| commit | d7850b299bed6af1598800b051c9efda155d97b0 | [log] [tgz] |
|---|---|---|
| author | deadbeef <deadbeef@webrtc.org> | Wed Aug 23 17:59:19 2017 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Aug 23 17:59:19 2017 |
| tree | 2d18149a55d47d874288e0239a0ae9d7f986b1da | |
| parent | 60e10c794e8d5740e7295a2e0f0bc697eecdaa3b [diff] |
Use fake audio device in peerconnectioninterface_unittest.cc. This test doesn't actually send/receive any audio; it's only testing the interface layer. But the fact that it was creating/destroying real audio devices repeatedly caused problems when tests were run in parallel. So switching to a fake audio device solves this. BUG=webrtc:7806 Review-Url: https://codereview.webrtc.org/2997383002 Cr-Commit-Position: refs/heads/master@{#19472}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.