Created a test that reports the statistics for the duration of APM stream processing API calls.

webrtc/modules/audio_processing/audio_processing_performance_unittest.cc

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
new file mode 100644
index 0000000000000000000000000000000000000000..0c8c060ea3b82bc30a2c79be664f7fe67716c61e
--- /dev/null
+++ b/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc
@@ -0,0 +1,724 @@
+/*
+ *  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/random.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/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,
+                          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, 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;
+  }
+
+  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_(
+            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));
+      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(rtc::kRealtimePriority);
+    ASSERT_NO_FATAL_FAILURE(capture_thread_->Start());
+    capture_thread_->SetPriority(rtc::kRealtimePriority);
+  }
+
+  // Event handler for the test.
+  const rtc::scoped_ptr<EventWrapper> test_complete_;
+
+  // Thread related variables.
+  rtc::scoped_ptr<rtc::PlatformThread> render_thread_;
+  rtc::scoped_ptr<rtc::PlatformThread> capture_thread_;
+  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);
+
+  return !test_->MaybeEndTest();
+}
+
+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