8f83b42946
Moving usage of bitrate configuration related interface from Call interface to the corresponding methods in the RtpSendTransportController interface. SetBitrateConfig was replaced with SetSdpBitrateParameters SetBitrateConfigMask was replaced with SetClientBitratePreferences OnNetworkRouteChanged was replaced with OnNetworkRouteChanged This makes it more clear that RtpSendTransportController owns bitrate configuration and fits a longer term ambition to reduce the scope of the Call class. Bug: webrtc:8415 Change-Id: I6d04eaad22a54ecd5ed60096e01689b0c67e9c65 Reviewed-on: https://webrtc-review.googlesource.com/54365 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22131}
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