| commit | 33397438780d5817ddfc0207e17ca27d6ed1a786 | [log] [tgz] |
|---|---|---|
| author | alessiob <alessiob@webrtc.org> | Tue Mar 28 12:39:59 2017 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Mar 28 12:39:59 2017 |
| tree | f2f304cb4603ba630fc83789533f6583205c7d3c | |
| parent | 83862e3c14a39065cb87e86a61536355f1c9284f [diff] |
MultiEndCall is responsible for analyzing and validating timing information and audiotracks with which a multi-end call can be simulated. The class creates one WavReaderInterface object for each unique audiotrack and builds the set of speaker names. Validating if the audiotrack lengths and the timing information are compatible (and hence valid) is not implemented yet. MultiEndCall is designed using dependency injection. This allows to use mock objects with which we can quickly simulate different timings and track lengths without needing actual wav files. BUG=webrtc:7218 Review-Url: https://codereview.webrtc.org/2761853002 Cr-Commit-Position: refs/heads/master@{#17421}
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.