Added a perf test for the residual echo detector.

webrtc/modules/audio_processing/residual_echo_detector_complexity_unittest.cc

Index: webrtc/modules/audio_processing/residual_echo_detector_complexity_unittest.cc
diff --git a/webrtc/modules/audio_processing/residual_echo_detector_complexity_unittest.cc b/webrtc/modules/audio_processing/residual_echo_detector_complexity_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..8f1f8543632486d2953134fdfa3ce8e0828185fd
--- /dev/null
+++ b/webrtc/modules/audio_processing/residual_echo_detector_complexity_unittest.cc
@@ -0,0 +1,168 @@
+/*
+ *  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/residual_echo_detector.h"
+#include "webrtc/modules/audio_processing/test/audio_buffer_tools.h"
+#include "webrtc/modules/audio_processing/test/performance_timer.h"
+#include "webrtc/modules/audio_processing/test/simulator_buffers.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;
+const int kSampleRate = AudioProcessing::kSampleRate48kHz;
+const int kNumberOfChannels = 1;
+
+std::string FormPerformanceMeasureString(const test::PerformanceTimer& timer) {
+  std::string s = std::to_string(timer.GetDurationAverage());
+  s += ", ";
+  s += std::to_string(timer.GetDurationStandardDeviation());
+  return s;
+}
+
+void RunStandaloneSubmodule() {
+  test::SimulatorBuffers buffers(
+      kSampleRate, kSampleRate, kSampleRate, kSampleRate, kNumberOfChannels,
+      kNumberOfChannels, kNumberOfChannels, kNumberOfChannels);
+  test::PerformanceTimer timer(kNumFramesToProcess);
+
+  ResidualEchoDetector echo_detector;
+  echo_detector.Initialize();
+
+  for (size_t frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
+    buffers.UpdateInputBuffers();
+
+    timer.StartTimer();
+    echo_detector.AnalyzeRenderAudio(rtc::ArrayView<const float>(
+        buffers.render_input_buffer->split_bands_const_f(0)[kBand0To8kHz],
+        buffers.render_input_buffer->num_frames_per_band()));
+    echo_detector.AnalyzeCaptureAudio(rtc::ArrayView<const float>(
+        buffers.capture_input_buffer->split_bands_const_f(0)[kBand0To8kHz],
+        buffers.capture_input_buffer->num_frames_per_band()));
+    timer.StopTimer();
+  }
+  webrtc::test::PrintResultMeanAndError(
+      "echo_detector_call_durations", "", "StandaloneEchoDetector",
+      FormPerformanceMeasureString(timer), "us", false);
+}
+
+void RunTogetherWithApm(std::string test_description,
+                        bool use_mobile_aec,
+                        bool include_default_apm_processing) {
+  test::SimulatorBuffers buffers(
+      kSampleRate, kSampleRate, kSampleRate, kSampleRate, kNumberOfChannels,
+      kNumberOfChannels, kNumberOfChannels, kNumberOfChannels);
+  test::PerformanceTimer render_timer(kNumFramesToProcess);
+  test::PerformanceTimer capture_timer(kNumFramesToProcess);
+  test::PerformanceTimer total_timer(kNumFramesToProcess);
+
+  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 = include_default_apm_processing;
+  apm_config.residual_echo_detector.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 stream_config(kSampleRate, kNumberOfChannels, 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], stream_config,
+                                  stream_config, &buffers.render_output[0]));
+
+    render_timer.StopTimer();
+
+    capture_timer.StartTimer();
+    ASSERT_EQ(AudioProcessing::kNoError, apm->set_stream_delay_ms(0));
+    if (include_default_apm_processing) {
+      apm->gain_control()->set_stream_analog_level(0);
+      if (!use_mobile_aec) {
+        apm->echo_cancellation()->set_stream_drift_samples(0);
+      }
+    }
+    ASSERT_EQ(AudioProcessing::kNoError,
+              apm->ProcessStream(&buffers.capture_input[0], stream_config,
+                                 stream_config, &buffers.capture_output[0]));
+
+    capture_timer.StopTimer();
+    total_timer.StopTimer();
+  }
+
+  webrtc::test::PrintResultMeanAndError(
+      "echo_detector_call_durations", "_render", test_description,
+      FormPerformanceMeasureString(render_timer), "us", false);
+  webrtc::test::PrintResultMeanAndError(
+      "echo_detector_call_durations", "_capture", test_description,
+      FormPerformanceMeasureString(capture_timer), "us", false);
+  webrtc::test::PrintResultMeanAndError(
+      "echo_detector_call_durations", "_total", test_description,
+      FormPerformanceMeasureString(total_timer), "us", false);
+}
+
+}  // namespace
+
+TEST(EchoDetectorPerformanceTest, StandaloneProcessing) {
+  RunStandaloneSubmodule();
+}
+
+TEST(EchoDetectorPerformanceTest, ProcessingViaApm) {
+  RunTogetherWithApm("SimpleEchoDetectorViaApm", false, false);
+}
+
+TEST(EchoDetectorPerformanceTest, InteractionWithDefaultApm) {
+  RunTogetherWithApm("EchoDetectorAndDefaultDesktopApm", false, true);
+  RunTogetherWithApm("EchoDetectorAndDefaultMobileApm", true, true);
+}
+
+}  // namespace webrtc