| commit | 4cd2790f170028b2864b7fa66d5e7400631b4273 | [log] [tgz] |
|---|---|---|
| author | mflodman <mflodman@webrtc.org> | Fri Aug 05 13:28:45 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Fri Aug 05 13:28:50 2016 |
| tree | 14d7fcf0705226de29322ddc0644c6adbc3c7b4d | |
| parent | b7b9dca4e43fba58bc1ec0fedbd0f92882af4c3a [diff] |
Move RTP for synchroninzation and rename classes, files and variables. This CL removes (almost) the last RTP references in VideoReceiveStream. There are still references to RTPFragmentationHeader and SSRCs, which will be dealt with later. There are also new GUARDED_BY and thred checker added to the synchronization class. When there are othre transports than RTP, there will instead be an interface + inheritance for RtpStreamReceiver and RtpStreamSynchronizattion in VideoReceiveStream. This work will be done when we actually know how we want to make thee transport interface. BUG=webrtc:5838 Review-Url: https://codereview.webrtc.org/2216533002 Cr-Commit-Position: refs/heads/master@{#13655}
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.