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 2379cd1be8ec06fd8a2626a32f14c82d2a46631e..4dffc549c318d3d713f3d8c64a7eba2bb445b617 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc |
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc |
@@ -1101,10 +1101,10 @@ 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 && |