Commit graph

14 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
Olivier Crête b5f2de3124 webrtc lib: Make the transceiver struct private
This will prevent any unsafe access.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2241>
2021-06-21 20:53:09 +00:00
Olivier Crête ed1f0f33a2 webrtcbin: Remember if a transceiver had a forced m-line
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2104>
2021-04-12 17:55:07 -04:00
Olivier Crête df8d29e9c3 webrtcbin: Remove remnant of non-rtcp-mux mode
There was some code left that wasn't used anymore.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1930>
2021-01-06 23:02:37 +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
Olivier Crête 0fbbdc5734 rtptransceiver: Store the SSRC of the current stream
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
2020-10-30 16:23:10 -04:00
Sebastian Dröge 849839ba97 Revert "rtptransceiver: Store the SSRC of the current stream"
This reverts commit d1da271f25.

It breaks the CI until the C# bindings are fixed.
2020-10-08 18:53:07 +03:00
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
Olivier Crête d1da271f25 rtptransceiver: Store the SSRC of the current stream
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
2020-10-06 16:49:08 -04:00
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 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