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Author SHA1 Message Date
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