gstreamer

mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-23 08:46:40 +00:00

Author	SHA1	Message	Date
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
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
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
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
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
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
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
Wim Taymans	c44a29bd53	bin: fix compilation	2013-07-24 14:17:45 +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
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
Sebastian Dröge	5a9f4a3cbc	rtpsession: Add support for group-id in the stream-start event	2013-07-22 15:30:13 +02:00
Olivier Crête	3aa20e7c8d	rtpmux: Enable proxy caps on the src pads	2013-07-11 17:21:22 -04:00
Olivier Crête	1997acc8b2	rtpmux: Keep caps order from the peer or the filter	2013-07-09 17:43:31 -04:00
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	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
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	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
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
Nicolas Dufresne	04c9f43567	rtpmux: Send stream-start before caps	2013-05-13 15:37:05 +02:00
Sebastian Dröge	b0b0557c48	gst: Add better support for static plugins	2013-04-15 15:54:11 +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
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
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
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
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	747447d298	rtpsession: avoid '...is used uninitialized'	2013-01-29 10:32:51 +01:00
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
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	8a3b116d1f	docs: add rtpmux and rtpdtmfmux to plugin docs https://bugzilla.gnome.org/show_bug.cgi?id=629117	2012-12-16 16:36:39 +00:00
Tim-Philipp Müller	3295b5d791	rtpmanager: move rtpmux and rtpdtmfmux elements from -bad https://bugzilla.gnome.org/show_bug.cgi?id=629117	2012-12-16 16:36:39 +00:00
Tim-Philipp Müller	de204ba754	rtpmux: Fix FSF address https://bugzilla.gnome.org/show_bug.cgi?id=687520	2012-12-16 16:36:39 +00:00
Tim-Philipp Müller	2778a1757f	rtpmux: Use gst_element_class_set_static_metadata() where possible. Avoids some string copies. Also re-indent some stuff. Also some indent fixes here and there.	2012-12-16 16:36:39 +00:00
Olivier Crête	15dfdc58d4	rtpmux: Misc fix for 0.11 Convert the incoming caps before proxying them Clear the last_pad when going to ready tests: Implement accept_caps, don't leak event	2012-12-16 16:36:38 +00:00
Wim Taymans	83262be703	rtpmux: update for RTP buffer api changes	2012-12-16 16:36:38 +00:00
Sebastian Dröge	f17064a8ea	rtpmux: Update for GST_PLUGIN_DEFINE() API changes	2012-12-16 16:36:34 +00:00
Wim Taymans	c86156ad8f	rtpmux: fix compilation	2012-12-16 16:35:36 +00:00
Wim Taymans	6826bbb6da	rtpmux: fix for caps api changes	2012-12-16 16:35:33 +00:00
Matej Knopp	bb345a584d	rtpmux: Fix compiler warnings	2012-12-16 16:35:29 +00:00
Olivier Crête	af4e999c59	rtpmux: Unref non-forwarded events Also, don't unref forwarded ones	2012-12-16 16:35:29 +00:00
Olivier Crête	a8789d1df1	rtpmux: resync iterator on resync	2012-12-16 16:35:29 +00:00
Olivier Crête	0c54079af5	rtpmux: Re-push sticky events on input pad change	2012-12-16 16:35:29 +00:00
Olivier Crête	21831b430f	rtpmux: Don't leak gvalue from iterator	2012-12-16 16:35:29 +00:00
Wim Taymans	ccc4b960fc	rtpmux: more porting	2012-12-16 16:35:26 +00:00
Olivier Crête	f20a6b1d16	rtpmux: port to 0.11	2012-12-16 16:35:26 +00:00
Wim Taymans	35b6668fb6	rtpmux: make request pads take _%u	2012-12-16 16:35:22 +00:00
Olivier Crête	aa3607ef5c	rtpdtmfmux: Add last-stop to dtmf-event upstream events Add the running time of the last outputted buffer to the upstream "dtmf-event" events so that the dtmf source does not leave a gap.	2012-12-16 16:35:22 +00:00
Edward Hervey	d137482fe5	rtpmux: Remove dead assignments	2012-12-16 16:35:22 +00:00
Stefan Kost	55aae6bfab	rtpmux: add missing G_PARAM_STATIC_STRINGS flags Canonicalize property names as needed.	2012-12-16 16:35:15 +00:00
Olivier Crête	9674d5cc23	rtpmux: Improve documentation Add an example pipeline, and try to explain a bit more what it does.	2012-12-16 16:35:15 +00:00
Stefan Kost	ca27a279ba	rtpdtmfmux: remove unused variable	2012-12-16 16:35:15 +00:00
Stefan Kost	c85dceeacb	rtpdtmfmux: remove unused signal boilerplate	2012-12-16 16:35:15 +00:00
Stefan Kost	2353f8d852	rtpmux: no need to ref pad in _chain()	2012-12-16 16:35:15 +00:00
Youness Alaoui	e42d2eebcb	rtpmux: Unlock the right mutex The mutex locked is for the 'mux' object, but we unlock the pad, which means that if the rtpmux gets a flush, then the object lock will stay locked forever, causing it to freeze the next time it tries to take it. Fixes bug #627991	2012-12-16 16:35:15 +00:00
Olivier Crête	78d1ebac9e	rtpmux: Add support for GstBufferList Factor out most of the buffer handling and implement a chain_list function. Also, the DTMF muxer has been modified to just have a function to accept or reject a buffer instead of having to subclass both chain and chain_list.	2012-12-16 16:35:15 +00:00
Olivier Crête	c00f14419b	rtpmux: Don't leak invalid buffers	2012-12-16 16:35:15 +00:00
Tim-Philipp Müller	a45429d81d	rtpmux: fix missing debug log message argument	2012-12-16 16:35:15 +00:00
Olivier Crête	4a8d0243b5	rtpdtmfmux: Add some debug messages	2012-12-16 16:35:14 +00:00
Olivier Crête	423ce98666	rtpdtmfmux: Remove stream-lock event handling	2012-12-16 16:35:14 +00:00
Olivier Crête	a4500c0e74	rtpdtmfmux: Update doc for simplification	2012-12-16 16:35:14 +00:00
Olivier Crête	70097866de	rtpdtmfmux: Drop buffers on non-priority sinks when something is incoming on the priority sink	2012-12-16 16:35:14 +00:00
Olivier Crête	f6548fe9b6	rtpdtmfmux: Add priority sink pads	2012-12-16 16:35:14 +00:00
Olivier Crête	2bcea1537b	rtpdtmfmux: Cleanup event function	2012-12-16 16:35:14 +00:00
Olivier Crête	8e58646f5c	rtpmux: Aggregate incoming segments	2012-12-16 16:35:14 +00:00
Olivier Crête	7be57cac3a	rtpdtmfmux: Update documentation	2012-12-16 16:35:14 +00:00
Olivier Crête	e590fc1f32	rtpmux: Simplify request pad creation	2012-12-16 16:35:14 +00:00
Benjamin Otte	2867e00225	rtpmux: gst_element_class_set_details => gst_element_class_set_details_simple	2012-12-16 16:35:10 +00:00
unknown	fb7266884d	rtpmux: update the current_ssrc from the caps Fixes #604101	2012-12-16 16:33:47 +00:00
Håvard Graff	eab65e84ca	rtpmux: release pads when disposing Because of an allocated priv (GstRTPMuxPadPrivate), the element will leak memory if not gst_rtp_mux_release_pad() is called. This would previously only happen if release_request_pad() was called explicitly, somthing that should not be neccesary. Fixes #604099	2012-12-16 16:33:46 +00:00
Wim Taymans	0d54122804	dtmfmux: method name cleanups	2012-12-16 16:33:46 +00:00
Olivier Crête	3841cc74cf	rtpmux: Don't ignore requested pad name	2012-12-16 16:33:46 +00:00
Olivier Crête	d93295ff9d	rtpmux: Remove empty finalize	2012-12-16 16:33:46 +00:00
Olivier Crête	5e90a4e86b	rtpmux: Free the pad private data on pad release Free the pad private data on pad release instead of using a weak ref, which is not thread safe. Also, lock the content of the pad private using the element's object lock.	2012-12-16 16:33:46 +00:00
Olivier Crête	4be63c9add	rtpmux: Reject wrong caps	2012-12-16 16:33:46 +00:00
Olivier Crête	0111bafb3a	rtpmux: Fix leak Fixed a leak discovered by Laurent Glayal <spegle@yahoo.fr>	2012-12-16 16:33:46 +00:00
Olivier Crête	fcc1522d2e	rtpmux: Fix leak Fixed a leak discovered by Laurent Glayal <spegle@yahoo.fr>	2012-12-16 16:33:46 +00:00
Olivier Crête	ff6686f1c7	rtpmux: Fix warning	2012-12-16 16:33:46 +00:00
Olivier Crête	00791f930b	rtpmux: Set different caps depending on the input	2012-12-16 16:33:46 +00:00
Olivier Crête	ed0b407038	rtpmux: Only free pad private when pad is disposed	2012-12-16 16:33:45 +00:00
Olivier Crête	92bb5199ac	rtpmux: Remove useless caps mangling	2012-12-16 16:33:45 +00:00
Olivier Crête	3ccf3217fe	rtpmux: Rename variable for more clarity	2012-12-16 16:33:45 +00:00
Olivier Crête	4b958f6d8d	rtpmux: Use GST_BOILERPLATE	2012-12-16 16:33:45 +00:00
Olivier Crête	abe57be248	rtpmux: Do the includes locally	2012-12-16 16:33:45 +00:00
Olivier Crête	05844c89e9	rtpmux: Add GST_DEBUG_FUNCPTRs	2012-12-16 16:33:45 +00:00
Olivier Crête	fd102b95ab	rtpdtmfmux: Release locked pad on release_pad Release the special pad if the pad is removed from the muxer.	2012-12-16 16:33:45 +00:00
Laurent Glayal	00f8bab712	rtpdtmfmux: Release special on pad dispose Fixes #577690	2012-12-16 16:33:45 +00:00
Stefan Kost	a4a22454dc	docs: various doc fixes No short-desc as we have them in the element details. Also keep things (Makefile.am and sections.txt) sorted. Reword ambigous returns. No text after since please.	2012-12-16 16:33:41 +00:00
Olivier Crête	7d4395a910	rtpmux: Move rtpmux from gst-plugins-farsight to -bad	2012-12-16 16:33:27 +00:00
Olivier Crête	68215752f4	rtpmux: Re-indent to Gst style	2012-12-16 16:33:24 +00:00
Olivier Crête	c7d0809434	rtpmux: Document rtp muxer a bit	2012-12-16 16:33:20 +00:00
Laurent Glayal	47c7a93df2	rtpmux: Add signals before stream lock and after unlocking	2012-12-16 16:33:17 +00:00
Olivier Crête	f1656ed8b0	rtpmux: Let ssrc through getcaps	2012-12-16 16:33:14 +00:00
Olivier Crête	1529dffaf9	rtpmux: Rename have_base to have_ts_base	2012-12-16 16:33:11 +00:00
Olivier Crête	57563517bd	rtpmux: Protect the seqnum with object lock in rtpmux	2012-12-16 16:33:08 +00:00
Olivier Crête	d3237eaf95	rtpmux: Remove unused sink_ts_base	2012-12-16 16:33:04 +00:00
Olivier Crête	cc23958183	rtpmux: Have getcaps to force the same clockrate on all pads	2012-12-16 16:33:01 +00:00
Olivier Crête	dc36590d0c	rtpmux: Validate RTP data in RTP Mux	2012-12-16 16:32:57 +00:00
Olivier Crête	360c8d4f1d	rtpmux: Remove unused clock-rate property	2012-12-16 16:32:54 +00:00
Olivier Crête	b86232d0dc	rtpmux: Clarify locking in rtpdtmfmux	2012-12-16 16:32:50 +00:00
Laurent Glayal	4b607cdda5	rtpmux: Missing format parameter	2012-12-16 16:32:47 +00:00
Håvard Graff	b313c80367	rtpmux: Update seqnum base in rtp muxer With help from Wim	2012-12-16 16:32:43 +00:00
Håvard Graff	c479f90274	rtpmux: Fix some more leaks	2012-12-16 16:32:40 +00:00
Håvard Graff	1b5e769e0b	rtpmux: Fix leak	2012-12-16 16:32:37 +00:00
Olivier Crête	5cbb0de823	rtpmux: Don't unref caps we don't know (thanks Wim)	2012-12-16 16:32:32 +00:00
Olivier Crête	cebf506949	rtpmux: Put per-buffer debug at level LOG	2012-12-16 16:32:29 +00:00
Olivier Crête	3c12a423b7	rtpmux: Make debug print accurate	2012-12-16 16:32:25 +00:00
Olivier Crête	c49f4c87c6	rtpmux: Set our caps on the buffers	2012-12-16 16:32:22 +00:00
Olivier Crête	ec63da9366	rtpmux: Take the clock-base stored from the last setcaps	2012-12-16 16:32:18 +00:00
Olivier Crête	674c074114	rtpmux: Store the clock-base on setcaps	2012-12-16 16:32:15 +00:00
Olivier Crête	90264b9686	rtpmux: Add padprivate to the request pads	2012-12-16 16:32:11 +00:00
Olivier Crête	15d661ba3e	rtpmux: Make indentation more correct	2012-12-16 16:31:56 +00:00
Olivier Crête	3a7d09a749	rtpmux: Fix typo	2012-12-16 16:31:53 +00:00
Olivier Crête	91aef3ec5e	rtpmux: Set seqnum-base and clock-base in caps from rtpmuxer	2012-12-16 16:31:50 +00:00
Zeeshan Ali	6ea5ca354d	rtpmux: more debug 20070815135038-f3f1e-9c7a5490a525c6e8753cb1b8c03354df99132b5c.gz	2012-12-16 16:31:46 +00:00
Youness Alaoui	f0e209b638	rtpmux: missing comment 20070820185032-4f0f6-0ab67b6ac40dd4e35a8fe53f3cb6daff65ce43b9.gz	2012-12-16 16:30:33 +00:00
Olivier Crete	3ed5590da6	rtpmux: Make buffer writable before writing into it 20070712195336-3e2dc-91a5fb797cfa4919d4e2f9a728c6d6fbd3b83d93.gz	2012-12-16 16:30:31 +00:00
Olivier Crete	dd13f7c8ef	rtpmux: Set pads active when adding them to a potentially running element 20070706202459-3e2dc-a3731f885725594def0a7be997fc7b3a739ee967.gz	2012-12-16 16:30:27 +00:00
Olivier Crete	1c5075f927	rtpmux: Fix multiple ref leaks (patches by SP GLE) 20070607120121-3e2dc-061e9ef7a47b1b84fa8f8092f4b8bcc0e6db8c8c.gz	2012-12-16 16:30:23 +00:00
Zeeshan Ali	42f455e902	rtpmux: send event to all src pads 20070528152505-f3f1e-039216c73dc93f64c49962c77a0253cb9cfec4d3.gz	2012-12-16 16:30:18 +00:00
Zeeshan Ali	dba101bb0f	rtpmux: print a warning if receive an error iterating sinkpads 20070528123749-f3f1e-4c1eb3f511b5610143610a65a94d117f2c3d2580.gz	2012-12-16 16:30:15 +00:00
Zeeshan Ali	baa48dc6bc	rtpmux: deal with all the gst_iterator_next() return values 20070528122808-f3f1e-d301644c3be7633ec6dc5e28596e9346d2da6a50.gz	2012-12-16 16:30:12 +00:00
Zeeshan Ali	de40874670	rtpmux: Return correct value from the event handler 20070525123116-f3f1e-131b37b5f4521618fe2f1320409a47e65b35ad2d.gz	2012-12-16 16:30:08 +00:00
Zeeshan Ali	ed76f67e96	rtpmux: Ville's original patch to fix the traversal of dtmf event 20070525102709-f3f1e-6c41d1ef934068a4f4e810e7e981b420075b0c98.gz	2012-12-16 16:30:05 +00:00
zeeshan.ali@nokia.com	94ebe07862	rtpmux: Set the correct ts-offset on the get_prop value 20070329135250-65035-a43e222d91d57c0a61cb3287586aaa29abf78674.gz	2012-12-16 16:30:01 +00:00
zeeshan.ali@nokia.com	1ee542c378	rtpmux: Refactorize state_change 20070329135223-65035-23a0107b2e397710f035c6e88cc0e49b65bb4d5d.gz	2012-12-16 16:29:58 +00:00
zeeshan.ali@nokia.com	2498ba671a	rtpmux: set SSRC on the packets 20070329133622-65035-1be6e0aa85a71389f7d257b9cd3e13a73d6b745b.gz	2012-12-16 16:29:55 +00:00
zeeshan.ali@nokia.com	ee69c2690d	rtpmux: Code clean-up and more debug output 20070329131936-65035-9d499e209e0d7a409c3aa0d1040778babf076179.gz	2012-12-16 16:29:52 +00:00
zeeshan.ali@nokia.com	1c799ce964	rtpmux: Use own clock-base 20070328112219-65035-1ba5fefbc65059e9b0c860528a31062ceb6a7331.gz	2012-12-16 16:29:48 +00:00
zeeshan.ali@nokia.com	b04630d7a2	rtpmux: Only accept RTP streams that have the same clock-rate 20070323163139-65035-fc0b17b0b8a7a041f48994c4f26e96568168bf95.gz	2012-12-16 16:29:45 +00:00
zeeshan.ali@nokia.com	6fe1e02efd	rtpmux: Some more code-cleanups 20070322161552-65035-bda96165e146b4f1d5fea1cc9576a7ab3abebc9e.gz	2012-12-16 16:29:42 +00:00
zeeshan.ali@nokia.com	1603223ee5	rtpmux: return newpad instead of NULL and warn if failed to create a pad 20070322154251-65035-cdb6651e61c2eb0205cc8c24693b43f98a2da718.gz	2012-12-16 16:29:38 +00:00
zeeshan.ali@nokia.com	23d3ed5c5f	rtpmux: Refactorize the RTPMux code 20070322124132-65035-0a3278147546e33f687097a43b775b3f6aa99f93.gz	2012-12-16 16:29:35 +00:00
zeeshan.ali@nokia.com	21e6e951f6	rtpmux: Some more doc fixing 20070322121453-65035-12d602272217b51bd97df4e5790024c399622dd3.gz	2012-12-16 16:29:32 +00:00
zeeshan.ali@nokia.com	0de7fb6f37	rtpmux: More Refactoring 20070322113228-65035-bae34a79599e7de5293ed77b022361ccff822bb9.gz	2012-12-16 16:29:29 +00:00
zeeshan.ali@nokia.com	0f755657ce	rtpmux: More documentation 20070322113154-65035-624850541a5b5fc3df231204be5a83d07239db28.gz	2012-12-16 16:29:26 +00:00
zeeshan.ali@nokia.com	5483c78ac0	rtpmux: Refactor the event handler function 20070321163311-65035-987e7f25d1ab5335b79f44b277abf15e4e37d317.gz	2012-12-16 16:29:23 +00:00
zeeshan.ali@nokia.com	db1523ae60	rtpmux: Add RTPDTMFMux element 20070321145244-65035-9a01390b0dee3398e53199a1fa1d9352004f338e.gz	2012-12-16 16:29:19 +00:00
zeeshan.ali@nokia.com	97ff54dce7	rtpmux: Remove DTMF-specific code from RTP muxer and make it extendable 20070321123149-65035-b8a8f55ff78eed8cbb0042e827885edfc5438242.gz	2012-12-16 16:29:16 +00:00
zeeshan.ali@nokia.com	1a227ac7e5	rtpmux: Put more helpful description 20070320120524-65035-db27a7cf6307b511aeb3d996d26e790e367a7bad.gz	2012-12-16 16:29:13 +00:00
zeeshan.ali@nokia.com	d876c0d8cc	rtpmux: remove the (commented-out) code for blocking the pads 20070316151641-65035-0123af387951f88594797c722e882cfe70240aff.gz	2012-12-16 16:29:10 +00:00
zeeshan.ali@nokia.com	209228c44d	rtpmux: Drop buffers instead of blocking the sinkpads 20070316131444-65035-9c1345ad96108881f455d4b55a7f623cd302d0ed.gz	2012-12-16 16:29:05 +00:00
zeeshan.ali@nokia.com	795822ffa5	rtpmux: Implement stream locking, needed for DTMF 20070314171618-65035-e4d24b1606ce0a3e2e739f01833f61e4d7555eac.gz	2012-12-16 16:29:02 +00:00
zeeshan.ali@nokia.com	fd209faa56	rtpmux: use GST__OBJECT instead of g_ 20070314102058-65035-e2442888f2e3e5a3a7659ad7954a4fba34749ce2.gz	2012-12-16 16:28:58 +00:00
zeeshan.ali@nokia.com	b0208cb0a6	rtpmux: No need to manage pads, parent does that for us 20070314101854-65035-ef5f4abde227102a1128835ab325905eae4c3726.gz	2012-12-16 16:28:55 +00:00
zeenix@gmail.com	74e9071dad	rtpmux: Fix copyright header 20070314090358-d014a-3a6d3eeeaaf5cb8ca3bca6a33e99a551f598bd48.gz	2012-12-16 16:28:51 +00:00
zeeshan.ali@nokia.com	3c4cdf1541	rtpmux: The first implementation of RTP muxer 20070307085307-65035-833402413f99cb3f8be4883e92bad4c8722510c9.gz	2012-12-16 16:28:41 +00:00
Havard Graff	9c94f1187c	jitterbuffer: bundle together late lost-events The scenario where you have a gap in a steady flow of packets of say 10 seconds (500 packets of with duration of 20ms), the jitterbuffer will idle up until it receives the first buffer after the gap, but will then go on to produce 499 lost-events, to "cover up" the gap. Now this is obviously wrong, since the last possible time for the earliest lost-events to be played out has obviously expired, but the fact that the jitterbuffer has a "length", represented with its own latency combined with the total latency downstream, allows for covering up at least some of this gap. So in the case of the "length" being 200ms, while having received packet 500, the jitterbuffer should still create a timeout for packet 491, which will have its time expire at 10,02 seconds, specially since it might actually arrive in time! But obviously, waiting for packet 100, that had its time expire at 2 seconds, (remembering that the current time is 10) is useless... The patch will create one "big" lost-event for the first 490 packets, and then go on to create single ones if they can reach their playout deadline. See https://bugzilla.gnome.org/show_bug.cgi?id=667838	2012-12-13 12:00:43 +01:00
Wim Taymans	c755af0cb0	rtpsource: protect against invalid RTP packets	2012-11-12 11:18:30 +01:00
Tim-Philipp Müller	230cf41cc9	Fix FSF address https://bugzilla.gnome.org/show_bug.cgi?id=687520	2012-11-04 00:07:18 +00:00
Wim Taymans	d5fd524a0c	rtsession: fix compiler warning	2012-10-17 13:55:45 +02:00
Wim Taymans	26a21e85e2	rtpbin: clarify the ntp-sync option	2012-10-17 13:35:07 +02:00
Wim Taymans	f17db5c4ed	rtpsession: update caps in the source Inform the source when caps changed. This was removed in the port to 1.0 leaving the source unaware of the clock-rate and unable to interpollate rtp timestamps for SR packets.	2012-10-17 13:22:40 +02:00
Wim Taymans	f4eef3f48d	rtpbin: set PTS and DTS in jitterbufffer	2012-10-17 12:46:32 +02:00
Wim Taymans	796c1d8029	rtpbin: disable check for ntp-sync Disable the check for the ntp-sync method. It is expected that a rather larger offset needs to be applied with this method.	2012-10-17 12:27:03 +02:00
Wim Taymans	1cebcfa8c2	rtpbin: use running-time for NTP time When use-pipeline-clock is set, use the running-time of the pipeline to calculate the NTP timestamps. This method would previously only work when the base-time is set to 0 but with this change it can also work with different offsets and we can also implement pause/resume of the sender and receiver now.	2012-10-17 12:26:05 +02:00
Wim Taymans	5b394385b9	session: also stop probatation on existing sources Receiving an RTCP packet should also stop probation on sources we have seen before. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=683065	2012-08-30 22:07:24 +02:00
Aleix Conchillo Flaque	4a200b670f	rtp: make rtp packet probation configurable (bug #682512 )	2012-08-30 21:49:57 +02:00
Tim-Philipp Müller	4bb52bbadf	docs: gst-launch -> gst-launch-1.0 and ffmpegcolorspace -> videoconvert	2012-08-27 21:20:30 +01:00
Aleix Conchillo Flaque	8d864dbbfc	rtspsrc: make jitterbuffer drop-on-latency available (fix #682055 ) Conflicts: gst/rtsp/gstrtspsrc.h	2012-08-22 10:39:19 +02:00
Tim-Philipp Müller	787c314ec3	Silence some 'variable may be used uninitialized' compiler warnings When compiling with -DG_DISABLE_ASSERT	2012-08-08 11:31:59 +01:00
Olivier Crête	2e21ace12c	rtpssrcdemux: Block pad while it is announced. Block the RTP pad and associated RTCP pads while they are being announced. This it to prevent a race where one is announced and before the callback has connected it, the other one gets a buffer. We can't use the "padlock" of ssrcdemux because it causes deadlocks.	2012-08-06 18:04:58 -07:00
Olivier Crête	2aa360c936	rtpssrcdemux: Release lock before signalling new pad This prevents a deadlock where something would try to push an event through the SSRC demux from the callback, causing the pads to be iterated and the lock taken.	2012-08-04 18:14:28 -07:00
Wim Taymans	51371d26ee	update for RTP buffer api changes	2012-07-17 16:38:27 +02:00
Tim-Philipp Müller	c22268b5d3	rtpsession: remove deprecated and unused "ntp-ns-base" property	2012-07-06 13:16:00 +01:00
Wim Taymans	30d3dfee36	update for task api change	2012-06-20 10:33:42 +02:00
Wim Taymans	dc04908412	update for clock api changes	2012-06-20 10:01:57 +02:00
Pascal Buhler	8161daef4a	rtpsession: creation should be signaled before validation https://bugzilla.gnome.org/show_bug.cgi?id=667850	2012-05-09 10:36:18 +02:00
Tim-Philipp Müller	e09ae5736d	Use new gst_element_class_set_static_metadata()	2012-04-10 00:51:41 +01:00
Sebastian Dröge	aa2cd462da	gst: Update for GST_PLUGIN_DEFINE() API changes	2012-04-05 17:36:38 +02:00
Sebastian Dröge	5cdd49bf25	gst: Update versioning	2012-04-04 14:37:47 +02:00
Mark Nauwelaerts	a34cbc7637	rtpbin: fix some lock management ... to avoid trying to take a non-recursive lock twice.	2012-03-26 18:38:34 +02:00
Wim Taymans	7f3a00decd	jitterbuffer: reply FALSe on serialized queries	2012-03-14 15:45:38 +01:00
Wim Taymans	af59f573b5	rtpsession: don't leak the address	2012-03-13 19:26:47 +01:00
Wim Taymans	b5f1969406	rtpbin: improve cleanup Reuse cleanup methods to make sure we remove all pads correctly	2012-03-07 15:22:36 +01:00
Wim Taymans	9942d3566e	rtpsession: set caps without the lock Release the lock before setting the caps on the srcpad, which triggers an event, which could eventually call back into us and cause a deadlock.	2012-03-07 15:02:44 +01:00
Wim Taymans	5cce960baa	ptdemux: set caps after activating the pad Set the caps after we activated the pad or else it will just fail.	2012-03-07 15:02:44 +01:00
Mark Nauwelaerts	f189f62b13	Merge branch 'master' into 0.11 Conflicts: ext/wavpack/gstwavpackenc.c tests/check/elements/audioiirfilter.c tests/examples/v4l2/probe.c	2012-03-01 11:29:50 +01:00
Edward Hervey	9beda57c3a	Suppress deprecation warnings in selected files, for g_value_array_* mostly	2012-02-27 14:47:25 +01:00
Tim-Philipp Müller	979431c034	rtpjitterbuffer: declare variables at the beginning of the block It's how we roll. Fixes 'ISO C90 forbids mixed declarations and code' compiler warning.	2012-02-16 11:21:28 +00:00
Wim Taymans	225e98d623	Merge branch 'master' into 0.11 Conflicts: ext/flac/gstflacenc.c ext/jack/gstjackaudioclient.c ext/jack/gstjackaudiosink.c ext/jack/gstjackaudiosrc.c ext/pulse/plugin.c ext/shout2/gstshout2.c gst/matroska/matroska-mux.c gst/rtp/gstrtph264pay.c	2012-02-10 16:23:14 +01:00
Wim Taymans	9365f12d6e	GST_FLOW_WRONG_STATE -> GST_FLOW_FLUSHING	2012-02-08 16:43:30 +01:00
Wim Taymans	ed8c0b7f63	jitterbuffer: fix caps after pt change	2012-02-06 09:23:07 +01:00
Wim Taymans	c94c06530e	jitterbuffer: fix caps leak	2012-02-06 09:18:17 +01:00
Tim-Philipp Müller	0f3b7b010e	build: ignore GValueArray deprecation warnings for the time being until this gets sorted out with the GLib folks and we have a viable alternative. https://bugzilla.gnome.org/show_bug.cgi?id=667228	2012-02-01 16:40:51 +00:00
Olivier Crête	87f2088303	rtpjitterbuffer: Don't leak caps event when not pushing	2012-01-27 19:05:24 +01:00
Olivier Crête	33a6d1921f	rtpptdemux: Forward sticky events	2012-01-27 19:05:24 +01:00
Olivier Crête	7b1f8cb8f0	rtpptdemux: Protect all uses pad list with OBJECT LOCK Actually protect the entire pad list and use it in a thread safe way.	2012-01-27 19:05:24 +01:00
Olivier Crête	b3f5cdd1f9	rtpssrcdemux: Forward sticky events to new pads	2012-01-27 19:05:24 +01:00
Olivier Crête	76c93af537	rtpssrcdemux: Add ssrc to forwarded CAPS events Also iterate the list of GstRtpSsrcDemuxPad safely	2012-01-27 19:05:23 +01:00
Olivier Crête	3285c45dbc	rtpssrccdemux: Factor out getting dpad by pad	2012-01-27 19:05:23 +01:00
Olivier Crête	b850741430	rtpsession: Keep the buffer mapped while it is being modified	2012-01-27 19:05:23 +01:00
Olivier Crête	aeec2d5f7e	rtpsession: Initialise the address pointer to NULL	2012-01-27 19:05:23 +01:00
Tim-Philipp Müller	5525e40970	rtpmanager: don't pretend our random hostnames are fully-qualified domain names	2012-01-25 13:19:12 +00:00
Sebastian Dröge	0b517ce9fb	Merge branch '0.11' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-good into 0.11	2012-01-25 12:49:34 +01:00
Sebastian Dröge	10554b271f	Merge branch 'master' into 0.11 Conflicts: ext/flac/gstflacdec.c ext/jpeg/gstjpegenc.c ext/pulse/pulsesink.c sys/v4l2/gstv4l2src.c	2012-01-25 12:49:11 +01:00
Wim Taymans	b4630dd3e0	more memory API porting	2012-01-25 12:30:29 +01:00
Tim-Philipp Müller	a476d529d2	rtpmanager: don't reveal the user's username, hostname or real name by default Send a randomly made-up user@hostname as CNAME and don't send a NAME at all by default. https://bugzilla.gnome.org/show_bug.cgi?id=668320	2012-01-23 13:47:08 +00:00
Tim-Philipp Müller	7cb9b7ab9d	Use new GLib API unconditionally	2012-01-22 23:15:19 +00:00
Mark Nauwelaerts	eff88a239f	rtpbin: arrange for initialized variables	2012-01-20 17:10:51 +01:00
Wim Taymans	1584806634	port to new gthread API	2012-01-19 11:33:53 +01:00
Sebastian Dröge	cb789e32ad	rtpmanager: Port to GIO	2012-01-17 13:08:42 +01:00
Tim-Philipp Müller	f10e8192fa	rtpptdemux: plug pad leak in error code path Based on patch by: Stig Sandnes <stig.sandnes@cisco.com> Don't leak srcpad if there are no caps. https://bugzilla.gnome.org/show_bug.cgi?id=667820	2012-01-13 11:02:24 +00:00
Vincent Penquerc'h	654a04f90c	gstrtpssrcdemux: fix element leak	2012-01-12 18:23:42 +00:00
Sebastian Dröge	93e3ed5a86	Merge branch 'master' into 0.11 Conflicts: ext/cairo/gsttextoverlay.c ext/pulse/pulseaudiosink.c gst/audioparsers/gstaacparse.c gst/avi/gstavimux.c gst/flv/gstflvmux.c gst/interleave/interleave.c gst/isomp4/gstqtmux.c gst/matroska/matroska-demux.c gst/matroska/matroska-mux.c gst/matroska/matroska-mux.h gst/matroska/matroska-read-common.c gst/multifile/gstmultifilesink.c gst/multipart/multipartmux.c gst/shapewipe/gstshapewipe.c gst/smpte/gstsmpte.c gst/udp/gstmultiudpsink.c gst/videobox/gstvideobox.c gst/videocrop/gstaspectratiocrop.c gst/videomixer/videomixer.c gst/videomixer/videomixer2.c gst/wavparse/gstwavparse.c po/ja.po po/lv.po po/sr.po tests/check/Makefile.am tests/check/elements/qtmux.c tests/check/elements/rgvolume.c	2012-01-10 14:32:32 +01:00
Wim Taymans	5fd2b7abe3	GST_FLOW_UNEXPECTED -> GST_FLOW_EOS	2012-01-03 15:26:21 +01:00
Tim-Philipp Müller	b8b8454bcb	Suppress deprecation warnings in selected files, for g_static_rec_mutex_* mostly GStaticRecMutex is part of our API/ABI, not much we can do here in 0.10 for most of these.	2011-12-12 09:46:27 +00:00
Tim-Philipp Müller	330d984288	Use g_thread_try_new() instead of g_thread_crate() with newer glib versions	2011-12-12 09:46:27 +00:00
Tim-Philipp Müller	66f6e12888	Work around deprecated thread API in glib master Add private replacements for deprecated functions such as g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly to avoid the deprecation warnings. We'll change these over to the new API once we depend on glib >= 2.32.	2011-12-12 09:46:27 +00:00
Wim Taymans	9e27b122d9	ssrcdemux: fix iterator and caps	2011-12-10 11:13:38 +01:00
Wim Taymans	da980884dd	rtpsession: forward the caps event	2011-12-10 11:13:38 +01:00
Wim Taymans	a705b2ec17	jitterbuffer: simply forward the caps event forward the caps event we get as input instead of making a new event etc..	2011-12-10 11:13:38 +01:00
Wim Taymans	68588c3f18	rtpsession: forward caps	2011-12-10 11:13:38 +01:00
Wim Taymans	6ac5e1ae16	rtp: pass parent to setcaps methods	2011-12-10 11:13:38 +01:00
Wim Taymans	439e2f1cfd	rtp: fix marshallers Remove custom marshallers for minobject. Init RTCP buffer correctly. Handle results from setcaps Remove asserts.	2011-12-09 10:51:14 +01:00
Edward Hervey	86a57e3546	rtpmanager: Initialize GstRTPBuffer before usage	2011-12-05 18:40:12 +01:00
Wim Taymans	71b615515a	update for clock provider API change	2011-11-28 17:52:06 +01:00
Vincent Penquerc'h	c0e101e93f	various: fix pad template leaks https://bugzilla.gnome.org/show_bug.cgi?id=662664	2011-11-28 13:30:27 +00:00
Tim-Philipp Müller	09ca5fa910	rtpmanager: rename gstrtp* -> rtp* This was done in 0.10 to avoid conflict with the rtp elements in farsight, but the gst-prefixing is no longer needed in 0.11	2011-11-24 00:54:08 +00:00
Matej Knopp	1e5dd9e315	Fix printf format compiler warnings on OS X / 64bit https://bugzilla.gnome.org/show_bug.cgi?id=662615	2011-11-22 01:28:22 +00:00
Wim Taymans	f8e988a94c	update for activation changes	2011-11-21 13:37:01 +01:00
Wim Taymans	b7aa7bca52	add parent to activate functions	2011-11-18 13:57:20 +01:00
Wim Taymans	07cc855b24	Merge branch 'master' into 0.11 Conflicts: ext/speex/gstspeexenc.c gst/rtpmanager/rtpsession.c	2011-11-17 17:17:11 +01:00
Wim Taymans	105650127e	add parent to pad functions	2011-11-17 15:02:55 +01:00
Wim Taymans	7cc4b72550	add parent to internal links	2011-11-16 17:54:49 +01:00
Wim Taymans	6190312214	add parent to query function	2011-11-16 17:27:13 +01:00
Wim Taymans	797523efbd	_peer_get_caps() -> _peer_query_caps()	2011-11-15 18:04:44 +01:00
Wim Taymans	75dc9634eb	change getcaps to query Chain up event function in payloaders.	2011-11-15 18:04:44 +01:00
Olivier Crête	1169bb05af	gstrtpsession: Add special mode to use FIR as repair as Google does https://bugzilla.gnome.org/show_bug.cgi?id=658419	2011-11-14 12:26:27 +01:00
Olivier Crête	79a9564c68	rtpsession: Send FIR requests in response to key unit requests with all-headers=TRUE https://bugzilla.gnome.org/show_bug.cgi?id=658419	2011-11-14 12:26:27 +01:00
Olivier Crête	12a6b9613b	rtpsession: Put the PLI requests in each RTPSource Also refactor a bit and put all the keyframe request code in one place inside rtpsession.c https://bugzilla.gnome.org/show_bug.cgi?id=658419	2011-11-14 12:26:27 +01:00
Olivier Crête	59c028a4ce	rtpsession: Hack to FIR because Google doesn't set the sender ssrc correctly https://bugzilla.gnome.org/show_bug.cgi?id=658419	2011-11-14 12:26:27 +01:00
Olivier Crête	0ad78db0a3	rtpsession: Process received Full Intra Requests Process FIR requests according to RFC 5104 https://bugzilla.gnome.org/show_bug.cgi?id=658419	2011-11-14 12:26:27 +01:00
Wim Taymans	a19a4a69ae	more template fixes	2011-11-04 13:12:37 +01:00
Wim Taymans	a95acb7122	make %u in all request pad templates	2011-11-04 11:58:22 +01:00
Wim Taymans	6cbd6afc0b	update for new net library	2011-11-03 16:43:00 +01:00
Wim Taymans	83ccefb24e	update for netbuffer api change	2011-11-02 09:06:38 +01:00
Wim Taymans	75e0c6052f	update for netaddress change	2011-11-02 09:06:38 +01:00
Wim Taymans	9a8a8e72c8	structure: fix for api update	2011-11-02 09:06:37 +01:00
Wim Taymans	161310fa23	bufferlist: update for new API	2011-11-02 09:06:37 +01:00
Tim-Philipp Müller	d18a578ba4	rtpmanager, v4l2: fix compiler warnings after gst_caps_new_simple() change	2011-10-28 09:06:41 +01:00
Wim Taymans	fc4684f4c6	fix compilation	2011-10-27 16:03:17 +02:00
Edward Hervey	d4a2a46606	rtpssrcdemux: Fix wrong usage of gst_iterator_filter It takes a GValue* as the user_data. And don't forget to unref the demuxer before returning.	2011-10-13 09:34:04 +02:00
Wim Taymans	87fbd1e784	Merge branch 'master' into 0.11 Conflicts: common ext/pulse/pulsesink.c ext/soup/gstsouphttpclientsink.c gst/audioparsers/gstaacparse.c gst/audioparsers/gstac3parse.c gst/rtp/gstrtph264depay.c gst/rtpmanager/gstrtpjitterbuffer.c gst/rtpmanager/rtpjitterbuffer.c gst/rtsp/gstrtspsrc.c sys/ximage/gstximagesrc.c	2011-09-28 12:44:59 +02:00
Ha Nguyen	931020158e	rtpbin: Fix a leaked clock for each buffering message Fixes bug #659237.	2011-09-19 14:05:26 +02:00
Mark Nauwelaerts	e2179cbb74	rtpsession: avoid source premature timing out Use slightly adjusted sender interval to determine sender timeout rather than our own sender side interval (which may have been forced small).	2011-09-19 11:56:44 +02:00
Mark Nauwelaerts	f65d4c8300	rtpsession: avoid timing out source too quickly ... following a PAUSE/PLAY cycle, particularly applicable when operating with a short RTCP interval (possibly forced so server-side).	2011-09-19 11:56:44 +02:00
Mark Nauwelaerts	77ebd33991	rtpjitterbuffer/rtpbin: relax dropping rtcp packets ... to at least having it trigger a/v synchronization, possibly without using provided values which are still not considered sane (as previously dropped).	2011-09-19 11:56:44 +02:00
Mark Nauwelaerts	adfe7d0467	rtpjitterbuffer: some more reset when clearing pt map ... which in particular caters for some more reset following a possible rtsp PLAY.	2011-09-19 11:56:44 +02:00
Mark Nauwelaerts	915db26029	rtpjitterbuffer: only reset skew on gap if input ts available	2011-09-19 11:56:44 +02:00
Mark Nauwelaerts	1e17e10f75	rtpjitterbuffer: check some more for possible rtp timestamp discontinuity ... when operating in non slave mode, and reset if detected. This should avoid some (large) bogus outgoing timestamp due to jumps in rtp time, as result of PAUSE/PLAY or seek or ...	2011-09-19 11:56:40 +02:00
Mark Nauwelaerts	9c95072048	rtpbin: alternative inter-stream syncing methods ... at least if not syncing to NPT time: * either sync using RTCP SR data (as currently) * only perform the above once using initial RTCP SR packets * discard RTCP and sync by equating provided stream's clock-base rtptime, as provided by jitterbuffer (typically obtained from RTP-Info in RTSP).	2011-09-19 11:52:03 +02:00
Mark Nauwelaerts	4b7301e4d1	rtpjitterbuffer: also provide clock-base to sync signal	2011-09-19 11:52:00 +02:00
Mark Nauwelaerts	f29c253934	rtpbin: allow configurable rtcp stream syncing interval ... rather than necessarily syncing at each RTCP SR.	2011-09-19 11:51:57 +02:00
Mark Nauwelaerts	afd26f0078	rtpsession: trigger reconsideration if rtcp interval set	2011-09-19 11:51:50 +02:00
Wim Taymans	33f18b8ea4	Merge branch 'master' into 0.11 Conflicts: gst/audioparsers/gstamrparse.c gst/isomp4/qtdemux.c	2011-09-06 16:06:25 +02:00
Olivier Crête	b2e8362767	rtpsession: Initialise the last_keyframe_request variable	2011-09-02 19:24:46 -04:00
Wim Taymans	4121021bb2	Merge branch 'master' into 0.11 Conflicts: ext/pulse/pulsesink.c ext/pulse/pulsesrc.c gst/audioparsers/gstac3parse.c gst/rtp/gstrtph264depay.c gst/rtp/gstrtph264pay.c gst/rtpmanager/gstrtpssrcdemux.c	2011-08-03 18:25:30 +02:00
Mark Nauwelaerts	c03648c8bb	rtpsession: properly init rtcp_min_interval	2011-07-29 12:08:42 +02:00
Mark Nauwelaerts	3a98f6f0fd	rtpssrcdemux: keep a ref on the src pad while using it Prevent a possible race if clear_ssrc() is called between getting the pad and doing the push. Based on patch by <olivier.crete@collabora.com> https://bugzilla.gnome.org/show_bug.cgi?id=650916	2011-07-28 14:51:01 +02:00
Olivier Crête	c7b9b98648	rtpssrcdemux: Make the pads lock recursive and hold it across the signal emit We need to keep the lock held because we don't want a push before the "new-ssrc-pad" handler has completed. But we may want to push an event from inside that handler, hence the recursive mutex. https://bugzilla.gnome.org/show_bug.cgi?id=650916	2011-07-28 14:50:59 +02:00
Olivier Crête	e26b5391c2	rtpssrcdemux: Use PADs lock https://bugzilla.gnome.org/show_bug.cgi?id=650916	2011-07-28 14:50:57 +02:00
Olivier Crête	6095d2a3f0	rtpsession: Always send application requested feedback in immediate mode Send as many application requested feedback messages in immediate mode, even if they have already been sent. https://bugzilla.gnome.org/show_bug.cgi?id=654583	2011-07-25 17:20:59 +02:00
Olivier Crête	354faabda0	rtpsession: Don't let the computed RTP bandwidth fall too low If it falls too low, the computed RTCP bandwidth will be near zero and the RTCP thread will be stopped. https://bugzilla.gnome.org/show_bug.cgi?id=654583	2011-07-25 16:19:00 +02:00
Olivier Crête	4d48109f9d	rtpsession: Wait longer to timeout SSRC collision Using the current RTCP interval to timeout SSRC collision can lead to collisions being timed out immediately if a BYE packet is sent because it is sent immediately, so the interval is 0. This is not what we want. So just set a static 10 times the default RTCP interval, it should be enough https://bugzilla.gnome.org/show_bug.cgi?id=648642	2011-07-25 16:18:58 +02:00
Mark Nauwelaerts	ef02634dc6	rtpmanager: port to 0.11 * use G_DEFINE_TYPE * adjust to new GstBuffer and corresponding rtp and rtcp buffer interfaces * misc caps and segment handling changes FIXME: also relies on being able to pass caps along with a buffer, which has no evident equivalent yet, so that either needs one, or still needs quite some code path modification to drag along caps.	2011-07-06 10:16:12 +02:00
Mark Nauwelaerts	d59a00aa1c	Merge branch 'master' into 0.11 Conflicts: ext/pulse/pulsesink.c	2011-07-04 11:48:13 +02:00
Miguel Angel Cabrera Moya	977a5eee7a	rtpjitterbuffer: return correct type when assertion fails	2011-06-24 11:59:01 +02:00
Wim Taymans	cc65bff7c1	Merge branch 'master' into 0.11 Conflicts: configure.ac docs/plugins/inspect/plugin-esdsink.xml docs/plugins/inspect/plugin-gconfelements.xml	2011-06-21 18:24:41 +02:00
Olivier Crête	581a30d892	rtpsession: The signal has 5 arguments, not 4	2011-06-20 16:47:36 -04:00
Wim Taymans	409f29700d	-good: port some more plugins	2011-06-13 17:51:40 +02:00
Wim Taymans	e15651816e	Merge branch 'master' into 0.11	2011-05-17 16:13:59 +02:00
Sebastian Dröge	b694bfeca3	ssrcdemux: Fix uninitialized variable compiler warning for (pre-) releases too	2011-05-17 10:47:32 +02:00
Sebastian Dröge	0f05d3e5a5	rtpssrcdemux: Fix uninitialized variable compiler warning	2011-05-17 09:24:08 +02:00
Olivier Crête	b6bfc512e8	ssrcdemux: Implement iterate internal links for sink pads https://bugzilla.gnome.org/show_bug.cgi?id=649617	2011-05-17 09:22:29 +02:00
Olivier Crête	23b6c8febc	rtpssrcdemux: iterate pad function is only valid for src pads The iterate function is only used for src pads, so mark it as such and remove dead code. https://bugzilla.gnome.org/show_bug.cgi?id=649617	2011-05-17 09:22:25 +02:00
Olivier Crête	1bf94a92b0	rtpssrcdemux: Release lock before emitting signal If the lock is not released before emitting a signal, it may cause a deadlock if any other function in the element is called. Also removed an unused timestamp parameter https://bugzilla.gnome.org/show_bug.cgi?id=649617	2011-05-17 09:22:20 +02:00
Wim Taymans	a1894ed363	Merge branch 'master' into 0.11	2011-04-25 11:38:28 +02:00
Olivier Crête	42531337f5	rtpsession: Remove incomplete support for RTCP FIR Remove bits that were meant to suppport RTCP FIR https://bugzilla.gnome.org/show_bug.cgi?id=648160	2011-04-20 07:50:43 +01:00
Wim Taymans	7555d0949f	Merge branch 'master' into 0.11 Conflicts: android/apetag.mk android/avi.mk android/flv.mk android/icydemux.mk android/id3demux.mk android/qtdemux.mk android/rtp.mk android/rtpmanager.mk android/rtsp.mk android/soup.mk android/udp.mk android/wavenc.mk android/wavparse.mk configure.ac	2011-04-18 10:23:45 +02:00
Robert Swain	5b18c652fb	rtp, rtpmanager: Address unused but set variables GCC 4.6.x spits warnings about variables that are unused but set. Such variables have been removed where trivial but with comments left behind for informational purposes in some cases. gst_rtp_session_chain_recv_rtcp () was changed in commit `490113d4` to always return GST_FLOW_OK instead of the return value of rtp_session_process_rtcp (), so we'll keep it that way.	2011-04-16 12:49:16 +01:00
Olivier Crête	9d9257916b	rtpsession: Use existing functions to parse RTCP FB packets Use existing functions to get the FCI from FB packets. https://bugzilla.gnome.org/show_bug.cgi?id=622553	2011-04-15 12:48:04 +01:00
Olivier Crête	5ccd964d86	rtpsession: marshal GstBuffer as a MiniObject instead of a pointer https://bugzilla.gnome.org/show_bug.cgi?id=622553	2011-04-15 12:47:40 +01:00
Pascal Buhler	0d2d52856f	rtpssrcdemux: Unknown SSRC is not fatal https://bugzilla.gnome.org/show_bug.cgi?id=646966	2011-04-11 17:37:58 -04:00
Pascal Buhler	58ef84846e	rtpsession: Number of active sources should be updated whenever the status of the source changes to active Forward-ported by Olivier Crête https://bugzilla.gnome.org/show_bug.cgi?id=646965	2011-04-11 17:37:36 -04:00
Havard Graff	53c88ae33e	rtpmanager: ignore a BYE if it is sent with our internal SSRC https://bugzilla.gnome.org/show_bug.cgi?id=646964	2011-04-11 17:34:12 -04:00
Thibault Saunier	b541208b77	android: Make it ready for androgenizer Remove the android/ top dir Fixe the Makefile.am to be androgenized To build gstreamer for android we are now using androgenizer which generates the needed Android.mk files. Androgenizer can be found here: http://git.collabora.co.uk/?p=user/derek/androgenizer.git	2011-04-11 01:20:11 +02:00
Havard Graff	93f022d6ab	rtpsession: fix wrongly applied patch Obviously recv_rtp_sink does not have much to do with send_rtcp_src... See commit 046ff170. https://bugzilla.gnome.org/show_bug.cgi?id=647263	2011-04-09 12:32:37 +01:00
Havard Graff	e71a908d96	jitterbuffer: Make src_query MT-safe It is possible that the element might be going down while the event arrives	2011-04-08 15:23:05 +02:00
Sebastian Dröge	4c36ca30b2	jitterbuffer: Unref event if the parent element disappeared	2011-04-08 15:22:19 +02:00
Havard Graff	342686bb02	jitterbuffer: Make upstream events MT-safe	2011-04-08 15:21:46 +02:00
Sebastian Dröge	31af4fe33e	rtp: Unref events if the parent element disappeared	2011-04-08 15:20:51 +02:00
Ole André Vadla Ravnås	046f170d6a	rtpmanager: fix pad callbacks so they handle when parent goes away 1) We need to lock and get a strong ref to the parent, if still there. 2) If it has gone away, we need to handle that gracefully. This is necessary in order to safely modify a running pipeline. Has been observed when a streaming thread is doing a buffer_alloc() while an application thread sends an event on a pad further downstream, and from within a pad probe (holding STREAM_LOCK) carries out the pipeline plumbing while the streaming thread has its buffer_alloc() in progress.	2011-04-08 15:16:56 +02:00
Havard Graff	f8370bb2a8	rtpsession: make iterate_internal_links MT-safe	2011-04-08 14:41:34 +02:00
Wim Taymans	c124ba1489	Merge branch 'master' into 0.11 Conflicts: gst/rtsp/gstrtspsrc.c	2011-04-05 17:20:08 +02:00
Mark Nauwelaerts	e5bcaa45e6	Revert "jitterbuffer: reset element base_time upon flush" This reverts commit `f84b8a69cb`. Fixes bug #646397.	2011-04-04 11:49:00 +02:00
Wim Taymans	8f22a09dc4	Merge branch 'master' into 0.11-fdo	2011-03-28 20:50:59 +02:00
Mark Nauwelaerts	6bc1aa0e59	jitterbuffer: handle position query	2011-03-09 17:18:08 +01:00
Wim Taymans	a4fdb8ee44	Merge branch 'master' into 0.11 Conflicts: tests/examples/cairo/Makefile.am	2011-03-08 10:14:20 +00:00
Mark Nauwelaerts	1f7f434df6	jitterbuffer: also estimate eos if very near eos	2011-03-07 16:56:43 +01:00
Mark Nauwelaerts	3c9a4239bf	jitterbuffer: avoid trying to buffer more than is available. That is, in case of short (or near eos of) stream, deadlock (until timeout) would occur trying to buffer more than is yet forthcoming.	2011-03-07 16:56:18 +01:00
Mark Nauwelaerts	f84b8a69cb	jitterbuffer: reset element base_time upon flush ... to arrange for properly scheduled timeout (following seek).	2011-03-07 11:07:12 +01:00
Wim Taymans	0a56b25882	rtpsession: use NetAddress metadata	2011-02-28 13:28:29 +01:00
Wim Taymans	d87c27fd2c	miniobject: use buffer private field for extra data Use the owner private field to store extra buffer data instead of using subclassing.	2011-02-28 11:58:48 +01:00
Blaise Gassend	0f88181f43	rtpbin: handle NULL demux elements When using gstrtpbin with ignore-pt=true, the free_stream function tries to call gst_element_set_locked_state and gst_element_set_state on a stream->demux which is NULL. fixes #642412	2011-02-22 13:31:35 +01:00
Wim Taymans	45ea930a99	rtpbin: fix setting the SDES property Only the sdes veriable is protected with the object lock. Use the right object when setting the sdes property.	2011-02-21 17:19:05 +01:00
Wim Taymans	61382aad28	source: fix type of ntpnstime	2011-02-02 18:30:47 +01:00
Wim Taymans	8598aaf81b	rtpbin: Get and use the NTP time when receiving RTCP When we receive an RTCP packet, get the current NTP time in nanseconds so that we can correctly calculate the round-trip time.	2011-02-02 18:30:46 +01:00
Olivier Crête	cd923223dd	rtpsession: Add action signal to request early RTCP	2011-02-01 18:28:51 +01:00
Olivier Crête	c0996e6b90	rtpsession: Add callback to get the current time	2011-02-01 18:28:51 +01:00
Olivier Crête	a630c68fc3	rtpsession: Don't relay more than one PLI request per RTT Drop PLI requests if one was relay in the last RTT, the other side may just not have received the keyframe yet.	2011-02-01 18:28:51 +01:00
Olivier Crête	a61bb9e94b	rtpsession: Send GstForceKeyUnit event in response to received RTCP PLI	2011-02-01 18:28:51 +01:00
Sjoerd Simons	7350d2adfa	gstrtpsession: Fallback for FIR to PLI if PLI isn't available	2011-02-01 18:28:51 +01:00
Olivier Crête	52f95fa7ee	rtpsession: Implement sending PLI packets in response to GstForceKeyUnit	2011-02-01 18:28:51 +01:00
Olivier Crête	db5150a23a	rtpsource: Retain RTCP Feedback packets for a specified amount of time	2011-02-01 18:28:51 +01:00
Olivier Crête	90354ecb49	rtpsession: Make rtcp buffer metadata writable after processing it Functions that process the rtcp buffer could decide to keep a ref on the buffer for further processing. So make the metadata writable only after they are done.	2011-02-01 18:28:50 +01:00
Olivier Crête	1643f427db	rtpsession: Emit signal on incoming RTCP FB packet	2011-02-01 18:28:50 +01:00
Wim Taymans	f399b6a641	rtpsession: fix compilation	2011-02-01 18:28:50 +01:00
Olivier Crête	1bde427250	rtpsession: Add method to request early RTCP packet Implement the early mode defined in RFC 4585. In this mode, RTCP feedback packets are sent early to notifier.	2011-02-01 17:03:39 +01:00
Olivier Crête	975e1fecb3	rtpsession: Add property for minimum interval between Regular RTCP messages This can be changed according to RFC 4585	2011-02-01 16:56:15 +01:00
Olivier Crête	cdb5465741	rtpsession: Emit signal when sending a compound RTCP packet This allows users to add extra RTCP packets to the compound RTCP packet.	2011-02-01 16:50:58 +01:00
Olivier Crête	589b254ce5	rtpptdemux: Tag upstream custom events with payload type	2011-02-01 16:50:25 +01:00
Olivier Crete	c7b1ce7310	rtpssrcdemux: Tag upstream custom events with SSRC	2011-02-01 16:49:10 +01:00
Olivier Crête	9f073459e0	rtpsession: Emit "on-ssrc-validated" when validating by RTCP Emit "on-ssrc-validated" if the SSRC is validated by receiving a RTCP SDES packet.	2011-02-01 16:45:58 +01:00
Stefan Kost	9f34b89245	rtpjitterbuffer: don't divide by 0	2011-01-25 21:57:57 +02:00
Wim Taymans	b5647685c4	rtpsource: use the right variable Use the right variable for specifying that we sent a receiver report.	2010-12-27 13:13:46 +01:00
Wim Taymans	7caad21a57	rtpsource: include last send RB block Only report RB values for non-internal sources. Report not only the RB blocks we last received from but also the last RB block we sent to a source.	2010-12-23 13:58:30 +01:00
Wim Taymans	8fa5ddab9a	rtpsession: remember last sent RB values.	2010-12-23 13:58:30 +01:00
Wim Taymans	6035ee08c0	rtpsource: include all stats and document Include all possible stats of a source in the stats structure because we might be interested in what happened in the past. Document the stats property and the fields.	2010-12-23 13:58:30 +01:00
Wim Taymans	10a5a795ea	rtpsession: also emit RTCP activity on SR Also emit RTCP activity signals when we receive an SR packet without RB blocks, such as from a sender that is not receiving anything.	2010-12-23 13:58:30 +01:00
Wim Taymans	1230258e6f	docs: add some more gstrtpbin docs	2010-12-23 13:58:29 +01:00
Wim Taymans	2b53cbe923	rtpsession: unlock before emitting signals	2010-12-22 11:46:21 +01:00
Wim Taymans	eb6d552353	jitterbuffer: get better buffering level When the jitterbuffer contains -1 timestamps, make sure we still calculate the buffer fill level by skipping the -1 buffers. Try to be more resilient to weird input timestamps.	2010-12-20 15:56:50 +01:00
Wim Taymans	6cb0efede4	jitterbuffer: provide a clock. since we are using the clock for sync, we need to also provide a clock for good measure. The reason is that even if downstream elements provide a clock, we don't want to have that clock selected because it might not be running yet.	2010-12-20 11:13:09 +01:00
Wim Taymans	210f1c44c7	rtpbin: copy buffering stats when we create an aggregate buffering message, copy the buffering stats form the last message. At least we get correct buffering mode then.	2010-12-20 11:13:09 +01:00
Wim Taymans	0c3333da04	session: fix average RTCP packet size some more. Fix stupid error in averaging macro. Include udp headers in packet length estimation.	2010-12-14 18:12:43 +01:00
Wim Taymans	7ebd374766	rtpbin: correctly calculate RTCP packet size	2010-12-14 17:15:23 +01:00
Wim Taymans	ffc7cd9803	jitterbuffer: avoid leaking sink events Avoid leaking the newsegment event when it has the wrong format.	2010-12-13 12:57:58 +01:00
Mark Nauwelaerts	46c91476eb	rtpssrcdemux: do not hold custom PAD_LOCK when pushing downstream	2010-12-03 15:50:21 +01:00
Olivier Crête	077a61932a	rtpbin: Use the right constant to define the "use-pipeline-clock" property The wrong #define was being used, now use the correct one.	2010-10-14 17:41:30 -04:00
Stefan Kost	d8167e3071	various (gst): add a missing G_PARAM_STATIC_STRINGS flags	2010-10-13 18:00:28 +03:00
Tim-Philipp Müller	d65eb2b91a	ext, gst: canonicalise property names where this wasn't the case ie. "foo_bar" -> "foo-bar"	2010-10-12 16:04:21 +01:00
Vladimir Eremeev	8bf7381385	rtpjitterbuffer: improve article reference in comment block https://bugzilla.gnome.org/show_bug.cgi?id=631082	2010-10-01 18:07:03 +01:00
Thijs Vermeir	2c2c90a723	rtpjitterbuffer: update link to documentation	2010-09-30 12:08:49 +02:00
Pascal Buhler	7a8c2a4b8a	rtpmanager: packet lost should not be a warning. It happens all the time...	2010-09-24 16:00:03 +02:00
Pascal Buhler	ca6a512b5e	rtpbin: Make cleaning up sources in rtp_session_on_timeout MT safe Using _foreach_remove on the hashtable, while releasing the lock protecting that table inside the callback is not a good idea. The hashtable might then change (a source removed or added) while signals like on_timeout are being sent. This solution makes a copy of the table, performs the _foreach without actually removing any sources, but marks them for removal on a second iteration with the real list, but this time not letting go of the lock. Fixes #630452	2010-09-24 15:38:00 +02:00
Pascal Buhler	bd8d80a8e4	rtpbin: Handle rysnc of iterator when looking for free pad name If a new pad was added while iterating then a pad could be returned that was already in use. Fixes #630451	2010-09-24 14:10:26 +02:00
Wim Taymans	8337c89c74	rtpsession: fix compilation	2010-09-24 14:10:26 +02:00
Trond Andersen	800b4bdb26	rtpbin: Unlock before adding pad in new_payload_found Holding internal locks while potentially calling out is a source of deadlocks, and in this case the application might subscribe to the pad-added signal. Fixes #630449	2010-09-24 14:00:11 +02:00
Havard Graff	062568a9f5	rtpsession: relax third-party collision detection If the source has been inactive for some time, we assume that it has simply changed its transport source address. Hence, there is no true third-party collision - only a simulated one. Fixes #630447	2010-09-24 13:56:56 +02:00
Wim Taymans	ce007b244e	rtpsource: whitespace fixes	2010-09-24 13:50:02 +02:00
Wim Taymans	c5203a479b	rtpsource: simplify the rate estimation some more	2010-09-24 13:48:50 +02:00
Havard Graff	0fa589a3dd	rtpmanager: provide additional statistics	2010-09-24 13:26:10 +02:00
Wim Taymans	2c8b725591	rtpstats: printf format fixes	2010-09-17 11:07:52 +02:00
Olivier Crête	8e73da10b3	gstrtpsession: Split getting the caps into its own function	2010-09-13 16:25:42 +02:00
Wim Taymans	8e1c9b5b33	rtpbin: small cleanup.	2010-09-13 16:25:42 +02:00
Wim Taymans	d541f5e24d	rtpsession: Small cleanups Make the property description prettier. Actually multiple the bandwidth with the fraction.	2010-09-13 15:51:20 +02:00
Olivier Crête	1f17b334ff	rtpsession: Calculate RTCP bandwidth as a fraction of the RTP bandwidth Calculate the RTCP bandwidth to be a fraction of the RTP bandwidth if it is specified as a value between 0 and 1.	2010-09-13 15:51:20 +02:00
Wim Taymans	8381d9788d	session: improve bandwidth recalculation Also recalculate bandwidth when one of the source bandwidths changed. Use the newly calculated bandwidth.	2010-09-13 15:51:20 +02:00
Olivier Crête	6f53a2b240	rtpsession: Add the option to auto-discover the RTP bandwidth	2010-09-13 15:51:19 +02:00
Thijs Vermeir	f38e37470a	rtpbin: set use-pipeline-clock on correct GObject	2010-09-13 14:39:51 +02:00
Olivier Crête	94e87ef8ee	rtpsession: Initialise the average scaled by 16	2010-09-13 13:10:19 +02:00
Wim Taymans	e6db74764b	rtpsession: add running_time argument docs	2010-09-13 12:41:56 +02:00
Olivier Crête	00fd89c074	rtpstats: Rectify description of current_time in RTPArrivalStats It is the current time, it is unrelated to when the packet was actually received.	2010-09-13 12:37:01 +02:00
Wim Taymans	cb6de429a0	rtpsession: compute the average correctly scaled	2010-09-13 12:31:40 +02:00
Olivier Crête	64e4ffa25b	rtpsession: Count sent RTCP packets after they have been finished If they are counted before calling gst_rtcp_buffer_end(), then the size is way too big.	2010-09-13 12:13:23 +02:00
Olivier Crête	306ee454c6	gstrtpsession: Don't unref pads in finalize The gstrtpsession object is not holding any reference to them directly	2010-09-13 12:10:11 +02:00
Wim Taymans	93228ccd52	rtpbin: add ntp-sync property Add an ntp-sync property that will sync the received streams to the server NTP time. This requires synchronized NTP times between the sender and receivers, like with ntpd. Based on patch from Thijs Vermeir. Fixes #627796	2010-09-06 11:01:57 +02:00
Wim Taymans	f03fd91400	jitterbuffer: rename a variable to avoid confusion	2010-09-06 11:01:57 +02:00
Wim Taymans	e3479630ae	rtpbin: rename some variables for less confusion	2010-09-06 11:01:57 +02:00
Wim Taymans	0f59664c6a	rtpjitterbuffer: move comment where it belongs	2010-09-06 11:01:57 +02:00
Wim Taymans	4fd81747f3	session: minor cleanups Make clock snapshots more accurate by only sampling the same clock once.	2010-09-06 11:01:57 +02:00
Thijs Vermeir	51020549f0	rtpbin: add use-pipeline-clock property With this property RTCP SR NTP times can be based on the system clock (maybe synced with ntpd) or the current pipeline clock. https://bugzilla.gnome.org/show_bug.cgi?id=627796	2010-09-06 11:01:57 +02:00
Thijs Vermeir	244a35a226	rtpptdemux: fix memleak on custom downstream events by not sending custom downstream event twice and fix memleak when not handling the event https://bugzilla.gnome.org/show_bug.cgi?id=623196	2010-06-30 12:39:09 +02:00
Sebastian Dröge	f16ed4a91c	gst: Don't use GST_DEBUG_FUNCPTR for GObject vfuncs	2010-06-06 17:52:40 +02:00
Thijs Vermeir	0bb2be3a7e	rtpjitterbuffer: fix compiler warning unused variable ‘estimated’	2010-06-02 15:32:36 +02:00
Alessandro Decina	4b6cb93025	rtpjitterbuffer: stop buffering and emit EOS at the end of a stream When using RTP_JITTER_BUFFER_MODE_BUFFER, make sure that the ringbuffer doesn't get stuck buffering forever when there isn't enough data left to fill the buffer.	2010-06-02 14:21:16 +02:00
Wim Taymans	dc2662e22b	rtpbin: fix docs Documentation error spotted by tony <caicai0119 at gmail.com> Fixes #618419	2010-05-13 13:01:26 +02:00
Wim Taymans	50f26c671b	rtpsession: fix return value	2010-05-07 19:06:35 +02:00
Wim Taymans	aadf4ddf7e	rtpsession: add properties to configure the bandwidth Add properties to proxy the bandwidth configuration to the session object.	2010-05-07 18:58:58 +02:00
Wim Taymans	69cde0e874	rtpsession: add properties to configure bandwidths Add properties to configure the sender and receiver bandwidths. Configure the bandwidths before calculating the RTCP timeout when we need to.	2010-05-07 18:57:13 +02:00
Wim Taymans	d84dc1112d	rtpstats: add some debug info	2010-05-07 18:56:30 +02:00
Wim Taymans	5690331c9e	rtpsession: small cleanups	2010-05-07 18:55:34 +02:00
Wim Taymans	0da5cf2e21	rtpstats: make bandwidths more configurable Add a method to configure the various bandwidths in the session.	2010-05-07 16:55:13 +02:00
Wim Taymans	6eee730c4a	rtpsession: handle NONE RTCP intervals Prepare for handling RTCP reporting intervals of GST_CLOCK_TIME_NONE, which means don't send RTCP at all.	2010-05-07 13:32:30 +02:00
Alessandro Decina	40899379c0	rtpjitterbuffer: move some initialization code from change_state to _init. Set ->active to TRUE in _init so it can be set to FALSE after creating the jitterbuffer and it won't be mistakenly reset to TRUE in the change_state function. This is needed to start the jitterbuffer as inactive when rtpbin is buffering.	2010-05-03 13:34:59 +02:00
Alessandro Decina	ffc2da30fc	rtpbin: fix a bug handling BUFFERING messages. If a session exists but has no streams, set the min buffering percent to 0 since it means that we haven't received anything for that session yet.	2010-05-03 11:56:58 +02:00
Alessandro Decina	f6e9f359b9	rtpbin: when a stream is created, pause the jitterbuffer if rtpbin is buffering.	2010-05-03 11:51:37 +02:00
Alessandro Decina	38a5b08ef2	rtpbin: fix a bug calculating stream offsets.	2010-05-03 11:23:59 +02:00
Stefan Kost	d6e9af2a11	docs: do proper escaping for "%"	2010-04-08 18:05:46 +03:00
Stefan Kost	9967a4112b	rtpsession: remove prototype for non existing function There is no function by that name anywhere.	2010-04-08 14:02:50 +03:00
Benjamin Otte	cccfeaa59c	gst_element_class_set_details => gst_element_class_set_details_simple	2010-03-18 14:32:00 +01:00
Benjamin Otte	1055aaa9cb	Add -Wredundant-decls warning flag Also fix compile issues	2010-03-17 19:35:10 +01:00
Benjamin Otte	21f66635e8	Update for recent changes to common submodule This just replaces every "$ERROR_CFLAGS" usage with a usage of "$WARNING_CFLAGS $ERROR_CFLAGS" to get the same functionality as previously. Actually using that separation will happen later.	2010-03-10 21:53:51 +01:00
Olivier Crête	a6dfe96169	rtpsession: Make it possible to favor new sources in case of SSRC conflict Add a "favor-new" property that tells the session to favor new sources when there is a SSRC conflict. This is useful for SIP calls and other such cases where a remote loop is extremely unlikely. Fixes #607615	2010-03-10 11:21:19 +01:00
Olivier Crête	f336ea283f	rtpsession: Move SSRC conflicts lists into RTPSource We will also need to track SSRC conflicts in remote sources. See #607615	2010-03-10 11:21:18 +01:00
Wim Taymans	529f443a61	rtpsource: use payload size to estimate bitrate Use the length of the payload for estimating the receiver bitrate so that it matches the calculations done on the sender side. Together with the number of packets one can scale the bitrate with the header overhead of the lower transport.	2010-03-08 17:48:04 +01:00
Wim Taymans	c971d1a9ab	rtpsource: refactor bitrate estimation Don't reuse the same variable we need for stats for the bitrate estimation because we're updating it. Refactor the bitrate estimation code so that both sender and receivers use the same code path.	2010-03-08 17:48:00 +01:00
Tristan Matthews	a0a6d4ff3b	added bitrate estimation to receiver-side stats, fixes #611213	2010-03-08 17:47:55 +01:00
Sebastian Dröge	bcd06ea527	rtpjitterbuffer: Reset skew detection after instantiating the jitterbuffer ...not only when going to READY. This sets high_level and friends to a more useful value.	2010-02-23 17:24:03 +01:00
Sebastian Dröge	0a12e69024	rtpjitterbuffer: Return 100 if high-level is 0 instead of dividing by zero	2010-02-23 17:20:02 +01:00
Tim-Philipp Müller	07fa73f199	docs: add Since: markers for new jitterbuffer properties	2010-02-19 12:13:07 +00:00
Wim Taymans	9d40d60960	rtpbin: remove use of ntp_ns_base	2010-02-15 21:36:29 +01:00
Wim Taymans	5a4ecc9da1	rtpbin: remove more ntpnstime and cleanups Remove some code where we pass ntpnstime around, we can do most things with the running_time just fine. Rename a variable in the ArrivalStats struct so that it's clear that this is the current system time.	2010-02-15 21:36:29 +01:00
Wim Taymans	74241e549f	rtpsource: use running_time for jitter Use the running_time to calculate the jitter instead of the ntp time. Part of the plan to get rid of ntpnsbase.	2010-02-15 21:36:29 +01:00
Wim Taymans	83cb1aecc8	rtpbin: change how NTP time is calculated in RTCP Don't calculate the NTP time based on the running_time of the pipeline but from the systemclock. This allows us to generate more accurate NTP timestamps in case the systemclock is synchronized with NTP or similar.	2010-02-15 21:36:29 +01:00
Tim-Philipp Müller	63c86ac3d8	raw1394, matroska, rtpmanager: remove padding from structures None of these element and class structures are in public headers, so don't need padding.	2010-02-15 00:50:10 +00:00
Wim Taymans	7f08081016	jitterbuffer: don't resync to invalid timestamps If we detect backward timestamps on the server, don't try to resync when we don't have an input timestamp (such as when using RTSP over TCP) instead, do nothing but assume the timestamp was ok, it will correct itself when time goes forwards.	2010-02-12 19:32:27 +01:00
Wim Taymans	d344754f03	rtpbin: fix typo	2010-02-12 17:22:56 +01:00
Wim Taymans	772eca5aff	jitterbuffer: start out active and not buffering There is no need to set the latency in the jittebuffer in _init, we will set that later when going to PAUSED. Set the jitterbuffer active and not buffering when starting.	2010-02-12 17:22:56 +01:00
Wim Taymans	8bbfd94c25	rtpbin: more buffering work When deactivating jitterbuffers when the buffering starts, keep the current percent of the jitterbuffer and also set the jitterbuffer in the buffering state so that we know when it's filled again. Add property to get the buffering percentage of the jitterbuffer.	2010-02-12 17:22:56 +01:00
Wim Taymans	e6e287cdcc	rtpjitterbuffer: adjust latency in buffer mode When we are in buffer mode, adjust the buffering low/high thresholds based on the total configured latency. If we don't and there is a huge queue or element with a big latency downstream we might drain the complete queue immediately and start buffering again.	2010-02-12 17:22:55 +01:00
Wim Taymans	ab73603031	jitterbuffer: add ts-offset to timestamp Add the ts-offset to the buffer timestamp to get the final output timestamp of the buffer.	2010-02-12 17:22:55 +01:00
Wim Taymans	74a3be350d	rtpbin: do more accurate buffer offsets Return the next timestamp in the jitterbuffer. Use the min-timestamp of the jitterbuffers to calculate an offset so that the next timestamp is pushed with a timestamp equal to running_time. Start producing timestamps from 0 in the buffering case too.	2010-02-12 17:22:55 +01:00
Wim Taymans	3efcc0fbc1	rtpbin: only start buffering when < 100% Only start buffering when the percentage message is < 100 %.	2010-02-12 17:22:55 +01:00
Wim Taymans	0348ebe651	rtpbin: keep track of elapsed pause time Keep track of the time we spend pausing the jitterbuffers when they were buffering and distribute this elapsed time to the jitterbuffers. Also keep the latency in nanosecond precision.	2010-02-12 17:22:54 +01:00
Wim Taymans	ecf6ed8fc1	jitterbuffer: keep track of offset Keep track of an outgoing offset that we add to each outgoing buffer to compensate for PAUSE when buffering. Adjust the offset when activating.	2010-02-12 17:22:54 +01:00
Wim Taymans	048e5b6fbe	jitterbuffer: report level using high watermark	2010-02-12 17:22:54 +01:00
Wim Taymans	8d814f3782	rtpbin: pass running_time to jitterbuffer pause Pass the current running time to the jitterbuffer when pausing or resuming so that it calculate the right offsets. Small cleanups and comments. Set the default rtspsrc latency to 2 seconds.	2010-02-12 17:22:54 +01:00
Wim Taymans	bf697b12e3	rtpbin: add some comments	2010-02-12 17:22:53 +01:00
Wim Taymans	20a27a545a	rtpbin: more buffering updates Add signal to pause the jitterbuffer. This will be emitted from gstrtpbin when one of the jitterbuffers is buffering. Make rtpbin collect the buffering messages and post a new buffering message with the min value. Remove the stats callback from jitterbuffer but pass a percent integer to functions that affect the buffering state of the jitterbuffer. This allows us then to post buffering messages from outside of the jitterbuffer lock.	2010-02-12 17:22:53 +01:00
Wim Taymans	a5b9d3f917	rtpbin: propagate buffer-mode property Propagate buffer-mode property to the jitterbuffers. Intercept BUFFERING messages in rtpbin	2010-02-12 17:22:53 +01:00
Wim Taymans	d3db9574a9	jitterbuffer: do more buffering implementation Add callback for buffering stats. Configure the latency in the jitterbuffer instead of passing it with _insert. Calculate buffering levels when pushing and popping Post buffering messages.	2010-02-12 17:22:52 +01:00
Wim Taymans	aeacbfed3e	jitterbuffer: flesh out buffering mode some more Add a buffering state to the jitterbuffer and wait until buffering ends before pushing out packets.	2010-02-12 17:22:52 +01:00
Wim Taymans	56b29c9a6b	jitterbuffer: hook up the mode property Expose a mode property on the jitterbuffer. Fix the case where timestamps are -1 in the check for outgoing timestamps.	2010-02-12 17:22:52 +01:00
Wim Taymans	be4517a6b8	jitterbuffer: add buffering mode options Add getters and setters for different buffering modes that the jitterbuffer will support. Default to the current slave mode.	2010-02-12 17:22:52 +01:00
Wim Taymans	99a581215f	jitterbuffer: add some more debug info	2010-02-12 13:53:57 +01:00
Wim Taymans	05418f1687	rtpbin: avoid some structure copies Don't make copied in the getter and setter for SDES in the RTPSource. This avoids a couple of copies of the SDES structure when generating RTCP packets.	2009-12-22 22:27:21 +01:00
Pascal Buhler	c3448f978e	rtpmanager: improve SDES handling Store SDES internally as a struct to support multiple PRIV values. Include all values set in SDES struct when sending RTCP SDES.	2009-12-22 21:43:25 +01:00
Wim Taymans	9734699788	rtpbin: add property to remove pads automatically Add a property called autoremove to automatically remove the pads of sources that timed out. Fixes #554839	2009-12-21 15:07:44 +01:00
Wim Taymans	c611bbaa8e	ssrcdemux: fix comparison A NULL means no pad was found.	2009-12-21 15:07:34 +01:00
Aurelien Grimaud	07f27f0efd	rtpsession: avoid buffer ref/unref pairs for CSRCs We ref the buffer before pushing it downstream in order to get the CSRCs of it after pushing. This causes performance problems when downstream elements want to change the metadata because the buffer needs to be subbuffered. Instead, read and store the CSRCs of the buffer in an array before pushing it and process the array after pushing the buffer. This allows us to remove the ref/unref pair. Fixes #603376	2009-11-30 15:59:50 +01:00
Wim Taymans	8070ae967b	jitterbuffer: avoid using wrong clock-rate Check for a valid clock-rate before attempting to estimate the npt stop time.	2009-11-25 10:38:23 -06:00
Wim Taymans	5682e2bf01	rtpbin: fix typo in comments	2009-11-25 10:37:30 -06:00
Stefan Kost	9ee0815e85	docs: more links and better short description Fix spelling of GstRtpSsrcDemux to get it linked. Add more links. Change the short description to be more meaningful.	2009-11-20 11:25:49 +02:00
Wim Taymans	f52859432f	jitterbuffer: release lock before emiting signals Release the jbuf lock before emiting the request-pt-map signal to avoid deadlocks. We also need to catch the shutdown case when locking again. Fixes #593354	2009-11-18 10:50:44 +01:00
Stefan Kost	e43eb89449	tests: add a jitterbuffer test Tests pushing a few buffers in various order and asserting the order sent by the jitterbuffer. Contains two disabled tests that need more work.	2009-10-22 13:35:57 +03:00
Stefan Kost	6904e46ef2	build: use gst-glib-gen.mak to fix the glib build rules. The build rules in glib-gen.mak were using pattern rules in a non save way.	2009-10-16 11:53:38 +03:00
Håvard Graff	58b9de4cca	rtpptdemux: only forward the lost-event to the last seen pt-number forward all events on all pads except for the PacketLost event, which we want to forward to the last seen pt pad. Fixes #598377	2009-10-14 12:28:55 +02:00
Stefan Kost	e0cdd879b4	build: fprintf, sprintf, sscanf need stdio.h	2009-10-07 14:03:20 +03:00
Wim Taymans	0040d01265	rtpbin: use locking around the sessions	2009-10-05 16:07:24 +02:00
Wim Taymans	8fb77403c5	jitterbuffer: cache latency in nanoseconds Cache the latency in nanoseconds units to avoid having to convert the milliseconds value to nanoseconds all the time.	2009-10-01 12:52:40 +02:00
Wim Taymans	c262735164	jitterbuffer: handle -1 input timestamps Don't try to check a -1 timestamp against the max delay.	2009-10-01 12:12:09 +02:00
Stefan Kost	0a7ef67ad0	docs: fix gtk-doc warnings	2009-09-10 10:28:48 +03:00
Marc Leeman	6b46aeb6a3	rtpbin: add ignore-pt parameter Add a parameter 'ignore-pt' that disables creating a gstrtpptdemux module and ghosts the pads of gstrtpjitterbuffer instead of the ones of gstrtpptdemux. Fixes #594490	2009-09-08 17:38:32 +02:00
Håvard Graff	2912b21d14	rtpbin: propagate payload-type-change signal from demuxer fixes #594254	2009-09-08 13:59:56 +02:00
Havard Graff	a52309eff7	jitterbuffer: change severity of clock-rate change debug Make log GST_DEBUG under normal circumstances, GST_WARNING otherwise. Fixes #594253	2009-09-08 13:44:49 +02:00

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1276 commits