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

webrtc/modules/audio_processing/audio_processing_impl.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 1083 matching lines @@
   MaybeUpdateHistograms();
 
   AudioBuffer* capture_buffer = capture_.capture_audio.get();  // For brevity.
 
   rms_.Analyze(rtc::ArrayView<const int16_t>(
       capture_buffer->channels_const()[0],
       capture_nonlocked_.capture_processing_format.num_frames()));
   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 &&
       public_submodules_->gain_control->is_enabled()) {
     private_submodules_->agc_manager->AnalyzePreProcess(
         capture_buffer->channels()[0], capture_buffer->num_channels(),
         capture_nonlocked_.capture_processing_format.num_frames());
   }
 
   if (submodule_states_.CaptureMultiBandSubModulesActive() &&
@@ skipping 773 matching lines @@
       capture_processing_format(kSampleRate16kHz),
       split_rate(kSampleRate16kHz) {}
 
 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
 
 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
 
 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
 
 }  // namespace webrtc