Pass initial video CSRCs via RtpConfig WebRtcVideoSendStream called SetCsrcs() right after creating the VideoSendStream to apply the CSRCs from the RTP parameters. Add a csrcs field to RtpConfig, like AudioSendStream::Config::Rtp::csrcs, and pass them at creation instead, so that RtpVideoSender is initialized with them. SetCsrcs() is still used for updates from setParameters(). Bug: webrtc:42223727 Change-Id: Id34d8c8d46b6ad63fa0150c7892b029f5fc6a30d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/506220 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Cr-Commit-Position: refs/heads/main@{#48778}
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.