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113 commits

Author SHA1 Message Date
Miguel París Díaz
389e0abeb0 rtpsource: complete warn log with SSRC 
https://bugzilla.gnome.org/show_bug.cgi?id=767195
 2016-06-06 10:47:17 +03:00
Nirbheek Chauhan
78847d03cf rtpmanager: Some comment and documentation clarifications/fixes 2016-03-15 09:32:47 +00:00
Luis de Bethencourt
d4f094f587 rtpmanager: use GST_STIME_ARGS for GstClockTimeDiff 
No need to manually handle negative values of diff, GST_STIME_ARGS does
exactly this.
 2015-11-03 14:26:32 +00:00
Miguel París Díaz
f321bfeaf4 rtpmanager: Take into account packet rate for max-dropout and max-misorder calculations 
https://bugzilla.gnome.org/show_bug.cgi?id=751311
 2015-10-07 12:07:18 +01:00
Miguel París Díaz
4c96094fbb rtpmanager: add "max-dropout-time" and "max-misorder-time" props 
https://bugzilla.gnome.org/show_bug.cgi?id=751311
 2015-10-07 12:06:47 +01:00
Hyunjun Ko
b814d7ed25 rtpsource: doesn't handle probation and rtp gap in case of sender 
https://bugzilla.gnome.org/show_bug.cgi?id=754548
 2015-10-02 16:42:36 +03:00
Stefan Sauer
12930c2f8c docs: fix "Symbol name not found at the start of the comment block" 
Add symbols or change comment into a regular comment.
 2015-07-07 17:12:02 +02:00
Jose Antonio Santos Cadenas
11f298a338 rtpsource: Do not try to push NULL buffers 
If update_receiver_stats() fails, we can't really do anything with this buffer
anymore and have to drop it. This happens if there's a big seqnum
discontinuity for example.

https://bugzilla.gnome.org/show_bug.cgi?id=751311
 2015-06-22 12:26:59 +02:00
Sebastian Dröge
dc059efa60 rtp: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP() 
The mix between all these in the RTP code is confusing, let's try to be
consistent.
 2015-06-10 14:34:47 +02:00
Ilya Konstantinov
c7e168ec70 rtpmanager: clarify negative lost packets in stats 
Also:
- Move notes on units before field documentation.
- Unify documentation style.

https://bugzilla.gnome.org/show_bug.cgi?id=750653
 2015-06-10 14:10:52 +02:00
Ilya Konstantinov
0a578c235a rtpmanager: document units of stats and arguments 
Also, minor spelling and style corrections.

https://bugzilla.gnome.org/show_bug.cgi?id=750653
 2015-06-09 18:21:59 +02:00
Sebastian Dröge
c60038f188 rtpsource: Queue bad packets instead of dropping them 
So we can send them out once we found the next, consecutive sequence number in
case one is following.
 2015-05-18 18:43:16 +03:00
Sebastian Dröge
9f18a271f3 rtpsource: Use g_queue_foreach() to unref all buffers in queues 2015-05-18 18:43:16 +03:00
Sebastian Dröge
54e924332e rtpsource: Refactor seqnum comparison code a bit 2015-05-18 18:43:16 +03:00
Sebastian Dröge
1974b24ef4 rtpsource: Allow sequence number wraparound during probation 2015-05-18 18:43:16 +03:00
Sebastian Dröge
3386de7a8a rtpsource: Make sequence number comparison code more readable 
... by using gst_rtp_buffer_compare_seqnum() and signed integers
instead of implictly using effects of integer over/underflows.
 2015-05-18 18:43:16 +03:00
Luis de Bethencourt
671b4d25cd remove unused enum items PROP_LAST 
This were probably added to the enums due to cargo cult programming and are
unused. Removing them.
 2015-04-24 17:01:12 +01:00
Sebastian Dröge
80268e7d37 rtpsource/rtprtxsend: Also pass correct seqnum-offset and payload to the RTX rtpsource 
https://bugzilla.gnome.org/show_bug.cgi?id=747394
 2015-04-16 17:33:37 +02:00
Santiago Carot-Nemesio
e05378ec16 rtp: Add Full Intra Request (FIR) packets to statistics 
https://bugzilla.gnome.org/show_bug.cgi?id=745587
 2015-03-04 12:04:40 +01:00
Santiago Carot-Nemesio
22791413f9 rtp: Add Packet Loss Indication (PLI) to statistics 
This is helpful to provide statistics in the format defined in
http://w3c.github.io/webrtc-stats/#dictionary-rtcrtpstreamstats-members.

https://bugzilla.gnome.org/show_bug.cgi?id=745587
 2015-03-04 12:04:07 +01:00
Olivier Crête
51a8bedced rtpsource: Rename seqnum-base to seqnum-offset in caps 
This was modified back in 1.0 in GstRtpBasePayload
 2014-10-10 18:33:34 -04:00
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
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
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
Wim Taymans
268a75e705 rtpsource: refactor receiver stats update 2013-11-07 16:24:30 +01:00
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
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
c795b72988 source: small cleanups 2013-09-13 14:34:27 +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
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
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
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
Aleix Conchillo Flaque
4a200b670f rtp: make rtp packet probation configurable (bug #682512) 2012-08-30 21:49:57 +02:00
Sebastian Dröge
cb789e32ad rtpmanager: Port to GIO 2012-01-17 13:08:42 +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
83ccefb24e update for netbuffer api 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
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
61382aad28 source: fix type of ntpnstime 2011-02-02 18:30:47 +01:00