| commit | d7f47b7ff82bdc23c87dc76b0c20ea5b215b94fd | [log] [tgz] |
|---|---|---|
| author | Helmer Nylen <helmern@google.com> | Mon Jun 09 15:36:00 2025 |
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Wed Jun 11 16:29:33 2025 |
| tree | eb9556351ca4dc08d4ba9699822d20d42704db70 | |
| parent | 9c086c7c7df28e72797115e81f3e0de63eb3ff10 [diff] |
Support setting a list of CSRCs in RtpEncodingParameters This is a reupload of https://webrtc-review.googlesource.com/c/src/+/392980. CSRCs are commonly used to indicate which participants have contributed to a media packet. This change makes it possible to specify the CSRCs for both audio and video tracks. As demonstrated in the integration test, the CSRCs can be set either when adding a track to a peer connection or via a SetParameters call on an existing sender. Bug: b/410811496 Change-Id: I3f8f0b1e22680abb7d83ec27ac8025414dcacaae Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/395800 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Helmer Nylén <helmern@google.com> Reviewed-by: Jonas Oreland <jonaso@webrtc.org> Cr-Commit-Position: refs/heads/main@{#44909}
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.