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Author SHA1 Message Date
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