Ilya Nikolaevskiy 6003e7afe0 Fix FakeEncoder to produce correct bitrate for several temporal layers
Also fix retransmission video send stream tests to not depend on actual frames sizes

Also, reduce key-frame scaling factor in FakeEncoder to better reflect real encoders behavior.

Bug: none
Change-Id: I33118160f3fec67ae8e732d9a85f0e9ee0784b7c
Reviewed-on: https://webrtc-review.googlesource.com/c/105642
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25165}
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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

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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