Gain metrics for digital adaptive AGC.

We add 2 metrics for measuring applied digital gain to
AgcManagerDirect. We also add an applied gain and an estimated noise
metric to Agc2.

Chromium histogram CL is
https://chromium-review.googlesource.com/c/chromium/src/+/1170833

Bug: webrtc:7494
Change-Id: Ie40873f9e43bc7d34d8f5473cd73bd47eb84e855
Reviewed-on: https://webrtc-review.googlesource.com/93468
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24290}

5 files changed, 23 insertions, 0 deletions

diff --git a/modules/audio_processing/agc/agc_manager_direct.cc b/modules/audio_processing/agc/agc_manager_direct.cc
index 5b5b9ea06b..dc6d45121c 100644
--- a/modules/audio_processing/agc/agc_manager_direct.cc
+++ b/modules/audio_processing/agc/agc_manager_direct.cc
@@ -454,6 +454,13 @@ void AgcManagerDirect::UpdateGain() {
 }
 
 void AgcManagerDirect::UpdateCompressor() {
+  calls_since_last_gain_log_++;
+  if (calls_since_last_gain_log_ == 100) {
+    calls_since_last_gain_log_ = 0;
+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc.DigitalGainApplied",
+                                compression_, 0, kMaxCompressionGain,
+                                kMaxCompressionGain + 1);
+  }
   if (compression_ == target_compression_) {
     return;
   }
@@ -478,6 +485,9 @@ void AgcManagerDirect::UpdateCompressor() {
 
   // Set the new compression gain.
   if (new_compression != compression_) {
+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc.DigitalGainUpdated",
+                                new_compression, 0, kMaxCompressionGain,
+                                kMaxCompressionGain + 1);
     compression_ = new_compression;
     compression_accumulator_ = new_compression;
     if (gctrl_->set_compression_gain_db(compression_) != 0) {
diff --git a/modules/audio_processing/agc/agc_manager_direct.h b/modules/audio_processing/agc/agc_manager_direct.h
index 7d2a33011f..cbfd6a1018 100644
--- a/modules/audio_processing/agc/agc_manager_direct.h
+++ b/modules/audio_processing/agc/agc_manager_direct.h
@@ -126,6 +126,7 @@ class AgcManagerDirect final {
   const bool disable_digital_adaptive_;
   int startup_min_level_;
   const int clipped_level_min_;
+  int calls_since_last_gain_log_ = 0;
 
   std::unique_ptr<DebugFile> file_preproc_;
   std::unique_ptr<DebugFile> file_postproc_;
diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn
index c56689b2a6..45ef968506 100644
--- a/modules/audio_processing/agc2/BUILD.gn
+++ b/modules/audio_processing/agc2/BUILD.gn
@@ -65,6 +65,7 @@ rtc_source_set("adaptive_digital") {
     "../../../rtc_base:checks",
     "../../../rtc_base:rtc_base_approved",
     "../../../rtc_base:safe_minmax",
+    "../../../system_wrappers:metrics_api",
   ]
 }
 
diff --git a/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc b/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc
index ca6ec5d4ec..f5342df829 100644
--- a/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc
+++ b/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc
@@ -16,6 +16,7 @@
 #include "modules/audio_processing/agc2/agc2_common.h"
 #include "modules/audio_processing/logging/apm_data_dumper.h"
 #include "rtc_base/numerics/safe_minmax.h"
+#include "system_wrappers/include/metrics.h"
 
 namespace webrtc {
 namespace {
@@ -76,6 +77,15 @@ void AdaptiveDigitalGainApplier::Process(
     float input_noise_level_dbfs,
     const VadWithLevel::LevelAndProbability vad_result,
     AudioFrameView<float> float_frame) {
+  calls_since_last_gain_log_++;
+  if (calls_since_last_gain_log_ == 100) {
+    calls_since_last_gain_log_ = 0;
+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.DigitalGainApplied",
+                                last_gain_db_, 0, kMaxGainDb, kMaxGainDb + 1);
+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.EstimatedNoiseLevel",
+                                input_noise_level_dbfs, 0, 100, 101);
+  }
+
   input_level_dbfs = std::min(input_level_dbfs, 0.f);
 
   RTC_DCHECK_GE(input_level_dbfs, -150.f);
diff --git a/modules/audio_processing/agc2/adaptive_digital_gain_applier.h b/modules/audio_processing/agc2/adaptive_digital_gain_applier.h
index 31f87f16e5..a3a1ff57f5 100644
--- a/modules/audio_processing/agc2/adaptive_digital_gain_applier.h
+++ b/modules/audio_processing/agc2/adaptive_digital_gain_applier.h
@@ -32,6 +32,7 @@ class AdaptiveDigitalGainApplier {
  private:
   float last_gain_db_ = kInitialAdaptiveDigitalGainDb;
   GainApplier gain_applier_;
+  int calls_since_last_gain_log_ = 0;
 
   // For some combinations of noise and speech probability, increasing
   // the level is not allowed. Since we may get VAD results in bursts,