Commit graph

339 commits

Author SHA1 Message Date
Sebastian Dröge
b07b7736b3 rtpsession: Delay the next regular RTCP packet after early RTCP
This is required to not exceed the short term average RTCP bitrate when
using early feedback as compared to without early feedback.
2015-01-26 18:49:31 +01:00
Sebastian Dröge
bc9111a03d rtpsession: Add new send-rtcp-full signal
This indicates with a boolean return value if scheduling a new RTCP packet
within the requested delay was possible. Otherwise it behaves exactly like
send-rtcp. The only reason for adding a new signal is ABI compatibility.
2015-01-26 18:49:31 +01:00
Sebastian Dröge
60e2d0c84f rtpsession: Fix indention 2015-01-22 11:03:25 +01:00
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
Sanjay NM
26a1344f37 Miscellaneous minor cleanups
Fix redundant variables and assignments,
and unreachable breaks.

https://bugzilla.gnome.org/show_bug.cgi?id=736875
https://bugzilla.gnome.org/show_bug.cgi?id=736876
https://bugzilla.gnome.org/show_bug.cgi?id=736879
https://bugzilla.gnome.org/show_bug.cgi?id=736880
https://bugzilla.gnome.org/show_bug.cgi?id=736881
https://bugzilla.gnome.org/show_bug.cgi?id=736888
https://bugzilla.gnome.org/show_bug.cgi?id=736890
https://bugzilla.gnome.org/show_bug.cgi?id=736892
https://bugzilla.gnome.org/show_bug.cgi?id=736893
https://bugzilla.gnome.org/show_bug.cgi?id=736894
2014-09-24 00:45:31 +01:00
Wim Taymans
d004eda79d rtpsession: update last_activity when sending RTP
Also update last_activity when doing something with the internal
source to make sure don't timeout early.

See https://bugzilla.gnome.org/show_bug.cgi?id=730217
2014-05-16 16:55:17 +02:00
George Kiagiadakis
7e2138794f rtpsession: remove unused if branch
1) sources that have sent BYE in the past cannot be senders, since
they would have timed out to being receivers in the meantime...
2) sources that have sent BYE are now being removed earlier inside
this function
2014-05-14 16:01:50 +02:00
George Kiagiadakis
85d4c031d4 rtpsession: cleanup sources that have sent BYE 2014-05-14 16:01:50 +02:00
George Kiagiadakis
7d7840cc4a rtpsession: unify nested if clauses 2014-05-14 16:01:50 +02:00
George Kiagiadakis
0e6a31411b rtpsession: timeout internal sources that are inactive for a long time and send BYE 2014-05-14 16:01:50 +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
d2f93e3afc session: small cleanups
It's nicer to explicitly check for NULL on pointer types to make it
clear that it's a pointer and not a boolean.
2014-03-05 14:28:26 +01:00
Wim Taymans
5818a0de1a session: handle unknown SSRC in FIR
https://bugzilla.gnome.org/show_bug.cgi?id=725712
2014-03-05 14:27:47 +01:00
Wim Taymans
03e4a180da session: place SSRC in Retransmission event 2014-01-03 20:48:29 +01:00
Wim Taymans
ee7f41ba2e rtpsession: internal-ssrc is no longer deprecated 2013-12-30 17:00:45 +01:00
George Kiagiadakis
17517ca491 rtpsession: allow setting internal-ssrc again 2013-12-30 14:03:05 +01:00
Olivier Crête
ada6ea668b rtpsession: Add error message if the app tries to set the internal-ssrc 2013-12-13 17:36:36 -05:00
Olivier Crête
d715010d78 rtpsession: Only count nacks when a nack packet is received
Not when any RTCP feedback packet is.
2013-12-13 16:08:35 -05:00
Olivier Crête
7af9fdbca6 rtpsession: Process PSFB FIR requests which lack the media ssrc
According to RFC 5104 section 4.3.1.2, RTCP PSFB FIR message SHALL
have a media_ssrc field set to 0. The actual media ssrc is in the FCI.
So in that case, we ignore the retained feedback and just let it through
to the rtp_session_process_fir() function which will check for the actual
SSRC inside the FCI.

Fixes a regression introduced by commit 57c27ec3
2013-12-13 16:01:07 -05:00
George Kiagiadakis
6a2de911fa rtpsession: fix rb blocks disappearing after the first rtcp cycle with multiple senders
Previously, when the session had multiple internal sender SSRCs, it would
issue SR reports with RB blocks only on the first RTCP timeout and afterwards
SR reports would be sent empty. This was because the "generation" number
in RTPSource would increase more than once during the same cycle and afterwards
it would always be greater than the session's generation, which would cause
it to be skipped from being included in RBs.

This commit fixes this problem by:
1) Increasing the RTPSource generation only at the end of each cycle,
which essentially fixes the problem but only when the internal senders
are less than GST_RTCP_MAX_RB_COUNT.
2) Keeping for each RTPSource a set of SSRCs which stores which SSRC's
SR the given RTPSource has been reported in, which also fixes the problem
when the internal senders are more than GST_RTCP_MAX_RB_COUNT. This is
necessary because of the fact that any RTPSource is marked as reported
in itself's SR and makes it impossible to know if it has been reported
in other SRs too or not, and which.
2013-12-12 16:44:27 +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
George Kiagiadakis
282028e753 rtpsession: when we schedule BYE, only deal with BYE sources
When we are doing the RTCP timeout to schedule BYE packets, don't
generate RTCP for all sources but only for the sources marked as BYE.
2013-12-12 10:34:38 +01:00
George Kiagiadakis
6a421c3d81 rtpsession: reset state after scheduling BYE
After we do RTCP, we are not scheduling bye anymore.
2013-12-12 10:32:48 +01:00
George Kiagiadakis
0a0ff100ef rtpsession: also count NACKS when no signal was pending 2013-12-12 10:31:38 +01:00
George Kiagiadakis
bec9c04ea0 session: ignore RTCP packets for the BYE sources
When we are scheduling BYE packets, ignore all RTCP for the sources that
are scheduling a BYE packet. Other sources that are not scheduling BYE
should continue receiving RTCP packets as usual.
2013-12-12 10:09:25 +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
Wim Taymans
e8edecc56e rtpsession: don't unref buffer twice
Cleaning the packet info will already unref the buffer.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715078
2013-11-28 16:51:13 +01:00
Tim-Philipp Müller
548e756e0a rtpmanager: fix Since markers
Should be next stable release series version
2013-11-16 12:15:14 +00:00
Torrie Fischer
acf74435e3 gstrtpsession: Implement a number of feedback packet statistics
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711693
2013-11-15 15:21:19 +01:00
Wim Taymans
c8db05d610 rtpsource: update receiver stats for sender
An internal sender in a session is also a receiver of its own packets so update
the receiver stats. Other senders in the session will use this info to generate
correct RB blocks in their SR reports.
2013-11-07 16:24:30 +01:00
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
28e5f90988 rtpbin: use PacketInfo for the sender
Avoid mapping the packet multiple times when sending RTP.
2013-09-13 14:34:28 +02:00
Wim Taymans
a02c9473d8 rtpbin: store more in the PacketInfo
Store all info in the PacketInfo so that we can avoid mapping the packet
multiple times.
2013-09-13 14:34:28 +02:00
Wim Taymans
e5c789abd6 session: store more in the PacketInfo structure 2013-09-13 14:34:28 +02:00
Wim Taymans
47662f9ca4 rtpbin: RTPArrivalStats -> RTPPacketInfo
Rename a structure because we are also going to use this for the sender
bits.
2013-09-13 14:34:28 +02:00
Wim Taymans
f1106cde66 session: only update next check time when reconsidering
Don't update the next RTCP check time in all cases but only when we
reconsidered. This avoids delaying sending a full RTCP packet when we
are doing early feedback.
2013-08-27 09:55:52 +02:00
Wim Taymans
47065db0b6 session: add more debug 2013-08-27 09:55:52 +02:00
Wim Taymans
4b7bcc2ec1 rtsession: fix locking
We need to take the session lock when getting and manipulating the
source.
2013-08-26 11:50:27 +02:00
Wim Taymans
3f46527f75 rtpsession: add some more debug 2013-08-26 11:50:13 +02:00
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
48174164eb session: add NACK feedback in RTCP 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
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
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
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
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
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
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
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
2971ed44ee rtpsession: remove dead code
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=668355
2013-02-07 15:06:40 +01:00
Wim Taymans
6b7d05ac57 rtpsession: improve debug 2013-01-07 14:22:48 +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
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
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
Wim Taymans
51371d26ee update for RTP buffer api changes 2012-07-17 16:38:27 +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
Wim Taymans
af59f573b5 rtpsession: don't leak the address 2012-03-13 19:26:47 +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
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
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
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
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
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
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
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
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
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
Wim Taymans
fc4684f4c6 fix compilation 2011-10-27 16:03:17 +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
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
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
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
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
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
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
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
Wim Taymans
0a56b25882 rtpsession: use NetAddress metadata 2011-02-28 13:28:29 +01:00
Wim Taymans
d87c27fd2c miniobject: use buffer private field for extra data
Use the owner private field to store extra buffer data instead of using
subclassing.
2011-02-28 11:58:48 +01:00
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