| commit | 70ab1a1ca89d280a7d51e3fadc51d4be9df209ca | [log] [tgz] |
|---|---|---|
| author | deadbeef <deadbeef@webrtc.org> | Mon Sep 28 16:53:55 2015 -0700 |
| committer | Commit bot <commit-bot@chromium.org> | Mon Sep 28 23:54:02 2015 +0000 |
| tree | f3dcf359eba295e94225e507929d626cb9933100 | |
| parent | 8e9cb09506b4a076bc097324e8b79a72d3124615 [diff] |
Exposing RtpSenders and RtpReceivers from PeerConnection. This CL essentially converts [Local|Remote]TrackHandler to Rtp[Sender|Receiver], and adds a "SetTrack" method for RtpSender. It also gets rid of MediaStreamHandler and MediaStreamHandlerContainer, since these classes weren't really anything more than containers. PeerConnection now manages the RtpSenders and RtpReceivers directly. Review URL: https://codereview.webrtc.org/1351803002 Cr-Commit-Position: refs/heads/master@{#10100}
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.