commit | 1a018dcda39691c7cb91ef524003482944bc8960 | [log] [tgz] |
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author | Taylor Brandstetter <deadbeef@webrtc.org> | Tue Mar 08 12:37:39 2016 -0800 |
committer | Taylor Brandstetter <deadbeef@webrtc.org> | Tue Mar 08 20:37:48 2016 +0000 |
tree | 126c9692b73fbbf8606534198c0e906eb7f2d340 | |
parent | 1ae6a459868917c8c2bb3cc8080192abc4f4bbb2 [diff] |
Prevent a voice channel from sending data before a source is set. 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}
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. This page is maintained by the Google Chrome team.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.