New module for the adaptive level controlling functionality in the audio processing module

webrtc/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc

+/*
+ *  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 "testing/gtest/include/gtest/gtest.h"
+#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/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_.resize(kNumFramesToProcess);

[hlundin-webrtc, 2016/06/28 11:29:01]

You must use reserve, not resize. reserve allocates memory, but the vector is
still empty, while resize also populates it with default-initialized elements.
This will not work at all now that you are using push_back below.

[peah-webrtc, 2016/06/28 22:19:38]

Fully true! Thanks!

Done.

+  }
+
+  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;
+
+  Config config;
+  if (include_default_apm_processing) {
+    config.Set<DelayAgnostic>(new DelayAgnostic(true));
+    config.Set<ExtendedFilter>(new ExtendedFilter(true));
+  }
+  config.Set<LevelControl>(new LevelControl(true));
+
+  std::unique_ptr<AudioProcessing> apm;
+  apm.reset(AudioProcessing::Create(config));
+  ASSERT_TRUE(apm.get());
+
+  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);
+    }
+  }
+}
+
+TEST(LevelControllerPerformanceTest, ProcessingViaApm) {
+  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);
+      }
+    }
+  }
+}
+
+TEST(LevelControllerPerformanceTest, InteractionWithDefaultApm) {
+  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