Sergey Silkin db3d1c611e Remove tests with non-zero packet loss.
Concealment is never used in WebRTC since we never feed decoders with
broken bitstream. If so, there is no need to evaluate concealment
quality.
But if we still want to evaluate it then the tests should be
redesigned: recovery frames should be generated with reasonable
interval and quality thresholds should be set to acceptable level.

Bug: webrtc:8524
Change-Id: Ie7197e0a5a88aafcb3b2698185edcb43b71fae3b
Reviewed-on: https://webrtc-review.googlesource.com/32303
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21230}
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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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