From: Wim Taymans Date: Fri, 17 Feb 2012 10:03:14 +0000 (+0100) Subject: RELEASE 0.11.2 X-Git-Tag: 1.19.3~511^2~6775 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=61c446f2efa0da14baff09cab8d4904ba14989b5;p=platform%2Fupstream%2Fgstreamer.git RELEASE 0.11.2 --- diff --git a/ChangeLog b/ChangeLog index a0b2b85..b4e3bd9 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,9 +1,7374 @@ +=== release 0.11.2 === + +2012-02-17 Wim Taymans + + * configure.ac: + releasing 0.11.2, "Drool Pool" + +2012-02-17 10:06:19 +0100 Wim Taymans + + * win32/common/libgstaudio.def: + defs: update + +2012-02-16 14:23:28 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + gst-libs/gst/audio/gstaudioencoder.c + gst-libs/gst/pbutils/gstdiscoverer.c + +2012-02-16 12:19:20 +0100 Mark Nauwelaerts + + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudiodecoder.h: + audiodecoder: add some properties to tweak baseclass behaviour + ... so subclass can also rely upon never being bothered with some NULL buffer + it can't do any interesting with, or with any data before it received + any format configuration (and setup properly). + +2012-02-16 12:18:03 +0100 Mark Nauwelaerts + + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/audio/gstaudioencoder.h: + audioencoder: add some properties to tweak baseclass behaviour + ... so subclass can also rely upon never being bothered with less data + than it desires or with some NULL buffer it can't do any interesting with. + +2012-02-16 12:15:47 +0100 Mark Nauwelaerts + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: assert some more that subclass parsed frame has proper len + +2012-02-15 13:42:19 +0100 Wim Taymans + + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudiodecoder.h: + audiodecoder: chain up to parent for defaults + Chain up to the parent instead of using the FALSE return value from + the event function (because it's otherwise impossible to return an error). + +2012-02-15 13:32:05 +0100 Wim Taymans + + * ext/vorbis/gstvorbisdec.c: + vorbisdec: remove old code + +2012-01-17 10:54:48 +0100 Olivier Aubert + + * gst/playback/gstplaybin2.c: + docs: fix playbin2 documentation about DVD URIs + and playbin => playbin2 in example pipelines. + https://bugzilla.gnome.org/show_bug.cgi?id=668081 + +2012-02-15 13:03:59 +0100 Wim Taymans + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: call default event handler + Call the default event handler for unknown events. + +2012-02-15 12:29:12 +0100 Wim Taymans + + * gst-libs/gst/tag/gsttagdemux.c: + tagdemux: refactor the tag find function + Move the code to find the tags and to typefind the data into a separate + function. Call this function from the loop function. + +2012-02-15 10:12:55 +0100 Wim Taymans + + * gst-libs/gst/tag/gsttagdemux.c: + tagdemux: don't to data processing in state change + Start a task to perform the pulling and typefind of the tags. + +2012-02-14 19:23:27 +0000 Tim-Philipp Müller + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: try harder to obtain a duration if we don't get one right away + If we don't get a duration right away, set the pipeline to playing + and sleep a bit, then try again. This is ugly, but the least worst + we can do right now. The alternative would be to make parsers etc. + return some bogus duration estimate even after only having pushed + a single frame, for example. + Fixes discoverer showing 0 durations for some mp3 and aac files + (e.g. soweto-adts.aac). + +2012-02-14 13:25:25 +0100 Wim Taymans + + * gst-libs/gst/tag/gsttagdemux.c: + tagdemux: fix src query handler + We don't want to blindly forward all queries. + +2012-02-14 10:50:45 +0100 Wim Taymans + + * tests/check/elements/decodebin.c: + tests: fix after baseparse api change + +2012-01-26 12:31:21 +0000 Vincent Penquerc'h + + * gst/playback/gstsubtitleoverlay.c: + subtitleoverlay: log why an overlay element cannot be used + +2012-01-25 16:02:04 +0000 Vincent Penquerc'h + + * gst/playback/gstplaybin2.c: + playbin2: fix old style raw A/V caps + They're now {audio,video}/x-raw, not {audio,video}/x-raw-* + https://bugzilla.gnome.org/show_bug.cgi?id=668682 + +2012-01-25 15:57:02 +0000 Vincent Penquerc'h + + * gst/playback/gstsubtitleoverlay.c: + subtitleoverlay: fix probing of raw video caps + They're now video/x-raw, not video/x-raw-* anymore. + https://bugzilla.gnome.org/show_bug.cgi?id=668682 + +2012-01-25 14:38:19 +0000 Vincent Penquerc'h + + * gst/playback/gstsubtitleoverlay.c: + subtitleoverlay: add a couple drive by const + https://bugzilla.gnome.org/show_bug.cgi?id=668682 + +2012-02-13 17:07:25 +0100 Wim Taymans + + * gst-libs/gst/video/gstvideometa.c: + videometa: adjust for memory api change + +2012-02-13 15:17:09 +0100 Wim Taymans + + * ext/vorbis/gstvorbisdeclib.h: + vorbis: port to new memory api + +2012-02-13 16:03:15 +0000 Christian Fredrik Kalager Schaller + + * gst-plugins-base.spec.in: + Add new file to spec file + +2012-02-13 16:03:03 +0000 Christian Fredrik Kalager Schaller + + * gst/tcp/Makefile.am: + Add missing header file to build file + +2012-02-12 22:28:31 +0100 Thomas Vander Stichele + + * tests/check/elements/multifdsink.c: + * tests/check/elements/multisocketsink.c: + fix up tests + +2012-02-12 22:04:02 +0100 Thomas Vander Stichele + + * configure.ac: + * gst/tcp/Makefile.am: + * gst/tcp/gsttcpplugin.c: + multifdsink: depends on sys/socket.h + +2012-01-28 18:07:46 +0100 Thomas Vander Stichele + + * gst/tcp/gstmultifdsink.c: + * gst/tcp/gstmultifdsink.h: + * gst/tcp/gstmultihandlesink.c: + * gst/tcp/gstmultihandlesink.h: + * gst/tcp/gstmultisocketsink.c: + * gst/tcp/gstmultisocketsink.h: + * gst/tcp/gsttcpserversink.c: + multihandlesink: finish refactor + +2012-01-28 18:06:02 +0100 Thomas Vander Stichele + + * gst/tcp/gstmultifdsink.c: + * gst/tcp/gstmultifdsink.h: + * gst/tcp/gstmultihandlesink.c: + * gst/tcp/gstmultihandlesink.h: + * gst/tcp/gstmultisocketsink.c: + * gst/tcp/gstmultisocketsink.h: + * tests/check/elements/multifdsink.c: + * tests/check/elements/multisocketsink.c: + multihandle: rename num-fds/-sockets to num-handles + +2012-01-28 11:02:21 +0100 Thomas Vander Stichele + + * gst/tcp/gstmultifdsink.c: + * gst/tcp/gstmultifdsink.h: + * gst/tcp/gstmultihandlesink.h: + * gst/tcp/gstmultisocketsink.c: + * gst/tcp/gstmultisocketsink.h: + multihandlesink: rework to use Handle + +2012-01-28 09:29:55 +0100 Thomas Vander Stichele + + * tests/check/elements/multifdsink.c: + * tests/check/elements/multisocketsink.c: + tests multihandle: verify number of handles + +2012-01-27 21:28:05 +0100 Thomas Vander Stichele + + * gst/tcp/gstmultifdsink.c: + * gst/tcp/gstmultifdsink.h: + * gst/tcp/gstmultihandlesink.h: + * gst/tcp/gstmultisocketsink.c: + * gst/tcp/gstmultisocketsink.h: + * gst/tcp/gsttcpserversink.c: + multihandlesink: introduce Handle union + +2012-01-27 18:44:04 +0100 Thomas Vander Stichele + + * gst/tcp/gstmultifdsink.c: + * gst/tcp/gstmultifdsink.h: + * gst/tcp/gstmultihandlesink.c: + * gst/tcp/gstmultihandlesink.h: + * gst/tcp/gstmultisocketsink.c: + * gst/tcp/gstmultisocketsink.h: + * tests/check/elements/multifdsink.c: + * tests/check/elements/multisocketsink.c: + multihandlesink: rework to use GST_TYPE_FORMAT + +2012-01-27 18:40:30 +0100 Thomas Vander Stichele + + * tests/check/elements/multisocketsink.c: + multisocketsink: fix tests by setting units properly + +2012-01-27 18:33:56 +0100 Thomas Vander Stichele + + * gst/tcp/gstmultifdsink.c: + * tests/check/elements/multifdsink.c: + test: use more literal enums + +2012-01-27 15:46:31 +0100 Thomas Vander Stichele + + * gst/tcp/gstmultifdsink.c: + * gst/tcp/gstmultifdsink.h: + * gst/tcp/gstmultihandlesink.c: + * gst/tcp/gstmultihandlesink.h: + * gst/tcp/gstmultisocketsink.c: + * gst/tcp/gstmultisocketsink.h: + * tests/check/elements/multifdsink.c: + * tests/check/elements/multisocketsink.c: + multihandlesink: further refactoring + +2012-01-27 12:58:12 +0100 Thomas Vander Stichele + + * gst/tcp/gstmultisocketsink.c: + * tests/check/elements/multisocketsink.c: + multisocketsink: fix refcounting bug + +2012-01-26 23:19:33 +0100 Thomas Vander Stichele + + * gst/tcp/gstmultifdsink.c: + * gst/tcp/gstmultifdsink.h: + * gst/tcp/gstmultihandlesink.c: + * gst/tcp/gstmultihandlesink.h: + * gst/tcp/gstmultioutputsink.c: + * gst/tcp/gstmultisocketsink.c: + * gst/tcp/gstmultisocketsink.h: + * gst/tcp/gsttcpserversink.c: + multihandlesink: further refactoring + +2012-01-26 19:34:47 +0100 Thomas Vander Stichele + + * gst/tcp/gstmultifdsink.c: + * gst/tcp/gstmultisocketsink.c: + * tests/check/elements/multisocketsink.c: + multihandlesink: fix one bug in multisocketsink refactoring + +2012-01-26 10:49:37 +0100 Thomas Vander Stichele + + * gst/tcp/Makefile.am: + * gst/tcp/gstmultifdsink.c: + * gst/tcp/gstmultifdsink.h: + * gst/tcp/gstmultihandlesink.c: + * gst/tcp/gstmultihandlesink.h: + * gst/tcp/gstmultisocketsink.c: + * gst/tcp/gstmultisocketsink.h: + multihandlesink: first stab at common base class + +2012-01-26 10:41:22 +0100 Thomas Vander Stichele + + * gst/tcp/gstmultifdsink.c: + * gst/tcp/gstmultifdsink.h: + * gst/tcp/gstmultihandlesink.c: + * gst/tcp/gstmultihandlesink.h: + * gst/tcp/gstmultisocketsink.h: + * gst/tcp/gsttcp-marshal.list: + * gst/tcp/gsttcpplugin.c: + * tests/check/elements/multifdsink.c: + gst/tcp: Factor out common symbols; fix tests. + +2012-01-26 10:08:47 +0100 Thomas Vander Stichele + + * gst/tcp/Makefile.am: + * gst/tcp/gstmultifdsink.c: + * gst/tcp/gstmultifdsink.h: + * tests/check/Makefile.am: + * tests/check/elements/multifdsink.c: + multifdsink: put back multifdsink before refactoring + +2012-01-26 12:30:21 +0100 Thomas Vander Stichele + + * tests/check/Makefile.am: + * tests/check/elements/multisocketsink.c: + multisocketsink: copy over multifdsink unit tests, with FIXME + +2012-02-12 16:54:56 +0000 Tim-Philipp Müller + + * gst-libs/gst/tag/gsttagmux.c: + tag: make GstTagMux base class a bit more functional + We can't use G_DEFINE_*TYPE here because we need the klass in the _init + method to get to the padtemplates. Fixes 'GstTagDemux subclass GstTagDemux + did not set up a {sink,src} pad template' warnings. + +2012-02-10 15:41:36 +0100 Wim Taymans + + * tests/check/elements/videoscale.c: + tests: don't run with unsupported formats + videoconvert does not work with GRAY formats yet so don't try to run the unit + test with it. + +2012-02-10 15:41:19 +0100 Wim Taymans + + * gst/videoconvert/videoconvert.c: + videoconvert: improve error reporting + +2012-02-10 15:41:06 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + ext/vorbis/gstvorbisparse.c + gst-libs/gst/video/video.c + gst/videoscale/gstvideoscale.c + sys/v4l/gstv4lxoverlay.c + sys/v4l/v4l_calls.c + sys/v4l/v4lsrc_calls.c + tests/check/libs/video.c + +2012-02-08 19:39:00 +0000 Tim-Philipp Müller + + * gst/typefind/gsttypefindfunctions.c: + typefindfunctions: make h264 typefinder more picky when returning "likely" probability + Only return LIKELY probability if we've seen an SPS, PPS and an + IDR slice nal, i.e. try harder to avoid false positives such + as with certain VC-1 files. + https://bugzilla.gnome.org/show_bug.cgi?id=668565 + +2012-02-09 16:03:35 +0100 Wim Taymans + + * gst-libs/gst/video/video.c: + video: add performance log for frame copy + +2012-02-09 16:00:59 +0100 Wim Taymans + + * gst/videoconvert/gstvideoconvert.c: + videoconvert: avoid using _CATEGORY_GET + +2012-02-09 15:51:10 +0100 Wim Taymans + + * gst/playback/gstsubtitleoverlay.c: + subtitleoverlay: fix merge error + Fix merge error from commit fb6d09055ae90979682fa4b1c6ee4f4abdaafd8f + +2012-02-09 15:28:54 +0100 Wim Taymans + + * gst-libs/gst/video/video.c: + * gst/videoconvert/gstvideoconvert.c: + * gst/videoscale/gstvideoscale.c: + debug: add some performance debug + +2012-02-08 19:34:57 +0000 Tim-Philipp Müller + + * gst/typefind/gsttypefindfunctions.c: + typefindfunctions: minor cosmetic change + Don't write < 1 when we mean == 0. + +2012-02-08 15:17:49 +0100 Wim Taymans + + * ext/ogg/gstoggmux.c: + * ext/ogg/gstogmparse.c: + * ext/pango/gstbasetextoverlay.c: + * gst-libs/gst/app/gstappsink.c: + * gst-libs/gst/app/gstappsrc.c: + * gst-libs/gst/audio/gstaudiobasesink.c: + * gst-libs/gst/audio/gstaudiobasesrc.c: + * gst/gio/gstgio.c: + * gst/tcp/gstmultisocketsink.c: + * gst/tcp/gsttcpclientsink.c: + * gst/tcp/gsttcpclientsrc.c: + * gst/tcp/gsttcpserversrc.c: + * tests/check/elements/textoverlay.c: + * tests/check/elements/videorate.c: + GST_FLOW_WRONG_STATE -> GST_FLOW_FLUSHING + +2012-02-07 23:42:48 +0000 Tim-Philipp Müller + + * gst-libs/gst/rtsp/Makefile.am: + rtsp: make g-ir-scanner include Gio-2.0 to suppress complaints about GSocket etc. + +2012-02-06 22:09:50 +0100 Mark Nauwelaerts + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: remove stray obsolete declaration + +2012-02-06 22:09:34 +0100 Mark Nauwelaerts + + * gst-libs/gst/audio/audio.c: + audio: correctly fill in fallback channel positions in stereo case + +2012-02-06 18:33:59 +0100 Wim Taymans + + * gst-libs/gst/video/video.c: + video: mark endianness correctly + +2012-02-06 16:08:24 +0100 Wim Taymans + + * gst/volume/gstvolume.c: + volume: use right info structure for setup + +2012-02-06 15:51:17 +0100 Wim Taymans + + * gst-libs/gst/tag/gsttagdemux.c: + tagdemux: push event in the right direction + Push the stored events in the right direction + +2012-02-06 13:49:12 +0000 Tim-Philipp Müller + + * gst-libs/gst/tag/Makefile.am: + tag: fix up define that tells code where to find the license translations too + Tell code about new location of translation dict. + +2012-02-06 13:22:14 +0100 Wim Taymans + + * gst-libs/gst/audio/gstaudiofilter.c: + * gst-libs/gst/audio/gstaudiofilter.h: + audiofilter: configure info after calling vmethod + First call the vmethod and then configure the audioinfo in the baseclass. This + allows subclasses to know about the old format. + +2012-02-06 09:45:10 +0100 Wim Taymans + + * win32/common/libgstaudio.def: + def: update + +2012-02-06 09:44:48 +0100 Wim Taymans + + * m4/Makefile.am: + fix for removed file + +2012-02-03 17:10:12 +0100 Wim Taymans + + * gst-libs/gst/video/gstvideofilter.c: + videofilter: take care of in_place transform + If the subclass doesn't implement a transform_frame function we need to force + the baseclass into in_place transform. + +2012-02-06 11:44:29 +0100 Sebastian Dröge + + * gst-libs/gst/tag/Makefile.am: + tag: Install license translations into $(pkgdatadir)/0.11 + This prevents file conflicts with GStreamer 0.10. + +2012-02-06 10:52:01 +0100 Mark Nauwelaerts + + * gst-libs/gst/video/video.h: + video: add GST_VIDEO_INFO_COMP_BITS + +2012-02-06 09:53:22 +0100 Sebastian Dröge + + * gst-libs/gst/video/video.h: + video: Add GST_VIDEO_INFO_COMP_WIDTH + +2012-02-05 10:56:44 +0000 Tim-Philipp Müller + + * ext/theora/gsttheoraenc.c: + * ext/theora/gsttheoraenc.h: + theoraenc: remove obsolete properties + https://bugzilla.gnome.org/show_bug.cgi?id=669328 + +2012-01-30 08:21:54 -0800 David Schleef + + * gst/videoscale/gstvideoscale.c: + * gst/videoscale/vs_image.c: + * gst/videoscale/vs_image.h: + * gst/videoscale/vs_scanline.c: + * gst/videoscale/vs_scanline.h: + videoscale: Add nearest/linear scaling for NV12 + +2012-01-25 15:49:00 -0800 David Schleef + + * gst/videoscale/gstvideoscale.c: + * gst/videoscale/vs_image.h: + * gst/videoscale/vs_lanczos.c: + videoscale: Add AYUV64 path to Lanczos + +2011-08-30 19:02:51 -0700 David Schleef + + * ext/theora/gsttheoraenc.c: + theoraenc: Use GAP flag when possible + Set TH_ENCCTL_SET_DUPLICATE_FLAG when we see a gap flag, to + indicate to the encoder that the current frame is a duplicate + of the previous frame. + +2012-02-03 15:01:50 +0100 Wim Taymans + + * tests/check/elements/volume.c: + tests: fix volume test + +2012-02-03 12:53:49 +0100 Wim Taymans + + * tests/check/elements/videotestsrc.c: + tests: video testsrc unit test + +2012-02-03 12:41:10 +0100 Wim Taymans + + * tests/check/elements/videorate.c: + * tests/check/elements/videoscale.c: + tests: fix more unit tests + +2012-02-03 12:09:34 +0100 Wim Taymans + + * tests/check/elements/textoverlay.c: + tests: don't set NULL caps + +2012-02-03 11:38:55 +0100 Wim Taymans + + * tests/check/elements/gdpdepay.c: + * tests/check/elements/gdppay.c: + gdp: fixup unit tests + +2012-02-03 11:38:15 +0100 Wim Taymans + + * gst/gdp/gstgdppay.c: + gdppay: fixup for changed caps + Try to send the streamheader after the first buffer. + +2012-02-03 11:37:21 +0100 Wim Taymans + + * gst/gdp/dataprotocol.c: + dataprotocol: don't define default Category + Since we now include this into the unit tests directly, don't define the default + category macro because it conflicts with check. + +2012-02-03 10:47:22 +0100 Wim Taymans + + * tests/check/elements/audioresample.c: + tests: fix audioresample test + +2012-02-03 09:57:21 +0100 Wim Taymans + + * tests/check/elements/audiorate.c: + tests: fix audiorate test + We need to add the layout to the audio caps. + +2012-02-03 09:56:56 +0100 Wim Taymans + + * gst/audiorate/gstaudiorate.c: + audiorate: use default event handler + Use the default event handler for unknown events. + +2012-02-03 09:48:22 +0100 Wim Taymans + + * gst-libs/gst/audio/gstaudioencoder.c: + audioencoder: don't unref caps parameter + Fix refcounting on incomming caps to make sure we don't unref it too much. + +2012-01-07 23:09:23 -0500 Ryan Lortie + + * autogen.sh: + autogen.sh: allow calling from out-of-tree + https://bugzilla.gnome.org/show_bug.cgi?id=667665 + +2012-02-02 16:10:45 +0000 Christian Fredrik Kalager Schaller + + * gst-plugins-base.spec.in: + Update spec file + +2012-02-01 15:28:45 +0000 Vincent Penquerc'h + + * ext/ogg/gstoggdemux.c: + * ext/ogg/gstoggstream.c: + * ext/ogg/gstoggstream.h: + oggdemux: fix granpos interpolation violating max keyframe distance + In case many packets fit on a page, we may not see a granpos for + a while, and granpos interpolation can wrap the 'frames since last + keyframe' part of the granpos, generating a granpos which is smaller + than what it should be. + This is fixed by detecting keyframe packets (at least for Theora), + and updating the last keyframe granpos from this. + This may still be generating potentially wrong granpos for streams + which have a Theora like granpos (keyframes, a max keyframe distance + and a count of frames since last keyframe), and which allow implicit + granules on packets. For these streams, a custom keyframe detection + routine should be plugged into their GstOggStream mapper. + https://bugzilla.gnome.org/show_bug.cgi?id=669164 + +2012-02-02 12:14:15 +0100 Wim Taymans + + * gst/playback/gstplaysinkconvertbin.c: + playsink: call the right default query handler + We need to call the default query handler of the proxy pad because only that one + will forward the query to the target pad in case of the allocation query. + +2012-02-02 01:35:21 +0000 Tim-Philipp Müller + + * gst/subparse/gstsubparse.c: + * gst/typefind/gsttypefindfunctions.c: + typefindfunctions, subparse: fix for gst_type_find_register() API change + +2012-02-01 19:26:29 +0000 Tim-Philipp Müller + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: mark GError argument of "discovered" signal with STATIC_SCOPE + So the error is passed to the callback as is without a copy being made. + +2012-02-01 16:46:13 +0000 Vincent Penquerc'h + + * ext/vorbis/gstvorbisparse.c: + vorbisparse: pedantically recognize undefined headers too + +2012-02-01 16:32:24 +0000 Vincent Penquerc'h + + * ext/vorbis/gstvorbisparse.c: + vorbisparse: fix header detection + It was matching non header packets. + This fixes various leaks, where buffers would be pushed onto a headers + list, but never popped. + Might also fix corruption as those buffers were dropped from the output + silently... + https://bugzilla.gnome.org/show_bug.cgi?id=669167 + +2012-01-29 00:21:19 +0000 Tim-Philipp Müller + + * gst/playback/gstdecodebin2.c: + * gst/playback/gstplaybin2.c: + * gst/playback/gsturidecodebin.c: + playback: suppress GValueArray deprecation warnings for the time being + until this gets sorted out and we have a viable alternative. + https://bugzilla.gnome.org/show_bug.cgi?id=667228 + +2012-02-01 16:33:30 +0100 Sebastian Dröge + + * gst-libs/gst/audio/gstaudioencoder.c: + audioencoder: gst_pad_get_pad_template_caps() now returns a new reference, don't forget to unref + +2012-02-01 16:32:53 +0100 Sebastian Dröge + + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudioencoder.c: + audio{enc,dec}oder: Check if srcpad caps are a subset of the template caps + +2012-02-01 16:04:03 +0100 Sebastian Dröge + + * ext/vorbis/gstvorbisdec.c: + * ext/vorbis/gstvorbisenc.c: + vorbis: Use new audio encoder/decoder base class API for srcpad caps + +2012-02-01 16:00:37 +0100 Sebastian Dröge + + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/audio/gstaudioencoder.h: + audioencoder: Add gst_audio_encoder_set_output_format() function for consistency + +2012-02-01 15:59:57 +0100 Sebastian Dröge + + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudiodecoder.h: + audiodecoder: Rename set_outcaps() to set_output_format() and take a GstAudioInfo as parameter + +2012-01-31 17:56:04 +0100 Wim Taymans + + * tests/check/elements/audioresample.c: + tests: fix audioresample formats + +2012-01-31 17:47:40 +0100 Wim Taymans + + * tests/check/elements/audiorate.c: + tests: improve tests + +2012-01-31 16:56:03 +0100 Wim Taymans + + * tests/check/elements/playbin-compressed.c: + * tests/check/elements/playbin.c: + tests: fix some more tests + +2012-01-31 16:12:33 +0100 Wim Taymans + + * tests/check/elements/volume.c: + tests: update after controller changes + +2012-01-31 16:12:16 +0100 Wim Taymans + + * win32/common/libgstrtsp.def: + defs: update for new API + +2012-01-31 12:28:30 +0100 Stefan Sauer + + * tests/check/elements/volume.c: + * tests/icles/audio-trickplay.c: + controller: adapt to control-source type changes + +2012-01-30 21:37:58 +0100 Stefan Sauer + + * tests/check/elements/volume.c: + * tests/icles/audio-trickplay.c: + controller: rename control-bindings + gst_control_binding_xxx -> gst_xxx_control_binding for consistency. + +2012-01-30 20:58:34 +0100 Wim Taymans + + * ext/ogg/gstoggdemux.c: + oggdemux: don't blindly forward all unknown events + It causes the caps event to be send downstream and cause negotiation failures. + +2012-01-30 17:16:17 +0100 Wim Taymans + + * ext/ogg/gstoggdemux.c: + * ext/ogg/gstoggmux.c: + * ext/ogg/gstoggparse.c: + * ext/theora/gsttheoraenc.c: + * ext/theora/gsttheoraparse.c: + * ext/vorbis/gstvorbisenc.c: + * ext/vorbis/gstvorbisparse.c: + * gst/gdp/dataprotocol.c: + * gst/gdp/gstgdppay.c: + * gst/tcp/gstmultisocketsink.c: + * tests/check/elements/gdpdepay.c: + * tests/check/elements/gdppay.c: + * tests/check/pipelines/oggmux.c: + * tests/check/pipelines/streamheader.c: + update for HEADER flag changes + +2012-01-10 21:17:58 +0200 George Kiagiadakis + + * tests/check/libs/video.c: + tests: test 16-bit rgb formats in test_parse_caps_rgb + https://bugzilla.gnome.org/show_bug.cgi?id=667681 + +2012-01-10 21:02:48 +0200 George Kiagiadakis + + * gst-libs/gst/video/video.c: + video: Use host endianness when generating caps for 16-bit rgb formats + This is necessary in order to match what the caps strings in + video.h contain for 16-bit rgb formats and also to match how + gst_video_format_parse_caps expects them. + https://bugzilla.gnome.org/show_bug.cgi?id=667681 + +2012-01-30 13:06:55 +0100 Wim Taymans + + * gst-libs/gst/video/gstvideopool.c: + * gst-libs/gst/video/gstvideopool.h: + videopool: update for allocator api update + +2012-01-26 10:35:51 +0100 Jonathan Matthew + + * tests/icles/playback/test7.c: + * tests/icles/playbin-text.c: + * tests/icles/position-formats.c: + * tests/icles/stress-playbin.c: + tests: use playbin, not playbin2 + +2012-01-28 14:53:21 +0000 Olivier Crête + + * gst-libs/gst/pbutils/install-plugins.c: + * gst-libs/gst/rtsp/gstrtspurl.c: + * gst/adder/gstadder.c: + Use macros to register boxed types thread safely + +2012-01-27 17:52:49 +0100 Olivier Crête + + * gst-libs/gst/rtp/gstrtcpbuffer.c: + * tests/check/libs/rtp.c: + rtcpbuffer: Set the map.size to the current size of the RTCP packet + maxsize is the maximum size + +2012-01-27 12:55:45 +0100 Olivier Crête + + * gst-libs/gst/rtp/gstrtcpbuffer.c: + rtpcbuffer: To write inside a RTCP buffer, you must be able to read + So always require read + +2012-01-26 18:24:44 +0100 Olivier Crête + + * gst-libs/gst/rtp/gstrtcpbuffer.c: + rtcpbuffer: Return errors if the map mode doesn't match the actions + +2012-01-26 18:24:20 +0100 Olivier Crête + + * gst-libs/gst/rtp/gstrtcpbuffer.c: + rtcpbuffer: Don't try to modify read-only buffers + +2012-01-27 18:25:38 +0100 Wim Taymans + + * gst-libs/gst/audio/gstaudiosrc.c: + audiosrc: wait on the right cond variable + This broke with a merge commit + +2012-01-27 17:55:34 +0100 Jason DeRose + + * gst/audiorate/gstaudiorate.c: + audiorate: Use the number of samples for the in and out properties as documented + +2012-01-27 17:10:35 +0100 Sebastian Dröge + + * ext/vorbis/gstvorbisenc.c: + vorbisenc: Properly generate the channel-mask on the sinkpad caps + +2012-01-27 13:52:30 +0000 Vincent Penquerc'h + + * sys/v4l/gstv4lxoverlay.c: + * sys/v4l/v4l_calls.c: + * sys/v4l/v4lsrc_calls.c: + v4l: include the glib compatiblity header for the deprecated mutex API + +2012-01-27 15:12:25 +0100 Sebastian Dröge + + Merge branch 'master' into 0.11 + Conflicts: + gst/adder/gstadder.c + +2012-01-27 12:08:33 +0100 Sebastian Dröge + + * ext/vorbis/gstvorbisparse.c: + * ext/vorbis/gstvorbisparse.h: + vorbisparse: Pass correct header buffer size to libvorbis and include channels/rate in the srcpad caps + +2012-01-26 19:47:38 +0100 Wim Taymans + + * gst-libs/gst/tag/gsttagdemux.c: + tagdemux: use default event handler for delayed events + +2012-01-26 15:25:18 +0100 Andoni Morales Alastruey + + * gst/tcp/gsttcpserversink.c: + tcpserversink: remove unused include + +2012-01-26 14:28:06 +0100 Wim Taymans + + * ext/alsa/gstalsa.c: + alsa: merge instead of appending structures + +2012-01-26 11:02:51 +0100 Sebastian Dröge + + * ext/theora/gsttheoraenc.c: + theoraenc: Add width/height/framerate to the srcpad caps + +2012-01-26 11:01:12 +0100 Sebastian Dröge + + * ext/vorbis/gstvorbisenc.c: + vorbisenc: Add samplerate and channels to the srcpad caps + +2012-01-26 10:27:00 +0100 Sebastian Dröge + + * gst/adder/gstadder.c: + adder: Update for new collectpads2 event handling API + +2012-01-25 18:24:07 +0100 Sebastian Dröge + + * ext/theora/gsttheoraenc.c: + theoraenc: Fix encoding of non-mod-16 widths/heights + The next higher multiple of 16 has to be passed + in the input buffers but Theora does never read + beyond the configured picture size. + +2012-01-25 16:42:43 +0100 Sebastian Dröge + + * ext/theora/gsttheoraparse.c: + theoraparse: Remove the synchronization points property + Is someone really using it? In that case it has to be + changed from a GValueArray property to something else. + +2012-01-25 14:31:34 +0100 Thomas Vander Stichele + + * 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-build.stamp: + * docs/plugins/inspect.stamp: + * docs/plugins/inspect/plugin-adder.xml: + * docs/plugins/inspect/plugin-alsa.xml: + * docs/plugins/inspect/plugin-app.xml: + * docs/plugins/inspect/plugin-audioconvert.xml: + * docs/plugins/inspect/plugin-audiorate.xml: + * docs/plugins/inspect/plugin-audioresample.xml: + * docs/plugins/inspect/plugin-audiotestsrc.xml: + * docs/plugins/inspect/plugin-cdparanoia.xml: + * docs/plugins/inspect/plugin-encoding.xml: + * docs/plugins/inspect/plugin-gdp.xml: + * docs/plugins/inspect/plugin-gio.xml: + * docs/plugins/inspect/plugin-libvisual.xml: + * docs/plugins/inspect/plugin-ogg.xml: + * docs/plugins/inspect/plugin-pango.xml: + * docs/plugins/inspect/plugin-playback.xml: + * docs/plugins/inspect/plugin-subparse.xml: + * docs/plugins/inspect/plugin-tcp.xml: + * docs/plugins/inspect/plugin-theora.xml: + * docs/plugins/inspect/plugin-typefindfunctions.xml: + * docs/plugins/inspect/plugin-uridecodebin.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/plugins: update docs, add multisocketsink + +2012-01-25 15:02:09 +0100 Edward Hervey + + * gst/adder/gstadder.c: + adder: Remove deprecation disabling + It's actually fixed in 0.11 + +2012-01-25 12:50:44 +0100 Edward Hervey + + * gst/adder/gstadder.c: + * tests/examples/audio/audiomix.c: + * tests/examples/audio/volume.c: + * tests/examples/seek/jsseek.c: + * tests/examples/seek/scrubby.c: + * tests/examples/seek/seek.c: + * tests/icles/test-colorkey.c: + * tests/icles/test-videooverlay.c: + Suppress deprecations in selected files + +2012-01-25 13:46:35 +0100 Thomas Vander Stichele + + * common: + Automatic update of common submodule + From c463bc0 to 7fda524 + +2012-01-25 12:50:44 +0100 Edward Hervey + + * gst/adder/gstadder.c: + * tests/examples/audio/audiomix.c: + * tests/examples/audio/volume.c: + * tests/examples/seek/jsseek.c: + * tests/examples/seek/scrubby.c: + * tests/examples/seek/seek.c: + * tests/icles/test-colorkey.c: + * tests/icles/test-xoverlay.c: + Suppress deprecations in selected files + +2012-01-24 17:44:21 +0000 Vincent Penquerc'h + + * gst/subparse/gstsubparse.c: + subparse: factor memory freeing + +2012-01-24 17:42:51 +0000 Vincent Penquerc'h + + * gst/subparse/gstsubparse.c: + subparse: fix parsing by not misusing non time segments + A simple filesrc ! subparse ! fakesink type pipeline now works again. + +2012-01-25 12:27:49 +0100 Wim Taymans + + * gst/playback/gstsubtitleoverlay.c: + subtitle: fix merge + +2012-01-24 14:37:12 +0100 Wim Taymans + + * gst-libs/gst/rtp/gstrtcpbuffer.c: + * gst-libs/gst/rtp/gstrtcpbuffer.h: + * gst-libs/gst/rtp/gstrtpbuffer.c: + * gst-libs/gst/rtp/gstrtpbuffer.h: + rtp: improve structures + Remove flags that is in the mapinfo now + +2012-01-20 16:11:54 +0100 Wim Taymans + + * ext/libvisual/visual.c: + * ext/ogg/gstoggaviparse.c: + * ext/ogg/gstoggmux.c: + * ext/ogg/gstoggparse.c: + * ext/ogg/gstoggstream.c: + * ext/ogg/gstogmparse.c: + * ext/pango/gstbasetextoverlay.c: + * ext/pango/gsttextrender.c: + * ext/theora/gsttheoradec.c: + * ext/theora/gsttheoraenc.c: + * ext/theora/gsttheoraparse.c: + * ext/vorbis/gstvorbisdec.c: + * ext/vorbis/gstvorbisdeclib.h: + * ext/vorbis/gstvorbisenc.c: + * ext/vorbis/gstvorbisparse.c: + * ext/vorbis/gstvorbistag.c: + * gst-libs/gst/audio/audio.c: + * gst-libs/gst/audio/gstaudiobasesink.c: + * gst-libs/gst/audio/gstaudiobasesrc.c: + * gst-libs/gst/riff/riff-media.c: + * gst-libs/gst/riff/riff-read.c: + * gst-libs/gst/rtp/gstrtcpbuffer.c: + * gst-libs/gst/rtp/gstrtcpbuffer.h: + * gst-libs/gst/rtp/gstrtpbuffer.c: + * gst-libs/gst/rtp/gstrtpbuffer.h: + * gst-libs/gst/tag/gstexiftag.c: + * gst-libs/gst/tag/gstvorbistag.c: + * gst-libs/gst/tag/gstxmptag.c: + * gst-libs/gst/tag/id3v2.c: + * gst-libs/gst/tag/tags.c: + * gst-libs/gst/video/gstvideometa.c: + * gst-libs/gst/video/gstvideometa.h: + * gst-libs/gst/video/video.c: + * gst-libs/gst/video/video.h: + * gst/adder/gstadder.c: + * gst/audioconvert/gstaudioconvert.c: + * gst/audiorate/gstaudiorate.c: + * gst/audioresample/gstaudioresample.c: + * gst/audiotestsrc/gstaudiotestsrc.c: + * gst/gdp/dataprotocol.c: + * gst/gdp/gstgdpdepay.c: + * gst/gio/gstgiobasesink.c: + * gst/gio/gstgiobasesrc.c: + * gst/subparse/gstssaparse.c: + * gst/subparse/gstsubparse.c: + * gst/tcp/gstmultisocketsink.c: + * gst/tcp/gsttcpclientsink.c: + * gst/tcp/gsttcpclientsrc.c: + * gst/tcp/gsttcpserversrc.c: + * gst/videoconvert/gstvideoconvert.c: + * gst/volume/gstvolume.c: + * tests/check/elements/audioresample.c: + * tests/check/elements/gdpdepay.c: + * tests/check/elements/gdppay.c: + * tests/check/elements/playbin.c: + * tests/check/elements/subparse.c: + * tests/check/elements/textoverlay.c: + * tests/check/elements/videoscale.c: + * tests/check/elements/videotestsrc.c: + * tests/check/elements/volume.c: + * tests/check/elements/vorbistag.c: + * tests/check/gst/typefindfunctions.c: + * tests/check/libs/audio.c: + * tests/check/libs/audiocdsrc.c: + * tests/check/libs/rtp.c: + * tests/check/libs/tag.c: + * tests/check/libs/video.c: + * tests/check/libs/xmpwriter.c: + * tests/check/pipelines/streamheader.c: + * tests/examples/app/appsrc_ex.c: + * tests/examples/seek/jsseek.c: + * tests/examples/seek/seek.c: + * tests/examples/snapshot/snapshot.c: + * tests/icles/playbin-text.c: + port to new map API + +2012-01-25 12:29:11 +0100 Sebastian Dröge + + Merge branch 'master' into 0.11 + Conflicts: + gst/playback/gstdecodebin2.c + +2012-01-25 12:25:05 +0100 Sebastian Dröge + + * gst/playback/gstdecodebin2.c: + Revert "decodebin2: Prune old groups before switching to the new one" + This reverts commit e2a038acee2969ed0b558093fa1c8b7422073e40. + This wasn't entirely correct yet and needs some changes here + and there. + +2012-01-25 12:03:31 +0100 Sebastian Dröge + + * gst/playback/gstdecodebin2.c: + decodebin2: Fix merge error + +2012-01-25 11:04:43 +0100 Olivier Crête + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + rtpbasepayload: Port to group-less GstBufferList + +2012-01-25 11:50:54 +0100 Sebastian Dröge + + Merge branch 'master' into 0.11 + Conflicts: + gst-libs/gst/interfaces/propertyprobe.c + sys/xvimage/xvimagesink.c + +2012-01-25 11:37:55 +0100 Sebastian Dröge + + * common: + Automatic update of common submodule + From 2a59016 to c463bc0 + +2012-01-23 09:28:18 -0800 David Schleef + + * gst-libs/gst/interfaces/propertyprobe.c: + propertyprobe: fix documentation + +2012-01-23 11:57:36 +0000 Tim-Philipp Müller + + * tests/icles/audio-trickplay.c: + tests: fix missing include in audio-trickplay + +2012-01-18 14:58:08 +0000 Vincent Penquerc'h + + * gst/playback/gstplaybin2.c: + playbin2: do not try to deactivate an inactive group + A group may have failed to activate due to an error (for instance, + having set the URI to a non existent location in about-to-finish). + https://bugzilla.gnome.org/show_bug.cgi?id=666395 + +2012-01-21 20:06:53 +0100 Stefan Sauer + + * tests/check/elements/volume.c: + * tests/icles/audio-trickplay.c: + controller: move from control-binding to control-binding-direct + +2012-01-22 22:52:28 +0000 Tim-Philipp Müller + + * ext/alsa/gstalsasink.c: + * ext/cdparanoia/gstcdparanoiasrc.c: + * tests/examples/seek/jsseek.c: + * tests/examples/seek/seek.c: + Replace deprecated GStaticMutex with GMutex + +2012-01-22 01:47:14 +0000 Tim-Philipp Müller + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: use G_TYPE_ERROR instead of GST_TYPE_G_ERROR + +2012-01-17 16:05:41 +0200 Anssi Hannula + + * gst/playback/gstsubtitleoverlay.c: + subtitleoverlay: fix state change stall on PAUSED->READY->PAUSED + After a PAUSED->READY change the sink pads are currently not set to + blocking state. When the element is set back to PAUSED, the change will + be done asynchronously, but as the _pad_blocked_cb() callback is now not + called, the state change never completes. + Fix that by setting the sink pads to blocking state on a PAUSED->READY + change, which ensures that the _pad_blocked_cb() is called when needed + on any future READY->PAUSED change. The sink pads are already put to + blocking state on NULL->READY change, so this behavior is consistent. + Fixes bug #668097. + +2012-01-20 14:44:19 +0100 Stefan Sauer + + * tests/check/elements/volume.c: + * tests/icles/audio-trickplay.c: + controller: adapt to control_binding changes + +2012-01-20 08:29:02 +0100 Stefan Sauer + + * gst/volume/gstvolume.c: + * tests/check/elements/volume.c: + * tests/icles/audio-trickplay.c: + controller: adapt to controller api changes + Don't use the convenience api for control sources. + +2012-01-19 16:40:22 +0100 Mark Nauwelaerts + + * gst/playback/gststreamsynchronizer.c: + streamsynchronizer: avoid unlikely NULL dereference + +2012-01-19 16:35:54 +0100 Mark Nauwelaerts + + * gst/videoscale/vs_fill_borders.c: + videoscale: prevent implicit upgrade to integer type and sign extension + +2012-01-19 16:35:04 +0100 Mark Nauwelaerts + + * tools/gst-discoverer.c: + gst-discoverer: remove extraneous variable + +2012-01-19 16:32:37 +0100 Mark Nauwelaerts + + * gst/playback/gstplaysink.c: + playsink: verify linking to overlay element + +2012-01-19 16:32:05 +0100 Mark Nauwelaerts + + * gst/playback/gstplaysink.c: + playsink: avoid finding sink in NULL bin in corner case + +2012-01-19 16:29:53 +0100 Mark Nauwelaerts + + * gst-libs/gst/tag/gstexiftag.c: + tag: exif: add missing break + +2012-01-19 15:32:52 +0100 Wim Taymans + + * tests/check/Makefile.am: + * tests/check/elements/appsink.c: + * tests/check/libs/rtp.c: + * tests/check/pipelines/streamheader.c: + tests: fix some tests + +2012-01-19 15:19:34 +0100 Wim Taymans + + * gst-libs/gst/rtp/gstrtcpbuffer.c: + rtcp: handle size update correctly + Do explicit resize to set the size of a buffer instead of setting a value in + unmap. + +2012-01-19 15:18:58 +0100 Wim Taymans + + * gst-libs/gst/app/gstappsrc.c: + appsrc: handle NULL caps correctly + +2012-01-19 14:07:34 +0000 Tim-Philipp Müller + + * common: + * configure.ac: + Add --disable-fatal-warnings configure option + +2012-01-19 09:17:07 +0100 Wim Taymans + + * gst-libs/gst/rtp/gstrtpbuffer.c: + * gst-libs/gst/video/gstvideometa.c: + * gst-libs/gst/video/gstvideometa.h: + * gst-libs/gst/video/video.c: + Update for memory API changes + +2012-01-19 09:48:38 +0100 Wim Taymans + + * ext/alsa/gstalsamixer.c: + * ext/alsa/gstalsamixer.h: + * ext/ogg/gstoggdemux.c: + * gst-libs/gst/audio/gstaudiobasesink.c: + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudiodecoder.h: + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/audio/gstaudioencoder.h: + * gst/adder/gstadder.c: + * gst/playback/gstdecodebin.c: + * gst/playback/gstdecodebin2.c: + * gst/playback/gstplaybin2.c: + * gst/playback/gstplaysink.c: + * gst/playback/gststreamsynchronizer.c: + * gst/tcp/gstmultisocketsink.c: + * gst/tcp/gstmultisocketsink.h: + port to new glib thread API + +2012-01-17 18:13:43 +0100 Robert Swain + + * docs/design/part-interlaced-video.txt: + docs: interlaced video: Update docs + +2012-01-19 09:17:31 +0100 Wim Taymans + + * gst/tcp/gsttcpclientsrc.c: + * gst/tcp/gsttcpserversrc.c: + tcp: work around compiler warnings + +2011-09-13 23:14:10 +0000 Youness Alaoui + + * gst/playback/gstdecodebin2.c: + decodebin2: Prune old groups before switching to the new one + In order to allow for proper functionality when a decoder only supports + one instance at a time (dsp), we must block the demuxer pads when they + get created if they are not part of the active group, preventing buffers + from being sent to the decoder (and initializing it through setcaps), + then after we switch to a new group, we unblock the demuxer pads for + the active groups. In the callback for the unblock, we prune the old + groups, making sure the previous decoder instance is destroyed before + we push a buffer to the new instance. + +2012-01-18 17:22:21 +0000 Tim-Philipp Müller + + * ext/alsa/gstalsamixer.c: + * ext/alsa/gstalsamixer.h: + * gst-libs/gst/audio/gstaudiosink.c: + * gst-libs/gst/audio/gstaudiosrc.c: + * gst-libs/gst/glib-compat-private.h: + * gst-libs/gst/tag/licenses.c: + * gst-libs/gst/tag/xmpwriter.c: + * gst-libs/gst/video/video-overlay-composition.c: + * gst/adder/gstadder.c: + * gst/audiorate/gstaudiorate.c: + * gst/tcp/gstmultisocketsink.c: + * gst/videorate/gstvideorate.c: + * sys/ximage/ximagesink.c: + * sys/xvimage/xvimagesink.c: + * tests/examples/encoding/encoding.c: + * tests/examples/overlay/gtk-videooverlay.c: + * tests/examples/overlay/qt-videooverlay.cpp: + * tests/examples/seek/jsseek.c: + * tests/examples/seek/scrubby.c: + * tests/examples/seek/seek.c: + * tests/icles/stress-playbin.c: + * tests/icles/test-colorkey.c: + * tests/icles/test-videooverlay.c: + * tools/gst-discoverer.c: + Remove compatibility code cruft for old GLib versions + +2012-01-18 17:21:57 +0000 Tim-Philipp Müller + + * Makefile.am: + Add ext/gio/ to CRUFT_DIRS + +2012-01-18 17:21:36 +0000 Tim-Philipp Müller + + * gst/encoding/gststreamcombiner.c: + * gst/encoding/gststreamcombiner.h: + * gst/encoding/gststreamsplitter.c: + * gst/encoding/gststreamsplitter.h: + encoding: port to new GLib threading API + +2012-01-18 17:21:02 +0000 Tim-Philipp Müller + + * ext/pango/gstbasetextoverlay.c: + * ext/pango/gstbasetextoverlay.h: + pango: port to new GLib threading API + +2012-01-18 16:55:45 +0100 Sebastian Dröge + + * configure.ac: + configure.ac: Remove GIO check, it's in gst-glib2.m4 now + +2012-01-18 16:46:01 +0100 Sebastian Dröge + + * common: + Automatic update of common submodule + From 0807187 to 2a59016 + +2012-01-18 16:19:12 +0100 Sebastian Dröge + + * configure.ac: + * docs/plugins/Makefile.am: + * ext/Makefile.am: + * ext/gio/Makefile.am: + * ext/gio/gstgio.c: + * ext/gio/gstgio.h: + * ext/gio/gstgiobasesink.c: + * ext/gio/gstgiobasesink.h: + * ext/gio/gstgiobasesrc.c: + * ext/gio/gstgiobasesrc.h: + * ext/gio/gstgiosink.c: + * ext/gio/gstgiosink.h: + * ext/gio/gstgiosrc.c: + * ext/gio/gstgiosrc.h: + * ext/gio/gstgiostreamsink.c: + * ext/gio/gstgiostreamsink.h: + * ext/gio/gstgiostreamsrc.c: + * ext/gio/gstgiostreamsrc.h: + * gst/gio/Makefile.am: + * gst/gio/gstgio.c: + * gst/gio/gstgio.h: + * gst/gio/gstgiobasesink.c: + * gst/gio/gstgiobasesink.h: + * gst/gio/gstgiobasesrc.c: + * gst/gio/gstgiobasesrc.h: + * gst/gio/gstgiosink.c: + * gst/gio/gstgiosink.h: + * gst/gio/gstgiosrc.c: + * gst/gio/gstgiosrc.h: + * gst/gio/gstgiostreamsink.c: + * gst/gio/gstgiostreamsink.h: + * gst/gio/gstgiostreamsrc.c: + * gst/gio/gstgiostreamsrc.h: + * tests/check/Makefile.am: + * tests/examples/Makefile.am: + * tests/examples/gio/Makefile.am: + gio: Move to gst subdirectory + It's a plugin without external dependencies now because we + unconditionally depend on GIO anyway. + +2012-01-18 16:15:30 +0100 Sebastian Dröge + + * configure.ac: + configure.ac: Require GLib 2.31.10 and improve GIO check + +2012-01-18 13:16:46 +0000 Christian Fredrik Kalager Schaller + + * gst-plugins-base.spec.in: + Update spec file with latest changes + +2012-01-18 01:57:41 +0000 Tim-Philipp Müller + + * po/POTFILES.in: + po: update POTFILES.in for recent changes + +2012-01-17 21:46:58 +0100 Mark Nauwelaerts + + * gst-libs/gst/audio/gstbaseaudiosink.c: + baseaudiosink: commit correct number of samples when not syncing + +2012-01-17 18:19:30 +0100 Mark Nauwelaerts + + * ext/ogg/gstoggstream.c: + oggstream: initialize variable + ... to help out challenged compiler. + +2012-01-17 16:55:54 +0100 Sebastian Dröge + + * configure.ac: + configure: Remove socket/winsock and related checks, not necessary anymore + +2012-01-17 16:38:45 +0100 Sebastian Dröge + + * gst-libs/gst/rtsp/Makefile.am: + * gst-libs/gst/rtsp/gstrtspconnection.c: + * gst-libs/gst/rtsp/gstrtspconnection.h: + * gst-libs/gst/rtsp/gstrtspdefs.c: + * pkgconfig/gstreamer-rtsp-uninstalled.pc.in: + * pkgconfig/gstreamer-rtsp.pc.in: + rtsp: Port to GIO + +2012-01-17 13:27:05 +0100 Sebastian Dröge + + * gst-libs/gst/sdp/Makefile.am: + * gst-libs/gst/sdp/gstsdpmessage.c: + * gst-libs/gst/sdp/gstsdpmessage.h: + * pkgconfig/gstreamer-sdp-uninstalled.pc.in: + * pkgconfig/gstreamer-sdp.pc.in: + sdp: Port to GIO for multicast address detection + +2012-01-17 12:21:54 +0100 Sebastian Dröge + + * gst/tcp/gsttcpclientsrc.c: + * gst/tcp/gsttcpserversrc.c: + tcp: Fix handling of closed connections + +2012-01-17 12:08:17 +0100 Sebastian Dröge + + * gst/tcp/gsttcpclientsink.c: + * gst/tcp/gsttcpclientsrc.c: + * gst/tcp/gsttcpserversink.c: + * gst/tcp/gsttcpserversrc.c: + tcp: Add support for IPv6 + +2012-01-17 11:52:49 +0100 Mark Nauwelaerts + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: register state change function + +2012-01-17 11:44:20 +0100 Sebastian Dröge + + * gst/tcp/gstmultisocketsink.c: + * gst/tcp/gsttcpclientsrc.c: + * gst/tcp/gsttcpserversrc.c: + tcp: Only read as much as is currently available from the socket + +2012-01-17 11:32:01 +0100 Sebastian Dröge + + * gst/tcp/gsttcpclientsink.c: + * gst/tcp/gsttcpclientsrc.c: + * gst/tcp/gsttcpserversink.c: + * gst/tcp/gsttcpserversrc.c: + tcp: Don't leak the resolver if name resolval failed + +2012-01-17 11:29:26 +0100 Sebastian Dröge + + * configure.ac: + configure: We require GIO now + +2012-01-16 11:43:25 +0000 Vincent Penquerc'h + + * ext/alsa/gstalsasink.c: + alsasink: fix high sample rates being rejected + An ALSA sink may select a different rate (as we use the _set_rate_near + API, which is not guaranteed to set the exact target rate). + The rest of the code seems to already handle this well, as output + from a 88200 Hz file seems to have the correct pitch when selecting + a 96 kHz rate. + +2012-01-16 11:40:47 +0000 Vincent Penquerc'h + + * ext/alsa/gstalsasink.c: + alsasink: fix rate match message mistaking error code for sample rate + +2012-01-16 11:40:16 +0000 Vincent Penquerc'h + + * ext/alsa/gstalsasink.c: + alsasink: log API errors along with the error code and string + +2012-01-16 12:29:35 +0100 Sebastian Dröge + + * gst/tcp/gstmultisocketsink.c: + multisocketsink: Fix possible GType namespace conflicts with the private element enums + +2012-01-16 12:17:00 +0100 Sebastian Dröge + + * gst/tcp/gstmultisocketsink.c: + * gst/tcp/gstmultisocketsink.h: + multisocketsink: Re-add QoS DSCP property + +2012-01-16 11:25:54 +0100 Sebastian Dröge + + * configure.ac: + * m4/gst-fionread.m4: + * tests/check/Makefile.am: + * tests/check/elements/multifdsink.c: + tcp: Remove remaining unused stuff + +2012-01-16 11:01:10 +0100 Sebastian Dröge + + * gst/tcp/Makefile.am: + * gst/tcp/gsttcp.c: + * gst/tcp/gsttcp.h: + tcp: Remove old socket helper functions + +2012-01-16 10:08:24 +0100 Sebastian Dröge + + * docs/plugins/Makefile.am: + * gst/tcp/Makefile.am: + * gst/tcp/gstmultifdsink.c: + * gst/tcp/gstmultifdsink.h: + * gst/tcp/gstmultisocketsink.c: + * gst/tcp/gstmultisocketsink.h: + * gst/tcp/gsttcp-marshal.list: + * gst/tcp/gsttcp.h: + * gst/tcp/gsttcpplugin.c: + * gst/tcp/gsttcpplugin.h: + * gst/tcp/gsttcpserversink.c: + * gst/tcp/gsttcpserversink.h: + tcpserversink: Port to GIO + And change multifdsink to GIO too and rename it to multisocketsink + because it only works on GSockets now, not generic fds. + +2012-01-11 16:06:22 +0100 Sebastian Dröge + + * gst/tcp/gsttcpserversrc.c: + * gst/tcp/gsttcpserversrc.h: + tcpserversrc: Port to GIO + +2012-01-11 15:43:11 +0100 Sebastian Dröge + + * gst/tcp/gsttcpclientsink.c: + * gst/tcp/gsttcpclientsink.h: + tcpclientsink: Port to GIO + +2012-01-11 15:09:46 +0100 Sebastian Dröge + + * gst/tcp/Makefile.am: + * gst/tcp/gsttcpclientsrc.c: + * gst/tcp/gsttcpclientsrc.h: + tcpclientsrc: Port to GIO + +2011-12-27 04:18:19 +0100 Matej Knopp + + * gst-libs/gst/video/gstvideopool.c: + videopool: fix printf warning in debug message + https://bugzilla.gnome.org/show_bug.cgi?id=662607 + +2012-01-13 16:57:15 -0300 Reynaldo H. Verdejo Pinochet + + * Android.mk: + Android, Add explicit path for zlib + This change fixes building gst-libs/gst/tag/ code with + the Android buildsystem. + +2012-01-13 14:50:49 -0300 Reynaldo H. Verdejo Pinochet + + * ext/vorbis/gstvorbisdec.c: + Fix wrong access to undefined struct member + For the USE_TREMOLO case, GstVorbisDec doesn't have + a vb member. Besides, Tremolo's vorbis_dsp_synthesis() + expects a vorbis_dsp_state to be passed as first + argument. Not a vorbis_block. + +2012-01-13 14:47:13 -0300 Reynaldo H. Verdejo Pinochet + + * ext/vorbis/gstvorbisdec.c: + Fix TREMELO -> TREMOLO typo + +2012-01-13 16:52:23 +0000 Vincent Penquerc'h + + * sys/xvimage/xvimagesink.c: + xvimagesink: fix leak when images are freed after the X context + I'm not 100% sure this is valid on any other X server than mine, + but since the XFree call does not take the context as a parameter, + it seems pretty certain it's the right thing to do, but I'll put + this caveat here in case someone checks in the future. + +2012-01-12 23:35:44 +0000 Tim-Philipp Müller + + * gst-libs/gst/tag/gstvorbistag.c: + * gst-libs/gst/tag/gstxmptag.c: + * gst-libs/gst/tag/id3v2frames.c: + * tests/check/libs/tag.c: + GST_TYPE_DATE -> G_TYPE_DATE + +2012-01-12 23:25:22 +0000 Tim-Philipp Müller + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: fix up for GstTagList != GstStructure + +2012-01-12 23:21:17 +0000 Tim-Philipp Müller + + Merge remote-tracking branch 'origin/master' into 0.11 + Conflicts: + gst-libs/gst/pbutils/gstdiscoverer-types.c + gst-libs/gst/pbutils/gstdiscoverer.c + tests/check/Makefile.am + +2012-01-12 17:31:44 +0000 Tim-Philipp Müller + + * tests/check/Makefile.am: + tests: discoverer test is now valgrind clean + +2012-01-12 16:24:01 +0000 Vincent Penquerc'h + + * ext/theora/gsttheoraparse.c: + theoraparse: fix array leak + +2012-01-12 14:26:05 +0000 Vincent Penquerc'h + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: fix structure leak + I hit the 'misc' one, but let's also make sure the topology + one get freed as well, though I do not know if this can happen + twice. + +2012-01-12 13:57:18 +0100 Mark Nauwelaerts + + * gst/playback/gstplaysinkconvertbin.c: + playsinkconvertbin: release extra ref on converter elements + +2012-01-11 20:47:00 -0300 Reynaldo H. Verdejo Pinochet + + * gst-libs/gst/video/Makefile.am: + Add missing DEFAULT_INCLUDES on androgenizer call + Fix building of the libgstvideo module on Android by adding the + missing and needed $(DEFAULT_INCLUDES) to CFLAGS for the + androgenizer call on gst-libs/gst/video/Makefile.am + Before this change, building was failing due to gst-plugins-base/ + and gst-plugins-base/gst-libs/gst/video being left out of the + include path. + +2012-01-11 16:17:42 +0000 Vincent Penquerc'h + + * ext/ogg/gstoggdemux.c: + oggdemux: fix push mode chain leak + When I first implemented push mode seeking, I removed the chain + freeing there as it could be used later. The current code does not + seem to do that though, so I'm restoring the previous freeing, + which plugs the leak while apparently not reintroducing use of + freed data with chained and normal files, both with gst-launch + playbin2 and Totem. + +2012-01-11 12:52:17 +0000 Vincent Penquerc'h + + * gst-libs/gst/pbutils/gstdiscoverer-types.c: + discoverer: fix leaks caused by some base class dtors not being called + +2012-01-11 12:16:28 +0000 Vincent Penquerc'h + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: fix caps and discoverer object ref leaks + +2012-01-11 11:55:59 +0000 Vincent Penquerc'h + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: add a few consts where appropriate + +2012-01-11 11:55:36 +0000 Vincent Penquerc'h + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: fix pad leak + +2012-01-11 10:49:49 +0100 Sebastian Dröge + + * gst-libs/gst/audio/audio.c: + audio: More UNPOSITION flag sanity checks + ..and turn the GST_WARNING() into a g_warning(). This is a programming + error and should be fixed. + +2012-01-11 10:44:37 +0100 Sebastian Dröge + + * gst-libs/gst/audio/audio.c: + audio: Add validity check for the UNPOSITIONED audio flag + Also reset the flag when parsing caps. + +2012-01-10 19:01:11 +0000 Vincent Penquerc'h + + * ext/ogg/gstoggmux.c: + Revert "oggmux: fix pad leak" + This reverts commit 5df30c1b905edce16f2258e414a0a4afb540d0f1. + I must have dreamt the Valgrind logs, reverting this reintroduces + no leak, and gets rid of the test failures it introduced :S + +2012-01-10 18:27:19 +0000 Tim-Philipp Müller + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: use GST_TYPE_TAG_LIST for tag lists + They may not be structures in 0.11/1.0. + +2012-01-10 18:07:19 +0000 Tim-Philipp Müller + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: fix potential tag list leaks + Not that I have ever seen these in practice, but if they + can't happen we may just as well just assign the new tag + list. Merge properly to be on the safe side, and also + avoid a useless tag list copy in the normal case where + there is no tag list yet. + +2012-01-10 17:48:44 +0000 Tim-Philipp Müller + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: fix potential caps leak + in last else chunk. + +2012-01-10 16:57:04 +0000 Vincent Penquerc'h + + * ext/ogg/gstoggstream.c: + oggstream: fix tag list leak + +2012-01-10 16:51:09 +0000 Vincent Penquerc'h + + * ext/ogg/gstoggdemux.c: + oggdemux: fix pad leak + +2012-01-10 16:14:29 +0000 Vincent Penquerc'h + + * ext/ogg/gstoggdemux.c: + oggdemux: fix hang on small truncated files + A first hang was happening when trying to locate a page backwards, + where we'd sync forever on the same page. + With that fixed, a second hang would happen after preparing an EOS + event, but with no chain created yet to send it to, the pipeline + would stay idle forever. + An element error is now emitted for this case. + +2012-01-10 14:35:31 +0000 Vincent Penquerc'h + + * ext/ogg/gstoggmux.c: + oggmux: fix pad leak + +2012-01-10 15:59:27 +0100 Sebastian Dröge + + * gst/playback/gststreamsynchronizer.c: + streamsynchronizer: Don't unref the parent in the event function + +2012-01-10 13:15:12 +0100 Sebastian Dröge + + Merge branch 'master' into 0.11 + Conflicts: + gst-libs/gst/app/gstappsrc.c + gst-libs/gst/audio/multichannel.h + gst-libs/gst/video/videooverlay.c + gst/playback/gstplaysink.c + gst/playback/gststreamsynchronizer.c + tests/check/Makefile.am + win32/common/libgstvideo.def + +2012-01-10 12:57:27 +0100 Sebastian Dröge + + * win32/common/libgstaudio.def: + win32: Add the new audio symbols to the list of exported symbols + +2012-01-10 12:46:05 +0100 Sebastian Dröge + + * gst-libs/gst/audio/gstaudiometa.c: + * gst-libs/gst/audio/gstaudiometa.h: + audiometa: Improve GstAudioDownmixMeta to be actually usable + This now has a two-dimensional array of coefficients + as required and also stores the source and destination + channel positions. + +2012-01-10 12:02:56 +0100 Sebastian Dröge + + * gst-libs/gst/audio/audio.c: + audio: Don't crash if NULL positions are passed to gst_audio_info_set_format() + +2012-01-09 14:19:54 +0100 Sebastian Dröge + + * gst-libs/gst/audio/gstaudiobasesink.c: + audiobasesink: Fix infinite recursion by chaining up to the correct parent class vfunc + +2012-01-09 12:31:02 +0100 Mark Nauwelaerts + + * gst/playback/gstplay-enum.h: + playback: document DEINTERLACE flag + +2012-01-09 08:24:23 +0100 Sebastian Dröge + + * gst-libs/gst/audio/audio.c: + audio: Don't check for channel positions in valid order when converting to a channel mask + +2012-01-07 20:12:17 +0000 Tim-Philipp Müller + + * gst-libs/gst/rtsp/gstrtspconnection.c: + rtspconnection: make hostname lookup more thread-safe + Don't write IP number string to return into a static + array which is shared amongst all threads (note: of + course a copy is returned). + https://bugzilla.gnome.org/show_bug.cgi?id=666711 + +2012-01-07 19:39:42 +0000 Tim-Philipp Müller + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: make is_subtitle_caps thread-safe + +2012-01-07 16:43:26 +0000 Tim-Philipp Müller + + * tests/check/Makefile.am: + * tests/check/libs/discoverer.c: + * tests/files/Makefile.am: + * tests/files/theora-vorbis.ogg: + tests: add ogg test file and some proper unit tests for discoverer + Leaks when re-used, so blacklisted for valgrind for now. + +2012-01-07 14:44:51 +0000 Tim-Philipp Müller + + * win32/common/libgstvideo.def: + win32: .def file should be sorted for make check-exports + +2012-01-06 16:15:40 +0100 Mark Nauwelaerts + + * ext/vorbis/gstvorbisdec.c: + vorbisdec: use right channel variable even more + +2012-01-06 16:13:35 +0100 Mark Nauwelaerts + + * gst/audioresample/gstaudioresample.c: + audioresample: fix debug message format specifier + +2012-01-06 15:40:06 +0100 Edward Hervey + + * gst/playback/gstdecodebin2.c: + Revert "decodebin2: Try harder to get initial topology caps" + This reverts commit 6b3e3544d41ce0bc42c3597b3eb2130719379917. + I really shouldn't put WIP commits in my main branch ... + +2012-01-06 15:16:00 +0100 Edward Hervey + + * tests/check/libs/gstlibscpp.cc: + * tests/check/libs/libsabi.c: + tests: Remove dead header include + +2012-01-06 15:14:59 +0100 Edward Hervey + + * gst-libs/gst/audio/audio.c: + audio: Fix size check + We fail (and return) if the size is *NOT* a multiple of samples. + +2012-01-05 08:29:43 +0100 Edward Hervey + + * gst/playback/gstdecodebin2.c: + decodebin2: Try harder to get initial topology caps + Since caps are no longer 'shared' between two pads (but forwarded from + source pad to sink pad) we end up with the first chain pad not having + specified caps (i.e. typefind:src). + This solves the issues by getting the pad's peer caps. + It is not optimal since it will (for most demuxers) return the pad + template caps, which might contain non-fixed caps (ex : with + qtdemux "video/quicktime; video/mj2; audio/x-m4a; application/x-3gp") + https://bugzilla.gnome.org/show_bug.cgi?id=667337 + +2012-01-06 12:06:00 +0000 Christian Fredrik Kalager Schaller + + * docs/design/Makefile.am: + Fix playbin2 -> playbin in Makefile + +2011-12-14 14:14:47 +0000 Vincent Penquerc'h + + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/video/video-blend.c: + * gst-libs/gst/video/video-blend.h: + * gst-libs/gst/video/video-overlay-composition.c: + * gst-libs/gst/video/video-overlay-composition.h: + * win32/common/libgstvideo.def: + video: overlays may now have premultiplied alpha + https://bugzilla.gnome.org/show_bug.cgi?id=666177 + +2011-11-01 17:57:59 +0100 Havard Graff + + * gst-libs/gst/pbutils/gstdiscoverer-types.c: + * gst-libs/gst/tag/tags.c: + * gst/audiotestsrc/gstaudiotestsrc.c: + * gst/encoding/gstsmartencoder.c: + * gst/playback/gstplaysink.c: + * tools/gst-discoverer.c: + Fix various unlikely, but still potential memoryleaks in error code paths + https://bugzilla.gnome.org/show_bug.cgi?id=667311 + +2011-10-22 16:41:23 +0200 Havard Graff + + * gst-libs/gst/app/gstappsrc.c: + appsrc: implement get_caps vfunc + This allows downstream elements to query what caps are available. + https://bugzilla.gnome.org/show_bug.cgi?id=667312 + +2012-01-05 13:59:32 +0100 Wim Taymans + + * gst-libs/gst/audio/audio.c: + * gst-libs/gst/audio/audio.h: + audio: expose API to convert channel array to a mask + +2012-01-05 12:23:08 +0000 Tim-Philipp Müller + + * tools/gst-discoverer.c: + tools: avoid unportable vararg macro construct in gst-discoverer + https://bugzilla.gnome.org/show_bug.cgi?id=667306 + +2012-01-05 12:32:06 +0100 Wim Taymans + + * ext/vorbis/gstvorbisdec.c: + vorbisdec: use right channel variable + +2012-01-05 12:31:51 +0100 Wim Taymans + + * gst-libs/gst/riff/riff-media.c: + riff: don't use NULL arrays + +2012-01-01 20:44:08 +0100 Idar Tollefsen + + * configure.ac: + build: Run platform check for platform specific configuration. + +2011-10-12 11:28:10 +0200 Pascal Buhler + + * gst-libs/gst/rtp/gstrtcpbuffer.c: + rtcpbuffer: prevent overflow of 16bit header length. + RTCP header can be (2^16 + 1) * 4 bytes long, so when validating a bogus + packet it was possible to get a 16bit overflow resulting in a length of 0. + This would put the gst_rtcp_buffer_validate_data function in a endless loop. + https://bugzilla.gnome.org/show_bug.cgi?id=667313 + +2011-09-24 14:05:42 +0200 Havard Graff + + * gst/videotestsrc/videotestsrc.c: + videotestsrc: keep the calculation fixed-point + https://bugzilla.gnome.org/show_bug.cgi?id=667315 + +2011-08-04 11:30:05 +0200 Idar Tollefsen + + * ext/pango/gstclockoverlay.c: + * ext/pango/gsttimeoverlay.c: + pango: changes includes from brackets to quotes for local files + https://bugzilla.gnome.org/show_bug.cgi?id=667316 + +2012-01-04 14:48:33 +0100 Sebastian Dröge + + * gst-libs/gst/audio/audio.c: + audio: Improve/fix handling of NONE layouts + +2012-01-04 14:35:48 +0100 Sebastian Dröge + + * gst-libs/gst/audio/audio.c: + audio: Add support again for more than 64 channels with NONE layouts + +2012-01-04 10:26:47 +0100 Sebastian Dröge + + * gst/audiotestsrc/gstaudiotestsrc.c: + audiotestsrc: Fix channel-mask handling + +2012-01-04 10:26:33 +0100 Sebastian Dröge + + * gst/audioconvert/gstaudioconvert.c: + audioconvert: Fix channel-mask handling + +2012-01-04 09:54:56 +0100 Sebastian Dröge + + * gst-libs/gst/audio/audio.h: + audio: Fix GST_AUDIO_CHANNEL_POSITION_MASK macro + +2011-12-31 14:32:45 +0100 Sebastian Dröge + + * ext/ogg/gstoggstream.c: + * ext/ogg/gstogmparse.c: + ogg: Update for the libgstriff API changes + Still needs to handle the raw audio channel reordering. + +2011-12-31 14:31:08 +0100 Sebastian Dröge + + * gst/adder/gstadder.c: + * gst/audiorate/gstaudiorate.c: + * gst/volume/gstvolume.c: + gst: Add new layout field to all raw audio caps + +2011-12-31 14:25:09 +0100 Sebastian Dröge + + * ext/alsa/gstalsasink.c: + * ext/alsa/gstalsasrc.c: + * ext/libvisual/visual.c: + * ext/ogg/gstoggstream.c: + * ext/vorbis/gstvorbisenc.c: + ext: Add new layout field to the raw audio caps + +2011-12-31 14:21:27 +0100 Sebastian Dröge + + * gst/audioconvert/gstaudioconvert.c: + * gst/audioresample/gstaudioresample.c: + * gst/audiotestsrc/gstaudiotestsrc.c: + gst: Add new layout field to the raw audio caps + +2011-12-31 14:15:41 +0100 Sebastian Dröge + + * gst-libs/gst/riff/riff-media.c: + * gst-libs/gst/riff/riff-media.h: + riff: Return a channel reorder map for raw audio when creating the caps + +2011-12-31 13:50:04 +0100 Sebastian Dröge + + * gst-libs/gst/riff/riff-media.c: + riff: Add the layout field to the raw audio caps + +2011-12-31 13:47:57 +0100 Sebastian Dröge + + * gst-libs/gst/audio/gstaudioencoder.c: + audioencoder: Proxy the channel mask field instead of the old channel-layout field + +2011-12-31 13:47:24 +0100 Sebastian Dröge + + * gst-libs/gst/audio/gstaudiocdsrc.c: + audiocdsrc: Add the layout field to the caps + +2011-12-31 13:46:53 +0100 Sebastian Dröge + + * gst-libs/gst/audio/audio.c: + * gst-libs/gst/audio/audio.h: + audio: Add "layout" field to the raw audio caps + This can be used to differentiate between interleaved + and non-interleaved audio and whatever comes in the future. + +2011-12-31 13:33:01 +0100 Sebastian Dröge + + * gst-libs/gst/audio/audio.c: + * gst-libs/gst/audio/audio.h: + audio: Add function to reorder channel positions from any order to the GStreamer order + +2011-12-24 10:54:20 +0100 Sebastian Dröge + + * gst-libs/gst/audio/gstaudioringbuffer.c: + audioringbuffer: Use new function to get a channel reordering map + +2011-12-24 10:50:20 +0100 Sebastian Dröge + + * gst-libs/gst/audio/audio.c: + audio: Add documentation for the new functions + +2011-12-24 10:37:28 +0100 Sebastian Dröge + + * gst-libs/gst/audio/audio.c: + * gst-libs/gst/audio/audio.h: + audio: Add public functions to check channel positions validity and to get a reorder map + +2011-12-20 16:55:34 +0100 Sebastian Dröge + + * gst-libs/gst/riff/riff-media.c: + riff: Port to the new multichannel caps + +2011-12-20 16:34:38 +0100 Sebastian Dröge + + * Makefile.am: + * tests/examples/audio/Makefile.am: + * tests/examples/audio/testchannels.c: + audio: Remove testchannels example + It's not really relevant anymore + +2011-12-20 12:08:53 +0100 Sebastian Dröge + + * ext/vorbis/gstvorbiscommon.c: + * ext/vorbis/gstvorbiscommon.h: + * ext/vorbis/gstvorbisdec.c: + * ext/vorbis/gstvorbisdeclib.c: + * ext/vorbis/gstvorbisenc.c: + vorbis: Port to the new multichannel caps + +2011-12-20 11:44:27 +0100 Sebastian Dröge + + * ext/alsa/gstalsa.c: + * ext/alsa/gstalsa.h: + * ext/alsa/gstalsasink.c: + * ext/alsa/gstalsasrc.c: + alsa: Port to the new multichannel caps + +2011-12-19 14:27:28 +0100 Sebastian Dröge + + * tests/check/elements/audioconvert.c: + audioconvert: Update unit test for the new multichannel caps + +2011-12-19 12:41:24 +0100 Sebastian Dröge + + * gst/audioconvert/gstaudioconvert.c: + * gst/audioconvert/gstchannelmix.c: + * gst/audioconvert/plugin.c: + audioconvert: Port to the new multichannel caps + audioconvert still needs support for mixing all the new + channel positions, see: + https://bugzilla.gnome.org/show_bug.cgi?id=666506 + +2011-12-20 16:20:06 +0100 Sebastian Dröge + + * gst-libs/gst/audio/gstaudioringbuffer.c: + * gst-libs/gst/audio/gstaudioringbuffer.h: + audioringbuffer: Add support for reordering of channels + +2011-12-19 10:04:30 +0100 Sebastian Dröge + + * tests/check/libs/audio.c: + audio: Add tests for the new multichannel caps and reordering function + +2011-12-16 10:55:13 +0100 Sebastian Dröge + + * gst-libs/gst/audio/Makefile.am: + * gst-libs/gst/audio/audio.c: + * gst-libs/gst/audio/audio.h: + * gst-libs/gst/audio/multichannel.c: + * gst-libs/gst/audio/multichannel.h: + audio: Add new channel positions and simplify channel expression in the caps + The available channel positions are all channels from SMPTE 2036-2-2008 + (in that order) and DTS Coherent Acoustics, which are basically all 28 + channels that currently can appear. + The channels are now expressed in the caps as a channel-mask, which + describes which of the channels are present, and an optional + channel-reorder-map, which must only be used after negotiation for + fixated caps. + For negotiation only the channel-mask and the channel count is relevant + and all elements are expected to handle all reorder maps. Elements that + don't can use the new API to reorder an audio buffer from any order to + another order. + This simplifies negotiation a lot while still having as few reorderings + necassary as possible and still allow all kinds of channel layouts. + +2012-01-05 01:51:35 +0000 Philip Flarsheim + + * gst-libs/gst/interfaces/xoverlay.c: + docs: add win32 code snippets to GstXOverlay Gtk+ example + +2012-01-04 19:50:58 +0000 Tim-Philipp Müller + + * common: + Automatic update of common submodule + From a62f3d4 to 0807187 + +2012-01-04 17:57:39 +0100 Wim Taymans + + * gst/tcp/gstmultifdsink.c: + multifdsink: use pad caps for streamheader + Instead of using the caps on the buffer, use the caps on the pad. + +2012-01-04 16:41:53 +0100 Wim Taymans + + * tests/check/Makefile.am: + * tests/check/elements/appsink.c: + * tests/check/elements/appsrc.c: + * tests/check/elements/audiorate.c: + * tests/check/elements/audioresample.c: + * tests/check/elements/gdpdepay.c: + * tests/check/elements/gdppay.c: + * tests/check/elements/multifdsink.c: + * tests/check/elements/playbin-compressed.c: + * tests/check/elements/playbin.c: + * tests/check/elements/subparse.c: + * tests/check/elements/textoverlay.c: + * tests/check/elements/videorate.c: + * tests/check/elements/videoscale.c: + * tests/check/elements/videotestsrc.c: + * tests/check/elements/volume.c: + * tests/check/pipelines/basetime.c: + * tests/check/pipelines/capsfilter-renegotiation.c: + * tests/check/pipelines/streamheader.c: + tests: port and enable more unit tests + +2012-01-03 21:20:04 +0000 Tim-Philipp Müller + + * gst/videotestsrc/Makefile.am: + videotestsrc: don't build generate_sine_table utility by default + +2012-01-03 11:04:23 +0100 Mark Nauwelaerts + + * gst/playback/gststreamsynchronizer.c: + streamsynchronizer: force fallback buffer_alloc when other pad not available + ... to avoid unnecessary spurious errors (upon e.g. shutdown). + If a real error is applicable in this unusual circumstance (missing other pad), + other (STREAM_LOCK protected) call paths can take care of that. + +2012-01-03 11:02:17 +0100 Mark Nauwelaerts + + * gst/playback/gststreamsynchronizer.c: + streamsynchronizer: avoid crashing when operating on released pad + +2012-01-03 10:41:51 +0100 Wim Taymans + + * gst-libs/gst/video/video.h: + video: add macro to check interlaced + Add a convenience macro to check if the video is interlaced. + +2012-01-02 18:31:16 +0100 Wim Taymans + + * win32/common/libgstvideo.def: + defs: update + +2012-01-02 18:31:05 +0100 Wim Taymans + + * tests/check/elements/encodebin.c: + tests: small cleanup + +2012-01-02 18:28:46 +0100 Wim Taymans + + * gst/encoding/gststreamcombiner.c: + streamcombiner: fix srcpad query caps + The caps query on the srcpad should return the template caps instead of + forwarding the query. + +2012-01-02 17:42:11 +0100 Wim Taymans + + * gst/videorate/gstvideorate.c: + videorate: chain up to parent event function + +2012-01-02 17:28:12 +0100 Wim Taymans + + * gst/videorate/gstvideorate.c: + videorate: fix caps negotiation function + +2012-01-02 16:13:51 +0100 Wim Taymans + + * gst-libs/gst/video/gstvideofilter.c: + videofilter: use caps of the allocation query + Use the caps from the allocation query to propose a video bufferpool instead of + our own negotiated caps. + +2012-01-02 15:59:09 +0100 Wim Taymans + + * gst/audioresample/gstaudioresample.c: + audioresample: truncate in fixation + +2012-01-02 15:40:35 +0100 Wim Taymans + + * tests/check/pipelines/oggmux.c: + tests: fix a unit test + The ogg muxer now has video and audio pads + +2012-01-02 15:39:58 +0100 Wim Taymans + + * gst-libs/gst/audio/gstaudioencoder.c: + audioencoder: turn assert into a real error + Post a real error instead of just asserting. Fixes a unit test. + +2012-01-02 14:30:53 +0000 Tim-Philipp Müller + + * gst-libs/gst/audio/mixerutils.c: + * gst/playback/gstdecodebin2.c: + * gst/playback/gstplaybin2.c: + * gst/playback/gstsubtitleoverlay.c: + * gst/playback/gsturidecodebin.c: + * tests/check/elements/decodebin.c: + * tests/check/elements/libvisual.c: + * tests/check/generic/states.c: + * tests/examples/seek/jsseek.c: + * tests/examples/seek/seek.c: + playback, mixerutils: gst_registry_get_default() -> gst_registry_get() + +2012-01-02 15:03:54 +0100 Wim Taymans + + * gst/audioconvert/audioconvert.c: + * gst/audioconvert/gstchannelmix.c: + audioconvert: handle unpositioned channels + Refuse to convert between unpositioned layouts. + +2012-01-02 15:01:58 +0100 Wim Taymans + + * gst-libs/gst/audio/audio.c: + * gst-libs/gst/audio/audio.h: + audio: add flag for unpositioned layout + Check if thr layout is explicitly unpositioned and set a flag in the + audio info structure. + +2012-01-02 15:00:51 +0100 Wim Taymans + + * tests/check/elements/audioconvert.c: + tests: remove unsupported formats + Remove tests for a format that is no longer supported + +2012-01-02 13:30:53 +0100 Wim Taymans + + * gst-libs/gst/video/video.c: + * gst-libs/gst/video/video.h: + * tests/check/libs/video.c: + video: fix some video formats + Rename the offset field in GstVideoFormatInfo to poffset to avoid confusion with + the offset of the plane in the buffer. The poffset is the offset in the plane + where the first byte of the component data can be found. + Properly implement the COMP_OFFSET calculations. + Fix YV12 and YVU9, simply use the same offsets as the regular I420 and YUV9 + variants, we use the plane info to reorder components already. + Improve the unit test. + +2012-01-02 00:59:39 +0000 Tim-Philipp Müller + + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/tag/lang.c: + * gst-libs/gst/tag/tag.h: + * tests/check/libs/tag.c: + * win32/common/libgsttag.def: + tag: add function to check whether a string is a valid language code + API: gst_tag_check_language_code() + +2011-12-20 21:48:29 +0000 Tim-Philipp Müller + + * gst-libs/gst/audio/multichannel.h: + * gst-libs/gst/rtsp/gstrtspdefs.h: + audio, rtsp: remove private/protected gtk-doc markup for enums + This confuses glib-mkenums, and is not really useful anyway. + https://bugzilla.gnome.org/show_bug.cgi?id=666618 + +2011-12-30 18:36:37 +0100 Stefan Sauer + + * tests/check/elements/volume.c: + * tests/icles/audio-trickplay.c: + controller: port to latest API changes + +2011-12-30 19:26:24 +0000 Tim-Philipp Müller + + * gst-libs/gst/video/gstvideofilter.h: + video: add some padding to GstVideoFilter + +2011-12-30 19:24:09 +0000 Tim-Philipp Müller + + * docs/libs/gst-plugins-base-libs-docs.sgml: + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudioringbuffer.h: + * gst-libs/gst/pbutils/encoding-profile.c: + * gst-libs/gst/video/gstvideofilter.h: + docs: make gtk-doc happier + +2011-12-30 16:47:13 +0000 Tim-Philipp Müller + + * tests/check/libs/audiocdsrc.c: + tests: disable direct structure access in audiocd test + +2011-12-30 16:26:47 +0000 Tim-Philipp Müller + + * gst-libs/gst/audio/gstaudiocdsrc.c: + * gst-libs/gst/audio/gstaudiocdsrc.h: + audiocdsrc: remove some probing-related vfuncs + GstPropertyProbe was removed, so these aren't actually used + and we probably want something different for the new API. + +2011-12-30 16:18:39 +0000 Tim-Philipp Müller + + * gst-libs/gst/audio/gstaudiocdsrc.c: + audiocdsrc: update for GstIndex removal + +2011-12-30 16:12:30 +0000 Tim-Philipp Müller + + * gst-libs/gst/audio/gstaudiocdsrc.c: + * gst-libs/gst/audio/gstaudiocdsrc.h: + audiocdsrc: make private bits private + +2011-12-30 13:21:35 +0100 Edward Hervey + + Merge remote-tracking branch 'origin/master' into 0.11 + Conflicts: + ext/theora/gsttheoraenc.c + gst-libs/gst/tag/gstexiftag.c + gst/adder/gstadder.c + gst/adder/gstadder.h + gst/playback/gstdecodebin2.c + gst/playback/gstsubtitleoverlay.c + tests/check/libs/tag.c + +2011-12-28 16:25:37 +0100 Edward Hervey + + * tests/check/libs/video.c: + check/video: Caps have "interlace-mode=progressive" by default + +2011-12-28 16:24:53 +0100 Edward Hervey + + * tests/check/elements/decodebin.c: + check/decodebin: Fix callback signature + The "gboolean last" argument is gone. + +2011-12-28 16:23:26 +0100 Edward Hervey + + * gst-libs/gst/pbutils/descriptions.c: + pbutils/descriptions: Handle "video/x-raw" without specified format + Without having it raise an assertion, which is valid when asking for + the description of the format. + +2011-12-25 18:07:10 +0100 Wim Taymans + + * gst-libs/gst/video/gstvideopool.c: + * gst-libs/gst/video/gstvideopool.h: + videopool: add support for custom allocators + +2011-12-27 14:37:26 -0300 Thiago Santos + + * ext/ogg/gstoggmux.c: + oggmux: fix leak when initializing pads + Pads are initialized twice: when requesting pads and when + initializing collectpads. Avoid double initialization by + checking if collectpads are still going to be initialized when + creating request pads. + +2011-12-25 23:19:57 +0000 Tim-Philipp Müller + + * android/gdp.mk: + * configure.ac: + * gst/gdp/Makefile.am: + * gst/gdp/dataprotocol.c: + * gst/gdp/dp-private.h: + * gst/gdp/gstgdp.c: + * gst/gdp/gstgdpdepay.c: + * gst/gdp/gstgdppay.c: + * tests/check/Makefile.am: + * tests/check/elements/gdpdepay.c: + * tests/check/elements/gdppay.c: + gdp: move dataprotocol library into gdp plugin and make private + We have removed things like protocol=gdp in the tcp elements + in favour of explicit gdppay/depay elements, so there's no need + to keep a public API and library for now. We can still add it + back later. Someone needs to think hard about 0.11 and gdp + anyway one of these days. + +2011-12-25 23:25:34 +0000 Tim-Philipp Müller + + Merge branch 'merge-dataprotocol-library-into-gdp-plugin' into 0.11 + +2011-12-25 23:10:23 +0000 Tim-Philipp Müller + + * android/tcp.mk: + * gst/tcp/Makefile.am: + * gst/tcp/gstmultifdsink.c: + * gst/tcp/gsttcp.c: + * gst/tcp/gsttcp.h: + * gst/tcp/gsttcpclientsink.c: + * gst/tcp/gsttcpclientsrc.c: + * gst/tcp/gsttcpplugin.c: + * gst/tcp/gsttcpserversink.c: + * gst/tcp/gsttcpserversrc.c: + tcp: remove some dataprotocol cruft + The protocol=gdp property has been removed in favour + of explicit gdppay/depay. + +2011-11-11 17:17:43 +0100 Wim Taymans + + * gst/gdp/dataprotocol.h: + gdp: fix header files + Ensure correct indentation and retab + Make sure all structure have padding + +2011-08-16 17:32:20 +0200 Wim Taymans + + * gst/gdp/dataprotocol.c: + gdp: rename buffer PREROLL -> LIVE flag + Rename the GST_BUFFER_FLAG_PREROLL to GST_BUFFER_FLAG_LIVE and give the new flag + a meaning. The old PREROLL flag never had a clear meaning. + +2011-06-10 13:40:57 +0200 Wim Taymans + + * gst/gdp/dataprotocol.c: + gdp: make new _buffer_allocate method + Make a new method to allocate a buffer + memory that takes the allocator and the + alignment as parameters. Provide a macro for the old method but prefer to use + the new method to encourage plugins to negotiate the allocator properly. + +2011-05-13 18:07:24 +0200 Wim Taymans + + * gst/gdp/dataprotocol.c: + gdp: Rework GstSegment handling + Improve GstSegment, rename some fields. The idea is to have the GstSegment + structure represent the timing structure of the buffers as they are generated by + the source or demuxer element. + gst_segment_set_seek() -> gst_segment_do_seek() + Rename the NEWSEGMENT event to SEGMENT. + Make parsing of the SEGMENT event into a GstSegment structure. + Pass a GstSegment structure when making a new SEGMENT event. This allows us to + pass the timing info directly to the next element. No accumulation is needed in + the receiving element, all the info is inside the element. + Remove gst_segment_set_newsegment(): This function as used to accumulate + segments received from upstream, which is now not needed anymore because the + segment event contains the complete timing information. + +2011-05-10 11:50:16 +0200 Wim Taymans + + * gst/gdp/dataprotocol.c: + gdp: Hide the GstStructure in GstEvent + Hide the GstStructure of the event in the implementation specific part so that + we can change it. + Add methods to check and make the event writable. + Add a new method to get a writable GstStructure of the element. + Avoid directly accising the event structure. + +2011-05-02 16:00:52 +0300 Stefan Kost + + * gst/gdp/dataprotocol.h: + gdp: add docs for GstDPPacketizer + +2011-03-21 18:13:55 +0100 Wim Taymans + + * gst/gdp/dataprotocol.c: + gdp: port code to new buffer data API + +2010-12-06 19:40:03 +0100 Wim Taymans + + * gst/gdp/dataprotocol.c: + * gst/gdp/dataprotocol.h: + gdp: remove deprecated code + +2010-10-08 09:34:47 +0100 Tim-Philipp Müller + + * gst/gdp/dataprotocol.c: + gdp: make public enum _get_type() functions thread-safe + Not that it is likely to matter in practice, but since these are public + API they should probably be thread-safe. + +2010-10-08 00:38:39 +0100 Tim-Philipp Müller + + * gst/gdp/dataprotocol.c: + gdp: dataprotocol, lfocontrolsource: fix enum value name in enums that are public API + So run-time bindings can introspect the names correctly (we abuse this + field as description field only in elements, not for public API + (where the description belongs into the gtk-doc chunk). + https://bugzilla.gnome.org/show_bug.cgi?id=629946 + +2010-03-02 22:58:06 +0100 Benjamin Otte + + * gst/gdp/dataprotocol.c: + gdp: Fixes for -Wmissing-declarations -Wmissing-prototypes + Also adds those flags to the configure warning flags + https://bugzilla.gnome.org/show_bug.cgi?id=611692 + +2010-03-02 23:51:18 +0100 Benjamin Otte + + * gst/gdp/dp-private.h: + gdp: Make code safe for -Wredundant-decls + Adds that warning to configure.ac + Includes a tiny change of the GST_BOILERPLATE_FULL() macro: + The get_type() function is no longer declared before being defined. + https://bugzilla.gnome.org/show_bug.cgi?id=611692 + +2009-11-27 16:39:37 +0200 Stefan Kost + + * gst/gdp/dataprotocol.c: + gdp: fix broken xrefs in docs + +2008-06-30 09:38:45 +0000 Sebastian Dröge + + gdp: Don't write to the same region of memory as a uint64 and uint16 as this breaks ... + Original commit message from CVS: + * libs/gst/dataprotocol/dataprotocol.c: + Don't write to the same region of memory as a uint64 and uint16 + as this breaks strict aliasing rules and apparantly breaks on PPC + and s390. Thanks to Sjoerd Simons for analysing. Fixes bug #348114. + +2008-03-27 15:23:55 +0000 Michael Smith + + gdp: When calculating GDP body CRC, use the correct pointer. + Original commit message from CVS: + * libs/gst/dataprotocol/dataprotocol.c: + (gst_dp_packet_from_event_1_0): + When calculating GDP body CRC, use the correct pointer. + Fixes part of #522401. + +2008-02-29 12:41:33 +0000 Sebastian Dröge + + gdp: Correct all relevant warnings found by the sparse semantic code analyzer. This include marking several symbols static... + Original commit message from CVS: + * gst/gstconfig.h.in: + * libs/gst/base/gstcollectpads.c: (gst_collect_pads_read_buffer): + * libs/gst/check/gstcheck.c: (gst_check_log_message_func), + (gst_check_log_critical_func), (gst_check_drop_buffers), + (gst_check_element_push_buffer_list): + * libs/gst/controller/gstcontroller.c: (gst_controller_get), + (gst_controller_get_type): + * libs/gst/controller/gsthelper.c: (gst_object_control_properties), + (gst_object_get_controller), (gst_object_get_control_source): + * libs/gst/controller/gstinterpolationcontrolsource.c: + (gst_interpolation_control_source_new): + * libs/gst/controller/gstlfocontrolsource.c: + (gst_lfo_control_source_new): + * libs/gst/dataprotocol/dataprotocol.c: + (gst_dp_event_from_packet_0_2): + * plugins/elements/gstfdsrc.c: + * plugins/elements/gstmultiqueue.c: + * plugins/elements/gsttee.c: + * plugins/elements/gsttypefindelement.c: + * plugins/indexers/gstfileindex.c: (_file_index_id_save_xml), + (gst_file_index_add_association): + * plugins/indexers/gstmemindex.c: + * tests/benchmarks/gstpollstress.c: (mess_some_more): + * tests/check/elements/queue.c: (setup_queue): + * tests/check/gst/gstpipeline.c: + * tests/check/libs/collectpads.c: (setup), (teardown), + (gst_collect_pads_suite): + * tests/examples/adapter/adapter_test.c: + * tests/examples/metadata/read-metadata.c: (make_pipeline): + * tests/examples/xml/createxml.c: + * tests/examples/xml/runxml.c: + * tools/gst-inspect.c: + * tools/gst-run.c: + Correct all relevant warnings found by the sparse semantic code + analyzer. This include marking several symbols static, using + NULL instead of 0 for pointers, not using variable sized arrays + on the stack, moving variable declarations to the beginning of + a block and using "foo (void)" instead of "foo ()" for declarations. + +2008-01-08 02:07:38 +0000 Damien Lespiau + + gdp: Fix empty prototypes. Fixes bug #507957. + Original commit message from CVS: + Patch by: Damien Lespiau + * libs/gst/controller/gstcontroller.h: + * libs/gst/controller/gstcontrolsource.h: + * libs/gst/controller/gstinterpolationcontrolsource.h: + * libs/gst/controller/gstlfocontrolsource.h: + * libs/gst/dataprotocol/dataprotocol.h: + Fix empty prototypes. Fixes bug #507957. + +2007-11-01 21:50:05 +0000 Tim-Philipp Müller + + gdp: g_type_class_ref() other types as well, see #349410 and #64764. + Original commit message from CVS: + * gst/gst.c: (init_post): + * gst/gstevent.c: (_gst_event_initialize): + * gst/gstquery.c: (_gst_query_initialize): + * libs/gst/dataprotocol/dataprotocol.c (gst_dp_init): + g_type_class_ref() other types as well, see #349410 and #64764. + * gst/gstbuffer.c: (_gst_buffer_initialize): + * gst/gstmessage.c: (_gst_message_initialize): + Simplify existing g_type_class_ref(). + +2006-10-05 14:26:08 +0000 Tim-Philipp Müller + + gdp: Printf fixes. + Original commit message from CVS: + * gst/gstpad.c: (pre_activate): + * gst/gstregistry.c: (gst_registry_scan_path_level): + * gst/gstregistryxml.c: (load_plugin): + * libs/gst/controller/gstcontroller.c: + (gst_controlled_property_set_interpolation_mode): + * libs/gst/dataprotocol/dataprotocol.c: + (gst_dp_packet_from_event_1_0): + * libs/gst/net/gstnetclientclock.c: + (gst_net_client_clock_observe_times): + * plugins/elements/gstfdsrc.c: (gst_fd_src_create): + Printf fixes. + +2006-08-11 15:26:33 +0000 Andy Wingo + + gdp: GST_DISABLE_DEPRECATED is only for users of API that don't want to see deprecated functions in the headers; people th... + Original commit message from CVS: + 2006-08-11 Andy Wingo + * configure.ac: + * libs/gst/dataprotocol/dataprotocol.c: (gst_dp_packetizer_new): + * tests/check/libs/gdp.c: (gst_dp_suite): GST_DISABLE_DEPRECATED + is only for users of API that don't want to see deprecated + functions in the headers; people that want to compile out + deprecated code should pass -DGST_REMOVE_DEPRECATED into the + CFLAGS. Fixes the build of multifdsink, or will soon.. + +2006-08-10 19:46:14 +0000 Stefan Kost + + gdp: add gst_object_{s,g}et_control_rate(), add private data section, fix docs + Original commit message from CVS: + * docs/libs/gstreamer-libs-sections.txt: + * libs/gst/controller/gstcontroller.c: + (_gst_controller_get_property), (_gst_controller_set_property), + (_gst_controller_init), (_gst_controller_class_init): + * libs/gst/controller/gstcontroller.h: + * libs/gst/controller/gsthelper.c: (gst_object_get_control_rate), + (gst_object_set_control_rate): + API: add gst_object_{s,g}et_control_rate(), add private data section, + fix docs + * libs/gst/dataprotocol/dataprotocol.c: (gst_dp_packetizer_new): + * libs/gst/dataprotocol/dataprotocol.h: + add deprecation guards to make gtk-doc happy and allow disabling cruft + +2006-08-02 15:19:30 +0000 Wim Taymans + + gdp: Make debug category static + Original commit message from CVS: + * libs/gst/dataprotocol/dataprotocol.c: + (gst_dp_header_from_buffer_any), (gst_dp_packet_from_caps_any), + (gst_dp_crc), (gst_dp_header_payload_length), + (gst_dp_header_payload_type), (gst_dp_packet_from_event), + (gst_dp_packet_from_event_1_0), (gst_dp_buffer_from_header), + (gst_dp_caps_from_packet), (gst_dp_event_from_packet_0_2), + (gst_dp_event_from_packet), (gst_dp_validate_header), + (gst_dp_validate_payload): + Make debug category static + Constify the crc table. + Do some more arg checking in public functions. + Fix some docs and do some small cleanups. + * tests/check/libs/gdp.c: (GST_START_TEST), (gst_dp_suite): + Add some more checks to see if GDP deals with bogus input. + +2006-07-13 14:02:16 +0000 Thomas Vander Stichele + + gdp: fix failure to deserialize event packets with empty payload (only ev... + Original commit message from CVS: + * libs/gst/dataprotocol/dataprotocol.c: + (gst_dp_event_from_packet_1_0): + Fixes #347337: failure to deserialize event packets with + empty payload (only event type) + +2006-06-13 19:24:34 +0000 Thomas Vander Stichele + + gdp: add a gdp image to the docs + Original commit message from CVS: + * docs/README: + * docs/images/gdp-header.svg: + add a gdp image + * docs/libs/Makefile.am: + * docs/libs/gdp-header.png: + * libs/gst/dataprotocol/dataprotocol.c: + add it to the API docs + * docs/manual/intro-motivation.xml: + fix typo + +2006-06-06 14:29:54 +0000 Thomas Vander Stichele + + * gst/gdp/dataprotocol.c: + gdp: add note to docs about GDP versioning; remove tmpl file + Original commit message from CVS: + add note to docs about GDP versioning; remove tmpl file + +2006-06-06 14:24:00 +0000 Thomas Vander Stichele + + gdp: add a GstDPPacketizer object, and create/free functions + Original commit message from CVS: + * libs/gst/dataprotocol/dataprotocol.c: + (gst_dp_header_from_buffer_any), (gst_dp_packet_from_caps_any), + (gst_dp_version_get_type), (gst_dp_init), + (gst_dp_header_from_buffer), (gst_dp_header_from_buffer_1_0), + (gst_dp_packet_from_caps), (gst_dp_packet_from_caps_1_0), + (gst_dp_packet_from_event), (gst_dp_packet_from_event_1_0), + (gst_dp_event_from_packet_0_2), (gst_dp_event_from_packet_1_0), + (gst_dp_event_from_packet), (gst_dp_packetizer_new), + (gst_dp_packetizer_free): + * libs/gst/dataprotocol/dataprotocol.h: + API: add a GstDPPacketizer object, and create/free functions + API: add GstDPVersion enum + Add 1.0 event function that uses the string serialization + Serialize more useful buffer flags + Fixes #343988 + +2006-06-02 16:46:19 +0000 Thomas Vander Stichele + + gdp: factor out CRC code + Original commit message from CVS: + * libs/gst/dataprotocol/dataprotocol.c: + (gst_dp_header_from_buffer), (gst_dp_packet_from_caps), + (gst_dp_packet_from_event): + factor out CRC code + +2006-06-02 10:58:47 +0000 Thomas Vander Stichele + + gdp: factor out some common header init code + Original commit message from CVS: + * libs/gst/dataprotocol/dataprotocol.c: + (gst_dp_header_from_buffer), (gst_dp_packet_from_caps), + (gst_dp_packet_from_event): + factor out some common header init code + +2006-06-02 10:08:31 +0000 Thomas Vander Stichele + + gdp: make gst_dp_crc() public + Original commit message from CVS: + * docs/libs/gstreamer-libs-sections.txt: + * docs/libs/tmpl/gstdataprotocol.sgml: + * libs/gst/dataprotocol/dataprotocol.c: (gst_dp_crc): + * libs/gst/dataprotocol/dataprotocol.h: + API: make gst_dp_crc() public + +2006-06-01 11:13:44 +0000 Thomas Vander Stichele + + gdp: make sure we zero the whole ABI-compatible area + Original commit message from CVS: + * libs/gst/dataprotocol/dataprotocol.c: + (gst_dp_header_from_buffer): + make sure we zero the whole ABI-compatible area + +2006-05-08 15:53:12 +0000 Thomas Vander Stichele + + * gst/gdp/dataprotocol.c: + gdp: whitespace, comment, doc fixup + Original commit message from CVS: + whitespace, comment, doc fixup + +2006-04-28 13:40:15 +0000 Michael Smith + + gdp: Fixes in reading/writing events over GDP (not currently used?) - dereferencing ... + Original commit message from CVS: + * libs/gst/dataprotocol/dataprotocol.c: (gst_dp_packet_from_event), + (gst_dp_event_from_packet): + Fixes in reading/writing events over GDP (not currently used?) - + dereferencing NULL events for unknown/invalid event types, memory + leak, and change g_warning to GST_WARNING. + +2006-03-10 15:30:27 +0000 Michael Smith + + gdp: Fix docs for dataprocotol to not get the return types completely wrong for a fe... + Original commit message from CVS: + * libs/gst/dataprotocol/dataprotocol.c: + Fix docs for dataprocotol to not get the return types completely + wrong for a few functions. + +2005-12-18 16:04:41 +0000 Wim Taymans + + gdp: Documentation updates. + Original commit message from CVS: + * libs/gst/base/gstadapter.c: + * libs/gst/base/gstadapter.h: + * libs/gst/base/gstbasesink.c: (gst_base_sink_class_init), + (gst_base_sink_get_position): + * libs/gst/base/gstbasesink.h: + * libs/gst/base/gstbasesrc.c: (gst_base_src_class_init), + (gst_base_src_default_query), (gst_base_src_default_do_seek), + (gst_base_src_do_seek), (gst_base_src_perform_seek), + (gst_base_src_send_event), (gst_base_src_update_length), + (gst_base_src_get_range), (gst_base_src_loop), + (gst_base_src_start): + * libs/gst/base/gstbasesrc.h: + * libs/gst/base/gstbasetransform.h: + * libs/gst/base/gstcollectpads.h: + * libs/gst/base/gstpushsrc.c: + * libs/gst/base/gstpushsrc.h: + * libs/gst/dataprotocol/dataprotocol.c: + * libs/gst/dataprotocol/dataprotocol.h: + * libs/gst/net/gstnetclientclock.h: + * libs/gst/net/gstnettimeprovider.h: + Documentation updates. + +2005-10-13 16:26:12 +0000 Andy Wingo + + gdp: Fix Timmeke Waymans bug. + Original commit message from CVS: + 2005-10-13 Andy Wingo + * libs/gst/dataprotocol/dataprotocol.c (gst_dp_packet_from_caps): + Fix Timmeke Waymans bug. + (gst_dp_caps_from_packet): Make sure we pass a NUL-terminated + string of the proper length to gst_caps_from_string. There's a + potential for, before this fix, that this could cause someone + connecting over the network to cause a segfault if the payload is + not NUL-terminated. + +2005-10-10 23:55:39 +0000 Thomas Vander Stichele + + * gst/gdp/dataprotocol.c: + gdp: fix more valgrind warnings before turning up the heat + Original commit message from CVS: + fix more valgrind warnings before turning up the heat + +2005-10-08 17:17:25 +0000 Wim Taymans + + gdp: It's about time we bump the version number. + Original commit message from CVS: + * libs/gst/dataprotocol/dataprotocol.c: + (gst_dp_header_from_buffer), (gst_dp_packet_from_caps), + (gst_dp_packet_from_event): + * libs/gst/dataprotocol/dataprotocol.h: + * libs/gst/dataprotocol/dp-private.h: + It's about time we bump the version number. + Since event types don't fit in the guint8 anymore describing + the payload type, make payload type 16 bits wide. + +2005-09-27 16:30:26 +0000 Andy Wingo + + gdp: Fix error-checking return values. + Original commit message from CVS: + 2005-09-27 Andy Wingo + * libs/gst/dataprotocol/dataprotocol.c: Fix error-checking return + values. + +2005-07-27 19:00:36 +0000 Wim Taymans + + gdp: Fix serialization of seek events. + Original commit message from CVS: + * libs/gst/dataprotocol/dataprotocol.c: (gst_dp_packet_from_event), + (gst_dp_event_from_packet): + Fix serialization of seek events. + +2005-07-27 18:33:03 +0000 Wim Taymans + + gdp: Some docs updates + Original commit message from CVS: + * CHANGES-0.9: + * docs/design/part-TODO.txt: + * docs/design/part-events.txt: + Some docs updates + * gst/base/gstbasesink.c: (gst_base_sink_handle_object), + (gst_base_sink_event), (gst_base_sink_do_sync), + (gst_base_sink_activate_push), (gst_base_sink_activate_pull): + * gst/base/gstbasesrc.c: (gst_base_src_send_discont), + (gst_base_src_do_seek), (gst_base_src_event_handler), + (gst_base_src_loop): + * gst/base/gstbasetransform.c: (gst_base_transform_transform_caps), + (gst_base_transform_configure_caps), (gst_base_transform_setcaps), + (gst_base_transform_get_size), (gst_base_transform_buffer_alloc), + (gst_base_transform_event), (gst_base_transform_handle_buffer), + (gst_base_transform_set_passthrough), + (gst_base_transform_is_passthrough): + * gst/elements/gstfakesink.c: (gst_fake_sink_event): + * gst/elements/gstfilesink.c: (gst_file_sink_event): + Event updates. + * gst/gstbuffer.h: + Use faster casts. + * gst/gstelement.c: (gst_element_seek): + * gst/gstelement.h: + Update gst_element_seek. + * gst/gstevent.c: (gst_event_finalize), (_gst_event_copy), + (gst_event_new), (gst_event_new_custom), (gst_event_get_structure), + (gst_event_new_flush_start), (gst_event_new_flush_stop), + (gst_event_new_eos), (gst_event_new_newsegment), + (gst_event_parse_newsegment), (gst_event_new_tag), + (gst_event_parse_tag), (gst_event_new_filler), (gst_event_new_qos), + (gst_event_parse_qos), (gst_event_new_seek), + (gst_event_parse_seek), (gst_event_new_navigation): + * gst/gstevent.h: + Make GstEvent use GstStructure. Add parsing code, make sure the + API is sufficiently generic. + Mark possible directions of events and serialization. + * gst/gstmessage.c: (gst_message_init), (gst_message_finalize), + (_gst_message_copy), (gst_message_new_segment_start), + (gst_message_new_segment_done), (gst_message_new_custom), + (gst_message_parse_segment_start), + (gst_message_parse_segment_done): + Small cleanups. + * gst/gstpad.c: (gst_pad_get_caps_unlocked), (gst_pad_accept_caps), + (gst_pad_set_caps), (gst_pad_send_event): + Update for new events. + Catch events sent in wrong directions. + * gst/gstqueue.c: (gst_queue_link_src), + (gst_queue_handle_sink_event), (gst_queue_chain), (gst_queue_loop), + (gst_queue_handle_src_query): + Event updates. + * gst/gsttag.c: + * gst/gsttag.h: + Remove event code from this file. + * libs/gst/dataprotocol/dataprotocol.c: (gst_dp_packet_from_event), + (gst_dp_event_from_packet): + Event updates. + +2005-07-05 10:20:14 +0000 Wim Taymans + + gdp: Ported dataprotol to 0.9. + Original commit message from CVS: + * configure.ac: + * libs/gst/dataprotocol/Makefile.am: + * libs/gst/dataprotocol/dataprotocol.c: (gst_dp_validate_packet): + * libs/gst/dataprotocol/dataprotocol.h: + * pkgconfig/Makefile.am: + * pkgconfig/gstreamer-dataprotocol-uninstalled.pc.in: + * pkgconfig/gstreamer-dataprotocol.pc.in: + Ported dataprotol to 0.9. + Added pkgconfig files. + +2005-05-16 20:21:55 +0000 David Schleef + + gdp: remove GstData checks + Original commit message from CVS: + * check/Makefile.am: remove GstData checks + * check/gst-libs/gdp.c: (START_TEST): fix for API changes + * gst/Makefile.am: add miniobject, remove data + * gst/gst.h: add miniobject, remove data + * gst/gstdata.c: remove + * gst/gstdata.h: remove + * gst/gstdata_private.h: remove + * gst/gsttypes.h: remove GstEvent and GstMessage + * gst/gstelement.c: (gst_element_post_message): fix for API changes + * gst/gstmarshal.list: change BOXED -> OBJECT + Implement GstMiniObject. + * gst/gstminiobject.c: + * gst/gstminiobject.h: + Modify to be subclasses of GstMiniObject. + * gst/gstbuffer.c: (_gst_buffer_initialize), (gst_buffer_get_type), + (gst_buffer_class_init), (gst_buffer_finalize), (_gst_buffer_copy), + (gst_buffer_init), (gst_buffer_new), (gst_buffer_new_and_alloc), + (gst_subbuffer_get_type), (gst_subbuffer_init), + (gst_buffer_create_sub), (gst_buffer_is_span_fast), + (gst_buffer_span): + * gst/gstbuffer.h: + * gst/gstevent.c: (_gst_event_initialize), (gst_event_get_type), + (gst_event_class_init), (gst_event_init), (gst_event_finalize), + (_gst_event_copy), (gst_event_new): + * gst/gstevent.h: + * gst/gstmessage.c: (_gst_message_initialize), + (gst_message_get_type), (gst_message_class_init), + (gst_message_init), (gst_message_finalize), (_gst_message_copy), + (gst_message_new), (gst_message_new_error), + (gst_message_new_warning), (gst_message_new_tag), + (gst_message_new_state_changed), (gst_message_new_application): + * gst/gstmessage.h: + * gst/gstprobe.c: (gst_probe_perform), + (gst_probe_dispatcher_dispatch): + * gst/gstprobe.h: + * gst/gstquery.c: (_gst_query_initialize), (gst_query_get_type), + (gst_query_class_init), (gst_query_finalize), (gst_query_init), + (_gst_query_copy), (gst_query_new): + Update elements for GstData -> GstMiniObject changes + * gst/gstquery.h: + * gst/gstqueue.c: (gst_queue_finalize), (gst_queue_locked_flush), + (gst_queue_chain), (gst_queue_loop): + * gst/elements/gstbufferstore.c: + (gst_buffer_store_add_buffer_func), + (gst_buffer_store_cleared_func), (gst_buffer_store_get_buffer): + * gst/elements/gstfakesink.c: (gst_fakesink_class_init), + (gst_fakesink_render): + * gst/elements/gstfakesrc.c: (gst_fakesrc_class_init): + * gst/elements/gstfilesrc.c: (gst_mmap_buffer_get_type), + (gst_mmap_buffer_class_init), (gst_mmap_buffer_init), + (gst_mmap_buffer_finalize), (gst_filesrc_map_region), + (gst_filesrc_create_read): + * gst/elements/gstidentity.c: (gst_identity_class_init): + * gst/elements/gsttypefindelement.c: + (gst_type_find_element_src_event), (free_entry_buffers), + (gst_type_find_element_handle_event): + * libs/gst/dataprotocol/dataprotocol.c: + (gst_dp_header_from_buffer): + * libs/gst/dataprotocol/dataprotocol.h: + * libs/gst/dataprotocol/dp-private.h: + +2005-05-04 21:29:44 +0000 Andy Wingo + + gdp: GCC 4 fixen. + Original commit message from CVS: + 2005-05-04 Andy Wingo + * check/Makefile.am: + * docs/gst/tmpl/gstatomic.sgml: + * docs/gst/tmpl/gstplugin.sgml: + * gst/base/gstbasesink.c: (gst_basesink_activate): + * gst/base/gstbasesrc.c: (gst_basesrc_class_init), + (gst_basesrc_init), (gst_basesrc_set_dataflow_funcs), + (gst_basesrc_query), (gst_basesrc_set_property), + (gst_basesrc_get_property), (gst_basesrc_check_get_range), + (gst_basesrc_activate): + * gst/base/gstbasesrc.h: + * gst/base/gstbasetransform.c: (gst_base_transform_sink_activate), + (gst_base_transform_src_activate): + * gst/elements/gstelements.c: + * gst/elements/gstfakesrc.c: (gst_fakesrc_class_init), + (gst_fakesrc_set_property), (gst_fakesrc_get_property): + * gst/elements/gsttee.c: (gst_tee_sink_activate): + * gst/elements/gsttypefindelement.c: (find_element_get_length), + (gst_type_find_element_checkgetrange), + (gst_type_find_element_activate): + * gst/gstbin.c: (gst_bin_save_thyself), (gst_bin_restore_thyself): + * gst/gstcaps.c: (gst_caps_do_simplify), (gst_caps_save_thyself), + (gst_caps_load_thyself): + * gst/gstelement.c: (gst_element_pads_activate), + (gst_element_save_thyself), (gst_element_restore_thyself): + * gst/gstpad.c: (gst_pad_load_and_link), (gst_pad_save_thyself), + (gst_ghost_pad_save_thyself), (gst_pad_check_pull_range): + * gst/gstpad.h: + * gst/gstxml.c: (gst_xml_write), (gst_xml_parse_doc), + (gst_xml_parse_file), (gst_xml_parse_memory), + (gst_xml_get_element), (gst_xml_make_element): + * gst/indexers/gstfileindex.c: (gst_file_index_load), + (_file_index_id_save_xml), (gst_file_index_commit): + * gst/registries/gstlibxmlregistry.c: (read_string), (read_uint), + (read_enum), (load_pad_template), (load_feature), (load_plugin), + (load_paths): + * libs/gst/dataprotocol/dataprotocol.c: (gst_dp_packet_from_caps), + (gst_dp_packet_from_event), (gst_dp_caps_from_packet): + * tools/gst-complete.c: (main): + * tools/gst-compprep.c: (main): + * tools/gst-inspect.c: (print_element_properties_info): + * tools/gst-launch.c: (xmllaunch_parse_cmdline): + * tools/gst-xmlinspect.c: (print_element_properties): + GCC 4 fixen. + +2005-03-21 17:34:02 +0000 Wim Taymans + + * gst/gdp/dataprotocol.c: + gdp: Next big merge. + Original commit message from CVS: + Next big merge. + Added GstBus for mainloop integration. + Added GstMessage for sending notifications on the bus. + Added GstTask as an abstraction for pipeline entry points. + Removed GstThread. + Removed Schedulers. + Simplified GstQueue for multithreaded core. + Made _link threadsafe, removed old capsnego. + Added STREAM_LOCK and PREROLL_LOCK in GstPad. + Added pad blocking functions. + Reworked scheduling functions in GstPad to prepare for + scheduling updates soon. + Moved events out of data stream. + Simplified GstEvent types. + Added return values to push/pull. + Removed clocking from GstElement. + Added prototypes for state change function for next merge. + Removed iterate from bins and state change management. + Fixed some elements, disabled others for now. + Fixed -inspect and -launch. + Added check for GstBus. + +2005-03-07 18:27:42 +0000 Wim Taymans + + * gst/gdp/dataprotocol.c: + gdp: First THREADED backport attempt, focusing on adding locks and making sure the API is threadsafe. Needs more work. Mor... + Original commit message from CVS: + First THREADED backport attempt, focusing on adding locks and + making sure the API is threadsafe. Needs more work. More docs + follow this week. + +2005-02-18 13:58:36 +0000 Zaheer Abbas Merali + + gdp: Allocate the 1 byte more memory that was forgotten!!!!! + Original commit message from CVS: + 2005-02-18 Zaheer Abbas Merali + * libs/gst/dataprotocol/dataprotocol.c: (gst_dp_dump_byte_array): + Allocate the 1 byte more memory that was forgotten!!!!! + +2004-10-01 16:49:01 +0000 Wim Taymans + + gdp: Fix threadsafety of the crc checking function. + Original commit message from CVS: + * libs/gst/dataprotocol/dataprotocol.c: (gst_dp_crc): + Fix threadsafety of the crc checking function. + +2004-08-16 10:35:36 +0000 Thomas Vander Stichele + + * gst/gdp/dataprotocol.c: + gdp: fix for #150242 + Original commit message from CVS: + fix for #150242 + +2004-07-28 10:22:07 +0000 Thomas Vander Stichele + + * gst/gdp/dataprotocol.c: + gdp: doc style fixes + Original commit message from CVS: + doc style fixes + +2004-06-09 16:24:19 +0000 Thomas Vander Stichele + + * gst/gdp/dataprotocol.c: + * gst/gdp/dataprotocol.h: + * gst/gdp/dp-private.h: + gdp: bump GDP to 0.1, add buffer flags + Original commit message from CVS: + bump GDP to 0.1, add buffer flags + +2004-05-24 16:38:15 +0000 Thomas Vander Stichele + + * gst/gdp/dataprotocol.h: + gdp: wrap header in _NEW + Original commit message from CVS: + wrap header in _NEW + +2004-05-19 17:22:53 +0000 Thomas Vander Stichele + + * gst/gdp/dataprotocol.c: + * gst/gdp/dp-private.h: + Original commit message from CVS: use GST macros; add asserts + +2004-05-19 16:59:39 +0000 Thomas Vander Stichele + + * gst/gdp/dp-private.h: + gdp: private prototype + Original commit message from CVS: + private prototype + +2004-05-19 16:37:53 +0000 Thomas Vander Stichele + + * gst/gdp/dataprotocol.c: + * gst/gdp/dataprotocol.h: + * gst/gdp/dp-private.h: + gdp: add dataprotocol + Original commit message from CVS: + clean up libs docs; add dataprotocol + +2011-12-25 21:39:39 +0000 Tim-Philipp Müller + + * win32/common/libgstapp.def: + * win32/common/libgstaudio.def: + * win32/common/libgstinterfaces.def: + * win32/common/libgsttag.def: + * win32/common/libgstvideo.def: + win32: update .def files for API changes + +2011-12-25 21:38:21 +0000 Tim-Philipp Müller + + * docs/libs/gst-plugins-base-libs-sections.txt: + * ext/alsa/gstalsasink.c: + * ext/alsa/gstalsasrc.c: + * gst-libs/gst/audio/gstaudioiec61937.c: + * gst-libs/gst/audio/gstaudioringbuffer.c: + * gst-libs/gst/audio/gstaudioringbuffer.h: + audioringbuffer: rename GST_BUFTYPE_* to GST_AUDIO_RING_BUFFER_FORMAT_TYPE_* + Bit unwieldy, but more appropriate. Could also be moved into + audio.h as GstAudioFormatType. + +2011-12-25 21:37:42 +0000 Tim-Philipp Müller + + * tests/check/elements/alsa.c: + * tests/check/libs/gstlibscpp.cc: + * tests/check/libs/libsabi.c: + * tests/check/libs/struct_arm.h: + * tests/check/libs/struct_i386.h: + * tests/check/libs/struct_i386_osx.h: + * tests/check/libs/struct_x86_64.h: + tests: remove more propertyprobe cruft + +2011-12-25 21:23:11 +0000 Tim-Philipp Müller + + * gst-libs/gst/audio/gstaudioringbuffer.h: + audioringbuffer: remove unused GstAudioRingBufferSegState enum and field + +2011-12-25 21:19:04 +0000 Tim-Philipp Müller + + * tests/icles/audio-trickplay.c: + tests: fix unused-variable compiler warning in audio trickplay test + +2011-12-25 21:18:47 +0000 Tim-Philipp Müller + + * docs/plugins/gst-plugins-base-plugins-docs.sgml: + docs: remove references to elements that don't exist any longer + +2011-12-25 19:14:55 +0100 Stefan Sauer + + * tests/icles/audio-trickplay.c: + controller: port to new controlsource api + +2011-12-23 22:51:59 +0000 Tim-Philipp Müller + + * ext/theora/gsttheoraenc.c: + theoraenc: fix template caps creation on big endian systems + +2011-12-23 22:24:44 +0000 Tim-Philipp Müller + + * gst-libs/gst/tag/gstexiftag.c: + * tests/check/libs/tag.c: + tag: fix writing of Exif tag payloads <= 4 bytes + When the payload for an Exif tag is less than or equal to 4 bytes, + the data is simply put into the offset field. Fix writing these + kinds of payloads on big endian systems (and possibly also on + little endian systems). The caller will have already formatted + the bytes in memory according to the writer's endianness, so just + write out the bytes as they are in this case. Fixes tags unit test + on big endian systems. + +2011-12-20 22:58:26 +0100 Stefan Sauer + + * gst/volume/gstvolume.c: + controller: port to new controlbinding api + +2011-12-23 16:09:13 +0100 Wim Taymans + + * ext/theora/gsttheoradec.c: + * ext/theora/gsttheoradec.h: + theoradec: improve cropping + Only add cropping metadata when needed + Remove some used code. + +2011-12-23 00:54:43 +0000 Tim-Philipp Müller + + * ext/alsa/gstalsasink.c: + alsasink: make work for raw audio formats by fixing template caps + +2011-12-22 16:54:18 +0100 Mark Nauwelaerts + + * gst-libs/gst/audio/gstaudioencoder.c: + audioencoder: add a few more debug statements + +2011-12-22 16:53:49 +0100 Mark Nauwelaerts + + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudiodecoder.h: + audiodecoder: tweak documentation + +2011-12-22 16:37:29 +0100 Wim Taymans + + * ext/alsa/gstalsadeviceprobe.h: + * ext/alsa/gstalsamixerelement.c: + * ext/alsa/gstalsasink.c: + * ext/alsa/gstalsasrc.c: + alsa: remove more property probe stuff + +2011-12-22 07:53:39 -0300 Thiago Santos + + * gst-libs/gst/tag/gstxmptag.c: + * tests/check/libs/tag.c: + tag: xmp: Keep compatibility with our old generated xmp + We used to add a trailing \n to the end of generated xmp packets. + Windows viewer was unhappy with it and we fixed it in + 96d2120c2bb0b29e1849098198f5fbef81939cdd + The problem is that this caused xmp generated before this fix + to not be recognized and parsed anymore. This patch makes it + recognize xmp with the trailing \n and without, fixing the + regression. Also adds tests for it. + +2011-12-21 23:46:53 +0100 Wim Taymans + + * gst-libs/gst/video/gstvideofilter.c: + * gst-libs/gst/video/gstvideofilter.h: + * gst/videoconvert/gstvideoconvert.c: + * gst/videoconvert/gstvideoconvert.h: + * gst/videoscale/gstvideoscale.c: + * gst/videoscale/gstvideoscale.h: + videofilter: improve video filter + Flesh out the video filter base class. Make it parse the input and output caps + and turn them into GstVideoInfo. Map buffers as video frames and pass them to + the transform functions. + This allows us to also implement the propose and decide_allocation vmethods. + Implement the transform size method as well. + Update subclasses with the new improvements. + +2011-12-21 18:58:42 +0100 Wim Taymans + + * gst/videoconvert/gstvideoconvert.c: + * gst/videoscale/gstvideoscale.c: + * gst/videoscale/gstvideoscale.h: + videofilter: implement propose_allocation + With the new video bufferpool we can now implement the propose_allocation + vmethod on some video filter elements so that we can also use video metadata and + bufferpools when not operating in passthrough mode. + +2011-12-21 18:58:08 +0100 Wim Taymans + + * docs/plugins/gst-plugins-base-plugins-sections.txt: + docs: small fixes + +2011-12-21 18:14:45 +0100 Wim Taymans + + * sys/ximage/ximagepool.c: + * sys/xvimage/xvimagepool.c: + x11: reset alignment + +2011-12-21 18:13:17 +0100 Wim Taymans + + * gst-libs/gst/video/gstvideopool.c: + * gst-libs/gst/video/gstvideopool.h: + videopool: add videopool implementation + Add a GstVideoPool object that can be used to allocate video frames with support + for metadata and alignment. + Add method to reset alignment info. + +2011-12-21 11:58:53 +0100 Wim Taymans + + * docs/libs/gst-plugins-base-libs-sections.txt: + * docs/libs/gst-plugins-base-libs.types: + * ext/alsa/gstalsadeviceprobe.c: + * gst-libs/gst/audio/mixerutils.c: + * gst-libs/gst/interfaces/Makefile.am: + * gst-libs/gst/interfaces/propertyprobe.c: + * gst-libs/gst/interfaces/propertyprobe.h: + * gst-libs/gst/pbutils/encoding-profile.c: + * gst-libs/gst/video/video-overlay-composition.c: + * gst-libs/gst/video/video.h: + * sys/xvimage/xvimagesink.c: + * tests/icles/test-colorkey.c: + propertyprobe: remove propertyprobe + Remove the propertyprobe interface + Improve docs + +2011-12-14 16:34:39 +0000 Vincent Penquerc'h + + * gst-libs/gst/video/video-blend.c: + gstvideo: fix a RGB ordering mixup in colorspace conversion code + +2011-12-19 17:41:23 +0100 Oleksij Rempel (Alexey Fisher) + + * ext/theora/gsttheoraenc.c: + * ext/theora/gsttheoraenc.h: + theoraenc: add "dup-on-gap" option + This option will produce duplicate frames if we get + a frame with GAP flag. This will reduce CPU load and file size. + This option should be disabled for real time applications, because it + collects GAP frames and waits until it gets a non GAP frame to start + encoding. + v30.06.2011: make some spell changes. + v03.07.2011: add handling of EOS and discontinuous for dup-on-gap. + v19.12.2011: fix pointer dangling in theora_timefifo_free + v20.12.2010: fix timestamp bug for dup-on-gap=0 + Bugzilla: https://bugzilla.gnome.org/show_bug.cgi?id=627459 + Signed-off-by: Oleksij Rempel (Alexey Fisher) + +2011-12-20 14:35:31 +0100 Sebastian Dröge + + * gst-libs/gst/audio/gstaudiobasesrc.c: + audiobasesrc: Use guint8 instead of guchar + +2011-12-20 14:34:50 +0100 Sebastian Dröge + + * gst-libs/gst/audio/gstaudioringbuffer.c: + * gst-libs/gst/audio/gstaudioringbuffer.h: + audioringbuffer: Use guint8 instead of guchar + +2011-12-20 13:26:10 +0100 Wim Taymans + + * docs/design/part-mediatype-audio-raw.txt: + docs: small update + +2011-12-20 12:53:16 +0100 Wim Taymans + + * gst/playback/gstsubtitleoverlay.c: + subtitle: don't use GST_CAPS_NONE macro + This macro returns a singleton. + +2011-12-20 12:42:18 +0100 Mark Nauwelaerts + + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudiodecoder.h: + audiodecoder: set a non-zero default maximum tolerated errors + Whereas the previous default 0 was backwards compatible in that it lead + to erroring out immediately upon any error, elements that are really + ported and using the base class error macro can be assumed to intend to + improve behaviour rather than maintaining the old one. So, make it easy + on those and any future one and tolerate some errors by default, as intended. + Fixes #666579. + +2011-12-20 12:02:25 +0100 Wim Taymans + + * docs/design/part-mediatype-audio-raw.txt: + * docs/design/part-mediatype-video-raw.txt: + * gst-libs/gst/audio/Makefile.am: + * gst-libs/gst/audio/gstaudiometa.c: + * gst-libs/gst/audio/gstaudiometa.h: + add audio metadata + Add some audio metadata to describe a downmix matrix. + Add metadata to media type document. + +2011-12-20 10:58:19 +0100 Wim Taymans + + * docs/design/part-mediatype-audio-raw.txt: + * docs/design/part-mediatype-video-raw.txt: + docs: update media design docs some more + Add audio media type design doc + +2011-12-20 10:08:46 +0100 Wim Taymans + + * docs/design/design-audiosinks.txt: + * docs/design/draft-media-types.txt: + * docs/design/part-interlaced-video.txt: + * docs/design/part-mediatype-video-raw.txt: + * docs/design/part-playbin.txt: + * docs/design/part-playbin2.txt: + docs: small update to design docs + +2011-12-19 23:41:25 +0100 Stefan Sauer + + * tests/check/elements/volume.c: + * tests/icles/audio-trickplay.c: + controller: port to new interpolation-mode api + +2011-12-19 22:51:47 +0100 Stefan Sauer + + * tests/check/elements/volume.c: + * tests/icles/audio-trickplay.c: + controller: port to new controller api + +2011-12-19 18:03:45 +0100 Wim Taymans + + * docs/design/draft-media-types.txt: + * gst-libs/gst/video/video.c: + * gst-libs/gst/video/video.h: + video: update interlace caps and docs + Remove interlaced boolean from caps and replace with an interlace-mode enum. + document this new property in the video caps document. With the enum we can + put fields into separate video meta. + Add enum for this interlace-mode in the VideoInfo. + Update the buffer flags. + +2011-12-19 11:03:55 +0100 Wim Taymans + + * gst-libs/gst/tag/gsttagdemux.c: + tagdemux: add FIXME + Add a FIXME because the EOS before-type case now has to be solved differently + because the srcpad is always available. + +2011-12-19 09:49:07 +0100 Wim Taymans + + * tests/examples/seek/jsseek.c: + * tests/examples/seek/seek.c: + * tests/examples/seek/stepping.c: + * tests/examples/seek/stepping2.c: + use playbin instead of playbin2 + +2011-12-16 17:32:41 +0000 Vincent Penquerc'h + + * gst/adder/gstadder.c: + adder: do not send too many flush-stop events + GstCollectPads2 now allows us to override the event function, + so we can withhold flush stop events if none are to be sent. + https://bugzilla.gnome.org/show_bug.cgi?id=666379 + +2011-12-16 17:31:06 +0000 Vincent Penquerc'h + + * gst/adder/gstadder.c: + adder: use the stream lock where appropriate + GstCollectPads2 locking was changed from GstCollectPads to use + the stream lock instead of the object lock for those cases, so + change it so here as well to match. + https://bugzilla.gnome.org/show_bug.cgi?id=666379 + +2011-12-16 17:25:10 +0000 Vincent Penquerc'h + + * gst/adder/gstadder.c: + adder: send a flush event before trying to get the stream lock + This avoids hanging when the streaming thread is busy in _chain + waiting for preroll. + https://bugzilla.gnome.org/show_bug.cgi?id=666379 + +2011-12-16 15:27:24 +0000 Vincent Penquerc'h + + * ext/ogg/gstoggdemux.c: + oggdemux: assume live stream if byte size cannot be determined + This prevents trying to seek and failing, then ending up unable + to stream because we can't get back at the headers. + A more robust way would be to find a good place to reinject the + headers when a seek fails, but I can't seem to get this to work. + +2011-12-15 11:01:01 -0300 Thiago Santos + + * gst-libs/gst/tag/gstexiftag.c: + tag: exif: do not include \0 in size passed to g_convert + When using g_convert, we should only pass the length + of the string content (without the \0) as g_convert will + only parse the real contents when changing formats. Including + the \0 causes it to add another \0, increasing the string + size when not needed. + For example, when writting a North geo location ref entry, that should + be a string with a single N letter, it would write: + "N\0\0", causing the string to have size 3, instead of 2 as expected. + In our case, we can pass -1 and let g_convert calculate the strlen as + we don't use the length anywhere else. + This fixes jifmux's tests on gst-plugins-bad. + +2011-12-14 18:26:07 +0000 Vincent Penquerc'h + + * gst/adder/gstadder.c: + * gst/adder/gstadder.h: + adder: port to GstCollectPads2 + +2011-12-14 17:34:55 +0000 Christian Fredrik Kalager Schaller + + * gst-libs/gst/pbutils/encoding-profile.c: + Fix 666168, add missing allow-None to encodebin APIs + +2011-10-03 14:51:56 +0200 Mark Nauwelaerts + + * gst/playback/gstdecodebin2.c: + decodebin2: tweak chain topology description + ... to also properly indicate chain's endpad if no elements are in the + chain (due to the endpad being a raw demuxer pad, or one setup without + decoders since uridecodebin or higher up decided not to need those). + +2011-12-14 12:28:26 +0000 Tim-Philipp Müller + + * gst-libs/gst/pbutils/encoding-profile.c: + encoding-profile: add some missing allow-none g-i annotations + Fix gst_encoding_container_profile_new() annotations. + https://bugzilla.gnome.org/show_bug.cgi?id=666096 + +2011-12-14 11:31:31 +0100 Stefan Sauer + + * gst-libs/gst/riff/riff-media.c: + riff-media: port GST_BUFFER_DATA to 0.11 in conditional code branch + +2011-12-13 12:55:45 +0000 Vincent Penquerc'h + + * gst-libs/gst/audio/gstbaseaudiosink.c: + baseaudiosink: fix late buffer leak + +2011-12-13 13:28:47 +0100 Sebastian Dröge + + * gst/playback/gstsubtitleoverlay.c: + subtitleoverlay: Refactor code to check if a property exists on an element + +2011-12-13 13:20:24 +0100 Sebastian Dröge + + * gst/playback/gstsubtitleoverlay.c: + subtitleoverlay: Refactor autoplugging code and select overlay element by rank too + Previously we always used textoverlay for rendering the output of + a parser, now the same code as for the renderers is used and the + element with the highest rank is used. + Fixes bug #663822. + +2011-12-12 11:54:56 +0100 Sebastian Dröge + + * gst-libs/gst/glib-compat-private.h: + glib-compat: Add license boilerplate for LGPL + +2011-12-12 17:27:10 +0000 Christian Fredrik Kalager Schaller + + * gst-plugins-base.spec.in: + Update file locations for 0.11 + +2011-12-12 13:02:01 +0000 Tim-Philipp Müller + + * po/cs.po: + * po/es.po: + * po/sr.po: + po: update translations + +2011-12-12 12:59:44 +0000 Tim-Philipp Müller + + * configure.ac: + Require gobject-introspection >= 1.31.1 + Same as core. + +2011-12-12 12:40:17 +0000 Tim-Philipp Müller + + Merge remote-tracking branch 'origin/master' into 0.11 + Conflicts: + gst-plugins-base.spec.in + po/LINGUAS + po/cs.po + po/eo.po + po/es.po + po/gl.po + po/lv.po + po/sr.po + +2011-12-10 01:36:14 +0000 Tim-Philipp Müller + + * po/LINGUAS: + * po/cs.po: + * po/eo.po: + * po/es.po: + * po/gl.po: + * po/lv.po: + * po/sr.po: + po: update translations + +2011-12-09 19:21:09 +0100 Wim Taymans + + * gst-libs/gst/rtsp/gstrtsptransport.c: + rtsp: use rtpbin + +2011-12-09 10:49:33 +0100 Wim Taymans + + * gst-libs/gst/rtp/gstrtcpbuffer.h: + rtp: add INIT macros + +2011-12-09 15:39:12 +0000 Christian Fredrik Kalager Schaller + + * gst-plugins-base.spec.in: + Add latest header file to spec file + +2011-12-09 15:06:33 +0000 Tim-Philipp Müller + + * tests/check/libs/video.c: + tests: disable composition tests in video unit test for now + +2011-12-09 15:03:41 +0000 Tim-Philipp Müller + + * gst-libs/gst/rtp/gstrtpbuffer.h: + rtpbuffer: add GST_RTP_BUFFER_INIT to initialize RTP buffers on the stack + Fixes build of -good. + +2011-12-09 12:08:37 +0000 Tim-Philipp Müller + + Merge remote-tracking branch 'origin/master' into 0.11 + +2011-12-09 01:31:20 +0000 Tim-Philipp Müller + + * gst/typefind/gsttypefindfunctions.c: + typefindfunctions: only typefind text with a BOM as text/utf16 or text/utf32 + We added the utf typefinder because the mp3 typefinder was a tad + overzealous when it came to typefinding things as mp3, and replaced + it with even more overzealous utf16/32 typefinders. + Fixes unit test. + +2011-12-08 01:20:24 +0000 Tim-Philipp Müller + + * tests/check/libs/audiocdsrc.c: + Revert "tests: fix audiocdsrc for changed preroll behaviour" + This reverts commit 2c9d442d51dd681463ae090c3c57320a90a4f888. + Behaviour changed again, so revert this. + +2011-12-08 01:19:03 +0000 Tim-Philipp Müller + + Merge remote-tracking branch 'origin/master' into 0.11 + Conflicts: + ext/alsa/gstalsadeviceprobe.c + ext/alsa/gstalsamixer.c + ext/pango/gsttextoverlay.c + ext/pango/gsttextoverlay.h + gst-libs/gst/audio/gstaudiobasesink.c + gst-libs/gst/audio/gstaudioringbuffer.c + gst-libs/gst/audio/gstaudiosrc.c + gst-libs/gst/video/Makefile.am + gst-libs/gst/video/video.c + gst/encoding/gststreamcombiner.c + gst/encoding/gststreamsplitter.c + gst/playback/gstplaybasebin.c + gst/playback/gststreamsynchronizer.c + gst/playback/gstsubtitleoverlay.c + gst/playback/gsturidecodebin.c + sys/xvimage/xvimagesink.c + tests/examples/Makefile.am + win32/common/libgstvideo.def + Video overlay composition disabled for now, needs + porting to buffer meta. + +2011-12-07 18:45:28 +0000 Tim-Philipp Müller + + * gst-libs/gst/video/video-overlay-composition.c: + * gst-libs/gst/video/video-overlay-composition.h: + video: make composition_blend() return a boolean + Not that anyone will ever check that, and it's not clear what + they're supposed to do if it fails, but at least it's there. + +2011-12-07 18:31:58 +0000 Tim-Philipp Müller + + * docs/libs/gst-plugins-base-libs-docs.sgml: + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/video/video-overlay-composition.c: + * gst-libs/gst/video/video-overlay-composition.h: + docs: add new API to docs + +2011-12-07 17:57:08 +0000 Tim-Philipp Müller + + * gst-libs/gst/video/video-overlay-composition.c: + * gst-libs/gst/video/video-overlay-composition.h: + * tests/check/libs/video.c: + * win32/common/libgstvideo.def: + video: add seqnum getters for overlay compositions and rectangles + API: gst_video_overlay_composition_get_seqnum() + API: gst_video_overlay_rectangle_get_seqnum() + +2011-11-23 15:45:57 -0300 Thibault Saunier + + * gst-libs/gst/video/video.c: + video: support any type of video in _parse_caps + Slight change in semantics for convenience. Shouldn't cause any + problems since this function is usually only used on pre-filtered + caps and not random caps, and it's hard to imagine a situation + where someone would want to rely on the previous behaviour. + +2011-12-06 21:57:32 +0000 Tim-Philipp Müller + + * gst/videorate/gstvideorate.c: + videorate: don't leak previous buffer when shutting down + Implement stop vfunc after port to basetransform, so we + can clean up properly. Fixes make elements/videorate.valgrind + +2011-12-06 20:30:55 +0000 Tim-Philipp Müller + + * tests/check/libs/video.c: + tests: fix calculation of last pixel offset in video unit test + And check the right buffer (pix2) in one case. + +2011-12-06 15:01:05 +0000 Tim-Philipp Müller + + * tests/examples/fft/Makefile.am: + examples: fix build of fft example + Should link against our own libgstfft-0.10. + +2011-12-06 14:55:38 +0000 Tim-Philipp Müller + + * gst-libs/gst/video/video.c: + video: fix leak in gst_video_format_new_template_caps() + g_value_reset() is not the same as g_value_unset() + +2011-12-06 15:06:12 +0100 Wim Taymans + + * gst-libs/gst/audio/gstaudiobasesink.c: + * gst-libs/gst/audio/gstaudioringbuffer.c: + * gst-libs/gst/audio/gstaudioringbuffer.h: + ringbuffer: remove old _full version + +2011-12-06 13:59:11 +0100 Wim Taymans + + * gst-libs/gst/audio/gstaudiocdsrc.c: + fix for basesrc changes + +2011-11-23 15:43:46 -0300 Thibault Saunier + + * gst/playback/gstsubtitleoverlay.c: + subtitleoverlay: add suport for hardware accelerated videos + Don't plug converters for non-raw video. + +2011-12-06 08:37:32 +0100 Stefan Sauer + + * gst/volume/gstvolume.c: + controller: port to GstValueArray removal API change + +2011-12-05 20:33:41 +0100 Wim Taymans + + * gst/videoconvert/gstvideoconvert.c: + Revert "videoconvert: We can handle GST_VIDEO_META_API" + This reverts commit bd539753eb098c37afa033065f122712bf85f53a. + Adding the supported metadata to the query does nothing at this stage. Proposing + allocation parameters and supported metadata for upstream should use the + propose_allocation vmethod. + +2011-12-05 18:42:24 +0100 Edward Hervey + + * gst-libs/gst/rtp/gstrtpbaseaudiopayload.c: + * gst-libs/gst/rtp/gstrtpbasedepayload.c: + rtp: Initialize GstRTPBuffer before usage + +2011-12-05 18:30:50 +0100 Edward Hervey + + * gst/videoconvert/gstvideoconvert.c: + videoconvert: We can handle GST_VIDEO_META_API + +2011-12-05 18:30:37 +0100 Edward Hervey + + * gst-libs/gst/rtp/gstrtpbasepayload.c: + rtp: Don't forget to initialize GstRTPBuffer + +2011-12-05 15:48:07 +0000 Tim-Philipp Müller + + * gst-libs/gst/video/video-overlay-composition.c: + video: don't use deprecated GStaticMutex with newer glib versions + +2011-12-05 15:34:42 +0000 Tim-Philipp Müller + + * tests/examples/Makefile.am: + examples: dist fft sub-directory + +2011-11-28 10:05:50 -0300 Thibault Saunier + + * ext/pango/gsttextoverlay.c: + textoverlay: unpremultiply text image + The GstVideoOverlayComposition only supports unpremultiplied ARGB + (for now anyway, support for pre-multiplied alpha is planned.) + +2011-11-23 12:49:02 -0300 Thibault Saunier + + * ext/pango/gsttextoverlay.c: + * ext/pango/gsttextoverlay.h: + textoverlay: Attach OverlayComposition to buffers when needed + Add video/x-surface support in the caps + We should then attach it whenever the sink supports it, but this + is working for the time being + +2011-11-18 13:22:52 -0300 Thibault Saunier + + * ext/pango/gsttextoverlay.c: + * ext/pango/gsttextoverlay.h: + textoverlay: Make the text_image data a buffer + This way we won't free data that would be attached to some buffer. + +2011-11-18 11:04:47 -0300 Thibault Saunier + + * ext/pango/gsttextoverlay.c: + textoverlay: Sync the caps with the new supported formats + Thanks to the use of the new video composition library, we gain support to + more colospaces and formats, let's state it. + +2011-11-16 17:54:43 -0300 Thibault Saunier + + * ext/pango/gsttextoverlay.c: + * ext/pango/gsttextoverlay.h: + textoverlay: Make use of the new video blending utility + +2011-11-25 16:46:09 +0000 Tim-Philipp Müller + + * tests/check/libs/video.c: + tests: add basic unit test for video overlay composition and rectangles + +2011-11-12 14:59:35 +0000 Tim-Philipp Müller + + * gst-libs/gst/video/Makefile.am: + * gst-libs/gst/video/video-overlay-composition.c: + * gst-libs/gst/video/video-overlay-composition.h: + * win32/common/libgstvideo.def: + video: add video overlay composition API for subtitles + Basic API to attach overlay rectangles to buffers, + or blend them directly onto raw video buffers. + To be used primarily for things like subtitles or + logo overlays, not meant to replace videomixer. + Allows us to associate subtitle overlays with + non-raw video surface buffers, so that subtitles + are not lost and can instead be rendered later + when those surfaces are displayed or converted, + whilst re-using all the existing overlay plugins + and not having to teach them about our special + video surfaces. Could also have been made part + of the surface buffer abstraction of course, but + a secondary goal was to consolidate the blending + code for raw video into libgstvideo, and this + kind of API allows us to do both in a way that's + minimally invasive to existing elements, and at + the same time is fairly intuitive. + More features and extensions like the ability to + pass the source data or text/markup directly will + be added later. + https://bugzilla.gnome.org/show_bug.cgi?id=665080 + API: gst_video_buffer_get_overlay_composition() + API: gst_video_buffer_set_overlay_composition() + API: gst_video_overlay_composition_new() + API: gst_video_overlay_composition_add_rectangle() + API: gst_video_overlay_composition_n_rectangles() + API: gst_video_overlay_composition_get_rectangle() + API: gst_video_overlay_composition_make_writable() + API: gst_video_overlay_composition_copy() + API: gst_video_overlay_composition_ref() + API: gst_video_overlay_composition_unref() + API: gst_video_overlay_composition_blend() + API: gst_video_overlay_rectangle_new_argb() + API: gst_video_overlay_rectangle_get_pixels_argb() + API: gst_video_overlay_rectangle_get_pixels_unscaled_argb() + API: gst_video_overlay_rectangle_get_render_rectangle() + API: gst_video_overlay_rectangle_set_render_rectangle() + API: gst_video_overlay_rectangle_copy() + API: gst_video_overlay_rectangle_ref() + API: gst_video_overlay_rectangle_unref() + +2011-11-23 00:31:18 +0000 Tim-Philipp Müller + + * gst-libs/gst/video/Makefile.am: + * gst-libs/gst/video/video-blend.h: + video: hide private video-blend.[ch] from gobject-introspection + And remove unused fields from helper structure. + +2011-11-15 18:00:00 +0000 Tim-Philipp Müller + + * gst-libs/gst/video/videoblendorc-dist.c: + * gst-libs/gst/video/videoblendorc-dist.h: + video: add fallbacks for compilation without orc + +2011-10-17 17:25:11 +0200 Thibault Saunier + + * gst-libs/gst/video/.gitignore: + * gst-libs/gst/video/Makefile.am: + * gst-libs/gst/video/video-blend.c: + * gst-libs/gst/video/video-blend.h: + * gst-libs/gst/video/videoblendorc.orc: + video: add some internal helper functions for image blending + This could be improved if we decide we don't need it to + be this generic/flexible. + +2011-12-05 09:38:33 +0100 Sebastian Dröge + + * gst-libs/gst/interfaces/xoverlay.c: + xoverlay: Fix mistakes in the sample code + Fixes bug #665430. + +2011-12-04 22:19:23 +0100 Matej Knopp + + * gst-libs/gst/app/gstappsink.c: + Appsink fixes + +2011-12-04 20:50:25 +0000 Tim-Philipp Müller + + * ext/alsa/gstalsamixer.c: + * ext/ogg/gstoggdemux.c: + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/audio/gstbaseaudiosink.c: + * gst/playback/gstdecodebin.c: + * gst/playback/gstdecodebin2.c: + * gst/playback/gstplaybin2.c: + * gst/playback/gstplaysink.c: + * gst/playback/gststreamsynchronizer.c: + * gst/tcp/gstmultifdsink.c: + Suppress deprecation warnings in selected files, for g_static_rec_mutex_* mostly + GStaticRecMutex is part of our API/ABI, not much we can do here + in 0.10 for most of these. + +2011-12-04 20:38:19 +0000 Tim-Philipp Müller + + * ext/alsa/gstalsamixer.c: + * ext/alsa/gstalsamixer.h: + alsamixer: use GRectMutext instead of GStaticRecMutex with newer glib versions + +2011-12-04 20:21:26 +0000 Tim-Philipp Müller + + * ext/alsa/gstalsamixer.c: + * ext/alsa/gstalsamixer.h: + alsamixer: embed static mutexes into the mixer structure + instead of allocating them dynamically + +2011-12-04 17:02:39 +0000 Tim-Philipp Müller + + * tests/examples/encoding/encoding.c: + * tests/examples/overlay/gtk-xoverlay.c: + * tests/examples/overlay/qt-xoverlay.cpp: + * tests/examples/seek/jsseek.c: + * tests/examples/seek/scrubby.c: + * tests/examples/seek/seek.c: + * tests/icles/stress-playbin.c: + * tests/icles/test-colorkey.c: + * tests/icles/test-xoverlay.c: + * tools/gst-discoverer.c: + tools, tests: g_thread_init() is deprecated in glib master + It's not needed any longer. + +2011-12-04 16:43:38 +0000 Tim-Philipp Müller + + * ext/alsa/gstalsadeviceprobe.c: + * ext/alsa/gstalsamixer.c: + * ext/alsa/gstalsasink.c: + * ext/alsa/gstalsasrc.c: + * ext/ogg/gstoggdemux.c: + * ext/pango/gsttextoverlay.c: + * gst-libs/gst/Makefile.am: + * gst-libs/gst/app/gstappsink.c: + * gst-libs/gst/app/gstappsrc.c: + * gst-libs/gst/audio/gstaudiosink.c: + * gst-libs/gst/audio/gstaudiosrc.c: + * gst-libs/gst/audio/gstringbuffer.c: + * gst-libs/gst/glib-compat-private.h: + * gst-libs/gst/pbutils/gstdiscoverer.c: + * gst-libs/gst/rtsp/gstrtspconnection.c: + * gst-libs/gst/video/convertframe.c: + * gst/encoding/gststreamcombiner.c: + * gst/encoding/gststreamsplitter.c: + * gst/playback/gstdecodebin.c: + * gst/playback/gstdecodebin2.c: + * gst/playback/gstplaybasebin.c: + * gst/playback/gstplaybin2.c: + * gst/playback/gstplaysinkconvertbin.c: + * gst/playback/gststreamsynchronizer.c: + * gst/playback/gstsubtitleoverlay.c: + * gst/playback/gsturidecodebin.c: + * gst/tcp/gstmultifdsink.c: + * sys/ximage/ximagesink.c: + * sys/xvimage/xvimagesink.c: + Work around deprecated thread API in glib master + Add private replacements for deprecated functions such as + g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly + to avoid the deprecation warnings. We'll change these + over to the new API once we depend on glib >= 2.32. + Replace g_thread_create() with g_thread_try_new(). + +2011-12-04 15:23:21 +0000 Tim-Philipp Müller + + * gst-libs/gst/tag/xmpwriter.c: + xmpwriter: update for thread API deprecations in glib master + +2011-12-04 13:43:06 +0100 Stefan Sauer + + * tests/examples/fft/Makefile.am: + fft-example: re-add Makefile.am + +2011-12-02 23:35:50 +0100 Stefan Sauer + + * configure.ac: + configure: trim trailing whitespace + +2011-12-02 23:34:47 +0100 Stefan Sauer + + * configure.ac: + * tests/examples/Makefile.am: + * tests/examples/fft/.gitignore: + * tests/examples/fft/fftrange.c: + tests: add a test for fft result value-ranges + Add a small example that uses ffts of various types and parameters and check the + result value ranges. + +2011-12-02 22:24:43 +0100 Wim Taymans + + * ext/gio/gstgiobasesink.c: + * gst-libs/gst/app/gstappsink.c: + * gst-libs/gst/audio/gstaudiobasesink.c: + * sys/ximage/ximagesink.c: + * sys/xvimage/xvimagesink.c: + update for basesink event handler changes + +2011-12-02 11:10:17 +0000 Tim-Philipp Müller + + Merge remote-tracking branch 'origin/master' into 0.11 + Conflicts: + gst-libs/gst/netbuffer/gstnetbuffer.c + gst/ffmpegcolorspace/avcodec.h + gst/ffmpegcolorspace/gstffmpegcodecmap.c + gst/ffmpegcolorspace/imgconvert.c + gst/ffmpegcolorspace/imgconvert_template.h + gst/ffmpegcolorspace/mem.c + gst/playback/README + gst/playback/gstplaybasebin.c + gst/playback/gstplaybasebin.h + gst/playback/gstplaybin.c + sys/v4l/v4lmjpegsrc_calls.c + sys/v4l/videodev_mjpeg.h + tests/check/elements/gnomevfssink.c + +2011-09-13 21:10:43 +0200 Piotr Fusik + + * docs/design/design-audiosinks.txt: + * docs/design/design-decodebin.txt: + * docs/design/design-encoding.txt: + * docs/design/design-orc-integration.txt: + * docs/design/draft-keyframe-force.txt: + * docs/design/draft-va.txt: + * ext/alsa/gstalsamixer.c: + * ext/libvisual/visual.c: + * ext/ogg/README: + * ext/ogg/gstoggdemux.c: + * ext/theora/gsttheoradec.c: + * ext/theora/gsttheoradec.h: + * ext/theora/gsttheoraparse.c: + * ext/vorbis/gstvorbisdec.c: + * gst-libs/gst/app/gstappsink.c: + * gst-libs/gst/app/gstappsrc.c: + * gst-libs/gst/app/gstappsrc.h: + * gst-libs/gst/audio/audio.c: + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/audio/gstbaseaudiosink.c: + * gst-libs/gst/audio/gstbaseaudiosrc.c: + * gst-libs/gst/audio/gstringbuffer.c: + * gst-libs/gst/audio/multichannel.h: + * gst-libs/gst/fft/gstfftf32.c: + * gst-libs/gst/fft/gstfftf64.c: + * gst-libs/gst/fft/gstffts16.c: + * gst-libs/gst/fft/gstffts32.c: + * gst-libs/gst/interfaces/navigation.c: + * gst-libs/gst/interfaces/xoverlay.c: + * gst-libs/gst/netbuffer/gstnetbuffer.c: + * gst-libs/gst/pbutils/descriptions.c: + * gst-libs/gst/pbutils/encoding-profile.c: + * gst-libs/gst/pbutils/encoding-target.h: + * gst-libs/gst/pbutils/gstdiscoverer-types.c: + * gst-libs/gst/pbutils/gstdiscoverer.c: + * gst-libs/gst/rtp/gstbasertpaudiopayload.c: + * gst-libs/gst/rtp/gstrtcpbuffer.c: + * gst-libs/gst/rtp/gstrtpbuffer.c: + * gst-libs/gst/rtsp/gstrtspconnection.c: + * gst-libs/gst/rtsp/gstrtsprange.c: + * gst-libs/gst/tag/gstexiftag.c: + * gst-libs/gst/tag/gstvorbistag.c: + * gst-libs/gst/tag/gstxmptag.c: + * gst-libs/gst/tag/id3v2.3.0.txt: + * gst-libs/gst/tag/id3v2.4.0-frames.txt: + * gst-libs/gst/tag/id3v2.4.0-structure.txt: + * gst/adder/gstadder.c: + * gst/audioconvert/audioconvert.c: + * gst/audiorate/gstaudiorate.c: + * gst/audioresample/gstaudioresample.c: + * gst/audioresample/resample.c: + * gst/encoding/gststreamsplitter.c: + * gst/ffmpegcolorspace/avcodec.h: + * gst/ffmpegcolorspace/gstffmpegcodecmap.c: + * gst/ffmpegcolorspace/imgconvert.c: + * gst/ffmpegcolorspace/imgconvert_template.h: + * gst/ffmpegcolorspace/mem.c: + * gst/playback/README: + * gst/playback/gstdecodebin.c: + * gst/playback/gstdecodebin2.c: + * gst/playback/gstplaybasebin.c: + * gst/playback/gstplaybasebin.h: + * gst/playback/gstplaybin.c: + * gst/playback/gstplaybin2.c: + * gst/playback/gstplaysink.c: + * gst/playback/gsturidecodebin.c: + * gst/tcp/gstmultifdsink.c: + * gst/tcp/gsttcp.c: + * gst/typefind/gsttypefindfunctions.c: + * gst/videotestsrc/gstvideotestsrc.c: + * m4/freetype2.m4: + * sys/v4l/v4lmjpegsrc_calls.c: + * sys/v4l/videodev_mjpeg.h: + * sys/ximage/ximagesink.c: + * sys/xvimage/xvimagesink.c: + * sys/xvimage/xvimagesink.h: + * tests/check/elements/adder.c: + * tests/check/elements/audioresample.c: + * tests/check/elements/gnomevfssink.c: + * tests/check/elements/textoverlay.c: + * tests/examples/encoding/encoding.c: + various: typo fixes + Fix typos in code and docs. Fixes. #658984 + +2011-12-02 00:07:39 +0000 Tim-Philipp Müller + + Merge remote-tracking branch 'origin/master' into 0.11 + Conflicts: + ext/alsa/gstalsasrc.c + ext/alsa/gstalsasrc.h + gst/adder/gstadder.c + gst/playback/gstplaybin2.c + gst/playback/gstplaysinkconvertbin.c + win32/common/libgstvideo.def + +2011-12-01 23:26:36 +0000 Tim-Philipp Müller + + * .gitignore: + Add {audio,video}-marshal.[ch] to .gitignore + +2011-12-01 18:51:51 +0100 Wim Taymans + + * gst-libs/gst/tag/gstid3tag.c: + * gst-libs/gst/tag/gstvorbistag.c: + * gst-libs/gst/tag/tag.h: + * gst-libs/gst/tag/tags.c: + tags: make the tag functions return GstSample + gst_tag_image_data_to_image_buffer() -> + gst_tag_image_data_to_image_sample() And make it return a GstSample. + Store the image-type into the extra sample info. + Remove a deprecated tag + +2011-12-01 16:48:49 +0100 Wim Taymans + + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/app/gstappsink.c: + * gst-libs/gst/app/gstappsink.h: + * gst-libs/gst/audio/gstaudiobasesink.c: + * gst-libs/gst/video/convertframe.c: + * gst-libs/gst/video/video.h: + * gst/playback/gstplaybin2.c: + * gst/playback/gstplaysink.c: + * gst/playback/gstplaysink.h: + * tests/check/libs/video.c: + * tests/examples/app/appsink-src.c: + * tests/examples/app/appsrc_ex.c: + * tests/examples/seek/seek.c: + Use the new GstSample for snapshots + Make appsink return a GstSample. Remove the pull_buffer_list method because it + is not very useful anymore. + Pass GstSample to the conversion function. + Update playbin2 and examples + +2011-12-01 15:54:49 +0100 Wim Taymans + + * gst-libs/gst/app/gstapp-marshal.list: + update marshal list + +2011-12-01 15:47:16 +0100 Wim Taymans + + * gst/videoconvert/gstvideoconvert.c: + videoconvert: fix the transform_size function + The output size of a buffer does not depend on the input size but simply on the + caps of the output buffers. Don't let the base implementation deal with + unit_sizes, because input buffers might not be a multiple of that when they have + padding or non-default strides. instead, implement a transform size function + that simply calculate the natural size of an output buffer based on the caps. + +2011-12-01 15:45:28 +0100 Wim Taymans + + * gst-libs/gst/video/gstvideometa.c: + videometa: add copy functions + Without copy functions, the metadata is lost when we make a buffer copy such as + when we make a buffer writable. + +2011-12-01 15:38:10 +0100 Wim Taymans + + * gst-libs/gst/app/gstappsrc.c: + appsrc: fix negotiation + Remove old useless caps code. + Make a negotiate function and use the configured caps as the caps on the appsrc + pad. If nothing was configured, fall back to the parent implementation. + +2011-12-01 11:59:17 +0100 Stefan Sauer + + * gst/adder/gstadder.c: + adder: be more graceful in the clipfunction + Doing dynamic pipelines is hard in 0.10. As we don't have the sticky events in + 0.10 and sending such events in special elements like adder and tee was outvoted + on last attempt, be graceful to the misbehaviour instead. + +2011-12-01 01:22:19 +0000 Tim-Philipp Müller + + * tests/check/elements/audioresample.c: + tests: fix caps leak in audioresample tests + +2011-12-01 01:07:26 +0000 Tim-Philipp Müller + + * tests/check/pipelines/basetime.c: + tests: fix memory leak in basetime test + +2011-11-30 23:58:19 +0000 Tim-Philipp Müller + + * gst/playback/gstplaybin2.c: + playbin2: tone down debug message about file URIs with spaces + Complain a bit less loudly about URIs that have not been + escaped properly. + +2011-11-30 23:15:35 +0000 Tim-Philipp Müller + + * ext/alsa/gstalsasrc.c: + * ext/alsa/gstalsasrc.h: + Revert "alsasrc: Improve timestamp accuracy" + This reverts commit 0b774e0b7cf7a8ef1780fb6100228ca6e8ca8bcf. + +2011-11-30 23:15:22 +0000 Tim-Philipp Müller + + * ext/alsa/gstalsasrc.c: + Revert "alsasrc: Fix some compilation errors" + This reverts commit 2b84f5bd74ddb50f7832917ea8b4dd38d005631b. + +2011-11-30 23:15:12 +0000 Tim-Philipp Müller + + * ext/alsa/gstalsasrc.c: + Revert "alsa: Remove unused but set variable" + This reverts commit e9aed7f31c7e9e415f733e147140ce3ef2f57a61. + +2011-11-30 23:15:03 +0000 Tim-Philipp Müller + + * ext/alsa/gstalsasrc.c: + * ext/alsa/gstalsasrc.h: + Revert "alsasrc: fail gracefully when ALSA does not give timestamps" + This reverts commit c7282a5718c7f31f84fb31b2c38fab0f9a38e2b0. + +2011-11-30 23:14:54 +0000 Tim-Philipp Müller + + * ext/alsa/gstalsasrc.c: + Revert "alsasrc: handle the case where the drivers don't supply timestamps" + This reverts commit 8154b69112cdc4830cd6002ec6c1f2917d30437b. + +2011-11-28 10:55:39 +0100 Stefan Sauer + + * ext/alsa/gstalsasrc.c: + Revert "alsasrc: style fix" + This reverts commit f70ca6d4cbfd2b672dcc7215814bf6b39ce2c3f8. + +2011-11-30 14:25:11 +0100 Sebastian Dröge + + * gst/playback/gstplaysinkconvertbin.c: + playsinkconvertbin: Don't send undefined NEWSEGMENT events to the internal elements + This happens when the internal elements are added before any NEWSEGMENT + event arrived and in that case we shouldn't send a NEWSEGMENT event + to the internal elements at all. They will get the NEWSEGMENT event + from upstream later. + +2011-11-30 11:34:23 +0100 Edward Hervey + + * tests/check/Makefile.am: + * tests/check/elements/alsa.c: + * tests/check/elements/playbin-compressed.c: + * tests/check/libs/gstlibscpp.cc: + * tests/check/libs/libsabi.c: + * tests/check/libs/mixer.c: + tests: More fixes for moved interfaces + +2011-11-30 11:34:04 +0100 Edward Hervey + + * win32/common/libgstaudio.def: + * win32/common/libgstinterfaces.def: + * win32/common/libgstvideo.def: + win32: update for API changes + +2011-11-30 11:33:41 +0100 Edward Hervey + + * gst-libs/gst/audio/Makefile.am: + audio: Add audio-marshal.list to dist-ed files + +2011-11-30 07:57:02 +0100 Wim Taymans + + * docs/libs/gst-plugins-base-libs-sections.txt: + * docs/libs/gst-plugins-base-libs.types: + * ext/alsa/gstalsamixer.h: + * ext/alsa/gstalsamixeroptions.h: + * ext/alsa/gstalsamixertrack.h: + * gst-libs/gst/audio/Makefile.am: + * gst-libs/gst/audio/audio-marshal.list: + * gst-libs/gst/audio/mixer.c: + * gst-libs/gst/audio/mixer.h: + * gst-libs/gst/audio/mixeroptions.c: + * gst-libs/gst/audio/mixeroptions.h: + * gst-libs/gst/audio/mixertrack.c: + * gst-libs/gst/audio/mixertrack.h: + * gst-libs/gst/audio/mixerutils.h: + * gst-libs/gst/audio/streamvolume.c: + * gst-libs/gst/audio/streamvolume.h: + * gst-libs/gst/interfaces/Makefile.am: + * gst-libs/gst/interfaces/interfaces-marshal.list: + * 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: + * gst-libs/gst/interfaces/streamvolume.c: + * gst-libs/gst/interfaces/streamvolume.h: + * gst/playback/Makefile.am: + * gst/playback/gstplaybin2.c: + * gst/volume/gstvolume.c: + * gst/volume/gstvolume.h: + audio: move audio interfaces + Move the audio related interfaces to the audio library. + +2011-11-30 07:23:47 +0100 Wim Taymans + + * tests/examples/overlay/gtk-videooverlay.c: + * tests/examples/seek/jsseek.c: + * tests/examples/seek/seek.c: + * tests/icles/test-videooverlay.c: + fix includes for moved interfaces + +2011-11-30 07:23:07 +0100 Wim Taymans + + * gst-libs/gst/pbutils/encoding-profile.c: + encoding-profile: small cleanup in docs + +2011-11-29 19:49:50 +0100 Edward Hervey + + * gst-libs/gst/video/Makefile.am: + video: Don't forget to install moved header files + +2011-11-29 19:31:55 +0100 Edward Hervey + + * tests/examples/seek/Makefile.am: + * tests/icles/Makefile.am: + * tests/icles/test-colorkey.c: + tests: More fixes for moved interfaces + +2011-11-29 19:10:01 +0100 Wim Taymans + + * docs/libs/gst-plugins-base-libs-sections.txt: + * docs/libs/gst-plugins-base-libs.types: + * gst-libs/gst/interfaces/Makefile.am: + * 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/videoorientation.c: + * gst-libs/gst/interfaces/videoorientation.h: + * gst-libs/gst/interfaces/videooverlay.c: + * gst-libs/gst/interfaces/videooverlay.h: + * gst-libs/gst/video/Makefile.am: + * gst-libs/gst/video/colorbalance.c: + * gst-libs/gst/video/colorbalance.h: + * gst-libs/gst/video/colorbalancechannel.c: + * gst-libs/gst/video/colorbalancechannel.h: + * gst-libs/gst/video/video-marshal.list: + * gst-libs/gst/video/videoorientation.c: + * gst-libs/gst/video/videoorientation.h: + * gst-libs/gst/video/videooverlay.c: + * gst-libs/gst/video/videooverlay.h: + * sys/ximage/ximagesink.c: + * sys/xvimage/xvimagesink.c: + * tests/check/libs/gstlibscpp.cc: + * tests/check/libs/libsabi.c: + * tests/examples/overlay/Makefile.am: + * tests/examples/overlay/qt-videooverlay.cpp: + * tests/examples/overlay/qtgv-videooverlay.cpp: + * tests/icles/Makefile.am: + * tests/icles/stress-videooverlay.c: + video: move some interfaces + Move some interfaces to the video library + +2011-11-29 14:47:37 +0100 Stefan Sauer + + * gst/adder/gstadder.c: + adder: fill the audio-info that we use and not some random other one + +2011-11-29 14:22:19 +0100 Stefan Sauer + + * gst/adder/gstadder.c: + adder: unbreak adder + There was one line too much removed when porting. + +2011-11-29 14:15:45 +0100 Sebastian Dröge + + * gst/playback/gstplaybin2.c: + playbin2: Fix decoder-sink compatibility check for raw audio/video formats + If the sink supports raw audio/video, we first check + if the decoder could output any raw audio/video format + and assume it is compatible with the sink then. We don't + do a complete compatibility check here if converters + are plugged between the decoder and the sink because + the converters will convert between raw formats and + even if the decoder format is not supported by the decoder + a converter will convert it. + We assume here that the converters can convert between + any raw format. + Fixes bug #665120. + +2011-11-29 10:40:40 +0100 Stefan Sauer + + * gst/adder/gstadder.c: + * gst/adder/gstadder.h: + adder: fix deadly setcaps recursion + Use a flag to avoid calling setcaps until our stack is exhausted. I don't see how this would be useful. + +2011-11-29 09:11:21 +0100 Alessandro Decina + + * ext/ogg/gstoggdemux.c: + oggdemux: fix compiler warning + +2011-11-29 08:49:53 +0100 Alessandro Decina + + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/video/video.c: + * gst-libs/gst/video/video.h: + * win32/common/libgstvideo.def: + libgstvideo: minor fixes to key unit events + Make out args to gst_video_event_parse_{downstream|upstream}_force_key_unit + optional, update libgstvideo.def and fix docs a bit. + API: gst_video_event_new_upstream_force_key_unit + API: gst_video_event_new_downstream_force_key_unit + API: gst_video_event_is_force_key_unit + API: gst_video_event_parse_upstream_force_key_unit + API: gst_video_event_parse_downstream_force_key_unit + https://bugzilla.gnome.org/show_bug.cgi?id=607742 + +2011-06-05 01:49:38 +0200 Andoni Morales Alastruey + + * gst-libs/gst/video/video.c: + * gst-libs/gst/video/video.h: + libgstvideo: Add force key unit events + +2011-11-28 21:25:11 +0000 Tim-Philipp Müller + + Merge remote-tracking branch 'origin/master' into 0.11 + Conflicts: + gst-libs/gst/fft/gstffts16.h + +2011-11-28 21:20:38 +0000 Tim-Philipp Müller + + Merge commit 'c5544630250ec434e4dafaf17274e83865415120' into 0.11 + +2011-11-28 21:20:10 +0000 Tim-Philipp Müller + + Merge commit '4a58223e4c824fedc024af435337a769e8ce593e' into 0.11 + +2011-11-28 20:11:09 +0100 Philippe Normand + + * gst-libs/gst/fft/gstfft.h: + * gst-libs/gst/fft/gstfftf32.h: + * gst-libs/gst/fft/gstfftf64.h: + * gst-libs/gst/fft/gstffts16.h: + * gst-libs/gst/fft/gstffts32.h: + fft: Bracket public headers + This is especially needed if the gstfftw library is used from C++ + code. + Fixes #665074 + +2011-11-28 20:10:18 +0100 Philippe Normand + + * gst/typefind/gsttypefindfunctions.c: + typefindfunctions: Fix compiler warning + +2011-11-28 19:03:50 +0100 Alexey Fisher + + * gst/typefind/gsttypefindfunctions.c: + typefind: fix build error + fix build errors: + gsttypefindfunctions.c:248:25: error: 'low' may be used uninitialized in this function [-Werror=uninitialized] + gsttypefindfunctions.c:239:24: error: 'high' may be used uninitialized in this function [-Werror=uninitialized] + Signed-off-by: Alexey Fisher + +2011-11-28 19:06:57 +0100 Sebastian Dröge + + * gst/playback/gstplaysinkconvertbin.c: + playsinkconvertbin: Fix stupid mistake in last commit + +2011-11-28 19:03:54 +0100 Sebastian Dröge + + * gst/playback/gstplaysinkconvertbin.c: + playsinkconvertbin: Only return the converter caps if we actually have raw caps + Fixes bug #664818 (hopefully). + +2011-11-28 18:24:03 +0100 Wim Taymans + + * gst-libs/gst/audio/gstaudiocdsrc.c: + Update for indexable change + +2011-11-28 17:59:32 +0100 Kipp Cannon + + * gst/audioresample/gstaudioresample.c: + audioresample: Don't emit DISCONT buffers if no discontinuity happened + audioresample is derived from GstBaseTransform, and one of + GstBaseTransform's traits is that if the derived element does not + produce an output buffer from some input buffer then the first output + buffer after that gets flaged as a discontinuity, whether or not the + buffer actually is discontinuous from the output buffer that preceded + it. When downsampling, the audioresample element requires more than + one input sample for each output sample, and if the ratio of input to + output sample rates is high enough and the input buffers short enough + it can come to pass that the resampler does not receive enough samples + on its input to produce any output. Currently the resampler returns + GST_BASE_TRANSFORM_FLOW_DROPPED from the transform() method in this case, + causing the next buffer to be flagged as a discontinuity. If subsequent + elements in the pipeline reset themselves on disconts, this can cause + clicks and other undesireable behaviour. + Fixes bug #665004. + +2011-11-28 17:51:41 +0100 Wim Taymans + + * gst-libs/gst/audio/gstaudiobasesink.c: + * gst-libs/gst/audio/gstaudiobasesink.h: + * gst-libs/gst/audio/gstaudiobasesrc.c: + audio: update for clock provider API change + +2011-09-30 20:00:50 +0100 Vincent Penquerc'h + + * gst/typefind/Makefile.am: + * gst/typefind/gsttypefindfunctions.c: + typefind: typefind UTF-16 and UTF-32 + This avoids the MP3 typefinder from getting the highest score + every time it thinks there's something it might possibly be + able to parse. + https://bugzilla.gnome.org/show_bug.cgi?id=607619 + +2011-11-28 16:55:32 +0100 Wim Taymans + + * gst/playback/gstplaysink.c: + * gst/playback/gsturidecodebin.c: + fix for element flag cleanups + +2011-11-28 13:27:29 +0000 Vincent Penquerc'h + + * ext/theora/gsttheoradec.c: + * ext/theora/gsttheoradec.h: + Revert "theoradec: move the QoS logic to libgstvideo" + This reverts commit 149a4ce390a78e21309b210f7daba9db5d42afe6. + *grumble* I managed to merge something I did not mean to. + +2011-11-28 13:26:53 +0000 Vincent Penquerc'h + + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/video/video.c: + * gst-libs/gst/video/video.h: + * win32/common/libgstvideo.def: + Revert "libgstvideo: add a new API to handle QoS events and dropping logic" + This reverts commit eb03323fb683e06ed8e7f557037f13252f150c25. + *grumble* I managed to merge something I did not mean to. + +2011-11-28 12:51:22 +0000 Vincent Penquerc'h + + * ext/alsa/gstalsasink.c: + * ext/alsa/gstalsasrc.c: + * ext/gio/gstgiobasesink.c: + * ext/gio/gstgiobasesrc.c: + * ext/gnomevfs/gstgnomevfssink.c: + * ext/gnomevfs/gstgnomevfssrc.c: + * ext/libvisual/visual.c: + * ext/ogg/gstoggaviparse.c: + * ext/ogg/gstoggdemux.c: + * ext/ogg/gstoggmux.c: + * ext/ogg/gstoggparse.c: + * ext/ogg/gstogmparse.c: + * ext/pango/gsttextoverlay.c: + * ext/pango/gsttextrender.c: + * ext/theora/gsttheoradec.c: + * ext/theora/gsttheoraenc.c: + * ext/theora/gsttheoraparse.c: + * ext/vorbis/gstvorbisdec.c: + * ext/vorbis/gstvorbisenc.c: + * ext/vorbis/gstvorbisparse.c: + * gst-libs/gst/app/gstappsink.c: + * gst-libs/gst/app/gstappsrc.c: + * gst-libs/gst/cdda/gstcddabasesrc.c: + * gst-libs/gst/tag/gsttagdemux.c: + * gst/adder/gstadder.c: + * gst/audioconvert/gstaudioconvert.c: + * gst/audiorate/gstaudiorate.c: + * gst/audioresample/gstaudioresample.c: + * gst/audiotestsrc/gstaudiotestsrc.c: + * gst/encoding/gstencodebin.c: + * gst/encoding/gstsmartencoder.c: + * gst/encoding/gststreamcombiner.c: + * gst/encoding/gststreamsplitter.c: + * gst/ffmpegcolorspace/gstffmpegcolorspace.c: + * gst/gdp/gstgdpdepay.c: + * gst/gdp/gstgdppay.c: + * gst/playback/gstdecodebin.c: + * gst/playback/gstdecodebin2.c: + * gst/playback/gstplaysink.c: + * gst/playback/gststreamselector.c: + * gst/playback/gststreamsynchronizer.c: + * gst/playback/gstsubtitleoverlay.c: + * gst/playback/gsturidecodebin.c: + * gst/subparse/gstssaparse.c: + * gst/subparse/gstsubparse.c: + * gst/tcp/gstmultifdsink.c: + * gst/tcp/gsttcpclientsink.c: + * gst/tcp/gsttcpclientsrc.c: + * gst/tcp/gsttcpserversrc.c: + * gst/videorate/gstvideorate.c: + * gst/videoscale/gstvideoscale.c: + * gst/videotestsrc/gstvideotestsrc.c: + * sys/v4l/gstv4lmjpegsink.c: + * sys/v4l/gstv4lmjpegsrc.c: + * sys/v4l/gstv4lsrc.c: + * sys/ximage/ximagesink.c: + * sys/xvimage/xvimagesink.c: + * tests/check/elements/audiorate.c: + * tests/check/elements/decodebin.c: + * tests/check/elements/decodebin2.c: + * tests/check/elements/playbin.c: + * tests/check/elements/playbin2-compressed.c: + * tests/check/elements/playbin2.c: + * tests/check/elements/videoscale.c: + various: fix pad template leaks + https://bugzilla.gnome.org/show_bug.cgi?id=662664 + +2011-09-07 16:04:14 +0100 Vincent Penquerc'h + + * ext/theora/gsttheoradec.c: + * ext/theora/gsttheoradec.h: + theoradec: move the QoS logic to libgstvideo + https://bugzilla.gnome.org/show_bug.cgi?id=658241 + +2011-09-05 13:56:05 +0100 Vincent Penquerc'h + + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/video/video.c: + * gst-libs/gst/video/video.h: + * win32/common/libgstvideo.def: + libgstvideo: add a new API to handle QoS events and dropping logic + https://bugzilla.gnome.org/show_bug.cgi?id=658241 + +2011-11-28 11:30:18 +0100 Mark Nauwelaerts + + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/audio/gstaudioencoder.h: + audioencoder: elaborate some documentation + +2011-11-28 11:28:06 +0100 Mark Nauwelaerts + + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudiodecoder.h: + audiodecoder: add some documentation + +2011-11-21 14:26:54 +0100 Mark Nauwelaerts + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: really discard NULL decoded frame altogether + ... including any timestamp, rather than having that one influence base_ts. + +2011-11-28 10:55:39 +0100 Stefan Sauer + + * ext/alsa/gstalsasrc.c: + alsasrc: style fix + Use timestamp==0 instead of mixing it with !timestamp style checks. + +2011-11-28 09:12:37 +0100 Stefan Sauer + + * ext/alsa/gstalsasrc.c: + alsasrc: handle the case where the drivers don't supply timestamps + If highres-timestamp is 0, try lowres and if that fails fallback to system clock + timestamps. + +2011-11-27 20:14:08 +0100 Matej Knopp + + * gst/playback/gsturidecodebin.c: + uridecodebin: fix debug message printf format compiler warning + https://bugzilla.gnome.org/show_bug.cgi?id=662607 + +2011-11-26 12:12:59 +0000 Tim-Philipp Müller + + Merge remote-tracking branch 'origin/master' into 0.11 + Conflicts: + ext/vorbis/gstvorbisenc.c + gst/playback/gstdecodebin2.c + gst/playback/gstplaysinkconvertbin.c + gst/videorate/gstvideorate.c + +2011-11-01 15:21:54 +0000 Vincent Penquerc'h + + * ext/ogg/gstoggmux.c: + oggmux: set collectpads2 not to wait on sparse streams + https://bugzilla.gnome.org/show_bug.cgi?id=663174 + +2011-11-25 15:35:39 +0100 Josep Torra + + * gst/playback/gstplaysinkconvertbin.c: + playsinkconvertbin: make identiy silent + +2011-11-25 13:11:54 +0000 Tim-Philipp Müller + + * ext/vorbis/Makefile.am: + * gst-libs/gst/audio/Makefile.am: + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudiodecoder.h: + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/audio/gstaudioencoder.h: + audio: remove unstable API guards from the audio decoder and encoder base classes + +2011-11-25 12:58:22 +0000 Tim-Philipp Müller + + * gst/playback/gstplaybin2.c: + docs: mention explicitly that playbin2 signals are emitted from a streaming thread + +2011-11-25 11:11:12 +0100 Sebastian Dröge + + * gst/playback/gstdecodebin2.c: + decodebin2: Set the multiqueue limits to the playing limits after overrun too + We don't expect any new pads anymore and prerolling is finished now. + +2011-11-25 11:08:58 +0100 Sebastian Dröge + + * gst/playback/gstdecodebin2.c: + decodebin2: Cache the upstream seekability for demuxer decode chains and use it for the non-preroll multiqueue limits + After preroll the multiqueue limits are still set to the preroll + limits if use-buffering is set to TRUE. In that case we only want + time limits on the multiqueue if upstream is seekable. + +2011-11-08 13:55:58 +0000 Vincent Penquerc'h + + * gst/playback/gstdecodebin2.c: + decodebin2: fix prerolling for low bitrate streams from hlsdemux + Such streams were detected as seekable, as the query on the typefind + element was testing the m3u8 file listing the actual streams, and + not going through the demuxer(s). + We now check for seekability for each multiqueue following a demuxer, + so the query will flow through the elements which might prevent seeking. + https://bugzilla.gnome.org/show_bug.cgi?id=647769 + +2011-11-25 10:31:38 +0100 Edward Hervey + + * gst-libs/gst/app/Makefile.am: + * gst-libs/gst/fft/Makefile.am: + * gst-libs/gst/interfaces/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: + gst-libs: Add --warn-all to introspection scanner + And let's get fixing those docs :) + +2011-11-24 21:39:14 +0100 René Stadler + + * tests/check/elements/audioconvert.c: + * tests/check/elements/audiotestsrc.c: + * tests/check/elements/vorbisdec.c: + * tests/check/elements/vorbistag.c: + tests: update for gstcheck API change + +2011-10-24 11:46:05 +0100 Vincent Penquerc'h + + * ext/ogg/gstoggdemux.c: + oggdemux: minor cleanup + +2011-09-27 16:45:26 +0100 Vincent Penquerc'h + + * gst-libs/gst/riff/riff-ids.h: + libgstriff: add a couple tags that need skipping + Found in a sample in the wild, appears to be ID3 tag. + https://bugzilla.gnome.org/show_bug.cgi?id=660249 + +2011-11-24 14:41:13 +0100 Sebastian Dröge + + * gst/videorate/gstvideorate.c: + videorate: Rename ARG_ enums to PROP_ + This is more consistent with other code and these are + properties anyway, not arguments + +2011-11-24 14:29:49 +0100 Sebastian Dröge + + * gst/videorate/gstvideorate.c: + * gst/videorate/gstvideorate.h: + videorate: Add property to force an output framerate + API: GstVideoRate:force-fps + Changing the framerate during playback is not possible + with a capsfilter downstream if upstream is not using + gst_pad_alloc_buffer(). In that case there's no way in + 0.10 to signal to videorate that the preferred framerate + has changed. + This new property will force the output framerate to + a specific value and can be changed during playback. + +2011-11-24 12:38:54 +0100 Sebastian Dröge + + * gst/playback/gstplaysinkconvertbin.c: + playsinkconvertbin: Reconfigure if we switch from raw to incompatible raw caps + We might need to add converters and worked in passthrough mode before. + +2011-11-24 12:37:58 +0100 Sebastian Dröge + + * gst/playback/gstplaysinkconvertbin.c: + playsinkconvertbin: Override acceptcaps function for the two ghostpads + The ghostpad acceptcaps functions are not valid in this case because + we don't only accept the caps accepted by the target but could also + insert converters. Fixes bug #663892. + +2011-11-24 11:34:12 +0100 Sebastian Dröge + + * gst/playback/gstplaysinkaudioconvert.c: + playsinkaudioconvert: use-volume and use-converters are no construct-only properties anymore + Fixes bug #663893. + +2011-11-24 11:09:20 +0100 Vincent Penquerc'h + + * gst/videoconvert/videoconvert.c: + videoconvert: fix width/height mismatches + https://bugzilla.gnome.org/show_bug.cgi?id=663238 + +2011-11-24 11:04:10 +0100 Mark Nauwelaerts + + * gst/videoconvert/videoconvert.c: + videoconvert: fix odd width and height handling in some fastpath cases + +2011-10-22 20:29:26 +0100 Vincent Penquerc'h + + * ext/ogg/gstoggdemux.c: + oggdemux: skip the second bisection when possible + If we already saw the keyframes that we need to find, + we do not need to bisect to find them. + This will always be the case for streams with audio only, + where each frame acts as a keyframe, but will occasionally + also happen for streams with video. + https://bugzilla.gnome.org/show_bug.cgi?id=662475 + +2011-10-22 20:20:38 +0100 Vincent Penquerc'h + + * ext/ogg/gstoggdemux.c: + * ext/ogg/gstoggdemux.h: + oggdemux: improve push time seeking + Various tweaks to improve convergence, in particular for + the worst case, which is now cut in about half. + https://bugzilla.gnome.org/show_bug.cgi?id=662475 + +2011-10-21 19:38:19 +0100 Vincent Penquerc'h + + * ext/ogg/gstoggdemux.c: + * ext/ogg/gstoggdemux.h: + oggdemux: gather some more stats about bisection + https://bugzilla.gnome.org/show_bug.cgi?id=662475 + +2011-11-24 01:30:50 +0000 Tim-Philipp Müller + + * gst/playback/gsturidecodebin.c: + uridecodebin: double-check property type before blindly setting/proxying values + +2011-11-24 01:18:38 +0000 Tim-Philipp Müller + + * gst/playback/gstplaybin2.c: + * gst/playback/gsturidecodebin.c: + playbin2, uridecodebin: make connection-speed property a guint64 + +2011-11-23 23:16:51 +0000 Tim-Philipp Müller + + * docs/libs/gst-plugins-base-libs-docs.sgml: + docs: update sgml for renames + +2011-11-23 16:09:13 +0000 Vincent Penquerc'h + + * ext/vorbis/gstvorbisenc.c: + vorbisenc: do not accept 256 channels, 255 is the max vorbis supports + +2011-11-23 11:10:31 +0100 Wim Taymans + + * ext/ogg/gstoggstream.c: + ogg: fix compilation + +2011-11-23 10:50:53 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + ext/ogg/gstoggmux.c + +2011-11-22 13:29:10 +0000 Vincent Penquerc'h + + * ext/ogg/gstoggstream.c: + oggstream: extract opus comments if available + +2011-11-22 13:15:33 +0000 Vincent Penquerc'h + + * ext/ogg/gstoggstream.c: + oggstream: recognize opus headers from data, not packet count + Opus streams outside of Ogg may not have headers, and oggstream + may be used by oggmux to mux an Opus stream which does not come + from Ogg - thus without headers. + Determining headerness by packet count would strip the first two + packets from such an Opus stream, leading to a very small amount + of audio being clipped at the beginning of the stream. + +2011-11-22 13:01:35 +0000 Vincent Penquerc'h + + * ext/ogg/gstoggdemux.c: + oggdemux: add some more debug info when determining start time + +2011-11-22 12:55:56 +0000 Vincent Penquerc'h + + * ext/ogg/gstoggstream.c: + oggstream: fix opus duration calculation + +2011-11-22 12:00:58 +0000 Vincent Penquerc'h + + * ext/ogg/gstoggstream.c: + oggstream: early out on headers when determining packet duration + +2011-11-21 17:03:21 +0000 Vincent Penquerc'h + + * ext/ogg/gstoggstream.c: + * ext/ogg/gstoggstream.h: + oggstream: account for opus pre-skip in granpos/time mapping + +2011-11-22 10:04:12 +0100 René Stadler + + * gst/playback/gstplaysinkconvertbin.c: + playsinkconvertbin: avoid removing children from bin twice + GstBin base class removes children in dispose, so we need to do the same. + +2011-11-22 01:21:04 +0000 Tim-Philipp Müller + + * ext/libvisual/visual.c: + * ext/vorbis/gstvorbisdec.c: + * ext/vorbis/gstvorbisenc.c: + Fix some more printf format warnings + +2011-11-21 19:28:01 +0100 Matej Knopp + + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudioencoder.c: + Fix printf format compiler warnings for OSX / 64bit + https://bugzilla.gnome.org/show_bug.cgi?id=662607 + +2011-11-21 13:35:34 +0100 Wim Taymans + + * ext/ogg/gstoggdemux.c: + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/tag/gsttagdemux.c: + update for activation changes + +2011-11-21 13:04:42 +0100 Edward Hervey + + * sys/ximage/ximagepool.c: + ximagebufferpool: Use the default ::free_buffer() implementation + Which does exactly the same thing + +2011-11-21 13:04:12 +0100 Edward Hervey + + * sys/xvimage/xvimagepool.c: + xvimagebufferpool: Use the default ::free_buffer() implementation + Which does exactly the same thing + +2011-11-19 16:06:09 +0000 Vincent Penquerc'h + + * ext/ogg/gstoggmux.c: + * ext/ogg/gstoggstream.c: + ogg: add opus support + +2011-11-18 17:58:58 +0100 Wim Taymans + + * ext/gio/gstgiosrc.c: + * ext/ogg/gstoggdemux.c: + * gst-libs/gst/app/gstappsrc.c: + * gst-libs/gst/audio/gstaudiobasesrc.c: + * gst-libs/gst/tag/gsttagdemux.c: + * gst/audiotestsrc/gstaudiotestsrc.c: + update for new scheduling query + +2011-11-18 13:56:04 +0100 Wim Taymans + + * ext/ogg/gstoggdemux.c: + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/tag/gsttagdemux.c: + add parent to activate functions + +2011-11-18 12:37:10 +0100 Wim Taymans + + * gst-libs/gst/audio/gstaudiobasesink.c: + fix for scheduling mode rename + +2011-11-17 17:07:41 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + gst-libs/gst/audio/gstaudiodecoder.c + +2011-11-17 16:15:46 +0100 Wim Taymans + + * gst-libs/gst/tag/gsttagdemux.c: + tag: update for new typefind + +2011-11-17 12:48:25 +0100 Wim Taymans + + * ext/libvisual/visual.c: + * ext/ogg/gstoggaviparse.c: + * ext/ogg/gstoggdemux.c: + * ext/ogg/gstoggmux.c: + * ext/ogg/gstoggparse.c: + * ext/ogg/gstogmparse.c: + * ext/pango/gstbasetextoverlay.c: + * ext/pango/gsttextrender.c: + * ext/theora/gsttheoradec.c: + * ext/theora/gsttheoraenc.c: + * ext/theora/gsttheoraparse.c: + * ext/vorbis/gstvorbisparse.c: + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/rtp/gstrtpbasedepayload.c: + * gst-libs/gst/rtp/gstrtpbasepayload.c: + * gst-libs/gst/tag/gsttagdemux.c: + * gst-libs/gst/tag/gsttagmux.c: + * gst/adder/gstadder.c: + * gst/audiorate/gstaudiorate.c: + * gst/encoding/gstsmartencoder.c: + * gst/encoding/gststreamcombiner.c: + * gst/encoding/gststreamsplitter.c: + * gst/gdp/gstgdpdepay.c: + * gst/gdp/gstgdppay.c: + * gst/playback/gstplaysinkconvertbin.c: + * gst/playback/gststreamsynchronizer.c: + * gst/playback/gstsubtitleoverlay.c: + * gst/subparse/gstssaparse.c: + * gst/subparse/gstsubparse.c: + add parent to pad functions + +2011-11-17 08:24:27 +0100 Stefan Sauer + + * gst/adder/gstadder.c: + collectpads: port API changes + +2011-11-16 19:00:44 +0100 Mark Nauwelaerts + + * ext/vorbis/gstvorbisenc.c: + vorbisenc: reset tag setter interface when appropriate + +2011-11-16 19:00:30 +0100 Mark Nauwelaerts + + * gst-libs/gst/audio/gstaudioencoder.c: + audioencoder: invalidate format info when setup negotiation failed + ... which ensures nothing subsequently tries to slip past _chain + and into a possibly improperly setup subclass. + +2011-11-15 13:29:31 +0000 Vincent Penquerc'h + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: accept dropped buffers before we know the format + This allows flacdec to not emit audio for headers, while allowing + the base audio decoder to keep its timestamps in sync. + +2011-11-16 17:50:03 +0100 Wim Taymans + + * gst/playback/gststreamsynchronizer.c: + add parent to internal links + +2011-11-16 17:25:17 +0100 Wim Taymans + + * ext/libvisual/visual.c: + * ext/ogg/gstoggdemux.c: + * ext/ogg/gstogmparse.c: + * ext/pango/gstbasetextoverlay.c: + * ext/theora/gsttheoradec.c: + * ext/theora/gsttheoraenc.c: + * ext/theora/gsttheoraparse.c: + * ext/vorbis/gstvorbisparse.c: + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/rtp/gstrtpbasepayload.c: + * gst-libs/gst/tag/gsttagdemux.c: + * gst/adder/gstadder.c: + * gst/audioresample/gstaudioresample.c: + * gst/encoding/gstsmartencoder.c: + * gst/encoding/gststreamcombiner.c: + * gst/encoding/gststreamsplitter.c: + * gst/playback/gstplaysinkconvertbin.c: + * gst/playback/gststreamsynchronizer.c: + * gst/playback/gstsubtitleoverlay.c: + * gst/subparse/gstsubparse.c: + add parent to query function + +2011-11-16 12:37:44 +0100 Wim Taymans + + * ext/libvisual/visual.c: + visual: update for renamed flags + Use the _check_reconfigure method instead of checking flags. + Don't need to ref the parent anymore, core does that. + +2011-11-15 17:58:19 +0100 Wim Taymans + + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/tag/gsttagdemux.c: + * gst/adder/gstadder.c: + * gst/playback/gstdecodebin2.c: + * gst/playback/gstplaybin2.c: + * gst/playback/gstsubtitleoverlay.c: + _query_peer_*() -> _peer_query_*() + +2011-11-15 17:17:53 +0100 Wim Taymans + + * ext/libvisual/visual.c: + * ext/pango/gstbasetextoverlay.c: + * ext/pango/gsttextrender.c: + * gst-libs/gst/rtp/gstrtpbasepayload.c: + * gst/adder/gstadder.c: + * gst/encoding/gstsmartencoder.c: + * gst/encoding/gststreamsplitter.c: + _peer_get_caps() -> _peer_query_caps() + +2011-11-15 16:48:15 +0100 Wim Taymans + + * ext/libvisual/visual.c: + * ext/ogg/gstoggmux.c: + * ext/ogg/gstoggparse.c: + * ext/pango/gsttextrender.c: + * ext/theora/gsttheoraenc.c: + * ext/theora/gsttheoraparse.c: + * ext/vorbis/gstvorbisparse.c: + * gst-libs/gst/pbutils/gstdiscoverer.c: + * gst/encoding/gstencodebin.c: + * gst/playback/gstdecodebin2.c: + * gst/playback/gstplaybin2.c: + * gst/playback/gstplaysink.c: + * gst/playback/gstplaysinkconvertbin.c: + * gst/playback/gstsubtitleoverlay.c: + * gst/playback/gsturidecodebin.c: + * tests/check/elements/audioconvert.c: + * tests/examples/encoding/encoding.c: + * tests/icles/playback/test.c: + * tests/icles/playback/test5.c: + * tests/icles/playback/test6.c: + update for _get_caps() -> _query_caps() + +2011-11-15 16:30:38 +0100 Wim Taymans + + * ext/libvisual/visual.c: + * ext/pango/gstbasetextoverlay.c: + * ext/theora/gsttheoraenc.c: + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/rtp/gstrtpbaseaudiopayload.c: + * gst-libs/gst/rtp/gstrtpbasepayload.c: + * gst-libs/gst/rtp/gstrtpbasepayload.h: + * gst/adder/gstadder.c: + * gst/audiorate/gstaudiorate.c: + * gst/encoding/gstsmartencoder.c: + * gst/encoding/gststreamcombiner.c: + * gst/encoding/gststreamsplitter.c: + * gst/playback/gstplaysinkconvertbin.c: + * gst/playback/gststreamsynchronizer.c: + * gst/playback/gstsubtitleoverlay.c: + change getcaps to query + Add sink and src event functions in rtpbasepayload + Add query vmethod to rtpbasepayload. + +2011-11-15 13:29:31 +0000 Vincent Penquerc'h + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: accept dropped buffers before we know the format + This allows flacdec to not emit audio for headers, while allowing + the base audio decoder to keep its timestamps in sync. + +2011-11-14 12:45:31 +0100 Robert Swain + + * gst-libs/gst/audio/gstaudiodecoder.c: + audio: Remove some unused variables + +2011-08-30 18:27:09 -0400 Olivier Crête + + * gst-libs/gst/rtp/gstrtcpbuffer.h: + rtcpbuffer: Add feedback message types from RFC 5104 + These are Codec Control messages (CCM) + https://bugzilla.gnome.org/show_bug.cgi?id=658419 + +2011-10-19 16:30:27 +0200 Mark Nauwelaerts + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: improve reverse playback + ... by doing some more (reverse) timestamp interpolating and + refactoring downstream pushing. + Fixes #661983. + +2011-11-14 09:59:36 +0000 Tim-Philipp Müller + + * gst-libs/gst/tag/gsttagdemux.c: + tag: convert GstTagDemux's sometimes source pad to an always source pad + Originally decodebin couldn't deal with that in 0.10, but now simply + setting the caps when we know them should be enough. Pad activation + mode switching might need some more testing/tweaking with the new + arrangement. + +2011-11-14 10:46:56 +0100 Wim Taymans + + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/rtp/gstrtcpbuffer.c: + * gst-libs/gst/rtp/gstrtcpbuffer.h: + * gst-libs/gst/rtp/gstrtpbuffer.c: + * gst-libs/gst/rtp/gstrtppayloads.h: + * gst-libs/gst/rtsp/gstrtsptransport.h: + fix docs + +2011-11-12 15:37:37 +0200 Stefan Sauer + + * tests/icles/audio-trickplay.c: + controller: no need to explicitely add controlled properties anymore + +2011-11-13 23:44:23 +0000 Tim-Philipp Müller + + * ext/gio/gstgio.c: + * gst-libs/gst/app/gstappsink.c: + * gst-libs/gst/app/gstappsrc.c: + * gst-libs/gst/audio/gstaudiocdsrc.c: + * tests/check/elements/playbin-compressed.c: + * tests/check/elements/playbin.c: + Update for GstURIHandler get_protocols() changes + +2011-11-13 18:22:06 +0000 Tim-Philipp Müller + + * ext/gio/gstgio.c: + * ext/gio/gstgiobasesink.c: + * ext/gio/gstgiobasesrc.c: + * gst-libs/gst/app/gstappsink.c: + * gst-libs/gst/app/gstappsrc.c: + * gst-libs/gst/audio/gstaudiocdsrc.c: + * tests/check/libs/audiocdsrc.c: + gio, appsrc, appsink, cdaudiosrc: update for GstURIHandler API changes + +2011-11-13 14:39:43 +0000 Tim-Philipp Müller + + * win32/common/libgstaudio.def: + * win32/common/libgstinterfaces.def: + * win32/common/libgstrtp.def: + * win32/common/libgstrtsp.def: + win32: update for API changes + +2011-11-13 13:32:30 +0000 Tim-Philipp Müller + + Merge remote-tracking branch 'origin/master' into 0.11 + Conflicts: + gst-libs/gst/audio/Makefile.am + gst-libs/gst/audio/audio.h + tests/examples/seek/jsseek.c + tests/examples/seek/seek.c + tests/icles/test-colorkey.c + +2011-11-13 13:18:16 +0000 Tim-Philipp Müller + + * gst-libs/gst/audio/audio.h: + * gst-libs/gst/audio/gstaudiodecoder.c: + audio: add GST_AUDIO_INFO_IS_VALID macro and use in audio decoder base class + API: GST_AUDIO_INFO_IS_VALID + +2011-11-12 15:51:52 +0000 Tim-Philipp Müller + + * configure.ac: + * tests/examples/seek/jsseek.c: + * tests/examples/seek/seek.c: + * tests/icles/test-colorkey.c: + * tests/icles/test-xoverlay.c: + tests: require Gtk+ 3.0 for examples and Gtk-based test apps + The Gtk+ dependency is entirely optional, we're just not + supporting Gtk+ 2.x any longer. + +2011-11-07 17:36:44 +0000 Tim-Philipp Müller + + * gst-libs/gst/audio/Makefile.am: + audio: fix order in LIBADD + Local libs must come first. + +2011-11-12 12:00:17 +0000 Tim-Philipp Müller + + * po/af.po: + * po/az.po: + * po/bg.po: + * po/ca.po: + * po/cs.po: + * po/da.po: + * po/de.po: + * po/el.po: + * po/en_GB.po: + * po/eo.po: + * po/es.po: + * po/eu.po: + * po/fi.po: + * po/fr.po: + * po/gl.po: + * po/hu.po: + * po/id.po: + * po/it.po: + * po/ja.po: + * po/lt.po: + * po/lv.po: + * po/nb.po: + * po/nl.po: + * po/or.po: + * po/pl.po: + * po/pt_BR.po: + * po/ro.po: + * po/ru.po: + * po/sk.po: + * po/sl.po: + * po/sq.po: + * po/sr.po: + * po/sv.po: + * po/tr.po: + * po/uk.po: + * po/vi.po: + * po/zh_CN.po: + po: update after library merge + +2011-11-12 11:56:06 +0000 Tim-Philipp Müller + + * tests/check/libs/gstlibscpp.cc: + * tests/check/libs/libsabi.c: + * tests/check/libs/struct_arm.h: + * tests/check/libs/struct_i386.h: + * tests/check/libs/struct_i386_osx.h: + * tests/check/libs/struct_x86_64.h: + tests: update after type renames + +2011-11-11 11:29:56 +0000 Tim-Philipp Müller + + * po/POTFILES.in: + po: update POTFILES.in for renamed source files + +2011-11-07 17:36:44 +0000 Tim-Philipp Müller + + * gst-libs/gst/audio/Makefile.am: + audio: fix order in LIBADD + Local libs must come first. + +2011-11-07 17:25:45 +0000 Tim-Philipp Müller + + * tests/check/libs/audiocdsrc.c: + tests: fix audiocdsrc for changed preroll behaviour + Previously, the source posted a TAG message before buffers would + even be pushed towards the sink, so we'd get the TAG message before + any ASYNC_DONE message. Now the tags get sent downstream to the sink + to get posted there, and the tag event will get queued and handled + later after preroll has finished, so now we get the ASYNC_DONE + message before the TAG message. + +2011-09-24 19:55:25 +0100 Tim-Philipp Müller + + * tests/check/Makefile.am: + * tests/check/libs/.gitignore: + * tests/check/libs/audiocdsrc.c: + * tests/check/libs/cddabasesrc.c: + * tests/check/libs/gstlibscpp.cc: + * tests/check/libs/libsabi.c: + * tests/check/libs/struct_arm.h: + * tests/check/libs/struct_i386.h: + * tests/check/libs/struct_i386_osx.h: + * tests/check/libs/struct_x86_64.h: + tests: fix up cddabasesrc unit test for GstCddaBaseSrc -> GstAudioCdSrc renaming + +2011-09-24 19:35:40 +0100 Tim-Philipp Müller + + * ext/cdparanoia/Makefile.am: + * ext/cdparanoia/gstcdparanoiasrc.c: + * ext/cdparanoia/gstcdparanoiasrc.h: + cdparanoia: update for GstCddaBaseSrc -> GstAudioCdSrc renaming + +2011-09-24 19:22:11 +0100 Tim-Philipp Müller + + * Android.mk: + * configure.ac: + * docs/libs/Makefile.am: + * docs/libs/gst-plugins-base-libs-docs.sgml: + * docs/libs/gst-plugins-base-libs-sections.txt: + * docs/libs/gst-plugins-base-libs.types: + * gst-libs/gst/Makefile.am: + * gst-libs/gst/audio/Makefile.am: + * gst-libs/gst/audio/gstaudiocdsrc.c: + * gst-libs/gst/audio/gstaudiocdsrc.h: + * gst-libs/gst/cdda/Makefile.am: + * gst-libs/gst/cdda/gstcddabasesrc.c: + * gst-libs/gst/cdda/gstcddabasesrc.h: + * gst-plugins-base.spec.in: + * pkgconfig/Makefile.am: + * pkgconfig/gstreamer-cdda-uninstalled.pc.in: + * pkgconfig/gstreamer-cdda.pc.in: + * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in: + * pkgconfig/gstreamer-plugins-base.pc.in: + * po/POTFILES.in: + * win32/MANIFEST: + * win32/common/libgstcdda.def: + * win32/vs6/libgstcdda.dsp: + cdda: rename GstCddaBaseSrc to GstAudioCdSrc and move to libgstaudio + Another mini-lib down, to make space for new mini libs. + Remove bogus copyright line while at it. + +2011-11-12 09:56:04 +0000 Christian Fredrik Kalager Schaller + + * gst-plugins-base.spec.in: + update spec file for latest 0.11 changes + +2011-11-12 01:38:37 +0100 René Stadler + + * gst/audioconvert/gstaudioconvert.c: + * gst/videoconvert/gstvideoconvert.c: + audioconvert, videoconvert: fix caps leak in transform_caps + +2011-11-11 20:19:53 +0100 René Stadler + + * gst/audioconvert/audioconvert.c: + audioconvert: fix leak of channel matrix + gst_channel_mix_unset_matrix relies on the channel count to free the matrix + array, so run it before resetting it to zero with gst_audio_info_init. + +2011-11-11 19:55:41 +0100 René Stadler + + * gst/videotestsrc/videotestsrc.c: + videotestsrc: fix crash with ARGB64 + This got broken when it was ported. + +2011-11-11 19:53:11 +0100 René Stadler + + * gst-libs/gst/video/video.c: + video: init chroma-size and colorimetry members even if missing from caps + This makes a TRUE return from gst_video_info_from_caps fully consistent with + gst_video_info_init. + +2011-11-11 19:36:23 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-11-11 19:35:33 +0100 Wim Taymans + + * gst-libs/gst/rtsp/gstrtspconnection.c: + * gst-libs/gst/rtsp/gstrtspconnection.h: + * gst-libs/gst/rtsp/gstrtspextension.h: + * gst-libs/gst/rtsp/gstrtspmessage.h: + * gst-libs/gst/rtsp/gstrtsprange.h: + * gst-libs/gst/rtsp/gstrtsptransport.h: + * gst-libs/gst/rtsp/gstrtspurl.h: + rtsp: cleanup headers + Add padding, fix indentation, remove deprecated stuff + +2011-11-11 19:21:09 +0100 Wim Taymans + + * gst-libs/gst/rtp/gstrtcpbuffer.h: + * gst-libs/gst/rtp/gstrtpbaseaudiopayload.c: + * gst-libs/gst/rtp/gstrtpbasedepayload.h: + * gst-libs/gst/rtp/gstrtpbasepayload.c: + * gst-libs/gst/rtp/gstrtpbasepayload.h: + * gst-libs/gst/rtp/gstrtpbuffer.h: + * gst-libs/gst/rtp/gstrtppayloads.h: + rtp: fix headers + indent, add padding, remove old abidata + +2011-11-11 19:16:54 +0100 Wim Taymans + + * gst-libs/gst/interfaces/colorbalance.h: + * gst-libs/gst/interfaces/mixer.h: + * gst-libs/gst/interfaces/navigation.h: + * gst-libs/gst/interfaces/propertyprobe.h: + * gst-libs/gst/interfaces/streamvolume.h: + * gst-libs/gst/interfaces/tuner.h: + * gst-libs/gst/interfaces/videoorientation.h: + remove padding from interfaces + +2011-11-11 19:16:12 +0100 Wim Taymans + + * gst-libs/gst/interfaces/tunernorm.h: + fix docs + +2011-11-11 19:14:26 +0100 Wim Taymans + + * gst-libs/gst/interfaces/mixertrack.h: + mixertrack: fix docs + +2011-11-11 19:13:52 +0100 Wim Taymans + + * gst-libs/gst/audio/audio.h: + audio: fix docs + +2011-11-11 19:01:56 +0100 Wim Taymans + + * gst-libs/gst/pbutils/encoding-profile.h: + * gst-libs/gst/pbutils/encoding-target.h: + * gst-libs/gst/pbutils/pbutils-private.h: + pbutils: clean up headers + Add padding + indent + +2011-11-11 18:49:09 +0100 Wim Taymans + + * gst-libs/gst/interfaces/colorbalance.h: + * gst-libs/gst/interfaces/colorbalancechannel.h: + * gst-libs/gst/interfaces/mixer.c: + * 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/streamvolume.h: + * gst-libs/gst/interfaces/tuner.h: + * gst-libs/gst/interfaces/tunerchannel.h: + * gst-libs/gst/interfaces/tunernorm.h: + * gst-libs/gst/interfaces/videoorientation.h: + * gst-libs/gst/interfaces/videooverlay.h: + interfaces: clean up + Remove deprecated bits + Fix FIXMES + Indent + Add padding + +2011-11-11 18:23:22 +0100 Wim Taymans + + * 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: + fft: fix headers + More fft structure into .c file + indent headers + +2011-11-11 17:53:03 +0100 Wim Taymans + + * gst-libs/gst/audio/audio.c: + * gst-libs/gst/audio/audio.h: + * gst-libs/gst/audio/gstaudiobasesrc.h: + * gst-libs/gst/audio/gstaudiodecoder.h: + * gst-libs/gst/audio/gstaudioencoder.h: + * gst-libs/gst/audio/gstaudioiec61937.h: + * gst-libs/gst/audio/gstaudiosink.h: + audio: fix headers + Add const to some methods. + Add padding. + Add GType for GstAudioInfo and GstAudioFormatInfo. + Add new/copy/free for GstAudioInfo. + +2011-11-11 17:52:36 +0100 Wim Taymans + + * gst-libs/gst/app/gstappsink.h: + * gst-libs/gst/app/gstappsrc.h: + app: fix headers + +2011-11-11 13:32:23 +0000 Tim-Philipp Müller + + * gst/playback/gstplaysinkconvertbin.c: + playsinkconvertbin: fix visualisations again + Make caps writable before merging other caps into them. + +2011-11-11 13:14:21 +0100 Wim Taymans + + * docs/design/draft-media-types.txt: + * gst-libs/gst/video/video.c: + * gst-libs/gst/video/video.h: + video: add support for max-framerate + Add support for max-framerate in the video helpers and update the video + caps document. + +2011-11-11 13:12:27 +0100 Wim Taymans + + * gst/playback/gstplaysinkconvertbin.c: + make the identity silent + +2011-11-11 12:35:50 +0100 Wim Taymans + + * gst-libs/gst/video/gstmetavideoclip.h: + remove bogus file + +2011-11-11 12:32:23 +0100 Wim Taymans + + * docs/libs/gst-plugins-base-libs-sections.txt: + * docs/libs/gst-plugins-base-libs.types: + * gst-libs/gst/rtp/Makefile.am: + * gst-libs/gst/rtp/gstbasertpaudiopayload.c: + * gst-libs/gst/rtp/gstbasertpaudiopayload.h: + * gst-libs/gst/rtp/gstbasertpdepayload.c: + * gst-libs/gst/rtp/gstbasertpdepayload.h: + * gst-libs/gst/rtp/gstbasertppayload.c: + * gst-libs/gst/rtp/gstbasertppayload.h: + * gst-libs/gst/rtp/gstrtpbaseaudiopayload.c: + * gst-libs/gst/rtp/gstrtpbaseaudiopayload.h: + * gst-libs/gst/rtp/gstrtpbasedepayload.c: + * gst-libs/gst/rtp/gstrtpbasedepayload.h: + * gst-libs/gst/rtp/gstrtpbasepayload.c: + * gst-libs/gst/rtp/gstrtpbasepayload.h: + rename files to match object names + +2011-11-11 12:24:08 +0100 Wim Taymans + + * docs/libs/gst-plugins-base-libs-sections.txt: + * docs/libs/gst-plugins-base-libs.types: + * gst-libs/gst/rtp/gstbasertpaudiopayload.c: + * gst-libs/gst/rtp/gstbasertpaudiopayload.h: + * gst-libs/gst/rtp/gstbasertpdepayload.c: + * gst-libs/gst/rtp/gstbasertpdepayload.h: + * gst-libs/gst/rtp/gstbasertppayload.c: + * gst-libs/gst/rtp/gstbasertppayload.h: + * gst-libs/gst/rtp/gstrtcpbuffer.c: + * gst-libs/gst/rtp/gstrtpbuffer.c: + rename BaseRTP -> RTPBase + +2011-11-11 12:00:52 +0100 Wim Taymans + + * docs/libs/gst-plugins-base-libs-sections.txt: + * docs/libs/gst-plugins-base-libs.types: + * gst-libs/gst/audio/Makefile.am: + * gst-libs/gst/audio/gstaudiobasesink.c: + * gst-libs/gst/audio/gstaudiobasesink.h: + * gst-libs/gst/audio/gstaudiobasesrc.c: + * gst-libs/gst/audio/gstaudiobasesrc.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: + rename baseaudio* -> audiobase* + +2011-11-11 11:52:47 +0100 Wim Taymans + + * docs/libs/gst-plugins-base-libs-sections.txt: + * docs/libs/gst-plugins-base-libs.types: + * ext/alsa/gstalsasrc.c: + * gst-libs/gst/audio/gstaudioclock.c: + * gst-libs/gst/audio/gstaudioringbuffer.c: + * 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: + rename GstBaseAudio* ->GstAudioBase* + +2011-11-11 11:33:15 +0100 Wim Taymans + + * docs/libs/gst-plugins-base-libs-sections.txt: + * docs/libs/gst-plugins-base-libs.types: + * gst-libs/gst/audio/Makefile.am: + * gst-libs/gst/audio/gstaudioiec61937.h: + * gst-libs/gst/audio/gstaudioringbuffer.c: + * gst-libs/gst/audio/gstaudioringbuffer.h: + * gst-libs/gst/audio/gstbaseaudiosink.h: + * gst-libs/gst/audio/gstbaseaudiosrc.h: + * gst-libs/gst/audio/gstringbuffer.c: + * gst-libs/gst/audio/gstringbuffer.h: + rename files to match contained objects + +2011-11-11 11:21:41 +0100 Wim Taymans + + * docs/libs/gst-plugins-base-libs-sections.txt: + * docs/libs/gst-plugins-base-libs.types: + * ext/alsa/gstalsasink.c: + * ext/alsa/gstalsasrc.c: + * gst-libs/gst/audio/gstaudioiec61937.c: + * gst-libs/gst/audio/gstaudioiec61937.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.c: + * gst-libs/gst/audio/gstringbuffer.h: + audio: GstRingBuffer -> GstAudioRingBuffer + +2011-11-11 10:54:39 +0100 Wim Taymans + + * gst-libs/gst/audio/gstaudiosink.c: + * gst-libs/gst/audio/gstaudiosrc.c: + audio: rename internal audio ringbuffer + +2011-11-11 10:27:27 +0100 Wim Taymans + + * gst-libs/gst/audio/gstaudioprocess.c: + * gst-libs/gst/audio/gstaudioprocess.h: + * gst-libs/gst/audio/gstaudioringbuffer.c: + * gst-libs/gst/audio/gstaudioringbuffer.h: + * gst-libs/gst/audio/gstbaseaudiosrc.c.orig: + * gst-libs/gst/audio/gstbaseaudiosrc.c.rej: + * gst-libs/gst/audio/gstringbufferthread.c: + * gst-libs/gst/audio/gstringbufferthread.h: + * gst-libs/gst/cdda/gst-plugins-base-sha1-2.patch: + * gst-libs/gst/cdda/gstcddabasesrc.c.orig: + * gst-libs/gst/rtp/gst-plugins-base-rtcp-feedback.patch: + * gst-libs/gst/rtp/gstbasertppayload.c.orig: + * gst-libs/gst/rtp/gstbasertppayload.c.rej: + * gst-libs/gst/rtp/gstrtpbuffer.c.new: + * gst-libs/gst/rtsp/gstrtspconnection.c.orig: + * gst-libs/gst/rtsp/rtsp-marshal.c: + * gst-libs/gst/rtsp/rtsp-marshal.h: + * gst-libs/gst/rtsp/rtspdefs.patch: + * gst/videorate/videorate-discont.patch: + remove bogus files + They got somehow commited in 7012e88090e69339c60a4eb9449f7a7e39ca6aa3 + +2011-11-10 23:02:35 +0200 Stefan Sauer + + * gst/volume/gstvolume.c: + * tests/icles/audio-trickplay.c: + controller: port controller api changes + +2011-11-10 18:32:39 +0100 Wim Taymans + + * ext/libvisual/visual.c: + * ext/theora/gsttheoraenc.c: + * gst-libs/gst/audio/gstaudioencoder.c: + * gst/gdp/gstgdpdepay.c: + * gst/subparse/gstsubparse.c: + update for adapter api changes + +2011-11-10 18:30:31 +0100 Wim Taymans + + * tests/check/libs/gstlibscpp.cc: + tests: fix build after removal of base64 lib + +2011-11-10 17:52:36 +0100 Wim Taymans + + * gst-libs/gst/video/gstvideosink.h: + videosink: reset padding + +2011-11-10 17:39:10 +0100 Wim Taymans + + * gst-libs/gst/rtsp/Makefile.am: + * gst-libs/gst/rtsp/gstrtspbase64.c: + * gst-libs/gst/rtsp/gstrtspbase64.h: + * gst-libs/gst/rtsp/gstrtspconnection.c: + rtsp: remove deprecated base64 library + +2011-11-10 17:26:12 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-11-10 17:18:00 +0100 Wim Taymans + + * docs/libs/gst-plugins-base-libs.types: + * gst-libs/gst/rtp/gstbasertpaudiopayload.c: + * gst-libs/gst/rtp/gstbasertpdepayload.c: + * gst-libs/gst/rtp/gstbasertpdepayload.h: + * gst-libs/gst/rtp/gstbasertppayload.c: + * gst-libs/gst/rtp/gstbasertppayload.h: + rtp: fix de/payloaders + gst_basertppayload -> gst_base_rtp_payload + Add pts/dts support in the depayloader + Remove old timestamp code + Add a default getcaps function so subclasses can chain up to it instead of + relying on the return value of the getcaps function. + +2011-11-10 15:55:31 +0000 Vincent Penquerc'h + + * gst-libs/gst/audio/gstbaseaudiosink.c: + baseaudiosink: make unsigned properties unsigned, not signed + +2011-11-10 16:24:12 +0100 Wim Taymans + + * gst-libs/gst/audio/gstbaseaudiosink.c: + * gst-libs/gst/audio/gstbaseaudiosrc.c: + audio: fix base class vmethods + +2011-11-10 16:02:01 +0100 Wim Taymans + + * ext/alsa/gstalsa.c: + * ext/alsa/gstalsasrc.c: + alsa: fix negotiation + Don't assume the format is a string because now it is a list of string in the + template. + Chain up to the parent class implementation of get_caps. + +2011-11-10 16:00:28 +0100 Wim Taymans + + * gst-libs/gst/audio/gstaudiosrc.c: + audiosrc: avoid deadlock + +2011-11-10 14:37:02 +0000 Vincent Penquerc'h + + * ext/vorbis/gstvorbisenc.c: + vorbisenc: fix getcaps ignoring filter caps + +2011-11-10 14:24:30 +0000 Vincent Penquerc'h + + * gst/audioconvert/gstaudioconvert.c: + audioconvert: truncate caps in _fixate + Otherwise the resulting caps may not be fixed. + +2011-11-10 14:18:54 +0000 Vincent Penquerc'h + + * ext/ogg/gstoggdemux.c: + oggdemux: do not try to write empty header buffers + Those are valid, and the EOS skeleton packet is actually empty. + +2011-11-10 13:02:13 +0000 Vincent Penquerc'h + + * ext/ogg/gstoggmux.c: + oggmux: split request pad templates into audio/video/subtitle + https://bugzilla.gnome.org/show_bug.cgi?id=663766 + +2011-11-10 13:50:08 +0100 Wim Taymans + + * gst-libs/gst/audio/gstaudioclock.c: + * gst-libs/gst/audio/gstaudioclock.h: + * gst-libs/gst/audio/gstbaseaudiosink.c: + * gst-libs/gst/audio/gstbaseaudiosrc.c: + audioclock: remove _full version + +2011-11-10 13:45:39 +0100 Wim Taymans + + * gst-libs/gst/app/gstappsink.h: + appsink: fix header + +2011-11-10 12:47:51 +0100 Edward Hervey + + * gst-libs/gst/pbutils/encoding-profile.c: + * gst-libs/gst/pbutils/encoding-target.c: + * gst-libs/gst/pbutils/gstdiscoverer-types.c: + pbutils: Fix introspection annotations + Fixes #663689 + +2011-11-10 11:42:10 +0100 Edward Hervey + + * tests/check/libs/struct_arm.h: + tests: Remove old structures from struct_arm.h + +2011-11-10 11:02:12 +0100 Wim Taymans + + * ext/libvisual/visual.c: + * ext/pango/gsttextrender.c: + update for removed fixate functions + +2011-11-09 17:37:31 +0100 Wim Taymans + + * gst/playback/gststreamsynchronizer.c: + * gst/playback/gstsubtitleoverlay.c: + upates for new ACCEPT_CAPS query + +2011-11-09 12:11:59 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + common + ext/pango/gsttextoverlay.c + gst-libs/gst/video/video.c + +2011-11-09 11:47:54 +0100 Wim Taymans + + * ext/ogg/gstoggdemux.c: + * ext/ogg/gstogmparse.c: + * ext/theora/gsttheoradec.c: + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/cdda/gstcddabasesrc.c: + * gst-libs/gst/tag/gsttagdemux.c: + * gst/audioresample/gstaudioresample.c: + remove query types + +2011-11-09 11:06:10 +0100 Wim Taymans + + * gst/playback/Makefile.am: + * gst/playback/gstplayback.c: + * gst/playback/gststreamselector.c: + * gst/playback/gststreamselector.h: + remove streamselector + It was only used by playbin, which is gone now + +2011-11-09 10:53:38 +0100 Wim Taymans + + * gst/playback/gststreamselector.c: + streamselector: GstSelectorPad -> GstStreamSelectorPad + Rename object to avoid conflicts with an object of the same name in core. + +2011-11-09 10:37:02 +0100 Wim Taymans + + * gst/playback/gststreamselector.c: + streamselector: cleanups + +2011-11-09 00:36:51 +0000 Tim-Philipp Müller + + * common: + * configure.ac: + configure: suppress warnings about unused variables if debugging system is disabled in core + https://bugzilla.gnome.org/show_bug.cgi?id=662952 + +2011-10-27 14:48:52 +0100 Vincent Penquerc'h + + * ext/pango/gsttextoverlay.c: + textoverlay: continue processing text when silent + This prevents playback wegding when text buffers are + left to pile up. + https://bugzilla.gnome.org/show_bug.cgi?id=662829 + +2011-11-08 11:07:18 +0100 Wim Taymans + + * gst-libs/gst/pbutils/gstdiscoverer.c: + * gst/playback/gstdecodebin2.c: + * gst/playback/gstplaybin2.c: + * gst/playback/gstplaysink.c: + * gst/playback/gstplaysinkconvertbin.c: + * gst/playback/gstsubtitleoverlay.c: + * gst/playback/gsturidecodebin.c: + * tests/check/elements/vorbistag.c: + * tests/check/pipelines/oggmux.c: + * tests/check/pipelines/theoraenc.c: + * tests/check/pipelines/vorbisenc.c: + * tests/icles/audio-trickplay.c: + update for pad probe api changes + +2011-11-08 08:22:56 +0100 Stefan Sauer + + * gst-libs/gst/video/video.c: + video: log important details and fix format strings + If we complain about wrong parameters passed, also log the actual value. + +2011-11-08 00:16:56 +0000 Tim-Philipp Müller + + * win32/common/libgstaudio.def: + win32: update .def file for new audiosink API + API: gst_base_audio_sink_get_alignment_threshold() + API: gst_base_audio_sink_set_alignment_threshold() + API: gst_base_audio_sink_get_discont_wait() + API: gst_base_audio_sink_set_discont_wait() + +2011-11-07 23:41:33 +0000 Tim-Philipp Müller + + * tests/examples/seek/seek.c: + examples: sprinkle GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS in seek test utility + https://bugzilla.gnome.org/show_bug.cgi?id=630497 + +2011-11-07 23:05:44 +0000 Tim-Philipp Müller + + * ext/pango/gsttextoverlay.c: + * gst-libs/gst/audio/gstaudioiec61937.c: + * gst-libs/gst/audio/gstbaseaudiosink.c: + * gst-libs/gst/audio/gstbaseaudiosink.h: + * gst-libs/gst/video/video.c: + docs: fix up some Since: markers + +2011-11-07 18:19:51 +0000 Vincent Penquerc'h + + * gst/videoconvert/videoconvert.c: + videoconvert: fix r210 writing only half a scanline + +2011-11-07 17:18:06 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-11-07 17:10:48 +0100 Wim Taymans + + * gst-libs/gst/pbutils/gstdiscoverer.c: + * gst/playback/gstdecodebin2.c: + * gst/playback/gstplaybin2.c: + * gst/playback/gstplaysink.c: + * gst/playback/gstplaysinkconvertbin.c: + * gst/playback/gstsubtitleoverlay.c: + * gst/playback/gsturidecodebin.c: + fix for new pad probe types + Restore the previous behaviour by only blocking downstream items and not + upstream events. + +2011-11-04 10:34:27 +0000 Vincent Penquerc'h + + * ext/theora/gsttheoraenc.c: + theoraenc: fix speed level failure test + It was testing the opposite of what it thought it was. + https://bugzilla.gnome.org/show_bug.cgi?id=663390 + +2011-11-04 10:57:40 +0000 Vincent Penquerc'h + + * ext/theora/gsttheoraenc.c: + theoraenc: make logically static const data just so + https://bugzilla.gnome.org/show_bug.cgi?id=663391 + +2011-11-04 10:58:15 +0000 Vincent Penquerc'h + + * ext/theora/gsttheoraenc.c: + theoraenc: use th_packet_iskeyframe instead of peeking at bits + https://bugzilla.gnome.org/show_bug.cgi?id=663391 + +2011-11-04 10:59:00 +0000 Vincent Penquerc'h + + * ext/theora/gsttheoraenc.c: + theoraenc: trivial comment typos fixes + https://bugzilla.gnome.org/show_bug.cgi?id=663391 + +2011-11-04 10:59:12 +0000 Vincent Penquerc'h + + * ext/theora/gsttheoraenc.c: + theoraenc: warn when trying to set an ignored obsolete property + https://bugzilla.gnome.org/show_bug.cgi?id=663391 + +2011-11-04 11:10:46 +0000 Vincent Penquerc'h + + * ext/theora/gsttheoraenc.c: + theoraenc: refuse to get to READY if the encoder was disabled + https://bugzilla.gnome.org/show_bug.cgi?id=663391 + +2011-10-18 17:58:49 +0100 Vincent Penquerc'h + + * ext/ogg/gstoggdemux.c: + oggdemux: survive skeleton finding length behind our backs in push mode + In push mode, we determine duration by doing a seek to the end of the + stream. However, a skeleton stream with an index will cause the duration + to be known already, and we end up never setting the push_time_duration + variable which we use to know duration has been determined. + https://bugzilla.gnome.org/show_bug.cgi?id=662049 + +2011-10-05 15:29:54 +0100 Vincent Penquerc'h + + * tests/check/gst-plugins-base.supp: + valgrind: add ALSA leaks fixed by snd_config_update_free_global + If they go when calling snd_config_update_free_global, they're + not really bug leaks, but more like intentional ones we don't + want to get told about. + https://bugzilla.gnome.org/show_bug.cgi?id=615342 + +2011-11-07 12:43:37 +0100 Wim Taymans + + * gst/playback/gstplaysinkconvertbin.c: + * gst/playback/gstplaysinkconvertbin.h: + convertbin: port to 0.11 again + +2011-11-07 12:23:15 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + common + configure.ac + gst-libs/gst/audio/gstbaseaudiosink.c + gst/playback/gstdecodebin2.c + gst/playback/gstplaysinkaudioconvert.c + gst/playback/gstplaysinkaudioconvert.h + gst/playback/gstplaysinkvideoconvert.c + gst/playback/gstplaysinkvideoconvert.h + +2011-05-02 13:05:28 +0300 Felipe Contreras + + * gst-libs/gst/audio/gstbaseaudiosink.c: + * gst-libs/gst/audio/gstbaseaudiosink.h: + baseaudiosink: make discont-wait configurable + Now we can configure how much time to wait before deciding that a + discont has happened. + Also, adds getter and setter to allow derived implementations to set + this value upon construction. + Suggestions and several improvements by Havard Graff. + Signed-off-by: Felipe Contreras + +2011-11-07 11:31:47 +0100 Felipe Contreras + + * gst-libs/gst/audio/gstbaseaudiosink.c: + baseaudiosink: delay the resyncing of timestamp vs ringbuffertime + A common problem for audio-playback is that the timestamps might not + be completely linear. This is specially common when doing streaming over + a network, where you can have jittery and/or bursty packettransmission, + which again will often be reflected on the buffertimestamps. + Now, the current implementation have a threshold that says how far the + buffertimestamp is allowed o drift from the ideal aligned time in the + ringbuffer. This was an instant reaction, and ment that if one buffer + arrived with a timestamp that would breach the drift-tolerance, a resync + would take place, and the result would be an audible gap for the + listener. + The annoying thing would be that in the case of a "timestamp-outlier", + you would first resync one way, say +100ms, and then, if the next + timestamp was "back on track", you would end up resyncing the other way + (-100ms) So in fact, when you had only one buffer with slightly off + timestamping, you would end up with *two* audible gaps. This is the + problem this patch addresses. + The way to "fix" this problem with the previous implementation, would + have been to increase the "drift-tolerance" to a value that was greater + than the largest timestamp-outlier one would normally expect. The big + problem with this approach, however, is that it will allow normal + operations with a huge offset timestamp vs running-time, which is + detrimental to lip-sync. If the drift-tolerance is set to 200ms, it + basically means that lip-sync can easily end up being off by that much. + This patch will basically start a timer when the first breach of + drift-tolerance is detected. If any following timestamp for the next n + nanoseconds gets "back on track" within the threshold, it has basically + eliminated the effect of an outlier, and the timer is stopped. If, + however, all timestamps within this time-limit are breaching the + threshold, we are probably facing a more permanent offset in the + timestamps, and a resync is allowed to happen. + So basically this patch offers something as rare as both higher + accuracy, it terms of allowing smaller drift-tolerances, as well as much + smoother, less glitchy playback! + Commit message and improvments by Havard Graff. + Fixes bug #640859. + +2011-11-07 11:18:34 +0100 Felipe Contreras + + * gst-libs/gst/audio/gstbaseaudiosink.c: + baseaudiosink: rename some variables + +2011-05-21 16:16:42 +0300 Felipe Contreras + + * gst-libs/gst/audio/gstbaseaudiosink.c: + baseaudiosink: use gst_util_uint64_scale_int when appropriate + It's probably safer this way. + +2011-05-21 15:49:20 +0300 Felipe Contreras + + * gst-libs/gst/audio/gstbaseaudiosink.c: + * gst-libs/gst/audio/gstbaseaudiosink.h: + baseaudiosink: split drift-tolerance into alignment-threshold + So that drift-tolerance is used for clock slaving resync, and + alignment-threshold is for timestamp drift. + +2011-05-21 16:02:36 +0300 Felipe Contreras + + * gst-libs/gst/audio/gstbaseaudiosink.c: + baseaudiosink: trivial comment fixes + Some found by Havard Graff. + Signed-off-by: Felipe Contreras + +2011-11-04 22:00:43 +0100 Stefan Sauer + + * gst/adder/gstadder.c: + adder: don't ref NULL caps + +2011-11-04 21:00:29 +0100 Stefan Sauer + + * gst/volume/gstvolume.c: + volume: use new api to check activity of a controller + +2011-11-04 15:23:25 +0100 Stefan Sauer + + * ext/pango/Makefile.am: + * ext/pango/gstbasetextoverlay.c: + * ext/pango/gstbasetextoverlay.h: + * gst/audiotestsrc/Makefile.am: + * gst/audiotestsrc/gstaudiotestsrc.c: + * gst/volume/Makefile.am: + * gst/volume/gstvolume.c: + * tests/check/elements/volume.c: + * tests/icles/audio-trickplay.c: + controller: port to new location and api changes + +2011-11-04 17:40:01 +0100 Wim Taymans + + * gst/playback/gstplaysinkaudioconvert.c: + * gst/playback/gstplaysinkvideoconvert.c: + playback: name conversion elements differently + +2011-11-04 15:36:25 +0100 Stefan Sauer + + * tests/examples/encoding/Makefile.am: + * tools/Makefile.am: + build: add audio libs (pulled by pbutils) to avoid linking against system version + +2011-11-04 13:21:24 +0100 Wim Taymans + + * gst-libs/gst/audio/gstringbuffer.c: + ringbuffer: store bpf in the right variable + +2011-11-04 13:01:52 +0100 Wim Taymans + + * docs/design/design-decodebin.txt: + * docs/design/design-encoding.txt: + docs: fix some docs + +2011-11-04 13:00:36 +0100 Wim Taymans + + * gst/playback/gsturidecodebin.c: + uridecodebin: fix template name + +2011-11-04 12:53:33 +0100 Wim Taymans + + * ext/ogg/gstoggdemux.c: + oggdemux: fix somtimes pad + +2011-11-04 10:48:50 +0100 Wim Taymans + + * ext/ogg/gstoggmux.c: + * gst/adder/gstadder.c: + * gst/encoding/gstencodebin.c: + * gst/encoding/gststreamcombiner.c: + * gst/encoding/gststreamsplitter.c: + * gst/playback/gstdecodebin.c: + * gst/playback/gstdecodebin2.c: + * gst/playback/gstplaysink.c: + * gst/playback/gststreamselector.c: + * gst/playback/gststreamsynchronizer.c: + * tests/check/elements/adder.c: + * tests/check/pipelines/oggmux.c: + * tests/examples/dynamic/sprinkle.c: + * tests/examples/dynamic/sprinkle2.c: + * tests/examples/dynamic/sprinkle3.c: + fix pad template names for request pads + +2011-11-04 10:37:12 +0100 Sebastian Dröge + + * gst/playback/gstsubtitleoverlay.c: + subtitleoverlay: Use gst_caps_merge() instead of gst_caps_union() + This keeps the caps order and is more efficient. + +2011-11-04 10:36:51 +0100 Sebastian Dröge + + * gst/playback/gstplaysinkconvertbin.c: + playsinkconvertbin: Use gst_caps_merge() instead of gst_caps_union() + This keeps the caps order and is more efficient. + +2011-11-04 08:41:00 +0100 Edward Hervey + + * gst-libs/gst/rtp/gstrtpbuffer.c: + rtpbuffer: Fix compilation issues with gcc 4.6.1 + +2011-11-04 08:58:23 +0100 Edward Hervey + + * win32/common/libgstvideo.def: + win32: Update for modified API + +2011-11-04 08:57:45 +0100 Edward Hervey + + * Android.mk: + * android/netbuffer.mk: + * docs/libs/gst-plugins-base-libs-docs.sgml: + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-plugins-base.spec.in: + * pkgconfig/Makefile.am: + * pkgconfig/gstreamer-plugins-base-uninstalled.pc.in: + * pkgconfig/gstreamer-plugins-base.pc.in: + * tests/check/Makefile.am: + * tests/check/libs/.gitignore: + * tests/check/libs/gstlibscpp.cc: + * tests/check/libs/libsabi.c: + * tests/check/libs/netbuffer.c: + * tests/check/libs/struct_arm.h: + * tests/check/libs/struct_i386_osx.h: + * tests/check/libs/struct_x86_64.h: + * win32/MANIFEST: + * win32/common/libgstnetbuffer.def: + * win32/vs6/gst_plugins_base.dsw: + * win32/vs6/libgstnetbuffer.dsp: + Really remove all mention of gstnetbuffer + +2011-11-03 21:35:38 -0300 Reynaldo H. Verdejo Pinochet + + * gst-libs/gst/tag/Makefile.am: + Add missing default include paths to androgenizer call + Fixes building tag/ with Android's NDK + +2011-11-03 17:58:57 +0100 Wim Taymans + + * gst/playback/gstdecodebin2.c: + * gst/playback/gstplaybin2.c: + * gst/playback/gstplaysink.c: + * tests/examples/dynamic/codec-select.c: + * tests/icles/output-selector-test.c: + update for request pads change. + +2011-11-03 16:48:51 +0100 Wim Taymans + + * configure.ac: + * gst-libs/gst/Makefile.am: + * gst-libs/gst/netbuffer/Makefile.am: + * gst-libs/gst/netbuffer/README: + * gst-libs/gst/netbuffer/gstnetbuffer.c: + * gst-libs/gst/netbuffer/gstnetbuffer.h: + * pkgconfig/gstreamer-netbuffer-uninstalled.pc.in: + * pkgconfig/gstreamer-netbuffer.pc.in: + net: remove net library, it's now in core + +2011-11-03 14:10:31 +0200 Mart Raudsepp + + * gst/playback/gstdecodebin2.c: + decodebin2: Post all source pads in stream-topology messages as "element-srcpad" values + This allows us to easily get ahold of all pads on a stream-topology message, including + pre-decoder ones, while "pad" only gives us access to the raw pads (as used by discoverer). + +2011-10-20 13:04:52 +0300 Mart Raudsepp + + * gst/playback/gstdecodebin2.c: + decodebin2: Use existing "caps" quark for one of the structure sets + +2011-11-03 14:19:50 +0100 Wim Taymans + + * tests/check/libs/netbuffer.c: + tests: fix netbuffer test + +2011-11-03 10:07:27 +0100 Sebastian Dröge + + * gst/playback/gstplaysinkconvertbin.c: + playsinkconvertbin: Don't add identity multiple times + +2011-10-19 14:13:39 +0100 Vincent Penquerc'h + + * gst/playback/gstplaysinkconvertbin.c: + playsink: send flush start/stop event when we switch elements + https://bugzilla.gnome.org/show_bug.cgi?id=661262 + +2011-10-19 14:13:30 +0100 Vincent Penquerc'h + + * gst/playback/gstplaysinkaudioconvert.c: + * gst/playback/gstplaysinkconvertbin.c: + * gst/playback/gstplaysinkconvertbin.h: + playsink: re-add identity where appropriate + https://bugzilla.gnome.org/show_bug.cgi?id=661262 + +2011-10-19 14:12:01 +0100 Vincent Penquerc'h + + * gst/playback/gstplaysinkaudioconvert.c: + playsink: lock the new {set,get}_property functions + https://bugzilla.gnome.org/show_bug.cgi?id=661262 + +2011-10-17 23:14:54 +0000 Thiago Santos + + * gst/playback/gstplaysinkconvertbin.c: + playsinkconvertbin: Be more consistent with ghostpad targets + Set up targets on READY->PAUSED state change to passthrough by + default. This prevents the targets from being unset on the + first run, while the 'raw' variable would mean that some + target is set. + +2011-10-17 22:41:49 +0000 Thiago Santos + + * gst/playback/gstplaysinkconvertbin.c: + playsinkconvertbin: No need to remove the identity + The identity element should be handled by the GstBin's cleanup, + removing it on the remove_elements function might remove it + too soon, as this function can be called directly from playsink + +2011-10-17 22:41:11 +0000 Thiago Santos + + * gst/playback/gstplaysinkconvertbin.c: + playsinkconvertbin: Adding some debug messages + Adds a couple debug messages and some g_assert to make debugging + easier + +2011-10-17 22:02:03 +0000 Thiago Santos + + * gst/playback/gstplaysinkvideoconvert.c: + playsink-videoconvert: Fix warning on build + Remove unused variable + +2011-10-17 21:05:30 +0000 Vincent Penquerc'h + + * gst/playback/gstplaysink.c: + * gst/playback/gstplaysinkaudioconvert.c: + * gst/playback/gstplaysinkaudioconvert.h: + * gst/playback/gstplaysinkconvertbin.c: + * gst/playback/gstplaysinkconvertbin.h: + * gst/playback/gstplaysinkvideoconvert.c: + * gst/playback/gstplaysinkvideoconvert.h: + playsink: handle after-the-fact changes in converters/volume booleans + The playsink was nastily poking a boolean in the structure. + Make those booleans properties, so we are told when they change, + and rebuild the conversion bin when they do. + Some cleanup to go with it too. + https://bugzilla.gnome.org/show_bug.cgi?id=661262 + +2011-10-17 18:43:06 +0000 Vincent Penquerc'h + + * gst/playback/gstplaysinkconvertbin.c: + playsink: handle NULL cached caps in getcaps + https://bugzilla.gnome.org/show_bug.cgi?id=661262 + +2011-10-17 18:06:00 +0000 Vincent Penquerc'h + + * gst/playback/gstplaysinkconvertbin.c: + playsink: consider both passthrough and converter caps in getcaps + Since we can switch between both modes. + https://bugzilla.gnome.org/show_bug.cgi?id=661262 + +2011-10-17 17:54:27 +0000 Vincent Penquerc'h + + * gst/playback/gstplaysinkconvertbin.c: + * gst/playback/gstplaysinkconvertbin.h: + playsink: cache inner converter bin caps + https://bugzilla.gnome.org/show_bug.cgi?id=661262 + +2011-10-17 17:26:48 +0000 Vincent Penquerc'h + + * gst/playback/gstplaysinkconvertbin.c: + playsink: keep both raw and non raw pipelines at all times + and switch between them as needed. + https://bugzilla.gnome.org/show_bug.cgi?id=661262 + +2011-10-17 17:29:50 +0000 Vincent Penquerc'h + + * gst/playback/gstplaysinkconvertbin.c: + playsink: only compare against the media type we expect + ie, audio/x-raw- for audio, video/x-raw- for video. + Add a trailing - to be more specific. I doubt there's anything + like audio/x-rawhide or something, but you never know. + https://bugzilla.gnome.org/show_bug.cgi?id=661262 + +2011-10-17 16:55:30 +0000 Vincent Penquerc'h + + * gst/playback/Makefile.am: + * gst/playback/gstplaysinkaudioconvert.c: + * gst/playback/gstplaysinkaudioconvert.h: + * gst/playback/gstplaysinkconvertbin.c: + * gst/playback/gstplaysinkconvertbin.h: + * gst/playback/gstplaysinkvideoconvert.c: + * gst/playback/gstplaysinkvideoconvert.h: + playsink: refactor the converter bins since they are almost identical + https://bugzilla.gnome.org/show_bug.cgi?id=661262 + +2011-10-17 13:00:05 +0000 Vincent Penquerc'h + + * gst/playback/gstplaysinkaudioconvert.c: + * gst/playback/gstplaysinkaudioconvert.h: + * gst/playback/gstplaysinkvideoconvert.c: + * gst/playback/gstplaysinkvideoconvert.h: + playsink: fix passthrough mode (hopefully) + The code was doing counterintuitive rewiring of pads when the + bin did not contain any elements. We now add an identity element + in that case, which makes it simpler, and should fix the AC3 + passthrough mode when using pulseaudio (but I don't see the bug + here so can't test). + https://bugzilla.gnome.org/show_bug.cgi?id=661262 + +2011-10-07 11:16:44 +0000 Vincent Penquerc'h + + * gst/playback/gstplaysinkaudioconvert.c: + * gst/playback/gstplaysinkvideoconvert.c: + playsink: handle NULL ghost pad target + For the src pad anyway. + https://bugzilla.gnome.org/show_bug.cgi?id=661262 + +2011-11-03 09:56:14 +0100 Sebastian Dröge + + * gst/playback/gstplaysinkaudioconvert.c: + Revert "playsinkaudioconvert: Fix warning when there is no target pad yet" + This reverts commit f35c51c14915729f0fdf2b348f351ea7e81027cc. + Better patch coming soon. + +2011-10-28 10:07:42 +0200 Sebastian Dröge + + * ext/ogg/gstoggmux.c: + oggmux: Remove obsolete #include + +2011-11-02 23:33:18 +0000 Tim-Philipp Müller + + * docs/design/draft-subtitle-overlays.txt: + docs: add draft for subtitle overlays to design docs + Main purpose is to provide a generic way to make subtitles work on + top of non-raw video (vaapi, vdpau, etc.). + +2011-11-02 15:31:11 -0400 Colin Walters + + * common: + * configure.ac: + configure: Allow setting GLIB_EXTRA_CFLAGS + Similar to gstreamer commit bb2020b1e794210cf7d44c6626122f611016a620 + +2011-11-02 12:08:22 +0100 Wim Taymans + + * ext/ogg/gstoggdemux.c: + * ext/ogg/gstogmparse.c: + * ext/theora/gsttheoradec.c: + * ext/vorbis/gstvorbisdec.c: + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/cdda/gstcddabasesrc.c: + * gst/subparse/gstssaparse.c: + * gst/subparse/gstsubparse.c: + update for tag API removal + +2011-11-02 11:24:05 +0100 Edward Hervey + + * gst-libs/gst/video/video.h: + video: Add convenience macros for accessing GstVideoInfo flags + +2011-10-31 02:39:48 +0100 Wim Taymans + + * gst-libs/gst/netbuffer/gstnetbuffer.c: + * gst-libs/gst/netbuffer/gstnetbuffer.h: + netbuffer: _netaddress_ -> _net_address_ + +2011-10-31 02:35:36 +0100 Wim Taymans + + * gst-libs/gst/netbuffer/gstnetbuffer.c: + * gst-libs/gst/netbuffer/gstnetbuffer.h: + netaddress: updata api + +2011-10-31 02:23:21 +0100 Wim Taymans + + * ext/theora/gsttheoradec.c: + * gst-libs/gst/video/Makefile.am: + * gst-libs/gst/video/gstmetavideo.c: + * gst-libs/gst/video/gstmetavideo.h: + * gst-libs/gst/video/gstvideometa.c: + * gst-libs/gst/video/gstvideometa.h: + * gst-libs/gst/video/gstvideopool.h: + * gst-libs/gst/video/video.c: + * gst/videoconvert/gstvideoconvert.c: + * gst/videoscale/gstvideoscale.c: + * gst/videotestsrc/gstvideotestsrc.c: + * gst/videotestsrc/gstvideotestsrc.h: + * sys/ximage/ximagepool.c: + * sys/ximage/ximagepool.h: + * sys/ximage/ximagesink.c: + * sys/xvimage/xvimagepool.c: + * sys/xvimage/xvimagepool.h: + * sys/xvimage/xvimagesink.c: + rename meta* -> *meta + +2011-10-29 09:28:57 +0200 Wim Taymans + + * ext/alsa/gstalsamixer.c: + alsa: update for new task api + +2011-10-29 09:03:07 +0200 Wim Taymans + + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/interfaces/videooverlay.c: + * gst-libs/gst/pbutils/gstdiscoverer.c: + * gst/encoding/gststreamsplitter.c: + * gst/playback/gstdecodebin2.c: + * gst/playback/gstplaybin2.c: + structure: fix for api update + +2011-10-29 08:25:07 +0200 Wim Taymans + + * gst-libs/gst/app/gstappsink.c: + * gst-libs/gst/rtp/gstbasertpaudiopayload.c: + bufferlist: update for new API + +2011-11-01 00:34:28 +0000 Tim-Philipp Müller + + * gst-libs/gst/audio/gstbaseaudiosink.c: + * gst-libs/gst/pbutils/gstdiscoverer.c: + * gst/playback/gstdecodebin2.c: + * gst/playback/gstplaybin2.c: + * gst/playback/gstplaysink.c: + * gst/playback/gstplaysinkaudioconvert.c: + * gst/playback/gstplaysinkvideoconvert.c: + * gst/playback/gstsubtitleoverlay.c: + * gst/playback/gsturidecodebin.c: + * tests/check/elements/vorbistag.c: + * tests/check/pipelines/oggmux.c: + * tests/check/pipelines/theoraenc.c: + * tests/check/pipelines/vorbisenc.c: + * tests/icles/audio-trickplay.c: + Update for pad API changes + GstProbeType, GstProbeReturn and GstActivateMode -> GstPad* + +2011-10-31 14:26:09 +0000 Tim-Philipp Müller + + * gst/playback/gstsubtitleoverlay.c: + subtitleoverlay: don't include header that's been removed + +2011-10-31 14:22:58 +0000 Tim-Philipp Müller + + Merge remote-tracking branch 'origin/master' into 0.11 + +2011-10-30 14:51:48 +0000 Tim-Philipp Müller + + * gst-libs/gst/audio/gstaudioencoder.c: + audioencoder: save audio info parsed in setcaps in encoder context + Otherwise we'll just error out when the first buffer gets pushed. + This is a porting artefact, in 0.10 the infos were allocated on the + heap, now we're doing everything with stack-allocated structs. + +2011-10-30 11:09:10 +0000 Tim-Philipp Müller + + * ext/ogg/gstoggdemux.c: + * ext/ogg/gstoggstream.c: + * ext/ogg/gstogmparse.c: + * ext/theora/gsttheoradec.c: + * ext/vorbis/gstvorbisdec.c: + * ext/vorbis/gstvorbisenc.c: + * gst-libs/gst/audio/gstaudioencoder.c: + * gst-libs/gst/cdda/gstcddabasesrc.c: + * gst-libs/gst/riff/riff-read.c: + * gst-libs/gst/tag/gstexiftag.c: + * gst-libs/gst/tag/gstid3tag.c: + * gst-libs/gst/tag/gstvorbistag.c: + * gst-libs/gst/tag/gstxmptag.c: + * gst-libs/gst/tag/id3v2.c: + * gst/audiotestsrc/gstaudiotestsrc.c: + * gst/subparse/gstssaparse.c: + * gst/subparse/gstsubparse.c: + * tests/check/elements/vorbistag.c: + * tests/check/libs/pbutils.c: + * tests/check/libs/tag.c: + * tests/check/libs/xmpwriter.c: + ext, gst, gst-libs, tests: update for tag list API changes + +2011-10-31 15:16:36 +0100 René Stadler + + * gst-libs/gst/audio/gstaudiofilterexample.c: + audio: remove old C file generated from template + Not sure how this one got pulled into a merge. In 0.10, it was moved away to + gst-template a long time ago. gstaudiofilterexample.c got generated from + gstaudiofiltertemplate.c. + +2011-10-30 20:00:47 +0000 Tim-Philipp Müller + + * gst/playback/gstsubtitleoverlay.c: + subtitleoverlay: don't use soon-to-be-deprecated gst_filter_run() + +2011-10-28 18:45:09 +0200 Edward Hervey + + * configure.ac: + configure.ac: Fix build + +2011-10-28 16:24:44 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-10-28 16:11:36 +0200 Wim Taymans + + * gst-libs/gst/tag/gsttagdemux.c: + fix compile for SEEK_TYPE_CUR removal + +2011-10-28 13:58:47 +0200 Mersad Jelacic + + * gst-libs/gst/audio/gstaudiosink.c: + audiosink: avoid deadlocking audioringbuffer thread + ... when it goes into wait for ringbuffer starting just after such + having been signalled. + Fixes #661738. + +2011-10-28 11:37:31 +0200 Wim Taymans + + * gst-libs/gst/audio/gstaudiofilter.c: + audiofilter: use BPF for unit_size + +2011-10-28 11:34:37 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-10-28 10:44:38 +0200 René Stadler + + * gst-libs/gst/audio/gstaudiofilter.c: + audiofilter: fix get_unit_size + +2011-10-28 11:13:52 +0200 René Stadler + + * gst-libs/gst/audio/gstaudiofilter.c: + audiofilter: init audio info sooner + +2011-10-28 11:11:55 +0200 René Stadler + + * gst-libs/gst/audio/audio.c: + * gst-libs/gst/video/video.c: + audio, video: init audio/video format info to UNKNOWN format + This is to prevent e.g. GST_AUDIO_INFO_FORMAT() from crashing on a NULL pointer + dereference when used with an unset info. + +2011-04-26 22:20:29 +0200 Philip Jägenstedt + + * gst/typefind/gsttypefindfunctions.c: + typefind: extract SOF marker in jpeg typefinder + The SOF types are defined by http://www.w3.org/Graphics/JPEG/itu-t81.pdf + This is needed to make sure that we plug a jpeg decoder that + can handle the type of JPEG we have (e.g. lossless JPEG) + https://bugzilla.gnome.org/show_bug.cgi?id=556648 + +2009-08-10 01:48:29 +0000 Thiago Santos + + * ext/ogg/gstoggmux.c: + * ext/ogg/gstoggmux.h: + oggmux: port to gstcollectpads2 + +2011-10-27 18:54:50 +0200 Wim Taymans + + * gst-libs/gst/rtp/gstbasertppayload.c: + basertppay: rename caps fields + Make the caps fields for timestamp and seqnum match the element + properties. + See #628773 + +2011-10-27 18:50:32 +0200 Wim Taymans + + * gst-libs/gst/rtp/gstbasertppayload.c: + * gst-libs/gst/rtp/gstbasertppayload.h: + basedepay: remove old fields + +2011-10-27 17:33:06 +0200 Wim Taymans + + * tests/check/elements/encodebin.c: + * tests/check/libs/pbutils.c: + * tests/check/libs/profile.c: + tests: fix compilation + +2011-10-27 17:26:58 +0200 Wim Taymans + + * ext/alsa/gstalsa.c: + * ext/ogg/gstoggaviparse.c: + * ext/ogg/gstoggdemux.c: + * ext/ogg/gstoggstream.c: + * ext/ogg/gstogmparse.c: + * ext/vorbis/gstvorbisenc.c: + * gst-libs/gst/riff/riff-media.c: + * gst-libs/gst/rtp/gstbasertppayload.c: + * gst/subparse/gstsubparse.c: + * gst/typefind/gsttypefindfunctions.c: + fix compilation + +2011-10-27 15:44:58 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + configure.ac + gst-libs/gst/audio/gstbaseaudiosink.c + gst/audioconvert/channelmixtest.c + gst/playback/gstplaybasebin.c + gst/playback/gstsubtitleoverlay.c + tests/examples/Makefile.am + tests/examples/audio/Makefile.am + +2011-10-27 15:29:36 +0200 Wim Taymans + + * gst-libs/gst/interfaces/videooverlay.c: + overlay: fix compilation + +2011-10-27 23:39:31 +1100 Jan Schmidt + + * tests/examples/Makefile.am: + build: Fix build for moved volume subdir + +2011-10-27 09:51:46 +0200 Stefan Sauer + + * Makefile.am: + * configure.ac: + * tests/examples/Makefile.am: + * tests/examples/audio/.gitignore: + * tests/examples/audio/Makefile.am: + * tests/examples/audio/volume.c: + * tests/examples/volume/.gitignore: + * tests/examples/volume/Makefile.am: + * tests/examples/volume/volume.c: + volume: move volume example to audio + +2011-10-27 09:42:36 +0200 Stefan Sauer + + * tests/examples/audio/Makefile.am: + audio examples. fix the makefile + +2011-10-27 09:33:55 +0200 Stefan Sauer + + * tests/examples/volume/volume.c: + volume: make global vars static + +2011-10-27 09:33:01 +0200 Stefan Sauer + + * tests/examples/audio/.gitignore: + * tests/examples/audio/Makefile.am: + * tests/examples/audio/audiomix.c: + audiomix: add a simple audiomix example + +2011-10-25 20:04:06 +1100 Jan Schmidt + + * gst/playback/gstplaysinkaudioconvert.c: + playsinkaudioconvert: Fix warning when there is no target pad yet + +2011-10-13 11:34:49 -0400 Nicolas Dufresne + + * gst/playback/gstdecodebin2.c: + decodebin2: Link elements before testing if they can reach the READY state + This is made possible by filtering errors. This is required to let + harware accelerated element query the video context. The video context + is used to determine if the HW is capable, and thus if the element is + supported or not. + Fixes bug #662330. + +2011-10-21 21:57:17 +0200 René Stadler + + * gst/playback/gstplaybasebin.c: + playbasebin: remove avoidable call to gst_object_set_name + +2011-10-21 21:41:03 +0200 René Stadler + + * ext/ogg/gstoggdemux.c: + oggdemux: remove avoidable call to gst_object_set_name + +2011-10-21 21:39:01 +0200 René Stadler + + * gst/audioconvert/Makefile.am: + * gst/audioconvert/channelmixtest.c: + audioconvert: bury dead test program + +2011-10-21 14:37:31 +0200 Stefan Sauer + + * docs/libs/gst-plugins-base-libs-sections.txt: + * ext/alsa/gstalsamixer.h: + * gst-libs/gst/audio/gstaudioprocess.c: + * gst-libs/gst/audio/gstaudioprocess.h: + * gst-libs/gst/interfaces/colorbalance.c: + * gst-libs/gst/interfaces/colorbalance.h: + * gst-libs/gst/interfaces/mixer.c: + * gst-libs/gst/interfaces/mixer.h: + * gst-libs/gst/interfaces/navigation.c: + * gst-libs/gst/interfaces/navigation.h: + * gst-libs/gst/interfaces/propertyprobe.c: + * gst-libs/gst/interfaces/propertyprobe.h: + * gst-libs/gst/interfaces/tuner.c: + * gst-libs/gst/interfaces/tuner.h: + * gst-libs/gst/interfaces/videoorientation.c: + * gst-libs/gst/interfaces/videoorientation.h: + * gst-libs/gst/interfaces/videooverlay.c: + * gst-libs/gst/interfaces/videooverlay.h: + * gst-libs/gst/rtsp/gstrtspextension.c: + * gst-libs/gst/rtsp/gstrtspextension.h: + * gst/volume/gstvolume.c: + * sys/ximage/ximagesink.c: + * sys/xvimage/xvimagesink.c: + * tests/check/libs/mixer.c: + * tests/check/libs/navigation.c: + * tests/check/libs/struct_arm.h: + * tests/check/libs/struct_i386.h: + * tests/check/libs/struct_i386_osx.h: + * tests/check/libs/struct_x86_64.h: + interfaces: clean up the use of iface and class/klass + +2011-10-20 10:13:46 -0300 Reynaldo H. Verdejo Pinochet + + * Android.mk: + Disable ext/vorbis for the android ndk build + It currently makes the build fail. Idea is to enable + it back again once its building problems get sorted + out. + +2011-10-19 19:44:06 +0200 René Stadler + + * gst/playback/gstsubtitleoverlay.c: + subtitleoverlay: fix leaks of pad templates and internal proxy pads + +2011-10-19 19:37:07 +0200 René Stadler + + * gst/playback/gstsubtitleoverlay.c: + subtitleoverlay: fix leak of element reference through pad block + If the pad block never happens because there is no data flow at all, the + callback is never fired and the reference is never released. This causes a + reference cycle between the pad and element, so valgrind is not very vocal + about it (memory is still reachable). + +2011-10-18 21:42:21 +0200 Mark Nauwelaerts + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: having gather queue contents implies some draining is in order + ... which ensures e.g. processing and sending last fragment of reverse playback + downstream at EOS. + +2011-10-19 15:28:44 +0100 Vincent Penquerc'h + + * ext/vorbis/gstvorbisdec.c: + vorbisdec: do not try to read past the buffer array + https://bugzilla.gnome.org/show_bug.cgi?id=662108 + +2011-10-18 21:40:54 +0200 Mark Nauwelaerts + + * ext/vorbis/gstvorbisdec.c: + vorbisdec: only finish header packet frame if received in-stream + ... rather than scaring audiodecoder with a frame extracted from caps. + Fixes #662108 (partially). + +2011-10-19 10:41:31 +0200 Stefan Sauer + + * sys/ximage/ximagesink.c: + * sys/xvimage/xvimagesink.c: + x(v)imagesink: make it more clean that "synchronous" props are not for avsync + +2011-10-19 00:32:13 +0100 Tim-Philipp Müller + + * gst-libs/gst/audio/gstbaseaudiosink.c: + baseaudiosink: fix unused variable compiler warning if debugging in core is disabled + https://bugzilla.gnome.org/show_bug.cgi?id=660150 + +2011-10-18 13:00:29 +0200 René Stadler + + * gst/playback/gstsubtitleoverlay.c: + subtitleoverlay: fix event unref in (rare) error case + +2011-10-17 15:41:58 +0100 Tim-Philipp Müller + + * gst/audiotestsrc/gstaudiotestsrc.c: + audiotestsrc: fix crash when setting the wave property before having negotiated a format + https://bugzilla.gnome.org/show_bug.cgi?id=661911 + +2011-10-07 17:41:32 +0100 Vincent Penquerc'h + + * gst/playback/gstdecodebin2.c: + decodebin2: fire drained signal where appropriate + This will allow playbin2 to send its about-to-finish signal. + Taken out (apparently by mistake) by the EOS rewrite in july. + https://bugzilla.gnome.org/show_bug.cgi?id=661202 + +2011-10-17 12:28:58 +0200 Edward Hervey + + * gst/audioconvert/gstaudioconvert.c: + audioconvert: We can handle channels conversion + +2011-10-17 12:00:55 +0200 Edward Hervey + + * gst-libs/gst/audio/audio.c: + audio: Add some default channel positions + +2011-10-17 12:00:16 +0200 Edward Hervey + + * gst-libs/gst/audio/audio.c: + * tests/check/libs/audio.c: + audio: Properly handle signedness in gst_audio_format_build_integer() + +2011-10-16 11:32:41 +0100 Vincent Penquerc'h + + * ext/ogg/gstoggdemux.c: + oggdemux: do not retry seeking indefinitely + https://bugzilla.gnome.org/show_bug.cgi?id=661897 + +2011-10-17 11:45:39 +0200 Edward Hervey + + * gst-libs/gst/audio/audio.c: + audio: Indent and doc fixes + +2011-10-13 08:53:34 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-10-11 17:42:35 +0200 Edward Hervey + + * gst-libs/gst/pbutils/gstdiscoverer.c: + discoverer: Only call gst_video_info_from_caps on raw video + +2011-10-10 12:15:37 -0300 Thiago Santos + + * gst/audiotestsrc/gstaudiotestsrc.c: + audiotestsrc: update blocksize when caps or samples-per-buffer change + Blocksize needs to be updated so we get a correct size buffer on + _fill function. + +2011-10-10 13:11:59 +0200 Brian Cameron + + * gst/videotestsrc/Makefile.am: + videotestsrc: fix LDADD missing GST_LIBS + +2011-10-10 11:45:49 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + ext/vorbis/gstvorbisenc.c + +2011-10-10 11:39:52 +0200 Wim Taymans + + * ext/gio/gstgiobasesrc.c: + * ext/ogg/gstoggdemux.c: + * ext/ogg/gstoggmux.c: + * ext/pango/gstbasetextoverlay.c: + * gst-libs/gst/app/gstappsrc.c: + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstbaseaudiosink.c: + * gst-libs/gst/cdda/gstcddabasesrc.c: + * gst-libs/gst/riff/riff-read.c: + * gst-libs/gst/tag/gsttagdemux.c: + * gst/adder/gstadder.c: + * gst/audiotestsrc/gstaudiotestsrc.c: + * gst/subparse/gstsubparse.c: + * gst/tcp/gsttcp.c: + * gst/videotestsrc/gstvideotestsrc.c: + update for UNEXPECTED -> EOS flowreturn + +2011-10-09 14:21:28 -0300 Thiago Santos + + * gst-libs/gst/video/video.c: + libs: video: Add protection against null strings + Check and assert if input for gst_video_format_from_string is null. + Return GST_VIDEO_FORMAT_UNKNOWN as a fallback + +2011-10-09 13:36:38 -0300 Thiago Santos + + * tests/check/libs/struct_arm.h: + * tests/check/libs/struct_i386.h: + * tests/check/libs/struct_i386_osx.h: + tests: Updating some tests with GstXOverlayClass -> GstVideoOverlayIface + +2011-10-09 21:19:32 +0200 Mark Nauwelaerts + + * ext/vorbis/gstvorbisenc.c: + * ext/vorbis/gstvorbisenc.h: + vorbisenc: only push header buffers following initial events + +2011-10-09 16:15:54 +0100 Tim-Philipp Müller + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: update to 0.11 API after merge + +2011-10-09 16:08:36 +0100 Tim-Philipp Müller + + Merge remote-tracking branch 'origin/master' into 0.11 + Conflicts: + tests/check/pipelines/vorbisdec.c + tests/check/pipelines/vorbisenc.c + +2011-10-09 16:48:18 +0200 Alessandro Decina + + * gst-libs/gst/audio/gstaudiodecoder.c: + audioencoder: fix compile warning + +2011-10-08 20:17:43 +0200 Mark Nauwelaerts + + * tests/check/pipelines/vorbisenc.c: + tests: vorbisenc: adjust discontinuity checking to audioencoder behaviour + ... which still detects gaps and marks DISCONT, depending on configuration, + but may come up with somewhat different timestamps when crossing the gap. + +2011-10-08 20:16:04 +0200 Mark Nauwelaerts + + * tests/check/pipelines/vorbisdec.c: + tests: vorbisdec: properly configure audiodecoder when requiring perfect ts + +2011-10-08 20:14:27 +0200 Mark Nauwelaerts + + * tests/check/elements/vorbisdec.c: + tests: vorbisdec: remove empty header buffer check + ... as empty buffers are discarded, and header buffers are now + also optionally retrieved from caps anyway. + +2011-10-08 20:13:11 +0200 Mark Nauwelaerts + + * gst-libs/gst/audio/gstaudioencoder.c: + audioencoder: only resync to upstream upon discont in perfect ts mode + ... as documented, where discont is marked here if tolerance has been + exceeded. + +2011-10-08 20:11:22 +0200 Mark Nauwelaerts + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: fix timestamp tolerance handling + +2011-10-08 20:09:09 +0200 Mark Nauwelaerts + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: handle empty input by discarding + +2011-10-08 11:05:29 +0200 Wim Taymans + + * ext/vorbis/gstvorbisdec.c: + * ext/vorbis/gstvorbisdeclib.h: + vorbisdec: report to 0.11 + +2011-10-08 10:19:06 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + ext/vorbis/gstvorbisdec.c + ext/vorbis/gstvorbisenc.c + ext/vorbis/gstvorbisenc.h + gst/audiotestsrc/gstaudiotestsrc.c + +2011-10-07 14:52:33 +0200 Mark Nauwelaerts + + * ext/vorbis/Makefile.am: + * ext/vorbis/gstvorbisdec.c: + * ext/vorbis/gstvorbisdec.h: + vorbisdec: port to audiodecoder + +2011-10-07 14:33:04 +0200 Mark Nauwelaerts + + * gst-libs/gst/audio/gstaudioencoder.c: + audioencoder: make upstream queries MT-safe + +2011-10-07 14:32:33 +0200 Mark Nauwelaerts + + * gst-libs/gst/audio/gstaudiodecoder.c: + audiodecoder: make upstream queries and events MT-safe + +2011-10-05 15:43:35 +0200 Mark Nauwelaerts + + * ext/vorbis/Makefile.am: + * ext/vorbis/gstvorbisenc.c: + * ext/vorbis/gstvorbisenc.h: + vorbisenc: port to audioencoder + +2011-10-07 14:05:19 +0200 René Stadler + + * ext/ogg/gstoggdemux.c: + oggdemux: don't leak scheduling query + +2011-10-06 18:21:29 +0100 Vincent Penquerc'h + + * tests/check/elements/audiotestsrc.c: + tests: actually test what we said we would + All tests were testing the default sine wave + https://bugzilla.gnome.org/show_bug.cgi?id=661106 + +2011-10-06 18:20:32 +0100 Vincent Penquerc'h + + * gst/audiotestsrc/gstaudiotestsrc.c: + audiotestsrc: add missing break + And make violet noise usable + https://bugzilla.gnome.org/show_bug.cgi?id=661105 + +2011-10-06 15:38:49 +0100 Vincent Penquerc'h + + * gst/playback/gstplaysinkaudioconvert.c: + * gst/playback/gstplaysinkvideoconvert.c: + playsink: fix caps negotiation through the new convenience bins + The bins' getcaps was bypassing the inner elements, and thus + failing to account for the caps transformations they allow, + which caused YUV video pipelines to fail with ximagesink, which + does not support YUV, even though the convenience bin includes + a colorspace converter for just this purpose. + https://bugzilla.gnome.org/show_bug.cgi?id=660816 + +2011-10-06 11:53:26 +0100 Vincent Penquerc'h + + * gst/playback/gstplaybin2.c: + playbin2: fix mismatch between video/ and video/x-dvd-subpicture + The new code was checking for a prefix, and would find video/ + first. Check in two passes, first checking for a perfect match, + and falling back to a prefix check if nothing was found. + https://bugzilla.gnome.org/show_bug.cgi?id=657261 + +2011-10-04 21:17:37 -0300 Thiago Santos + + * gst/encoding/gstencodebin.c: + encodebin: Re-enable parsers + Re-enable parsers in encodebin to allow more passthrough scenarios + to work. Specially the ones that require changing 'stream formats'. + i.e. h264 in mkv to mpegts. + +2011-10-05 12:45:19 +0200 Robert Swain + + * gst/playback/gstplaysink.c: + playsink: Add audio- and text-sink props + +2011-10-05 11:57:54 +0200 Edward Hervey + + * gst-libs/gst/audio/audio.c: + audio: Make sure 'channels' and 'channel-positions' are coherent + If channel-positions are present, check they match the reported + 'channels' value. + +2011-10-05 11:51:07 +0200 Edward Hervey + + * gst-libs/gst/audio/audio.c: + audio: Fix overread in channel positions + The array we're writing to is limited to 64 ... but the amount of + input positions might be lower than 64. Therefore use MIN and not + MAX to know how many values to read from the array. + +2011-10-04 23:09:42 +0200 Stefan Sauer + + * gst/audiotestsrc/gstaudiotestsrc.c: + auditestsrc: indent fix + +2011-10-04 18:06:07 +0200 Wim Taymans + + * gst/playback/gstplaybin2.c: + playbin2: port new bits to 0.11 + +2011-10-04 17:58:49 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-10-04 17:56:19 +0200 Wim Taymans + + * tests/check/Makefile.am: + Makefile: remove 0.11 fixme + +2011-10-04 16:22:55 +0200 Robert Swain + + * gst/playback/gstplaysink.c: + playsink: Add video-sink property + The video-sink property allows manual specification via g_object_set () + of the video sink element to be used. + +2011-10-03 15:20:06 +0200 Sebastian Dröge + + * gst/playback/gstplaybin2.c: + playbin2: Minor cleanup of decoder-sink compatibility checking code + +2011-09-30 12:29:34 -0300 Thibault Saunier + + * gst/playback/gstplaybin2.c: + playbin2: Make sure that the decoders we plug are compatible with the fixed sink + The fact that a decoder is not compatible with the fixed sink + is currently happenning in the case where we have hardware accelerated + video decoders on the system (especially vaapi elements that are actually plugged), + and the user is providing a sink that doesn't support the surface. + A simple example that shows how it used to crash on a system where gstreamer-vaapi + is installed: + gst-launch playbin2 video-sink=xvimagesink uri=/codec/supported/by/vaapi + What we are now doing in this case, is avoid using the accelerated + decoder and plug a "normal" decoder instead (if avalaible). + This commit doesn't handle the case where we have hardware accelerated + demuxing. + +2011-02-18 11:48:37 +0000 Vincent Penquerc'h + + * docs/libs/gst-plugins-base-libs-sections.txt: + * gst-libs/gst/pbutils/encoding-profile.c: + * gst-libs/gst/pbutils/encoding-profile.h: + * win32/common/libgstpbutils.def: + encoding-profile: add a function to create a profile from a discoverer info + Only A/V streams are added at the moment, there does not seem to be + a similar way to add other streams (eg, subtitles). + https://bugzilla.gnome.org/show_bug.cgi?id=642878 + +2011-09-27 00:26:29 +0100 Vincent Penquerc'h + + * ext/alsa/gstalsasrc.c: + * ext/alsa/gstalsasrc.h: + alsasrc: fail gracefully when ALSA does not give timestamps + https://bugzilla.gnome.org/show_bug.cgi?id=660170 + +2011-10-03 10:55:53 +0200 Sebastian Dröge + + * gst/playback/gstdecodebin2.c: + decodebin2: Use a TIME limit for pre-rolling in live streams and not in non-live streams + Fixes bug #647769 for real. + +2011-10-03 10:11:19 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + ext/pango/gsttextoverlay.c + gst/encoding/gstencodebin.c + +2011-10-03 10:02:21 +0200 Wim Taymans + + * gst-libs/gst/video/video.h: + video: add h264 transfer functions + +2011-10-01 01:05:00 +0100 Vincent Penquerc'h + + * ext/pango/gsttextoverlay.c: + textoverlay: add YV12 support + Basically the same as I420, just with chroma planes swapped. + https://bugzilla.gnome.org/show_bug.cgi?id=660604 + +2011-09-30 09:44:12 -0300 Thiago Santos + + * gst/encoding/gstencodebin.c: + encodebin: Fix typo on formatter adding condition + The condition is if the muxer doesn't have tag setter *and* isn't + a formatter itself. Any of those two conditions makes the muxer + good enough to not need a formatter. + +2011-09-30 10:54:26 +0100 Tim-Philipp Müller + + * gst-libs/gst/audio/gstaudiodecoder.c: + * gst-libs/gst/audio/gstaudioencoder.c: + audio: don't use GST_PTR_FORMAT for segments + Avoids crashes with debugging output enabled. + +2011-09-30 11:45:51 +0200 Wim Taymans + + * gst/playback/gstsubtitleoverlay.c: + Revert "sbutitleoverlay: fix compiler warning" + This reverts commit ed792293e7fc2bd54f4627649bb836a05709b5ab. + Not needed anymore because of another commit + +2011-09-30 11:00:31 +0200 Wim Taymans + + * gst-libs/gst/video/video.h: + video: add another color matrix for mpeg2 + +2011-09-30 11:00:15 +0200 Wim Taymans + + * gst/playback/gstsubtitleoverlay.c: + sbutitleoverlay: fix compiler warning + +2011-09-30 10:59:52 +0200 Wim Taymans + + * gst-libs/gst/video/video.h: + video: fix docs + +2011-09-29 21:50:59 +0100 Tim-Philipp Müller + + * ext/vorbis/gstvorbisdec.c: + vorbisdec: set channel positions + +2011-09-29 21:30:52 +0100 Tim-Philipp Müller + + * gst/playback/gstsubtitleoverlay.c: + subitleoverlay: fix compiler warning + gstsubtitleoverlay.c: In function 'gst_subtitle_overlay_video_sink_event': + gstsubtitleoverlay.c:1736:22: error: 'target' may be used uninitialized in this function + +2011-09-29 17:43:57 +0200 Wim Taymans + + * configure.ac: + back to development + === release 0.11.1 === -2011-09-29 Wim Taymans +2011-09-29 17:43:00 +0200 Wim Taymans + * ChangeLog: + * NEWS: + * RELEASE: * configure.ac: - releasing 0.11.1, "A handful sometimes, A heartful always" + * gst-plugins-base.doap: + * win32/common/_stdint.h: + * win32/common/config.h: + * win32/common/gstrtsp-enumtypes.c: + RELEASE 0.11.1 + +2011-09-29 17:41:34 +0200 Wim Taymans + + * po/af.po: + * po/az.po: + * po/bg.po: + * po/ca.po: + * po/cs.po: + * po/da.po: + * po/de.po: + * po/el.po: + * po/en_GB.po: + * po/eo.po: + * po/es.po: + * po/eu.po: + * po/fi.po: + * po/fr.po: + * po/gl.po: + * po/hu.po: + * po/id.po: + * po/it.po: + * po/ja.po: + * po/lt.po: + * po/lv.po: + * po/nb.po: + * po/nl.po: + * po/or.po: + * po/pl.po: + * po/pt_BR.po: + * po/ro.po: + * po/ru.po: + * po/sk.po: + * po/sl.po: + * po/sq.po: + * po/sr.po: + * po/sv.po: + * po/tr.po: + * po/uk.po: + * po/vi.po: + * po/zh_CN.po: + Update .po files 2011-09-29 13:46:36 +0200 Wim Taymans diff --git a/NEWS b/NEWS index d5d4ab8..0b4ac02 100644 --- a/NEWS +++ b/NEWS @@ -1,9 +1,12 @@ -This is GStreamer Base Plug-ins 0.11.1, "A handful sometimes, A heartful always" +This is GStreamer Base Plug-ins 0.11.2, "Drool Pool" -New in 0.11.1: +New in 0.11.2: - * Ported to 0.11.0 core API changes - * Rename GstXOverlay -> GstVideoOverlay - * Reworked audio caps - * Support for multiple frames in buffers - * Add video colorimetry support + * Parallel installability with 0.10.x series + * Many API cleanups + * Ported to new 0.11 core API changes + * Use new GstSample for snapshots + * Improved video filter base class + * New multichannel caps with mask + * Port network elements to GIO + * Many fixes and improvements diff --git a/RELEASE b/RELEASE index 7f2b749..803d1ad 100644 --- a/RELEASE +++ b/RELEASE @@ -1,5 +1,5 @@ -Release notes for GStreamer Base Plug-ins 0.11.1 "A handful sometimes, A heartful always" +Release notes for GStreamer Base Plug-ins 0.11.2 "Drool Pool" @@ -8,12 +8,12 @@ in the 0.11.x unstable series of the GStreamer Base Plug-ins. -The 0.11.x series is an unstable series targeted at developers. +The 0.11.x series is an unstable series targeted at developers and will +eventually lead up to the stable 1.0 series. It is not API or ABI compatible with the stable 0.10.x series. It is, however, parallel installable with the 0.10.x series. - This module contains a set of reference plugins, base classes for other plugins, and helper libraries. @@ -23,18 +23,18 @@ their development. This module contains elements for, among others: - device plugins: x(v)imagesink, alsa, v4lsrc, cdparanoia + device plugins: x(v)imagesink, alsa, cdparanoia containers: ogg codecs: vorbis, theora text: textoverlay, subparse - sources: audiotestsrc, videotestsrc, gnomevfssrc, giosrc + sources: audiotestsrc, videotestsrc, giosrc network: tcp typefind functions audio processing: audioconvert, adder, audiorate, audioresample, volume visualisation: libvisual video processing: videoconvert aggregate elements: uridecodebin, playbin, decodebin, encodebin - libraries: app, audio, cdda, fft, interfaces, netbuffer, pbutils, riff, rtp, rtsp, sdp, tag, video + libraries: app, audio, cdda, fft, interfaces, pbutils, riff, rtp, rtsp, sdp, tag, video Other modules containing plug-ins are: @@ -56,12 +56,23 @@ contains a set of less supported plug-ins that haven't passed the Features of this release * Parallel installability with 0.10.x series - * Rename GstXOverlay -> GstVideoOverlay - * Reworked audio caps - * Support for multiple frames in buffers - * Add video colorimetry support -There were no bugs fixed in this release - + * Many API cleanups + * Ported to new 0.11 core API changes + * Use new GstSample for snapshots + * Improved video filter base class + * New multichannel caps with mask + * Port network elements to GIO + * Many fixes and improvements + +Bugs fixed in this release + + * 651222 : [0.11] [playbin(2)/uridecodebin] " connection-speed " should be a guint64 instead of a guint + * 656264 : tag: no-return-in-nonvoid-function compiler warning gstxmptag.c:178 + * 659692 : [0.11] circular dependency between libgstpbutils and libgstaudio + * 661911 : Audiotestsrc segfaults when trying to changing 'wave' parameter + * 662607 : printf format mismatch causes compiler warnings + * 663689 : Missing DiscoverInfo API in 0.11 + * 669328 : [0.11] Remove deprecated theoraenc properties Download @@ -90,38 +101,54 @@ Applications Contributors to this release - * Age Bosma * Alessandro Decina - * Alex Lancaster + * Alexey Fisher + * Andoni Morales Alastruey + * Andy Wingo + * Anssi Hannula * Benjamin Otte + * Brian Cameron * Christian Fredrik Kalager Schaller + * Colin Walters + * Damien Lespiau * David Schleef * Edward Hervey - * Erich Schubert - * Iago Toral + * Felipe Contreras + * George Kiagiadakis + * Havard Graff + * Idar Tollefsen * Jan Schmidt - * Jason Kivlighn + * Jason DeRose + * Jonathan Matthew * Josep Torra - * LoneStar + * Kipp Cannon * Mark Nauwelaerts - * Monty Montgomery + * Mart Raudsepp + * Matej Knopp + * Mersad Jelacic + * Michael Smith * Nicolas Dufresne + * Oleksij Rempel (Alexey Fisher) + * Olivier Aubert * Olivier Crête - * Raimo Järvi + * Pascal Buhler + * Philip Flarsheim + * Philip Jägenstedt + * Philippe Normand + * Piotr Fusik * René Stadler + * Reynaldo H. Verdejo Pinochet * Robert Swain + * Ryan Lortie * Sebastian Dröge - * Sergey Scobich - * Sergey Scobich) - * Sjoerd Simons * Stefan Kost * Stefan Sauer - * Sébastien Moutte * Thiago Santos + * Thibault Saunier * Thomas Vander Stichele * Tim-Philipp Müller - * Tommi Myöhänen * Vincent Penquerc'h * Wim Taymans * Youness Alaoui + * Zaheer Abbas Merali   \ No newline at end of file diff --git a/configure.ac b/configure.ac index 22643b5..e69ee3c 100644 --- a/configure.ac +++ b/configure.ac @@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file dnl initialize autoconf dnl releases only do -Wall, git and prerelease does -Werror too dnl use a three digit version number for releases, and four for git/prerelease -AC_INIT(GStreamer Base Plug-ins, 0.11.1.1, +AC_INIT(GStreamer Base Plug-ins, 0.11.2, http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer, gst-plugins-base) @@ -49,7 +49,7 @@ dnl - interfaces added/removed/changed -> increment CURRENT, REVISION = 0 dnl - interfaces added -> increment AGE dnl - interfaces removed -> AGE = 0 dnl sets GST_LT_LDFLAGS -AS_LIBTOOL(GST, 26, 0, 0) +AS_LIBTOOL(GST, 27, 0, 0) dnl FIXME: this macro doesn't actually work; dnl the generated libtool script has no support for the listed tags. @@ -60,7 +60,7 @@ AC_LIBTOOL_WIN32_DLL AM_PROG_LIBTOOL dnl *** required versions of GStreamer stuff *** -GST_REQ=0.11.1 +GST_REQ=0.11.2 dnl *** autotools stuff **** diff --git a/docs/plugins/gst-plugins-base-plugins.hierarchy b/docs/plugins/gst-plugins-base-plugins.hierarchy index 3834b1a..d6e5300 100644 --- a/docs/plugins/gst-plugins-base-plugins.hierarchy +++ b/docs/plugins/gst-plugins-base-plugins.hierarchy @@ -26,8 +26,10 @@ GObject GstGioBaseSink GstGioSink GstGioStreamSink - GstMultiSocketSink - GstTCPServerSink + GstMultiHandleSink + GstMultiFdSink + GstMultiSocketSink + GstTCPServerSink GstTCPClientSink GstVideoSink GstXImageSink @@ -84,6 +86,15 @@ GObject GstTheoraDec GstTheoraEnc GstTheoraParse + GstVisual + GstVisualbumpscope + GstVisualcorona + GstVisualinfinite + GstVisualjakdaw + GstVisualjess + GstVisuallv_analyzer + GstVisuallv_scope + GstVisualoinksie GstVorbisParse GstVorbisTag GstPad @@ -98,8 +109,6 @@ GObject GInputStream GOutputStream GSocket - GTypeModule - PangoModule GstColorBalanceChannel GstEncodingProfile GstMixerTrack diff --git a/docs/plugins/gst-plugins-base-plugins.signals b/docs/plugins/gst-plugins-base-plugins.signals index 5a6ba71..3b78ae0 100644 --- a/docs/plugins/gst-plugins-base-plugins.signals +++ b/docs/plugins/gst-plugins-base-plugins.signals @@ -27,12 +27,12 @@ gint arg1 l GstMultiFdSink *gstmultifdsink gint arg1 -GstClientStatus arg2 +GstMultiHandleSinkClientStatus arg2 GstMultiFdSink::get-stats -GValueArray* +GstStructure* la GstMultiFdSink *gstmultifdsink gint arg1 @@ -60,10 +60,10 @@ gint arg1 la GstMultiFdSink *gstmultifdsink gint arg1 -GstSyncMethod arg2 -GstTCPUnitType arg3 +GstMultiHandleSinkSyncMethod arg2 +GstFormat arg3 guint64 arg4 -GstTCPUnitType arg5 +GstFormat arg5 guint64 arg6 @@ -76,6 +76,14 @@ gint arg1 +GstMultiFdSink::client-handle-removed +void +l +GstMultiFdSink *gstmultifdsink +gint arg1 + + + GstDecodeBin::new-decoded-pad void l @@ -586,7 +594,7 @@ GSocket *arg1 la GstMultiSocketSink *gstmultisocketsink GSocket *arg1 -GstMultiSocketSinkSyncMethod arg2 +GstMultiHandleSinkSyncMethod arg2 GstFormat arg3 guint64 arg4 GstFormat arg5 @@ -614,7 +622,7 @@ GObject *arg1 l GstMultiSocketSink *gstmultisocketsink gint arg1 -GstMultiSocketSinkClientStatus arg2 +GstMultiHandleSinkClientStatus arg2 @@ -650,6 +658,14 @@ GSocket *arg1 +GstMultiSocketSink::client-handle-removed +void +l +GstMultiSocketSink *gstmultisocketsink +GSocket *arg1 + + + GstPlayBin::about-to-finish void l diff --git a/docs/plugins/inspect/plugin-adder.xml b/docs/plugins/inspect/plugin-adder.xml index f818886..69440a4 100644 --- a/docs/plugins/inspect/plugin-adder.xml +++ b/docs/plugins/inspect/plugin-adder.xml @@ -3,10 +3,10 @@ Adds multiple streams ../../gst/adder/.libs/libgstadder.so libgstadder.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-alsa.xml b/docs/plugins/inspect/plugin-alsa.xml index e015efe..50fbc9b 100644 --- a/docs/plugins/inspect/plugin-alsa.xml +++ b/docs/plugins/inspect/plugin-alsa.xml @@ -3,10 +3,10 @@ ALSA plugin library ../../ext/alsa/.libs/libgstalsa.so libgstalsa.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-app.xml b/docs/plugins/inspect/plugin-app.xml index d163242..b97eca1 100644 --- a/docs/plugins/inspect/plugin-app.xml +++ b/docs/plugins/inspect/plugin-app.xml @@ -3,10 +3,10 @@ Elements used to communicate with applications ../../gst/app/.libs/libgstapp.so libgstapp.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-audioconvert.xml b/docs/plugins/inspect/plugin-audioconvert.xml index 946bc3c..e96ce3f 100644 --- a/docs/plugins/inspect/plugin-audioconvert.xml +++ b/docs/plugins/inspect/plugin-audioconvert.xml @@ -3,10 +3,10 @@ Convert audio to different formats ../../gst/audioconvert/.libs/libgstaudioconvert.so libgstaudioconvert.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-audiorate.xml b/docs/plugins/inspect/plugin-audiorate.xml index 841167e..7541cf0 100644 --- a/docs/plugins/inspect/plugin-audiorate.xml +++ b/docs/plugins/inspect/plugin-audiorate.xml @@ -3,10 +3,10 @@ Adjusts audio frames ../../gst/audiorate/.libs/libgstaudiorate.so libgstaudiorate.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-audioresample.xml b/docs/plugins/inspect/plugin-audioresample.xml index 05316af..49e853c 100644 --- a/docs/plugins/inspect/plugin-audioresample.xml +++ b/docs/plugins/inspect/plugin-audioresample.xml @@ -3,10 +3,10 @@ Resamples audio ../../gst/audioresample/.libs/libgstaudioresample.so libgstaudioresample.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-audiotestsrc.xml b/docs/plugins/inspect/plugin-audiotestsrc.xml index 0e95b9d..b504f04 100644 --- a/docs/plugins/inspect/plugin-audiotestsrc.xml +++ b/docs/plugins/inspect/plugin-audiotestsrc.xml @@ -3,10 +3,10 @@ Creates audio test signals of given frequency and volume ../../gst/audiotestsrc/.libs/libgstaudiotestsrc.so libgstaudiotestsrc.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-cdparanoia.xml b/docs/plugins/inspect/plugin-cdparanoia.xml index 03d1870..3941a25 100644 --- a/docs/plugins/inspect/plugin-cdparanoia.xml +++ b/docs/plugins/inspect/plugin-cdparanoia.xml @@ -3,10 +3,10 @@ Read audio from CD in paranoid mode ../../ext/cdparanoia/.libs/libgstcdparanoia.so libgstcdparanoia.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-encoding.xml b/docs/plugins/inspect/plugin-encoding.xml index af0c838..3bac8e6 100644 --- a/docs/plugins/inspect/plugin-encoding.xml +++ b/docs/plugins/inspect/plugin-encoding.xml @@ -3,10 +3,10 @@ various encoding-related elements ../../gst/encoding/.libs/libgstencodebin.so libgstencodebin.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-gdp.xml b/docs/plugins/inspect/plugin-gdp.xml index e36db21..3327701 100644 --- a/docs/plugins/inspect/plugin-gdp.xml +++ b/docs/plugins/inspect/plugin-gdp.xml @@ -3,10 +3,10 @@ Payload/depayload GDP packets ../../gst/gdp/.libs/libgstgdp.so libgstgdp.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-gio.xml b/docs/plugins/inspect/plugin-gio.xml index 5569f52..7afa746 100644 --- a/docs/plugins/inspect/plugin-gio.xml +++ b/docs/plugins/inspect/plugin-gio.xml @@ -3,10 +3,10 @@ GIO elements ../../gst/gio/.libs/libgstgio.so libgstgio.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-libvisual.xml b/docs/plugins/inspect/plugin-libvisual.xml index f86521e..f7ded87 100644 --- a/docs/plugins/inspect/plugin-libvisual.xml +++ b/docs/plugins/inspect/plugin-libvisual.xml @@ -3,11 +3,179 @@ libvisual visualization plugins ../../ext/libvisual/.libs/libgstlibvisual.so libgstlibvisual.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin + + libvisual_bumpscope + libvisual Bumpscope plugin plugin v.0.0.1 + Visualization + Bumpscope visual plugin + Benjamin Otte <otte@gnome.org> + + + sink + sink + always +
audio/x-raw, format=(string)S16LE, layout=(string)interleaved, channels=(int){ 1, 2 }, rate=(int){ 8000, 11250, 22500, 32000, 44100, 48000, 96000 }
+ + + src + source + always +
video/x-raw, format=(string){ BGRx, BGR, RGB16 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
+ + + + + libvisual_corona + libvisual libvisual corona plugin plugin v.0.1 + Visualization + Libvisual corona plugin + Benjamin Otte <otte@gnome.org> + + + sink + sink + always +
audio/x-raw, format=(string)S16LE, layout=(string)interleaved, channels=(int){ 1, 2 }, rate=(int){ 8000, 11250, 22500, 32000, 44100, 48000, 96000 }
+ + + src + source + always +
video/x-raw, format=(string){ BGRx, BGR, RGB16 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
+ + + + + libvisual_infinite + libvisual infinite plugin plugin v.0.1 + Visualization + Infinite visual plugin + Benjamin Otte <otte@gnome.org> + + + sink + sink + always +
audio/x-raw, format=(string)S16LE, layout=(string)interleaved, channels=(int){ 1, 2 }, rate=(int){ 8000, 11250, 22500, 32000, 44100, 48000, 96000 }
+ + + src + source + always +
video/x-raw, format=(string){ BGRx, BGR, RGB16 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
+ + + + + libvisual_jakdaw + libvisual Jakdaw plugin plugin v.0.0.1 + Visualization + jakdaw visual plugin + Benjamin Otte <otte@gnome.org> + + + sink + sink + always +
audio/x-raw, format=(string)S16LE, layout=(string)interleaved, channels=(int){ 1, 2 }, rate=(int){ 8000, 11250, 22500, 32000, 44100, 48000, 96000 }
+ + + src + source + always +
video/x-raw, format=(string){ BGRx, BGR, RGB16 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
+ + + + + libvisual_jess + libvisual jess plugin plugin v.0.1 + Visualization + Jess visual plugin + Benjamin Otte <otte@gnome.org> + + + sink + sink + always +
audio/x-raw, format=(string)S16LE, layout=(string)interleaved, channels=(int){ 1, 2 }, rate=(int){ 8000, 11250, 22500, 32000, 44100, 48000, 96000 }
+ + + src + source + always +
video/x-raw, format=(string){ BGRx, BGR, RGB16 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
+ + + + + libvisual_lv_analyzer + libvisual libvisual analyzer plugin v.1.0 + Visualization + Libvisual analyzer plugin + Benjamin Otte <otte@gnome.org> + + + sink + sink + always +
audio/x-raw, format=(string)S16LE, layout=(string)interleaved, channels=(int){ 1, 2 }, rate=(int){ 8000, 11250, 22500, 32000, 44100, 48000, 96000 }
+ + + src + source + always +
video/x-raw, format=(string){ BGRx, BGR, RGB16 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
+ + + + + libvisual_lv_scope + libvisual libvisual scope plugin v.0.1 + Visualization + Libvisual scope plugin + Benjamin Otte <otte@gnome.org> + + + sink + sink + always +
audio/x-raw, format=(string)S16LE, layout=(string)interleaved, channels=(int){ 1, 2 }, rate=(int){ 8000, 11250, 22500, 32000, 44100, 48000, 96000 }
+ + + src + source + always +
video/x-raw, format=(string){ BGRx, BGR, RGB16 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
+ + + + + libvisual_oinksie + libvisual oinksie plugin plugin v.0.1 + Visualization + Libvisual Oinksie visual plugin + Benjamin Otte <otte@gnome.org> + + + sink + sink + always +
audio/x-raw, format=(string)S16LE, layout=(string)interleaved, channels=(int){ 1, 2 }, rate=(int){ 8000, 11250, 22500, 32000, 44100, 48000, 96000 }
+ + + src + source + always +
video/x-raw, format=(string){ BGRx, BGR, RGB16 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
+ + + \ No newline at end of file diff --git a/docs/plugins/inspect/plugin-ogg.xml b/docs/plugins/inspect/plugin-ogg.xml index 7e757dd..315eb40 100644 --- a/docs/plugins/inspect/plugin-ogg.xml +++ b/docs/plugins/inspect/plugin-ogg.xml @@ -3,10 +3,10 @@ ogg stream manipulation (info about ogg: http://xiph.org) ../../ext/ogg/.libs/libgstogg.so libgstogg.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-pango.xml b/docs/plugins/inspect/plugin-pango.xml index 50739b1..a514ad3 100644 --- a/docs/plugins/inspect/plugin-pango.xml +++ b/docs/plugins/inspect/plugin-pango.xml @@ -3,10 +3,10 @@ Pango-based text rendering and overlay ../../ext/pango/.libs/libgstpango.so libgstpango.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-playback.xml b/docs/plugins/inspect/plugin-playback.xml index e9dd9a8..89452c8 100644 --- a/docs/plugins/inspect/plugin-playback.xml +++ b/docs/plugins/inspect/plugin-playback.xml @@ -3,10 +3,10 @@ various playback elements ../../gst/playback/.libs/libgstplayback.so libgstplayback.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-subparse.xml b/docs/plugins/inspect/plugin-subparse.xml index 8905899..776e675 100644 --- a/docs/plugins/inspect/plugin-subparse.xml +++ b/docs/plugins/inspect/plugin-subparse.xml @@ -3,10 +3,10 @@ Subtitle parsing ../../gst/subparse/.libs/libgstsubparse.so libgstsubparse.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-tcp.xml b/docs/plugins/inspect/plugin-tcp.xml index c949a53..e159d83 100644 --- a/docs/plugins/inspect/plugin-tcp.xml +++ b/docs/plugins/inspect/plugin-tcp.xml @@ -3,13 +3,28 @@ transfer data over the network via TCP ../../gst/tcp/.libs/libgsttcp.so libgsttcp.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin + multifdsink + Multi filedescriptor sink + Sink/Network + Send data to multiple filedescriptors + Thomas Vander Stichele <thomas at apestaart dot org>, Wim Taymans <wim@fluendo.com> + + + sink + sink + always +
ANY
+ + + + multisocketsink Multi socket sink Sink/Network diff --git a/docs/plugins/inspect/plugin-theora.xml b/docs/plugins/inspect/plugin-theora.xml index 6407d32..d04df27 100644 --- a/docs/plugins/inspect/plugin-theora.xml +++ b/docs/plugins/inspect/plugin-theora.xml @@ -3,10 +3,10 @@ Theora plugin library ../../ext/theora/.libs/libgsttheora.so libgsttheora.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin @@ -47,7 +47,7 @@ src source always -
video/x-theora
+
video/x-theora, framerate=(fraction)[ 1/2147483647, 2147483647/1 ], width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ]
 diff --git a/docs/plugins/inspect/plugin-typefindfunctions.xml b/docs/plugins/inspect/plugin-typefindfunctions.xml index 9eb8fdf..fa6a62e 100644 --- a/docs/plugins/inspect/plugin-typefindfunctions.xml +++ b/docs/plugins/inspect/plugin-typefindfunctions.xml @@ -3,10 +3,10 @@ default typefind functions ../../gst/typefind/.libs/libgsttypefindfunctions.so libgsttypefindfunctions.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-videorate.xml b/docs/plugins/inspect/plugin-videorate.xml index 7cc2ea1..e233843 100644 --- a/docs/plugins/inspect/plugin-videorate.xml +++ b/docs/plugins/inspect/plugin-videorate.xml @@ -3,10 +3,10 @@ Adjusts video frames ../../gst/videorate/.libs/libgstvideorate.so libgstvideorate.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-videoscale.xml b/docs/plugins/inspect/plugin-videoscale.xml index 5c60a5c..f740941 100644 --- a/docs/plugins/inspect/plugin-videoscale.xml +++ b/docs/plugins/inspect/plugin-videoscale.xml @@ -3,10 +3,10 @@ Resizes video ../../gst/videoscale/.libs/libgstvideoscale.so libgstvideoscale.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin @@ -20,13 +20,13 @@ sink sink always -
video/x-raw, format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, GRAY8, GRAY16_BE, GRAY16_LE, v308, Y800, Y16, RGB16, RGB15, ARGB64, AYUV64 }, width=(int)[ 1, 32767 ], height=(int)[ 1, 32767 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
+
video/x-raw, format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, GRAY8, GRAY16_BE, GRAY16_LE, v308, Y800, Y16, RGB16, RGB15, ARGB64, AYUV64, NV12 }, width=(int)[ 1, 32767 ], height=(int)[ 1, 32767 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
src source always -
video/x-raw, format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, GRAY8, GRAY16_BE, GRAY16_LE, v308, Y800, Y16, RGB16, RGB15, ARGB64, AYUV64 }, width=(int)[ 1, 32767 ], height=(int)[ 1, 32767 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
+
video/x-raw, format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, GRAY8, GRAY16_BE, GRAY16_LE, v308, Y800, Y16, RGB16, RGB15, ARGB64, AYUV64, NV12 }, width=(int)[ 1, 32767 ], height=(int)[ 1, 32767 ], framerate=(fraction)[ 0/1, 2147483647/1 ]
 diff --git a/docs/plugins/inspect/plugin-videotestsrc.xml b/docs/plugins/inspect/plugin-videotestsrc.xml index 6a40986..1e775b2 100644 --- a/docs/plugins/inspect/plugin-videotestsrc.xml +++ b/docs/plugins/inspect/plugin-videotestsrc.xml @@ -3,10 +3,10 @@ Creates a test video stream ../../gst/videotestsrc/.libs/libgstvideotestsrc.so libgstvideotestsrc.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-volume.xml b/docs/plugins/inspect/plugin-volume.xml index a8018f9..c82cb69 100644 --- a/docs/plugins/inspect/plugin-volume.xml +++ b/docs/plugins/inspect/plugin-volume.xml @@ -3,10 +3,10 @@ plugin for controlling audio volume ../../gst/volume/.libs/libgstvolume.so libgstvolume.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-vorbis.xml b/docs/plugins/inspect/plugin-vorbis.xml index aa22e56..bb8e1f7 100644 --- a/docs/plugins/inspect/plugin-vorbis.xml +++ b/docs/plugins/inspect/plugin-vorbis.xml @@ -3,10 +3,10 @@ Vorbis plugin library ../../ext/vorbis/.libs/libgstvorbis.so libgstvorbis.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin @@ -47,7 +47,7 @@ src source always -
audio/x-vorbis
+
audio/x-vorbis, rate=(int)[ 1, 200000 ], channels=(int)[ 1, 255 ]
 diff --git a/docs/plugins/inspect/plugin-ximagesink.xml b/docs/plugins/inspect/plugin-ximagesink.xml index 316fe5e..7a763da 100644 --- a/docs/plugins/inspect/plugin-ximagesink.xml +++ b/docs/plugins/inspect/plugin-ximagesink.xml @@ -3,10 +3,10 @@ X11 video output element based on standard Xlib calls ../../sys/ximage/.libs/libgstximagesink.so libgstximagesink.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/docs/plugins/inspect/plugin-xvimagesink.xml b/docs/plugins/inspect/plugin-xvimagesink.xml index f3aa346..693cfa3 100644 --- a/docs/plugins/inspect/plugin-xvimagesink.xml +++ b/docs/plugins/inspect/plugin-xvimagesink.xml @@ -3,10 +3,10 @@ XFree86 video output plugin using Xv extension ../../sys/xvimage/.libs/libgstxvimagesink.so libgstxvimagesink.so - 0.11.1.1 + 0.11.2 LGPL gst-plugins-base - GStreamer Base Plug-ins git + GStreamer Base Plug-ins source release Unknown package origin diff --git a/gst-plugins-base.doap b/gst-plugins-base.doap index b0e4670..ca3b699 100644 --- a/gst-plugins-base.doap +++ b/gst-plugins-base.doap @@ -36,6 +36,17 @@ A wide range of video and audio decoders, encoders, and filters are included. + 0.11.2 + 0.11 + Drool Pool + 2012-02-16 + + + + + + + 0.11.1 0.11 A handful sometimes, A heartful always diff --git a/gst/videoconvert/gstvideoconvertorc-dist.c b/gst/videoconvert/gstvideoconvertorc-dist.c index 6645465..7cb87bb 100644 --- a/gst/videoconvert/gstvideoconvertorc-dist.c +++ b/gst/videoconvert/gstvideoconvertorc-dist.c @@ -142,10 +142,10 @@ void orc_matrix3_000_u8 (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, const guint8 * ORC_RESTRICT s2, const guint8 * ORC_RESTRICT s3, int p1, int p2, int p3, int p4, int p5, int n); -void orc_pack_123x (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, +void orc_pack_123x (guint32 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, const guint8 * ORC_RESTRICT s2, const guint8 * ORC_RESTRICT s3, int p1, int n); -void orc_pack_x123 (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, +void orc_pack_x123 (guint32 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, const guint8 * ORC_RESTRICT s2, const guint8 * ORC_RESTRICT s3, int p1, int n); void cogorc_combine2_u8 (guint8 * ORC_RESTRICT d1, @@ -4660,7 +4660,7 @@ orc_matrix3_000_u8 (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, /* orc_pack_123x */ #ifdef DISABLE_ORC void -orc_pack_123x (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, +orc_pack_123x (guint32 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, const guint8 * ORC_RESTRICT s2, const guint8 * ORC_RESTRICT s3, int p1, int n) { @@ -4780,7 +4780,7 @@ _backup_orc_pack_123x (OrcExecutor * ORC_RESTRICT ex) } void -orc_pack_123x (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, +orc_pack_123x (guint32 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, const guint8 * ORC_RESTRICT s2, const guint8 * ORC_RESTRICT s3, int p1, int n) { @@ -4834,7 +4834,7 @@ orc_pack_123x (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, /* orc_pack_x123 */ #ifdef DISABLE_ORC void -orc_pack_x123 (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, +orc_pack_x123 (guint32 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, const guint8 * ORC_RESTRICT s2, const guint8 * ORC_RESTRICT s3, int p1, int n) { @@ -4954,7 +4954,7 @@ _backup_orc_pack_x123 (OrcExecutor * ORC_RESTRICT ex) } void -orc_pack_x123 (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, +orc_pack_x123 (guint32 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, const guint8 * ORC_RESTRICT s2, const guint8 * ORC_RESTRICT s3, int p1, int n) { diff --git a/gst/videoconvert/gstvideoconvertorc-dist.h b/gst/videoconvert/gstvideoconvertorc-dist.h index 6ba5ae6..3437b2c 100644 --- a/gst/videoconvert/gstvideoconvertorc-dist.h +++ b/gst/videoconvert/gstvideoconvertorc-dist.h @@ -92,8 +92,8 @@ void orc_matrix3_u8 (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, c void orc_matrix3_100_u8 (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, const guint8 * ORC_RESTRICT s2, const guint8 * ORC_RESTRICT s3, int p1, int p2, int p3, int n); void orc_matrix3_100_offset_u8 (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, const guint8 * ORC_RESTRICT s2, const guint8 * ORC_RESTRICT s3, int p1, int p2, int p3, int p4, int p5, int n); void orc_matrix3_000_u8 (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, const guint8 * ORC_RESTRICT s2, const guint8 * ORC_RESTRICT s3, int p1, int p2, int p3, int p4, int p5, int n); -void orc_pack_123x (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, const guint8 * ORC_RESTRICT s2, const guint8 * ORC_RESTRICT s3, int p1, int n); -void orc_pack_x123 (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, const guint8 * ORC_RESTRICT s2, const guint8 * ORC_RESTRICT s3, int p1, int n); +void orc_pack_123x (guint32 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, const guint8 * ORC_RESTRICT s2, const guint8 * ORC_RESTRICT s3, int p1, int n); +void orc_pack_x123 (guint32 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, const guint8 * ORC_RESTRICT s2, const guint8 * ORC_RESTRICT s3, int p1, int n); void cogorc_combine2_u8 (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, const guint8 * ORC_RESTRICT s2, int p1, int p2, int n); void cogorc_convert_I420_UYVY (guint8 * ORC_RESTRICT d1, guint8 * ORC_RESTRICT d2, const guint8 * ORC_RESTRICT s1, const guint8 * ORC_RESTRICT s2, const guint8 * ORC_RESTRICT s3, const guint8 * ORC_RESTRICT s4, int n); void cogorc_convert_I420_YUY2 (guint8 * ORC_RESTRICT d1, guint8 * ORC_RESTRICT d2, const guint8 * ORC_RESTRICT s1, const guint8 * ORC_RESTRICT s2, const guint8 * ORC_RESTRICT s3, const guint8 * ORC_RESTRICT s4, int n); diff --git a/po/af.po b/po/af.po index da7c6e0..a3a0419 100644 --- a/po/af.po +++ b/po/af.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins 0.7.6\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2004-03-18 14:16+0200\n" "Last-Translator: Petri Jooste \n" "Language-Team: Afrikaans \n" @@ -192,10 +192,6 @@ msgstr "" msgid "Error while sending data to \"%s:%d\"." msgstr "Fout tydens toemaak van lêer \"%s\"." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "" - msgid "This CD has no audio tracks" msgstr "" diff --git a/po/az.po b/po/az.po index 461caf6..c2d0411 100644 --- a/po/az.po +++ b/po/az.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-0.8.0\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2004-03-19 18:29+0200\n" "Last-Translator: Metin Amiroff \n" "Language-Team: Azerbaijani \n" @@ -193,10 +193,6 @@ msgstr "" msgid "Error while sending data to \"%s:%d\"." msgstr "\"%s\" faylı bağlana bilmədi." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "" - msgid "This CD has no audio tracks" msgstr "" diff --git a/po/bg.po b/po/bg.po index f93421f..ae17074 100644 --- a/po/bg.po +++ b/po/bg.po @@ -8,7 +8,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.32.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2011-04-26 22:31+0300\n" "Last-Translator: Alexander Shopov \n" "Language-Team: Bulgarian \n" @@ -189,10 +189,6 @@ msgstr "Елементът-източник е грешен." msgid "Error while sending data to \"%s:%d\"." msgstr "Грешка при запис във файла „%s:%d“." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Връзката към %s:%d е отказана." - msgid "This CD has no audio tracks" msgstr "В това CD липсва аудио" @@ -507,6 +503,9 @@ msgstr "Непознат елемент-кодер" msgid "Plugin or element of unknown type" msgstr "Приставка или елемент от непознат вид" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Връзката към %s:%d е отказана." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "Некомпресирано видео по равнини YUV 4:2:0" diff --git a/po/ca.po b/po/ca.po index ff15fcb..77e473d 100644 --- a/po/ca.po +++ b/po/ca.po @@ -2,19 +2,22 @@ # Copyright © 2005, 2010 Free Software Foundation, Inc. # This file is put in the public domain. # Jordi Mallach , 2005, 2010. +# Jordi Estrada , 2011. +# Gil Forcada , 2012. # msgid "" msgstr "" -"Project-Id-Version: gst-plugins-base 0.10.30.3\n" +"Project-Id-Version: gst-plugins-base 0.10.32.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" -"PO-Revision-Date: 2010-11-04 23:13+0100\n" -"Last-Translator: Jordi Mallach \n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" +"PO-Revision-Date: 2012-01-01 14:19+0100\n" +"Last-Translator: Gil Forcada \n" "Language-Team: Catalan \n" "Language: ca\n" "MIME-Version: 1.0\n" "Content-Type: text/plain; charset=UTF-8\n" "Content-Transfer-Encoding: 8bit\n" +"Plural-Forms: nplurals=2; plural=n != 1;\n" msgid "Master" msgstr "Mestre" @@ -23,7 +26,7 @@ msgid "Bass" msgstr "Baixos" msgid "Treble" -msgstr "Treble" +msgstr "Aguts" msgid "PCM" msgstr "PCM" @@ -41,7 +44,7 @@ msgid "Microphone" msgstr "Micròfon" msgid "PC Speaker" -msgstr "Altaveu PC" +msgstr "Altaveu del PC" msgid "Playback" msgstr "Reproducció" @@ -50,48 +53,51 @@ msgid "Capture" msgstr "Captura" msgid "Could not open device for playback in mono mode." -msgstr "No s'ha pogut obrir el dispositiu per a reproduir en mode mono." +msgstr "No s'ha pogut obrir el dispositiu per a la reproducció en mode mono." msgid "Could not open device for playback in stereo mode." -msgstr "No s'ha pogut obrir el dispositiu per a reproduir en mode estèreo." +msgstr "" +"No s'ha pogut obrir el dispositiu per a la reproducció en mode estèreo." #, c-format msgid "Could not open device for playback in %d-channel mode." -msgstr "No s'ha pogut obrir el dispositiu per a reproduir en mode %d-canals." +msgstr "" +"No s'ha pogut obrir el dispositiu per a la reproducció en mode %d-canals." msgid "" "Could not open audio device for playback. Device is being used by another " "application." msgstr "" -"No s'ha pogut obrir el dispositiu d'àudio per a reproduir. El dispositiu " -"està en ús per una altra aplicació." +"No s'ha pogut obrir el dispositiu d'àudio per a la reproducció. El " +"dispositiu està en ús per una altra aplicació." msgid "Could not open audio device for playback." -msgstr "No s'ha pogut obrir el dispositiu d'àudio per a reproduir." +msgstr "No s'ha pogut obrir el dispositiu d'àudio per a la reproducció." msgid "Could not open device for recording in mono mode." -msgstr "No s'ha pogut obrir el dispositiu per a enregistrar en mode mono." +msgstr "No s'ha pogut obrir el dispositiu per a l'enregistrament en mode mono." msgid "Could not open device for recording in stereo mode." -msgstr "No s'ha pogut obrir el dispositiu per a enregistrar en mode estèreo." +msgstr "" +"No s'ha pogut obrir el dispositiu per a l'enregistrament en mode estèreo." -# FIXME Trailing dot. jm #, c-format msgid "Could not open device for recording in %d-channel mode" -msgstr "No s'ha pogut obrir el dispositiu per a enregistrar en mode %d-canals" +msgstr "" +"No s'ha pogut obrir el dispositiu per a l'enregistrament en mode %d-canals" msgid "" "Could not open audio device for recording. Device is being used by another " "application." msgstr "" -"No s'ha pogut obrir el dispositiu d'àudio per a enregistrar. El dispositiu " -"està en ús per una altra aplicació." +"No s'ha pogut obrir el dispositiu d'àudio per a l'enregistrament. El " +"dispositiu està en ús per una altra aplicació." msgid "Could not open audio device for recording." -msgstr "No s'ha pogut obrir el dispositiu d'àudio per a enregistrar." +msgstr "No s'ha pogut obrir el dispositiu d'àudio per a l'enregistrament." msgid "Could not open CD device for reading." -msgstr "No s'ha pogut obrir el dispositiu del CD per a lectura." +msgstr "No s'ha pogut obrir el dispositiu del CD per a la lectura." msgid "Could not seek CD." msgstr "No s'ha pogut cercar el CD." @@ -112,7 +118,7 @@ msgid "Could not determine type of stream" msgstr "No s'ha pogut determinar el tipus de flux" msgid "This appears to be a text file" -msgstr "Açò sembla ser un fitxer de text" +msgstr "Això sembla ser un fitxer de text" #, c-format msgid "Missing element '%s' - check your GStreamer installation." @@ -131,13 +137,13 @@ msgstr "El videosink configurat %s no funciona." #, c-format msgid "Both autovideosink and %s elements are not working." -msgstr "Tant l'element autovideosink com el %s no funcionen." +msgstr "Els elements autovideosink i %s no funcionen." msgid "The autovideosink element is not working." msgstr "L'element autovideosink no funciona." msgid "Custom text sink element is not usable." -msgstr "L'element sortida personalitzat de text no es pot emprar." +msgstr "L'element sortida de text personalitzat no es pot utilitzar." msgid "No volume control found" msgstr "No s'ha trobat un control de volum" @@ -155,7 +161,7 @@ msgstr "L'audiosink configurat %s no funciona." #, c-format msgid "Both autoaudiosink and %s elements are not working." -msgstr "No funcionen tant l'element autoaudiosink com el %s." +msgstr "Els elements autoaudiosink i %s no funcionen." msgid "The autoaudiosink element is not working." msgstr "L'element autoaudiosink no funciona." @@ -165,14 +171,14 @@ msgstr "No es pot reproduir un fitxer de text sense vídeo o visualitzacions." #, c-format msgid "No decoder available for type '%s'." -msgstr "No hi ha un decodificador disponible per al tipus «%s»." +msgstr "No hi ha cap descodificador disponible per al tipus «%s»." msgid "No URI specified to play from." -msgstr "No s'ha especificat cap URL des d'on reproduir." +msgstr "No s'ha especificat cap URI des d'on reproduir." #, c-format msgid "Invalid URI \"%s\"." -msgstr "L'URI «%s» és invàlid." +msgstr "L'URI «%s» no és vàlid." msgid "This stream type cannot be played yet." msgstr "Encara no es pot reproduir aquest tipus de flux." @@ -182,16 +188,12 @@ msgid "No URI handler implemented for \"%s\"." msgstr "No hi ha cap gestor d'URI implementat per a «%s»." msgid "Source element is invalid." -msgstr "L'element font és invàlid." +msgstr "L'element font no és vàlid." #, c-format msgid "Error while sending data to \"%s:%d\"." msgstr "S'ha produït un error en enviar dades a «%s:%d»." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "S'ha refusat la connexió amb %s:%d." - msgid "This CD has no audio tracks" msgstr "Aquest CD no té pistes d'àudio" @@ -232,125 +234,132 @@ msgid "MusicBrainz TRM ID" msgstr "ID TRM al MusicBrainz" msgid "capturing shutter speed" -msgstr "velocitat de l'obturador de la captura" +msgstr "velocitat d'obturació de la captura" msgid "Shutter speed used when capturing an image, in seconds" -msgstr "" +msgstr "Velocitat d'obturació utilitzat en capturar una imatge, en segons" msgid "capturing focal ratio" -msgstr "" +msgstr "relació focal de la captura" msgid "Focal ratio (f-number) used when capturing the image" -msgstr "" +msgstr "Relació focal (nombre f) utilitzat en capturar la imatge" msgid "capturing focal length" -msgstr "" +msgstr "longitud focal de la captura" msgid "Focal length of the lens used capturing the image, in mm" -msgstr "" +msgstr "Longitud focal de la lent utilitzada en capturar la imatge, en mm" msgid "capturing digital zoom ratio" -msgstr "" +msgstr "relació de zoom digital de la captura" msgid "Digital zoom ratio used when capturing an image" -msgstr "" +msgstr "Relació de zoom digital utilitzat en capturar una imatge" msgid "capturing iso speed" -msgstr "" +msgstr "velocitat iso de la captura" msgid "The ISO speed used when capturing an image" -msgstr "" +msgstr "La velocitat ISO utilitzada en capturar una imatge" msgid "capturing exposure program" -msgstr "" +msgstr "programa d'exposició de la captura" msgid "The exposure program used when capturing an image" -msgstr "" +msgstr "El programa d'exposició utilitzar en capturar una imatge" msgid "capturing exposure mode" -msgstr "" +msgstr "mode d'exposició de la captura" msgid "The exposure mode used when capturing an image" -msgstr "" +msgstr "El mode d'exposició utilitzat en capturar una imatge" +#, fuzzy msgid "capturing exposure compensation" -msgstr "" +msgstr "mode d'exposició de la captura" +#, fuzzy msgid "The exposure compensation used when capturing an image" -msgstr "" +msgstr "El mode d'exposició utilitzat en capturar una imatge" msgid "capturing scene capture type" -msgstr "" +msgstr "tipus de captura d'escena de la captura" msgid "The scene capture mode used when capturing an image" -msgstr "" +msgstr "El mode de captura d'escena utilitzat en capturar una imatge" msgid "capturing gain adjustment" -msgstr "" +msgstr "ajust de guany de la captura" msgid "The overall gain adjustment applied on an image" -msgstr "" +msgstr "L'ajust del guany general aplicat a una imatge" msgid "capturing white balance" -msgstr "" +msgstr "balanç de blancs de la captura" msgid "The white balance mode set when capturing an image" -msgstr "" +msgstr "El mode de balanç de blancs utilitzat en capturar una imatge" msgid "capturing contrast" -msgstr "" +msgstr "contrast de la captura" msgid "The direction of contrast processing applied when capturing an image" msgstr "" +"La direcció del processament del contrast aplicada en capturar una imatge" msgid "capturing saturation" -msgstr "" +msgstr "saturació de la captura" msgid "The direction of saturation processing applied when capturing an image" msgstr "" +"La direcció del processament de la saturació aplicada en capturar una imatge" msgid "capturing sharpness" -msgstr "" +msgstr "nitidesa de la captura" msgid "The direction of sharpness processing applied when capturing an image" msgstr "" +"La direcció del processament de la nitidesa aplicada en capturar una imatge" msgid "capturing flash fired" -msgstr "" +msgstr "flaix de la captura" msgid "If the flash fired while capturing an image" -msgstr "" +msgstr "Si el flaix s'ha disparat en capturar una imatge" msgid "capturing flash mode" -msgstr "" +msgstr "mode de flaix de la captura" msgid "The selected flash mode while capturing an image" -msgstr "" +msgstr "El mode de flaix seleccionat en capturar una imatge" msgid "capturing metering mode" -msgstr "" +msgstr "mode de mesurament de la captura" msgid "" "The metering mode used while determining exposure for capturing an image" msgstr "" +"El mode de mesurament utilitzat en determinar l'exposició per a capturar una " +"imatge" msgid "capturing source" -msgstr "" +msgstr "font de la captura" msgid "The source or type of device used for the capture" -msgstr "" +msgstr "La font o tipus de dispositiu utilitzat per a la captura" msgid "image horizontal ppi" -msgstr "ppi horitzontal de la imatge" +msgstr "ppp horitzontal de la imatge" msgid "Media (image/video) intended horizontal pixel density in ppi" -msgstr "Densitat horitzontal de píxels del medi (imatge/vídeo), en ppi" +msgstr "Densitat horitzontal de píxels del multimèdia (imatge/vídeo), en ppp" msgid "image vertical ppi" -msgstr "ppi vertical de la imatge" +msgstr "ppp vertical de la imatge" msgid "Media (image/video) intended vertical pixel density in ppi" -msgstr "Densitat vertical de píxels del medi (imatge/vídeo), en ppi" +msgstr "Densitat vertical de píxels del multimèdia (imatge/vídeo), en ppp" msgid "ID3v2 frame" msgstr "" @@ -365,31 +374,31 @@ msgid "APE tag" msgstr "Etiqueta APE" msgid "ICY internet radio" -msgstr "Emisora de ràdio per Internet ICY" +msgstr "Emissora de ràdio per Internet ICY" msgid "Apple Lossless Audio (ALAC)" -msgstr "Apple Lossless Audio (ALAC)" +msgstr "Àudio sense pèrdues d'Apple (ALAC)" msgid "Free Lossless Audio Codec (FLAC)" -msgstr "Free Lossless Audio Codec (FLAC)" +msgstr "Còdec d'àudio sense pèrdues lliure (FLAC)" msgid "Lossless True Audio (TTA)" -msgstr "Lossless True Audio (TTA)" +msgstr "Àudio sense pèrdues real (TTA)" msgid "Windows Media Speech" -msgstr "Windows Media Speech" +msgstr "Parla de Windows Media" msgid "CYUV Lossless" -msgstr "CYUV sense pèrdua" +msgstr "CYUV sense pèrdues" msgid "FFMpeg v1" msgstr "FFMpeg v1" msgid "Lossless MSZH" -msgstr "MSZH sense pèrdua" +msgstr "MSZH sense pèrdues" msgid "Run-length encoding" -msgstr "" +msgstr "Codificació Run-length" msgid "Sami subtitle format" msgstr "Format de subtítols Sami" @@ -406,7 +415,7 @@ msgstr "YUV sense comprimir" #, fuzzy msgid "Uncompressed gray" -msgstr "Imatge en escala de grisos no comprimida" +msgstr "Imatge en escala de grisos sense comprimir" #, fuzzy, c-format msgid "Uncompressed %s YUV %s" @@ -414,7 +423,7 @@ msgstr "YUV sense comprimir" #, fuzzy, c-format msgid "Uncompressed %s%d-bit %s" -msgstr "YUV sense comprimir" +msgstr "Paletitzat %d-bit %s sense comprimir" #, c-format msgid "DivX MPEG-4 Version %d" @@ -426,7 +435,7 @@ msgstr "YUV sense comprimir" #, fuzzy, c-format msgid "Raw %d-bit %s audio" -msgstr "Àudio en cru de %d-bit" +msgstr "Àudio PCM en cru de %d-bit" msgid "Audio CD source" msgstr "Font de CD d'àudio" @@ -435,10 +444,10 @@ msgid "DVD source" msgstr "Font de DVD" msgid "Real Time Streaming Protocol (RTSP) source" -msgstr "Font del Real Time Streaming Protocol (RTSP)" +msgstr "Font del protocol de transmissió en temps real (RTSP)" msgid "Microsoft Media Server (MMS) protocol source" -msgstr "Font del protocol Microsoft Media Server (MMS)" +msgstr "Font del protocol de servidor Microsoft Media (MMS)" #, c-format msgid "%s protocol source" @@ -458,11 +467,11 @@ msgstr "Descarregador RTP %s" #, c-format msgid "%s demuxer" -msgstr "Demultiplexor %s" +msgstr "Desmultiplexor %s" #, c-format msgid "%s decoder" -msgstr "Decodificador %s" +msgstr "Descodificador %s" #, c-format msgid "%s video RTP payloader" @@ -498,7 +507,7 @@ msgid "Unknown element" msgstr "L'element és desconegut" msgid "Unknown decoder element" -msgstr "L'element decodificador és desconegut" +msgstr "L'element descodificador és desconegut" msgid "Unknown encoder element" msgstr "L'element codificador és desconegut" @@ -506,15 +515,6 @@ msgstr "L'element codificador és desconegut" msgid "Plugin or element of unknown type" msgstr "El connector o element és de tipus desconegut" -#~ msgid "Raw PCM audio" -#~ msgstr "Àudio PCM en cru" - -#~ msgid "Raw %d-bit floating-point audio" -#~ msgstr "Àudio en cru de %d-bit en coma flotant" - -#~ msgid "Raw floating-point audio" -#~ msgstr "Àudio en cru en coma flotant" - #~ msgid "Could not open vfs file \"%s\" for writing: %s." #~ msgstr "No s'ha pogut obrir el fitxer vfs «%s» per a l'escriptura: %s." @@ -528,7 +528,7 @@ msgstr "El connector o element és de tipus desconegut" #~ msgstr "S'ha produït un error en escriure al fitxer «%s»." #~ msgid "Invalid subtitle URI \"%s\", subtitles disabled." -#~ msgstr "L'URI de subtítols «%s» és invàlida, s'inhabiliten els subtítols." +#~ msgstr "L'URI de subtítols «%s» no és vàlid, s'inhabiliten els subtítols." #~ msgid "RTSP streams cannot be played yet." #~ msgstr "Encara no es poden reproduir els fluxes RTSP." @@ -543,26 +543,77 @@ msgstr "El connector o element és de tipus desconegut" #~ msgstr "" #~ "Només s'ha detectat un flux de subtítols. O bé esteu carregant un fitxer " #~ "de subtítols o qualsevol altre tipus de fitxer de text, o no s'ha " -#~ "reconegut el fitxer de medi." +#~ "reconegut el fitxer multimèdia." #~ msgid "" #~ "You do not have a decoder installed to handle this file. You might need " #~ "to install the necessary plugins." #~ msgstr "" -#~ "No teniu un decodificador instaŀlat per a gestionar aquest fitxer. És " +#~ "No teniu un descodificador instaŀlat per a gestionar aquest fitxer. És " #~ "possible que necessiteu instaŀlar els connectors necessaris." #~ msgid "This is not a media file" -#~ msgstr "Aquest no és un fitxer de medi" +#~ msgstr "Això no és un fitxer multimèdia" #~ msgid "A subtitle stream was detected, but no video stream." #~ msgstr "S'ha detectat un flux de subtítols, però no un flux de vídeo." #~ msgid "Both autovideosink and xvimagesink elements are missing." -#~ msgstr "Manquen l'element autovideosink i el xvimagesink." +#~ msgstr "Manquen els elements autovideosink i xvimagesink." #~ msgid "Both autoaudiosink and alsasink elements are missing." -#~ msgstr "Manquen l'element autoaudiosink i l'alsasink." +#~ msgstr "Manquen els elements autoaudiosink i alsasink." + +#~ msgid "Error while sending gdp header data to \"%s:%d\"." +#~ msgstr "" +#~ "S'ha produït un error en enviar dades de la capçalera gdp a «%s:%d»." + +# Payload -> càrrega en molts àmbits. jm +#~ msgid "Error while sending gdp payload data to \"%s:%d\"." +#~ msgstr "" +#~ "S'ha produït un error en enviar la càrrega de dades de gdp a «%s:%d»." + +#~ msgid "Connection to %s:%d refused." +#~ msgstr "S'ha refusat la connexió amb %s:%d." + +#~ msgid "Uncompressed planar YUV 4:2:0" +#~ msgstr "YUV 4:2:0 planar sense comprimir" + +#~ msgid "Uncompressed planar YVU 4:2:0" +#~ msgstr "YVU 4:2:0 planar sense comprimir" + +#~ msgid "Uncompressed packed YUV 4:2:2" +#~ msgstr "YUV 4:2:2 empaquetat sense comprimir" + +#~ msgid "Uncompressed packed YUV 4:1:0" +#~ msgstr "YUV 4:1:0 empaquetat sense comprimir" + +#~ msgid "Uncompressed packed YVU 4:1:0" +#~ msgstr "YVU 4:1:0 empaquetat sense comprimir" + +#~ msgid "Uncompressed packed YUV 4:1:1" +#~ msgstr "YUV 4:1:1 empaquetat sense comprimir" + +#~ msgid "Uncompressed packed YUV 4:4:4" +#~ msgstr "YUV 4:4:4 empaquetat sense comprimir" + +#~ msgid "Uncompressed planar YUV 4:2:2" +#~ msgstr "YUV 4:2:2 planar sense comprimir" + +#~ msgid "Uncompressed planar YUV 4:1:1" +#~ msgstr "YUV 4:1:1 planar sense comprimir" + +#~ msgid "Uncompressed black and white Y-plane" +#~ msgstr "Pla Y blanc i negre sense comprimir" + +#~ msgid "Raw PCM audio" +#~ msgstr "Àudio PCM en cru" + +#~ msgid "Raw %d-bit floating-point audio" +#~ msgstr "Àudio en cru de %d-bit en coma flotant" + +#~ msgid "Raw floating-point audio" +#~ msgstr "Àudio en cru en coma flotant" #~ msgid "No device specified." #~ msgstr "No s'ha especificat un dispositiu." @@ -574,16 +625,7 @@ msgstr "El connector o element és de tipus desconegut" #~ msgstr "El dispositiu «%s» ja és en ús." #~ msgid "Could not open device \"%s\" for reading and writing." -#~ msgstr "No s'ha pogut obrir el dispositiu «%s» per a llegir i escriure." - -#~ msgid "Error while sending gdp header data to \"%s:%d\"." -#~ msgstr "" -#~ "S'ha produït un error en enviar dades de la capçalera gdp a «%s:%d»." - -# Payload -> càrrega en molts àmbits. jm -#~ msgid "Error while sending gdp payload data to \"%s:%d\"." -#~ msgstr "" -#~ "S'ha produït un error en enviar la càrrega de dades de gdp a «%s:%d»." +#~ msgstr "No s'ha pogut obrir el dispositiu «%s» per a lectura o escriptura." #~ msgid "discid" #~ msgstr "id del disc" diff --git a/po/cs.po b/po/cs.po index b3df912..4e9eb5e 100644 --- a/po/cs.po +++ b/po/cs.po @@ -8,7 +8,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base-0.10.32.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-12-12 12:57+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2011-05-29 16:49+0200\n" "Last-Translator: Petr Kovar \n" "Language-Team: Czech \n" @@ -190,10 +190,6 @@ msgstr "Zdrojový prvek je neplatný." msgid "Error while sending data to \"%s:%d\"." msgstr "Chyba při odesílání dat na \"%s:%d\"." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Spojení s %s:%d bylo odmítnuto." - msgid "This CD has no audio tracks" msgstr "Toto CD nemá žádné zvukové stopy" @@ -511,6 +507,9 @@ msgstr "Neznámý kodérový prvek" msgid "Plugin or element of unknown type" msgstr "Zásuvný modul nebo prvek neznámého typu" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Spojení s %s:%d bylo odmítnuto." + #~ msgid "Could not open vfs file \"%s\" for writing: %s." #~ msgstr "Nezdařilo se otevření souboru vfs \"%s\" k zápisu: %s." diff --git a/po/da.po b/po/da.po index 591f228..8a05fad 100644 --- a/po/da.po +++ b/po/da.po @@ -33,7 +33,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.32.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2011-04-28 16:38+0100\n" "Last-Translator: Joe Hansen \n" "Language-Team: Danish \n" @@ -213,10 +213,6 @@ msgstr "Kildeelement er ugyldigt." msgid "Error while sending data to \"%s:%d\"." msgstr "Der opstod en fejl under data-overførsel til \"%s:%d\"." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Opkobling til %s:%d afvist." - msgid "This CD has no audio tracks" msgstr "Denne cd har ingen lydspor" @@ -548,6 +544,9 @@ msgstr "Ukendt indkodeelement" msgid "Plugin or element of unknown type" msgstr "Plugin eller element af ukendt type" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Opkobling til %s:%d afvist." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "Ukomprimeret plan-YUV 4:2:0" diff --git a/po/de.po b/po/de.po index 688f4f3..3c05d21 100644 --- a/po/de.po +++ b/po/de.po @@ -9,7 +9,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.32.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2011-04-28 00:04+0200\n" "Last-Translator: Christian Kirbach \n" "Language-Team: German \n" @@ -193,10 +193,6 @@ msgstr "Das Quellelement ist ungültig." msgid "Error while sending data to \"%s:%d\"." msgstr "Fehler beim Senden der Daten nach »%s:%d«." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Verbindung nach %s:%d wurde verweigert." - msgid "This CD has no audio tracks" msgstr "Auf dieser CD befinden sich keine Audio-Titel" @@ -517,6 +513,9 @@ msgstr "Unbekanntes Encoder-Element" msgid "Plugin or element of unknown type" msgstr "Plugin oder Element unbekannten Typs" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Verbindung nach %s:%d wurde verweigert." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "Unkomprimiertes ungepacktes YUV 4:2:0" diff --git a/po/el.po b/po/el.po index fe033f6..e8f5616 100644 --- a/po/el.po +++ b/po/el.po @@ -8,7 +8,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.30.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2010-10-27 12:05+0200\n" "Last-Translator: Michael Kotsarinis \n" "Language-Team: Greek \n" @@ -191,10 +191,6 @@ msgstr "Το στοιχείο προέλευσης δεν είναι έγκυρ msgid "Error while sending data to \"%s:%d\"." msgstr "Σφάλμα κατά την αποστολή δεδομένων σε «%s:%d»." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Άρνηση σύνδεσης σε %s:%d." - msgid "This CD has no audio tracks" msgstr "Αυτό το CD δεν έχει ηχητικά κομμάτια" @@ -526,6 +522,9 @@ msgstr "Άγνωστο στοιχείο κωδικοποιητή" msgid "Plugin or element of unknown type" msgstr "Πρόσθετη λειτουργία ή στοιχείο άγνωστου τύπου" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Άρνηση σύνδεσης σε %s:%d." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "Ασυμπίεστο planar YUV 4:2:0" diff --git a/po/en_GB.po b/po/en_GB.po index 7913497..1f97860 100644 --- a/po/en_GB.po +++ b/po/en_GB.po @@ -6,7 +6,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins 0.8.1\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2004-04-26 10:41-0400\n" "Last-Translator: Gareth Owen \n" "Language-Team: English (British) \n" @@ -191,10 +191,6 @@ msgstr "" msgid "Error while sending data to \"%s:%d\"." msgstr "Error closing file \"%s\"." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "" - msgid "This CD has no audio tracks" msgstr "" diff --git a/po/eo.po b/po/eo.po index 23bad19..fdcef98 100644 --- a/po/eo.po +++ b/po/eo.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.32.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2011-06-04 21:11+0100\n" "Last-Translator: Kristjan SCHMIDT \n" "Language-Team: Esperanto \n" @@ -186,10 +186,6 @@ msgstr "Font-elemento estas nevalide." msgid "Error while sending data to \"%s:%d\"." msgstr "Eraro dum sendado de datumoj al \"%s:%d\"." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "" - msgid "This CD has no audio tracks" msgstr "" diff --git a/po/es.po b/po/es.po index 7492ac1..4ffe700 100644 --- a/po/es.po +++ b/po/es.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.32.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-12-12 12:57+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2011-10-02 15:46+0200\n" "Last-Translator: Jorge González González \n" "Language-Team: Spanish \n" @@ -189,10 +189,6 @@ msgstr "El elemento fuente no es válido." msgid "Error while sending data to \"%s:%d\"." msgstr "Error al enviar los datos a «%s:%d»." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Se rechazó la conexión con %s:%d." - msgid "This CD has no audio tracks" msgstr "Este CD no tiene pistas de sonido" @@ -508,6 +504,9 @@ msgstr "Elemento codificador desconocido" msgid "Plugin or element of unknown type" msgstr "Complemento o elemento de tipo desconocido" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Se rechazó la conexión con %s:%d." + #~ msgid "Could not open vfs file \"%s\" for writing: %s." #~ msgstr "No se pudo abrir el archivo VFS «%s» para escribir: %s." diff --git a/po/eu.po b/po/eu.po index fdcfe34..2593614 100644 --- a/po/eu.po +++ b/po/eu.po @@ -8,7 +8,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base-0.10.26.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2010-03-25 12:32+0100\n" "Last-Translator: Mikel Olasagasti Uranga \n" "Language-Team: Basque \n" @@ -191,10 +191,6 @@ msgstr "Iturburuko elementua baliogabea da." msgid "Error while sending data to \"%s:%d\"." msgstr "Errorea gertatu da datuak \"%s:%d\"(e)ra bidaltzean." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "%s:%d(e)ra konektatzea ukatu da." - msgid "This CD has no audio tracks" msgstr "CD honek ez du audio-pistarik" @@ -509,6 +505,9 @@ msgstr "Kodetzailearen elementu ezezaguna" msgid "Plugin or element of unknown type" msgstr "Mota ezezaguneko plugina edo elementua" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "%s:%d(e)ra konektatzea ukatu da." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "Konprimitu gabeko YUV 4:2:0 planarra" diff --git a/po/fi.po b/po/fi.po index 90567f8..4f0c7d1 100644 --- a/po/fi.po +++ b/po/fi.po @@ -12,7 +12,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.30.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2010-12-31 23:21+0200\n" "Last-Translator: Tommi Vainikainen \n" "Language-Team: Finnish \n" @@ -191,10 +191,6 @@ msgstr "Lähde-elementti on virheellinen." msgid "Error while sending data to \"%s:%d\"." msgstr "Virhe lähetettäessä tietoa kohteeseen \"%s:%d\"." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Yhteys kohteeseen %s:%d estettiin." - msgid "This CD has no audio tracks" msgstr "Tällä CD-levyllä ei ole ääniraitoja" @@ -511,6 +507,9 @@ msgstr "Tuntematon kodekkielementti" msgid "Plugin or element of unknown type" msgstr "Liitännäisen tai elementin tyyppi on tuntematon" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Yhteys kohteeseen %s:%d estettiin." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "Pakkaamaton tasollinen YUV 4:2:0" diff --git a/po/fr.po b/po/fr.po index f4c9b4d..e267a2e 100644 --- a/po/fr.po +++ b/po/fr.po @@ -9,7 +9,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.32.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2011-04-28 09:19+0200\n" "Last-Translator: Claude Paroz \n" "Language-Team: French \n" @@ -194,10 +194,6 @@ msgstr "Élément source non valide." msgid "Error while sending data to \"%s:%d\"." msgstr "Erreur lors de l'envoi de données vers « %s:%d »." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Connexion refusée vers %s:%d." - msgid "This CD has no audio tracks" msgstr "Ce CD ne contient aucune piste audio" @@ -527,6 +523,9 @@ msgstr "Élément codeur inconnu" msgid "Plugin or element of unknown type" msgstr "Greffon ou élément de type inconnu" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Connexion refusée vers %s:%d." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "YUV planaire 4:2:0 non compressé" diff --git a/po/gl.po b/po/gl.po index b996e9b..5a9b4e8 100644 --- a/po/gl.po +++ b/po/gl.po @@ -8,7 +8,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.32.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2011-09-05 12:48+0200\n" "Last-Translator: Fran Dieguez \n" "Language-Team: Galician \n" @@ -191,10 +191,6 @@ msgstr "O elemento fonte é incorrecto." msgid "Error while sending data to \"%s:%d\"." msgstr "Produciuse un erro mentres se enviaban datos a \"%s:%d\"." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Rexeitouse a conexión a %s:%d." - msgid "This CD has no audio tracks" msgstr "Este CD non contén pistas de son" @@ -510,6 +506,9 @@ msgstr "O elemento codificador é descoñecido" msgid "Plugin or element of unknown type" msgstr "Engadido ou elemento de tipo descoñecido" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Rexeitouse a conexión a %s:%d." + #~ msgid "Could not open vfs file \"%s\" for writing: %s." #~ msgstr "Non foi posíbel abrir o ficheiro vfs «%s» para escribir: %s." diff --git a/po/hu.po b/po/hu.po index 0bfc5c7..df9bd27 100644 --- a/po/hu.po +++ b/po/hu.po @@ -8,7 +8,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.30.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2010-11-03 02:48+0100\n" "Last-Translator: Gabor Kelemen \n" "Language-Team: Hungarian \n" @@ -191,10 +191,6 @@ msgstr "A forráselem érvénytelen." msgid "Error while sending data to \"%s:%d\"." msgstr "Hiba adatok küldése során a következőnek: „%s:%d”." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "A kapcsolat visszautasítva a következőhöz: %s:%d." - msgid "This CD has no audio tracks" msgstr "Ez a CD nem rendelkezik hangsávokkal" @@ -511,6 +507,9 @@ msgstr "Ismeretlen kódolóelem" msgid "Plugin or element of unknown type" msgstr "Ismeretlen típusú bővítmény vagy elem" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "A kapcsolat visszautasítva a következőhöz: %s:%d." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "Tömörítetlen síkbeli YUV 4:2:0" diff --git a/po/id.po b/po/id.po index 4289cce..019b593 100644 --- a/po/id.po +++ b/po/id.po @@ -1,13 +1,13 @@ # Indonesian translations for gst-plugins-base package. # This file is put in the public domain. -# Andhika Padmawan , 2010. +# Andhika Padmawan , 2010-2012. # msgid "" msgstr "" -"Project-Id-Version: gst-plugins-base 0.10.28.2\n" +"Project-Id-Version: gst-plugins-base 0.10.32.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" -"PO-Revision-Date: 2010-04-26 22:01+0700\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" +"PO-Revision-Date: 2012-01-28 11:31+0700\n" "Last-Translator: Andhika Padmawan \n" "Language-Team: Indonesian \n" "Language: id\n" @@ -184,10 +184,6 @@ msgstr "Elemen sumber tidak sah." msgid "Error while sending data to \"%s:%d\"." msgstr "Galat ketika mengirim data ke \"%s:%d\"." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Koneksi ke %s:%d ditolak." - msgid "This CD has no audio tracks" msgstr "CD ini tidak memiliki jalur audio" @@ -228,125 +224,127 @@ msgid "MusicBrainz TRM ID" msgstr "ID TRM MusicBrainz" msgid "capturing shutter speed" -msgstr "" +msgstr "menangkap kecepatan rana" msgid "Shutter speed used when capturing an image, in seconds" -msgstr "" +msgstr "Kecepatan rana digunakan saat menangkap gambar, dalam detik" msgid "capturing focal ratio" -msgstr "" +msgstr "menangkap bukaan" msgid "Focal ratio (f-number) used when capturing the image" -msgstr "" +msgstr "Bukaan (angka-f) digunakan ketika menangkap gambar" msgid "capturing focal length" -msgstr "" +msgstr "menangkap panjang fokal" msgid "Focal length of the lens used capturing the image, in mm" -msgstr "" +msgstr "Panjang fokal dari lensa digunakan ketika menangkap gambar, dalam mm" msgid "capturing digital zoom ratio" -msgstr "" +msgstr "menangkap rasio pembesaran digital" msgid "Digital zoom ratio used when capturing an image" -msgstr "" +msgstr "Rasio pembesaran digital digunakan ketika menangkap gambar" msgid "capturing iso speed" -msgstr "" +msgstr "menangkap kecepatan iso" msgid "The ISO speed used when capturing an image" -msgstr "" +msgstr "Kecepatan ISO digunakan menangkap gambar" msgid "capturing exposure program" -msgstr "" +msgstr "menangkap program pajanan" msgid "The exposure program used when capturing an image" -msgstr "" +msgstr "Program pajanan digunakan ketika menangkap gambar" msgid "capturing exposure mode" -msgstr "" +msgstr "menangkap mode pajanan" msgid "The exposure mode used when capturing an image" -msgstr "" +msgstr "Mode pajanan digunakan ketika menangkap gambar" msgid "capturing exposure compensation" -msgstr "" +msgstr "menangkap kompensasi pajanan" msgid "The exposure compensation used when capturing an image" -msgstr "" +msgstr "Kompensasi pajanan digunakan ketika menangkap gambar" msgid "capturing scene capture type" -msgstr "" +msgstr "menangkap tipe pengambilan scene" msgid "The scene capture mode used when capturing an image" -msgstr "" +msgstr "Mode penangkapan scene digunakan ketika menangkap gambar" msgid "capturing gain adjustment" -msgstr "" +msgstr "menangkap penyesuaian bati" msgid "The overall gain adjustment applied on an image" -msgstr "" +msgstr "Keseluruhan penyesuaian bati diterapkan pada gambar" msgid "capturing white balance" -msgstr "" +msgstr "menangkap white balance" msgid "The white balance mode set when capturing an image" -msgstr "" +msgstr "Pengaturan mode white balance ketika menangkap gambar" msgid "capturing contrast" -msgstr "" +msgstr "menangkap kontras" msgid "The direction of contrast processing applied when capturing an image" -msgstr "" +msgstr "Arah pemrosesan kontras diterapkan ketika menangkap gambar" msgid "capturing saturation" -msgstr "" +msgstr "menangkap saturasi" msgid "The direction of saturation processing applied when capturing an image" -msgstr "" +msgstr "Arah pemrosesan saturasi diterapkan ketika menangkap gambar" msgid "capturing sharpness" -msgstr "" +msgstr "menangkap ketajaman" msgid "The direction of sharpness processing applied when capturing an image" -msgstr "" +msgstr "Arah pemrosesan ketajaman diterapkan ketika menangkap gambar" msgid "capturing flash fired" -msgstr "" +msgstr "menangkap blitz ditembakkan" msgid "If the flash fired while capturing an image" -msgstr "" +msgstr "Apakah blitz ditembakkan ketika menangkap gambar" msgid "capturing flash mode" -msgstr "" +msgstr "menangkap mode blitz" msgid "The selected flash mode while capturing an image" -msgstr "" +msgstr "Mode blitz terpilih ketika menangkap gambar" msgid "capturing metering mode" -msgstr "" +msgstr "menangkap mode meter" msgid "" "The metering mode used while determining exposure for capturing an image" -msgstr "" +msgstr "Mode meter digunakan untuk menentukan pajanan ketika menangkap gambar" msgid "capturing source" -msgstr "" +msgstr "menangkap sumber" msgid "The source or type of device used for the capture" -msgstr "" +msgstr "Sumber atau tipe divais digunakan untuk menangkap" msgid "image horizontal ppi" -msgstr "" +msgstr "ppi horizontal gambar" msgid "Media (image/video) intended horizontal pixel density in ppi" msgstr "" +"Kerapatan pixel horizontal yang diinginkan media (gambar/video) dalam ppi" msgid "image vertical ppi" -msgstr "" +msgstr "ppi vertikal gambar" msgid "Media (image/video) intended vertical pixel density in ppi" msgstr "" +"Kerapatan pixel vertikal yang diinginkan media (gambar/video) dalam ppi" msgid "ID3v2 frame" msgstr "" @@ -556,6 +554,9 @@ msgstr "Plugin atau elemen dari tipe yang tak diketahui" #~ msgid "Error while sending gdp payload data to \"%s:%d\"." #~ msgstr "Galat ketika mengirim data pemuat gdp ke \"%s:%d\"." +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Koneksi ke %s:%d ditolak." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "YUV 4:2:0 planar tak dikompresi" diff --git a/po/it.po b/po/it.po index 7028588..0fb9a05 100644 --- a/po/it.po +++ b/po/it.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.28.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2010-04-28 14:27+0200\n" "Last-Translator: Luca Ferretti \n" "Language-Team: Italian \n" @@ -192,10 +192,6 @@ msgstr "L'elemento sorgente non è valido." msgid "Error while sending data to \"%s:%d\"." msgstr "Errore durante l'invio dei dati a \"%s:%d\"." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Connessione a %s:%d rifiutata." - msgid "This CD has no audio tracks" msgstr "Questo CD non presenta alcuna traccia audio" @@ -521,6 +517,9 @@ msgstr "Elemento di codifica sconosciuto" msgid "Plugin or element of unknown type" msgstr "Plugin o elemento di tipo sconosciuto" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Connessione a %s:%d rifiutata." + # cfr http://en.wikipedia.org/wiki/YUV # http://support.microsoft.com/kb/281188/it # http://support.microsoft.com/kb/294880/it (traduz automatica) :-( diff --git a/po/ja.po b/po/ja.po index 7086bd7..e53b55a 100644 --- a/po/ja.po +++ b/po/ja.po @@ -6,7 +6,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.30.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2010-10-25 10:27+0900\n" "Last-Translator: Makoto Kato \n" "Language-Team: Japanese \n" @@ -190,10 +190,6 @@ msgstr "ソースエレメントが不正です。" msgid "Error while sending data to \"%s:%d\"." msgstr "データを \"%s:%d\" へ送信中にエラーが発生しました" -#, c-format -msgid "Connection to %s:%d refused." -msgstr "%s:%d への接続が拒否されました" - msgid "This CD has no audio tracks" msgstr "この CD にはオーディオトラックがありません" @@ -510,6 +506,9 @@ msgstr "不明なエンコーダーエレメント" msgid "Plugin or element of unknown type" msgstr "不明な種類のプラグインまたはエレメント" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "%s:%d への接続が拒否されました" + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "非圧縮 planar YUV 4:2:0" diff --git a/po/lt.po b/po/lt.po index c0e8393..a2d7384 100644 --- a/po/lt.po +++ b/po/lt.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base-0.10.15.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2008-03-07 23:43+0200\n" "Last-Translator: Gintautas Miliauskas \n" "Language-Team: Lithuanian \n" @@ -193,10 +193,6 @@ msgstr "Šaltinio elementas nekorektiškas." msgid "Error while sending data to \"%s:%d\"." msgstr "Klaida siunčiant duomenis į „%s:%d“." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Prisijungimas prie %s:%d atmestas." - msgid "This CD has no audio tracks" msgstr "Šiame CD nėra audio takelių" @@ -512,6 +508,9 @@ msgstr "Nežinomas kodavimo elementas" msgid "Plugin or element of unknown type" msgstr "Nežinomo tipo įskiepis ar elementas" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Prisijungimas prie %s:%d atmestas." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "Nekompresuotas planarinis YUV 4:2:0" diff --git a/po/lv.po b/po/lv.po index 7dccaf8..bc7fa07 100644 --- a/po/lv.po +++ b/po/lv.po @@ -8,7 +8,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.32.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2011-09-02 12:02-0000\n" "Last-Translator: Rihards Priedītis \n" "Language-Team: Latvian \n" @@ -192,10 +192,6 @@ msgstr "Avota elements ir nederīgs." msgid "Error while sending data to \"%s:%d\"." msgstr "Radās kļūda nosūtot datus uz \"%s:%d\"." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Savienojums ar %s:%d noraidīts." - msgid "This CD has no audio tracks" msgstr "Šajā CD nav neviena audio celiņa" @@ -510,6 +506,9 @@ msgstr "Nezināms kodētāja elements" msgid "Plugin or element of unknown type" msgstr "Nezināma veida spraudnis vai elements" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Savienojums ar %s:%d noraidīts." + #~ msgid "Could not open vfs file \"%s\" for writing: %s." #~ msgstr "Nevarēja atvērt vfs failu \"%s\" rakstīšanai: %s." diff --git a/po/nb.po b/po/nb.po index ddcc96c..85cc444 100644 --- a/po/nb.po +++ b/po/nb.po @@ -6,7 +6,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.30.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2010-10-24 21:44+0200\n" "Last-Translator: Kjartan Maraas \n" "Language-Team: Norwegian Bokmaal \n" @@ -187,10 +187,6 @@ msgstr "Kildeelement er ugyldig." msgid "Error while sending data to \"%s:%d\"." msgstr "" -#, c-format -msgid "Connection to %s:%d refused." -msgstr "" - msgid "This CD has no audio tracks" msgstr "Denne CDen har ingen lydspor" diff --git a/po/nl.po b/po/nl.po index 7b6eed6..31d8afd 100644 --- a/po/nl.po +++ b/po/nl.po @@ -6,7 +6,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.32.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2011-04-27 00:12+0200\n" "Last-Translator: Freek de Kruijf \n" "Language-Team: Dutch \n" @@ -188,10 +188,6 @@ msgstr "Bronelement is ongeldig." msgid "Error while sending data to \"%s:%d\"." msgstr "Fout bij het zenden van gegevens naar \"%s:%d\"." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Verbinding naar %s:%d is geweigerd." - msgid "This CD has no audio tracks" msgstr "Deze CD heeft geen audiotracks" @@ -515,6 +511,9 @@ msgstr "Onbekend encoder-element" msgid "Plugin or element of unknown type" msgstr "Plugin of element van onbekend type" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Verbinding naar %s:%d is geweigerd." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "Niet-gecomprimeerde planar YUV 4:2:0" diff --git a/po/or.po b/po/or.po index 8905288..b641a61 100644 --- a/po/or.po +++ b/po/or.po @@ -8,7 +8,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-0.8.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2004-09-27 13:32+0530\n" "Last-Translator: Gora Mohanty \n" "Language-Team: Oriya \n" @@ -193,10 +193,6 @@ msgstr "" msgid "Error while sending data to \"%s:%d\"." msgstr "\"%s\" ଫାଇଲ ବନ୍ଦ କରିବାରେ ତ୍ରୁଟି." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "" - msgid "This CD has no audio tracks" msgstr "" diff --git a/po/pl.po b/po/pl.po index 065b5d4..bdd13f2 100644 --- a/po/pl.po +++ b/po/pl.po @@ -6,7 +6,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.32.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2011-04-26 18:01+0200\n" "Last-Translator: Jakub Bogusz \n" "Language-Team: Polish \n" @@ -194,10 +194,6 @@ msgstr "Element źródłowy jest niepoprawny." msgid "Error while sending data to \"%s:%d\"." msgstr "Błąd podczas wysyłania danych do \"%s:%d\"." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Połączenie z %s:%d odrzucone." - msgid "This CD has no audio tracks" msgstr "Ta płyta CD nie ma ścieżek dźwiękowych" @@ -512,6 +508,9 @@ msgstr "Nieznany element kodujący" msgid "Plugin or element of unknown type" msgstr "Wtyczka lub element nieznanego typu" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Połączenie z %s:%d odrzucone." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "Nieskompresowany planarny YUV 4:2:0" diff --git a/po/pt_BR.po b/po/pt_BR.po index 25f1408..a36e2dc 100644 --- a/po/pt_BR.po +++ b/po/pt_BR.po @@ -9,7 +9,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base-0.10.31.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2011-01-08 01:02-0300\n" "Last-Translator: Fabrício Godoy \n" "Language-Team: Brazilian Portuguese \n" @@ -191,10 +191,6 @@ msgstr "O elemente de origem é inválido." msgid "Error while sending data to \"%s:%d\"." msgstr "Erro ao enviar dados para \"%s:%d\"." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "A conexão para %s:%d foi recusada." - msgid "This CD has no audio tracks" msgstr "Este CD não tem trilhas de áudio" @@ -518,6 +514,9 @@ msgstr "Elemento codificador desconhecido" msgid "Plugin or element of unknown type" msgstr "Elemento ou plug-in de tipo desconhecido" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "A conexão para %s:%d foi recusada." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "YUV 4:2:0 plano sem compressão" diff --git a/po/ro.po b/po/ro.po index bed3947..f417ebe 100644 --- a/po/ro.po +++ b/po/ro.po @@ -5,7 +5,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.29.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2010-08-16 01:21+0300\n" "Last-Translator: Lucian Adrian Grijincu \n" "Language-Team: Romanian \n" @@ -189,10 +189,6 @@ msgstr "Element sursă nevalid." msgid "Error while sending data to \"%s:%d\"." msgstr "Eroare la trimiterea datelor către „%s:%d”." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Conexiunea la %s:%d a fost refuzată." - msgid "This CD has no audio tracks" msgstr "Acest CD nu conține piste audio" @@ -507,6 +503,9 @@ msgstr "Element codor necunoscut" msgid "Plugin or element of unknown type" msgstr "Modul de extensie sau element de tip necunoscut" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Conexiunea la %s:%d a fost refuzată." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "Planar necomprimat YUV 4:2:0" diff --git a/po/ru.po b/po/ru.po index c8e54f9..dba124c 100644 --- a/po/ru.po +++ b/po/ru.po @@ -8,7 +8,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.32.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2011-04-26 20:30+0400\n" "Last-Translator: Yuri Kozlov \n" "Language-Team: Russian \n" @@ -192,10 +192,6 @@ msgstr "Неверный элемент источника." msgid "Error while sending data to \"%s:%d\"." msgstr "Ошибка отправки данных в «%s:%d»." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "В соединении с %s:%d отказано." - msgid "This CD has no audio tracks" msgstr "На CD нет звуковых дорожек" @@ -512,6 +508,9 @@ msgstr "Неизвестный элемент-кодировщик" msgid "Plugin or element of unknown type" msgstr "Модуль или элемент неизвестного типа" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "В соединении с %s:%d отказано." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "Несжатый планарный YUV 4:2:0" diff --git a/po/sk.po b/po/sk.po index d8ac415..93b9d61 100644 --- a/po/sk.po +++ b/po/sk.po @@ -6,7 +6,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.30.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2010-11-08 15:34+0100\n" "Last-Translator: Peter Tuhársky \n" "Language-Team: Slovak \n" @@ -191,10 +191,6 @@ msgstr "Zdrojový prvok je chybný." msgid "Error while sending data to \"%s:%d\"." msgstr "Chyba pri posielaní údajov do \"%s:%d\"." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Spojenie s %s:%d bolo odmietnuté." - msgid "This CD has no audio tracks" msgstr "Toto CD nemá zvukové stopy" @@ -511,6 +507,9 @@ msgstr "Neznámy prvok enkodéra" msgid "Plugin or element of unknown type" msgstr "Neznámy typ zásuvného modulu alebo prvku" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Spojenie s %s:%d bolo odmietnuté." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "Nekomprimovaný planar YUV 4:2:0" diff --git a/po/sl.po b/po/sl.po index 60e554f..f456de1 100644 --- a/po/sl.po +++ b/po/sl.po @@ -8,7 +8,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.32.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2011-04-26 15:56+0100\n" "Last-Translator: Klemen Košir \n" "Language-Team: Slovenian \n" @@ -190,10 +190,6 @@ msgstr "Izvorni predmet je neveljaven." msgid "Error while sending data to \"%s:%d\"." msgstr "Napaka med pošiljanjem podatkov na \"%s:%d\"." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Povezava s %s:%d zavrnjena." - msgid "This CD has no audio tracks" msgstr "Ta CD nima zvočnih sledi" @@ -509,6 +505,9 @@ msgstr "Neznan predmet kodirnika" msgid "Plugin or element of unknown type" msgstr "Vstavek ali predmet neznane vrste" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Povezava s %s:%d zavrnjena." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "Nestisnjen ploskovni YUV 4:2:0" diff --git a/po/sq.po b/po/sq.po index a648366..b72709a 100644 --- a/po/sq.po +++ b/po/sq.po @@ -6,7 +6,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins 0.8.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2004-08-07 20:29+0200\n" "Last-Translator: Laurent Dhima \n" "Language-Team: Albanian \n" @@ -191,10 +191,6 @@ msgstr "" msgid "Error while sending data to \"%s:%d\"." msgstr "Gabim gjatë mbylljes së file \"%s\"." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "" - msgid "This CD has no audio tracks" msgstr "" diff --git a/po/sr.po b/po/sr.po index dca4cc5..53e5b12 100644 --- a/po/sr.po +++ b/po/sr.po @@ -7,7 +7,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base-0.10.32.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-12-12 12:57+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2011-12-05 09:10+0200\n" "Last-Translator: Мирослав Николић \n" "Language-Team: Serbian \n" @@ -189,10 +189,6 @@ msgstr "Изворни елемент је неисправан." msgid "Error while sending data to \"%s:%d\"." msgstr "Грешка приликом слања података у „%s:%d“." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Повезивање са %s:%d је одбијено." - msgid "This CD has no audio tracks" msgstr "Овај ЦД нема звучних нумера" @@ -507,6 +503,9 @@ msgstr "Непознати елемент кодера" msgid "Plugin or element of unknown type" msgstr "Прикључак или елемент непознате врсте" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Повезивање са %s:%d је одбијено." + #~ msgid "Could not open vfs file \"%s\" for writing: %s." #~ msgstr "Не могу да отворим всд датотеку „%s“ ради уписа: %s." diff --git a/po/sv.po b/po/sv.po index f40548b..912c7f1 100644 --- a/po/sv.po +++ b/po/sv.po @@ -8,7 +8,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.28.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2010-06-07 18:17+0100\n" "Last-Translator: Daniel Nylander \n" "Language-Team: Swedish \n" @@ -188,10 +188,6 @@ msgstr "Källelementet är ogiltigt." msgid "Error while sending data to \"%s:%d\"." msgstr "Fel vid sändning av data till \"%s:%d\"." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Anslutningen till %s:%d nekades." - msgid "This CD has no audio tracks" msgstr "Den här cd-skivan saknar ljudspår" @@ -506,6 +502,9 @@ msgstr "Okänt kodarelement" msgid "Plugin or element of unknown type" msgstr "Insticksmodul eller element av okänd typ" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Anslutningen till %s:%d nekades." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "Okomprimerad plan YUV 4:2:0" diff --git a/po/tr.po b/po/tr.po index 556bf7f..2936ec9 100644 --- a/po/tr.po +++ b/po/tr.po @@ -5,7 +5,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.32.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2011-04-26 19:22+0200\n" "Last-Translator: Server Acim \n" "Language-Team: Turkish \n" @@ -183,10 +183,6 @@ msgstr "Kaynak öğesi geçersiz." msgid "Error while sending data to \"%s:%d\"." msgstr "Dosyayı şuraya gönderirken hata \"%s:%d\"." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Şuraya bağlantı %s:%d reddedildi." - msgid "This CD has no audio tracks" msgstr "Bu CD hiç ses izi içermiyor" @@ -555,6 +551,9 @@ msgstr "Bilinmeyen türde eklenti veya öğe" #~ msgid "Error while sending gdp payload data to \"%s:%d\"." #~ msgstr "Bir gdp verisini şuraya gönderirken hata \"%s:%d\"." +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Şuraya bağlantı %s:%d reddedildi." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "Sıkıştırılmamış planar YUV 4:2:0" diff --git a/po/uk.po b/po/uk.po index 2200a7d..3c83317 100644 --- a/po/uk.po +++ b/po/uk.po @@ -8,7 +8,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.32.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2011-04-27 17:49+0300\n" "Last-Translator: Yuri Chornoivan \n" "Language-Team: Ukrainian \n" @@ -190,10 +190,6 @@ msgstr "Неправильний вхідний елемент." msgid "Error while sending data to \"%s:%d\"." msgstr "Помилка при надсиланні даних до \"%s:%d\"." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "У з'єднанні з %s:%d відмовлено." - msgid "This CD has no audio tracks" msgstr "На цьому компакт-диску немає звукових доріжок" @@ -519,6 +515,9 @@ msgstr "Невідомий елемент кодера" msgid "Plugin or element of unknown type" msgstr "Модуль або елемент невідомого типу" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "У з'єднанні з %s:%d відмовлено." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "Нестиснений планарний YUV 4:2:0" diff --git a/po/vi.po b/po/vi.po index 810065f..7235107 100644 --- a/po/vi.po +++ b/po/vi.po @@ -6,7 +6,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.28.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2010-04-27 22:51+0930\n" "Last-Translator: Clytie Siddall \n" "Language-Team: Vietnamese \n" @@ -185,10 +185,6 @@ msgstr "Phần tử nguồn không phải hợp lệ." msgid "Error while sending data to \"%s:%d\"." msgstr "Lỗi khi gởi dữ liệu cho « %s:%d »." -#, c-format -msgid "Connection to %s:%d refused." -msgstr "Kết nối tới « %s:%d » bị từ chối." - msgid "This CD has no audio tracks" msgstr "Đĩa CD này không có rãnh âm thanh nào" @@ -508,6 +504,9 @@ msgstr "Không rõ phần tử mã hoá" msgid "Plugin or element of unknown type" msgstr "Không rõ kiểu phần bổ sung hay phần tử" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "Kết nối tới « %s:%d » bị từ chối." + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "YUV 4:2:0 phẳng không nén" diff --git a/po/zh_CN.po b/po/zh_CN.po index e33a97c..fec336b 100644 --- a/po/zh_CN.po +++ b/po/zh_CN.po @@ -6,7 +6,7 @@ msgid "" msgstr "" "Project-Id-Version: gst-plugins-base 0.10.21.3\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-11-12 00:54+0000\n" +"POT-Creation-Date: 2012-02-17 09:59+0100\n" "PO-Revision-Date: 2009-01-14 12:41+0800\n" "Last-Translator: Ji ZhengYu \n" "Language-Team: Chinese (simplified) \n" @@ -186,10 +186,6 @@ msgstr "无效的源组件。" msgid "Error while sending data to \"%s:%d\"." msgstr "发送数据至“%s:%d”时出错。" -#, c-format -msgid "Connection to %s:%d refused." -msgstr "拒绝连接至 %s:%d。" - msgid "This CD has no audio tracks" msgstr "此 CD 无音轨" @@ -563,6 +559,9 @@ msgstr "未知类型的插件或组件" #~ msgid "Error while sending gdp payload data to \"%s:%d\"." #~ msgstr "发送 gdp 负载数据至“%s:%d”时出错。" +#~ msgid "Connection to %s:%d refused." +#~ msgstr "拒绝连接至 %s:%d。" + #~ msgid "Uncompressed planar YUV 4:2:0" #~ msgstr "未压缩的 planar YUV 4:2:0" diff --git a/win32/common/_stdint.h b/win32/common/_stdint.h index 54baf84..d7f28c8 100644 --- a/win32/common/_stdint.h +++ b/win32/common/_stdint.h @@ -1,8 +1,8 @@ #ifndef _GST_PLUGINS_BASE__STDINT_H #define _GST_PLUGINS_BASE__STDINT_H 1 #ifndef _GENERATED_STDINT_H -#define _GENERATED_STDINT_H "gst-plugins-base 0.11.1" -/* generated using gnu compiler gcc (Ubuntu/Linaro 4.5.2-8ubuntu4) 4.5.2 */ +#define _GENERATED_STDINT_H "gst-plugins-base 0.11.2" +/* generated using gnu compiler gcc (Ubuntu/Linaro 4.6.1-9ubuntu3) 4.6.1 */ #define _STDINT_HAVE_STDINT_H 1 #include #endif diff --git a/win32/common/audio-enumtypes.c b/win32/common/audio-enumtypes.c index 84b8a80..8e28beb 100644 --- a/win32/common/audio-enumtypes.c +++ b/win32/common/audio-enumtypes.c @@ -3,60 +3,163 @@ #include "audio-enumtypes.h" -#include "multichannel.h" -#include "gstringbuffer.h" +#include "audio.h" +#include "gstaudioringbuffer.h" + +/* enumerations from "audio.h" */ +GType +gst_audio_format_get_type (void) +{ + static volatile gsize g_define_type_id__volatile = 0; + if (g_once_init_enter (&g_define_type_id__volatile)) { + static const GEnumValue values[] = { + {GST_AUDIO_FORMAT_UNKNOWN, "GST_AUDIO_FORMAT_UNKNOWN", "unknown"}, + {GST_AUDIO_FORMAT_S8, "GST_AUDIO_FORMAT_S8", "s8"}, + {GST_AUDIO_FORMAT_U8, "GST_AUDIO_FORMAT_U8", "u8"}, + {GST_AUDIO_FORMAT_S16LE, "GST_AUDIO_FORMAT_S16LE", "s16le"}, + {GST_AUDIO_FORMAT_S16BE, "GST_AUDIO_FORMAT_S16BE", "s16be"}, + {GST_AUDIO_FORMAT_U16LE, "GST_AUDIO_FORMAT_U16LE", "u16le"}, + {GST_AUDIO_FORMAT_U16BE, "GST_AUDIO_FORMAT_U16BE", "u16be"}, + {GST_AUDIO_FORMAT_S24_32LE, "GST_AUDIO_FORMAT_S24_32LE", "s24-32le"}, + {GST_AUDIO_FORMAT_S24_32BE, "GST_AUDIO_FORMAT_S24_32BE", "s24-32be"}, + {GST_AUDIO_FORMAT_U24_32LE, "GST_AUDIO_FORMAT_U24_32LE", "u24-32le"}, + {GST_AUDIO_FORMAT_U24_32BE, "GST_AUDIO_FORMAT_U24_32BE", "u24-32be"}, + {GST_AUDIO_FORMAT_S32LE, "GST_AUDIO_FORMAT_S32LE", "s32le"}, + {GST_AUDIO_FORMAT_S32BE, "GST_AUDIO_FORMAT_S32BE", "s32be"}, + {GST_AUDIO_FORMAT_U32LE, "GST_AUDIO_FORMAT_U32LE", "u32le"}, + {GST_AUDIO_FORMAT_U32BE, "GST_AUDIO_FORMAT_U32BE", "u32be"}, + {GST_AUDIO_FORMAT_S24LE, "GST_AUDIO_FORMAT_S24LE", "s24le"}, + {GST_AUDIO_FORMAT_S24BE, "GST_AUDIO_FORMAT_S24BE", "s24be"}, + {GST_AUDIO_FORMAT_U24LE, "GST_AUDIO_FORMAT_U24LE", "u24le"}, + {GST_AUDIO_FORMAT_U24BE, "GST_AUDIO_FORMAT_U24BE", "u24be"}, + {GST_AUDIO_FORMAT_S20LE, "GST_AUDIO_FORMAT_S20LE", "s20le"}, + {GST_AUDIO_FORMAT_S20BE, "GST_AUDIO_FORMAT_S20BE", "s20be"}, + {GST_AUDIO_FORMAT_U20LE, "GST_AUDIO_FORMAT_U20LE", "u20le"}, + {GST_AUDIO_FORMAT_U20BE, "GST_AUDIO_FORMAT_U20BE", "u20be"}, + {GST_AUDIO_FORMAT_S18LE, "GST_AUDIO_FORMAT_S18LE", "s18le"}, + {GST_AUDIO_FORMAT_S18BE, "GST_AUDIO_FORMAT_S18BE", "s18be"}, + {GST_AUDIO_FORMAT_U18LE, "GST_AUDIO_FORMAT_U18LE", "u18le"}, + {GST_AUDIO_FORMAT_U18BE, "GST_AUDIO_FORMAT_U18BE", "u18be"}, + {GST_AUDIO_FORMAT_F32LE, "GST_AUDIO_FORMAT_F32LE", "f32le"}, + {GST_AUDIO_FORMAT_F32BE, "GST_AUDIO_FORMAT_F32BE", "f32be"}, + {GST_AUDIO_FORMAT_F64LE, "GST_AUDIO_FORMAT_F64LE", "f64le"}, + {GST_AUDIO_FORMAT_F64BE, "GST_AUDIO_FORMAT_F64BE", "f64be"}, + {GST_AUDIO_FORMAT_S16, "GST_AUDIO_FORMAT_S16", "s16"}, + {GST_AUDIO_FORMAT_U16, "GST_AUDIO_FORMAT_U16", "u16"}, + {GST_AUDIO_FORMAT_S24_32, "GST_AUDIO_FORMAT_S24_32", "s24-32"}, + {GST_AUDIO_FORMAT_U24_32, "GST_AUDIO_FORMAT_U24_32", "u24-32"}, + {GST_AUDIO_FORMAT_S32, "GST_AUDIO_FORMAT_S32", "s32"}, + {GST_AUDIO_FORMAT_U32, "GST_AUDIO_FORMAT_U32", "u32"}, + {GST_AUDIO_FORMAT_S24, "GST_AUDIO_FORMAT_S24", "s24"}, + {GST_AUDIO_FORMAT_U24, "GST_AUDIO_FORMAT_U24", "u24"}, + {GST_AUDIO_FORMAT_S20, "GST_AUDIO_FORMAT_S20", "s20"}, + {GST_AUDIO_FORMAT_U20, "GST_AUDIO_FORMAT_U20", "u20"}, + {GST_AUDIO_FORMAT_S18, "GST_AUDIO_FORMAT_S18", "s18"}, + {GST_AUDIO_FORMAT_U18, "GST_AUDIO_FORMAT_U18", "u18"}, + {GST_AUDIO_FORMAT_F32, "GST_AUDIO_FORMAT_F32", "f32"}, + {GST_AUDIO_FORMAT_F64, "GST_AUDIO_FORMAT_F64", "f64"}, + {0, NULL, NULL} + }; + GType g_define_type_id = g_enum_register_static ("GstAudioFormat", values); + g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); + } + return g_define_type_id__volatile; +} + +GType +gst_audio_format_flags_get_type (void) +{ + static volatile gsize g_define_type_id__volatile = 0; + if (g_once_init_enter (&g_define_type_id__volatile)) { + static const GFlagsValue values[] = { + {GST_AUDIO_FORMAT_FLAG_INTEGER, "GST_AUDIO_FORMAT_FLAG_INTEGER", + "integer"}, + {GST_AUDIO_FORMAT_FLAG_FLOAT, "GST_AUDIO_FORMAT_FLAG_FLOAT", "float"}, + {GST_AUDIO_FORMAT_FLAG_SIGNED, "GST_AUDIO_FORMAT_FLAG_SIGNED", "signed"}, + {GST_AUDIO_FORMAT_FLAG_COMPLEX, "GST_AUDIO_FORMAT_FLAG_COMPLEX", + "complex"}, + {0, NULL, NULL} + }; + GType g_define_type_id = + g_flags_register_static ("GstAudioFormatFlags", values); + g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); + } + return g_define_type_id__volatile; +} -/* enumerations from "multichannel.h" */ GType gst_audio_channel_position_get_type (void) { static volatile gsize g_define_type_id__volatile = 0; if (g_once_init_enter (&g_define_type_id__volatile)) { static const GEnumValue values[] = { + {GST_AUDIO_CHANNEL_POSITION_NONE, "GST_AUDIO_CHANNEL_POSITION_NONE", + "none"}, + {GST_AUDIO_CHANNEL_POSITION_MONO, "GST_AUDIO_CHANNEL_POSITION_MONO", + "mono"}, {GST_AUDIO_CHANNEL_POSITION_INVALID, "GST_AUDIO_CHANNEL_POSITION_INVALID", "invalid"}, - {GST_AUDIO_CHANNEL_POSITION_FRONT_MONO, - "GST_AUDIO_CHANNEL_POSITION_FRONT_MONO", "front-mono"}, {GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, "GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT", "front-left"}, {GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT, "GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT", "front-right"}, - {GST_AUDIO_CHANNEL_POSITION_REAR_CENTER, - "GST_AUDIO_CHANNEL_POSITION_REAR_CENTER", "rear-center"}, + {GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, + "GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER", "front-center"}, + {GST_AUDIO_CHANNEL_POSITION_LFE1, "GST_AUDIO_CHANNEL_POSITION_LFE1", + "lfe1"}, {GST_AUDIO_CHANNEL_POSITION_REAR_LEFT, "GST_AUDIO_CHANNEL_POSITION_REAR_LEFT", "rear-left"}, {GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT, "GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT", "rear-right"}, - {GST_AUDIO_CHANNEL_POSITION_LFE, "GST_AUDIO_CHANNEL_POSITION_LFE", "lfe"}, - {GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, - "GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER", "front-center"}, {GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER, "GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT_OF_CENTER", "front-left-of-center"}, {GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER, "GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT_OF_CENTER", "front-right-of-center"}, + {GST_AUDIO_CHANNEL_POSITION_REAR_CENTER, + "GST_AUDIO_CHANNEL_POSITION_REAR_CENTER", "rear-center"}, + {GST_AUDIO_CHANNEL_POSITION_LFE2, "GST_AUDIO_CHANNEL_POSITION_LFE2", + "lfe2"}, {GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT, "GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT", "side-left"}, {GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT, "GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT", "side-right"}, - {GST_AUDIO_CHANNEL_POSITION_TOP_CENTER, - "GST_AUDIO_CHANNEL_POSITION_TOP_CENTER", "top-center"}, {GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT, "GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_LEFT", "top-front-left"}, {GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT, "GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_RIGHT", "top-front-right"}, {GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER, "GST_AUDIO_CHANNEL_POSITION_TOP_FRONT_CENTER", "top-front-center"}, + {GST_AUDIO_CHANNEL_POSITION_TOP_CENTER, + "GST_AUDIO_CHANNEL_POSITION_TOP_CENTER", "top-center"}, {GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT, "GST_AUDIO_CHANNEL_POSITION_TOP_REAR_LEFT", "top-rear-left"}, {GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT, "GST_AUDIO_CHANNEL_POSITION_TOP_REAR_RIGHT", "top-rear-right"}, + {GST_AUDIO_CHANNEL_POSITION_TOP_SIDE_LEFT, + "GST_AUDIO_CHANNEL_POSITION_TOP_SIDE_LEFT", "top-side-left"}, + {GST_AUDIO_CHANNEL_POSITION_TOP_SIDE_RIGHT, + "GST_AUDIO_CHANNEL_POSITION_TOP_SIDE_RIGHT", "top-side-right"}, {GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER, "GST_AUDIO_CHANNEL_POSITION_TOP_REAR_CENTER", "top-rear-center"}, - {GST_AUDIO_CHANNEL_POSITION_NONE, "GST_AUDIO_CHANNEL_POSITION_NONE", - "none"}, - {GST_AUDIO_CHANNEL_POSITION_NUM, "GST_AUDIO_CHANNEL_POSITION_NUM", "num"}, + {GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_CENTER, + "GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_CENTER", + "bottom-front-center"}, + {GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_LEFT, + "GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_LEFT", + "bottom-front-left"}, + {GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_RIGHT, + "GST_AUDIO_CHANNEL_POSITION_BOTTOM_FRONT_RIGHT", + "bottom-front-right"}, + {GST_AUDIO_CHANNEL_POSITION_WIDE_LEFT, + "GST_AUDIO_CHANNEL_POSITION_WIDE_LEFT", "wide-left"}, + {GST_AUDIO_CHANNEL_POSITION_WIDE_RIGHT, + "GST_AUDIO_CHANNEL_POSITION_WIDE_RIGHT", "wide-right"}, + {GST_AUDIO_CHANNEL_POSITION_SURROUND_LEFT, + "GST_AUDIO_CHANNEL_POSITION_SURROUND_LEFT", "surround-left"}, + {GST_AUDIO_CHANNEL_POSITION_SURROUND_RIGHT, + "GST_AUDIO_CHANNEL_POSITION_SURROUND_RIGHT", "surround-right"}, {0, NULL, NULL} }; GType g_define_type_id = @@ -66,68 +169,97 @@ gst_audio_channel_position_get_type (void) return g_define_type_id__volatile; } -/* enumerations from "gstringbuffer.h" */ GType -gst_ring_buffer_state_get_type (void) +gst_audio_flags_get_type (void) +{ + static volatile gsize g_define_type_id__volatile = 0; + if (g_once_init_enter (&g_define_type_id__volatile)) { + static const GFlagsValue values[] = { + {GST_AUDIO_FLAG_NONE, "GST_AUDIO_FLAG_NONE", "none"}, + {GST_AUDIO_FLAG_UNPOSITIONED, "GST_AUDIO_FLAG_UNPOSITIONED", + "unpositioned"}, + {0, NULL, NULL} + }; + GType g_define_type_id = g_flags_register_static ("GstAudioFlags", values); + g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); + } + return g_define_type_id__volatile; +} + +GType +gst_audio_layout_get_type (void) { static volatile gsize g_define_type_id__volatile = 0; if (g_once_init_enter (&g_define_type_id__volatile)) { static const GEnumValue values[] = { - {GST_RING_BUFFER_STATE_STOPPED, "GST_RING_BUFFER_STATE_STOPPED", - "stopped"}, - {GST_RING_BUFFER_STATE_PAUSED, "GST_RING_BUFFER_STATE_PAUSED", "paused"}, - {GST_RING_BUFFER_STATE_STARTED, "GST_RING_BUFFER_STATE_STARTED", - "started"}, + {GST_AUDIO_LAYOUT_INTERLEAVED, "GST_AUDIO_LAYOUT_INTERLEAVED", + "interleaved"}, + {GST_AUDIO_LAYOUT_NON_INTERLEAVED, "GST_AUDIO_LAYOUT_NON_INTERLEAVED", + "non-interleaved"}, {0, NULL, NULL} }; - GType g_define_type_id = - g_enum_register_static ("GstRingBufferState", values); + GType g_define_type_id = g_enum_register_static ("GstAudioLayout", values); g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); } return g_define_type_id__volatile; } +/* enumerations from "gstaudioringbuffer.h" */ GType -gst_ring_buffer_seg_state_get_type (void) +gst_audio_ring_buffer_state_get_type (void) { static volatile gsize g_define_type_id__volatile = 0; if (g_once_init_enter (&g_define_type_id__volatile)) { static const GEnumValue values[] = { - {GST_SEGSTATE_INVALID, "GST_SEGSTATE_INVALID", "invalid"}, - {GST_SEGSTATE_EMPTY, "GST_SEGSTATE_EMPTY", "empty"}, - {GST_SEGSTATE_FILLED, "GST_SEGSTATE_FILLED", "filled"}, - {GST_SEGSTATE_PARTIAL, "GST_SEGSTATE_PARTIAL", "partial"}, + {GST_AUDIO_RING_BUFFER_STATE_STOPPED, + "GST_AUDIO_RING_BUFFER_STATE_STOPPED", "stopped"}, + {GST_AUDIO_RING_BUFFER_STATE_PAUSED, "GST_AUDIO_RING_BUFFER_STATE_PAUSED", + "paused"}, + {GST_AUDIO_RING_BUFFER_STATE_STARTED, + "GST_AUDIO_RING_BUFFER_STATE_STARTED", "started"}, {0, NULL, NULL} }; GType g_define_type_id = - g_enum_register_static ("GstRingBufferSegState", values); + g_enum_register_static ("GstAudioRingBufferState", values); g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); } return g_define_type_id__volatile; } GType -gst_buffer_format_type_get_type (void) +gst_audio_ring_buffer_format_type_get_type (void) { static volatile gsize g_define_type_id__volatile = 0; if (g_once_init_enter (&g_define_type_id__volatile)) { static const GEnumValue values[] = { - {GST_BUFTYPE_RAW, "GST_BUFTYPE_RAW", "raw"}, - {GST_BUFTYPE_MU_LAW, "GST_BUFTYPE_MU_LAW", "mu-law"}, - {GST_BUFTYPE_A_LAW, "GST_BUFTYPE_A_LAW", "a-law"}, - {GST_BUFTYPE_IMA_ADPCM, "GST_BUFTYPE_IMA_ADPCM", "ima-adpcm"}, - {GST_BUFTYPE_MPEG, "GST_BUFTYPE_MPEG", "mpeg"}, - {GST_BUFTYPE_GSM, "GST_BUFTYPE_GSM", "gsm"}, - {GST_BUFTYPE_IEC958, "GST_BUFTYPE_IEC958", "iec958"}, - {GST_BUFTYPE_AC3, "GST_BUFTYPE_AC3", "ac3"}, - {GST_BUFTYPE_EAC3, "GST_BUFTYPE_EAC3", "eac3"}, - {GST_BUFTYPE_DTS, "GST_BUFTYPE_DTS", "dts"}, - {GST_BUFTYPE_MPEG2_AAC, "GST_BUFTYPE_MPEG2_AAC", "mpeg2-aac"}, - {GST_BUFTYPE_MPEG4_AAC, "GST_BUFTYPE_MPEG4_AAC", "mpeg4-aac"}, + {GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW, + "GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW", "raw"}, + {GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW, + "GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW", "mu-law"}, + {GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW, + "GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW", "a-law"}, + {GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM, + "GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IMA_ADPCM", "ima-adpcm"}, + {GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG, + "GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG", "mpeg"}, + {GST_AUDIO_RING_BUFFER_FORMAT_TYPE_GSM, + "GST_AUDIO_RING_BUFFER_FORMAT_TYPE_GSM", "gsm"}, + {GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958, + "GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958", "iec958"}, + {GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3, + "GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3", "ac3"}, + {GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3, + "GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3", "eac3"}, + {GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS, + "GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS", "dts"}, + {GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC, + "GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG2_AAC", "mpeg2-aac"}, + {GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC, + "GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG4_AAC", "mpeg4-aac"}, {0, NULL, NULL} }; GType g_define_type_id = - g_enum_register_static ("GstBufferFormatType", values); + g_enum_register_static ("GstAudioRingBufferFormatType", values); g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); } return g_define_type_id__volatile; diff --git a/win32/common/audio-enumtypes.h b/win32/common/audio-enumtypes.h index 3ace3cb..ed54c1b 100644 --- a/win32/common/audio-enumtypes.h +++ b/win32/common/audio-enumtypes.h @@ -8,17 +8,23 @@ G_BEGIN_DECLS -/* enumerations from "multichannel.h" */ +/* enumerations from "audio.h" */ +GType gst_audio_format_get_type (void); +#define GST_TYPE_AUDIO_FORMAT (gst_audio_format_get_type()) +GType gst_audio_format_flags_get_type (void); +#define GST_TYPE_AUDIO_FORMAT_FLAGS (gst_audio_format_flags_get_type()) GType gst_audio_channel_position_get_type (void); #define GST_TYPE_AUDIO_CHANNEL_POSITION (gst_audio_channel_position_get_type()) - -/* enumerations from "gstringbuffer.h" */ -GType gst_ring_buffer_state_get_type (void); -#define GST_TYPE_RING_BUFFER_STATE (gst_ring_buffer_state_get_type()) -GType gst_ring_buffer_seg_state_get_type (void); -#define GST_TYPE_RING_BUFFER_SEG_STATE (gst_ring_buffer_seg_state_get_type()) -GType gst_buffer_format_type_get_type (void); -#define GST_TYPE_BUFFER_FORMAT_TYPE (gst_buffer_format_type_get_type()) +GType gst_audio_flags_get_type (void); +#define GST_TYPE_AUDIO_FLAGS (gst_audio_flags_get_type()) +GType gst_audio_layout_get_type (void); +#define GST_TYPE_AUDIO_LAYOUT (gst_audio_layout_get_type()) + +/* enumerations from "gstaudioringbuffer.h" */ +GType gst_audio_ring_buffer_state_get_type (void); +#define GST_TYPE_AUDIO_RING_BUFFER_STATE (gst_audio_ring_buffer_state_get_type()) +GType gst_audio_ring_buffer_format_type_get_type (void); +#define GST_TYPE_AUDIO_RING_BUFFER_FORMAT_TYPE (gst_audio_ring_buffer_format_type_get_type()) G_END_DECLS #endif /* __GST_AUDIO_ENUM_TYPES_H__ */ diff --git a/win32/common/config.h b/win32/common/config.h index 170fce4..a737431 100644 --- a/win32/common/config.h +++ b/win32/common/config.h @@ -56,12 +56,18 @@ /* set to disable libxml2-dependent code in subparse */ #undef GST_DISABLE_XML +/* Extra platform specific plugin suffix */ +#undef GST_EXTRA_MODULE_SUFFIX + /* macro to use to show function name */ #undef GST_FUNCTION /* Defined if gcov is enabled to force a rebuild due to config.h changing */ #undef GST_GCOV_ENABLED +/* Defined when registry scanning through fork is unsafe */ +#undef GST_HAVE_UNSAFE_FORK + /* plugin install helper script */ #define GST_INSTALL_PLUGINS_HELPER PREFIX "\\libexec\\gst-install-plugins-helper.exe" @@ -81,7 +87,7 @@ #define GST_PACKAGE_ORIGIN "Unknown package origin" /* GStreamer package release date/time for plugins as YYYY-MM-DD */ -#define GST_PACKAGE_RELEASE_DATETIME "2011-09-29" +#define GST_PACKAGE_RELEASE_DATETIME "2012-02-16" /* Define to enable ALSA (used by alsa). */ #undef HAVE_ALSA @@ -152,12 +158,6 @@ /* Define to enable building of plug-ins with external deps. */ #undef HAVE_EXTERNAL -/* FIONREAD ioctl found in sys/filio.h */ -#undef HAVE_FIONREAD_IN_SYS_FILIO - -/* FIONREAD ioctl found in sys/ioclt.h */ -#undef HAVE_FIONREAD_IN_SYS_IOCTL - /* Define to 1 if fseeko (and presumably ftello) exists and is declared. */ #undef HAVE_FSEEKO @@ -170,9 +170,6 @@ /* Define if the GNU gettext() function is already present or preinstalled. */ #undef HAVE_GETTEXT -/* Define to enable GIO library (used by gio). */ -#undef HAVE_GIO - /* Define to 1 if you have the `gmtime_r' function. */ #undef HAVE_GMTIME_R @@ -191,15 +188,6 @@ /* Define to 1 if you have the `asound' library (-lasound). */ #undef HAVE_LIBASOUND -/* Define to 1 if you have the `nsl' library (-lnsl). */ -#undef HAVE_LIBNSL - -/* Define to 1 if you have the `resolv' library (-lresolv). */ -#undef HAVE_LIBRESOLV - -/* Define to 1 if you have the `socket' library (-lsocket). */ -#undef HAVE_LIBSOCKET - /* Define to enable libvisual visualization library (used by libvisual). */ #undef HAVE_LIBVISUAL @@ -224,6 +212,9 @@ /* Use Orc */ #undef HAVE_ORC +/* Defined if compiling for OSX */ +#undef HAVE_OSX + /* Define to enable Pango font rendering (used by pango). */ #undef HAVE_PANGO @@ -275,8 +266,8 @@ /* defined if vorbis_synthesis_restart is present */ #undef HAVE_VORBIS_SYNTHESIS_RESTART -/* Define to 1 if you have the header file. */ -#define HAVE_WINSOCK2_H 1 +/* Defined if compiling for Windows */ +#define HAVE_WIN32 1 /* Define to enable X libraries and plugins (used by ximagesink). */ #undef HAVE_X @@ -309,9 +300,6 @@ */ #undef LT_OBJDIR -/* Define if you have no native hstrerror() function. */ -#undef NO_HSTRERROR - /* Define to 1 if your C compiler doesn't accept -c and -o together. */ #undef NO_MINUS_C_MINUS_O @@ -325,7 +313,7 @@ #define PACKAGE_NAME "GStreamer Base Plug-ins" /* Define to the full name and version of this package. */ -#define PACKAGE_STRING "GStreamer Base Plug-ins 0.11.1" +#define PACKAGE_STRING "GStreamer Base Plug-ins 0.11.2" /* Define to the one symbol short name of this package. */ #define PACKAGE_TARNAME "gst-plugins-base" @@ -334,7 +322,7 @@ #undef PACKAGE_URL /* Define to the version of this package. */ -#define PACKAGE_VERSION "0.11.1" +#define PACKAGE_VERSION "0.11.2" /* directory where plugins are located */ #ifdef _DEBUG @@ -365,7 +353,7 @@ #undef USE_TREMOLO /* Version number of package */ -#define VERSION "0.11.1" +#define VERSION "0.11.2" /* Define WORDS_BIGENDIAN to 1 if your processor stores words with the most significant byte first (like Motorola and SPARC, unlike Intel). */ @@ -390,3 +378,6 @@ /* Define for large files, on AIX-style hosts. */ #undef _LARGE_FILES + +/* We need at least WinXP SP2 for __stat64 */ +#undef __MSVCRT_VERSION__ diff --git a/win32/common/interfaces-enumtypes.c b/win32/common/interfaces-enumtypes.c index 84e968e..aa388e4 100644 --- a/win32/common/interfaces-enumtypes.c +++ b/win32/common/interfaces-enumtypes.c @@ -3,129 +3,10 @@ #include "interfaces-enumtypes.h" -#include "colorbalance.h" -#include "colorbalancechannel.h" -#include "mixer.h" -#include "mixeroptions.h" -#include "mixertrack.h" #include "navigation.h" -#include "propertyprobe.h" -#include "streamvolume.h" #include "tuner.h" #include "tunernorm.h" #include "tunerchannel.h" -#include "videoorientation.h" -#include "videooverlay.h" - -/* enumerations from "colorbalance.h" */ -GType -gst_color_balance_type_get_type (void) -{ - static volatile gsize g_define_type_id__volatile = 0; - if (g_once_init_enter (&g_define_type_id__volatile)) { - static const GEnumValue values[] = { - {GST_COLOR_BALANCE_HARDWARE, "GST_COLOR_BALANCE_HARDWARE", "hardware"}, - {GST_COLOR_BALANCE_SOFTWARE, "GST_COLOR_BALANCE_SOFTWARE", "software"}, - {0, NULL, NULL} - }; - GType g_define_type_id = - g_enum_register_static ("GstColorBalanceType", values); - g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); - } - return g_define_type_id__volatile; -} - -/* enumerations from "mixer.h" */ -GType -gst_mixer_type_get_type (void) -{ - static volatile gsize g_define_type_id__volatile = 0; - if (g_once_init_enter (&g_define_type_id__volatile)) { - static const GEnumValue values[] = { - {GST_MIXER_HARDWARE, "GST_MIXER_HARDWARE", "hardware"}, - {GST_MIXER_SOFTWARE, "GST_MIXER_SOFTWARE", "software"}, - {0, NULL, NULL} - }; - GType g_define_type_id = g_enum_register_static ("GstMixerType", values); - g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); - } - return g_define_type_id__volatile; -} - -GType -gst_mixer_message_type_get_type (void) -{ - static volatile gsize g_define_type_id__volatile = 0; - if (g_once_init_enter (&g_define_type_id__volatile)) { - static const GEnumValue values[] = { - {GST_MIXER_MESSAGE_INVALID, "GST_MIXER_MESSAGE_INVALID", "invalid"}, - {GST_MIXER_MESSAGE_MUTE_TOGGLED, "GST_MIXER_MESSAGE_MUTE_TOGGLED", - "mute-toggled"}, - {GST_MIXER_MESSAGE_RECORD_TOGGLED, "GST_MIXER_MESSAGE_RECORD_TOGGLED", - "record-toggled"}, - {GST_MIXER_MESSAGE_VOLUME_CHANGED, "GST_MIXER_MESSAGE_VOLUME_CHANGED", - "volume-changed"}, - {GST_MIXER_MESSAGE_OPTION_CHANGED, "GST_MIXER_MESSAGE_OPTION_CHANGED", - "option-changed"}, - {GST_MIXER_MESSAGE_OPTIONS_LIST_CHANGED, - "GST_MIXER_MESSAGE_OPTIONS_LIST_CHANGED", "options-list-changed"}, - {GST_MIXER_MESSAGE_MIXER_CHANGED, "GST_MIXER_MESSAGE_MIXER_CHANGED", - "mixer-changed"}, - {0, NULL, NULL} - }; - GType g_define_type_id = - g_enum_register_static ("GstMixerMessageType", values); - g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); - } - return g_define_type_id__volatile; -} - -GType -gst_mixer_flags_get_type (void) -{ - static volatile gsize g_define_type_id__volatile = 0; - if (g_once_init_enter (&g_define_type_id__volatile)) { - static const GFlagsValue values[] = { - {GST_MIXER_FLAG_NONE, "GST_MIXER_FLAG_NONE", "none"}, - {GST_MIXER_FLAG_AUTO_NOTIFICATIONS, "GST_MIXER_FLAG_AUTO_NOTIFICATIONS", - "auto-notifications"}, - {GST_MIXER_FLAG_HAS_WHITELIST, "GST_MIXER_FLAG_HAS_WHITELIST", - "has-whitelist"}, - {GST_MIXER_FLAG_GROUPING, "GST_MIXER_FLAG_GROUPING", "grouping"}, - {0, NULL, NULL} - }; - GType g_define_type_id = g_flags_register_static ("GstMixerFlags", values); - g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); - } - return g_define_type_id__volatile; -} - -/* enumerations from "mixertrack.h" */ -GType -gst_mixer_track_flags_get_type (void) -{ - static volatile gsize g_define_type_id__volatile = 0; - if (g_once_init_enter (&g_define_type_id__volatile)) { - static const GFlagsValue values[] = { - {GST_MIXER_TRACK_INPUT, "GST_MIXER_TRACK_INPUT", "input"}, - {GST_MIXER_TRACK_OUTPUT, "GST_MIXER_TRACK_OUTPUT", "output"}, - {GST_MIXER_TRACK_MUTE, "GST_MIXER_TRACK_MUTE", "mute"}, - {GST_MIXER_TRACK_RECORD, "GST_MIXER_TRACK_RECORD", "record"}, - {GST_MIXER_TRACK_MASTER, "GST_MIXER_TRACK_MASTER", "master"}, - {GST_MIXER_TRACK_SOFTWARE, "GST_MIXER_TRACK_SOFTWARE", "software"}, - {GST_MIXER_TRACK_NO_RECORD, "GST_MIXER_TRACK_NO_RECORD", "no-record"}, - {GST_MIXER_TRACK_NO_MUTE, "GST_MIXER_TRACK_NO_MUTE", "no-mute"}, - {GST_MIXER_TRACK_WHITELIST, "GST_MIXER_TRACK_WHITELIST", "whitelist"}, - {GST_MIXER_TRACK_READONLY, "GST_MIXER_TRACK_READONLY", "readonly"}, - {GST_MIXER_TRACK_WRITEONLY, "GST_MIXER_TRACK_WRITEONLY", "writeonly"}, - {0, NULL, NULL} - }; - GType g_define_type_id = - g_flags_register_static ("GstMixerTrackFlags", values); - g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); - } - return g_define_type_id__volatile; -} /* enumerations from "navigation.h" */ GType @@ -232,27 +113,6 @@ gst_navigation_event_type_get_type (void) return g_define_type_id__volatile; } -/* enumerations from "streamvolume.h" */ -GType -gst_stream_volume_format_get_type (void) -{ - static volatile gsize g_define_type_id__volatile = 0; - if (g_once_init_enter (&g_define_type_id__volatile)) { - static const GEnumValue values[] = { - {GST_STREAM_VOLUME_FORMAT_LINEAR, "GST_STREAM_VOLUME_FORMAT_LINEAR", - "linear"}, - {GST_STREAM_VOLUME_FORMAT_CUBIC, "GST_STREAM_VOLUME_FORMAT_CUBIC", - "cubic"}, - {GST_STREAM_VOLUME_FORMAT_DB, "GST_STREAM_VOLUME_FORMAT_DB", "db"}, - {0, NULL, NULL} - }; - GType g_define_type_id = - g_enum_register_static ("GstStreamVolumeFormat", values); - g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); - } - return g_define_type_id__volatile; -} - /* enumerations from "tunerchannel.h" */ GType gst_tuner_channel_flags_get_type (void) diff --git a/win32/common/interfaces-enumtypes.h b/win32/common/interfaces-enumtypes.h index 77fd7eb..703cdf0 100644 --- a/win32/common/interfaces-enumtypes.h +++ b/win32/common/interfaces-enumtypes.h @@ -8,22 +8,6 @@ G_BEGIN_DECLS -/* enumerations from "colorbalance.h" */ -GType gst_color_balance_type_get_type (void); -#define GST_TYPE_COLOR_BALANCE_TYPE (gst_color_balance_type_get_type()) - -/* enumerations from "mixer.h" */ -GType gst_mixer_type_get_type (void); -#define GST_TYPE_MIXER_TYPE (gst_mixer_type_get_type()) -GType gst_mixer_message_type_get_type (void); -#define GST_TYPE_MIXER_MESSAGE_TYPE (gst_mixer_message_type_get_type()) -GType gst_mixer_flags_get_type (void); -#define GST_TYPE_MIXER_FLAGS (gst_mixer_flags_get_type()) - -/* enumerations from "mixertrack.h" */ -GType gst_mixer_track_flags_get_type (void); -#define GST_TYPE_MIXER_TRACK_FLAGS (gst_mixer_track_flags_get_type()) - /* enumerations from "navigation.h" */ GType gst_navigation_command_get_type (void); #define GST_TYPE_NAVIGATION_COMMAND (gst_navigation_command_get_type()) @@ -34,10 +18,6 @@ GType gst_navigation_message_type_get_type (void); GType gst_navigation_event_type_get_type (void); #define GST_TYPE_NAVIGATION_EVENT_TYPE (gst_navigation_event_type_get_type()) -/* enumerations from "streamvolume.h" */ -GType gst_stream_volume_format_get_type (void); -#define GST_TYPE_STREAM_VOLUME_FORMAT (gst_stream_volume_format_get_type()) - /* enumerations from "tunerchannel.h" */ GType gst_tuner_channel_flags_get_type (void); #define GST_TYPE_TUNER_CHANNEL_FLAGS (gst_tuner_channel_flags_get_type()) diff --git a/win32/common/video-enumtypes.c b/win32/common/video-enumtypes.c index 1ad3740..87f269f 100644 --- a/win32/common/video-enumtypes.c +++ b/win32/common/video-enumtypes.c @@ -4,6 +4,7 @@ #include "video-enumtypes.h" #include "video.h" +#include "colorbalance.h" /* enumerations from "video.h" */ GType @@ -89,6 +90,29 @@ gst_video_format_flags_get_type (void) } GType +gst_video_interlace_mode_get_type (void) +{ + static volatile gsize g_define_type_id__volatile = 0; + if (g_once_init_enter (&g_define_type_id__volatile)) { + static const GEnumValue values[] = { + {GST_VIDEO_INTERLACE_MODE_PROGRESSIVE, + "GST_VIDEO_INTERLACE_MODE_PROGRESSIVE", "progressive"}, + {GST_VIDEO_INTERLACE_MODE_INTERLEAVED, + "GST_VIDEO_INTERLACE_MODE_INTERLEAVED", "interleaved"}, + {GST_VIDEO_INTERLACE_MODE_MIXED, "GST_VIDEO_INTERLACE_MODE_MIXED", + "mixed"}, + {GST_VIDEO_INTERLACE_MODE_FIELDS, "GST_VIDEO_INTERLACE_MODE_FIELDS", + "fields"}, + {0, NULL, NULL} + }; + GType g_define_type_id = + g_enum_register_static ("GstVideoInterlaceMode", values); + g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); + } + return g_define_type_id__volatile; +} + +GType gst_video_flags_get_type (void) { static volatile gsize g_define_type_id__volatile = 0; @@ -99,8 +123,8 @@ gst_video_flags_get_type (void) {GST_VIDEO_FLAG_TFF, "GST_VIDEO_FLAG_TFF", "tff"}, {GST_VIDEO_FLAG_RFF, "GST_VIDEO_FLAG_RFF", "rff"}, {GST_VIDEO_FLAG_ONEFIELD, "GST_VIDEO_FLAG_ONEFIELD", "onefield"}, - {GST_VIDEO_FLAG_TELECINE, "GST_VIDEO_FLAG_TELECINE", "telecine"}, - {GST_VIDEO_FLAG_PROGRESSIVE, "GST_VIDEO_FLAG_PROGRESSIVE", "progressive"}, + {GST_VIDEO_FLAG_VARIABLE_FPS, "GST_VIDEO_FLAG_VARIABLE_FPS", + "variable-fps"}, {0, NULL, NULL} }; GType g_define_type_id = g_flags_register_static ("GstVideoFlags", values); @@ -166,6 +190,7 @@ gst_video_color_matrix_get_type (void) {GST_VIDEO_COLOR_MATRIX_UNKNOWN, "GST_VIDEO_COLOR_MATRIX_UNKNOWN", "unknown"}, {GST_VIDEO_COLOR_MATRIX_RGB, "GST_VIDEO_COLOR_MATRIX_RGB", "rgb"}, + {GST_VIDEO_COLOR_MATRIX_FCC, "GST_VIDEO_COLOR_MATRIX_FCC", "fcc"}, {GST_VIDEO_COLOR_MATRIX_BT709, "GST_VIDEO_COLOR_MATRIX_BT709", "bt709"}, {GST_VIDEO_COLOR_MATRIX_BT601, "GST_VIDEO_COLOR_MATRIX_BT601", "bt601"}, {GST_VIDEO_COLOR_MATRIX_SMPTE240M, "GST_VIDEO_COLOR_MATRIX_SMPTE240M", @@ -195,6 +220,8 @@ gst_video_transfer_function_get_type (void) "smpte240m"}, {GST_VIDEO_TRANSFER_SRGB, "GST_VIDEO_TRANSFER_SRGB", "srgb"}, {GST_VIDEO_TRANSFER_GAMMA28, "GST_VIDEO_TRANSFER_GAMMA28", "gamma28"}, + {GST_VIDEO_TRANSFER_LOG100, "GST_VIDEO_TRANSFER_LOG100", "log100"}, + {GST_VIDEO_TRANSFER_LOG316, "GST_VIDEO_TRANSFER_LOG316", "log316"}, {0, NULL, NULL} }; GType g_define_type_id = @@ -237,12 +264,12 @@ gst_video_buffer_flags_get_type (void) static volatile gsize g_define_type_id__volatile = 0; if (g_once_init_enter (&g_define_type_id__volatile)) { static const GFlagsValue values[] = { + {GST_VIDEO_BUFFER_FLAG_INTERLACED, "GST_VIDEO_BUFFER_FLAG_INTERLACED", + "interlaced"}, {GST_VIDEO_BUFFER_FLAG_TFF, "GST_VIDEO_BUFFER_FLAG_TFF", "tff"}, {GST_VIDEO_BUFFER_FLAG_RFF, "GST_VIDEO_BUFFER_FLAG_RFF", "rff"}, {GST_VIDEO_BUFFER_FLAG_ONEFIELD, "GST_VIDEO_BUFFER_FLAG_ONEFIELD", "onefield"}, - {GST_VIDEO_BUFFER_FLAG_PROGRESSIVE, "GST_VIDEO_BUFFER_FLAG_PROGRESSIVE", - "progressive"}, {GST_VIDEO_BUFFER_FLAG_LAST, "GST_VIDEO_BUFFER_FLAG_LAST", "last"}, {0, NULL, NULL} }; @@ -252,3 +279,21 @@ gst_video_buffer_flags_get_type (void) } return g_define_type_id__volatile; } + +/* enumerations from "colorbalance.h" */ +GType +gst_color_balance_type_get_type (void) +{ + static volatile gsize g_define_type_id__volatile = 0; + if (g_once_init_enter (&g_define_type_id__volatile)) { + static const GEnumValue values[] = { + {GST_COLOR_BALANCE_HARDWARE, "GST_COLOR_BALANCE_HARDWARE", "hardware"}, + {GST_COLOR_BALANCE_SOFTWARE, "GST_COLOR_BALANCE_SOFTWARE", "software"}, + {0, NULL, NULL} + }; + GType g_define_type_id = + g_enum_register_static ("GstColorBalanceType", values); + g_once_init_leave (&g_define_type_id__volatile, g_define_type_id); + } + return g_define_type_id__volatile; +} diff --git a/win32/common/video-enumtypes.h b/win32/common/video-enumtypes.h index dd1805d..4a9db78 100644 --- a/win32/common/video-enumtypes.h +++ b/win32/common/video-enumtypes.h @@ -13,6 +13,8 @@ GType gst_video_format_get_type (void); #define GST_TYPE_VIDEO_FORMAT (gst_video_format_get_type()) GType gst_video_format_flags_get_type (void); #define GST_TYPE_VIDEO_FORMAT_FLAGS (gst_video_format_flags_get_type()) +GType gst_video_interlace_mode_get_type (void); +#define GST_TYPE_VIDEO_INTERLACE_MODE (gst_video_interlace_mode_get_type()) GType gst_video_flags_get_type (void); #define GST_TYPE_VIDEO_FLAGS (gst_video_flags_get_type()) GType gst_video_chroma_site_get_type (void); @@ -27,6 +29,10 @@ GType gst_video_color_primaries_get_type (void); #define GST_TYPE_VIDEO_COLOR_PRIMARIES (gst_video_color_primaries_get_type()) GType gst_video_buffer_flags_get_type (void); #define GST_TYPE_VIDEO_BUFFER_FLAGS (gst_video_buffer_flags_get_type()) + +/* enumerations from "colorbalance.h" */ +GType gst_color_balance_type_get_type (void); +#define GST_TYPE_COLOR_BALANCE_TYPE (gst_color_balance_type_get_type()) G_END_DECLS #endif /* __GST_VIDEO_ENUM_TYPES_H__ */