Google Git
Sign in
webrtc / src / / 76d265f48179b135694b667241b9ff8e17d36425 / . / modules / audio_processing / audio_processing_impl_unittest.cc
blob: e03f966b0655c1fbe010d2fb49e78ad9511803e2 [file] [log] [blame]
/*
* Copyright (c) 2014 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/audio_processing_impl.h"
#include <algorithm>
#include <array>
#include <memory>
#include <tuple>
#include "absl/types/optional.h"
#include "api/make_ref_counted.h"
#include "api/scoped_refptr.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "modules/audio_processing/optionally_built_submodule_creators.h"
#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
#include "modules/audio_processing/test/echo_canceller_test_tools.h"
#include "modules/audio_processing/test/echo_control_mock.h"
#include "modules/audio_processing/test/test_utils.h"
#include "rtc_base/checks.h"
#include "rtc_base/random.h"
#include "rtc_base/strings/string_builder.h"
#include "test/field_trial.h"
#include "test/gmock.h"
#include "test/gtest.h"
namespace webrtc {
namespace {
using ::testing::Invoke;
using ::testing::NotNull;
class MockInitialize : public AudioProcessingImpl {
public:
MockInitialize() : AudioProcessingImpl() {}
MOCK_METHOD(void, InitializeLocked, (), (override));
void RealInitializeLocked() {
AssertLockedForTest();
AudioProcessingImpl::InitializeLocked();
}
MOCK_METHOD(void, AddRef, (), (const, override));
MOCK_METHOD(RefCountReleaseStatus, Release, (), (const, override));
};
// Creates MockEchoControl instances and provides a raw pointer access to
// the next created one. The raw pointer is meant to be used with gmock.
// Returning a pointer of the next created MockEchoControl instance is necessary
// for the following reasons: (i) gmock expectations must be set before any call
// occurs, (ii) APM is initialized the first time that
// AudioProcessingImpl::ProcessStream() is called and the initialization leads
// to the creation of a new EchoControl object.
class MockEchoControlFactory : public EchoControlFactory {
public:
MockEchoControlFactory() : next_mock_(std::make_unique<MockEchoControl>()) {}
// Returns a pointer to the next MockEchoControl that this factory creates.
MockEchoControl* GetNext() const { return next_mock_.get(); }
std::unique_ptr<EchoControl> Create(int sample_rate_hz,
int num_render_channels,
int num_capture_channels) override {
std::unique_ptr<EchoControl> mock = std::move(next_mock_);
next_mock_ = std::make_unique<MockEchoControl>();
return mock;
}
private:
std::unique_ptr<MockEchoControl> next_mock_;
};
// Mocks EchoDetector and records the first samples of the last analyzed render
// stream frame. Used to check what data is read by an EchoDetector
// implementation injected into an APM.
class TestEchoDetector : public EchoDetector {
public:
TestEchoDetector()
: analyze_render_audio_called_(false),
last_render_audio_first_sample_(0.f) {}
~TestEchoDetector() override = default;
void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio) override {
last_render_audio_first_sample_ = render_audio[0];
analyze_render_audio_called_ = true;
}
void AnalyzeCaptureAudio(rtc::ArrayView<const float> capture_audio) override {
}
void Initialize(int capture_sample_rate_hz,
int num_capture_channels,
int render_sample_rate_hz,
int num_render_channels) override {}
EchoDetector::Metrics GetMetrics() const override { return {}; }
// Returns true if AnalyzeRenderAudio() has been called at least once.
bool analyze_render_audio_called() const {
return analyze_render_audio_called_;
}
// Returns the first sample of the last analyzed render frame.
float last_render_audio_first_sample() const {
return last_render_audio_first_sample_;
}
private:
bool analyze_render_audio_called_;
float last_render_audio_first_sample_;
};
// Mocks CustomProcessing and applies ProcessSample() to all the samples.
// Meant to be injected into an APM to modify samples in a known and detectable
// way.
class TestRenderPreProcessor : public CustomProcessing {
public:
TestRenderPreProcessor() = default;
~TestRenderPreProcessor() = default;
void Initialize(int sample_rate_hz, int num_channels) override {}
void Process(AudioBuffer* audio) override {
for (size_t k = 0; k < audio->num_channels(); ++k) {
rtc::ArrayView<float> channel_view(audio->channels()[k],
audio->num_frames());
std::transform(channel_view.begin(), channel_view.end(),
channel_view.begin(), ProcessSample);
}
}
std::string ToString() const override { return "TestRenderPreProcessor"; }
void SetRuntimeSetting(AudioProcessing::RuntimeSetting setting) override {}
// Modifies a sample. This member is used in Process() to modify a frame and
// it is publicly visible to enable tests.
static constexpr float ProcessSample(float x) { return 2.f * x; }
};
// Runs `apm` input processing for volume adjustments for `num_frames` random
// frames starting from the volume `initial_volume`. This includes three steps:
// 1) Set the input volume 2) Process the stream 3) Set the new recommended
// input volume. Returns the new recommended input volume.
int ProcessInputVolume(AudioProcessing& apm,
int num_frames,
int initial_volume) {
constexpr int kSampleRateHz = 48000;
constexpr int kNumChannels = 1;
std::array<float, kSampleRateHz / 100> buffer;
float* channel_pointers[] = {buffer.data()};
StreamConfig stream_config(/*sample_rate_hz=*/kSampleRateHz,
/*num_channels=*/kNumChannels);
int recommended_input_volume = initial_volume;
for (int i = 0; i < num_frames; ++i) {
Random random_generator(2341U);
RandomizeSampleVector(&random_generator, buffer);
apm.set_stream_analog_level(recommended_input_volume);
apm.ProcessStream(channel_pointers, stream_config, stream_config,
channel_pointers);
recommended_input_volume = apm.recommended_stream_analog_level();
}
return recommended_input_volume;
}
} // namespace
TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) {
MockInitialize mock;
ON_CALL(mock, InitializeLocked)
.WillByDefault(Invoke(&mock, &MockInitialize::RealInitializeLocked));
EXPECT_CALL(mock, InitializeLocked).Times(1);
mock.Initialize();
constexpr size_t kMaxSampleRateHz = 32000;
constexpr size_t kMaxNumChannels = 2;
std::array<int16_t, kMaxNumChannels * kMaxSampleRateHz / 100> frame;
frame.fill(0);
StreamConfig config(16000, 1);
// Call with the default parameters; there should be an init.
EXPECT_CALL(mock, InitializeLocked).Times(0);
EXPECT_NOERR(mock.ProcessStream(frame.data(), config, config, frame.data()));
EXPECT_NOERR(
mock.ProcessReverseStream(frame.data(), config, config, frame.data()));
// New sample rate. (Only impacts ProcessStream).
config = StreamConfig(32000, 1);
EXPECT_CALL(mock, InitializeLocked).Times(1);
EXPECT_NOERR(mock.ProcessStream(frame.data(), config, config, frame.data()));
// New number of channels.
config = StreamConfig(32000, 2);
EXPECT_CALL(mock, InitializeLocked).Times(2);
EXPECT_NOERR(mock.ProcessStream(frame.data(), config, config, frame.data()));
EXPECT_NOERR(
mock.ProcessReverseStream(frame.data(), config, config, frame.data()));
// A new sample rate passed to ProcessReverseStream should cause an init.
config = StreamConfig(16000, 2);
EXPECT_CALL(mock, InitializeLocked).Times(1);
EXPECT_NOERR(
mock.ProcessReverseStream(frame.data(), config, config, frame.data()));
}
TEST(AudioProcessingImplTest, UpdateCapturePreGainRuntimeSetting) {
rtc::scoped_refptr<AudioProcessing> apm =
AudioProcessingBuilderForTesting().Create();
webrtc::AudioProcessing::Config apm_config;
apm_config.pre_amplifier.enabled = true;
apm_config.pre_amplifier.fixed_gain_factor = 1.f;
apm->ApplyConfig(apm_config);
constexpr int kSampleRateHz = 48000;
constexpr int16_t kAudioLevel = 10000;
constexpr size_t kNumChannels = 2;
std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
StreamConfig config(kSampleRateHz, kNumChannels);
frame.fill(kAudioLevel);
apm->ProcessStream(frame.data(), config, config, frame.data());
EXPECT_EQ(frame[100], kAudioLevel)
<< "With factor 1, frame shouldn't be modified.";
constexpr float kGainFactor = 2.f;
apm->SetRuntimeSetting(
AudioProcessing::RuntimeSetting::CreateCapturePreGain(kGainFactor));
// Process for two frames to have time to ramp up gain.
for (int i = 0; i < 2; ++i) {
frame.fill(kAudioLevel);
apm->ProcessStream(frame.data(), config, config, frame.data());
}
EXPECT_EQ(frame[100], kGainFactor * kAudioLevel)
<< "Frame should be amplified.";
}
TEST(AudioProcessingImplTest,
LevelAdjustmentUpdateCapturePreGainRuntimeSetting) {
rtc::scoped_refptr<AudioProcessing> apm =
AudioProcessingBuilderForTesting().Create();
webrtc::AudioProcessing::Config apm_config;
apm_config.capture_level_adjustment.enabled = true;
apm_config.capture_level_adjustment.pre_gain_factor = 1.f;
apm->ApplyConfig(apm_config);
constexpr int kSampleRateHz = 48000;
constexpr int16_t kAudioLevel = 10000;
constexpr size_t kNumChannels = 2;
std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
StreamConfig config(kSampleRateHz, kNumChannels);
frame.fill(kAudioLevel);
apm->ProcessStream(frame.data(), config, config, frame.data());
EXPECT_EQ(frame[100], kAudioLevel)
<< "With factor 1, frame shouldn't be modified.";
constexpr float kGainFactor = 2.f;
apm->SetRuntimeSetting(
AudioProcessing::RuntimeSetting::CreateCapturePreGain(kGainFactor));
// Process for two frames to have time to ramp up gain.
for (int i = 0; i < 2; ++i) {
frame.fill(kAudioLevel);
apm->ProcessStream(frame.data(), config, config, frame.data());
}
EXPECT_EQ(frame[100], kGainFactor * kAudioLevel)
<< "Frame should be amplified.";
}
TEST(AudioProcessingImplTest,
LevelAdjustmentUpdateCapturePostGainRuntimeSetting) {
rtc::scoped_refptr<AudioProcessing> apm =
AudioProcessingBuilderForTesting().Create();
webrtc::AudioProcessing::Config apm_config;
apm_config.capture_level_adjustment.enabled = true;
apm_config.capture_level_adjustment.post_gain_factor = 1.f;
apm->ApplyConfig(apm_config);
constexpr int kSampleRateHz = 48000;
constexpr int16_t kAudioLevel = 10000;
constexpr size_t kNumChannels = 2;
std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
StreamConfig config(kSampleRateHz, kNumChannels);
frame.fill(kAudioLevel);
apm->ProcessStream(frame.data(), config, config, frame.data());
EXPECT_EQ(frame[100], kAudioLevel)
<< "With factor 1, frame shouldn't be modified.";
constexpr float kGainFactor = 2.f;
apm->SetRuntimeSetting(
AudioProcessing::RuntimeSetting::CreateCapturePostGain(kGainFactor));
// Process for two frames to have time to ramp up gain.
for (int i = 0; i < 2; ++i) {
frame.fill(kAudioLevel);
apm->ProcessStream(frame.data(), config, config, frame.data());
}
EXPECT_EQ(frame[100], kGainFactor * kAudioLevel)
<< "Frame should be amplified.";
}
TEST(AudioProcessingImplTest, EchoControllerObservesSetCaptureUsageChange) {
// Tests that the echo controller observes that the capture usage has been
// updated.
auto echo_control_factory = std::make_unique<MockEchoControlFactory>();
const MockEchoControlFactory* echo_control_factory_ptr =
echo_control_factory.get();
rtc::scoped_refptr<AudioProcessing> apm =
AudioProcessingBuilderForTesting()
.SetEchoControlFactory(std::move(echo_control_factory))
.Create();
constexpr int16_t kAudioLevel = 10000;
constexpr int kSampleRateHz = 48000;
constexpr int kNumChannels = 2;
std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
StreamConfig config(kSampleRateHz, kNumChannels);
frame.fill(kAudioLevel);
MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext();
// Ensure that SetCaptureOutputUsage is not called when no runtime settings
// are passed.
EXPECT_CALL(*echo_control_mock, SetCaptureOutputUsage(testing::_)).Times(0);
apm->ProcessStream(frame.data(), config, config, frame.data());
// Ensure that SetCaptureOutputUsage is called with the right information when
// a runtime setting is passed.
EXPECT_CALL(*echo_control_mock,
SetCaptureOutputUsage(/*capture_output_used=*/false))
.Times(1);
EXPECT_TRUE(apm->PostRuntimeSetting(
AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
/*capture_output_used=*/false)));
apm->ProcessStream(frame.data(), config, config, frame.data());
EXPECT_CALL(*echo_control_mock,
SetCaptureOutputUsage(/*capture_output_used=*/true))
.Times(1);
EXPECT_TRUE(apm->PostRuntimeSetting(
AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
/*capture_output_used=*/true)));
apm->ProcessStream(frame.data(), config, config, frame.data());
// The number of positions to place items in the queue is equal to the queue
// size minus 1.
constexpr int kNumSlotsInQueue = RuntimeSettingQueueSize();
// Ensure that SetCaptureOutputUsage is called with the right information when
// many runtime settings are passed.
for (int k = 0; k < kNumSlotsInQueue - 1; ++k) {
EXPECT_TRUE(apm->PostRuntimeSetting(
AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
/*capture_output_used=*/false)));
}
EXPECT_CALL(*echo_control_mock,
SetCaptureOutputUsage(/*capture_output_used=*/false))
.Times(kNumSlotsInQueue - 1);
apm->ProcessStream(frame.data(), config, config, frame.data());
// Ensure that SetCaptureOutputUsage is properly called with the fallback
// value when the runtime settings queue becomes full.
for (int k = 0; k < kNumSlotsInQueue; ++k) {
EXPECT_TRUE(apm->PostRuntimeSetting(
AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
/*capture_output_used=*/false)));
}
EXPECT_FALSE(apm->PostRuntimeSetting(
AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
/*capture_output_used=*/false)));
EXPECT_FALSE(apm->PostRuntimeSetting(
AudioProcessing::RuntimeSetting::CreateCaptureOutputUsedSetting(
/*capture_output_used=*/false)));
EXPECT_CALL(*echo_control_mock,
SetCaptureOutputUsage(/*capture_output_used=*/false))
.Times(kNumSlotsInQueue);
EXPECT_CALL(*echo_control_mock,
SetCaptureOutputUsage(/*capture_output_used=*/true))
.Times(1);
apm->ProcessStream(frame.data(), config, config, frame.data());
}
TEST(AudioProcessingImplTest,
EchoControllerObservesPreAmplifierEchoPathGainChange) {
// Tests that the echo controller observes an echo path gain change when the
// pre-amplifier submodule changes the gain.
auto echo_control_factory = std::make_unique<MockEchoControlFactory>();
const auto* echo_control_factory_ptr = echo_control_factory.get();
rtc::scoped_refptr<AudioProcessing> apm =
AudioProcessingBuilderForTesting()
.SetEchoControlFactory(std::move(echo_control_factory))
.Create();
// Disable AGC.
webrtc::AudioProcessing::Config apm_config;
apm_config.gain_controller1.enabled = false;
apm_config.gain_controller2.enabled = false;
apm_config.pre_amplifier.enabled = true;
apm_config.pre_amplifier.fixed_gain_factor = 1.f;
apm->ApplyConfig(apm_config);
constexpr int16_t kAudioLevel = 10000;
constexpr size_t kSampleRateHz = 48000;
constexpr size_t kNumChannels = 2;
std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
StreamConfig config(kSampleRateHz, kNumChannels);
frame.fill(kAudioLevel);
MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext();
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock,
ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
.Times(1);
apm->ProcessStream(frame.data(), config, config, frame.data());
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock,
ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/true))
.Times(1);
apm->SetRuntimeSetting(
AudioProcessing::RuntimeSetting::CreateCapturePreGain(2.f));
apm->ProcessStream(frame.data(), config, config, frame.data());
}
TEST(AudioProcessingImplTest,
EchoControllerObservesLevelAdjustmentPreGainEchoPathGainChange) {
// Tests that the echo controller observes an echo path gain change when the
// pre-amplifier submodule changes the gain.
auto echo_control_factory = std::make_unique<MockEchoControlFactory>();
const auto* echo_control_factory_ptr = echo_control_factory.get();
rtc::scoped_refptr<AudioProcessing> apm =
AudioProcessingBuilderForTesting()
.SetEchoControlFactory(std::move(echo_control_factory))
.Create();
// Disable AGC.
webrtc::AudioProcessing::Config apm_config;
apm_config.gain_controller1.enabled = false;
apm_config.gain_controller2.enabled = false;
apm_config.capture_level_adjustment.enabled = true;
apm_config.capture_level_adjustment.pre_gain_factor = 1.f;
apm->ApplyConfig(apm_config);
constexpr int16_t kAudioLevel = 10000;
constexpr size_t kSampleRateHz = 48000;
constexpr size_t kNumChannels = 2;
std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
StreamConfig config(kSampleRateHz, kNumChannels);
frame.fill(kAudioLevel);
MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext();
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock,
ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
.Times(1);
apm->ProcessStream(frame.data(), config, config, frame.data());
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock,
ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/true))
.Times(1);
apm->SetRuntimeSetting(
AudioProcessing::RuntimeSetting::CreateCapturePreGain(2.f));
apm->ProcessStream(frame.data(), config, config, frame.data());
}
TEST(AudioProcessingImplTest,
EchoControllerObservesAnalogAgc1EchoPathGainChange) {
// Tests that the echo controller observes an echo path gain change when the
// AGC1 analog adaptive submodule changes the analog gain.
auto echo_control_factory = std::make_unique<MockEchoControlFactory>();
const auto* echo_control_factory_ptr = echo_control_factory.get();
rtc::scoped_refptr<AudioProcessing> apm =
AudioProcessingBuilderForTesting()
.SetEchoControlFactory(std::move(echo_control_factory))
.Create();
webrtc::AudioProcessing::Config apm_config;
// Enable AGC1.
apm_config.gain_controller1.enabled = true;
apm_config.gain_controller1.analog_gain_controller.enabled = true;
apm_config.gain_controller2.enabled = false;
apm_config.pre_amplifier.enabled = false;
apm->ApplyConfig(apm_config);
constexpr int16_t kAudioLevel = 1000;
constexpr size_t kSampleRateHz = 48000;
constexpr size_t kNumChannels = 2;
std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
StreamConfig stream_config(kSampleRateHz, kNumChannels);
frame.fill(kAudioLevel);
MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext();
constexpr int kInitialStreamAnalogLevel = 123;
apm->set_stream_analog_level(kInitialStreamAnalogLevel);
// When the first fame is processed, no echo path gain change must be
// detected.
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock,
ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
.Times(1);
apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
// Simulate the application of the recommended analog level.
int recommended_analog_level = apm->recommended_stream_analog_level();
if (recommended_analog_level == kInitialStreamAnalogLevel) {
// Force an analog gain change if it did not happen.
recommended_analog_level++;
}
apm->set_stream_analog_level(recommended_analog_level);
// After the first fame and with a stream analog level change, the echo path
// gain change must be detected.
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock,
ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/true))
.Times(1);
apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
}
TEST(AudioProcessingImplTest,
ProcessWithAgc2AndTransientSuppressorVadModeDefault) {
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-GainController2/Disabled/");
auto apm = AudioProcessingBuilder()
.SetConfig({.gain_controller1{.enabled = false}})
.Create();
ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError);
webrtc::AudioProcessing::Config apm_config;
apm_config.gain_controller1.enabled = false;
apm_config.gain_controller2.enabled = true;
apm_config.gain_controller2.adaptive_digital.enabled = true;
apm_config.transient_suppression.enabled = true;
apm->ApplyConfig(apm_config);
constexpr int kSampleRateHz = 48000;
constexpr int kNumChannels = 1;
std::array<float, kSampleRateHz / 100> buffer;
float* channel_pointers[] = {buffer.data()};
StreamConfig stream_config(/*sample_rate_hz=*/kSampleRateHz,
/*num_channels=*/kNumChannels);
Random random_generator(2341U);
constexpr int kFramesToProcess = 10;
for (int i = 0; i < kFramesToProcess; ++i) {
RandomizeSampleVector(&random_generator, buffer);
ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
channel_pointers),
kNoErr);
}
}
TEST(AudioProcessingImplTest,
ProcessWithAgc2AndTransientSuppressorVadModeRnnVad) {
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/");
rtc::scoped_refptr<AudioProcessing> apm = AudioProcessingBuilder().Create();
ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError);
webrtc::AudioProcessing::Config apm_config;
apm_config.gain_controller1.enabled = false;
apm_config.gain_controller2.enabled = true;
apm_config.gain_controller2.adaptive_digital.enabled = true;
apm_config.transient_suppression.enabled = true;
apm->ApplyConfig(apm_config);
constexpr int kSampleRateHz = 48000;
constexpr int kNumChannels = 1;
std::array<float, kSampleRateHz / 100> buffer;
float* channel_pointers[] = {buffer.data()};
StreamConfig stream_config(/*sample_rate_hz=*/kSampleRateHz,
/*num_channels=*/kNumChannels);
Random random_generator(2341U);
constexpr int kFramesToProcess = 10;
for (int i = 0; i < kFramesToProcess; ++i) {
RandomizeSampleVector(&random_generator, buffer);
ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
channel_pointers),
kNoErr);
}
}
TEST(AudioProcessingImplTest, EchoControllerObservesPlayoutVolumeChange) {
// Tests that the echo controller observes an echo path gain change when a
// playout volume change is reported.
auto echo_control_factory = std::make_unique<MockEchoControlFactory>();
const auto* echo_control_factory_ptr = echo_control_factory.get();
rtc::scoped_refptr<AudioProcessing> apm =
AudioProcessingBuilderForTesting()
.SetEchoControlFactory(std::move(echo_control_factory))
.Create();
// Disable AGC.
webrtc::AudioProcessing::Config apm_config;
apm_config.gain_controller1.enabled = false;
apm_config.gain_controller2.enabled = false;
apm->ApplyConfig(apm_config);
constexpr int16_t kAudioLevel = 10000;
constexpr size_t kSampleRateHz = 48000;
constexpr size_t kNumChannels = 2;
std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
StreamConfig stream_config(kSampleRateHz, kNumChannels);
frame.fill(kAudioLevel);
MockEchoControl* echo_control_mock = echo_control_factory_ptr->GetNext();
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock,
ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
.Times(1);
apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock,
ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
.Times(1);
apm->SetRuntimeSetting(
AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(50));
apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock,
ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/false))
.Times(1);
apm->SetRuntimeSetting(
AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(50));
apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
EXPECT_CALL(*echo_control_mock, AnalyzeCapture(testing::_)).Times(1);
EXPECT_CALL(*echo_control_mock,
ProcessCapture(NotNull(), testing::_, /*echo_path_change=*/true))
.Times(1);
apm->SetRuntimeSetting(
AudioProcessing::RuntimeSetting::CreatePlayoutVolumeChange(100));
apm->ProcessStream(frame.data(), stream_config, stream_config, frame.data());
}
TEST(AudioProcessingImplTest, RenderPreProcessorBeforeEchoDetector) {
// Make sure that signal changes caused by a render pre-processing sub-module
// take place before any echo detector analysis.
auto test_echo_detector = rtc::make_ref_counted<TestEchoDetector>();
std::unique_ptr<CustomProcessing> test_render_pre_processor(
new TestRenderPreProcessor());
// Create APM injecting the test echo detector and render pre-processor.
rtc::scoped_refptr<AudioProcessing> apm =
AudioProcessingBuilderForTesting()
.SetEchoDetector(test_echo_detector)
.SetRenderPreProcessing(std::move(test_render_pre_processor))
.Create();
webrtc::AudioProcessing::Config apm_config;
apm_config.pre_amplifier.enabled = true;
apm->ApplyConfig(apm_config);
constexpr int16_t kAudioLevel = 1000;
constexpr int kSampleRateHz = 16000;
constexpr size_t kNumChannels = 1;
// Explicitly initialize APM to ensure no render frames are discarded.
const ProcessingConfig processing_config = {{
{kSampleRateHz, kNumChannels},
{kSampleRateHz, kNumChannels},
{kSampleRateHz, kNumChannels},
{kSampleRateHz, kNumChannels},
}};
apm->Initialize(processing_config);
std::array<int16_t, kNumChannels * kSampleRateHz / 100> frame;
StreamConfig stream_config(kSampleRateHz, kNumChannels);
constexpr float kAudioLevelFloat = static_cast<float>(kAudioLevel);
constexpr float kExpectedPreprocessedAudioLevel =
TestRenderPreProcessor::ProcessSample(kAudioLevelFloat);
ASSERT_NE(kAudioLevelFloat, kExpectedPreprocessedAudioLevel);
// Analyze a render stream frame.
frame.fill(kAudioLevel);
ASSERT_EQ(AudioProcessing::Error::kNoError,
apm->ProcessReverseStream(frame.data(), stream_config,
stream_config, frame.data()));
// Trigger a call to in EchoDetector::AnalyzeRenderAudio() via
// ProcessStream().
frame.fill(kAudioLevel);
ASSERT_EQ(AudioProcessing::Error::kNoError,
apm->ProcessStream(frame.data(), stream_config, stream_config,
frame.data()));
// Regardless of how the call to in EchoDetector::AnalyzeRenderAudio() is
// triggered, the line below checks that the call has occurred. If not, the
// APM implementation may have changed and this test might need to be adapted.
ASSERT_TRUE(test_echo_detector->analyze_render_audio_called());
// Check that the data read in EchoDetector::AnalyzeRenderAudio() is that
// produced by the render pre-processor.
EXPECT_EQ(kExpectedPreprocessedAudioLevel,
test_echo_detector->last_render_audio_first_sample());
}
// Disabling build-optional submodules and trying to enable them via the APM
// config should be bit-exact with running APM with said submodules disabled.
// This mainly tests that SetCreateOptionalSubmodulesForTesting has an effect.
TEST(ApmWithSubmodulesExcludedTest, BitexactWithDisabledModules) {
auto apm = rtc::make_ref_counted<AudioProcessingImpl>();
ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError);
ApmSubmoduleCreationOverrides overrides;
overrides.transient_suppression = true;
apm->OverrideSubmoduleCreationForTesting(overrides);
AudioProcessing::Config apm_config = apm->GetConfig();
apm_config.transient_suppression.enabled = true;
apm->ApplyConfig(apm_config);
rtc::scoped_refptr<AudioProcessing> apm_reference =
AudioProcessingBuilder().Create();
apm_config = apm_reference->GetConfig();
apm_config.transient_suppression.enabled = false;
apm_reference->ApplyConfig(apm_config);
constexpr int kSampleRateHz = 16000;
constexpr int kNumChannels = 1;
std::array<float, kSampleRateHz / 100> buffer;
std::array<float, kSampleRateHz / 100> buffer_reference;
float* channel_pointers[] = {buffer.data()};
float* channel_pointers_reference[] = {buffer_reference.data()};
StreamConfig stream_config(/*sample_rate_hz=*/kSampleRateHz,
/*num_channels=*/kNumChannels);
Random random_generator(2341U);
constexpr int kFramesToProcessPerConfiguration = 10;
for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
RandomizeSampleVector(&random_generator, buffer);
std::copy(buffer.begin(), buffer.end(), buffer_reference.begin());
ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
channel_pointers),
kNoErr);
ASSERT_EQ(
apm_reference->ProcessStream(channel_pointers_reference, stream_config,
stream_config, channel_pointers_reference),
kNoErr);
for (int j = 0; j < kSampleRateHz / 100; ++j) {
EXPECT_EQ(buffer[j], buffer_reference[j]);
}
}
}
// Disable transient suppressor creation and run APM in ways that should trigger
// calls to the transient suppressor API.
TEST(ApmWithSubmodulesExcludedTest, ReinitializeTransientSuppressor) {
auto apm = rtc::make_ref_counted<AudioProcessingImpl>();
ASSERT_EQ(apm->Initialize(), kNoErr);
ApmSubmoduleCreationOverrides overrides;
overrides.transient_suppression = true;
apm->OverrideSubmoduleCreationForTesting(overrides);
AudioProcessing::Config config = apm->GetConfig();
config.transient_suppression.enabled = true;
apm->ApplyConfig(config);
// 960 samples per frame: 10 ms of <= 48 kHz audio with <= 2 channels.
float buffer[960];
float* channel_pointers[] = {&buffer[0], &buffer[480]};
Random random_generator(2341U);
constexpr int kFramesToProcessPerConfiguration = 3;
StreamConfig initial_stream_config(/*sample_rate_hz=*/16000,
/*num_channels=*/1);
for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
RandomizeSampleVector(&random_generator, buffer);
EXPECT_EQ(apm->ProcessStream(channel_pointers, initial_stream_config,
initial_stream_config, channel_pointers),
kNoErr);
}
StreamConfig stereo_stream_config(/*sample_rate_hz=*/16000,
/*num_channels=*/2);
for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
RandomizeSampleVector(&random_generator, buffer);
EXPECT_EQ(apm->ProcessStream(channel_pointers, stereo_stream_config,
stereo_stream_config, channel_pointers),
kNoErr);
}
StreamConfig high_sample_rate_stream_config(/*sample_rate_hz=*/48000,
/*num_channels=*/2);
for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
RandomizeSampleVector(&random_generator, buffer);
EXPECT_EQ(
apm->ProcessStream(channel_pointers, high_sample_rate_stream_config,
high_sample_rate_stream_config, channel_pointers),
kNoErr);
}
}
// Disable transient suppressor creation and run APM in ways that should trigger
// calls to the transient suppressor API.
TEST(ApmWithSubmodulesExcludedTest, ToggleTransientSuppressor) {
auto apm = rtc::make_ref_counted<AudioProcessingImpl>();
ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError);
ApmSubmoduleCreationOverrides overrides;
overrides.transient_suppression = true;
apm->OverrideSubmoduleCreationForTesting(overrides);
// 960 samples per frame: 10 ms of <= 48 kHz audio with <= 2 channels.
float buffer[960];
float* channel_pointers[] = {&buffer[0], &buffer[480]};
Random random_generator(2341U);
constexpr int kFramesToProcessPerConfiguration = 3;
StreamConfig stream_config(/*sample_rate_hz=*/16000,
/*num_channels=*/1);
AudioProcessing::Config config = apm->GetConfig();
config.transient_suppression.enabled = true;
apm->ApplyConfig(config);
for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
RandomizeSampleVector(&random_generator, buffer);
EXPECT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
channel_pointers),
kNoErr);
}
config = apm->GetConfig();
config.transient_suppression.enabled = false;
apm->ApplyConfig(config);
for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
RandomizeSampleVector(&random_generator, buffer);
EXPECT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
channel_pointers),
kNoErr);
}
config = apm->GetConfig();
config.transient_suppression.enabled = true;
apm->ApplyConfig(config);
for (int i = 0; i < kFramesToProcessPerConfiguration; ++i) {
RandomizeSampleVector(&random_generator, buffer);
EXPECT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
channel_pointers),
kNoErr);
}
}
class StartupInputVolumeParameterizedTest
: public ::testing::TestWithParam<int> {};
// Tests that, when no input volume controller is used, the startup input volume
// is never modified.
TEST_P(StartupInputVolumeParameterizedTest,
WithNoInputVolumeControllerStartupVolumeNotModified) {
webrtc::AudioProcessing::Config config;
config.gain_controller1.enabled = false;
config.gain_controller2.enabled = false;
auto apm = AudioProcessingBuilder().SetConfig(config).Create();
int startup_volume = GetParam();
int recommended_volume = ProcessInputVolume(
*apm, /*num_frames=*/1, /*initial_volume=*/startup_volume);
EXPECT_EQ(recommended_volume, startup_volume);
}
INSTANTIATE_TEST_SUITE_P(AudioProcessingImplTest,
StartupInputVolumeParameterizedTest,
::testing::Values(0, 5, 15, 50, 100));
// Tests that, when no input volume controller is used, the recommended input
// volume always matches the applied one.
TEST(AudioProcessingImplTest,
WithNoInputVolumeControllerAppliedAndRecommendedVolumesMatch) {
webrtc::AudioProcessing::Config config;
config.gain_controller1.enabled = false;
config.gain_controller2.enabled = false;
auto apm = AudioProcessingBuilder().SetConfig(config).Create();
Random rand_gen(42);
for (int i = 0; i < 32; ++i) {
SCOPED_TRACE(i);
int32_t applied_volume = rand_gen.Rand(/*low=*/0, /*high=*/255);
int recommended_volume =
ProcessInputVolume(*apm, /*num_frames=*/1, applied_volume);
EXPECT_EQ(recommended_volume, applied_volume);
}
}
class ApmInputVolumeControllerParametrizedTest
: public ::testing::TestWithParam<
std::tuple<int, int, AudioProcessing::Config>> {
protected:
ApmInputVolumeControllerParametrizedTest()
: sample_rate_hz_(std::get<0>(GetParam())),
num_channels_(std::get<1>(GetParam())),
channels_(num_channels_),
channel_pointers_(num_channels_) {
const int frame_size = sample_rate_hz_ / 100;
for (int c = 0; c < num_channels_; ++c) {
channels_[c].resize(frame_size);
channel_pointers_[c] = channels_[c].data();
std::fill(channels_[c].begin(), channels_[c].end(), 0.0f);
}
}
int sample_rate_hz() const { return sample_rate_hz_; }
int num_channels() const { return num_channels_; }
AudioProcessing::Config GetConfig() const { return std::get<2>(GetParam()); }
float* const* channel_pointers() { return channel_pointers_.data(); }
private:
const int sample_rate_hz_;
const int num_channels_;
std::vector<std::vector<float>> channels_;
std::vector<float*> channel_pointers_;
};
TEST_P(ApmInputVolumeControllerParametrizedTest,
EnforceMinInputVolumeAtStartupWithZeroVolume) {
const StreamConfig stream_config(sample_rate_hz(), num_channels());
auto apm = AudioProcessingBuilder().SetConfig(GetConfig()).Create();
apm->set_stream_analog_level(0);
apm->ProcessStream(channel_pointers(), stream_config, stream_config,
channel_pointers());
EXPECT_GT(apm->recommended_stream_analog_level(), 0);
}
TEST_P(ApmInputVolumeControllerParametrizedTest,
EnforceMinInputVolumeAtStartupWithNonZeroVolume) {
const StreamConfig stream_config(sample_rate_hz(), num_channels());
auto apm = AudioProcessingBuilder().SetConfig(GetConfig()).Create();
constexpr int kStartupVolume = 3;
apm->set_stream_analog_level(kStartupVolume);
apm->ProcessStream(channel_pointers(), stream_config, stream_config,
channel_pointers());
EXPECT_GT(apm->recommended_stream_analog_level(), kStartupVolume);
}
TEST_P(ApmInputVolumeControllerParametrizedTest,
EnforceMinInputVolumeAfterManualVolumeAdjustment) {
const auto config = GetConfig();
if (config.gain_controller1.enabled) {
// After a downward manual adjustment, AGC1 slowly converges to the minimum
// input volume.
GTEST_SKIP() << "Does not apply to AGC1";
}
const StreamConfig stream_config(sample_rate_hz(), num_channels());
auto apm = AudioProcessingBuilder().SetConfig(GetConfig()).Create();
apm->set_stream_analog_level(20);
apm->ProcessStream(channel_pointers(), stream_config, stream_config,
channel_pointers());
constexpr int kManuallyAdjustedVolume = 3;
apm->set_stream_analog_level(kManuallyAdjustedVolume);
apm->ProcessStream(channel_pointers(), stream_config, stream_config,
channel_pointers());
EXPECT_GT(apm->recommended_stream_analog_level(), kManuallyAdjustedVolume);
}
TEST_P(ApmInputVolumeControllerParametrizedTest,
DoNotEnforceMinInputVolumeAtStartupWithHighVolume) {
const StreamConfig stream_config(sample_rate_hz(), num_channels());
auto apm = AudioProcessingBuilder().SetConfig(GetConfig()).Create();
constexpr int kStartupVolume = 200;
apm->set_stream_analog_level(kStartupVolume);
apm->ProcessStream(channel_pointers(), stream_config, stream_config,
channel_pointers());
EXPECT_EQ(apm->recommended_stream_analog_level(), kStartupVolume);
}
TEST_P(ApmInputVolumeControllerParametrizedTest,
DoNotEnforceMinInputVolumeAfterManualVolumeAdjustmentToZero) {
const StreamConfig stream_config(sample_rate_hz(), num_channels());
auto apm = AudioProcessingBuilder().SetConfig(GetConfig()).Create();
apm->set_stream_analog_level(100);
apm->ProcessStream(channel_pointers(), stream_config, stream_config,
channel_pointers());
apm->set_stream_analog_level(0);
apm->ProcessStream(channel_pointers(), stream_config, stream_config,
channel_pointers());
EXPECT_EQ(apm->recommended_stream_analog_level(), 0);
}
INSTANTIATE_TEST_SUITE_P(
AudioProcessingImplTest,
ApmInputVolumeControllerParametrizedTest,
::testing::Combine(
::testing::Values(8000, 16000, 32000, 48000), // Sample rates.
::testing::Values(1, 2), // Number of channels.
::testing::Values(
// Full AGC1.
AudioProcessing::Config{
.gain_controller1 = {.enabled = true,
.analog_gain_controller =
{.enabled = true,
.enable_digital_adaptive = true}},
.gain_controller2 = {.enabled = false}},
// Hybrid AGC.
AudioProcessing::Config{
.gain_controller1 = {.enabled = true,
.analog_gain_controller =
{.enabled = true,
.enable_digital_adaptive = false}},
.gain_controller2 = {.enabled = true,
.adaptive_digital = {.enabled = true}}})));
// When the input volume is not emulated and no input volume controller is
// active, the recommended volume must always be the applied volume.
TEST(AudioProcessingImplTest,
RecommendAppliedInputVolumeWithNoAgcWithNoEmulation) {
auto apm = AudioProcessingBuilder()
.SetConfig({.capture_level_adjustment = {.enabled = false},
.gain_controller1 = {.enabled = false}})
.Create();
constexpr int kOneFrame = 1;
EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/123), 123);
EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/59), 59);
EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/135), 135);
}
// When the input volume is emulated, the recommended volume must always be the
// applied volume and at any time it must not be that set in the input volume
// emulator.
// TODO(bugs.webrtc.org/14581): Enable when APM fixed to let this test pass.
TEST(AudioProcessingImplTest,
DISABLED_RecommendAppliedInputVolumeWithNoAgcWithEmulation) {
auto apm =
AudioProcessingBuilder()
.SetConfig({.capture_level_adjustment = {.enabled = true,
.analog_mic_gain_emulation{
.enabled = true,
.initial_level = 255}},
.gain_controller1 = {.enabled = false}})
.Create();
constexpr int kOneFrame = 1;
EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/123), 123);
EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/59), 59);
EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/135), 135);
}
// Even if there is an enabled input volume controller, when the input volume is
// emulated, the recommended volume is always the applied volume because the
// active controller must only adjust the internally emulated volume and leave
// the externally applied volume unchanged.
// TODO(bugs.webrtc.org/14581): Enable when APM fixed to let this test pass.
TEST(AudioProcessingImplTest,
DISABLED_RecommendAppliedInputVolumeWithAgcWithEmulation) {
auto apm =
AudioProcessingBuilder()
.SetConfig({.capture_level_adjustment = {.enabled = true,
.analog_mic_gain_emulation{
.enabled = true}},
.gain_controller1 = {.enabled = true,
.analog_gain_controller{
.enabled = true,
}}})
.Create();
constexpr int kOneFrame = 1;
EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/123), 123);
EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/59), 59);
EXPECT_EQ(ProcessInputVolume(*apm, kOneFrame, /*initial_volume=*/135), 135);
}
TEST(AudioProcessingImplTest,
Agc2FieldTrialDoNotSwitchToFullAgc2WhenNoAgcIsActive) {
constexpr AudioProcessing::Config kOriginal{
.gain_controller1{.enabled = false},
.gain_controller2{.enabled = false},
};
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/");
// Test config application via `AudioProcessing` ctor.
auto adjusted =
AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1);
EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2);
// Test config application via `AudioProcessing::ApplyConfig()`.
auto apm = AudioProcessingBuilder().Create();
apm->ApplyConfig(kOriginal);
adjusted = apm->GetConfig();
EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1);
EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2);
}
TEST(AudioProcessingImplTest,
Agc2FieldTrialDoNotSwitchToFullAgc2WithAgc1Agc2InputVolumeControllers) {
constexpr AudioProcessing::Config kOriginal{
.gain_controller1{.enabled = true,
.analog_gain_controller{.enabled = true}},
.gain_controller2{.enabled = true,
.input_volume_controller{.enabled = true}},
};
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/");
// Test config application via `AudioProcessing` ctor.
auto adjusted =
AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1);
EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2);
// Test config application via `AudioProcessing::ApplyConfig()`.
auto apm = AudioProcessingBuilder().Create();
apm->ApplyConfig(kOriginal);
adjusted = apm->GetConfig();
EXPECT_EQ(adjusted.gain_controller1, kOriginal.gain_controller1);
EXPECT_EQ(adjusted.gain_controller2, kOriginal.gain_controller2);
}
class Agc2FieldTrialParametrizedTest
: public ::testing::TestWithParam<AudioProcessing::Config> {};
TEST_P(Agc2FieldTrialParametrizedTest, DoNotChangeConfigIfDisabled) {
const AudioProcessing::Config original = GetParam();
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-GainController2/Disabled/");
// Test config application via `AudioProcessing` ctor.
auto adjusted =
AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
// Test config application via `AudioProcessing::ApplyConfig()`.
auto apm = AudioProcessingBuilder().Create();
apm->ApplyConfig(original);
adjusted = apm->GetConfig();
EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
}
TEST_P(Agc2FieldTrialParametrizedTest, DoNotChangeConfigIfNoOverride) {
const AudioProcessing::Config original = GetParam();
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-GainController2/Enabled,"
"switch_to_agc2:false,"
"disallow_transient_suppressor_usage:false/");
// Test config application via `AudioProcessing` ctor.
auto adjusted =
AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
// Test config application via `AudioProcessing::ApplyConfig()`.
auto apm = AudioProcessingBuilder().Create();
apm->ApplyConfig(original);
adjusted = apm->GetConfig();
EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
}
TEST_P(Agc2FieldTrialParametrizedTest, DoNotSwitchToFullAgc2) {
const AudioProcessing::Config original = GetParam();
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-GainController2/Enabled,switch_to_agc2:false/");
// Test config application via `AudioProcessing` ctor.
auto adjusted =
AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
// Test config application via `AudioProcessing::ApplyConfig()`.
auto apm = AudioProcessingBuilder().Create();
apm->ApplyConfig(original);
adjusted = apm->GetConfig();
EXPECT_EQ(adjusted.gain_controller1, original.gain_controller1);
EXPECT_EQ(adjusted.gain_controller2, original.gain_controller2);
}
TEST_P(Agc2FieldTrialParametrizedTest, SwitchToFullAgc2) {
const AudioProcessing::Config original = GetParam();
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true/");
// Test config application via `AudioProcessing` ctor.
auto adjusted =
AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
EXPECT_FALSE(adjusted.gain_controller1.enabled);
EXPECT_TRUE(adjusted.gain_controller2.enabled);
EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
// Test config application via `AudioProcessing::ApplyConfig()`.
auto apm = AudioProcessingBuilder().Create();
apm->ApplyConfig(original);
adjusted = apm->GetConfig();
EXPECT_FALSE(adjusted.gain_controller1.enabled);
EXPECT_TRUE(adjusted.gain_controller2.enabled);
EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
}
TEST_P(Agc2FieldTrialParametrizedTest,
SwitchToFullAgc2AndOverrideInputVolumeControllerParameters) {
const AudioProcessing::Config original = GetParam();
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true,"
"min_input_volume:123,"
"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:80,"
"speech_probability_threshold:0.9,"
"speech_ratio_threshold:1.0/");
// Test config application via `AudioProcessing` ctor.
auto adjusted =
AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
EXPECT_FALSE(adjusted.gain_controller1.enabled);
EXPECT_TRUE(adjusted.gain_controller2.enabled);
EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
// Test config application via `AudioProcessing::ApplyConfig()`.
auto apm = AudioProcessingBuilder().Create();
apm->ApplyConfig(original);
adjusted = apm->GetConfig();
EXPECT_FALSE(adjusted.gain_controller1.enabled);
EXPECT_TRUE(adjusted.gain_controller2.enabled);
EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
}
TEST_P(Agc2FieldTrialParametrizedTest,
SwitchToFullAgc2AndOverrideAdaptiveDigitalControllerParameters) {
const AudioProcessing::Config original = GetParam();
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-GainController2/Enabled,switch_to_agc2:true,"
"headroom_db:10,"
"max_gain_db:20,"
"initial_gain_db:7,"
"max_gain_change_db_per_second:5,"
"max_output_noise_level_dbfs:-40/");
// Test config application via `AudioProcessing` ctor.
auto adjusted =
AudioProcessingBuilder().SetConfig(original).Create()->GetConfig();
EXPECT_FALSE(adjusted.gain_controller1.enabled);
EXPECT_TRUE(adjusted.gain_controller2.enabled);
EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
ASSERT_NE(adjusted.gain_controller2.adaptive_digital,
original.gain_controller2.adaptive_digital);
EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.headroom_db, 10);
EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.max_gain_db, 20);
EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.initial_gain_db, 7);
EXPECT_EQ(
adjusted.gain_controller2.adaptive_digital.max_gain_change_db_per_second,
5);
EXPECT_EQ(
adjusted.gain_controller2.adaptive_digital.max_output_noise_level_dbfs,
-40);
// Test config application via `AudioProcessing::ApplyConfig()`.
auto apm = AudioProcessingBuilder().Create();
apm->ApplyConfig(original);
adjusted = apm->GetConfig();
EXPECT_FALSE(adjusted.gain_controller1.enabled);
EXPECT_TRUE(adjusted.gain_controller2.enabled);
EXPECT_TRUE(adjusted.gain_controller2.input_volume_controller.enabled);
EXPECT_TRUE(adjusted.gain_controller2.adaptive_digital.enabled);
ASSERT_NE(adjusted.gain_controller2.adaptive_digital,
original.gain_controller2.adaptive_digital);
EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.headroom_db, 10);
EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.max_gain_db, 20);
EXPECT_EQ(adjusted.gain_controller2.adaptive_digital.initial_gain_db, 7);
EXPECT_EQ(
adjusted.gain_controller2.adaptive_digital.max_gain_change_db_per_second,
5);
EXPECT_EQ(
adjusted.gain_controller2.adaptive_digital.max_output_noise_level_dbfs,
-40);
}
TEST_P(Agc2FieldTrialParametrizedTest, ProcessSucceedsWithTs) {
AudioProcessing::Config config = GetParam();
if (!config.transient_suppression.enabled) {
GTEST_SKIP() << "TS is disabled, skip.";
}
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-GainController2/Disabled/");
auto apm = AudioProcessingBuilder().SetConfig(config).Create();
constexpr int kSampleRateHz = 48000;
constexpr int kNumChannels = 1;
std::array<float, kSampleRateHz / 100> buffer;
float* channel_pointers[] = {buffer.data()};
StreamConfig stream_config(kSampleRateHz, kNumChannels);
Random random_generator(2341U);
constexpr int kFramesToProcess = 10;
int volume = 100;
for (int i = 0; i < kFramesToProcess; ++i) {
SCOPED_TRACE(i);
RandomizeSampleVector(&random_generator, buffer);
apm->set_stream_analog_level(volume);
ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
channel_pointers),
kNoErr);
volume = apm->recommended_stream_analog_level();
}
}
TEST_P(Agc2FieldTrialParametrizedTest, ProcessSucceedsWithoutTs) {
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-GainController2/Enabled,"
"switch_to_agc2:false,"
"disallow_transient_suppressor_usage:true/");
auto apm = AudioProcessingBuilder().SetConfig(GetParam()).Create();
constexpr int kSampleRateHz = 48000;
constexpr int kNumChannels = 1;
std::array<float, kSampleRateHz / 100> buffer;
float* channel_pointers[] = {buffer.data()};
StreamConfig stream_config(kSampleRateHz, kNumChannels);
Random random_generator(2341U);
constexpr int kFramesToProcess = 10;
int volume = 100;
for (int i = 0; i < kFramesToProcess; ++i) {
SCOPED_TRACE(i);
RandomizeSampleVector(&random_generator, buffer);
apm->set_stream_analog_level(volume);
ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
channel_pointers),
kNoErr);
volume = apm->recommended_stream_analog_level();
}
}
TEST_P(Agc2FieldTrialParametrizedTest,
ProcessSucceedsWhenSwitchToFullAgc2WithTs) {
AudioProcessing::Config config = GetParam();
if (!config.transient_suppression.enabled) {
GTEST_SKIP() << "TS is disabled, skip.";
}
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-GainController2/Enabled,"
"switch_to_agc2:true,"
"disallow_transient_suppressor_usage:false/");
auto apm = AudioProcessingBuilder().SetConfig(config).Create();
constexpr int kSampleRateHz = 48000;
constexpr int kNumChannels = 1;
std::array<float, kSampleRateHz / 100> buffer;
float* channel_pointers[] = {buffer.data()};
StreamConfig stream_config(kSampleRateHz, kNumChannels);
Random random_generator(2341U);
constexpr int kFramesToProcess = 10;
int volume = 100;
for (int i = 0; i < kFramesToProcess; ++i) {
SCOPED_TRACE(i);
RandomizeSampleVector(&random_generator, buffer);
apm->set_stream_analog_level(volume);
ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
channel_pointers),
kNoErr);
volume = apm->recommended_stream_analog_level();
}
}
TEST_P(Agc2FieldTrialParametrizedTest,
ProcessSucceedsWhenSwitchToFullAgc2WithoutTs) {
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-GainController2/Enabled,"
"switch_to_agc2:true,"
"disallow_transient_suppressor_usage:true/");
auto apm = AudioProcessingBuilder().SetConfig(GetParam()).Create();
constexpr int kSampleRateHz = 48000;
constexpr int kNumChannels = 1;
std::array<float, kSampleRateHz / 100> buffer;
float* channel_pointers[] = {buffer.data()};
StreamConfig stream_config(kSampleRateHz, kNumChannels);
Random random_generator(2341U);
constexpr int kFramesToProcess = 10;
int volume = 100;
for (int i = 0; i < kFramesToProcess; ++i) {
SCOPED_TRACE(i);
RandomizeSampleVector(&random_generator, buffer);
apm->set_stream_analog_level(volume);
ASSERT_EQ(apm->ProcessStream(channel_pointers, stream_config, stream_config,
channel_pointers),
kNoErr);
volume = apm->recommended_stream_analog_level();
}
}
INSTANTIATE_TEST_SUITE_P(
AudioProcessingImplTest,
Agc2FieldTrialParametrizedTest,
::testing::Values(
// Full AGC1, TS disabled.
AudioProcessing::Config{
.transient_suppression = {.enabled = false},
.gain_controller1 =
{.enabled = true,
.analog_gain_controller = {.enabled = true,
.enable_digital_adaptive = true}},
.gain_controller2 = {.enabled = false}},
// Full AGC1, TS enabled.
AudioProcessing::Config{
.transient_suppression = {.enabled = true},
.gain_controller1 =
{.enabled = true,
.analog_gain_controller = {.enabled = true,
.enable_digital_adaptive = true}},
.gain_controller2 = {.enabled = false}},
// Hybrid AGC, TS disabled.
AudioProcessing::Config{
.transient_suppression = {.enabled = false},
.gain_controller1 =
{.enabled = true,
.analog_gain_controller = {.enabled = true,
.enable_digital_adaptive = false}},
.gain_controller2 = {.enabled = true,
.adaptive_digital = {.enabled = true}}},
// Hybrid AGC, TS enabled.
AudioProcessing::Config{
.transient_suppression = {.enabled = true},
.gain_controller1 =
{.enabled = true,
.analog_gain_controller = {.enabled = true,
.enable_digital_adaptive = false}},
.gain_controller2 = {.enabled = true,
.adaptive_digital = {.enabled = true}}}));
TEST(AudioProcessingImplTest, CanDisableTransientSuppressor) {
constexpr AudioProcessing::Config kOriginal = {
.transient_suppression = {.enabled = false}};
// Test config application via `AudioProcessing` ctor.
auto adjusted =
AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
EXPECT_FALSE(adjusted.transient_suppression.enabled);
// Test config application via `AudioProcessing::ApplyConfig()`.
auto apm = AudioProcessingBuilder().Create();
apm->ApplyConfig(kOriginal);
adjusted = apm->GetConfig();
EXPECT_FALSE(apm->GetConfig().transient_suppression.enabled);
}
TEST(AudioProcessingImplTest, CanEnableTs) {
constexpr AudioProcessing::Config kOriginal = {
.transient_suppression = {.enabled = true}};
// Test config application via `AudioProcessing` ctor.
auto adjusted =
AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
EXPECT_TRUE(adjusted.transient_suppression.enabled);
// Test config application via `AudioProcessing::ApplyConfig()`.
auto apm = AudioProcessingBuilder().Create();
apm->ApplyConfig(kOriginal);
adjusted = apm->GetConfig();
EXPECT_TRUE(adjusted.transient_suppression.enabled);
}
TEST(AudioProcessingImplTest, CanDisableTsWithAgc2FieldTrialDisabled) {
constexpr AudioProcessing::Config kOriginal = {
.transient_suppression = {.enabled = false}};
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-GainController2/Disabled/");
// Test config application via `AudioProcessing` ctor.
auto adjusted =
AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
EXPECT_FALSE(adjusted.transient_suppression.enabled);
// Test config application via `AudioProcessing::ApplyConfig()`.
auto apm = AudioProcessingBuilder().Create();
apm->ApplyConfig(kOriginal);
adjusted = apm->GetConfig();
EXPECT_FALSE(apm->GetConfig().transient_suppression.enabled);
}
TEST(AudioProcessingImplTest, CanEnableTsWithAgc2FieldTrialDisabled) {
constexpr AudioProcessing::Config kOriginal = {
.transient_suppression = {.enabled = true}};
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-GainController2/Disabled/");
// Test config application via `AudioProcessing` ctor.
auto adjusted =
AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
EXPECT_TRUE(adjusted.transient_suppression.enabled);
// Test config application via `AudioProcessing::ApplyConfig()`.
auto apm = AudioProcessingBuilder().Create();
apm->ApplyConfig(kOriginal);
adjusted = apm->GetConfig();
EXPECT_TRUE(adjusted.transient_suppression.enabled);
}
TEST(AudioProcessingImplTest,
CanDisableTsWithAgc2FieldTrialEnabledAndUsageAllowed) {
constexpr AudioProcessing::Config kOriginal = {
.transient_suppression = {.enabled = false}};
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-GainController2/Enabled,"
"disallow_transient_suppressor_usage:false/");
// Test config application via `AudioProcessing` ctor.
auto adjusted =
AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
EXPECT_FALSE(adjusted.transient_suppression.enabled);
// Test config application via `AudioProcessing::ApplyConfig()`.
auto apm = AudioProcessingBuilder().Create();
apm->ApplyConfig(kOriginal);
adjusted = apm->GetConfig();
EXPECT_FALSE(adjusted.transient_suppression.enabled);
}
TEST(AudioProcessingImplTest,
CanEnableTsWithAgc2FieldTrialEnabledAndUsageAllowed) {
constexpr AudioProcessing::Config kOriginal = {
.transient_suppression = {.enabled = true}};
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-GainController2/Enabled,"
"disallow_transient_suppressor_usage:false/");
// Test config application via `AudioProcessing` ctor.
auto adjusted =
AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
EXPECT_TRUE(adjusted.transient_suppression.enabled);
// Test config application via `AudioProcessing::ApplyConfig()`.
auto apm = AudioProcessingBuilder().Create();
apm->ApplyConfig(kOriginal);
adjusted = apm->GetConfig();
EXPECT_TRUE(adjusted.transient_suppression.enabled);
}
TEST(AudioProcessingImplTest,
CannotEnableTsWithAgc2FieldTrialEnabledAndUsageDisallowed) {
constexpr AudioProcessing::Config kOriginal = {
.transient_suppression = {.enabled = true}};
webrtc::test::ScopedFieldTrials field_trials(
"WebRTC-Audio-GainController2/Enabled,"
"disallow_transient_suppressor_usage:true/");
// Test config application via `AudioProcessing` ctor.
auto adjusted =
AudioProcessingBuilder().SetConfig(kOriginal).Create()->GetConfig();
EXPECT_FALSE(adjusted.transient_suppression.enabled);
// Test config application via `AudioProcessing::ApplyConfig()`.
auto apm = AudioProcessingBuilder().Create();
apm->ApplyConfig(kOriginal);
adjusted = apm->GetConfig();
EXPECT_FALSE(apm->GetConfig().transient_suppression.enabled);
}
} // namespace webrtc