=== 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