| commit | 6b648c4697cede14605fd2b89425866eec5f7c79 | [log] [tgz] |
|---|---|---|
| author | alessiob <alessiob@webrtc.org> | Wed Jun 07 18:04:35 2017 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Jun 07 18:04:35 2017 |
| tree | 329f845b03a8bef8207d0cb09541ad30944f3f7a | |
| parent | bb28b35922b195971ecf7f76b3bb9e0a03f57f6e [diff] |
The simulator puts into action the schedule of speech turns encoded in a MultiEndCall instance. The output is a set of audio track pairs. There is one set for each speaker and each set contains one near-end and one far-end audio track. The tracks are directly written into wav files instead of creating them in memory. To speed up the creation of the output wav files, *all* the source audio tracks (i.e., the atomic speech turns) are pre-loaded. The ConversationalSpeechTest.MultiEndCallSimulator unit test defines a conversational speech sequence and creates two wav files (with pure tones at 440 and 880 Hz) that are used as atomic speech turn tracks. This CL also patches MultiEndCall in order to allow input audio tracks with same sample rate and single channel only. BUG=webrtc:7218 Review-Url: https://codereview.webrtc.org/2790933002 Cr-Commit-Position: refs/heads/master@{#18480}
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.