/* * 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 #include #include #include "api/array_view.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/gtest.h" namespace webrtc { namespace test { namespace { void SetAudioBufferSamples(float value, AudioBuffer* ab) { // Sets all the samples in |ab| to |value|. 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, size_t num_frames, int sample_rate) { const int num_samples = rtc::CheckedDivExact(sample_rate, 100); AudioBuffer ab(sample_rate, 1, sample_rate, 1, sample_rate, 1); // Give time to the level estimator to converge. for (size_t i = 0; i < num_frames + 1; ++i) { SetAudioBufferSamples(input_level, &ab); agc2->Process(&ab); } // Return the last sample from the last processed frame. return ab.channels()[0][num_samples - 1]; } AudioProcessing::Config::GainController2 CreateAgc2FixedDigitalModeConfig( float fixed_gain_db) { AudioProcessing::Config::GainController2 config; config.adaptive_digital.enabled = false; config.fixed_digital.gain_db = fixed_gain_db; // TODO(alessiob): Check why ASSERT_TRUE() below does not compile. EXPECT_TRUE(GainController2::Validate(config)); return config; } std::unique_ptr CreateAgc2FixedDigitalMode( float fixed_gain_db, size_t sample_rate_hz) { auto agc2 = std::make_unique(); agc2->ApplyConfig(CreateAgc2FixedDigitalModeConfig(fixed_gain_db)); agc2->Initialize(sample_rate_hz); return agc2; } float GainDbAfterProcessingFile(GainController2& gain_controller, int max_duration_ms) { // Set up an AudioBuffer to be filled from the speech file. constexpr size_t kStereo = 2u; const StreamConfig capture_config(AudioProcessing::kSampleRate48kHz, kStereo, false); AudioBuffer ab(capture_config.sample_rate_hz(), capture_config.num_channels(), capture_config.sample_rate_hz(), capture_config.num_channels(), capture_config.sample_rate_hz(), capture_config.num_channels()); test::InputAudioFile capture_file( test::GetApmCaptureTestVectorFileName(AudioProcessing::kSampleRate48kHz)); std::vector capture_input(capture_config.num_frames() * capture_config.num_channels()); // Process the input file which must be long enough to cover // `max_duration_ms`. RTC_DCHECK_GT(max_duration_ms, 0); const int num_frames = rtc::CheckedDivExact(max_duration_ms, 10); for (int i = 0; i < num_frames; ++i) { ReadFloatSamplesFromStereoFile(capture_config.num_frames(), capture_config.num_channels(), &capture_file, capture_input); test::CopyVectorToAudioBuffer(capture_config, capture_input, &ab); gain_controller.Process(&ab); } // Send in a last frame with minimum dBFS level. constexpr float sample_value = 1.f; SetAudioBufferSamples(sample_value, &ab); gain_controller.Process(&ab); // Measure the RMS level after processing. float rms = 0.0f; for (size_t i = 0; i < capture_config.num_frames(); ++i) { rms += ab.channels()[0][i] * ab.channels()[0][i]; } // Return the applied gain in dB. return 20.0f * std::log10(std::sqrt(rms / capture_config.num_frames())); } } // namespace TEST(GainController2, CheckDefaultConfig) { AudioProcessing::Config::GainController2 config; EXPECT_TRUE(GainController2::Validate(config)); } TEST(GainController2, CheckFixedDigitalConfig) { AudioProcessing::Config::GainController2 config; // Attenuation is not allowed. config.fixed_digital.gain_db = -5.f; EXPECT_FALSE(GainController2::Validate(config)); // No gain is allowed. config.fixed_digital.gain_db = 0.f; EXPECT_TRUE(GainController2::Validate(config)); // Positive gain is allowed. config.fixed_digital.gain_db = 15.f; EXPECT_TRUE(GainController2::Validate(config)); } TEST(GainController2, CheckAdaptiveDigitalVadProbabilityAttackConfig) { AudioProcessing::Config::GainController2 config; // Reject invalid attack. config.adaptive_digital.vad_probability_attack = -123.f; EXPECT_FALSE(GainController2::Validate(config)); config.adaptive_digital.vad_probability_attack = 0.f; EXPECT_FALSE(GainController2::Validate(config)); config.adaptive_digital.vad_probability_attack = 42.f; EXPECT_FALSE(GainController2::Validate(config)); // Accept valid attack. config.adaptive_digital.vad_probability_attack = 0.1f; EXPECT_TRUE(GainController2::Validate(config)); config.adaptive_digital.vad_probability_attack = 1.f; EXPECT_TRUE(GainController2::Validate(config)); } TEST(GainController2, CheckAdaptiveDigitalLevelEstimatorSpeechFramesThresholdConfig) { AudioProcessing::Config::GainController2 config; config.adaptive_digital.level_estimator_adjacent_speech_frames_threshold = 0; EXPECT_FALSE(GainController2::Validate(config)); config.adaptive_digital.level_estimator_adjacent_speech_frames_threshold = 1; EXPECT_TRUE(GainController2::Validate(config)); config.adaptive_digital.level_estimator_adjacent_speech_frames_threshold = 7; EXPECT_TRUE(GainController2::Validate(config)); } TEST(GainController2, CheckAdaptiveDigitalInitialSaturationMarginConfig) { AudioProcessing::Config::GainController2 config; config.adaptive_digital.initial_saturation_margin_db = -1.f; EXPECT_FALSE(GainController2::Validate(config)); config.adaptive_digital.initial_saturation_margin_db = 0.f; EXPECT_TRUE(GainController2::Validate(config)); config.adaptive_digital.initial_saturation_margin_db = 50.f; EXPECT_TRUE(GainController2::Validate(config)); } TEST(GainController2, CheckAdaptiveDigitalExtraSaturationMarginConfig) { AudioProcessing::Config::GainController2 config; config.adaptive_digital.extra_saturation_margin_db = -1.f; EXPECT_FALSE(GainController2::Validate(config)); config.adaptive_digital.extra_saturation_margin_db = 0.f; EXPECT_TRUE(GainController2::Validate(config)); config.adaptive_digital.extra_saturation_margin_db = 50.f; EXPECT_TRUE(GainController2::Validate(config)); } TEST(GainController2, CheckAdaptiveDigitalGainApplierSpeechFramesThresholdConfig) { AudioProcessing::Config::GainController2 config; config.adaptive_digital.gain_applier_adjacent_speech_frames_threshold = 0; EXPECT_FALSE(GainController2::Validate(config)); config.adaptive_digital.gain_applier_adjacent_speech_frames_threshold = 1; EXPECT_TRUE(GainController2::Validate(config)); config.adaptive_digital.gain_applier_adjacent_speech_frames_threshold = 7; EXPECT_TRUE(GainController2::Validate(config)); } TEST(GainController2, CheckAdaptiveDigitalMaxGainChangeSpeedConfig) { AudioProcessing::Config::GainController2 config; config.adaptive_digital.max_gain_change_db_per_second = -1.f; EXPECT_FALSE(GainController2::Validate(config)); config.adaptive_digital.max_gain_change_db_per_second = 0.f; EXPECT_FALSE(GainController2::Validate(config)); config.adaptive_digital.max_gain_change_db_per_second = 5.f; EXPECT_TRUE(GainController2::Validate(config)); } TEST(GainController2, CheckAdaptiveDigitalMaxOutputNoiseLevelConfig) { AudioProcessing::Config::GainController2 config; config.adaptive_digital.max_output_noise_level_dbfs = 5.f; EXPECT_FALSE(GainController2::Validate(config)); config.adaptive_digital.max_output_noise_level_dbfs = 0.f; EXPECT_TRUE(GainController2::Validate(config)); config.adaptive_digital.max_output_noise_level_dbfs = -5.f; EXPECT_TRUE(GainController2::Validate(config)); } // Checks that the default config is applied. TEST(GainController2, ApplyDefaultConfig) { auto gain_controller2 = std::make_unique(); AudioProcessing::Config::GainController2 config; gain_controller2->ApplyConfig(config); } TEST(GainController2FixedDigital, GainShouldChangeOnSetGain) { constexpr float kInputLevel = 1000.f; constexpr size_t kNumFrames = 5; constexpr size_t kSampleRateHz = 8000; constexpr float kGain0Db = 0.f; constexpr float kGain20Db = 20.f; auto agc2_fixed = CreateAgc2FixedDigitalMode(kGain0Db, kSampleRateHz); // Signal level is unchanged with 0 db gain. EXPECT_FLOAT_EQ(RunAgc2WithConstantInput(agc2_fixed.get(), kInputLevel, kNumFrames, kSampleRateHz), kInputLevel); // +20 db should increase signal by a factor of 10. agc2_fixed->ApplyConfig(CreateAgc2FixedDigitalModeConfig(kGain20Db)); EXPECT_FLOAT_EQ(RunAgc2WithConstantInput(agc2_fixed.get(), 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.f; constexpr size_t kSampleRateHz = 8000; constexpr float kGainDbLow = 0.f; constexpr float kGainDbHigh = 25.f; static_assert(kGainDbLow < kGainDbHigh, ""); auto agc2_fixed = CreateAgc2FixedDigitalMode(kGainDbLow, kSampleRateHz); // Start with a lower gain. const float output_level_pre = RunAgc2WithConstantInput( agc2_fixed.get(), kInputLevel, kNumFrames, kSampleRateHz); // Increase gain. agc2_fixed->ApplyConfig(CreateAgc2FixedDigitalModeConfig(kGainDbHigh)); static_cast(RunAgc2WithConstantInput(agc2_fixed.get(), kInputLevel, kNumFrames, kSampleRateHz)); // Back to the lower gain. agc2_fixed->ApplyConfig(CreateAgc2FixedDigitalModeConfig(kGainDbLow)); const float output_level_post = RunAgc2WithConstantInput( agc2_fixed.get(), kInputLevel, kNumFrames, kSampleRateHz); EXPECT_EQ(output_level_pre, output_level_post); } struct FixedDigitalTestParams { FixedDigitalTestParams(float gain_db_min, float gain_db_max, size_t sample_rate, bool saturation_expected) : gain_db_min(gain_db_min), gain_db_max(gain_db_max), sample_rate(sample_rate), saturation_expected(saturation_expected) {} float gain_db_min; float gain_db_max; size_t sample_rate; bool saturation_expected; }; class FixedDigitalTest : public ::testing::Test, public ::testing::WithParamInterface {}; TEST_P(FixedDigitalTest, CheckSaturationBehaviorWithLimiter) { const float kInputLevel = 32767.f; const size_t kNumFrames = 5; const auto params = GetParam(); const auto gains_db = test::LinSpace(params.gain_db_min, params.gain_db_max, 10); for (const auto gain_db : gains_db) { SCOPED_TRACE(std::to_string(gain_db)); auto agc2_fixed = CreateAgc2FixedDigitalMode(gain_db, params.sample_rate); const float processed_sample = RunAgc2WithConstantInput( agc2_fixed.get(), kInputLevel, kNumFrames, params.sample_rate); if (params.saturation_expected) { EXPECT_FLOAT_EQ(processed_sample, 32767.f); } else { EXPECT_LT(processed_sample, 32767.f); } } } 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). FixedDigitalTestParams(0.1f, test::kLimiterMaxInputLevelDbFs - 0.01f, 8000, false), FixedDigitalTestParams(0.1, test::kLimiterMaxInputLevelDbFs - 0.01f, 48000, false), // When gain > |test::kLimiterMaxInputLevelDbFs|, the limiter will // saturate the signal (at any sample rate). FixedDigitalTestParams(test::kLimiterMaxInputLevelDbFs + 0.01f, 10.f, 8000, true), FixedDigitalTestParams(test::kLimiterMaxInputLevelDbFs + 0.01f, 10.f, 48000, true))); // Checks that the gain applied at the end of a PCM samples file is close to the // expected value. TEST(GainController2, CheckGainAdaptiveDigital) { constexpr float kExpectedGainDb = 4.3f; constexpr float kToleranceDb = 0.5f; GainController2 gain_controller2; gain_controller2.Initialize(AudioProcessing::kSampleRate48kHz); AudioProcessing::Config::GainController2 config; config.fixed_digital.gain_db = 0.f; config.adaptive_digital.enabled = true; gain_controller2.ApplyConfig(config); EXPECT_NEAR( GainDbAfterProcessingFile(gain_controller2, /*max_duration_ms=*/2000), kExpectedGainDb, kToleranceDb); } } // namespace test } // namespace webrtc