| commit | 709ed67c38d0a942f3bf3e68e337cc27a27bc353 | [log] [tgz] |
|---|---|---|
| author | Fredrik Solenberg <solenberg@webrtc.org> | Tue Sep 15 10:26:33 2015 |
| committer | Fredrik Solenberg <solenberg@webrtc.org> | Tue Sep 15 10:26:45 2015 |
| tree | 3ae0b9a103e22f53136118731b450a32888c117a | |
| parent | 4ae28a10701c82e7bcdd32f075bb89f43d745c95 [diff] |
Move instantiation of webrtc::Call into a MediaController class so that it can be used for both audio and video media channels. I'm not super happy with the GetVoE() function added on MediaEngineInterface, but this will eventually be gone, once webrtc::Call owns the shared VoE state (or initially, maps ADM* to an implicitly created VoE). BUG=webrtc:4690 R=pbos@webrtc.org, pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1269863005 . Cr-Commit-Position: refs/heads/master@{#9939}
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.