diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 061495d9a2..2379cd1be8 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -1095,6 +1095,18 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
 
   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);
+  }
+
   if (constants_.use_experimental_agc &&
       public_submodules_->gain_control->is_enabled()) {
     private_submodules_->agc_manager->AnalyzePreProcess(
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index d171715a96..4db2272c81 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -25,6 +25,7 @@
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/render_queue_item_verifier.h"
+#include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
@@ -406,6 +407,9 @@ class AudioProcessingImpl : public AudioProcessing {
   std::vector<float> red_render_queue_buffer_ GUARDED_BY(crit_render_);
   std::vector<float> red_capture_queue_buffer_ GUARDED_BY(crit_capture_);
 
+  RmsLevel rms_ GUARDED_BY(crit_capture_);
+  int rms_interval_counter_ GUARDED_BY(crit_capture_) = 0;
+
   // Lock protection not needed.
   std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>>
       aec_render_signal_queue_;
diff --git a/webrtc/modules/audio_processing/rms_level.cc b/webrtc/modules/audio_processing/rms_level.cc
index f77357a000..8a77b3f7d9 100644
--- a/webrtc/modules/audio_processing/rms_level.cc
+++ b/webrtc/modules/audio_processing/rms_level.cc
@@ -19,7 +19,6 @@
 namespace webrtc {
 namespace {
 static constexpr float kMaxSquaredLevel = 32768 * 32768;
-static constexpr int kMinLevelDb = 127;
 // kMinLevel is the level corresponding to kMinLevelDb, that is 10^(-127/10).
 static constexpr float kMinLevel = 1.995262314968883e-13f;
 
@@ -31,7 +30,7 @@ static constexpr float kMinLevel = 1.995262314968883e-13f;
 int ComputeRms(float mean_square) {
   if (mean_square <= kMinLevel * kMaxSquaredLevel) {
     // Very faint; simply return the minimum value.
-    return kMinLevelDb;
+    return RmsLevel::kMinLevelDb;
   }
   // Normalize by the max level.
   const float mean_square_norm = mean_square / kMaxSquaredLevel;
@@ -39,7 +38,7 @@ int ComputeRms(float mean_square) {
   // 20log_10(x^0.5) = 10log_10(x)
   const float rms = 10.f * log10(mean_square_norm);
   RTC_DCHECK_LE(rms, 0.f);
-  RTC_DCHECK_GT(rms, -kMinLevelDb);
+  RTC_DCHECK_GT(rms, -RmsLevel::kMinLevelDb);
   // Return the negated value.
   return static_cast<int>(-rms + 0.5f);
 }
@@ -81,7 +80,7 @@ void RmsLevel::AnalyzeMuted(size_t length) {
 }
 
 int RmsLevel::Average() {
-  int rms = (sample_count_ == 0) ? kMinLevelDb
+  int rms = (sample_count_ == 0) ? RmsLevel::kMinLevelDb
                                  : ComputeRms(sum_square_ / sample_count_);
   Reset();
   return rms;
@@ -92,7 +91,7 @@ RmsLevel::Levels RmsLevel::AverageAndPeak() {
   // sample_count_ != 0. Also, the * operator of rtc::Optional enforces this
   // with a DCHECK.
   Levels levels = (sample_count_ == 0)
-                      ? Levels{kMinLevelDb, kMinLevelDb}
+                      ? Levels{RmsLevel::kMinLevelDb, RmsLevel::kMinLevelDb}
                       : Levels{ComputeRms(sum_square_ / sample_count_),
                                ComputeRms(max_sum_square_ / *block_size_)};
   Reset();
diff --git a/webrtc/modules/audio_processing/rms_level.h b/webrtc/modules/audio_processing/rms_level.h
index 8c3aa78eb7..a186bcf608 100644
--- a/webrtc/modules/audio_processing/rms_level.h
+++ b/webrtc/modules/audio_processing/rms_level.h
@@ -32,6 +32,8 @@ class RmsLevel {
     int peak;
   };
 
+  static constexpr int kMinLevelDb = 127;
+
   RmsLevel();
   ~RmsLevel();