Jesús de Vicente Peña 9558192711 AEC3: Removing the need of a buffer for the stationarity estimator of the render signal.
Change-Id: I6983e1d8bdd048a5d92209e3023c687f82d383d5

Bug: webrtc:9193,chromium:836790
Change-Id: I6983e1d8bdd048a5d92209e3023c687f82d383d5
Reviewed-on: https://webrtc-review.googlesource.com/72760
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23065}
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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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