| commit | 84ef615a5da13ffd3d8ac150485afa94ca3343c9 | [log] [tgz] |
|---|---|---|
| author | aleloi <aleloi@webrtc.org> | Thu Aug 04 05:28:21 2016 -0700 |
| committer | Commit bot <commit-bot@chromium.org> | Thu Aug 04 12:28:28 2016 +0000 |
| tree | c8c27098ce5d7fbb8d91f395716c7cf0496e365d | |
| parent | cb56065c62a31d83919abcd4e343ea3dbe029e9f [diff] |
Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine. This is part of rewriting the ConferenceMixer and OutputMixer. Calls are instead routed through AudioReceiveStream::Start/Stop. NOTRY=True Review-Url: https://codereview.webrtc.org/2206223002 Cr-Commit-Position: refs/heads/master@{#13636}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.