commit | 63e6072a4356a478dee8dce6cc87878dd77295cb | [log] [tgz] |
---|---|---|
author | Fredrik Solenberg <solenberg@webrtc.org> | Mon Nov 20 21:12:21 2017 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Nov 21 10:51:02 2017 |
tree | 818f0f3777f0b9c696fe92bae289fd3d38c75651 | |
parent | d6c98c020aac5f9d07f0218b5bf34b5eb0029f44 [diff] |
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}
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.
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.