gstreamer

mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-04 15:36:35 +00:00

Author	SHA1	Message	Date
Sebastian Dröge	d585bd7bbd	rtptheorapay: Don't send headers twice if we got them from the caps already	2013-12-04 21:58:29 +01:00
Sebastian Dröge	d105de6e0f	rtptheorapay: Don't leak config data when receiving a second CAPS event	2013-12-04 21:58:29 +01:00
Sebastian Dröge	0915d696c7	rtpvorbispay: Don't send headers twice if we got them from the caps already	2013-12-04 21:58:29 +01:00
Sebastian Dröge	967280df42	rtpvorbispay: Don't leak config data when receiving a second CAPS event	2013-12-04 21:58:29 +01:00
Sebastian Dröge	d87f6cf483	rtpstreamdepay: Add RFC4571 RTP stream depayloading element https://bugzilla.gnome.org/show_bug.cgi?id=719829	2013-12-04 21:58:29 +01:00
Sebastian Dröge	c5284dc047	rtpstreampay: Add RFC4571 RTP stream payloading element https://bugzilla.gnome.org/show_bug.cgi?id=719829	2013-12-04 21:58:29 +01:00
Thiago Santos	1fd094d96b	qtdemux: improve fragment-start tracking Some buffers can have multiple moov atoms inside and the strategy of using the gst_adapter_prev_pts timestamp to get the base timestamp for the media of the fragment would fail as it would reuse the same base timestamp for all moofs in the buffer instead of accumulating the durations for all of them. Heres a better explanation of the issue: qtdemux receives a buffer where PTS(buf) = X buf -> moofA \| moofB \| moofC The problem was that PTS(buf) was used as the base timestamp for all 3 moofs, causing all buffers to be X based. In this case we want only moofA to be X based as it is what the PTS on buf means, and the other moofB and moofC just use the accumulated timestamp from the previous moofs durations. To solve this, this patch uses gst_adapter_prev_pts distance result, this allows qtdemux to calculate if it should use the resulting pts or just accumulate the samples as it can identify if the moofs belong to the same upstream buffer or not. https://bugzilla.gnome.org/show_bug.cgi?id=719783	2013-12-04 10:36:38 -03:00
Wim Taymans	0d55724a2b	audioparsers: don't leak template caps	2013-12-04 09:12:07 +01:00
Wim Taymans	e0a5c07e8d	audioparsers: use ACCEPT_INTERSECT flag The parser can accept input that is not completely specified. Use the ACCEPT_INTERSECT flag on the sinkpad to tweak the acceptcaps function to check for intersection only. This allows us to proxy downstream constraints while still allowing non-subset caps as input. We can then also remove the appended template caps workaround. Make a unit-test to check the new feature. This reverts commit `26040ee38c` Fixes https://bugzilla.gnome.org/show_bug.cgi?id=705024	2013-12-03 22:26:44 +01:00
Wim Taymans	e3f393f7e6	audioparsers: remove fields from filter We need to remove the fields from the filter when we can convert between them.	2013-12-03 21:39:57 +01:00
Wim Taymans	e8313a1e70	audioparsers: refactor code to remove caps fields	2013-12-03 21:29:13 +01:00
Tim-Philipp Müller	a424fb289b	deinterlace: microoptimisation: avoid some unnecessary GValue copies	2013-12-02 00:10:43 +00:00
Tim-Philipp Müller	63b0e84add	deinterlace: fix off-by-one crash when downstream caps contain a list of framerates https://bugzilla.gnome.org/show_bug.cgi?id=719544	2013-12-01 23:33:04 +00:00
Thiago Santos	079dde49ed	qtdemux: Use the timestamp of the moof as the base fragment start In SmoothStreaming fragmented scenario, the timestamps are calculated starting from the fragment buffer timestamp. When there is a not-linked return from downstream, qtdemux will return upstream and will keep the non-pushed data into its adapter. On a new fragment buffer pushed to qtdemux, the new buffer timestamp would overwrite the previous one that should be used on the still to be pushed buffers. Because of this, this patch will also update the fragment_start timestamp from the adapter last pts to make sure the moof and timestamps are in sync and will result in correct timestamps for all fragments.	2013-11-29 17:28:48 -03:00
Thiago Santos	45c16599ff	qtdemux: avoid re-reading the same moov and entering into loop In the scenario of "mdat \| moov (with fragmented artifacts)" qtdemux could read the moov again after the mdat because it was considering the media as a fragmented one. To avoid this loop this patch makes it store the last processed moov_offset to avoid parsing it again. And it also checks if there are any samples to play before resturning to the mdat, so that it knows there is new data to be played. https://bugzilla.gnome.org/show_bug.cgi?id=691570	2013-11-29 17:28:48 -03:00
Thiago Santos	fcc78aa3bd	qtdemux: do not free streams if they were not created locally When parsing a trak only free streams on failures if those streams were created locally. They could have been created from a previous fragment, in this case we they have valid info from the other fragment. Including pads. https://bugzilla.gnome.org/show_bug.cgi?id=691570	2013-11-29 17:28:48 -03:00
Sebastian Dröge	220a947dc7	videomixer: Simplify NV12/21 blending code macros	2013-11-29 19:57:46 +01:00
Sebastian Dröge	b0529e0fe8	videomixer: Fix segfault when filling the background of a UYVY frame https://bugzilla.gnome.org/show_bug.cgi?id=712401	2013-11-29 19:52:34 +01:00
Tim-Philipp Müller	4278ab18ff	qtdemux: fix compilation with gst debuging disabled qtdemux.c:9452:1: error: label at end of compound statement	2013-11-29 09:21:52 +00:00
Jonas Holmberg	0ab0421759	rtph264pay: Map inbuffer once only Do not call gst_buffer_extract() twice since each call will map and unmap the biffer. https://bugzilla.gnome.org/show_bug.cgi?id=719434	2013-11-28 16:08:40 -05:00
Tim-Philipp Müller	b8f689a9d9	videoflip: don't crash on tag events without orientation tag Would crash in g_free() trying to free an uninitialised pointer. https://bugzilla.gnome.org/show_bug.cgi?id=719497	2013-11-28 16:09:04 +00:00
Wim Taymans	e8edecc56e	rtpsession: don't unref buffer twice Cleaning the packet info will already unref the buffer. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715078	2013-11-28 16:51:13 +01:00
Jan Schmidt	b3b89dfec1	qtdemux: Add HydrogenAudio ReplayGain tags Identical to the itunes (tm) version, but labelled with org.hydrogenaudio.replaygain as the producer.	2013-11-28 22:36:44 +11:00
Mathieu Duponchelle	532598e360	videomixer: explicitly fail when alpha information would have been lost.	2013-11-27 16:35:46 +01:00
Sebastian Dröge	fb14f66696	matroska-demux: Allow a bit more variation when detecting common framerates Instead of +/- 1ns we allow 2ns now. Due to rounding errors there are some Matroska files out there with 33.333331ms per frame for 30fps.	2013-11-26 11:17:42 +01:00
Sebastian Dröge	20ad174679	matroska-demux: Use gst_util_double_to_fraction() instead of GValue magic	2013-11-26 10:21:04 +01:00
Nicolas Dufresne	c42bc9efa0	videoflip: Set default method at contruction Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712333	2013-11-25 14:03:21 -05:00
Wim Taymans	710d1f3a2a	rtpjitterbuffer: improve clear-pt-map handling Don't reset the expected output seqnum when clearing the pt map because this could stall the jitterbuffer forever. Add a unit test for this. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=709800	2013-11-25 15:52:22 +01:00
Jan Schmidt	fdfc6a2a86	qtdemux: Discard 2 byte subpicture packets As for text subtitles and as suggested in #712643, throw away the 2 byte terminator packets that some encoders insert. This will make things better when remuxing and causes generation of gap events.	2013-11-25 12:24:22 +11:00
Tim-Philipp Müller	901ec63462	rtpjitterbuffer: fix wake-up when new buffers come in after running empty Spotted by 'gratias' on IRC. Probably introduced in recent refactoring. https://bugzilla.gnome.org/show_bug.cgi?id=715039	2013-11-25 00:37:50 +00:00
Mark Nauwelaerts	643e6fdc36	matroskamux: correctly handle negative relative timestamps ... rather than scaling these as unsigned. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712744 Based on patch by Krzysztof Kotlenga <pocek@users.sf.net>	2013-11-23 12:25:05 +01:00
MathieuDuponchelle	83f8ee1d41	videomixer2: Merge tag events to send them in collected. Otherwise there were race conditions where we would send tags on a flushing srcpad. We have a test for that in GES, but this should be tested systematically with harness in the future as I believe it is useful for exactly that kind of cases. https://bugzilla.gnome.org/show_bug.cgi?id=708165	2013-11-22 18:54:35 -03:00
Thibault Saunier	a45d470236	qtdemux: Use GstVideoInfo helper to create caps for raw video This way we do not miss mandatory fields in caps. At the same time use the gst_pb_utils_get_codec_description helper to get codec description. https://bugzilla.gnome.org/show_bug.cgi?id=712335	2013-11-22 18:52:54 -03:00
Thibault Saunier	6ff7522ba2	matroskademux: Use GstVideoInfo helper to create caps for raw video This way we do not miss mandatory fields in caps. At the same time use the gst_pb_utils_get_codec_description helper to get codec description. https://bugzilla.gnome.org/show_bug.cgi?id=712328	2013-11-22 18:52:54 -03:00
Thibault Saunier	1fc591238b	multifilesrc: Implement seeking in case of multiple images https://bugzilla.gnome.org/show_bug.cgi?id=712254	2013-11-22 18:52:54 -03:00
Wim Taymans	4c9474905b	rtpjitterbuffer: pass downstream flowreturn to upstream Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712722	2013-11-22 12:27:31 +01:00
Tim-Philipp Müller	d9c2914c90	g_memmove() is deprecated Just use plain memmove(), g_memmove() is deprecated in recent GLib versions. https://bugzilla.gnome.org/show_bug.cgi?id=712811	2013-11-21 15:30:34 +00:00
Wim Taymans	3a1199c2f7	rtpvorbisdepay: handle packets > 0xffff Handle input packet sizes larger than 16 bits in the depayloader. Remove size restrictions on the payloader.	2013-11-21 11:32:15 +01:00
Wim Taymans	43e9b56122	rtptheoradepay: handle packets > 0xffff Reorganize some things in the depayloader so that it can handle packets larger than 16 bits. Remove the size restriction on the payloader.	2013-11-21 11:30:28 +01:00
Jan Schmidt	81e2c8130a	isomp4: Handle mp4s subpicture streams better. Clean up the handling of mp4s streams. Use the generic esds descriptor function to extract the palette, instead of hard coding a wrong magic offset. Add some more size safety checks when parsing ES descriptors, and replace magic numbers with the descriptive constants that are already defined. Enhance dump output for stsd atoms. Streams from both bug 712643 and historic bug 568278 now both work correctly. Fixes: #712643	2013-11-21 02:28:27 +11:00
Jan Schmidt	217d2d8deb	qtdemux: Sort fourcc declarations and remove duplicates	2013-11-20 22:08:25 +11:00
Jan Schmidt	b6f581eecc	qtdemux: Merge all the fourcc headers into one Remove qtdemux_fourcc.h and ftypcc.h and put it all in fourcc.h	2013-11-20 21:48:03 +11:00
Wim Taymans	0c6f4efe4a	rtpjitterbuffer: avoid mapping the buffer Reuse the parsed structure to get the timestamps.	2013-11-19 10:12:00 +01:00
Tim-Philipp Müller	28f524a551	rtspsrc: fix 'make check' Fix generic/states check. Also, g_return_if_fail() is not for internal state checking.	2013-11-18 17:13:49 +00:00
Tim-Philipp Müller	d506409af5	docs: get rid of 'Since: 0.10.x' markers And some gtk-doc markup fixes.	2013-11-18 14:47:35 +00:00
Tim-Philipp Müller	548e756e0a	rtpmanager: fix Since markers Should be next stable release series version	2013-11-16 12:15:14 +00:00
George Kiagiadakis	387e3b918a	rtpjitterbuffer: Fix stats property field names and documentation	2013-11-15 16:23:34 +02:00
Torrie Fischer	acf74435e3	gstrtpsession: Implement a number of feedback packet statistics Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711693	2013-11-15 15:21:19 +01:00
Thiago Santos	cfdadd4114	qtdemux: remove math operation from loop The elst_offset doesn't change inside the loop, so compute it outside	2013-11-14 18:15:20 -03:00
Stefan Sauer	1a4e7338d9	qtmux: fix playback regression In `ae1150e85c` flipping a condition misaligned the else branch, where for there condition that was change there is none. Fixes #712303	2013-11-14 20:56:36 +01:00
Wim Taymans	b450d31503	rtpjitterbuffer: rename property to 'stats' This makes the unit test work. We can later also add more stats, not specific to retransmission. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711411	2013-11-14 09:24:26 +01:00
Torrie Fischer	22ceb80ba9	rtpjitterbuffer: implement rtx statistics	2013-11-14 09:24:26 +01:00
Wim Taymans	2e5b462ae3	jitterbuffer: advance expected seqnum after dropping After dropping a buffer, move our expected seqnum Conflicts: gst/rtpmanager/gstrtpjitterbuffer.c	2013-11-13 12:02:57 +01:00
Wim Taymans	a065b4fcde	gstpay: only send one caps Only send one caps in a packet. Two caps can happen when setcaps is called and the config-interval expires at the same time.	2013-11-13 12:02:57 +01:00
Sebastian Dröge	9ae6981578	rtspsrc: Use the synced buffer mode in auto mode if a clock provider is in the SDP	2013-11-13 10:54:19 +01:00
Wim Taymans	e4bc81d7d2	rtpsession: remove collision reconfigure event Remove bogus reconfigure event on collision, we don't want to send the event on the receiving RTP pad and the collision event is now handling this case. See https://bugzilla.gnome.org/show_bug.cgi?id=711560	2013-11-11 15:27:18 +01:00
Julien Isorce	b32fc6f416	gstrtpsession: send custom upstream event "GstRTPCollision" on send_rtp_sink pad See https://bugzilla.gnome.org/show_bug.cgi?id=711560	2013-11-11 15:25:52 +01:00
Mark Nauwelaerts	49d52a64d6	ac3parse: correctly handle timestamps when parsing x-private1-ac3 ... the way it has always worked fine in a52dec.	2013-11-11 13:35:29 +01:00
George Kiagiadakis	b81b2efa3e	rtpjitterbuffer: fix crash when do-retransmission=true and a lot of buffers are lost The problem here was that the jitterbuffer lock was unlocked to push the event, but that caused another thread to remove the timer currently being processed, probably because the amount of rtx events (and therefore timers) was getting too high. The solution is to unlock and push the event only after timer processing has finished. fixes https://bugzilla.gnome.org/show_bug.cgi?id=711131	2013-11-11 11:51:45 +01:00
Per x Johansson	b3e0b1dbca	matroskademux: Avoid division by zero assert in gst_matroska_demux_search_pos https://bugzilla.gnome.org/show_bug.cgi?id=711829	2013-11-11 11:30:54 +01:00
Philippe Normand	0ee332378b	wavenc: generate a non-empty data header Restore the behavior of the element to the state before commit `db29522a43`. A non-empty header is generated and when the EOS event is received the header is generated again, this time with the correct size. https://bugzilla.gnome.org/show_bug.cgi?id=711699	2013-11-09 11:22:12 +01:00
Wim Taymans	c8db05d610	rtpsource: update receiver stats for sender An internal sender in a session is also a receiver of its own packets so update the receiver stats. Other senders in the session will use this info to generate correct RB blocks in their SR reports.	2013-11-07 16:24:30 +01:00
Wim Taymans	268a75e705	rtpsource: refactor receiver stats update	2013-11-07 16:24:30 +01:00
Thiago Santos	33ebda8ecf	qtdemux: handle fragmented files with mdat before moofs Assume a file with atoms in the following order: moov, mdat, moof, mdat, moof ... The first moov usually doesn't contain any sample entries atoms (or they are all set to 0 length), because the real samples are signaled at the moofs. In push mode, qtdemux parses the moov and then finds the mdat, but then it has 0 entries and assumes it is EOS. This patch makes it continue parsing in case it is a fragmented file so that it might find the moofs and play the media. https://bugzilla.gnome.org/show_bug.cgi?id=710623	2013-11-07 11:22:04 -03:00
Thiago Santos	0e78ffc9d6	qtdemux: When using a buffered mdat, store all received data for later use In push mode, when qtdemux can't use a seek to skip the mdat buffer it has to buffer it for later use. The issue is that after parsing the next moov/moof, there might be some trailing bytes from the next atom in the file. This data was being discarded along with the already parsed moov/moof and playback would fail to continue after the contents of this moov/moof are played. This is particularly bad on fragmented files that have the mdat before the corresponding moof. So you'd get: mdat\|moof\|mdat\|moof ... When a moof was received, it usually came with some extra bytes that would belong to the next mdat (because upstream doesn't care about atoms alignment). So those bytes were being discarded and playback would fail. This patch makes qtdemux store those extra bytes to reuse them later after the mdat is emptied. https://bugzilla.gnome.org/show_bug.cgi?id=710623	2013-11-07 11:22:03 -03:00
Sebastian Dröge	fd89e36c8a	multiudpsink: Also use the bind-port property if no bind-address was given	2013-11-07 09:50:39 +01:00
Sebastian Dröge	111982de28	rtpvp8pay: Make Picture ID mode configurable and default to no picture ID Some implementations (linphone) only support no picture at all in the stream and will fail if one is provided. https://bugzilla.gnome.org/show_bug.cgi?id=711497	2013-11-05 17:26:49 +01:00
Paul HENRYS	8eceb8f327	Add call to gst_rtp_h264_pay_clear_sps_pps() when receiving a STREAM_START event https://bugzilla.gnome.org/show_bug.cgi?id=692787	2013-11-04 14:36:28 -05:00
Rico Tzschichholz	b137f79581	rtsp: Add missing gio-2.0 deps and includes	2013-11-02 23:12:13 +01:00
Sebastian Dröge	f180f3d1ba	audioiirfilter: Fix initialization coefficient handling Broke unit test.	2013-11-01 18:31:36 +01:00
Aleix Conchillo Flaque	82b8374af8	rtspsrc: allow setting tls certificate validation flags Added a new property "tls-validation-flags". If the url transport is TLS, the validation flags will be set to the rtsp connection. https://bugzilla.gnome.org/show_bug.cgi?id=711230	2013-11-01 16:47:36 +01:00
Sebastian Dröge	2559557ff1	audioiirfilter: Don't crash if no filter coefficients are provided ...and by default use a identity filter. https://bugzilla.gnome.org/show_bug.cgi?id=710215	2013-10-31 22:43:49 +01:00
Wim Taymans	e96f8f519c	rtspsrc: proxy new buffer mode	2013-10-31 10:38:35 +01:00
Wim Taymans	43645d5981	jitterbuffer: add new timestamp mode Add a new timestamp mode that assumes the local and remote clock are synchronized. It takes the first timestamp as a base time and then uses the RTP timestamps for the output PTS.	2013-10-31 10:15:25 +01:00
Sebastian Dröge	4a8082856a	matroska-demux: Fix compiler warning matroska-demux.c: In function 'gst_matroska_demux_add_stream': matroska-demux.c:1379:7: error: format '%u' expects argument of type 'unsigned int', but argument 4 has type 'guint64' [-Werror=format=] "%03u", context->uid); ^	2013-10-30 22:13:06 +01:00
Matthieu Bouron	52d0588c21	videomixer: remove unneeded guint comparaison https://bugzilla.gnome.org/show_bug.cgi?id=711010	2013-10-29 16:38:26 +00:00
Matthieu Bouron	ec8c141d6a	y4menc: fix uninitialized variable warning https://bugzilla.gnome.org/show_bug.cgi?id=711011	2013-10-28 14:20:13 +00:00
Thiago Santos	2eec7909aa	qtdemux: check if the end_time is defined before using it Avoids sending EOS too soon because of overflow. Can happen on fragmented mp4 playback.	2013-10-25 11:30:36 -03:00
Thiago Santos	673301ef48	qtdemux: use correct unref function Events aren't GstObjects, but GstMiniObjects	2013-10-23 13:38:56 -03:00
Stefan Sauer	ae1150e85c	qtdemux: rename chunks_are_chunks to chunks_are_samples and flip the logic As the variable name suggests, sometimes chunks are chunks. Rename the variable to tell what they are when they are not chunks.	2013-10-15 09:53:30 +02:00
Stefan Sauer	6789ba1ece	qtdemux: fix typos and add more logging for unhandled parts	2013-10-15 09:53:30 +02:00
Ognyan Tonchev	c81ce6b152	multiudpsink: Fix memory leak Unmap all GstMemory of the current buffer when flushing. https://bugzilla.gnome.org/show_bug.cgi?id=710110	2013-10-14 18:21:54 +02:00
Tim-Philipp Müller	771ffe5609	flvmux: fix broken sample pipeline which was muxing raw audio and video into flvmux, which won't work, even if there were converters.	2013-10-12 20:44:31 +01:00
Tim-Philipp Müller	29effb522a	flvmux: require stream-format=raw for mpeg-2 too, but don't require framed field raw implies that it's framed already. Fixes .. ! faac ! flvmux	2013-10-12 20:37:41 +01:00
Sebastian Dröge	b8f9e966d5	wavenc: A-Law and Mu-Law don't have width/depth/signed caps fields https://bugzilla.gnome.org/show_bug.cgi?id=709614	2013-10-08 11:28:04 +02:00
Sebastian Dröge	a5bf9f24c9	deinterlace: Fix handling of planar video formats in greedyh method https://bugzilla.gnome.org/show_bug.cgi?id=709507	2013-10-07 12:54:11 +02:00
Reynaldo H. Verdejo Pinochet	38c5e5efdc	matroska: Trivial grammar fix on debug msg	2013-10-06 10:02:09 -07:00
Reynaldo H. Verdejo Pinochet	1cb31eeacc	matroskamux: Add context flag for WebM WebM has a couple of specific requirements we need to handle. Idea is to set this flag once and just rely on mux->is_webm at run time instead of repeatedly figuring this out from GST_MATROSKA_DOCTYPE_WEBM (which requires a strcmp()).	2013-10-06 09:54:28 -07:00
Reynaldo H. Verdejo Pinochet	edeed575ae	matroska: Do not write SegmentUID for WebM mux WebM spec states SegmentUID is Unsupported. Files produced with gstreamer without this change will spit an error like this when passed to mkvalidator: ERR201: Invalid 'SegmentUID' for profile 'webm' in Info at 192	2013-10-06 08:12:50 -07:00
Matej Knopp	cf12017ef8	matroskademux: make dvd palette change event sticky So they don't get lost. https://bugzilla.gnome.org/show_bug.cgi?id=709454	2013-10-05 10:55:03 +01:00
Nicolas Dufresne	ed77b22f2b	videoflip: Add automatic flip mode driven by image-orientation tag https://bugzilla.gnome.org/show_bug.cgi?id=709312	2013-10-04 14:52:57 -04:00
Wim Taymans	d4892859d4	jitterbuffer: fix race in flush-start/flush-stop When flush-stop arrives before we process the result of the _push() in the loop function, we might pause even though we are not flushing anymore. Fix this race by waiting for the srcpad loop function to completely pause after doing the flush-start.	2013-10-04 12:35:18 +02:00
Mathieu Duponchelle	ef548c2b28	videomixer: Update videoconvert copy https://bugzilla.gnome.org/show_bug.cgi?id=709390	2013-10-04 10:57:36 +02:00
Mathieu Duponchelle	3d780c5c6d	videomixer: Check if the pad needs reconfiguration in collected https://bugzilla.gnome.org/show_bug.cgi?id=709384	2013-10-04 10:53:26 +02:00
Sebastian Dröge	21947f9d13	qtdemux: Add support for the mp2v fourcc for MPEG-2 video https://bugzilla.gnome.org/show_bug.cgi?id=709270	2013-10-03 11:59:25 +02:00
Ognyan Tonchev	30f62a2eec	matroskademux: Fix memory leak https://bugzilla.gnome.org/show_bug.cgi?id=709266	2013-10-02 16:17:33 +02:00
Sreerenj Balachandran	e779b6587b	qtdemux: Add HEVC support https://bugzilla.gnome.org/show_bug.cgi?id=709093	2013-10-02 11:54:24 +02:00
Ognyan Tonchev	93d5e182d2	rtpgstpay: Fix memory leak We were leaking the GList nodes of the pending buffers. https://bugzilla.gnome.org/show_bug.cgi?id=709079	2013-10-02 11:07:16 +02:00
Wim Taymans	00056965e8	rtpjitterbuffer: fix race when updating the next_seqnum If we were not waiting for the missing seqnum when we insert the lost packet event in the jitterbuffer, we end up not updating the next_seqnum and wait forever for the lost packets to arrive. Instead, keep track of the amount of packets contained by the jitterbuffer item and update the next expected seqnum only after pushing the buffer/event. This makes sure we correctly handle GAPS in the sequence numbers.	2013-09-30 12:31:00 +02:00
Wim Taymans	fde438791e	rtpjitterbuffer: small debug improvement	2013-09-30 12:30:23 +02:00
Wim Taymans	6e7d547be4	rtpjitterbuffer: reset skew does not reset clock-rate Don't reset the clock-rate when we reset the skew correction algorithm. Reset the skew correction algorithm when we change the clock-rate.	2013-09-30 11:53:08 +02:00
Wim Taymans	03d520eb69	rtpjitterbuffer: pause timer when PAUSED Also pause the timer when we go to the PAUSED state. It is possible that we don't have a clock or base-time in PAUSED to perform the timeouts.	2013-09-30 11:16:32 +02:00
Wim Taymans	4a31aec513	rtpjitterbuffer: improve debug	2013-09-30 11:15:25 +02:00
Hans Månsson	041946423a	mp4mux: Do not require framerate in peer video caps Remove the framerate restriction on the caps. Reference: https://bugzilla.gnome.org/show_bug.cgi?id=708864	2013-09-28 13:02:11 +02:00
Wim Taymans	8c5ce0dbdc	rtspsrc: also go into the loop function after connect When we have opened the stream, go into the loop function so that we can receive messages from the server.	2013-09-27 15:08:31 +02:00
Matej Knopp	40c0586c17	matroskademux: move the check for subtitle buffer being null terminated before validating UTF-8 https://bugzilla.gnome.org/show_bug.cgi?id=707933	2013-09-27 14:38:19 +02:00
Wim Taymans	d4b4b4e924	rtpjitterbuffer: don't calculate skew without rtptime Skip trying to calculate the skew when we don't have an rtptime. It causes problems when lost packet events are placed in the jitterbuffer.	2013-09-26 16:21:33 +02:00
Wim Taymans	6095e2e859	rtspsrc: disable checks when linking pads We know the pad links will work (and we don't check the return value anyway).	2013-09-25 17:42:02 +02:00
Wim Taymans	2efd58fc84	rtpbin: avoid some pad link checks Link pads without checks, we know it will work.	2013-09-25 17:38:31 +02:00
Sebastian Dröge	4a91a93d4e	qtmux: Don't error out if downstream is not seekable for non-fragmented variants Doing so would be a regression over 1.0 and breaks the unit test. However the result will be most likely unusable, so let's post a warning message on the bus.	2013-09-25 13:25:34 +02:00
Wim Taymans	97f4674655	rtpjitterbuffer: calculate some stats	2013-09-25 10:50:05 +02:00
Wim Taymans	b1d29483bb	rtpjitterbuffer: move send_lost_event function Move the send_lost_event function to the do_lost_event handling, there is no need to have a separate function.	2013-09-25 10:50:05 +02:00
Thiago Santos	dc02d91c14	qtdemux: add code to parse creation time earlier than 1970 Use g_date_time seconds manipulation to allow to cover the quicktime spec for creation_time. It uses seconds since 1904. Both paths could be done using the generic approach of seconds since 1904 with GDateTime handling, but the first path using seconds from 1970 should be more commonly found and avoids a few objects creation and ref/unref, so keep it there for performance. Additionally, the code for handling seconds since 1970 changed from > to >= because having 0 seconds since 1970 is also a valid case for that path to handle. https://bugzilla.gnome.org/show_bug.cgi?id=707975	2013-09-24 15:16:54 -07:00
Matej Knopp	a1a493dae4	matroskademux: update stream->pos when sending buffers so that gap events are not sent unnecessarily https://bugzilla.gnome.org/show_bug.cgi?id=708505	2013-09-24 15:12:44 -07:00
Wim Taymans	adf5d96044	rtpmanager: update docs	2013-09-23 16:34:15 +02:00
Wim Taymans	e5019de80d	docs: update docs with 1.0 element names	2013-09-23 15:36:47 +02:00
Wim Taymans	8ce674da87	rtpjitterbuffer: always store lost event in jitterbuffer Always prepare a lost event in the jitterbuffer, it is to wake up and make the pushing thread continue. We drop the event when we are not supposed to push lost events downstream.	2013-09-23 14:45:27 +02:00
Wim Taymans	9f3345fcc2	rtpjitterbuffer: schedule lost event differently Schedule the lost event by placing it inside the jitterbuffer with the seqnum that was lost so that the pushing thread can interleave and push it properly.	2013-09-23 14:45:27 +02:00
Wim Taymans	ae389aeb0c	rtpjitterbuffer: remove list debug	2013-09-23 14:45:26 +02:00
Wim Taymans	28641e3145	rtpjitterbuffer: add type to the item So that the upper layer can know what data is contained in the item.	2013-09-23 14:45:26 +02:00
Wim Taymans	479c7642fd	rtpjitterbuffer: fix flush Pass function to flush to properly free the queue items.	2013-09-23 14:45:25 +02:00
Wim Taymans	0cc887eb98	rtpjitterbuffer: append seqnum -1 packets	2013-09-23 14:45:25 +02:00
Wim Taymans	39a2ba669d	rtpjitterbuffer: use structure to hold packet information Make the jitterbuffer operate on a structure containing all the packet information. This avoids mapping the buffer multiple times just to get the RTP information. It will also make it possible to store other miniobjects such as events later.	2013-09-23 14:45:25 +02:00
Wim Taymans	1760817005	rtpjitterbuffer: update expected timer when possible When we receive a packet and we have some missing packets, we can update their estimated arrival times based on the timestamp difference.	2013-09-23 14:45:25 +02:00
Wim Taymans	fdc1ed1680	rtpjitterbuffer: fix order of timeout events Improve the order of the timeout events, if there are timers with the same timeout, we want to trigger the lowest seqnum first. For this we need to loop over the complete array of timers to find the best one before triggering the timeout.	2013-09-23 14:45:25 +02:00
Wim Taymans	0b1a7edfea	rtpjitterbuffer: send lost event before signaling next buffer First send the lost event, then update the next_seqnum counter and then send the signal to the pushing thread that it can retry to push a buffer. This avoids pushing out buffers before the lost event is pushed.	2013-09-23 14:45:25 +02:00
Wim Taymans	5051f51f0a	jitterbuffer: configure clock-rate on jitterbuffer Add a get and setter to configure the clock-rate in the jitterbuffer instead of passing it as an argument to the insert method.	2013-09-23 14:45:24 +02:00
Wim Taymans	3c421e7e48	rtpjitterbuffer: add option to reset retransmission timers	2013-09-23 14:45:24 +02:00
Wim Taymans	6f4deab298	rtpjitterbuffer: stop the timer thread The timeout code could release the lock so we need to check if we are allowed to wait for the clock some more.	2013-09-23 14:45:24 +02:00
Wim Taymans	cba4e6a707	rtpjitterbuffer: unlock only once	2013-09-23 14:45:24 +02:00
Wim Taymans	5dc207948c	rtpjitterbuffer: improve flush and shutdown There is no need to unschedule the timer in flush-start, flush-stop will remove the timers and unschedule. Unschedule the current timer before attempting to join the timer thread.	2013-09-23 14:45:23 +02:00
Wim Taymans	a512cc2d3c	rtpjitterbuffer: set correct expected time When we already have a timer for a packet, skip it but don't forget to adjust the dts to the expected dts of the next packet.	2013-09-23 14:45:23 +02:00
Wim Taymans	517ea0f4d9	jitterbuffer: improve debug	2013-09-23 14:45:23 +02:00
Wim Taymans	c395bf62dd	alpha: use POFFSET instead of OFFSET Use the more correct POFFSET macro to get the offset of a component in its plane. The offset macro gives the offset of the component relative to the start of the frame.	2013-09-23 14:45:23 +02:00
Sebastian Dröge	94ad6724ba	goom: Fix MMX assembly compilation with clang clang does not want or need a clobber list for emms: error: clobbers must be last on the x87 stack Patch taken from the FreeBSD ports, provided by Dan McGregor <dan.mcgregor@usask.ca>	2013-09-21 18:48:19 +02:00
Sebastian Dröge	d8841b4832	matroska-demux: Make sure that subtitle buffers are \0-terminated https://bugzilla.gnome.org/show_bug.cgi?id=707933	2013-09-20 10:22:40 +02:00
Andoni Morales Alastruey	cfefdaebb6	qtmux: handle issues correctly when downstream is not seekable The streamable property only make sense for fragmented formats. For regular MP4, when downstream is not seekable we can't rewrite the headers, so qtmux can only work with fast-start=TRUE, where the headers are written finishing the file. For fragmented MP4, when streamable is not seekable and the streamable property is FALSE, we must enforce streamable=TRUE warning the user about this change https://bugzilla.gnome.org/show_bug.cgi?id=707242	2013-09-20 10:09:48 +02:00
Andoni Morales Alastruey	9ae5082204	qtmux: make "streamable" TRUE as default The most common use case for fragmented MP4 (Dash and Smooth Streaming) is producing streamable content (even for VOD). streamable=FALSE would only be used to generate fragmented MP4 with and index of MOOF's that could be reproduced without a playlist/manifest https://bugzilla.gnome.org/show_bug.cgi?id=707242	2013-09-20 10:09:48 +02:00
Andoni Morales Alastruey	5732684e18	qtmux: deprecate the streamable property for non-fragmented MP4 The streamable property only makes sense for fragmented MP4. https://bugzilla.gnome.org/show_bug.cgi?id=707242	2013-09-20 10:09:48 +02:00
Wim Taymans	926e2fa93b	alpha: don't assume planar formats have just 1 block Don't assume planar formats have just one memory block with the data but use the macros to access the right memory block where a component can be found.	2013-09-19 16:50:44 +02:00
Wim Taymans	fd6c57cf10	rtpjitterbuffer: keep delay as a separate variable in timer Keep a separate delay in the timer so that we still know the original timestamp of the packet that this timer refers to. We can then place the correct running-time in the Retransmission event.	2013-09-19 14:32:48 +02:00
Wim Taymans	d34184dd03	rtpjitterbuffer: fix writability of properties	2013-09-19 14:32:48 +02:00
Wim Taymans	6bb2626498	rtpjitterbuffer: reevaluate the current timer after timeout When we trigger the timeout logic of a timer, reevaluate it because it is possible that it still has the lowest timeout.	2013-09-18 16:33:40 +02:00
Wim Taymans	8d021b6ede	rtpjitterbuffer: don't update time when unscheduled Don't try to estimate the current time when we got unscheduled.	2013-09-18 16:31:26 +02:00
Wim Taymans	65606a25bf	rtpjitterbuffer: init packet spacing on first buffer Already init the packet spacing variables on the first buffer so that we can calculate the spacing on the second buffer already.	2013-09-18 16:29:37 +02:00
Wim Taymans	f2efdf28f5	rtpjitterbuffer: push the lost event from the timer thread Instead of pushing the lost event from the chain function, schedule a timeout that will push the lost event from the timer thread. This avoid blocking the upstream thread while we push and sync the event.	2013-09-18 14:57:09 +02:00
Wim Taymans	5d5fc03e04	rtpjitterbuffer: round gap duration to multiple of duration Make sure the gap duration in the lost event is a multiple of the packet duration. Enable another test.	2013-09-18 14:12:47 +02:00
Wim Taymans	6e4a051d40	rtpjitterbuffer: keep track of duration Keep track of the estimated duration of missing packets and use it in the lost event. Enable another unit test	2013-09-18 12:29:38 +02:00
Wim Taymans	ac3bb3acf6	rtpjitterbuffer: handle large gaps with one lost event When we have a large number of missing packets, generate one lost event for all the packets that have no chance of being pushed out in time. Fix and activate unit test for large gaps.	2013-09-18 11:59:28 +02:00
Wim Taymans	26402e1c21	rtpjitterbuffer: refactor lost event sending Also make sure we only increment the expected seqnum and last output timestamp.	2013-09-18 11:57:06 +02:00
Wim Taymans	f49981a597	jitterbuffer: refactor timeout triggers	2013-09-17 23:29:56 +02:00
Wim Taymans	047021c443	jitterbuffer: simplify the timeout code Keep track of the current time in the timeout loop. Loop over all timers and trigger all the expired ones, we can do this in the same loop that selects the new best timer.	2013-09-17 23:29:56 +02:00
Wim Taymans	fa1ef3328b	jitterbuffer: rearrange timer update code Also update the timers when retransmission is disabled. We need to do this because when we added LOST timers when we detected missing packets and we need to remove those timers when the packet finally arrives.	2013-09-17 23:29:56 +02:00
Tim-Philipp Müller	7a76595b22	videomixer: link to libm for maths stuff Fixes undefined references to rint and pow on ubuntu build bot.	2013-09-17 22:02:04 +01:00
Wim Taymans	232fdd8b56	jitterbuffer: release lock on shutdown	2013-09-17 15:19:42 +02:00
Matej Knopp	b2982bb749	qtmux: remove MAX_TOLERATED_LATENESS https://bugzilla.gnome.org/show_bug.cgi?id=707411	2013-09-16 11:11:12 -03:00
Wim Taymans	4de919a17a	jitterbuffer: use separate thread for timeouts Use a separate thread for scheduling the timeouts instead of using the downstream streaming thread that might block at any time.	2013-09-16 15:55:55 +02:00
Matej Knopp	b363832c2c	qtmux: set first_ts to DTS for streams that have DTS https://bugzilla.gnome.org/show_bug.cgi?id=707340	2013-09-16 12:14:00 +02:00
Matej Knopp	39f7e52266	qtmux: make sure duration is a valid number for last buffer https://bugzilla.gnome.org/show_bug.cgi?id=707340	2013-09-16 12:14:00 +02:00
Matej Knopp	4e3c13c87c	qtmux: use segment.start or last buffer end time in case of missing DTS https://bugzilla.gnome.org/show_bug.cgi?id=707340	2013-09-16 12:14:00 +02:00
Matej Knopp	85728c04c4	Revert qtmux: Use buffer PTS if DTS is not set" This reverts commit f72c3cf71fde622067f41f31a53978ba4c94469d. https://bugzilla.gnome.org/show_bug.cgi?id=707340	2013-09-16 12:13:54 +02:00
Sebastian Dröge	d646a34681	videomixer: Update orc generated files https://bugzilla.gnome.org/show_bug.cgi?id=708131	2013-09-16 11:03:06 +02:00
Olivier Crête	b9ceafe5af	rtpsession: Demux RTCP buffers from the RTP stream If there are RTCP buffers in the RTP stream, process them as RTCP. This way, we want receive streams following RFC 5761 https://bugzilla.gnome.org/show_bug.cgi?id=687657	2013-09-13 16:25:49 +02:00
Jan Schmidt	299d3f5c42	rtp: Remove bogus extra caps from L24 template. The extra caps entry in the template was making it sometimes get plugged for any dynamically allocated payload type.	2013-09-13 23:27:49 +10:00
Wim Taymans	28e5f90988	rtpbin: use PacketInfo for the sender Avoid mapping the packet multiple times when sending RTP.	2013-09-13 14:34:28 +02:00
Wim Taymans	a02c9473d8	rtpbin: store more in the PacketInfo Store all info in the PacketInfo so that we can avoid mapping the packet multiple times.	2013-09-13 14:34:28 +02:00
Wim Taymans	e5c789abd6	session: store more in the PacketInfo structure	2013-09-13 14:34:28 +02:00
Wim Taymans	47662f9ca4	rtpbin: RTPArrivalStats -> RTPPacketInfo Rename a structure because we are also going to use this for the sender bits.	2013-09-13 14:34:28 +02:00
Wim Taymans	c795b72988	source: small cleanups	2013-09-13 14:34:27 +02:00
Thiago Santos	566b0dce40	qtdemux: only update stop position if seek requests it Check for GST_SEEK_TYPE_NONE for stop poistion and only update the stop time if it is requested. Otherwise just maintain whatever was stored at the segment https://bugzilla.gnome.org/show_bug.cgi?id=707530	2013-09-13 09:21:12 -03:00
Rico Tzschichholz	8ed1ff6821	rtp: Add missing headers tp fix make dist In addition to `a956a6ceb2`	2013-09-13 14:06:13 +02:00
Sebastian Dröge	b95ddd55cd	flacparse: Make sure we have enough data to read image tags Thanks to iputinei for reporting this on IRC.	2013-09-12 15:39:51 +02:00
Wim Taymans	9f9ba21404	jitterbuffer: handle segments with non-0 start We keep the DTS and PTS in running-time inside the jitterbuffer. Make sure to transform it back to a buffer timestamp before pushing out the buffer. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707931	2013-09-12 15:04:30 +02:00
Seán de Búrca	9d3dbd6581	matroskademux: Fix off-by-one in validation of UTF-8 https://bugzilla.gnome.org/show_bug.cgi?id=707933	2013-09-12 09:19:15 +02:00
Thibault Saunier	9f4a8ccdf4	videomixer: Do not check if caps are empty when they are NULL In the case the caps are actually NULL, we should just concider it the same way as empty caps in that case.	2013-09-11 14:33:31 -03:00
Seán de Búrca	268058eb37	videomixer: fix build if orc is not installed https://bugzilla.gnome.org/show_bug.cgi?id=707886	2013-09-11 00:17:44 +01:00
Thiago Santos	193ce9110e	matroskademux: Preserve seqnum when pushing seek upstream After converting a seek from time to bytes, use the same seqnum on the event that goes upstream	2013-09-10 17:57:49 -03:00
Thiago Santos	be0eeae491	qtdemux: track streams that are EOS on push mode to finish earlier When the segment has a defined stop position, qtdemux should check when streams reach this position and mark those as EOS. When all streams are EOS it will return GST_FLOW_EOS to upstream to allow the pipeline to finish instead of continuously consume buffers from upstream that are not useful for the segment. https://bugzilla.gnome.org/show_bug.cgi?id=707530	2013-09-10 16:43:17 -03:00
Thiago Santos	33cf8b679d	qtdemux: preserve stop of segment when doing seeks in push mode When handling seeks in push mode, qtdemux converts the seek to bytes and pushes upstream. It needs to keep track of the seek and the subsequent segment to be able to map them back to the requested seek time and properly preserve the segment stop of the seek. This is done by using the start offset in bytes of the seek, that should be the same of the segment from upstream. And this is also backwards compatible with what qtdemux already was using. https://bugzilla.gnome.org/show_bug.cgi?id=707530	2013-09-10 16:42:36 -03:00
Mathieu Duponchelle	8db40a8c7f	videomixer: Add colorspace conversion https://bugzilla.gnome.org/show_bug.cgi?id=704950	2013-09-10 10:37:23 +02:00
Mathieu Duponchelle	707e39fe7a	videomixer: Don't send reconfigure event when formats or PAR are different It is racy with multiple pads. https://bugzilla.gnome.org/show_bug.cgi?id=704950	2013-09-10 10:36:48 +02:00
Mathieu Duponchelle	8db3648544	videomixer: Bundle private copies of videoconvert code Ideally, this would be part of libgstvideo. Prefixes videoconvert symbols with videomixer_. https://bugzilla.gnome.org/show_bug.cgi?id=704950	2013-09-10 10:36:30 +02:00
Wim Taymans	9f9bcbc405	rtspsrc: only wait if we flushed Only wait for the STREAM_LOCK when we flushed something when sending a command for PAUSED or PLAYING. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707611	2013-09-09 15:13:46 +02:00
Wim Taymans	7b2e002879	rtspsrc: return when a flush was issued Make gst_rtspsrc_loop_send_cmd() return TRUE when the current action has been flushed	2013-09-09 15:13:46 +02:00
David Holroyd	a956a6ceb2	rtp: add L24 pay and depayloader Fixes https://bugzilla.gnome.org/show_bug.cgi?id=707734	2013-09-09 15:13:46 +02:00
Matej Knopp	a5ceab82dd	matroskademux: fix leaking buffer and caps https://bugzilla.gnome.org/show_bug.cgi?id=707688	2013-09-07 15:50:36 +01:00
Tim-Philipp Müller	60e72b0254	udpsrc: fix build on win32 gstudpsrc.c:855:15: error: #if with no expression	2013-09-05 19:46:37 +01:00
Wim Taymans	5d2ff288b3	avidemux: handle unseekable streams Handle streams that we can't seek in and ignore them in the seek logic.	2013-09-04 15:53:05 +02:00
Wim Taymans	6f0e8a8b87	avidemux: only check video compression for video streams Or else we might deref a stream with a NULL strf.vids and segfault	2013-09-04 15:53:05 +02:00
Alex Ashley	a965185dee	qtdemux: Add support for the avc3 sample entry format of the AVC file format Amendment 2 of ISO/IEC 14496-15 (AVC file format) is defining a new structure for fragmented MP4 called "avc3". The principal difference between AVC1 and AVC3 is the location of the codec initialisation data (e.g. SPS, PPS). In AVC1 this data is placed in the initial MOOV box (moov.trak.mdia.minf.stbl.stsd.avc1) but in AVC3 this data goes in the first sample of every fragment (i.e. the first sample in each mdat box). The principal reason for avc3 is to make it easier for client implementations, because it removes the requirement to insert the SPS+PPS in to the decoder pipeline every time there is a representation change. This commit adds support for the "avc3" atom, which is almost identical to the "avc1" atom, except it does not contain any SPS or PPS data. https://bugzilla.gnome.org/show_bug.cgi?id=702004	2013-09-04 13:33:22 +02:00
Mathieu Duponchelle	b68f419b6f	videomixer: Don't set EOS to FALSE when the collectpad is EOS https://bugzilla.gnome.org/show_bug.cgi?id=707238	2013-09-04 11:09:04 +02:00
Matej Knopp	349afc633a	flacparse: cleanup on error after state change https://bugzilla.gnome.org/show_bug.cgi?id=707229	2013-09-03 18:06:18 +02:00
Sebastian Dröge	7f59436979	udpsrc: Bind to multicast addresses on non-Windows systems On Windows it's not possible to bind to a multicast address but the OS will make sure to filter out all packets that arrive not for the multicast address the socket joined. On Linux and others it is necessary to bind to a multicast address to let the OS filter out all packets that are received on the same port but for different addresses than the multicast address And deprecate the multicast-group property and replace it with the address property. https://bugzilla.gnome.org/show_bug.cgi?id=707042	2013-09-03 11:23:24 +02:00
Matej Knopp	73751dbbe7	flacparse: Free GstBaseParseFrame if pushing a header failed	2013-09-03 10:10:49 +02:00
Sebastian Dröge	edf6d28765	udpsrc: Refactor address resolval into its own function	2013-09-03 10:10:49 +02:00
Tim-Philipp Müller	966f848edb	replaygain: fix taglist leak in rganalysis And add some FIXMEs.	2013-09-02 23:00:29 +01:00
Sebastian Dröge	1971c43279	flacparse: Properly propagate downstream flow returns upstream https://bugzilla.gnome.org/show_bug.cgi?id=707229	2013-09-02 11:56:33 +02:00
Tim-Philipp Müller	1dfc1f2686	Don't use setlocale in plugins() Only apps should call setlocale(), not libraries.	2013-09-01 21:18:38 +01:00
Wim Taymans	d851b8a8b4	rtpmpvpay: Fix RTP buffer allocation in rtpmpvpay RTP buffer allocation should not be done with padding for the specific MPEG2 header as the padding is done at the end of the buffer and the last byte is the size of the padding. https://bugzilla.gnome.org/show_bug.cgi?id=706970	2013-08-29 13:15:15 +02:00
Bernhard Miller	f7528d274b	autovideosink: add sync property https://bugzilla.gnome.org/show_bug.cgi?id=706955	2013-08-29 12:23:24 +02:00
Bernhard Miller	2fa68fce07	autoaudiosink: introduce sync property https://bugzilla.gnome.org/show_bug.cgi?id=706955	2013-08-29 12:23:23 +02:00
Thiago Santos	9549289a18	qtdemux: push buffers after segment stop until reaching a keyframe This should make decoders able to precisely push buffers until the stop time in case they need the next keyframe to do it. Also, according to gst_segment_clip, it should only push a buffer that the starting ts is strictly smaller than the segment stop, so we change the min < comparison for <=	2013-08-28 12:58:56 -03:00
Sebastian Dröge	76293efd72	Release 1.1.4	2013-08-28 12:52:25 +02:00
Wim Taymans	2a8566ddec	matroska-mux: remove framerate restriction Remove the framerate restriction on the caps.	2013-08-27 15:25:16 +02:00
Wim Taymans	f1106cde66	session: only update next check time when reconsidering Don't update the next RTCP check time in all cases but only when we reconsidered. This avoids delaying sending a full RTCP packet when we are doing early feedback.	2013-08-27 09:55:52 +02:00
Wim Taymans	47065db0b6	session: add more debug	2013-08-27 09:55:52 +02:00
Wim Taymans	454d75951e	jitterbuffer: fix types of the retransmission event	2013-08-27 09:55:52 +02:00
Wim Taymans	dd4af0d11c	jitterbuffer: only timeout EXPECTED timers on gap Only timeout the EXPECTED timers when we detect a large seqnum gap.	2013-08-27 09:44:18 +02:00
Wim Taymans	4b7bcc2ec1	rtsession: fix locking We need to take the session lock when getting and manipulating the source.	2013-08-26 11:50:27 +02:00
Wim Taymans	3f46527f75	rtpsession: add some more debug	2013-08-26 11:50:13 +02:00
Mathieu Duponchelle	5d21f8f2e3	videomixer: don't send flush_stop twice. If we get flush start and a seek we need to only send flush_stop once. More info at #706441	2013-08-23 20:17:11 -04:00
Tim-Philipp Müller	9b0bcc01a0	multipartdemux: propagate discont	2013-08-23 15:57:46 +01:00
Tim-Philipp Müller	c3af414cbf	multipartdemux: remove dynamic sourcpads when going from PAUSED to READY	2013-08-23 15:57:46 +01:00
Tim-Philipp Müller	7d78a68c8d	multipartdemux: timestamp output buffers based on first input buffer that provided bytes not last https://bugzilla.gnome.org/show_bug.cgi?id=637754	2013-08-23 15:57:46 +01:00
Wim Taymans	54e7e7547a	rtxqueue: add property to configure queue size	2013-08-23 15:47:25 +02:00
Wim Taymans	84833bed11	rtpbin: proxy jitterbuffer do-retransmission property	2013-08-23 12:10:19 +02:00
Michael Olbrich	23d4044e2c	avimux: unmap the correct buffer The audio buffer was mapped so unmap it and not the video buffer https://bugzilla.gnome.org/show_bug.cgi?id=706642	2013-08-23 11:32:52 +02:00
Wim Taymans	89b9019e3e	rtx: various improvements Use locking Don't push from the event handler, collected packets in a queue and push from the chain function. Clear queues on shutdown.	2013-08-21 17:02:27 +02:00
Wim Taymans	ee15bc9284	session: generate events correctly Do correct shifting of the bitmask for lost packets.	2013-08-21 17:02:27 +02:00
Wim Taymans	67523d3ecb	rtp: register rtx element better	2013-08-21 17:02:26 +02:00
Wim Taymans	f626e29897	jpegdepay: add some more debug	2013-08-21 12:56:35 +02:00
Wim Taymans	77ed44a88a	rtpgstdepay: only push events when they changed Keep track of the STREAM_START and TAG events and only push them when they changed.	2013-08-21 12:10:00 +02:00
Wim Taymans	b144809b7c	rtpgstpay: taglists should not be merged in 1.0	2013-08-21 10:52:59 +02:00
Wim Taymans	69b0dcd7df	rtpgstdepay: flush on FLUSH_STOP event	2013-08-21 10:28:50 +02:00
Wim Taymans	5ff9093843	rtpgstpay: reset on state change Do full reset on state change to READY	2013-08-21 10:03:52 +02:00
Wim Taymans	ae9239aac7	rtpgstpay: reset on FLUSH_STOP Clear the adapter and pending buffer list on FLUSH_STOP.	2013-08-21 09:55:20 +02:00
Wim Taymans	2e8955df39	rtpgstpay: don't use clock for config interval We can't use the clock to time our config-interval because we are not live (or there might not be a clock or the clock might not be running). Instead just simply take the timestamp diff.	2013-08-21 09:39:30 +02:00
Wim Taymans	182f96ff79	rtpgstay: don't use // comments	2013-08-21 09:33:04 +02:00
Youness Alaoui	e22f7e91c4	rtspsrc: Fix response argument in handle-request signal	2013-08-21 09:06:02 +02:00
Youness Alaoui	6636efd31a	rtspsrc: Add sdes property and proxy it to rtpbin	2013-08-21 09:06:02 +02:00
Youness Alaoui	62a6f58697	Send a stream-start whenever we send tags This is to make sure tags are cleared on the client if the stream-start was previously lost, otherwise, the client may end up with a merged taglist of multiple songs	2013-08-21 09:06:01 +02:00
Youness Alaoui	05bcfee5a3	rtpgstpay: Add a config-interval property to resend the caps/tags at a regular interval This is useful in case the packet containing the inlined caps was lost or if new client joins an already running RTP stream and they missed the previous tag events. This also makes the payloader keep a list of merged tags so the retransmitted tag event contains all previously received. A STREAM_START event will flush the list of tags.	2013-08-21 09:06:01 +02:00
Youness Alaoui	1f4ca28868	rtpgstpay: Refactor the setcaps and use new method to send arbitrary caps at any time	2013-08-21 09:06:01 +02:00
Youness Alaoui	9257409613	rtpgstpay: Do not flush events for stream-start and avoid conflict between event and pending inline caps	2013-08-21 09:06:01 +02:00
Youness Alaoui	2d53289b6b	rtpgstpay: Add a create_from_adapter API and use a list of GstBufferList This is necessary to fix event/caps sending. If we send a STREAM_START packet, it will cause an error because the stream didn't receive its caps and new-segment events, so we must wait for the first buffer before sending the stream-start event buffer. However, the caps will be sent at the same time and so the 'inline caps' will be set for the event. We need to be able to payload individual packets (data, caps or events) and only send them when we call flush.	2013-08-21 09:06:01 +02:00
Youness Alaoui	0070ba76f2	rtpgstpay: Add etype=4 for payloading GST_EVENT_STREAM_START	2013-08-21 09:06:01 +02:00
Youness Alaoui	6155b27971	rtpgstpay: Fix typo, GST_EVENT_CUSTOM_BOTH has etype of 3	2013-08-21 09:06:01 +02:00
Wim Taymans	587dc055e9	jitterbuffer: handle EOS When the queue is empty, and we received EOS, pause and push an EOS event downstream. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706387	2013-08-20 14:36:59 +02:00
Wim Taymans	533f26fc99	jitterbuffer: update docs	2013-08-20 10:26:15 +02:00
Wim Taymans	c7f9ef8012	jitterbuffer: update all timers Keep looping over all registered timers so that we can mark them lost instead of stopping as soon as we find the timer for the current seqnum.	2013-08-20 10:25:17 +02:00
Wim Taymans	5debda9ca1	jitterbuffer: remove unused variables	2013-08-20 08:55:50 +02:00
Wim Taymans	a88db5fa2c	jitterbuffer: reorganize timer handling Restructure handling of incomming packet and the gap with the expected seqnum and register all timers from the _chain function. Convert a timer to a LOST packet timer when the max amount of retransmission requests has been reached.	2013-08-19 22:04:51 +02:00
Wim Taymans	d9d6eac4bb	jitterbuffer: refactor packet spacing calculation	2013-08-19 22:04:50 +02:00
Wim Taymans	c4dc159656	jitterbuffer: keep track of last seqnum and dts	2013-08-19 22:04:50 +02:00
Wim Taymans	652ce95ca6	jitterbuffer: small cleanups	2013-08-19 22:04:50 +02:00
Wim Taymans	b4a35bbe82	jitterbuffer: reset retransmission timers in add/reschedule Reset the retransmission timers when adding and rescheduling a timer.	2013-08-19 22:04:50 +02:00
Wim Taymans	cf8a0652f3	jitterbuffer: rename variables for packet spacing	2013-08-19 22:04:50 +02:00
Wim Taymans	ec82e4ab7c	jitterbuffer: remove lost timer when we get the packet When we receive a packet, also remove the LOST timer for it.	2013-08-19 22:04:50 +02:00
Wim Taymans	2f03b43b21	jitterbuffer: expected seqnum must increase Only update the expected seqnum when it is bigger than the previous expected seqnum.	2013-08-19 22:04:50 +02:00
Wim Taymans	c5bf376bb5	jitterbuffer: add more debug	2013-08-19 22:04:50 +02:00
Wim Taymans	ff825a2919	rtxqueue: add retransmission queue element	2013-08-19 22:04:50 +02:00
Wim Taymans	5fe18ee432	session: add some docs	2013-08-19 22:04:49 +02:00
Wim Taymans	482dacfb54	session: handle NACK feedback and generate events Handle and parse the feedback NACK packets and generate a Retransmission event for each NACKed packet	2013-08-19 22:04:49 +02:00
Thibault Saunier	e47ffb203b	videomixer: Do not send flush_stop ourself after a flush_start When we receive a flush_start, we should wait for the next flush_stop and foward it, not create a flush_stop ourself.	2013-08-17 11:40:27 +02:00
Wim Taymans	db90f6e68d	h264depay: init debug category early Init the debug variable when we register the element because it is also used by the payloader element when it calls the add_sps_pps method.	2013-08-16 17:12:19 +02:00
Chris Bass	3e9dea3f8c	qtdemux: check denominator isn't zero before scaling duration. When gst_qtdemux_configure_stream sets fps_d, check that n_samples is non-zero before using it as a denominator to scale the stream duration. https://bugzilla.gnome.org/show_bug.cgi?id=706076	2013-08-16 10:14:30 +02:00
Wim Taymans	f11c2c9b3b	jitterbuffer: forward flush before stopping dataflow First forward the flush event and then stop our loop function.	2013-08-14 16:19:32 +02:00
Olivier Crête	4c6e636720	rtph264pay: Use the SPS/PPS handling function from the depayloader Remove duplicated copies https://bugzilla.gnome.org/show_bug.cgi?id=705553	2013-08-13 10:38:23 -04:00
Olivier Crête	742b90747d	rtph264depay: Make the SPS/PPS deduplication function generic Make it not touch any internals of the depayloader https://bugzilla.gnome.org/show_bug.cgi?id=705553	2013-08-13 10:38:23 -04:00
Chris Bass	b40bf67526	aacparse: allow conversion from raw AAC to ADTS This patch will prepend ADTS headers to raw AAC audio frames, allowing upstream elements to link to decoders that only support AAC in ADTS format. Note that no error correction bits are added to ADTS frames in this code. https://bugzilla.gnome.org/show_bug.cgi?id=615740	2013-08-13 15:58:23 +02:00
Sebastian Dröge	282afae244	rtspsrc: Only free GCheckSum after its last usage https://bugzilla.gnome.org/show_bug.cgi?id=705760	2013-08-13 12:44:11 +02:00
Matej Knopp	2269ac8f28	qtdemux: elst should offset samples instead of buffers The current approach where buffers are offset is not ideal, as during seek and loop current time is compared to sample times. https://bugzilla.gnome.org/show_bug.cgi?id=700264	2013-08-12 13:48:04 +02:00
Thibault Saunier	6c349d6ec3	videomixer: Send EOS if buf_end >= segment.stop That means the whole segment is already played, and we are sure we are EOS at that point. Also handle segment seeks, and do not send EOS in that case.	2013-08-11 19:05:18 +02:00
Matej Knopp	96afba915a	avidemux: send proper stream_start event https://bugzilla.gnome.org//show_bug.cgi?id=705449	2013-08-08 11:57:32 +02:00
Sebastian Dröge	9863e08839	matroskademux: Don't print warnings during flushing and stop as soon as possible https://bugzilla.gnome.org//show_bug.cgi?id=705442	2013-08-08 11:53:15 +02:00
Tim-Philipp Müller	957c8e3e61	rtpvp8depay: mark key frames and delta frames properly https://bugzilla.gnome.org/show_bug.cgi?id=705550	2013-08-07 11:14:38 +01:00
Wim Taymans	48174164eb	session: add NACK feedback in RTCP	2013-08-06 15:50:19 +02:00
Wim Taymans	4379ed1dee	source: add methods to register NACK Add a method to register a missing packet for an ssrc along with methods to get the missing packets and clear them.	2013-08-06 15:50:19 +02:00
Wim Taymans	50638b8106	session: handle Retransmission event and schedule NACK Handle the retransmission event from downstream and use it to schedule a NACK request.	2013-08-06 15:50:19 +02:00
Wim Taymans	0bddbd682d	session: pass data to remove func Pass the data to the remove function because we are going to deref it when there is pli or fir.	2013-08-06 15:50:19 +02:00
Thibault Saunier	38946bd9f4	qtdemux: Fix compilation	2013-08-06 15:31:38 +02:00
Thibault Saunier	593a31f2b4	qtdemux: Raw buffer DTS should always be CLOCK_TIME_NONE	2013-08-06 15:17:44 +02:00
Thibault Saunier	c5fa4666b7	videomixer: Make sure to send EOS if the buffer end time equals the segment end time Otherwize EOS never gets sent in that particular case.	2013-08-06 12:21:33 +02:00
Sjoerd Simons	d14d4c436c	goom: Ensure src caps are writable In some cases the src caps determined by goom weren't writable, causing a bunch of assertion failures and failed caps. Fixed by always explicitely making the caps writable https://bugzilla.gnome.org/show_bug.cgi?id=705475	2013-08-05 15:33:39 +02:00
Wim Taymans	3c82de59f9	session: use common send_rtcp method Reuse the send_rtcp method that already asks for the current time when requesting a keyframe.	2013-08-05 15:02:59 +02:00
Wim Taymans	3c14c6021c	session: Don't use ClockTimeDiff for unsigned delays	2013-08-05 15:02:59 +02:00
Edward Hervey	4f4f6432cc	qtmux: Use buffer PTS if DTS is not set Avoids ending up with completely bogus scaled duration/pts when new buffers have invalid DTS.	2013-08-04 17:15:38 +02:00
Tim-Philipp Müller	7272dec5fe	rtpdec: use generic marshaller	2013-08-04 11:20:41 +01:00
Tim-Philipp Müller	fe098e3aff	udp: remove unused marshal and enumtypes files	2013-08-04 11:03:07 +01:00
Tim-Philipp Müller	7469cd3a4c	rtpmanager: use generic marshaller	2013-08-04 11:03:07 +01:00
Wim Taymans	7584f91f31	jitterbuffer: send event in right direction	2013-08-04 00:24:36 +02:00
Wim Taymans	9613e481ad	session: add FIR and PLI like other RTCP packets Add the FIR and PLI packets like the other RTCP packet instead of from the on-sending-rtcp default signal handler.	2013-08-03 00:33:24 +02:00
Wim Taymans	743e1b1191	jitterbuffer: fix property ranges	2013-08-02 17:22:55 +02:00
Wim Taymans	cd0164f4cc	jitterbuffer: push retransmission events	2013-08-02 16:43:59 +02:00
Wim Taymans	9a13267e85	jitterbuffer: add support for retransmission retry When we didn't receive a packet after requesting retransmission, retry asking for retransmission for a certain period.	2013-08-02 14:54:56 +02:00
Wim Taymans	e9ad5126db	jitterbuffer: add properties Add properties to control retransmission parameters	2013-08-02 14:47:56 +02:00
Wim Taymans	a8c7ff7489	jitterbuffer: use corrected timeout when rescheduling When we recalculate the timeout, use the corrected timeout value depending on the timer type.	2013-08-02 12:44:58 +02:00
Wim Taymans	9c7e3e3455	jitterbuffer: update timers after queueing Else we might update the timer needlessly for duplicates.	2013-08-02 12:43:00 +02:00
Wim Taymans	ebd6b8f8ab	jitterbuffer: move method up	2013-08-02 12:42:08 +02:00
Wim Taymans	f6b6797874	jitterbuffer: small cleanup	2013-08-02 06:28:32 +02:00
Wim Taymans	0e41414926	jitterbuffer: unschedule old expected packets When we receive a new packet, unschedule old outstanding packets when their seqnum is too far away.	2013-08-01 23:36:07 +02:00
Wim Taymans	70695466ed	jitterbuffer: refactor timer update	2013-08-01 23:32:00 +02:00
Wim Taymans	4ab3f5d3da	jitterbuffer: update timers when removing Update the timers when we remove a timer. Handle canceled timers, make them unschedule the current timer and trigger the timeout code.	2013-08-01 23:24:29 +02:00
Wim Taymans	b983cf675b	jitterbuffer: fix typo	2013-08-01 23:22:02 +02:00
Wim Taymans	f3c658cbe6	jitterbuffer: improve timeout management If we change the seqnum of an existing timer and we were waiting for that timer, unschedule it. If we change the timeout of an existing timer and we were waiting on it, only unschedule when the new time is smaller.	2013-08-01 15:40:52 +02:00
Wim Taymans	77e5d320ab	jitterbuffer: install timer for expected arrival Install a timer that is triggered when the expected arrival time of a packet expired.	2013-08-01 15:11:13 +02:00
Wim Taymans	f08d98404e	jitterbuffer: improve unschedule of timers Conflicts: gst/rtpmanager/gstrtpjitterbuffer.c	2013-08-01 14:57:11 +02:00
Wim Taymans	9d3b824e2a	jitterbuffer: move code around	2013-08-01 12:21:53 +02:00
Wim Taymans	fe32e80c92	jitterbuffer: estimate inter packet spacing When we see two packets with consecutive seqnums and a different RTP time, use the DTS difference as the inter packet spacing estimate.	2013-08-01 12:07:11 +02:00
Wim Taymans	255b7106f5	jitterbuffer: keep track of current timeout	2013-08-01 12:01:15 +02:00
Wim Taymans	7e43dba19b	jitterbuffer: cleanup timer handling	2013-08-01 11:49:10 +02:00
Wim Taymans	9d88ac9cbb	jitterbuffer: reset is only possible with a GAP	2013-08-01 11:40:41 +02:00
Wim Taymans	f864131227	jitterbuffer: operate on DTS Make the jitterbuffer schedule the timeouts based on the DTS instead of the PTS. This makes it all smoother with reordered frames and gives the decoder time to reorder the frames in time.	2013-08-01 11:36:56 +02:00
Wim Taymans	80c5934290	jitterbuffer: rename timout variable	2013-08-01 11:14:12 +02:00
Wim Taymans	aa951433ee	jitterbuffer: small cleanup	2013-07-31 17:08:58 +02:00
Wim Taymans	69c78f72d5	jitterbuffer: block output in paused or buffering	2013-07-31 16:59:58 +02:00
Wim Taymans	4fbbc53a49	jitterbuffer: store pts in timer Only store the pts in the timer so that we can both do timeouts with timings on the input and output of the jitterbuffer.	2013-07-31 16:59:09 +02:00
Wim Taymans	77846d35c6	rtpjitterbuffer: refactor jitterbuffer Refactor the jitterbuffer code. Make separate function for peeking a buffer, pushing the next buffer, waiting for timeouts and handling the timeouts. The main loop now tries to push as many buffers as it can until it runs out of buffers or when it detects a seqnum discont. Then it will wait for some event to happen before attempting to push more buffers. Make methods to register timeouts in an array. These timeouts are registered when we detect a missing packet, sync for the first packet or when we find an estimation for the end-of-stream. This greatly simplifies and clarifies the code and also makes it possible to register more complicated timeout schemes later.	2013-07-30 23:24:23 +02:00
Wim Taymans	ea931d4f57	rtpjitterbuffer: use NULL to ignore percent If we pass NULL to pop and push we ignore the percent result.	2013-07-30 23:24:23 +02:00
Wim Taymans	b3e8a85a54	jitterbuffer: refactor Move eos estimation into separate function	2013-07-30 23:24:22 +02:00
Tim-Philipp Müller	a5532b4510	flvdemux: don't leak stream_id string https://bugzilla.gnome.org/show_bug.cgi?id=705142	2013-07-30 14:28:19 +01:00
Sebastian Dröge	2e35b36aab	gst: Don't swap start/stop for negative rates in the SEGMENT query	2013-07-29 12:12:41 +02:00
Matej Knopp	47ed79fb1c	qtdemux: Check for data size when parsing h264 codec data from strf atom	2013-07-29 11:53:07 +02:00
Sebastian Dröge	722ef42196	matroskademux: Implement SEGMENT query	2013-07-29 10:53:54 +02:00
Sebastian Dröge	d135373beb	flvdemux: Implement SEGMENT query	2013-07-29 10:53:47 +02:00
Sebastian Dröge	4e78974c87	avidemux: Implement SEGMENT query	2013-07-29 10:50:59 +02:00
Matej Knopp	2dcdfe07f7	qtdemux: Support H264 fourcc https://bugzilla.gnome.org/show_bug.cgi?id=704996	2013-07-29 09:11:39 +02:00
Sebastian Dröge	1fbb6d30a6	avidemux: Fix duration reporting in push mode https://bugzilla.gnome.org/show_bug.cgi?id=700933	2013-07-28 17:38:56 +02:00
Sebastian Dröge	89a3dc2ecd	avidemux: Don't forget unmapping and unreffing buffer	2013-07-28 17:32:59 +02:00
Matej Knopp	1947587784	avidemux: unmap buffer https://bugzilla.gnome.org/show_bug.cgi?id=704951	2013-07-28 17:32:59 +02:00
Wim Taymans	02359f9219	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 22:31:41 +02:00
Wim Taymans	0261199fc4	session: ignore RTCP for inactive sources	2013-07-26 22:31:23 +02:00
Wim Taymans	a4b4ca53c0	session: small cleanup	2013-07-26 22:25:17 +02:00
Wim Taymans	e0abd2e9b5	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 17:29:10 +02:00
Wim Taymans	6cce6fb04c	session: create SSRC before doing session cleanup Make the internal source before we do session cleanup	2013-07-26 17:29:10 +02:00
Wim Taymans	5b0298c63e	session: reorganize the report block code	2013-07-26 17:29:10 +02:00
Matej Knopp	7335b81c47	matroskademux: fix memory leak in check_subtitle_buffer https://bugzilla.gnome.org/show_bug.cgi?id=704921	2013-07-26 17:11:31 +02:00
Wim Taymans	3c44cd7c83	session: refactor active and sender checks	2013-07-26 14:21:40 +02:00
Wim Taymans	e952f7ba43	session: remove internal sources on timeout When an internal source times out and becomes a receiver, remove it.	2013-07-26 12:18:01 +02:00
Wim Taymans	e9e2fe3950	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 12:18:01 +02:00
Wim Taymans	bd0709c15c	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 12:18:01 +02:00
Wim Taymans	88f5a5f355	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 12:18:01 +02:00
Wim Taymans	abc90da1dc	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 12:18:01 +02:00
Wim Taymans	e0a1ce1291	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 12:18:01 +02:00
Wim Taymans	495d43c089	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 12:18:00 +02:00
Wim Taymans	9505fd4150	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 12:17:59 +02:00
Wim Taymans	3d6ee1fb5e	session: internal sources don't use probation	2013-07-26 12:17:59 +02:00
Wim Taymans	0e53e9109e	session: give caps to session Let the session parse the caps and update its SSRC when needed.	2013-07-26 12:17:59 +02:00
Wim Taymans	c06482a2cb	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 12:17:59 +02:00
Wim Taymans	33ce50e8b1	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 12:17:59 +02:00
Wim Taymans	5652f02b76	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 12:17:59 +02:00
Wim Taymans	7f83927c95	session: count internal sources and how many are senders	2013-07-26 12:17:58 +02:00
Wim Taymans	719343c206	rtpsession: separate BYE marking and scheduling First mark sources with BYE and then schedule the BYE RTCP message.	2013-07-26 12:17:58 +02:00
Wim Taymans	391943ba82	session: get SSRC from RTCP packet itself Get the SSRC from the RTCP packet instead.	2013-07-26 12:17:57 +02:00
Wim Taymans	a3f75a17ef	session: fix bandwidth calculation We iterate over all sources and the internal one is also in the hashtable so avoid adding it twice.	2013-07-26 12:17:57 +02:00
Wim Taymans	9eaef9d332	session: add some docs	2013-07-26 12:17:56 +02:00
Wim Taymans	2163355a47	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.	2013-07-26 12:17:56 +02:00
Wim Taymans	a3bf374351	session: move check for is_early around Move the check for the early RTCP to where it is needed and used.	2013-07-26 12:17:56 +02:00
Wim Taymans	b069db6a2e	session: parse packet outside of the session lock	2013-07-26 12:17:56 +02:00
Wim Taymans	57c27ec319	session: do nicer checks for internal sources	2013-07-26 12:17:56 +02:00
Wim Taymans	93d07298ff	session: let source keep track if it sent BYE	2013-07-26 12:17:56 +02:00
Wim Taymans	0c9c1434a8	source: reset more	2013-07-26 12:17:56 +02:00
Wim Taymans	1d02496d15	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.	2013-07-26 12:17:56 +02:00
Wim Taymans	ddd071e54c	session: configure sdes with structure only Remove code to configure the SDES with methods and types, only allow configuration with GstStructure	2013-07-26 12:17:55 +02:00
Wim Taymans	0060e1d45d	session: refactor add and find source Make functions to find and add a source to the hashtable.	2013-07-26 12:17:55 +02:00
Wim Taymans	adb0d68c07	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.	2013-07-26 12:17:55 +02:00
Wim Taymans	bf7d8173b3	jitterbuffer: add some more debug	2013-07-26 12:17:55 +02:00
Vincent Penquerc'h	91d4abceaa	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).	2013-07-26 09:44:11 +01:00
Vincent Penquerc'h	55e9338846	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.	2013-07-26 09:44:10 +01:00
Thiago Santos	7eac4c7c03	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 15:24:31 -03:00
Thiago Santos	8bd12e12b3	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 15:24:31 -03:00
Thiago Santos	e49b6e7c35	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	2013-07-25 15:24:31 -03:00
Wim Taymans	c44a29bd53	bin: fix compilation	2013-07-24 14:17:45 +02:00
Wim Taymans	cc92ef1db2	vrawdepay: fix UYVP format	2013-07-24 12:42:31 +02:00
Wim Taymans	8191b6fcd2	vrawpay: fix UYVP format	2013-07-24 12:41:58 +02:00
Wim Taymans	37af93c361	vrawpay: fix caps	2013-07-24 12:41:44 +02:00
Wim Taymans	f87875e35b	rtpjitterbuffer: fix locking Take the lock earlier so that we do things that follow with the right locking.	2013-07-24 10:49:03 +02:00
Wim Taymans	dece8413ef	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 17:41:48 +02:00
Wim Taymans	25e0f0d6b6	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 17:41:48 +02:00
Wim Taymans	c337265ee4	session: reset some RTCP variables The early_send time was set to 0 and always triggering an early RTCP packet.	2013-07-23 17:41:48 +02:00
Edward Hervey	3d48d25756	qtdemux: Add all the mpeg XDCAM variants This should cover all known XDCAM variants (which are all mpeg2 video) Fixes #672227	2013-07-23 15:03:31 +02:00
Carlos Rafael Giani	95429f1d4b	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>	2013-07-23 06:25:20 +02:00
Tim-Philipp Müller	f18b1f7e80	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 18:00:16 +01:00
Sebastian Dröge	0c2ff91a5c	wavparse: Add support for group-id in the stream-start event	2013-07-22 15:30:13 +02:00
Sebastian Dröge	169b490664	rtspsrc: Add support for group-id in the stream-start event	2013-07-22 15:30:13 +02:00
Sebastian Dröge	5a9f4a3cbc	rtpsession: Add support for group-id in the stream-start event	2013-07-22 15:30:13 +02:00
Sebastian Dröge	57dd1189d5	matroskademux: Add support for group-id in the stream-start event	2013-07-22 15:30:13 +02:00
Sebastian Dröge	1a0278ed64	qtdemux: Add support for group-id in the stream-start event	2013-07-22 15:30:13 +02:00
Sebastian Dröge	1122698491	flvdemux: Add support for group-id in the stream-start event	2013-07-22 15:30:12 +02:00
Sebastian Dröge	6cc16da531	avidemux: Add support for group-id in the stream-start event	2013-07-22 15:30:12 +02:00
Mathieu Duponchelle	d67a671bfb	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.	2013-07-21 19:21:57 -04:00
Olivier Crête	96a8fb92e2	qtdemux: Add WRLE support	2013-07-19 14:58:30 -04:00
Tim-Philipp Müller	aa7d597120	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 19:38:30 +01:00
Tim-Philipp Müller	ce52b319ff	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 17:31:38 +01:00
Wim Taymans	bdd3c31902	qtdemux: simplify some helpers Some helper functions are not needed anymore or can be simplified.	2013-07-19 17:26:54 +02:00
Wim Taymans	61a8937ced	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 17:14:46 +02:00
Arnaud Vrac	40ab78825c	qtdemux: nitpicking in esds parsing	2013-07-19 14:26:18 +02:00
Arnaud Vrac	d0d25a5e1f	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	2013-07-19 14:26:08 +02:00
Arnaud Vrac	5def061d20	qtdemux: remove chapter stream Remove all streams that are actually table of contents, since we will never need the data after parsing them.	2013-07-18 11:48:12 +02:00
Arnaud Vrac	ae67c13416	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.	2013-07-18 11:48:11 +02:00
Arnaud Vrac	1237898351	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.	2013-07-18 11:48:11 +02:00
Arnaud Vrac	e561d12655	qtdemux: advertise subtitle streams as sparse	2013-07-18 11:48:11 +02:00
Arnaud Vrac	6e26f1d067	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 18:10:11 +01:00
Wim Taymans	4c97701650	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 15:17:19 +02:00
Sebastian Dröge	9f73447229	goom2k1: Fix event handling and negotiate as soon as possible	2013-07-17 14:30:16 +02:00
Sebastian Dröge	78c7c16e9e	goom: Fix event handling and negotiate as soon as possible	2013-07-17 14:28:43 +02:00
Wim Taymans	6b82c89562	qtdemux: add support for WRAW Fixes https://bugzilla.gnome.org/show_bug.cgi?id=704292	2013-07-17 09:57:17 +02:00
Wim Taymans	f698483bb3	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	2013-07-17 09:57:16 +02:00
Olivier Crête	54c5a7f690	rtp: Use gst_adapter_take_buffer_fast() where possible in RTP payloaders	2013-07-16 15:37:49 -04:00
Arnaud Vrac	54bba4f60c	qtdemux: reset segment on flush stop `cca2f555d1` 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	2013-07-16 10:47:20 +02:00
Matej Knopp	ca32442f86	qtdemux: offset samples according to edit list https://bugzilla.gnome.org/show_bug.cgi?id=700264	2013-07-15 09:59:23 +02:00
Matej Knopp	ae92ea21a1	aacparse: be less verbose when parsing LOAS streams https://bugzilla.gnome.org/show_bug.cgi?id=704162	2013-07-15 07:55:08 +02:00
Matej Knopp	3111161e8a	qtdemux: unselect instead of ignoring disabled track, detect chapter track https://bugzilla.gnome.org/show_bug.cgi?id=704007	2013-07-12 11:45:33 +02:00
Kyosuke Nekomura	4d517e94ef	audioecho: Fix handling of delay property in PLAYING/PAUSED state https://bugzilla.gnome.org/show_bug.cgi?id=703901	2013-07-12 09:36:16 +02:00
Olivier Crête	3aa20e7c8d	rtpmux: Enable proxy caps on the src pads	2013-07-11 17:21:22 -04:00
Matej Knopp	7b69f427f1	qtdemux: correct argument order in gst_util_uint64_scale_int_round https://bugzilla.gnome.org/show_bug.cgi?id=703350	2013-07-10 09:20:17 +02:00
Olivier Crête	1997acc8b2	rtpmux: Keep caps order from the peer or the filter	2013-07-09 17:43:31 -04:00
Sebastian Dröge	3d0988f46f	videomixer: Fix handling of buffers without a duration We'll have to pop buffer from collectpads and store it internally only to get the timestamp of the next buffer. If we continue to keep it in collectpads, no new buffer to calculate the end time will ever arrive. https://bugzilla.gnome.org/show_bug.cgi?id=703743	2013-07-09 12:42:17 +02:00
Sebastian Dröge	9e9d2ce098	videomixer: Fix negotiation with 0/1 framerates https://bugzilla.gnome.org/show_bug.cgi?id=703743	2013-07-09 11:53:28 +02:00
Jonas Holmberg	beebe2b7af	matroskademux: Unlock stream lock after use Stream lock of sink pad was not unlocked after non-updating seek.	2013-07-09 11:25:14 +02:00
Ognyan Tonchev	aa2d96c46b	multipartmux: Re-set need_segment flag after FLUSH_STOP https://bugzilla.gnome.org/show_bug.cgi?id=703182	2013-07-09 09:16:20 +02:00
Sebastian Dröge	0cc77d8e30	rtph263ppay: Don't pass upstream filter caps to downstream Downstream usually can't accept video/x-h263 but only application/x-rtp, so we would always get an empty intersection here. https://bugzilla.gnome.org/show_bug.cgi?id=702632	2013-07-08 14:10:44 +02:00
Wim Taymans	ab24598443	rtspsrc: avoid some strdup	2013-07-02 11:13:25 +02:00
Wim Taymans	7c950ef3f2	rtspsrc: add select-stream signal Add a signal to let the app select what streams will be selected. See https://bugzilla.gnome.org/show_bug.cgi?id=634419	2013-07-02 10:40:35 +02:00
Wim Taymans	2d276e1bcb	rtspsrc: avoid strdup	2013-07-02 10:40:35 +02:00
J. Rick Ramstetter	f01b751e52	rtp: Fix documentation and comments to use rtpbin instead of old gstrtpbin https://bugzilla.gnome.org/show_bug.cgi?id=703426	2013-07-02 10:12:17 +02:00
Wim Taymans	1db7e62060	rtspsrc: add signal to notify of the SDP This way, the app can look and modify the SDP.	2013-07-01 17:31:30 +02:00
Matej Knopp	4053e1d6ac	qtdemux: compute framerate from average sample duration https://bugzilla.gnome.org/show_bug.cgi?id=703350	2013-07-01 12:53:17 +02:00
Alban Browaeys	97015d3c93	flvdemux: Add flvversion 1 to the flash-video caps This allows using avdec_flv which requires this field to be present in the caps. FLV only supports flash-video version 1 right now. https://bugzilla.gnome.org/show_bug.cgi?id=703076	2013-07-01 11:43:46 +02:00
Sebastian Dröge	5f6469fe2a	deinterleave: Don't hold object lock while sending events downstream Based on a patch by Kishore Arepalli <kishore.arepalli@gmail.com> https://bugzilla.gnome.org/show_bug.cgi?id=703114	2013-07-01 11:37:00 +02:00
Sebastian Dröge	75b5a54f17	matroskademux: Add MPEG4 video profile/level to the caps	2013-07-01 11:01:13 +02:00
Sebastian Dröge	423bddac6a	matroskademux: Add AAC profile/level to the caps https://bugzilla.gnome.org/show_bug.cgi?id=703312	2013-07-01 11:01:13 +02:00
Wim Taymans	c469434ea8	vorbispay: add support for config-interval Align code with the theora payloader and add support for the config-interval to periodically send out the config headers.	2013-06-28 15:21:56 +02:00
Wim Taymans	006562c9f4	theorapay: small cleanups	2013-06-28 15:21:12 +02:00
Wim Taymans	cdc66462ce	theorapay: handle streamheaders as well	2013-06-28 12:08:19 +02:00
Wim Taymans	3169432ed4	vorbispay: always collect headers on data When we see a data packet, always check if we need to collect any previous headers.	2013-06-28 12:07:58 +02:00
Wim Taymans	6c716dfc25	vorbispay: handle streamheader as well Take config strings from the streamheader when we can Fixes https://bugzilla.gnome.org/show_bug.cgi?id=664312	2013-06-28 11:43:17 +02:00
David Svensson Fors	692206d3a7	rtph264pay: avoid double buffer unmap on error Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703171	2013-06-27 17:14:11 +02:00
Wim Taymans	3289a2963b	rtspsrc: reset-sync before play Call reset-sync on the rtpbin before we go to playing. This makes us require SR packets for all streams again before we attempt to sync them. If we don't reset, it might be that we combine SR packets from before and after the PAUSE/PLAYING state change and end up with huge bogus offsets.	2013-06-27 17:02:14 +02:00
Wim Taymans	519305d14d	jitterbuffer: improve sync on first packets Don't throw away the first RTCP packet if it arrives before the first RTP packet but remember and use it to signal sync once we get the RTP packet. See https://bugzilla.gnome.org/show_bug.cgi?id=691400	2013-06-27 16:23:20 +02:00
Wim Taymans	8969f00661	jitterbuffer: only signal loop when active Only signal the loop function when it is active.	2013-06-27 16:15:45 +02:00
Wim Taymans	4bd2ffb26e	jitterbuffer: signal timestamp discont We can now use the RESYNC buffer flag to mark a timestamp discont when we update the ts-offset property.	2013-06-27 16:13:37 +02:00
Wim Taymans	4258ddcc36	jpegpay: turn some errors into warnings Turn some errors into warnings, we can continue processing so this should not be fatal. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=657079	2013-06-26 20:49:41 +02:00
Wim Taymans	bb9d42b976	rtspsrc: avoid some flushes	2013-06-26 14:58:53 +02:00
Wim Taymans	f39ef2ab68	rtspsrc: handle data message when waiting for reply When we are waiting for a server reply, handle data messages instead of ignoring them.	2013-06-26 14:41:36 +02:00
Wim Taymans	61219dc6ed	rtspsrc: handle data messages in separate method Refactor and make a method to handle a data message.	2013-06-26 14:41:36 +02:00
Wim Taymans	a4be0c6de3	rtspsrc: add some more docs to handle-request signal See https://bugzilla.gnome.org/show_bug.cgi?id=702705	2013-06-25 20:36:18 +02:00
Youness Alaoui	52e440c91b	Send a clock_provide message on the bus when we get a netclock	2013-06-25 14:50:47 +02:00
Youness Alaoui	547df8e14f	rtspsrc: Expose use-pipeline-clock property	2013-06-25 14:50:33 +02:00
Wim Taymans	35f6e79b94	udpsink: bind to the given interface Actually call BINDTODEVICE to bind to the interface as given by the property. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702819	2013-06-24 17:13:05 +02:00
Sebastian Dröge	3c9aba91dc	matroska: Add initial VP9 support	2013-06-21 18:22:13 +02:00
Youness Alaoui	95906b8f1c	rtsp: go back into the loop after doing pause After we do a pause request, go back to loop mode so that we can listen for server messages again. See https://bugzilla.gnome.org/show_bug.cgi?id=702705	2013-06-21 10:42:20 +02:00
Olivier Crête	2cd6f53e24	rtpptdemux: Wait after the caps to forward the other events First forward the stream-start, then the caps, then the rest	2013-06-20 23:16:59 -04:00
Wim Taymans	b96d931bf4	rtspsrc: fix race in state change to paused When we go to paused, we first flush the connection and then send the pause command. As a result of the flushing, the scheduled paused command can get lost. Wait until the connection is completely flushed and the rtsp task is waiting before issuing the paused or playing request. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702705	2013-06-20 14:43:47 +02:00
Wim Taymans	8428423c04	qtdemux: handle SEGMENT query	2013-06-20 11:31:22 +02:00
Kishore Arepalli	5b32891ae1	avidemux: duration query returns zero for DV video in avi https://bugzilla.gnome.org/show_bug.cgi?id=702625	2013-06-19 11:17:22 +02:00
Sebastian Dröge	b001da2926	qtdemux: Disable usage of allocation queries This can only reliably work if demuxers have a separate streaming thread per srcpad. This should be done in a demuxer base class, which integrates parts of multiqueue https://bugzilla.gnome.org/show_bug.cgi?id=701856	2013-06-19 11:07:48 +02:00
Alex Ashley	46a137c810	Avoid skipping moov atoms for fragmented MP4 files. bug #700505 Following a representation change that causes a resolution change, the video decoder fails to decode correctly. Dashdemux detects the representation change and pushes a new caps event and an initialization segment (a new moov atom) to the downstream qtdemux, but it doesn't handle this new moov yet, it will only parse the first one it receives. This commit changes qtdemux to accept a new moov in a dash bitstream switching scenario.	2013-06-19 01:44:22 -03:00
Thiago Santos	384e8f6c34	qtdemux: send stream-start only once for each stream Do not send stream start again when reconfiguring a pad for new caps. That is common for adaptive streams	2013-06-19 00:55:30 -03:00
Jens Georg	745be945ce	rtpmp2tdepay: accept mislabelled streams from GStreamer 0.10 as well The mp2t payloader in 0.10 mislabelled the streams as MP2T-ES instead of MP2T, so accept that as well for compatibility reasons. https://bugzilla.gnome.org/show_bug.cgi?id=702457	2013-06-17 15:39:17 +01:00
Wim Taymans	d9bc48edc9	rtspsrc: manage element state ourselves Lock the state of the all our elements and manage their states outselves. Because we are working async, we can't rely on the state change function to set the state at the right time or to return the right return value from the state change function. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702046	2013-06-16 05:40:13 +02:00
Bruno Gonzalez	e89a48616b	matroskademux: Don't unlock stream lock without locking it first https://bugzilla.gnome.org/show_bug.cgi?id=702167	2013-06-14 14:10:13 +02:00
Wim Taymans	51c9f7989f	rtpsession: Use the right hashtable to calculate bandwidth Don't use an unused hashtable to iterate source to calculate bandwidth. Remove unused code.	2013-06-13 16:02:19 +02:00
Sebastian Dröge	01cc493944	Revert "videomixer: When all sinkpads are eos, update output segment stop and forward it" This reverts commit `2d3910fc79`. It's not solving any problem and instead causes code to fall apart. https://bugzilla.gnome.org/show_bug.cgi?id=701519	2013-06-12 18:25:59 +02:00
Tim-Philipp Müller	213cd2777b	matroskademux: mark subtitle streams as sparse in stream-start event And also mark the streams that should be selected by default if marked so in the headers. https://bugzilla.gnome.org/show_bug.cgi?id=600648	2013-06-12 15:31:22 +01:00
Stefan Sauer	39c4c5f251	audiopanorama: add prebuilt files	2013-06-11 22:14:33 +02:00
Stefan Sauer	349a60e164	audiopanorama: cleanup of transform() Only map input if we are reading it. Cleanup the logging and the comments a bit.	2013-06-11 21:48:18 +02:00
Stefan Sauer	1dc06932a2	audiopanorama: use orc to speedup processing Use special variants for the case when we don't change the panorama (pan=0.0). Simplify the processing functions by passing the panorama value directy instead of the instance. Use orc for clearing buffers too.	2013-06-11 21:48:18 +02:00
Mathieu Duponchelle	6e23f1fec4	videomixer: check last end_time after conversion to running segment The last end_time was saved after conversion, so the comparison had to be made after conversion for it to make sense. https://bugzilla.gnome.org/show_bug.cgi?id=701385	2013-06-11 21:03:35 +02:00
Mathieu Duponchelle	4243714301	videomixer: add mix->segment.start to output_end_time When the segment start is not 0, this created a situation where the output_end_time is inferior to output_start_time, and the duration of the next buffer ended up underflowing. https://bugzilla.gnome.org/show_bug.cgi?id=701385	2013-06-11 21:03:03 +02:00
Sebastian Dröge	e2b46a776f	matroskademux: Send stream headers after the segment event https://bugzilla.gnome.org/show_bug.cgi?id=700799	2013-06-11 13:54:53 +02:00
Sebastian Dröge	adc9f0bd10	qtdemux: Do allocation query after exposing all pads and no-more-pads Also configure video streams as early as possible. Related https://bugzilla.gnome.org/show_bug.cgi?id=701856 but not fixing that.	2013-06-11 12:27:19 +02:00
Sebastian Dröge	ab275b62a8	flvdemux: Don't forward CAPS events from upstream Just use the default pad event handler. https://bugzilla.gnome.org/show_bug.cgi?id=701976	2013-06-11 12:27:19 +02:00
Stefan Sauer	4ef27eb0f9	audiopanorama: move the enum to the header and use instead of gint Move the enum for the processing method to the header so that we can use the type for the instance struct.	2013-06-09 20:39:48 +02:00
Sebastian Dröge	1ba08e331c	wavenc: Link with libgstbase for GstByteWriter	2013-06-07 15:15:15 +02:00
Sebastian Dröge	db1c2a28a6	wavparse: Push stream-start event in pull mode before anything else	2013-06-07 13:27:07 +02:00
Sebastian Dröge	048866f1b1	Release 1.1.1	2013-06-05 18:31:40 +02:00
Sebastian Dröge	ea75b890dc	wavenc: Fix taglist ref handling that made the unit test fail	2013-06-05 15:50:04 +02:00
Wim Taymans	0d27829a6b	udpsink: avoid leaking the host Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701586	2013-06-05 12:14:01 +02:00
Thiago Santos	7c12435f9b	qtdemux: make sure taglist is writable before adding tags Avoids assertions	2013-06-02 15:37:06 -03:00
Thiago Santos	78dfdee2aa	qtdemux: effectively skip tracks that weren't listed on the 1st moov Without this, stream is NULL and the code will try to access it, leading to segfaults.	2013-06-02 13:06:15 -03:00
Thiago Santos	70fca21c28	qtdemux: skip redundant check !got_moov is already checked the line above	2013-06-02 13:06:15 -03:00
Stefan Sauer	bcf1bba689	level: remove unused variables in instance struct	2013-06-01 21:34:37 +02:00
Anton Belka	db29522a43	wavenc: add tags & toc support Write tags as LIST INFO chunk. Format the toc as cue + LIST adtl chunk. Remove old #ifdef'ed code.	2013-06-01 21:34:37 +02:00
Wim Taymans	1f0600ee6f	Revert "rtph264pay: Restructuring to allow for adding optional caps" This reverts commit `61666898cf`. This commit changes what the set_sps_pps() function does, not it doesn't set caps anymore (and should have been renamed). The main problem is that not all call sites are updated and thus leak the string.	2013-05-31 15:18:48 +02:00
Wim Taymans	1516c14881	Revert "rtph264pay/depay: Add frame dimensions a payloaded caps" This reverts commit `3dca756a5d`. The H264 RTP spec has no attributes for width and height.	2013-05-31 15:11:12 +02:00
Wim Taymans	b79d217396	Revert "rtph264pay/depay: Add optional framerate caps for use in SDP" This reverts commit `d8825e2a5c`. There is no framerate attribute in the h264 RTP spec.	2013-05-31 15:09:51 +02:00
Wim Taymans	190b3d6688	Revert "rtpjpegpay/depay: Replace framesize caps with width/height" This reverts commit `0075d111b4`. Extra application/x-rtp are SDP fields, which are strings.	2013-05-31 15:08:16 +02:00
Wim Taymans	f870cef8bc	Revert "rtpjpegpay/depay: Replace framerate caps field with fraction" This reverts commit `9fd25a810b`. We deal with sdp attributes in application/sdp, which are always strings.	2013-05-31 15:05:51 +02:00
Wim Taymans	25082a50b9	rtspsrc: add extra TLS url protocols We also support TLS protocols now.	2013-05-31 12:34:22 +02:00
Sebastian Dröge	e2e1d1a158	videomixer: Add FIXME comment about the DURATION query from adder Currently the code just takes with maximum upstream duration, which is wrong. It should be the maximum upstream duration in running time.	2013-05-30 23:56:38 +02:00
Mathieu Duponchelle	5223868caa	videomixer: Set a reference to mix->current_caps as the QUERY_CAPS result.	2013-05-30 15:36:48 -04:00
Stefan Sauer	6feaf69bec	level: misc cleanups Fix some oudated comments. Sort out some confusion of interval_frames and num_frames.	2013-05-30 17:38:55 +02:00
Stefan Sauer	52282b5faa	level: fix discontinuities in timestamps	2013-05-28 19:09:12 +02:00
Wim Taymans	80850df711	rtspsrc: create and push stream-start in TCP mode	2013-05-28 15:45:49 +02:00
Wim Taymans	4fc1f3088b	rtspsrc: remove some obsolete code It is not needed to do a state change from the _play() function on ourselves. The state change function already did that and we don't want to interfere with that (or use hacks to avoid interference).	2013-05-28 15:10:07 +02:00
Wim Taymans	e6f850996b	rtspsrc: set RTCP caps on the RTCP pads	2013-05-28 12:26:25 +02:00
Wim Taymans	63f0ecbbe7	rtpsession: send stream-start and segment events Also send stream-start and segment event on the RTCP pad. We don't need to send anything on the sync_src pad because we already forwarded all incomming events.	2013-05-28 12:26:25 +02:00
Wim Taymans	779bcc093c	rtspsrc: add signal to handle server requests Add a signal to be notified of a server request. The signal handler can then construct the response message for the server. See https://bugzilla.gnome.org/show_bug.cgi?id=632207	2013-05-28 12:26:24 +02:00
Nicolas Dufresne	cd30a81ee3	videomixer: Maintain z-order when new pad are added https://bugzilla.gnome.org/show_bug.cgi?id=701109	2013-05-27 22:43:25 -04:00
Thibault Saunier	7a3df1ab31	videomixer: Always handle flush_stop_pending atomically It is not protected with the COLLECT_PADS_STREAM_LOCK anymore	2013-05-25 12:20:08 -04:00
Thibault Saunier	608bd3e2db	videomixer: Do not take COLLECT_PADS_STREAM_LOCK when unnecessary Collectpad takes the lock itself when receiving serialized events and we should not take it for not serialized ones	2013-05-25 11:03:31 -04:00
Sebastian Dröge	1b5a8ac41c	flxdec: Properly skip non-frame chunks	2013-05-24 19:34:05 +02:00
Sebastian Dröge	ae3ee32f42	flxdec: Flush data from adapter after reading it Otherwise we're going in an infinite loop, reading the same data over and over again.	2013-05-24 19:31:14 +02:00
Andoni Morales Alastruey	a62af107ae	goom2k1: fix more duplicated symbols	2013-05-24 09:29:23 +02:00
Sebastian Rasmussen	9fd25a810b	rtpjpegpay/depay: Replace framerate caps field with fraction The previous implementation had the formatting of SDP attributes happen in each RTP payloader, now instead the constituent values are propagated as caps fields. This allows for applications to do SDP offer/answer based on caps negotiation. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748	2013-05-23 21:05:49 +02:00
Sebastian Rasmussen	0075d111b4	rtpjpegpay/depay: Replace framesize caps with width/height The previous implementation had the formatting of SDP attributes happen in each RTP payloader, now instead the constituent values are propagated as caps fields. This allows for applications to do SDP offer/answer based on caps negotiation. Keep parsing a-framerate, x-framerate and x-dimensions in rtpjpegdepay to be backwards compatible with previous payloaders. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748	2013-05-23 21:05:43 +02:00
Sebastian Rasmussen	d8825e2a5c	rtph264pay/depay: Add optional framerate caps for use in SDP This allows for applications to format SDP attributes and still do SDP offer/answer based on caps negotiation. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749	2013-05-23 21:04:17 +02:00
Sebastian Rasmussen	3dca756a5d	rtph264pay/depay: Add frame dimensions a payloaded caps This allows for applications to format SDP attributes and still do SDP offer/answer based on caps negotiation. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749	2013-05-23 21:04:11 +02:00
Sebastian Rasmussen	61666898cf	rtph264pay: Restructuring to allow for adding optional caps Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700749	2013-05-23 21:04:00 +02:00
Sebastian Dröge	e26b8c2832	(dyn\|multi)udpsink: Add properties to specify the bind address and port By default we use the any addresses and a random port for binding the socket.	2013-05-23 18:42:09 +02:00
Sebastian Dröge	5b79b8ff3c	(dyn\|multi)udpsink: Bind socket before using it https://bugzilla.gnome.org/show_bug.cgi?id=700878	2013-05-23 18:05:07 +02:00
Sebastian Dröge	1ed7f7a6a8	(multi)udpsink: Add missing getters for socket-v6 and used-socket-v6 properties	2013-05-23 17:26:31 +02:00
Nicolas Dufresne	d8c5e31657	videomixer: Don't hold stream-lock while pushing non-serialized events https://bugzilla.gnome.org/show_bug.cgi?id=700868	2013-05-23 09:20:04 -04:00
Nicolas Dufresne	a7e0f251ca	videomixer: Don't hold object lock while sending events https://bugzilla.gnome.org/show_bug.cgi?id=700868	2013-05-23 09:20:04 -04:00
Sebastian Dröge	ecc6c607ff	deinterlace: The return value of gst_pad_set_caps() is not relevant anymore Caps can fail to be set because the pad is not linked yet for example.	2013-05-22 17:34:07 +02:00
David Schleef	318cd39c3e	qtdemux: Add error if file has playready drm	2013-05-21 18:21:49 -07:00
Thibault Saunier	18ef4f18d0	videomixer: Send a reconfigure event upstream if sinkpad caps are not usable https://bugzilla.gnome.org/show_bug.cgi?id=684237	2013-05-21 12:15:36 -04:00
Alexander Schrab	a1df050356	mulawdec: Handle NULL buffers in handle_frame https://bugzilla.gnome.org/show_bug.cgi?id=698894	2013-05-21 15:18:04 +02:00
Sebastian Rasmussen	2361567bae	rtpjpegpay/depay: Add framesize caps for use in SDP The format of the value adheres to RFC6064 and it is meant to be parsed and included in the SDP sent by gst-rtsp-server to its clients. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748	2013-05-21 09:09:03 +02:00
Sebastian Rasmussen	919eed0787	rtpjpegpay: Add optional framerate caps for use in SDP The format of the value adheres to RFC4566 and it is meant to be parsed and included in the SDP sent by gst-rtsp-server to its clients. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700748	2013-05-21 09:08:21 +02:00
Mathieu Duponchelle	2d3910fc79	videomixer: When all sinkpads are eos, update output segment stop and forward it https://bugzilla.gnome.org/show_bug.cgi?id=699793	2013-05-20 21:06:56 +02:00
Mathieu Duponchelle	521c9a7b5d	videomixer: Don't reset the output segment on flush stop Only init it when getting from READY to PAUSED, and change it on seek events. https://bugzilla.gnome.org/show_bug.cgi?id=699793	2013-05-20 21:03:03 +02:00
Thibault Saunier	86b106091c	videomixer: Send caps event from the streaming thread This way we avoid races in caps negotiation and we make sure that the caps are sent after stream-start. https://bugzilla.gnome.org/show_bug.cgi?id=684237	2013-05-19 09:28:04 -04:00
Thibault Saunier	718f9004d0	videomixer: Do not send flush_stop when receiving a seek There is no reason to send a flush-stop when receiving a seek event. In the case of a flushing seek, we could eventually want to, but in the code path were we check if the seek is "flushing", we have the following comment that makes sense: "we can't send FLUSH_STOP here since upstream could start pushing data after we unlock mix->collect. We set flush_stop_pending to TRUE instead and send FLUSH_STOP after forwarding the seek upstream or from gst_videomixer_collected, whichever happens first." https://bugzilla.gnome.org/show_bug.cgi?id=684237	2013-05-19 09:28:04 -04:00
Thibault Saunier	85b6852deb	videomixer2: Protect flush_stop_pending with the collectpad stream lock And make sure to expect a flush-stop after a flush-start https://bugzilla.gnome.org/show_bug.cgi?id=684237	2013-05-19 09:28:04 -04:00
Michael Olbrich	d1c56376d6	rtpmp4apay: clear config buffer before using it This is necessary because parts of the memory are only modified with "\|=" https://bugzilla.gnome.org/show_bug.cgi?id=700514	2013-05-18 10:57:56 +01:00
Thiago Santos	55caa99ccd	qtdemux: Do not expect EOS after a segment event if upstream is mss In case qtdemux is handling a mss stream, do not mark the stream to wait for EOS after a segment. Even if it seems to be the last one according to the current streams information. MSS handling is different here because there is another demuxer driving the pipeline	2013-05-16 16:50:49 -03:00
Thiago Santos	5517e352ab	qtdemux: only set channels and rate if qtdemux knows it Setting both of those to 0 is pointless and means that qtdemux doesn't know the real value. Avoid setting it in this case.	2013-05-16 16:50:49 -03:00
Arnaud Vrac	6edcc564ba	qtdemux: set alac caps using info from codec buffer The samplerate field in the STSD atom is not right for some ALAC files (usually when audio is 96kHz/24bits), so the audio caps must be extracted from the codec data. https://bugzilla.gnome.org/show_bug.cgi?id=700382	2013-05-15 18:42:11 +01:00
Arnaud Vrac	8ed611cdbc	avidemux: do not push discont buffers if they aren't discont https://bugzilla.gnome.org/show_bug.cgi?id=682110	2013-05-15 13:16:11 +01:00
Joshua M. Doe	837dcfb363	videocrop: Add support for GRAY16_LE/GRAY16_BE https://bugzilla.gnome.org/show_bug.cgi?id=700331	2013-05-15 09:29:30 +02:00
Sebastian Dröge	41e1af3751	rgvolume: Send all events through the proxypads instead of just sending to the target Otherwise the sticky events are missing on the proxypads.	2013-05-14 17:29:58 +02:00
Sebastian Dröge	4fdbf88a65	matroskaparse: Make sure to send a segment event before dataflow	2013-05-14 13:52:18 +02:00
Sebastian Dröge	5c8bb90262	deinterlace: Improve handling of min/max buffer numbers of the buffer pool	2013-05-14 09:45:12 +02:00
Matej Knopp	30c00f4fb7	deinterlace: set caps for buffer pool config	2013-05-14 09:38:24 +02:00
Olivier Crête	4f0fdabf10	multifilesink: Let the base class do get_times This will make sync=TRUE work, the default is still sync=FALSE	2013-05-13 13:34:22 -04:00
Nicolas Dufresne	f67c227878	interleave: Send stream-start before caps event	2013-05-13 15:37:38 +02:00
Nicolas Dufresne	04c9f43567	rtpmux: Send stream-start before caps	2013-05-13 15:37:05 +02:00
Sebastian Dröge	6dee7d3a06	icydemux: Fix sticky event handling	2013-05-13 15:19:25 +02:00
Sebastian Dröge	9ac456bd43	flvmux: Push sticky events in the right order	2013-05-13 15:06:03 +02:00
Sebastian Dröge	0ab23ef5c9	deinterleave: Fix sticky event handling	2013-05-13 14:54:35 +02:00
Sebastian Dröge	c94fbf6206	deinterleave: Code style fixes	2013-05-13 13:55:44 +02:00
Sebastian Dröge	f28ab45f3e	rtpgstpay: First let baseclass handle events, then put them into the stream Fixes handling of sticky events. https://bugzilla.gnome.org/show_bug.cgi?id=700213	2013-05-13 13:44:35 +02:00
Tim-Philipp Müller	8359b6bff1	multipartdemux: fix example pipeline Need jpegparse.	2013-05-10 14:01:14 +01:00
Nicolas Dufresne	0b737fba0d	shapewipe: Can't map twice the same buffer for writing I took the opportunity to simplify that code a bit. We now use gst_buffer_make_writable() to make the buffer writable and map twice the same buffer, with first map being read/write, and second read only. This get rid of the critical: GStreamer-CRITICAL **: gst_structure_set_name: assertion `IS_MUTABLE https://bugzilla.gnome.org/show_bug.cgi?id=700044	2013-05-10 09:27:02 +02:00
Nicolas Dufresne	13a5d0304d	shapewipe: Ensure caps are writable The exist one case where that we endup with original caps in ret, in which case we are not guaratied to have writable caps. Simply ensure this is the caps are writable before entering the loop. https://bugzilla.gnome.org/show_bug.cgi?id=700044	2013-05-10 09:26:07 +02:00
Nicolas Dufresne	59c2f459de	shapewipe: Fix sample pipeline in documentation https://bugzilla.gnome.org/show_bug.cgi?id=700044	2013-05-10 09:26:00 +02:00
Sebastian Dröge	3110b7cc31	Revert "videomixer2: Take into account new segments" This reverts commit `84ae670ab4`. Actually this is not how it is supposed to work. videomixer creates a [0,-1] segment and then puts frames of the different streams there based on their running times in their own segments.	2013-05-09 16:26:19 +02:00
Mathieu Duponchelle	84ae670ab4	videomixer2: Take into account new segments Also forward the event downstream on the next opportunity. https://bugzilla.gnome.org/show_bug.cgi?id=699793	2013-05-09 16:18:54 +02:00
Tim-Philipp Müller	643450c9b8	Revert "gstrtspsrc: set buffer-size for multicast buffers" This reverts commit `2481e95d03`. This is already done five lines above, it was added a year ago in commit `561b131e`.	2013-05-09 09:09:59 +01:00
Nicolas Dufresne	2d53229a86	audiowsinclimit: Frequence property renamed cutoff Updating the documentation to reflect this change. See: https://bugzilla.gnome.org/show_bug.cgi?id=699964	2013-05-09 08:46:04 +02:00
Aha Unsworth	2481e95d03	gstrtspsrc: set buffer-size for multicast buffers For receiving video data via RTSP when the video is sent via multicast there is no way to specify the udpsrc buffer-size. On windows the native network buffer is not large and with video i-frames being huge the buffer is to small and you get i-frame corruption, it looks terrible, and there is no (easy) way to set the udpsrc buffer-size. https://bugs.freedesktop.org/show_bug.cgi?id=52264	2013-05-08 16:57:53 -03:00
Sebastian Dröge	1588cda9a1	videomixer2: Send stream-start before caps event https://bugzilla.gnome.org/show_bug.cgi?id=699895	2013-05-08 16:02:46 +02:00
Thiago Santos	a0e934e72e	qtdemux: push new caps events when caps change Whenever the demuxer has a new caps on a stream, it should set the new_caps variable to true and a new caps event will be pushed before the next buffer	2013-05-07 19:29:17 -03:00
Thiago Santos	725faab590	qtdemux: do not push discont buffers if they aren't discont qtdemux takes its buffers from a GstAdapter. Those buffers are created from the larger buffer that it obtained from upstream and they carry the same flags, including DISCONT if it is set. In these cases, all buffers that qtdemux is going to push would be marked as DISCONT. This scenario can make parsers/decoders flush on every buffer leading to no decoding at all hapenning. This patch prevents this by unsetting the flag if it shouldn't be set.	2013-05-07 19:29:17 -03:00
Thiago Santos	4d073beeee	qtdemux: some code cleanup for mss handling code * Explicitly init variables for fragmented formats at init * Do not use GstClockTime type if the variable isn't a timestamp * Fix a style/readability issue at an if block * Group 2 mss mode conditional blocks together to improve readability Conflicts: gst/isomp4/qtdemux.c	2013-05-07 19:29:17 -03:00
Thiago Santos	d1b91c755c	qtdemux: avoid storing non-time newsegments to push later This can confuse downstream when they get a byte segment after receiving the natural time segment from qtdemux that it sends when starting to push buffers. This is specially the case with parsers that try to convert the position from byte to time format and might miss the correct position for playback to start.	2013-05-07 19:29:17 -03:00
Thiago Santos	895525b5cb	qtdemux: avoid setting fields to non-writable caps	2013-05-07 19:29:17 -03:00
Wim Taymans	544d926732	qtdemux: don't send so many segment events Only send one segment event in the beginning of the stream, not after each moov and moof atom. Conflicts: gst/isomp4/qtdemux.c	2013-05-07 19:29:17 -03:00
Wim Taymans	d9cd4fcc17	qtdemux: place incomming timestamps on output Place the incomming timestamp (if any) directly onto the outgoing buffers and interpollate other timestamps. Conflicts: gst/isomp4/qtdemux.c	2013-05-07 19:29:17 -03:00
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

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8013 commits