Adding metrics to AEC3.
This CL adds metrics reporting to AEC3.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2722453002
Cr-Commit-Position: refs/heads/master@{#16929}
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index bbcc8d0..84f25b5 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -38,6 +38,8 @@
"aec3/block_framer.h",
"aec3/block_processor.cc",
"aec3/block_processor.h",
+ "aec3/block_processor_metrics.cc",
+ "aec3/block_processor_metrics.h",
"aec3/cascaded_biquad_filter.cc",
"aec3/cascaded_biquad_filter.h",
"aec3/comfort_noise_generator.cc",
@@ -52,6 +54,8 @@
"aec3/echo_path_variability.h",
"aec3/echo_remover.cc",
"aec3/echo_remover.h",
+ "aec3/echo_remover_metrics.cc",
+ "aec3/echo_remover_metrics.h",
"aec3/erl_estimator.cc",
"aec3/erl_estimator.h",
"aec3/erle_estimator.cc",
@@ -75,6 +79,8 @@
"aec3/render_delay_buffer.h",
"aec3/render_delay_controller.cc",
"aec3/render_delay_controller.h",
+ "aec3/render_delay_controller_metrics.cc",
+ "aec3/render_delay_controller_metrics.h",
"aec3/render_signal_analyzer.cc",
"aec3/render_signal_analyzer.h",
"aec3/residual_echo_estimator.cc",
@@ -563,6 +569,7 @@
"aec3/aec3_fft_unittest.cc",
"aec3/aec_state_unittest.cc",
"aec3/block_framer_unittest.cc",
+ "aec3/block_processor_metrics_unittest.cc",
"aec3/block_processor_unittest.cc",
"aec3/cascaded_biquad_filter_unittest.cc",
"aec3/comfort_noise_generator_unittest.cc",
@@ -570,6 +577,7 @@
"aec3/echo_canceller3_unittest.cc",
"aec3/echo_path_delay_estimator_unittest.cc",
"aec3/echo_path_variability_unittest.cc",
+ "aec3/echo_remover_metrics_unittest.cc",
"aec3/echo_remover_unittest.cc",
"aec3/erl_estimator_unittest.cc",
"aec3/erle_estimator_unittest.cc",
@@ -582,6 +590,7 @@
"aec3/output_selector_unittest.cc",
"aec3/power_echo_model_unittest.cc",
"aec3/render_delay_buffer_unittest.cc",
+ "aec3/render_delay_controller_metrics_unittest.cc",
"aec3/render_delay_controller_unittest.cc",
"aec3/render_signal_analyzer_unittest.cc",
"aec3/residual_echo_estimator_unittest.cc",
diff --git a/webrtc/modules/audio_processing/aec3/aec3_common.h b/webrtc/modules/audio_processing/aec3/aec3_common.h
index 3a5e835..1d4a9fe 100644
--- a/webrtc/modules/audio_processing/aec3/aec3_common.h
+++ b/webrtc/modules/audio_processing/aec3/aec3_common.h
@@ -26,6 +26,11 @@
enum class Aec3Optimization { kNone, kSse2 };
+constexpr int kMetricsReportingIntervalBlocks = 10 * 250;
+constexpr int kMetricsComputationBlocks = 9;
+constexpr int kMetricsCollectionBlocks =
+ kMetricsReportingIntervalBlocks - kMetricsComputationBlocks;
+
constexpr size_t kFftLengthBy2 = 64;
constexpr size_t kFftLengthBy2Plus1 = kFftLengthBy2 + 1;
constexpr size_t kFftLengthBy2Minus1 = kFftLengthBy2 - 1;
diff --git a/webrtc/modules/audio_processing/aec3/block_processor.cc b/webrtc/modules/audio_processing/aec3/block_processor.cc
index 223b693..5055b3f 100644
--- a/webrtc/modules/audio_processing/aec3/block_processor.cc
+++ b/webrtc/modules/audio_processing/aec3/block_processor.cc
@@ -13,9 +13,9 @@
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/optional.h"
#include "webrtc/modules/audio_processing/aec3/aec3_common.h"
+#include "webrtc/modules/audio_processing/aec3/block_processor_metrics.h"
#include "webrtc/modules/audio_processing/aec3/echo_path_variability.h"
#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
-#include "webrtc/system_wrappers/include/logging.h"
namespace webrtc {
namespace {
@@ -44,6 +44,7 @@
std::unique_ptr<RenderDelayBuffer> render_buffer_;
std::unique_ptr<RenderDelayController> delay_controller_;
std::unique_ptr<EchoRemover> echo_remover_;
+ BlockProcessorMetrics metrics_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(BlockProcessorImpl);
};
@@ -88,8 +89,9 @@
delay_controller_->AlignmentHeadroomSamples(),
EchoPathVariability(echo_path_gain_change, render_delay_change),
capture_signal_saturation, render_block, capture_block);
+ metrics_.UpdateCapture(false);
} else {
- LOG(LS_INFO) << "AEC3 empty render buffer";
+ metrics_.UpdateCapture(true);
}
}
@@ -102,10 +104,10 @@
!delay_controller_->AnalyzeRender((*block)[0]);
const bool render_buffer_overrun = !render_buffer_->Insert(block);
if (delay_controller_overrun || render_buffer_overrun) {
- LOG(LS_INFO) << "AEC3 buffer overrrun";
+ metrics_.UpdateRender(true);
return false;
}
-
+ metrics_.UpdateRender(false);
return true;
}
diff --git a/webrtc/modules/audio_processing/aec3/block_processor_metrics.cc b/webrtc/modules/audio_processing/aec3/block_processor_metrics.cc
new file mode 100644
index 0000000..3eeafa2
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec3/block_processor_metrics.cc
@@ -0,0 +1,103 @@
+/*
+ * Copyright (c) 2017 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/aec3/block_processor_metrics.h"
+
+#include "webrtc/modules/audio_processing/aec3/aec3_common.h"
+#include "webrtc/system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+
+enum class RenderUnderrunCategory {
+ kNone,
+ kFew,
+ kSeveral,
+ kMany,
+ kConstant,
+ kNumCategories
+};
+
+enum class RenderOverrunCategory {
+ kNone,
+ kFew,
+ kSeveral,
+ kMany,
+ kConstant,
+ kNumCategories
+};
+
+} // namespace
+
+void BlockProcessorMetrics::UpdateCapture(bool underrun) {
+ ++capture_block_counter_;
+ if (underrun) {
+ ++render_buffer_underruns_;
+ }
+
+ if (capture_block_counter_ == kMetricsReportingIntervalBlocks) {
+ metrics_reported_ = true;
+
+ RenderUnderrunCategory underrun_category;
+ if (render_buffer_underruns_ == 0) {
+ underrun_category = RenderUnderrunCategory::kNone;
+ } else if (render_buffer_underruns_ > (capture_block_counter_ >> 1)) {
+ underrun_category = RenderUnderrunCategory::kConstant;
+ } else if (render_buffer_underruns_ > 100) {
+ underrun_category = RenderUnderrunCategory::kMany;
+ } else if (render_buffer_underruns_ > 10) {
+ underrun_category = RenderUnderrunCategory::kSeveral;
+ } else {
+ underrun_category = RenderUnderrunCategory::kFew;
+ }
+ RTC_HISTOGRAM_ENUMERATION(
+ "WebRTC.Audio.EchoCanceller.RenderUnderruns",
+ static_cast<int>(underrun_category),
+ static_cast<int>(RenderUnderrunCategory::kNumCategories));
+
+ RenderOverrunCategory overrun_category;
+ if (render_buffer_overruns_ == 0) {
+ overrun_category = RenderOverrunCategory::kNone;
+ } else if (render_buffer_overruns_ > (buffer_render_calls_ >> 1)) {
+ overrun_category = RenderOverrunCategory::kConstant;
+ } else if (render_buffer_overruns_ > 100) {
+ overrun_category = RenderOverrunCategory::kMany;
+ } else if (render_buffer_overruns_ > 10) {
+ overrun_category = RenderOverrunCategory::kSeveral;
+ } else {
+ overrun_category = RenderOverrunCategory::kFew;
+ }
+ RTC_HISTOGRAM_ENUMERATION(
+ "WebRTC.Audio.EchoCanceller.RenderOverruns",
+ static_cast<int>(overrun_category),
+ static_cast<int>(RenderOverrunCategory::kNumCategories));
+
+ ResetMetrics();
+ capture_block_counter_ = 0;
+ } else {
+ metrics_reported_ = false;
+ }
+}
+
+void BlockProcessorMetrics::UpdateRender(bool overrun) {
+ ++buffer_render_calls_;
+ if (overrun) {
+ ++render_buffer_overruns_;
+ }
+}
+
+void BlockProcessorMetrics::ResetMetrics() {
+ render_buffer_underruns_ = 0;
+ render_buffer_overruns_ = 0;
+ buffer_render_calls_ = 0;
+}
+
+} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/aec3/block_processor_metrics.h b/webrtc/modules/audio_processing/aec3/block_processor_metrics.h
new file mode 100644
index 0000000..6c278cd
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec3/block_processor_metrics.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright (c) 2017 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_METRICS_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_METRICS_H_
+
+#include "webrtc/base/constructormagic.h"
+
+namespace webrtc {
+
+// Handles the reporting of metrics for the block_processor.
+class BlockProcessorMetrics {
+ public:
+ BlockProcessorMetrics() = default;
+
+ // Updates the metric with new capture data.
+ void UpdateCapture(bool underrun);
+
+ // Updates the metric with new render data.
+ void UpdateRender(bool overrun);
+
+ // Returns true if the metrics have just been reported, otherwise false.
+ bool MetricsReported() { return metrics_reported_; }
+
+ private:
+ // Resets the metrics.
+ void ResetMetrics();
+
+ int capture_block_counter_ = 0;
+ bool metrics_reported_ = false;
+ int render_buffer_underruns_ = 0;
+ int render_buffer_overruns_ = 0;
+ int buffer_render_calls_ = 0;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(BlockProcessorMetrics);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_BLOCK_PROCESSOR_METRICS_H_
diff --git a/webrtc/modules/audio_processing/aec3/block_processor_metrics_unittest.cc b/webrtc/modules/audio_processing/aec3/block_processor_metrics_unittest.cc
new file mode 100644
index 0000000..8ab1048
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec3/block_processor_metrics_unittest.cc
@@ -0,0 +1,34 @@
+/*
+ * Copyright (c) 2017 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/aec3/aec3_common.h"
+#include "webrtc/modules/audio_processing/aec3/block_processor_metrics.h"
+
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+
+// Verify the general functionality of BlockProcessorMetrics.
+TEST(BlockProcessorMetrics, NormalUsage) {
+ BlockProcessorMetrics metrics;
+
+ for (int j = 0; j < 3; ++j) {
+ for (int k = 0; k < kMetricsReportingIntervalBlocks - 1; ++k) {
+ metrics.UpdateRender(false);
+ metrics.UpdateRender(false);
+ metrics.UpdateCapture(false);
+ EXPECT_FALSE(metrics.MetricsReported());
+ }
+ metrics.UpdateCapture(false);
+ EXPECT_TRUE(metrics.MetricsReported());
+ }
+}
+
+} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/aec3/echo_remover.cc b/webrtc/modules/audio_processing/aec3/echo_remover.cc
index 2d63286..a0e03fb 100644
--- a/webrtc/modules/audio_processing/aec3/echo_remover.cc
+++ b/webrtc/modules/audio_processing/aec3/echo_remover.cc
@@ -21,6 +21,7 @@
#include "webrtc/modules/audio_processing/aec3/aec_state.h"
#include "webrtc/modules/audio_processing/aec3/comfort_noise_generator.h"
#include "webrtc/modules/audio_processing/aec3/echo_path_variability.h"
+#include "webrtc/modules/audio_processing/aec3/echo_remover_metrics.h"
#include "webrtc/modules/audio_processing/aec3/fft_buffer.h"
#include "webrtc/modules/audio_processing/aec3/fft_data.h"
#include "webrtc/modules/audio_processing/aec3/output_selector.h"
@@ -90,6 +91,7 @@
bool echo_leakage_detected_ = false;
std::array<float, kBlockSize> x_old_;
AecState aec_state_;
+ EchoRemoverMetrics metrics_;
RTC_DISALLOW_COPY_AND_ASSIGN(EchoRemoverImpl);
};
@@ -209,6 +211,9 @@
doubletalk ? 0.001f : 0.0001f, &G);
suppression_filter_.ApplyGain(comfort_noise, high_band_comfort_noise, G, y);
+ // Update the metrics.
+ metrics_.Update(aec_state_, cng_.NoiseSpectrum(), G);
+
// Debug outputs for the purpose of development and analysis.
data_dumper_->DumpRaw("aec3_N2", cng_.NoiseSpectrum());
data_dumper_->DumpRaw("aec3_suppressor_gain", G);
diff --git a/webrtc/modules/audio_processing/aec3/echo_remover_metrics.cc b/webrtc/modules/audio_processing/aec3/echo_remover_metrics.cc
new file mode 100644
index 0000000..16a36f4
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec3/echo_remover_metrics.cc
@@ -0,0 +1,284 @@
+/*
+ * Copyright (c) 2017 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/aec3/echo_remover_metrics.h"
+
+#include <math.h>
+#include <algorithm>
+#include <numeric>
+
+#include "webrtc/system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+
+constexpr float kOneByMetricsCollectionBlocks = 1.f / kMetricsCollectionBlocks;
+
+} // namespace
+
+EchoRemoverMetrics::DbMetric::DbMetric() : DbMetric(0.f, 0.f, 0.f) {}
+EchoRemoverMetrics::DbMetric::DbMetric(float sum_value,
+ float floor_value,
+ float ceil_value)
+ : sum_value(sum_value), floor_value(floor_value), ceil_value(ceil_value) {}
+
+void EchoRemoverMetrics::DbMetric::Update(float value) {
+ sum_value += value;
+ floor_value = std::min(floor_value, value);
+ ceil_value = std::max(ceil_value, value);
+}
+
+EchoRemoverMetrics::EchoRemoverMetrics() {
+ ResetMetrics();
+}
+
+void EchoRemoverMetrics::ResetMetrics() {
+ erl_.fill(DbMetric(0.f, 10000.f, 0.000f));
+ erle_.fill(DbMetric(0.f, 0.f, 1000.f));
+ comfort_noise_.fill(DbMetric(0.f, 100000000.f, 0.f));
+ suppressor_gain_.fill(DbMetric(0.f, 1.f, 0.f));
+ active_render_count_ = 0;
+ saturated_capture_ = false;
+}
+
+void EchoRemoverMetrics::Update(
+ const AecState& aec_state,
+ const std::array<float, kFftLengthBy2Plus1>& comfort_noise_spectrum,
+ const std::array<float, kFftLengthBy2Plus1>& suppressor_gain) {
+ metrics_reported_ = false;
+ if (++block_counter_ <= kMetricsCollectionBlocks) {
+ aec3::UpdateDbMetric(aec_state.Erl(), &erl_);
+ aec3::UpdateDbMetric(aec_state.Erle(), &erle_);
+ aec3::UpdateDbMetric(comfort_noise_spectrum, &comfort_noise_);
+ aec3::UpdateDbMetric(suppressor_gain, &suppressor_gain_);
+ active_render_count_ += (aec_state.ActiveRender() ? 1 : 0);
+ saturated_capture_ = saturated_capture_ || aec_state.SaturatedCapture();
+ } else {
+ // Report the metrics over several frames in order to lower the impact of
+ // the logarithms involved on the computational complexity.
+ constexpr int kMetricsCollectionBlocksBy2 = kMetricsCollectionBlocks / 2;
+ constexpr float kComfortNoiseScaling = 1.f / (kBlockSize * kBlockSize);
+ switch (block_counter_) {
+ case kMetricsCollectionBlocks + 1:
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.ErleBand0.Average",
+ aec3::TransformDbMetricForReporting(true, 0.f, 19.f, 0.f,
+ kOneByMetricsCollectionBlocks,
+ erle_[0].sum_value),
+ 0, 19, 20);
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.ErleBand0.Max",
+ aec3::TransformDbMetricForReporting(true, 0.f, 19.f, 0.f, 1.f,
+ erle_[0].ceil_value),
+ 0, 19, 20);
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.ErleBand0.Min",
+ aec3::TransformDbMetricForReporting(true, 0.f, 19.f, 0.f, 1.f,
+ erle_[0].floor_value),
+ 0, 19, 20);
+ break;
+ case kMetricsCollectionBlocks + 2:
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.ErleBand1.Average",
+ aec3::TransformDbMetricForReporting(true, 0.f, 19.f, 0.f,
+ kOneByMetricsCollectionBlocks,
+ erle_[1].sum_value),
+ 0, 19, 20);
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.ErleBand1.Max",
+ aec3::TransformDbMetricForReporting(true, 0.f, 19.f, 0.f, 1.f,
+ erle_[1].ceil_value),
+ 0, 19, 20);
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.ErleBand1.Min",
+ aec3::TransformDbMetricForReporting(true, 0.f, 19.f, 0.f, 1.f,
+ erle_[1].floor_value),
+ 0, 19, 20);
+ break;
+ case kMetricsCollectionBlocks + 3:
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.ErlBand0.Average",
+ aec3::TransformDbMetricForReporting(true, 0.f, 59.f, 30.f,
+ kOneByMetricsCollectionBlocks,
+ erl_[0].sum_value),
+ 0, 59, 30);
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.ErlBand0.Max",
+ aec3::TransformDbMetricForReporting(true, 0.f, 59.f, 30.f, 1.f,
+ erl_[0].ceil_value),
+ 0, 59, 30);
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.ErlBand0.Min",
+ aec3::TransformDbMetricForReporting(true, 0.f, 59.f, 30.f, 1.f,
+ erl_[0].floor_value),
+ 0, 59, 30);
+ break;
+ case kMetricsCollectionBlocks + 4:
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.ErlBand1.Average",
+ aec3::TransformDbMetricForReporting(true, 0.f, 59.f, 30.f,
+ kOneByMetricsCollectionBlocks,
+ erl_[1].sum_value),
+ 0, 59, 30);
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.ErlBand1.Max",
+ aec3::TransformDbMetricForReporting(true, 0.f, 59.f, 30.f, 1.f,
+ erl_[1].ceil_value),
+ 0, 59, 30);
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.ErlBand1.Min",
+ aec3::TransformDbMetricForReporting(true, 0.f, 59.f, 30.f, 1.f,
+ erl_[1].floor_value),
+ 0, 59, 30);
+ break;
+ case kMetricsCollectionBlocks + 5:
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.ComfortNoiseBand0.Average",
+ aec3::TransformDbMetricForReporting(
+ true, 0.f, 89.f, -90.3f,
+ kComfortNoiseScaling * kOneByMetricsCollectionBlocks,
+ comfort_noise_[0].sum_value),
+ 0, 89, 45);
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.ComfortNoiseBand0.Max",
+ aec3::TransformDbMetricForReporting(true, 0.f, 89.f, -90.3f,
+ kComfortNoiseScaling,
+ comfort_noise_[0].ceil_value),
+ 0, 89, 45);
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.ComfortNoiseBand0.Min",
+ aec3::TransformDbMetricForReporting(true, 0.f, 89.f, -90.3f,
+ kComfortNoiseScaling,
+ comfort_noise_[0].floor_value),
+ 0, 89, 45);
+ break;
+ case kMetricsCollectionBlocks + 6:
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.ComfortNoiseBand1.Average",
+ aec3::TransformDbMetricForReporting(
+ true, 0.f, 89.f, -90.3f,
+ kComfortNoiseScaling * kOneByMetricsCollectionBlocks,
+ comfort_noise_[1].sum_value),
+ 0, 89, 45);
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.ComfortNoiseBand1.Max",
+ aec3::TransformDbMetricForReporting(true, 0.f, 89.f, -90.3f,
+ kComfortNoiseScaling,
+ comfort_noise_[1].ceil_value),
+ 0, 89, 45);
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.ComfortNoiseBand1.Min",
+ aec3::TransformDbMetricForReporting(true, 0.f, 89.f, -90.3f,
+ kComfortNoiseScaling,
+ comfort_noise_[1].floor_value),
+ 0, 89, 45);
+ break;
+ case kMetricsCollectionBlocks + 7:
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.SuppressorGainBand0.Average",
+ aec3::TransformDbMetricForReporting(true, 0.f, 59.f, 0.f,
+ kOneByMetricsCollectionBlocks,
+ suppressor_gain_[0].sum_value),
+ 0, 59, 30);
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.SuppressorGainBand0.Max",
+ aec3::TransformDbMetricForReporting(true, 0.f, 59.f, 0.f, 1.f,
+ suppressor_gain_[0].ceil_value),
+ 0, 59, 30);
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.SuppressorGainBand0.Min",
+ aec3::TransformDbMetricForReporting(
+ true, 0.f, 59.f, 0.f, 1.f, suppressor_gain_[0].floor_value),
+ 0, 59, 30);
+ break;
+ case kMetricsCollectionBlocks + 8:
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.SuppressorGainBand1.Average",
+ aec3::TransformDbMetricForReporting(true, 0.f, 59.f, 0.f,
+ kOneByMetricsCollectionBlocks,
+ suppressor_gain_[1].sum_value),
+ 0, 59, 30);
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.SuppressorGainBand1.Max",
+ aec3::TransformDbMetricForReporting(true, 0.f, 59.f, 0.f, 1.f,
+ suppressor_gain_[1].ceil_value),
+ 0, 59, 30);
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.SuppressorGainBand1.Min",
+ aec3::TransformDbMetricForReporting(
+ true, 0.f, 59.f, 0.f, 1.f, suppressor_gain_[1].floor_value),
+ 0, 59, 30);
+ break;
+ case kMetricsCollectionBlocks + 9:
+ RTC_HISTOGRAM_BOOLEAN(
+ "WebRTC.Audio.EchoCanceller.UsableLinearEstimate",
+ static_cast<int>(aec_state.UsableLinearEstimate() ? 1 : 0));
+ RTC_HISTOGRAM_BOOLEAN(
+ "WebRTC.Audio.EchoCanceller.ModelBasedAecFeasible",
+ static_cast<int>(aec_state.ModelBasedAecFeasible() ? 1 : 0));
+ RTC_HISTOGRAM_BOOLEAN(
+ "WebRTC.Audio.EchoCanceller.ActiveRender",
+ static_cast<int>(
+ active_render_count_ > kMetricsCollectionBlocksBy2 ? 1 : 0));
+ RTC_HISTOGRAM_COUNTS_LINEAR(
+ "WebRTC.Audio.EchoCanceller.FilterDelay",
+ aec_state.FilterDelay() ? *aec_state.FilterDelay() + 1 : 0, 0, 30,
+ 31);
+ RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.EchoCanceller.CaptureSaturation",
+ static_cast<int>(saturated_capture_ ? 1 : 0));
+ metrics_reported_ = true;
+ RTC_DCHECK_EQ(kMetricsReportingIntervalBlocks, block_counter_);
+ block_counter_ = 0;
+ ResetMetrics();
+ break;
+ default:
+ RTC_NOTREACHED();
+ break;
+ }
+ }
+}
+
+namespace aec3 {
+
+void UpdateDbMetric(const std::array<float, kFftLengthBy2Plus1>& value,
+ std::array<EchoRemoverMetrics::DbMetric, 2>* statistic) {
+ RTC_DCHECK(statistic);
+ // Truncation is intended in the band width computation.
+ constexpr int kNumBands = 2;
+ constexpr int kBandWidth = 65 / kNumBands;
+ constexpr float kOneByBandWidth = 1.f / kBandWidth;
+ RTC_DCHECK_EQ(kNumBands, statistic->size());
+ RTC_DCHECK_EQ(65, value.size());
+ for (size_t k = 0; k < statistic->size(); ++k) {
+ float average_band =
+ std::accumulate(value.begin() + kBandWidth * k,
+ value.begin() + kBandWidth * (k + 1), 0.f) *
+ kOneByBandWidth;
+ (*statistic)[k].Update(average_band);
+ }
+}
+
+int TransformDbMetricForReporting(bool negate,
+ float min_value,
+ float max_value,
+ float offset,
+ float scaling,
+ float value) {
+ float new_value = 10.f * log10(value * scaling + 1e-10f) + offset;
+ if (negate) {
+ new_value = -new_value;
+ }
+ return static_cast<int>(std::max(min_value, std::min(max_value, new_value)));
+}
+
+} // namespace aec3
+
+} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/aec3/echo_remover_metrics.h b/webrtc/modules/audio_processing/aec3/echo_remover_metrics.h
new file mode 100644
index 0000000..aaca159
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec3/echo_remover_metrics.h
@@ -0,0 +1,77 @@
+/*
+ * Copyright (c) 2017 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_METRICS_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_METRICS_H_
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_processing/aec3/aec_state.h"
+
+namespace webrtc {
+
+// Handles the reporting of metrics for the echo remover.
+class EchoRemoverMetrics {
+ public:
+ struct DbMetric {
+ DbMetric();
+ DbMetric(float sum_value, float floor_value, float ceil_value);
+ void Update(float value);
+ float sum_value;
+ float floor_value;
+ float ceil_value;
+ };
+
+ EchoRemoverMetrics();
+
+ // Updates the metric with new data.
+ void Update(
+ const AecState& aec_state,
+ const std::array<float, kFftLengthBy2Plus1>& comfort_noise_spectrum,
+ const std::array<float, kFftLengthBy2Plus1>& suppressor_gain);
+
+ // Returns true if the metrics have just been reported, otherwise false.
+ bool MetricsReported() { return metrics_reported_; }
+
+ private:
+ // Resets the metrics.
+ void ResetMetrics();
+
+ int block_counter_ = 0;
+ std::array<DbMetric, 2> erl_;
+ std::array<DbMetric, 2> erle_;
+ std::array<DbMetric, 2> comfort_noise_;
+ std::array<DbMetric, 2> suppressor_gain_;
+ int active_render_count_ = 0;
+ bool saturated_capture_ = false;
+ bool metrics_reported_ = false;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(EchoRemoverMetrics);
+};
+
+namespace aec3 {
+
+// Updates a banded metric of type DbMetric with the values in the supplied
+// array.
+void UpdateDbMetric(const std::array<float, kFftLengthBy2Plus1>& value,
+ std::array<EchoRemoverMetrics::DbMetric, 2>* statistic);
+
+// Transforms a DbMetric from the linear domain into the logarithmic domain.
+int TransformDbMetricForReporting(bool negate,
+ float min_value,
+ float max_value,
+ float offset,
+ float scaling,
+ float value);
+
+} // namespace aec3
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_ECHO_REMOVER_METRICS_H_
diff --git a/webrtc/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc b/webrtc/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc
new file mode 100644
index 0000000..8030f81
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc
@@ -0,0 +1,144 @@
+/*
+ * Copyright (c) 2017 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/aec3/echo_remover_metrics.h"
+
+#include <math.h>
+
+#include "webrtc/modules/audio_processing/aec3/aec_state.h"
+#include "webrtc/modules/audio_processing/aec3/aec3_fft.h"
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+
+// Verifies the check for non-null input.
+TEST(UpdateDbMetric, NullValue) {
+ std::array<float, kFftLengthBy2Plus1> value;
+ value.fill(0.f);
+ EXPECT_DEATH(aec3::UpdateDbMetric(value, nullptr), "");
+}
+
+#endif
+
+// Verifies the updating functionality of UpdateDbMetric.
+TEST(UpdateDbMetric, Updating) {
+ std::array<float, kFftLengthBy2Plus1> value;
+ std::array<EchoRemoverMetrics::DbMetric, 2> statistic;
+ statistic.fill(EchoRemoverMetrics::DbMetric(0.f, 100.f, -100.f));
+ constexpr float kValue0 = 10.f;
+ constexpr float kValue1 = 20.f;
+ std::fill(value.begin(), value.begin() + 32, kValue0);
+ std::fill(value.begin() + 32, value.begin() + 64, kValue1);
+
+ aec3::UpdateDbMetric(value, &statistic);
+ EXPECT_FLOAT_EQ(kValue0, statistic[0].sum_value);
+ EXPECT_FLOAT_EQ(kValue0, statistic[0].ceil_value);
+ EXPECT_FLOAT_EQ(kValue0, statistic[0].floor_value);
+ EXPECT_FLOAT_EQ(kValue1, statistic[1].sum_value);
+ EXPECT_FLOAT_EQ(kValue1, statistic[1].ceil_value);
+ EXPECT_FLOAT_EQ(kValue1, statistic[1].floor_value);
+
+ aec3::UpdateDbMetric(value, &statistic);
+ EXPECT_FLOAT_EQ(2.f * kValue0, statistic[0].sum_value);
+ EXPECT_FLOAT_EQ(kValue0, statistic[0].ceil_value);
+ EXPECT_FLOAT_EQ(kValue0, statistic[0].floor_value);
+ EXPECT_FLOAT_EQ(2.f * kValue1, statistic[1].sum_value);
+ EXPECT_FLOAT_EQ(kValue1, statistic[1].ceil_value);
+ EXPECT_FLOAT_EQ(kValue1, statistic[1].floor_value);
+}
+
+// Verifies that the TransformDbMetricForReporting method produces the desired
+// output for values for dBFS.
+TEST(TransformDbMetricForReporting, DbFsScaling) {
+ std::array<float, kBlockSize> x;
+ FftData X;
+ std::array<float, kFftLengthBy2Plus1> X2;
+ Aec3Fft fft;
+ x.fill(1000.f);
+ fft.ZeroPaddedFft(x, &X);
+ X.Spectrum(Aec3Optimization::kNone, &X2);
+
+ float offset = -10.f * log10(32768.f * 32768.f);
+ EXPECT_NEAR(offset, -90.3f, 0.1f);
+ EXPECT_EQ(
+ static_cast<int>(30.3f),
+ aec3::TransformDbMetricForReporting(
+ true, 0.f, 90.f, offset, 1.f / (kBlockSize * kBlockSize), X2[0]));
+}
+
+// Verifies that the TransformDbMetricForReporting method is able to properly
+// limit the output.
+TEST(TransformDbMetricForReporting, Limits) {
+ EXPECT_EQ(
+ 0,
+ aec3::TransformDbMetricForReporting(false, 0.f, 10.f, 0.f, 1.f, 0.001f));
+ EXPECT_EQ(
+ 10,
+ aec3::TransformDbMetricForReporting(false, 0.f, 10.f, 0.f, 1.f, 100.f));
+}
+
+// Verifies that the TransformDbMetricForReporting method is able to properly
+// negate output.
+TEST(TransformDbMetricForReporting, Negate) {
+ EXPECT_EQ(
+ 10,
+ aec3::TransformDbMetricForReporting(true, -20.f, 20.f, 0.f, 1.f, 0.1f));
+ EXPECT_EQ(
+ -10,
+ aec3::TransformDbMetricForReporting(true, -20.f, 20.f, 0.f, 1.f, 10.f));
+}
+
+// Verify the Update functionality of DbMetric.
+TEST(DbMetric, Update) {
+ EchoRemoverMetrics::DbMetric metric(0.f, 20.f, -20.f);
+ constexpr int kNumValues = 100;
+ constexpr float kValue = 10.f;
+ for (int k = 0; k < kNumValues; ++k) {
+ metric.Update(kValue);
+ }
+ EXPECT_FLOAT_EQ(kValue * kNumValues, metric.sum_value);
+ EXPECT_FLOAT_EQ(kValue, metric.ceil_value);
+ EXPECT_FLOAT_EQ(kValue, metric.floor_value);
+}
+
+// Verify the constructor functionality of DbMetric.
+TEST(DbMetric, Constructor) {
+ EchoRemoverMetrics::DbMetric metric;
+ EXPECT_FLOAT_EQ(0.f, metric.sum_value);
+ EXPECT_FLOAT_EQ(0.f, metric.ceil_value);
+ EXPECT_FLOAT_EQ(0.f, metric.floor_value);
+
+ metric = EchoRemoverMetrics::DbMetric(1.f, 2.f, 3.f);
+ EXPECT_FLOAT_EQ(1.f, metric.sum_value);
+ EXPECT_FLOAT_EQ(2.f, metric.floor_value);
+ EXPECT_FLOAT_EQ(3.f, metric.ceil_value);
+}
+
+// Verify the general functionality of EchoRemoverMetrics.
+TEST(EchoRemoverMetrics, NormalUsage) {
+ EchoRemoverMetrics metrics;
+ AecState aec_state;
+ std::array<float, kFftLengthBy2Plus1> comfort_noise_spectrum;
+ std::array<float, kFftLengthBy2Plus1> suppressor_gain;
+ comfort_noise_spectrum.fill(10.f);
+ suppressor_gain.fill(1.f);
+ for (int j = 0; j < 3; ++j) {
+ for (int k = 0; k < kMetricsReportingIntervalBlocks - 1; ++k) {
+ metrics.Update(aec_state, comfort_noise_spectrum, suppressor_gain);
+ EXPECT_FALSE(metrics.MetricsReported());
+ }
+ metrics.Update(aec_state, comfort_noise_spectrum, suppressor_gain);
+ EXPECT_TRUE(metrics.MetricsReported());
+ }
+}
+
+} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/aec3/render_delay_controller.cc b/webrtc/modules/audio_processing/aec3/render_delay_controller.cc
index 834e1e7..195d8cd 100644
--- a/webrtc/modules/audio_processing/aec3/render_delay_controller.cc
+++ b/webrtc/modules/audio_processing/aec3/render_delay_controller.cc
@@ -18,7 +18,7 @@
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/audio_processing/aec3/aec3_common.h"
#include "webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h"
-#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h"
namespace webrtc {
@@ -81,6 +81,7 @@
int echo_path_delay_samples_ = 0;
size_t align_call_counter_ = 0;
rtc::Optional<size_t> headroom_samples_;
+ RenderDelayControllerMetrics metrics_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderDelayControllerImpl);
};
@@ -151,6 +152,8 @@
headroom_samples_ = rtc::Optional<size_t>();
}
+ metrics_.Update(echo_path_delay_samples, delay_);
+
data_dumper_->DumpRaw("aec3_render_delay_controller_delay", 1,
&echo_path_delay_samples_);
data_dumper_->DumpRaw("aec3_render_delay_controller_buffer_delay", delay_);
diff --git a/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc b/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc
new file mode 100644
index 0000000..b84b916
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc
@@ -0,0 +1,117 @@
+/*
+ * Copyright (c) 2017 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/aec3/render_delay_controller_metrics.h"
+
+#include <algorithm>
+
+#include "webrtc/modules/audio_processing/aec3/aec3_common.h"
+#include "webrtc/system_wrappers/include/metrics.h"
+
+namespace webrtc {
+
+namespace {
+
+enum class DelayReliabilityCategory {
+ kNone,
+ kPoor,
+ kMedium,
+ kGood,
+ kExcellent,
+ kNumCategories
+};
+enum class DelayChangesCategory {
+ kNone,
+ kFew,
+ kSeveral,
+ kMany,
+ kConstant,
+ kNumCategories
+};
+
+} // namespace
+
+void RenderDelayControllerMetrics::Update(rtc::Optional<size_t> delay_samples,
+ size_t buffer_delay_blocks) {
+ ++call_counter_;
+
+ if (!initial_update) {
+ if (delay_samples) {
+ ++reliable_delay_estimate_counter_;
+ size_t delay_blocks = (*delay_samples) / kBlockSize;
+
+ if (delay_blocks != delay_blocks_) {
+ ++delay_change_counter_;
+ delay_blocks_ = delay_blocks;
+ }
+ }
+ } else if (++initial_call_counter_ == 5 * 250) {
+ initial_update = false;
+ }
+
+ if (call_counter_ == kMetricsReportingIntervalBlocks) {
+ int value_to_report = static_cast<int>(delay_blocks_);
+ value_to_report = std::min(124, value_to_report);
+ RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.EchoPathDelay",
+ value_to_report, 0, 124, 125);
+
+ value_to_report = static_cast<int>(buffer_delay_blocks);
+ value_to_report = std::min(124, value_to_report);
+ RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.BufferDelay",
+ value_to_report, 0, 124, 125);
+
+ DelayReliabilityCategory delay_reliability;
+ if (reliable_delay_estimate_counter_ == 0) {
+ delay_reliability = DelayReliabilityCategory::kNone;
+ } else if (reliable_delay_estimate_counter_ > (call_counter_ >> 1)) {
+ delay_reliability = DelayReliabilityCategory::kExcellent;
+ } else if (reliable_delay_estimate_counter_ > 100) {
+ delay_reliability = DelayReliabilityCategory::kGood;
+ } else if (reliable_delay_estimate_counter_ > 10) {
+ delay_reliability = DelayReliabilityCategory::kMedium;
+ } else {
+ delay_reliability = DelayReliabilityCategory::kPoor;
+ }
+ RTC_HISTOGRAM_ENUMERATION(
+ "WebRTC.Audio.EchoCanceller.ReliableDelayEstimates",
+ static_cast<int>(delay_reliability),
+ static_cast<int>(DelayReliabilityCategory::kNumCategories));
+
+ DelayChangesCategory delay_changes;
+ if (delay_change_counter_ == 0) {
+ delay_changes = DelayChangesCategory::kNone;
+ } else if (delay_change_counter_ > 10) {
+ delay_changes = DelayChangesCategory::kConstant;
+ } else if (delay_change_counter_ > 5) {
+ delay_changes = DelayChangesCategory::kMany;
+ } else if (delay_change_counter_ > 2) {
+ delay_changes = DelayChangesCategory::kSeveral;
+ } else {
+ delay_changes = DelayChangesCategory::kFew;
+ }
+ RTC_HISTOGRAM_ENUMERATION(
+ "WebRTC.Audio.EchoCanceller.DelayChanges",
+ static_cast<int>(delay_changes),
+ static_cast<int>(DelayChangesCategory::kNumCategories));
+
+ metrics_reported_ = true;
+ call_counter_ = 0;
+ ResetMetrics();
+ } else {
+ metrics_reported_ = false;
+ }
+}
+
+void RenderDelayControllerMetrics::ResetMetrics() {
+ delay_change_counter_ = 0;
+ reliable_delay_estimate_counter_ = 0;
+}
+
+} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h b/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h
new file mode 100644
index 0000000..9b9ea86
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright (c) 2017 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_METRICS_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_METRICS_H_
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/optional.h"
+
+namespace webrtc {
+
+// Handles the reporting of metrics for the render delay controller.
+class RenderDelayControllerMetrics {
+ public:
+ RenderDelayControllerMetrics() = default;
+
+ // Updates the metric with new data.
+ void Update(rtc::Optional<size_t> delay_samples, size_t buffer_delay_blocks);
+
+ // Returns true if the metrics have just been reported, otherwise false.
+ bool MetricsReported() { return metrics_reported_; }
+
+ private:
+ // Resets the metrics.
+ void ResetMetrics();
+
+ size_t delay_blocks_ = 0;
+ int reliable_delay_estimate_counter_ = 0;
+ int delay_change_counter_ = 0;
+ int call_counter_ = 0;
+ int initial_call_counter_ = 0;
+ bool metrics_reported_ = false;
+ bool initial_update = true;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(RenderDelayControllerMetrics);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_METRICS_H_
diff --git a/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc b/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
new file mode 100644
index 0000000..2551e8a
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
@@ -0,0 +1,33 @@
+/*
+ * Copyright (c) 2017 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/base/optional.h"
+#include "webrtc/modules/audio_processing/aec3/aec3_common.h"
+#include "webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h"
+
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+
+// Verify the general functionality of RenderDelayControllerMetrics.
+TEST(RenderDelayControllerMetrics, NormalUsage) {
+ RenderDelayControllerMetrics metrics;
+
+ for (int j = 0; j < 3; ++j) {
+ for (int k = 0; k < kMetricsReportingIntervalBlocks - 1; ++k) {
+ metrics.Update(rtc::Optional<size_t>(), 0);
+ EXPECT_FALSE(metrics.MetricsReported());
+ }
+ metrics.Update(rtc::Optional<size_t>(), 0);
+ EXPECT_TRUE(metrics.MetricsReported());
+ }
+}
+
+} // namespace webrtc