| commit | b84ad63b0a189a33abaf78a527b04f881d881e91 | [log] [tgz] |
|---|---|---|
| author | sprang <sprang@webrtc.org> | Tue Nov 01 09:50:12 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Nov 01 09:50:17 2016 |
| tree | 505386054065fc735949209e4a067ab91ccc3392 | |
| parent | 6ded19086432b395a85c9a0206fa8a3bc6d33caf [diff] |
Add RTCP packet class for signaling encoder target bitrate. This is a proposal for a new RTCP message. Feel free to comment on the message structure, selected type ids etc, as well as code for serialization/deserialization. Once we agree on this, I'll continue with wiring it up in the actual rtcp sender and receiver. BUG=webrtc:6301 Review-Url: https://codereview.webrtc.org/2306873003 Cr-Commit-Position: refs/heads/master@{#14867}
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.