Commit graph

7 commits

Author SHA1 Message Date
Sebastian Dröge
303a094779 webrtc: Actually make use of promise created one line before
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1042
2019-12-31 11:19:20 +00:00
Tim-Philipp Müller
f218ec2794 Remove autotools build system 2019-10-14 13:54:27 +01:00
Sebastian Dröge
28b0be4036 rtptransceiver: Remove direction setter and vfunc and replace it by a property
It was changed from a function to a property in the latest WebRTC spec.
2019-08-06 12:22:21 +00:00
Matthew Waters
177aa22bcd webrtc: Initial support for stream addition/removal
Limitations:
- No transport changes at all (ICE, DTLS)
- Codec changes are untested and probably don't work
- Stream removal doesn't remove transports (i.e. non-bundled transports
  will stay around until webrtcbin is shutdown)
- Unified Plan SDP only. No Plan-B support.
2019-05-30 21:33:09 +10:00
Nicolas Dufresne
06f7d1f6c5 examples: Don't install non-prefixed examples 2018-06-06 14:32:56 -04:00
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
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