| commit | 4496809b28d8f10506b1b622a6cfd9d930c60617 | [log] [tgz] |
|---|---|---|
| author | aleloi <aleloi@webrtc.org> | Mon Aug 08 10:18:58 2016 -0700 |
| committer | Commit bot <commit-bot@chromium.org> | Mon Aug 08 17:19:03 2016 +0000 |
| tree | 61b8810b9650096ee677e1be81dd05a51aa72a3a | |
| parent | 0727f15186fb05f55d69d87ff9633ccdade10258 [diff] |
Changed mixing api and moved resampler. Removed resampler from NewAudioConferenceMixer and AudioMixer (which started as a copy of former OutputMixer). This is part of the mixer rewrite project. In particular, this is one of the steps required to have a single mixing component instead of two doing the same thing. The next planned change (which is not part of this CL) is to plug in the new mixer (NewAudioConferenceMixer) into AudioState and AudioDeviceModule. NOTRY=True Review-Url: https://codereview.webrtc.org/2221443002 Cr-Commit-Position: refs/heads/master@{#13674}
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.