| commit | 6d4dd593a824cfc9c67f6e9fd3a08014d08350a0 | [log] [tgz] |
|---|---|---|
| author | nisse <nisse@webrtc.org> | Wed Feb 01 03:06:58 2017 -0800 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Feb 01 11:06:58 2017 +0000 |
| tree | e6e2dfa1aa9b55bf56133c69590930c5f6129b95 | |
| parent | 803dc29bb6773b60dadf5b630ef6a00b8e72f77f [diff] |
Always call RemoteBitrateEstimator::IncomingPacket from Call. Delete the calls from RtpStreamReceiver (for video) and AudioReceiveStream. BUG=webrtc:6847 Review-Url: https://codereview.webrtc.org/2659563002 Cr-Commit-Position: refs/heads/master@{#16393}
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.