GNOME Bugzilla – Bug 792523
new webrtc plugin
Last modified: 2018-02-02 04:04:05 UTC
Contributing a webrtcbin element that will send/receive rtp payloaded streams using a WebRTC-like API. Code is available from https://github.com/ystreet/gst-plugins-bad/commits/webrtc and will be squashed before pushing.
Cool! Should the headers in the lib be guarded by GST_USE_UNSTABLE_API like in the other -bad libs? The class and instance structs in the lib should probably have padding?
Done!
Let's get it in then!
commit 1894293d6378c69548d974d2965e9decc1527654 Author: Matthew Waters <matthew@centricular.com> Date: Tue Jan 31 20:56:59 2017 +1100 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