Fredrik Solenberg 73276ad7ed - Removes voe_conference_test.
- Adds a new AudioStatsTest, with better coverage of the same features, based on call_test.
- Adds an AudioEndToEndTest utility, which AudioStatsTest and LowBandwidthAudioTest uses.

BUG=webrtc:4690
R=kwiberg@webrtc.org

Review-Url: https://codereview.webrtc.org/3008273002 .
Cr-Commit-Position: refs/heads/master@{#19833}
2017-09-14 12:46:50 +00:00
2017-09-14 12:46:50 +00:00
2017-06-30 10:04:59 +00:00
2017-06-30 10:04:59 +00:00
2017-01-20 20:45:07 +00:00

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.

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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