Commit graph

18 commits

Author SHA1 Message Date
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
Olivier Crête
78c687da3e webrtc: Document more objects
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
2020-10-30 16:23:10 -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
Sebastian Dröge
f12265d9c5 Revert "webrtc: Document more objects"
This reverts commit ad68c6b1eb.

It breaks the CI until the C# bindings are fixed.
2020-10-08 18:52:50 +03:00
Sebastian Dröge
74a42c5ba8 Revert "webrtc: Add hotdoc style since tags"
This reverts commit 63a5fa818c.

It breaks the CI until the C# bindings are fixed.
2020-10-08 18:51:37 +03:00
Olivier Crête
63a5fa818c webrtc: Add hotdoc style since tags
We're stuck having to add a separate comment for now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
2020-10-06 16:52:48 -04:00
Olivier Crête
ad68c6b1eb webrtc: Document more objects
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1425>
2020-10-06 16:49:08 -04:00
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
Niels De Graef
7af1a4566f Use G_DEFINE_AUTOPTR_CLEANUP_FUNC unconditionally
Since we started depending on GLib 2.44, we can be sure this macro is
defined (it will be a no-op on compilers that don't support it).
2019-06-05 08:12:10 +02:00
Thibault Saunier
7fe3f36ac8 Minor documentation fixes 2019-05-13 11:36:27 -04:00
Niels De Graef
39c8c206be webrtc: Add g_autoptr() support for public types 2019-05-08 15:47:06 +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
Tim-Philipp Müller
333f636555 webrtc: GST_EXPORT -> GST_WEBRTC_API
We need different export decorators for the different libs.
For now no actual change though, just rename before the release,
and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:36:33 +00:00
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