commit | 66a29b9953c0959cc9639f43323befb22fce8ec7 | [log] [tgz] |
---|---|---|
author | Artem Titov <titovartem@webrtc.org> | Tue Jan 15 13:43:20 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Jan 15 15:06:55 2019 |
tree | aaa47109abe40a2a10727bb351ab2f9ade9f2187 | |
parent | ccc1b57e32bd99e4f220a3db0e540713f4349ad9 [diff] |
Introduce CopyToFileAudioCapturer. It will be used to dump generated audio from TestAudioDeviceModule into user defuned file in peer connection level test framework. Bug: webrtc:10138 Change-Id: I6e3db36aaf1303ab148e8812937c4f9cd1b49315 Reviewed-on: https://webrtc-review.googlesource.com/c/117220 Commit-Queue: Artem Titov <titovartem@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26267}
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.