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12 commits

Author SHA1 Message Date
Thibault Saunier 019971a3c7 Move files from gst-plugins-bad into the "subprojects/gst-plugins-bad/" subdir 2021-09-24 16:14:36 -03:00
Mathieu Duponchelle 86c009e7aa webrtc: expose transport property on sender and receiver 
As advised by !1366#note_629558 , the nice transport should be
accessed through:

> transceiver->sender/receiver->transport/rtcp_transport->icetransport

All the objects on the path can be accessed through properties
except sender/receiver->transport. This patch addresses that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1952>
 2021-01-13 19:22:42 +00:00
Olivier Crête 5d5417f271 webrtc: Remove non rtcp-mux code 
RTCP mux is now always required by the WebRTC spec

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765>
 2020-11-24 01:59:55 +00:00
Olivier Crête 80a56c25a6 webrtc: Set the DSCP markings based on the priority 
This matches how the WebRTC javascript API works and the Chrome implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
 2020-10-30 16:24:40 -04:00
Sebastian Dröge b565a7ef66 Revert "webrtc: Set the DSCP markings based on the priority" 
This reverts commit 8ba08598bb.

It breaks the CI until the C# bindings are fixed.
 2020-10-08 18:53:00 +03:00
Olivier Crête 8ba08598bb webrtc: Set the DSCP markings based on the priority 
This matches how the WebRTC javascript API works and the Chrome implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
 2020-10-06 16:49:08 -04:00
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
Matthew Waters c3c4b07ad3 webrtc/transceiver: add a set_direction function 
Matches the setDirection() from the W3C spec and allows changing the
transceiver direction at the next negotiation cycle.
 2019-05-30 21:33:09 +10:00
Matthew Waters 5ecca0bb22 webrtc: move some functions to the appropriate files 2018-11-26 16:07:57 +11:00
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