| commit | 307a0922c536317f04ff05fe8a8d4bc3ea75b487 | [log] [tgz] |
|---|---|---|
| author | Tze Kwang Chin <tkchin@webrtc.org> | Mon Mar 21 13:57:40 2016 -0700 |
| committer | Tze Kwang Chin <tkchin@webrtc.org> | Mon Mar 21 20:58:01 2016 +0000 |
| tree | 2380028fc55a142ad332b855ec241bbe72a9f58d | |
| parent | bc73fe1aadb87b2fe6f844cc14df2f1b52f0e22e [diff] |
Support delayed AudioUnit initialization. Applications can choose to decide when to give up control of the AVAudioSession to WebRTC. Otherwise, behavior should be unchanged. Adds a toggle to AppRTCDemo so developers can see the different paths. BUG= R=haysc@webrtc.org Review URL: https://codereview.webrtc.org/1822543002 . Cr-Commit-Position: refs/heads/master@{#12080}
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.