Commit graph

143 commits

Author SHA1 Message Date
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
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
da980884dd rtpsession: forward the caps event 2011-12-10 11:13:38 +01:00
Wim Taymans
68588c3f18 rtpsession: forward caps 2011-12-10 11:13:38 +01:00
Wim Taymans
6ac5e1ae16 rtp: pass parent to setcaps methods 2011-12-10 11:13:38 +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
Vincent Penquerc'h
c0e101e93f various: fix pad template leaks
https://bugzilla.gnome.org/show_bug.cgi?id=662664
2011-11-28 13:30:27 +00:00
Matej Knopp
1e5dd9e315 Fix printf format compiler warnings on OS X / 64bit
https://bugzilla.gnome.org/show_bug.cgi?id=662615
2011-11-22 01:28:22 +00:00
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
Wim Taymans
105650127e add parent to pad functions 2011-11-17 15:02:55 +01:00
Wim Taymans
7cc4b72550 add parent to internal links 2011-11-16 17:54:49 +01:00
Wim Taymans
6190312214 add parent to query function 2011-11-16 17:27:13 +01:00
Wim Taymans
75dc9634eb change getcaps to query
Chain up event function in payloaders.
2011-11-15 18:04:44 +01:00
Olivier Crête
1169bb05af gstrtpsession: Add special mode to use FIR as repair as Google does
https://bugzilla.gnome.org/show_bug.cgi?id=658419
2011-11-14 12:26:27 +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
Wim Taymans
9a8a8e72c8 structure: fix for api update 2011-11-02 09:06:37 +01:00
Tim-Philipp Müller
d18a578ba4 rtpmanager, v4l2: fix compiler warnings after gst_caps_new_simple() change 2011-10-28 09:06:41 +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
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
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
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
Robert Swain
5b18c652fb rtp, rtpmanager: Address unused but set variables
GCC 4.6.x spits warnings about variables that are unused but set. Such
variables have been removed where trivial but with comments left behind
for informational purposes in some cases.

gst_rtp_session_chain_recv_rtcp () was changed in commit 490113d4
to always return GST_FLOW_OK instead of the return value of
rtp_session_process_rtcp (), so we'll keep it that way.
2011-04-16 12:49:16 +01:00
Havard Graff
93f022d6ab rtpsession: fix wrongly applied patch
Obviously recv_rtp_sink does not have much to do with send_rtcp_src...
See commit 046ff170.

https://bugzilla.gnome.org/show_bug.cgi?id=647263
2011-04-09 12:32:37 +01:00
Sebastian Dröge
31af4fe33e rtp: Unref events if the parent element disappeared 2011-04-08 15:20:51 +02:00
Ole André Vadla Ravnås
046f170d6a rtpmanager: fix pad callbacks so they handle when parent goes away
1) We need to lock and get a strong ref to the parent, if still there.
2) If it has gone away, we need to handle that gracefully.

This is necessary in order to safely modify a running pipeline. Has been
observed when a streaming thread is doing a buffer_alloc() while an
application thread sends an event on a pad further downstream, and from
within a pad probe (holding STREAM_LOCK) carries out the pipeline plumbing
while the streaming thread has its buffer_alloc() in progress.
2011-04-08 15:16:56 +02:00
Havard Graff
f8370bb2a8 rtpsession: make iterate_internal_links MT-safe 2011-04-08 14:41:34 +02:00