| commit | eb45981165f982dd51425fad5ecb7ea9619063d3 | [log] [tgz] |
|---|---|---|
| author | deadbeef <deadbeef@webrtc.org> | Wed Dec 16 03:24:43 2015 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Dec 16 03:24:50 2015 |
| tree | 09ba18d4f72f585892a0fb7962cd6cc950303a2c | |
| parent | 44f0819978c2ba1f765835bca91e3243eb9f638b [diff] |
Restoring behavior where PeerConnection tracks changes to MediaStreams. If a MediaStream is added to a PeerConnection, and later a track is added to the MediaStream, a new RtpSender will now be created for that track, and it will appear in subsequent offers. Similarly, removed tracks will remove RtpSenders. BUG=webrtc:5265 Review URL: https://codereview.webrtc.org/1507973003 Cr-Commit-Position: refs/heads/master@{#11040}
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.