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
Olivier Crête
a630c68fc3
rtpsession: Don't relay more than one PLI request per RTT
...
Drop PLI requests if one was relay in the last RTT, the other side may
just not have received the keyframe yet.
2011-02-01 18:28:51 +01:00
Olivier Crête
a61bb9e94b
rtpsession: Send GstForceKeyUnit event in response to received RTCP PLI
2011-02-01 18:28:51 +01:00
Olivier Crête
52f95fa7ee
rtpsession: Implement sending PLI packets in response to GstForceKeyUnit
2011-02-01 18:28:51 +01:00
Olivier Crête
db5150a23a
rtpsource: Retain RTCP Feedback packets for a specified amount of time
2011-02-01 18:28:51 +01:00
Olivier Crête
90354ecb49
rtpsession: Make rtcp buffer metadata writable after processing it
...
Functions that process the rtcp buffer could decide to keep a ref
on the buffer for further processing. So make the metadata writable
only after they are done.
2011-02-01 18:28:50 +01:00
Olivier Crête
1643f427db
rtpsession: Emit signal on incoming RTCP FB packet
2011-02-01 18:28:50 +01:00
Wim Taymans
f399b6a641
rtpsession: fix compilation
2011-02-01 18:28:50 +01:00
Olivier Crête
1bde427250
rtpsession: Add method to request early RTCP packet
...
Implement the early mode defined in RFC 4585. In this mode, RTCP feedback
packets are sent early to notifier.
2011-02-01 17:03:39 +01:00
Olivier Crête
975e1fecb3
rtpsession: Add property for minimum interval between Regular RTCP messages
...
This can be changed according to RFC 4585
2011-02-01 16:56:15 +01:00
Olivier Crête
cdb5465741
rtpsession: Emit signal when sending a compound RTCP packet
...
This allows users to add extra RTCP packets to the compound
RTCP packet.
2011-02-01 16:50:58 +01:00
Olivier Crête
9f073459e0
rtpsession: Emit "on-ssrc-validated" when validating by RTCP
...
Emit "on-ssrc-validated" if the SSRC is validated by receiving
a RTCP SDES packet.
2011-02-01 16:45:58 +01:00
Wim Taymans
8fa5ddab9a
rtpsession: remember last sent RB values.
2010-12-23 13:58:30 +01:00
Wim Taymans
10a5a795ea
rtpsession: also emit RTCP activity on SR
...
Also emit RTCP activity signals when we receive an SR packet without RB blocks,
such as from a sender that is not receiving anything.
2010-12-23 13:58:30 +01:00
Wim Taymans
0c3333da04
session: fix average RTCP packet size some more.
...
Fix stupid error in averaging macro.
Include udp headers in packet length estimation.
2010-12-14 18:12:43 +01:00
Wim Taymans
7ebd374766
rtpbin: correctly calculate RTCP packet size
2010-12-14 17:15:23 +01:00
Stefan Kost
d8167e3071
various (gst): add a missing G_PARAM_STATIC_STRINGS flags
2010-10-13 18:00:28 +03:00
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
8337c89c74
rtpsession: fix compilation
2010-09-24 14:10:26 +02:00
Havard Graff
062568a9f5
rtpsession: relax third-party collision detection
...
If the source has been inactive for some time, we assume that it has
simply changed its transport source address. Hence, there is no true
third-party collision - only a simulated one.
Fixes #630447
2010-09-24 13:56:56 +02:00
Olivier Crête
1f17b334ff
rtpsession: Calculate RTCP bandwidth as a fraction of the RTP bandwidth
...
Calculate the RTCP bandwidth to be a fraction of the RTP bandwidth if it is
specified as a value between 0 and 1.
2010-09-13 15:51:20 +02:00
Wim Taymans
8381d9788d
session: improve bandwidth recalculation
...
Also recalculate bandwidth when one of the source bandwidths changed.
Use the newly calculated bandwidth.
2010-09-13 15:51:20 +02:00
Olivier Crête
6f53a2b240
rtpsession: Add the option to auto-discover the RTP bandwidth
2010-09-13 15:51:19 +02:00
Olivier Crête
94e87ef8ee
rtpsession: Initialise the average scaled by 16
2010-09-13 13:10:19 +02:00
Wim Taymans
e6db74764b
rtpsession: add running_time argument docs
2010-09-13 12:41:56 +02:00
Wim Taymans
cb6de429a0
rtpsession: compute the average correctly scaled
2010-09-13 12:31:40 +02:00
Olivier Crête
64e4ffa25b
rtpsession: Count sent RTCP packets after they have been finished
...
If they are counted before calling gst_rtcp_buffer_end(), then the
size is way too big.
2010-09-13 12:13:23 +02:00
Wim Taymans
50f26c671b
rtpsession: fix return value
2010-05-07 19:06:35 +02:00
Wim Taymans
69cde0e874
rtpsession: add properties to configure bandwidths
...
Add properties to configure the sender and receiver bandwidths.
Configure the bandwidths before calculating the RTCP timeout when we need to.
2010-05-07 18:57:13 +02:00
Wim Taymans
6eee730c4a
rtpsession: handle NONE RTCP intervals
...
Prepare for handling RTCP reporting intervals of GST_CLOCK_TIME_NONE, which
means don't send RTCP at all.
2010-05-07 13:32:30 +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
5a4ecc9da1
rtpbin: remove more ntpnstime and cleanups
...
Remove some code where we pass ntpnstime around, we can do most things with the
running_time just fine.
Rename a variable in the ArrivalStats struct so that it's clear that this is the
current system time.
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
Wim Taymans
05418f1687
rtpbin: avoid some structure copies
...
Don't make copied in the getter and setter for SDES in the RTPSource. This
avoids a couple of copies of the SDES structure when generating RTCP
packets.
2009-12-22 22:27:21 +01:00
Pascal Buhler
c3448f978e
rtpmanager: improve SDES handling
...
Store SDES internally as a struct to support multiple PRIV values.
Include all values set in SDES struct when sending RTCP SDES.
2009-12-22 21:43:25 +01:00
Aurelien Grimaud
07f27f0efd
rtpsession: avoid buffer ref/unref pairs for CSRCs
...
We ref the buffer before pushing it downstream in order to get the CSRCs of it
after pushing. This causes performance problems when downstream elements want to
change the metadata because the buffer needs to be subbuffered.
Instead, read and store the CSRCs of the buffer in an array before pushing it
and process the array after pushing the buffer. This allows us to remove the
ref/unref pair.
Fixes #603376
2009-11-30 15:59:50 +01:00
Wim Taymans
3f629f6001
rtpsession: whitespace fixes
2009-09-03 19:16:59 +02:00
Wim Taymans
3091137217
rtpsession: avoid doing lip-sync in BYE
...
When we get a BYE packet, don't do lip-sync with the SR inside because some
senders have trouble constructing valid SR packets after BYE.
2009-08-11 02:30:47 +01:00
Wim Taymans
3747ede14a
rtpbin: don't do lip-sync after a BYE
...
After a BYE packet from a source, stop forwarding the SR packets for lip-sync
to rtpbin. Some senders don't update their SR packets correctly after sending a
BYE and then we break lip-sync. We prefer to let the jitterbuffers drain with
the current lip-sync instead.
2009-08-11 02:30:47 +01:00
Wim Taymans
d2ef095b80
rtpbin: only reconsider once for BYE
...
When iterating the sources of a BYE packet, don't signal a reconsideration for
each of them but signal after we handled all sources.
2009-08-11 02:30:47 +01:00