Pipe CSRCs down through the audio and video send streams This CL is part of a chain. It exposes methods on the audio and video send streams to set the CSRCs on the underlying senders (support for this is added in https://webrtc-review.googlesource.com/c/src/+/392940). These methods are used in https://webrtc-review.googlesource.com/c/src/+/392980. Bug: b/410811496 Change-Id: I5e2445c70152724a9837634112e148e71d180ef5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392961 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Helmer Nylén <helmern@google.com> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#44825}
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.