Commit graph

119 commits

Author SHA1 Message Date
Matthew Waters
7bf18ad258 webrtc: start in the closed state
This means that we will reject all operations before we've transitioned
into READY.

This also fixes the tests using the default GMainContext in the NULL
state instead of the webrtcbin internal GMainContext and thread.  Also
removes a potential ordering race where on the element transitioning to
READY, an operations could have been queued on two different threads and
removing a guarentee on operation ordering.
2018-10-08 21:56:31 +11:00
Aleix Conchillo Flaqué
c4fe52395b webrtcbin: start and stop thread when changing state
It might be possible that if we set webrtcbin to the NULL state some
tasks (idle sources) are still executed and they might even freeze. The freeze
is caused because the webrtcbin tasks don't hold a reference to webrtcbin and
if it's last unref inside the idle source itself this will not allow the main
loop to finish because the main loop is waiting on the idle source to finish.

We now start and stop webrtcbin thread when changing states. This will allow
the idle sources to finish properly.

https://bugzilla.gnome.org/show_bug.cgi?id=797251
2018-10-08 13:46:55 +11:00
Matthew Waters
07e9374eff webrtcbin: add support for data channels based on SCTP
Mostly follows the W3C specification
https://www.w3.org/TR/webrtc/#peer-to-peer-data-api

With contributions from:
Mathieu Duponchelle <mathieu@centricular.com>

https://bugzilla.gnome.org/show_bug.cgi?id=794351
2018-09-21 19:45:12 +10:00
Matthew Waters
cf46d49b1e webrtcbin: functionify dependent element checks 2018-09-21 19:36:52 +10:00
Mathieu Duponchelle
1d6160d59c webrtcbin: New add-turn-server API
It is possible and often desirable to pass multiple ICE relays
to libnice agents, the "turn-server" property, while convenient
to use from the command line, does not allow that.

This adds a new action signal, "add-turn-server" to address that.

https://bugzilla.gnome.org/show_bug.cgi?id=797012
2018-09-19 16:17:24 +02:00
Sam Gigliotti
1b095e9609 webrtc: fix memory leak
When it parses SDP, it doesn't free the error object.

https://bugzilla.gnome.org/show_bug.cgi?id=796830
2018-07-19 14:30:55 +01:00
Jan Schmidt
e6a564216d webrtc: Add a warning in sdp_media_from_transceiver()
When generating caps with no ssrc, at least throw a
warning instead of using an uninitialised stack variable

https://bugzilla.gnome.org/show_bug.cgi?id=796810
2018-07-15 23:07:21 +10:00
Jan Schmidt
76a93da2a7 webrtc: Fix memory leak
Fix a leaked string when building RTX info.
2018-07-14 23:20:13 +10:00
Mathieu Duponchelle
6fd3e2a400 find_codec_preferences: use received caps
When negotiation is triggered by receiving caps on our sink pad
probes, we could encounter a race condition where need-negotiation
is emitted and the application requires the creation of an offer
before the current caps were actually updated.

This led to retrieving incomplete caps when creating the offer,
using find_codec_preferences -> pad_get_current_caps.

Instead, as we save the caps in the probe callback anyway, it is better
and thread safe to use these if they were set.

https://bugzilla.gnome.org/show_bug.cgi?id=796801
2018-07-12 21:39:41 +02:00
Jan Schmidt
27b28f3aec webrtc: Explicitly initialise mutex and condition
Fixes random crashes when an allocated webrtcbin isn't
given fresh 0-filled memory in its allocation. It works
mostly because GMutex and GCond are automatically initialised
in that case.
2018-07-01 10:44:45 +10:00
Tim-Philipp Müller
6f46792f0f webrtc: Update for g_type_class_add_private() deprecation in recent GLib 2018-06-24 00:17:26 +02:00
Mathieu Duponchelle
33c7af8845 webrtcbin: copy sticky events on our ghostpads
This lets users call gst_pad_get_current_caps on newly-added
pads to easily determine what to plug them into.

We cannot copy sticky events unconditionally in core,
see #719437

https://bugzilla.gnome.org/show_bug.cgi?id=796387
2018-05-29 13:07:30 +02:00
Tim-Philipp Müller
ed7a98d45b webrtcbin: rtpstorage takes a 64-bit integer for "size-time" property
https://bugzilla.gnome.org/show_bug.cgi?id=796429
2018-05-28 10:43:37 +01:00
Mathieu Duponchelle
5c450c5992 webrtcbin: implement support for FEC and RTX
https://bugzilla.gnome.org/show_bug.cgi?id=795044
2018-05-09 14:46:14 +02:00
Sebastian Dröge
8375e33965 webrtcbin: Remove parameter from gst_webrtc_rtp_sender_new() 2018-03-16 11:07:01 +02:00
Matthew Waters
6f50d35246 webrtc: Fix ffeb09e4 conditional
Fixes ffeb09e4ab

if (sscanf(...)) {  // != 0
  error;
}

Is not correct where != 0 indicates some kind of success.

Check instead that the correct number of elements were slurped.
2018-02-08 15:51:35 +11:00
Matthew Waters
7e6b4dcb49 webrtc: change dead code to an assert
CID #1429140
2018-02-08 15:47:33 +11:00
Matthew Waters
ffeb09e4ab webrtc: bail on invalid rtpbin names
If we fail parsing rtpbin pad names, someone has screwed up so critical
and return.

CID #1429142
2018-02-08 15:29:56 +11:00
Matthew Waters
1894293d63 webrtcbin: an element that handles the transport aspects of webrtc connections
SDP's are generated and consumed according to the W3C PeerConnection API
available from https://www.w3.org/TR/webrtc/

The SDP is either created initially from the connected
sink pads/attached transceivers as in the case of generating an offer or
intersected with the connected sink pads/attached transceivers as in
the case for creating an answer.  In both cases, the rtp payloaded streams
sent by the peer are exposed as separate src pads.

The implementation supports trickle ICE, RTCP muxing, reduced size RTCP.

With contributions from:
Nirbheek Chauhan <nirbheek@centricular.com>
Mathieu Duponchelle <mathieu@centricular.com>
Edward Hervey <edward@centricular.com>

https://bugzilla.gnome.org/show_bug.cgi?id=792523
2018-02-02 15:02:21 +11:00