diff --git a/webrtc/modules/audio_processing/agc/agc_manager_direct.cc b/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
index 70a0e1f2b2..a4ac45af85 100644
--- a/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
+++ b/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
@@ -413,7 +413,8 @@ void AgcManagerDirect::UpdateGain() {
   SetLevel(LevelFromGainError(residual_gain, level_));
   if (old_level != level_) {
     // level_ was updated by SetLevel; log the new value.
-    RTC_HISTOGRAM_ENUMERATION("WebRTC.Audio.AgcLevel", level_, kMaxMicLevel);
+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.AgcSetLevel", level_, 1,
+                                kMaxMicLevel, 50);
   }
 }
 
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index a490b0a9e0..5ca9275f0e 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -1102,10 +1102,10 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
   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 &&