| /* | 
 |  *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. | 
 |  * | 
 |  *  Use of this source code is governed by a BSD-style license | 
 |  *  that can be found in the LICENSE file in the root of the source | 
 |  *  tree. An additional intellectual property rights grant can be found | 
 |  *  in the file PATENTS.  All contributing project authors may | 
 |  *  be found in the AUTHORS file in the root of the source tree. | 
 |  */ | 
 |  | 
 | #include "modules/audio_processing/gain_controller2.h" | 
 |  | 
 | #include <algorithm> | 
 | #include <cmath> | 
 | #include <memory> | 
 | #include <numeric> | 
 | #include <tuple> | 
 |  | 
 | #include "api/environment/environment.h" | 
 | #include "api/environment/environment_factory.h" | 
 | #include "modules/audio_processing/agc2/agc2_testing_common.h" | 
 | #include "modules/audio_processing/audio_buffer.h" | 
 | #include "modules/audio_processing/test/audio_buffer_tools.h" | 
 | #include "modules/audio_processing/test/bitexactness_tools.h" | 
 | #include "rtc_base/checks.h" | 
 | #include "test/gmock.h" | 
 | #include "test/gtest.h" | 
 |  | 
 | namespace webrtc { | 
 | namespace test { | 
 | namespace { | 
 |  | 
 | using ::testing::Eq; | 
 | using ::testing::Optional; | 
 |  | 
 | using Agc2Config = AudioProcessing::Config::GainController2; | 
 | using InputVolumeControllerConfig = InputVolumeController::Config; | 
 |  | 
 | // Sets all the samples in `ab` to `value`. | 
 | void SetAudioBufferSamples(float value, AudioBuffer& ab) { | 
 |   for (size_t k = 0; k < ab.num_channels(); ++k) { | 
 |     std::fill(ab.channels()[k], ab.channels()[k] + ab.num_frames(), value); | 
 |   } | 
 | } | 
 |  | 
 | float RunAgc2WithConstantInput(GainController2& agc2, | 
 |                                float input_level, | 
 |                                int num_frames, | 
 |                                int sample_rate_hz, | 
 |                                int num_channels = 1, | 
 |                                int applied_initial_volume = 0) { | 
 |   const int num_samples = CheckedDivExact(sample_rate_hz, 100); | 
 |   AudioBuffer ab(sample_rate_hz, num_channels, sample_rate_hz, num_channels, | 
 |                  sample_rate_hz, num_channels); | 
 |  | 
 |   // Give time to the level estimator to converge. | 
 |   for (int i = 0; i < num_frames + 1; ++i) { | 
 |     SetAudioBufferSamples(input_level, ab); | 
 |     const auto applied_volume = agc2.recommended_input_volume(); | 
 |     agc2.Analyze(applied_volume.value_or(applied_initial_volume), ab); | 
 |  | 
 |     agc2.Process(/*speech_probability=*/std::nullopt, | 
 |                  /*input_volume_changed=*/false, &ab); | 
 |   } | 
 |  | 
 |   // Return the last sample from the last processed frame. | 
 |   return ab.channels()[0][num_samples - 1]; | 
 | } | 
 |  | 
 | std::unique_ptr<GainController2> CreateAgc2FixedDigitalMode( | 
 |     float fixed_gain_db, | 
 |     int sample_rate_hz) { | 
 |   Agc2Config config; | 
 |   config.adaptive_digital.enabled = false; | 
 |   config.fixed_digital.gain_db = fixed_gain_db; | 
 |   EXPECT_TRUE(GainController2::Validate(config)); | 
 |   return std::make_unique<GainController2>(CreateEnvironment(), config, | 
 |                                            InputVolumeControllerConfig{}, | 
 |                                            sample_rate_hz, | 
 |                                            /*num_channels=*/1, | 
 |                                            /*use_internal_vad=*/true); | 
 | } | 
 |  | 
 | constexpr InputVolumeControllerConfig kTestInputVolumeControllerConfig{ | 
 |     .clipped_level_min = 20, | 
 |     .clipped_level_step = 30, | 
 |     .clipped_ratio_threshold = 0.4, | 
 |     .clipped_wait_frames = 50, | 
 |     .enable_clipping_predictor = true, | 
 |     .target_range_max_dbfs = -6, | 
 |     .target_range_min_dbfs = -70, | 
 |     .update_input_volume_wait_frames = 100, | 
 |     .speech_probability_threshold = 0.9, | 
 |     .speech_ratio_threshold = 1, | 
 | }; | 
 |  | 
 | }  // namespace | 
 |  | 
 | TEST(GainController2, CheckDefaultConfig) { | 
 |   Agc2Config config; | 
 |   EXPECT_TRUE(GainController2::Validate(config)); | 
 | } | 
 |  | 
 | TEST(GainController2, CheckFixedDigitalConfig) { | 
 |   Agc2Config config; | 
 |   // Attenuation is not allowed. | 
 |   config.fixed_digital.gain_db = -5.0f; | 
 |   EXPECT_FALSE(GainController2::Validate(config)); | 
 |   // No gain is allowed. | 
 |   config.fixed_digital.gain_db = 0.0f; | 
 |   EXPECT_TRUE(GainController2::Validate(config)); | 
 |   // Positive gain is allowed. | 
 |   config.fixed_digital.gain_db = 15.0f; | 
 |   EXPECT_TRUE(GainController2::Validate(config)); | 
 | } | 
 |  | 
 | TEST(GainController2, CheckHeadroomDb) { | 
 |   Agc2Config config; | 
 |   config.adaptive_digital.headroom_db = -1.0f; | 
 |   EXPECT_FALSE(GainController2::Validate(config)); | 
 |   config.adaptive_digital.headroom_db = 0.0f; | 
 |   EXPECT_TRUE(GainController2::Validate(config)); | 
 |   config.adaptive_digital.headroom_db = 5.0f; | 
 |   EXPECT_TRUE(GainController2::Validate(config)); | 
 | } | 
 |  | 
 | TEST(GainController2, CheckMaxGainDb) { | 
 |   Agc2Config config; | 
 |   config.adaptive_digital.max_gain_db = -1.0f; | 
 |   EXPECT_FALSE(GainController2::Validate(config)); | 
 |   config.adaptive_digital.max_gain_db = 0.0f; | 
 |   EXPECT_FALSE(GainController2::Validate(config)); | 
 |   config.adaptive_digital.max_gain_db = 5.0f; | 
 |   EXPECT_TRUE(GainController2::Validate(config)); | 
 | } | 
 |  | 
 | TEST(GainController2, CheckInitialGainDb) { | 
 |   Agc2Config config; | 
 |   config.adaptive_digital.initial_gain_db = -1.0f; | 
 |   EXPECT_FALSE(GainController2::Validate(config)); | 
 |   config.adaptive_digital.initial_gain_db = 0.0f; | 
 |   EXPECT_TRUE(GainController2::Validate(config)); | 
 |   config.adaptive_digital.initial_gain_db = 5.0f; | 
 |   EXPECT_TRUE(GainController2::Validate(config)); | 
 | } | 
 |  | 
 | TEST(GainController2, CheckAdaptiveDigitalMaxGainChangeSpeedConfig) { | 
 |   Agc2Config config; | 
 |   config.adaptive_digital.max_gain_change_db_per_second = -1.0f; | 
 |   EXPECT_FALSE(GainController2::Validate(config)); | 
 |   config.adaptive_digital.max_gain_change_db_per_second = 0.0f; | 
 |   EXPECT_FALSE(GainController2::Validate(config)); | 
 |   config.adaptive_digital.max_gain_change_db_per_second = 5.0f; | 
 |   EXPECT_TRUE(GainController2::Validate(config)); | 
 | } | 
 |  | 
 | TEST(GainController2, CheckAdaptiveDigitalMaxOutputNoiseLevelConfig) { | 
 |   Agc2Config config; | 
 |   config.adaptive_digital.max_output_noise_level_dbfs = 5.0f; | 
 |   EXPECT_FALSE(GainController2::Validate(config)); | 
 |   config.adaptive_digital.max_output_noise_level_dbfs = 0.0f; | 
 |   EXPECT_TRUE(GainController2::Validate(config)); | 
 |   config.adaptive_digital.max_output_noise_level_dbfs = -5.0f; | 
 |   EXPECT_TRUE(GainController2::Validate(config)); | 
 | } | 
 |  | 
 | TEST(GainController2, | 
 |      CheckGetRecommendedInputVolumeWhenInputVolumeControllerNotEnabled) { | 
 |   constexpr float kHighInputLevel = 32767.0f; | 
 |   constexpr float kLowInputLevel = 1000.0f; | 
 |   constexpr int kInitialInputVolume = 100; | 
 |   constexpr int kNumChannels = 2; | 
 |   constexpr int kNumFrames = 5; | 
 |   constexpr int kSampleRateHz = 16000; | 
 |  | 
 |   Agc2Config config; | 
 |   config.input_volume_controller.enabled = false; | 
 |  | 
 |   auto gain_controller = std::make_unique<GainController2>( | 
 |       CreateEnvironment(), config, InputVolumeControllerConfig{}, kSampleRateHz, | 
 |       kNumChannels, | 
 |       /*use_internal_vad=*/true); | 
 |  | 
 |   EXPECT_FALSE(gain_controller->recommended_input_volume().has_value()); | 
 |  | 
 |   // Run AGC for a signal with no clipping or detected speech. | 
 |   RunAgc2WithConstantInput(*gain_controller, kLowInputLevel, kNumFrames, | 
 |                            kSampleRateHz, kNumChannels, kInitialInputVolume); | 
 |  | 
 |   EXPECT_FALSE(gain_controller->recommended_input_volume().has_value()); | 
 |  | 
 |   // Run AGC for a signal with clipping. | 
 |   RunAgc2WithConstantInput(*gain_controller, kHighInputLevel, kNumFrames, | 
 |                            kSampleRateHz, kNumChannels, kInitialInputVolume); | 
 |  | 
 |   EXPECT_FALSE(gain_controller->recommended_input_volume().has_value()); | 
 | } | 
 |  | 
 | TEST( | 
 |     GainController2, | 
 |     CheckGetRecommendedInputVolumeWhenInputVolumeControllerNotEnabledAndSpecificConfigUsed) { | 
 |   constexpr float kHighInputLevel = 32767.0f; | 
 |   constexpr float kLowInputLevel = 1000.0f; | 
 |   constexpr int kInitialInputVolume = 100; | 
 |   constexpr int kNumChannels = 2; | 
 |   constexpr int kNumFrames = 5; | 
 |   constexpr int kSampleRateHz = 16000; | 
 |  | 
 |   Agc2Config config; | 
 |   config.input_volume_controller.enabled = false; | 
 |  | 
 |   auto gain_controller = std::make_unique<GainController2>( | 
 |       CreateEnvironment(), config, kTestInputVolumeControllerConfig, | 
 |       kSampleRateHz, kNumChannels, | 
 |       /*use_internal_vad=*/true); | 
 |  | 
 |   EXPECT_FALSE(gain_controller->recommended_input_volume().has_value()); | 
 |  | 
 |   // Run AGC for a signal with no clipping or detected speech. | 
 |   RunAgc2WithConstantInput(*gain_controller, kLowInputLevel, kNumFrames, | 
 |                            kSampleRateHz, kNumChannels, kInitialInputVolume); | 
 |  | 
 |   EXPECT_FALSE(gain_controller->recommended_input_volume().has_value()); | 
 |  | 
 |   // Run AGC for a signal with clipping. | 
 |   RunAgc2WithConstantInput(*gain_controller, kHighInputLevel, kNumFrames, | 
 |                            kSampleRateHz, kNumChannels, kInitialInputVolume); | 
 |  | 
 |   EXPECT_FALSE(gain_controller->recommended_input_volume().has_value()); | 
 | } | 
 |  | 
 | TEST(GainController2, | 
 |      CheckGetRecommendedInputVolumeWhenInputVolumeControllerEnabled) { | 
 |   constexpr float kHighInputLevel = 32767.0f; | 
 |   constexpr float kLowInputLevel = 1000.0f; | 
 |   constexpr int kInitialInputVolume = 100; | 
 |   constexpr int kNumChannels = 2; | 
 |   constexpr int kNumFrames = 5; | 
 |   constexpr int kSampleRateHz = 16000; | 
 |  | 
 |   Agc2Config config; | 
 |   config.input_volume_controller.enabled = true; | 
 |   config.adaptive_digital.enabled = true; | 
 |  | 
 |   auto gain_controller = std::make_unique<GainController2>( | 
 |       CreateEnvironment(), config, InputVolumeControllerConfig{}, kSampleRateHz, | 
 |       kNumChannels, | 
 |       /*use_internal_vad=*/true); | 
 |  | 
 |   EXPECT_FALSE(gain_controller->recommended_input_volume().has_value()); | 
 |  | 
 |   // Run AGC for a signal with no clipping or detected speech. | 
 |   RunAgc2WithConstantInput(*gain_controller, kLowInputLevel, kNumFrames, | 
 |                            kSampleRateHz, kNumChannels, kInitialInputVolume); | 
 |  | 
 |   EXPECT_TRUE(gain_controller->recommended_input_volume().has_value()); | 
 |  | 
 |   // Run AGC for a signal with clipping. | 
 |   RunAgc2WithConstantInput(*gain_controller, kHighInputLevel, kNumFrames, | 
 |                            kSampleRateHz, kNumChannels, kInitialInputVolume); | 
 |  | 
 |   EXPECT_TRUE(gain_controller->recommended_input_volume().has_value()); | 
 | } | 
 |  | 
 | TEST( | 
 |     GainController2, | 
 |     CheckGetRecommendedInputVolumeWhenInputVolumeControllerEnabledAndSpecificConfigUsed) { | 
 |   constexpr float kHighInputLevel = 32767.0f; | 
 |   constexpr float kLowInputLevel = 1000.0f; | 
 |   constexpr int kInitialInputVolume = 100; | 
 |   constexpr int kNumChannels = 2; | 
 |   constexpr int kNumFrames = 5; | 
 |   constexpr int kSampleRateHz = 16000; | 
 |  | 
 |   Agc2Config config; | 
 |   config.input_volume_controller.enabled = true; | 
 |   config.adaptive_digital.enabled = true; | 
 |  | 
 |   auto gain_controller = std::make_unique<GainController2>( | 
 |       CreateEnvironment(), config, kTestInputVolumeControllerConfig, | 
 |       kSampleRateHz, kNumChannels, | 
 |       /*use_internal_vad=*/true); | 
 |  | 
 |   EXPECT_FALSE(gain_controller->recommended_input_volume().has_value()); | 
 |  | 
 |   // Run AGC for a signal with no clipping or detected speech. | 
 |   RunAgc2WithConstantInput(*gain_controller, kLowInputLevel, kNumFrames, | 
 |                            kSampleRateHz, kNumChannels, kInitialInputVolume); | 
 |  | 
 |   EXPECT_TRUE(gain_controller->recommended_input_volume().has_value()); | 
 |  | 
 |   // Run AGC for a signal with clipping. | 
 |   RunAgc2WithConstantInput(*gain_controller, kHighInputLevel, kNumFrames, | 
 |                            kSampleRateHz, kNumChannels, kInitialInputVolume); | 
 |  | 
 |   EXPECT_TRUE(gain_controller->recommended_input_volume().has_value()); | 
 | } | 
 |  | 
 | // Checks that the default config is applied. | 
 | TEST(GainController2, ApplyDefaultConfig) { | 
 |   auto gain_controller2 = std::make_unique<GainController2>( | 
 |       CreateEnvironment(), Agc2Config{}, InputVolumeControllerConfig{}, | 
 |       /*sample_rate_hz=*/16000, /*num_channels=*/2, | 
 |       /*use_internal_vad=*/true); | 
 |   EXPECT_TRUE(gain_controller2.get()); | 
 | } | 
 |  | 
 | TEST(GainController2FixedDigital, GainShouldChangeOnSetGain) { | 
 |   constexpr float kInputLevel = 1000.0f; | 
 |   constexpr size_t kNumFrames = 5; | 
 |   constexpr size_t kSampleRateHz = 8000; | 
 |   constexpr float kGain0Db = 0.0f; | 
 |   constexpr float kGain20Db = 20.0f; | 
 |  | 
 |   auto agc2_fixed = CreateAgc2FixedDigitalMode(kGain0Db, kSampleRateHz); | 
 |  | 
 |   // Signal level is unchanged with 0 db gain. | 
 |   EXPECT_FLOAT_EQ(RunAgc2WithConstantInput(*agc2_fixed, kInputLevel, kNumFrames, | 
 |                                            kSampleRateHz), | 
 |                   kInputLevel); | 
 |  | 
 |   // +20 db should increase signal by a factor of 10. | 
 |   agc2_fixed->SetFixedGainDb(kGain20Db); | 
 |   EXPECT_FLOAT_EQ(RunAgc2WithConstantInput(*agc2_fixed, kInputLevel, kNumFrames, | 
 |                                            kSampleRateHz), | 
 |                   kInputLevel * 10); | 
 | } | 
 |  | 
 | TEST(GainController2FixedDigital, ChangeFixedGainShouldBeFastAndTimeInvariant) { | 
 |   // Number of frames required for the fixed gain controller to adapt on the | 
 |   // input signal when the gain changes. | 
 |   constexpr size_t kNumFrames = 5; | 
 |  | 
 |   constexpr float kInputLevel = 1000.0f; | 
 |   constexpr size_t kSampleRateHz = 8000; | 
 |   constexpr float kGainDbLow = 0.0f; | 
 |   constexpr float kGainDbHigh = 25.0f; | 
 |   static_assert(kGainDbLow < kGainDbHigh, ""); | 
 |  | 
 |   auto agc2_fixed = CreateAgc2FixedDigitalMode(kGainDbLow, kSampleRateHz); | 
 |  | 
 |   // Start with a lower gain. | 
 |   const float output_level_pre = RunAgc2WithConstantInput( | 
 |       *agc2_fixed, kInputLevel, kNumFrames, kSampleRateHz); | 
 |  | 
 |   // Increase gain. | 
 |   agc2_fixed->SetFixedGainDb(kGainDbHigh); | 
 |   static_cast<void>(RunAgc2WithConstantInput(*agc2_fixed, kInputLevel, | 
 |                                              kNumFrames, kSampleRateHz)); | 
 |  | 
 |   // Back to the lower gain. | 
 |   agc2_fixed->SetFixedGainDb(kGainDbLow); | 
 |   const float output_level_post = RunAgc2WithConstantInput( | 
 |       *agc2_fixed, kInputLevel, kNumFrames, kSampleRateHz); | 
 |  | 
 |   EXPECT_EQ(output_level_pre, output_level_post); | 
 | } | 
 |  | 
 | class FixedDigitalTest | 
 |     : public ::testing::TestWithParam<std::tuple<float, float, int, bool>> { | 
 |  protected: | 
 |   float gain_db_min() const { return std::get<0>(GetParam()); } | 
 |   float gain_db_max() const { return std::get<1>(GetParam()); } | 
 |   int sample_rate_hz() const { return std::get<2>(GetParam()); } | 
 |   bool saturation_expected() const { return std::get<3>(GetParam()); } | 
 | }; | 
 |  | 
 | TEST_P(FixedDigitalTest, CheckSaturationBehaviorWithLimiter) { | 
 |   for (const float gain_db : test::LinSpace(gain_db_min(), gain_db_max(), 10)) { | 
 |     SCOPED_TRACE(gain_db); | 
 |     auto agc2_fixed = CreateAgc2FixedDigitalMode(gain_db, sample_rate_hz()); | 
 |     const float processed_sample = | 
 |         RunAgc2WithConstantInput(*agc2_fixed, /*input_level=*/32767.0f, | 
 |                                  /*num_frames=*/5, sample_rate_hz()); | 
 |     if (saturation_expected()) { | 
 |       EXPECT_FLOAT_EQ(processed_sample, 32767.0f); | 
 |     } else { | 
 |       EXPECT_LT(processed_sample, 32767.0f); | 
 |     } | 
 |   } | 
 | } | 
 |  | 
 | static_assert(test::kLimiterMaxInputLevelDbFs < 10, ""); | 
 | INSTANTIATE_TEST_SUITE_P( | 
 |     GainController2, | 
 |     FixedDigitalTest, | 
 |     ::testing::Values( | 
 |         // When gain < `test::kLimiterMaxInputLevelDbFs`, the limiter will not | 
 |         // saturate the signal (at any sample rate). | 
 |         std::make_tuple(0.1f, | 
 |                         test::kLimiterMaxInputLevelDbFs - 0.01f, | 
 |                         8000, | 
 |                         false), | 
 |         std::make_tuple(0.1, | 
 |                         test::kLimiterMaxInputLevelDbFs - 0.01f, | 
 |                         48000, | 
 |                         false), | 
 |         // When gain > `test::kLimiterMaxInputLevelDbFs`, the limiter will | 
 |         // saturate the signal (at any sample rate). | 
 |         std::make_tuple(test::kLimiterMaxInputLevelDbFs + 0.01f, | 
 |                         10.0f, | 
 |                         8000, | 
 |                         true), | 
 |         std::make_tuple(test::kLimiterMaxInputLevelDbFs + 0.01f, | 
 |                         10.0f, | 
 |                         48000, | 
 |                         true))); | 
 |  | 
 | // Processes a test audio file and checks that the gain applied at the end of | 
 | // the recording is close to the expected value. | 
 | TEST(GainController2, CheckFinalGainWithAdaptiveDigitalController) { | 
 |   constexpr int kSampleRateHz = AudioProcessing::kSampleRate48kHz; | 
 |   constexpr int kStereo = 2; | 
 |  | 
 |   // Create AGC2 enabling only the adaptive digital controller. | 
 |   Agc2Config config; | 
 |   config.fixed_digital.gain_db = 0.0f; | 
 |   config.adaptive_digital.enabled = true; | 
 |   GainController2 agc2(CreateEnvironment(), config, | 
 |                        /*input_volume_controller_config=*/{}, kSampleRateHz, | 
 |                        kStereo, | 
 |                        /*use_internal_vad=*/true); | 
 |  | 
 |   test::InputAudioFile input_file( | 
 |       test::GetApmCaptureTestVectorFileName(kSampleRateHz), | 
 |       /*loop_at_end=*/true); | 
 |   const StreamConfig stream_config(kSampleRateHz, kStereo); | 
 |  | 
 |   // Init buffers. | 
 |   constexpr int kFrameDurationMs = 10; | 
 |   std::vector<float> frame(kStereo * stream_config.num_frames()); | 
 |   AudioBuffer audio_buffer(kSampleRateHz, kStereo, kSampleRateHz, kStereo, | 
 |                            kSampleRateHz, kStereo); | 
 |  | 
 |   // Simulate. | 
 |   constexpr float kGainDb = -6.0f; | 
 |   const float gain = std::pow(10.0f, kGainDb / 20.0f); | 
 |   constexpr int kDurationMs = 10000; | 
 |   constexpr int kNumFramesToProcess = kDurationMs / kFrameDurationMs; | 
 |   for (int i = 0; i < kNumFramesToProcess; ++i) { | 
 |     ReadFloatSamplesFromStereoFile(stream_config.num_frames(), | 
 |                                    stream_config.num_channels(), &input_file, | 
 |                                    frame); | 
 |     // Apply a fixed gain to the input audio. | 
 |     for (float& x : frame) { | 
 |       x *= gain; | 
 |     } | 
 |     test::CopyVectorToAudioBuffer(stream_config, frame, &audio_buffer); | 
 |     agc2.Process(/*speech_probability=*/std::nullopt, | 
 |                  /*input_volume_changed=*/false, &audio_buffer); | 
 |   } | 
 |  | 
 |   // Estimate the applied gain by processing a probing frame. | 
 |   SetAudioBufferSamples(/*value=*/1.0f, audio_buffer); | 
 |   agc2.Process(/*speech_probability=*/std::nullopt, | 
 |                /*input_volume_changed=*/false, &audio_buffer); | 
 |   const float applied_gain_db = | 
 |       20.0f * std::log10(audio_buffer.channels_const()[0][0]); | 
 |  | 
 |   constexpr float kExpectedGainDb = 7.0f; | 
 |   constexpr float kToleranceDb = 0.3f; | 
 |   EXPECT_NEAR(applied_gain_db, kExpectedGainDb, kToleranceDb); | 
 | } | 
 |  | 
 | #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) | 
 | // Checks that `GainController2` crashes in debug mode if it runs its internal | 
 | // VAD and the speech probability values are provided by the caller. | 
 | TEST(GainController2DeathTest, | 
 |      DebugCrashIfUseInternalVadAndSpeechProbabilityGiven) { | 
 |   constexpr int kSampleRateHz = AudioProcessing::kSampleRate48kHz; | 
 |   constexpr int kStereo = 2; | 
 |   AudioBuffer audio_buffer(kSampleRateHz, kStereo, kSampleRateHz, kStereo, | 
 |                            kSampleRateHz, kStereo); | 
 |   // Create AGC2 so that the interval VAD is also created. | 
 |   GainController2 agc2( | 
 |       CreateEnvironment(), /*config=*/{.adaptive_digital = {.enabled = true}}, | 
 |       /*input_volume_controller_config=*/{}, kSampleRateHz, kStereo, | 
 |       /*use_internal_vad=*/true); | 
 |  | 
 |   EXPECT_DEATH(agc2.Process(/*speech_probability=*/0.123f, | 
 |                             /*input_volume_changed=*/false, &audio_buffer), | 
 |                ""); | 
 | } | 
 | #endif | 
 |  | 
 | // Processes a test audio file and checks that the injected speech probability | 
 | // is not ignored when the internal VAD is not used. | 
 | TEST(GainController2, | 
 |      CheckInjectedVadProbabilityUsedWithAdaptiveDigitalController) { | 
 |   constexpr int kSampleRateHz = AudioProcessing::kSampleRate48kHz; | 
 |   constexpr int kStereo = 2; | 
 |  | 
 |   // Create AGC2 enabling only the adaptive digital controller. | 
 |   const Environment env = CreateEnvironment(); | 
 |   Agc2Config config; | 
 |   config.fixed_digital.gain_db = 0.0f; | 
 |   config.adaptive_digital.enabled = true; | 
 |   GainController2 agc2(env, config, /*input_volume_controller_config=*/{}, | 
 |                        kSampleRateHz, kStereo, | 
 |                        /*use_internal_vad=*/false); | 
 |   GainController2 agc2_reference(env, config, | 
 |                                  /*input_volume_controller_config=*/{}, | 
 |                                  kSampleRateHz, kStereo, | 
 |                                  /*use_internal_vad=*/true); | 
 |  | 
 |   test::InputAudioFile input_file( | 
 |       test::GetApmCaptureTestVectorFileName(kSampleRateHz), | 
 |       /*loop_at_end=*/true); | 
 |   const StreamConfig stream_config(kSampleRateHz, kStereo); | 
 |  | 
 |   // Init buffers. | 
 |   constexpr int kFrameDurationMs = 10; | 
 |   std::vector<float> frame(kStereo * stream_config.num_frames()); | 
 |   AudioBuffer audio_buffer(kSampleRateHz, kStereo, kSampleRateHz, kStereo, | 
 |                            kSampleRateHz, kStereo); | 
 |   AudioBuffer audio_buffer_reference(kSampleRateHz, kStereo, kSampleRateHz, | 
 |                                      kStereo, kSampleRateHz, kStereo); | 
 |   // Simulate. | 
 |   constexpr float kGainDb = -6.0f; | 
 |   const float gain = std::pow(10.0f, kGainDb / 20.0f); | 
 |   constexpr int kDurationMs = 10000; | 
 |   constexpr int kNumFramesToProcess = kDurationMs / kFrameDurationMs; | 
 |   constexpr float kSpeechProbabilities[] = {1.0f, 0.3f}; | 
 |   constexpr float kEpsilon = 0.0001f; | 
 |   bool all_samples_zero = true; | 
 |   bool all_samples_equal = true; | 
 |   for (int i = 0, j = 0; i < kNumFramesToProcess; ++i, j = 1 - j) { | 
 |     ReadFloatSamplesFromStereoFile(stream_config.num_frames(), | 
 |                                    stream_config.num_channels(), &input_file, | 
 |                                    frame); | 
 |     // Apply a fixed gain to the input audio. | 
 |     for (float& x : frame) { | 
 |       x *= gain; | 
 |     } | 
 |     test::CopyVectorToAudioBuffer(stream_config, frame, &audio_buffer); | 
 |     agc2.Process(kSpeechProbabilities[j], /*input_volume_changed=*/false, | 
 |                  &audio_buffer); | 
 |     test::CopyVectorToAudioBuffer(stream_config, frame, | 
 |                                   &audio_buffer_reference); | 
 |     agc2_reference.Process(/*speech_probability=*/std::nullopt, | 
 |                            /*input_volume_changed=*/false, | 
 |                            &audio_buffer_reference); | 
 |     // Check the output buffers. | 
 |     for (int channel = 0; channel < kStereo; ++channel) { | 
 |       for (int frame_num = 0; | 
 |            frame_num < static_cast<int>(audio_buffer.num_frames()); | 
 |            ++frame_num) { | 
 |         all_samples_zero &= | 
 |             fabs(audio_buffer.channels_const()[channel][frame_num]) < kEpsilon; | 
 |         all_samples_equal &= | 
 |             fabs(audio_buffer.channels_const()[channel][frame_num] - | 
 |                  audio_buffer_reference.channels_const()[channel][frame_num]) < | 
 |             kEpsilon; | 
 |       } | 
 |     } | 
 |   } | 
 |   EXPECT_FALSE(all_samples_zero); | 
 |   EXPECT_FALSE(all_samples_equal); | 
 | } | 
 |  | 
 | // Processes a test audio file and checks that the output is equal when | 
 | // an injected speech probability from `VoiceActivityDetectorWrapper` and | 
 | // the speech probability computed by the internal VAD are the same. | 
 | TEST(GainController2, | 
 |      CheckEqualResultFromInjectedVadProbabilityWithAdaptiveDigitalController) { | 
 |   constexpr int kSampleRateHz = AudioProcessing::kSampleRate48kHz; | 
 |   constexpr int kStereo = 2; | 
 |  | 
 |   // Create AGC2 enabling only the adaptive digital controller. | 
 |   const Environment env = CreateEnvironment(); | 
 |   Agc2Config config; | 
 |   config.fixed_digital.gain_db = 0.0f; | 
 |   config.adaptive_digital.enabled = true; | 
 |   GainController2 agc2(env, config, /*input_volume_controller_config=*/{}, | 
 |                        kSampleRateHz, kStereo, | 
 |                        /*use_internal_vad=*/false); | 
 |   GainController2 agc2_reference(env, config, | 
 |                                  /*input_volume_controller_config=*/{}, | 
 |                                  kSampleRateHz, kStereo, | 
 |                                  /*use_internal_vad=*/true); | 
 |   VoiceActivityDetectorWrapper vad(GetAvailableCpuFeatures(), kSampleRateHz); | 
 |   test::InputAudioFile input_file( | 
 |       test::GetApmCaptureTestVectorFileName(kSampleRateHz), | 
 |       /*loop_at_end=*/true); | 
 |   const StreamConfig stream_config(kSampleRateHz, kStereo); | 
 |  | 
 |   // Init buffers. | 
 |   constexpr int kFrameDurationMs = 10; | 
 |   std::vector<float> frame(kStereo * stream_config.num_frames()); | 
 |   AudioBuffer audio_buffer(kSampleRateHz, kStereo, kSampleRateHz, kStereo, | 
 |                            kSampleRateHz, kStereo); | 
 |   AudioBuffer audio_buffer_reference(kSampleRateHz, kStereo, kSampleRateHz, | 
 |                                      kStereo, kSampleRateHz, kStereo); | 
 |  | 
 |   // Simulate. | 
 |   constexpr float kGainDb = -6.0f; | 
 |   const float gain = std::pow(10.0f, kGainDb / 20.0f); | 
 |   constexpr int kDurationMs = 10000; | 
 |   constexpr int kNumFramesToProcess = kDurationMs / kFrameDurationMs; | 
 |   for (int i = 0; i < kNumFramesToProcess; ++i) { | 
 |     ReadFloatSamplesFromStereoFile(stream_config.num_frames(), | 
 |                                    stream_config.num_channels(), &input_file, | 
 |                                    frame); | 
 |     // Apply a fixed gain to the input audio. | 
 |     for (float& x : frame) { | 
 |       x *= gain; | 
 |     } | 
 |     test::CopyVectorToAudioBuffer(stream_config, frame, | 
 |                                   &audio_buffer_reference); | 
 |     agc2_reference.Process(std::nullopt, /*input_volume_changed=*/false, | 
 |                            &audio_buffer_reference); | 
 |     test::CopyVectorToAudioBuffer(stream_config, frame, &audio_buffer); | 
 |     float speech_probability = vad.Analyze(audio_buffer.view()); | 
 |     agc2.Process(speech_probability, /*input_volume_changed=*/false, | 
 |                  &audio_buffer); | 
 |     // Check the output buffer. | 
 |     for (int channel = 0; channel < kStereo; ++channel) { | 
 |       for (int frame_num = 0; | 
 |            frame_num < static_cast<int>(audio_buffer.num_frames()); | 
 |            ++frame_num) { | 
 |         EXPECT_FLOAT_EQ( | 
 |             audio_buffer.channels_const()[channel][frame_num], | 
 |             audio_buffer_reference.channels_const()[channel][frame_num]); | 
 |       } | 
 |     } | 
 |   } | 
 | } | 
 |  | 
 | }  // namespace test | 
 | }  // namespace webrtc |