modules/audio_processing/audio_processing_performance_unittest.cc - src/ - Git at Google

 /*
  *  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 <math.h>

 #include <algorithm>
 #include <memory>
 #include <vector>

 #include "api/array_view.h"
 #include "modules/audio_processing/audio_processing_impl.h"
 #include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
 #include "modules/audio_processing/test/test_utils.h"
 #include "rtc_base/atomic_ops.h"
 #include "rtc_base/event.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 "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_);
     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);
   }

   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_DCHECK_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()
       : rand_gen_(42U),
         simulation_config_(static_cast<SimulationConfig>(GetParam())) {}

   // Run the call simulation with a timeout.
   bool Run() {
     StartThreads();

     bool 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;

   // Stop all running threads.
   void StopThreads() {
     render_thread_.Finalize();
     capture_thread_.Finalize();
   }

   // 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) {
       AudioProcessing::Config apm_config = apm->GetConfig();
       apm_config.echo_canceller.enabled = true;
       apm_config.echo_canceller.mobile_mode = false;
       apm_config.noise_suppression.enabled = true;
       apm_config.gain_controller1.enabled = true;
       apm_config.gain_controller1.mode =
           AudioProcessing::Config::GainController1::kAdaptiveDigital;
       apm->ApplyConfig(apm_config);
     };

     // Lambda function for setting the default APM runtime settings for mobile.
     auto set_default_mobile_apm_runtime_settings = [](AudioProcessing* apm) {
       AudioProcessing::Config apm_config = apm->GetConfig();
       apm_config.echo_canceller.enabled = true;
       apm_config.echo_canceller.mobile_mode = true;
       apm_config.noise_suppression.enabled = true;
       apm_config.gain_controller1.mode =
           AudioProcessing::Config::GainController1::kAdaptiveDigital;
       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) {
       AudioProcessing::Config apm_config = apm->GetConfig();
       apm_config.echo_canceller.enabled = false;
       apm_config.gain_controller1.enabled = false;
       apm_config.noise_suppression.enabled = false;
       apm->ApplyConfig(apm_config);
     };

     int num_capture_channels = 1;
     switch (simulation_config_.simulation_settings) {
       case SettingsType::kDefaultApmMobile: {
         apm_ = AudioProcessingBuilderForTesting().Create();
         ASSERT_TRUE(!!apm_);
         set_default_mobile_apm_runtime_settings(apm_.get());
         break;
       }
       case SettingsType::kDefaultApmDesktop: {
         apm_ = AudioProcessingBuilderForTesting().Create();
         ASSERT_TRUE(!!apm_);
         set_default_desktop_apm_runtime_settings(apm_.get());
         break;
       }
       case SettingsType::kAllSubmodulesTurnedOff: {
         apm_ = AudioProcessingBuilderForTesting().Create();
         ASSERT_TRUE(!!apm_);
         turn_off_default_apm_runtime_settings(apm_.get());
         break;
       }
       case SettingsType::kDefaultApmDesktopWithoutDelayAgnostic: {
         apm_ = AudioProcessingBuilderForTesting().Create();
         ASSERT_TRUE(!!apm_);
         set_default_desktop_apm_runtime_settings(apm_.get());
         break;
       }
       case SettingsType::kDefaultApmDesktopWithoutExtendedFilter: {
         apm_ = AudioProcessingBuilderForTesting().Create();
         ASSERT_TRUE(!!apm_);
         set_default_desktop_apm_runtime_settings(apm_.get());
         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));
   }

   // Start the threads used in the test.
   void StartThreads() {
     const auto attributes =
         rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime);
     render_thread_ = rtc::PlatformThread::SpawnJoinable(
         [this] {
           while (render_thread_state_->Process()) {
           }
         },
         "render", attributes);
     capture_thread_ = rtc::PlatformThread::SpawnJoinable(
         [this] {
           while (capture_thread_state_->Process()) {
           }
         },
         "capture", attributes);
   }

   // Event handler for the test.
   rtc::Event test_complete_;

   // Thread related variables.
   Random rand_gen_;

   rtc::scoped_refptr<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_;
   rtc::PlatformThread render_thread_;
   rtc::PlatformThread capture_thread_;
 };

 // 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_TRUE(Run());
 }

 INSTANTIATE_TEST_SUITE_P(
     AudioProcessingPerformanceTest,
     CallSimulator,
     ::testing::ValuesIn(SimulationConfig::GenerateSimulationConfigs()));

 }  // namespace webrtc
	/*
	* 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 <math.h>

	#include <algorithm>
	#include <memory>
	#include <vector>

	#include "api/array_view.h"
	#include "modules/audio_processing/audio_processing_impl.h"
	#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
	#include "modules/audio_processing/test/test_utils.h"
	#include "rtc_base/atomic_ops.h"
	#include "rtc_base/event.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 "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_);
	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);
	}

	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_DCHECK_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()
	: rand_gen_(42U),
	simulation_config_(static_cast<SimulationConfig>(GetParam())) {}

	// Run the call simulation with a timeout.
	bool Run() {
	StartThreads();

	bool 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;

	// Stop all running threads.
	void StopThreads() {
	render_thread_.Finalize();
	capture_thread_.Finalize();
	}

	// 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) {
	AudioProcessing::Config apm_config = apm->GetConfig();
	apm_config.echo_canceller.enabled = true;
	apm_config.echo_canceller.mobile_mode = false;
	apm_config.noise_suppression.enabled = true;
	apm_config.gain_controller1.enabled = true;
	apm_config.gain_controller1.mode =
	AudioProcessing::Config::GainController1::kAdaptiveDigital;
	apm->ApplyConfig(apm_config);
	};

	// Lambda function for setting the default APM runtime settings for mobile.
	auto set_default_mobile_apm_runtime_settings = [](AudioProcessing* apm) {
	AudioProcessing::Config apm_config = apm->GetConfig();
	apm_config.echo_canceller.enabled = true;
	apm_config.echo_canceller.mobile_mode = true;
	apm_config.noise_suppression.enabled = true;
	apm_config.gain_controller1.mode =
	AudioProcessing::Config::GainController1::kAdaptiveDigital;
	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) {
	AudioProcessing::Config apm_config = apm->GetConfig();
	apm_config.echo_canceller.enabled = false;
	apm_config.gain_controller1.enabled = false;
	apm_config.noise_suppression.enabled = false;
	apm->ApplyConfig(apm_config);
	};

	int num_capture_channels = 1;
	switch (simulation_config_.simulation_settings) {
	case SettingsType::kDefaultApmMobile: {
	apm_ = AudioProcessingBuilderForTesting().Create();
	ASSERT_TRUE(!!apm_);
	set_default_mobile_apm_runtime_settings(apm_.get());
	break;
	}
	case SettingsType::kDefaultApmDesktop: {
	apm_ = AudioProcessingBuilderForTesting().Create();
	ASSERT_TRUE(!!apm_);
	set_default_desktop_apm_runtime_settings(apm_.get());
	break;
	}
	case SettingsType::kAllSubmodulesTurnedOff: {
	apm_ = AudioProcessingBuilderForTesting().Create();
	ASSERT_TRUE(!!apm_);
	turn_off_default_apm_runtime_settings(apm_.get());
	break;
	}
	case SettingsType::kDefaultApmDesktopWithoutDelayAgnostic: {
	apm_ = AudioProcessingBuilderForTesting().Create();
	ASSERT_TRUE(!!apm_);
	set_default_desktop_apm_runtime_settings(apm_.get());
	break;
	}
	case SettingsType::kDefaultApmDesktopWithoutExtendedFilter: {
	apm_ = AudioProcessingBuilderForTesting().Create();
	ASSERT_TRUE(!!apm_);
	set_default_desktop_apm_runtime_settings(apm_.get());
	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));
	}

	// Start the threads used in the test.
	void StartThreads() {
	const auto attributes =
	rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime);
	render_thread_ = rtc::PlatformThread::SpawnJoinable(
	[this] {
	while (render_thread_state_->Process()) {
	}
	},
	"render", attributes);
	capture_thread_ = rtc::PlatformThread::SpawnJoinable(
	[this] {
	while (capture_thread_state_->Process()) {
	}
	},
	"capture", attributes);
	}

	// Event handler for the test.
	rtc::Event test_complete_;

	// Thread related variables.
	Random rand_gen_;

	rtc::scoped_refptr<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_;
	rtc::PlatformThread render_thread_;
	rtc::PlatformThread capture_thread_;
	};

	// 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_TRUE(Run());
	}

	INSTANTIATE_TEST_SUITE_P(
	AudioProcessingPerformanceTest,
	CallSimulator,
	::testing::ValuesIn(SimulationConfig::GenerateSimulationConfigs()));

	} // namespace webrtc