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This separates the bitrate configuration logic from other call specific logic, creating a greater separation of concern and simplifying testing. The old call tests are kept but can be removed in the future reducing the dependencies on rtp transport control interface and congestion control in the system, which will simplify future refactoring. This also prepares for moving the bitrate configuration responsibility to the rtp transport controller in a later CL. Bug: webrtc:8415 Change-Id: I97126e89f30b63fc9b5d98a0bed1c29f18a6ed44 Reviewed-on: https://webrtc-review.googlesource.com/54401 Reviewed-by: Zach Stein <zstein@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22124}
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