Commit graph

12 commits

Author SHA1 Message Date
Olivier Crête
a6593753a5 webrtc lib: Make the rtpsender 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
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
52c676546d webrtc: Also remove rtcp_transport from the structure
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765>
2020-11-24 01:59:55 +00:00
Olivier Crête
c5d76d944e webrtc: Remove APIs to set transport on sender/receiver
They're not not used ever.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765>
2020-11-24 01:59:55 +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
cca313ecd8 rtpsender: Add API to set the priority
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
2020-10-30 16:24:40 -04:00
Sebastian Dröge
a40d6f4994 Revert "rtpsender: Add API to set the priority"
This reverts commit a8b287c764.

It breaks the CI until the C# bindings are fixed.
2020-10-08 18:53:10 +03:00
Olivier Crête
a8b287c764 rtpsender: Add API to set the priority
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
2020-10-06 16:49:08 -04:00
Mathieu Duponchelle
42adb02a10 docstrings: port ulinks to markdown links 2019-08-23 20:14:12 +02:00
Sebastian Dröge
b1ca76377f webrtc: Remove unused parameter from rtpsender constructor
https://bugzilla.gnome.org/show_bug.cgi?id=794363
2018-03-16 10:37:24 +02:00
Sebastian Dröge
950ead9215 webrtc: Add some locks to setters and remove non-existing functions from headers
https://bugzilla.gnome.org/show_bug.cgi?id=794363
2018-03-16 10:37:24 +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