Bind default sink to existing unsignaled receive stream WebRtcVideoReceiveChannel::SetDefaultSink() previously only updated the cached default_sink_ pointer. If the first RTP packet for an unsignaled SSRC arrived before the application attached a sink (e.g. the remote track exists but AddSink had not yet been called), the default receive stream was created with a null sink, and the later SetDefaultSink call had no way to reach it. Frames were then silently dropped until the unsignaled SSRC changed and the stream was recreated. Propagate the new sink to the already-existing unsignaled receive stream, matching the behavior of WebRtcVoiceReceiveChannel:: SetDefaultRawAudioSink(). Bug: None Change-Id: I2c0e90c054b7118e752ca637dab709b5b97440aa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/475840 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#47823}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
See here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.