47a03e8743
This enables send side bandwidth estimation for audio and removes field trial "WebRTC-Audio-SendSideBwe" which this was controlled through. Transport-cc extension still needs to be negotiated. Bug: webrtc:12222 Change-Id: Ie2268fad13703eeb0f0d38fcf484baaa29715b7c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194142 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32681}
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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
- Reporting bugs