commit | 290d43aa1406790132db794f81103922442c24c4 | [log] [tgz] |
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author | henrik.lundin <henrik.lundin@webrtc.org> | Tue Nov 29 08:09:09 2016 -0800 |
committer | Commit bot <commit-bot@chromium.org> | Tue Nov 29 16:09:17 2016 +0000 |
tree | ad3c2a99f3c5a3003c4dca4dd27a1f6df5b29d00 | |
parent | 939e08f5f4b167451253d3acd5663dd9ead25875 [diff] |
Add a new UMA metric in APM to track incoming capture-side audio level This CL adds WebRTC.Audio.ApmCaptureInputLevelAverage and WebRTC.Audio.ApmCaptureInputLevelPeak. The metrics are updated once every 10 seconds. BUG=webrtc:6622 Review-Url: https://codereview.webrtc.org/2534473004 Cr-Commit-Position: refs/heads/master@{#15300}
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.