modules/audio_processing/gain_controller2_unittest.cc - chromiumos/third_party/webrtc-apm - Git at Google

 /*
  *  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 <algorithm>

 #include "absl/memory/memory.h"
 #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/gain_controller2.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_f()[k], ab->channels_f()[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(num_samples, 1, num_samples, 1, num_samples);

   // 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_f()[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<GainController2> CreateAgc2FixedDigitalMode(
     float fixed_gain_db,
     size_t sample_rate_hz) {
   auto agc2 = absl::make_unique<GainController2>();
   agc2->ApplyConfig(CreateAgc2FixedDigitalModeConfig(fixed_gain_db));
   agc2->Initialize(sample_rate_hz);
   return agc2;
 }

 float GainAfterProcessingFile(GainController2* gain_controller) {
   // 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.num_frames(), capture_config.num_channels(),
                  capture_config.num_frames(), capture_config.num_channels(),
                  capture_config.num_frames());
   test::InputAudioFile capture_file(
       test::GetApmCaptureTestVectorFileName(AudioProcessing::kSampleRate48kHz));
   std::vector<float> capture_input(capture_config.num_frames() *
                                    capture_config.num_channels());

   // The file should contain at least this many frames. Every iteration, we put
   // a frame through the gain controller.
   const int kNumFramesToProcess = 100;
   for (int frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
     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 values constant 1 (It's low enough to detect high
   // gain, and for ease of computation). The applied gain is the result.
   constexpr float sample_value = 1.f;
   SetAudioBufferSamples(sample_value, &ab);
   gain_controller->Process(&ab);
   return ab.channels_f()[0][0];
 }

 }  // namespace

 TEST(GainController2, CreateApplyConfig) {
   // Instances GainController2 and applies different configurations.
   std::unique_ptr<GainController2> gain_controller2(new GainController2());

   // Check that the default config is valid.
   AudioProcessing::Config::GainController2 config;
   EXPECT_TRUE(GainController2::Validate(config));
   gain_controller2->ApplyConfig(config);

   // Check that attenuation is not allowed.
   config.fixed_digital.gain_db = -5.f;
   EXPECT_FALSE(GainController2::Validate(config));

   // Check that valid configurations are applied.
   for (const float& fixed_gain_db : {0.f, 5.f, 10.f, 40.f}) {
     config.fixed_digital.gain_db = fixed_gain_db;
     EXPECT_TRUE(GainController2::Validate(config));
     gain_controller2->ApplyConfig(config);
   }
 }

 TEST(GainController2, ToString) {
   // Tests GainController2::ToString(). Only test the enabled property.
   AudioProcessing::Config::GainController2 config;

   config.enabled = false;
   EXPECT_EQ("{enabled: false", GainController2::ToString(config).substr(0, 15));

   config.enabled = true;
   EXPECT_EQ("{enabled: true", GainController2::ToString(config).substr(0, 14));
 }

 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<void>(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<FixedDigitalTestParams> {};

 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)));

 TEST(GainController2, UsageSaturationMargin) {
   GainController2 gain_controller2;
   gain_controller2.Initialize(AudioProcessing::kSampleRate48kHz);

   AudioProcessing::Config::GainController2 config;
   // Check that samples are not amplified as much when extra margin is
   // high. They should not be amplified at all, but only after convergence. GC2
   // starts with a gain, and it takes time until it's down to 0 dB.
   config.fixed_digital.gain_db = 0.f;
   config.adaptive_digital.enabled = true;
   config.adaptive_digital.extra_saturation_margin_db = 50.f;
   gain_controller2.ApplyConfig(config);

   EXPECT_LT(GainAfterProcessingFile(&gain_controller2), 2.f);
 }

 TEST(GainController2, UsageNoSaturationMargin) {
   GainController2 gain_controller2;
   gain_controller2.Initialize(AudioProcessing::kSampleRate48kHz);

   AudioProcessing::Config::GainController2 config;
   // Check that some gain is applied if there is no margin.
   config.fixed_digital.gain_db = 0.f;
   config.adaptive_digital.enabled = true;
   config.adaptive_digital.extra_saturation_margin_db = 0.f;
   gain_controller2.ApplyConfig(config);

   EXPECT_GT(GainAfterProcessingFile(&gain_controller2), 2.f);
 }

 }  // namespace test
 }  // namespace webrtc
	/*
	* 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 <algorithm>

	#include "absl/memory/memory.h"
	#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/gain_controller2.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_f()[k], ab->channels_f()[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(num_samples, 1, num_samples, 1, num_samples);

	// 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_f()[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<GainController2> CreateAgc2FixedDigitalMode(
	float fixed_gain_db,
	size_t sample_rate_hz) {
	auto agc2 = absl::make_unique<GainController2>();
	agc2->ApplyConfig(CreateAgc2FixedDigitalModeConfig(fixed_gain_db));
	agc2->Initialize(sample_rate_hz);
	return agc2;
	}

	float GainAfterProcessingFile(GainController2* gain_controller) {
	// 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.num_frames(), capture_config.num_channels(),
	capture_config.num_frames(), capture_config.num_channels(),
	capture_config.num_frames());
	test::InputAudioFile capture_file(
	test::GetApmCaptureTestVectorFileName(AudioProcessing::kSampleRate48kHz));
	std::vector<float> capture_input(capture_config.num_frames() *
	capture_config.num_channels());

	// The file should contain at least this many frames. Every iteration, we put
	// a frame through the gain controller.
	const int kNumFramesToProcess = 100;
	for (int frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
	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 values constant 1 (It's low enough to detect high
	// gain, and for ease of computation). The applied gain is the result.
	constexpr float sample_value = 1.f;
	SetAudioBufferSamples(sample_value, &ab);
	gain_controller->Process(&ab);
	return ab.channels_f()[0][0];
	}

	} // namespace

	TEST(GainController2, CreateApplyConfig) {
	// Instances GainController2 and applies different configurations.
	std::unique_ptr<GainController2> gain_controller2(new GainController2());

	// Check that the default config is valid.
	AudioProcessing::Config::GainController2 config;
	EXPECT_TRUE(GainController2::Validate(config));
	gain_controller2->ApplyConfig(config);

	// Check that attenuation is not allowed.
	config.fixed_digital.gain_db = -5.f;
	EXPECT_FALSE(GainController2::Validate(config));

	// Check that valid configurations are applied.
	for (const float& fixed_gain_db : {0.f, 5.f, 10.f, 40.f}) {
	config.fixed_digital.gain_db = fixed_gain_db;
	EXPECT_TRUE(GainController2::Validate(config));
	gain_controller2->ApplyConfig(config);
	}
	}

	TEST(GainController2, ToString) {
	// Tests GainController2::ToString(). Only test the enabled property.
	AudioProcessing::Config::GainController2 config;

	config.enabled = false;
	EXPECT_EQ("{enabled: false", GainController2::ToString(config).substr(0, 15));

	config.enabled = true;
	EXPECT_EQ("{enabled: true", GainController2::ToString(config).substr(0, 14));
	}

	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<void>(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<FixedDigitalTestParams> {};

	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)));

	TEST(GainController2, UsageSaturationMargin) {
	GainController2 gain_controller2;
	gain_controller2.Initialize(AudioProcessing::kSampleRate48kHz);

	AudioProcessing::Config::GainController2 config;
	// Check that samples are not amplified as much when extra margin is
	// high. They should not be amplified at all, but only after convergence. GC2
	// starts with a gain, and it takes time until it's down to 0 dB.
	config.fixed_digital.gain_db = 0.f;
	config.adaptive_digital.enabled = true;
	config.adaptive_digital.extra_saturation_margin_db = 50.f;
	gain_controller2.ApplyConfig(config);

	EXPECT_LT(GainAfterProcessingFile(&gain_controller2), 2.f);
	}

	TEST(GainController2, UsageNoSaturationMargin) {
	GainController2 gain_controller2;
	gain_controller2.Initialize(AudioProcessing::kSampleRate48kHz);

	AudioProcessing::Config::GainController2 config;
	// Check that some gain is applied if there is no margin.
	config.fixed_digital.gain_db = 0.f;
	config.adaptive_digital.enabled = true;
	config.adaptive_digital.extra_saturation_margin_db = 0.f;
	gain_controller2.ApplyConfig(config);

	EXPECT_GT(GainAfterProcessingFile(&gain_controller2), 2.f);
	}

	} // namespace test
	} // namespace webrtc