Commit graph

402 commits

Author SHA1 Message Date
Mark Nauwelaerts
3a98f6f0fd rtpssrcdemux: keep a ref on the src pad while using it
Prevent a possible race if clear_ssrc() is called between getting the pad and
doing the push.

Based on patch by <olivier.crete@collabora.com>

https://bugzilla.gnome.org/show_bug.cgi?id=650916
2011-07-28 14:51:01 +02:00
Olivier Crête
c7b9b98648 rtpssrcdemux: Make the pads lock recursive and hold it across the signal emit
We need to keep the lock held because we don't want a push before the "new-ssrc-pad"
handler has completed. But we may want to push an event from inside that handler, hence
the recursive mutex.

https://bugzilla.gnome.org/show_bug.cgi?id=650916
2011-07-28 14:50:59 +02:00
Olivier Crête
e26b5391c2 rtpssrcdemux: Use PADs lock
https://bugzilla.gnome.org/show_bug.cgi?id=650916
2011-07-28 14:50:57 +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
Miguel Angel Cabrera Moya
977a5eee7a rtpjitterbuffer: return correct type when assertion fails 2011-06-24 11:59:01 +02:00
Olivier Crête
581a30d892 rtpsession: The signal has 5 arguments, not 4 2011-06-20 16:47:36 -04:00
Sebastian Dröge
b694bfeca3 ssrcdemux: Fix uninitialized variable compiler warning for (pre-) releases too 2011-05-17 10:47:32 +02:00
Sebastian Dröge
0f05d3e5a5 rtpssrcdemux: Fix uninitialized variable compiler warning 2011-05-17 09:24:08 +02:00
Olivier Crête
b6bfc512e8 ssrcdemux: Implement iterate internal links for sink pads
https://bugzilla.gnome.org/show_bug.cgi?id=649617
2011-05-17 09:22:29 +02:00
Olivier Crête
23b6c8febc rtpssrcdemux: iterate pad function is only valid for src pads
The iterate function is only used for src pads, so mark it as such and remove
dead code.

https://bugzilla.gnome.org/show_bug.cgi?id=649617
2011-05-17 09:22:25 +02:00
Olivier Crête
1bf94a92b0 rtpssrcdemux: Release lock before emitting signal
If the lock is not released before emitting a signal, it may cause a deadlock
if any other function in the element is called.

Also removed an unused timestamp parameter

https://bugzilla.gnome.org/show_bug.cgi?id=649617
2011-05-17 09:22:20 +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
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
0d2d52856f rtpssrcdemux: Unknown SSRC is not fatal
https://bugzilla.gnome.org/show_bug.cgi?id=646966
2011-04-11 17:37:58 -04:00
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
Thibault Saunier
b541208b77 android: Make it ready for androgenizer
Remove the android/ top dir
Fixe the Makefile.am to be androgenized

To build gstreamer for android we are now using androgenizer which generates the needed Android.mk files.
Androgenizer can be found here: http://git.collabora.co.uk/?p=user/derek/androgenizer.git
2011-04-11 01:20:11 +02:00
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
Havard Graff
e71a908d96 jitterbuffer: Make src_query MT-safe
It is possible that the element might be going down while the event arrives
2011-04-08 15:23:05 +02:00
Sebastian Dröge
4c36ca30b2 jitterbuffer: Unref event if the parent element disappeared 2011-04-08 15:22:19 +02:00
Havard Graff
342686bb02 jitterbuffer: Make upstream events MT-safe 2011-04-08 15:21:46 +02:00
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
Mark Nauwelaerts
e5bcaa45e6 Revert "jitterbuffer: reset element base_time upon flush"
This reverts commit f84b8a69cb.

Fixes bug #646397.
2011-04-04 11:49:00 +02:00
Mark Nauwelaerts
6bc1aa0e59 jitterbuffer: handle position query 2011-03-09 17:18:08 +01:00
Mark Nauwelaerts
1f7f434df6 jitterbuffer: also estimate eos if very near eos 2011-03-07 16:56:43 +01:00
Mark Nauwelaerts
3c9a4239bf jitterbuffer: avoid trying to buffer more than is available.
That is, in case of short (or near eos of) stream, deadlock (until timeout)
would occur trying to buffer more than is yet forthcoming.
2011-03-07 16:56:18 +01:00
Mark Nauwelaerts
f84b8a69cb jitterbuffer: reset element base_time upon flush
... to arrange for properly scheduled timeout (following seek).
2011-03-07 11:07:12 +01:00
Blaise Gassend
0f88181f43 rtpbin: handle NULL demux elements
When using gstrtpbin with ignore-pt=true, the free_stream function tries to
call gst_element_set_locked_state and gst_element_set_state on a stream->demux
which is NULL.

fixes #642412
2011-02-22 13:31:35 +01:00
Wim Taymans
45ea930a99 rtpbin: fix setting the SDES property
Only the sdes veriable is protected with the object lock.
Use the right object when setting the sdes property.
2011-02-21 17:19:05 +01:00
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
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
Sjoerd Simons
7350d2adfa gstrtpsession: Fallback for FIR to PLI if PLI isn't available 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