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If the bandwidth is just on the edge of being able to enable a new stream, the keyframe generated when it is enabled might be large enough to trigger an overuse and force the stream off again. To avoid toggling, this CL adds hysteresis so that the available bandwidth needs to be above X% to start bitrate in order to enable the stream. It will be shut down once available bitrate falls below the original enabling bitrate. For screen content, X defaults to 35. For realtime content, X defaults to 0. Both can be individually modified via field trials. Bug: webrtc:9734 Change-Id: I941332d7be7f2a801d13d9202b2076d330e7df32 Reviewed-on: https://webrtc-review.googlesource.com/100308 Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24745}
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