Pipe CSRCs down through the audio and video send streams This is a reupload of https://webrtc-review.googlesource.com/c/src/+/392961. This CL exposes methods on the audio and video send streams to set CSRCs on the underlying senders. The necessary support for this is added in the parent CL. Bug: b/410811496 Change-Id: I8f8a7237c3348801ec43402dfa42b2835f557ff1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/395760 Commit-Queue: Helmer Nylén <helmern@google.com> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#44908}
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.
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Authoritative list of directories that contain the native API header files.