d92288f5babdafe409880ebd170d9a6feb0b9600

Also adjust to base-layer fraction for the shortened 3-tl pattern to be 60%, just like the 2-tl setting. This CL removes direct use of the allocation matrix and moves it behind a static getter. Bug: webrtc:9477 Change-Id: Ifd7d1edffa0555024fd252834357b926997d13b5 Reviewed-on: https://webrtc-review.googlesource.com/86681 Commit-Queue: Erik Språng <sprang@webrtc.org> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23834}
Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream.
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.
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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.
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