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/*
* 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/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 {
using Agc2Config = AudioProcessing::Config::GainController2;
// 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) {
const int num_samples = rtc::CheckedDivExact(sample_rate_hz, 100);
AudioBuffer ab(sample_rate_hz, 1, sample_rate_hz, 1, sample_rate_hz, 1);
// Give time to the level estimator to converge.
for (int i = 0; i < num_frames + 1; ++i) {
SetAudioBufferSamples(input_level, ab);
agc2.Process(/*speech_probability=*/absl::nullopt, &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>(config, sample_rate_hz,
/*num_channels=*/1,
/*use_internal_vad=*/true);
}
} // 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));
}
// Checks that the default config is applied.
TEST(GainController2, ApplyDefaultConfig) {
auto gain_controller2 = std::make_unique<GainController2>(
Agc2Config{}, /*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(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=*/absl::nullopt, &audio_buffer);
}
// Estimate the applied gain by processing a probing frame.
SetAudioBufferSamples(/*value=*/1.0f, audio_buffer);
agc2.Process(/*speech_probability=*/absl::nullopt, &audio_buffer);
const float applied_gain_db =
20.0f * std::log10(audio_buffer.channels_const()[0][0]);
constexpr float kExpectedGainDb = 5.6f;
constexpr float kToleranceDb = 0.3f;
EXPECT_NEAR(applied_gain_db, kExpectedGainDb, kToleranceDb);
}
// Processes a test audio file and checks that the injected speech probability
// is ignored when the internal VAD is used.
TEST(GainController2,
CheckInjectedVadProbabilityNotUsedWithAdaptiveDigitalController) {
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(config, kSampleRateHz, kStereo,
/*use_internal_vad=*/true);
GainController2 agc2_reference(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;
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], &audio_buffer);
test::CopyVectorToAudioBuffer(stream_config, frame,
&audio_buffer_reference);
agc2_reference.Process(absl::nullopt, &audio_buffer_reference);
// Check the output buffers.
for (int i = 0; i < kStereo; ++i) {
for (int j = 0; j < static_cast<int>(audio_buffer.num_frames()); ++j) {
all_samples_zero &=
fabs(audio_buffer.channels_const()[i][j]) < kEpsilon;
EXPECT_FLOAT_EQ(audio_buffer.channels_const()[i][j],
audio_buffer_reference.channels_const()[i][j]);
}
}
}
EXPECT_FALSE(all_samples_zero);
}
// 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.
Agc2Config config;
config.fixed_digital.gain_db = 0.0f;
config.adaptive_digital.enabled = true;
GainController2 agc2(config, kSampleRateHz, kStereo,
/*use_internal_vad=*/false);
GainController2 agc2_reference(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], &audio_buffer);
test::CopyVectorToAudioBuffer(stream_config, frame,
&audio_buffer_reference);
agc2_reference.Process(absl::nullopt, &audio_buffer_reference);
// Check the output buffers.
for (int i = 0; i < kStereo; ++i) {
for (int j = 0; j < static_cast<int>(audio_buffer.num_frames()); ++j) {
all_samples_zero &=
fabs(audio_buffer.channels_const()[i][j]) < kEpsilon;
all_samples_equal &=
fabs(audio_buffer.channels_const()[i][j] -
audio_buffer_reference.channels_const()[i][j]) < 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.
Agc2Config config;
config.fixed_digital.gain_db = 0.0f;
config.adaptive_digital.enabled = true;
GainController2 agc2(config, kSampleRateHz, kStereo,
/*use_internal_vad=*/false);
GainController2 agc2_reference(config, kSampleRateHz, kStereo,
/*use_internal_vad=*/true);
VoiceActivityDetectorWrapper vad(config.adaptive_digital.vad_reset_period_ms,
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(absl::nullopt, &audio_buffer_reference);
test::CopyVectorToAudioBuffer(stream_config, frame, &audio_buffer);
agc2.Process(vad.Analyze(AudioFrameView<const float>(
audio_buffer.channels(), audio_buffer.num_channels(),
audio_buffer.num_frames())),
&audio_buffer);
// Check the output buffer.
for (int i = 0; i < kStereo; ++i) {
for (int j = 0; j < static_cast<int>(audio_buffer.num_frames()); ++j) {
EXPECT_FLOAT_EQ(audio_buffer.channels_const()[i][j],
audio_buffer_reference.channels_const()[i][j]);
}
}
}
}
} // namespace test
} // namespace webrtc