8b5d9d8650
This is a follow up of a comment in https://webrtc-review.googlesource.com/c/src/+/110105 It was not very useful to split the audio and video report interval since the RTCP module can only either be audio or video. The recent it was written that way in https://webrtc-review.googlesource.com/c/src/+/43201/ was because that was a straightforward transition from two global constants to two variable. Bug: webrtc:8789 Change-Id: I2293de14ba5f363351f379a02022ed5dc7b8d458 Reviewed-on: https://webrtc-review.googlesource.com/c/110824 Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Jiawei Ou <ouj@fb.com> Cr-Commit-Position: refs/heads/master@{#25741}
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 http://www.webrtc.org/native-code/development 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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct