82fad3d513
As the rate allocation has been moved into entirely into SimulcastRateAllocator, and the listeners are thus no longer needed, this class doesn't fill any other purpose than to determine if ScreenshareLayers or TemporalLayers should be created for a given simulcast stream. This can however be done just from looking at the VideoCodec instance, so changing this into a static factory method. Due to dependencies from upstream projects, keep the class name and field in VideoCodec around for now. Bug: webrtc:9012 Change-Id: I028fe6b2a19e0d16b35956cc2df01dcf5bfa7979 Reviewed-on: https://webrtc-review.googlesource.com/63264 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22529}
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