Reason for revert: The downstream problem is now fixed, and this should be good to land again. Original issue's description: > Revert of APM: Change 3 UMA metrics to fewer but linearly distributed buckets (patchset #2 id:20001 of https://codereview.webrtc.org/2547593002/ ) > > Reason for revert: > Breaks down-stream dependencies. > > Original issue's description: > > 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 > > > > Committed: https://crrev.com/49715fe3be17d8579586d5bc954d626126d53415 > > Cr-Commit-Position: refs/heads/master@{#15418} > > TBR=peah@webrtc.org,rkaplow@chromium.org > # Skipping CQ checks because original CL landed less than 1 days ago. > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > BUG=webrtc:6622 > > Committed: https://crrev.com/63407a9b6ae6f3fc096e01d64e46c6d21d86b517 > Cr-Commit-Position: refs/heads/master@{#15420} TBR=peah@webrtc.org,rkaplow@chromium.org BUG=webrtc:6622 CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_rel_ng Review-Url: https://codereview.webrtc.org/2551863003 Cr-Commit-Position: refs/heads/master@{#15442}
Revert of CQ: Disable android_more_configs trybot (patchset #1 id:1 of https://codereview.webrtc.org/2522953003/ )
Reland of APM: Change 3 UMA metrics to fewer but linearly distributed buckets (patchset #1 id:1 of https://codereview.webrtc.org/2548333002/ )
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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.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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