Sergey Ulanov 0182f85fd1 More reliable ALR detection
Previously AlrDetector was measuring amount of data sent in each 100ms
interval and would enter ALR mode after 5 consecutive intervals when
average bandwidth usage doesn't exceed 30% of the current estimate
estimate. This meant that an application that uses only slightely more
than 6% of total bandwidth may stay out of ALR mode, e.g. if it sends
a frame of size BW*30ms every 0.5 seconds. 100ms is too short interval
to average over, particularly when frame-rate falls below 10fps.

With this change AlrDetector averages BW usage over last 500ms. It then
enters ALR state when usage falls below 30% and exits it when usage
exceeds 50%.

BUG=webrtc:6332
R=philipel@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/2503643003 .

Cr-Commit-Position: refs/heads/master@{#15109}
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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. This page is maintained by the Google Chrome team.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

More info

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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