| commit | 65a1e77202ce4c03a8416965a5d8446949f12544 | [log] [tgz] |
|---|---|---|
| author | brandtr <brandtr@webrtc.org> | Mon Dec 12 09:54:58 2016 |
| committer | Commit bot <commit-bot@chromium.org> | Mon Dec 12 09:55:09 2016 |
| tree | b1dc156a76554bed1baf809109b039afc3dea8e3 | |
| parent | e448dd53554b4ecd3dd62a46a08b25044bc2e210 [diff] |
Try to deflake VideoSendStream tests with ULPFEC. The changes here are the same as in https://codereview.webrtc.org/2523993002/: - reduce packet loss from 50% to 5%, to allow the BWE to ramp up better. - Do not wait for 100 packets to be sent before letting the test pass. BUG=webrtc:6851 Review-Url: https://codereview.webrtc.org/2558303002 Cr-Commit-Position: refs/heads/master@{#15542}
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.