610c76323e
This CL plumbs an additional signal from VideoSendStream down to VideoStreamEncoder, namely the amount of headroom that's left between the encoder max bitrate and the current bitrate allocation for the media track. This will be used in follow-up CLs to tune encoder rate adjustment and some codec specific paramaters a bit differently, based on the knowledge if we are network constrained or not. Bug: webrtc:10155 Change-Id: Ic6ccc79be5c6845468bab65b4ca9918b56923fa4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125981 Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27008}
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