From 63407a9b6ae6f3fc096e01d64e46c6d21d86b517 Mon Sep 17 00:00:00 2001 From: "henrik.lundin" Date: Mon, 5 Dec 2016 05:11:27 -0800 Subject: [PATCH] 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 Review-Url: https://codereview.webrtc.org/2548333002 Cr-Commit-Position: refs/heads/master@{#15420} --- webrtc/modules/audio_processing/agc/agc_manager_direct.cc | 3 +-- webrtc/modules/audio_processing/audio_processing_impl.cc | 8 ++++---- 2 files changed, 5 insertions(+), 6 deletions(-) diff --git a/webrtc/modules/audio_processing/agc/agc_manager_direct.cc b/webrtc/modules/audio_processing/agc/agc_manager_direct.cc index 576bf2d56a..f8fc310c57 100644 --- a/webrtc/modules/audio_processing/agc/agc_manager_direct.cc +++ b/webrtc/modules/audio_processing/agc/agc_manager_direct.cc @@ -411,8 +411,7 @@ void AgcManagerDirect::UpdateGain() { SetLevel(LevelFromGainError(residual_gain, level_)); if (old_level != level_) { // level_ was updated by SetLevel; log the new value. - RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.AgcSetLevel", level_, 1, - kMaxMicLevel, 50); + RTC_HISTOGRAM_ENUMERATION("WebRTC.Audio.AgcLevel", level_, kMaxMicLevel); } } diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc index 4dffc549c3..2379cd1be8 100644 --- a/webrtc/modules/audio_processing/audio_processing_impl.cc +++ b/webrtc/modules/audio_processing/audio_processing_impl.cc @@ -1101,10 +1101,10 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() { if (++rms_interval_counter_ >= 1000) { rms_interval_counter_ = 0; RmsLevel::Levels levels = rms_.AverageAndPeak(); - RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureInputLevelAverageRms", - levels.average, 1, RmsLevel::kMinLevelDb, 64); - RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.ApmCaptureInputLevelPeakRms", - levels.peak, 1, RmsLevel::kMinLevelDb, 64); + RTC_HISTOGRAM_COUNTS("WebRTC.Audio.ApmCaptureInputLevelAverage", + levels.average, 1, RmsLevel::kMinLevelDb, 100); + RTC_HISTOGRAM_COUNTS("WebRTC.Audio.ApmCaptureInputLevelPeak", levels.peak, + 1, RmsLevel::kMinLevelDb, 100); } if (constants_.use_experimental_agc &&