gstreamer

mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-18 15:51:11 +00:00

Author	SHA1	Message	Date
Thiago Santos	cca2f555d1	qtdemux: improve reset of internal status Reset different variables on state changes to ready and when handling a flush-stop. For handling flush stops we should check if there is an upstream adaptive demuxer driving the pipeline as this means that qtdemux will get a new moov atom. For 'standard' isomedia streams this isn't true and qtdemux should keep the previous moov information around. Conflicts: gst/isomp4/qtdemux.c	2013-05-07 19:29:17 -03:00
Thiago Santos	6c69e59677	qtdemux: prepare qtdemux to accept multiple dash moovs in a row Whenever dashdemux switches bitrates it sends a new moov with the new stream configuration. qtdemux should now handle this by splitting the exposing and configuration of streams into separate functions. When the stream is new it is configured and exposed, when it is a new bitrate of an existing stream it is only reconfigured. Conflicts: gst/isomp4/qtdemux.c	2013-05-07 19:25:30 -03:00
Andre Moreira Magalhaes (andrunko)	2a7d3d1598	qtdemux: Move FLUSH_STOP/PAUSED_TO_READY handling to a reset method. Conflicts: gst/isomp4/qtdemux.c	2013-05-07 19:18:03 -03:00
Louis-Francis Ratté-Boulianne	d499b461da	qtdemux: Remove old pads when exposing streams and other general fixes. Conflicts: gst/isomp4/qtdemux.c	2013-05-07 19:18:03 -03:00
Thiago Santos	a3c19eeea1	qtdemux: handle mss streams smoothstreaming streams should be handled as a special kind of fragmented isomedia. In MSS the fragments will not contain a 'moov' atom with the media descriptions, this has to be extracted from the caps. Additionally, there should be another demuxer upstream that is likely going to be the one to answer/act on queries and events, so qtdemux has to forward those upstream.	2013-05-07 19:18:03 -03:00
Sebastian Rasmussen	9532b04947	rtpgstpay: fix invalid memory access in event handler First process event in payloader, then hand it to the base class which takes ownership of the event. https://bugzilla.gnome.org/show_bug.cgi?id=699637	2013-05-04 10:49:23 +01:00
Tim-Philipp Müller	68ac392e8f	ac3parse, dcaparse: check buffer size before trimming and unref old buffer as soon as possible.	2013-05-04 10:08:47 +01:00
Andoni Morales Alastruey	3462282b83	dcaparse: add support for "audio/x-private1-dts"	2013-05-03 13:44:23 +02:00
Andoni Morales Alastruey	4531381541	ac3parse: add support for "audio/x-private1-ac3"	2013-05-03 13:44:23 +02:00
Andoni Morales Alastruey	4a78a77e65	rtp: fix duplicated symbols with libvpx	2013-05-02 14:03:33 +02:00
Andoni Morales Alastruey	584fdbad84	goom2k1: fix duplicated symbols with goom	2013-05-02 14:03:26 +02:00
Sebastian Dröge	ae05ed4f05	rtph264pay: If the adapter is empty on EOS don't try to map its content https://bugzilla.gnome.org/show_bug.cgi?id=699314	2013-05-01 15:49:45 +02:00
Ognyan Tonchev	0584d5c4c9	matroskademux: add stream-format=raw to aac caps https://bugzilla.gnome.org/show_bug.cgi?id=699303	2013-05-01 15:47:15 +02:00
Tim-Philipp Müller	7ccb387e85	udp: log WARNING debug message if UDP multicast is likely to be broken	2013-04-27 11:25:12 +01:00
Tim-Philipp Müller	4273eccace	udpsrc: add includes to get socklen_t defined on Windows https://bugzilla.gnome.org/show_bug.cgi?id=692400	2013-04-27 11:16:54 +01:00
Yury Delendik	4bc06859d1	qtdemux: add support for VP6F VP6 flash codec https://bugzilla.gnome.org/show_bug.cgi?id=699010	2013-04-27 09:39:45 +01:00
Edward Hervey	3e5ad52c0d	monoscope: Fix debug statement	2013-04-26 12:16:49 +02:00
Alexander Schrab	3ec9673dfc	mulawdec: change base class to GstAudioDecoder https://bugzilla.gnome.org/show_bug.cgi?id=698894	2013-04-26 08:46:34 +02:00
Mathieu Duponchelle	6b153ce385	videomixer: send stream-start event.	2013-04-25 16:09:34 -03:00
Wim Taymans	1df2e623b5	docs: add some pay/depayloaders See https://bugzilla.gnome.org/show_bug.cgi?id=551631	2013-04-25 14:05:55 +02:00
Sebastian Dröge	fb0384fa0d	mulaw: Some minor memleak fixes and cleanup	2013-04-25 12:44:15 +02:00
Alexander Schrab	f0edb5fb70	mulawenc: change to gstaudioencoder base, added bitrate tags	2013-04-25 12:36:15 +02:00
Sebastian Dröge	b1af93f791	(multi)udpsink: Use separate sockets for IPv4 and IPv6 https://bugzilla.gnome.org/show_bug.cgi?id=534243	2013-04-25 12:12:23 +02:00
Sebastian Dröge	0b552150ce	dynudpsink: Use separate sockets for IPv4 and IPv6 https://bugzilla.gnome.org/show_bug.cgi?id=534243	2013-04-25 12:09:27 +02:00
Sebastian Dröge	ed8ea46424	udp: Don't include removed gstudp.h in noinst_HEADERS	2013-04-25 10:43:56 +02:00
Sebastian Dröge	afb284e3a9	udp: Remove unused enum type	2013-04-25 09:16:14 +02:00
Sebastian Dröge	a957457cc1	udp: Use the generic marshaller instead of generating marshallers	2013-04-25 09:13:51 +02:00
Sebastian Dröge	07d3363436	udpsrc: Rename instance variable from host to multi_group This is more consistent as it's used for the multicast-group property.	2013-04-25 09:07:41 +02:00
Sebastian Dröge	427673d283	udpsrc: Add bind-address property This is equivalent to multicast-group currently for backwards compatibility. In 2.0 this should be handled separately, the former only being the multicast group and the latter always being the address the socket is bound to, even if a multicast group is given.	2013-04-25 09:05:12 +02:00
Wim Taymans	5ba3fd3c63	vrawdepay: return output buffer from process Return the output buffer from the process function instead of pushing it ourselves. This way, the subclass can actually deal with the return value of the push. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=693727	2013-04-24 16:24:25 +02:00
Wim Taymans	eac9efb92e	rtp: a marker bit should translate to RESYNC A marker bit on an audio packet does not mean a DISCONT (in the GStreamer sense of missing data) but it means that the packet is the end of a talkspurt and thus a good opportunity to resync to the clock. Use the RESYNC buffer flag to note this. Real discontinuities are marked with DISCONT still when the seqnum has a GAP or when the input buffer has the DISCONT flag set. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=627204	2013-04-24 15:42:45 +02:00
Sebastian Dröge	fdb667ae00	rtpjpegdepay: Drop frame if it's less than 2 bytes large https://bugzilla.gnome.org/show_bug.cgi?id=677560	2013-04-22 10:19:29 +02:00
Sreerenj Balachandran	504360fe36	autodetect: use _plugin_feature_rank_compare API instead of duplicating the code.	2013-04-18 14:00:06 +02:00
Olivier Crête	24bb263d54	videomixer: Don't unref query, we don't own it Fixes double-unref bug. Bug found by Youness Alaoui	2013-04-16 19:29:48 -04:00
Sebastian Dröge	b0b0557c48	gst: Add better support for static plugins	2013-04-15 15:54:11 +02:00
Andoni Morales Alastruey	2ea9a66dd5	goom2k1: fix duplicated symbol with goom	2013-04-15 08:43:05 +02:00
Wim Taymans	9d7519f66e	rtp: register tag image types The rtpgstdepay needs the type to be available in order to deserialize the event.	2013-04-12 16:18:42 +01:00
Wim Taymans	b1f4587d75	rtpgstdepay: handle event parse failures better	2013-04-12 16:18:42 +01:00
Anton Belka	b959d827be	wavenc: add TOC setter support	2013-04-12 14:35:47 +02:00
Stefan Sauer	f4577ff492	wavenc: small cleanups for toc handling Don't add empty labl/note chunks. Always pass instance as the first param. Add more logging.	2013-04-12 14:35:47 +02:00
Sebastian Dröge	b17750ed9e	rtspsrc: Proxy the ntp-sync property of rtpbin	2013-04-12 12:58:50 +02:00
Sebastian Dröge	53dae1585e	rtspsrc: Give the manager always the name "manager" This allows to use the GstChildProxy interface to adjust properties on it.	2013-04-12 12:51:05 +02:00
Anton Belka	bda2703e88	wavenc: add 'note' chunk support	2013-04-11 20:47:18 +02:00
Wim Taymans	f8013487c9	rtspsrc: add support for NetClientClock When the server suggests a GstNetTimeProvider in the SDP, set up a GstNetClientClock that slaves to the remote clock and suggest this clock in provide_clock.	2013-04-11 15:00:05 +01:00
Wim Taymans	f96aa414e1	udpsink: avoid alloc and free in render function Avoid doing alloc and free in the render function for each buffer. Instead, allocate the needed arrays in _init and use those.	2013-04-11 14:57:11 +01:00
Stefan Sauer	48b9919e31	waveparse: remove superfluous g_list_first() calls The variables already point to the start of the list.	2013-04-10 14:25:24 +02:00
Andreas Fenkart	20d3ec8810	rtpsbcdepay: fix sbc frame length calculation for mono and stereo modes https://bugzilla.gnome.org/show_bug.cgi?id=697463	2013-04-09 23:17:57 +01:00
Anton Belka	5ae92ce770	wavparse: add 'note' chunk support Add 'note' chunk support in TOC as GST_TAG_COMMENT https://bugzilla.gnome.org/show_bug.cgi?id=696549	2013-04-09 22:58:27 +02:00
David Schleef	a55ccff854	qtdemux: check value inside enda to set endianness	2013-04-09 13:30:17 -07:00
Wim Taymans	ece73b786a	icydemux: avoid copy when we can	2013-04-09 17:34:12 +02:00
Wim Taymans	91a3afc4dc	gstpay: use bufferlist to avoid memcpy	2013-04-09 16:53:31 +02:00
Wim Taymans	3d7d757521	udpsink: improve debug	2013-04-09 16:53:31 +02:00
Alexander Schrab	79d5a7d03c	wavparse: error out if we receive eos before any valid data https://bugzilla.gnome.org/show_bug.cgi?id=696684	2013-04-09 00:27:31 +01:00
Matej Knopp	67c2219687	deinterlace: force deinterlacing in "interlaced" mode https://bugzilla.gnome.org/show_bug.cgi?id=697467	2013-04-07 20:48:21 +01:00
Nicola Murino	c41c16424d	rtpsbcdepay: fix printf format compiler warnings https://bugzilla.gnome.org/show_bug.cgi?id=697343	2013-04-05 13:50:19 +01:00
Stefan Sauer	b79f667ef4	level: resync on discont Drop pending data on discont and start a new cycle with a new base timestamp. Cleanup some variables.	2013-04-04 22:49:49 +02:00
Olivier Crête	f8831c0cd2	rtpsbcdepay: Rank as secondary This way, it will be selected by decodebin Bug reported by andreas.fenkart@streamunlimited.com https://bugzilla.gnome.org/show_bug.cgi?id=697227	2013-04-03 18:25:36 -04:00
Stefan Sauer	2e56032031	level: subdivide buffers for sample accurate interval handling Previously we would skip level message when processing buffers > the requested interval. Also the message frequency would contain quite some jitter due to only considering them at the end of buffers. Cleanup the tests while we're at it.	2013-04-03 21:40:17 +02:00
Stefan Sauer	b062171dda	spectrum: remove old since comment	2013-04-03 20:30:08 +02:00
Sebastian Dröge	d80ff8e7f3	rtspsrc: Proxy the multicast-iface property of udpsrc	2013-04-03 17:53:13 +02:00
Olivier Crête	6f3734c305	rtpssrcdemux: Only forward stick events while holding the sinkpad stream lock Otherwise we get a race where if the RTCP packet comes in first and while it is added the pads, the segment event arrives on the RTP stream, the event may be lost completely and never forwarded.	2013-04-02 23:42:42 -04:00
Olivier Crête	76679f9ae9	rtpssrcdemux: No need to explicitely forward the caps They are forwarded with the other events	2013-04-02 23:42:41 -04:00
Olivier Crête	4ad8693f3c	rtpssrcdemux: Remove unused GstSegment	2013-04-02 23:42:41 -04:00
Olivier Crête	7293b0eff7	rtpssrcdemux: Simplify event forwarding Use the gst_pad_forward() mechanic, this way we won't miss pads that are added while we are pushing	2013-04-02 23:42:41 -04:00
Olivier Crête	f4c3aef13a	rtpssrcdemux: Don't cross the internal links We had the wrong condition to check for the internal links, so RTP and RTCP pads got crossed!	2013-04-02 23:42:41 -04:00
Tim-Philipp Müller	078ff16abe	matroskademux: fix some debug messages	2013-04-03 00:49:37 +01:00
Arnaud Vrac	00b46b4744	matroskademux: handle TrueHD audio codec id https://bugzilla.gnome.org/show_bug.cgi?id=697113	2013-04-02 22:47:54 +01:00
Wim Taymans	ac2bcfa833	theorapay: add delta-unit to output frames	2013-03-31 19:14:04 +02:00
Matej Knopp	5686512b77	qtmux: use timestamp delta as duration if possible https://bugzilla.gnome.org/show_bug.cgi?id=696437	2013-03-30 15:18:45 -07:00
Josep Torra	509631f60b	rtp: fixes debug message printf related compiler warnings in SBC depayloader	2013-03-30 09:44:41 +01:00
Arun Raghavan	87bdad4bfc	rtp: Add an rtpsbcdepay element Pretty straightforward - takes SBC encapsulated in RTP, depayloads, and pushes out SBC buffers. https://bugzilla.gnome.org/show_bug.cgi?id=690582	2013-03-28 17:22:33 +00:00
Tim-Philipp Müller	477cc51fe7	rtp: fix SBC payloader Init RTP buffer on stack correctly, so mapping it works without criticals and the payloader actually works.	2013-03-27 22:18:34 +00:00
David Schleef	53f8b05b08	Use %03u for format in gst_pad_create_stream_id_printf()	2013-03-25 18:57:08 -07:00
Sebastian Dröge	56062768af	capssetter: Prevent unneeded caps copying and allocation	2013-03-25 10:12:03 +01:00
Dirk Van Haerenborgh	766c5b22ed	capssetter: Pass any or filter caps upstream capsetter accepts anything and just forwards different caps, as such it should return ANY caps on the sinkpad. https://bugzilla.gnome.org/show_bug.cgi?id=693005	2013-03-25 10:11:32 +01:00
Tim-Philipp Müller	35769f7c5d	wavparse: expose CUE sheet items as tracks not chapter entries in TOC https://bugzilla.gnome.org/show_bug.cgi?id=677306	2013-03-24 17:55:55 +00:00
Tim-Philipp Müller	163a7afa1a	wavenc: add some example pipelines	2013-03-23 12:59:26 +00:00
Anton Belka	e808173483	wavenc: add TOC support https://bugzilla.gnome.org/show_bug.cgi?id=680998	2013-03-23 12:55:08 +00:00
Matej Knopp	f29e62c131	qtdemux: make empty subtitle buffer recognition more robust https://bugzilla.gnome.org/show_bug.cgi?id=696244	2013-03-23 11:24:23 +00:00
David Schleef	c0443a17c4	qtmux: Fix test regression with one buffer streams	2013-03-22 15:14:15 -07:00
David Schleef	5bd2864101	qtdemux: split large raw audio samples In order to deal with a file that has samples that are 24 seconds long. Seeking still doesn't work with such files.	2013-03-22 14:14:05 -07:00
David Schleef	364433c105	qtmux: Remove documentation for dts-method	2013-03-22 14:14:04 -07:00
David Schleef	6571e388be	qtmux: deprecate dts-method property	2013-03-22 14:14:04 -07:00
David Schleef	ee56a7cb99	qtmux: Fix problems causing bad durations in file - Fix up out-of-order incoming DTS values. - Fix duration of initial sample.	2013-03-22 14:14:04 -07:00
David Schleef	816e186029	qtmux: fix all timestamps once first_ts is determined	2013-03-22 14:14:04 -07:00
David Schleef	258c40c6dd	qtmux: Use PTS/DTS from incoming buffers Remove old DTS guessing code.	2013-03-22 14:14:04 -07:00
Nicola Murino	709f05234f	qtmux: expose mulaw caps https://bugzilla.gnome.org/show_bug.cgi?id=696052	2013-03-22 20:08:06 +00:00
Rodolfo Schulz de Lima	874808fd2c	qtdemux: fix sample leak when processing private qt tags https://bugzilla.gnome.org/show_bug.cgi?id=696355	2013-03-22 08:47:17 +00:00
Matej Knopp	d8ac666137	qtmux: set stream language code from tag https://bugzilla.gnome.org/show_bug.cgi?id=696358	2013-03-22 08:40:26 +00:00
Matej Knopp	49d9050e9a	qtdemux: send GAP events for subtitle streams https://bugzilla.gnome.org/show_bug.cgi?id=696244	2013-03-21 10:03:37 +00:00
Matej Knopp	516a0b8acb	qtdemux: ignore empty subtitle buffers https://bugzilla.gnome.org/show_bug.cgi?id=696244	2013-03-21 10:03:34 +00:00
Matej Knopp	f494635126	qtdemux: recognize SBTL subtype for subtitles https://bugzilla.gnome.org/show_bug.cgi?id=696244	2013-03-21 10:03:14 +00:00
Anton Belka	0f97b6f978	flacparse: add support for the toc-select event Select tracks from the CUE sheet by sending a toc-select event based on the uid in the TOC. https://bugzilla.gnome.org/show_bug.cgi?id=540891	2013-03-21 00:38:48 +00:00
Michael Smith	b85c5f236b	mp4mux: in faststart mode, don't output up to 4 kB of garbage at the end.	2013-03-19 18:09:31 -07:00
Tim-Philipp Müller	5240b7453c	sbcparse: pack multiple frames into one output buffer Don't output a single buffer for every tiny SBC frame	2013-03-20 00:35:17 +00:00
Kishore Arepalli	288e05c99d	deinterlace: fix infinite loop on EOS with non-default methods or fields Fixes problem of infinite loop in gst_deinterlace_reset_history. Last field in the history was never deinterlaced because idx becomes negative. Happens e.g. with method=scalerbob fields=bottom or method=greedyl fields=top https://bugzilla.gnome.org/show_bug.cgi?id=695644 https://bugzilla.gnome.org/show_bug.cgi?id=693173	2013-03-17 14:47:26 +00:00
Tim-Philipp Müller	dfde4179e8	avimux: change raw video caps order so that GRAY8 is last People like colours. https://bugzilla.gnome.org/show_bug.cgi?id=695543	2013-03-12 00:16:18 +00:00
Ognyan Tonchev	3f8ad30cee	rtph264pay: Don't use upstream caps with peer_query_caps () Calling gst_pad_peer_query_caps () on the src pad with the caps upstream can produce as a filter from gst_rtp_h264_pay_getcaps () is wrong and makes caps negotiation fail if upstream caps are not NULL. https://bugzilla.gnome.org/show_bug.cgi?id=695629	2013-03-11 16:55:13 -04:00
Dirk Van Haerenborgh	065bdf5925	avimux: support raw BGR https://bugzilla.gnome.org/show_bug.cgi?id=695543	2013-03-11 14:51:00 +01:00
Dirk Van Haerenborgh	d7743cf7c6	avidemux: support raw video with negative height https://bugzilla.gnome.org/show_bug.cgi?id=695541	2013-03-11 14:23:46 +01:00
Tim-Philipp Müller	694dbcc5a0	dtmf: move dtmf plugin from -bad to -good https://bugzilla.gnome.org/show_bug.cgi?id=687416	2013-03-09 01:18:30 +00:00
Tim-Philipp Müller	a4c5aa38ec	Merge branch 'dtmf-moved-from-bad' https://bugzilla.gnome.org/show_bug.cgi?id=687416	2013-03-09 00:30:38 +00:00
Sebastian Dröge	539126c097	matroska: Include config.h, it's needed for _stdint.h	2013-03-03 11:59:31 +01:00
Sebastian Dröge	1810786083	flacparse: Fix (wrong) use of uninitialized variable compiler warning	2013-03-03 11:53:04 +01:00
Tim-Philipp Müller	677bfecc6f	qtdemux: add variant field to H.263 caps avdec_h263 won't get plugged otherwise.	2013-03-02 13:59:52 +00:00
Arnaud Vrac	1cff6427f1	qtdemux: skip disabled tracks ISO/IEC 14496-12 specifies disabled tracks should be completely ignored, so just do it. Avoids deadlock during prerolling for some files. Also prevents 'chapter' subtitle tracks from showing up. https://bugzilla.gnome.org/show_bug.cgi?id=693993 https://bugzilla.gnome.org/show_bug.cgi?id=628790	2013-03-02 13:54:23 +00:00
Stefan Sauer	15a81baea5	spectrum: remove the since doc-comment from 0.10	2013-02-28 09:43:12 +01:00
Stefan Sauer	b62cb3edcd	level: add a "post-messages" property and deprecate "message" In spectrum this was changed from 0.10 to 1.0, lets do this here too.	2013-02-28 09:43:12 +01:00
Olivier Crête	df5ca83baf	rtpmp4gdepay: streamtype is not put by all RTSP server, not make it optional Specific case here is Wowza 3.5.0	2013-02-26 14:19:10 -05:00
Thomas Vander Stichele	df8f5f2f83	level: put back deprecation warnings	2013-02-25 00:35:58 +01:00
Thomas Vander Stichele	52b7aab711	level: send last message on EOS	2013-02-25 00:19:22 +01:00
Mark Nauwelaerts	56e2767c20	avidemux: push mode: handle some more 0-size buffer cases Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684944	2013-02-24 19:28:07 +01:00
Tim-Philipp Müller	8004ae0369	matroskamux: fix up example pipeline in docs	2013-02-23 18:50:52 +00:00
Paul HENRYS	10802cae73	rtpsession: Fix wrong code organisation in case of collision change_ssrc field of RTPSession should be set before calling rtp_session_schedule_bye_locked () as this function will call reconsider function that will wake up rtcp_thread which will call rtp_session_on_timeout () that will check change_ssrc to change the ssrc. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=694184	2013-02-22 09:28:07 +02:00
Jean-François Fortin Tam	f5cb19e287	alpha: improve descriptions of chroma keying-related properties and enums https://bugzilla.gnome.org/show_bug.cgi?id=694374	2013-02-22 00:09:56 +00:00
Youness Alaoui	a65fd146f8	alpha: Do not override the method with custom r/g/b values Depending on the order g_object_set() calls aare made, the target r/g/b settings will override the method if set to green/blue. Change that so we do not use the target-r/g/b values unless the method is set to custom. https://bugzilla.gnome.org/show_bug.cgi?id=694374	2013-02-22 00:04:51 +00:00
Ognyan Tonchev	42d8b96f2d	auparse: do not leak src_caps https://bugzilla.gnome.org/show_bug.cgi?id=694275	2013-02-21 19:31:59 +00:00
Wim Taymans	a61055809f	rtpsession: only delay RTCP when we are a sender Only delay the RTCP thread when we are a sender, which we can know because we have a send_rtp_src pad. Otherwise we might delay the RTCP thread if we are only a receiver and then there is no code path that wakes up the RTCP thread and we end up without RTCP packets.	2013-02-20 21:07:41 +02:00
Tim-Philipp Müller	5b19be933b	qtdemux: fix up dodgy code that tries to fix up a broken moov atom After gst_buffer_new_and_alloc() gst_buffer_copy_into() will likely append to the already-existing memory instead of filling it.	2013-02-18 20:04:05 +00:00
Tim-Philipp Müller	34b81f7c93	qtdemux: fix potential crash on short MOOV atom Don't unmap short MOOV atom buffer twice, which happened in the case where we don't fix up the MOOV atom. Fixes crashes when thumbnailing partial mp4 file where the MOOV atom is still incomplete. https://bugzilla.gnome.org/show_bug.cgi?id=694010	2013-02-18 16:35:08 +00:00
Stefan Sauer	99f84b8c4c	audiopanorama: remove channel-mask from caps The channel-mask is only needed for channels>2 which we don't do.	2013-02-15 21:30:15 +01:00
Tim-Philipp Müller	01c6512d5f	udpsrc: use g_socket_set_option() to set buffer size with newer GLib versions So we have to worry less about portability. https://bugzilla.gnome.org/show_bug.cgi?id=692400	2013-02-15 14:11:36 +00:00
Sebastian Dröge	a7ddbc03fe	rtp-payloading: Fix unit test caps and AMR depayloader sink template caps Fields were missing from the actual caps, or too many fields existed in the template caps.	2013-02-13 12:02:46 +01:00
Michael Smith	e3430b0d07	qtdemux: extract codec_data for ProRes	2013-02-12 13:19:53 -08:00
Tim 'mithro' Ansell	c499a81848	avimux: Fixing buffer leak in gst_avi_mux_do_buffer gst_avi_mux_do_buffer was leaking data from gst_collect_pads_pop.	2013-02-12 10:09:05 +01:00
Mark Nauwelaerts	bf81dce432	avidemux: correct duration for audio VBR buffers in pull mode	2013-02-10 15:10:32 +01:00
Mark Nauwelaerts	f0645b79c5	avidemux: proper position reporting and push mode timestamping ... and align current_total semantics in push and pull mode, which tracks bytes for CBR and blocks for VBR. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=691481	2013-02-08 21:41:55 +01:00
Wim Taymans	2d5319c1fa	rtpsession: delay RTCP until first RTP packet Delay sending the first RTCP packet until we have sent the first RTP packet. Otherwise we will send out a Receiver Report instead of a sender report. See https://bugzilla.gnome.org/show_bug.cgi?id=691400	2013-02-08 17:05:27 +01:00
Wim Taymans	2971ed44ee	rtpsession: remove dead code Fixes https://bugzilla.gnome.org/show_bug.cgi?id=668355	2013-02-07 15:06:40 +01:00
Paul HENRYS	0e91c949d8	rtpptdemux: forward sticky events and then set caps When a new src pad is added, first forward the sticky events and then set the caps on the src pad Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692786	2013-02-07 14:38:20 +01:00
Markovtsev Vadim	7cebe2fc41	rtpjitterbuffer: improve debug output Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688935	2013-02-07 14:32:26 +01:00
Wim Taymans	978cc9f538	rtpbin: rework cleanup of streams Move the work of cleaning up the client streams in the free_stream function. This allows us to properly clean up the client streams when we remove an RTP stream as well. Based on patch by Sujay <sdatar@cisco.com> Fixes https://bugzilla.gnome.org/show_bug.cgi?id=660156	2013-02-07 13:02:34 +01:00
Tim 'mithro' Ansell	3a5d17e852	videomixer2: avoid caps leak Fixes https://bugzilla.gnome.org/show_bug.cgi?id=693307	2013-02-07 11:40:35 +01:00
Wim Taymans	c3077012c0	jitterbuffer: do skew estimation only for new timestamps Only run the skew estimation code when we have a new RTP timestamp. If we have the same RTP timestamp, we simply use the previous estimation. This works because the new observation with the same RTP timestamp has to have a bigger receiver time and is thus not going to influence the estimation except for causing more jitter. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=640023	2013-02-06 17:15:11 +01:00
Wim Taymans	640de61740	rtspsrc: only EOS when our source sends BYE Only EOS when we receive a BYE event from the SSRC of our stream. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=675453	2013-02-06 14:01:16 +01:00
Wim Taymans	0540492ab2	rtspsrc: save the stream SSRC Conflicts: gst/rtsp/gstrtspsrc.c	2013-02-06 14:00:56 +01:00
Wim Taymans	c8fb1c720c	rtspsrc: flush connection when stopping When we stop, we can flush all pending commands so that we can stop and join the task. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684924	2013-02-06 13:18:18 +01:00
Stefan Sauer	96f8775a0d	spectrum: remove outdates readme Lets remove the readme from pre-0.1.0 that is completely irrelevant now.	2013-02-05 22:02:13 +01:00
Stefan Sauer	86ae581928	audiopanorama: add more debug logging	2013-02-05 18:51:27 +01:00
Rico Tzschichholz	682e49a752	audioparsers: fix typo in noinst_headers	2013-02-04 18:38:41 +00:00
Stefan Sauer	1f1fe47cb6	audiopanorama: further port to 1.0 Transformcaps is not called with caps containing single structures anymore. Also add missing filter handling. Still does not negotiate though.	2013-02-04 11:08:23 +01:00
Stefan Sauer	d187b96ee2	audiopanorama: fix caps We don't turn float into 32bit pcm. Looks like a typo from updating the caps.	2013-02-03 22:45:52 +01:00
Olivier Crête	fe3e535853	level: Add missing coma between formats	2013-02-03 13:14:50 +01:00
Matthew Waters	b9151a9c28	videomixer: fix eos timestamp check fixes hang in videotestsrc num-buffers=20 ! videomixer ! fakesink Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692935	2013-01-31 16:45:38 +01:00
Dirk Van Haerenborgh	18ff57d6b3	avimux: add support for raw monochrome 8-bit video Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692932	2013-01-31 13:00:17 +01:00
Wim Taymans	747447d298	rtpsession: avoid '...is used uninitialized'	2013-01-29 10:32:51 +01:00
Youness Alaoui	f6a00ad6e9	qtdemux: set interleaved layout correctly for LPCM audio https://bugzilla.gnome.org/show_bug.cgi?id=663458	2013-01-28 23:44:01 +00:00
Youness Alaoui	a76524ea08	qtdemux: add support for LPCM fourcc (uncompressed audio in Quicktime7) https://bugzilla.gnome.org/show_bug.cgi?id=663458	2013-01-28 23:43:57 +00:00
Youness Alaoui	69b814546a	qtdemux: print all debug for sound sample description v2 https://bugzilla.gnome.org/show_bug.cgi?id=663458	2013-01-28 23:43:49 +00:00
Youness Alaoui	92ff8a9b09	qtdemux: sound sample description v2 doesn't override samples_per_packet https://bugzilla.gnome.org/show_bug.cgi?id=663458	2013-01-28 23:43:42 +00:00
Youness Alaoui	ee3d9cbd98	qtdemux: pass stsd data to qtdemux_audio_caps() We will need that later for LPCM format support. Disable QDM2 parsing of stsd data which dead code before as well because data was always NULL. https://bugzilla.gnome.org/show_bug.cgi?id=663458	2013-01-28 23:43:38 +00:00
Youness Alaoui	6d3ff78575	qtdemux: add len check for sound sample descriptions v1 and v2 https://bugzilla.gnome.org/show_bug.cgi?id=663458	2013-01-28 23:43:28 +00:00
Tim-Philipp Müller	629772f735	rtpmanager: use C89-style comments	2013-01-28 23:07:34 +00:00
Olivier Crête	451217c437	gstrtpsession: Fix double-declared variable	2013-01-28 18:06:15 -05:00
Olivier Crête	7300d489fe	rtp: Fix compilation errors in previous patches	2013-01-28 17:58:20 -05:00
Haakon Sporsheim	86c13ceae6	rtpsession: Ensure MT safe event handling and plug event leak. https://bugzilla.gnome.org/show_bug.cgi?id=667826	2013-01-28 17:44:31 -05:00
Idar Tollefsen	268c998a32	rtpsession: mt-safe event-push By taking a ref of the sink-pad under lock, it won't dissappear while the push is taking place https://bugzilla.gnome.org/show_bug.cgi?id=667816	2013-01-28 17:34:50 -05:00
Pascal Buhler	f459fe2673	rtpssrcdemux: Safely push on pads that might be removed due to a RTCP BYE https://bugzilla.gnome.org/show_bug.cgi?id=667815	2013-01-28 17:01:27 -05:00
Tim-Philipp Müller	721dd1ab26	sbcparse: init some variables to avoid bogus compiler warnings	2013-01-28 11:58:50 +00:00
Wim Taymans	4397c8ffbf	rtpdepay: remove payload type restrictions Remove the pt restrictions for all the depayloaders that have an encoding-name. We can use this to autoplug decoders. Remove the encoding-name for all the payloaders with a fixed payload type. We now either have an encoding-name or a pt in the sinkpad caps of a depayloader. See https://bugzilla.gnome.org/show_bug.cgi?id=639292	2013-01-28 12:41:04 +01:00
Marc Leeman	bab2f3c92b	rtp: remove payload requirements from selected depayloaders encoding name is required in the caps and is a better fit for autoplugging than the pt value. Hardware manufacturers have a bad habit of skimming through RFCs and in this case; use unassigned numbers for encoders instead of dynamic numbers. In essence, this patch will add support for a lot of Bosch hardware encoders without breaking autoplugging. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=639292	2013-01-28 12:23:41 +01:00
Mark Nauwelaerts	a1a579afeb	qtdemux: push mode: only parse moov 1 once Fixes https://bugzilla.gnome.org/show_bug.cgi?id=691570	2013-01-27 12:54:20 +01:00
Tim-Philipp Müller	47fccbe635	rtpdtmfsrc: fix compiler warning gstrtpdtmfsrc.c: In function 'gst_dtmf_src_prepare_message.isra.1': gstrtpdtmfsrc.c:669:3: error: 's' may be used uninitialized in this function	2013-01-26 22:58:29 +00:00
Olivier Crête	db5c3f4048	rtpdtmfdepay: Fix missing work in doc	2013-01-25 21:06:05 -05:00
Olivier Crête	92f9a9d9ff	rtpdtmfsrc: Post the messages after the clock wait This way, the messages will be closer in time to when the packets are sent out	2013-01-25 20:45:43 -05:00
Olivier Crête	0d316b4f43	rtpdtmfsrc: Only set the duration when starting to send The duration depends on the clock rate, which could change due to renegotiation	2013-01-25 20:45:43 -05:00
Olivier Crête	90497aa3cd	rtpdtmfsrc: remove "ssrc" from caps ssrc is uint and we don't have a uint range type	2013-01-25 20:45:43 -05:00
Tim-Philipp Müller	d62019fff2	qtmux: set language to 'undefined' instead of English by default	2013-01-24 21:08:51 +00:00
Mark Nauwelaerts	0777a600e3	audioparsers: sbc: fix bogus compiler warning gst-plugins-good/gst/audioparsers/gstsbcparse.c: In function 'gst_sbc_parse_handle_frame': gst-plugins-good/gst/audioparsers/gstsbcparse.c:210:32: error: 'ch_mode' may be used uninitialized i	2013-01-22 19:26:09 +01:00
Thijs Vermeir	16128f0234	autoparsers: use appropriate printf format for gsize	2013-01-16 14:32:56 +01:00
Tim-Philipp Müller	9455a3aee1	rtpsbcpay: update some fields in the caps to their new name and to match the parser. "mode" got renamed to "channel-mode" and "allocation" to "allocation-method".	2013-01-16 10:19:36 +00:00
Tim-Philipp Müller	9f7a949773	audioparsers: add SBC audio parser From-scratch rewrite, the bluez one was useless and broken. https://bugzilla.gnome.org/show_bug.cgi?id=690582	2013-01-15 17:45:30 +00:00
Tim-Philipp Müller	39ef892938	rtp: import rtpsbcpay from bluez and port to 1.0 Compiles, but not tested yet (sbc elements still need to be ported). https://bugzilla.gnome.org/show_bug.cgi?id=690582	2013-01-10 12:43:50 +00:00
Olivier Crête	c6dea5d09c	dtmf/spandsp: Move dtmfdetect to use libspandsp Remove our copy of the tone_detect.c file and use the original from libspandsp. Also move the element to the spandsp plugin.	2013-01-09 20:05:16 -05:00
Marcel Holtmann	4196feb659	rtpsbcpay: Remove workaround for compiler warnings	2013-01-10 00:18:03 +00:00
Marcel Holtmann	fe79c60d74	rtpsbcpay: Add pragma based workaround for GStreamer warnings	2013-01-10 00:18:03 +00:00
Marcel Holtmann	08e95e7249	rtpsbcpay: Update copyright information	2013-01-10 00:15:36 +00:00
Marcel Holtmann	7fa03c0076	rtpsbcpay: Fix signed/unsigned comparison issue within GStreamer plugin	2013-01-10 00:15:35 +00:00
Marcel Holtmann	27a6b0abfe	rtpsbcpay: Update copyright information	2013-01-10 00:15:35 +00:00
Marcel Holtmann	f890079aae	rtpsbcpay: First attempt in fixing compiler warnings (still needs cleanup)	2013-01-10 00:15:35 +00:00
Johan Hedberg	7d4f846112	rtpsbcpay: More coding style fixes	2013-01-10 00:15:35 +00:00
Luiz Augusto von Dentz	151ad9b28d	rtpsbcpay: Remove possible extra memcpy for gstreamer plugin.	2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz	69c8374b7c	rtpsbcpay: Fix bug sending empty packages and remove a buffer copy.	2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz	7b3e4356ea	rtpsbcpay: Fix runtime warnings of gstreamer plugin.	2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz	f74f061f3b	rtpsbcpay: Update gstreamer plugin to use new sbc API.	2013-01-10 00:13:14 +00:00
Marcel Holtmann	b9be04f07b	rtpsbcpay: Update copyright information	2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz	687400ecf4	rtpsbcpay: Fixes gstreamer caps and code cleanup.	2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz	a4f9624261	rtpsbcpay: Fix gtreamer payloader sending fragmented frames.	2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz	41e2f4f544	rtpsbcpay: Fix use of gstreamer plugin with rhythmbox and banshee and rtp timestamps.	2013-01-10 00:13:14 +00:00
Luiz Augusto von Dentz	96971cd323	rtpsbcpay: Make a2dpsink to act like a bin and split the payloader.	2013-01-10 00:13:14 +00:00
Wim Taymans	72402cc649	rtp: small improvements	2013-01-08 16:27:42 +01:00
Wim Taymans	af055d9574	jitterbuffer: refactor handle sync code Move the code that combines the last SR packet and the current jitterbuffer sync values into a sync structure, into its own function. We want to reuse this bit later.	2013-01-07 15:50:33 +01:00
Wim Taymans	87f7d6b9bf	rtp: include downstream latency in SR calculations When we make a mapping between an RTP timestamp and an NTP timestamp, include the downstream latency applied to the sinks. This makes it possible to have both sinks run with different latencies and still have correct sync on the client. It also is more correct because the RTP timestamp in the SR report will actually correspond more closely to the NTP time it was sent on the server. For pipelines with high latency on the sender side, this actually allows a GStreamer receiver to perform synchronisation instead of dropping the RTCP packets.	2013-01-07 15:45:10 +01:00
Wim Taymans	c631ed3300	rtpsession: don't cast event functions There is no need to cast the event functions and only causes problems later when we change the signature later and things silently compiles wrong code.	2013-01-07 14:25:14 +01:00
Wim Taymans	8dcde8b3ea	rtp: more debug	2013-01-07 14:23:34 +01:00
Wim Taymans	6b7d05ac57	rtpsession: improve debug	2013-01-07 14:22:48 +01:00
Tim-Philipp Müller	cf1f6aff0d	udpsrc: sanity check size of available packet data for reading to avoid memory waste On Windows and OS/X, _get_available_bytes() may not return the size of the next pending packet, but the size of all pending packets in the kernel-side buffer, which might be rather large depending on configuration. Sanity-check the size returned by _get_available_bytes() to make sure we never allocate more memory than the max. size for a packet, if it's an IPv4 socket. https://bugzilla.gnome.org/show_bug.cgi?id=610364	2013-01-04 14:00:55 +00:00
Tim-Philipp Müller	95a37196b3	rtspsrc: add "proxy-id" and "proxy-pw" properties to match souphttpsrc. user/password passed via the URI will still take precedence though. https://bugzilla.gnome.org/show_bug.cgi?id=395427	2012-12-31 00:22:27 +00:00
Wim Taymans	8cfec6a88d	rtspsrc: fix cmd comparison Fixes https://bugzilla.gnome.org/show_bug.cgi?id=690476	2012-12-20 17:12:30 +01:00
Wim Taymans	75616fac9a	rtspsrc: add some more debug	2012-12-20 17:12:20 +01:00
Jonas Holmberg	e12457f138	rtpjpegpay: handle width and height > 2040 If width or height is greater than 2040 set width and height to zero in the rtp header and add x-dimensions to outcaps. Solves #684955	2012-12-20 15:40:49 +01:00
Wim Taymans	dcb0e0af93	avidemux: cleanup in flag define	2012-12-20 13:04:52 +01:00
Wim Taymans	0e522bc69a	avidemux: improve debug	2012-12-20 13:04:52 +01:00
Thijs Vermeir	de41376231	rtp: use appropriate printf format for gsize	2012-12-18 16:02:09 +01:00
Thijs Vermeir	df88341ffb	deinterlace: use appropriate printf format for gsize	2012-12-18 16:02:09 +01:00
Philippe Normand	2bd77e1c8a	interleave: set src pad caps upon last sink pad CAPS event Gather caps on all sink pads before setting the src pad caps. This is specially needed when the audio channel mapping is set on the sink pads and the element needs to preserve it on its src pad. https://bugzilla.gnome.org/show_bug.cgi?id=690267	2012-12-18 12:58:43 +01:00
Tim-Philipp Müller	f4cb0c4315	matroskademux: skip empty tags instead of trying to add tags with empty strings, which causes criticals at runtime. https://bugzilla.gnome.org/show_bug.cgi?id=690358	2012-12-17 22:55:12 +00:00
Sebastian Dröge	c49dede772	audioparsers: Make sure the caps are actually writable before changing them	2012-12-17 15:17:12 +01:00
Sebastian Dröge	26040ee38c	audioparsers: Use the peer caps for restrictions instead of the srcpad allowed caps Otherwise we will intersect with the srcpad template caps and add all the caps fields that the parser will ever set, no matter if downstream restricts this field or not. This requires upstream to set this field on the caps to successfully negotiate. https://bugzilla.gnome.org/show_bug.cgi?id=690184	2012-12-17 15:01:02 +01:00
Alexey Fisher	7e47e3b92d	matroskamux: set appropriate block header flag for VP8 invisible frames Useful for debugging mostly. https://bugzilla.gnome.org/show_bug.cgi?id=654259	2012-12-16 23:30:13 +00:00
Tim-Philipp Müller	8a3b116d1f	docs: add rtpmux and rtpdtmfmux to plugin docs https://bugzilla.gnome.org/show_bug.cgi?id=629117	2012-12-16 16:36:39 +00:00
Tim-Philipp Müller	3295b5d791	rtpmanager: move rtpmux and rtpdtmfmux elements from -bad https://bugzilla.gnome.org/show_bug.cgi?id=629117	2012-12-16 16:36:39 +00:00
Tim-Philipp Müller	de204ba754	rtpmux: Fix FSF address https://bugzilla.gnome.org/show_bug.cgi?id=687520	2012-12-16 16:36:39 +00:00
Tim-Philipp Müller	2778a1757f	rtpmux: Use gst_element_class_set_static_metadata() where possible. Avoids some string copies. Also re-indent some stuff. Also some indent fixes here and there.	2012-12-16 16:36:39 +00:00
Olivier Crête	15dfdc58d4	rtpmux: Misc fix for 0.11 Convert the incoming caps before proxying them Clear the last_pad when going to ready tests: Implement accept_caps, don't leak event	2012-12-16 16:36:38 +00:00
Wim Taymans	83262be703	rtpmux: update for RTP buffer api changes	2012-12-16 16:36:38 +00:00
Sebastian Dröge	f17064a8ea	rtpmux: Update for GST_PLUGIN_DEFINE() API changes	2012-12-16 16:36:34 +00:00
Wim Taymans	c86156ad8f	rtpmux: fix compilation	2012-12-16 16:35:36 +00:00
Wim Taymans	6826bbb6da	rtpmux: fix for caps api changes	2012-12-16 16:35:33 +00:00
Matej Knopp	bb345a584d	rtpmux: Fix compiler warnings	2012-12-16 16:35:29 +00:00
Olivier Crête	af4e999c59	rtpmux: Unref non-forwarded events Also, don't unref forwarded ones	2012-12-16 16:35:29 +00:00
Olivier Crête	a8789d1df1	rtpmux: resync iterator on resync	2012-12-16 16:35:29 +00:00
Olivier Crête	0c54079af5	rtpmux: Re-push sticky events on input pad change	2012-12-16 16:35:29 +00:00
Olivier Crête	21831b430f	rtpmux: Don't leak gvalue from iterator	2012-12-16 16:35:29 +00:00
Wim Taymans	ccc4b960fc	rtpmux: more porting	2012-12-16 16:35:26 +00:00
Olivier Crête	f20a6b1d16	rtpmux: port to 0.11	2012-12-16 16:35:26 +00:00
Wim Taymans	35b6668fb6	rtpmux: make request pads take _%u	2012-12-16 16:35:22 +00:00
Olivier Crête	aa3607ef5c	rtpdtmfmux: Add last-stop to dtmf-event upstream events Add the running time of the last outputted buffer to the upstream "dtmf-event" events so that the dtmf source does not leave a gap.	2012-12-16 16:35:22 +00:00
Edward Hervey	d137482fe5	rtpmux: Remove dead assignments	2012-12-16 16:35:22 +00:00
Stefan Kost	55aae6bfab	rtpmux: add missing G_PARAM_STATIC_STRINGS flags Canonicalize property names as needed.	2012-12-16 16:35:15 +00:00
Olivier Crête	9674d5cc23	rtpmux: Improve documentation Add an example pipeline, and try to explain a bit more what it does.	2012-12-16 16:35:15 +00:00
Stefan Kost	ca27a279ba	rtpdtmfmux: remove unused variable	2012-12-16 16:35:15 +00:00
Stefan Kost	c85dceeacb	rtpdtmfmux: remove unused signal boilerplate	2012-12-16 16:35:15 +00:00
Stefan Kost	2353f8d852	rtpmux: no need to ref pad in _chain()	2012-12-16 16:35:15 +00:00
Youness Alaoui	e42d2eebcb	rtpmux: Unlock the right mutex The mutex locked is for the 'mux' object, but we unlock the pad, which means that if the rtpmux gets a flush, then the object lock will stay locked forever, causing it to freeze the next time it tries to take it. Fixes bug #627991	2012-12-16 16:35:15 +00:00
Olivier Crête	78d1ebac9e	rtpmux: Add support for GstBufferList Factor out most of the buffer handling and implement a chain_list function. Also, the DTMF muxer has been modified to just have a function to accept or reject a buffer instead of having to subclass both chain and chain_list.	2012-12-16 16:35:15 +00:00
Olivier Crête	c00f14419b	rtpmux: Don't leak invalid buffers	2012-12-16 16:35:15 +00:00
Tim-Philipp Müller	a45429d81d	rtpmux: fix missing debug log message argument	2012-12-16 16:35:15 +00:00
Olivier Crête	4a8d0243b5	rtpdtmfmux: Add some debug messages	2012-12-16 16:35:14 +00:00
Olivier Crête	423ce98666	rtpdtmfmux: Remove stream-lock event handling	2012-12-16 16:35:14 +00:00
Olivier Crête	a4500c0e74	rtpdtmfmux: Update doc for simplification	2012-12-16 16:35:14 +00:00
Olivier Crête	70097866de	rtpdtmfmux: Drop buffers on non-priority sinks when something is incoming on the priority sink	2012-12-16 16:35:14 +00:00
Olivier Crête	f6548fe9b6	rtpdtmfmux: Add priority sink pads	2012-12-16 16:35:14 +00:00
Olivier Crête	2bcea1537b	rtpdtmfmux: Cleanup event function	2012-12-16 16:35:14 +00:00
Olivier Crête	8e58646f5c	rtpmux: Aggregate incoming segments	2012-12-16 16:35:14 +00:00
Olivier Crête	7be57cac3a	rtpdtmfmux: Update documentation	2012-12-16 16:35:14 +00:00
Olivier Crête	e590fc1f32	rtpmux: Simplify request pad creation	2012-12-16 16:35:14 +00:00
Benjamin Otte	2867e00225	rtpmux: gst_element_class_set_details => gst_element_class_set_details_simple	2012-12-16 16:35:10 +00:00
unknown	fb7266884d	rtpmux: update the current_ssrc from the caps Fixes #604101	2012-12-16 16:33:47 +00:00
Håvard Graff	eab65e84ca	rtpmux: release pads when disposing Because of an allocated priv (GstRTPMuxPadPrivate), the element will leak memory if not gst_rtp_mux_release_pad() is called. This would previously only happen if release_request_pad() was called explicitly, somthing that should not be neccesary. Fixes #604099	2012-12-16 16:33:46 +00:00
Wim Taymans	0d54122804	dtmfmux: method name cleanups	2012-12-16 16:33:46 +00:00
Olivier Crête	3841cc74cf	rtpmux: Don't ignore requested pad name	2012-12-16 16:33:46 +00:00
Olivier Crête	d93295ff9d	rtpmux: Remove empty finalize	2012-12-16 16:33:46 +00:00
Olivier Crête	5e90a4e86b	rtpmux: Free the pad private data on pad release Free the pad private data on pad release instead of using a weak ref, which is not thread safe. Also, lock the content of the pad private using the element's object lock.	2012-12-16 16:33:46 +00:00
Olivier Crête	4be63c9add	rtpmux: Reject wrong caps	2012-12-16 16:33:46 +00:00
Olivier Crête	0111bafb3a	rtpmux: Fix leak Fixed a leak discovered by Laurent Glayal <spegle@yahoo.fr>	2012-12-16 16:33:46 +00:00
Olivier Crête	fcc1522d2e	rtpmux: Fix leak Fixed a leak discovered by Laurent Glayal <spegle@yahoo.fr>	2012-12-16 16:33:46 +00:00
Olivier Crête	ff6686f1c7	rtpmux: Fix warning	2012-12-16 16:33:46 +00:00
Olivier Crête	00791f930b	rtpmux: Set different caps depending on the input	2012-12-16 16:33:46 +00:00
Olivier Crête	ed0b407038	rtpmux: Only free pad private when pad is disposed	2012-12-16 16:33:45 +00:00
Olivier Crête	92bb5199ac	rtpmux: Remove useless caps mangling	2012-12-16 16:33:45 +00:00
Olivier Crête	3ccf3217fe	rtpmux: Rename variable for more clarity	2012-12-16 16:33:45 +00:00
Olivier Crête	4b958f6d8d	rtpmux: Use GST_BOILERPLATE	2012-12-16 16:33:45 +00:00
Olivier Crête	abe57be248	rtpmux: Do the includes locally	2012-12-16 16:33:45 +00:00
Olivier Crête	05844c89e9	rtpmux: Add GST_DEBUG_FUNCPTRs	2012-12-16 16:33:45 +00:00
Olivier Crête	fd102b95ab	rtpdtmfmux: Release locked pad on release_pad Release the special pad if the pad is removed from the muxer.	2012-12-16 16:33:45 +00:00
Laurent Glayal	00f8bab712	rtpdtmfmux: Release special on pad dispose Fixes #577690	2012-12-16 16:33:45 +00:00
Stefan Kost	a4a22454dc	docs: various doc fixes No short-desc as we have them in the element details. Also keep things (Makefile.am and sections.txt) sorted. Reword ambigous returns. No text after since please.	2012-12-16 16:33:41 +00:00
Olivier Crête	7d4395a910	rtpmux: Move rtpmux from gst-plugins-farsight to -bad	2012-12-16 16:33:27 +00:00
Olivier Crête	68215752f4	rtpmux: Re-indent to Gst style	2012-12-16 16:33:24 +00:00
Olivier Crête	c7d0809434	rtpmux: Document rtp muxer a bit	2012-12-16 16:33:20 +00:00
Laurent Glayal	47c7a93df2	rtpmux: Add signals before stream lock and after unlocking	2012-12-16 16:33:17 +00:00
Olivier Crête	f1656ed8b0	rtpmux: Let ssrc through getcaps	2012-12-16 16:33:14 +00:00
Olivier Crête	1529dffaf9	rtpmux: Rename have_base to have_ts_base	2012-12-16 16:33:11 +00:00
Olivier Crête	57563517bd	rtpmux: Protect the seqnum with object lock in rtpmux	2012-12-16 16:33:08 +00:00
Olivier Crête	d3237eaf95	rtpmux: Remove unused sink_ts_base	2012-12-16 16:33:04 +00:00
Olivier Crête	cc23958183	rtpmux: Have getcaps to force the same clockrate on all pads	2012-12-16 16:33:01 +00:00
Olivier Crête	dc36590d0c	rtpmux: Validate RTP data in RTP Mux	2012-12-16 16:32:57 +00:00
Olivier Crête	360c8d4f1d	rtpmux: Remove unused clock-rate property	2012-12-16 16:32:54 +00:00
Olivier Crête	b86232d0dc	rtpmux: Clarify locking in rtpdtmfmux	2012-12-16 16:32:50 +00:00
Laurent Glayal	4b607cdda5	rtpmux: Missing format parameter	2012-12-16 16:32:47 +00:00
Håvard Graff	b313c80367	rtpmux: Update seqnum base in rtp muxer With help from Wim	2012-12-16 16:32:43 +00:00
Håvard Graff	c479f90274	rtpmux: Fix some more leaks	2012-12-16 16:32:40 +00:00
Håvard Graff	1b5e769e0b	rtpmux: Fix leak	2012-12-16 16:32:37 +00:00
Olivier Crête	5cbb0de823	rtpmux: Don't unref caps we don't know (thanks Wim)	2012-12-16 16:32:32 +00:00
Olivier Crête	cebf506949	rtpmux: Put per-buffer debug at level LOG	2012-12-16 16:32:29 +00:00
Olivier Crête	3c12a423b7	rtpmux: Make debug print accurate	2012-12-16 16:32:25 +00:00
Olivier Crête	c49f4c87c6	rtpmux: Set our caps on the buffers	2012-12-16 16:32:22 +00:00
Olivier Crête	ec63da9366	rtpmux: Take the clock-base stored from the last setcaps	2012-12-16 16:32:18 +00:00
Olivier Crête	674c074114	rtpmux: Store the clock-base on setcaps	2012-12-16 16:32:15 +00:00
Olivier Crête	90264b9686	rtpmux: Add padprivate to the request pads	2012-12-16 16:32:11 +00:00
Olivier Crête	15d661ba3e	rtpmux: Make indentation more correct	2012-12-16 16:31:56 +00:00
Olivier Crête	3a7d09a749	rtpmux: Fix typo	2012-12-16 16:31:53 +00:00
Olivier Crête	91aef3ec5e	rtpmux: Set seqnum-base and clock-base in caps from rtpmuxer	2012-12-16 16:31:50 +00:00
Zeeshan Ali	6ea5ca354d	rtpmux: more debug 20070815135038-f3f1e-9c7a5490a525c6e8753cb1b8c03354df99132b5c.gz	2012-12-16 16:31:46 +00:00
Youness Alaoui	f0e209b638	rtpmux: missing comment 20070820185032-4f0f6-0ab67b6ac40dd4e35a8fe53f3cb6daff65ce43b9.gz	2012-12-16 16:30:33 +00:00
Olivier Crete	3ed5590da6	rtpmux: Make buffer writable before writing into it 20070712195336-3e2dc-91a5fb797cfa4919d4e2f9a728c6d6fbd3b83d93.gz	2012-12-16 16:30:31 +00:00
Olivier Crete	dd13f7c8ef	rtpmux: Set pads active when adding them to a potentially running element 20070706202459-3e2dc-a3731f885725594def0a7be997fc7b3a739ee967.gz	2012-12-16 16:30:27 +00:00
Olivier Crete	1c5075f927	rtpmux: Fix multiple ref leaks (patches by SP GLE) 20070607120121-3e2dc-061e9ef7a47b1b84fa8f8092f4b8bcc0e6db8c8c.gz	2012-12-16 16:30:23 +00:00
Zeeshan Ali	42f455e902	rtpmux: send event to all src pads 20070528152505-f3f1e-039216c73dc93f64c49962c77a0253cb9cfec4d3.gz	2012-12-16 16:30:18 +00:00
Zeeshan Ali	dba101bb0f	rtpmux: print a warning if receive an error iterating sinkpads 20070528123749-f3f1e-4c1eb3f511b5610143610a65a94d117f2c3d2580.gz	2012-12-16 16:30:15 +00:00
Zeeshan Ali	baa48dc6bc	rtpmux: deal with all the gst_iterator_next() return values 20070528122808-f3f1e-d301644c3be7633ec6dc5e28596e9346d2da6a50.gz	2012-12-16 16:30:12 +00:00
Zeeshan Ali	de40874670	rtpmux: Return correct value from the event handler 20070525123116-f3f1e-131b37b5f4521618fe2f1320409a47e65b35ad2d.gz	2012-12-16 16:30:08 +00:00
Zeeshan Ali	ed76f67e96	rtpmux: Ville's original patch to fix the traversal of dtmf event 20070525102709-f3f1e-6c41d1ef934068a4f4e810e7e981b420075b0c98.gz	2012-12-16 16:30:05 +00:00
zeeshan.ali@nokia.com	94ebe07862	rtpmux: Set the correct ts-offset on the get_prop value 20070329135250-65035-a43e222d91d57c0a61cb3287586aaa29abf78674.gz	2012-12-16 16:30:01 +00:00
zeeshan.ali@nokia.com	1ee542c378	rtpmux: Refactorize state_change 20070329135223-65035-23a0107b2e397710f035c6e88cc0e49b65bb4d5d.gz	2012-12-16 16:29:58 +00:00
zeeshan.ali@nokia.com	2498ba671a	rtpmux: set SSRC on the packets 20070329133622-65035-1be6e0aa85a71389f7d257b9cd3e13a73d6b745b.gz	2012-12-16 16:29:55 +00:00
zeeshan.ali@nokia.com	ee69c2690d	rtpmux: Code clean-up and more debug output 20070329131936-65035-9d499e209e0d7a409c3aa0d1040778babf076179.gz	2012-12-16 16:29:52 +00:00
zeeshan.ali@nokia.com	1c799ce964	rtpmux: Use own clock-base 20070328112219-65035-1ba5fefbc65059e9b0c860528a31062ceb6a7331.gz	2012-12-16 16:29:48 +00:00
zeeshan.ali@nokia.com	b04630d7a2	rtpmux: Only accept RTP streams that have the same clock-rate 20070323163139-65035-fc0b17b0b8a7a041f48994c4f26e96568168bf95.gz	2012-12-16 16:29:45 +00:00
zeeshan.ali@nokia.com	6fe1e02efd	rtpmux: Some more code-cleanups 20070322161552-65035-bda96165e146b4f1d5fea1cc9576a7ab3abebc9e.gz	2012-12-16 16:29:42 +00:00
zeeshan.ali@nokia.com	1603223ee5	rtpmux: return newpad instead of NULL and warn if failed to create a pad 20070322154251-65035-cdb6651e61c2eb0205cc8c24693b43f98a2da718.gz	2012-12-16 16:29:38 +00:00
zeeshan.ali@nokia.com	23d3ed5c5f	rtpmux: Refactorize the RTPMux code 20070322124132-65035-0a3278147546e33f687097a43b775b3f6aa99f93.gz	2012-12-16 16:29:35 +00:00
zeeshan.ali@nokia.com	21e6e951f6	rtpmux: Some more doc fixing 20070322121453-65035-12d602272217b51bd97df4e5790024c399622dd3.gz	2012-12-16 16:29:32 +00:00
zeeshan.ali@nokia.com	0de7fb6f37	rtpmux: More Refactoring 20070322113228-65035-bae34a79599e7de5293ed77b022361ccff822bb9.gz	2012-12-16 16:29:29 +00:00
zeeshan.ali@nokia.com	0f755657ce	rtpmux: More documentation 20070322113154-65035-624850541a5b5fc3df231204be5a83d07239db28.gz	2012-12-16 16:29:26 +00:00
zeeshan.ali@nokia.com	5483c78ac0	rtpmux: Refactor the event handler function 20070321163311-65035-987e7f25d1ab5335b79f44b277abf15e4e37d317.gz	2012-12-16 16:29:23 +00:00
zeeshan.ali@nokia.com	db1523ae60	rtpmux: Add RTPDTMFMux element 20070321145244-65035-9a01390b0dee3398e53199a1fa1d9352004f338e.gz	2012-12-16 16:29:19 +00:00
zeeshan.ali@nokia.com	97ff54dce7	rtpmux: Remove DTMF-specific code from RTP muxer and make it extendable 20070321123149-65035-b8a8f55ff78eed8cbb0042e827885edfc5438242.gz	2012-12-16 16:29:16 +00:00
zeeshan.ali@nokia.com	1a227ac7e5	rtpmux: Put more helpful description 20070320120524-65035-db27a7cf6307b511aeb3d996d26e790e367a7bad.gz	2012-12-16 16:29:13 +00:00
zeeshan.ali@nokia.com	d876c0d8cc	rtpmux: remove the (commented-out) code for blocking the pads 20070316151641-65035-0123af387951f88594797c722e882cfe70240aff.gz	2012-12-16 16:29:10 +00:00
zeeshan.ali@nokia.com	209228c44d	rtpmux: Drop buffers instead of blocking the sinkpads 20070316131444-65035-9c1345ad96108881f455d4b55a7f623cd302d0ed.gz	2012-12-16 16:29:05 +00:00
zeeshan.ali@nokia.com	795822ffa5	rtpmux: Implement stream locking, needed for DTMF 20070314171618-65035-e4d24b1606ce0a3e2e739f01833f61e4d7555eac.gz	2012-12-16 16:29:02 +00:00
zeeshan.ali@nokia.com	fd209faa56	rtpmux: use GST__OBJECT instead of g_ 20070314102058-65035-e2442888f2e3e5a3a7659ad7954a4fba34749ce2.gz	2012-12-16 16:28:58 +00:00
zeeshan.ali@nokia.com	b0208cb0a6	rtpmux: No need to manage pads, parent does that for us 20070314101854-65035-ef5f4abde227102a1128835ab325905eae4c3726.gz	2012-12-16 16:28:55 +00:00
zeenix@gmail.com	74e9071dad	rtpmux: Fix copyright header 20070314090358-d014a-3a6d3eeeaaf5cb8ca3bca6a33e99a551f598bd48.gz	2012-12-16 16:28:51 +00:00
zeeshan.ali@nokia.com	3c4cdf1541	rtpmux: The first implementation of RTP muxer 20070307085307-65035-833402413f99cb3f8be4883e92bad4c8722510c9.gz	2012-12-16 16:28:41 +00:00
Tim-Philipp Müller	b19122bac8	scaletempo: no need for a private struct	2012-12-15 21:27:01 +00:00
Tim-Philipp Müller	61913ab7b4	audiofx: move scaletempo element from -bad https://bugzilla.gnome.org/show_bug.cgi?id=687262	2012-12-14 13:16:17 +00:00
Sebastian Dröge	314765c294	scaletempo: Fix event leak	2012-12-14 13:16:17 +00:00
Sebastian Dröge	490e408991	scaletempo: Fix timestamp tracking	2012-12-14 13:16:17 +00:00
Sebastian Dröge	502eb8d1b7	scaletempo: Implement LATENCY query	2012-12-14 13:16:17 +00:00
Sebastian Dröge	c7589817cb	scaletempo: Store instance private data in the instance struct Getting it over and over again via G_TYPE_INSTANCE_GET_PRIVATE() is really slow.	2012-12-14 13:16:17 +00:00
Tim-Philipp Müller	e552bd484f	scaletempo: use gst_element_class_set_static_metadata() where possible. Avoids some string copies. Also re-indent some stuff. Also some indent fixes here and there.	2012-12-14 13:16:17 +00:00
Mark Nauwelaerts	d2dd91ac47	scaletempo: replace gst_element_class_set_details_simple with gst_element_class_set_metadata	2012-12-14 13:16:17 +00:00
Wim Taymans	cb1743d578	scaletempo: ffmpegcolorspace is no more	2012-12-14 13:16:17 +00:00
Sebastian Dröge	93e1091d7f	scaletempo: Update for GST_PLUGIN_DEFINE() API changes	2012-12-14 13:16:17 +00:00
Mark Nauwelaerts	3286cdd542	scaletempo: port to 0.11	2012-12-14 13:16:16 +00:00
Stefan Kost	62d780cd51	scaletempo: improve the docs Fix the syntax, add more explanation and xref the properties.	2012-12-14 13:16:16 +00:00
Chris E Jones	caf2b6cb5c	scaletempo: Correctly handle newsegment events with stop==-1 Fixes bug #645420.	2012-12-14 13:16:16 +00:00
Stefan Kost	6d54058982	scaletempo: add missing G_PARAM_STATIC_STRINGS flags Canonicalize property names as needed.	2012-12-14 13:16:16 +00:00
Benjamin Otte	38bc2dfb4a	scaletempo: gst_element_class_set_details => gst_element_class_set_details_simple	2012-12-14 13:16:16 +00:00
Thiago Santos	2d72ec153a	scaletempo: properly update new segments Scaletempo was missing an update of 'stop' in new segment parameters when pushing it downstream, which caused files to end earlier when rate < 1. Fixes #599903 Based on patch by: Bastian Hecht <hechtb@gmail.com>	2012-12-14 13:16:16 +00:00
Maximilian Högner	2fe7a97f1c	scaletempo: Explicitely cast to signed integers to fix a segfault Fixes bug #585660.	2012-12-14 13:16:16 +00:00
Michael Smith	1b1f6f56d6	scaletempo: Do not use void pointer arithmetic.	2012-12-14 13:16:16 +00:00
Stefan Kost	9284c85b33	scaletempo: Return the result of parent_class->event() Original commit message from CVS: * gst/audiofx/gstscaletempo.c: Return the result of parent_class->event().	2012-12-14 13:16:16 +00:00
Rov Juvano	43e79f7769	Add scaletempo plugin, which allows to scale the speed of audio without changing the pitch by handling seeks with a r... Original commit message from CVS: Patch by: Rov Juvano <rovjuvano at users dot sourceforge dot net> * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/inspect/plugin-scaletempo.xml: * examples/scaletempo/Makefile.am: * examples/scaletempo/demo-gui.c: (pop_status_bar), (status_bar_printf), (demo_gui_seek_bar_format), (update_position), (demo_gui_seek_bar_change), (demo_gui_do_change_rate), (demo_gui_do_set_rate), (demo_gui_do_rate_entered), (demo_gui_do_toggle_advanced), (demo_gui_do_toggle_disabled), (demo_gui_do_seek), (demo_gui_do_play), (demo_gui_do_pause), (demo_gui_do_play_pause), (demo_gui_do_open_file), (demo_gui_do_playlist_prev), (demo_gui_do_playlist_next), (demo_gui_do_about_dialog), (demo_gui_do_quit), (demo_gui_request_set_stride), (demo_gui_request_set_overlap), (demo_gui_request_set_search), (demo_gui_rate_changed), (demo_gui_playing_started), (demo_gui_playing_paused), (demo_gui_playing_ended), (demo_gui_player_errored), (demo_gui_stride_changed), (demo_gui_overlap_changed), (demo_gui_search_changed), (demo_gui_set_player_func), (demo_gui_set_playlist_func), (build_gvalue_array), (create_action), (demo_gui_show_func), (demo_gui_set_player), (demo_gui_set_playlist), (demo_gui_show), (demo_gui_get_property), (demo_gui_set_property), (demo_gui_init), (demo_gui_class_init), (demo_gui_get_type): * examples/scaletempo/demo-gui.h: * examples/scaletempo/demo-main.c: (handle_error_message), (handle_quit), (main): * examples/scaletempo/demo-player.c: (no_pipeline), (demo_player_event_listener), (demo_player_state_changed_cb), (demo_player_eos_cb), (demo_player_build_pipeline), (_set_rate), (demo_player_scale_rate_func), (demo_player_set_rate_func), (_set_state_and_wait), (demo_player_load_uri_func), (demo_player_play_func), (demo_player_pause_func), (_seek_to), (demo_player_seek_by_func), (demo_player_seek_to_func), (demo_player_get_position_func), (demo_player_get_duration_func), (demo_player_scale_rate), (demo_player_set_rate), (demo_player_load_uri), (demo_player_play), (demo_player_pause), (demo_player_seek_by), (demo_player_seek_to), (demo_player_get_position), (demo_player_get_duration), (demo_player_get_property), (demo_player_set_property), (demo_player_init), (demo_player_class_init), (demo_player_get_type): * examples/scaletempo/demo-player.h: * gst/audiofx/Makefile.am: * gst/audiofx/gstscaletempo.c: (best_overlap_offset_float), (best_overlap_offset_s16), (output_overlap_float), (output_overlap_s16), (fill_queue), (reinit_buffers), (gst_scaletempo_transform), (gst_scaletempo_transform_size), (gst_scaletempo_sink_event), (gst_scaletempo_set_caps), (gst_scaletempo_get_property), (gst_scaletempo_set_property), (gst_scaletempo_base_init), (gst_scaletempo_class_init), (gst_scaletempo_init): * gst/audiofx/gstscaletempo.h: * gst/audiofx/gstscaletempoplugin.c: (plugin_init): Add scaletempo plugin, which allows to scale the speed of audio without changing the pitch by handling seeks with a rate!=1.0. Integrate it into the docs and add the example application for it. Fixes bug #537700.	2012-12-14 13:16:15 +00:00
Havard Graff	9c94f1187c	jitterbuffer: bundle together late lost-events The scenario where you have a gap in a steady flow of packets of say 10 seconds (500 packets of with duration of 20ms), the jitterbuffer will idle up until it receives the first buffer after the gap, but will then go on to produce 499 lost-events, to "cover up" the gap. Now this is obviously wrong, since the last possible time for the earliest lost-events to be played out has obviously expired, but the fact that the jitterbuffer has a "length", represented with its own latency combined with the total latency downstream, allows for covering up at least some of this gap. So in the case of the "length" being 200ms, while having received packet 500, the jitterbuffer should still create a timeout for packet 491, which will have its time expire at 10,02 seconds, specially since it might actually arrive in time! But obviously, waiting for packet 100, that had its time expire at 2 seconds, (remembering that the current time is 10) is useless... The patch will create one "big" lost-event for the first 490 packets, and then go on to create single ones if they can reach their playout deadline. See https://bugzilla.gnome.org/show_bug.cgi?id=667838	2012-12-13 12:00:43 +01:00
Wim Taymans	a858bf46db	rtspsrc: fix TCP reconnect Ignore other commands when reconnecting, otherwise the loop function would pause and the reconnection would not happen. Continue looping after doing a reconnect so that we have a chance to actually read the new data.	2012-12-13 09:30:59 +01:00
Philippe Normand	a8fa9f2b47	deinterleave: properly set srcpad channel position The src pad caps always describe a single audio channel so only the first position matters if deinterleave is configured to keep channel positions in its src pads.	2012-12-12 11:20:56 +00:00
Wim Taymans	b1dc816772	rtspsrc: timeout on udpsrc is in nanoseconds	2012-12-12 11:09:42 +01:00
Wim Taymans	32bd981303	udpsrc: improve timeouts Make it possible to set the timeout after we went to the READY state by using the timeout when checking the condition. This also makes it possible to set the timeout with a higher granularity than seconds.	2012-12-12 11:08:13 +01:00
Wim Taymans	abd7e33db6	deinterlace: add support for strides Implement stride support correctly by taking it from the GstVideoFrame. Propose a bufferpool upstream when not operating in passthrough.	2012-12-11 13:00:46 +01:00
Aleix Conchillo Flaque	3503aef946	rtspsrc: do not change state to PLAYING if currently chaning state * gst/rtsp/gstrtspsrc.c (gst_rtspsrc_play): state change might be happening in the application thread, so we don't change the state to PLAYING in the gstrtspsrc thread unless it is safe. A specific case is when chaning the state to NULL from the application thread. This will synchronously try to stop the task (with the element state lock acquired), but we will try a gst_element_set_state from gstrtspsrc thread which will block on the element state lock causing a deadlock. https://bugzilla.gnome.org/show_bug.cgi?id=684312	2012-12-10 15:13:22 +01:00
Tim-Philipp Müller	672ab8fb5b	webmux: fix linking with shout2send element Shout2send only accepts webm format, not matroska, but due to a bug in matroskamux, webmmux's source pad is also created with the matroska source pad template as pad template, which makes the link function think it can't link webmmux to shout2send. Also add unit test. https://bugzilla.gnome.org/show_bug.cgi?id=689336	2012-11-30 17:22:34 +00:00
Wim Taymans	64cdbb77a9	rtspsrc: use new option parser function	2012-11-27 11:13:37 +01:00
Tim-Philipp Müller	5dee61a8d5	law: fix accidental file permissions change https://bugzilla.gnome.org/show_bug.cgi?id=687469	2012-11-26 15:17:13 +00:00
Tim-Philipp Müller	314efb684b	qtdemux: avoid criticals if unknown fourcc has space at beginning or end https://bugzilla.gnome.org/show_bug.cgi?id=682936	2012-11-25 14:16:09 +00:00
Tim-Philipp Müller	efaa80fbc6	videobox: fix border filling for planar YUV formats We would get a green border instead of a black one, for example. https://bugzilla.gnome.org/show_bug.cgi?id=684991	2012-11-24 19:32:51 +00:00
Tim-Philipp Müller	ef6c16a32e	mulaw: const-ify some arrays	2012-11-24 14:27:33 +00:00
Roland Krikava	3be45f7022	mulawdec: fix integer overrun There might be more than 65535 samples in a chunk of data. https://bugzilla.gnome.org/show_bug.cgi?id=687469	2012-11-24 14:24:41 +00:00
Wim Taymans	5d0507c09e	rtspsrc: pause the task instead of spinning Actually pause the loop task instead of spinning forever.	2012-11-22 11:34:31 +01:00
Joshua M. Doe	fe9fb8d8a7	videoflip: Add gray 8/16 support	2012-11-20 12:49:49 +01:00
Wim Taymans	c28bfa8902	rtspsrc: handle segment event Make a segment event when we send a new range header to a client (first PLAY request or after a seek). Send the segment event in interleaved mode. Clean the segment event on cleanup Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688382	2012-11-16 15:38:29 +01:00
Wim Taymans	bd91bd3193	rtspsrc: fix check for active streams A stream can be active without a srcpad yet and we want to send events on those streams as well.	2012-11-16 15:22:46 +01:00
Wim Taymans	11cf4d4fd3	rtspsrc: create and add pads outside of lock Create and add the ghostpad for the new stream outside of the lock because it is not needed and causes deadlocks.	2012-11-16 13:33:44 +01:00
Aleix Conchillo Flaque	6c855edf03	rtspsrc: allow client to disable reconnection * gst/rtsp/gstrtspsrc.[ch]: added new "udp-reconnect" property. Before, rtspsrc always tried to reconnect to the server when the RTSP connection was closed by the server. This property lets the user decide whether it wants rtspsrc to reconnect or not. https://bugzilla.gnome.org/show_bug.cgi?id=683912	2012-11-16 12:55:10 +01:00
Wim Taymans	e2a4d28c1f	rtspsrc: clear variables before retrying Else we might unref an old udpsrc twice in cleanup.	2012-11-16 12:17:37 +01:00
Wim Taymans	cc9cb26be1	rtspsrc: propose ports in multicast When the user configured a port-range, propose ports from this range as the multicast ports. The server is free to ignore this request but if it honours it, increment our ports so that we suggest the next port pair for the next stream. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=639420	2012-11-16 12:17:37 +01:00
Wim Taymans	5025b3f1b3	rtspsrc: add more debug	2012-11-16 12:17:37 +01:00
Tim-Philipp Müller	6f1aa3e4d5	multifilesink: post messages in max-size mode as well No reason not to really.	2012-11-16 09:13:22 +00:00
Wim Taymans	c33507f186	udpsrc: post error before stopping	2012-11-15 14:48:59 +01:00
Tim-Philipp Müller	bdf3c77828	gst_adapter_prev_timestamp -> gst_adapter_prev_pts https://bugzilla.gnome.org/show_bug.cgi?id=675598	2012-11-14 00:13:36 +00:00
Nicolas Dufresne	673d2d24b8	videoflip: Add NV12/NV21 support https://bugzilla.gnome.org/show_bug.cgi?id=688225	2012-11-13 14:25:04 +01:00
Wim Taymans	c755af0cb0	rtpsource: protect against invalid RTP packets	2012-11-12 11:18:30 +01:00
Tim-Philipp Müller	35fafae241	videocrop: add support for YV12 We can do I420, so we can do YV12 as well.	2012-11-10 18:21:28 +00:00
Alessandro Decina	b916d2b398	multifilesink: don't write stream headers with key-unit-event Don't write stream headers, let upstream elements insert them in the stream if all_headers=true is set in key unit events.	2012-11-10 12:41:33 +01:00
Nicolas Dufresne	e111068f7b	videocrop: Add NV12/NV21 support https://bugzilla.gnome.org/show_bug.cgi?id=687964	2012-11-10 01:52:44 +01:00
Sebastian Dröge	c70ba7765a	udpsrc: Also clear GError	2012-11-09 11:22:30 +01:00
Sebastian Dröge	b86d20e45b	udpsrc: Don't error out if we get an ICMP destination-unreachable message when trying to read packets See bug #529454 and #687782 and commit `751f2bb364`	2012-11-09 11:20:27 +01:00
Christian Fredrik Kalager Schaller	485505f323	Fix vp8rtp header names in Makefile	2012-11-07 13:36:33 +01:00
Nicolas Dufresne	1ad8ebac44	videocrop: Add support for automatic cropping This change enable automatic cropping using -1 set to left, top, right or bottom property. In the case both side are set to automatic cropping, the croping will be done equally on both side (in the odd case, right and bottom cropping will be 1 pixel more). https://bugzilla.gnome.org/show_bug.cgi?id=687761	2012-11-07 11:20:24 +01:00
Marc Leeman	7cbca3dcd1	rtsp: the RTCP port number is inclusive The configured port number pair has its upper bound set to the maximum allowed RTCP port, inclusive. See https://bugzilla.gnome.org/show_bug.cgi?id=639420	2012-11-06 13:22:58 +01:00
Tim-Philipp Müller	beb3c9c9be	Fix FSF address https://bugzilla.gnome.org/show_bug.cgi?id=687520	2012-11-04 00:09:59 +00:00
Tim-Philipp Müller	230cf41cc9	Fix FSF address https://bugzilla.gnome.org/show_bug.cgi?id=687520	2012-11-04 00:07:18 +00:00
Wim Taymans	9857e6af4d	vrawdepay: don't access rtp buffer after unmap Read the marker bit before we unmap the rtp packet.	2012-11-02 18:48:17 +00:00
Douglas Bagnall	0b898ab911	videoconvert: Compare y offset with height, not width, when testing for overlap This could have prevented images showing that should have when the source height is greater than its width. When width exceeds height, as is common, it probably only caused a miniscule amount of unnecessary work. I haven't tested.	2012-11-02 09:29:30 +01:00
Tim-Philipp Müller	5ac789408b	rtpvp8: include config.h and minor style fixes	2012-11-01 21:10:21 +00:00
Tim-Philipp Müller	4a849d6690	rtp: fix tabs/space mess in Makefile.am	2012-11-01 20:53:48 +00:00
Tim-Philipp Müller	321acd14dc	rtp: move VP8 payloader and depayloader from -bad Spec is still in draft state, but should hopefully not change much now. Besides, we announce things as VP8-DRAFT-IETF-01 in our caps, so even if things change in incompatible ways it should not break anything. https://bugzilla.gnome.org/show_bug.cgi?id=687263	2012-11-01 20:53:48 +00:00
Tim-Philipp Müller	44efab8e3d	rtpvp8: use gst_element_class_set_static_metadata() where possible. Avoids some string copies. Also re-indent some stuff. Also some indent fixes here and there.	2012-11-01 20:53:48 +00:00
Mark Nauwelaerts	bc7dbbbd4f	rtpvp8: replace gst_element_class_set_details_simple with gst_element_class_set_metadata	2012-11-01 20:53:48 +00:00
Sebastian Dröge	4853001547	rtpvp8: update for GST_PLUGIN_DEFINE() API changes	2012-11-01 20:53:48 +00:00
Wim Taymans	fccfca38d4	rtpvp8: update for buffer changes	2012-11-01 20:53:48 +00:00
Danilo Cesar Lemes de Paula	3edffb13e3	rtpvp8; fix compatibility with the third draft https://bugzilla.gnome.org/show_bug.cgi?id=671073	2012-11-01 20:53:48 +00:00
Mark Nauwelaerts	d9581832a0	rtpvp8: port some more to new memory API	2012-11-01 20:53:47 +00:00
Olivier Crête	c6761daa27	rtpvp8: port to 0.11	2012-11-01 20:53:47 +00:00
Sebastian Dröge	2c5ea76bdc	rtpvp8pay: Fix typo	2012-11-01 20:53:47 +00:00
Youness Alaoui	1cf155d70d	rtpvp8: Update the pay/depay to the ietf-draft-01 spec	2012-11-01 20:53:47 +00:00
Vincent Penquerc'h	88aade4150	rtpvp8: fix bitstream parsing using the wrong kind of bitreader VP8 uses a probabilistic bool coder, not a straight bit coder. This fixes parsing when error-resilient is set. This commit includes a copy of libvpx's bool coder, BSD licensed. https://bugzilla.gnome.org/show_bug.cgi?id=652694	2012-11-01 20:53:47 +00:00
Olivier Crête	97c3f3617c	rtpvp8: Reject unknown bitstream versions	2012-11-01 20:53:47 +00:00
Edward Hervey	74a1a704bf	rtpvp8: Fix unitialized variable Makes macosx compiler happy.	2012-11-01 20:53:47 +00:00
Sjoerd Simons	6ed6318076	rtpvp8depay: Accept packets with only one byte of data When fragmenting partions it can happen that an RTP packet only caries 1 byte of RTP data.	2012-11-01 20:53:47 +00:00
Sjoerd Simons	a45e7a3fc0	rtpvp8pay: Treat the frame header just like any other partition When setting up the initial mapping just act as if the global frame information is another partition. This saves special-casing it later in the actual packetizing code.	2012-11-01 20:53:47 +00:00
Sjoerd Simons	e9f4e9342f	rtpvp8: Add simple payloaders and depayloaders for VP8 Minimal implementation of http://www.webmproject.org/code/specs/rtp/, version 0.3.2	2012-11-01 20:53:47 +00:00
Wim Taymans	d6fd0ebd04	gstpay: fix for 1.0 events Caps events are sometimes not followed by a buffer but by an event. Flush any pending caps before we make a packet with the event. Chain up to the parent event handler before we attempt to push RTP packets, it might be a segment event.	2012-11-01 18:42:39 +00:00
Wim Taymans	05232c55a5	gstdepay: fix small leak	2012-11-01 18:42:24 +00:00
Wim Taymans	08e5a197b4	gstdepay: add support for events Conflicts: gst/rtp/gstrtpgstdepay.c	2012-11-01 18:18:19 +00:00
Wim Taymans	54b783b5a3	rtpgstpay: add support for sending events We currently only send tags and custom events. The other events might interfere with the receiver timings or are otherwise handled by RTP. Conflicts: gst/rtp/gstrtpgstpay.c	2012-11-01 18:06:11 +00:00
Wim Taymans	6502d08e43	gstpay: rewrite payloader Use adapter to assemble the payload and make a flush function to turn this payload into (fragmented) packets. Conflicts: gst/rtp/gstrtpgstpay.c gst/rtp/gstrtpgstpay.h	2012-11-01 17:57:52 +00:00
Douglas Bagnall	e3c77ba709	videomixer: get height via GST_VIDEO_FRAME_HEIGHT, not _WIDTH https://bugzilla.gnome.org/show_bug.cgi?id=687330	2012-11-01 13:03:44 +00:00
Douglas Bagnall	79403bcb0c	videbox: fix border filling for gray formats Get the height via GST_VIDEO_FRAME_HEIGHT, not _WIDTH. https://bugzilla.gnome.org/show_bug.cgi?id=687330	2012-11-01 13:02:16 +00:00
Wim Taymans	c0713e4b80	gstdepay: check for correct fragment offset Make sure we only insert the rtp packet in the adapter when the frag_offset matches. When the first packet of a fragment is dropped, it avoids putting the remaining packets in the adapter and processing the partial fragment. Conflicts: gst/rtp/gstrtpgstdepay.c	2012-11-01 12:09:47 +00:00
Wim Taymans	8a402e0c06	gstpay: set C flag on all buffers of the fragment Set the C flags on all the fragments instead of only those with caps in them. This makes it easier in the receiver to check if there is a caps in the assembled fragments just by looking at the last RTP packet flags.	2012-11-01 12:06:08 +00:00
Wim Taymans	d78ff07f7d	gstdepay: use the capsversion Take the caps from the input caps and store it in the slot given by capsversion.	2012-11-01 11:37:44 +00:00
Wim Taymans	936c3819b5	gstpay: send caps inline Place the capsversion on the outgoing caps so that they end up in an SDP as well. Receivers need to know what capsversion a particular caps is for to be able to match the caps to the CV in the RTP packets. Place the caps inside the RTP packet whenever the caps change. Based on patch by Andrzej Bieniek <andrzej.bieniek@pure.com> Conflicts: gst/rtp/gstrtpgstpay.c gst/rtp/gstrtpgstpay.h	2012-11-01 11:34:33 +00:00
Andrzej Bieniek	3b1931a039	gstpay: add debug Conflicts: gst/rtp/gstrtpgstpay.c	2012-11-01 11:28:50 +00:00
Andrzej Bieniek	ee5ecc7773	depay: correctly skip caps header size Conflicts: gst/rtp/gstrtpgstdepay.c	2012-11-01 11:27:13 +00:00
Tim-Philipp Müller	ef0805ea14	matroskademux: put streamheaders on vorbis/speex/flac/theora caps to make remuxing work https://bugzilla.gnome.org/show_bug.cgi?id=640589	2012-10-30 23:29:46 +00:00
Tim-Philipp Müller	752cf98745	gst: fix variable order in some Makefile.am https://bugzilla.gnome.org/show_bug.cgi?id=687013	2012-10-27 23:27:38 +01:00
Antoine Tremblay	a1c86de09a	gst: add various missing GST_PLUGINS_BASE_LIBS in Makefile.am Those plugins depend on either libgstaudio or libgstvideo, which are in gst-plugins-base. https://bugzilla.gnome.org/show_bug.cgi?id=687013	2012-10-27 23:26:41 +01:00
Alexey Fisher	29cd24bc41	matroskademux: mark invisible VP8 frames with the DECODE_ONLY flag https://bugzilla.gnome.org/show_bug.cgi?id=654259	2012-10-27 14:46:02 +01:00
Stas Sergeev	238a5ec826	multifilesrc: fix stop index handling Make sure the stop index is always honoured. Avoids endless loop if one wants to read and output the same file N times, for example. https://bugzilla.gnome.org/show_bug.cgi?id=654853	2012-10-26 11:04:01 +01:00
Руслан Ижбулатов	78193dfe71	matroskademux: Support recursive SimpleTags Fixes #682644 Depends on #682615	2012-10-26 10:16:42 +02:00
Руслан Ижбулатов	cd719bb808	matroskademux: Expand the tag mapping. * Also expose unknown tags as key=value pairs. * Arrange tag map in the same order tags are listed in Matroska spec, leaving unmapped tags as comments. * More specific TODOs. * Remove duplicate DATE define. Fixes #682615 Depends on #682524	2012-10-26 10:12:52 +02:00
Sebastian Dröge	6c635ce64f	matroskademux: Fix uninitialized variable compiler warning	2012-10-26 10:09:39 +02:00
Руслан Ижбулатов	71fd688ef0	matroskademux: Matroska tag TargetType support * Reads TargetType and TargetTypeValue from a Tag. * After Tag is completely read, processes taglist, substituting some of the tags depending on target type value and the presence of video/subtitle streams. * Supports reading two new simpletags - PART_NUMBER and TOTAL_PARTS Depends on #682448 Fixes #682524	2012-10-26 10:08:18 +02:00
Руслан Ижбулатов	b75628f041	matroskademux: Per-track tags for Matroska Requires Matroska file to have sane layout (track info before tag info). Uses replace-merge. Makes track UIDs 64-bit. Fixes #682448	2012-10-26 10:03:55 +02:00
Tim-Philipp Müller	fe7236230c	multifilesrc: fix typo in property description	2012-10-25 20:19:44 +01:00
Michael Smith	a88caf84b4	qtdemux: read video format header fully (so we can find 'pasp' atoms) for more fourccs. Fixes aspect ratio of prores files.	2012-10-25 12:18:50 -07:00
Thiago Santos	02d91dcd24	imagefreeze: the new get_caps already does the filter intersection It should be faster to pass the caps to intersect as the filter caps, rather than using NULL and intersecting 'manually' later. https://bugzilla.gnome.org/show_bug.cgi?id=686837	2012-10-25 10:32:17 -03:00
Thiago Santos	115581eb2e	imagefreeze: avoid assertion when using accept caps query This query must receive a fixed caps, so imagefreeze should fixate its framerate before sending the query downstream. https://bugzilla.gnome.org/show_bug.cgi?id=686837	2012-10-25 09:39:36 -03:00
Arnaud Vrac	bc79fe565c	qtdemux: use correct type for channel-mask bitmask Fixes crash on 32-bit systems.	2012-10-24 12:54:08 +01:00
Tim-Philipp Müller	7275860bdd	flacparse: fix coverart extraction if vorbis comments come after picture header See sample file for bug #684701.	2012-10-23 16:02:05 +01:00
Tim-Philipp Müller	7c41f42eec	flacparse: ignore bad headers if we have a valid STREAMINFO header If we run into any header parsing issues and we have a valid STREAMINFO header already, don't error out, but just stop header parsing and try to find some audio frames. https://bugzilla.gnome.org/show_bug.cgi?id=684701	2012-10-23 13:56:54 +01:00
Tim-Philipp Müller	49cc719809	flacparse: post proper error message and fix buffer leak on header parsing error https://bugzilla.gnome.org/show_bug.cgi?id=684701	2012-10-23 13:56:54 +01:00
Michael Smith	150bd97e96	qtdemux: with raw audio, set a default channel-mask for multichannel audio. This doesn't actually parse 'chan' because it's absurdly complex.	2012-10-22 22:34:43 -07:00
Sebastian Rasmussen	9fc62a58e3	updsrc: fix typo causing compilation error gstudpsrc.c: In function 'gst_udpsrc_create': gstudpsrc.c:365: error: 'ret' may be used uninitialized in this function https://bugzilla.gnome.org/show_bug.cgi?id=686642	2012-10-22 23:19:28 +01:00
Wim Taymans	a2eead3d60	avi_ fix invert function Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686550	2012-10-22 11:55:59 +02:00
Wim Taymans	0e3ef30c31	avi: fix debug	2012-10-22 11:55:22 +02:00
Wim Taymans	199aaa4021	qtdemux: add support for 'generic' samples Add support for stuffing a complete stream into 1 sample. See https://bugzilla.gnome.org/show_bug.cgi?id=686550	2012-10-22 11:39:37 +02:00
Tim-Philipp Müller	aa3ba65eb5	qtdemux: don't leak gst_riff_strf_auds in case of MS/RIFF audio https://bugzilla.gnome.org/show_bug.cgi?id=681192	2012-10-19 19:26:45 +01:00
Mark Nauwelaerts	35cd53867c	matroskamux: unsigned subtitle template	2012-10-19 16:14:01 +02:00
Youness Alaoui	13328bc129	videomixer2: Fix race condition where a src setcaps is ignored If both pads receive data at the same time, they will both get their sink_setcaps called which will call the src_setcaps, but there is a race condition where the second one might not be called. Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=683842	2012-10-19 12:10:31 +02:00
Mark Nauwelaerts	5742352e10	matroskamux: do not use unoffical V_MJPEG codec id Since it's not spec'ed, consider it a VfW compatibility case. Many applications (e.g. avidemux) don't understand the unofficial V_MJPEG id. Fixes #659837. Conflicts: gst/matroska/matroska-mux.c	2012-10-18 18:29:40 +01:00
Tim-Philipp Müller	488549bb54	Use gst_element_class_set_static_metadata() where possible. Avoids some string copies. Also re-indent some stuff. Also some indent fixes here and there.	2012-10-17 17:46:34 +01:00
Wim Taymans	e9040e90a5	jpegdepay: store quant tables in zigzag order	2012-10-17 14:23:01 +02:00
Wim Taymans	d5fd524a0c	rtsession: fix compiler warning	2012-10-17 13:55:45 +02:00
Wim Taymans	26a21e85e2	rtpbin: clarify the ntp-sync option	2012-10-17 13:35:07 +02:00
Wim Taymans	f17db5c4ed	rtpsession: update caps in the source Inform the source when caps changed. This was removed in the port to 1.0 leaving the source unaware of the clock-rate and unable to interpollate rtp timestamps for SR packets.	2012-10-17 13:22:40 +02:00
Wim Taymans	f4eef3f48d	rtpbin: set PTS and DTS in jitterbufffer	2012-10-17 12:46:32 +02:00
Wim Taymans	796c1d8029	rtpbin: disable check for ntp-sync Disable the check for the ntp-sync method. It is expected that a rather larger offset needs to be applied with this method.	2012-10-17 12:27:03 +02:00
Wim Taymans	1cebcfa8c2	rtpbin: use running-time for NTP time When use-pipeline-clock is set, use the running-time of the pipeline to calculate the NTP timestamps. This method would previously only work when the base-time is set to 0 but with this change it can also work with different offsets and we can also implement pause/resume of the sender and receiver now.	2012-10-17 12:26:05 +02:00
Wim Taymans	5ec642d0c3	videocrop: port to videofilter	2012-10-17 10:20:12 +02:00
Wim Taymans	3ef7c8ab93	videobox: use out_info for out properties	2012-10-17 09:36:50 +02:00
Wim Taymans	f701d980e6	median: small cleanups	2012-10-16 14:40:19 +02:00
Wim Taymans	0e21e80e9b	median: remove now that it is in videofilter	2012-10-16 13:56:19 +02:00
Wim Taymans	9e67891f72	videomedian: copy media to videomedian Copy the median video filter to videofilters and rename to videomedian.	2012-10-16 13:47:24 +02:00
Wim Taymans	b893197317	media: port to 1.0	2012-10-16 13:16:29 +02:00
Tim-Philipp Müller	f94572fb36	avidemux: append palette data to paletted 8-bit RGB frames Fixes playback of 8-bit indexed RGB videos, with fixes in -base. https://bugzilla.gnome.org/show_bug.cgi?id=686046	2012-10-16 01:09:05 +01:00
Tim-Philipp Müller	e9682b938a	qtdemux: don't assert if upstream size is not available when guessing bitrates Fixes abort in push mode where the source is not seekable and the size of the file is not available, as with cat foo.mp4 \| gst-launch-1.0 playbin uri=fd://0 Less noticable with releases, since we disable all g_assert() there. https://bugzilla.gnome.org/show_bug.cgi?id=686008	2012-10-13 00:08:01 +01:00
Michael Smith	3a3a7c38aa	qtdemux: allow more streams. Bump this constant to 32, which should be enough for real-world files.	2012-10-12 14:38:33 -07:00
Michael Smith	d60c9ce2a4	qtdemux: support more different fourcc values for other ProRes variants.	2012-10-12 14:35:49 -07:00
Wim Taymans	adb70e89f9	rtspsrc: remove unused include	2012-10-10 12:05:34 +02:00
Rasmus Rohde	11ed7c0373	multiudpsink: add multicast-iface property udpsrc already has support for setting the multicast interface, which is useful for multi-homed machines. This patch adds the same code to the multiudpsink. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685864	2012-10-10 11:48:25 +02:00
Wim Taymans	54f049c355	multiudpsink: don't error on send errors but only warn Don't error on send errors but simply post a warning, it's possible that the next packet will be fine.	2012-10-10 11:32:17 +02:00
Rasmus Rohde	6c169312d1	multiudpsink: add force-ipv4 option Add an option to the multiudpsink that makes it possible to force the use of an IPv4 socket. This can e.g. be used to handle the issue described in https://bugzilla.gnome.org/show_bug.cgi?id=682481	2012-10-10 10:28:24 +02:00
Wim Taymans	2955f0e10c	multiudpsink: remove unused field	2012-10-10 10:18:52 +02:00
Wim Taymans	f4e1bb02b7	udpsrc: use negotiated allocator or pool Use the base class to allocate a buffer for us because it knows how to use the negotiated allocator or bufferpool.	2012-10-10 10:10:26 +02:00
Wim Taymans	e8d951ed68	multiudpsink: post error when something goes wrong	2012-10-10 10:09:37 +02:00
Wim Taymans	15c2b997e9	spectrum: elements post element messages	2012-10-10 10:09:10 +02:00
Michael Smith	7aed5a4b4b	deinterleave: output channels should be marked as MONO, not FRONT_LEFT, if we're not preserving input channel positions.	2012-10-05 15:12:27 -07:00
Michael Smith	7522cd1595	interleave: use gst_audio_channel_positions_to_mask instead of a local copy of half of it. Handles some values more correctly.	2012-10-04 15:13:20 -07:00
Rasmus Rohde	47a8eb7ca8	gstrtpdepay: don't leak input buffer The rtp buffer is never unmapped in the normal code exit path of gst_rtp_gst_depay_process(..) resulting in a memory leak. https://bugzilla.gnome.org/show_bug.cgi?id=685512	2012-10-04 19:44:28 +01:00
Sebastian Dröge	1ac6a782c3	videobalance: Add support for NV12 and NV21	2012-10-04 18:37:48 +02:00
Patricia Muscalu	7a863e4d8d	rtph264pay: do not push unmapped data Also do not use a GstBuffer after it has been pushed into the adapter. https://bugzilla.gnome.org/show_bug.cgi?id=685213	2012-10-04 09:22:50 +01:00
Michael Smith	b04b1b5089	meta info: threadsafe registration using g_once	2012-10-03 10:51:45 -07:00
Mark Nauwelaerts	b10829d6c8	avidemux: push mode; handle some initial junk before hdrl list Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685059	2012-10-01 15:50:53 +02:00
Tim-Philipp Müller	e6d37eb30a	Purge references to liboil https://bugzilla.gnome.org/show_bug.cgi?id=673285	2012-09-29 12:41:37 +01:00
Mark Nauwelaerts	cb0e4b2059	avidemux: recognize all xsub frames as keyframes Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684977	2012-09-28 17:04:42 +02:00
Mark Nauwelaerts	511dfa5ee5	avidemux: push mode: find the correct chunk for segment following seek Fixes https://bugzilla.gnome.org/show_bug.cgi?id=684977	2012-09-28 17:04:42 +02:00
Arnaud Vrac	f0db4a8213	qtdemux: fix parsing in push mode when moov atom is at the end When playing an mp4 file with the MOOV atom at the end of the file, playback fails with the error message "no 'moov' atom within the first 10 MB". This is due to a mistake in the upstream_size typing, making the seek to the end of file never happening. https://bugzilla.gnome.org/show_bug.cgi?id=684972	2012-09-27 22:20:19 +01:00
Andre Moreira Magalhaes (andrunko)	25803d651b	gamma: remove duplicate entries at format at caps Avoids extra caps/structures processing	2012-09-27 15:50:49 -03:00
Wim Taymans	dbe941338d	rtpvrawdepay: negotiate pool with srcpad caps	2012-09-27 14:15:50 +02:00
Tim-Philipp Müller	f5e0321dfc	videomixer: clear video frame more correctly Make sure not to touch memory that doesn't belong to our frame, we might be one part of a side-by-side 3D frame, or in a picture-in-picture scenario.	2012-09-26 09:28:59 +01:00
Tim-Philipp Müller	c203ce2dbe	flvdemux: minor clean-up Use GstByteWriter, because we can, and g_value_take_boxed.	2012-09-26 00:44:59 +01:00
Dmitriy Samonenko	7d4b6f655e	flvdemux: fix speex audio decoding by creating fake stream header https://bugzilla.gnome.org/show_bug.cgi?id=683622	2012-09-26 00:16:06 +01:00
Tim-Philipp Müller	626e0258e3	videomixer: fix warnings when using transparent background gst_video_frame_map() increases the refcount, which makes the buffer not writable any more technically, so calling gst_buffer_memset() on it will cause nasty warnings. Unit test disabled because it very rarely (for me) fails, possibly negotiation-related. https://bugzilla.gnome.org/show_bug.cgi?id=684398	2012-09-25 23:31:34 +01:00
Robert Swain	03e5376827	deinterlace: Add some useful debug logging	2012-09-25 17:05:37 +02:00
Robert Swain	33dd81569f	deinterlace: Fix telecine This only affects behaviour in telecine cases with pattern locking enabled. The default case should be untouched. This works with the output from fieldanalysis at least, but the field order looks swapped for telecine mixed buffers with the David_slides_Schleef clip.	2012-09-25 17:04:54 +02:00
Edward Hervey	ac9394de29	videomixer: Fix leak	2012-09-25 14:18:35 +02:00
Tim-Philipp Müller	ebe0b1887a	smpte: send stream-start event	2012-09-23 16:51:31 +01:00
Tim-Philipp Müller	8e3c7fa799	multipartmux: send stream-start event	2012-09-23 16:51:24 +01:00
Tim-Philipp Müller	154404fa43	matroskamux: send stream-start	2012-09-23 16:33:35 +01:00
Tim-Philipp Müller	bc37b9f4fc	qtmux: send stream-start event	2012-09-23 16:33:35 +01:00
Tim-Philipp Müller	ea7f8a919c	interleave: add a bunch of FIXMEs Needs some more work, so stream-start, caps and tags are sent in the right order.	2012-09-23 16:33:35 +01:00
Tim-Philipp Müller	1c3c8c64e6	flvmux: send stream-start event	2012-09-23 16:33:34 +01:00
Tim-Philipp Müller	c3f62d7ead	avimux: send stream-start event	2012-09-23 16:33:34 +01:00
Olivier Crête	0363c1cebf	rtpdtmfdepay: Use 1.0-style caps negotiation and audio/x-raw	2012-09-22 15:00:27 -04:00
Tim-Philipp Müller	8b20603f8b	rtspsrc: answer URI query Without this, something also answered the query with TRUE but without setting a uri, not sure what that was..	2012-09-21 23:33:47 +01:00

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7212 commits