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through streams related to a call object. The Call object does not know directly when packets pass through it, only which AudioSendStreams are used. Each AudioSendStream has a pointer to the Transport object through which its packets are send. This CL: By registering an internal wrapper class, TimedTransport, the AudioSendStream can stay up-to-date on when packets have passed through its Transport. This lifetime (as an interval) is then queried by the Call when the AudioSendStream is destroyed. When Call is destroyed, all streams are guaranteed to have been destroyed and hence Call is up-to-date on packet activity. The class TimeInterval keeps the code in Call and AudioSendStream smaller, with fewer get methods in their APIs and less code for updating values. Also modifies the unit test for AudioSendStream: it previously enforced that the stream registers (with its channel proxy) the same transport that it was constructed with. BUG=webrtc:7882 Review-Url: https://codereview.webrtc.org/2979833002 Cr-Commit-Position: refs/heads/master@{#19087}
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.