| commit | 657bab24558648a9851e8b237df6ac245bd02f66 | [log] [tgz] |
|---|---|---|
| author | nisse <nisse@webrtc.org> | Tue Feb 21 14:28:10 2017 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Feb 21 14:28:10 2017 |
| tree | c610086cca283f13488ff3fa5b812ccc3750c745 | |
| parent | b94491d7904feb9b072b1adaa9454c9615fba5f8 [diff] |
Replace AudioReceiveStream::DeliverRtp with OnRtpPacket. This avoids redoing RTP header parsing already done in Call. BUG=webrtc:7135 Review-Url: https://codereview.webrtc.org/2697833002 Cr-Commit-Position: refs/heads/master@{#16750}
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.