| /* |
| * 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 <memory> |
| #include <vector> |
| |
| #include "webrtc/modules/audio_processing/test/test_utils.h" |
| #include "webrtc/modules/include/module_common_types.h" |
| #include "webrtc/rtc_base/array_view.h" |
| #include "webrtc/rtc_base/atomicops.h" |
| #include "webrtc/rtc_base/platform_thread.h" |
| #include "webrtc/rtc_base/random.h" |
| #include "webrtc/rtc_base/safe_conversions.h" |
| #include "webrtc/system_wrappers/include/clock.h" |
| #include "webrtc/system_wrappers/include/event_wrapper.h" |
| #include "webrtc/test/gtest.h" |
| #include "webrtc/test/testsupport/perf_test.h" |
| |
| // Check to verify that the define for the intelligibility enhancer is properly |
| // set. |
| #if !defined(WEBRTC_INTELLIGIBILITY_ENHANCER) || \ |
| (WEBRTC_INTELLIGIBILITY_ENHANCER != 0 && \ |
| WEBRTC_INTELLIGIBILITY_ENHANCER != 1) |
| #error "Set WEBRTC_INTELLIGIBILITY_ENHANCER to either 0 or 1" |
| #endif |
| |
| 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, |
| 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)); |
| } |
| } |
| |
| #if WEBRTC_INTELLIGIBILITY_ENHANCER == 1 |
| 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)); |
| } |
| } |
| #endif |
| |
| 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::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"; |
| |
| // 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(); |
| |
| case ProcessorType::kCapture: |
| return ReadyToProcessCapture(); |
| } |
| |
| // Should not be reached, but the return statement is needed for the code to |
| // build successfully on Android. |
| RTC_NOTREACHED(); |
| return false; |
| } |
| |
| Random* rand_gen_ = nullptr; |
| FrameCounters* frame_counters_ = nullptr; |
| LockedFlag* capture_call_checker_ = nullptr; |
| CallSimulator* test_ = nullptr; |
| const SimulationConfig* const simulation_config_ = nullptr; |
| AudioProcessing* apm_ = nullptr; |
| AudioFrameData frame_data_; |
| webrtc::Clock* clock_; |
| const size_t num_durations_to_store_; |
| std::vector<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::kDefaultApmDesktopWithoutDelayAgnostic: { |
| 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::kDefaultApmDesktopWithoutExtendedFilter: { |
| 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 std::unique_ptr<EventWrapper> test_complete_; |
| |
| // Thread related variables. |
| std::unique_ptr<rtc::PlatformThread> render_thread_; |
| std::unique_ptr<rtc::PlatformThread> capture_thread_; |
| Random rand_gen_; |
| |
| std::unique_ptr<AudioProcessing> apm_; |
| const SimulationConfig simulation_config_; |
| FrameCounters frame_counters_; |
| LockedFlag capture_call_checker_; |
| std::unique_ptr<TimedThreadApiProcessor> render_thread_state_; |
| std::unique_ptr<TimedThreadApiProcessor> capture_thread_state_; |
| }; |
| |
| // Implements the callback functionality for the threads. |
| bool TimedThreadApiProcessor::Process() { |
| PrepareFrame(); |
| |
| // Wait in a spinlock manner until it is ok to start processing. |
| // Note that SleepMs is not applicable since it only allows sleeping |
| // on a millisecond basis which is too long. |
| // TODO(tommi): This loop may affect the performance of the test that it's |
| // meant to measure. See if we could use events instead to signal readiness. |
| while (!ReadyToProcess()) { |
| } |
| |
| int result = AudioProcessing::kNoError; |
| switch (processor_type_) { |
| case ProcessorType::kRender: |
| result = ProcessRender(); |
| break; |
| case ProcessorType::kCapture: |
| result = ProcessCapture(); |
| break; |
| } |
| |
| EXPECT_EQ(result, AudioProcessing::kNoError); |
| |
| return !test_->MaybeEndTest(); |
| } |
| |
| const float CallSimulator::kRenderInputFloatLevel = 0.5f; |
| const float CallSimulator::kCaptureInputFloatLevel = 0.03125f; |
| } // anonymous namespace |
| |
| // TODO(peah): Reactivate once issue 7712 has been resolved. |
| TEST_P(CallSimulator, DISABLED_ApiCallDurationTest) { |
| // Run test and verify that it did not time out. |
| EXPECT_EQ(kEventSignaled, Run()); |
| } |
| |
| INSTANTIATE_TEST_CASE_P( |
| AudioProcessingPerformanceTest, |
| CallSimulator, |
| ::testing::ValuesIn(SimulationConfig::GenerateSimulationConfigs())); |
| |
| } // namespace webrtc |