| commit | 433b95a68585313fb5d607639e6a6f6d3de70427 | [log] [tgz] |
|---|---|---|
| author | Taylor Brandstetter <deadbeef@webrtc.org> | Fri Mar 18 18:41:03 2016 |
| committer | Taylor Brandstetter <deadbeef@webrtc.org> | Fri Mar 18 18:41:15 2016 |
| tree | 16edaa62afdb17d9654df67425ad1d8ccca2dc8f | |
| parent | f5629ad44f33cc7de1920635b19756d218558265 [diff] |
Fixing issues with timestamps in video_quality_test.cc. The fundamental issue is that RTCP packet timestamps were accidentally being fed into wrap_handler_, causing it to think the 32-bit timestamp had wrapped around when it actually hadn't. Was also using a 32-bit timestamp instead of a 64-bit timestamp in one place, meaning that if wrapping actually DID occur, the test would still fail due to a 64-bit value being cast to a 32-bit value. BUG=webrtc:5668 R=pbos@webrtc.org, sprang@webrtc.org Review URL: https://codereview.webrtc.org/1814023003 . Cr-Commit-Position: refs/heads/master@{#12055}
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.