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}
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 4dffc54..2379cd1 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -1101,10 +1101,10 @@
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 &&