mollyim/webrtc
1
0
Fork 0
mirror of https://github.com/mollyim/webrtc.git synced 2025-05-13 05:40:42 +01:00
Code Issues Projects Releases Packages Wiki Activity Actions
webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
Alessio Bazzica 40b5bd72d0 APM: fix TS initialization bugs with WebRTC-Audio-GainController2 
When the `WebRTC-Audio-GainController2` field trial is used, the
initial APM configuration is adjusted depending on its original
values and the field trial parameters.

This CL fixes two cases when the code crashes:
1. when, in the initial APM config, AGC1 is enabled, AGC2 is
   disabled and TS is enabled
2. when the initial APM sample rate is different from the
   capture one and the VAD APM sub-module is not re-initialized

This CL also improves the unit tests coverage and it has been
tested offline to check that the VAD sub-module is created only
when expected and that AGC2 uses its internal VAD when expected.
The tests ran on a few Wav files with different sample rates and
one AEC dump and on 16 different APM and field trial
configurations.

Bug: chromium:1407341, b/265112132
Change-Id: I7cc267ea81cb02be92c1f37f273b7ae93b6e4634
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290988
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39118}
2023-01-16 20:30:12 +00:00

1569 lines
64 KiB
C++
Raw Blame History

/*
* 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(rtc::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());
}
// Tests that a stream is successfully processed when AGC2 adaptive digital is
// used and when the field trial
// `WebRTC-Audio-TransientSuppressorVadMode/Enabled-Default/` is set.
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);
}
}
// Tests that a stream is successfully processed when AGC2 adaptive digital is
// used and when the field trial
// `WebRTC-Audio-TransientSuppressorVadMode/Enabled-RnnVad/` is set.
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
Reference in a new issue View git blame Copy permalink