| commit | a9cc40b7d263cfce28bd3f126481e99283623d75 | [log] [tgz] |
|---|---|---|
| author | peah <peah@webrtc.org> | Thu Jun 29 15:32:09 2017 |
| committer | Commit Bot <commit-bot@chromium.org> | Thu Jun 29 15:32:09 2017 |
| tree | 4bb326429dd6fae5b020bae7a40193db5e3caf50 | |
| parent | 3dbfac3515db2da6a69c00c5061add0e2eb97ead [diff] |
Allow an external audio processing module to be used in WebRTC [This CL is a rebase of an original CL by solenberg@: https://codereview.webrtc.org/2948763002/ which in turn was a rebase of an original CL by peah@: https://chromium-review.googlesource.com/c/527032/] Allow an external audio processing module to be used in WebRTC This CL adds support for optionally using an externally created audio processing module in a peerconnection. The ownership is shared between the peerconnection and the external creator of the module. As part of this the internal ownership of the audio processing module is moved from VoiceEngine to WebRtcVoiceEngine. BUG=webrtc:7775 Review-Url: https://codereview.webrtc.org/2961723004 Cr-Commit-Position: refs/heads/master@{#18837}
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.