Fix WebRTC.Audio.Apm.RecommendedInputVolume.OnChangeToMatchTarget tests

- Reset the tested metrics to avoid interactions between tests that
  depend on the execution order
- Address the comment in [1] by adding a function to log the same
  histogram in two different places

[1] https://chromium-review.googlesource.com/c/chromium/src/+/4087426/4/tools/metrics/histograms/metadata/web_rtc/histograms.xml#179

Bug: webrtc:7494
Change-Id: Ia4d339b03c8078eb63626c91579f8a9547f087f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287681
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38873}

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" ]

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_);
   }
 }
 

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",

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_);
   }
 }
 

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();

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

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_