d3416972dd
Record the RMSE of encoding bitrate diff from output stream to target bitrate set by WebRTC when the AV1/VP9 stream is removed. Design doc: https://docs.google.com/document/d/1tE7Fquj1U6CwZ01QxxjoCaJS2o0LFYyIxvstLDx0qB0/edit#heading=h.c11ebhaja9tg Bug: chromium:1358750 Change-Id: Ifa6f5aacfa8f6b0835f4609f102ae3ae1ee2897e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298684 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39907}
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.
Development
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.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation