commit | 0335e6c4bfa9a3254a392c003344c71238ee832f | [log] [tgz] |
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author | solenberg <solenberg@webrtc.org> | Wed Feb 22 07:07:04 2017 -0800 |
committer | Commit bot <commit-bot@chromium.org> | Wed Feb 22 15:07:04 2017 +0000 |
tree | 3566d0bc5e350a01cb0dfe1dde34be89b96087c1 | |
parent | 1d4e3d8a2ec0a138d1340e0d8643d69313a2194b [diff] |
Fix flaky test WebRtcMediaRecorderTest.PeerConnection A previous CL changed from logging to DCHECKing when setting minimum playout delay on a VoE channel: https://codereview.webrtc.org/2452163004/ I thought it safe at the time, since the input parameter range is capped, but apparently I didn't dig deep enough, as ultimately a failure may be returned for other reasons: https://chromium.googlesource.com/external/webrtc/+/master/webrtc/modules/audio_coding/neteq/delay_manager.cc#381 Thus, reverting to old behavior. BUG=694373 Review-Url: https://codereview.webrtc.org/2704933008 Cr-Commit-Position: refs/heads/master@{#16775}
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.