
|
|
* ext/pango/gsttextoverlay.h:
|
|
* ext/pango/gsttextrender.h:
|
|
* ext/pango/gsttimeoverlay.h:
|
|
* ext/theora/gsttheoradec.h:
|
|
* ext/theora/gsttheoraenc.h:
|
|
* ext/theora/theoradec.c:
|
|
* ext/theora/theoraenc.c:
|
|
* gst/audioconvert/gstaudioconvert.h:
|
|
* gst/audiotestsrc/gstaudiotestsrc.h:
|
|
* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
|
|
* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init):
|
|
* gst/tcp/gstmultifdsink.h:
|
|
Updated/added documentation.
|
|
* ext/pango/gsttextoverlay.c: (gst_text_overlay_valign_get_type),
|
|
(gst_text_overlay_halign_get_type),
|
|
(gst_text_overlay_wrap_mode_get_type),
|
|
(gst_text_overlay_base_init), (gst_text_overlay_class_init),
|
|
(gst_text_overlay_init), (gst_text_overlay_set_property),
|
|
(gst_text_overlay_get_property):
|
|
Fix up properties to be enums instead of string to make bindings,
|
|
introspection and automatic GUI creation possible.
|
|
Add getters for the properties.
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|
|
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2006-02-28 21:21:07 +0000 Sébastien Moutte <sebastien@moutte.net>
|
|
|
|
gst/audiotestsrc/gstaudiotestsrc.c: added defines of M_PI and M_PI_2
|
|
Original commit message from CVS:
|
|
* gst/audiotestsrc/gstaudiotestsrc.c:
|
|
added defines of M_PI and M_PI_2
|
|
* gst/ffmpegcolorspace/avcodec.h:
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|
removed #include "stdint.h" for win32 as _stdint.h is
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|
autogenerated to win32/common
|
|
* win32/common/libgstaudio.def:
|
|
* win32/common/libgsttag.def:
|
|
added some exports
|
|
* win32/vs6:
|
|
some project files bugs corrected
|
|
* win32/vs7:
|
|
project files are reset to the default vs7 configuration
|
|
(they link to msvcr71.dll using default optimizations)
|
|
|
|
2006-02-28 19:08:12 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/gnomevfs/gstgnomevfssink.c: Fix some docs.
|
|
Original commit message from CVS:
|
|
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init):
|
|
Fix some docs.
|
|
|
|
2006-02-28 13:52:04 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
ext/alsa/gstalsasrc.c: Set proper class on the ElementDetails:
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsasrc.c:
|
|
Set proper class on the ElementDetails:
|
|
Source/Audio instead of Src/Audio
|
|
|
|
2006-02-28 12:19:11 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
gst/videoscale/vs_scanline.c: Revert optimization in videoscale. It should go in liboil and have an appropriate liboi...
|
|
Original commit message from CVS:
|
|
* gst/videoscale/vs_scanline.c:
|
|
(vs_scanline_resample_nearest_RGBA):
|
|
Revert optimization in videoscale. It should go in liboil and have
|
|
an appropriate liboil function.
|
|
|
|
2006-02-28 11:06:24 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock in the NULL state.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_provide_clock):
|
|
Don't try to provide a clock in the NULL state.
|
|
|
|
2006-02-28 11:04:47 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/ogg/gstoggdemux.c: Use GstSegment infrastructure to remove duplicated code and handle more seek cases correctly.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_receive_event),
|
|
(gst_ogg_pad_event), (gst_ogg_pad_internal_chain),
|
|
(gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
|
|
(gst_ogg_demux_deactivate_current_chain),
|
|
(gst_ogg_demux_activate_chain), (gst_ogg_demux_do_seek),
|
|
(gst_ogg_demux_perform_seek), (gst_ogg_demux_collect_info),
|
|
(gst_ogg_demux_find_chains), (gst_ogg_demux_chain),
|
|
(gst_ogg_demux_loop), (gst_ogg_demux_change_state):
|
|
Use GstSegment infrastructure to remove duplicated code
|
|
and handle more seek cases correctly.
|
|
|
|
2006-02-28 10:39:19 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/ffmpegcolorspace/gstffmpegcolorspace.c: Don't ignore return code from ffmpeg convert function.
|
|
Original commit message from CVS:
|
|
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
|
|
(gst_ffmpegcsp_transform):
|
|
Don't ignore return code from ffmpeg convert function.
|
|
* gst/ffmpegcolorspace/imgconvert.c: (img_convert):
|
|
Split out some long statements to ease debugging.
|
|
|
|
2006-02-27 12:08:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
ext/libvisual/visual.c: Don't use gst_pad_use_fixed_caps, because it prevents downstream from being able to renegotia...
|
|
Original commit message from CVS:
|
|
* ext/libvisual/visual.c: (gst_visual_init),
|
|
(gst_vis_src_negotiate), (get_buffer), (plugin_init):
|
|
Don't use gst_pad_use_fixed_caps, because it prevents downstream from
|
|
being able to renegotiate the size. Instead, use the negotiation
|
|
algorithm from the goom plugin to pick an initial output caps.
|
|
Also, allow theoretical libvisual plugins that might support non-GL
|
|
output even if they also do GL.
|
|
|
|
2006-02-26 21:05:46 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
ext/libvisual/visual.c: Load only non GL plugins. Fix some memleaks and possible negotiation issues.
|
|
Original commit message from CVS:
|
|
2006-02-26 Julien MOUTTE <julien@moutte.net>
|
|
* ext/libvisual/visual.c: (gst_visual_init),
|
|
(gst_visual_src_setcaps), (get_buffer), (gst_visual_chain),
|
|
(plugin_init): Load only non GL plugins. Fix some memleaks and
|
|
possible negotiation issues.
|
|
|
|
2006-02-24 23:19:44 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
gst-libs/gst/tag/tag.h: Adding Annodex tags here.
|
|
Original commit message from CVS:
|
|
2006-02-25 Julien MOUTTE <julien@moutte.net>
|
|
* gst-libs/gst/tag/tag.h: Adding Annodex tags here.
|
|
|
|
2006-02-24 18:55:27 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Fix CMML type find function to not require a specific minor version of the CMML ...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (mpeg4_video_type_find),
|
|
(cmml_type_find), (plugin_init):
|
|
Fix CMML type find function to not require a specific minor version
|
|
of the CMML header.
|
|
Add an MPEG4 video elementary stream typefind function.
|
|
|
|
2006-02-24 17:31:53 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
ext/ogg/gstoggdemux.c: Annodex support in ogg demuxer. Doesn't do very much without the other annodex patches (to come).
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_parse_skeleton_fishead),
|
|
(gst_ogg_pad_parse_skeleton_fisbone), (gst_ogg_pad_query_convert),
|
|
(gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet),
|
|
(gst_ogg_demux_perform_seek), (gst_ogg_demux_read_chain),
|
|
(gst_ogg_demux_read_end_chain), (gst_ogg_demux_collect_chain_info),
|
|
(gst_ogg_demux_change_state), (gst_annodex_granule_to_time):
|
|
Annodex support in ogg demuxer. Doesn't do very much without the
|
|
other annodex patches (to come).
|
|
|
|
2006-02-24 16:21:34 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/riff/riff-media.c:
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
|
|
Pick up palette for MS video v1 (#327028, patch by:
|
|
Fabrizio Gennari <fabrizio dot get at tiscali dot it>)
|
|
|
|
2006-02-24 13:54:04 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/ffmpegcolorspace/gstffmpegcolorspace.c: The 'palette_data' field from incoming RGB caps shouldn't be proxied on o...
|
|
Original commit message from CVS:
|
|
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
|
|
(gst_ffmpegcsp_caps_remove_format_info),
|
|
(gst_ffmpegcsp_get_unit_size):
|
|
The 'palette_data' field from incoming RGB caps shouldn't be
|
|
proxied on outgoing YUV caps; also, restrict unit size
|
|
adjustment in case of paletted data only to the unit that
|
|
actually has a palette. Fixes #330711.
|
|
|
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2006-02-24 12:18:14 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/ffmpegcolorspace/gstffmpegcolorspace.c: Plug some memory leaks.
|
|
Original commit message from CVS:
|
|
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
|
|
(gst_ffmpegcsp_transform_caps), (gst_ffmpegcsp_set_caps),
|
|
(gst_ffmpegcsp_finalize), (gst_ffmpegcsp_class_init),
|
|
(gst_ffmpegcsp_get_unit_size):
|
|
Plug some memory leaks.
|
|
|
|
2006-02-24 10:18:52 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
sys/: Add some _CFLAGS and _LIBS that seem to be missing and/or required for Cygwin (see #317048).
|
|
Original commit message from CVS:
|
|
* sys/ximage/Makefile.am:
|
|
* sys/xvimage/Makefile.am:
|
|
Add some _CFLAGS and _LIBS that seem to be missing
|
|
and/or required for Cygwin (see #317048).
|
|
|
|
2006-02-24 00:07:18 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
* ChangeLog:
|
|
ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
|
|
Original commit message from CVS:
|
|
ChangeLog surgery: use UTF-8 encoding in stead of ISO-8859-15
|
|
|
|
2006-02-22 18:46:46 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
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ext/alsa/gstalsasrc.c: Fix description as pointed out by caugier.
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Original commit message from CVS:
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* ext/alsa/gstalsasrc.c:
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Fix description as pointed out by caugier.
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2006-02-22 10:29:22 +0000 Edward Hervey <bilboed@bilboed.com>
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gst/typefind/gsttypefindfunctions.c: Better 3gp typefinding.
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Original commit message from CVS:
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Reviewed by : Edward Hervey <edward@fluendo.com>
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* gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
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(qt_type_find):
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Better 3gp typefinding.
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2006-02-21 12:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/gnomevfs/gstgnomevfssrc.c: Don't send EOS event here, the base class will send one for us.
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Original commit message from CVS:
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* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
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Don't send EOS event here, the base class will send one for us.
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* gst/playback/gstplaybasebin.c: (prepare_output):
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Subpictures without video stream aren't allowed either.
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* gst/subparse/gstsubparse.c: (gst_subparse_type_find):
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Fix debug statement copy'n'paste-o.
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2006-02-21 12:05:18 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/alsa/gstalsamixer.c: Fix issues with mixer keeping state when muting/unmuting and when changing the volume whilst...
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Original commit message from CVS:
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* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_get_volume):
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Fix issues with mixer keeping state when muting/unmuting
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and when changing the volume whilst muted (see #331763
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and #331765).
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2006-02-20 18:27:06 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/subparse/gstsubparse.c: Set right caps given that we send escaped text. Also, honour <i></i>, <b></b> and <u></u>...
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Original commit message from CVS:
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* gst/subparse/gstsubparse.c: (subrip_unescape_formatting),
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(parse_subrip), (gst_sub_parse_format_autodetect):
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Set right caps given that we send escaped text. Also,
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honour <i></i>, <b></b> and <u></u> markers that can be found
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in .srt files (fixes #310202).
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2006-02-20 16:21:14 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/audio/mixerutils.c: Make order in which elements are tried more determinable.
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Original commit message from CVS:
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* gst-libs/gst/audio/mixerutils.c:
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(element_factory_rank_compare_func):
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Make order in which elements are tried more determinable.
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2006-02-20 15:57:51 +0000 Julien Moutte <julien@moutte.net>
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gst/playback/gstdecodebin.c: Make decodebin reusable by fixing remove_element_chain first and then introduce a cleane...
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Original commit message from CVS:
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* gst/playback/gstdecodebin.c: (get_our_ghost_pad),
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(remove_element_chain), (cleanup_decodebin),
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(gst_decode_bin_change_state): Make decodebin reusable by
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fixing remove_element_chain first and then introduce a
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cleaner in state change to ->NULL. (Closes #331678)
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------------------------------------------------------
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2006-02-19 14:32:35 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/gnomevfs/gstgnomevfssink.c: use 0666 mask when creating files so umask gets applied correctly. Fixes #331295.
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Original commit message from CVS:
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* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_open_file):
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use 0666 mask when creating files so umask gets applied
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correctly. Fixes #331295.
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2006-02-19 14:16:16 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/subparse/: Add very basic parser for SSA subtitle streams (as often found in matroska files).
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Original commit message from CVS:
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* gst/subparse/Makefile.am:
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* gst/subparse/gstssaparse.c: (gst_ssa_parse_base_init),
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(gst_ssa_parse_dispose), (gst_ssa_parse_init),
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(gst_ssa_parse_class_init), (gst_ssa_parse_src_event),
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(gst_ssa_parse_sink_event), (gst_ssa_parse_setcaps),
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(gst_ssa_parse_remove_override_codes), (gst_ssa_parse_parse_line),
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(gst_ssa_parse_chain), (gst_ssa_parse_change_state):
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* gst/subparse/gstssaparse.h:
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* gst/subparse/gstsubparse.c: (plugin_init):
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Add very basic parser for SSA subtitle streams (as often
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found in matroska files).
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2006-02-19 14:09:40 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gstdecodebin.c: That should be text/x-pango-markup, not text/x-pango-layout.
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Original commit message from CVS:
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* gst/playback/gstdecodebin.c: (mimetype_is_raw):
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That should be text/x-pango-markup, not text/x-pango-layout.
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2006-02-19 12:41:03 +0000 Julien Moutte <julien@moutte.net>
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ext/pango/gsttextoverlay.c: Polishing.
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Original commit message from CVS:
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2006-02-19 Julien MOUTTE <julien@moutte.net>
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* ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize):
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Polishing.
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2006-02-19 12:05:23 +0000 Julien Moutte <julien@moutte.net>
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ext/pango/gsttextoverlay.c: Fix state change deadlock.
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Original commit message from CVS:
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2006-02-19 Julien MOUTTE <julien@moutte.net>
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* ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
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(gst_text_overlay_finalize), (gst_text_overlay_init),
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(gst_text_overlay_setcaps), (gst_text_overlay_src_event),
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(gst_text_overlay_render_text),
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(gst_text_overlay_text_pad_link),
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(gst_text_overlay_text_event), (gst_text_overlay_video_event),
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(gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
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(gst_text_overlay_video_chain), (gst_text_overlay_change_state):
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Fix state change deadlock.
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2006-02-19 11:56:28 +0000 Julien Moutte <julien@moutte.net>
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ext/pango/gsttextoverlay.*: Fix seeking both for muxed formats and subtitles files.
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Original commit message from CVS:
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2006-02-19 Julien MOUTTE <julien@moutte.net>
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* ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
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(gst_text_overlay_finalize), (gst_text_overlay_init),
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(gst_text_overlay_setcaps), (gst_text_overlay_src_event),
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(gst_text_overlay_render_text),
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(gst_text_overlay_text_pad_link),
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(gst_text_overlay_text_event), (gst_text_overlay_video_event),
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(gst_text_overlay_pop_text), (gst_text_overlay_text_chain),
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(gst_text_overlay_video_chain), (gst_text_overlay_change_state):
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* ext/pango/gsttextoverlay.h: Fix seeking both for muxed formats
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and subtitles files.
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2006-02-19 00:40:38 +0000 Julien Moutte <julien@moutte.net>
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gst/playback/gstdecodebin.c: pango layout should be considered as row.
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Original commit message from CVS:
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2006-02-19 Julien MOUTTE <julien@moutte.net>
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* gst/playback/gstdecodebin.c: (mimetype_is_raw): pango layout
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should be considered as row.
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2006-02-19 00:25:16 +0000 Julien Moutte <julien@moutte.net>
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gst/playback/gststreaminfo.*: Introduce language informations.
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Original commit message from CVS:
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2006-02-19 Julien MOUTTE <julien@moutte.net>
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* gst/playback/gststreaminfo.c: (gst_stream_type_get_type),
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(cb_probe):
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* gst/playback/gststreaminfo.h: Introduce language informations.
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2006-02-18 22:41:31 +0000 Jan Schmidt <thaytan@mad.scientist.com>
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sys/: Set shared memory segments to be deleted as soon as we have attached, that way they get cleaned up automaticall...
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Original commit message from CVS:
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* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls),
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(gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy):
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* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
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(gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new):
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Set shared memory segments to be deleted as soon as we have attached,
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that way they get cleaned up automatically if we crash.
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2006-02-18 19:53:48 +0000 Julien Moutte <julien@moutte.net>
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ext/pango/: Those functions are called with lock held.
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Original commit message from CVS:
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2006-02-18 Julien MOUTTE <julien@moutte.net>
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* ext/pango/gstclockoverlay.c: (gst_clock_overlay_get_text):
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* ext/pango/gsttimeoverlay.c: (gst_time_overlay_get_text): Those
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functions are called with lock held.
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2006-02-18 19:51:47 +0000 Julien Moutte <julien@moutte.net>
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* ChangeLog:
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Forgot Changelog.
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Original commit message from CVS:
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Forgot Changelog.
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2006-02-18 19:10:35 +0000 Julien Moutte <julien@moutte.net>
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ext/pango/gsttextoverlay.c: Refactoring of textoverlay without collectpads. This now supports sparse subtitles coming...
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Original commit message from CVS:
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2006-02-18 Julien MOUTTE <julien@moutte.net>
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* ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
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(gst_text_overlay_finalize), (gst_text_overlay_init),
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(gst_text_overlay_setcaps), (gst_text_overlay_src_event),
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(gst_text_overlay_render_text),
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(gst_text_overlay_text_pad_link),
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(gst_text_overlay_text_pad_unlink),
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(gst_text_overlay_text_event),
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(gst_text_overlay_video_event), (gst_text_overlay_pop_text),
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(gst_text_overlay_text_chain), (gst_text_overlay_video_chain),
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(gst_text_overlay_change_state): Refactoring of textoverlay
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without collectpads. This now supports sparse subtitles coming
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from a demuxer instead of a sub file. Seeking is still broken
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though. Need to discuss with wtay some more on how to handle
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seeking correctly.
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* ext/pango/gsttextoverlay.h:
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* gst/playback/gstplaybin.c: (setup_sinks): Support linking with
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subtitles coming from the demuxer.
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2006-02-17 19:31:12 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/vorbis/vorbisenc.c: Use some more scaling functions.
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|
Original commit message from CVS:
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|
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_convert_src),
|
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(gst_vorbisenc_convert_sink):
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Use some more scaling functions.
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2006-02-17 16:12:11 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/cdparanoia/gstcdparanoiasrc.*: Add back 'transport-error' and 'uncorrected-error' signals and make them actually ...
|
|
Original commit message from CVS:
|
|
* ext/cdparanoia/gstcdparanoiasrc.c:
|
|
(gst_cd_paranoia_src_class_init), (gst_cd_paranoia_dummy_callback),
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(gst_cd_paranoia_paranoia_callback),
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(gst_cd_paranoia_src_signal_is_being_watched),
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(gst_cd_paranoia_src_read_sector):
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* ext/cdparanoia/gstcdparanoiasrc.h:
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Add back 'transport-error' and 'uncorrected-error' signals and
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make them actually be fired when bad stuff happens (#319340).
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2006-02-17 14:07:01 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstringbuffer.c: Small cleanups.
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|
Original commit message from CVS:
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|
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_get_type),
|
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(gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
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(gst_ring_buffer_device_is_open), (gst_ring_buffer_acquire),
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(gst_ring_buffer_release), (gst_ring_buffer_set_flushing),
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(gst_ring_buffer_start), (gst_ring_buffer_pause_unlocked),
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(gst_ring_buffer_pause), (gst_ring_buffer_stop),
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(gst_ring_buffer_delay), (gst_ring_buffer_samples_done),
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(gst_ring_buffer_set_sample), (gst_ring_buffer_clear_all),
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(gst_ring_buffer_commit), (gst_ring_buffer_prepare_read),
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(gst_ring_buffer_clear):
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Small cleanups.
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Added some G_LIKELY.
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2006-02-17 10:15:52 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/TODO: Update TODO
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|
Original commit message from CVS:
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* gst-libs/gst/audio/TODO:
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Update TODO
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_get_offset):
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When trying to play samples ASAP and we don't have a
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previous sample, try to play at position 0 instead of
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an invalid position.
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2006-02-17 09:24:56 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/alsa/gstalsasink.c: Also release lock when we get an error in _reset(); fix an error message.
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Original commit message from CVS:
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* ext/alsa/gstalsasink.c: (gst_alsasink_open),
|
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(gst_alsasink_reset):
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Also release lock when we get an error in _reset();
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fix an error message.
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2006-02-16 21:01:23 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/alsa/gstalsasink.*: Add support for more than 2 channels (#326720).
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|
Original commit message from CVS:
|
|
* ext/alsa/gstalsasink.c: (gst_alsasink_class_init),
|
|
(gst_alsasink_init), (get_channel_free_structure),
|
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(caps_add_channel_configuration), (gst_alsasink_getcaps),
|
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(gst_alsasink_close):
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* ext/alsa/gstalsasink.h:
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Add support for more than 2 channels (#326720).
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2006-02-16 20:19:51 +0000 Tim-Philipp Müller <tim@centricular.net>
|
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gst-libs/gst/riff/riff-media.c: Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM' with 4 or 6 channe...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
|
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Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM'
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with 4 or 6 channels, assume a default channel layout to make things
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work (not sure there's anything else we can do in those cases).
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2006-02-16 19:18:46 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/audio/multichannel.c: Minor docs fix.
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Original commit message from CVS:
|
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* gst-libs/gst/audio/multichannel.c:
|
|
Minor docs fix.
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* gst-libs/gst/riff/Makefile.am:
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* gst-libs/gst/riff/riff-ids.h:
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* gst-libs/gst/riff/riff-media.c:
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(gst_riff_wavext_add_channel_layout), (gst_riff_create_audio_caps):
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Add support for WAVEFORMATEX, eg. PCM audio with more than two
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channels and a channel layout map.
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2006-02-16 17:06:46 +0000 Edward Hervey <bilboed@bilboed.com>
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gst/videoscale/vs_scanline.c: C-level optimization of the RGBA nearest neighbour function.
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|
Original commit message from CVS:
|
|
Reviewed by Edward Hervey <edward@fluendo.com>
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* gst/videoscale/vs_scanline.c: (vs_scanline_resample_nearest_RGBA):
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C-level optimization of the RGBA nearest neighbour function.
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Eventually this might end up in liboil with vectorized versions.
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2006-02-16 11:44:43 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/audio/multichannel.c: When we have more than 2 channels, but no channel layout is specified in the caps,...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/multichannel.c:
|
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(gst_audio_get_channel_positions):
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When we have more than 2 channels, but no channel layout is
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specified in the caps, return some default channel layout
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to the caller and warn about about a possibly buggy element
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(could be buggy filtercaps as well of course) (#317038).
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2006-02-16 09:29:38 +0000 Tim-Philipp Müller <tim@centricular.net>
|
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pkgconfig/gstreamer-plugins-base-uninstalled.pc.in: Add gst-libs/gst/cdda to list of lib search paths.
|
|
Original commit message from CVS:
|
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* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
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Add gst-libs/gst/cdda to list of lib search paths.
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|
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2006-02-15 12:20:47 +0000 Andy Wingo <wingo@pobox.com>
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ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating timestamp, update timestamp_end as well. Fixes a bugaboo. ...
|
|
Original commit message from CVS:
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|
2006-02-15 Andy Wingo <wingo@pobox.com>
|
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* ext/ogg/gstoggmux.c (gst_ogg_mux_collected): When updating
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timestamp, update timestamp_end as well. Fixes a bugaboo. I hope
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to the Lord Jesus that I do not have to touch the ogg muxer ever
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again.
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2006-02-15 12:07:57 +0000 Edward Hervey <bilboed@bilboed.com>
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gst/typefind/gsttypefindfunctions.c: quicktime movie files can also contain 'uuid' atoms.
|
|
Original commit message from CVS:
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* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
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quicktime movie files can also contain 'uuid' atoms.
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|
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2006-02-14 18:52:52 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/audioconvert/plugin.c: Register the GstAudioChannelPosition enum type with the type system in the plugin_init fun...
|
|
Original commit message from CVS:
|
|
* gst/audioconvert/plugin.c: (plugin_init):
|
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Register the GstAudioChannelPosition enum type with the type
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system in the plugin_init function, so that it is known before
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any element actually makes use of multi-channel stuff. This is
|
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required for example if one wants to be able to deserialise/use
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a caps string with channel positions before any pipeline has
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been setup and started, like with gst-launch.
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2006-02-14 13:45:35 +0000 Wim Taymans <wim.taymans@gmail.com>
|
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gst-libs/gst/audio/gstringbuffer.c: Add some compiler G_(UN_)LIKELY help.
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|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstringbuffer.c:
|
|
(gst_ring_buffer_pause_unlocked), (gst_ring_buffer_delay),
|
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(gst_ring_buffer_samples_done), (wait_segment),
|
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(gst_ring_buffer_commit), (gst_ring_buffer_clear):
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Add some compiler G_(UN_)LIKELY help.
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SIGNAL the ringbuffer waiters when going to PAUSED as well to
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make sure they can exit their functions. Should fix #330748
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2006-02-13 20:49:07 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
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Windows does not have long long; copy the generated _stdint.h
|
|
Original commit message from CVS:
|
|
* Makefile.am:
|
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* configure.ac:
|
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* win32/MANIFEST:
|
|
* win32/common/_stdint.h:
|
|
Windows does not have long long; copy the generated _stdint.h
|
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* win32/common/interfaces-enumtypes.c:
|
|
(gst_color_balance_type_get_type), (gst_mixer_type_get_type),
|
|
(gst_mixer_track_flags_get_type),
|
|
(gst_tuner_channel_flags_get_type):
|
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* win32/common/multichannel-enumtypes.c:
|
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(gst_audio_channel_position_get_type):
|
|
update
|
|
|
|
2006-02-13 18:49:02 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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gst-libs/gst/audio/gstbaseaudiosink.c: Always sync on first sample we receive when starting.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_get_time), (gst_base_audio_sink_preroll),
|
|
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
|
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Always sync on first sample we receive when starting.
|
|
|
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2006-02-13 15:59:48 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst/playback/gstplaybin.c: Update vis bin docs.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin.c: (gen_vis_element):
|
|
Update vis bin docs.
|
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Move queue after tee so we don't queue video buffers but
|
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audio samples instead. Fixes problems where the video queue
|
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is filled and the audio queue empty.
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|
|
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2006-02-13 15:17:34 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
|
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gst-libs/gst/cdda/gstcddabasesrc.c: No need to push an EOS event here, GstBaseSrc will do that for us when we return ...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
|
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No need to push an EOS event here, GstBaseSrc will do that for us
|
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when we return FLOW_UNEXPECTED.
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|
|
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2006-02-12 14:54:55 +0000 Wim Taymans <wim.taymans@gmail.com>
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|
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gst-libs/gst/audio/gstbaseaudiosink.c: Use scale functions when possible.
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|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_get_time), (gst_base_audio_sink_setcaps),
|
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(gst_base_audio_sink_drain), (gst_base_audio_sink_preroll),
|
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(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
|
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Use scale functions when possible.
|
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Fix error messages.
|
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Free clockid when after waiting for EOS.
|
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Use G_(UN_)LIKLY when it makes sense.
|
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Fix sample clipping bug found by Arwed v. Merkatz fixes #330789.
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2006-02-12 14:26:55 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
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gst/playback/gstplaybasebin.c: Remove stray semi-colon (fixes #330888).
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (prepare_output):
|
|
Remove stray semi-colon (fixes #330888).
|
|
|
|
2006-02-11 23:35:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
sys/: Fix up the XShm call testing so that we catch errors, and don't cause new ones by attempting to detach from a s...
|
|
Original commit message from CVS:
|
|
* sys/ximage/ximagesink.c: (gst_ximagesink_check_xshm_calls):
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_check_xshm_calls):
|
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Fix up the XShm call testing so that we catch errors, and don't
|
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cause new ones by attempting to detach from a segment we failed
|
|
to attach to. Fixes #312439.
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|
|
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2006-02-10 11:29:55 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
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gst/typefind/gsttypefindfunctions.c: Added flv file typefind (video/x-flv).
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
|
|
Added flv file typefind (video/x-flv).
|
|
|
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2006-02-10 10:53:33 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
gst-libs/gst/riff/riff-media.c: Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
|
|
(gst_riff_create_video_template_caps):
|
|
Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
|
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Also added the caps to the default set of riff video caps.
|
|
|
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2006-02-09 19:05:23 +0000 Andy Wingo <wingo@pobox.com>
|
|
|
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ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start time and the end time of the last packet in the page.
|
|
Original commit message from CVS:
|
|
2006-02-09 Andy Wingo <wingo@pobox.com>
|
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* ext/ogg/gstoggmux.c (GstOggPad): Keep track of both the start
|
|
time and the end time of the last packet in the page.
|
|
(gst_ogg_mux_pad_queue_page): In addition to setting the timestamp
|
|
on the pages in our queue, set the duration as well. Reflow a
|
|
debug statement.
|
|
(gst_ogg_mux_collected): Keep track of GstOggPad->timestamp_end.
|
|
Fixes bad muxing order.
|
|
|
|
2006-02-09 17:04:18 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
gst-libs/gst/rtp/gstbasertppayload.c: update seqnum before setting it on the packet; this makes sure that the timesta...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/rtp/gstbasertppayload.c:
|
|
(gst_basertppayload_setcaps), (gst_basertppayload_push):
|
|
update seqnum before setting it on the packet; this makes sure
|
|
that the timestamp and seqnum properties match after pushing
|
|
a buffer
|
|
|
|
2006-02-09 12:16:35 +0000 Andy Wingo <wingo@pobox.com>
|
|
|
|
* ChangeLog:
|
|
changelog foo
|
|
Original commit message from CVS:
|
|
changelog foo
|
|
|
|
2006-02-09 11:46:03 +0000 Andy Wingo <wingo@pobox.com>
|
|
|
|
* gst-libs/gst/audio/gstringbuffer.c:
|
|
* win32/common/config.h:
|
|
kapowpowpow
|
|
Original commit message from CVS:
|
|
kapowpowpow
|
|
|
|
2006-02-09 11:36:18 +0000 Andy Wingo <wingo@pobox.com>
|
|
|
|
gst-libs/gst/audio/gstringbuffer.c
|
|
Original commit message from CVS:
|
|
2006-02-09 Andy Wingo <wingo@pobox.com>
|
|
* gst-libs/gst/audio/gstringbuffer.c
|
|
(gst_ring_buffer_samples_done): Cast to guint64, fixes an integer
|
|
overflow after 13.5 hours of recording. Kapow!
|
|
* ext/alsa/gstalsasrc.c (gst_alsasrc_delay): Clamp the delay to
|
|
the buffer size -- we don't care about underrun/overrun reporting
|
|
right now, just need to return a useful value.
|
|
|
|
2006-02-09 11:21:33 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
configure.ac: Back to CVS
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Back to CVS
|
|
|
|
=== release 0.10.3 ===
|
|
|
|
2006-02-09 11:18:22 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
* ChangeLog:
|
|
* NEWS:
|
|
* RELEASE:
|
|
* configure.ac:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.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-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.xml:
|
|
* docs/plugins/inspect/plugin-libvisual.xml:
|
|
* docs/plugins/inspect/plugin-ogg.xml:
|
|
* docs/plugins/inspect/plugin-pango.xml:
|
|
* docs/plugins/inspect/plugin-playbin.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-video4linux.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:
|
|
* win32/common/config.h:
|
|
Releasing 0.10.3
|
|
Original commit message from CVS:
|
|
Releasing 0.10.3
|
|
|
|
2006-02-08 18:37:38 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
configure.ac: Drat. Bump libtool version number for new API.
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
Drat. Bump libtool version number for new API.
|
|
Prelease 0.10.2.3 (of 0.10.3)
|
|
|
|
2006-02-08 15:57:53 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
0.10.2.2 prerelease (of 0.10.3).
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* win32/common/config.h:
|
|
0.10.2.2 prerelease (of 0.10.3).
|
|
|
|
2006-02-08 15:50:08 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst/tcp/gsttcpclientsrc.c: Revert Andy's newsegment change pending a more correct fix.
|
|
Original commit message from CVS:
|
|
* gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_create):
|
|
Revert Andy's newsegment change pending a more correct
|
|
fix.
|
|
|
|
2006-02-08 12:46:14 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/cs.po:
|
|
* po/en_GB.po:
|
|
* po/hu.po:
|
|
* po/it.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
Update .po files
|
|
Original commit message from CVS:
|
|
Update .po files
|
|
|
|
2006-02-08 11:04:09 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* gst/tcp/gstmultifdsink.c:
|
|
doc fixes
|
|
Original commit message from CVS:
|
|
doc fixes
|
|
|
|
2006-02-08 09:20:23 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: detect more files as 3gp group and reorder the iso file formats
|
|
Original commit message from CVS:
|
|
:
|
|
* gst/typefind/gsttypefindfunctions.c: (q3gp_type_find),
|
|
(qt_type_find), (plugin_init):
|
|
detect more files as 3gp
|
|
group and reorder the iso file formats
|
|
|
|
2006-02-07 18:32:00 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/vorbis/vorbis.c: Register musicbrainz tags, so apps don't have to.
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbis.c: (plugin_init):
|
|
Register musicbrainz tags, so apps don't have to.
|
|
|
|
2006-02-07 17:44:53 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/tag/gstvorbistag.c: Make sure we called gst_tag_register_musicbrainz_tags() before possibly mapping a vo...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/gstvorbistag.c: (gst_tag_from_vorbis_tag),
|
|
(gst_tag_to_vorbis_tag):
|
|
Make sure we called gst_tag_register_musicbrainz_tags()
|
|
before possibly mapping a vorbiscomment string from/to a
|
|
musicbrainz tag.
|
|
|
|
2006-02-07 16:16:41 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: In case we can't find the required number of consecutive mpeg audio frames to po...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
|
|
In case we can't find the required number of consecutive
|
|
mpeg audio frames to positively identify an MPEG audio
|
|
stream, check if there's at least a valid mpeg audio
|
|
frame right at offset 0 and if so suggest mpeg/audio
|
|
caps with a very low probability (#153004).
|
|
|
|
2006-02-07 15:52:26 +0000 Andy Wingo <wingo@pobox.com>
|
|
|
|
gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to a TIME segment if we get timestamped buffers. Requir...
|
|
Original commit message from CVS:
|
|
2006-02-07 Andy Wingo <wingo@pobox.com>
|
|
* gst/tcp/gsttcpclientsrc.c (gst_tcp_client_src_create): Switch to
|
|
a TIME segment if we get timestamped buffers. Requires recent
|
|
fixes in core to work properly.
|
|
|
|
2006-02-07 14:57:46 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/playback/gstplaybasebin.c: Don't print the URI as part of the error message, it makes error dialogs look rather u...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (prepare_output):
|
|
Don't print the URI as part of the error message, it
|
|
makes error dialogs look rather ugly, especially if
|
|
the URI is very long or has characters in it that
|
|
need escaping.
|
|
|
|
2006-02-07 13:11:31 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/playback/gstplaybasebin.c: Error out if we have only text or subtitles, but nothing else. Also error out if we ha...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (prepare_output):
|
|
Error out if we have only text or subtitles, but nothing
|
|
else. Also error out if we have subtitles but no video
|
|
stream.
|
|
|
|
2006-02-07 11:44:39 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/gnomevfs/gstgnomevfssrc.c: Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
|
|
Original commit message from CVS:
|
|
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create):
|
|
Treat GNOME_VFS_RESULT_EOF as EOS, not as error (#329194).
|
|
Post an error message on the bus when we encounter an
|
|
error, which will hopefully be more meaningful than the
|
|
'Internal Flow Error' message users get to see if we
|
|
just return GST_FLOW_ERROR.
|
|
|
|
2006-02-07 11:28:04 +0000 Andy Wingo <wingo@pobox.com>
|
|
|
|
configure.ac (GST_MAJORMINOR): Update core version req to 0.10.2.2, for the collectpads API addition (#330244).
|
|
Original commit message from CVS:
|
|
2006-02-07 Andy Wingo <wingo@pobox.com>
|
|
* configure.ac (GST_MAJORMINOR): Update core version req to
|
|
0.10.2.2, for the collectpads API addition (#330244).
|
|
|
|
2006-02-06 19:09:26 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/gnomevfs/gstgnomevfs.c: Return FALSE from plugin_init() when GnomeVFS can't be initialised for some reason (#3284...
|
|
Original commit message from CVS:
|
|
* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
|
|
Return FALSE from plugin_init() when GnomeVFS can't
|
|
be initialised for some reason (#328423).
|
|
|
|
2006-02-06 13:26:54 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
ext/pango/gsttextoverlay.c: Stick to seeking theory until i find the bug.
|
|
Original commit message from CVS:
|
|
2006-02-06 Julien MOUTTE <julien@moutte.net>
|
|
* ext/pango/gsttextoverlay.c: (gst_text_overlay_src_event):
|
|
Stick to seeking theory until i find the bug.
|
|
* gst/subparse/gstsubparse.c: (parse_subrip): Fix debug.
|
|
|
|
2006-02-06 12:38:48 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
Make theoraenc and the tests leak free. Like, really.
|
|
Original commit message from CVS:
|
|
* ext/theora/theoraenc.c: (gst_theora_enc_class_init),
|
|
(theora_enc_finalize), (theora_enc_sink_setcaps),
|
|
(theora_set_header_on_caps), (theora_enc_chain),
|
|
(theora_enc_change_state):
|
|
* tests/check/pipelines/theoraenc.c: (GST_START_TEST):
|
|
Make theoraenc and the tests leak free. Like, really.
|
|
|
|
2006-02-05 23:31:05 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
Add a finalize method to ensure we clean up state even if someone omitted the state change back to NULL.
|
|
Original commit message from CVS:
|
|
(theora_enc_finalize), (theora_enc_sink_setcaps):
|
|
Add a finalize method to ensure we clean up state even if
|
|
someone omitted the state change back to NULL.
|
|
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1),
|
|
(gst_vorbisenc_chain):
|
|
Free some more leaked bits.
|
|
* tests/check/pipelines/theoraenc.c: (start_pipeline),
|
|
(stop_pipeline):
|
|
Wait for state changes to happen if they're ASYNC.
|
|
This ought to teach those fancy pants buildbots a lesson.
|
|
|
|
2006-02-05 22:47:41 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/tag/gstid3tag.c: Add mapping for ID3 International Standard Recording Code tag "TSRC"
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/tag/gstid3tag.c:
|
|
Add mapping for ID3 International Standard Recording Code
|
|
tag "TSRC"
|
|
|
|
2006-02-05 22:44:55 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
ext/vorbis/vorbisenc.c: Don't leak tag names.
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_metadata_set1):
|
|
Don't leak tag names.
|
|
|
|
2006-02-05 18:22:01 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
Split libgsttag docs into multiple sections.
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-docs.sgml:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/tag/gstid3tag.c:
|
|
* gst-libs/gst/tag/gstvorbistag.c:
|
|
* gst-libs/gst/tag/tags.c:
|
|
Split libgsttag docs into multiple sections.
|
|
|
|
2006-02-05 18:01:33 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
Add libgsttag to the docs.
|
|
Original commit message from CVS:
|
|
* docs/libs/Makefile.am:
|
|
* docs/libs/gst-plugins-base-libs-docs.sgml:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_tag):
|
|
* gst-libs/gst/tag/gstvorbistag.c:
|
|
* gst-libs/gst/tag/tag.h:
|
|
* gst-libs/gst/tag/tags.c:
|
|
Add libgsttag to the docs.
|
|
|
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2006-02-05 17:21:23 +0000 Julien Moutte <julien@moutte.net>
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ext/pango/gsttextoverlay.c: Fix clockoverlay.
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Original commit message from CVS:
|
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2006-02-05 Julien MOUTTE <julien@moutte.net>
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* ext/pango/gsttextoverlay.c: (gst_text_overlay_finalize),
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(gst_text_overlay_init), (gst_text_overlay_src_event),
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(gst_text_overlay_collected): Fix clockoverlay.
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2006-02-05 17:15:17 +0000 Tim-Philipp Müller <tim@centricular.net>
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docs/libs/compiling.sgml: Fix typo: it's pkg-config, not pkg-gconfig
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Original commit message from CVS:
|
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* docs/libs/compiling.sgml:
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Fix typo: it's pkg-config, not pkg-gconfig
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* docs/libs/gst-plugins-base-libs-docs.sgml:
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* docs/libs/gst-plugins-base-libs-sections.txt:
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* docs/libs/tmpl/gstgconf.sgml:
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There is no libgstgconf in 0.10, remove it
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from the docs.
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2006-02-05 16:03:48 +0000 Julien Moutte <julien@moutte.net>
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docs/libs/tmpl/gstcolorbalance.sgml: Updated.
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Original commit message from CVS:
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2006-02-05 Julien MOUTTE <julien@moutte.net>
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* docs/libs/tmpl/gstcolorbalance.sgml: Updated.
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* ext/pango/gsttextoverlay.c: (gst_text_overlay_init),
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(gst_text_overlay_src_event), (gst_text_overlay_collected):
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* gst/subparse/gstsubparse.c: (gst_sub_parse_dispose),
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(gst_sub_parse_class_init), (gst_sub_parse_init),
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(gst_sub_parse_src_event), (parse_mdvdsub), (parse_subrip),
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(parse_mpsub), (parser_state_init), (handle_buffer),
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(gst_sub_parse_chain), (gst_sub_parse_sink_event),
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(plugin_init):
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* gst/subparse/gstsubparse.h: Introduce seeking code.
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2006-02-05 15:14:06 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/tag/gstvorbistag.c: Add comment about LANGUAGE tag inconsistency (we want
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Original commit message from CVS:
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* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
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Add comment about LANGUAGE tag inconsistency (we want
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ISO-639-1, but extract three-letter identifiers?)
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* po/POTFILES.in:
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Add two translatable files.
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2006-02-05 14:59:28 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst-libs/gst/tag/: Forward-port some tags stuff from the 0.8 branch. This is mostly the addition of musicbrainz tags ...
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Original commit message from CVS:
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* gst-libs/gst/tag/Makefile.am:
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* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
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* gst-libs/gst/tag/tag.h:
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* gst-libs/gst/tag/tags.c:
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(gst_tag_register_musicbrainz_tags_internal),
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(gst_tag_register_musicbrainz_tags):
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Forward-port some tags stuff from the 0.8 branch. This is
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mostly the addition of musicbrainz tags and their mapping
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to vorbistags, and a vorbistag mapping of the language tag.
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2006-02-05 12:06:25 +0000 Julien Moutte <julien@moutte.net>
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gst/playback/gstplaybin.c: Fix broken code refactoring.
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Original commit message from CVS:
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2006-02-05 Julien MOUTTE <julien@moutte.net>
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* gst/playback/gstplaybin.c: (gen_text_element): Fix broken code
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refactoring.
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2006-02-05 03:05:41 +0000 David Schleef <ds@schleef.org>
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Add Dirac typefinding and add dirac format to oggmux.
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Original commit message from CVS:
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* ext/ogg/gstoggmux.c:
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* gst/typefind/gsttypefindfunctions.c:
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Add Dirac typefinding and add dirac format to oggmux.
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2006-02-04 07:49:03 +0000 Michael Smith <msmith@xiph.org>
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* configure.ac:
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Improve error message for liboil missingness.
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Original commit message from CVS:
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Improve error message for liboil missingness.
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2006-02-03 19:23:41 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gstdecodebin.c: Don't put essential function call into g_return_*() macro, otherwise it'll all be replac...
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Original commit message from CVS:
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* gst/playback/gstdecodebin.c: (try_to_link_1):
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Don't put essential function call into
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g_return_*() macro, otherwise it'll all be
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replaced by NOOPs when compiling with
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G_DISABLE_CHECKS defined.
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2006-02-03 17:45:44 +0000 Edgard Lima <edgard.lima@indt.org.br>
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* ChangeLog:
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* ext/ogg/gstoggdemux.c:
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* ext/ogg/gstoggparse.c:
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* gst/tcp/gsttcpserversink.c:
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* sys/v4l/v4lsrc_calls.c:
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* sys/v4l/v4lsrc_calls.h:
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Just make it compile with --disable-gst-debug.
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Original commit message from CVS:
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Just make it compile with --disable-gst-debug.
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|
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2006-02-03 12:51:47 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/alsa/gstalsasink.*: Add lock to protect alsa calls.
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Original commit message from CVS:
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* ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
|
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(gst_alsasink_class_init), (gst_alsasink_init),
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(gst_alsasink_write), (gst_alsasink_reset):
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* ext/alsa/gstalsasink.h:
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Add lock to protect alsa calls.
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Implement reset to flush samples ASAP, does not work
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with dmix though.
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2006-02-02 18:18:31 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstbaseaudiosink.c: Ugh.. getting late I guess...
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|
Original commit message from CVS:
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_provide_clock):
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Ugh.. getting late I guess...
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2006-02-02 18:13:26 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/gstbaseaudiosink.c: Don't try to provide a clock when we are not negotiated since we might not be ...
|
|
Original commit message from CVS:
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_provide_clock),
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(gst_base_audio_sink_set_property),
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(gst_base_audio_sink_get_property), (gst_base_audio_sink_render):
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Don't try to provide a clock when we are not negotiated since
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we might not be able to make it run.
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2006-02-02 17:51:48 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/gstdecodebin.c: Unlinking two source pads is ... hard.
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|
Original commit message from CVS:
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* gst/playback/gstdecodebin.c: (try_to_link_1):
|
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Unlinking two source pads is ... hard.
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|
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2006-02-02 12:14:35 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst-libs/gst/audio/TODO: Updated.
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|
Original commit message from CVS:
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* gst-libs/gst/audio/TODO:
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|
Updated.
|
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_drain), (gst_base_audio_sink_event):
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On EOS, wait till the last sample is played before posting EOS.
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2006-02-02 08:53:27 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* tests/check/pipelines/theoraenc.c:
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|
comment on my understanding
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|
Original commit message from CVS:
|
|
comment on my understanding
|
|
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2006-02-02 08:47:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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* common:
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* tests/check/pipelines/theoraenc.c:
|
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reformat to fit 80 chars
|
|
Original commit message from CVS:
|
|
reformat to fit 80 chars
|
|
|
|
2006-02-02 00:04:37 +0000 Kai Vehmanen <kv2004@eca.cx>
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|
|
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gst-libs/gst/rtp/gstbasertpdepayload.c: setting queue_delay to zero. Also avoid thread being started if queue_delay i...
|
|
Original commit message from CVS:
|
|
2006-02-01 Philippe Kalaf <burger at speedy dot org>
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
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Patch by Kai Vehmanen : Adds ability to enable newsegment bypass by
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setting queue_delay to zero. Also avoid thread being started if
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queue_delay is zero.
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2006-02-01 14:51:29 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/test6.c: Make test work again by connecting fakesinks to each decoded pad, which makes the pipeline wait...
|
|
Original commit message from CVS:
|
|
* gst/playback/test6.c: (new_decoded_pad_cb), (show_error), (main):
|
|
Make test work again by connecting fakesinks to each decoded pad,
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which makes the pipeline wait until each fakesink has a buffer
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queued before going to PAUSED state. At that point we know the
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decodebin pads are negotiated.
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2006-02-01 11:59:47 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/: Pass unhandled queries to the parent class's query function.
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|
Original commit message from CVS:
|
|
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_query),
|
|
(gst_cdda_base_src_handle_event):
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* gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_query):
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Pass unhandled queries to the parent class's query function.
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2006-02-01 11:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
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Pass unhandled queries upstream instead of just dropping them (#326447). Also, fix supported query types list for som...
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_query_types),
|
|
(gst_ogg_pad_src_query):
|
|
* ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
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* ext/theora/theoradec.c: (theora_dec_src_query),
|
|
(theora_dec_sink_query):
|
|
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
|
|
(vorbis_dec_sink_query):
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|
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_src_query),
|
|
(gst_vorbisenc_sink_query):
|
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* gst/adder/gstadder.c: (gst_adder_query):
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Pass unhandled queries upstream instead of just
|
|
dropping them (#326447). Also, fix supported
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query types list for some elements.
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|
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2006-02-01 09:58:15 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Fix typefinding for audio/x-au, audio/x-paris and audio/iLBC-sh. We cannot use t...
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (au_type_find),
|
|
(paris_type_find), (ilbc_type_find), (plugin_init):
|
|
Fix typefinding for audio/x-au, audio/x-paris and
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audio/iLBC-sh. We cannot use the START_WITH macros
|
|
here, because there can only be one typefind factory
|
|
with the same name (caps), so the second one would
|
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replace the first one and the first one would never
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be called when doing typefinding (see #161712).
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|
|
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2006-01-31 19:25:10 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/vorbis/vorbisdec.c: Use scale_int when we can, add some more scaling.
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisdec.c: (vorbis_dec_convert),
|
|
(vorbis_handle_header_packet), (vorbis_dec_push),
|
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(vorbis_handle_data_packet):
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Use scale_int when we can, add some more scaling.
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Check packettype before parsing it.
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|
|
|
2006-01-31 17:44:35 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/theora/theoradec.c: Call right _scale functions.
|
|
Original commit message from CVS:
|
|
* ext/theora/theoradec.c: (_theora_granule_time),
|
|
(theora_dec_src_convert), (theora_dec_sink_convert):
|
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Call right _scale functions.
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Use parameter instead of some other random value.
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|
|
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2006-01-31 17:27:00 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
ext/theora/theoradec.c: Use higher precision timestamps calculation.
|
|
Original commit message from CVS:
|
|
* ext/theora/theoradec.c: (_theora_granule_frame),
|
|
(_theora_granule_time), (_inc_granulepos),
|
|
(theora_dec_src_convert), (theora_dec_sink_convert),
|
|
(theora_handle_type_packet), (theora_handle_data_packet),
|
|
(theora_dec_chain):
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Use higher precision timestamps calculation.
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Convert some other conversions to _scale.
|
|
|
|
2006-01-31 17:19:09 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/: initialize gst_controller before using
|
|
Original commit message from CVS:
|
|
* gst/audiotestsrc/gstaudiotestsrc.c:
|
|
(gst_audio_test_src_create_sine_table), (plugin_init):
|
|
* gst/volume/gstvolume.c: (plugin_init):
|
|
initialize gst_controller before using
|
|
|
|
2006-01-31 16:26:57 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
tests/check/pipelines/: Define constant using G_GINT64_CONSTANT to avoid errors when passing it around - otherwise it...
|
|
Original commit message from CVS:
|
|
* tests/check/pipelines/theoraenc.c:
|
|
* tests/check/pipelines/vorbisenc.c:
|
|
Define constant using G_GINT64_CONSTANT to avoid errors when
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passing it around - otherwise it gets truncated to 32 bits.
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Fixes failing tests.
|
|
|
|
2006-01-31 15:36:13 +0000 Andy Wingo <wingo@pobox.com>
|
|
|
|
sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the caps being set doesn't have a framerate value. Basic...
|
|
Original commit message from CVS:
|
|
2006-01-31 Andy Wingo <wingo@pobox.com>
|
|
* sys/v4l/gstv4lsrc.c (gst_v4lsrc_set_caps): Don't segfault if the
|
|
caps being set doesn't have a framerate value. Basically a stopgap
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|
measure.
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* ext/ogg/gstoggmux.c (GST_BUFFER_END_TIME): New macro. Not
|
|
technically correct enough to put into core though.
|
|
(gst_ogg_mux_dequeue_page): Use END_TIME instead of TIMESTAMP +
|
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DURATION. Fixes theoraenc ! oggmux.
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* sys/v4l/gstv4lsrc.c (gst_v4lsrc_fixate): Fixate to the nearest
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|
fraction, not double.
|
|
|
|
2006-01-31 12:23:35 +0000 Christian Schaller <uraeus@gnome.org>
|
|
|
|
* gst-plugins-base.spec.in:
|
|
update with latest files
|
|
Original commit message from CVS:
|
|
update with latest files
|
|
|
|
2006-01-30 23:42:54 +0000 Sébastien Moutte <sebastien@moutte.net>
|
|
|
|
win32/vs7: add vs7 project files created by Sergey Scobich
|
|
Original commit message from CVS:
|
|
* win32/vs7:
|
|
add vs7 project files created by Sergey Scobich
|
|
|
|
2006-01-30 22:18:53 +0000 Sébastien Moutte <sebastien@moutte.net>
|
|
|
|
win32/vs8: add vs8 project files created by Sergey Scobich
|
|
Original commit message from CVS:
|
|
* win32/vs8:
|
|
add vs8 project files created by Sergey Scobich
|
|
|
|
2006-01-30 19:22:22 +0000 Andy Wingo <wingo@pobox.com>
|
|
|
|
ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare timestamp + duration, not just timestamp -- ogg pages should ...
|
|
Original commit message from CVS:
|
|
2006-01-30 Andy Wingo <wingo@pobox.com>
|
|
* ext/ogg/gstoggmux.c (gst_ogg_mux_dequeue_page): Compare
|
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timestamp + duration, not just timestamp -- ogg pages should be
|
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ordered by stop time. Necessary fix given the change in vorbis
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timestamps.
|
|
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2006-01-30 19:21:07 +0000 Andy Wingo <wingo@pobox.com>
|
|
|
|
* ChangeLog:
|
|
* ext/theora/gsttheoraenc.h:
|
|
* ext/theora/theoraenc.c:
|
|
* tests/check/pipelines/theoraenc.c:
|
|
ext/theora/theoraenc.c (theora_enc_sink_setcaps)
|
|
Original commit message from CVS:
|
|
2006-01-30 Andy Wingo <wingo@pobox.com>
|
|
* ext/theora/theoraenc.c (theora_enc_sink_setcaps)
|
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(gst_theora_enc_init): Pull the granule shift out of the encoder.
|
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(granulepos_add): New function, handles the messiness of adjusting
|
|
granulepos values.
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(theora_buffer_from_packet):
|
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(theora_enc_chain):
|
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(theora_enc_sink_event): Use granulepos_add, not +.
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* tests/check/pipelines/theoraenc.c
|
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(check_buffer_granulepos_from_starttime): Just check the frame
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count, not the actual granulepos -- we can't dictate to the
|
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encoder when it should be placing keyframes.
|
|
|
|
2006-01-30 18:17:19 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
ext/gnomevfs/gstgnomevfssrc.c: SERVICE_NOT_AVAILABLE happens for example when you're trying to play an http:// stream...
|
|
Original commit message from CVS:
|
|
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_start):
|
|
SERVICE_NOT_AVAILABLE happens for example when you're trying to
|
|
play an http:// stream from a server that's not serving
|
|
|
|
2006-01-30 17:08:11 +0000 Andy Wingo <wingo@pobox.com>
|
|
|
|
tests/check/pipelines/: Totally remove the UINT64_CONSTANT macro, doesn't appear to be needed or available.
|
|
Original commit message from CVS:
|
|
2006-01-30 Andy Wingo <wingo@pobox.com>
|
|
* tests/check/pipelines/vorbisenc.c (TIMESTAMP_OFFSET):
|
|
* tests/check/pipelines/theoraenc.c (TIMESTAMP_OFFSET): Totally
|
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remove the UINT64_CONSTANT macro, doesn't appear to be needed or
|
|
available.
|
|
|
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2006-01-30 17:01:54 +0000 Andy Wingo <wingo@pobox.com>
|
|
|
|
ext/theora/: Same changes as were done to vorbisenc, although theoraenc was timestamping correctly. Added handling of...
|
|
Original commit message from CVS:
|
|
2006-01-30 Andy Wingo <wingo@pobox.com>
|
|
* ext/theora/gsttheoraenc.h:
|
|
* ext/theora/theoraenc.c: Same changes as were done to vorbisenc,
|
|
although theoraenc was timestamping correctly. Added handling of
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streams that start with nonzero timestamps.
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* tests/check/Makefile.am:
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* tests/check/pipelines/theoraenc.c: New file, basically does same
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tests as vorbisenc.
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* tests/check/pipelines/vorbisenc.c: I claim these bugs.
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|
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2006-01-30 16:19:33 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstaudiosink.c: Implement pause that does not wait for completion.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstaudiosink.c:
|
|
(gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
|
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(gst_audioringbuffer_pause):
|
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Implement pause that does not wait for completion.
|
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* gst-libs/gst/audio/gstbaseaudiosink.c:
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(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
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Don't drop buffers when going to PAUSED but perform preroll on
|
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remaining samples now that core base class supports this.
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* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
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(gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
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(gst_ring_buffer_commit):
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Pause should not signal waiters.
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Implement return value of _commit correctly.
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|
|
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2006-01-30 15:01:28 +0000 Andy Wingo <wingo@pobox.com>
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|
|
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tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
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|
Original commit message from CVS:
|
|
2006-01-30 Andy Wingo <wingo@pobox.com>
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* tests/check/Makefile.am (check_vorbis): Add pipelines/vorbisenc.
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* ext/vorbis/vorbisenc.c (gst_vorbisenc_buffer_from_packet): Logic
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updated to timestamp from the first sample, not the last.
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(gst_vorbisenc_buffer_from_header_packet): New function, takes
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special care of granulepos and timestamp for header packets.
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(gst_vorbisenc_chain): Reflow, fix some leaks, and handle the case
|
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when the first buffer has a nonzero timestamp.
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* ext/vorbis/vorbisenc.h (GstVorbisEnc.granulepos_offset)
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(GstVorbisEnc.subgranule_offset): New members. Take care of the
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case when the first audio buffer we get has a nonzero timestamp.
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(GstVorbisEnc.next_ts): Renamed from prev_ts, because now we
|
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properly timestamp vorbis buffers with the time of the first
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sample, not the last.
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* ext/vorbis/vorbisenc.c (granulepos_to_clocktime): Renamed from
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vorbis_granule_time_copy -- now it takes the granule/subgranule
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offset into account.
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* tests/check/pipelines/vorbisenc.c: New test for correctness of
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timestamps, durations, and granulepos on buffers produced by
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vorbisenc.
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2006-01-30 14:42:28 +0000 Eric Jonas <jonas@mit.edu>
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|
|
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gst/ffmpegcolorspace/gstffmpegcodecmap.c: Patch from Eric Jonas to support conversions to/from UYVY (Fixes: #324626)
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|
Original commit message from CVS:
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
|
|
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt):
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Patch from Eric Jonas to support conversions to/from UYVY
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(Fixes: #324626)
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2006-01-30 08:11:14 +0000 Julien Moutte <julien@moutte.net>
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|
|
|
gst/playback/: Implement subtitles.
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|
Original commit message from CVS:
|
|
2006-01-30 Julien MOUTTE <julien@moutte.net>
|
|
* gst/playback/gstplaybasebin.c: (group_commit),
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(queue_overrun),
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(setup_subtitle), (setup_source), (set_active_source):
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* gst/playback/gstplaybin.c: (gst_play_bin_dispose),
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(gen_text_element), (gen_audio_element), (gen_vis_element),
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(remove_sinks), (add_sink), (setup_sinks): Implement subtitles.
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2006-01-29 19:13:39 +0000 Sébastien Moutte <sebastien@moutte.net>
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|
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gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
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|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
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use of gst_guint64_to_gdouble to be compliant with vs6
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* gst/playback/gstdecodebin.c: (try_to_link_1)
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* gst/videorate/videorate.c: (gst_video_rate_blank_data)
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use of G_GINT64_CONSTANT for int64 constants
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* win32/common/libgstinterfaces.def:
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export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
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* win32/vs6:
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update and add new project files
|
|
|
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2006-01-29 18:21:12 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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|
|
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add a win32-update rule like in core, and copy over enumtypes files
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|
Original commit message from CVS:
|
|
* Makefile.am:
|
|
* win32/MANIFEST:
|
|
* win32/common/interfaces-enumtypes.c:
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|
(gst_color_balance_type_get_type), (gst_mixer_type_get_type),
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(gst_mixer_track_flags_get_type),
|
|
(gst_tuner_channel_flags_get_type):
|
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* win32/common/interfaces-enumtypes.h:
|
|
* win32/common/multichannel-enumtypes.c:
|
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(gst_audio_channel_position_get_type):
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* win32/common/multichannel-enumtypes.h:
|
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add a win32-update rule like in core, and copy over enumtypes files
|
|
|
|
2006-01-29 18:07:51 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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|
|
|
* configure.ac:
|
|
generate win32/common/config.h
|
|
Original commit message from CVS:
|
|
generate win32/common/config.h
|
|
|
|
2006-01-29 18:05:42 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
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win32/: add config files just like in core
|
|
Original commit message from CVS:
|
|
* win32/MANIFEST:
|
|
* win32/common/config.h:
|
|
* win32/common/config.h.in:
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|
add config files just like in core
|
|
|
|
2006-01-28 18:22:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
ext/alsa/: Update all error messages. All of them should either use the default translated message, or actually prov...
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsasink.c: (gst_alsasink_init), (set_hwparams),
|
|
(set_swparams), (gst_alsasink_prepare), (gst_alsasink_unprepare),
|
|
(gst_alsasink_close), (gst_alsasink_write), (gst_alsasink_reset):
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* ext/alsa/gstalsasrc.c: (gst_alsasrc_init), (set_hwparams),
|
|
(set_swparams), (gst_alsasrc_open), (gst_alsasrc_prepare),
|
|
(gst_alsasrc_unprepare), (gst_alsasrc_read):
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Update all error messages. All of them should either use
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the default translated message, or actually provide a
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|
translatable string.
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Make the string for channel count problems meaningful.
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|
|
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2006-01-28 18:19:18 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/audio/gstringbuffer.c: Make gcc-4.1 happy (part of #327357).
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstringbuffer.c: (build_linear_format):
|
|
Make gcc-4.1 happy (part of #327357).
|
|
|
|
2006-01-28 16:35:47 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
sys/v4l/v4l_calls.c: check for and throw RESOURCE_BUSY
|
|
Original commit message from CVS:
|
|
* sys/v4l/v4l_calls.c: (gst_v4l_open):
|
|
check for and throw RESOURCE_BUSY
|
|
|
|
2006-01-28 02:13:14 +0000 David Schleef <ds@schleef.org>
|
|
|
|
gst/videoscale/vs_scanline.c: Oops, *that's* why I never checked in this change -- it requires liboil features not in...
|
|
Original commit message from CVS:
|
|
* gst/videoscale/vs_scanline.c: Oops, *that's* why I never
|
|
checked in this change -- it requires liboil features not
|
|
in 0.3.6. Revert parts.
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|
|
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2006-01-27 23:40:19 +0000 David Schleef <ds@schleef.org>
|
|
|
|
update liboil requirement to 0.3.6
|
|
Original commit message from CVS:
|
|
* REQUIREMENTS:
|
|
* configure.ac: update liboil requirement to 0.3.6
|
|
* gst/videoscale/Makefile.am:
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* gst/videoscale/vs_scanline.c: liboilify
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|
|
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2006-01-27 17:00:09 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
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ext/libvisual/visual.c: When pad_alloc returns a GstFlowReturn other than GST_FLOW_OK, make sure it is passed upstream.
|
|
Original commit message from CVS:
|
|
* ext/libvisual/visual.c: (get_buffer):
|
|
When pad_alloc returns a GstFlowReturn other
|
|
than GST_FLOW_OK, make sure it is passed upstream.
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|
|
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2006-01-27 01:36:01 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
ext/alsa/gstalsasink.c: Free the device name string.
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsasink.c: (gst_alsasink_finalise),
|
|
(gst_alsasink_class_init):
|
|
Free the device name string.
|
|
* ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init),
|
|
(gst_ogg_mux_request_new_pad), (gst_ogg_mux_release_pad),
|
|
(gst_ogg_mux_handle_src_event), (gst_ogg_mux_clear_collectpads):
|
|
Don't remove a pad from the collectpads structure until it
|
|
is released - it's a request pad, and may receive data again
|
|
if the element gets moved back to PLAYING state.
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
|
|
Ensure we turn on double buffering on the Xv port, and
|
|
set the colour key to something dark and mysterious that
|
|
isn't black.
|
|
|
|
2006-01-27 01:06:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
ext/: - a library should not call setlocale. see Libraries node in gettext manual
|
|
Original commit message from CVS:
|
|
* ext/alsa/gstalsaplugin.c: (plugin_init):
|
|
* ext/cdparanoia/gstcdparanoiasrc.c:
|
|
(gst_cd_paranoia_src_base_init), (plugin_init):
|
|
* ext/gnomevfs/gstgnomevfs.c: (plugin_init):
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_plugin_init):
|
|
- a library should not call setlocale. see Libraries node in
|
|
gettext manual
|
|
- make sure all plugins that use translation do bindtextdomain
|
|
to point to the localedir
|
|
* gst/playback/gstplaybin.c: (gen_vis_element), (add_sink),
|
|
(setup_sinks), (plugin_init):
|
|
all this, and check for NULL when creating sinks
|
|
|
|
2006-01-26 23:21:31 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
gst/subparse/gstsubparse.c: Make typefinding of subtitles work again.
|
|
Original commit message from CVS:
|
|
2006-01-27 Julien MOUTTE <julien@moutte.net>
|
|
* gst/subparse/gstsubparse.c: (gst_subparse_type_find),
|
|
(plugin_init): Make typefinding of subtitles work again.
|
|
|
|
2006-01-26 20:40:20 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/typefind/gsttypefindfunctions.c: Backport a bunch of typefinding fixes from the 0.8 branch.
|
|
Original commit message from CVS:
|
|
* gst/typefind/gsttypefindfunctions.c: (aac_type_find),
|
|
(mp3_type_frame_length_from_header), (mp3_type_find),
|
|
(wavpack_type_find), (m4a_type_find), (ircam_type_find),
|
|
(plugin_init):
|
|
Backport a bunch of typefinding fixes from the 0.8 branch.
|
|
Also, improve wavpack typefinding: if we can't peek the
|
|
entire wavpack block, try to parse the bits we can get and
|
|
see if we find what we're looking for in those.
|
|
|
|
2006-01-26 19:17:38 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
sys/: Handle some more cases of pixel aspect ratio.
|
|
Original commit message from CVS:
|
|
2006-01-26 Julien MOUTTE <julien@moutte.net>
|
|
* sys/ximage/ximagesink.c:
|
|
(gst_ximagesink_calculate_pixel_aspect_ratio):
|
|
* sys/xvimage/xvimagesink.c:
|
|
(gst_xvimagesink_calculate_pixel_aspect_ratio): Handle some
|
|
more cases of pixel aspect ratio.
|
|
|
|
2006-01-26 13:09:24 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
gst/playback/gstdecodebin.c: Also consider the flush-start and tag events as unblockers for the pad probes.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (pad_probe):
|
|
Also consider the flush-start and tag events as unblockers
|
|
for the pad probes.
|
|
|
|
2006-01-26 12:32:58 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
gst/playback/gstplaybin.c: On the fly visualisation switch, works disabling, enabling as well but it won't be able to...
|
|
Original commit message from CVS:
|
|
2006-01-26 Julien MOUTTE <julien@moutte.net>
|
|
* gst/playback/gstplaybin.c: (gst_play_bin_init),
|
|
(gst_play_bin_dispose), (gst_play_bin_vis_unblocked),
|
|
(gst_play_bin_vis_blocked), (gst_play_bin_set_property):
|
|
On the fly visualisation switch, works disabling, enabling as
|
|
well but it won't be able to enable vis in a playbin that was
|
|
created with no visualisation.
|
|
|
|
2006-01-25 10:50:32 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.c: Undo previous commit, it breaks resume after pause.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
|
|
Undo previous commit, it breaks resume after pause.
|
|
|
|
2006-01-25 09:27:01 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.c: Improve debugging.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
|
|
(gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
|
|
Improve debugging.
|
|
Post error when caps cannot be parsed.
|
|
Resync on discontinuity in the stream.
|
|
Clip samples to segment boundaries.
|
|
return WRONG_STATE sooner when we are flushing.
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
|
|
(gst_base_audio_src_get_time), (gst_base_audio_src_create):
|
|
Make audiosrc operate in TIME.
|
|
Set TIMESTAMP and DURATION on buffers.
|
|
|
|
2006-01-24 21:55:21 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/examples/seek/seek.c: Output tag messages as well.
|
|
Original commit message from CVS:
|
|
* tests/examples/seek/seek.c: (main):
|
|
Output tag messages as well.
|
|
|
|
2006-01-23 15:05:24 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
gst/playback/gstdecodebin.c: Replace GstPadBlockCallback with pad probes that detect first buffer AND eos before remo...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (gst_decode_bin_init),
|
|
(free_pad_probes), (remove_fakesink), (pad_probe),
|
|
(close_pad_link), (gst_decode_bin_change_state):
|
|
Replace GstPadBlockCallback with pad probes that detect
|
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first buffer AND eos before removing fakesink.
|
|
Fixes hang with demuxers doing EOS while pre-rolling.
|
|
Solves #328279
|
|
|
|
2006-01-23 10:10:36 +0000 Jens Granseuer <jensgr@gmx.net>
|
|
|
|
GCC 2.95 fixes (#328263).
|
|
Original commit message from CVS:
|
|
2006-01-23 Andy Wingo <wingo@pobox.com>
|
|
* ext/alsa/gstalsasink.c:
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_setcaps),
|
|
(gst_base_rtp_depayload_add_to_queue),
|
|
(gst_base_rtp_depayload_queue_release): GCC 2.95 fixes (#328263).
|
|
Patch by: Jens Granseuer <jensgr at gmx dot net>
|
|
|
|
2006-01-22 17:24:02 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
sys/: Playbin keeps some ref to some frames. We might get a frame destroyed after changing state to
|
|
Original commit message from CVS:
|
|
2006-01-22 Julien MOUTTE <julien@moutte.net>
|
|
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_destroy):
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
|
|
(gst_xvimagesink_buffer_alloc): Playbin keeps some ref to some
|
|
frames. We might get a frame destroyed after changing state to
|
|
NULL, adding a safety check on xcontext.
|
|
|
|
2006-01-22 14:50:53 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/interfaces/xoverlay.c: Fix prepare-xwindow-id code example in the docs - we need to ignore all messages ...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/interfaces/xoverlay.c:
|
|
Fix prepare-xwindow-id code example in the docs - we need to
|
|
ignore all messages that aren't element messages as well.
|
|
|
|
2006-01-21 22:40:03 +0000 Julien Moutte <julien@moutte.net>
|
|
|
|
sys/xvimage/xvimagesink.c: I think one day i'll completely undestand how caps negotiation is supposed to work. This r...
|
|
Original commit message from CVS:
|
|
2006-01-21 Julien MOUTTE <julien@moutte.net>
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_buffer_alloc):
|
|
I think one day i'll completely undestand how caps negotiation
|
|
is supposed to work. This refactoring handles buffer_alloc
|
|
called with caps we can't handle. We definitely don't want a
|
|
set_caps with those caps, so we define and allocate a buffer
|
|
we would like to receive.
|
|
|
|
2006-01-20 19:10:26 +0000 Christian Schaller <uraeus@gnome.org>
|
|
|
|
* autogen.sh:
|
|
* common:
|
|
up automake requirement to 1.7
|
|
Original commit message from CVS:
|
|
up automake requirement to 1.7
|
|
|
|
2006-01-19 10:59:51 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/playback/gstplaybasebin.c: Free iterator when done.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (setup_source):
|
|
Free iterator when done.
|
|
|
|
2006-01-17 11:43:49 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst-libs/gst/audio/gstbaseaudiosink.c: Fix playback of non-synchronised streams by assuming a rate of 1.0 instead of ...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_render):
|
|
Fix playback of non-synchronised streams by assuming a rate
|
|
of 1.0 instead of a random one.
|
|
Makes this work again:
|
|
gst-launch filesrc location=raw_audio.file ! 'audio/x-raw-int,
|
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endianness=(int)4321, signed=(boolean)true, width=(int)16,
|
|
depth=(int)16, rate=(int)44100, channels=(int)2' ! audioconvert !
|
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audioresample ! alsasink
|
|
|
|
2006-01-16 21:01:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* configure.ac:
|
|
back to HEAD
|
|
Original commit message from CVS:
|
|
back to HEAD
|
|
|
|
=== release 0.10.2 ===
|
|
|
|
2006-01-16 20:59:32 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* ChangeLog:
|
|
* NEWS:
|
|
* RELEASE:
|
|
* configure.ac:
|
|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.xml:
|
|
* docs/plugins/inspect/plugin-audioconvert.xml:
|
|
* docs/plugins/inspect/plugin-audiorate.xml:
|
|
* docs/plugins/inspect/plugin-audioresample.xml:
|
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* docs/plugins/inspect/plugin-audiotestsrc.xml:
|
|
* docs/plugins/inspect/plugin-cdparanoia.xml:
|
|
* docs/plugins/inspect/plugin-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.xml:
|
|
* docs/plugins/inspect/plugin-libvisual.xml:
|
|
* docs/plugins/inspect/plugin-ogg.xml:
|
|
* docs/plugins/inspect/plugin-pango.xml:
|
|
* docs/plugins/inspect/plugin-playbin.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-video4linux.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:
|
|
releasing 0.10.2
|
|
Original commit message from CVS:
|
|
releasing 0.10.2
|
|
|
|
2006-01-16 16:38:15 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
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gst/playback/: Comment out broken code that connects to the state-changed signal.
|
|
Original commit message from CVS:
|
|
* gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
|
|
* gst/playback/gststreamselector.c:
|
|
(gst_stream_selector_set_property):
|
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Comment out broken code that connects to the state-changed signal.
|
|
At this point, changing current stream selection is broken, but
|
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stuff like gst-launch playbin current-audio=1 works and filters
|
|
to the chosen stream.
|
|
|
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2006-01-16 15:31:14 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
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ext/vorbis/vorbisdec.c: Fix #327216 (null dereference in vorbisdec)
|
|
Original commit message from CVS:
|
|
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
|
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Fix #327216 (null dereference in vorbisdec)
|
|
|
|
2006-01-16 15:19:55 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/theora/theoradec.c: Post taglist actually on bus instead of just freeing it (fixes #327114 and totem bug #327080).
|
|
Original commit message from CVS:
|
|
* ext/theora/theoradec.c: (theora_handle_comment_packet):
|
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Post taglist actually on bus instead of just freeing it
|
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(fixes #327114 and totem bug #327080).
|
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* ext/vorbis/vorbisdec.c: (vorbis_handle_comment_packet):
|
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Use gst_element_found_tags_for_pad(), so that the tags
|
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are sent downstream as an event as well.
|
|
|
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2006-01-15 10:06:40 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
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sys/: move all regularly occurring messages to GST_LOG level add some more object logs
|
|
Original commit message from CVS:
|
|
* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
|
|
(gst_ximagesink_ximage_new), (gst_ximagesink_ximage_put),
|
|
(gst_ximagesink_buffer_alloc):
|
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* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize),
|
|
(gst_xvimagesink_xvimage_put), (gst_xvimagesink_show_frame),
|
|
(gst_xvimagesink_buffer_alloc):
|
|
move all regularly occurring messages to GST_LOG level
|
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add some more object logs
|
|
|
|
2006-01-14 22:59:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* configure.ac:
|
|
* po/af.po:
|
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* po/az.po:
|
|
* po/cs.po:
|
|
* po/en_GB.po:
|
|
* po/hu.po:
|
|
* po/it.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
prerelease
|
|
Original commit message from CVS:
|
|
prerelease
|
|
|
|
2006-01-14 20:46:25 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
ext/ogg/gstoggmux.c: fix a silly segfault
|
|
Original commit message from CVS:
|
|
2006-01-14 Thomas Vander Stichele <thomas at apestaart dot org>
|
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* ext/ogg/gstoggmux.c: (gst_ogg_mux_collected):
|
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fix a silly segfault
|
|
|
|
2006-01-14 12:52:22 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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Add docs for mixerutils stuff.
|
|
Original commit message from CVS:
|
|
* docs/libs/gst-plugins-base-libs-docs.sgml:
|
|
* docs/libs/gst-plugins-base-libs-sections.txt:
|
|
* gst-libs/gst/audio/mixerutils.c:
|
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* gst-libs/gst/audio/mixerutils.h:
|
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Add docs for mixerutils stuff.
|
|
|
|
2006-01-13 17:17:07 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/playback/gstplaybasebin.c: Fix playback for sources that emit raw audio or raw video streams (e.g.: cd audio sour...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (setup_source):
|
|
Fix playback for sources that emit raw audio or
|
|
raw video streams (e.g.: cd audio sources) (#325984).
|
|
|
|
2006-01-13 16:45:50 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
gst-libs/gst/audio/mixerutils.c: actually save the element we create
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/mixerutils.c:
|
|
(gst_audio_mixer_filter_do_filter):
|
|
actually save the element we create
|
|
|
|
2006-01-13 16:17:50 +0000 Christian Schaller <uraeus@gnome.org>
|
|
|
|
* gst-plugins-base.spec.in:
|
|
remove version suffix
|
|
Original commit message from CVS:
|
|
remove version suffix
|
|
|
|
2006-01-12 14:56:11 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/cdda/gstcddabasesrc.c: No need to post a tag message on the bus when seeking within the same track, only...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/cdda/gstcddabasesrc.c:
|
|
(gst_cdda_base_src_handle_track_seek):
|
|
No need to post a tag message on the bus when seeking
|
|
within the same track, only post it when the current
|
|
track changes.
|
|
|
|
2006-01-11 18:30:25 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst/playback/: Reenable stream selection. These mechanisms need a complete overhaul in the face of 0.8->0.10 changes ...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (group_destroy),
|
|
(probe_triggered), (new_decoded_pad), (mute_group_type),
|
|
(set_active_source):
|
|
* gst/playback/gststreaminfo.c: (gst_stream_info_set_mute):
|
|
* gst/playback/gststreamselector.c:
|
|
(gst_stream_selector_base_init),
|
|
(gst_stream_selector_set_property),
|
|
(gst_stream_selector_request_new_pad):
|
|
Reenable stream selection. These mechanisms need a complete overhaul
|
|
in the face of 0.8->0.10 changes though.
|
|
|
|
2006-01-11 18:03:24 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
ext/ogg/gstoggdemux.c: Change the pad template to src_%d to match the pads that are created from it. decodebin needs ...
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c:
|
|
Change the pad template to src_%d to match the pads that
|
|
are created from it. decodebin needs this information in order
|
|
to decide that oggdemux is capable of producing multiple pads
|
|
(and hence needs queues inserted).
|
|
* ext/ogg/gstoggmux.c: (gst_ogg_mux_queue_pads),
|
|
(gst_ogg_mux_collected):
|
|
Make debug output more useful by using GST_PTR_FORMAT.
|
|
|
|
2006-01-11 17:38:35 +0000 Christian Schaller <uraeus@gnome.org>
|
|
|
|
* gst-plugins-base.spec.in:
|
|
update spec.in file
|
|
Original commit message from CVS:
|
|
update spec.in file
|
|
|
|
2006-01-11 15:11:20 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/audio/gstringbuffer.c: Set depth and width for alaw/mulaw (fixes #326601).
|
|
Original commit message from CVS:
|
|
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
|
|
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
|
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Set depth and width for alaw/mulaw (fixes #326601).
|
|
|
|
2006-01-10 23:58:36 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
tests/icles/Makefile.am: don't build the tests if we don't have the libs
|
|
Original commit message from CVS:
|
|
* tests/icles/Makefile.am:
|
|
don't build the tests if we don't have the libs
|
|
|
|
2006-01-10 18:06:56 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
ext/cdparanoia/gstcdparanoiasrc.c: Don't try to free NULL pointers.
|
|
Original commit message from CVS:
|
|
* ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_close),
|
|
(gst_cd_paranoia_paranoia_callback):
|
|
Don't try to free NULL pointers.
|
|
|
|
2006-01-10 15:47:48 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
gst/audiorate/gstaudiorate.c: Add debugging category.
|
|
Original commit message from CVS:
|
|
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain),
|
|
(gst_audio_rate_change_state), (plugin_init):
|
|
Add debugging category.
|
|
Fix type issues.
|
|
Add case for incoming buffers without valid offset/offset_end.
|
|
|
|
2006-01-10 12:25:59 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst-libs/gst/audio/gstaudiosrc.c: Don't leak GCond in audio sources.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_dispose):
|
|
Don't leak GCond in audio sources.
|
|
|
|
2006-01-10 11:49:28 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
gst/playback/gstplaybin.c: Don't leak an autoaudiosink/alsasink when we generate a new audio element. (old code, I gu...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybin.c: (gen_audio_element):
|
|
Don't leak an autoaudiosink/alsasink when we generate
|
|
a new audio element. (old code, I guess)
|
|
|
|
2006-01-10 11:04:21 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/audiorate/gstaudiorate.c: Support float audio in audiorate.
|
|
Original commit message from CVS:
|
|
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_setcaps):
|
|
Support float audio in audiorate.
|
|
Use width rather than depth for selecting sample width.
|
|
|
|
2006-01-10 10:06:53 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/videotestsrc/videotestsrc.h: Use GLib types here (that way we don't have to include the generated _stdint.h heade...
|
|
Original commit message from CVS:
|
|
* gst/videotestsrc/videotestsrc.h:
|
|
Use GLib types here (that way we don't have to include the
|
|
generated _stdint.h header, which makes life easier for win32
|
|
folks that don't use autotools for the build) (#325990, patch
|
|
by: Sergey Scobich).
|
|
|
|
2006-01-10 09:38:44 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/audio/gstringbuffer.*: Name (private) union, makes Forte compiler happy (this time for real) (#324900).
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
|
|
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
|
|
(gst_ring_buffer_pause), (wait_segment):
|
|
* gst-libs/gst/audio/gstringbuffer.h:
|
|
Name (private) union, makes Forte compiler happy (this time
|
|
for real) (#324900).
|
|
|
|
2006-01-09 10:52:33 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/audio/Makefile.am: Link against libgstinterfaces, needed for mixer and property probe stuff.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/Makefile.am:
|
|
Link against libgstinterfaces, needed for mixer
|
|
and property probe stuff.
|
|
|
|
2006-01-09 10:46:52 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
gst-libs/gst/Makefile.am:
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/Makefile.am:
|
|
|
|
2006-01-09 09:38:34 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/audio/: Add gst_audio_default_registry_mixer_filter() utility function.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/Makefile.am:
|
|
* gst-libs/gst/audio/mixerutils.c:
|
|
(gst_audio_mixer_filter_do_filter),
|
|
(gst_audio_mixer_filter_check_element),
|
|
(gst_audio_mixer_filter_probe_feature),
|
|
(element_factory_rank_compare_func),
|
|
(gst_audio_default_registry_mixer_filter):
|
|
* gst-libs/gst/audio/mixerutils.h:
|
|
Add gst_audio_default_registry_mixer_filter() utility
|
|
function.
|
|
|
|
2006-01-03 17:33:16 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/audioresample/resample.h: As before, but for o_buf
|
|
Original commit message from CVS:
|
|
* gst/audioresample/resample.h:
|
|
As before, but for o_buf
|
|
|
|
2006-01-03 17:27:13 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/audioresample/resample.h: Declare struct _ResampleState.buffer as unsigned char *, not void *, since we do arithm...
|
|
Original commit message from CVS:
|
|
* gst/audioresample/resample.h:
|
|
Declare struct _ResampleState.buffer as unsigned char *, not void *,
|
|
since we do arithmetic on it.
|
|
|
|
2006-01-02 23:37:38 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst-libs/gst/audio/gstringbuffer.*: Sun's Forte compiler doesn't seem to like anonymous structs, so use same setup as...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
|
|
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
|
|
(gst_ring_buffer_pause), (wait_segment):
|
|
* gst-libs/gst/audio/gstringbuffer.h:
|
|
Sun's Forte compiler doesn't seem to like anonymous structs,
|
|
so use same setup as in GstBaseSrc (fixes #324900).
|
|
|
|
2005-12-30 14:54:06 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
move old example to tests/examples/volume/volune.c
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* gst/volume/Makefile.am:
|
|
* gst/volume/demo.c:
|
|
move old example to tests/examples/volume/volune.c
|
|
* tests/examples/Makefile.am:
|
|
* tests/examples/seek/seek.c: (main):
|
|
change window-close event from "delete-event" to "destroy"
|
|
* tests/examples/volume/Makefile.am:
|
|
* tests/examples/volume/volume.c: (value_changed_callback),
|
|
(setup_gui), (message_received), (eos_message_received), (main):
|
|
fix event handling and bus usage
|
|
|
|
2005-12-29 20:37:23 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
gst/audiotestsrc/gstaudiotestsrc.*: update to basesrc changes, implement segmented seeking and eos handling, add a 's...
|
|
Original commit message from CVS:
|
|
* gst/audiotestsrc/gstaudiotestsrc.c:
|
|
(gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init),
|
|
(gst_audio_test_src_init), (gst_audio_test_src_src_fixate),
|
|
(gst_audio_test_src_query), (gst_audio_test_src_create_sine),
|
|
(gst_audio_test_src_create_square),
|
|
(gst_audio_test_src_create_saw),
|
|
(gst_audio_test_src_create_triangle),
|
|
(gst_audio_test_src_create_silence),
|
|
(gst_audio_test_src_create_white_noise),
|
|
(gst_audio_test_src_create_pink_noise),
|
|
(gst_audio_test_src_init_sine_table),
|
|
(gst_audio_test_src_create_sine_table),
|
|
(gst_audio_test_src_change_wave),
|
|
(gst_audio_test_src_change_volume), (gst_audio_test_src_do_seek),
|
|
(gst_audio_test_src_create), (gst_audio_test_src_set_property):
|
|
* gst/audiotestsrc/gstaudiotestsrc.h:
|
|
update to basesrc changes, implement segmented seeking and eos handling,
|
|
add a 'sine-tab' waveform for performance critical playback
|
|
|
|
2005-12-29 16:17:55 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
po/POTFILES.in: ... and this time the other modified file that I missed last time.
|
|
Original commit message from CVS:
|
|
* po/POTFILES.in:
|
|
... and this time the other modified file that I missed last time.
|
|
|
|
2005-12-29 14:31:49 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/playback/gstdecodebin.c: Fix non-C89 variable declaration not at the start of a block. Should help some compilers.
|
|
Original commit message from CVS:
|
|
* gst/playback/gstdecodebin.c: (new_pad):
|
|
Fix non-C89 variable declaration not at the start of a block. Should
|
|
help some compilers.
|
|
|
|
2005-12-29 12:43:22 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/Makefile.am: And now fix 'make distcheck' (builddir != srcdir)
|
|
Original commit message from CVS:
|
|
* tests/check/Makefile.am:
|
|
And now fix 'make distcheck' (builddir != srcdir)
|
|
|
|
2005-12-29 12:22:24 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
New cdparanoiasrc element based on cddabasesrc; enable cdparanoia plugin again (there are still fixes required to pla...
|
|
Original commit message from CVS:
|
|
* configure.ac:
|
|
* ext/cdparanoia/Makefile.am:
|
|
* ext/cdparanoia/gstcdparanoia.c:
|
|
* ext/cdparanoia/gstcdparanoia.h:
|
|
* ext/cdparanoia/gstcdparanoiasrc.c:
|
|
(gst_cd_paranoia_mode_get_type), (gst_cd_paranoia_src_base_init),
|
|
(gst_cd_paranoia_src_init), (gst_cd_paranoia_src_class_init),
|
|
(gst_cd_paranoia_src_open), (gst_cd_paranoia_src_close),
|
|
(gst_cd_paranoia_paranoia_callback),
|
|
(gst_cd_paranoia_src_read_sector), (gst_cd_paranoia_src_finalize),
|
|
(gst_cd_paranoia_src_set_property),
|
|
(gst_cd_paranoia_src_get_property), (plugin_init):
|
|
* ext/cdparanoia/gstcdparanoiasrc.h:
|
|
New cdparanoiasrc element based on cddabasesrc; enable cdparanoia
|
|
plugin again (there are still fixes required to playbin to make
|
|
cdda:// uris work there).
|
|
|
|
2005-12-29 12:13:57 +0000 Tim-Philipp Müller <tim@centricular.net>
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tests/check/Makefile.am: Fix test case compilation.
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Original commit message from CVS:
|
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* tests/check/Makefile.am:
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Fix test case compilation.
|
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|
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2005-12-29 11:49:11 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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gst-libs/gst/cdda/gstcddabasesrc.c: An integer is not a string. Fix access to uninitialised variable.
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Original commit message from CVS:
|
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* gst-libs/gst/cdda/gstcddabasesrc.c:
|
|
(gst_cdda_base_src_update_duration),
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(gst_cdda_base_src_calculate_cddb_id):
|
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An integer is not a string. Fix access to uninitialised variable.
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* tests/check/Makefile.am:
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Add cddabasesrc unit test; also actually enable the vorbis test.
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* tests/check/generic/states.c:
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Blacklist new cd audio elements as well.
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* tests/check/libs/cddabasesrc.c:
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Unit test for GstCddaBaseSrc (discid calculation mostly).
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|
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2005-12-28 18:19:25 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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docs/libs/: Add docs for libgstcdda/GstCddaBaseSrc.
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Original commit message from CVS:
|
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* docs/libs/Makefile.am:
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* docs/libs/gst-plugins-base-libs-docs.sgml:
|
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* docs/libs/gst-plugins-base-libs-sections.txt:
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* docs/libs/gst-plugins-base-libs.types:
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Add docs for libgstcdda/GstCddaBaseSrc.
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* gst-libs/gst/interfaces/mixertrack.h:
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Do one struct member per line with a semicolon at the end, that way
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even gtk-doc might parse it without complaining.
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2005-12-28 18:06:50 +0000 Tim-Philipp Müller <tim@centricular.net>
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Add new libgstcdda with GstCddaBaseSrc class.
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Original commit message from CVS:
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* configure.ac:
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* gst-libs/gst/Makefile.am:
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* gst-libs/gst/cdda/Makefile.am:
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* gst-libs/gst/cdda/base64.c:
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* gst-libs/gst/cdda/base64.h:
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* gst-libs/gst/cdda/gstcddabasesrc.c:
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(gst_cdda_base_src_mode_get_type), (gst_cdda_base_src_base_init),
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(gst_cdda_base_src_class_init), (gst_cdda_base_src_init),
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(gst_cdda_base_src_finalize), (gst_cdda_base_src_set_property),
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(gst_cdda_base_src_get_property),
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(gst_cdda_base_src_get_track_from_sector),
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(gst_cdda_base_src_get_query_types), (gst_cdda_base_src_convert),
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(gst_cdda_base_src_query), (gst_cdda_base_src_is_seekable),
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(gst_cdda_base_src_do_seek), (gst_cdda_base_src_handle_track_seek),
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(gst_cdda_base_src_handle_event), (gst_cdda_base_src_uri_get_type),
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(gst_cdda_base_src_uri_get_protocols),
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(gst_cdda_base_src_uri_get_uri), (gst_cdda_base_src_uri_set_uri),
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(gst_cdda_base_src_uri_handler_init),
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(gst_cdda_base_src_setup_interfaces),
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(gst_cdda_base_src_add_track), (gst_cdda_base_src_update_duration),
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(cddb_sum), (gst_cddabasesrc_calculate_musicbrainz_discid),
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(lba_to_msf), (gst_cdda_base_src_calculate_cddb_id),
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(gst_cdda_base_src_add_tags),
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(gst_cdda_base_src_add_index_associations),
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(gst_cdda_base_src_set_index), (gst_cdda_base_src_get_index),
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(gst_cdda_base_src_track_sort_func), (gst_cdda_base_src_start),
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(gst_cdda_base_src_clear_tracks), (gst_cdda_base_src_stop),
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(gst_cdda_base_src_create):
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* gst-libs/gst/cdda/gstcddabasesrc.h:
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* gst-libs/gst/cdda/sha1.c:
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* gst-libs/gst/cdda/sha1.h:
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Add new libgstcdda with GstCddaBaseSrc class.
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|
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2005-12-28 14:59:41 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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ext/gnomevfs/gstgnomevfssink.h: Use GstBaseSinkClass as parent_class member for class struct, not
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|
Original commit message from CVS:
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* ext/gnomevfs/gstgnomevfssink.h:
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Use GstBaseSinkClass as parent_class member for class struct, not
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GstBaseSink.
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2005-12-27 22:29:43 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/videotestsrc/gstvideotestsrc.c: Add start method to reset running time and number of frames sent
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|
Original commit message from CVS:
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* gst/videotestsrc/gstvideotestsrc.c:
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|
(gst_video_test_src_class_init), (gst_video_test_src_start):
|
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Add start method to reset running time and number of frames sent
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when starting up (fixes #324696; patch by: Michal Benes).
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|
|
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2005-12-27 21:58:28 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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docs/plugins/: Add docs stuff for gnomevfssrc and gnomevfssink.
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|
Original commit message from CVS:
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* docs/plugins/Makefile.am:
|
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* docs/plugins/gst-plugins-base-plugins-docs.sgml:
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* docs/plugins/gst-plugins-base-plugins-sections.txt:
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* docs/plugins/gst-plugins-base-plugins.args:
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* docs/plugins/gst-plugins-base-plugins.hierarchy:
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* docs/plugins/gst-plugins-base-plugins.signals:
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Add docs stuff for gnomevfssrc and gnomevfssink.
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* ext/gnomevfs/gstgnomevfssrc.c:
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Fix example pipeline in gtk-doc blurb.
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2005-12-27 21:42:23 +0000 Tim-Philipp Müller <tim@centricular.net>
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|
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ext/gnomevfs/: Port gnomevfssink; add gtk-doc blurb.
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|
Original commit message from CVS:
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* ext/gnomevfs/Makefile.am:
|
|
* ext/gnomevfs/gstgnomevfs.c: (gst_gnome_vfs_uri_get_type),
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|
(gst_gnome_vfs_handle_copy), (gst_gnome_vfs_handle_free),
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|
(gst_gnome_vfs_handle_get_type), (plugin_init):
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* ext/gnomevfs/gstgnomevfs.h:
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* ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_do_init),
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|
(gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init),
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|
(gst_gnome_vfs_sink_finalize), (gst_gnome_vfs_sink_init),
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(gst_gnome_vfs_sink_set_property),
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(gst_gnome_vfs_sink_get_property), (gst_gnome_vfs_sink_open_file),
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|
(gst_gnome_vfs_sink_close_file), (gst_gnome_vfs_sink_start),
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|
(gst_gnome_vfs_sink_stop), (gst_gnome_vfs_sink_handle_event),
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(gst_gnome_vfs_sink_query), (gst_gnome_vfs_sink_render),
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(gst_gnome_vfs_sink_uri_get_type),
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|
(gst_gnome_vfs_sink_uri_get_protocols),
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|
(gst_gnome_vfs_sink_uri_get_uri), (gst_gnome_vfs_sink_uri_set_uri),
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(gst_gnome_vfs_sink_uri_handler_init):
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* ext/gnomevfs/gstgnomevfssink.h:
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Port gnomevfssink; add gtk-doc blurb.
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* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_get_type),
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(gst_gnome_vfs_src_base_init), (gst_gnome_vfs_src_class_init),
|
|
(gst_gnome_vfs_src_init), (gst_gnome_vfs_src_finalize),
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|
(gst_gnome_vfs_src_uri_get_type),
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|
(gst_gnome_vfs_src_uri_get_protocols),
|
|
(gst_gnome_vfs_src_uri_get_uri), (gst_gnome_vfs_src_uri_set_uri),
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|
(gst_gnome_vfs_src_uri_handler_init),
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|
(gst_gnome_vfs_src_set_property), (gst_gnome_vfs_src_get_property),
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(gst_gnome_vfs_src_unicodify), (audiocast_thread_run),
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(gst_gnome_vfs_src_send_additional_headers_callback),
|
|
(gst_gnome_vfs_src_received_headers_callback),
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(gst_gnome_vfs_src_push_callbacks),
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|
(gst_gnome_vfs_src_pop_callbacks),
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|
(gst_gnome_vfs_src_get_icy_metadata), (gst_gnome_vfs_src_create),
|
|
(gst_gnome_vfs_src_is_seekable), (gst_gnome_vfs_src_get_size),
|
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(gst_gnome_vfs_src_start), (gst_gnome_vfs_src_stop):
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* ext/gnomevfs/gstgnomevfssrc.h:
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s/gst_gnomevfssrc/gst_gnome_vfs_src/; move header stuff to header
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file; add gtk-doc blurb with example pipelines.
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2005-12-23 18:16:22 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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|
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* configure.ac:
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|
back to HEAD
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|
Original commit message from CVS:
|
|
back to HEAD
|
|
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|
=== release 0.10.1 ===
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|
|
2005-12-23 18:08:39 +0000 Thomas Vander Stichele <thomas@apestaart.org>
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|
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* ChangeLog:
|
|
* NEWS:
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|
* RELEASE:
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|
* configure.ac:
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|
* docs/libs/tmpl/gstcolorbalance.sgml:
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|
* docs/plugins/gst-plugins-base-plugins.args:
|
|
* docs/plugins/gst-plugins-base-plugins.signals:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.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-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.xml:
|
|
* docs/plugins/inspect/plugin-libvisual.xml:
|
|
* docs/plugins/inspect/plugin-ogg.xml:
|
|
* docs/plugins/inspect/plugin-pango.xml:
|
|
* docs/plugins/inspect/plugin-playbin.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-video4linux.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:
|
|
releasing 0.10.1
|
|
Original commit message from CVS:
|
|
releasing 0.10.1
|
|
|
|
2005-12-21 20:59:52 +0000 Edgard Lima <edgard.lima@indt.org.br>
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|
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|
* ChangeLog:
|
|
* gst/typefind/gsttypefindfunctions.c:
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|
iLBC30 and iLBC20 added to typefind.
|
|
Original commit message from CVS:
|
|
iLBC30 and iLBC20 added to typefind.
|
|
|
|
2005-12-20 15:57:06 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* ChangeLog:
|
|
* configure.ac:
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|
* docs/libs/tmpl/gstcolorbalance.sgml:
|
|
* po/af.po:
|
|
* po/az.po:
|
|
* po/cs.po:
|
|
* po/en_GB.po:
|
|
* po/hu.po:
|
|
* po/it.po:
|
|
* po/nb.po:
|
|
* po/nl.po:
|
|
* po/or.po:
|
|
* po/sq.po:
|
|
* po/sr.po:
|
|
* po/sv.po:
|
|
* po/uk.po:
|
|
* po/vi.po:
|
|
prereleasing
|
|
Original commit message from CVS:
|
|
prereleasing
|
|
|
|
2005-12-20 12:24:29 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
stop making fun of older compilers
|
|
Original commit message from CVS:
|
|
stop making fun of older compilers
|
|
|
|
2005-12-20 12:00:26 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
gst-libs/gst/audio/: update strings, values are in microseconds change the default sink buffer time to something that...
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
(gst_base_audio_sink_class_init):
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
(gst_base_audio_src_class_init):
|
|
update strings, values are in microseconds
|
|
change the default sink buffer time to something that is smaller
|
|
(to help software volume mixing have a slightly lower delay) but
|
|
still be acceptable on Wim's laptop
|
|
|
|
2005-12-20 10:13:05 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
gst-libs/gst/riff/riff-media.c: Made a quack, forgot to add DUCK to the riff video template.
|
|
Original commit message from CVS:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_template_caps):
|
|
Made a quack, forgot to add DUCK to the riff video template.
|
|
|
|
2005-12-19 15:00:38 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
ext/ogg/gstogmparse.c: Make sure pads are initialized correctly.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstogmparse.c: (gst_ogm_text_parse_base_init),
|
|
(gst_ogm_parse_init), (gst_ogm_audio_parse_init),
|
|
(gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
|
|
(gst_ogm_parse_chain):
|
|
Make sure pads are initialized correctly.
|
|
* gst-libs/gst/riff/riff-ids.h:
|
|
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
|
|
(gst_riff_create_video_template_caps):
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Add a whole bunch of FOURCC <=> MimeType.
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Extend the riff video pad template to support the newly added fourcc.
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|
|
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2005-12-18 15:04:21 +0000 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
ext/ogg/gstoggdemux.c: Extra debug output when activating/deactivating chains.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_deactivate_current_chain),
|
|
(gst_ogg_demux_activate_chain):
|
|
Extra debug output when activating/deactivating chains.
|
|
* gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter),
|
|
(is_demuxer_element), (try_to_link_1), (remove_element_chain),
|
|
(unlinked):
|
|
Remove a queue from our list when it becomes unlinked.
|
|
Don't add queues to elements in class 'Demux' if they
|
|
can only produce one pad
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|
|
|
2005-12-18 10:54:33 +0000 Julien Moutte <julien@moutte.net>
|
|
|
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gst-libs/gst/video/gstvideosink.c: Add a debug category.
|
|
Original commit message from CVS:
|
|
2005-12-18 Julien MOUTTE <julien@moutte.net>
|
|
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_base_init),
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|
(gst_video_sink_get_type): Add a debug category.
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|
|
|
2005-12-18 00:56:07 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
|
|
|
|
gst-libs/gst/rtp/gstbasertpdepayload.c: Handle downstream newsegment by sending our own newsegment before the next bu...
|
|
Original commit message from CVS:
|
|
2005-12-17 Philippe Khalaf <burger@speedy.org>
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_init), (gst_base_rtp_depayload_handle_sink_event):
|
|
Handle downstream newsegment by sending our own newsegment before the
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|
next buffer to be released. (#323900)
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|
|
|
2005-12-18 00:41:10 +0000 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
|
|
|
|
gst-libs/gst/rtp/gstbasertpdepayload.c: add queue delay to new segment as well (as opposed to just the first buffer)....
|
|
Original commit message from CVS:
|
|
2005-12-17 Philippe Khalaf <burger@speedy.org>
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.c:
|
|
(gst_base_rtp_depayload_set_gst_timestamp):
|
|
add queue delay to new segment as well (as opposed to just the first
|
|
buffer). (bug #322347)
|
|
|
|
2005-12-16 22:00:07 +0000 Stefan Kost <ensonic@users.sourceforge.net>
|
|
|
|
ext/libvisual/visual.c: change some char* into char[]
|
|
Original commit message from CVS:
|
|
* ext/libvisual/visual.c: (make_valid_name):
|
|
change some char* into char[]
|
|
* gst/audiotestsrc/gstaudiotestsrc.c:
|
|
(gst_audio_test_src_class_init), (gst_audio_test_src_do_seek),
|
|
(gst_audio_test_src_create):
|
|
* gst/audiotestsrc/gstaudiotestsrc.h:
|
|
prepare to handle EOS and SEGMENT_DONE
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|
|
|
2005-12-16 12:32:37 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
tests/check/generic/states.c: Blacklist cdparanoia element in state test.
|
|
Original commit message from CVS:
|
|
* tests/check/generic/states.c: (GST_START_TEST):
|
|
Blacklist cdparanoia element in state test.
|
|
|
|
2005-12-16 11:25:51 +0000 Benjamin Pineau <ben.pineau@gmail.com>
|
|
|
|
gst/tcp/: Add <string.h> includes for memset and FD_ZERO (fixes #323878;
|
|
Original commit message from CVS:
|
|
* gst/tcp/gsttcp.c:
|
|
* gst/tcp/gsttcpclientsink.c:
|
|
* gst/tcp/gsttcpserversink.c:
|
|
* gst/tcp/gsttcpserversrc.c:
|
|
Add <string.h> includes for memset and FD_ZERO (fixes #323878;
|
|
patch by: Benjamin Pineau).
|
|
|
|
2005-12-15 14:43:38 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/videorate/gstvideorate.c: Fix timestamping for videorate when the first buffer it sees has a non-zero timestamp. ...
|
|
Original commit message from CVS:
|
|
* gst/videorate/gstvideorate.c: (gst_video_rate_blank_data),
|
|
(gst_video_rate_chain):
|
|
Fix timestamping for videorate when the first buffer it sees has a
|
|
non-zero timestamp. Fix some misleading debug output.
|
|
|
|
2005-12-15 10:30:14 +0000 Michael Smith <msmith@xiph.org>
|
|
|
|
gst/audioresample/gstaudioresample.c: Don't leak all input buffers to audioresample.
|
|
Original commit message from CVS:
|
|
* gst/audioresample/gstaudioresample.c:
|
|
Don't leak all input buffers to audioresample.
|
|
|
|
2005-12-15 10:15:10 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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ext/pango/gsttextoverlay.c: Don't operate on empty text buffers. Strip newlines and tabs only from the end of the tex...
|
|
Original commit message from CVS:
|
|
* ext/pango/gsttextoverlay.c: (gst_text_overlay_collected):
|
|
Don't operate on empty text buffers. Strip newlines and
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tabs only from the end of the text, but leave them intact
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in the middle. Fix typo in gtk-doc description.
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2005-12-15 09:48:19 +0000 Tim-Philipp Müller <tim@centricular.net>
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gst/playback/: Make sure the video frame buffer we return to apps via the "frame" property always has caps set on it....
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|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c:
|
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* gst/playback/gstplaybin.c: (handoff):
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Make sure the video frame buffer we return to apps via the
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"frame" property always has caps set on it. Modify
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_gst_gvalue_set_object() macro to handle NULL objects
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gracefully too.
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2005-12-14 20:42:11 +0000 Stefan Kost <ensonic@users.sourceforge.net>
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gst/audiotestsrc/gstaudiotestsrc.*: Adjust to some recent api changes and add wtays new cool seeking capabillities
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|
Original commit message from CVS:
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* gst/audiotestsrc/gstaudiotestsrc.c:
|
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(gst_audio_test_src_class_init), (gst_audio_test_src_init),
|
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(gst_audio_test_src_setcaps), (gst_audio_test_src_src_query),
|
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(gst_audio_test_src_do_seek), (gst_audio_test_src_is_seekable),
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(gst_audio_test_src_create):
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* gst/audiotestsrc/gstaudiotestsrc.h:
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Adjust to some recent api changes and add wtays new cool seeking
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capabillities
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2005-12-14 17:58:48 +0000 Tim-Philipp Müller <tim@centricular.net>
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ext/alsa/: Helper functions to add device probing via the GstPropertyProbe interface to a class.
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|
Original commit message from CVS:
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|
* ext/alsa/Makefile.am:
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* ext/alsa/gstalsadeviceprobe.c:
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* ext/alsa/gstalsadeviceprobe.h:
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Helper functions to add device probing via the GstPropertyProbe
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interface to a class.
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* ext/alsa/gstalsamixer.h:
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Comment out GST_ALSA_MIXER, it returns a struct that's not
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used.
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* ext/alsa/gstalsamixer.c: (gst_alsa_mixer_open):
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Add some debug info.
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* ext/alsa/gstalsamixerelement.c:
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(gst_alsa_mixer_element_interface_supported),
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(gst_implements_interface_init),
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(gst_alsa_mixer_element_init_interfaces),
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(gst_alsa_mixer_element_class_init),
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(gst_alsa_mixer_element_finalize), (gst_alsa_mixer_element_init),
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(gst_alsa_mixer_element_set_property),
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(gst_alsa_mixer_element_get_property),
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(gst_alsa_mixer_element_change_state):
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* ext/alsa/gstalsamixerelement.h:
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Add 'device' and 'device-name' properties. Add GstPropertyProbe
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for device handling (gnome-volume-control will need that).
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2005-12-12 20:31:24 +0000 Christian Schaller <uraeus@gnome.org>
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* ChangeLog:
|
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* ext/Makefile.am:
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* gst-plugins-base.spec.in:
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updates to activate cdparanoia plugin
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|
Original commit message from CVS:
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updates to activate cdparanoia plugin
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|
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2005-12-12 19:13:09 +0000 Michael Smith <msmith@xiph.org>
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ext/ogg/gstoggdemux.c: Use the correct function to free list of typefind factories.
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Original commit message from CVS:
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* ext/ogg/gstoggdemux.c: (gst_ogg_type_find):
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Use the correct function to free list of typefind factories.
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|
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2005-12-12 15:09:55 +0000 Wim Taymans <wim.taymans@gmail.com>
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gst/videotestsrc/gstvideotestsrc.*: Implement seeking in videotestsrc.
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|
Original commit message from CVS:
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* gst/videotestsrc/gstvideotestsrc.c:
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(gst_video_test_src_class_init), (gst_video_test_src_init),
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(gst_video_test_src_parse_caps), (gst_video_test_src_query),
|
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(gst_video_test_src_do_seek), (gst_video_test_src_is_seekable),
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(gst_video_test_src_create):
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* gst/videotestsrc/gstvideotestsrc.h:
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Implement seeking in videotestsrc.
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Small cleanups.
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2005-12-12 15:06:46 +0000 Wim Taymans <wim.taymans@gmail.com>
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ext/cdparanoia/: Partially ported cdparanoia now that basesrc can support a plugin like this..
|
|
Original commit message from CVS:
|
|
* ext/cdparanoia/Makefile.am:
|
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* ext/cdparanoia/gstcdparanoia.c: (gst_paranoia_mode_get_type),
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(gst_paranoia_endian_get_type), (_do_init),
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(cdparanoia_class_init), (cdparanoia_init),
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(cdparanoia_set_property), (cdparanoia_get_property),
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(cdparanoia_do_seek), (cdparanoia_is_seekable),
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(cdparanoia_create), (cdparanoia_start), (cdparanoia_stop),
|
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(cdparanoia_convert), (cdparanoia_get_query_types),
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(cdparanoia_query), (cdparanoia_set_index),
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(cdparanoia_uri_set_uri):
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* ext/cdparanoia/gstcdparanoia.h:
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Partially ported cdparanoia now that basesrc can support a
|
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plugin like this..
|
|
|
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2005-12-12 13:03:50 +0000 Wim Taymans <wim.taymans@gmail.com>
|
|
|
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tests/examples/seek/scrubby.c: Set higher priority for bus events so they don't get reordered with gtk gui events.
|
|
Original commit message from CVS:
|
|
* tests/examples/seek/scrubby.c: (main):
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Set higher priority for bus events so they don't get reordered with
|
|
gtk gui events.
|
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* tests/examples/seek/seek.c: (do_seek), (start_seek), (stop_seek),
|
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(flush_toggle_cb), (main):
|
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Added checkbox do disable flushing seeks.
|
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Disable scrubbing when doing non flushing seeks.
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|
|
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2005-12-12 09:52:37 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
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gst/subparse/gstsubparse.c: Implement some sort of event handling that doesn't rely on g_return_if_fail; make sure we...
|
|
Original commit message from CVS:
|
|
* gst/subparse/gstsubparse.c: (gst_sub_parse_init),
|
|
(gst_sub_parse_do_seek), (gst_sub_parse_src_event), (parse_subrip),
|
|
(parser_state_init), (handle_buffer), (gst_sub_parse_chain),
|
|
(gst_sub_parse_sink_event), (gst_sub_parse_change_state):
|
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Implement some sort of event handling that doesn't rely on
|
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g_return_if_fail; make sure we always push the last chunk of an
|
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.srt out when we receive an EOS; use gst_pad_alloc_buffer; fix
|
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state change function; remove some old cruft. Seeking is still
|
|
rather unlikely to work though.
|
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* tools/.cvsignore:
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|
Ignore more.
|
|
|
|
2005-12-11 22:26:08 +0000 Julien Moutte <julien@moutte.net>
|
|
|
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sys/xvimage/xvimagesink.c: Fixed a leak of the current image reference when cleaning up.
|
|
Original commit message from CVS:
|
|
2005-12-11 Julien MOUTTE <julien@moutte.net>
|
|
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_change_state):
|
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Fixed a leak of the current image reference when cleaning up.
|
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Thanks to Arwed von Merkatz (alley_cat) for pointing it out.
|
|
|
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2005-12-09 10:23:42 +0000 Michael Smith <msmith@xiph.org>
|
|
|
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tools/: Remove gst-launch-ext. It doesn't work, and is no longer particularly useful.
|
|
Original commit message from CVS:
|
|
* tools/Makefile.am:
|
|
* tools/gst-launch-ext-m.m:
|
|
Remove gst-launch-ext. It doesn't work, and is no longer
|
|
particularly useful.
|
|
|
|
2005-12-08 18:53:57 +0000 Luca Ognibene <luogni@tin.it>
|
|
|
|
ext/ogg/gstogmparse.c: don't pass random values to ogmparse convert function.
|
|
Original commit message from CVS:
|
|
* ext/ogg/gstogmparse.c: (gst_ogm_parse_sink_query):
|
|
don't pass random values to ogmparse convert function.
|
|
Make seeking possible in the exile1.ogm file.
|
|
|
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2005-12-07 18:51:35 +0000 Tim-Philipp Müller <tim@centricular.net>
|
|
|
|
gst/playback/: Work around refcount problem with g_value_set_object() that occur if the core has been compiled agains...
|
|
Original commit message from CVS:
|
|
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_get_property):
|
|
* gst/playback/gstplaybin.c: (gst_play_bin_get_property):
|
|
Work around refcount problem with g_value_set_object() that occur
|
|
if the core has been compiled against GLib-2.6 (g_value_set_object()
|
|
will only g_object_ref() the element, but the caller will
|
|
gst_object_unref() it and bad things will happen due to the way
|
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GstObjects are refcounted in the GLib-2.6 case). Fixes problems with
|
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totem for people on FC4 using Thomas's 0.10 RPMs.
|
|
|
|
2005-12-07 11:34:37 +0000 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
Time to welcome ogm to 0.10 :)
|
|
Original commit message from CVS:
|
|
Time to welcome ogm to 0.10 :)
|
|
* ext/ogg/gstoggdemux.c: (internal_element_pad_added_cb),
|
|
(gst_ogg_pad_typefind):
|
|
Oggdemux can now properly typefind elements with dynamic pads.
|
|
* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
|
|
Properly set caps on src pad, and set caps on outgoing buffers.
|
|
|
|
2005-12-06 19:42:02 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* ChangeLog:
|
|
* ext/alsa/gstalsamixer.h:
|
|
* ext/alsa/gstalsamixerelement.h:
|
|
* ext/alsa/gstalsamixeroptions.h:
|
|
* ext/alsa/gstalsamixertrack.h:
|
|
* ext/alsa/gstalsasink.c:
|
|
* ext/alsa/gstalsasink.h:
|
|
* ext/alsa/gstalsasrc.c:
|
|
* ext/alsa/gstalsasrc.h:
|
|
* ext/cdparanoia/gstcdparanoia.h:
|
|
* ext/gnomevfs/gstgnomevfsuri.h:
|
|
* ext/ogg/gstoggdemux.c:
|
|
* ext/ogg/gstoggmux.c:
|
|
* ext/pango/gsttextoverlay.h:
|
|
* ext/theora/theoradec.c:
|
|
* ext/theora/theoraenc.c:
|
|
* ext/vorbis/vorbisdec.h:
|
|
* ext/vorbis/vorbisenc.c:
|
|
* ext/vorbis/vorbisenc.h:
|
|
* ext/vorbis/vorbisparse.h:
|
|
* gst-libs/gst/audio/gstaudioclock.h:
|
|
* gst-libs/gst/audio/gstaudiosink.c:
|
|
* gst-libs/gst/audio/gstaudiosink.h:
|
|
* gst-libs/gst/audio/gstaudiosrc.c:
|
|
* gst-libs/gst/audio/gstaudiosrc.h:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.c:
|
|
* gst-libs/gst/audio/gstbaseaudiosink.h:
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.c:
|
|
* gst-libs/gst/audio/gstbaseaudiosrc.h:
|
|
* gst-libs/gst/audio/gstringbuffer.h:
|
|
* gst-libs/gst/audio/multichannel.h:
|
|
* gst-libs/gst/floatcast/floatcast.h:
|
|
* gst-libs/gst/interfaces/colorbalance.c:
|
|
* gst-libs/gst/interfaces/colorbalance.h:
|
|
* gst-libs/gst/interfaces/colorbalancechannel.h:
|
|
* gst-libs/gst/interfaces/mixer.h:
|
|
* gst-libs/gst/interfaces/mixeroptions.h:
|
|
* gst-libs/gst/interfaces/mixertrack.h:
|
|
* gst-libs/gst/interfaces/navigation.h:
|
|
* gst-libs/gst/interfaces/propertyprobe.h:
|
|
* gst-libs/gst/interfaces/tuner.h:
|
|
* gst-libs/gst/interfaces/tunerchannel.h:
|
|
* gst-libs/gst/interfaces/tunernorm.h:
|
|
* gst-libs/gst/interfaces/xoverlay.h:
|
|
* gst-libs/gst/netbuffer/gstnetbuffer.h:
|
|
* gst-libs/gst/riff/riff-ids.h:
|
|
* gst-libs/gst/riff/riff-media.h:
|
|
* gst-libs/gst/riff/riff-read.h:
|
|
* gst-libs/gst/rtp/gstbasertpdepayload.h:
|
|
* gst-libs/gst/rtp/gstbasertppayload.c:
|
|
* gst-libs/gst/rtp/gstbasertppayload.h:
|
|
* gst-libs/gst/rtp/gstrtpbuffer.c:
|
|
* gst-libs/gst/rtp/gstrtpbuffer.h:
|
|
* gst-libs/gst/tag/gsttageditingprivate.h:
|
|
* gst-libs/gst/tag/gstvorbistag.c:
|
|
* gst-libs/gst/tag/tag.h:
|
|
* gst-libs/gst/video/video.h:
|
|
* gst/adder/gstadder.c:
|
|
* gst/adder/gstadder.h:
|
|
* gst/audioconvert/audioconvert.c:
|
|
* gst/audioconvert/audioconvert.h:
|
|
* gst/audioconvert/gstaudioconvert.c:
|
|
* gst/audioconvert/gstchannelmix.c:
|
|
* gst/audioconvert/gstchannelmix.h:
|
|
* gst/audiorate/gstaudiorate.c:
|
|
* gst/audioresample/buffer.h:
|
|
* gst/audioresample/functable.h:
|
|
* gst/audioresample/gstaudioresample.c:
|
|
* gst/audioresample/resample.h:
|
|
* gst/ffmpegcolorspace/avcodec.h:
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
|
|
* gst/ffmpegcolorspace/gstffmpegcodecmap.h:
|
|
* gst/ffmpegcolorspace/imgconvert.c:
|
|
* gst/ffmpegcolorspace/imgconvert_template.h:
|
|
* gst/playback/gstdecodebin.c:
|
|
* gst/playback/gstplaybasebin.h:
|
|
* gst/playback/gstplaybin.c:
|
|
* gst/playback/gststreaminfo.h:
|
|
* gst/tcp/gstfdset.c:
|
|
* gst/tcp/gstfdset.h:
|
|
* gst/tcp/gstmultifdsink.c:
|
|
* gst/tcp/gstmultifdsink.h:
|
|
* gst/tcp/gsttcp.h:
|
|
* gst/tcp/gsttcpclientsrc.c:
|
|
* gst/tcp/gsttcpclientsrc.h:
|
|
* gst/tcp/gsttcpplugin.h:
|
|
* gst/tcp/gsttcpserversink.c:
|
|
* gst/tcp/gsttcpserversrc.c:
|
|
* gst/typefind/gsttypefindfunctions.c:
|
|
* gst/videorate/gstvideorate.c:
|
|
* gst/videotestsrc/gstvideotestsrc.h:
|
|
* gst/videotestsrc/videotestsrc.h:
|
|
* sys/v4l/gstv4lcolorbalance.h:
|
|
* sys/v4l/gstv4ltuner.h:
|
|
* sys/v4l/gstv4lxoverlay.h:
|
|
* sys/v4l/v4l_calls.h:
|
|
* sys/v4l/videodev_mjpeg.h:
|
|
* tests/check/elements/audioconvert.c:
|
|
* tests/check/elements/audioresample.c:
|
|
* tests/check/elements/audiotestsrc.c:
|
|
* tests/check/elements/videotestsrc.c:
|
|
* tests/check/elements/volume.c:
|
|
* tests/examples/seek/scrubby.c:
|
|
* tests/examples/seek/seek.c:
|
|
expand tabs
|
|
Original commit message from CVS:
|
|
expand tabs
|
|
|
|
2005-12-06 19:28:24 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* docs/libs/tmpl/gstaudio.sgml:
|
|
* docs/libs/tmpl/gstcolorbalance.sgml:
|
|
* docs/libs/tmpl/gstgconf.sgml:
|
|
* docs/libs/tmpl/gstmixer.sgml:
|
|
* docs/libs/tmpl/gstringbuffer.sgml:
|
|
* docs/libs/tmpl/gsttuner.sgml:
|
|
* docs/libs/tmpl/gstxoverlay.sgml:
|
|
put back stability level
|
|
Original commit message from CVS:
|
|
put back stability level
|
|
|
|
2005-12-05 18:11:49 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* configure.ac:
|
|
back to HEAD
|
|
Original commit message from CVS:
|
|
back to HEAD
|
|
|
|
=== release 0.10.0 ===
|
|
|
|
2005-12-05 18:02:48 +0000 Thomas Vander Stichele <thomas@apestaart.org>
|
|
|
|
* ChangeLog:
|
|
* NEWS:
|
|
* RELEASE:
|
|
* configure.ac:
|
|
* docs/libs/tmpl/gstcolorbalance.sgml:
|
|
* docs/plugins/inspect/plugin-adder.xml:
|
|
* docs/plugins/inspect/plugin-alsa.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-decodebin.xml:
|
|
* docs/plugins/inspect/plugin-ffmpegcolorspace.xml:
|
|
* docs/plugins/inspect/plugin-gnomevfs.xml:
|
|
* docs/plugins/inspect/plugin-libvisual.xml:
|
|
* docs/plugins/inspect/plugin-ogg.xml:
|
|
* docs/plugins/inspect/plugin-pango.xml:
|
|
* docs/plugins/inspect/plugin-playbin.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-video4linux.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:
|
|
releasing 0.10.0
|
|
Original commit message from CVS:
|
|
releasing 0.10.0
|
|
|