Commit graph

170 commits

Author SHA1 Message Date
Tim-Philipp Müller
c9597298f9 docs: remove outdated and pointless 'Last reviewed' lines from docs
They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
2014-04-26 23:35:17 +01:00
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
bbe6d9a258 rtpsession: proxy caps and allocation on RTP pads
recv_rtp_sink: allow proxying of the allocation query.
send_rtp_sink: allow proxying of caps and allocation. This allows us to
query caps downstream as well as get an allocator from downstream.
send_rtp_src: allow proxy of caps, this makes the caps query do
upstream.

See https://bugzilla.gnome.org/show_bug.cgi?id=723850
2014-02-14 12:05:55 +01:00
Wim Taymans
679b5a8682 session: also push EOS event to RTCP srcpad 2014-01-03 20:48:29 +01:00
Wim Taymans
03e4a180da session: place SSRC in Retransmission event 2014-01-03 20:48:29 +01:00
George Kiagiadakis
5ddf6a5e32 gstrtpsession: suggest upstream to use the new "internal-ssrc" after a collision
When a collision is found on the internal ssrc, we have to change it.
Ideally, we want also the payloader upstream to follow this change and use
the new internal ssrc. Ideally we want this condition to be always met:
if there is one payloader sending on this session, its ssrc should match the
internal ssrc.
2013-12-30 14:03:05 +01:00
Wim Taymans
7181a21ca9 rtpsession: use RTT from the Retransmission event
Place the estimated RTT in the Retransmission event and let the session
manager use that instead of the hardcoded value.
2013-12-30 11:18:50 +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
e4bc81d7d2 rtpsession: remove collision reconfigure event
Remove bogus reconfigure event on collision, we don't want to send the event on
the receiving RTP pad and the collision event is now handling this
case.

See https://bugzilla.gnome.org/show_bug.cgi?id=711560
2013-11-11 15:27:18 +01:00
Julien Isorce
b32fc6f416 gstrtpsession: send custom upstream event "GstRTPCollision" on send_rtp_sink pad
See https://bugzilla.gnome.org/show_bug.cgi?id=711560
2013-11-11 15:25:52 +01:00
Wim Taymans
adf5d96044 rtpmanager: update docs 2013-09-23 16:34:15 +02:00
Wim Taymans
e5019de80d docs: update docs with 1.0 element names 2013-09-23 15:36:47 +02:00
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
454d75951e jitterbuffer: fix types of the retransmission event 2013-08-27 09:55:52 +02:00
Wim Taymans
ee15bc9284 session: generate events correctly
Do correct shifting of the bitmask for lost packets.
2013-08-21 17:02:27 +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
Tim-Philipp Müller
7469cd3a4c rtpmanager: use generic marshaller 2013-08-04 11:03:07 +01:00
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
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
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
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
Sebastian Dröge
5a9f4a3cbc rtpsession: Add support for group-id in the stream-start event 2013-07-22 15:30:13 +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
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
747447d298 rtpsession: avoid '...is used uninitialized' 2013-01-29 10:32:51 +01:00
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
Wim Taymans
72402cc649 rtp: small improvements 2013-01-08 16:27:42 +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
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
Wim Taymans
1cebcfa8c2 rtpbin: use running-time for NTP time
When use-pipeline-clock is set, use the running-time of the
pipeline to calculate the NTP timestamps. This method would previously
only work when the base-time is set to 0 but with this change it can
also work with different offsets and we can also implement pause/resume
of the sender and receiver now.
2012-10-17 12:26:05 +02:00
Aleix Conchillo Flaque
4a200b670f rtp: make rtp packet probation configurable (bug #682512) 2012-08-30 21:49:57 +02:00
Tim-Philipp Müller
4bb52bbadf docs: gst-launch -> gst-launch-1.0 and ffmpegcolorspace -> videoconvert 2012-08-27 21:20:30 +01:00
Tim-Philipp Müller
c22268b5d3 rtpsession: remove deprecated and unused "ntp-ns-base" property 2012-07-06 13:16:00 +01:00
Tim-Philipp Müller
e09ae5736d Use new gst_element_class_set_static_metadata() 2012-04-10 00:51:41 +01:00
Wim Taymans
9942d3566e rtpsession: set caps without the lock
Release the lock before setting the caps on the srcpad, which triggers an event,
which could eventually call back into us and cause a deadlock.
2012-03-07 15:02:44 +01:00
Tim-Philipp Müller
7cb9b7ab9d Use new GLib API unconditionally 2012-01-22 23:15:19 +00:00
Wim Taymans
1584806634 port to new gthread API 2012-01-19 11:33:53 +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
330d984288 Use g_thread_try_new() instead of g_thread_crate() with newer glib versions 2011-12-12 09:46:27 +00:00