| commit | 9c6a0c7f6dc4446431f5138170cc95751ee5af8b | [log] [tgz] |
|---|---|---|
| author | danilchap <danilchap@webrtc.org> | Wed Feb 10 18:54:47 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Wed Feb 10 18:54:52 2016 |
| tree | 0623d578b88c376c3c5b1e40a3c0976fc3efcf33 | |
| parent | a75339c40d5e68ede8f0fb7d6a2b0f207e295043 [diff] |
Added A/V sync tests with drifting clocks. adding 30% drift to media generator (e.g. audio frame generated every 7ms instead of promised 10ms) works fine adding 2% drift to video ntp-timestamp-stamper makes A/V sync fail. BUG=webrtc:5504 R=pbos@webrtc.org,stefan@webrtc.org Review URL: https://codereview.webrtc.org/1674413004 Cr-Commit-Position: refs/heads/master@{#11556}
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.