diff --git a/modules/audio_processing/agc/BUILD.gn b/modules/audio_processing/agc/BUILD.gn
index 24402e559a..508f901b08 100644
--- a/modules/audio_processing/agc/BUILD.gn
+++ b/modules/audio_processing/agc/BUILD.gn
@@ -36,6 +36,7 @@ rtc_library("agc") {
     "../../../system_wrappers:metrics",
     "../agc2:clipping_predictor",
     "../agc2:gain_map",
+    "../agc2:input_volume_stats_reporter",
     "../vad",
   ]
   absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
diff --git a/modules/audio_processing/agc/agc_manager_direct.cc b/modules/audio_processing/agc/agc_manager_direct.cc
index 67c9c6ddf9..b8ad4a8bb9 100644
--- a/modules/audio_processing/agc/agc_manager_direct.cc
+++ b/modules/audio_processing/agc/agc_manager_direct.cc
@@ -17,6 +17,7 @@
 #include "common_audio/include/audio_util.h"
 #include "modules/audio_processing/agc/gain_control.h"
 #include "modules/audio_processing/agc2/gain_map_internal.h"
+#include "modules/audio_processing/agc2/input_volume_stats_reporter.h"
 #include "modules/audio_processing/include/audio_frame_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
@@ -650,9 +651,8 @@ void AgcManagerDirect::Process(const AudioBuffer& audio_buffer,
   if (volume_after_clipping_handling != recommended_input_volume_) {
     // The recommended input volume was adjusted in order to match the target
     // level.
-    RTC_HISTOGRAM_COUNTS_LINEAR(
-        "WebRTC.Audio.Apm.RecommendedInputVolume.OnChangeToMatchTarget",
-        recommended_input_volume_, 1, kMaxMicLevel, 50);
+    UpdateHistogramOnRecommendedInputVolumeChangeToMatchTarget(
+        recommended_input_volume_);
   }
 }
 
diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn
index 220f4d0e9a..bd59ad3dae 100644
--- a/modules/audio_processing/agc2/BUILD.gn
+++ b/modules/audio_processing/agc2/BUILD.gn
@@ -208,6 +208,7 @@ rtc_library("input_volume_controller") {
   deps = [
     ":clipping_predictor",
     ":gain_map",
+    ":input_volume_stats_reporter",
     "..:api",
     "..:audio_buffer",
     "..:audio_frame_view",
diff --git a/modules/audio_processing/agc2/input_volume_controller.cc b/modules/audio_processing/agc2/input_volume_controller.cc
index 549fb3e48e..46962ae220 100644
--- a/modules/audio_processing/agc2/input_volume_controller.cc
+++ b/modules/audio_processing/agc2/input_volume_controller.cc
@@ -15,6 +15,7 @@
 
 #include "api/array_view.h"
 #include "modules/audio_processing/agc2/gain_map_internal.h"
+#include "modules/audio_processing/agc2/input_volume_stats_reporter.h"
 #include "modules/audio_processing/include/audio_frame_view.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
@@ -536,9 +537,8 @@ void InputVolumeController::Process(float speech_probability,
   if (volume_after_clipping_handling != recommended_input_volume_) {
     // The recommended input volume was adjusted in order to match the target
     // level.
-    RTC_HISTOGRAM_COUNTS_LINEAR(
-        "WebRTC.Audio.Apm.RecommendedInputVolume.OnChangeToMatchTarget",
-        recommended_input_volume_, 1, kMaxInputVolume, 50);
+    UpdateHistogramOnRecommendedInputVolumeChangeToMatchTarget(
+        recommended_input_volume_);
   }
 }
 
diff --git a/modules/audio_processing/agc2/input_volume_controller_unittest.cc b/modules/audio_processing/agc2/input_volume_controller_unittest.cc
index cb1c7de230..29349872a4 100644
--- a/modules/audio_processing/agc2/input_volume_controller_unittest.cc
+++ b/modules/audio_processing/agc2/input_volume_controller_unittest.cc
@@ -1420,6 +1420,8 @@ TEST(InputVolumeControllerTest, SpeechProbabilityThresholdIsEffective) {
 
 TEST(InputVolumeControllerTest,
      DoNotLogRecommendedInputVolumeOnChangeToMatchTarget) {
+  metrics::Reset();
+
   SpeechSamplesReader reader;
   auto controller = CreateInputVolumeController();
   controller->Initialize();
@@ -1439,6 +1441,8 @@ TEST(InputVolumeControllerTest,
 
 TEST(InputVolumeControllerTest,
      LogRecommendedInputVolumeOnUpwardChangeToMatchTarget) {
+  metrics::Reset();
+
   SpeechSamplesReader reader;
   auto controller = CreateInputVolumeController();
   controller->Initialize();
@@ -1457,6 +1461,8 @@ TEST(InputVolumeControllerTest,
 
 TEST(InputVolumeControllerTest,
      LogRecommendedInputVolumeOnDownwardChangeToMatchTarget) {
+  metrics::Reset();
+
   SpeechSamplesReader reader;
   auto controller = CreateInputVolumeController();
   controller->Initialize();
diff --git a/modules/audio_processing/agc2/input_volume_stats_reporter.cc b/modules/audio_processing/agc2/input_volume_stats_reporter.cc
index dfeb2185a9..05624b1f92 100644
--- a/modules/audio_processing/agc2/input_volume_stats_reporter.cc
+++ b/modules/audio_processing/agc2/input_volume_stats_reporter.cc
@@ -162,4 +162,10 @@ void InputVolumeStatsReporter::LogVolumeUpdateStats() const {
   }
 }
 
+void UpdateHistogramOnRecommendedInputVolumeChangeToMatchTarget(int volume) {
+  RTC_HISTOGRAM_COUNTS_LINEAR(
+      "WebRTC.Audio.Apm.RecommendedInputVolume.OnChangeToMatchTarget", volume,
+      1, kMaxInputVolume, 50);
+}
+
 }  // namespace webrtc
diff --git a/modules/audio_processing/agc2/input_volume_stats_reporter.h b/modules/audio_processing/agc2/input_volume_stats_reporter.h
index 0037a9a5ab..31b110031c 100644
--- a/modules/audio_processing/agc2/input_volume_stats_reporter.h
+++ b/modules/audio_processing/agc2/input_volume_stats_reporter.h
@@ -85,6 +85,12 @@ class InputVolumeStatsReporter {
   int log_volume_update_stats_counter_ = 0;
   absl::optional<int> previous_input_volume_ = absl::nullopt;
 };
+
+// Updates the histogram that keeps track of recommended input volume changes
+// required in order to match the target level in the input volume adaptation
+// process.
+void UpdateHistogramOnRecommendedInputVolumeChangeToMatchTarget(int volume);
+
 }  // namespace webrtc
 
 #endif  // MODULES_AUDIO_PROCESSING_AGC2_INPUT_VOLUME_STATS_REPORTER_H_