commit 49715fe3be17d8579586d5bc954d626126d53415
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

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/modules/audio_processing/audio_processing_impl.cc

diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 2379cd1..4dffc54 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("WebRTC.Audio.ApmCaptureInputLevelAverage",
-                         levels.average, 1, RmsLevel::kMinLevelDb, 100);
-    RTC_HISTOGRAM_COUNTS("WebRTC.Audio.ApmCaptureInputLevelPeak", levels.peak,
-                         1, RmsLevel::kMinLevelDb, 100);
+    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);
   }
 
   if (constants_.use_experimental_agc &&