+=== release 1.1.90 ===
+
+2013-09-19 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * configure.ac:
+ releasing 1.1.90
+
+2013-09-19 09:45:18 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * po/cs.po:
+ * po/nl.po:
+ * po/pl.po:
+ * po/uk.po:
+ * po/vi.po:
+ po: Update translations
+
+2013-09-11 14:27:02 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: dmabuf is not a singleton anymore
+ https://bugzilla.gnome.org/show_bug.cgi?id=707793
+
+2013-09-16 13:53:45 -0300 Thiago Santos <thiago.sousa.santos@collabora.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: do not do http requests in READY
+ HEAD requests to discover if the server is seekable shouldn't be done in
+ READY as it might lock the main thread that is doing the state change.
+ https://bugzilla.gnome.org/show_bug.cgi?id=705371
+
+2013-09-18 16:32:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: reevaluate the current timer after timeout
+ When we trigger the timeout logic of a timer, reevaluate it because it is
+ possible that it still has the lowest timeout.
+
+2013-09-18 16:31:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: don't update time when unscheduled
+ Don't try to estimate the current time when we got unscheduled.
+
+2013-09-18 16:29:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: init packet spacing on first buffer
+ Already init the packet spacing variables on the first buffer so that we can
+ calculate the spacing on the second buffer already.
+
+2013-09-18 15:08:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ tests: fix comments
+
+2013-09-18 14:57:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: push the lost event from the timer thread
+ Instead of pushing the lost event from the chain function, schedule a timeout
+ that will push the lost event from the timer thread. This avoid blocking the
+ upstream thread while we push and sync the event.
+
+2013-09-18 14:23:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: add another test
+ The test is modified slightly because the late lost packets are only
+ generated now when a large gap is received.
+
+2013-09-18 14:12:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: round gap duration to multiple of duration
+ Make sure the gap duration in the lost event is a multiple of the packet
+ duration.
+ Enable another test.
+
+2013-09-18 12:29:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: keep track of duration
+ Keep track of the estimated duration of missing packets and use it in the lost
+ event.
+ Enable another unit test
+
+2013-09-18 11:59:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: handle large gaps with one lost event
+ When we have a large number of missing packets, generate one lost event for all
+ the packets that have no chance of being pushed out in time.
+ Fix and activate unit test for large gaps.
+
+2013-09-18 11:56:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: refactor lost event sending
+ Also make sure we only increment the expected seqnum and last
+ output timestamp.
+
+2013-09-17 23:21:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: refactor timeout triggers
+
+2013-09-17 23:03:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: simplify the timeout code
+ Keep track of the current time in the timeout loop.
+ Loop over all timers and trigger all the expired ones, we can do this in the
+ same loop that selects the new best timer.
+
+2013-09-17 23:01:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: rearrange timer update code
+ Also update the timers when retransmission is disabled. We need to
+ do this because when we added LOST timers when we detected missing packets and
+ we need to remove those timers when the packet finally arrives.
+
+2013-09-17 22:02:04 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/videomixer/Makefile.am:
+ videomixer: link to libm for maths stuff
+ Fixes undefined references to rint and pow on ubuntu
+ build bot.
+
+2013-09-17 15:19:42 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: release lock on shutdown
+
+2013-09-17 15:11:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/check/Makefile.am:
+ check: change for videomixer renamed orc file
+
+2013-09-14 16:03:20 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: remove MAX_TOLERATED_LATENESS
+ https://bugzilla.gnome.org/show_bug.cgi?id=707411
+
+2013-09-16 15:54:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * tests/examples/rtp/client-H264-rtx.sh:
+ examples: we don't need the queue anymore
+
+2013-09-16 15:53:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: use separate thread for timeouts
+ Use a separate thread for scheduling the timeouts instead of using the
+ downstream streaming thread that might block at any time.
+
+2013-09-14 15:56:04 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: set first_ts to DTS for streams that have DTS
+ https://bugzilla.gnome.org/show_bug.cgi?id=707340
+
+2013-09-14 15:55:22 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: make sure duration is a valid number for last buffer
+ https://bugzilla.gnome.org/show_bug.cgi?id=707340
+
+2013-09-14 15:54:29 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: use segment.start or last buffer end time in case of missing DTS
+ https://bugzilla.gnome.org/show_bug.cgi?id=707340
+
+2013-09-03 18:14:04 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/gstqtmux.c:
+ Revert qtmux: Use buffer PTS if DTS is not set"
+ This reverts commit f72c3cf71fde622067f41f31a53978ba4c94469d.
+ https://bugzilla.gnome.org/show_bug.cgi?id=707340
+
+2013-09-16 11:03:06 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/videomixer/videomixerorc-dist.c:
+ * gst/videomixer/videomixerorc-dist.h:
+ videomixer: Update orc generated files
+ https://bugzilla.gnome.org/show_bug.cgi?id=708131
+
+2013-09-13 16:25:49 +0200 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ rtpsession: Demux RTCP buffers from the RTP stream
+ If there are RTCP buffers in the RTP stream, process them as
+ RTCP. This way, we want receive streams following RFC 5761
+ https://bugzilla.gnome.org/show_bug.cgi?id=687657
+
+2013-09-13 23:26:21 +1000 Jan Schmidt <thaytan@noraisin.net>
+
+ * gst/rtp/gstrtpL24depay.c:
+ rtp: Remove bogus extra caps from L24 template.
+ The extra caps entry in the template was making it sometimes
+ get plugged for any dynamically allocated payload type.
+
+2013-09-13 12:40:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ * gst/rtpmanager/rtpstats.h:
+ rtpbin: use PacketInfo for the sender
+ Avoid mapping the packet multiple times when sending RTP.
+
+2013-09-13 12:22:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ * gst/rtpmanager/rtpstats.h:
+ rtpbin: store more in the PacketInfo
+ Store all info in the PacketInfo so that we can avoid mapping the packet
+ multiple times.
+
+2013-09-13 11:32:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpstats.h:
+ session: store more in the PacketInfo structure
+
+2013-09-13 11:08:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ * gst/rtpmanager/rtpstats.h:
+ rtpbin: RTPArrivalStats -> RTPPacketInfo
+ Rename a structure because we are also going to use this for the sender
+ bits.
+
+2013-09-13 10:55:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ source: small cleanups
+
+2013-09-12 13:31:01 -0300 Thiago Santos <thiago.sousa.santos@collabora.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: only update stop position if seek requests it
+ Check for GST_SEEK_TYPE_NONE for stop poistion and only update
+ the stop time if it is requested. Otherwise just maintain whatever
+ was stored at the segment
+ https://bugzilla.gnome.org/show_bug.cgi?id=707530
+
+2013-09-13 08:53:25 +0200 Rico Tzschichholz <ricotz@ubuntu.com>
+
+ * gst/rtp/Makefile.am:
+ rtp: Add missing headers tp fix make dist
+ In addition to a956a6ceb2deb87cc1361aee1d6626449f46dab2
+
+2013-09-12 15:07:48 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: Make sure we have enough data to read image tags
+ Thanks to iputinei for reporting this on IRC.
+
+2013-09-12 15:01:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: handle segments with non-0 start
+ We keep the DTS and PTS in running-time inside the jitterbuffer. Make sure to
+ transform it back to a buffer timestamp before pushing out the buffer.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707931
+
+2013-09-11 13:11:58 -0600 Seán de Búrca <leftmostcat@gmail.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Fix off-by-one in validation of UTF-8
+ https://bugzilla.gnome.org/show_bug.cgi?id=707933
+
+2013-09-11 14:32:17 -0300 Thibault Saunier <thibault.saunier@collabora.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: Do not check if caps are empty when they are NULL
+ In the case the caps are actually NULL, we should just concider it the
+ same way as empty caps in that case.
+
+2013-09-10 16:44:53 -0600 Seán de Búrca <leftmostcat@gmail.com>
+
+ * gst/videomixer/blendorc-dist.c:
+ * gst/videomixer/blendorc-dist.h:
+ * gst/videomixer/videomixerorc-dist.c:
+ * gst/videomixer/videomixerorc-dist.h:
+ videomixer: fix build if orc is not installed
+ https://bugzilla.gnome.org/show_bug.cgi?id=707886
+
+2013-09-10 17:57:49 -0300 Thiago Santos <thiago.sousa.santos@collabora.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Preserve seqnum when pushing seek upstream
+ After converting a seek from time to bytes, use the same seqnum
+ on the event that goes upstream
+
+2013-09-05 00:17:16 -0300 Thiago Santos <thiago.sousa.santos@collabora.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: track streams that are EOS on push mode to finish earlier
+ When the segment has a defined stop position, qtdemux should check
+ when streams reach this position and mark those as EOS. When all
+ streams are EOS it will return GST_FLOW_EOS to upstream to allow
+ the pipeline to finish instead of continuously consume buffers
+ from upstream that are not useful for the segment.
+ https://bugzilla.gnome.org/show_bug.cgi?id=707530
+
+2013-09-04 15:34:35 -0300 Thiago Santos <thiago.sousa.santos@collabora.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: preserve stop of segment when doing seeks in push mode
+ When handling seeks in push mode, qtdemux converts the seek to bytes
+ and pushes upstream. It needs to keep track of the seek and the
+ subsequent segment to be able to map them back to the requested
+ seek time and properly preserve the segment stop of the seek.
+ This is done by using the start offset in bytes of the seek,
+ that should be the same of the segment from upstream. And this
+ is also backwards compatible with what qtdemux already was using.
+ https://bugzilla.gnome.org/show_bug.cgi?id=707530
+
+2013-07-26 19:40:53 +0200 Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>
+
+ * gst/videomixer/videomixer2.c:
+ * gst/videomixer/videomixer2pad.h:
+ videomixer: Add colorspace conversion
+ https://bugzilla.gnome.org/show_bug.cgi?id=704950
+
+2013-08-06 15:38:39 +0200 Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: Don't send reconfigure event when formats or PAR are different
+ It is racy with multiple pads.
+ https://bugzilla.gnome.org/show_bug.cgi?id=704950
+
+2013-07-25 13:49:57 +0200 Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>
+
+ * gst/videomixer/Makefile.am:
+ * gst/videomixer/blend.c:
+ * gst/videomixer/blendorc.orc:
+ * gst/videomixer/gstcms.c:
+ * gst/videomixer/gstcms.h:
+ * gst/videomixer/videoconvert.c:
+ * gst/videomixer/videoconvert.h:
+ * gst/videomixer/videomixer2.c:
+ * gst/videomixer/videomixerorc.orc:
+ videomixer: Bundle private copies of videoconvert code
+ Ideally, this would be part of libgstvideo.
+ Prefixes videoconvert symbols with videomixer_.
+ https://bugzilla.gnome.org/show_bug.cgi?id=704950
+
+2013-08-22 00:03:48 +0200 Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2: Use newly #defined metadata names.
+
+2013-09-09 15:11:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: only wait if we flushed
+ Only wait for the STREAM_LOCK when we flushed something when sending
+ a command for PAUSED or PLAYING.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707611
+
+2013-09-09 15:09:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: return when a flush was issued
+ Make gst_rtspsrc_loop_send_cmd() return TRUE when the current
+ action has been flushed
+
+2013-09-09 11:16:40 +0200 David Holroyd <dave@badgers-in-foil.co.uk>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpL24depay.c:
+ * gst/rtp/gstrtpL24depay.h:
+ * gst/rtp/gstrtpL24pay.c:
+ * gst/rtp/gstrtpL24pay.h:
+ * tests/check/elements/rtp-payloading.c:
+ rtp: add L24 pay and depayloader
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707734
+
+2013-09-09 14:46:42 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Fix missing condition in previous commit
+
+2013-09-09 14:44:58 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Also fix strides for other semi-planar video formats
+
+2013-09-09 14:41:42 +0200 Andreea Fulger <andreea.fulger@parrot.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Fix stride for NV12/NV21
+ https://bugzilla.gnome.org/show_bug.cgi?id=707758
+
+2013-09-07 16:37:03 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/matroska/matroska-read-common.c:
+ matroskademux: fix leaking buffer and caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=707688
+
+2013-09-05 19:46:37 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: fix build on win32
+ gstudpsrc.c:855:15: error: #if with no expression
+
+2013-09-04 15:50:42 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: handle unseekable streams
+ Handle streams that we can't seek in and ignore them in the
+ seek logic.
+
+2013-09-04 15:25:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: only check video compression for video streams
+ Or else we might deref a stream with a NULL strf.vids and segfault
+
+2013-06-18 13:27:20 +0100 Alex Ashley <bugzilla@ashley-family.net>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/ftypcc.h:
+ * gst/isomp4/gstrtpxqtdepay.c:
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux_fourcc.h:
+ * gst/isomp4/qtdemux_types.c:
+ qtdemux: Add support for the avc3 sample entry format of the AVC file format
+ Amendment 2 of ISO/IEC 14496-15 (AVC file format) is defining a new
+ structure for fragmented MP4 called "avc3". The principal difference
+ between AVC1 and AVC3 is the location of the codec initialisation
+ data (e.g. SPS, PPS). In AVC1 this data is placed in the initial
+ MOOV box (moov.trak.mdia.minf.stbl.stsd.avc1) but in AVC3 this data
+ goes in the first sample of every fragment (i.e. the first sample in
+ each mdat box). The principal reason for avc3 is to make it easier
+ for client implementations, because it removes the requirement to
+ insert the SPS+PPS in to the decoder pipeline every time there is a
+ representation change.
+ This commit adds support for the "avc3" atom, which is almost identical
+ to the "avc1" atom, except it does not contain any SPS or PPS data.
+ https://bugzilla.gnome.org/show_bug.cgi?id=702004
+
+2013-09-04 00:27:50 +0200 Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: Don't set EOS to FALSE when the collectpad *is* EOS
+ https://bugzilla.gnome.org/show_bug.cgi?id=707238
+
+2013-09-03 17:32:41 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: cleanup on error after state change
+ https://bugzilla.gnome.org/show_bug.cgi?id=707229
+
+2013-09-03 11:23:24 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/udp/gstudpsrc.c:
+ * gst/udp/gstudpsrc.h:
+ udpsrc: Bind to multicast addresses on non-Windows systems
+ On Windows it's not possible to bind to a multicast address
+ but the OS will make sure to filter out all packets that
+ arrive not for the multicast address the socket joined.
+ On Linux and others it is necessary to bind to a multicast
+ address to let the OS filter out all packets that are received
+ on the same port but for different addresses than the multicast
+ address
+ And deprecate the multicast-group property and replace it with the
+ address property.
+ https://bugzilla.gnome.org/show_bug.cgi?id=707042
+
+2013-09-03 10:10:01 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: Free GstBaseParseFrame if pushing a header failed
+
+2013-09-02 16:02:37 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: Refactor address resolval into its own function
+
+2013-09-02 23:00:29 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/replaygain/gstrganalysis.c:
+ replaygain: fix taglist leak in rganalysis
+ And add some FIXMEs.
+
+2013-09-02 22:50:58 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * tests/check/elements/rganalysis.c:
+ tests: rganalysis: rename function for clarity
+
+2013-03-18 14:32:07 +0100 Christoph Reiter <reiter.christoph@gmail.com>
+
+ * tests/check/elements/rganalysis.c:
+ tests: fix skipped rganalysis tests
+ In 0.10 elements would post tag messages on the bus
+ directly, and rganalysis would only post a tag message
+ when it changed tags. In 1.0, only sinks post tag
+ messages when they receive the serialised tag event.
+ This means that we get an additional tag message on
+ the bus now where we didn't expect one before.
+ https://bugzilla.gnome.org/show_bug.cgi?id=695090
+
+2013-09-02 11:46:52 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: Properly propagate downstream flow returns upstream
+ https://bugzilla.gnome.org/show_bug.cgi?id=707229
+
+2013-09-01 21:18:38 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * ext/shout2/gstshout2.c:
+ * gst/avi/gstavi.c:
+ * gst/isomp4/isomp4-plugin.c:
+ * gst/rtsp/gstrtsp.c:
+ * sys/sunaudio/gstsunaudio.c:
+ * sys/v4l2/gstv4l2.c:
+ Don't use setlocale in plugins()
+ Only apps should call setlocale(), not libraries.
+
+2013-08-29 13:15:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst/rtp/gstrtpmpvpay.c:
+ rtpmpvpay: Fix RTP buffer allocation in rtpmpvpay
+ RTP buffer allocation should not be done with padding for the specific MPEG2
+ header as the padding is done at the end of the buffer and the last byte is
+ the size of the padding.
+ https://bugzilla.gnome.org/show_bug.cgi?id=706970
+
+2013-08-28 10:51:32 +0200 Bernhard Miller <bernhard.miller@streamunlimited.com>
+
+ * gst/autodetect/gstautovideosink.c:
+ * gst/autodetect/gstautovideosink.h:
+ autovideosink: add sync property
+ https://bugzilla.gnome.org/show_bug.cgi?id=706955
+
+2013-08-28 07:15:00 +0200 Bernhard Miller <bernhard.miller@streamunlimited.com>
+
+ * gst/autodetect/gstautoaudiosink.c:
+ * gst/autodetect/gstautoaudiosink.h:
+ autoaudiosink: introduce sync property
+ https://bugzilla.gnome.org/show_bug.cgi?id=706955
+
+2013-08-27 17:33:40 -0300 Thiago Santos <thiago.sousa.santos@collabora.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: push buffers after segment stop until reaching a keyframe
+ This should make decoders able to precisely push buffers until the stop
+ time in case they need the next keyframe to do it.
+ Also, according to gst_segment_clip, it should only push a buffer that
+ the starting ts is strictly smaller than the segment stop, so we change
+ the min < comparison for <=
+
+2013-08-28 13:26:47 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * configure.ac:
+ Back to development
+
=== release 1.1.4 ===
-2013-08-28 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2013-08-28 12:52:25 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 1.1.4
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-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-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ * gst-plugins-good.doap:
+ * gst/audiofx/audiopanoramaorc-dist.c:
+ * win32/common/config.h:
+ Release 1.1.4
+
+2013-08-28 12:52:16 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ Update .po files
2013-08-28 12:32:10 +0200 Sebastian Dröge <slomo@circular-chaos.org>
-This is GStreamer Good Plugins 1.1.4
+This is GStreamer Good Plugins 1.1.90
-Release notes for GStreamer Good Plugins 1.1.4
+Release notes for GStreamer Good Plugins 1.1.90
The GStreamer team is proud to announce a new bug-fix release
Bugs fixed in this release
+ * 646963 : rtpmanager: Only update last_rtcp_send_time when actually sending a report
+ * 687657 : rtpsession: Demux RTCP buffers from the RTP stream
+ * 695090 : rganalysis: fix tests
+ * 702004 : qtdemux: add support for the avc3 sample entry format of the AVC file format
+ * 704950 : videomixer: add colorspace conversion
* 705371 : souphttpsrc: Does network operations from the state change thread
- * 590768 : GstPulseSrc should allow swapping the device used by the stream
- * 637754 : multipartdemux: time stamp output buffer based on first input buffer not last
- * 694445 : pulsesink: add support for AAC pass-through
- * 700264 : qtdemux: ignores first editlist
- * 702988 : gdkpixbufoverlay: crashes if any property changes during playback when location property is not set
- * 705108 : rtpgstpay: Add a config-interval property
- * 705442 : matroskademux: prints warnings on seek
- * 705449 : avidemux: sends stream_start event without group_id
- * 705553 : rtph264pay: Entries of SPS and PPS duplicated
- * 705760 : rtspsrc produces GLib error
- * 705959 : souphttpsrc deprecated symbols
- * 706076 : qtdemux: failed assertion for fragmented mp4 (DASH) streams
- * 706642 : avimux: unmap the correct buffer
+ * 706955 : autoaudiosink/autovideosink: Introduce " sync " property
+ * 707042 : udpsrc binds to INADDR_ANY instead of multicast group address
+ * 707303 : flacenc: 'Got data flow before stream-start event' warnings
+ * 707340 : qtmux: should NOT use PTS if DTS is missing
+ * 707411 : qtmux: what is purpose of MAX_TOLERATED_LATENESS?
+ * 707530 : qtdemux: Handle segments correctly in push mode seeks
+ * 707688 : matroskademux: leaking buffer and caps when parsing attachments
+ * 707734 : rtp: add payloader and depayloader for 24bit raw audio
+ * 707758 : v4l2: Incorrect UV plane stride value for NV12/NV21 formats
+ * 707886 : videomixer: build fails due to unrenamed files if orc is not installed
+ * 707933 : matroskademux: Wrong UTF8 detection causes wrong detection of subtitle encoding
+ * 708131 : videomixer: undefined reference to `videomixer_video_convert_orc_convert_I420_BGRA'
==== Download ====
Contributors to this release
- * Akihiro Tsukada
- * Andoni Morales Alastruey
- * Chris Bass
- * David Schleef
- * Edward Hervey
- * Kishore Arepalli
- * Lubosz Sarnecki
+ * Alex Ashley
+ * Andreea Fulger
+ * Bernhard Miller
+ * Christoph Reiter
+ * David Holroyd
+ * Jan Schmidt
* Matej Knopp
* Mathieu Duponchelle
- * Michael Olbrich
* Olivier Crête
+ * Rico Tzschichholz
* Sebastian Dröge
- * Sjoerd Simons
+ * Seán de Búrca
+ * Thiago Santos
* Thibault Saunier
* Tim-Philipp Müller
* Wim Taymans
- * Youness Alaoui
\ 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],[1.1.4.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-good])
+AC_INIT([GStreamer Good Plug-ins],[1.1.90],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-good])
AG_GST_INIT
[GStreamer API Version])
AG_GST_LIBTOOL_PREPARE
-AS_LIBTOOL(GST, 104, 0, 104)
+AS_LIBTOOL(GST, 190, 0, 190)
dnl *** required versions of GStreamer stuff ***
-GST_REQ=1.1.4.1
-GSTPB_REQ=1.1.4.1
+GST_REQ=1.1.90
+GSTPB_REQ=1.1.90
dnl *** autotools stuff ****
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Multicast Group</NICK>
-<BLURB>The Address of multicast group to join.</BLURB>
+<BLURB>The Address of multicast group to join. DEPRECATED: Use address property instead.</BLURB>
<DEFAULT>"0.0.0.0"</DEFAULT>
</ARG>
</ARG>
<ARG>
+<NAME>GstUDPSrc::address</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Address</NICK>
+<BLURB>Address to receive packets for. This is equivalent to the multicast-group property for now.</BLURB>
+<DEFAULT>"0.0.0.0"</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstSMPTE::border</NAME>
<TYPE>gint</TYPE>
<RANGE>>= 0</RANGE>
</ARG>
<ARG>
+<NAME>GstAutoAudioSink::sync</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Sync</NICK>
+<BLURB>Sync on the clock.</BLURB>
+<DEFAULT>TRUE</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstAutoVideoSink::filter-caps</NAME>
<TYPE>GstCaps*</TYPE>
<RANGE></RANGE>
</ARG>
<ARG>
+<NAME>GstAutoVideoSink::sync</NAME>
+<TYPE>gboolean</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Sync</NICK>
+<BLURB>Sync on the clock.</BLURB>
+<DEFAULT>TRUE</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstOsxAudioSink::device</NAME>
<TYPE>gint</TYPE>
<RANGE>>= 0</RANGE>
GstRtpJ2KDepay
GstRtpJPEGDepay
GstRtpL16Depay
+ GstRtpL24Depay
GstRtpMP1SDepay
GstRtpMP2TDepay
GstRtpMP4ADepay
GstRtpG722Pay
GstRtpG726Pay
GstRtpL16Pay
+ GstRtpL24Pay
GstRtpPcmaPay
GstRtpPcmuPay
GstRTPDVPay
<description>Source for video data via IEEE1394 interface</description>
<filename>../../ext/raw1394/.libs/libgst1394.so</filename>
<basename>libgst1394.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>ASCII Art video sink</description>
<filename>../../ext/aalib/.libs/libgstaasink.so</filename>
<basename>libgstaasink.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>ALaw audio conversion routines</description>
<filename>../../gst/law/.libs/libgstalaw.so</filename>
<basename>libgstalaw.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<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>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>RGBA from/to AYUV colorspace conversion preserving the alpha channel</description>
<filename>../../gst/alpha/.libs/libgstalphacolor.so</filename>
<basename>libgstalphacolor.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>APEv1/2 tag reader</description>
<filename>../../gst/apetag/.libs/libgstapetag.so</filename>
<basename>libgstapetag.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Audio effects plugin</description>
<filename>../../gst/audiofx/.libs/libgstaudiofx.so</filename>
<basename>libgstaudiofx.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Parsers for various audio formats</description>
<filename>../../gst/audioparsers/.libs/libgstaudioparsers.so</filename>
<basename>libgstaudioparsers.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>parses au streams</description>
<filename>../../gst/auparse/.libs/libgstauparse.so</filename>
<basename>libgstauparse.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<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>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>AVI stream handling</description>
<filename>../../gst/avi/.libs/libgstavi.so</filename>
<basename>libgstavi.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Colored ASCII Art video sink</description>
<filename>../../ext/libcaca/.libs/libgstcacasink.so</filename>
<basename>libgstcacasink.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Cairo-based elements</description>
<filename>../../ext/cairo/.libs/libgstcairo.so</filename>
<basename>libgstcairo.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Audio Cutter to split audio into non-silent bits</description>
<filename>../../gst/cutter/.libs/libgstcutter.so</filename>
<basename>libgstcutter.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>elements for testing and debugging</description>
<filename>../../gst/debugutils/.libs/libgstdebug.so</filename>
<basename>libgstdebug.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Deinterlacer</description>
<filename>../../gst/deinterlace/.libs/libgstdeinterlace.so</filename>
<basename>libgstdeinterlace.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>DTMF plugins</description>
<filename>../../gst/dtmf/.libs/libgstdtmf.so</filename>
<basename>libgstdtmf.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>DV demuxer and decoder based on libdv (libdv.sf.net)</description>
<filename>../../ext/dv/.libs/libgstdv.so</filename>
<basename>libgstdv.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>effect plugins from the effectv project</description>
<filename>../../gst/effectv/.libs/libgsteffectv.so</filename>
<basename>libgsteffectv.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>GStreamer audio equalizers</description>
<filename>../../gst/equalizer/.libs/libgstequalizer.so</filename>
<basename>libgstequalizer.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>The FLAC Lossless compressor Codec</description>
<filename>../../ext/flac/.libs/libgstflac.so</filename>
<basename>libgstflac.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>FLV muxing and demuxing plugin</description>
<filename>../../gst/flv/.libs/libgstflv.so</filename>
<basename>libgstflv.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>FLC/FLI/FLX video decoder</description>
<filename>../../gst/flx/.libs/libgstflxdec.so</filename>
<basename>libgstflxdec.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>GdkPixbuf-based image decoder, overlay and sink</description>
<filename>../../ext/gdk_pixbuf/.libs/libgstgdkpixbuf.so</filename>
<basename>libgstgdkpixbuf.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>GOOM visualization filter</description>
<filename>../../gst/goom/.libs/libgstgoom.so</filename>
<basename>libgstgoom.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>GOOM 2k1 visualization filter</description>
<filename>../../gst/goom2k1/.libs/libgstgoom2k1.so</filename>
<basename>libgstgoom2k1.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Demux ICY tags from a stream</description>
<filename>../../gst/icydemux/.libs/libgsticydemux.so</filename>
<basename>libgsticydemux.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Demux ID3v1 and ID3v2 tags from a file</description>
<filename>../../gst/id3demux/.libs/libgstid3demux.so</filename>
<basename>libgstid3demux.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Still frame stream generator</description>
<filename>../../gst/imagefreeze/.libs/libgstimagefreeze.so</filename>
<basename>libgstimagefreeze.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Audio interleaver/deinterleaver</description>
<filename>../../gst/interleave/.libs/libgstinterleave.so</filename>
<basename>libgstinterleave.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>ISO base media file format support (mp4, 3gpp, qt, mj2)</description>
<filename>../../gst/isomp4/.libs/libgstisomp4.so</filename>
<basename>libgstisomp4.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>JACK audio elements</description>
<filename>../../ext/jack/.libs/libgstjack.so</filename>
<basename>libgstjack.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>JPeg plugin library</description>
<filename>../../ext/jpeg/.libs/libgstjpeg.so</filename>
<basename>libgstjpeg.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Audio level plugin</description>
<filename>../../gst/level/.libs/libgstlevel.so</filename>
<basename>libgstlevel.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Matroska and WebM stream handling</description>
<filename>../../gst/matroska/.libs/libgstmatroska.so</filename>
<basename>libgstmatroska.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>MuLaw audio conversion routines</description>
<filename>../../gst/law/.libs/libgstmulaw.so</filename>
<basename>libgstmulaw.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Reads/Writes buffers from/to sequentially named files</description>
<filename>../../gst/multifile/.libs/libgstmultifile.so</filename>
<basename>libgstmultifile.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>multipart stream manipulation</description>
<filename>../../gst/multipart/.libs/libgstmultipart.so</filename>
<basename>libgstmultipart.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Template for a video filter</description>
<filename>../../gst/debugutils/.libs/libgstnavigationtest.so</filename>
<basename>libgstnavigationtest.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Open Sound System (OSS) version 4 support for GStreamer</description>
<filename>../../sys/oss4/.libs/libgstoss4audio.so</filename>
<basename>libgstoss4audio.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>OSS (Open Sound System) support for GStreamer</description>
<filename>../../sys/oss/.libs/libgstossaudio.so</filename>
<basename>libgstossaudio.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>PNG plugin library</description>
<filename>../../ext/libpng/.libs/libgstpng.so</filename>
<basename>libgstpng.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>PulseAudio plugin library</description>
<filename>../../ext/pulse/.libs/libgstpulse.so</filename>
<basename>libgstpulse.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>ReplayGain volume normalization</description>
<filename>../../gst/replaygain/.libs/libgstreplaygain.so</filename>
<basename>libgstreplaygain.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Real-time protocol plugins</description>
<filename>../../gst/rtp/.libs/libgstrtp.so</filename>
<basename>libgstrtp.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
</pads>
</element>
<element>
+ <name>rtpL24depay</name>
+ <longname>RTP audio depayloader</longname>
+ <class>Codec/Depayloader/Network/RTP</class>
+ <description>Extracts raw 24-bit audio from RTP packets</description>
+ <author>Zeeshan Ali <zak147@yahoo.com>,Wim Taymans <wim.taymans@gmail.com>,David Holroyd <dave@badgers-in-foil.co.uk></author>
+ <pads>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>application/x-rtp, media=(string)audio, clock-rate=(int)[ 1, 2147483647 ], encoding-name=(string)L24</details>
+ </caps>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>audio/x-raw, format=(string)S24BE, layout=(string)interleaved, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
+ </caps>
+ </pads>
+ </element>
+ <element>
+ <name>rtpL24pay</name>
+ <longname>RTP audio payloader</longname>
+ <class>Codec/Payloader/Network/RTP</class>
+ <description>Payload-encode Raw 24-bit audio into RTP packets (RFC 3190)</description>
+ <author>Wim Taymans <wim.taymans@gmail.com>,David Holroyd <dave@badgers-in-foil.co.uk></author>
+ <pads>
+ <caps>
+ <name>sink</name>
+ <direction>sink</direction>
+ <presence>always</presence>
+ <details>audio/x-raw, format=(string)S24BE, layout=(string)interleaved, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ]</details>
+ </caps>
+ <caps>
+ <name>src</name>
+ <direction>source</direction>
+ <presence>always</presence>
+ <details>application/x-rtp, media=(string)audio, payload=(int)[ 96, 127 ], clock-rate=(int)[ 1, 2147483647 ], encoding-name=(string)L24, channels=(int)[ 1, 2147483647 ]</details>
+ </caps>
+ </pads>
+ </element>
+ <element>
<name>rtpac3depay</name>
<longname>RTP AC3 depayloader</longname>
<class>Codec/Depayloader/Network/RTP</class>
<description>RTP session management plugin library</description>
<filename>../../gst/rtpmanager/.libs/libgstrtpmanager.so</filename>
<basename>libgstrtpmanager.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>transfer data via RTSP</description>
<filename>../../gst/rtsp/.libs/libgstrtsp.so</filename>
<basename>libgstrtsp.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Shape Wipe transition filter</description>
<filename>../../gst/shapewipe/.libs/libgstshapewipe.so</filename>
<basename>libgstshapewipe.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Sends data to an icecast server using libshout2</description>
<filename>../../ext/shout2/.libs/libgstshout2.so</filename>
<basename>libgstshout2.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</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>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>libsoup HTTP client src/sink</description>
<filename>../../ext/soup/.libs/libgstsouphttpsrc.so</filename>
<basename>libgstsouphttpsrc.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Run an FFT on the audio signal, output spectrum data</description>
<filename>../../gst/spectrum/.libs/libgstspectrum.so</filename>
<basename>libgstspectrum.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Speex plugin library</description>
<filename>../../ext/speex/.libs/libgstspeex.so</filename>
<basename>libgstspeex.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Tag writing plug-in based on taglib</description>
<filename>../../ext/taglib/.libs/libgsttaglib.so</filename>
<basename>libgsttaglib.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>transfer data via UDP</description>
<filename>../../gst/udp/.libs/libgstudp.so</filename>
<basename>libgstudp.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>elements for Video 4 Linux</description>
<filename>../../sys/v4l2/.libs/libgstvideo4linux2.so</filename>
<basename>libgstvideo4linux2.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>resizes a video by adding borders or cropping</description>
<filename>../../gst/videobox/.libs/libgstvideobox.so</filename>
<basename>libgstvideobox.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Crops video into a user-defined region</description>
<filename>../../gst/videocrop/.libs/libgstvideocrop.so</filename>
<basename>libgstvideocrop.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Video filters plugin</description>
<filename>../../gst/videofilter/.libs/libgstvideofilter.so</filename>
<basename>libgstvideofilter.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Video mixer</description>
<filename>../../gst/videomixer/.libs/libgstvideomixer.so</filename>
<basename>libgstvideomixer.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>VP8 plugin</description>
<filename>../../ext/vpx/.libs/libgstvpx.so</filename>
<basename>libgstvpx.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Encode raw audio into WAV</description>
<filename>../../gst/wavenc/.libs/libgstwavenc.so</filename>
<basename>libgstwavenc.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Wavpack lossless/lossy audio format handling</description>
<filename>../../ext/wavpack/.libs/libgstwavpack.so</filename>
<basename>libgstwavpack.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Parse a .wav file into raw audio</description>
<filename>../../gst/wavparse/.libs/libgstwavparse.so</filename>
<basename>libgstwavparse.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>X11 video input plugin using standard Xlib calls</description>
<filename>../../sys/ximage/.libs/libgstximagesrc.so</filename>
<basename>libgstximagesrc.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Encodes a YUV frame into the yuv4mpeg format (mjpegtools)</description>
<filename>../../gst/y4m/.libs/libgsty4menc.so</filename>
<basename>libgsty4menc.so</basename>
- <version>1.1.4</version>
+ <version>1.1.90</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<release>
<Version>
+ <revision>1.1.90</revision>
+ <branch>1.1</branch>
+ <name></name>
+ <created>2013-09-19</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-plugins-good/gst-plugins-good-1.1.90.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.1.4</revision>
<branch>1.1</branch>
<name></name>
#define GST_PACKAGE_ORIGIN "Unknown package origin"
/* GStreamer package release date/time for plugins as YYYY-MM-DD */
-#define GST_PACKAGE_RELEASE_DATETIME "2013-08-28"
+#define GST_PACKAGE_RELEASE_DATETIME "2013-09-19"
/* Define if static plugins should be built */
#undef GST_PLUGIN_BUILD_STATIC
*/
#undef LT_OBJDIR
-/* Define to 1 if your C compiler doesn't accept -c and -o together. */
-#undef NO_MINUS_C_MINUS_O
-
/* Name of package */
#define PACKAGE "gst-plugins-good"
#define PACKAGE_NAME "GStreamer Good Plug-ins"
/* Define to the full name and version of this package. */
-#define PACKAGE_STRING "GStreamer Good Plug-ins 1.1.4"
+#define PACKAGE_STRING "GStreamer Good Plug-ins 1.1.90"
/* 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 "1.1.4"
+#define PACKAGE_VERSION "1.1.90"
/* directory where plugins are located */
#ifdef _DEBUG
#undef TARGET_CPU
/* Version number of package */
-#define VERSION "1.1.4"
+#define VERSION "1.1.90"
/* old wavpack API */
#undef WAVPACK_OLD_API
projects / platform / upstream / gst-plugins-good.git / commitdiff