=== release 1.1.3 === 2013-07-29 Sebastian Dröge <sebastian.droege@collabora.co.uk> * configure.ac: releasing 1.1.3 2013-07-29 12:12:41 +0200 Sebastian Dröge <slomo@circular-chaos.org> * gst/avi/gstavidemux.c: * gst/flv/gstflvdemux.c: * gst/isomp4/qtdemux.c: * gst/matroska/matroska-demux.c: gst: Don't swap start/stop for negative rates in the SEGMENT query 2013-07-29 11:18:40 +0200 Matej Knopp <matej.knopp@gmail.com> * gst/isomp4/qtdemux.c: qtdemux: Check for data size when parsing h264 codec data from strf atom 2013-07-29 10:53:54 +0200 Sebastian Dröge <slomo@circular-chaos.org> * gst/matroska/matroska-demux.c: matroskademux: Implement SEGMENT query 2013-07-29 10:53:47 +0200 Sebastian Dröge <slomo@circular-chaos.org> * gst/flv/gstflvdemux.c: flvdemux: Implement SEGMENT query 2013-07-29 10:50:59 +0200 Sebastian Dröge <slomo@circular-chaos.org> * gst/avi/gstavidemux.c: avidemux: Implement SEGMENT query 2013-07-27 18:10:22 +0200 Matej Knopp <matej.knopp@gmail.com> * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_fourcc.h: qtdemux: Support H264 fourcc https://bugzilla.gnome.org/show_bug.cgi?id=704996 2013-07-28 18:09:33 +0200 Sebastian Dröge <slomo@circular-chaos.org> * ext/flac/gstflacenc.c: flacenc: Fix handling of image tags The caps should be used to get the mimetype and there is only an info structure for the GstSample if the image-type is not NONE. 2013-07-28 18:04:32 +0200 Sebastian Dröge <slomo@circular-chaos.org> * ext/flac/gstflacenc.c: flacenc: Don't crash if there is no image tag information https://bugzilla.gnome.org/show_bug.cgi?id=705018 2013-07-28 17:38:56 +0200 Sebastian Dröge <slomo@circular-chaos.org> * gst/avi/gstavidemux.c: avidemux: Fix duration reporting in push mode https://bugzilla.gnome.org/show_bug.cgi?id=700933 2013-07-28 17:32:27 +0200 Sebastian Dröge <slomo@circular-chaos.org> * gst/avi/gstavidemux.c: avidemux: Don't forget unmapping and unreffing buffer 2013-07-26 21:06:17 +0200 Matej Knopp <matej.knopp@gmail.com> * gst/avi/gstavidemux.c: avidemux: unmap buffer https://bugzilla.gnome.org/show_bug.cgi?id=704951 2013-07-26 22:31:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: session: don't make buffer writable prematurely There is no reason to make the SR buffer writable at this point. This is better delayed until needed. 2013-07-26 22:25:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: session: ignore RTCP for inactive sources 2013-07-26 22:25:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: session: small cleanup 2013-07-26 17:17:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.h: session: handle partial RTCP report blocks When we have more SSRCs to report than what fit in an RTCP packet, use a generation counter to make sure all of them end up in a packet eventually. 2013-07-26 17:23:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: session: create SSRC before doing session cleanup Make the internal source before we do session cleanup 2013-07-26 17:21:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: session: reorganize the report block code 2013-07-26 16:02:01 +0200 Matej Knopp <matej.knopp@gmail.com> * gst/matroska/matroska-demux.c: matroskademux: fix memory leak in check_subtitle_buffer https://bugzilla.gnome.org/show_bug.cgi?id=704921 2013-07-26 14:21:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: session: refactor active and sender checks 2013-07-26 12:06:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: session: remove internal sources on timeout When an internal source times out and becomes a receiver, remove it. 2013-07-26 11:47:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: session: create an internal source for RTCP When we need to do RTCP and we don't have an internal source yet, make one. 2013-07-26 10:47:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: session: remove old code to change SSRC Remove code used to change the SSRC after a collision. We now send a RECONFIGURE event upstream to make the upstream element change the SSRC. 2013-07-26 10:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsource.c: source: don't update packet SSRC Remove the code to update the SSRC in packets, it can never be called now that we always use a source with matching packet SSRC. 2013-07-26 10:24:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: delay allocation of internal source Allocate the internal source when we receive a caps with the SSRC or when we see a buffer with the SSRC. 2013-07-26 10:00:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: session: generate reconfigure on collision When we detect a collision, change the SSRC that we suggest upstream and trigger RECONFIGURE. This should make upstream select a new SSRC. 2013-07-26 09:37:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: produce RTCP for all internal sources Loop over all the internal sources and produce RTCP. We also need to queue the RTCP packets and send them when we are finished. 2013-07-26 01:40:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: deprecate internal source and ssrc properties Deprecate the internal source and internal ssrc properties. There might be more than one internal source. 2013-07-26 01:29:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: session: internal sources don't use probation 2013-07-26 01:24:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: session: give caps to session Let the session parse the caps and update its SSRC when needed. 2013-07-26 01:14:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: make method to suggest available SSRC Make a method to suggest the best available SSRC. This is the SSRC of the last created internal source and is used to instruct upstream to produce this SSRC. 2013-07-26 01:01:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: keep SDES and set on new internal sources Keep track of the SDES ourselves and set it on all newly created internal sources. 2013-07-26 00:48:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: session: make method to make internal sources Add a method to obtain an internal source and use it to create our internal source 2013-07-26 00:29:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpstats.h: session: count internal sources and how many are senders 2013-07-26 00:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: rtpsession: separate BYE marking and scheduling First mark sources with BYE and then schedule the BYE RTCP message. 2013-07-25 23:56:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: session: get SSRC from RTCP packet itself Get the SSRC from the RTCP packet instead. 2013-07-25 23:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: session: fix bandwidth calculation We iterate over all sources and the internal one is also in the hashtable so avoid adding it twice. 2013-07-25 23:38:08 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: session: add some docs 2013-07-25 23:11:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: session: Rearrange RTCP reporting a little Make a function to generate an RTCP packet for a source, pass the source as a parameter. Move timeout of collisions to session cleanup phase. 2013-07-25 22:39:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: session: move check for is_early around Move the check for the early RTCP to where it is needed and used. 2013-07-25 17:35:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: session: parse packet outside of the session lock 2013-07-25 17:34:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: session: do nicer checks for internal sources 2013-07-25 17:15:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: session: let source keep track if it sent BYE 2013-07-25 17:06:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsource.c: source: reset more 2013-07-25 16:49:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: source: also use the source for bye_reason Store the BYE reason in our internal source object. Rename the methods on the source object a little because now the BYE can be received in RTCP or set when the session wants to send BYE. 2013-07-25 16:24:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: session: configure sdes with structure only Remove code to configure the SDES with methods and types, only allow configuration with GstStructure 2013-07-25 15:56:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: session: refactor add and find source Make functions to find and add a source to the hashtable. 2013-07-25 15:43:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: remove source from sync_rtcp We don't need to know the sender source of the session in the callback, the SR packet is for all participants in the session. 2013-07-24 14:18:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: add some more debug 2013-07-15 17:11:45 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> * gst/audioparsers/Makefile.am: * gst/audioparsers/gstaacparse.c: * gst/audioparsers/gstaacparse.h: aacparse: allow conversion from ADTS to raw AAC Some muxers (eg, qtmux) only support raw AAC, so this allows linking an encoder that outputs ADTS only to those muxers. The conversion is simple (omit the first 7 or 9 bytes of the frame), but has to be done in pre_push instead of handle_frame as 1.0 does not seem to allow skipping bytes there as 0.10 used to. Other conversions are not supported (yet). 2013-07-15 17:15:44 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> * gst/audioparsers/gstaacparse.c: aacparse: fix object_type parsing off-by-one in ADTS frame According to http://wiki.multimedia.cx/index.php?title=ADTS, the value stored in ADTS headers is one less than the object type of the AAC stream. A look at ffmpeg shows it also adds 1 to the value read off the ADTS header. Note that this might break other things that happen to have an inverse off by one to match the existing code. 2013-07-25 11:13:01 -0300 Thiago Santos <thiago.sousa.santos@collabora.com> * gst/avi/gstavidemux.c: avidemux: fix seqnum handling for seeks Use the same seqnum as the seek for flushes/segments that are caused by the seek. Also do the same for segment events Fixes #676242 2013-07-25 01:39:58 -0300 Thiago Santos <thiago.sousa.santos@collabora.com> * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: fix seqnum handling for seeks Use the same seqnum as the seek for flushes/segments that are caused by the seek. Also do the same for segment events Fixes #676242 2013-07-25 01:11:31 -0300 Thiago Santos <thiago.sousa.santos@collabora.com> * gst/isomp4/qtdemux.c: qtdemux: correctly handle seqnum for seeks and segments Use the same seqnum on messages and events for derived events. Fixed for flushes / stream-start / segment after a seek, and segment after a segment. Fixes #676242 2013-07-12 20:01:42 +0200 Arnaud Vrac <avrac@freebox.fr> * ext/soup/gstsouphttpsrc.c: souphttpsrc: always ignore HEAD errors https://bugzilla.gnome.org/show_bug.cgi?id=704241 2013-07-25 14:26:07 +0200 Sebastian Dröge <slomo@circular-chaos.org> * ext/jpeg/gstjpegenc.c: jpegenc: Clean up reset/start/stop handling 2013-07-25 14:13:10 +0200 Sebastian Dröge <slomo@circular-chaos.org> * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegdec.h: jpegdec: Use base class error handling function instead of replicating it here 2013-07-25 14:12:56 +0200 Sebastian Dröge <slomo@circular-chaos.org> * ext/jpeg/gstjpegdec.c: jpegdec: Clean up handling of reset/start/stop 2013-07-25 10:41:22 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * tests/files/id3-407349-1.tag: * tests/files/id3-407349-2.tag: * tests/files/id3-447000-wcop.tag: tests: fix test ID3 tags up not to rely on dodgy typefinding code Change 0xff 0xfb 'mp3' marker to 'fLaC' marker, so we can fix the typefinder. https://bugzilla.gnome.org/show_bug.cgi?id=681368 2013-07-25 08:22:45 +0200 Alessandro Decina <alessandro.d@gmail.com> * sys/osxaudio/gstosxaudiosink.c: osxaudiosink: intersect the probed caps with the filter passed to get_caps() 2013-07-24 14:17:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/gstrtpbin.c: bin: fix compilation 2013-07-24 12:42:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtp/gstrtpvrawdepay.c: vrawdepay: fix UYVP format 2013-07-24 12:41:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtp/gstrtpvrawpay.c: vrawpay: fix UYVP format 2013-07-24 12:41:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtp/gstrtpvrawpay.c: vrawpay: fix caps 2013-07-24 10:49:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: fix locking Take the lock earlier so that we do things that follow with the right locking. 2013-07-23 17:40:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: rtpsession: don't use invalid times in RTCP timeouts An invalid timeout can be calculated when we disabled RTCP by setting the bandwidth to 0. Make sure all code can handle this case. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674626 2013-07-23 17:38:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: rtpsession: lock session when changing bandwidth Take the session lock when changing the bandwidth properties so that we don't end up with inconsistent behaviour. 2013-07-23 17:37:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/rtpmanager/rtpsession.c: session: reset some RTCP variables The early_send time was set to 0 and always triggering an early RTCP packet. 2013-07-23 15:03:31 +0200 Edward Hervey <edward@collabora.com> * gst/isomp4/qtdemux.c: qtdemux: Add all the mpeg XDCAM variants This should cover all known XDCAM variants (which are all mpeg2 video) Fixes #672227 2013-07-03 18:41:42 +0200 Carlos Rafael Giani <dv@pseudoterminal.org> * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: added custom downstream sync event rtpbin can now send a custom in-band downstream event which informs downstream that the bin has received an RTCP SR packet. This is useful for applications which want to drop the initial unsynchronized received RTP packets. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703560 Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org> 2013-07-22 18:00:16 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * gst/deinterlace/gstdeinterlace.c: deinterlace: fix on-the-fly changing of "mode" and "fields" properties We call setcaps() to reconfigure ourselves, but we need to pass the current *sink* caps, not the source caps then. Also fix a caps leak. https://bugzilla.gnome.org/show_bug.cgi?id=641599 2013-07-22 15:23:39 +0200 Sebastian Dröge <slomo@circular-chaos.org> * gst/wavparse/gstwavparse.c: wavparse: Add support for group-id in the stream-start event 2013-07-22 15:23:20 +0200 Sebastian Dröge <slomo@circular-chaos.org> * gst/rtsp/gstrtspsrc.c: rtspsrc: Add support for group-id in the stream-start event 2013-07-22 15:23:11 +0200 Sebastian Dröge <slomo@circular-chaos.org> * gst/rtpmanager/gstrtpsession.c: rtpsession: Add support for group-id in the stream-start event 2013-07-22 15:22:55 +0200 Sebastian Dröge <slomo@circular-chaos.org> * gst/matroska/matroska-demux.c: * gst/matroska/matroska-demux.h: matroskademux: Add support for group-id in the stream-start event 2013-07-22 15:22:47 +0200 Sebastian Dröge <slomo@circular-chaos.org> * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: Add support for group-id in the stream-start event 2013-07-22 15:22:36 +0200 Sebastian Dröge <slomo@circular-chaos.org> * gst/flv/gstflvdemux.c: * gst/flv/gstflvdemux.h: flvdemux: Add support for group-id in the stream-start event 2013-07-22 15:22:16 +0200 Sebastian Dröge <slomo@circular-chaos.org> * gst/avi/gstavidemux.c: * gst/avi/gstavidemux.h: avidemux: Add support for group-id in the stream-start event 2013-07-22 15:21:49 +0200 Sebastian Dröge <slomo@circular-chaos.org> * ext/dv/gstdvdemux.c: * ext/dv/gstdvdemux.h: dvdemux: Add support for group-id in the stream-start event 2013-07-19 22:59:15 +0200 Mathieu Duponchelle <mathieu.duponchelle@epitech.eu> * gst/videomixer/videomixer2.c: videomixer: use gst_util_uint64_scale*_round. There could be a case where: 1) you do a new set_caps after buffers have been processed. 2) ts_offset gets set to a different value, eg 0.033333333 3) your pads get EOS, but the check dor that doesn't work because you use ts_offset + a truncated value < segment.stop 4) so in the next collected, you end up comparing for example: 0.9999999999 > 1., which is false and means you don't send EOS. Also adds scale_round in two other places where it potentially could have caused problems. 2013-07-15 17:55:19 -0400 Olivier Crête <olivier.crete@collabora.com> * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_fourcc.h: qtdemux: Add WRLE support 2013-07-19 19:35:26 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux_fourcc.h: qtdemux: make files from Vivotek camera play Skip tracks of 'vivo' subtype with empty stsd instead of erroring out saying that the file is broken. https://bugzilla.gnome.org/show_bug.cgi?id=699791 2013-07-19 17:14:06 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk> * gst/isomp4/gstqtmux.c: qtmux: when streaming don't try to seek when stopping It might cause errors in sinks that are not seekable and have reported this (like e.g. fdsink) https://bugzilla.gnome.org/show_bug.cgi?id=696228 2013-07-19 17:26:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/isomp4/qtdemux.c: qtdemux: simplify some helpers Some helper functions are not needed anymore or can be simplified. 2013-07-19 17:12:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/isomp4/qtdemux.c: qtdemux: for non-raw video, move palette in caps We only need to append the palette to raw video buffers, non-raw video has the palette in the caps still. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=704292 2013-07-19 01:49:20 +0200 Arnaud Vrac <avrac@freebox.fr> * gst/isomp4/qtdemux.c: qtdemux: nitpicking in esds parsing 2013-07-19 01:49:07 +0200 Arnaud Vrac <avrac@freebox.fr> * gst/isomp4/qtdemux.c: qtdemux: set proper caps for mpeg-1 audio Remove AAC specific fields from mpeg-1 audio caps, remove assumption that the mpeg1 audio layer is 3, and set `parsed' field. https://bugzilla.gnome.org/show_bug.cgi?id=704548 2013-06-17 21:27:37 +0200 Arnaud Vrac <avrac@freebox.fr> * ext/vpx/gstvp8dec.h: * ext/vpx/gstvp8enc.h: * ext/vpx/gstvp9dec.h: * ext/vpx/gstvp9enc.h: vpx: fix compilation when encoder or decoder headers are not installed https://bugzilla.gnome.org/show_bug.cgi?id=704547 2013-07-16 20:41:15 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com> * tests/check/elements/videocrop.c: videocrop: Fix unit for GRAY16 formats 2013-07-16 22:17:17 +0200 Arnaud Vrac <avrac@freebox.fr> * gst/isomp4/qtdemux.c: qtdemux: remove chapter stream Remove all streams that are actually table of contents, since we will never need the data after parsing them. 2013-07-16 21:59:37 +0200 Arnaud Vrac <avrac@freebox.fr> * gst/isomp4/qtdemux.c: qtdemux: send gap event for sparse streams in push mode This allows to pre-roll at least if the next subtitle buffer is far away. 2013-07-16 21:56:07 +0200 Arnaud Vrac <avrac@freebox.fr> * gst/isomp4/qtdemux.c: qtdemux: do not use indexes from sparse stream when seeking in push mode This makes seeking more accurate in push mode, since the previous keyframe on a sparse stream might be far away. 2013-07-16 21:04:07 +0200 Arnaud Vrac <avrac@freebox.fr> * gst/isomp4/qtdemux.c: qtdemux: advertise subtitle streams as sparse 2013-07-17 17:11:44 +0200 Arnaud Vrac <avrac@freebox.fr> * gst/matroska/matroska-demux.c: mastrokademux: do not push discont buffers if they aren't discont Unset the discont flag instead of posssibly pushing a buffer with a flag that's still set. https://bugzilla.gnome.org/show_bug.cgi?id=682110 2013-07-17 15:10:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/isomp4/qtdemux.c: qtdemux: extract the palette from stsd Sometimes a palette is inside the stsd, extract it instead of always using the default one 2013-07-17 14:30:16 +0200 Sebastian Dröge <slomo@circular-chaos.org> * gst/goom2k1/gstgoom.c: goom2k1: Fix event handling and negotiate as soon as possible 2013-07-17 14:27:57 +0200 Sebastian Dröge <slomo@circular-chaos.org> * gst/goom/gstgoom.c: goom: Fix event handling and negotiate as soon as possible 2013-07-11 19:45:17 +0200 Andoni Morales Alastruey <ylatuya@gmail.com> * sys/osxvideo/osxvideosink.m: osxvideosink: warn about the future deprecation of the "embed" property 2013-07-17 09:56:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/isomp4/qtdemux.c: qtdemux: add support for WRAW Fixes https://bugzilla.gnome.org/show_bug.cgi?id=704292 2013-07-17 09:54:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * gst/isomp4/qtdemux.c: qtdemux: palette is appended to buffers, not in caps Fix the palette handling, in 1.0 we append the palette to the buffer instead of placing it on the caps. See also https://bugzilla.gnome.org/show_bug.cgi?id=704292 2013-07-16 15:37:49 -0400 Olivier Crête <olivier.crete@collabora.com> * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpmp2tpay.c: * gst/rtp/gstrtpmp4gpay.c: * gst/rtp/gstrtpmp4vpay.c: * gst/rtp/gstrtpmpapay.c: * gst/rtp/gstrtpmpvpay.c: rtp: Use gst_adapter_take_buffer_fast() where possible in RTP payloaders 2013-07-15 16:24:07 +0200 Arnaud Vrac <avrac@freebox.fr> * gst/isomp4/qtdemux.c: qtdemux: reset segment on flush stop cca2f555d14 introduces a regression, where the demux segment is not reset on flush stop, so the next upstream segment event will calculate an invalid base time on the new segment to be sent downstream. https://bugzilla.gnome.org/show_bug.cgi?id=704255 2013-07-06 17:20:49 +0200 Matej Knopp <matej.knopp@gmail.com> * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: qtdemux: offset samples according to edit list https://bugzilla.gnome.org/show_bug.cgi?id=700264 2013-07-14 12:50:13 +1200 Douglas Bagnall <douglas@halo.gen.nz> * tests/examples/spectrum/spectrum-example.c: level: Fix the spectrum example for 1.0 The "message" property has been replaced by "post-messages". Pre-patch output: (test_spectrum:23101): GLib-GObject-WARNING **: g_object_set_valist: object class `GstSpectrum' has no property named `message' New spectrum message, endtime 0:00:00.100000000 (test_spectrum:23101): GStreamer-CRITICAL **: gst_value_list_get_value: assertion `GST_VALUE_HOLDS_LIST (value)' failed [...] Post-patch: New spectrum message, endtime 0:00:00.100000000 band 0 (freq 400): magnitude -65.988777 dB phase 1.533397 band 1 (freq 1200): magnitude -65.545563 dB phase -0.780900 band 2 (freq 2000): magnitude -64.791946 dB phase -0.799611 band 3 (freq 2800): magnitude -64.556175 dB phase -0.063615 [...] https://bugzilla.gnome.org/show_bug.cgi?id=704179 2013-07-13 20:56:26 +0200 Matej Knopp <matej.knopp@gmail.com> * gst/audioparsers/gstaacparse.c: aacparse: be less verbose when parsing LOAS streams https://bugzilla.gnome.org/show_bug.cgi?id=704162 2013-07-12 12:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * ext/pulse/pulsesink.h: sink: alaw/mulaw caps don't have a layout property 2013-07-12 12:27:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk> * ext/pulse/pulseutil.c: pulse: relax mulaw and alaw format checks The audio library considers them as encoded formats and does not fill in the sample width. The audio ringbuffers identifies the format as alaw/mulaw and that is always 8 bits. 2013-07-11 16:13:05 +0200 Matej Knopp <matej.knopp@gmail.com> * gst/isomp4/qtdemux.c: * gst/isomp4/qtdemux.h: * gst/isomp4/qtdemux_fourcc.h: * gst/isomp4/qtdemux_types.c: qtdemux: unselect instead of ignoring disabled track, detect chapter track https://bugzilla.gnome.org/show_bug.cgi?id=704007 2013-07-11 20:41:23 -0300 Thiago Santos <thiago.sousa.santos@collabora.com> * ext/soup/gstsouphttpsrc.c: souphttpsrc: ignore errors from HEAD request HEAD requests are used to check the server headers to see if it seekable. Ignore errors from those requests as they shouldn't be critical. https://bugzilla.gnome.org/show_bug.cgi?id=704053 2013-07-12 03:24:08 +0800 Kyosuke Nekomura <supercatexpert@gmail.com> * gst/audiofx/audioecho.c: audioecho: Fix handling of delay property in PLAYING/PAUSED state https://bugzilla.gnome.org/show_bug.cgi?id=703901 2013-07-09 17:56:57 -0400 Olivier Crête <olivier.crete@collabora.com> * gst/rtpmanager/gstrtpmux.c: rtpmux: Enable proxy caps on the src pads 2013-07-11 16:57:15 +0200 Sebastian Dröge <slomo@circular-chaos.org> * configure.ac: Back to development