| commit | 116ec6da50c7c82b2341b97e2d6d512ade3f6e00 | [log] [tgz] |
|---|---|---|
| author | aleloi <aleloi@webrtc.org> | Wed Oct 12 13:07:07 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Oct 12 13:07:13 2016 |
| tree | 5f7f8dac580f8caf5eba8952cf6eafa778cf7d78 | |
| parent | 7e30432b3663ca64aa4776becc536b4a64fc45c9 [diff] |
Implemented further mixer interface change suggestions from https://codereview.webrtc.org/2386383003/ Changed mixability status into AddSource/RemoveSource. Added 'ssrc()' method to the MixerSource interface. Removed unnecessary member 'num_audio_sources_' and made the mixer be refcounted. BUG=webrtc:6346 NOTRY=True Review-Url: https://codereview.webrtc.org/2408683002 Cr-Commit-Position: refs/heads/master@{#14612}
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.