commit | 97c392935411398b506861601c82e31d95c591f0 | [log] [tgz] |
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author | deadbeef <deadbeef@webrtc.org> | Tue Oct 13 20:23:41 2015 |
committer | Commit bot <commit-bot@chromium.org> | Tue Oct 13 20:23:48 2015 |
tree | f1a691c85e691612f58063de02da4c3fc9908c13 | |
parent | a0751c5c068ee76aaeeac56173ca043da1d568ff [diff] |
Moving MediaStreamSignaling logic into PeerConnection. This needs to happen because in the future, m-lines will be offered based on the set of RtpSenders/RtpReceivers, rather than the set of tracks that MediaStreamSignaling knows about. Besides that, MediaStreamSignaling was a "glue class" without a clearly defined role, so it going away is good for other reasons as well. Review URL: https://codereview.webrtc.org/1393563002 Cr-Commit-Position: refs/heads/master@{#10268}
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.