Henrik Boström fd4ddd1fb1 Add a simulcast test that verifies media is flowing on all layers.
Previous tests only asserted that O/A succeeded and that the number of
encodings was as expected. This test goes further and also asserts that
bytesSent eventually becomes non-zero (after an initial ramp-up time).

Let's get testing straight before we add VP9 simulcast support.

Bug: webrtc:14885, webrtc:14884
Change-Id: Idccce66698a077264fa0df2c448c8474d2439aea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291960
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39271}
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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 here 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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