| Index: webrtc/modules/audio_processing/audio_processing_performance_unittest.cc
|
| diff --git a/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc b/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc
|
| deleted file mode 100644
|
| index 9da3cd499be1c18e731b587ca095ce5320299e84..0000000000000000000000000000000000000000
|
| --- a/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc
|
| +++ /dev/null
|
| @@ -1,720 +0,0 @@
|
| -/*
|
| - * 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 "webrtc/modules/audio_processing/audio_processing_impl.h"
|
| -
|
| -#include <math.h>
|
| -
|
| -#include <algorithm>
|
| -#include <vector>
|
| -
|
| -#include "testing/gtest/include/gtest/gtest.h"
|
| -#include "webrtc/base/array_view.h"
|
| -#include "webrtc/base/criticalsection.h"
|
| -#include "webrtc/base/platform_thread.h"
|
| -#include "webrtc/base/safe_conversions.h"
|
| -#include "webrtc/config.h"
|
| -#include "webrtc/modules/audio_processing/test/test_utils.h"
|
| -#include "webrtc/modules/include/module_common_types.h"
|
| -#include "webrtc/system_wrappers/include/clock.h"
|
| -#include "webrtc/system_wrappers/include/event_wrapper.h"
|
| -#include "webrtc/system_wrappers/include/sleep.h"
|
| -#include "webrtc/test/random.h"
|
| -#include "webrtc/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,
|
| - kDefaultApmDesktopAndBeamformer,
|
| - kDefaultApmDesktopAndIntelligibilityEnhancer,
|
| - kAllSubmodulesTurnedOff,
|
| - kDefaultDesktopApmWithoutDelayAgnostic,
|
| - kDefaultDesktopApmWithoutExtendedFilter
|
| -};
|
| -
|
| -// 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::kDefaultDesktopApmWithoutDelayAgnostic,
|
| - SettingsType::kDefaultDesktopApmWithoutExtendedFilter};
|
| -
|
| - 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));
|
| - }
|
| - }
|
| -
|
| - const SettingsType intelligibility_enhancer_settings[] = {
|
| - SettingsType::kDefaultApmDesktopAndIntelligibilityEnhancer};
|
| -
|
| - const int intelligibility_enhancer_sample_rates[] = {8000, 16000, 32000,
|
| - 48000};
|
| -
|
| - for (auto sample_rate : intelligibility_enhancer_sample_rates) {
|
| - for (auto settings : intelligibility_enhancer_settings) {
|
| - simulation_configs.push_back(SimulationConfig(sample_rate, settings));
|
| - }
|
| - }
|
| -
|
| - const SettingsType beamformer_settings[] = {
|
| - SettingsType::kDefaultApmDesktopAndBeamformer};
|
| -
|
| - const int beamformer_sample_rates[] = {8000, 16000, 32000, 48000};
|
| -
|
| - for (auto sample_rate : beamformer_sample_rates) {
|
| - for (auto settings : beamformer_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::kDefaultApmDesktopAndBeamformer:
|
| - description = "DefaultApmDesktopAndBeamformer";
|
| - break;
|
| - case SettingsType::kDefaultApmDesktopAndIntelligibilityEnhancer:
|
| - description = "DefaultApmDesktopAndIntelligibilityEnhancer";
|
| - break;
|
| - case SettingsType::kAllSubmodulesTurnedOff:
|
| - description = "AllSubmodulesOff";
|
| - break;
|
| - case SettingsType::kDefaultDesktopApmWithoutDelayAgnostic:
|
| - description = "DefaultDesktopApmWithoutDelayAgnostic";
|
| - break;
|
| - case SettingsType::kDefaultDesktopApmWithoutExtendedFilter:
|
| - description = "DefaultDesktopApmWithoutExtendedFilter";
|
| - break;
|
| - }
|
| - return description;
|
| - }
|
| -
|
| - int sample_rate_hz = 16000;
|
| - SettingsType simulation_settings = SettingsType::kDefaultApmDesktop;
|
| -};
|
| -
|
| -// Handler for the frame counters.
|
| -class FrameCounters {
|
| - public:
|
| - void IncreaseRenderCounter() {
|
| - rtc::CritScope cs(&crit_);
|
| - render_count_++;
|
| - }
|
| -
|
| - void IncreaseCaptureCounter() {
|
| - rtc::CritScope cs(&crit_);
|
| - capture_count_++;
|
| - }
|
| -
|
| - int GetCaptureCounter() const {
|
| - rtc::CritScope cs(&crit_);
|
| - return capture_count_;
|
| - }
|
| -
|
| - int GetRenderCounter() const {
|
| - rtc::CritScope cs(&crit_);
|
| - return render_count_;
|
| - }
|
| -
|
| - int CaptureMinusRenderCounters() const {
|
| - rtc::CritScope cs(&crit_);
|
| - return capture_count_ - render_count_;
|
| - }
|
| -
|
| - int RenderMinusCaptureCounters() const {
|
| - return -CaptureMinusRenderCounters();
|
| - }
|
| -
|
| - bool BothCountersExceedeThreshold(int threshold) const {
|
| - rtc::CritScope cs(&crit_);
|
| - return (render_count_ > threshold && capture_count_ > threshold);
|
| - }
|
| -
|
| - private:
|
| - mutable rtc::CriticalSection crit_;
|
| - int render_count_ GUARDED_BY(crit_) = 0;
|
| - int capture_count_ GUARDED_BY(crit_) = 0;
|
| -};
|
| -
|
| -// Class that protects a flag using a lock.
|
| -class LockedFlag {
|
| - public:
|
| - bool get_flag() const {
|
| - rtc::CritScope cs(&crit_);
|
| - return flag_;
|
| - }
|
| -
|
| - void set_flag() {
|
| - rtc::CritScope cs(&crit_);
|
| - flag_ = true;
|
| - }
|
| -
|
| - private:
|
| - mutable rtc::CriticalSection crit_;
|
| - bool flag_ GUARDED_BY(crit_) = false;
|
| -};
|
| -
|
| -// Parent class for the thread processors.
|
| -class TimedThreadApiProcessor {
|
| - public:
|
| - TimedThreadApiProcessor(ProcessorType processor_type,
|
| - test::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(std::string processor_name) const {
|
| - const std::string modifier = "_api_call_duration";
|
| -
|
| - // Lambda function for creating a test printout string.
|
| - auto create_mean_and_std_string = [](int64_t average,
|
| - int64_t standard_dev) {
|
| - std::string s = std::to_string(average);
|
| - s += ", ";
|
| - s += std::to_string(standard_dev);
|
| - return s;
|
| - };
|
| -
|
| - 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,
|
| - create_mean_and_std_string(GetDurationAverage(),
|
| - GetDurationStandardDeviation()),
|
| - "us", false);
|
| -
|
| - if (kPrintAllDurations) {
|
| - std::string value_string = "";
|
| - for (int64_t duration : api_call_durations_) {
|
| - value_string += std::to_string(duration) + ",";
|
| - }
|
| - webrtc::test::PrintResultList("apm_call_durations", sample_rate_name,
|
| - processor_name, value_string, "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,
|
| - test::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();
|
| - break;
|
| - case ProcessorType::kCapture:
|
| - return ReadyToProcessCapture();
|
| - break;
|
| - }
|
| -
|
| - // Should not be reached, but the return statement is needed for the code to
|
| - // build successfully on Android.
|
| - RTC_NOTREACHED();
|
| - return false;
|
| - }
|
| -
|
| - test::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<int64_t> 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_(PlatformThread::CreateThread(RenderProcessorThreadFunc,
|
| - this,
|
| - "render")),
|
| - capture_thread_(PlatformThread::CreateThread(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);
|
| -
|
| - 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.
|
| - void MaybeEndTest() {
|
| - if (frame_counters_.BothCountersExceedeThreshold(kMinNumFramesToProcess)) {
|
| - test_complete_->Set();
|
| - }
|
| - }
|
| -
|
| - 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 {
|
| - 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));
|
| - ASSERT_EQ(apm->kNoError, apm->echo_control_mobile()->Enable(false));
|
| - ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
|
| - ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->enable_metrics(true));
|
| - ASSERT_EQ(apm->kNoError,
|
| - apm->echo_cancellation()->enable_delay_logging(true));
|
| - };
|
| -
|
| - // 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));
|
| - ASSERT_EQ(apm->kNoError, apm->echo_control_mobile()->Enable(true));
|
| - ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(false));
|
| - };
|
| -
|
| - // 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));
|
| - ASSERT_EQ(apm->kNoError, apm->echo_control_mobile()->Enable(false));
|
| - ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(false));
|
| - ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->enable_metrics(false));
|
| - ASSERT_EQ(apm->kNoError,
|
| - apm->echo_cancellation()->enable_delay_logging(false));
|
| - };
|
| -
|
| - // 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));
|
| - };
|
| -
|
| - // Lambda function for adding beamformer settings to a config.
|
| - auto add_beamformer_config = [](Config* config) {
|
| - const size_t num_mics = 2;
|
| - const std::vector<Point> array_geometry =
|
| - ParseArrayGeometry("0 0 0 0.05 0 0", num_mics);
|
| - RTC_CHECK_EQ(array_geometry.size(), num_mics);
|
| -
|
| - config->Set<Beamforming>(
|
| - new Beamforming(true, array_geometry,
|
| - SphericalPointf(DegreesToRadians(90), 0.f, 1.f)));
|
| - };
|
| -
|
| - int num_capture_channels = 1;
|
| - switch (simulation_config_.simulation_settings) {
|
| - case SettingsType::kDefaultApmMobile: {
|
| - apm_.reset(AudioProcessingImpl::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(AudioProcessingImpl::Create(config));
|
| - ASSERT_TRUE(!!apm_);
|
| - set_default_desktop_apm_runtime_settings(apm_.get());
|
| - apm_->SetExtraOptions(config);
|
| - break;
|
| - }
|
| - case SettingsType::kDefaultApmDesktopAndBeamformer: {
|
| - Config config;
|
| - add_beamformer_config(&config);
|
| - add_default_desktop_config(&config);
|
| - apm_.reset(AudioProcessingImpl::Create(config));
|
| - ASSERT_TRUE(!!apm_);
|
| - set_default_desktop_apm_runtime_settings(apm_.get());
|
| - apm_->SetExtraOptions(config);
|
| - num_capture_channels = 2;
|
| - break;
|
| - }
|
| - case SettingsType::kDefaultApmDesktopAndIntelligibilityEnhancer: {
|
| - Config config;
|
| - config.Set<Intelligibility>(new Intelligibility(true));
|
| - add_default_desktop_config(&config);
|
| - apm_.reset(AudioProcessingImpl::Create(config));
|
| - ASSERT_TRUE(!!apm_);
|
| - set_default_desktop_apm_runtime_settings(apm_.get());
|
| - apm_->SetExtraOptions(config);
|
| - break;
|
| - }
|
| - case SettingsType::kAllSubmodulesTurnedOff: {
|
| - apm_.reset(AudioProcessingImpl::Create());
|
| - ASSERT_TRUE(!!apm_);
|
| - turn_off_default_apm_runtime_settings(apm_.get());
|
| - break;
|
| - }
|
| - case SettingsType::kDefaultDesktopApmWithoutDelayAgnostic: {
|
| - Config config;
|
| - config.Set<ExtendedFilter>(new ExtendedFilter(true));
|
| - config.Set<DelayAgnostic>(new DelayAgnostic(false));
|
| - apm_.reset(AudioProcessingImpl::Create(config));
|
| - ASSERT_TRUE(!!apm_);
|
| - set_default_desktop_apm_runtime_settings(apm_.get());
|
| - apm_->SetExtraOptions(config);
|
| - break;
|
| - }
|
| - case SettingsType::kDefaultDesktopApmWithoutExtendedFilter: {
|
| - Config config;
|
| - config.Set<ExtendedFilter>(new ExtendedFilter(false));
|
| - config.Set<DelayAgnostic>(new DelayAgnostic(true));
|
| - apm_.reset(AudioProcessingImpl::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(kRealtimePriority);
|
| - ASSERT_NO_FATAL_FAILURE(capture_thread_->Start());
|
| - capture_thread_->SetPriority(kRealtimePriority);
|
| - }
|
| -
|
| - // Event handler for the test.
|
| - const rtc::scoped_ptr<EventWrapper> test_complete_;
|
| -
|
| - // Thread related variables.
|
| - rtc::scoped_ptr<PlatformThread> render_thread_;
|
| - rtc::scoped_ptr<PlatformThread> capture_thread_;
|
| - test::Random rand_gen_;
|
| -
|
| - rtc::scoped_ptr<AudioProcessing> apm_;
|
| - const SimulationConfig simulation_config_;
|
| - FrameCounters frame_counters_;
|
| - LockedFlag capture_call_checker_;
|
| - rtc::scoped_ptr<TimedThreadApiProcessor> render_thread_state_;
|
| - rtc::scoped_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.
|
| - 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);
|
| -
|
| - test_->MaybeEndTest();
|
| -
|
| - return true;
|
| -}
|
| -
|
| -const float CallSimulator::kRenderInputFloatLevel = 0.5f;
|
| -const float CallSimulator::kCaptureInputFloatLevel = 0.03125f;
|
| -} // anonymous namespace
|
| -
|
| -TEST_P(CallSimulator, 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
|
|
|