| commit | 49715fe3be17d8579586d5bc954d626126d53415 | [log] [tgz] |
|---|---|---|
| author | henrik.lundin <henrik.lundin@webrtc.org> | Mon Dec 05 04:12:59 2016 -0800 |
| committer | Commit bot <commit-bot@chromium.org> | Mon Dec 05 12:13:05 2016 +0000 |
| tree | 7bc5c01fa69957f63c8dbd31265142f80e82cb3a | |
| parent | fa07b910a90336ff3a5986226352b6590d3eafa9 [diff] |
APM: Change 3 UMA metrics to fewer but linearly distributed buckets
In this change WebRTC.Audio.ApmCaptureInputLevel{Average,Peak} are
changed to 64 buckets, while WebRTC.Audio.AgcLevel is changed to 50
buckets. All three are changed to have linear spacing between buckets.
Also, the metrics are renamed to avoid stats conflicts because of different bucket schemes:
- WebRTC.Audio.AgcLevel -> WebRTC.Audio.AgcSetLevel
- WebRTC.Audio.ApmCaptureInputLevelAverage -> WebRTC.Audio.ApmCaptureInputLevelAverageRms
- WebRTC.Audio.ApmCaptureInputLevelPeakRms -> WebRTC.Audio.ApmCaptureInputLevelPeakRms
BUG=webrtc:6622
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng
Review-Url: https://codereview.webrtc.org/2547593002
Cr-Commit-Position: refs/heads/master@{#15418}
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.