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At the top level, setting a track on an RtpSender is equivalent to setting a source (previously called a renderer) on a voice send stream. An RtpSender without a track is not supposed to send data (not even muted data), so a send stream without a source shouldn't send data. Also replacing SendFlags with a boolean and implementing "Start" and "Stop" methods on AudioSendStream, which was planned anyway and simplifies this CL. R=pthatcher@webrtc.org, solenberg@webrtc.org Review URL: https://codereview.webrtc.org/1741933002 . Cr-Commit-Position: refs/heads/master@{#11918}
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.