| commit | 7411061982ed43190ca16432f498802e00924d37 | [log] [tgz] |
|---|---|---|
| author | danilchap <danilchap@webrtc.org> | Sun Oct 02 17:54:29 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Sun Oct 02 17:54:52 2016 |
| tree | 3e58511a9386acc41f22aa351392ea5424220381 | |
| parent | ac9f876bc06b8152ecc3a328147cdab19136361b [diff] |
Use RtpPacketToSend in RtpSenderVideo. This reduce reparsing rtp packet while sending. BUG=webrtc:5261 Review-Url: https://codereview.webrtc.org/2217383002 Cr-Commit-Position: refs/heads/master@{#14465}
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.