| commit | dc7d0d2ef037dc02d7380925bf4e2133507539aa | [log] [tgz] |
|---|---|---|
| author | mflodman <mflodman@webrtc.org> | Fri May 06 12:32:22 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Fri May 06 12:32:30 2016 |
| tree | ce4f82285b7d05abb9f08baacc33a49f1cd69669 | |
| parent | b56069e650210ac7b11b5e5507ac517270f1a793 [diff] |
Move, almost, all receive side references to RTP to RtpStreamReceiver. There are still a few places in VideoReceiveStream where the RTP module is explicitly used, e.g. setting up a/v sync, but it's a bigger task to change and that will be done in a follow up instead of in this CL. BUG=webrtc:5838 Review-Url: https://codereview.webrtc.org/1947913002 Cr-Commit-Position: refs/heads/master@{#12642}
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.