commit | 0a9fc05583c15438ca381498ee2db1da718c0c57 | [log] [tgz] |
---|---|---|
author | Peter Boström <pbos@webrtc.org> | Wed Mar 02 15:24:10 2016 |
committer | Peter Boström <pbos@webrtc.org> | Wed Mar 02 15:24:21 2016 |
tree | 447004d64491d3ccd3e847befa2e20012c2b77bd | |
parent | 7b19b08c18ac9b797d2ef4477c96270c5a87af17 [diff] |
Move RTP module send status outside of ViEChannel. Removes StartSend, StopSend and SetSendCodec from ViEChannel and into VideoSendStream which uses the payload router to configure them directly. BUG=webrtc:5494 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/1758603003 . Cr-Commit-Position: refs/heads/master@{#11845}
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.