Commit graph

97 commits

Author SHA1 Message Date
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
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
db5150a23a rtpsource: Retain RTCP Feedback packets for a specified amount of time 2011-02-01 18:28:51 +01:00
Wim Taymans
b5647685c4 rtpsource: use the right variable
Use the right variable for specifying that we sent a receiver report.
2010-12-27 13:13:46 +01:00
Wim Taymans
7caad21a57 rtpsource: include last send RB block
Only report RB values for non-internal sources.
Report not only the RB blocks we last received from but also the last RB
block we sent to a source.
2010-12-23 13:58:30 +01:00
Wim Taymans
6035ee08c0 rtpsource: include all stats and document
Include all possible stats of a source in the stats structure because we might
be interested in what happened in the past.
Document the stats property and the fields.
2010-12-23 13:58:30 +01:00
Pascal Buhler
ca6a512b5e rtpbin: Make cleaning up sources in rtp_session_on_timeout MT safe
Using _foreach_remove on the hashtable, while releasing the lock protecting
that table inside the callback is not a good idea. The hashtable might
then change (a source removed or added) while signals like on_timeout
are being sent.

This solution makes a copy of the table, performs the _foreach without
actually removing any sources, but marks them for removal on a second
iteration with the real list, but this time not letting go of the lock.

Fixes #630452
2010-09-24 15:38:00 +02:00
Wim Taymans
ce007b244e rtpsource: whitespace fixes 2010-09-24 13:50:02 +02:00
Wim Taymans
c5203a479b rtpsource: simplify the rate estimation some more 2010-09-24 13:48:50 +02:00
Havard Graff
0fa589a3dd rtpmanager: provide additional statistics 2010-09-24 13:26:10 +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
Wim Taymans
529f443a61 rtpsource: use payload size to estimate bitrate
Use the length of the payload for estimating the receiver bitrate so that it
matches the calculations done on the sender side. Together with the number of
packets one can scale the bitrate with the header overhead of the lower
transport.
2010-03-08 17:48:04 +01:00
Wim Taymans
c971d1a9ab rtpsource: refactor bitrate estimation
Don't reuse the same variable we need for stats for the bitrate estimation
because we're updating it.
Refactor the bitrate estimation code so that both sender and receivers use the
same code path.
2010-03-08 17:48:00 +01:00
Tristan Matthews
a0a6d4ff3b added bitrate estimation to receiver-side stats, fixes #611213 2010-03-08 17:47:55 +01:00
Wim Taymans
74241e549f rtpsource: use running_time for jitter
Use the running_time to calculate the jitter instead of the ntp time. Part of
the plan to get rid of ntpnsbase.
2010-02-15 21:36:29 +01:00
Wim Taymans
83cb1aecc8 rtpbin: change how NTP time is calculated in RTCP
Don't calculate the NTP time based on the running_time of the pipeline but from
the systemclock. This allows us to generate more accurate NTP timestamps in case
the systemclock is synchronized with NTP or similar.
2010-02-15 21:36:29 +01:00