modules/audio_processing/level_controller/level_controller_complexity_unittest.cc - src/webrtc - Git at Google

 /*
  *  Copyright (c) 2016 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 <numeric>
 #include <vector>

 #include "webrtc/base/array_view.h"
 #include "webrtc/base/random.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/level_controller/level_controller.h"
 #include "webrtc/modules/audio_processing/test/audio_buffer_tools.h"
 #include "webrtc/modules/audio_processing/test/bitexactness_tools.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/perf_test.h"

 namespace webrtc {
 namespace {

 const size_t kNumFramesToProcess = 100;

 struct SimulatorBuffers {
   SimulatorBuffers(int render_input_sample_rate_hz,
                    int capture_input_sample_rate_hz,
                    int render_output_sample_rate_hz,
                    int capture_output_sample_rate_hz,
                    size_t num_render_input_channels,
                    size_t num_capture_input_channels,
                    size_t num_render_output_channels,
                    size_t num_capture_output_channels) {
     Random rand_gen(42);
     CreateConfigAndBuffer(render_input_sample_rate_hz,
                           num_render_input_channels, &rand_gen,
                           &render_input_buffer, &render_input_config,
                           &render_input, &render_input_samples);

     CreateConfigAndBuffer(render_output_sample_rate_hz,
                           num_render_output_channels, &rand_gen,
                           &render_output_buffer, &render_output_config,
                           &render_output, &render_output_samples);

     CreateConfigAndBuffer(capture_input_sample_rate_hz,
                           num_capture_input_channels, &rand_gen,
                           &capture_input_buffer, &capture_input_config,
                           &capture_input, &capture_input_samples);

     CreateConfigAndBuffer(capture_output_sample_rate_hz,
                           num_capture_output_channels, &rand_gen,
                           &capture_output_buffer, &capture_output_config,
                           &capture_output, &capture_output_samples);

     UpdateInputBuffers();
   }

   void CreateConfigAndBuffer(int sample_rate_hz,
                              size_t num_channels,
                              Random* rand_gen,
                              std::unique_ptr<AudioBuffer>* buffer,
                              StreamConfig* config,
                              std::vector<float*>* buffer_data,
                              std::vector<float>* buffer_data_samples) {
     int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
     *config = StreamConfig(sample_rate_hz, num_channels, false);
     buffer->reset(new AudioBuffer(config->num_frames(), config->num_channels(),
                                   config->num_frames(), config->num_channels(),
                                   config->num_frames()));

     buffer_data_samples->resize(samples_per_channel * num_channels);
     for (auto& v : *buffer_data_samples) {
       v = rand_gen->Rand<float>();
     }

     buffer_data->resize(num_channels);
     for (size_t ch = 0; ch < num_channels; ++ch) {
       (*buffer_data)[ch] = &(*buffer_data_samples)[ch * samples_per_channel];
     }
   }

   void UpdateInputBuffers() {
     test::CopyVectorToAudioBuffer(capture_input_config, capture_input_samples,
                                   capture_input_buffer.get());
     test::CopyVectorToAudioBuffer(render_input_config, render_input_samples,
                                   render_input_buffer.get());
   }

   std::unique_ptr<AudioBuffer> render_input_buffer;
   std::unique_ptr<AudioBuffer> capture_input_buffer;
   std::unique_ptr<AudioBuffer> render_output_buffer;
   std::unique_ptr<AudioBuffer> capture_output_buffer;
   StreamConfig render_input_config;
   StreamConfig capture_input_config;
   StreamConfig render_output_config;
   StreamConfig capture_output_config;
   std::vector<float*> render_input;
   std::vector<float> render_input_samples;
   std::vector<float*> capture_input;
   std::vector<float> capture_input_samples;
   std::vector<float*> render_output;
   std::vector<float> render_output_samples;
   std::vector<float*> capture_output;
   std::vector<float> capture_output_samples;
 };

 class SubmodulePerformanceTimer {
  public:
   SubmodulePerformanceTimer() : clock_(webrtc::Clock::GetRealTimeClock()) {
     timestamps_us_.reserve(kNumFramesToProcess);
   }

   void StartTimer() {
     start_timestamp_us_ = rtc::Optional<int64_t>(clock_->TimeInMicroseconds());
   }
   void StopTimer() {
     RTC_DCHECK(start_timestamp_us_);
     timestamps_us_.push_back(clock_->TimeInMicroseconds() -
                              *start_timestamp_us_);
   }

   double GetDurationAverage() const {
     RTC_DCHECK(!timestamps_us_.empty());
     return static_cast<double>(std::accumulate(timestamps_us_.begin(),
                                                timestamps_us_.end(), 0)) /
            timestamps_us_.size();
   }

   double GetDurationStandardDeviation() const {
     RTC_DCHECK(!timestamps_us_.empty());
     double average_duration = GetDurationAverage();

     int64_t variance =
         std::accumulate(timestamps_us_.begin(), timestamps_us_.end(), 0,
                         [average_duration](const int64_t& a, const int64_t& b) {
                           return a + (b - average_duration);
                         });

     return sqrt(variance / timestamps_us_.size());
   }

  private:
   webrtc::Clock* clock_;
   rtc::Optional<int64_t> start_timestamp_us_;
   std::vector<int64_t> timestamps_us_;
 };

 std::string FormPerformanceMeasureString(
     const SubmodulePerformanceTimer& timer) {
   std::string s = std::to_string(timer.GetDurationAverage());
   s += ", ";
   s += std::to_string(timer.GetDurationStandardDeviation());
   return s;
 }

 void RunStandaloneSubmodule(int sample_rate_hz, size_t num_channels) {
   SimulatorBuffers buffers(sample_rate_hz, sample_rate_hz, sample_rate_hz,
                            sample_rate_hz, num_channels, num_channels,
                            num_channels, num_channels);
   SubmodulePerformanceTimer timer;

   LevelController level_controller;
   level_controller.Initialize(sample_rate_hz);

   for (size_t frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
     buffers.UpdateInputBuffers();

     timer.StartTimer();
     level_controller.Process(buffers.capture_input_buffer.get());
     timer.StopTimer();
   }
   webrtc::test::PrintResultMeanAndError(
       "level_controller_call_durations",
       "_" + std::to_string(sample_rate_hz) + "Hz_" +
           std::to_string(num_channels) + "_channels",
       "StandaloneLevelControl", FormPerformanceMeasureString(timer), "us",
       false);
 }

 void RunTogetherWithApm(std::string test_description,
                         int render_input_sample_rate_hz,
                         int render_output_sample_rate_hz,
                         int capture_input_sample_rate_hz,
                         int capture_output_sample_rate_hz,
                         size_t num_channels,
                         bool use_mobile_aec,
                         bool include_default_apm_processing) {
   SimulatorBuffers buffers(
       render_input_sample_rate_hz, capture_input_sample_rate_hz,
       render_output_sample_rate_hz, capture_output_sample_rate_hz, num_channels,
       num_channels, num_channels, num_channels);
   SubmodulePerformanceTimer render_timer;
   SubmodulePerformanceTimer capture_timer;
   SubmodulePerformanceTimer total_timer;

   webrtc::Config config;
   AudioProcessing::Config apm_config;
   if (include_default_apm_processing) {
     config.Set<DelayAgnostic>(new DelayAgnostic(true));
     config.Set<ExtendedFilter>(new ExtendedFilter(true));
   }
   apm_config.level_controller.enabled = true;

   std::unique_ptr<AudioProcessing> apm;
   apm.reset(AudioProcessing::Create(config));
   ASSERT_TRUE(apm.get());
   apm->ApplyConfig(apm_config);

   ASSERT_EQ(AudioProcessing::kNoError,
             apm->gain_control()->Enable(include_default_apm_processing));
   if (use_mobile_aec) {
     ASSERT_EQ(AudioProcessing::kNoError,
               apm->echo_cancellation()->Enable(false));
     ASSERT_EQ(AudioProcessing::kNoError, apm->echo_control_mobile()->Enable(
                                              include_default_apm_processing));
   } else {
     ASSERT_EQ(AudioProcessing::kNoError,
               apm->echo_cancellation()->Enable(include_default_apm_processing));
     ASSERT_EQ(AudioProcessing::kNoError,
               apm->echo_control_mobile()->Enable(false));
   }
   ASSERT_EQ(AudioProcessing::kNoError,
             apm->high_pass_filter()->Enable(include_default_apm_processing));
   ASSERT_EQ(AudioProcessing::kNoError,
             apm->noise_suppression()->Enable(include_default_apm_processing));
   ASSERT_EQ(AudioProcessing::kNoError,
             apm->voice_detection()->Enable(include_default_apm_processing));
   ASSERT_EQ(AudioProcessing::kNoError,
             apm->level_estimator()->Enable(include_default_apm_processing));

   StreamConfig render_input_config(render_input_sample_rate_hz, num_channels,
                                    false);
   StreamConfig render_output_config(render_output_sample_rate_hz, num_channels,
                                     false);
   StreamConfig capture_input_config(capture_input_sample_rate_hz, num_channels,
                                     false);
   StreamConfig capture_output_config(capture_output_sample_rate_hz,
                                      num_channels, false);

   for (size_t frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
     buffers.UpdateInputBuffers();

     total_timer.StartTimer();
     render_timer.StartTimer();
     ASSERT_EQ(AudioProcessing::kNoError,
               apm->ProcessReverseStream(
                   &buffers.render_input[0], render_input_config,
                   render_output_config, &buffers.render_output[0]));

     render_timer.StopTimer();

     capture_timer.StartTimer();
     ASSERT_EQ(AudioProcessing::kNoError, apm->set_stream_delay_ms(0));
     ASSERT_EQ(
         AudioProcessing::kNoError,
         apm->ProcessStream(&buffers.capture_input[0], capture_input_config,
                            capture_output_config, &buffers.capture_output[0]));

     capture_timer.StopTimer();
     total_timer.StopTimer();
   }

   webrtc::test::PrintResultMeanAndError(
       "level_controller_call_durations",
       "_" + std::to_string(render_input_sample_rate_hz) + "_" +
           std::to_string(render_output_sample_rate_hz) + "_" +
           std::to_string(capture_input_sample_rate_hz) + "_" +
           std::to_string(capture_output_sample_rate_hz) + "Hz_" +
           std::to_string(num_channels) + "_channels" + "_render",
       test_description, FormPerformanceMeasureString(render_timer), "us",
       false);
   webrtc::test::PrintResultMeanAndError(
       "level_controller_call_durations",
       "_" + std::to_string(render_input_sample_rate_hz) + "_" +
           std::to_string(render_output_sample_rate_hz) + "_" +
           std::to_string(capture_input_sample_rate_hz) + "_" +
           std::to_string(capture_output_sample_rate_hz) + "Hz_" +
           std::to_string(num_channels) + "_channels" + "_capture",
       test_description, FormPerformanceMeasureString(capture_timer), "us",
       false);
   webrtc::test::PrintResultMeanAndError(
       "level_controller_call_durations",
       "_" + std::to_string(render_input_sample_rate_hz) + "_" +
           std::to_string(render_output_sample_rate_hz) + "_" +
           std::to_string(capture_input_sample_rate_hz) + "_" +
           std::to_string(capture_output_sample_rate_hz) + "Hz_" +
           std::to_string(num_channels) + "_channels" + "_total",
       test_description, FormPerformanceMeasureString(total_timer), "us", false);
 }

 }  // namespace

 TEST(LevelControllerPerformanceTest, StandaloneProcessing) {
   int sample_rates_to_test[] = {
       AudioProcessing::kSampleRate8kHz, AudioProcessing::kSampleRate16kHz,
       AudioProcessing::kSampleRate32kHz, AudioProcessing::kSampleRate48kHz};
   for (auto sample_rate : sample_rates_to_test) {
     for (size_t num_channels = 1; num_channels <= 2; ++num_channels) {
       RunStandaloneSubmodule(sample_rate, num_channels);
     }
   }
 }

 #if !defined(WEBRTC_ANDROID)
 TEST(LevelControllerPerformanceTest, ProcessingViaApm) {
 #else
 TEST(LevelControllerPerformanceTest, DISABLED_ProcessingViaApm) {
 #endif
   int sample_rates_to_test[] = {AudioProcessing::kSampleRate8kHz,
                                 AudioProcessing::kSampleRate16kHz,
                                 AudioProcessing::kSampleRate32kHz,
                                 AudioProcessing::kSampleRate48kHz, 44100};
   for (auto capture_input_sample_rate_hz : sample_rates_to_test) {
     for (auto capture_output_sample_rate_hz : sample_rates_to_test) {
       for (size_t num_channels = 1; num_channels <= 2; ++num_channels) {
         RunTogetherWithApm("SimpleLevelControlViaApm", 48000, 48000,
                            capture_input_sample_rate_hz,
                            capture_output_sample_rate_hz, num_channels, false,
                            false);
       }
     }
   }
 }

 #if !defined(WEBRTC_ANDROID)
 TEST(LevelControllerPerformanceTest, InteractionWithDefaultApm) {
 #else
 TEST(LevelControllerPerformanceTest, DISABLED_InteractionWithDefaultApm) {
 #endif
   int sample_rates_to_test[] = {AudioProcessing::kSampleRate8kHz,
                                 AudioProcessing::kSampleRate16kHz,
                                 AudioProcessing::kSampleRate32kHz,
                                 AudioProcessing::kSampleRate48kHz, 44100};
   for (auto capture_input_sample_rate_hz : sample_rates_to_test) {
     for (auto capture_output_sample_rate_hz : sample_rates_to_test) {
       for (size_t num_channels = 1; num_channels <= 2; ++num_channels) {
         RunTogetherWithApm("LevelControlAndDefaultDesktopApm", 48000, 48000,
                            capture_input_sample_rate_hz,
                            capture_output_sample_rate_hz, num_channels, false,
                            true);
         RunTogetherWithApm("LevelControlAndDefaultMobileApm", 48000, 48000,
                            capture_input_sample_rate_hz,
                            capture_output_sample_rate_hz, num_channels, true,
                            true);
       }
     }
   }
 }

 }  // namespace webrtc
	/*
	* Copyright (c) 2016 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 <numeric>
	#include <vector>

	#include "webrtc/base/array_view.h"
	#include "webrtc/base/random.h"
	#include "webrtc/modules/audio_processing/audio_buffer.h"
	#include "webrtc/modules/audio_processing/include/audio_processing.h"
	#include "webrtc/modules/audio_processing/level_controller/level_controller.h"
	#include "webrtc/modules/audio_processing/test/audio_buffer_tools.h"
	#include "webrtc/modules/audio_processing/test/bitexactness_tools.h"
	#include "webrtc/system_wrappers/include/clock.h"
	#include "webrtc/test/gtest.h"
	#include "webrtc/test/testsupport/perf_test.h"

	namespace webrtc {
	namespace {

	const size_t kNumFramesToProcess = 100;

	struct SimulatorBuffers {
	SimulatorBuffers(int render_input_sample_rate_hz,
	int capture_input_sample_rate_hz,
	int render_output_sample_rate_hz,
	int capture_output_sample_rate_hz,
	size_t num_render_input_channels,
	size_t num_capture_input_channels,
	size_t num_render_output_channels,
	size_t num_capture_output_channels) {
	Random rand_gen(42);
	CreateConfigAndBuffer(render_input_sample_rate_hz,
	num_render_input_channels, &rand_gen,
	&render_input_buffer, &render_input_config,
	&render_input, &render_input_samples);

	CreateConfigAndBuffer(render_output_sample_rate_hz,
	num_render_output_channels, &rand_gen,
	&render_output_buffer, &render_output_config,
	&render_output, &render_output_samples);

	CreateConfigAndBuffer(capture_input_sample_rate_hz,
	num_capture_input_channels, &rand_gen,
	&capture_input_buffer, &capture_input_config,
	&capture_input, &capture_input_samples);

	CreateConfigAndBuffer(capture_output_sample_rate_hz,
	num_capture_output_channels, &rand_gen,
	&capture_output_buffer, &capture_output_config,
	&capture_output, &capture_output_samples);

	UpdateInputBuffers();
	}

	void CreateConfigAndBuffer(int sample_rate_hz,
	size_t num_channels,
	Random* rand_gen,
	std::unique_ptr<AudioBuffer>* buffer,
	StreamConfig* config,
	std::vector<float> buffer_data,
	std::vector<float>* buffer_data_samples) {
	int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
	*config = StreamConfig(sample_rate_hz, num_channels, false);
	buffer->reset(new AudioBuffer(config->num_frames(), config->num_channels(),
	config->num_frames(), config->num_channels(),
	config->num_frames()));

	buffer_data_samples->resize(samples_per_channel * num_channels);
	for (auto& v : *buffer_data_samples) {
	v = rand_gen->Rand<float>();
	}

	buffer_data->resize(num_channels);
	for (size_t ch = 0; ch < num_channels; ++ch) {
	(buffer_data)[ch] = &(buffer_data_samples)[ch * samples_per_channel];
	}
	}

	void UpdateInputBuffers() {
	test::CopyVectorToAudioBuffer(capture_input_config, capture_input_samples,
	capture_input_buffer.get());
	test::CopyVectorToAudioBuffer(render_input_config, render_input_samples,
	render_input_buffer.get());
	}

	std::unique_ptr<AudioBuffer> render_input_buffer;
	std::unique_ptr<AudioBuffer> capture_input_buffer;
	std::unique_ptr<AudioBuffer> render_output_buffer;
	std::unique_ptr<AudioBuffer> capture_output_buffer;
	StreamConfig render_input_config;
	StreamConfig capture_input_config;
	StreamConfig render_output_config;
	StreamConfig capture_output_config;
	std::vector<float*> render_input;
	std::vector<float> render_input_samples;
	std::vector<float*> capture_input;
	std::vector<float> capture_input_samples;
	std::vector<float*> render_output;
	std::vector<float> render_output_samples;
	std::vector<float*> capture_output;
	std::vector<float> capture_output_samples;
	};

	class SubmodulePerformanceTimer {
	public:
	SubmodulePerformanceTimer() : clock_(webrtc::Clock::GetRealTimeClock()) {
	timestamps_us_.reserve(kNumFramesToProcess);
	}

	void StartTimer() {
	start_timestamp_us_ = rtc::Optional<int64_t>(clock_->TimeInMicroseconds());
	}
	void StopTimer() {
	RTC_DCHECK(start_timestamp_us_);
	timestamps_us_.push_back(clock_->TimeInMicroseconds() -
	*start_timestamp_us_);
	}

	double GetDurationAverage() const {
	RTC_DCHECK(!timestamps_us_.empty());
	return static_cast<double>(std::accumulate(timestamps_us_.begin(),
	timestamps_us_.end(), 0)) /
	timestamps_us_.size();
	}

	double GetDurationStandardDeviation() const {
	RTC_DCHECK(!timestamps_us_.empty());
	double average_duration = GetDurationAverage();

	int64_t variance =
	std::accumulate(timestamps_us_.begin(), timestamps_us_.end(), 0,
	[average_duration](const int64_t& a, const int64_t& b) {
	return a + (b - average_duration);
	});

	return sqrt(variance / timestamps_us_.size());
	}

	private:
	webrtc::Clock* clock_;
	rtc::Optional<int64_t> start_timestamp_us_;
	std::vector<int64_t> timestamps_us_;
	};

	std::string FormPerformanceMeasureString(
	const SubmodulePerformanceTimer& timer) {
	std::string s = std::to_string(timer.GetDurationAverage());
	s += ", ";
	s += std::to_string(timer.GetDurationStandardDeviation());
	return s;
	}

	void RunStandaloneSubmodule(int sample_rate_hz, size_t num_channels) {
	SimulatorBuffers buffers(sample_rate_hz, sample_rate_hz, sample_rate_hz,
	sample_rate_hz, num_channels, num_channels,
	num_channels, num_channels);
	SubmodulePerformanceTimer timer;

	LevelController level_controller;
	level_controller.Initialize(sample_rate_hz);

	for (size_t frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
	buffers.UpdateInputBuffers();

	timer.StartTimer();
	level_controller.Process(buffers.capture_input_buffer.get());
	timer.StopTimer();
	}
	webrtc::test::PrintResultMeanAndError(
	"level_controller_call_durations",
	"_" + std::to_string(sample_rate_hz) + "Hz_" +
	std::to_string(num_channels) + "_channels",
	"StandaloneLevelControl", FormPerformanceMeasureString(timer), "us",
	false);
	}

	void RunTogetherWithApm(std::string test_description,
	int render_input_sample_rate_hz,
	int render_output_sample_rate_hz,
	int capture_input_sample_rate_hz,
	int capture_output_sample_rate_hz,
	size_t num_channels,
	bool use_mobile_aec,
	bool include_default_apm_processing) {
	SimulatorBuffers buffers(
	render_input_sample_rate_hz, capture_input_sample_rate_hz,
	render_output_sample_rate_hz, capture_output_sample_rate_hz, num_channels,
	num_channels, num_channels, num_channels);
	SubmodulePerformanceTimer render_timer;
	SubmodulePerformanceTimer capture_timer;
	SubmodulePerformanceTimer total_timer;

	webrtc::Config config;
	AudioProcessing::Config apm_config;
	if (include_default_apm_processing) {
	config.Set<DelayAgnostic>(new DelayAgnostic(true));
	config.Set<ExtendedFilter>(new ExtendedFilter(true));
	}
	apm_config.level_controller.enabled = true;

	std::unique_ptr<AudioProcessing> apm;
	apm.reset(AudioProcessing::Create(config));
	ASSERT_TRUE(apm.get());
	apm->ApplyConfig(apm_config);

	ASSERT_EQ(AudioProcessing::kNoError,
	apm->gain_control()->Enable(include_default_apm_processing));
	if (use_mobile_aec) {
	ASSERT_EQ(AudioProcessing::kNoError,
	apm->echo_cancellation()->Enable(false));
	ASSERT_EQ(AudioProcessing::kNoError, apm->echo_control_mobile()->Enable(
	include_default_apm_processing));
	} else {
	ASSERT_EQ(AudioProcessing::kNoError,
	apm->echo_cancellation()->Enable(include_default_apm_processing));
	ASSERT_EQ(AudioProcessing::kNoError,
	apm->echo_control_mobile()->Enable(false));
	}
	ASSERT_EQ(AudioProcessing::kNoError,
	apm->high_pass_filter()->Enable(include_default_apm_processing));
	ASSERT_EQ(AudioProcessing::kNoError,
	apm->noise_suppression()->Enable(include_default_apm_processing));
	ASSERT_EQ(AudioProcessing::kNoError,
	apm->voice_detection()->Enable(include_default_apm_processing));
	ASSERT_EQ(AudioProcessing::kNoError,
	apm->level_estimator()->Enable(include_default_apm_processing));

	StreamConfig render_input_config(render_input_sample_rate_hz, num_channels,
	false);
	StreamConfig render_output_config(render_output_sample_rate_hz, num_channels,
	false);
	StreamConfig capture_input_config(capture_input_sample_rate_hz, num_channels,
	false);
	StreamConfig capture_output_config(capture_output_sample_rate_hz,
	num_channels, false);

	for (size_t frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
	buffers.UpdateInputBuffers();

	total_timer.StartTimer();
	render_timer.StartTimer();
	ASSERT_EQ(AudioProcessing::kNoError,
	apm->ProcessReverseStream(
	&buffers.render_input[0], render_input_config,
	render_output_config, &buffers.render_output[0]));

	render_timer.StopTimer();

	capture_timer.StartTimer();
	ASSERT_EQ(AudioProcessing::kNoError, apm->set_stream_delay_ms(0));
	ASSERT_EQ(
	AudioProcessing::kNoError,
	apm->ProcessStream(&buffers.capture_input[0], capture_input_config,
	capture_output_config, &buffers.capture_output[0]));

	capture_timer.StopTimer();
	total_timer.StopTimer();
	}

	webrtc::test::PrintResultMeanAndError(
	"level_controller_call_durations",
	"_" + std::to_string(render_input_sample_rate_hz) + "_" +
	std::to_string(render_output_sample_rate_hz) + "_" +
	std::to_string(capture_input_sample_rate_hz) + "_" +
	std::to_string(capture_output_sample_rate_hz) + "Hz_" +
	std::to_string(num_channels) + "_channels" + "_render",
	test_description, FormPerformanceMeasureString(render_timer), "us",
	false);
	webrtc::test::PrintResultMeanAndError(
	"level_controller_call_durations",
	"_" + std::to_string(render_input_sample_rate_hz) + "_" +
	std::to_string(render_output_sample_rate_hz) + "_" +
	std::to_string(capture_input_sample_rate_hz) + "_" +
	std::to_string(capture_output_sample_rate_hz) + "Hz_" +
	std::to_string(num_channels) + "_channels" + "_capture",
	test_description, FormPerformanceMeasureString(capture_timer), "us",
	false);
	webrtc::test::PrintResultMeanAndError(
	"level_controller_call_durations",
	"_" + std::to_string(render_input_sample_rate_hz) + "_" +
	std::to_string(render_output_sample_rate_hz) + "_" +
	std::to_string(capture_input_sample_rate_hz) + "_" +
	std::to_string(capture_output_sample_rate_hz) + "Hz_" +
	std::to_string(num_channels) + "_channels" + "_total",
	test_description, FormPerformanceMeasureString(total_timer), "us", false);
	}

	} // namespace

	TEST(LevelControllerPerformanceTest, StandaloneProcessing) {
	int sample_rates_to_test[] = {
	AudioProcessing::kSampleRate8kHz, AudioProcessing::kSampleRate16kHz,
	AudioProcessing::kSampleRate32kHz, AudioProcessing::kSampleRate48kHz};
	for (auto sample_rate : sample_rates_to_test) {
	for (size_t num_channels = 1; num_channels <= 2; ++num_channels) {
	RunStandaloneSubmodule(sample_rate, num_channels);
	}
	}
	}

	#if !defined(WEBRTC_ANDROID)
	TEST(LevelControllerPerformanceTest, ProcessingViaApm) {
	#else
	TEST(LevelControllerPerformanceTest, DISABLED_ProcessingViaApm) {
	#endif
	int sample_rates_to_test[] = {AudioProcessing::kSampleRate8kHz,
	AudioProcessing::kSampleRate16kHz,
	AudioProcessing::kSampleRate32kHz,
	AudioProcessing::kSampleRate48kHz, 44100};
	for (auto capture_input_sample_rate_hz : sample_rates_to_test) {
	for (auto capture_output_sample_rate_hz : sample_rates_to_test) {
	for (size_t num_channels = 1; num_channels <= 2; ++num_channels) {
	RunTogetherWithApm("SimpleLevelControlViaApm", 48000, 48000,
	capture_input_sample_rate_hz,
	capture_output_sample_rate_hz, num_channels, false,
	false);
	}
	}
	}
	}

	#if !defined(WEBRTC_ANDROID)
	TEST(LevelControllerPerformanceTest, InteractionWithDefaultApm) {
	#else
	TEST(LevelControllerPerformanceTest, DISABLED_InteractionWithDefaultApm) {
	#endif
	int sample_rates_to_test[] = {AudioProcessing::kSampleRate8kHz,
	AudioProcessing::kSampleRate16kHz,
	AudioProcessing::kSampleRate32kHz,
	AudioProcessing::kSampleRate48kHz, 44100};
	for (auto capture_input_sample_rate_hz : sample_rates_to_test) {
	for (auto capture_output_sample_rate_hz : sample_rates_to_test) {
	for (size_t num_channels = 1; num_channels <= 2; ++num_channels) {
	RunTogetherWithApm("LevelControlAndDefaultDesktopApm", 48000, 48000,
	capture_input_sample_rate_hz,
	capture_output_sample_rate_hz, num_channels, false,
	true);
	RunTogetherWithApm("LevelControlAndDefaultMobileApm", 48000, 48000,
	capture_input_sample_rate_hz,
	capture_output_sample_rate_hz, num_channels, true,
	true);
	}
	}
	}
	}

	} // namespace webrtc