Fredrik Solenberg 63e6072a43 Add AudioState::audio_transport() to prepare clients for moving ADM initialization out of VoiceEngine.
(See: https://webrtc-review.googlesource.com/c/src/+/23820)

Bug: webrtc:4690
Change-Id: I474a327303aa0c9b5b34c2055ae3a35e466a7d9f
Reviewed-on: https://webrtc-review.googlesource.com/24720
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20810}
2017-11-21 10:51:02 +00:00
2017-11-16 18:25:33 +00:00
2017-11-20 23:18:22 +00:00
2017-11-15 13:31:51 +00:00
2017-11-20 23:18:22 +00:00
2017-11-16 08:50:44 +00:00
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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
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