+=== release 0.10.18 ===
+
+2010-02-10 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ releasing 0.10.18, "Short Circuit"
+
+2010-02-10 20:36:56 +0000 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: temporary safety check to avoid crashes with a certain file
+ Add temporary check to avoid crashes with a certain file when seeking
+ until the real cause of this is figured out. See #609405.
+
+2010-02-05 18:05:39 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ * gst/qtdemux/qtdemux.h:
+ qtdemux: skip unknown atoms when looking for moov
+ Fixes bug #609107
+
+2010-02-05 02:13:33 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ * win32/common/config.h:
+ 0.10.17.3 pre-release
+
+2010-02-04 19:10:36 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * po/bg.po:
+ * po/hu.po:
+ po: update translations
+
+2010-02-04 14:46:56 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ * gst/qtdemux/qtdemux.h:
+ qtdemux: Set the segment start time to the requested seek time for non-keyframe seeks
+
+2010-02-04 12:00:03 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Fix time returned for index at a byte offset
+ The logic for searching forwards/backwards was swapped
+
+2010-02-01 19:22:24 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * ext/speex/gstspeexdec.c:
+ speexdec: initialize stereo decoding state
+
+2010-01-28 18:58:08 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: improve stream synchronization
+ In particular, do not make it send newsegment updates that
+ sort-of contradict the indented playback segment (e.g. start time).
+
+2010-01-28 18:53:18 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: fix bridging (time) gaps in streams
+ As a side effect, avoid sending newsegment updates with start times
+ that go back and forth, which leads to bogus downstream running_time.
+ Also fixes seeking in bug #606744.
+
+2010-01-28 18:49:57 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: fix stream synchronization
+ .. by initializing streams starting at 0, as that is basically
+ where we 'seek to' at the start and assume streams to start elsewhere.
+ Also enables newsegment update events for subtitle streams.
+
+2010-02-02 13:41:03 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * ext/jpeg/gstjpegdec.c:
+ jpeg: don't directly access message, some message have args
+ This caused bogus messages, such as reported in bug #607471.
+
+2010-02-02 00:02:34 +0000 David Hoyt <dhoyt@llnl.gov>
+
+ * ext/libpng/gstpngdec.c:
+ png: fix compilation with libpng 1.4
+ png_set_gray_1_2_4_to_8() has been deprecated for a while and was
+ finally removed in libpng 1.4.x. Use png_set_expand_gray_1_2_4_to_8()
+ instead.
+ Fixes #608629.
+
+2010-02-01 16:46:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: free transports on errors
+ See #608564
+
+2010-02-01 09:18:53 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * sys/v4l2/v4l2_calls.c:
+ v4l2: fix unportable printf format
+
+2010-01-30 15:18:48 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 15d47a6 to 96dc793
+
+2010-01-27 17:53:07 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: index timestamps should be in seconds, not milliseconds
+
+2010-01-27 15:24:52 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * ext/speex/gstspeexdec.c:
+ speexdec: free some more when resetting
+ Fixes #608255.
+
+2010-01-27 15:24:24 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/rtp/gstrtpspeexpay.c:
+ rtpspeexpay: fix occasional buffer leak
+ Fixes #608255.
+
+2010-01-27 15:22:46 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * ext/speex/gstspeexenc.c:
+ speexenc: prevent invalid arithmetic if not setup yet
+ Fixes #608255.
+
+2010-01-27 16:34:21 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/blend_mmx.h:
+ videomixer: Fix assembly register constraints
+ Fixes bug #608209.
+
+2010-01-27 01:56:03 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ * win32/common/config.h:
+ 0.10.17.2 pre-release
+
+2010-01-27 01:52:59 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * po/LINGUAS:
+ * 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/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ po: update translations
+
+2010-01-27 01:49:49 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * tests/check/elements/.gitignore:
+ checks: ignore deinterlace check binary
+
+2010-01-27 01:18:51 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: purge all mention of CVS
+
+2010-01-26 11:18:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: ignore streams that finished
+ When we receive an UNEXPECTED from a stream, move to the next stream and only go
+ EOS when all streams are EOS. When selecting a stream to push, ignore streams
+ that went EOS.
+ Fixes #607949
+
+2010-01-25 17:23:43 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * sys/v4l2/v4l2src_calls.c:
+ v4l2src: don't deref NULL
+ Error out when the pool gets shutdown.
+
+2010-01-25 17:21:13 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * ext/jpeg/gstjpegenc.c:
+ * sys/v4l2/v4l2src_calls.c:
+ * tests/check/Makefile.am:
+ Revert "v4l2src: don't deref NULL"
+ This reverts commit 3d9d34bd60faeb940b36d992a47168fc895036ba.
+
+2010-01-25 14:16:22 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * ext/jpeg/gstjpegenc.c:
+ * sys/v4l2/v4l2src_calls.c:
+ * tests/check/Makefile.am:
+ v4l2src: don't deref NULL
+ Error out when the pool gets shutdown.
+
+2010-01-23 15:32:48 -0800 Michael Smith <msmith@xiph.org>
+
+ * ext/jpeg/gstjpegenc.c:
+ jpegenc: when creating an overflow buffer, copy timestamps.
+
+2010-01-23 14:47:55 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: dmb1 is a valid fourcc for Motion-JPEG
+
+2010-01-23 14:20:02 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdeux: IV32 is also used for Indeo 3 video streams
+
+2010-01-22 16:48:01 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * tests/icles/ximagesrc-test.c:
+ build: no unused variables when disabling asserts
+
+2010-01-21 23:17:40 -0300 Roland Krikava <rkrikava@gmail.com>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Avoid negative overflow on keyframe search
+ Do not overflow negatively when searching a previous
+ "keyframe" on audio streams. Could cause infinite loops
+ on backwards playback
+ Fixes #607718
+
+2010-01-21 17:22:38 -0800 Peter van Hardenberg <pvh@songbirdnest.com>
+
+ * ext/jpeg/gstjpegenc.c:
+ * ext/jpeg/gstjpegenc.h:
+ jpegenc: enlarge buffer if libjpeg tells us it's out of space. Fixes buffer overflow on some high-quality, low-resolution jpeg encodes.
+
+2010-01-21 19:24:22 +0100 Alessandro Decina <alessandro.d@gmail.com>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: fix compiler warnings under OS X.
+
+2010-01-21 17:57:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: don't parse NULL indexes
+ for some streams we might fail to fetch the index offsets. Don't try to parse
+ NULL indexes in those cases.
+
+2010-01-18 21:15:51 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtp/gstrtpg729pay.c:
+ rtpg729pay: ptime should is in nanoseconds
+ https://bugzilla.gnome.org/show_bug.cgi?id=607403
+
+2010-01-20 15:11:15 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/wavenc/gstwavenc.c:
+ * gst/wavenc/gstwavenc.h:
+ wavenc: Post warning if file isnt finished properly
+ When the pipeline is shut down and the file isn't
+ finished properly, wavenc should post a warning.
+ Fixes #607440
+
+2009-05-27 13:51:44 +0200 Arnout Vandecappelle <arnout@mind.be>
+
+ * gst/matroska/matroska-mux.c:
+ * gst/matroska/matroska-mux.h:
+ matroskamux: make index size configurable.
+ Added the 'min-index-interval' property to matroskamux,
+ which determines how much time (nanoseconds) is left
+ between keyframes stored in the index.
+ Fixes #583985.
+
+2010-01-20 16:28:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: scale spspps_interval to milliseconds
+ The spspps_interval is kept in seconds. Convert it to milliseconds before
+ comparing it to another value in milliseconds.
+
+2010-01-20 15:18:47 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: always keep media segments within total duration
+ ... as opposed to only doing so following a seek.
+
+2010-01-20 15:44:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: rename spspps-interval property
+ Rename the spspps-interval property to config-interval because it is nicer.
+
+2010-01-19 18:37:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: skip RIFF and index in push mode
+ When we are in push mode, we can encounter RIFF and idx tags in the data chunk
+ when we are dealing with ODML files. In these cases, simply skip the chunks and
+ continue streaming instead of going EOS.
+
+2010-01-20 11:27:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: more DISCONT handling
+ Add some debug in the DISCONT handling code.
+ When we receive a DISCONT in push mode, mark all streams as DISCONT.
+
+2010-01-20 11:26:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: reset on flush events
+ When we receive a flush event on the sinkpad, reset the EOS state and the
+ flowreturn of all streams. Also mark the streams with a DISCONT.
+
+2010-01-20 11:22:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ * gst/avi/gstavidemux.h:
+ avidemux: rename some variable
+ Rename the seek_event variable to seg_event because it really contains the
+ newsegment event that needs to be pushed.
+
+2010-01-20 00:54:03 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 14cec89 to 15d47a6
+
+2010-01-18 14:49:26 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtp/gstrtph264pay.c:
+ * gst/rtp/gstrtph264pay.h:
+ rtph264pay: Don't set profile-level-id in out caps
+ The profile-level-id represents restrictions on what can be sent, it does not
+ describe the stream. So it should be reflected in the sink caps of the
+ payloader, not the src caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=607353
+
+2010-01-18 14:41:10 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: Don't ignore the return value from set_outcaps
+ https://bugzilla.gnome.org/show_bug.cgi?id=607353
+
+2010-01-18 17:43:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/deinterlace/tvtime/greedyhmacros.h:
+ * gst/deinterlace/tvtime/linear.c:
+ * gst/deinterlace/tvtime/linearblend.c:
+ * gst/deinterlace/tvtime/tomsmocomp.c:
+ * gst/deinterlace/tvtime/weave.c:
+ * gst/deinterlace/tvtime/weavebff.c:
+ * gst/deinterlace/tvtime/weavetff.c:
+ deinterlace: Fix license and copyright headers
+
+2010-01-18 14:57:42 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * sys/v4l2/gstv4l2bufferpool.h:
+ v4l2: move G_END_DECLS to the end
+
+2010-01-18 14:55:38 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2bufferpool.h:
+ v4l2: fix bufferpool file names in header comment
+
+2010-01-15 18:15:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: avoid some typecasting
+
+2010-01-15 18:13:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: avoid some type checks
+
+2010-01-15 18:09:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ * gst/avi/gstavidemux.h:
+ avidemux: fallback to avih duration
+ when we have not yet parsed the indexes (in push mode, for example) use
+ the duration as given in the avih header instead of -1.
+
+2010-01-15 13:32:32 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: g_free is NULL safe
+
+2010-01-15 13:27:40 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: use DEMUX errors, instead of DECODE
+ qtdemux should use DEMUX errors, and not DECODE
+ Conflicts:
+ gst/qtdemux/qtdemux.c
+
+2010-01-14 19:16:19 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Minor refactor
+ Replace repeated code with a function call
+
+2010-01-14 17:11:13 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ * gst/qtdemux/qtdemux_fourcc.h:
+ qtdemux: Handle another kind of redirect trak
+ Some traks might contain a redirect rtsp uri inside
+ hndl atom (which is a dref atom entry). This commit makes qtdemux
+ post a message when it finds one of these traks and there are
+ no other traks.
+ Fixes #597497
+
+2010-01-14 16:13:08 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ * gst/qtdemux/qtdemux.h:
+ qtdemux: Post error when reaching EOS without pads
+ Post an error when EOS is reached and there are no src pads
+
+2010-01-14 14:13:50 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Do not post empty redirect messages
+ Some misinterpreted data could result in posting redirect messages
+ with empty redirect strings. It is better not to post them.
+ An example is the file on bug #597497
+
+2010-01-14 18:19:25 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: polish last buffer end time usage
+ That is, reset it upon seek, and note that (rarely) last pushed buffer
+ time might precede segment start.
+
+2010-01-13 16:48:46 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/videomixer/blend_mmx.h:
+ videomixer: use 'q' constraint instead of 'r'
+ This avoids the "bad register name `%dil'" compilation errors on 32bit where
+ because of 'r' gcc puts the value in a general purpose register and then tries
+ to access the lower part as %dil/%sil which is not existing on 32bit. 'q' requests
+ a-d registers
+
+2010-01-13 16:44:58 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/avi/gstavidemux.c:
+ avi: add missing include for sscanf
+
+2010-01-13 09:36:03 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/equalizer/gstiirequalizer10bands.c:
+ equalizer: Fix property description for the 3rd band of the 10band equalizer
+ The frequency is actually 237 Hz, not 227 Hz.
+ Fixes bug #606692.
+
+2010-01-13 09:22:20 +0100 Kipp Cannon <kcannon@ligo.caltech.edu>
+
+ * gst/audiofx/audioamplify.c:
+ audioamplify: Allow negative amplifications
+ Fixes bug #606807.
+
+2010-01-13 09:17:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/taglib/gstapev2mux.cc:
+ apev2mux: Don't call constructors directly, this leads to compiler errors with gcc 4.5
+
+2010-01-12 17:39:05 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: use G_GSIZE_FORMAT for platform independent gsize qualifier
+ Fixes build on macosx
+
+2010-01-11 19:02:34 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: refactor eos sending when pausing loop
+ Also, prevent hanging if no pads yet on which to send eos by
+ posting a message instead.
+
+2010-01-11 17:50:35 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: standardize seek handling
+ ... which implies fixing some corner cases.
+
+2010-01-11 15:14:06 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: use more generic xiphN_streamheader_to_codecdata helper
+
+2010-01-11 17:50:04 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: reflow audio and video setcaps and improve logging
+ Also ensure width and height are available as they are mandatory
+ in matroska specs.
+
+2010-01-11 11:42:43 -0800 Michael Smith <msmith@songbirdnest.com>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: fix offset for type 2 mp4a sound sample descriptions.
+ Allows us to correctly find the esds (and thus the codec data) for such
+ mp4a files.
+
+2010-01-11 15:45:49 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/rtp/gstrtpmp4gdepay.c:
+ * gst/rtp/gstrtpmp4gpay.c:
+ rtpmp4g(de)pay: Only handle raw aac
+ rtpmp4g(de)pay should only handle raw AAC streams
+
+2010-01-11 18:59:43 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/videomixer.c:
+ * gst/videomixer/videomixer.h:
+ videomixer: Implement basic QoS
+ This drops frames if they're too late anyway before blending and all
+ that starts but QoS events are not forwarded upstream. In the future
+ the QoS events should be transformed somehow and forwarded upstream.
+
+2010-01-11 14:48:26 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/rtp/gstrtpmp4adepay.c:
+ * gst/rtp/gstrtpmp4apay.c:
+ rtpmp4a(de)pay: Only accept raw aac
+ rtpmp4a(de)pay should only handle raw aac to conform to the RFC
+
+2010-01-11 18:35:47 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/blend.c:
+ * gst/videomixer/blend_mmx.h:
+ videomixer: Add MMX implementations for I420 and all non-alpha RGB formats
+
+2010-01-04 10:24:45 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/Makefile.am:
+ * gst/videomixer/blend.c:
+ * gst/videomixer/blend.h:
+ * gst/videomixer/blend_ayuv.c:
+ * gst/videomixer/blend_bgra.c:
+ * gst/videomixer/blend_i420.c:
+ * gst/videomixer/blend_mmx.h:
+ * gst/videomixer/blend_rgb.c:
+ * gst/videomixer/videomixer.c:
+ * gst/videomixer/videomixer.h:
+ videomixer: Refactor processing functions
+ This allows easier plugging of optimized processing functions
+ in the future, like for SSE or AltiVec.
+
+2010-01-11 13:26:32 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/avi/gstavimux.c:
+ * gst/matroska/matroska-mux.c:
+ avimux: matroskamux: rename aac's stream-format to raw
+ AAC's none stream-format has been renamed to raw, rename
+ on avimux and matroskamux as well
+
+2010-01-11 12:07:29 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: Only accept raw aac
+ makes matroskamux reject aac streams that are not
+ in raw format (stream-format=none)
+ Fixes #598350
+
+2010-01-11 12:08:55 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/avi/gstavimux.c:
+ avimux: Only accept raw aac
+ makes avimux reject aac streams that are not
+ in raw format (stream-format=none)
+ Fixes #598350
+
+2010-01-11 10:38:10 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Oops. The gpointer cast is needed because of the const qualifiers on the data elements
+
+2010-01-11 10:17:54 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Debug -> info level for a message for benchmarking index parsing
+ The extra message output at higher levels affects the accuracy of the
+ benchmark.
+
+2010-01-11 10:05:10 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Don't check for NULL pointers or cast to gpointer as this is not needed
+
+2010-01-08 13:55:05 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Refactor stbl sub-atom freeing. Free when index has been completely parsed.
+
+2010-01-08 14:32:06 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Avoid whitespace commits due to inconsistent GNU indent behaviour
+
+2010-01-11 00:10:34 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: remove newline at end of debug statement
+
+2010-01-08 19:26:21 +0100 Havard Graff <havard.graff@tandberg.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: Compiler warning fixes for Windows
+ Just simple missing casts
+ Fixes bug #606438.
+
+2010-01-08 18:04:14 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: fix seekpoints property copy-and-paste documentation
+
+2010-01-06 17:06:53 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * ext/flac/gstflacenc.c:
+ * ext/flac/gstflacenc.h:
+ flacenc: optionally add a seek table
+ API: GstFlacEnc:seekpoints
+ Fixes #351595.
+
+2010-01-08 11:33:02 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Use more glib and be safer
+ Be safer on sscanf by limiting string format sizes.
+ Remove useless parameter and use g_strndup.
+
+2010-01-08 10:44:44 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Simplifying code
+ Greatly simplify the IDIT chunk handling by using sscanf
+ instead of 'manually' parsing. Also replaces strncasecmp and
+ is_alpha/is_digit with glib versions.
+
+2010-01-08 10:18:30 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: it's feb for february
+ Fix typo in last commit.
+
+2010-01-08 09:17:22 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Parse and post IDIT dates
+ Parses and post date tags contained in IDIT chunks.
+ Fixes #503582
+
+2010-01-07 17:25:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiofx/audiofirfilter.c:
+ * gst/audiofx/audiofxbasefirfilter.c:
+ * gst/audiofx/audiofxbasefirfilter.h:
+ audiofxbasefirfilter: Add property for not draining the history on kernel changes
+ Currently this only works if the kernel size doesn't change, in the future
+ it will be possible to change the kernel size too without draining
+ the complete history and without loosing anything.
+ Partially based on a patch by
+ Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+2010-01-07 16:58:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: remove weird memcmp code
+ Use plain memcmp for comparing memory instead of the custom buggy one.
+ Fixes #606198
+
+2010-01-07 15:38:36 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/level/gstlevel.c:
+ level: fix typo in 'message' property description
+
+2010-01-06 14:06:14 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * ext/flac/gstflacdec.c:
+ flacdec: really use upstream timestamp if there is one
+ See/fixes #603471.
+
+2010-01-06 13:45:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpg729pay.c:
+ rtpg728pay: remove unused adapter peek
+
+2010-01-05 19:00:35 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * tests/check/elements/deinterlace.c:
+ deinterlace: Improve passthrough tests
+ Improve passthrough tests by forcing more specific
+ interlaced/deinterlaced caps to be tested
+
+2010-01-05 18:22:49 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * tests/check/elements/deinterlace.c:
+ deinterlace: Adds some docs to the new tests
+ Adds some docs explaining the utility functions of the check
+ tests of deinterlace
+
+2010-01-05 18:14:08 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * tests/check/elements/deinterlace.c:
+ deinterlace: Adds tests for passthrough
+ Adds tests for checking if the element really does
+ passthrough in disabled mode and in auto (if the input is
+ not interlaced)
+
+2010-01-05 07:50:51 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/deinterlace.c:
+ deinterlace: Adds tests for caps acceptance
+ Adds check unit tests for deinterlace for validating
+ caps accepting and the expected caps output on the
+ other pad
+
+2010-01-04 13:43:00 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/deinterlace.c:
+ deinterlace: Adds basic check test
+ Adds a basic check test for deinterlace element
+
+2010-01-04 15:44:28 -0800 Michael Smith <msmith@songbirdnest.com>
+
+ * gst/qtdemux/Makefile.am:
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Add support for wave-style audio in qt.
+ Uses gstriff to parse the wave headers appropriately. Tested with MS-ADPCM
+ content.
+
+2009-12-31 17:09:03 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * tests/check/elements/rtp-payloading.c:
+ tests: Add G.729 RTP payloader/depayloader test
+ https://bugzilla.gnome.org/show_bug.cgi?id=606050
+
+2009-12-31 16:52:30 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtp/gstrtpg729pay.c:
+ rtpg729pay: Simplify adapter usage
+ https://bugzilla.gnome.org/show_bug.cgi?id=606050
+
+2009-12-31 16:27:30 -0500 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtp/gstrtpg729pay.c:
+ rtpg729pay: Support ptime from caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=606050
+
+2009-12-02 19:35:21 +0530 Olivier Crête <olivier.crete@collabora.co.uk>
+
+ * gst/rtp/README:
+ rtp: Add maxptime to the README
+ https://bugzilla.gnome.org/show_bug.cgi?id=606050
+
+2010-01-05 19:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpg723depay.c:
+ * gst/rtp/gstrtpg723depay.h:
+ rtpg723depay: add G723 depayloader
+
+2010-01-05 19:02:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpg729depay.c:
+ * gst/rtp/gstrtpg729depay.h:
+ rtpg729depay: remove unused variable
+
+2010-01-05 18:33:25 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpg723pay.c:
+ * gst/rtp/gstrtpg723pay.h:
+ rtpg723pay: rewrite payloader
+ Handle all 3 packet sizes according to RFC 3551.
+ Totally untested, we don't have a G723 encoder.
+ Fixes #605882
+
+2010-01-05 11:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: fix chunk counter
+
+2010-01-04 19:44:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: more work at reducing loop overhead
+ Try to avoid derefs when parsing the index. Save the state into the structures
+ when we exit the loop instead of for each iteration.
+
+2010-01-04 16:33:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: cleanups and make duration more accurate
+ Make the QtDemuxSample struct smaller by keeping the duration and the pts_offset
+ as their 32 bit values.
+ Make some macros to calculate PTS, DTS and duration of a sample.
+ Deref the sample index less often by keeping a ref to the sample we're dealing
+ with.
+
+2010-01-04 13:41:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: simplify logic to calculate duration
+ Since we no longer store the timestamp and duration in nanoseconds, we can now
+ simply store the duration as-is.
+
+2010-01-01 16:42:57 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Store timestamps in mov format in the index
+ This allows faster building of the index upon seeks so that scaling of
+ timestamps only occurs when actually needed.
+
+2009-12-18 13:54:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: make seeking in push mode work
+ Move sample position checks into qtdemux_parse_samples where we can protect it
+ with a lock.
+ Refactor and make an qtdemux_ensure_index function.
+ Rename qtdemux_do_push_seek to qtdemux_seek_offset in order to avoid confusion
+ with gst_qtdemux_do_push_seek.
+
+2009-12-18 12:44:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: move error code out of normal flow
+
+2009-11-24 16:27:26 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ * gst/qtdemux/qtdemux.h:
+ qtdemux: Add push mode seek support for seeking to obtain the moov atom
+
+2010-01-05 12:22:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: fix on-npt-stop signal warnings for RDT
+ The RDT manager does not implement this signal so we need to check for it before
+ trying to connect to it.
+
+2010-01-05 09:47:00 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2src.c:
+ v4l2src: fix memory leak in new uri handler code
+ Don't leak a string everytime get_uri() is called and a device
+ has been set. There's a limited number of devices, so just
+ intern the string instead of doing more elaborate housekeeping
+ and storing it in the instance struct or so.
+
+2010-01-01 14:10:49 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/avi/gstavimux.c:
+ avimux: fix typo in warning message
+
+2010-01-04 09:28:36 -0300 Robert Weidlich <gnomebugzilla@robert.weidlich.cc>
+
+ * ext/shout2/gstshout2.c:
+ * ext/shout2/gstshout2.h:
+ shout2send: Add 'public' property
+ Adds a property to set 'public' flag on libshout, making
+ the stream listed on the server's stream directory.
+ Fixes #605269
+
+2009-12-30 14:14:55 +0530 Arun Raghavan <arun.raghavan@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Add tags for average and maximum bitrate
+ Fixes #599300.
+
+2009-12-26 16:59:14 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/audiofx/audiofxbasefirfilter.c:
+ audiofxbasefirfilter: do not try to alloc really large buffers
+ When nsamples_out is larger than nsamples_in, using unsigned
+ ints lead to a overflow and the resulting value is wrong and
+ way too large for allocating a buffer. Use signed integers
+ and returning immediatelly when that happens.
+
+2009-12-25 12:38:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videomixer/blend_ayuv.c:
+ videomixer: optimize blend code some more
+ Use more efficient formula that uses less multiplies.
+ Reduce the amount of scalar code, use MMX to calculate the desired
+ alpha value.
+ Unroll and handle 2 pixels in one iteration for improved pairing.
+
+2009-12-24 22:59:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videomixer/blend_ayuv.c:
+ * gst/videomixer/blend_bgra.c:
+ * gst/videomixer/blend_i420.c:
+ * gst/videomixer/blend_rgb.c:
+ videomixer: scale and clamp
+ Scale and clamp to the max alpha values.
+
+2009-12-24 22:50:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/alpha/gstalpha.c:
+ alpha: scale and clamp alpha to its full extend
+ Convert the alpha value to 0->255 when setting and to 0->256 when using as
+ a scaling factor. This makes sure we can reach the full opacity value of 0xff in
+ all cases.
+
+2009-12-24 22:23:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: fix some comments, remove property check
+ Fix some comments, clarify some FIXMEs
+ Remove the on-ntp-stop signal check now that the jitterbuffer is in
+ -good and we know that it supports this signal.
+
+2009-12-24 20:27:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/videomixer/videomixer.c:
+ videomixer: some trivial cleanups
+
+2009-12-24 17:04:28 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Parse all rtpinfo entries
+ Do not forget to parse all rtp-info entries, instead of
+ parsing the first one only.
+ Fixes #605222
+
+2009-12-22 12:44:50 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: perf tag should map to GST_TAG_ARTIST
+
+2009-12-24 17:03:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/interleave/interleave.c:
+ interleave: fix weird indentation
+
+2009-12-24 17:01:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtph263ppay.c:
+ rtph263ppay: use faster _adapter_copy() whem possible
+
+2009-12-24 17:01:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/examples/audiofx/firfilter-example.c:
+ tests: use right type when passing vararg value
+
+2009-12-23 17:50:34 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * ext/flac/gstflacdec.c:
+ * ext/flac/gstflacdec.h:
+ flacdec: use a single decoder field for both push and pull mode
+
+2009-12-23 17:03:32 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * ext/flac/gstflacdec.c:
+ flacdec: fix possible hanging in pull mode seeking
+ A seek in multi-sink pipeline typically leads to several seek events in a row,
+ which could lead to sending several newsegments in a row without intermediate
+ flushing. These would then accumulate, distort rendering times and as such
+ lead to 'hanging'.
+
+2009-12-23 19:39:05 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: fix uninitialized variable
+
+2009-12-23 13:09:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstasteriskh263.c:
+ * gst/rtp/gstrtpL16depay.c:
+ * gst/rtp/gstrtpac3depay.c:
+ * gst/rtp/gstrtpamrdepay.c:
+ * gst/rtp/gstrtpamrpay.c:
+ * gst/rtp/gstrtpbvpay.c:
+ * gst/rtp/gstrtpdepay.c:
+ * gst/rtp/gstrtpg729depay.c:
+ * gst/rtp/gstrtpgsmdepay.c:
+ * gst/rtp/gstrtpgsmpay.c:
+ * gst/rtp/gstrtph263depay.c:
+ * gst/rtp/gstrtph263pay.c:
+ * gst/rtp/gstrtph263pdepay.c:
+ * gst/rtp/gstrtph263ppay.c:
+ * gst/rtp/gstrtpilbcpay.c:
+ * gst/rtp/gstrtpjpegdepay.c:
+ * gst/rtp/gstrtpmp1sdepay.c:
+ * gst/rtp/gstrtpmp2tdepay.c:
+ * gst/rtp/gstrtpmp4apay.c:
+ * gst/rtp/gstrtpmp4gdepay.c:
+ * gst/rtp/gstrtpmp4gpay.c:
+ * gst/rtp/gstrtpmp4vpay.c:
+ * gst/rtp/gstrtpmpadepay.c:
+ * gst/rtp/gstrtpmpapay.c:
+ * gst/rtp/gstrtpmpvdepay.c:
+ * gst/rtp/gstrtppcmadepay.c:
+ * gst/rtp/gstrtppcmudepay.c:
+ * gst/rtp/gstrtppcmupay.c:
+ * gst/rtp/gstrtpqdmdepay.c:
+ * gst/rtp/gstrtpsirenpay.c:
+ * gst/rtp/gstrtpsv3vdepay.c:
+ * gst/rtp/gstrtptheorapay.c:
+ * gst/rtp/gstrtpvorbispay.c:
+ * gst/rtp/gstrtpvrawdepay.c:
+ * gst/rtp/gstrtpvrawpay.c:
+ rtp: use boilerplate
+
+2009-12-23 00:38:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpL16pay.c:
+ * gst/rtp/gstrtpL16pay.h:
+ rtpL16pay: convert to baseaudiopayload
+ Use GstRTPBaseAudioPayload as the base class. This saves a lot of code and fixes
+ a bunch of problems that were already solved in the base class.
+ Fixes #853367
+
+2009-12-23 00:30:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtppcmapay.c:
+ rtppcmapay: the boilerplate macro sets parent_class
+
+2009-12-22 22:27:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ rtpbin: avoid some structure copies
+ Don't make copied in the getter and setter for SDES in the RTPSource. This
+ avoids a couple of copies of the SDES structure when generating RTCP
+ packets.
+
+2009-08-31 18:42:25 +0200 Pascal Buhler <pascal.buhler@tandberg.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ rtpmanager: improve SDES handling
+ Store SDES internally as a struct to support multiple PRIV values.
+ Include all values set in SDES struct when sending RTCP SDES.
+
+2009-12-22 14:41:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtph263depay.c:
+ rtph263depay: add some fixmes
+
+2009-12-22 14:35:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtph263depay.c:
+ rtph263depay: baseclass handles timestamps for us
+
+2009-12-22 14:27:40 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtph263depay.c:
+ rtph263depay: reset start variable properly
+
+2009-05-29 15:49:27 +0300 Marco Ballesio <marco.ballesio@nokia.com>
+
+ * gst/rtp/gstrtph263depay.c:
+ * gst/rtp/gstrtph263depay.h:
+ Drop the whole frame if a packet is lost.
+ Fixes #582575
+
+2009-12-21 20:39:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtph264pay.c:
+ * gst/rtp/gstrtph264pay.h:
+ rtph264pay: add option to insert PPS/SPS in streams
+ Add a new spspps-interval property to instruct the payloader to insert
+ SPS and PPS at periodic intervals in the stream.
+ Rework the SPS/PPS handling so that bytestream and AVC sample code both use the
+ same code paths to handle sprop-parameter-sets. This also allows to have the AVC
+ code to insert SPS/PPS like the bytestream code.
+ Fixes #604913
+
+2009-12-21 19:12:22 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 47cb23a to 14cec89
+
+2009-12-21 12:01:53 -0300 Jonathan Conder <j@skurvy.no-ip.org>
+
+ * gst/qtdemux/qtdemux.c:
+ * gst/qtdemux/qtdemux_fourcc.h:
+ * gst/qtdemux/qtdemux_types.c:
+ qtdemux: Adds new tags
+ Adds some new tags mapping to qtdemux.
+ Fixes #599759
+
+2009-12-21 15:05:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: add property to remove pads automatically
+ Add a property called autoremove to automatically remove the pads of sources
+ that timed out.
+ Fixes #554839
+
+2009-12-21 14:55:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpssrcdemux.c:
+ ssrcdemux: fix comparison
+ A NULL means no pad was found.
+
+2009-11-08 11:49:14 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * sys/v4l2/gstv4l2src.c:
+ v4l2src: Add GstURIHandler interface. Fixes #601143
+ This allows using v4l2://[<device>]
+
+2009-12-20 17:24:47 -0800 Michael Smith <msmith@xiph.org>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: pass length parameter to g_convert
+
+2009-12-18 12:44:50 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroska: Fix unitialized variable.
+ Yes, it's stupid, but macosx compilers are even more stupid.
+
+2009-12-17 16:01:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/blend_ayuv.c:
+ videomixer: Fix assembly compilation on x86
+ Fixes bug #604814.
+
+2009-12-17 17:37:03 +0100 Branko Čibej <brane at xbc.nu>
+
+ * gst/replaygain/rganalysis.c:
+ rganalysis: fix timestamp rounding
+ Use scaling function to round and avoid overflows.
+ Fixes #604352
+
+2009-12-17 17:27:42 +0100 Tiago Katcipis <tiago.katcipis@digitro.com.br>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpg723pay.c:
+ * gst/rtp/gstrtpg723pay.h:
+ rtp: add G723 payloader
+ Fixes #597823
+
+2009-12-17 16:22:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ * gst/qtdemux/qtdemux_types.c:
+ qtdemux: Fix ALAC codec_data parsing
+ Fixes #604611
+
+2009-12-16 17:28:30 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Remove cpp style coments
+ Removes // comments and replace them with /* */ comments
+
+2009-12-16 12:48:02 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-demux.h:
+ matroskademux: also consider BlockNumber indicated in index when seeking
+
+2009-12-16 12:43:27 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/matroska/ebml-read.c:
+ * gst/matroska/ebml-read.h:
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-demux.h:
+ matroskademux: support push based mode
+ Fixes #598610.
+
+2009-12-16 12:44:36 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/matroska/ebml-read.c:
+ matroskademux: fix ebml read cache usage
+
+2009-12-16 10:50:32 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/blend_ayuv.c:
+ videomixer: Use movzbl instead of movzxb for moving one byte to a l register
+ For some reason latest gcc/binutils accept movzxb here while
+ movzbl would be correct and is the only thing accepted by older
+ gcc/binutils.
+ Fixes bug #604679.
+
+2009-12-16 06:59:01 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/blend_ayuv.c:
+ videomixer: src/dest are input and output of the AYUV blending MMX assembler
+
+2009-12-15 18:18:54 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiofx/audiowsincband.c:
+ audiowsincband: Use the same upper length limit as audiowsinclimit
+
+2009-12-12 17:00:50 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiofx/audiowsincband.c:
+ * gst/audiofx/audiowsinclimit.c:
+ audiowsinc{limit,band}: Allow much larger filter lengths now
+
+2009-12-11 12:27:32 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiofx/audiofxbasefirfilter.c:
+ audiofxbasefirfilter: Fix frequency response calculation
+
+2009-12-08 14:57:02 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiofx/audiofxbasefirfilter.c:
+ audiofxbasefirfilter: Remove dead assignments
+
+2009-12-06 16:58:51 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiofx/audiofxbasefirfilter.c:
+ audiofxbasefirfilter: Add special processing functions for Mono/Stereo
+ This provides another 7% speedup for the time domain convolution and 1.5%
+ speedup for the FFT convolution on Mono input.
+ This optimization assumes that the compiler simplifies calculations
+ and conditions on constant numbers and unrolls loops with a constant
+ number of repeats.
+
+2009-12-04 09:25:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiofx/audiofxbasefirfilter.c:
+ * gst/audiofx/audiofxbasefirfilter.h:
+ audiofxbasefirfilter: Add a "low-latency" mode
+ This will always use time-domain convolution, which lowers the latency.
+ With FFT convolution it's always a multiple of the kernel length,
+ with time domain convolution it's only the pre-latency of the filter kernel.
+
+2009-12-04 09:00:22 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiofx/audiofxbasefirfilter.c:
+ audiofxbasefirfilter: Remove obsolete TODO comments
+
+2009-12-03 20:12:01 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiofx/audiofxbasefirfilter.c:
+ audiofxbasefirfilter: Use samples everywhere instead of samples*channels sometimes
+
+2009-12-03 17:27:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiofx/Makefile.am:
+ * gst/audiofx/audiofxbasefirfilter.c:
+ * gst/audiofx/audiofxbasefirfilter.h:
+ audiofxbasefirfilter: FFT convolution implementation
+ This provides a great speedup, especially the relationship between kernel
+ length and processing size is now logarithmic instead of linear. Below a
+ kernel size of 32 it's a bit slower, afterwards it's much faster:
+ 17 0.788000 -> 0.950000
+ 33 1.208000 -> 1.146000
+ 65 2.166000 -> 1.146000
+ ...
+ 4097 107.444000 -> 1.508000
+ For sizes smaller 32 the normal time-domain convolution is chosen,
+ for larger sizes the FFT convolution is automatically used.
+ Fixes bug #594381.
+
+2009-11-27 20:33:14 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiofx/audiofxbasefirfilter.c:
+ * gst/audiofx/audiofxbasefirfilter.h:
+ audiofxbasefirfilter: Make most code parts independent of the processing functions and used convolution algorithm
+ Only remaining part is the residue pushing, which will be fixed later.
+
+2009-11-26 15:17:27 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiofx/audiofxbasefirfilter.c:
+ audiofxbasefirfilter: Optimize time-domain convolution
+ Remove some redundant calculations, move comparisions out of
+ inner loops, etc.
+ This makes the convolution about 3 (!) times faster but
+ processing time is of course still proportional to the
+ filter size.
+
+2009-11-26 10:45:37 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiofx/audiofxbasefirfilter.c:
+ audiofxbasefirfilter: Use _CAST macros in some places and do some calculations only once
+
+2009-11-25 18:12:05 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiofx/audiofxbasefirfilter.c:
+ * gst/audiofx/audiofxbasefirfilter.h:
+ audiofxbasefirfilter: Rewrite timestamp tracking
+ It's much simpler now and doesn't introduce accumulating rounding
+ errors.
+
+2009-11-25 17:39:53 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiofx/audiofxbasefirfilter.c:
+ * gst/audiofx/audiofxbasefirfilter.h:
+ audiofxbasefirfilter: Rename some variables and change comments
+
+2009-11-24 20:06:25 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/audiofx/audiofxbasefirfilter.c:
+ * gst/audiofx/audiofxbasefirfilter.h:
+ audiofxbasefirfilter: Add const qualifier to the source data array
+
+2009-12-14 20:08:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/Makefile.am:
+ * gst/videomixer/blend_ayuv.c:
+ * gst/videomixer/videomixer.c:
+ videomixer: Add MMX implementations of the AYUV blending and color filling functions
+ This provides a 20% speedup for blending and 100% for color filling.
+ The blending can probably be optimized even more.
+
+2009-12-13 13:19:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/id3demux/id3v2frames.c:
+ id3demux: prefer two letter ISO 639-1 code for extended comment
+
+2009-12-13 13:10:12 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: fix up language code extraction some more
+ Quicktime uses ISO 639-2 for language codes, but GST_TAG_LANGUAGE
+ is supposed to hold a ISO 639-1 code, so convert as needed using
+ the new API from -base.
+ See #602126.
+
+2009-12-13 12:45:22 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-mux.c:
+ matroska: fix language code writing and extraction
+ Matroska uses three-letter ISO 639-2B codes, but GST_TAG_LANGUAGE is
+ supposed to contain two-letter ISO 639-1 codes, so use new language
+ code mapping functions in -base to convert between those two as
+ needed.
+ Fixes #505823.
+
+2009-12-07 20:54:07 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: minor debug message changes
+ Fix up a few debug messages so that it's clearer what they mean.
+
+2009-12-12 17:44:04 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ Revert "qtdemux: Correctly parse classification tags"
+ This reverts commit cd883aa60c1133196a6ae052884d15c295c37dde.
+ Previous code was correct, 4 is due to table and language code,
+ not only language code
+
+2009-12-12 16:28:36 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Correctly parse classification tags
+ In clsf atoms, the language code is 2 bytes long, not 4.
+
+2009-12-12 16:55:13 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/videomixer.c:
+ videomixer: Dequeue current buffer on FLUSH_STOP and don't unref NULL buffers
+ ... NULL buffers shouldn't really happen anymore when popping the
+ buffer from GstCollectPads but better check for this and print a warning.
+
+2009-12-11 13:11:12 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/blend_i420.c:
+ videomixer: Fix stupid mistake in last commit
+
+2009-12-11 12:35:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/blend_i420.c:
+ videomixer: Don't do floating point math in the inner processing loop for I420 blending
+
+2009-12-10 18:43:44 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: handle NULL and empty transport strings
+ When an RTSP extension returns NULL or an empty transport string, just ignore it
+ and try to get the next possible transport. Fixes playback of RealMedia streams.
+
+2009-12-10 18:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: install event function on internal RTCP pad
+ Install a custom event function on the internal RTCP pad so that we can reply
+ TRUE to a latency event.
+
+2009-12-10 10:48:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/blend_ayuv.c:
+ * gst/videomixer/blend_bgra.c:
+ * gst/videomixer/blend_rgb.c:
+ videomixer: Remove wrong comments, copied from the I420 blend function
+
+2009-12-09 21:15:07 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/videomixer.c:
+ videomixer: The queued duration is a signed integer
+ ...and it will really be negative sometimes.
+
+2009-12-09 21:03:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/videomixer.c:
+ videomixer: Only pop buffers from collectpads after they're fully consumed
+ This decreases latency and memory usage because new buffers are only
+ accepted by collectpads if there's no queued buffer.
+
+2009-12-09 20:42:44 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-demux.h:
+ matroskademux: Clean up position/duration handling
+ Also use the last end time for closing the segment, not the
+ start time of the last buffer.
+
+2009-12-09 16:50:02 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Close the segment on EOS if the real duration is known
+
+2009-12-09 16:46:18 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Update duration if current buffer is already after the old duration
+
+2009-12-09 16:43:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Drop buffers that are after segment stop
+ ...and if this happened for all streams go EOS.
+
+2009-12-09 16:41:04 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Fix position tracking and sending of filler segments
+
+2009-12-09 16:15:09 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/videomixer/videomixer.c:
+ videomixer: Use gst_util_uint64_scale_int() for fps to seconds per frame calculations
+
+2009-12-08 17:34:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Keep the segment stop position for update newsegment events
+
+2009-12-04 14:42:49 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * configure.ac:
+ * ext/Makefile.am:
+ * ext/ladspa/Makefile.am:
+ * ext/ladspa/gstladspa.c:
+ * ext/ladspa/gstladspa.h:
+ * ext/ladspa/gstsignalprocessor.c:
+ * ext/ladspa/gstsignalprocessor.h:
+ * ext/ladspa/load.c:
+ * ext/ladspa/search.c:
+ * ext/ladspa/utils.h:
+ ladspa: Remove the sources from gst-plugins-good
+ It's disabled anyway and the latest version of it is in
+ gst-plugins-bad. Fixes bug #603779.
+
+2009-12-04 13:50:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: init current_entry in push mode
+ Set the current_entry to 0 (instead of -1) in push mode so that we correctly
+ calculate the current frame number and timestamp.
+ Add some more debug info and fic the duration debug.
+
+2009-12-04 11:14:03 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: fix major memory leak when playing back rtsp video streams
+ Don't forget to unref QoS, navigation and latency events when
+ dropping them.
+
+2009-12-03 08:58:08 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: only send pending tags with newsegment events
+ Send pending tags only from the streaming thread, just after we've sent
+ the newsegment event, not with e.g. flush-start. This not only does the
+ right thing, but also makes sure we're not trampling over variables set
+ up in the streaming thread from the seeking thread in case someone tries
+ to issue a seek just as the demuxer is parsing the headers.
+ Fixes #601617. Spotted by Ognyan Tonchev.
+
+2009-12-03 17:49:55 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: fix debug message printf args
+ Fixes debug message printf format to make it build in mac's gcc
+
+2009-12-02 13:33:20 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * ext/shout2/gstshout2.c:
+ shout2: Convert delay correctly
+ Use GST_MSECOND to convert delay in msecs to nanosecs
+ Fixes #603547
+
+2009-12-01 19:24:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/jpeg/gstjpegdec.c:
+ jpegdec: reset segment info after flush
+ Reset the segment info after a flush. We use the segment for handling QoS and if
+ we don't reset the segment, QoS is basically disabled after a flushing seek.
+
+2009-12-01 15:07:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 87bf428 to 47cb23a
+
+2009-12-01 14:15:46 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From da4c75c to 87bf428
+
+2009-11-30 15:59:50 +0100 Aurelien Grimaud <gstelzz at yahoo dot fr>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: avoid buffer ref/unref pairs for CSRCs
+ We ref the buffer before pushing it downstream in order to get the CSRCs of it
+ after pushing. This causes performance problems when downstream elements want to
+ change the metadata because the buffer needs to be subbuffered.
+ Instead, read and store the CSRCs of the buffer in an array before pushing it
+ and process the array after pushing the buffer. This allows us to remove the
+ ref/unref pair.
+ Fixes #603376
+
+2009-11-28 19:23:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/shout2/gstshout2.c:
+ * ext/shout2/gstshout2.h:
+ shout2: use gstpoll for timeouts
+ Use our own GstPoll based timeout instead of the shout sleep so that we can
+ interrupt when doing a state change and shutting down.
+ Fixes #602887
+
+2009-11-28 12:25:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ check: fix jitterbuffer check
+ Make sure we set a base_time on the element.
+ Fix the timeout to at least twice the jitterbuffer latency.
+ Enable previously failing tests.
+ Remove impossible checks.
+
+2009-11-27 18:55:20 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From 53a2485 to da4c75c
+
+2009-11-26 16:14:30 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtph264depay.h:
+ rtph264depay: optionally merge NALUs into Access Units
+ ... which may be expected/desired by some downstream decoders
+ (and spec-wise highly recommended for at least non-bytestream mode).
+
+2009-11-26 17:29:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: fix timestamp datatype
+
+2009-11-25 10:38:23 -0600 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: avoid using wrong clock-rate
+ Check for a valid clock-rate before attempting to estimate the npt
+ stop time.
+
+2009-11-25 10:37:30 -0600 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: fix typo in comments
+
+2009-11-25 16:05:10 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffertest: add one more test and file a bug now
+ CHange the backwards test to always send first buffer first to have a define
+ basetime. Add another test that sends buffers backwards to assert that only
+ first sent buffer is keep and used as basetime. Disabled those tests still,
+ as its not passing/failing consitently and file a bug for jitterbuffer.
+
+2009-11-25 10:17:34 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ jitterbuffertest: improve the test
+ the tests are a bit more solid now but still not produce reliable results.
+ Wonder if they are still flawky or if its a bug in jitterbuffer.
+
+2009-11-24 11:13:06 -0800 Michael Smith <msmith@songbirdnest.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: return error message on windows too.
+
+2009-11-24 10:58:49 -0800 Michael Smith <msmith@songbirdnest.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: first phase of fixing up error reporting for windows.
+
+2009-10-30 03:13:54 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/avi/gstavimux.c:
+ avimux: also set the suggested buf size for audio
+ We were only setting the suggested buf size for video,
+ we can set it for audio as well.
+ This and 195e14529d80ef318ce3a778c1995efb11f266cd
+ fix an issue that prevented seeking on large avi files
+ on WMP (non-recent versions).
+
+2009-11-04 16:10:23 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/avi/gstavimux.c:
+ * gst/avi/gstavimux.h:
+ avimux: fix indx duration for PCM audio
+ GstBuffers for PCM audio usually contains more than
+ 1 sample, we need to get the total number of samples to set
+ the indx duration.
+
+2009-11-04 16:04:10 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/avi/gstavimux.c:
+ avimux: Audio buffers should be picked earlier
+ Adds a 0.5s advantage for audio buffers to being
+ picked earlier for muxing.
+
+2009-11-24 16:40:19 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Fix push mode by making sure stbl information is available in next_entry_size ()
+
+2009-11-24 16:35:20 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Fix order of arguments in log message
+
+2009-11-24 15:51:21 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * ext/jpeg/gstjpegenc.c:
+ jpegenc: fix spelling in comment
+
+2009-11-23 17:58:17 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * common:
+ build system: Fix wrongly committed change to common/
+
+2009-11-10 10:26:07 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Ease debugging by removing a goto for an error message
+
+2009-11-14 15:52:09 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * common:
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Parse per sample rather than all at once but build complete index when seeking
+
+2009-11-04 17:31:15 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Save atom data for later use so it doesn't get freed after initial parsing
+
+2009-11-06 11:00:04 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Parse from the previously parsed sample up to sample n
+
+2009-11-04 17:04:22 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Make qtdemux_parse_samples () parse up to n samples
+
+2009-10-28 17:49:02 +0000 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Separate off stbl sub-atom initialisation
+
+2009-10-26 22:42:36 +0000 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Move variables into context in preparation for refactorisation
+
+2009-10-26 20:36:08 +0000 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Fix bug where stps is never parsed due to logic error
+
+2009-11-04 17:31:15 +0100 Robert Swain <robert.swain@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: Port ctts from Gnode * to GstByteReader
+
+2009-10-23 13:06:44 +0100 Robert Swain <robert.swain@gmail.com>
+
+ * gst/qtdemux/qtatomparser.h:
+ * gst/qtdemux/qtdemux.c:
+ * gst/qtdemux/qtdemux_dump.c:
+ * gst/qtdemux/qtdemux_dump.h:
+ * gst/qtdemux/qtdemux_types.h:
+ qtdemux: Switch from QtAtomParser to GstByteReader
+
+2009-11-23 12:53:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: fix typo and grammar
+
+2009-11-20 10:30:00 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: fix typo in mode enum description
+
+2009-11-20 11:25:49 +0200 Stefan Kost <ensonic@users.sf.net>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ docs: more links and better short description
+ Fix spelling of GstRtpSsrcDemux to get it linked. Add more links. Change
+ the short description to be more meaningful.
+
+2009-11-20 09:58:26 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * tests/check/elements/wavpackparse.c:
+ wavpackparse: Fix unit test for recent position reporting changes
+
+2009-11-19 16:09:38 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/wavpack/gstwavpackparse.c:
+ wavpackparse: After pushing a frame, update last_stop to the end of the frame
+ This improves position reporting, especially because of the fact that
+ WavPack frames are usually 0.5-1.0 seconds long.
+
+2009-11-19 16:08:33 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * ext/wavpack/gstwavpackparse.c:
+ wavpackparse: Allow pulling the last WavPack frame of a file
+ Because of a >= instead of a >, that last frame of a WavPack file
+ would never be parsed in pull mode.
+
+2009-11-19 10:30:43 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * common:
+ Automatic update of common submodule
+ From 0702fe1 to 53a2485
+
+2009-10-29 08:29:38 -0300 Thiago Santos <thiago.sousa.santos@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ * gst/qtdemux/qtdemux_fourcc.h:
+ qtdemux: Add more fields to SVQ3 caps
+ qtdemux only added the whole stsd atom as 'codec_data'
+ in its output caps for SVQ3. This patch makes it add
+ the SEQH (inside a SMI atom) and a gamma field (taken
+ from the gama atom) if available.
+ Fixes #587922
+
+2009-11-18 17:55:42 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/wavenc/gstwavenc.c:
+ wavenc: Raise rank of muxer to PRIMARY
+
+2009-11-18 17:54:16 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/y4m/gsty4mencode.c:
+ y4m: Raise rank of encoder to PRIMARY
+
+2009-11-18 17:54:02 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/law/alaw.c:
+ * gst/law/mulaw.c:
+ law: Raise rank of encoders to PRIMARY
+
+2009-11-12 19:11:18 +0000 Bastien Nocera <hadess@hadess.net>
+
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ Add user-id and user-pw properties
+ So that one doesn't need to modify the URL to have access
+ to authenticated RTSP streams.
+ fixes #601728
+
+2009-11-18 12:22:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: use acquired flag when checking valid state
+ Use the acquired field of the ringbuffer in get_time to know when we are in an
+ invalid state. We don't clear the rate flag when releasing the ringbuffer so
+ this values is not usable.
+ Avoids some error messages being posted because the pulseaudio connection is
+ down.
+
+2009-11-18 10:17:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+ * configure.ac:
+ configure: bump core requirement to 0.10.25.1 as well
+ Make implicit requirement explicit.
+
+2009-11-18 12:53:44 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: fix bogus memory chunk size check
+
+2009-11-18 12:01:52 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: implement some more callbacks
+ Implement some more callbacks for debugging purposes.
+
+2009-11-11 15:50:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: release lock before emiting signals
+ Release the jbuf lock before emiting the request-pt-map signal to avoid
+ deadlocks. We also need to catch the shutdown case when locking again.
+ Fixes #593354
+
+2009-11-11 11:59:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpbvdepay.c:
+ * gst/rtp/gstrtpbvdepay.h:
+ rtp: add BroadcomVoice depayloader
+
+2009-11-11 11:38:36 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpbvpay.c:
+ rtpbvpay: add rfc reference
+
+2009-11-11 11:37:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpbvpay.c:
+ * gst/rtp/gstrtpbvpay.h:
+ rtp: add BroadcomVoice payloader
+
+2009-11-09 12:17:34 +0100 Jan Urbański <wulczer@wulczer.org>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: properly finish the ECMA array
+ The ECMA array with the file index was missing a mandatory end marker.
+ Fixes bug #601242.
+
+2009-11-18 02:15:15 +0000 Jan Schmidt <thaytan@noraisin.net>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ Use new still-frame API from gst-plugins-base
+
+2009-11-18 02:14:46 +0000 Jan Schmidt <thaytan@noraisin.net>
+
+ * configure.ac:
+ Bump gst-plugins-base requirement to 0.10.25.1
+
+2009-11-17 17:59:13 -0800 Michael Smith <msmith@songbirdnest.com>
+
+ * gst/qtdemux/qtdemux.c:
+ qtdemux: identify IMA adpcm in qt properly.
+
+2009-11-18 01:27:37 +0000 Jan Schmidt <thaytan@noraisin.net>
+
+ * configure.ac:
+ * win32/common/config.h:
+ Back to development -> 0.10.17.1
+
+2009-11-17 01:53:08 +0000 Jan Schmidt <thaytan@noraisin.net>
+
+ * gst-plugins-good.doap:
+ Add release 0.10.17 to the doap file
+
=== release 0.10.17 ===
-2009-11-17 Jan Schmidt <jan.schmidt@sun.com>
+2009-11-17 01:25:30 +0000 Jan Schmidt <thaytan@noraisin.net>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 0.10.17, "They used to sparkle"
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/gst-plugins-good-plugins.interfaces:
+ * docs/plugins/gst-plugins-good-plugins.prerequisites:
+ * docs/plugins/gst-plugins-good-plugins.signals:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-annodex.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-efence.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-esdsink.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gamma.xml:
+ * docs/plugins/inspect/plugin-gconfelements.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-gstrtpmanager.xml:
+ * docs/plugins/inspect/plugin-halelements.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-monoscope.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-quicktime.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobalance.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videoflip.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ Release 0.10.17
2009-11-17 00:18:22 +0000 Jan Schmidt <thaytan@noraisin.net>
-This is GStreamer Good Plug-ins 0.10.17, "They used to sparkle"
+This is GStreamer Good Plug-ins 0.10.18, "Short Circuit"
+
+Changes since 0.10.17:
+
+ * v4l2src: implement GstURIHandler interface
+ * matroskamux: make index size configurable
+ * matroskademux: support push based mode
+ * matroskademux: improve stream synchronization
+ * flacdec: fix possible hanging in pull mode seeking
+ * flacdec: use a single decoder field for both push and pull mode
+ * flacenc: optionally add a seek table
+ * rtp: add BroadcomVoice payloader and depayloader
+ * rtp: add G.723 payloader and depayloader
+ * rtph264pay: add option to insert PPS/SPS in streams
+ * rtph264depay: optionally merge NALUs into Access Units
+ * rtspsrc: add user-id and user-pw properties; fix major memory leak
+ * avimux: many fixes, also better compatibility with Windows Media Player
+ * avidemux: per-stream index parsing (= much faster startup)
+ * qtdemux: progressive download support / seeking in push mode
+ * qtdemux: per sample parsing (= much faster start up)
+ * wavenc: Post warning if file hasn't been finalised properly
+ * videomixer: MMX optimisations and other improvements; implement basic QoS
+ * matroska, qtdemux, id3demux: fix language code writing and extraction
+
+Bugs fixed since 0.10.17:
+
+ * 609405 : [qtdemux] Issues when seeking with file with lots of tracks and edit lists
+ * 503582 : [avidemux] Extract date tag (contained in the IDIT chunk)
+ * 351595 : [flacenc] write seek tables
+ * 505823 : [matroskademux] language tags have wrong iso code
+ * 515073 : [goom] Update to goom2k4
+ * 539858 : not enough NEWSEGMENT events from matroskademux
+ * 554839 : [rtpbin] Automaticaly remove pads
+ * 582575 : [rtph263depay] dropping only part of key frames on lost fragmets
+ * 583367 : gstrtpL16pay ignores max-ptime property
+ * 583985 : [matroskamux] make index size configurable
+ * 587323 : rtpmp4vpay does not payload mp4v stream depayloaded with rtpmp4vdepay
+ * 593354 : rtpjitterbuffer sometimes outputs packets with timestamps in the past
+ * 595265 : SDES handling in RTPSource
+ * 597497 : can't play a redirecting .mov file via playbin
+ * 597823 : Add rtpg723pay plugin
+ * 599300 : [qtdemux] Doesn't populate video bitrate field
+ * 601143 : v4l2src: add GstURIHandler interface
+ * 601242 : [flvmux] ECMA array with file index lacks final 0x09 byte
+ * 601728 : [rtspsrc] Add username/password properties
+ * 602231 : Deadlock between rtpjitterbuffer and gstrtpbin
+ * 602508 : qtdemux: Parse stbl atom per sample instead of all at once
+ * 602887 : shout2send element won't change from PLAYING state to NULL
+ * 602940 : jitterbuffer is racy determining basetime
+ * 603376 : rtpsession : g_type_create_instance performance issue : avoid buffer ref
+ * 603471 : [flacdec] not timestamping output buffers
+ * 603547 : shout2send plugin sends data too fast
+ * 603779 : [ladspa] Remove ladspa plugin code
+ * 604352 : [rganalysis] miscomputes timestamps
+ * 604611 : [qtdemux] Provides invalid ALAC codec data
+ * 604679 : videomixer MMX code doesn't build on fedora12
+ * 604814 : videomixer make error
+ * 604872 : [udpsink] Add missing 'gssize len' parameter to g_convert()
+ * 604913 : rtph264pay/NALU/rtph264depay
+ * 605222 : Mobile Youtube RTSP streams time out at EOS
+ * 605269 : [shout2][patch] Setting public flag
+ * 605447 : Unable to play Real Audio stream for radioBERLIN.
+ * 605882 : rtpg723pay is incorrect
+ * 606198 : rtph264pay is causing alignment trap on ARM arch
+ * 606438 : multiudpsink: warningfixes for windows
+ * 606692 : Incorrect Center Frequency For Band3
+ * 606807 : audioamplify: allow negative amplifications
+ * 607353 : rtph264pay & base: Don't crash if the other side specifies the profile-level-id
+ * 607440 : [wavenc] should post warning if the file isn't finished properly on pipeline shutting down
+ * 607718 : [qtdemux] Infinite loop doing negative rate playback for single audio stream
+ * 607949 : [avidemux] regression in stop position for mp3 streams
+ * 608209 : [videomixer] blend_mmx.h:173: Error: can't encode register '%ah' in an instruction requiring REX prefix
+ * 608255 : [speex] speexenc crash and leaks in rtpspeexpay and speexdec
+ * 608268 : [flvmux] index timestamps should be in seconds, not milliseconds
+ * 608629 : [pngdec] png_set_gray_1_2_4_to_8() removed in libpng > = 1.4.0
+ * 608671 : [mkv] issues when seeking
+ * 608990 : [qtdemux] Segment start timestamps can be broken
+ * 609107 : [qtdemux] Unknown atoms should also be skipped when looking for moov
+ * 598610 : [matroskademux] Support push mode operation
+ * 594381 : audiofirfilter: Implement FFT convolution
Changes since 0.10.16:
-Release notes for GStreamer Good Plug-ins 0.10.17 "They used to sparkle"
+Release notes for GStreamer Good Plug-ins 0.10.18 "Short Circuit"
Features of this release
- * RTP improvements
- * Support automatic cropping in videobox
- * Add TTL multicast UDP property
- * AVI demux push mode fixes and performance improvements
- * Support large and unusual chunks sizes in wav
- * Quicktime demuxer improvements
- * JPEG decode fixes and speedups
- * Support interlaced Y4M file output
- * DV demuxer improvements
- * Pulseaudio fixes and improvements
- * Support Pulseaudio PLAY/PAUSE requests
- * speexdec improvements
- * FLV demuxer improvements
- * Fix audio noise in the Equalizer plugin, and other improvements
- * Fix compilation on OS/X Snow Leopard
- * AVI muxer fixes
- * Support MPEG V4L2 devices and improve timestamping
- * Better jpeg2k support
- * Many other bug fixes and improvements
+ * v4l2src: implement GstURIHandler interface
+ * matroskamux: make index size configurable
+ * matroskademux: support push based mode
+ * matroskademux: improve stream synchronization
+ * flacdec: fix possible hanging in pull mode seeking
+ * flacdec: use a single decoder field for both push and pull mode
+ * flacenc: optionally add a seek table
+ * rtp: add BroadcomVoice payloader and depayloader
+ * rtp: add G.723 payloader and depayloader
+ * rtph264pay: add option to insert PPS/SPS in streams
+ * rtph264depay: optionally merge NALUs into Access Units
+ * rtspsrc: add user-id and user-pw properties; fix major memory leak
+ * avimux: many fixes, also better compatibility with Windows Media Player
+ * avidemux: per-stream index parsing (= much faster startup)
+ * qtdemux: progressive download support / seeking in push mode
+ * qtdemux: per sample parsing (= much faster start up)
+ * wavenc: Post warning if file hasn't been finalised properly
+ * videomixer: MMX optimisations and other improvements; implement basic QoS
+ * matroska, qtdemux, id3demux: fix language code writing and extraction
Bugs fixed in this release
- * 597848 : " Media Player Classic " won't play certain files produced by avimux.
- * 588245 : TTL is never applied with udpsink/udpmultisink
- * 368681 : avimux + vbr lame always out of sync
- * 458629 : [avidemux] high memory usage for many index entries
- * 561825 : Problem with RTCP thread using freed objects
- * 581334 : [qtdemux] Add support for embedded subtitles
- * 582238 : [videobox] Add support for autocrop to caps
- * 590362 : [v4l2src] x264enc ! qtmux fails because of missing frame duration
- * 591713 : [y4menc] interlaced support
+ * 609405 : [qtdemux] Issues when seeking with file with lots of tracks and edit lists
+ * 503582 : [avidemux] Extract date tag (contained in the IDIT chunk)
+ * 351595 : [flacenc] write seek tables
+ * 505823 : [matroskademux] language tags have wrong iso code
+ * 515073 : [goom] Update to goom2k4
+ * 539858 : not enough NEWSEGMENT events from matroskademux
+ * 554839 : [rtpbin] Automaticaly remove pads
+ * 582575 : [rtph263depay] dropping only part of key frames on lost fragmets
+ * 583367 : gstrtpL16pay ignores max-ptime property
+ * 583985 : [matroskamux] make index size configurable
+ * 587323 : rtpmp4vpay does not payload mp4v stream depayloaded with rtpmp4vdepay
* 593354 : rtpjitterbuffer sometimes outputs packets with timestamps in the past
- * 593391 : [rtpsession] : rtp_session_on_timeout : Invalid read of size 4
- * 593688 : effectv can no longer be compiled with gcc 3
- * 593757 : [qtdemux] Lack of support for QualComm PureVoice
- * 593764 : [v4l2src] format ordering: put emulated formats behind native formats
- * 593955 : rtpjitterbuffer: clock_rate can change between its check and its use
- * 594039 : missing unref in rtpsource / leak
- * 594133 : [rtspsrc] leaks authentication info
- * 594247 : missing math.h include in rtpjpegdepay
- * 594248 : Use locked-state on internal rtp-bin to avoid shutdown-state-race
- * 594251 : Avoid throwing out reordered packets with the same timestamp
- * 594253 : jitterbuf: Only post a warning of clock-rate changed if it is changed from something initialized
- * 594254 : propagate the pt-type-changed signal
- * 594283 : rtpbin: make free_session() remove dangling stream references
- * 594298 : Check if libsoup has SSL support before running HTTPS test in souphttpsrc testsuite
- * 594490 : gstrtpbin always uses pt to demux
- * 594520 : multipartmux: mark data buffer as delta-unit
- * 594599 : videobox: converts AYUV to I420 incorrectly
- * 594663 : Patch for multifilesink
- * 594691 : rtph263pay: leak
- * 595029 : pulse elements fail to connect to pulse 0.9.9
- * 595220 : gstreamer crashes on pulseaudio latency change
- * 595231 : [pulsesink] Lowers volume after every new track
- * 595888 : qtdemux plugin should not return value from void function
- * 595897 : Problem linking videomixer
- * 595942 : [qtdemux] issue with corrupted 3gp file
- * 596319 : [qtdemux] fails to parse pixel aspect ratio data
- * 597091 : [flvdemux] not outputting no-more-pads causes playbin2 to fail badly on streamed single-stream flv
- * 597214 : [avidemux] Fix printf formats to avoid warnings in avidemux
- * 597348 : [qtdemux] Cast variables passed to printf to avoid warnings about incorrect formats
- * 597351 : [jpegdec] segfaults on a specific picture
- * 597397 : equalizer is non deterministic
- * 597463 : [pulsesrc] has no lower bound for fragment size
- * 597601 : [pulsesink] needs to take control of minreq value
- * 597730 : osssrc rank should be secondary, just like osssink
- * 597847 : Windows Media Player won't play large files produced by avimux
- * 597867 : Plugins good do not build on Ubuntu Hardy (kernel 2.6.24)
- * 598377 : rtpmanager: only forward the lost event to the last seen payloadnumber
- * 598517 : [jpegdec] Regression supporting 4:2:2 jpeg videos
- * 598810 : wavenc: Fix buffer offset by moving length incrementation
- * 598933 : [pulse] Fix the StreamVolume interface not being advertised
- * 601381 : v4l2: Make sure to initialize variables before using them
+ * 595265 : SDES handling in RTPSource
+ * 597497 : can't play a redirecting .mov file via playbin
+ * 597823 : Add rtpg723pay plugin
+ * 599300 : [qtdemux] Doesn't populate video bitrate field
+ * 601143 : v4l2src: add GstURIHandler interface
+ * 601242 : [flvmux] ECMA array with file index lacks final 0x09 byte
+ * 601728 : [rtspsrc] Add username/password properties
+ * 602231 : Deadlock between rtpjitterbuffer and gstrtpbin
+ * 602508 : qtdemux: Parse stbl atom per sample instead of all at once
+ * 602887 : shout2send element won't change from PLAYING state to NULL
+ * 602940 : jitterbuffer is racy determining basetime
+ * 603376 : rtpsession : g_type_create_instance performance issue : avoid buffer ref
+ * 603471 : [flacdec] not timestamping output buffers
+ * 603547 : shout2send plugin sends data too fast
+ * 603779 : [ladspa] Remove ladspa plugin code
+ * 604352 : [rganalysis] miscomputes timestamps
+ * 604611 : [qtdemux] Provides invalid ALAC codec data
+ * 604679 : videomixer MMX code doesn't build on fedora12
+ * 604814 : videomixer make error
+ * 604872 : [udpsink] Add missing 'gssize len' parameter to g_convert()
+ * 604913 : rtph264pay/NALU/rtph264depay
+ * 605222 : Mobile Youtube RTSP streams time out at EOS
+ * 605269 : [shout2][patch] Setting public flag
+ * 605447 : Unable to play Real Audio stream for radioBERLIN.
+ * 605882 : rtpg723pay is incorrect
+ * 606198 : rtph264pay is causing alignment trap on ARM arch
+ * 606438 : multiudpsink: warningfixes for windows
+ * 606692 : Incorrect Center Frequency For Band3
+ * 606807 : audioamplify: allow negative amplifications
+ * 607353 : rtph264pay & base: Don't crash if the other side specifies the profile-level-id
+ * 607440 : [wavenc] should post warning if the file isn't finished properly on pipeline shutting down
+ * 607718 : [qtdemux] Infinite loop doing negative rate playback for single audio stream
+ * 607949 : [avidemux] regression in stop position for mp3 streams
+ * 608209 : [videomixer] blend_mmx.h:173: Error: can't encode register '%ah' in an instruction requiring REX prefix
+ * 608255 : [speex] speexenc crash and leaks in rtpspeexpay and speexdec
+ * 608268 : [flvmux] index timestamps should be in seconds, not milliseconds
+ * 608629 : [pngdec] png_set_gray_1_2_4_to_8() removed in libpng > = 1.4.0
+ * 608671 : [mkv] issues when seeking
+ * 608990 : [qtdemux] Segment start timestamps can be broken
+ * 609107 : [qtdemux] Unknown atoms should also be skipped when looking for moov
+ * 598610 : [matroskademux] Support push mode operation
+ * 594381 : audiofirfilter: Implement FFT convolution
Download
Contributors to this release
* Alessandro Decina
- * Andy Wingo
* Arnout Vandecappelle
+ * Arun Raghavan
+ * Aurelien Grimaud
* Bastien Nocera
- * Brian Cameron
- * Christian F.K. Schaller
- * David Henningsson
- * David Schleef
+ * Branko Čibej
+ * David Hoyt
* Edward Hervey
- * Gabriel Millaire
* Havard Graff
- * Håvard Graff
* Jan Schmidt
- * Jarkko Palviainen
- * Josep Torra
- * Laurent Glayal
- * Lennart Poettering
- * Marc Leeman
- * Marc-André Lureau
+ * Jan Urbański
+ * Jonathan Conder
+ * Kipp Cannon
+ * Marco Ballesio
* Mark Nauwelaerts
- * Marvin Schmidt
* Michael Smith
* Olivier Crête
- * Pau Garcia i Quiles
- * Peter Kjellerstedt
- * Priit Laes
- * René Stadler
+ * Pascal Buhler
+ * Peter van Hardenberg
* Robert Swain
+ * Robert Weidlich
+ * Roland Krikava
* Sebastian Dröge
- * Sjoerd Simons
* Stefan Kost
- * Stephen Jungels
- * Stig Sandnes
* Thiago Santos
+ * Tiago Katcipis
* Tim-Philipp Müller
* Wim Taymans
- * Zaheer Merali
\ No newline at end of file
dnl initialize autoconf
dnl releases only do -Wall, git and prerelease does -Werror too
dnl use a three digit version number for releases, and four for git/pre
-AC_INIT(GStreamer Good Plug-ins, 0.10.17.3,
+AC_INIT(GStreamer Good Plug-ins, 0.10.18,
http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer,
gst-plugins-good)
AM_PROG_LIBTOOL
dnl *** required versions of GStreamer stuff ***
-GST_REQ=0.10.25.2
-GSTPB_REQ=0.10.25.2
+GST_REQ=0.10.26
+GSTPB_REQ=0.10.26
dnl *** autotools stuff ****
<TYPE>gdouble</TYPE>
<RANGE>[-24,12]</RANGE>
<FLAGS>rw</FLAGS>
-<NICK>227 Hz</NICK>
-<BLURB>gain for the frequency band 227 Hz, ranging from -24 dB to +12 dB.</BLURB>
+<NICK>237 Hz</NICK>
+<BLURB>gain for the frequency band 237 Hz, ranging from -24 dB to +12 dB.</BLURB>
<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
<NAME>GstUDPSrc::sockfd</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Socket Handle</NICK>
<BLURB>Socket to use for UDP reception. (-1 == allocate).</BLURB>
<ARG>
<NAME>GstUDPSrc::sock</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>r</FLAGS>
<NICK>Socket Handle</NICK>
<BLURB>Socket currently in use for UDP reception. (-1 = no socket).</BLURB>
</ARG>
<ARG>
+<NAME>GstRTSPSrc::user-id</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>user-id</NICK>
+<BLURB>RTSP location URI user id for authentication.</BLURB>
+<DEFAULT>NULL</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstRTSPSrc::user-pw</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>user-pw</NICK>
+<BLURB>RTSP location URI user password for authentication.</BLURB>
+<DEFAULT>NULL</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstRTPDec::skip</NAME>
<TYPE>gint</TYPE>
<RANGE></RANGE>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
-<NICK>mesage</NICK>
+<NICK>message</NICK>
<BLURB>Post a level message for each passed interval.</BLURB>
<DEFAULT>TRUE</DEFAULT>
</ARG>
</ARG>
<ARG>
+<NAME>GstShout2send::public</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>public</NICK>
+<BLURB>If the stream should be listed on the server's stream directory.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstSpeexDec::enh</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<ARG>
<NAME>GstDV1394Src::port</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,16]</RANGE>
+<RANGE>[G_MAXULONG,16]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Port</NICK>
<BLURB>Port number (-1 automatic).</BLURB>
</ARG>
<ARG>
+<NAME>GstFlacEnc::seekpoints</NAME>
+<TYPE>gint</TYPE>
+<RANGE>>= -2147483647</RANGE>
+<FLAGS>rwx</FLAGS>
+<NICK>Seekpoints</NICK>
+<BLURB>Add SEEKTABLE metadata (if > 0, number of entries, if < 0, interval in sec).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstOssSink::device</NAME>
<TYPE>gchar*</TYPE>
<RANGE></RANGE>
</ARG>
<ARG>
+<NAME>GstMatroskaMux::min-index-interval</NAME>
+<TYPE>gint64</TYPE>
+<RANGE>>= 0</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Minimum time between index entries</NICK>
+<BLURB>An index entry is created every so many nanoseconds.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstTest::allowed-timestamp-deviation</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>allowed timestamp deviation</NICK>
<BLURB>allowed average difference in usec between timestamp of next buffer and expected timestamp from analyzing last buffer.</BLURB>
<ARG>
<NAME>GstTest::buffer-count</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>r</FLAGS>
<NICK>buffer count</NICK>
<BLURB>number of buffers in stream.</BLURB>
<ARG>
<NAME>GstTest::expected-buffer-count</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>expected buffer count</NICK>
<BLURB>expected number of buffers in stream.</BLURB>
<ARG>
<NAME>GstTest::expected-length</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>expected length</NICK>
<BLURB>expected length of stream.</BLURB>
<ARG>
<NAME>GstTest::length</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>r</FLAGS>
<NICK>length</NICK>
<BLURB>length of stream.</BLURB>
<ARG>
<NAME>GstTest::timestamp-deviation</NAME>
<TYPE>gint64</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>r</FLAGS>
<NICK>timestamp deviation</NICK>
<BLURB>average difference in usec between timestamp of next buffer and expected timestamp from analyzing last buffer.</BLURB>
<ARG>
<NAME>GstBreakMyData::set-to</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,255]</RANGE>
+<RANGE>[G_MAXULONG,255]</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>set-to</NICK>
<BLURB>set changed bytes to this value (-1 means random value.</BLURB>
<ARG>
<NAME>GstDynUDPSink::sockfd</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,32767]</RANGE>
+<RANGE>[G_MAXULONG,32767]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>socket handle</NICK>
<BLURB>Socket to use for UDP sending. (-1 == allocate).</BLURB>
<ARG>
<NAME>GstMultiUDPSink::sock</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>r</FLAGS>
<NICK>Socket Handle</NICK>
<BLURB>Socket currently in use for UDP sending. (-1 == no socket).</BLURB>
<ARG>
<NAME>GstMultiUDPSink::sockfd</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Socket Handle</NICK>
<BLURB>Socket to use for UDP sending. (-1 == allocate).</BLURB>
<ARG>
<NAME>GstMultiUDPSink::qos-dscp</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,63]</RANGE>
+<RANGE>[G_MAXULONG,63]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>QoS diff srv code point</NICK>
<BLURB>Quality of Service, differentiated services code point (-1 default).</BLURB>
<ARG>
<NAME>GstAudioAmplify::amplification</NAME>
<TYPE>gfloat</TYPE>
-<RANGE>>= 0</RANGE>
+<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Amplification</NICK>
<BLURB>Factor of amplification.</BLURB>
<ARG>
<NAME>GstV4l2Src::device-fd</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>r</FLAGS>
<NICK>File descriptor</NICK>
<BLURB>File descriptor of the device.</BLURB>
<ARG>
<NAME>GstAudioWSincBand::length</NAME>
<TYPE>gint</TYPE>
-<RANGE>[3,50000]</RANGE>
+<RANGE>[3,256000]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Length</NICK>
<BLURB>Filter kernel length, will be rounded to the next odd number.</BLURB>
<ARG>
<NAME>GstAudioWSincLimit::length</NAME>
<TYPE>gint</TYPE>
-<RANGE>[3,50000]</RANGE>
+<RANGE>[3,256000]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Length</NICK>
<BLURB>Filter kernel length, will be rounded to the next odd number.</BLURB>
<ARG>
<NAME>GstRndBufferSize::max</NAME>
<TYPE>glong</TYPE>
-<RANGE>>= 1</RANGE>
+<RANGE>[1,G_MAXINT]</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>maximum</NICK>
<BLURB>maximum buffer size.</BLURB>
<ARG>
<NAME>GstRndBufferSize::min</NAME>
<TYPE>glong</TYPE>
-<RANGE>>= 0</RANGE>
+<RANGE>[0,G_MAXINT]</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>mininum</NICK>
<BLURB>mininum buffer size.</BLURB>
<ARG>
<NAME>GstRndBufferSize::seed</NAME>
<TYPE>gulong</TYPE>
-<RANGE></RANGE>
+<RANGE><= G_MAXUINT</RANGE>
<FLAGS>rwx</FLAGS>
<NICK>random number seed</NICK>
<BLURB>seed for randomness (initialized when going from READY to PAUSED).</BLURB>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>profile-level-id</NICK>
-<BLURB>The base64 profile-level-id to set in out caps (set to NULL to extract from stream).</BLURB>
+<BLURB>The base64 profile-level-id to set in the sink caps (deprecated).</BLURB>
<DEFAULT>NULL</DEFAULT>
</ARG>
</ARG>
<ARG>
+<NAME>GstRtpH264Pay::config-interval</NAME>
+<TYPE>guint</TYPE>
+<RANGE><= 3600</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>SPS PPS Send Interval</NICK>
+<BLURB>Send SPS and PPS Insertion Interval in seconds (sprop parameter sets will be multiplexed in the data stream when detected.) (0 = disabled).</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>ladspa-hardLimiter::Residue-level</NAME>
<TYPE>gfloat</TYPE>
<RANGE>[0,1]</RANGE>
</ARG>
<ARG>
+<NAME>GstRtpH264Depay::access-unit</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Access Unit</NICK>
+<BLURB>Merge NALU into AU (picture).</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstAudioKaraoke::filter-band</NAME>
<TYPE>gfloat</TYPE>
<RANGE>[0,441]</RANGE>
<ARG>
<NAME>GstHDV1394Src::port</NAME>
<TYPE>gint</TYPE>
-<RANGE>[-1,16]</RANGE>
+<RANGE>[G_MAXULONG,16]</RANGE>
<FLAGS>rw</FLAGS>
<NICK>Port</NICK>
<BLURB>Port number (-1 automatic).</BLURB>
<FLAGS>rw</FLAGS>
<NICK>Mode</NICK>
<BLURB>Deinterlace Mode.</BLURB>
-<DEFAULT>Enfore deinterlacing</DEFAULT>
+<DEFAULT>Force deinterlacing</DEFAULT>
</ARG>
<ARG>
</ARG>
<ARG>
+<NAME>GstRtpBin::autoremove</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Auto Remove</NICK>
+<BLURB>Automatically removed timed out sources.</BLURB>
+<DEFAULT>FALSE</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstRtpJitterBuffer::do-lost</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<ARG>
<NAME>GstV4l2Sink::device-fd</NAME>
<TYPE>gint</TYPE>
-<RANGE>>= -1</RANGE>
+<RANGE>>= G_MAXULONG</RANGE>
<FLAGS>r</FLAGS>
<NICK>File descriptor</NICK>
<BLURB>File descriptor of the device.</BLURB>
GstElement
GstBin
GstPipeline
- GstHalAudioSink
- GstHalAudioSrc
GstGConfVideoSink
GstGConfVideoSrc
GstSwitchSink
GstGConfAudioSink
GstGConfAudioSrc
+ GstHalAudioSink
+ GstHalAudioSrc
+ GstRtpBin
GstAutoVideoSink
GstAutoVideoSrc
GstAutoAudioSink
GstAutoAudioSrc
- GstRgVolume
- GstRtpBin
- GstRTSPSrc
GstPushFileSrc
+ GstRTSPSrc
+ GstRgVolume
GstAspectRatioCrop
+ GstCmmlEnc
+ GstCmmlDec
GstBaseSink
+ GstAASink
GstBaseAudioSink
+ GstPulseSink
GstAudioSink
GstEsdSink
GstOssSink
- GstPulseSink
- GstShout2send
GstCACASink
GstVideoSink
GstGdkPixbufSink
GstV4l2Sink
- GstAASink
+ GstShout2send
+ GstTest
+ GstMultiFileSink
GstMultiUDPSink
GstUDPSink
GstDynUDPSink
- GstMultiFileSink
- GstTest
- GstJpegEnc
- GstJpegDec
- GstSmokeEnc
- GstSmokeDec
GstBaseSrc
GstPushSrc
+ GstDV1394Src
+ GstHDV1394Src
+ GstSoupHTTPSrc
GstBaseAudioSrc
GstAudioSrc
GstPulseSrc
GstOssSrc
- GstSoupHTTPSrc
- GstDV1394Src
- GstHDV1394Src
GstV4l2Src
GstXImageSrc
- GstUDPSrc
GstMultiFileSrc
- GstPulseMixer
- GstFlacEnc
- GstFlacDec
- GstFlacTag
+ GstUDPSrc
+ GstWavpackParse
+ GstWavpackDec
+ GstWavpackEnc
+ GstDVDemux
+ GstDVDec
GstTagLibMux
GstId3v2Mux
GstApev2Mux
+ GstFlacEnc
+ GstFlacDec
+ GstFlacTag
GstCairoTextOverlay
GstBaseTransform
GstCairoTimeOverlay
GstPixbufScale
- GstLevel
GstVideoFilter
GstEdgeTV
GstAgingTV
GstRadioacTV
GstStreakTV
GstRippleTV
- GstVideoBalance
+ GstNavigationtest
GstVideoFlip
+ GstVideoBalance
GstGamma
GstSMPTEAlpha
- GstNavigationtest
- GstRgAnalysis
- GstRgLimiter
+ GstVideoBox
+ GstBreakMyData
+ GstNavSeek
+ GstProgressReport
+ GstTagInject
+ GstLevel
+ GstAlpha
+ GstAlphaColor
GstAudioFilter
- GstSpectrum
GstIirEqualizer
GstIirEqualizerNBands
GstIirEqualizer3Bands
GstIirEqualizer10Bands
+ GstSpectrum
GstAudioInvert
GstAudioKaraoke
GstAudioAmplify
GstAudioWSincBand
GstAudioFIRFilter
GstAudioEcho
- GstAlpha
- GstAlphaColor
- GstAudioPanorama
- GstVideoBox
- GstBreakMyData
- GstNavSeek
- GstProgressReport
- GstTagInject
+ GstRgAnalysis
+ GstRgLimiter
GstVideoCrop
+ GstAudioPanorama
GstCairoRender
+ GstPulseMixer
+ GstSpeexEnc
+ GstSpeexDec
+ GstJpegEnc
+ GstJpegDec
+ GstSmokeEnc
+ GstSmokeDec
GstPngDec
GstPngEnc
- GstWavpackParse
- GstWavpackDec
- GstWavpackEnc
GstGdkPixbuf
- GstCmmlEnc
- GstCmmlDec
- GstSpeexEnc
- GstSpeexDec
- GstDVDemux
- GstDVDec
GstOssMixerElement
- GstWavEnc
+ GstAviDemux
+ GstAviMux
+ GstAviSubtitle
GstRTPDepay
GstBaseRTPDepayload
GstRtpAC3Depay
+ GstRTPBVDepay
GstRtpCELTDepay
GstRTPDVDepay
GstRTPiLBCDepay
+ GstRtpG723Depay
GstRtpG726Depay
GstRtpG729Depay
GstRTPGSMDepay
GstRtpVRawDepay
GstRtpXQTDepay
GstBaseRTPPayload
- GstRtpCELTPay
- GstRTPDVPay
GstBaseRTPAudioPayload
+ GstRTPBVPay
GstRTPILBCPay
GstRtpG726Pay
GstRTPG729Pay
GstRtpPcmuPay
GstRtpPmcaPay
+ GstRtpL16Pay
GstRTPSirenPay
+ GstRtpCELTPay
+ GstRTPDVPay
+ GstRTPG723Pay
GstRTPGSMPay
GstRtpAMRPay
GstRtpMPAPay
GstRtpH264Pay
GstRtpJ2KPay
GstRtpJPEGPay
- GstRtpL16Pay
GstRTPMP2TPay
GstRtpMP4VPay
GstRtpMP4APay
GstRtpVorbisPay
GstRtpVRawPay
GstAsteriskh263
- GstTagDemux
- GstID3Demux
- GstApeDemux
- GstICYDemux
- GstALawEnc
- GstALawDec
- GstMuLawEnc
- GstMuLawDec
+ GstGoom
+ GstGoom2k1
+ GstWavEnc
GstRtpJitterBuffer
GstRtpPtDemux
GstRtpSession
GstRtpSsrcDemux
- GstQTDemux
- GstWavParse
+ GstRndBufferSize
+ GstEFence
GstCutter
+ GstEbmlRead
+ GstMatroskaDemux
+ GstMatroskaMux
+ GstRTPDec
+ GstQTDemux
+ GstSMPTE
+ GstAuParse
GstMultipartDemux
GstMultipartMux
- GstGoom
- GstY4mEncode
- GstSMPTE
- GstRTPDec
+ GstALawEnc
+ GstALawDec
+ GstMuLawEnc
+ GstMuLawDec
+ GstTagDemux
+ GstApeDemux
+ GstID3Demux
+ GstFlxDec
GstDeinterlace
+ GstY4mEncode
GstInterleave
GstDeinterleave
- GstEFence
- GstRndBufferSize
- GstVideoMixer
- GstAuParse
- GstMonoscope
- GstFlxDec
- GstEbmlRead
- GstMatroskaDemux
- GstMatroskaMux
- GstGoom2k1
+ GstWavParse
GstFLVDemux
GstFlvMux
- GstAviDemux
- GstAviMux
- GstAviSubtitle
+ GstMonoscope
+ GstICYDemux
+ GstVideoMixer
GstBus
GstTask
GstTaskPool
GstPlugin
GstRegistry
GstRingBuffer
- GstAudioSrcRingBuffer
GstAudioSinkRingBuffer
+ GstAudioSrcRingBuffer
GstSignalObject
GstCmmlTagStream
GstCmmlTagHead
GTypePlugin
GstChildProxy
GstURIHandler
+ GstPropertyProbe
+ GstPreset
+ GstTagSetter
GstStreamVolume
GstImplementsInterface
- GstPropertyProbe
GstMixer
- GstTagSetter
- GstPreset
GstTuner
GstColorBalance
GstVideoOrientation
GstBin GstChildProxy
GstPipeline GstChildProxy
-GstHalAudioSink GstChildProxy
-GstHalAudioSrc GstChildProxy
GstGConfVideoSink GstChildProxy
GstGConfVideoSrc GstChildProxy
GstSwitchSink GstChildProxy
GstGConfAudioSink GstChildProxy
GstGConfAudioSrc GstChildProxy
+GstHalAudioSink GstChildProxy
+GstHalAudioSrc GstChildProxy
+GstRtpBin GstChildProxy
GstAutoVideoSink GstChildProxy
GstAutoVideoSrc GstChildProxy
GstAutoAudioSink GstChildProxy
GstAutoAudioSrc GstChildProxy
+GstPushFileSrc GstChildProxy GstURIHandler
+GstRTSPSrc GstChildProxy GstURIHandler
GstRgVolume GstChildProxy
-GstRtpBin GstChildProxy
-GstRTSPSrc GstURIHandler GstChildProxy
-GstPushFileSrc GstURIHandler GstChildProxy
GstAspectRatioCrop GstChildProxy
GstPulseSink GstStreamVolume GstImplementsInterface GstPropertyProbe
+GstV4l2Sink GstImplementsInterface GstColorBalance GstVideoOrientation GstPropertyProbe
GstShout2send GstTagSetter
-GstV4l2Sink GstImplementsInterface GstPropertyProbe GstColorBalance GstVideoOrientation
GstUDPSink GstURIHandler
-GstPulseSrc GstImplementsInterface GstPropertyProbe GstMixer
-GstOssSrc GstImplementsInterface GstMixer
-GstSoupHTTPSrc GstURIHandler
GstDV1394Src GstURIHandler GstPropertyProbe
GstHDV1394Src GstURIHandler GstPropertyProbe
-GstV4l2Src GstImplementsInterface GstPropertyProbe GstTuner GstColorBalance GstVideoOrientation
+GstSoupHTTPSrc GstURIHandler
+GstPulseSrc GstImplementsInterface GstMixer GstPropertyProbe
+GstOssSrc GstImplementsInterface GstMixer
+GstV4l2Src GstURIHandler GstImplementsInterface GstTuner GstColorBalance GstVideoOrientation GstPropertyProbe
GstUDPSrc GstURIHandler
-GstPulseMixer GstImplementsInterface GstPropertyProbe GstMixer
-GstFlacEnc GstTagSetter GstPreset
-GstFlacTag GstTagSetter
+GstWavpackEnc GstPreset
GstTagLibMux GstTagSetter
GstId3v2Mux GstTagSetter
GstApev2Mux GstTagSetter
+GstFlacEnc GstTagSetter GstPreset
+GstFlacTag GstTagSetter
GstVideoBalance GstImplementsInterface GstColorBalance
GstIirEqualizer GstChildProxy
GstIirEqualizerNBands GstChildProxy
GstIirEqualizer3Bands GstChildProxy GstPreset
GstIirEqualizer10Bands GstChildProxy GstPreset
-GstWavpackEnc GstPreset
+GstPulseMixer GstImplementsInterface GstMixer GstPropertyProbe
GstSpeexEnc GstTagSetter GstPreset
GstOssMixerElement GstImplementsInterface GstMixer
-GstDeinterlace GstChildProxy
-GstVideoMixer GstChildProxy
+GstAviMux GstTagSetter
GstMatroskaMux GstTagSetter
+GstDeinterlace GstChildProxy
GstFlvMux GstTagSetter
-GstAviMux GstTagSetter
+GstVideoMixer GstChildProxy
GstChildProxy GstObject
+GstTagSetter GstObject GstElement
GstStreamVolume GObject
-GstImplementsInterface GstElement
-GstMixer GstImplementsInterface GstElement
-GstTagSetter GstElement
-GstTuner GstImplementsInterface GstElement
-GstColorBalance GstImplementsInterface GstElement
-GstVideoOrientation GstImplementsInterface GstElement
+GstImplementsInterface GstObject GstElement
+GstMixer GstObject GstImplementsInterface GstElement
+GstTuner GstObject GstImplementsInterface GstElement
+GstColorBalance GstObject GstImplementsInterface GstElement
+GstVideoOrientation GstObject GstImplementsInterface GstElement
<description>Source for video data via IEEE1394 interface</description>
<filename>../../ext/raw1394/.libs/libgst1394.so</filename>
<basename>libgst1394.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>ASCII Art video sink</description>
<filename>../../ext/aalib/.libs/libgstaasink.so</filename>
<basename>libgstaasink.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>ALaw audio conversion routines</description>
<filename>../../gst/law/.libs/libgstalaw.so</filename>
<basename>libgstalaw.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>adds an alpha channel to video - constant or via chroma-keying</description>
<filename>../../gst/alpha/.libs/libgstalpha.so</filename>
<basename>libgstalpha.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>RGBA to AYUV colorspace conversion preserving the alpha channel</description>
<filename>../../gst/alpha/.libs/libgstalphacolor.so</filename>
<basename>libgstalphacolor.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>annodex stream manipulation (info about annodex: http://www.annodex.net)</description>
<filename>../../ext/annodex/.libs/libgstannodex.so</filename>
<basename>libgstannodex.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>APEv1/2 tag reader</description>
<filename>../../gst/apetag/.libs/libgstapetag.so</filename>
<basename>libgstapetag.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Audio effects plugin</description>
<filename>../../gst/audiofx/.libs/libgstaudiofx.so</filename>
<basename>libgstaudiofx.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>parses au streams</description>
<filename>../../gst/auparse/.libs/libgstauparse.so</filename>
<basename>libgstauparse.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Plugin contains auto-detection plugins for video/audio in- and outputs</description>
<filename>../../gst/autodetect/.libs/libgstautodetect.so</filename>
<basename>libgstautodetect.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>AVI stream handling</description>
<filename>../../gst/avi/.libs/libgstavi.so</filename>
<basename>libgstavi.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<name>video_%02d</name>
<direction>source</direction>
<presence>sometimes</presence>
- <details>video/x-3ivx, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-asus, asusversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-asus, asusversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-cirrus-logic-accupak, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-compressed-yuv, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-raw-rgb, bpp=(int){ 8, 24 }, depth=(int){ 8, 24 }, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-divx, divxversion=(int)3, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-divx, divxversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-truemotion, trueversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-dv, systemstream=(boolean)false, dvversion=(int)25, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-dv, systemstream=(boolean)false, dvversion=(int)50, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-divx, divxversion=(int)5, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/mpeg, mpegversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-flash-video, flvversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h263, variant=(string)itu, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h263, variant=(string)lucent, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h264, variant=(string)itu, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-huffyuv, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-intel-h263, variant=(string)intel, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-raw-yuv, format=(fourcc)I420, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-indeo, indeoversion=(int)3, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-indeo, indeoversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-indeo, indeoversion=(int)5, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h263, variant=(string)lead, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h264, variant=(string)lead, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h263, variant=(string)microsoft, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/mpeg, mpegversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; image/jpeg, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-msmpeg, msmpegversion=(int)42, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-msmpeg, msmpegversion=(int)43, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/mpeg, systemstream=(boolean)false, mpegversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/mpeg, systemstream=(boolean)false, mpegversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-msmpeg, msmpegversion=(int)41, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-mszh, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; image/png, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-rle, layout=(string)microsoft, depth=(int)[ 1, 64 ], framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-indeo, indeoversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/sp5x, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-truemotion, trueversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-camtasia, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-ultimotion, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-raw-yuv, format=(fourcc)UYVY, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-ati-vcr, vcrversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-ati-vcr, vcrversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h263, variant=(string)vdolive, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h263, variant=(string)vivo, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-vmnc, version=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-vp3, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h264, variant=(string)videosoft, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-wmv, wmvversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-wmv, wmvversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-wmv, wmvversion=(int)3, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-xvid, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-xan, wcversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-raw-yuv, format=(fourcc)YUY2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-raw-yuv, format=(fourcc)YVU9, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-zlib, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-cinepak, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h264, variant=(string)itu, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-msvideocodec, msvideoversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h263, variant=(string)xirlink, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-dirac, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-ffv, ffvversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-kmvc, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-vp6, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-mimic, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-apple-video, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-theora, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-fraps, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-dv, systemstream=(boolean)true; video/x-avi-unknown</details>
+ <details>video/x-3ivx, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-asus, asusversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-asus, asusversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-cirrus-logic-accupak, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-compressed-yuv, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-raw-rgb, bpp=(int){ 8, 24, 32 }, depth=(int){ 8, 24 }, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-divx, divxversion=(int)3, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-divx, divxversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-truemotion, trueversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-dv, systemstream=(boolean)false, dvversion=(int)25, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-dv, systemstream=(boolean)false, dvversion=(int)50, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-divx, divxversion=(int)5, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/mpeg, mpegversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-flash-video, flvversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h263, variant=(string)itu, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h263, variant=(string)lucent, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h264, variant=(string)itu, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-huffyuv, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-intel-h263, variant=(string)intel, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-raw-yuv, format=(fourcc)I420, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-indeo, indeoversion=(int)3, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-indeo, indeoversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-indeo, indeoversion=(int)5, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h263, variant=(string)lead, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h264, variant=(string)lead, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h263, variant=(string)microsoft, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/mpeg, mpegversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; image/jpeg, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-msmpeg, msmpegversion=(int)42, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-msmpeg, msmpegversion=(int)43, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/mpeg, systemstream=(boolean)false, mpegversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/mpeg, systemstream=(boolean)false, mpegversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-msmpeg, msmpegversion=(int)41, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-mszh, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; image/png, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-rle, layout=(string)microsoft, depth=(int)[ 1, 64 ], framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-indeo, indeoversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/sp5x, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-truemotion, trueversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-camtasia, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-ultimotion, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-raw-yuv, format=(fourcc)UYVY, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-ati-vcr, vcrversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-ati-vcr, vcrversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h263, variant=(string)vdolive, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h263, variant=(string)vivo, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-vmnc, version=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-vp3, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h264, variant=(string)videosoft, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-wmv, wmvversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-wmv, wmvversion=(int)2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-wmv, wmvversion=(int)3, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-xvid, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-xan, wcversion=(int)4, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-raw-yuv, format=(fourcc)YUY2, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-raw-yuv, format=(fourcc)YVU9, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-zlib, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-cinepak, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h264, variant=(string)itu, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-msvideocodec, msvideoversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-h263, variant=(string)xirlink, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-dirac, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-ffv, ffvversion=(int)1, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-kmvc, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-vp5, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-vp6, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-mimic, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-apple-video, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-theora, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-fraps, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-aasc, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-raw-yuv, format=(fourcc)YV12, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-loco, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-zmbv, framerate=(fraction)[ 0/1, 2147483647/1 ], width=(int)[ 16, 4096 ], height=(int)[ 16, 4096 ]; video/x-dv, systemstream=(boolean)true; video/x-avi-unknown</details>
</caps>
<caps>
<name>audio_%02d</name>
<name>audio_%d</name>
<direction>sink</direction>
<presence>request</presence>
- <details>audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int){ 8, 16 }, depth=(int){ 8, 16 }, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ]; audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ], rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ]; audio/mpeg, mpegversion=(int)4, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ]; audio/x-ac3, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ]; audio/x-alaw, rate=(int)[ 1000, 48000 ], channels=(int)[ 1, 2 ]; audio/x-mulaw, rate=(int)[ 1000, 48000 ], channels=(int)[ 1, 2 ]; audio/x-wma, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ], wmaversion=(int)[ 1, 2 ]</details>
+ <details>audio/x-raw-int, endianness=(int)1234, signed=(boolean){ true, false }, width=(int){ 8, 16 }, depth=(int){ 8, 16 }, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ]; audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ], rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ]; audio/mpeg, mpegversion=(int)4, stream-format=(string)raw, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ]; audio/x-ac3, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ]; audio/x-alaw, rate=(int)[ 1000, 48000 ], channels=(int)[ 1, 2 ]; audio/x-mulaw, rate=(int)[ 1000, 48000 ], channels=(int)[ 1, 2 ]; audio/x-wma, rate=(int)[ 1000, 96000 ], channels=(int)[ 1, 2 ], wmaversion=(int)[ 1, 2 ]</details>
</caps>
<caps>
<name>src</name>
<description>Colored ASCII Art video sink</description>
<filename>../../ext/libcaca/.libs/libgstcacasink.so</filename>
<basename>libgstcacasink.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Cairo-based elements</description>
<filename>../../ext/cairo/.libs/libgstcairo.so</filename>
<basename>libgstcairo.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Audio Cutter to split audio into non-silent bits</description>
<filename>../../gst/cutter/.libs/libgstcutter.so</filename>
<basename>libgstcutter.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>elements for testing and debugging</description>
<filename>../../gst/debugutils/.libs/libgstdebug.so</filename>
<basename>libgstdebug.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Deinterlacer</description>
<filename>../../gst/deinterlace/.libs/libgstdeinterlace.so</filename>
<basename>libgstdeinterlace.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>DV demuxer and decoder based on libdv (libdv.sf.net)</description>
<filename>../../ext/dv/.libs/libgstdv.so</filename>
<basename>libgstdv.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>This element converts a stream of normal GStreamer buffers into a stream of buffers that are allocated in such a way that out-of-bounds access to data in the buffer is more likely to cause segmentation faults. This allocation method is very similar to the debugging tool "Electric Fence".</description>
<filename>../../gst/debugutils/.libs/libgstefence.so</filename>
<basename>libgstefence.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>effect plugins from the effectv project</description>
<filename>../../gst/effectv/.libs/libgsteffectv.so</filename>
<basename>libgsteffectv.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>GStreamer audio equalizers</description>
<filename>../../gst/equalizer/.libs/libgstequalizer.so</filename>
<basename>libgstequalizer.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>ESD Element Plugins</description>
<filename>../../ext/esd/.libs/libgstesd.so</filename>
<basename>libgstesd.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>The FLAC Lossless compressor Codec</description>
<filename>../../ext/flac/.libs/libgstflac.so</filename>
<basename>libgstflac.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>FLV muxing and demuxing plugin</description>
<filename>../../gst/flv/.libs/libgstflv.so</filename>
<basename>libgstflv.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>FLC/FLI/FLX video decoder</description>
<filename>../../gst/flx/.libs/libgstflxdec.so</filename>
<basename>libgstflxdec.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Changes gamma on video images</description>
<filename>../../gst/videofilter/.libs/libgstgamma.so</filename>
<basename>libgstgamma.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>elements wrapping the GStreamer/GConf audio/video output settings</description>
<filename>../../ext/gconf/.libs/libgstgconfelements.so</filename>
<basename>libgstgconfelements.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>GdkPixbuf-based image decoder, scaler and sink</description>
<filename>../../ext/gdk_pixbuf/.libs/libgstgdkpixbuf.so</filename>
<basename>libgstgdkpixbuf.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>GOOM visualization filter</description>
<filename>../../gst/goom/.libs/libgstgoom.so</filename>
<basename>libgstgoom.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>GOOM 2k1 visualization filter</description>
<filename>../../gst/goom2k1/.libs/libgstgoom2k1.so</filename>
<basename>libgstgoom2k1.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>RTP session management plugin library</description>
<filename>../../gst/rtpmanager/.libs/libgstrtpmanager.so</filename>
<basename>libgstrtpmanager.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<name>gstrtpbin</name>
<longname>RTP Bin</longname>
<class>Filter/Network/RTP</class>
- <description>Implement an RTP bin</description>
+ <description>Real-Time Transport Protocol bin</description>
<author>Wim Taymans <wim.taymans@gmail.com></author>
<pads>
<caps>
<description>elements wrapping the GStreamer/HAL audio input/output devices</description>
<filename>../../ext/hal/.libs/libgsthalelements.so</filename>
<basename>libgsthalelements.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Demux ICY tags from a stream</description>
<filename>../../gst/icydemux/.libs/libgsticydemux.so</filename>
<basename>libgsticydemux.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Demux ID3v1 and ID3v2 tags from a file</description>
<filename>../../gst/id3demux/.libs/libgstid3demux.so</filename>
<basename>libgstid3demux.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Audio interleaver/deinterleaver</description>
<filename>../../gst/interleave/.libs/libgstinterleave.so</filename>
<basename>libgstinterleave.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>JPeg plugin library</description>
<filename>../../ext/jpeg/.libs/libgstjpeg.so</filename>
<basename>libgstjpeg.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Audio level plugin</description>
<filename>../../gst/level/.libs/libgstlevel.so</filename>
<basename>libgstlevel.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Matroska stream handling</description>
<filename>../../gst/matroska/.libs/libgstmatroska.so</filename>
<basename>libgstmatroska.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<name>audio_%d</name>
<direction>sink</direction>
<presence>request</presence>
- <details>audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ], channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int){ 2, 4 }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-ac3, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-vorbis, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-flac, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-speex, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)8, depth=(int)8, signed=(boolean)false, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int){ 4321, 1234 }, signed=(boolean)true, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)24, depth=(int)24, endianness=(int){ 4321, 1234 }, signed=(boolean)true, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)32, depth=(int)32, endianness=(int){ 4321, 1234 }, signed=(boolean)true, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-float, width=(int)[ 32, 64 ], endianness=(int)1234, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-tta, width=(int){ 8, 16, 24 }, channels=(int){ 1, 2 }, rate=(int)[ 8000, 96000 ]; audio/x-pn-realaudio, raversion=(int){ 1, 2, 8 }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-wma, wmaversion=(int)[ 1, 3 ], block_align=(int)[ 0, 65535 ], bitrate=(int)[ 0, 524288 ], channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]</details>
+ <details>audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ], stream-format=(string){ raw }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int){ 2, 4 }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-ac3, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-vorbis, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-flac, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-speex, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)8, depth=(int)8, signed=(boolean)false, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)16, depth=(int)16, endianness=(int){ 4321, 1234 }, signed=(boolean)true, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)24, depth=(int)24, endianness=(int){ 4321, 1234 }, signed=(boolean)true, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-int, width=(int)32, depth=(int)32, endianness=(int){ 4321, 1234 }, signed=(boolean)true, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw-float, width=(int)[ 32, 64 ], endianness=(int)1234, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-tta, width=(int){ 8, 16, 24 }, channels=(int){ 1, 2 }, rate=(int)[ 8000, 96000 ]; audio/x-pn-realaudio, raversion=(int){ 1, 2, 8 }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-wma, wmaversion=(int)[ 1, 3 ], block_align=(int)[ 0, 65535 ], bitrate=(int)[ 0, 524288 ], channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]</details>
</caps>
<caps>
<name>video_%d</name>
<description>Monoscope visualization</description>
<filename>../../gst/monoscope/.libs/libgstmonoscope.so</filename>
<basename>libgstmonoscope.so</basename>
- <version>0.10.17</version>
+ <version>0.10.16.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins git/prerelease</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>MuLaw audio conversion routines</description>
<filename>../../gst/law/.libs/libgstmulaw.so</filename>
<basename>libgstmulaw.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Reads/Writes buffers from/to sequentially named files</description>
<filename>../../gst/multifile/.libs/libgstmultifile.so</filename>
<basename>libgstmultifile.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>multipart stream manipulation</description>
<filename>../../gst/multipart/.libs/libgstmultipart.so</filename>
<basename>libgstmultipart.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Template for a video filter</description>
<filename>../../gst/debugutils/.libs/libgstnavigationtest.so</filename>
<basename>libgstnavigationtest.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>OSS (Open Sound System) support for GStreamer</description>
<filename>../../sys/oss/.libs/libgstossaudio.so</filename>
<basename>libgstossaudio.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>PNG plugin library</description>
<filename>../../ext/libpng/.libs/libgstpng.so</filename>
<basename>libgstpng.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>PulseAudio plugin library</description>
<filename>../../ext/pulse/.libs/libgstpulse.so</filename>
<basename>libgstpulse.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Quicktime support</description>
<filename>../../gst/qtdemux/.libs/libgstqtdemux.so</filename>
<basename>libgstqtdemux.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>ReplayGain volume normalization</description>
<filename>../../gst/replaygain/.libs/libgstreplaygain.so</filename>
<basename>libgstreplaygain.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Real-time protocol plugins</description>
<filename>../../gst/rtp/.libs/libgstrtp.so</filename>
<basename>libgstrtp.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
</pads>
</element>
<element>
+ <name>rtpbvdepay</name>
+ <longname>RTP BroadcomVoice depayloader</longname>
+ <class>Codec/Depayloader/Network</class>
+ <description>Extracts BroadcomVoice audio from RTP packets (RFC 4298)</description>
+ <author>Wim Taymans <wim.taymans@collabora.co.uk></author>
+ <pads>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)8000, encoding-name=(string)BV16; application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)16000, encoding-name=(string)BV32</details>
+ </caps>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>audio/x-bv, mode=(int){ 16, 32 }</details>
+ </caps>
+ </pads>
+ </element>
+ <element>
+ <name>rtpbvpay</name>
+ <longname>RTP BV Payloader</longname>
+ <class>Codec/Payloader/Network</class>
+ <description>Packetize BroadcomVoice audio streams into RTP packets (RFC 4298)</description>
+ <author>Wim Taymans <wim.taymans@collabora.co.uk></author>
+ <pads>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)8000, encoding-name=(string)BV16; application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)16000, encoding-name=(string)BV32</details>
+ </caps>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>audio/x-bv, mode=(int){ 16, 32 }</details>
+ </caps>
+ </pads>
+ </element>
+ <element>
<name>rtpceltdepay</name>
<longname>RTP CELT depayloader</longname>
<class>Codec/Depayloader/Network</class>
</pads>
</element>
<element>
+ <name>rtpg723depay</name>
+ <longname>RTP G.723 depayloader</longname>
+ <class>Codec/Depayloader/Network</class>
+ <description>Extracts G.723 audio from RTP packets (RFC 3551)</description>
+ <author>Wim Taymans <wim.taymans@gmail.com></author>
+ <pads>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)8000, encoding-name=(string)G723; application/x-rtp, media=(string)audio, payload=(int)4, clock-rate=(int)8000</details>
+ </caps>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>audio/G723, channels=(int)1, rate=(int)8000</details>
+ </caps>
+ </pads>
+ </element>
+ <element>
+ <name>rtpg723pay</name>
+ <longname>RTP G.723 payloader</longname>
+ <class>Codec/Payloader/Network</class>
+ <description>Packetize G.723 audio into RTP packets</description>
+ <author>Wim Taymans <wim.taymans@gmail.com></author>
+ <pads>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>application/x-rtp, media=(string)audio, payload=(int)4, clock-rate=(int)8000, encoding-name=(string)G723; application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)8000, encoding-name=(string)G723</details>
+ </caps>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>audio/G723, channels=(int)1, rate=(int)8000</details>
+ </caps>
+ </pads>
+ </element>
+ <element>
<name>rtpg726depay</name>
<longname>RTP G.726 depayloader</longname>
<class>Codec/Depayloader/Network</class>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
- <details>audio/mpeg, mpegversion=(int)4, framed=(boolean)true</details>
+ <details>audio/mpeg, mpegversion=(int)4, framed=(boolean)true, stream-format=(string)raw</details>
</caps>
</pads>
</element>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
- <details>audio/mpeg, mpegversion=(int)4</details>
+ <details>audio/mpeg, mpegversion=(int)4, stream-format=(string)raw</details>
</caps>
<caps>
<name>src</name>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
- <details>video/mpeg, mpegversion=(int)4, systemstream=(boolean)false; audio/mpeg, mpegversion=(int)4</details>
+ <details>video/mpeg, mpegversion=(int)4, systemstream=(boolean)false; audio/mpeg, mpegversion=(int)4, stream-format=(string)raw</details>
</caps>
</pads>
</element>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
- <details>video/mpeg, mpegversion=(int)4, systemstream=(boolean)false; audio/mpeg, mpegversion=(int)4</details>
+ <details>video/mpeg, mpegversion=(int)4, systemstream=(boolean)false; audio/mpeg, mpegversion=(int)4, stream-format=(string)raw</details>
</caps>
<caps>
<name>src</name>
<description>transfer data via RTSP</description>
<filename>../../gst/rtsp/.libs/libgstrtsp.so</filename>
<basename>libgstrtsp.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Sends data to an icecast server using libshout2</description>
<filename>../../ext/shout2/.libs/libgstshout2.so</filename>
<basename>libgstshout2.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>libshout2</package>
<description>Apply the standard SMPTE transitions on video images</description>
<filename>../../gst/smpte/.libs/libgstsmpte.so</filename>
<basename>libgstsmpte.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>libsoup HTTP client src</description>
<filename>../../ext/soup/.libs/libgstsouphttpsrc.so</filename>
<basename>libgstsouphttpsrc.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Run an FFT on the audio signal, output spectrum data</description>
<filename>../../gst/spectrum/.libs/libgstspectrum.so</filename>
<basename>libgstspectrum.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Speex plugin library</description>
<filename>../../ext/speex/.libs/libgstspeex.so</filename>
<basename>libgstspeex.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Tag writing plug-in based on taglib</description>
<filename>../../ext/taglib/.libs/libgsttaglib.so</filename>
<basename>libgsttaglib.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>transfer data via UDP</description>
<filename>../../gst/udp/.libs/libgstudp.so</filename>
<basename>libgstudp.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>elements for Video 4 Linux</description>
<filename>../../sys/v4l2/.libs/libgstvideo4linux2.so</filename>
<basename>libgstvideo4linux2.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Changes hue, saturation, brightness etc. on video images</description>
<filename>../../gst/videofilter/.libs/libgstvideobalance.so</filename>
<basename>libgstvideobalance.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>resizes a video by adding borders or cropping</description>
<filename>../../gst/videobox/.libs/libgstvideobox.so</filename>
<basename>libgstvideobox.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Crops video into a user-defined region</description>
<filename>../../gst/videocrop/.libs/libgstvideocrop.so</filename>
<basename>libgstvideocrop.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Flips and rotates video</description>
<filename>../../gst/videofilter/.libs/libgstvideoflip.so</filename>
<basename>libgstvideoflip.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Video mixer</description>
<filename>../../gst/videomixer/.libs/libgstvideomixer.so</filename>
<basename>libgstvideomixer.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Encode raw audio into WAV</description>
<filename>../../gst/wavenc/.libs/libgstwavenc.so</filename>
<basename>libgstwavenc.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Wavpack lossless/lossy audio format handling</description>
<filename>../../ext/wavpack/.libs/libgstwavpack.so</filename>
<basename>libgstwavpack.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Parse a .wav file into raw audio</description>
<filename>../../gst/wavparse/.libs/libgstwavparse.so</filename>
<basename>libgstwavparse.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>X11 video input plugin using standard Xlib calls</description>
<filename>../../sys/ximage/.libs/libgstximagesrc.so</filename>
<basename>libgstximagesrc.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<description>Encodes a YUV frame into the yuv4mpeg format (mjpegtools)</description>
<filename>../../gst/y4m/.libs/libgsty4menc.so</filename>
<basename>libgsty4menc.so</basename>
- <version>0.10.17</version>
+ <version>0.10.18</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
- <package>GStreamer Good Plug-ins prerelease</package>
+ <package>GStreamer Good Plug-ins source release</package>
<origin>Unknown package origin</origin>
<elements>
<element>
<release>
<Version>
+ <revision>0.10.18</revision>
+ <branch>0.10</branch>
+ <name>Short Circuit</name>
+ <created>2010-02-10</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-plugins-good/gst-plugins-good-0.10.18.tar.bz2" />
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-plugins-good/gst-plugins-good-0.10.18.tar.gz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>0.10.17</revision>
<branch>0.10</branch>
<name>They used to sparkle</name>
#define GST_LICENSE "LGPL"
/* package name in plugins */
-#define GST_PACKAGE_NAME "GStreamer Good Plug-ins prerelease"
+#define GST_PACKAGE_NAME "GStreamer Good Plug-ins source release"
/* package origin */
#define GST_PACKAGE_ORIGIN "Unknown package origin"
#define PACKAGE_NAME "GStreamer Good Plug-ins"
/* Define to the full name and version of this package. */
-#define PACKAGE_STRING "GStreamer Good Plug-ins 0.10.17.3"
+#define PACKAGE_STRING "GStreamer Good Plug-ins 0.10.18"
/* Define to the one symbol short name of this package. */
#define PACKAGE_TARNAME "gst-plugins-good"
#undef PACKAGE_URL
/* Define to the version of this package. */
-#define PACKAGE_VERSION "0.10.17.3"
+#define PACKAGE_VERSION "0.10.18"
/* directory where plugins are located */
#ifdef _DEBUG
#undef STDC_HEADERS
/* Version number of package */
-#define VERSION "0.10.17.3"
+#define VERSION "0.10.18"
/* old wavpack API */
#undef WAVPACK_OLD_API
projects / platform / upstream / gst-plugins-good.git / commitdiff