3c24ea8340
SetTransportOverhead was used by send streams to signal the packet overhead that they received from Call. However, call receives the value from OnNetworkRouteChanged in WebRtcVideoChannel and WebRtcVoiceMediaChannel which is already propagated to RtpTransportControllerSend. By skipping the round trip, the interface on the rtp transport controller can be reduced. Bug: None Change-Id: I759b1207aab214bbc2b993106f6ff7cc24e177f7 Reviewed-on: https://webrtc-review.googlesource.com/57182 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22226}
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