diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index bbcc8d018f..84f25b5e8e 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -38,6 +38,8 @@ rtc_static_library("audio_processing") {
     "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 @@ rtc_static_library("audio_processing") {
     "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 @@ rtc_static_library("audio_processing") {
     "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 @@ if (rtc_include_tests) {
         "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 @@ if (rtc_include_tests) {
         "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 @@ if (rtc_include_tests) {
         "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 3a5e835e21..1d4a9feba1 100644
--- a/webrtc/modules/audio_processing/aec3/aec3_common.h
+++ b/webrtc/modules/audio_processing/aec3/aec3_common.h
@@ -26,6 +26,11 @@ namespace webrtc {
 
 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 223b693813..5055b3f77d 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 @@ class BlockProcessorImpl final : public BlockProcessor {
   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 @@ void BlockProcessorImpl::ProcessCapture(
         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 @@ bool BlockProcessorImpl::BufferRender(std::vector<std::vector<float>>* block) {
       !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 0000000000..3eeafa2e45
--- /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 0000000000..6c278cd793
--- /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 0000000000..8ab1048556
--- /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 2d6328655c..a0e03fbeab 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 @@ class EchoRemoverImpl final : public EchoRemover {
   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 @@ void EchoRemoverImpl::ProcessBlock(
                             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 0000000000..16a36f4fb9
--- /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 0000000000..aaca1596d7
--- /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 0000000000..8030f812f6
--- /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 834e1e7641..195d8cd161 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 @@ class RenderDelayControllerImpl final : public RenderDelayController {
   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 @@ size_t RenderDelayControllerImpl::GetDelay(
     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 0000000000..b84b9160a6
--- /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 0000000000..9b9ea86437
--- /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 0000000000..2551e8a704
--- /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