Commit graph

100 commits

Author SHA1 Message Date
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
Olivier Crête 581a30d892 rtpsession: The signal has 5 arguments, not 4 2011-06-20 16:47:36 -04:00
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
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 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