AGC2: AdaptiveAgc ctor with sample rate and # of channels

The class has also been renamed to better reflect its purpose.

Bug: webrtc:7494
Change-Id: I223a364ab4f8b8a5fef765848bf05675d045cefd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236343
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35277}

modules/audio_processing/agc2/BUILD.gn

@@ -17,10 +17,10 @@ group("agc2") {
 
 rtc_library("adaptive_digital") {
   sources = [
-    "adaptive_agc.cc",
-    "adaptive_agc.h",
     "adaptive_digital_gain_applier.cc",
     "adaptive_digital_gain_applier.h",
+    "adaptive_digital_gain_controller.cc",
+    "adaptive_digital_gain_controller.h",
     "adaptive_mode_level_estimator.cc",
     "adaptive_mode_level_estimator.h",
     "saturation_protector.cc",

modules/audio_processing/agc2/adaptive_digital_gain_applier.cc

@@ -118,7 +118,9 @@ void CopyAudio(AudioFrameView<const float> src,
 
 AdaptiveDigitalGainApplier::AdaptiveDigitalGainApplier(
     ApmDataDumper* apm_data_dumper,
-    const AudioProcessing::Config::GainController2::AdaptiveDigital& config)
+    const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
+    int sample_rate_hz,
+    int num_channels)
     : apm_data_dumper_(apm_data_dumper),
       gain_applier_(
           /*hard_clip_samples=*/false,
@@ -134,7 +136,7 @@ AdaptiveDigitalGainApplier::AdaptiveDigitalGainApplier(
   RTC_DCHECK_GE(frames_to_gain_increase_allowed_, 1);
   RTC_DCHECK_GE(config_.max_output_noise_level_dbfs, -90.0f);
   RTC_DCHECK_LE(config_.max_output_noise_level_dbfs, 0.0f);
-  Initialize(/*sample_rate_hz=*/48000, /*num_channels=*/1);
+  Initialize(sample_rate_hz, num_channels);
 }
 
 void AdaptiveDigitalGainApplier::Initialize(int sample_rate_hz,

modules/audio_processing/agc2/adaptive_digital_gain_applier.h

@@ -38,7 +38,9 @@ class AdaptiveDigitalGainApplier {
 
   AdaptiveDigitalGainApplier(
       ApmDataDumper* apm_data_dumper,
-      const AudioProcessing::Config::GainController2::AdaptiveDigital& config);
+      const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
+      int sample_rate_hz,
+      int num_channels);
   AdaptiveDigitalGainApplier(const AdaptiveDigitalGainApplier&) = delete;
   AdaptiveDigitalGainApplier& operator=(const AdaptiveDigitalGainApplier&) =
       delete;

modules/audio_processing/agc2/adaptive_digital_gain_applier_unittest.cc

@@ -50,11 +50,15 @@ constexpr AdaptiveDigitalConfig kDefaultConfig{};
 
 // Helper to create initialized `AdaptiveDigitalGainApplier` objects.
 struct GainApplierHelper {
-  explicit GainApplierHelper(const AdaptiveDigitalConfig& config)
+  GainApplierHelper(const AdaptiveDigitalConfig& config,
+                    int sample_rate_hz,
+                    int num_channels)
       : apm_data_dumper(0),
         gain_applier(
             std::make_unique<AdaptiveDigitalGainApplier>(&apm_data_dumper,
-                                                         config)) {}
+                                                         config,
+                                                         sample_rate_hz,
+                                                         num_channels)) {}
   ApmDataDumper apm_data_dumper;
   std::unique_ptr<AdaptiveDigitalGainApplier> gain_applier;
 };
@@ -76,8 +80,7 @@ AdaptiveDigitalGainApplier::FrameInfo GetFrameInfoToNotAdapt(
 }
 
 TEST(GainController2AdaptiveGainApplier, GainApplierShouldNotCrash) {
-  GainApplierHelper helper(kDefaultConfig);
-  helper.gain_applier->Initialize(/*sample_rate_hz=*/48000, kStereo);
+  GainApplierHelper helper(kDefaultConfig, /*sample_rate_hz=*/48000, kStereo);
   // Make one call with reasonable audio level values and settings.
   VectorFloatFrame fake_audio(kStereo, kFrameLen10ms48kHz, 10000.0f);
   helper.gain_applier->Process(GetFrameInfoToNotAdapt(kDefaultConfig),
@@ -92,8 +95,7 @@ TEST(GainController2AdaptiveGainApplier, MaxGainApplied) {
                            kDefaultConfig.max_gain_change_db_per_second)) +
       kNumExtraFrames;
 
-  GainApplierHelper helper(kDefaultConfig);
-  helper.gain_applier->Initialize(/*sample_rate_hz=*/8000, kMono);
+  GainApplierHelper helper(kDefaultConfig, /*sample_rate_hz=*/8000, kMono);
   AdaptiveDigitalGainApplier::FrameInfo info =
       GetFrameInfoToNotAdapt(kDefaultConfig);
   info.speech_level_dbfs = -60.0f;
@@ -108,8 +110,7 @@ TEST(GainController2AdaptiveGainApplier, MaxGainApplied) {
 }
 
 TEST(GainController2AdaptiveGainApplier, GainDoesNotChangeFast) {
-  GainApplierHelper helper(kDefaultConfig);
-  helper.gain_applier->Initialize(/*sample_rate_hz=*/8000, kMono);
+  GainApplierHelper helper(kDefaultConfig, /*sample_rate_hz=*/8000, kMono);
 
   constexpr float initial_level_dbfs = -25.0f;
   constexpr float kMaxGainChangeDbPerFrame =
@@ -150,8 +151,7 @@ TEST(GainController2AdaptiveGainApplier, GainDoesNotChangeFast) {
 }
 
 TEST(GainController2AdaptiveGainApplier, GainIsRampedInAFrame) {
-  GainApplierHelper helper(kDefaultConfig);
-  helper.gain_applier->Initialize(/*sample_rate_hz=*/48000, kMono);
+  GainApplierHelper helper(kDefaultConfig, /*sample_rate_hz=*/48000, kMono);
 
   constexpr float initial_level_dbfs = -25.0f;
 
@@ -177,8 +177,7 @@ TEST(GainController2AdaptiveGainApplier, GainIsRampedInAFrame) {
 }
 
 TEST(GainController2AdaptiveGainApplier, NoiseLimitsGain) {
-  GainApplierHelper helper(kDefaultConfig);
-  helper.gain_applier->Initialize(/*sample_rate_hz=*/48000, kMono);
+  GainApplierHelper helper(kDefaultConfig, /*sample_rate_hz=*/48000, kMono);
 
   constexpr float initial_level_dbfs = -25.0f;
   constexpr int num_initial_frames =
@@ -209,8 +208,7 @@ TEST(GainController2AdaptiveGainApplier, NoiseLimitsGain) {
 }
 
 TEST(GainController2GainApplier, CanHandlePositiveSpeechLevels) {
-  GainApplierHelper helper(kDefaultConfig);
-  helper.gain_applier->Initialize(/*sample_rate_hz=*/48000, kStereo);
+  GainApplierHelper helper(kDefaultConfig, /*sample_rate_hz=*/48000, kStereo);
 
   // Make one call with positive audio level values and settings.
   VectorFloatFrame fake_audio(kStereo, kFrameLen10ms48kHz, 10000.0f);
@@ -221,8 +219,7 @@ TEST(GainController2GainApplier, CanHandlePositiveSpeechLevels) {
 }
 
 TEST(GainController2GainApplier, AudioLevelLimitsGain) {
-  GainApplierHelper helper(kDefaultConfig);
-  helper.gain_applier->Initialize(/*sample_rate_hz=*/48000, kMono);
+  GainApplierHelper helper(kDefaultConfig, /*sample_rate_hz=*/48000, kMono);
 
   constexpr float initial_level_dbfs = -25.0f;
   constexpr int num_initial_frames =
@@ -262,8 +259,7 @@ TEST_P(AdaptiveDigitalGainApplierTest,
        DoNotIncreaseGainWithTooFewSpeechFrames) {
   AdaptiveDigitalConfig config;
   config.adjacent_speech_frames_threshold = adjacent_speech_frames_threshold();
-  GainApplierHelper helper(config);
-  helper.gain_applier->Initialize(/*sample_rate_hz=*/48000, kMono);
+  GainApplierHelper helper(config, /*sample_rate_hz=*/48000, kMono);
 
   // Lower the speech level so that the target gain will be increased.
   AdaptiveDigitalGainApplier::FrameInfo info = GetFrameInfoToNotAdapt(config);
@@ -285,8 +281,7 @@ TEST_P(AdaptiveDigitalGainApplierTest,
 TEST_P(AdaptiveDigitalGainApplierTest, IncreaseGainWithEnoughSpeechFrames) {
   AdaptiveDigitalConfig config;
   config.adjacent_speech_frames_threshold = adjacent_speech_frames_threshold();
-  GainApplierHelper helper(config);
-  helper.gain_applier->Initialize(/*sample_rate_hz=*/48000, kMono);
+  GainApplierHelper helper(config, /*sample_rate_hz=*/48000, kMono);
 
   // Lower the speech level so that the target gain will be increased.
   AdaptiveDigitalGainApplier::FrameInfo info = GetFrameInfoToNotAdapt(config);
@@ -316,7 +311,7 @@ INSTANTIATE_TEST_SUITE_P(GainController2,
 TEST(GainController2GainApplier, DryRunDoesNotChangeInput) {
   AdaptiveDigitalConfig config;
   config.dry_run = true;
-  GainApplierHelper helper(config);
+  GainApplierHelper helper(config, /*sample_rate_hz=*/8000, kMono);
 
   // Simulate an input signal with log speech level.
   AdaptiveDigitalGainApplier::FrameInfo info = GetFrameInfoToNotAdapt(config);
@@ -328,7 +323,6 @@ TEST(GainController2GainApplier, DryRunDoesNotChangeInput) {
       kNumExtraFrames;
   constexpr float kPcmSamples = 123.456f;
   // Run the gain applier and check that the PCM samples are not modified.
-  helper.gain_applier->Initialize(/*sample_rate_hz=*/8000, kMono);
   for (int i = 0; i < num_frames_to_adapt; ++i) {
     SCOPED_TRACE(i);
     VectorFloatFrame fake_audio(kMono, kFrameLen10ms8kHz, kPcmSamples);
@@ -341,13 +335,12 @@ TEST(GainController2GainApplier, DryRunDoesNotChangeInput) {
 TEST(GainController2GainApplier, DryRunHandlesSampleRateChange) {
   AdaptiveDigitalConfig config;
   config.dry_run = true;
-  GainApplierHelper helper(config);
+  GainApplierHelper helper(config, /*sample_rate_hz=*/8000, kMono);
 
   AdaptiveDigitalGainApplier::FrameInfo info = GetFrameInfoToNotAdapt(config);
   info.speech_level_dbfs = -60.0f;
   constexpr float kPcmSamples = 123.456f;
   VectorFloatFrame fake_audio_8k(kMono, kFrameLen10ms8kHz, kPcmSamples);
-  helper.gain_applier->Initialize(/*sample_rate_hz=*/8000, kMono);
   helper.gain_applier->Process(info, fake_audio_8k.float_frame_view());
   EXPECT_FLOAT_EQ(fake_audio_8k.float_frame_view().channel(0)[0], kPcmSamples);
   helper.gain_applier->Initialize(/*sample_rate_hz=*/48000, kMono);
@@ -361,13 +354,12 @@ TEST(GainController2GainApplier, DryRunHandlesSampleRateChange) {
 TEST(GainController2GainApplier, DryRunHandlesNumChannelsChange) {
   AdaptiveDigitalConfig config;
   config.dry_run = true;
-  GainApplierHelper helper(config);
+  GainApplierHelper helper(config, /*sample_rate_hz=*/8000, kMono);
 
   AdaptiveDigitalGainApplier::FrameInfo info = GetFrameInfoToNotAdapt(config);
   info.speech_level_dbfs = -60.0f;
   constexpr float kPcmSamples = 123.456f;
   VectorFloatFrame fake_audio_8k(kMono, kFrameLen10ms8kHz, kPcmSamples);
-  helper.gain_applier->Initialize(/*sample_rate_hz=*/8000, kMono);
   helper.gain_applier->Process(info, fake_audio_8k.float_frame_view());
   EXPECT_FLOAT_EQ(fake_audio_8k.float_frame_view().channel(0)[0], kPcmSamples);
   VectorFloatFrame fake_audio_48k(kStereo, kFrameLen10ms8kHz, kPcmSamples);

modules/audio_processing/agc2/adaptive_digital_gain_controller.cc

@@ -8,7 +8,9 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#include "modules/audio_processing/agc2/adaptive_agc.h"
+#include "modules/audio_processing/agc2/adaptive_digital_gain_controller.h"
+
+#include <algorithm>
 
 #include "common_audio/include/audio_util.h"
 #include "modules/audio_processing/agc2/vad_wrapper.h"
@@ -39,11 +41,13 @@ AudioLevels ComputeAudioLevels(AudioFrameView<float> frame) {
 
 }  // namespace
 
-AdaptiveAgc::AdaptiveAgc(
+AdaptiveDigitalGainController::AdaptiveDigitalGainController(
     ApmDataDumper* apm_data_dumper,
-    const AudioProcessing::Config::GainController2::AdaptiveDigital& config)
+    const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
+    int sample_rate_hz,
+    int num_channels)
     : speech_level_estimator_(apm_data_dumper, config),
-      gain_controller_(apm_data_dumper, config),
+      gain_controller_(apm_data_dumper, config, sample_rate_hz, num_channels),
       apm_data_dumper_(apm_data_dumper),
       noise_level_estimator_(CreateNoiseFloorEstimator(apm_data_dumper)),
       saturation_protector_(
@@ -55,15 +59,16 @@ AdaptiveAgc::AdaptiveAgc(
   RTC_DCHECK(saturation_protector_);
 }
 
-AdaptiveAgc::~AdaptiveAgc() = default;
+AdaptiveDigitalGainController::~AdaptiveDigitalGainController() = default;
 
-void AdaptiveAgc::Initialize(int sample_rate_hz, int num_channels) {
+void AdaptiveDigitalGainController::Initialize(int sample_rate_hz,
+                                               int num_channels) {
   gain_controller_.Initialize(sample_rate_hz, num_channels);
 }
 
-void AdaptiveAgc::Process(AudioFrameView<float> frame,
-                          float speech_probability,
-                          float limiter_envelope) {
+void AdaptiveDigitalGainController::Process(AudioFrameView<float> frame,
+                                            float speech_probability,
+                                            float limiter_envelope) {
   AudioLevels levels = ComputeAudioLevels(frame);
   apm_data_dumper_->DumpRaw("agc2_input_rms_dbfs", levels.rms_dbfs);
   apm_data_dumper_->DumpRaw("agc2_input_peak_dbfs", levels.peak_dbfs);
@@ -95,7 +100,7 @@ void AdaptiveAgc::Process(AudioFrameView<float> frame,
   gain_controller_.Process(info, frame);
 }
 
-void AdaptiveAgc::HandleInputGainChange() {
+void AdaptiveDigitalGainController::HandleInputGainChange() {
   speech_level_estimator_.Reset();
   saturation_protector_->Reset();
 }

modules/audio_processing/agc2/adaptive_digital_gain_controller.h

@@ -8,8 +8,8 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
-#ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
-#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_
 
 #include <memory>
 
@@ -23,22 +23,26 @@
 namespace webrtc {
 class ApmDataDumper;
 
-// Adaptive digital gain controller.
-// TODO(crbug.com/webrtc/7494): Rename to `AdaptiveDigitalGainController`.
-class AdaptiveAgc {
+// Gain controller that adapts and applies a variable digital gain to meet the
+// target level, which is determined by the given configuration.
+class AdaptiveDigitalGainController {
  public:
-  AdaptiveAgc(
+  AdaptiveDigitalGainController(
       ApmDataDumper* apm_data_dumper,
-      const AudioProcessing::Config::GainController2::AdaptiveDigital& config);
-  ~AdaptiveAgc();
+      const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
+      int sample_rate_hz,
+      int num_channels);
+  AdaptiveDigitalGainController(const AdaptiveDigitalGainController&) = delete;
+  AdaptiveDigitalGainController& operator=(
+      const AdaptiveDigitalGainController&) = delete;
+  ~AdaptiveDigitalGainController();
 
+  // Detects and handles changes of sample rate and or number of channels.
   void Initialize(int sample_rate_hz, int num_channels);
 
-  // TODO(crbug.com/webrtc/7494): Add `SetLimiterEnvelope()`.
-
-  // Analyzes `frame` and applies a digital adaptive gain to it. Takes into
-  // account the speech probability and the envelope measured by the limiter.
-  // TODO(crbug.com/webrtc/7494): Remove `limiter_envelope`.
+  // Analyzes `frame`, adapts the current digital gain and applies it to
+  // `frame`.
+  // TODO(bugs.webrtc.org/7494): Remove `limiter_envelope`.
   void Process(AudioFrameView<float> frame,
                float speech_probability,
                float limiter_envelope);
@@ -56,4 +60,4 @@ class AdaptiveAgc {
 
 }  // namespace webrtc
 
-#endif  // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_

modules/audio_processing/gain_controller2.cc

@@ -51,18 +51,14 @@ AvailableCpuFeatures GetAllowedCpuFeatures() {
 }
 
 // Creates an adaptive digital gain controller if enabled.
-std::unique_ptr<AdaptiveAgc> CreateAdaptiveDigitalController(
+std::unique_ptr<AdaptiveDigitalGainController> CreateAdaptiveDigitalController(
     const Agc2Config::AdaptiveDigital& config,
     int sample_rate_hz,
     int num_channels,
     ApmDataDumper* data_dumper) {
   if (config.enabled) {
-    // TODO(bugs.webrtc.org/7494): Also init with sample rate and num
-    // channels.
-    auto controller = std::make_unique<AdaptiveAgc>(data_dumper, config);
-    // TODO(bugs.webrtc.org/7494): Remove once passed to the ctor.
-    controller->Initialize(sample_rate_hz, num_channels);
-    return controller;
+    return std::make_unique<AdaptiveDigitalGainController>(
+        data_dumper, config, sample_rate_hz, num_channels);
   }
   return nullptr;
 }

modules/audio_processing/gain_controller2.h

@@ -14,7 +14,7 @@
 #include <memory>
 #include <string>
 
-#include "modules/audio_processing/agc2/adaptive_agc.h"
+#include "modules/audio_processing/agc2/adaptive_digital_gain_controller.h"
 #include "modules/audio_processing/agc2/cpu_features.h"
 #include "modules/audio_processing/agc2/gain_applier.h"
 #include "modules/audio_processing/agc2/limiter.h"
@@ -57,7 +57,7 @@ class GainController2 {
   ApmDataDumper data_dumper_;
   GainApplier fixed_gain_applier_;
   std::unique_ptr<VoiceActivityDetectorWrapper> vad_;
-  std::unique_ptr<AdaptiveAgc> adaptive_digital_controller_;
+  std::unique_ptr<AdaptiveDigitalGainController> adaptive_digital_controller_;
   Limiter limiter_;
   int calls_since_last_limiter_log_;
   int analog_level_;