Revert of APM: Change 3 UMA metrics to fewer but linearly distributed buckets

webrtc/modules/audio_processing/audio_processing_impl.cc

Index: 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 4dffc549c318d3d713f3d8c64a7eba2bb445b617..2379cd1be8ec06fd8a2626a32f14c82d2a46631e 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 &&