Edward Lemur 98d4036f5a Make it possible to run low_bandwidth_audio_test on Android swarming.
R=phoglund@webrtc.org, solenberg@webrtc.org

Bug: chromium:755660
Change-Id: I8755a9c9df92fe8157c870cc7519130291441b25
Reviewed-on: https://webrtc-review.googlesource.com/39780
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21634}
2018-01-16 11:42:17 +00:00
2018-01-14 09:35:07 +00:00
2018-01-12 08:15:01 +00:00
2018-01-12 11:31:52 +00:00
2017-09-15 04:25:06 +00:00
2018-01-12 11:31:52 +00:00
2017-09-15 04:25:06 +00:00
2018-01-12 11:31:52 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +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.

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
Languages
C++ 90.3%
Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%