Revert of Created a test that reports the statistics for the duration of APM stream processing API calls. (patchset #15 id:280001 of https://codereview.webrtc.org/1436553004/ )
Reason for revert:
This breaks the Win32 Release [large tests] bot (webrtc_perf_tests times out after 1h23m): https://build.chromium.org/p/client.webrtc/builders/Win32%20Release%20%5Blarge%20tests%5D
The Mac64 Release [large tests] bot's runtime also increased with +20 minutes.
These bot configs are not a part of the default trybot set, so please run them manually or add this to the CL description:
CQ_EXTRA_TRYBOTS=tryserver.webrtc:win_baremetal,mac_baremetal,linux_baremetal
Original issue's description:
> A unittest that reports the statistics for the duration of an APM stream processing API call.
>
> BUG=webrtc:5099
>
> Committed: https://crrev.com/880896ab0976bbf86a6753d0c900c70e51f421cb
> Cr-Commit-Position: refs/heads/master@{#10786}
TBR=henrik.lundin@webrtc.org,solenberg@webrtc.org,peah@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5099
Review URL: https://codereview.webrtc.org/1473733004
Cr-Original-Commit-Position: refs/heads/master@{#10791}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 6e004a44e8e8f3b9e9727f454ccf179911700bb3
diff --git a/modules/audio_processing/audio_processing_performance_unittest.cc b/modules/audio_processing/audio_processing_performance_unittest.cc
deleted file mode 100644
index 9da3cd4..0000000
--- a/modules/audio_processing/audio_processing_performance_unittest.cc
+++ /dev/null
@@ -1,720 +0,0 @@
-/*
- * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-#include "webrtc/modules/audio_processing/audio_processing_impl.h"
-
-#include <math.h>
-
-#include <algorithm>
-#include <vector>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/array_view.h"
-#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/platform_thread.h"
-#include "webrtc/base/safe_conversions.h"
-#include "webrtc/config.h"
-#include "webrtc/modules/audio_processing/test/test_utils.h"
-#include "webrtc/modules/include/module_common_types.h"
-#include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/system_wrappers/include/event_wrapper.h"
-#include "webrtc/system_wrappers/include/sleep.h"
-#include "webrtc/test/random.h"
-#include "webrtc/test/testsupport/perf_test.h"
-
-namespace webrtc {
-
-namespace {
-
-static const bool kPrintAllDurations = false;
-
-class CallSimulator;
-
-// Type of the render thread APM API call to use in the test.
-enum class ProcessorType { kRender, kCapture };
-
-// Variant of APM processing settings to use in the test.
-enum class SettingsType {
- kDefaultApmDesktop,
- kDefaultApmMobile,
- kDefaultApmDesktopAndBeamformer,
- kDefaultApmDesktopAndIntelligibilityEnhancer,
- kAllSubmodulesTurnedOff,
- kDefaultDesktopApmWithoutDelayAgnostic,
- kDefaultDesktopApmWithoutExtendedFilter
-};
-
-// Variables related to the audio data and formats.
-struct AudioFrameData {
- explicit AudioFrameData(size_t max_frame_size) {
- // Set up the two-dimensional arrays needed for the APM API calls.
- input_framechannels.resize(2 * max_frame_size);
- input_frame.resize(2);
- input_frame[0] = &input_framechannels[0];
- input_frame[1] = &input_framechannels[max_frame_size];
-
- output_frame_channels.resize(2 * max_frame_size);
- output_frame.resize(2);
- output_frame[0] = &output_frame_channels[0];
- output_frame[1] = &output_frame_channels[max_frame_size];
- }
-
- std::vector<float> output_frame_channels;
- std::vector<float*> output_frame;
- std::vector<float> input_framechannels;
- std::vector<float*> input_frame;
- StreamConfig input_stream_config;
- StreamConfig output_stream_config;
-};
-
-// The configuration for the test.
-struct SimulationConfig {
- SimulationConfig(int sample_rate_hz, SettingsType simulation_settings)
- : sample_rate_hz(sample_rate_hz),
- simulation_settings(simulation_settings) {}
-
- static std::vector<SimulationConfig> GenerateSimulationConfigs() {
- std::vector<SimulationConfig> simulation_configs;
-#ifndef WEBRTC_ANDROID
- const SettingsType desktop_settings[] = {
- SettingsType::kDefaultApmDesktop, SettingsType::kAllSubmodulesTurnedOff,
- SettingsType::kDefaultDesktopApmWithoutDelayAgnostic,
- SettingsType::kDefaultDesktopApmWithoutExtendedFilter};
-
- const int desktop_sample_rates[] = {8000, 16000, 32000, 48000};
-
- for (auto sample_rate : desktop_sample_rates) {
- for (auto settings : desktop_settings) {
- simulation_configs.push_back(SimulationConfig(sample_rate, settings));
- }
- }
-
- const SettingsType intelligibility_enhancer_settings[] = {
- SettingsType::kDefaultApmDesktopAndIntelligibilityEnhancer};
-
- const int intelligibility_enhancer_sample_rates[] = {8000, 16000, 32000,
- 48000};
-
- for (auto sample_rate : intelligibility_enhancer_sample_rates) {
- for (auto settings : intelligibility_enhancer_settings) {
- simulation_configs.push_back(SimulationConfig(sample_rate, settings));
- }
- }
-
- const SettingsType beamformer_settings[] = {
- SettingsType::kDefaultApmDesktopAndBeamformer};
-
- const int beamformer_sample_rates[] = {8000, 16000, 32000, 48000};
-
- for (auto sample_rate : beamformer_sample_rates) {
- for (auto settings : beamformer_settings) {
- simulation_configs.push_back(SimulationConfig(sample_rate, settings));
- }
- }
-#endif
-
- const SettingsType mobile_settings[] = {SettingsType::kDefaultApmMobile};
-
- const int mobile_sample_rates[] = {8000, 16000};
-
- for (auto sample_rate : mobile_sample_rates) {
- for (auto settings : mobile_settings) {
- simulation_configs.push_back(SimulationConfig(sample_rate, settings));
- }
- }
-
- return simulation_configs;
- }
-
- std::string SettingsDescription() const {
- std::string description;
- switch (simulation_settings) {
- case SettingsType::kDefaultApmMobile:
- description = "DefaultApmMobile";
- break;
- case SettingsType::kDefaultApmDesktop:
- description = "DefaultApmDesktop";
- break;
- case SettingsType::kDefaultApmDesktopAndBeamformer:
- description = "DefaultApmDesktopAndBeamformer";
- break;
- case SettingsType::kDefaultApmDesktopAndIntelligibilityEnhancer:
- description = "DefaultApmDesktopAndIntelligibilityEnhancer";
- break;
- case SettingsType::kAllSubmodulesTurnedOff:
- description = "AllSubmodulesOff";
- break;
- case SettingsType::kDefaultDesktopApmWithoutDelayAgnostic:
- description = "DefaultDesktopApmWithoutDelayAgnostic";
- break;
- case SettingsType::kDefaultDesktopApmWithoutExtendedFilter:
- description = "DefaultDesktopApmWithoutExtendedFilter";
- break;
- }
- return description;
- }
-
- int sample_rate_hz = 16000;
- SettingsType simulation_settings = SettingsType::kDefaultApmDesktop;
-};
-
-// Handler for the frame counters.
-class FrameCounters {
- public:
- void IncreaseRenderCounter() {
- rtc::CritScope cs(&crit_);
- render_count_++;
- }
-
- void IncreaseCaptureCounter() {
- rtc::CritScope cs(&crit_);
- capture_count_++;
- }
-
- int GetCaptureCounter() const {
- rtc::CritScope cs(&crit_);
- return capture_count_;
- }
-
- int GetRenderCounter() const {
- rtc::CritScope cs(&crit_);
- return render_count_;
- }
-
- int CaptureMinusRenderCounters() const {
- rtc::CritScope cs(&crit_);
- return capture_count_ - render_count_;
- }
-
- int RenderMinusCaptureCounters() const {
- return -CaptureMinusRenderCounters();
- }
-
- bool BothCountersExceedeThreshold(int threshold) const {
- rtc::CritScope cs(&crit_);
- return (render_count_ > threshold && capture_count_ > threshold);
- }
-
- private:
- mutable rtc::CriticalSection crit_;
- int render_count_ GUARDED_BY(crit_) = 0;
- int capture_count_ GUARDED_BY(crit_) = 0;
-};
-
-// Class that protects a flag using a lock.
-class LockedFlag {
- public:
- bool get_flag() const {
- rtc::CritScope cs(&crit_);
- return flag_;
- }
-
- void set_flag() {
- rtc::CritScope cs(&crit_);
- flag_ = true;
- }
-
- private:
- mutable rtc::CriticalSection crit_;
- bool flag_ GUARDED_BY(crit_) = false;
-};
-
-// Parent class for the thread processors.
-class TimedThreadApiProcessor {
- public:
- TimedThreadApiProcessor(ProcessorType processor_type,
- test::Random* rand_gen,
- FrameCounters* shared_counters_state,
- LockedFlag* capture_call_checker,
- CallSimulator* test_framework,
- const SimulationConfig* simulation_config,
- AudioProcessing* apm,
- int num_durations_to_store,
- float input_level,
- int num_channels)
- : rand_gen_(rand_gen),
- frame_counters_(shared_counters_state),
- capture_call_checker_(capture_call_checker),
- test_(test_framework),
- simulation_config_(simulation_config),
- apm_(apm),
- frame_data_(kMaxFrameSize),
- clock_(webrtc::Clock::GetRealTimeClock()),
- num_durations_to_store_(num_durations_to_store),
- input_level_(input_level),
- processor_type_(processor_type),
- num_channels_(num_channels) {
- api_call_durations_.reserve(num_durations_to_store_);
- }
-
- // Implements the callback functionality for the threads.
- bool Process();
-
- // Method for printing out the simulation statistics.
- void print_processor_statistics(std::string processor_name) const {
- const std::string modifier = "_api_call_duration";
-
- // Lambda function for creating a test printout string.
- auto create_mean_and_std_string = [](int64_t average,
- int64_t standard_dev) {
- std::string s = std::to_string(average);
- s += ", ";
- s += std::to_string(standard_dev);
- return s;
- };
-
- const std::string sample_rate_name =
- "_" + std::to_string(simulation_config_->sample_rate_hz) + "Hz";
-
- webrtc::test::PrintResultMeanAndError(
- "apm_timing", sample_rate_name, processor_name,
- create_mean_and_std_string(GetDurationAverage(),
- GetDurationStandardDeviation()),
- "us", false);
-
- if (kPrintAllDurations) {
- std::string value_string = "";
- for (int64_t duration : api_call_durations_) {
- value_string += std::to_string(duration) + ",";
- }
- webrtc::test::PrintResultList("apm_call_durations", sample_rate_name,
- processor_name, value_string, "us", false);
- }
- }
-
- void AddDuration(int64_t duration) {
- if (api_call_durations_.size() < num_durations_to_store_) {
- api_call_durations_.push_back(duration);
- }
- }
-
- private:
- static const int kMaxCallDifference = 10;
- static const int kMaxFrameSize = 480;
- static const int kNumInitializationFrames = 5;
-
- int64_t GetDurationStandardDeviation() const {
- double variance = 0;
- const int64_t average_duration = GetDurationAverage();
- for (size_t k = kNumInitializationFrames; k < api_call_durations_.size();
- k++) {
- int64_t tmp = api_call_durations_[k] - average_duration;
- variance += static_cast<double>(tmp * tmp);
- }
- const int denominator = rtc::checked_cast<int>(api_call_durations_.size()) -
- kNumInitializationFrames;
- return (denominator > 0
- ? rtc::checked_cast<int64_t>(sqrt(variance / denominator))
- : -1);
- }
-
- int64_t GetDurationAverage() const {
- int64_t average_duration = 0;
- for (size_t k = kNumInitializationFrames; k < api_call_durations_.size();
- k++) {
- average_duration += api_call_durations_[k];
- }
- const int denominator = rtc::checked_cast<int>(api_call_durations_.size()) -
- kNumInitializationFrames;
- return (denominator > 0 ? average_duration / denominator : -1);
- }
-
- int ProcessCapture() {
- // Set the stream delay.
- apm_->set_stream_delay_ms(30);
-
- // Call and time the specified capture side API processing method.
- const int64_t start_time = clock_->TimeInMicroseconds();
- const int result = apm_->ProcessStream(
- &frame_data_.input_frame[0], frame_data_.input_stream_config,
- frame_data_.output_stream_config, &frame_data_.output_frame[0]);
- const int64_t end_time = clock_->TimeInMicroseconds();
-
- frame_counters_->IncreaseCaptureCounter();
-
- AddDuration(end_time - start_time);
-
- if (first_process_call_) {
- // Flag that the capture side has been called at least once
- // (needed to ensure that a capture call has been done
- // before the first render call is performed (implicitly
- // required by the APM API).
- capture_call_checker_->set_flag();
- first_process_call_ = false;
- }
- return result;
- }
-
- bool ReadyToProcessCapture() {
- return (frame_counters_->CaptureMinusRenderCounters() <=
- kMaxCallDifference);
- }
-
- int ProcessRender() {
- // Call and time the specified render side API processing method.
- const int64_t start_time = clock_->TimeInMicroseconds();
- const int result = apm_->ProcessReverseStream(
- &frame_data_.input_frame[0], frame_data_.input_stream_config,
- frame_data_.output_stream_config, &frame_data_.output_frame[0]);
- const int64_t end_time = clock_->TimeInMicroseconds();
- frame_counters_->IncreaseRenderCounter();
-
- AddDuration(end_time - start_time);
-
- return result;
- }
-
- bool ReadyToProcessRender() {
- // Do not process until at least one capture call has been done.
- // (implicitly required by the APM API).
- if (first_process_call_ && !capture_call_checker_->get_flag()) {
- return false;
- }
-
- // Ensure that the number of render and capture calls do not differ too
- // much.
- if (frame_counters_->RenderMinusCaptureCounters() > kMaxCallDifference) {
- return false;
- }
-
- first_process_call_ = false;
- return true;
- }
-
- void PrepareFrame() {
- // Lambda function for populating a float multichannel audio frame
- // with random data.
- auto populate_audio_frame = [](float amplitude, size_t num_channels,
- size_t samples_per_channel,
- test::Random* rand_gen, float** frame) {
- for (size_t ch = 0; ch < num_channels; ch++) {
- for (size_t k = 0; k < samples_per_channel; k++) {
- // Store random float number with a value between +-amplitude.
- frame[ch][k] = amplitude * (2 * rand_gen->Rand<float>() - 1);
- }
- }
- };
-
- // Prepare the audio input data and metadata.
- frame_data_.input_stream_config.set_sample_rate_hz(
- simulation_config_->sample_rate_hz);
- frame_data_.input_stream_config.set_num_channels(num_channels_);
- frame_data_.input_stream_config.set_has_keyboard(false);
- populate_audio_frame(input_level_, num_channels_,
- (simulation_config_->sample_rate_hz *
- AudioProcessing::kChunkSizeMs / 1000),
- rand_gen_, &frame_data_.input_frame[0]);
-
- // Prepare the float audio output data and metadata.
- frame_data_.output_stream_config.set_sample_rate_hz(
- simulation_config_->sample_rate_hz);
- frame_data_.output_stream_config.set_num_channels(1);
- frame_data_.output_stream_config.set_has_keyboard(false);
- }
-
- bool ReadyToProcess() {
- switch (processor_type_) {
- case ProcessorType::kRender:
- return ReadyToProcessRender();
- break;
- case ProcessorType::kCapture:
- return ReadyToProcessCapture();
- break;
- }
-
- // Should not be reached, but the return statement is needed for the code to
- // build successfully on Android.
- RTC_NOTREACHED();
- return false;
- }
-
- test::Random* rand_gen_ = nullptr;
- FrameCounters* frame_counters_ = nullptr;
- LockedFlag* capture_call_checker_ = nullptr;
- CallSimulator* test_ = nullptr;
- const SimulationConfig* const simulation_config_ = nullptr;
- AudioProcessing* apm_ = nullptr;
- AudioFrameData frame_data_;
- webrtc::Clock* clock_;
- const size_t num_durations_to_store_;
- std::vector<int64_t> api_call_durations_;
- const float input_level_;
- bool first_process_call_ = true;
- const ProcessorType processor_type_;
- const int num_channels_ = 1;
-};
-
-// Class for managing the test simulation.
-class CallSimulator : public ::testing::TestWithParam<SimulationConfig> {
- public:
- CallSimulator()
- : test_complete_(EventWrapper::Create()),
- render_thread_(PlatformThread::CreateThread(RenderProcessorThreadFunc,
- this,
- "render")),
- capture_thread_(PlatformThread::CreateThread(CaptureProcessorThreadFunc,
- this,
- "capture")),
- rand_gen_(42U),
- simulation_config_(static_cast<SimulationConfig>(GetParam())) {}
-
- // Run the call simulation with a timeout.
- EventTypeWrapper Run() {
- StartThreads();
-
- EventTypeWrapper result = test_complete_->Wait(kTestTimeout);
-
- render_thread_state_->print_processor_statistics(
- simulation_config_.SettingsDescription() + "_render");
- capture_thread_state_->print_processor_statistics(
- simulation_config_.SettingsDescription() + "_capture");
-
- return result;
- }
-
- // Tests whether all the required render and capture side calls have been
- // done.
- void MaybeEndTest() {
- if (frame_counters_.BothCountersExceedeThreshold(kMinNumFramesToProcess)) {
- test_complete_->Set();
- }
- }
-
- private:
- static const float kCaptureInputFloatLevel;
- static const float kRenderInputFloatLevel;
- static const int kMinNumFramesToProcess = 150;
- static const int32_t kTestTimeout = 3 * 10 * kMinNumFramesToProcess;
-
- // ::testing::TestWithParam<> implementation.
- void TearDown() override {
- render_thread_->Stop();
- capture_thread_->Stop();
- }
-
- // Simulator and APM setup.
- void SetUp() override {
- // Lambda function for setting the default APM runtime settings for desktop.
- auto set_default_desktop_apm_runtime_settings = [](AudioProcessing* apm) {
- ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(true));
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->set_mode(GainControl::kAdaptiveDigital));
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
- ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
- ASSERT_EQ(apm->kNoError, apm->voice_detection()->Enable(true));
- ASSERT_EQ(apm->kNoError, apm->echo_control_mobile()->Enable(false));
- ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
- ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->enable_metrics(true));
- ASSERT_EQ(apm->kNoError,
- apm->echo_cancellation()->enable_delay_logging(true));
- };
-
- // Lambda function for setting the default APM runtime settings for mobile.
- auto set_default_mobile_apm_runtime_settings = [](AudioProcessing* apm) {
- ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(true));
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->set_mode(GainControl::kAdaptiveDigital));
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
- ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
- ASSERT_EQ(apm->kNoError, apm->voice_detection()->Enable(true));
- ASSERT_EQ(apm->kNoError, apm->echo_control_mobile()->Enable(true));
- ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(false));
- };
-
- // Lambda function for turning off all of the APM runtime settings
- // submodules.
- auto turn_off_default_apm_runtime_settings = [](AudioProcessing* apm) {
- ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(false));
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(false));
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->set_mode(GainControl::kAdaptiveDigital));
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(false));
- ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(false));
- ASSERT_EQ(apm->kNoError, apm->voice_detection()->Enable(false));
- ASSERT_EQ(apm->kNoError, apm->echo_control_mobile()->Enable(false));
- ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(false));
- ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->enable_metrics(false));
- ASSERT_EQ(apm->kNoError,
- apm->echo_cancellation()->enable_delay_logging(false));
- };
-
- // Lambda function for adding default desktop APM settings to a config.
- auto add_default_desktop_config = [](Config* config) {
- config->Set<ExtendedFilter>(new ExtendedFilter(true));
- config->Set<DelayAgnostic>(new DelayAgnostic(true));
- };
-
- // Lambda function for adding beamformer settings to a config.
- auto add_beamformer_config = [](Config* config) {
- const size_t num_mics = 2;
- const std::vector<Point> array_geometry =
- ParseArrayGeometry("0 0 0 0.05 0 0", num_mics);
- RTC_CHECK_EQ(array_geometry.size(), num_mics);
-
- config->Set<Beamforming>(
- new Beamforming(true, array_geometry,
- SphericalPointf(DegreesToRadians(90), 0.f, 1.f)));
- };
-
- int num_capture_channels = 1;
- switch (simulation_config_.simulation_settings) {
- case SettingsType::kDefaultApmMobile: {
- apm_.reset(AudioProcessingImpl::Create());
- ASSERT_TRUE(!!apm_);
- set_default_mobile_apm_runtime_settings(apm_.get());
- break;
- }
- case SettingsType::kDefaultApmDesktop: {
- Config config;
- add_default_desktop_config(&config);
- apm_.reset(AudioProcessingImpl::Create(config));
- ASSERT_TRUE(!!apm_);
- set_default_desktop_apm_runtime_settings(apm_.get());
- apm_->SetExtraOptions(config);
- break;
- }
- case SettingsType::kDefaultApmDesktopAndBeamformer: {
- Config config;
- add_beamformer_config(&config);
- add_default_desktop_config(&config);
- apm_.reset(AudioProcessingImpl::Create(config));
- ASSERT_TRUE(!!apm_);
- set_default_desktop_apm_runtime_settings(apm_.get());
- apm_->SetExtraOptions(config);
- num_capture_channels = 2;
- break;
- }
- case SettingsType::kDefaultApmDesktopAndIntelligibilityEnhancer: {
- Config config;
- config.Set<Intelligibility>(new Intelligibility(true));
- add_default_desktop_config(&config);
- apm_.reset(AudioProcessingImpl::Create(config));
- ASSERT_TRUE(!!apm_);
- set_default_desktop_apm_runtime_settings(apm_.get());
- apm_->SetExtraOptions(config);
- break;
- }
- case SettingsType::kAllSubmodulesTurnedOff: {
- apm_.reset(AudioProcessingImpl::Create());
- ASSERT_TRUE(!!apm_);
- turn_off_default_apm_runtime_settings(apm_.get());
- break;
- }
- case SettingsType::kDefaultDesktopApmWithoutDelayAgnostic: {
- Config config;
- config.Set<ExtendedFilter>(new ExtendedFilter(true));
- config.Set<DelayAgnostic>(new DelayAgnostic(false));
- apm_.reset(AudioProcessingImpl::Create(config));
- ASSERT_TRUE(!!apm_);
- set_default_desktop_apm_runtime_settings(apm_.get());
- apm_->SetExtraOptions(config);
- break;
- }
- case SettingsType::kDefaultDesktopApmWithoutExtendedFilter: {
- Config config;
- config.Set<ExtendedFilter>(new ExtendedFilter(false));
- config.Set<DelayAgnostic>(new DelayAgnostic(true));
- apm_.reset(AudioProcessingImpl::Create(config));
- ASSERT_TRUE(!!apm_);
- set_default_desktop_apm_runtime_settings(apm_.get());
- apm_->SetExtraOptions(config);
- break;
- }
- }
-
- render_thread_state_.reset(new TimedThreadApiProcessor(
- ProcessorType::kRender, &rand_gen_, &frame_counters_,
- &capture_call_checker_, this, &simulation_config_, apm_.get(),
- kMinNumFramesToProcess, kRenderInputFloatLevel, 1));
- capture_thread_state_.reset(new TimedThreadApiProcessor(
- ProcessorType::kCapture, &rand_gen_, &frame_counters_,
- &capture_call_checker_, this, &simulation_config_, apm_.get(),
- kMinNumFramesToProcess, kCaptureInputFloatLevel, num_capture_channels));
- }
-
- // Thread callback for the render thread.
- static bool RenderProcessorThreadFunc(void* context) {
- return reinterpret_cast<CallSimulator*>(context)
- ->render_thread_state_->Process();
- }
-
- // Thread callback for the capture thread.
- static bool CaptureProcessorThreadFunc(void* context) {
- return reinterpret_cast<CallSimulator*>(context)
- ->capture_thread_state_->Process();
- }
-
- // Start the threads used in the test.
- void StartThreads() {
- ASSERT_NO_FATAL_FAILURE(render_thread_->Start());
- render_thread_->SetPriority(kRealtimePriority);
- ASSERT_NO_FATAL_FAILURE(capture_thread_->Start());
- capture_thread_->SetPriority(kRealtimePriority);
- }
-
- // Event handler for the test.
- const rtc::scoped_ptr<EventWrapper> test_complete_;
-
- // Thread related variables.
- rtc::scoped_ptr<PlatformThread> render_thread_;
- rtc::scoped_ptr<PlatformThread> capture_thread_;
- test::Random rand_gen_;
-
- rtc::scoped_ptr<AudioProcessing> apm_;
- const SimulationConfig simulation_config_;
- FrameCounters frame_counters_;
- LockedFlag capture_call_checker_;
- rtc::scoped_ptr<TimedThreadApiProcessor> render_thread_state_;
- rtc::scoped_ptr<TimedThreadApiProcessor> capture_thread_state_;
-};
-
-// Implements the callback functionality for the threads.
-bool TimedThreadApiProcessor::Process() {
- PrepareFrame();
-
- // Wait in a spinlock manner until it is ok to start processing.
- // Note that SleepMs is not applicable since it only allows sleeping
- // on a millisecond basis which is too long.
- while (!ReadyToProcess()) {
- }
-
- int result = AudioProcessing::kNoError;
- switch (processor_type_) {
- case ProcessorType::kRender:
- result = ProcessRender();
- break;
- case ProcessorType::kCapture:
- result = ProcessCapture();
- break;
- }
-
- EXPECT_EQ(result, AudioProcessing::kNoError);
-
- test_->MaybeEndTest();
-
- return true;
-}
-
-const float CallSimulator::kRenderInputFloatLevel = 0.5f;
-const float CallSimulator::kCaptureInputFloatLevel = 0.03125f;
-} // anonymous namespace
-
-TEST_P(CallSimulator, ApiCallDurationTest) {
- // Run test and verify that it did not time out.
- EXPECT_EQ(kEventSignaled, Run());
-}
-
-INSTANTIATE_TEST_CASE_P(
- AudioProcessingPerformanceTest,
- CallSimulator,
- ::testing::ValuesIn(SimulationConfig::GenerateSimulationConfigs()));
-
-} // namespace webrtc
diff --git a/webrtc_tests.gypi b/webrtc_tests.gypi
index 1a2d86c..cd51bc9 100644
--- a/webrtc_tests.gypi
+++ b/webrtc_tests.gypi
@@ -207,7 +207,6 @@
'sources': [
'call/call_perf_tests.cc',
'modules/audio_coding/neteq/test/neteq_performance_unittest.cc',
- 'modules/audio_processing/audio_processing_performance_unittest.cc',
'modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc',
'video/full_stack.cc',
'video/rampup_tests.cc',
@@ -216,8 +215,6 @@
'dependencies': [
'<(DEPTH)/testing/gmock.gyp:gmock',
'<(DEPTH)/testing/gtest.gyp:gtest',
- '<(webrtc_root)/modules/modules.gyp:audio_processing',
- '<(webrtc_root)/modules/modules.gyp:audioproc_test_utils',
'<(webrtc_root)/modules/modules.gyp:video_capture',
'<(webrtc_root)/test/test.gyp:channel_transport',
'<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
