| commit | 37a2111d7ce430a32c35183442747868ca9a28c9 | [log] [tgz] |
|---|---|---|
| author | ehmaldonado <ehmaldonado@webrtc.org> | Thu Nov 24 03:13:16 2016 -0800 |
| committer | Commit bot <commit-bot@chromium.org> | Thu Nov 24 11:13:24 2016 +0000 |
| tree | 7c65c37ed098ea2f6ef699de088d77b7058746a1 | |
| parent | 3edc7f05f5a6d94b8ea2d29faa5ed65ae0a226d8 [diff] |
Increase the threshold for RunPlayoutAndRecordingInFullDuplex. Again. RunPlayoutAndRecordingInFullDuplex fails sometimes on Android swarming bots, presumably because the timing is hardware dependent. This test ensures that audio starts pumping. The exact performance is not that important. R=kjellander@webrtc.org, henrika@webrtc.org BUG=webrtc:6464 NOTRY=True Review-Url: https://codereview.webrtc.org/2525943003 Cr-Commit-Position: refs/heads/master@{#15223}
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.