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The conversational_speech::Timing class models a list of turns. Each turn, is identified by a speaker, the audiotrack name, and an offset in milliseconds. The unit test checks that an instance of Timing is correctly populated and that save/reload leads to the same data. BUG=webrtc:7218 Review-Url: https://codereview.webrtc.org/2750353002 Cr-Commit-Position: refs/heads/master@{#17346}
Name: WebRTC URL: http://www.webrtc.org Version: 90 License: BSD License File: LICENSE Description: WebRTC provides real time voice and video processing functionality to enable the implementation of PeerConnection/MediaStream. Third party code used in this project is described in the file LICENSE_THIRD_PARTY.