blob: 95f736c22f168bfdb87267839d7b042ea3fe9916 [file] [log] [blame]
/*
* Copyright (c) 2015 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 <math.h>
#include <algorithm>
#include <memory>
#include <vector>
#include "api/array_view.h"
#include "modules/audio_processing/test/test_utils.h"
#include "rtc_base/atomicops.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/platform_thread.h"
#include "rtc_base/random.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/event_wrapper.h"
#include "test/gtest.h"
#include "test/testsupport/perf_test.h"
namespace webrtc {
namespace {
static const bool kPrintAllDurations = false;
class CallSimulator;
// Type of the render thread APM API call to use in the test.
enum class ProcessorType { kRender, kCapture };
// Variant of APM processing settings to use in the test.
enum class SettingsType {
kDefaultApmDesktop,
kDefaultApmMobile,
kAllSubmodulesTurnedOff,
kDefaultApmDesktopWithoutDelayAgnostic,
kDefaultApmDesktopWithoutExtendedFilter
};
// Variables related to the audio data and formats.
struct AudioFrameData {
explicit AudioFrameData(size_t max_frame_size) {
// Set up the two-dimensional arrays needed for the APM API calls.
input_framechannels.resize(2 * max_frame_size);
input_frame.resize(2);
input_frame[0] = &input_framechannels[0];
input_frame[1] = &input_framechannels[max_frame_size];
output_frame_channels.resize(2 * max_frame_size);
output_frame.resize(2);
output_frame[0] = &output_frame_channels[0];
output_frame[1] = &output_frame_channels[max_frame_size];
}
std::vector<float> output_frame_channels;
std::vector<float*> output_frame;
std::vector<float> input_framechannels;
std::vector<float*> input_frame;
StreamConfig input_stream_config;
StreamConfig output_stream_config;
};
// The configuration for the test.
struct SimulationConfig {
SimulationConfig(int sample_rate_hz, SettingsType simulation_settings)
: sample_rate_hz(sample_rate_hz),
simulation_settings(simulation_settings) {}
static std::vector<SimulationConfig> GenerateSimulationConfigs() {
std::vector<SimulationConfig> simulation_configs;
#ifndef WEBRTC_ANDROID
const SettingsType desktop_settings[] = {
SettingsType::kDefaultApmDesktop, SettingsType::kAllSubmodulesTurnedOff,
SettingsType::kDefaultApmDesktopWithoutDelayAgnostic,
SettingsType::kDefaultApmDesktopWithoutExtendedFilter};
const int desktop_sample_rates[] = {8000, 16000, 32000, 48000};
for (auto sample_rate : desktop_sample_rates) {
for (auto settings : desktop_settings) {
simulation_configs.push_back(SimulationConfig(sample_rate, settings));
}
}
#endif
const SettingsType mobile_settings[] = {SettingsType::kDefaultApmMobile};
const int mobile_sample_rates[] = {8000, 16000};
for (auto sample_rate : mobile_sample_rates) {
for (auto settings : mobile_settings) {
simulation_configs.push_back(SimulationConfig(sample_rate, settings));
}
}
return simulation_configs;
}
std::string SettingsDescription() const {
std::string description;
switch (simulation_settings) {
case SettingsType::kDefaultApmMobile:
description = "DefaultApmMobile";
break;
case SettingsType::kDefaultApmDesktop:
description = "DefaultApmDesktop";
break;
case SettingsType::kAllSubmodulesTurnedOff:
description = "AllSubmodulesOff";
break;
case SettingsType::kDefaultApmDesktopWithoutDelayAgnostic:
description = "DefaultApmDesktopWithoutDelayAgnostic";
break;
case SettingsType::kDefaultApmDesktopWithoutExtendedFilter:
description = "DefaultApmDesktopWithoutExtendedFilter";
break;
}
return description;
}
int sample_rate_hz = 16000;
SettingsType simulation_settings = SettingsType::kDefaultApmDesktop;
};
// Handler for the frame counters.
class FrameCounters {
public:
void IncreaseRenderCounter() { rtc::AtomicOps::Increment(&render_count_); }
void IncreaseCaptureCounter() { rtc::AtomicOps::Increment(&capture_count_); }
int CaptureMinusRenderCounters() const {
// The return value will be approximate, but that's good enough since
// by the time we return the value, it's not guaranteed to be correct
// anyway.
return rtc::AtomicOps::AcquireLoad(&capture_count_) -
rtc::AtomicOps::AcquireLoad(&render_count_);
}
int RenderMinusCaptureCounters() const {
return -CaptureMinusRenderCounters();
}
bool BothCountersExceedeThreshold(int threshold) const {
// TODO(tommi): We could use an event to signal this so that we don't need
// to be polling from the main thread and possibly steal cycles.
const int capture_count = rtc::AtomicOps::AcquireLoad(&capture_count_);
const int render_count = rtc::AtomicOps::AcquireLoad(&render_count_);
return (render_count > threshold && capture_count > threshold);
}
private:
int render_count_ = 0;
int capture_count_ = 0;
};
// Class that represents a flag that can only be raised.
class LockedFlag {
public:
bool get_flag() const { return rtc::AtomicOps::AcquireLoad(&flag_); }
void set_flag() {
if (!get_flag()) // read-only operation to avoid affecting the cache-line.
rtc::AtomicOps::CompareAndSwap(&flag_, 0, 1);
}
private:
int flag_ = 0;
};
// Parent class for the thread processors.
class TimedThreadApiProcessor {
public:
TimedThreadApiProcessor(ProcessorType processor_type,
Random* rand_gen,
FrameCounters* shared_counters_state,
LockedFlag* capture_call_checker,
CallSimulator* test_framework,
const SimulationConfig* simulation_config,
AudioProcessing* apm,
int num_durations_to_store,
float input_level,
int num_channels)
: rand_gen_(rand_gen),
frame_counters_(shared_counters_state),
capture_call_checker_(capture_call_checker),
test_(test_framework),
simulation_config_(simulation_config),
apm_(apm),
frame_data_(kMaxFrameSize),
clock_(webrtc::Clock::GetRealTimeClock()),
num_durations_to_store_(num_durations_to_store),
input_level_(input_level),
processor_type_(processor_type),
num_channels_(num_channels) {
api_call_durations_.reserve(num_durations_to_store_);
}
// Implements the callback functionality for the threads.
bool Process();
// Method for printing out the simulation statistics.
void print_processor_statistics(const std::string& processor_name) const {
const std::string modifier = "_api_call_duration";
const std::string sample_rate_name =
"_" + std::to_string(simulation_config_->sample_rate_hz) + "Hz";
webrtc::test::PrintResultMeanAndError(
"apm_timing", sample_rate_name, processor_name, GetDurationAverage(),
GetDurationStandardDeviation(), "us", false);
if (kPrintAllDurations) {
webrtc::test::PrintResultList("apm_call_durations", sample_rate_name,
processor_name, api_call_durations_, "us",
false);
}
}
void AddDuration(int64_t duration) {
if (api_call_durations_.size() < num_durations_to_store_) {
api_call_durations_.push_back(duration);
}
}
private:
static const int kMaxCallDifference = 10;
static const int kMaxFrameSize = 480;
static const int kNumInitializationFrames = 5;
int64_t GetDurationStandardDeviation() const {
double variance = 0;
const int64_t average_duration = GetDurationAverage();
for (size_t k = kNumInitializationFrames; k < api_call_durations_.size();
k++) {
int64_t tmp = api_call_durations_[k] - average_duration;
variance += static_cast<double>(tmp * tmp);
}
const int denominator = rtc::checked_cast<int>(api_call_durations_.size()) -
kNumInitializationFrames;
return (denominator > 0
? rtc::checked_cast<int64_t>(sqrt(variance / denominator))
: -1);
}
int64_t GetDurationAverage() const {
int64_t average_duration = 0;
for (size_t k = kNumInitializationFrames; k < api_call_durations_.size();
k++) {
average_duration += api_call_durations_[k];
}
const int denominator = rtc::checked_cast<int>(api_call_durations_.size()) -
kNumInitializationFrames;
return (denominator > 0 ? average_duration / denominator : -1);
}
int ProcessCapture() {
// Set the stream delay.
apm_->set_stream_delay_ms(30);
// Call and time the specified capture side API processing method.
const int64_t start_time = clock_->TimeInMicroseconds();
const int result = apm_->ProcessStream(
&frame_data_.input_frame[0], frame_data_.input_stream_config,
frame_data_.output_stream_config, &frame_data_.output_frame[0]);
const int64_t end_time = clock_->TimeInMicroseconds();
frame_counters_->IncreaseCaptureCounter();
AddDuration(end_time - start_time);
if (first_process_call_) {
// Flag that the capture side has been called at least once
// (needed to ensure that a capture call has been done
// before the first render call is performed (implicitly
// required by the APM API).
capture_call_checker_->set_flag();
first_process_call_ = false;
}
return result;
}
bool ReadyToProcessCapture() {
return (frame_counters_->CaptureMinusRenderCounters() <=
kMaxCallDifference);
}
int ProcessRender() {
// Call and time the specified render side API processing method.
const int64_t start_time = clock_->TimeInMicroseconds();
const int result = apm_->ProcessReverseStream(
&frame_data_.input_frame[0], frame_data_.input_stream_config,
frame_data_.output_stream_config, &frame_data_.output_frame[0]);
const int64_t end_time = clock_->TimeInMicroseconds();
frame_counters_->IncreaseRenderCounter();
AddDuration(end_time - start_time);
return result;
}
bool ReadyToProcessRender() {
// Do not process until at least one capture call has been done.
// (implicitly required by the APM API).
if (first_process_call_ && !capture_call_checker_->get_flag()) {
return false;
}
// Ensure that the number of render and capture calls do not differ too
// much.
if (frame_counters_->RenderMinusCaptureCounters() > kMaxCallDifference) {
return false;
}
first_process_call_ = false;
return true;
}
void PrepareFrame() {
// Lambda function for populating a float multichannel audio frame
// with random data.
auto populate_audio_frame = [](float amplitude, size_t num_channels,
size_t samples_per_channel, Random* rand_gen,
float** frame) {
for (size_t ch = 0; ch < num_channels; ch++) {
for (size_t k = 0; k < samples_per_channel; k++) {
// Store random float number with a value between +-amplitude.
frame[ch][k] = amplitude * (2 * rand_gen->Rand<float>() - 1);
}
}
};
// Prepare the audio input data and metadata.
frame_data_.input_stream_config.set_sample_rate_hz(
simulation_config_->sample_rate_hz);
frame_data_.input_stream_config.set_num_channels(num_channels_);
frame_data_.input_stream_config.set_has_keyboard(false);
populate_audio_frame(input_level_, num_channels_,
(simulation_config_->sample_rate_hz *
AudioProcessing::kChunkSizeMs / 1000),
rand_gen_, &frame_data_.input_frame[0]);
// Prepare the float audio output data and metadata.
frame_data_.output_stream_config.set_sample_rate_hz(
simulation_config_->sample_rate_hz);
frame_data_.output_stream_config.set_num_channels(1);
frame_data_.output_stream_config.set_has_keyboard(false);
}
bool ReadyToProcess() {
switch (processor_type_) {
case ProcessorType::kRender:
return ReadyToProcessRender();
case ProcessorType::kCapture:
return ReadyToProcessCapture();
}
// Should not be reached, but the return statement is needed for the code to
// build successfully on Android.
RTC_NOTREACHED();
return false;
}
Random* rand_gen_ = nullptr;
FrameCounters* frame_counters_ = nullptr;
LockedFlag* capture_call_checker_ = nullptr;
CallSimulator* test_ = nullptr;
const SimulationConfig* const simulation_config_ = nullptr;
AudioProcessing* apm_ = nullptr;
AudioFrameData frame_data_;
webrtc::Clock* clock_;
const size_t num_durations_to_store_;
std::vector<double> api_call_durations_;
const float input_level_;
bool first_process_call_ = true;
const ProcessorType processor_type_;
const int num_channels_ = 1;
};
// Class for managing the test simulation.
class CallSimulator : public ::testing::TestWithParam<SimulationConfig> {
public:
CallSimulator()
: test_complete_(EventWrapper::Create()),
render_thread_(
new rtc::PlatformThread(RenderProcessorThreadFunc, this, "render")),
capture_thread_(new rtc::PlatformThread(CaptureProcessorThreadFunc,
this,
"capture")),
rand_gen_(42U),
simulation_config_(static_cast<SimulationConfig>(GetParam())) {}
// Run the call simulation with a timeout.
EventTypeWrapper Run() {
StartThreads();
EventTypeWrapper result = test_complete_->Wait(kTestTimeout);
StopThreads();
render_thread_state_->print_processor_statistics(
simulation_config_.SettingsDescription() + "_render");
capture_thread_state_->print_processor_statistics(
simulation_config_.SettingsDescription() + "_capture");
return result;
}
// Tests whether all the required render and capture side calls have been
// done.
bool MaybeEndTest() {
if (frame_counters_.BothCountersExceedeThreshold(kMinNumFramesToProcess)) {
test_complete_->Set();
return true;
}
return false;
}
private:
static const float kCaptureInputFloatLevel;
static const float kRenderInputFloatLevel;
static const int kMinNumFramesToProcess = 150;
static const int32_t kTestTimeout = 3 * 10 * kMinNumFramesToProcess;
// ::testing::TestWithParam<> implementation.
void TearDown() override { StopThreads(); }
// Stop all running threads.
void StopThreads() {
render_thread_->Stop();
capture_thread_->Stop();
}
// Simulator and APM setup.
void SetUp() override {
// Lambda function for setting the default APM runtime settings for desktop.
auto set_default_desktop_apm_runtime_settings = [](AudioProcessing* apm) {
ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(true));
ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
ASSERT_EQ(apm->kNoError,
apm->gain_control()->set_mode(GainControl::kAdaptiveDigital));
ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
ASSERT_EQ(apm->kNoError, apm->voice_detection()->Enable(true));
AudioProcessing::Config apm_config = apm->GetConfig();
apm_config.echo_canceller.enabled = true;
apm_config.echo_canceller.mobile_mode = false;
apm->ApplyConfig(apm_config);
};
// Lambda function for setting the default APM runtime settings for mobile.
auto set_default_mobile_apm_runtime_settings = [](AudioProcessing* apm) {
ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(true));
ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
ASSERT_EQ(apm->kNoError,
apm->gain_control()->set_mode(GainControl::kAdaptiveDigital));
ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
ASSERT_EQ(apm->kNoError, apm->voice_detection()->Enable(true));
AudioProcessing::Config apm_config = apm->GetConfig();
apm_config.echo_canceller.enabled = true;
apm_config.echo_canceller.mobile_mode = true;
apm->ApplyConfig(apm_config);
};
// Lambda function for turning off all of the APM runtime settings
// submodules.
auto turn_off_default_apm_runtime_settings = [](AudioProcessing* apm) {
ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(false));
ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(false));
ASSERT_EQ(apm->kNoError,
apm->gain_control()->set_mode(GainControl::kAdaptiveDigital));
ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(false));
ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(false));
ASSERT_EQ(apm->kNoError, apm->voice_detection()->Enable(false));
AudioProcessing::Config apm_config = apm->GetConfig();
apm_config.echo_canceller.enabled = false;
apm->ApplyConfig(apm_config);
};
// Lambda function for adding default desktop APM settings to a config.
auto add_default_desktop_config = [](Config* config) {
config->Set<ExtendedFilter>(new ExtendedFilter(true));
config->Set<DelayAgnostic>(new DelayAgnostic(true));
};
int num_capture_channels = 1;
switch (simulation_config_.simulation_settings) {
case SettingsType::kDefaultApmMobile: {
apm_.reset(AudioProcessingBuilder().Create());
ASSERT_TRUE(!!apm_);
set_default_mobile_apm_runtime_settings(apm_.get());
break;
}
case SettingsType::kDefaultApmDesktop: {
Config config;
add_default_desktop_config(&config);
apm_.reset(AudioProcessingBuilder().Create(config));
ASSERT_TRUE(!!apm_);
set_default_desktop_apm_runtime_settings(apm_.get());
apm_->SetExtraOptions(config);
break;
}
case SettingsType::kAllSubmodulesTurnedOff: {
apm_.reset(AudioProcessingBuilder().Create());
ASSERT_TRUE(!!apm_);
turn_off_default_apm_runtime_settings(apm_.get());
break;
}
case SettingsType::kDefaultApmDesktopWithoutDelayAgnostic: {
Config config;
config.Set<ExtendedFilter>(new ExtendedFilter(true));
config.Set<DelayAgnostic>(new DelayAgnostic(false));
apm_.reset(AudioProcessingBuilder().Create(config));
ASSERT_TRUE(!!apm_);
set_default_desktop_apm_runtime_settings(apm_.get());
apm_->SetExtraOptions(config);
break;
}
case SettingsType::kDefaultApmDesktopWithoutExtendedFilter: {
Config config;
config.Set<ExtendedFilter>(new ExtendedFilter(false));
config.Set<DelayAgnostic>(new DelayAgnostic(true));
apm_.reset(AudioProcessingBuilder().Create(config));
ASSERT_TRUE(!!apm_);
set_default_desktop_apm_runtime_settings(apm_.get());
apm_->SetExtraOptions(config);
break;
}
}
render_thread_state_.reset(new TimedThreadApiProcessor(
ProcessorType::kRender, &rand_gen_, &frame_counters_,
&capture_call_checker_, this, &simulation_config_, apm_.get(),
kMinNumFramesToProcess, kRenderInputFloatLevel, 1));
capture_thread_state_.reset(new TimedThreadApiProcessor(
ProcessorType::kCapture, &rand_gen_, &frame_counters_,
&capture_call_checker_, this, &simulation_config_, apm_.get(),
kMinNumFramesToProcess, kCaptureInputFloatLevel, num_capture_channels));
}
// Thread callback for the render thread.
static bool RenderProcessorThreadFunc(void* context) {
return reinterpret_cast<CallSimulator*>(context)
->render_thread_state_->Process();
}
// Thread callback for the capture thread.
static bool CaptureProcessorThreadFunc(void* context) {
return reinterpret_cast<CallSimulator*>(context)
->capture_thread_state_->Process();
}
// Start the threads used in the test.
void StartThreads() {
ASSERT_NO_FATAL_FAILURE(render_thread_->Start());
render_thread_->SetPriority(rtc::kRealtimePriority);
ASSERT_NO_FATAL_FAILURE(capture_thread_->Start());
capture_thread_->SetPriority(rtc::kRealtimePriority);
}
// Event handler for the test.
const std::unique_ptr<EventWrapper> test_complete_;
// Thread related variables.
std::unique_ptr<rtc::PlatformThread> render_thread_;
std::unique_ptr<rtc::PlatformThread> capture_thread_;
Random rand_gen_;
std::unique_ptr<AudioProcessing> apm_;
const SimulationConfig simulation_config_;
FrameCounters frame_counters_;
LockedFlag capture_call_checker_;
std::unique_ptr<TimedThreadApiProcessor> render_thread_state_;
std::unique_ptr<TimedThreadApiProcessor> capture_thread_state_;
};
// Implements the callback functionality for the threads.
bool TimedThreadApiProcessor::Process() {
PrepareFrame();
// Wait in a spinlock manner until it is ok to start processing.
// Note that SleepMs is not applicable since it only allows sleeping
// on a millisecond basis which is too long.
// TODO(tommi): This loop may affect the performance of the test that it's
// meant to measure. See if we could use events instead to signal readiness.
while (!ReadyToProcess()) {
}
int result = AudioProcessing::kNoError;
switch (processor_type_) {
case ProcessorType::kRender:
result = ProcessRender();
break;
case ProcessorType::kCapture:
result = ProcessCapture();
break;
}
EXPECT_EQ(result, AudioProcessing::kNoError);
return !test_->MaybeEndTest();
}
const float CallSimulator::kRenderInputFloatLevel = 0.5f;
const float CallSimulator::kCaptureInputFloatLevel = 0.03125f;
} // anonymous namespace
// TODO(peah): Reactivate once issue 7712 has been resolved.
TEST_P(CallSimulator, DISABLED_ApiCallDurationTest) {
// Run test and verify that it did not time out.
EXPECT_EQ(kEventSignaled, Run());
}
INSTANTIATE_TEST_CASE_P(
AudioProcessingPerformanceTest,
CallSimulator,
::testing::ValuesIn(SimulationConfig::GenerateSimulationConfigs()));
} // namespace webrtc