Commit graph

74 commits

Author SHA1 Message Date
Miguel París Díaz
e6504e3a65 rtpsession: fix Early Feedback Transmission
In early retransmission we are allowed to schedule 1 regular RTCP packet
at an earlier time. When we do that, we need to set allow_early to FALSE
and ignore/drop (or merge) all future requests for early transmission.
We now first check if we can schedule an early RTCP and if we can,
actually prepare the data for the next RTCP interval.

After we send the next regular RTCP after the early RTCP, we set
allow_early to TRUE again to allow more early requests.

Remove the condition for the immediate feedback for now.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738319
2014-10-22 13:13:47 +02:00
Olivier Crête
2e54d38dd0 rtpsession: Keep local conflicting addresses in the session
As we now replace the local RTPSource on a conflict, it's no longer possible
to keep local conflicts in the RTPSource, so they instead need to be kept
in the RTPSession.

Also fix the rtpcollision test to generate multiple collisions instead of
one by change the address, as otherwise we detected that it was a single one.
2014-05-03 18:30:20 -04:00
Wim Taymans
38a486b374 rtpsession: send reconfigure when internal-ssrc changes
When the internal-ssrc property changes, we want to send a reconfigure
upstream to make payloaders use the new suggested ssrc.
Using the internal-ssrc property to change the SSRC of a stream is not a
good idea and doesn't work when there are multiple senders, we want to
set the SSRC directly on the payloaders. Therefore, deprecate this
property.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725361
2014-04-18 10:21:27 +02:00
Wim Taymans
03e4a180da session: place SSRC in Retransmission event 2014-01-03 20:48:29 +01:00
George Kiagiadakis
c78a115154 rtpsession: keep extra stats for scheduling BYE
Keep an extra stats structure for scheduling the BYE packets. When we
decide to schedule BYE, make a copy of the current stats into the
bye_stats. Then while we schedule the BYE, update and use only the
bye_stats. When we finished scheduling the BYE packet, we use the
regular stats again.
2013-12-12 10:38:43 +01:00
Julien Isorce
33b398e345 rtpsession: determine if the session is doing point-to-point
In this case T_dither_max is set to 0 according to RFC 4585
2013-12-10 16:57:56 +01:00
Olivier Crête
b9ceafe5af rtpsession: Demux RTCP buffers from the RTP stream
If there are RTCP buffers in the RTP stream, process them as
RTCP. This way, we want receive streams following RFC 5761

https://bugzilla.gnome.org/show_bug.cgi?id=687657
2013-09-13 16:25:49 +02:00
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
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
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
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
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
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
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
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
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
93d07298ff session: let source keep track if it sent BYE 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
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
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
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
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
Aleix Conchillo Flaque
4a200b670f rtp: make rtp packet probation configurable (bug #682512) 2012-08-30 21:49:57 +02:00
Wim Taymans
1584806634 port to new gthread API 2012-01-19 11:33:53 +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
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
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
6cbd6afc0b update for new net library 2011-11-03 16:43:00 +01:00
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
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
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
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
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
1643f427db rtpsession: Emit signal on incoming RTCP FB packet 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
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
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
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
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