Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 568 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_performance_unittest.cc b/third_party/libwebrtc/modules/audio_processing/audio_processing_performance_unittest.cc
new file mode 100644
index 0000000000..10d3d84951
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_performance_unittest.cc
@@ -0,0 +1,568 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include <math.h>
+
+#include <algorithm>
+#include <atomic>
+#include <memory>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "api/array_view.h"
+#include "api/numerics/samples_stats_counter.h"
+#include "api/test/metrics/global_metrics_logger_and_exporter.h"
+#include "api/test/metrics/metric.h"
+#include "modules/audio_processing/audio_processing_impl.h"
+#include "modules/audio_processing/test/audio_processing_builder_for_testing.h"
+#include "modules/audio_processing/test/test_utils.h"
+#include "rtc_base/event.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/platform_thread.h"
+#include "rtc_base/random.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::webrtc::test::GetGlobalMetricsLogger;
+using ::webrtc::test::ImprovementDirection;
+using ::webrtc::test::Metric;
+using ::webrtc::test::Unit;
+
+class CallSimulator;
+
+// Type of the render thread APM API call to use in the test.
+enum class ProcessorType { kRender, kCapture };
+
+// Variant of APM processing settings to use in the test.
+enum class SettingsType {
+  kDefaultApmDesktop,
+  kDefaultApmMobile,
+  kAllSubmodulesTurnedOff,
+  kDefaultApmDesktopWithoutDelayAgnostic,
+  kDefaultApmDesktopWithoutExtendedFilter
+};
+
+// Variables related to the audio data and formats.
+struct AudioFrameData {
+  explicit AudioFrameData(size_t max_frame_size) {
+    // Set up the two-dimensional arrays needed for the APM API calls.
+    input_framechannels.resize(2 * max_frame_size);
+    input_frame.resize(2);
+    input_frame[0] = &input_framechannels[0];
+    input_frame[1] = &input_framechannels[max_frame_size];
+
+    output_frame_channels.resize(2 * max_frame_size);
+    output_frame.resize(2);
+    output_frame[0] = &output_frame_channels[0];
+    output_frame[1] = &output_frame_channels[max_frame_size];
+  }
+
+  std::vector<float> output_frame_channels;
+  std::vector<float*> output_frame;
+  std::vector<float> input_framechannels;
+  std::vector<float*> input_frame;
+  StreamConfig input_stream_config;
+  StreamConfig output_stream_config;
+};
+
+// The configuration for the test.
+struct SimulationConfig {
+  SimulationConfig(int sample_rate_hz, SettingsType simulation_settings)
+      : sample_rate_hz(sample_rate_hz),
+        simulation_settings(simulation_settings) {}
+
+  static std::vector<SimulationConfig> GenerateSimulationConfigs() {
+    std::vector<SimulationConfig> simulation_configs;
+#ifndef WEBRTC_ANDROID
+    const SettingsType desktop_settings[] = {
+        SettingsType::kDefaultApmDesktop, SettingsType::kAllSubmodulesTurnedOff,
+        SettingsType::kDefaultApmDesktopWithoutDelayAgnostic,
+        SettingsType::kDefaultApmDesktopWithoutExtendedFilter};
+
+    const int desktop_sample_rates[] = {8000, 16000, 32000, 48000};
+
+    for (auto sample_rate : desktop_sample_rates) {
+      for (auto settings : desktop_settings) {
+        simulation_configs.push_back(SimulationConfig(sample_rate, settings));
+      }
+    }
+#endif
+
+    const SettingsType mobile_settings[] = {SettingsType::kDefaultApmMobile};
+
+    const int mobile_sample_rates[] = {8000, 16000};
+
+    for (auto sample_rate : mobile_sample_rates) {
+      for (auto settings : mobile_settings) {
+        simulation_configs.push_back(SimulationConfig(sample_rate, settings));
+      }
+    }
+
+    return simulation_configs;
+  }
+
+  std::string SettingsDescription() const {
+    std::string description;
+    switch (simulation_settings) {
+      case SettingsType::kDefaultApmMobile:
+        description = "DefaultApmMobile";
+        break;
+      case SettingsType::kDefaultApmDesktop:
+        description = "DefaultApmDesktop";
+        break;
+      case SettingsType::kAllSubmodulesTurnedOff:
+        description = "AllSubmodulesOff";
+        break;
+      case SettingsType::kDefaultApmDesktopWithoutDelayAgnostic:
+        description = "DefaultApmDesktopWithoutDelayAgnostic";
+        break;
+      case SettingsType::kDefaultApmDesktopWithoutExtendedFilter:
+        description = "DefaultApmDesktopWithoutExtendedFilter";
+        break;
+    }
+    return description;
+  }
+
+  int sample_rate_hz = 16000;
+  SettingsType simulation_settings = SettingsType::kDefaultApmDesktop;
+};
+
+// Handler for the frame counters.
+class FrameCounters {
+ public:
+  void IncreaseRenderCounter() { render_count_.fetch_add(1); }
+
+  void IncreaseCaptureCounter() { capture_count_.fetch_add(1); }
+
+  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 capture_count_.load(std::memory_order_acquire) -
+           render_count_.load(std::memory_order_acquire);
+  }
+
+  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 = capture_count_.load(std::memory_order_acquire);
+    const int render_count = render_count_.load(std::memory_order_acquire);
+    return (render_count > threshold && capture_count > threshold);
+  }
+
+ private:
+  std::atomic<int> render_count_{0};
+  std::atomic<int> capture_count_{0};
+};
+
+// Class that represents a flag that can only be raised.
+class LockedFlag {
+ public:
+  bool get_flag() const { return flag_.load(std::memory_order_acquire); }
+
+  void set_flag() {
+    if (!get_flag()) {
+      // read-only operation to avoid affecting the cache-line.
+      int zero = 0;
+      flag_.compare_exchange_strong(zero, 1);
+    }
+  }
+
+ private:
+  std::atomic<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),
+        api_call_durations_(num_durations_to_store_ - kNumInitializationFrames),
+        samples_count_(0),
+        input_level_(input_level),
+        processor_type_(processor_type),
+        num_channels_(num_channels) {}
+
+  // Implements the callback functionality for the threads.
+  bool Process();
+
+  // Method for printing out the simulation statistics.
+  void print_processor_statistics(absl::string_view processor_name) const {
+    const std::string modifier = "_api_call_duration";
+
+    const std::string sample_rate_name =
+        "_" + std::to_string(simulation_config_->sample_rate_hz) + "Hz";
+
+    GetGlobalMetricsLogger()->LogMetric(
+        "apm_timing" + sample_rate_name, processor_name, api_call_durations_,
+        Unit::kMilliseconds, ImprovementDirection::kNeitherIsBetter);
+  }
+
+  void AddDuration(int64_t duration) {
+    if (samples_count_ >= kNumInitializationFrames &&
+        samples_count_ < num_durations_to_store_) {
+      api_call_durations_.AddSample(duration);
+    }
+    samples_count_++;
+  }
+
+ private:
+  static const int kMaxCallDifference = 10;
+  static const int kMaxFrameSize = 480;
+  static const int kNumInitializationFrames = 5;
+
+  int ProcessCapture() {
+    // Set the stream delay.
+    apm_->set_stream_delay_ms(30);
+
+    // Call and time the specified capture side API processing method.
+    const int64_t start_time = clock_->TimeInMicroseconds();
+    const int result = apm_->ProcessStream(
+        &frame_data_.input_frame[0], frame_data_.input_stream_config,
+        frame_data_.output_stream_config, &frame_data_.output_frame[0]);
+    const int64_t end_time = clock_->TimeInMicroseconds();
+
+    frame_counters_->IncreaseCaptureCounter();
+
+    AddDuration(end_time - start_time);
+
+    if (first_process_call_) {
+      // Flag that the capture side has been called at least once
+      // (needed to ensure that a capture call has been done
+      // before the first render call is performed (implicitly
+      // required by the APM API).
+      capture_call_checker_->set_flag();
+      first_process_call_ = false;
+    }
+    return result;
+  }
+
+  bool ReadyToProcessCapture() {
+    return (frame_counters_->CaptureMinusRenderCounters() <=
+            kMaxCallDifference);
+  }
+
+  int ProcessRender() {
+    // Call and time the specified render side API processing method.
+    const int64_t start_time = clock_->TimeInMicroseconds();
+    const int result = apm_->ProcessReverseStream(
+        &frame_data_.input_frame[0], frame_data_.input_stream_config,
+        frame_data_.output_stream_config, &frame_data_.output_frame[0]);
+    const int64_t end_time = clock_->TimeInMicroseconds();
+    frame_counters_->IncreaseRenderCounter();
+
+    AddDuration(end_time - start_time);
+
+    return result;
+  }
+
+  bool ReadyToProcessRender() {
+    // Do not process until at least one capture call has been done.
+    // (implicitly required by the APM API).
+    if (first_process_call_ && !capture_call_checker_->get_flag()) {
+      return false;
+    }
+
+    // Ensure that the number of render and capture calls do not differ too
+    // much.
+    if (frame_counters_->RenderMinusCaptureCounters() > kMaxCallDifference) {
+      return false;
+    }
+
+    first_process_call_ = false;
+    return true;
+  }
+
+  void PrepareFrame() {
+    // Lambda function for populating a float multichannel audio frame
+    // with random data.
+    auto populate_audio_frame = [](float amplitude, size_t num_channels,
+                                   size_t samples_per_channel, Random* rand_gen,
+                                   float** frame) {
+      for (size_t ch = 0; ch < num_channels; ch++) {
+        for (size_t k = 0; k < samples_per_channel; k++) {
+          // Store random float number with a value between +-amplitude.
+          frame[ch][k] = amplitude * (2 * rand_gen->Rand<float>() - 1);
+        }
+      }
+    };
+
+    // Prepare the audio input data and metadata.
+    frame_data_.input_stream_config.set_sample_rate_hz(
+        simulation_config_->sample_rate_hz);
+    frame_data_.input_stream_config.set_num_channels(num_channels_);
+    populate_audio_frame(input_level_, num_channels_,
+                         (simulation_config_->sample_rate_hz *
+                          AudioProcessing::kChunkSizeMs / 1000),
+                         rand_gen_, &frame_data_.input_frame[0]);
+
+    // Prepare the float audio output data and metadata.
+    frame_data_.output_stream_config.set_sample_rate_hz(
+        simulation_config_->sample_rate_hz);
+    frame_data_.output_stream_config.set_num_channels(1);
+  }
+
+  bool ReadyToProcess() {
+    switch (processor_type_) {
+      case ProcessorType::kRender:
+        return ReadyToProcessRender();
+
+      case ProcessorType::kCapture:
+        return ReadyToProcessCapture();
+    }
+
+    // Should not be reached, but the return statement is needed for the code to
+    // build successfully on Android.
+    RTC_DCHECK_NOTREACHED();
+    return false;
+  }
+
+  Random* rand_gen_ = nullptr;
+  FrameCounters* frame_counters_ = nullptr;
+  LockedFlag* capture_call_checker_ = nullptr;
+  CallSimulator* test_ = nullptr;
+  const SimulationConfig* const simulation_config_ = nullptr;
+  AudioProcessing* apm_ = nullptr;
+  AudioFrameData frame_data_;
+  webrtc::Clock* clock_;
+  const size_t num_durations_to_store_;
+  SamplesStatsCounter api_call_durations_;
+  size_t samples_count_ = 0;
+  const float input_level_;
+  bool first_process_call_ = true;
+  const ProcessorType processor_type_;
+  const int num_channels_ = 1;
+};
+
+// Class for managing the test simulation.
+class CallSimulator : public ::testing::TestWithParam<SimulationConfig> {
+ public:
+  CallSimulator()
+      : rand_gen_(42U),
+        simulation_config_(static_cast<SimulationConfig>(GetParam())) {}
+
+  // Run the call simulation with a timeout.
+  bool Run() {
+    StartThreads();
+
+    bool result = test_complete_.Wait(kTestTimeout);
+
+    StopThreads();
+
+    render_thread_state_->print_processor_statistics(
+        simulation_config_.SettingsDescription() + "_render");
+    capture_thread_state_->print_processor_statistics(
+        simulation_config_.SettingsDescription() + "_capture");
+
+    return result;
+  }
+
+  // Tests whether all the required render and capture side calls have been
+  // done.
+  bool MaybeEndTest() {
+    if (frame_counters_.BothCountersExceedeThreshold(kMinNumFramesToProcess)) {
+      test_complete_.Set();
+      return true;
+    }
+    return false;
+  }
+
+ private:
+  static const float kCaptureInputFloatLevel;
+  static const float kRenderInputFloatLevel;
+  static const int kMinNumFramesToProcess = 150;
+  static constexpr TimeDelta kTestTimeout =
+      TimeDelta::Millis(3 * 10 * kMinNumFramesToProcess);
+
+  // Stop all running threads.
+  void StopThreads() {
+    render_thread_.Finalize();
+    capture_thread_.Finalize();
+  }
+
+  // Simulator and APM setup.
+  void SetUp() override {
+    // Lambda function for setting the default APM runtime settings for desktop.
+    auto set_default_desktop_apm_runtime_settings = [](AudioProcessing* apm) {
+      AudioProcessing::Config apm_config = apm->GetConfig();
+      apm_config.echo_canceller.enabled = true;
+      apm_config.echo_canceller.mobile_mode = false;
+      apm_config.noise_suppression.enabled = true;
+      apm_config.gain_controller1.enabled = true;
+      apm_config.gain_controller1.mode =
+          AudioProcessing::Config::GainController1::kAdaptiveDigital;
+      apm->ApplyConfig(apm_config);
+    };
+
+    // Lambda function for setting the default APM runtime settings for mobile.
+    auto set_default_mobile_apm_runtime_settings = [](AudioProcessing* apm) {
+      AudioProcessing::Config apm_config = apm->GetConfig();
+      apm_config.echo_canceller.enabled = true;
+      apm_config.echo_canceller.mobile_mode = true;
+      apm_config.noise_suppression.enabled = true;
+      apm_config.gain_controller1.mode =
+          AudioProcessing::Config::GainController1::kAdaptiveDigital;
+      apm->ApplyConfig(apm_config);
+    };
+
+    // Lambda function for turning off all of the APM runtime settings
+    // submodules.
+    auto turn_off_default_apm_runtime_settings = [](AudioProcessing* apm) {
+      AudioProcessing::Config apm_config = apm->GetConfig();
+      apm_config.echo_canceller.enabled = false;
+      apm_config.gain_controller1.enabled = false;
+      apm_config.noise_suppression.enabled = false;
+      apm->ApplyConfig(apm_config);
+    };
+
+    int num_capture_channels = 1;
+    switch (simulation_config_.simulation_settings) {
+      case SettingsType::kDefaultApmMobile: {
+        apm_ = AudioProcessingBuilderForTesting().Create();
+        ASSERT_TRUE(!!apm_);
+        set_default_mobile_apm_runtime_settings(apm_.get());
+        break;
+      }
+      case SettingsType::kDefaultApmDesktop: {
+        apm_ = AudioProcessingBuilderForTesting().Create();
+        ASSERT_TRUE(!!apm_);
+        set_default_desktop_apm_runtime_settings(apm_.get());
+        break;
+      }
+      case SettingsType::kAllSubmodulesTurnedOff: {
+        apm_ = AudioProcessingBuilderForTesting().Create();
+        ASSERT_TRUE(!!apm_);
+        turn_off_default_apm_runtime_settings(apm_.get());
+        break;
+      }
+      case SettingsType::kDefaultApmDesktopWithoutDelayAgnostic: {
+        apm_ = AudioProcessingBuilderForTesting().Create();
+        ASSERT_TRUE(!!apm_);
+        set_default_desktop_apm_runtime_settings(apm_.get());
+        break;
+      }
+      case SettingsType::kDefaultApmDesktopWithoutExtendedFilter: {
+        apm_ = AudioProcessingBuilderForTesting().Create();
+        ASSERT_TRUE(!!apm_);
+        set_default_desktop_apm_runtime_settings(apm_.get());
+        break;
+      }
+    }
+
+    render_thread_state_.reset(new TimedThreadApiProcessor(
+        ProcessorType::kRender, &rand_gen_, &frame_counters_,
+        &capture_call_checker_, this, &simulation_config_, apm_.get(),
+        kMinNumFramesToProcess, kRenderInputFloatLevel, 1));
+    capture_thread_state_.reset(new TimedThreadApiProcessor(
+        ProcessorType::kCapture, &rand_gen_, &frame_counters_,
+        &capture_call_checker_, this, &simulation_config_, apm_.get(),
+        kMinNumFramesToProcess, kCaptureInputFloatLevel, num_capture_channels));
+  }
+
+  // Start the threads used in the test.
+  void StartThreads() {
+    const auto attributes =
+        rtc::ThreadAttributes().SetPriority(rtc::ThreadPriority::kRealtime);
+    render_thread_ = rtc::PlatformThread::SpawnJoinable(
+        [this] {
+          while (render_thread_state_->Process()) {
+          }
+        },
+        "render", attributes);
+    capture_thread_ = rtc::PlatformThread::SpawnJoinable(
+        [this] {
+          while (capture_thread_state_->Process()) {
+          }
+        },
+        "capture", attributes);
+  }
+
+  // Event handler for the test.
+  rtc::Event test_complete_;
+
+  // Thread related variables.
+  Random rand_gen_;
+
+  rtc::scoped_refptr<AudioProcessing> apm_;
+  const SimulationConfig simulation_config_;
+  FrameCounters frame_counters_;
+  LockedFlag capture_call_checker_;
+  std::unique_ptr<TimedThreadApiProcessor> render_thread_state_;
+  std::unique_ptr<TimedThreadApiProcessor> capture_thread_state_;
+  rtc::PlatformThread render_thread_;
+  rtc::PlatformThread capture_thread_;
+};
+
+// Implements the callback functionality for the threads.
+bool TimedThreadApiProcessor::Process() {
+  PrepareFrame();
+
+  // Wait in a spinlock manner until it is ok to start processing.
+  // Note that SleepMs is not applicable since it only allows sleeping
+  // on a millisecond basis which is too long.
+  // TODO(tommi): This loop may affect the performance of the test that it's
+  // meant to measure.  See if we could use events instead to signal readiness.
+  while (!ReadyToProcess()) {
+  }
+
+  int result = AudioProcessing::kNoError;
+  switch (processor_type_) {
+    case ProcessorType::kRender:
+      result = ProcessRender();
+      break;
+    case ProcessorType::kCapture:
+      result = ProcessCapture();
+      break;
+  }
+
+  EXPECT_EQ(result, AudioProcessing::kNoError);
+
+  return !test_->MaybeEndTest();
+}
+
+const float CallSimulator::kRenderInputFloatLevel = 0.5f;
+const float CallSimulator::kCaptureInputFloatLevel = 0.03125f;
+}  // anonymous namespace
+
+// TODO(peah): Reactivate once issue 7712 has been resolved.
+TEST_P(CallSimulator, DISABLED_ApiCallDurationTest) {
+  // Run test and verify that it did not time out.
+  EXPECT_TRUE(Run());
+}
+
+INSTANTIATE_TEST_SUITE_P(
+    AudioProcessingPerformanceTest,
+    CallSimulator,
+    ::testing::ValuesIn(SimulationConfig::GenerateSimulationConfigs()));
+
+}  // namespace webrtc