diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index b36e76b1cc..b0b425329d 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -550,6 +550,7 @@ if (rtc_include_tests) {
       "aec_dump:mock_aec_dump_unittests",
       "agc2:adaptive_digital_unittests",
       "agc2:fixed_digital_unittests",
+      "agc2:noise_estimator_unittests",
       "test/conversational_speech:unittest",
       "vad:vad_unittests",
       "//testing/gtest",
diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn
index ca71b9278f..f9051257e4 100644
--- a/modules/audio_processing/agc2/BUILD.gn
+++ b/modules/audio_processing/agc2/BUILD.gn
@@ -23,8 +23,6 @@ rtc_source_set("adaptive_digital") {
     "adaptive_digital_gain_applier.h",
     "adaptive_mode_level_estimator.cc",
     "adaptive_mode_level_estimator.h",
-    "noise_level_estimator.cc",
-    "noise_level_estimator.h",
     "saturation_protector.cc",
     "saturation_protector.h",
   ]
@@ -33,6 +31,7 @@ rtc_source_set("adaptive_digital") {
 
   deps = [
     ":common",
+    ":noise_level_estimator",
     "..:aec_core",
     "..:apm_logging",
     "..:audio_frame_view",
@@ -83,6 +82,32 @@ rtc_source_set("common") {
   ]
 }
 
+rtc_source_set("noise_level_estimator") {
+  sources = [
+    "biquad_filter.cc",
+    "biquad_filter.h",
+    "down_sampler.cc",
+    "down_sampler.h",
+    "noise_level_estimator.cc",
+    "noise_level_estimator.h",
+    "noise_spectrum_estimator.cc",
+    "noise_spectrum_estimator.h",
+    "signal_classifier.cc",
+    "signal_classifier.h",
+  ]
+  deps = [
+    "..:aec_core",
+    "..:apm_logging",
+    "..:audio_frame_view",
+    "../../../api:array_view",
+    "../../../common_audio",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:macromagic",
+  ]
+
+  configs += [ "..:apm_debug_dump" ]
+}
+
 rtc_source_set("test_utils") {
   testonly = true
   visibility = [ ":*" ]
@@ -151,3 +176,23 @@ rtc_source_set("adaptive_digital_unittests") {
     "../vad:vad_with_level",
   ]
 }
+
+rtc_source_set("noise_estimator_unittests") {
+  testonly = true
+  configs += [ "..:apm_debug_dump" ]
+
+  sources = [
+    "noise_level_estimator_unittest.cc",
+    "signal_classifier_unittest.cc",
+  ]
+  deps = [
+    ":noise_level_estimator",
+    ":test_utils",
+    "..:apm_logging",
+    "..:audio_frame_view",
+    "../../../api:array_view",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:rtc_base_approved",
+    "../../../rtc_base:rtc_base_tests_utils",
+  ]
+}
diff --git a/modules/audio_processing/agc2/adaptive_agc.cc b/modules/audio_processing/agc2/adaptive_agc.cc
index dff38fb44b..0de27a41a6 100644
--- a/modules/audio_processing/agc2/adaptive_agc.cc
+++ b/modules/audio_processing/agc2/adaptive_agc.cc
@@ -22,7 +22,8 @@ namespace webrtc {
 AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper)
     : speech_level_estimator_(apm_data_dumper),
       gain_applier_(apm_data_dumper),
-      apm_data_dumper_(apm_data_dumper) {
+      apm_data_dumper_(apm_data_dumper),
+      noise_level_estimator_(apm_data_dumper) {
   RTC_DCHECK(apm_data_dumper);
 }
 
diff --git a/modules/audio_processing/agc2/agc2_testing_common.h b/modules/audio_processing/agc2/agc2_testing_common.h
index a176282ede..8c4f400a98 100644
--- a/modules/audio_processing/agc2/agc2_testing_common.h
+++ b/modules/audio_processing/agc2/agc2_testing_common.h
@@ -11,9 +11,13 @@
 #ifndef MODULES_AUDIO_PROCESSING_AGC2_AGC2_TESTING_COMMON_H_
 #define MODULES_AUDIO_PROCESSING_AGC2_AGC2_TESTING_COMMON_H_
 
+#include <math.h>
+
+#include <limits>
 #include <vector>
 
 #include "rtc_base/basictypes.h"
+#include "rtc_base/checks.h"
 
 namespace webrtc {
 
@@ -26,8 +30,49 @@ constexpr float kDecayMs = 500.f;
 constexpr float kLimiterMaxInputLevelDbFs = 1.f;
 constexpr float kLimiterKneeSmoothnessDb = 1.f;
 constexpr float kLimiterCompressionRatio = 5.f;
+constexpr float kPi = 3.1415926536f;
 
 std::vector<double> LinSpace(const double l, const double r, size_t num_points);
+
+class SineGenerator {
+ public:
+  SineGenerator(float frequency, int rate)
+      : frequency_(frequency), rate_(rate) {}
+  float operator()() {
+    x_radians_ += frequency_ / rate_ * 2 * kPi;
+    if (x_radians_ > 2 * kPi) {
+      x_radians_ -= 2 * kPi;
+    }
+    return 1000.f * sinf(x_radians_);
+  }
+
+ private:
+  float frequency_;
+  int rate_;
+  float x_radians_ = 0.f;
+};
+
+class PulseGenerator {
+ public:
+  PulseGenerator(float frequency, int rate)
+      : samples_period_(
+            static_cast<int>(static_cast<float>(rate) / frequency)) {
+    RTC_DCHECK_GT(rate, frequency);
+  }
+  float operator()() {
+    sample_counter_++;
+    if (sample_counter_ >= samples_period_) {
+      sample_counter_ -= samples_period_;
+    }
+    return static_cast<float>(
+        sample_counter_ == 0 ? std::numeric_limits<int16_t>::max() : 10.f);
+  }
+
+ private:
+  int samples_period_;
+  int sample_counter_ = 0;
+};
+
 }  // namespace test
 }  // namespace webrtc
 
diff --git a/modules/audio_processing/agc2/biquad_filter.cc b/modules/audio_processing/agc2/biquad_filter.cc
new file mode 100644
index 0000000000..c15c6449ad
--- /dev/null
+++ b/modules/audio_processing/agc2/biquad_filter.cc
@@ -0,0 +1,35 @@
+/*
+ *  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 "modules/audio_processing/agc2/biquad_filter.h"
+
+namespace webrtc {
+
+// This method applies a biquad filter to an input signal x to produce an
+// output signal y. The biquad coefficients are specified at the construction
+// of the object.
+void BiQuadFilter::Process(rtc::ArrayView<const float> x,
+                           rtc::ArrayView<float> y) {
+  for (size_t k = 0; k < x.size(); ++k) {
+    // Use temporary variable for x[k] to allow in-place function call
+    // (that x and y refer to the same array).
+    const float tmp = x[k];
+    y[k] = coefficients_.b[0] * tmp + coefficients_.b[1] * biquad_state_.b[0] +
+           coefficients_.b[2] * biquad_state_.b[1] -
+           coefficients_.a[0] * biquad_state_.a[0] -
+           coefficients_.a[1] * biquad_state_.a[1];
+    biquad_state_.b[1] = biquad_state_.b[0];
+    biquad_state_.b[0] = tmp;
+    biquad_state_.a[1] = biquad_state_.a[0];
+    biquad_state_.a[0] = y[k];
+  }
+}
+
+}  // namespace webrtc
diff --git a/modules/audio_processing/agc2/biquad_filter.h b/modules/audio_processing/agc2/biquad_filter.h
new file mode 100644
index 0000000000..4fd5e2e392
--- /dev/null
+++ b/modules/audio_processing/agc2/biquad_filter.h
@@ -0,0 +1,56 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_BIQUAD_FILTER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_BIQUAD_FILTER_H_
+
+#include "api/array_view.h"
+#include "rtc_base/arraysize.h"
+#include "rtc_base/constructormagic.h"
+
+namespace webrtc {
+
+class BiQuadFilter {
+ public:
+  struct BiQuadCoefficients {
+    float b[3];
+    float a[2];
+  };
+
+  BiQuadFilter() = default;
+
+  void Initialize(const BiQuadCoefficients& coefficients) {
+    coefficients_ = coefficients;
+  }
+
+  // Produces a filtered output y of the input x. Both x and y need to
+  // have the same length.
+  void Process(rtc::ArrayView<const float> x, rtc::ArrayView<float> y);
+
+ private:
+  struct BiQuadState {
+    BiQuadState() {
+      std::fill(b, b + arraysize(b), 0.f);
+      std::fill(a, a + arraysize(a), 0.f);
+    }
+
+    float b[2];
+    float a[2];
+  };
+
+  BiQuadState biquad_state_;
+  BiQuadCoefficients coefficients_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(BiQuadFilter);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_BIQUAD_FILTER_H_
diff --git a/modules/audio_processing/agc2/down_sampler.cc b/modules/audio_processing/agc2/down_sampler.cc
new file mode 100644
index 0000000000..50486e0a36
--- /dev/null
+++ b/modules/audio_processing/agc2/down_sampler.cc
@@ -0,0 +1,98 @@
+/*
+ *  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 "modules/audio_processing/agc2/down_sampler.h"
+
+#include <string.h>
+#include <algorithm>
+
+#include "modules/audio_processing/agc2/biquad_filter.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace {
+
+constexpr int kChunkSizeMs = 10;
+constexpr int kSampleRate8kHz = 8000;
+constexpr int kSampleRate16kHz = 16000;
+constexpr int kSampleRate32kHz = 32000;
+constexpr int kSampleRate48kHz = 48000;
+
+// Bandlimiter coefficients computed based on that only
+// the first 40 bins of the spectrum for the downsampled
+// signal are used.
+// [B,A] = butter(2,(41/64*4000)/8000)
+const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_16kHz = {
+    {0.1455f, 0.2911f, 0.1455f},
+    {-0.6698f, 0.2520f}};
+
+// [B,A] = butter(2,(41/64*4000)/16000)
+const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_32kHz = {
+    {0.0462f, 0.0924f, 0.0462f},
+    {-1.3066f, 0.4915f}};
+
+// [B,A] = butter(2,(41/64*4000)/24000)
+const BiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients_48kHz = {
+    {0.0226f, 0.0452f, 0.0226f},
+    {-1.5320f, 0.6224f}};
+
+}  // namespace
+
+DownSampler::DownSampler(ApmDataDumper* data_dumper)
+    : data_dumper_(data_dumper) {
+  Initialize(48000);
+}
+void DownSampler::Initialize(int sample_rate_hz) {
+  RTC_DCHECK(
+      sample_rate_hz == kSampleRate8kHz || sample_rate_hz == kSampleRate16kHz ||
+      sample_rate_hz == kSampleRate32kHz || sample_rate_hz == kSampleRate48kHz);
+
+  sample_rate_hz_ = sample_rate_hz;
+  down_sampling_factor_ = rtc::CheckedDivExact(sample_rate_hz_, 8000);
+
+  /// Note that the down sampling filter is not used if the sample rate is 8
+  /// kHz.
+  if (sample_rate_hz_ == kSampleRate16kHz) {
+    low_pass_filter_.Initialize(kLowPassFilterCoefficients_16kHz);
+  } else if (sample_rate_hz_ == kSampleRate32kHz) {
+    low_pass_filter_.Initialize(kLowPassFilterCoefficients_32kHz);
+  } else if (sample_rate_hz_ == kSampleRate48kHz) {
+    low_pass_filter_.Initialize(kLowPassFilterCoefficients_48kHz);
+  }
+}
+
+void DownSampler::DownSample(rtc::ArrayView<const float> in,
+                             rtc::ArrayView<float> out) {
+  data_dumper_->DumpWav("lc_down_sampler_input", in, sample_rate_hz_, 1);
+  RTC_DCHECK_EQ(sample_rate_hz_ * kChunkSizeMs / 1000, in.size());
+  RTC_DCHECK_EQ(kSampleRate8kHz * kChunkSizeMs / 1000, out.size());
+  const size_t kMaxNumFrames = kSampleRate48kHz * kChunkSizeMs / 1000;
+  float x[kMaxNumFrames];
+
+  // Band-limit the signal to 4 kHz.
+  if (sample_rate_hz_ != kSampleRate8kHz) {
+    low_pass_filter_.Process(in, rtc::ArrayView<float>(x, in.size()));
+
+    // Downsample the signal.
+    size_t k = 0;
+    for (size_t j = 0; j < out.size(); ++j) {
+      RTC_DCHECK_GT(kMaxNumFrames, k);
+      out[j] = x[k];
+      k += down_sampling_factor_;
+    }
+  } else {
+    std::copy(in.data(), in.data() + in.size(), out.data());
+  }
+
+  data_dumper_->DumpWav("lc_down_sampler_output", out, kSampleRate8kHz, 1);
+}
+
+}  // namespace webrtc
diff --git a/modules/audio_processing/agc2/down_sampler.h b/modules/audio_processing/agc2/down_sampler.h
new file mode 100644
index 0000000000..a609ea8e0c
--- /dev/null
+++ b/modules/audio_processing/agc2/down_sampler.h
@@ -0,0 +1,40 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_DOWN_SAMPLER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_DOWN_SAMPLER_H_
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/biquad_filter.h"
+#include "rtc_base/constructormagic.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+
+class DownSampler {
+ public:
+  explicit DownSampler(ApmDataDumper* data_dumper);
+  void Initialize(int sample_rate_hz);
+
+  void DownSample(rtc::ArrayView<const float> in, rtc::ArrayView<float> out);
+
+ private:
+  ApmDataDumper* data_dumper_;
+  int sample_rate_hz_;
+  int down_sampling_factor_;
+  BiQuadFilter low_pass_filter_;
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DownSampler);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_DOWN_SAMPLER_H_
diff --git a/modules/audio_processing/agc2/noise_level_estimator.cc b/modules/audio_processing/agc2/noise_level_estimator.cc
index ede431c799..d9aaf1f9bd 100644
--- a/modules/audio_processing/agc2/noise_level_estimator.cc
+++ b/modules/audio_processing/agc2/noise_level_estimator.cc
@@ -10,11 +10,102 @@
 
 #include "modules/audio_processing/agc2/noise_level_estimator.h"
 
+#include <math.h>
+
+#include <algorithm>
+#include <numeric>
+
+#include "common_audio/include/audio_util.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+
 namespace webrtc {
 
-float NoiseLevelEstimator::Analyze(AudioFrameView<const float> frame) {
-  // TODO(webrtc:7494): This is a stub. Add implementation.
-  return -50.f;
+namespace {
+constexpr int kFramesPerSecond = 100;
+
+float FrameEnergy(const AudioFrameView<const float>& audio) {
+  float energy = 0.f;
+  for (size_t k = 0; k < audio.num_channels(); ++k) {
+    float channel_energy =
+        std::accumulate(audio.channel(k).begin(), audio.channel(k).end(), 0.f,
+                        [](float a, float b) -> float { return a + b * b; });
+    energy = std::max(channel_energy, energy);
+  }
+  return energy;
+}
+
+float EnergyToDbfs(float signal_energy, size_t num_samples) {
+  const float rms = std::sqrt(signal_energy / num_samples);
+  return FloatS16ToDbfs(rms);
+}
+}  // namespace
+
+NoiseLevelEstimator::NoiseLevelEstimator(ApmDataDumper* data_dumper)
+    : signal_classifier_(data_dumper) {
+  Initialize(48000);
+}
+
+NoiseLevelEstimator::~NoiseLevelEstimator() {}
+
+void NoiseLevelEstimator::Initialize(int sample_rate_hz) {
+  sample_rate_hz_ = sample_rate_hz;
+  noise_energy_ = 1.f;
+  first_update_ = true;
+  min_noise_energy_ = sample_rate_hz * 2.f * 2.f / kFramesPerSecond;
+  noise_energy_hold_counter_ = 0;
+  signal_classifier_.Initialize(sample_rate_hz);
+}
+
+float NoiseLevelEstimator::Analyze(const AudioFrameView<const float>& frame) {
+  const int rate =
+      static_cast<int>(frame.samples_per_channel() * kFramesPerSecond);
+  if (rate != sample_rate_hz_) {
+    Initialize(rate);
+  }
+  const float frame_energy = FrameEnergy(frame);
+  if (frame_energy <= 0.f) {
+    RTC_DCHECK_GE(frame_energy, 0.f);
+    return EnergyToDbfs(noise_energy_, frame.samples_per_channel());
+  }
+
+  if (first_update_) {
+    // Initialize the noise energy to the frame energy.
+    first_update_ = false;
+    return EnergyToDbfs(
+        noise_energy_ = std::max(frame_energy, min_noise_energy_),
+        frame.samples_per_channel());
+  }
+
+  const SignalClassifier::SignalType signal_type =
+      signal_classifier_.Analyze(frame.channel(0));
+
+  // Update the noise estimate in a minimum statistics-type manner.
+  if (signal_type == SignalClassifier::SignalType::kStationary) {
+    if (frame_energy > noise_energy_) {
+      // Leak the estimate upwards towards the frame energy if no recent
+      // downward update.
+      noise_energy_hold_counter_ = std::max(noise_energy_hold_counter_ - 1, 0);
+
+      if (noise_energy_hold_counter_ == 0) {
+        noise_energy_ = std::min(noise_energy_ * 1.01f, frame_energy);
+      }
+    } else {
+      // Update smoothly downwards with a limited maximum update magnitude.
+      noise_energy_ =
+          std::max(noise_energy_ * 0.9f,
+                   noise_energy_ + 0.05f * (frame_energy - noise_energy_));
+      noise_energy_hold_counter_ = 1000;
+    }
+  } else {
+    // For a non-stationary signal, leak the estimate downwards in order to
+    // avoid estimate locking due to incorrect signal classification.
+    noise_energy_ = noise_energy_ * 0.99f;
+  }
+
+  // Ensure a minimum of the estimate.
+  return EnergyToDbfs(
+      noise_energy_ = std::max(noise_energy_, min_noise_energy_),
+      frame.samples_per_channel());
 }
 
 }  // namespace webrtc
diff --git a/modules/audio_processing/agc2/noise_level_estimator.h b/modules/audio_processing/agc2/noise_level_estimator.h
index f9e4abc8f5..24067a1665 100644
--- a/modules/audio_processing/agc2/noise_level_estimator.h
+++ b/modules/audio_processing/agc2/noise_level_estimator.h
@@ -11,19 +11,30 @@
 #ifndef MODULES_AUDIO_PROCESSING_AGC2_NOISE_LEVEL_ESTIMATOR_H_
 #define MODULES_AUDIO_PROCESSING_AGC2_NOISE_LEVEL_ESTIMATOR_H_
 
+#include "modules/audio_processing/agc2/signal_classifier.h"
 #include "modules/audio_processing/include/audio_frame_view.h"
 #include "rtc_base/constructormagic.h"
 
 namespace webrtc {
+class ApmDataDumper;
 
 class NoiseLevelEstimator {
  public:
-  NoiseLevelEstimator() {}
-
+  NoiseLevelEstimator(ApmDataDumper* data_dumper);
+  ~NoiseLevelEstimator();
   // Returns the estimated noise level in dBFS.
-  float Analyze(AudioFrameView<const float> frame);
+  float Analyze(const AudioFrameView<const float>& frame);
 
  private:
+  void Initialize(int sample_rate_hz);
+
+  int sample_rate_hz_;
+  float min_noise_energy_;
+  bool first_update_;
+  float noise_energy_;
+  int noise_energy_hold_counter_;
+  SignalClassifier signal_classifier_;
+
   RTC_DISALLOW_COPY_AND_ASSIGN(NoiseLevelEstimator);
 };
 
diff --git a/modules/audio_processing/agc2/noise_level_estimator_unittest.cc b/modules/audio_processing/agc2/noise_level_estimator_unittest.cc
new file mode 100644
index 0000000000..c4fd33b0a0
--- /dev/null
+++ b/modules/audio_processing/agc2/noise_level_estimator_unittest.cc
@@ -0,0 +1,83 @@
+/*
+ *  Copyright (c) 2018 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 "modules/audio_processing/agc2/noise_level_estimator.h"
+
+#include <array>
+#include <functional>
+#include <limits>
+
+#include "modules/audio_processing/agc2/agc2_testing_common.h"
+#include "modules/audio_processing/agc2/vector_float_frame.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/gunit.h"
+#include "rtc_base/random.h"
+
+namespace webrtc {
+namespace {
+Random rand_gen(42);
+ApmDataDumper data_dumper(0);
+constexpr int kNumIterations = 200;
+constexpr int kFramesPerSecond = 100;
+
+// Runs the noise estimator on audio generated by 'sample_generator'
+// for kNumIterations. Returns the last noise level estimate.
+float RunEstimator(std::function<float()> sample_generator, int rate) {
+  NoiseLevelEstimator estimator(&data_dumper);
+  const size_t samples_per_channel =
+      rtc::CheckedDivExact(rate, kFramesPerSecond);
+  VectorFloatFrame signal(1, static_cast<int>(samples_per_channel), 0.f);
+
+  for (int i = 0; i < kNumIterations; ++i) {
+    AudioFrameView<float> frame_view = signal.float_frame_view();
+    for (size_t j = 0; j < samples_per_channel; ++j) {
+      frame_view.channel(0)[j] = sample_generator();
+    }
+    estimator.Analyze(frame_view);
+  }
+  return estimator.Analyze(signal.float_frame_view());
+}
+
+float WhiteNoiseGenerator() {
+  return static_cast<float>(rand_gen.Rand(std::numeric_limits<int16_t>::min(),
+                                          std::numeric_limits<int16_t>::max()));
+}
+}  // namespace
+
+// White random noise is stationary, but does not trigger the detector
+// every frame due to the randomness.
+TEST(AutomaticGainController2NoiseEstimator, RandomNoise) {
+  for (const auto rate : {8000, 16000, 32000, 48000}) {
+    const float noise_level = RunEstimator(WhiteNoiseGenerator, rate);
+    EXPECT_NEAR(noise_level, -5.f, 1.f);
+  }
+}
+
+// Sine curves are (very) stationary. They trigger the detector all
+// the time. Except for a few initial frames.
+TEST(AutomaticGainController2NoiseEstimator, SineTone) {
+  for (const auto rate : {8000, 16000, 32000, 48000}) {
+    test::SineGenerator gen(600.f, rate);
+    const float noise_level = RunEstimator(gen, rate);
+    EXPECT_NEAR(noise_level, -33.f, 1.f);
+  }
+}
+
+// Pulses are transient if they are far enough apart. They shouldn't
+// trigger the noise detector.
+TEST(AutomaticGainController2NoiseEstimator, PulseTone) {
+  for (const auto rate : {8000, 16000, 32000, 48000}) {
+    test::PulseGenerator gen(20.f, rate);
+    const int noise_level = RunEstimator(gen, rate);
+    EXPECT_NEAR(noise_level, -79.f, 1.f);
+  }
+}
+
+}  // namespace webrtc
diff --git a/modules/audio_processing/agc2/noise_spectrum_estimator.cc b/modules/audio_processing/agc2/noise_spectrum_estimator.cc
new file mode 100644
index 0000000000..9e08126e89
--- /dev/null
+++ b/modules/audio_processing/agc2/noise_spectrum_estimator.cc
@@ -0,0 +1,68 @@
+/*
+ *  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 "modules/audio_processing/agc2/noise_spectrum_estimator.h"
+
+#include <string.h>
+#include <algorithm>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/arraysize.h"
+
+namespace webrtc {
+namespace {
+constexpr float kMinNoisePower = 100.f;
+}  // namespace
+
+NoiseSpectrumEstimator::NoiseSpectrumEstimator(ApmDataDumper* data_dumper)
+    : data_dumper_(data_dumper) {
+  Initialize();
+}
+
+void NoiseSpectrumEstimator::Initialize() {
+  std::fill(noise_spectrum_, noise_spectrum_ + arraysize(noise_spectrum_),
+            kMinNoisePower);
+}
+
+void NoiseSpectrumEstimator::Update(rtc::ArrayView<const float> spectrum,
+                                    bool first_update) {
+  RTC_DCHECK_EQ(65, spectrum.size());
+
+  if (first_update) {
+    // Initialize the noise spectral estimate with the signal spectrum.
+    std::copy(spectrum.data(), spectrum.data() + spectrum.size(),
+              noise_spectrum_);
+  } else {
+    // Smoothly update the noise spectral estimate towards the signal spectrum
+    // such that the magnitude of the updates are limited.
+    for (size_t k = 0; k < spectrum.size(); ++k) {
+      if (noise_spectrum_[k] < spectrum[k]) {
+        noise_spectrum_[k] = std::min(
+            1.01f * noise_spectrum_[k],
+            noise_spectrum_[k] + 0.05f * (spectrum[k] - noise_spectrum_[k]));
+      } else {
+        noise_spectrum_[k] = std::max(
+            0.99f * noise_spectrum_[k],
+            noise_spectrum_[k] + 0.05f * (spectrum[k] - noise_spectrum_[k]));
+      }
+    }
+  }
+
+  // Ensure that the noise spectal estimate does not become too low.
+  for (auto& v : noise_spectrum_) {
+    v = std::max(v, kMinNoisePower);
+  }
+
+  data_dumper_->DumpRaw("lc_noise_spectrum", 65, noise_spectrum_);
+  data_dumper_->DumpRaw("lc_signal_spectrum", spectrum);
+}
+
+}  // namespace webrtc
diff --git a/modules/audio_processing/agc2/noise_spectrum_estimator.h b/modules/audio_processing/agc2/noise_spectrum_estimator.h
new file mode 100644
index 0000000000..fd1cc13a3f
--- /dev/null
+++ b/modules/audio_processing/agc2/noise_spectrum_estimator.h
@@ -0,0 +1,40 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_NOISE_SPECTRUM_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_NOISE_SPECTRUM_ESTIMATOR_H_
+
+#include "api/array_view.h"
+#include "rtc_base/constructormagic.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+
+class NoiseSpectrumEstimator {
+ public:
+  explicit NoiseSpectrumEstimator(ApmDataDumper* data_dumper);
+  void Initialize();
+  void Update(rtc::ArrayView<const float> spectrum, bool first_update);
+
+  rtc::ArrayView<const float> GetNoiseSpectrum() const {
+    return rtc::ArrayView<const float>(noise_spectrum_);
+  }
+
+ private:
+  ApmDataDumper* data_dumper_;
+  float noise_spectrum_[65];
+
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(NoiseSpectrumEstimator);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_NOISE_SPECTRUM_ESTIMATOR_H_
diff --git a/modules/audio_processing/agc2/signal_classifier.cc b/modules/audio_processing/agc2/signal_classifier.cc
new file mode 100644
index 0000000000..0ec34148b9
--- /dev/null
+++ b/modules/audio_processing/agc2/signal_classifier.cc
@@ -0,0 +1,167 @@
+/*
+ *  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 "modules/audio_processing/agc2/signal_classifier.h"
+
+#include <algorithm>
+#include <numeric>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/down_sampler.h"
+#include "modules/audio_processing/agc2/noise_spectrum_estimator.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/constructormagic.h"
+
+namespace webrtc {
+namespace {
+
+void RemoveDcLevel(rtc::ArrayView<float> x) {
+  RTC_DCHECK_LT(0, x.size());
+  float mean = std::accumulate(x.data(), x.data() + x.size(), 0.f);
+  mean /= x.size();
+
+  for (float& v : x) {
+    v -= mean;
+  }
+}
+
+void PowerSpectrum(const OouraFft* ooura_fft,
+                   rtc::ArrayView<const float> x,
+                   rtc::ArrayView<float> spectrum) {
+  RTC_DCHECK_EQ(65, spectrum.size());
+  RTC_DCHECK_EQ(128, x.size());
+  float X[128];
+  std::copy(x.data(), x.data() + x.size(), X);
+  ooura_fft->Fft(X);
+
+  float* X_p = X;
+  RTC_DCHECK_EQ(X_p, &X[0]);
+  spectrum[0] = (*X_p) * (*X_p);
+  ++X_p;
+  RTC_DCHECK_EQ(X_p, &X[1]);
+  spectrum[64] = (*X_p) * (*X_p);
+  for (int k = 1; k < 64; ++k) {
+    ++X_p;
+    RTC_DCHECK_EQ(X_p, &X[2 * k]);
+    spectrum[k] = (*X_p) * (*X_p);
+    ++X_p;
+    RTC_DCHECK_EQ(X_p, &X[2 * k + 1]);
+    spectrum[k] += (*X_p) * (*X_p);
+  }
+}
+
+webrtc::SignalClassifier::SignalType ClassifySignal(
+    rtc::ArrayView<const float> signal_spectrum,
+    rtc::ArrayView<const float> noise_spectrum,
+    ApmDataDumper* data_dumper) {
+  int num_stationary_bands = 0;
+  int num_highly_nonstationary_bands = 0;
+
+  // Detect stationary and highly nonstationary bands.
+  for (size_t k = 1; k < 40; k++) {
+    if (signal_spectrum[k] < 3 * noise_spectrum[k] &&
+        signal_spectrum[k] * 3 > noise_spectrum[k]) {
+      ++num_stationary_bands;
+    } else if (signal_spectrum[k] > 9 * noise_spectrum[k]) {
+      ++num_highly_nonstationary_bands;
+    }
+  }
+
+  data_dumper->DumpRaw("lc_num_stationary_bands", 1, &num_stationary_bands);
+  data_dumper->DumpRaw("lc_num_highly_nonstationary_bands", 1,
+                       &num_highly_nonstationary_bands);
+
+  // Use the detected number of bands to classify the overall signal
+  // stationarity.
+  if (num_stationary_bands > 15) {
+    return SignalClassifier::SignalType::kStationary;
+  } else {
+    return SignalClassifier::SignalType::kNonStationary;
+  }
+}
+
+}  // namespace
+
+SignalClassifier::FrameExtender::FrameExtender(size_t frame_size,
+                                               size_t extended_frame_size)
+    : x_old_(extended_frame_size - frame_size, 0.f) {}
+
+SignalClassifier::FrameExtender::~FrameExtender() = default;
+
+void SignalClassifier::FrameExtender::ExtendFrame(
+    rtc::ArrayView<const float> x,
+    rtc::ArrayView<float> x_extended) {
+  RTC_DCHECK_EQ(x_old_.size() + x.size(), x_extended.size());
+  std::copy(x_old_.data(), x_old_.data() + x_old_.size(), x_extended.data());
+  std::copy(x.data(), x.data() + x.size(), x_extended.data() + x_old_.size());
+  std::copy(x_extended.data() + x_extended.size() - x_old_.size(),
+            x_extended.data() + x_extended.size(), x_old_.data());
+}
+
+SignalClassifier::SignalClassifier(ApmDataDumper* data_dumper)
+    : data_dumper_(data_dumper),
+      down_sampler_(data_dumper_),
+      noise_spectrum_estimator_(data_dumper_) {
+  Initialize(48000);
+}
+SignalClassifier::~SignalClassifier() {}
+
+void SignalClassifier::Initialize(int sample_rate_hz) {
+  down_sampler_.Initialize(sample_rate_hz);
+  noise_spectrum_estimator_.Initialize();
+  frame_extender_.reset(new FrameExtender(80, 128));
+  sample_rate_hz_ = sample_rate_hz;
+  initialization_frames_left_ = 2;
+  consistent_classification_counter_ = 3;
+  last_signal_type_ = SignalClassifier::SignalType::kNonStationary;
+}
+
+SignalClassifier::SignalType SignalClassifier::Analyze(
+    rtc::ArrayView<const float> signal) {
+  RTC_DCHECK_EQ(signal.size(), sample_rate_hz_ / 100);
+
+  // Compute the signal power spectrum.
+  float downsampled_frame[80];
+  down_sampler_.DownSample(signal, downsampled_frame);
+  float extended_frame[128];
+  frame_extender_->ExtendFrame(downsampled_frame, extended_frame);
+  RemoveDcLevel(extended_frame);
+  float signal_spectrum[65];
+  PowerSpectrum(&ooura_fft_, extended_frame, signal_spectrum);
+
+  // Classify the signal based on the estimate of the noise spectrum and the
+  // signal spectrum estimate.
+  const SignalType signal_type = ClassifySignal(
+      signal_spectrum, noise_spectrum_estimator_.GetNoiseSpectrum(),
+      data_dumper_);
+
+  // Update the noise spectrum based on the signal spectrum.
+  noise_spectrum_estimator_.Update(signal_spectrum,
+                                   initialization_frames_left_ > 0);
+
+  // Update the number of frames until a reliable signal spectrum is achieved.
+  initialization_frames_left_ = std::max(0, initialization_frames_left_ - 1);
+
+  if (last_signal_type_ == signal_type) {
+    consistent_classification_counter_ =
+        std::max(0, consistent_classification_counter_ - 1);
+  } else {
+    last_signal_type_ = signal_type;
+    consistent_classification_counter_ = 3;
+  }
+
+  if (consistent_classification_counter_ > 0) {
+    return SignalClassifier::SignalType::kNonStationary;
+  }
+  return signal_type;
+}
+
+}  // namespace webrtc
diff --git a/modules/audio_processing/agc2/signal_classifier.h b/modules/audio_processing/agc2/signal_classifier.h
new file mode 100644
index 0000000000..23fe315977
--- /dev/null
+++ b/modules/audio_processing/agc2/signal_classifier.h
@@ -0,0 +1,67 @@
+/*
+ *  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.
+ */
+
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_SIGNAL_CLASSIFIER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_SIGNAL_CLASSIFIER_H_
+
+#include <memory>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/agc2/down_sampler.h"
+#include "modules/audio_processing/agc2/noise_spectrum_estimator.h"
+#include "modules/audio_processing/utility/ooura_fft.h"
+#include "rtc_base/constructormagic.h"
+
+namespace webrtc {
+
+class ApmDataDumper;
+class AudioBuffer;
+
+class SignalClassifier {
+ public:
+  enum class SignalType { kNonStationary, kStationary };
+
+  explicit SignalClassifier(ApmDataDumper* data_dumper);
+  ~SignalClassifier();
+
+  void Initialize(int sample_rate_hz);
+  SignalType Analyze(rtc::ArrayView<const float> signal);
+
+ private:
+  class FrameExtender {
+   public:
+    FrameExtender(size_t frame_size, size_t extended_frame_size);
+    ~FrameExtender();
+
+    void ExtendFrame(rtc::ArrayView<const float> x,
+                     rtc::ArrayView<float> x_extended);
+
+   private:
+    std::vector<float> x_old_;
+
+    RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FrameExtender);
+  };
+
+  ApmDataDumper* const data_dumper_;
+  DownSampler down_sampler_;
+  std::unique_ptr<FrameExtender> frame_extender_;
+  NoiseSpectrumEstimator noise_spectrum_estimator_;
+  int sample_rate_hz_;
+  int initialization_frames_left_;
+  int consistent_classification_counter_;
+  SignalType last_signal_type_;
+  const OouraFft ooura_fft_;
+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SignalClassifier);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_SIGNAL_CLASSIFIER_H_
diff --git a/modules/audio_processing/agc2/signal_classifier_unittest.cc b/modules/audio_processing/agc2/signal_classifier_unittest.cc
new file mode 100644
index 0000000000..62171b32e6
--- /dev/null
+++ b/modules/audio_processing/agc2/signal_classifier_unittest.cc
@@ -0,0 +1,82 @@
+/*
+ *  Copyright (c) 2018 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 "modules/audio_processing/agc2/signal_classifier.h"
+
+#include <array>
+#include <functional>
+#include <limits>
+
+#include "modules/audio_processing/agc2/agc2_testing_common.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/gunit.h"
+#include "rtc_base/random.h"
+
+namespace webrtc {
+
+namespace {
+Random rand_gen(42);
+ApmDataDumper data_dumper(0);
+constexpr int kNumIterations = 100;
+
+// Runs the signal classifier on audio generated by 'sample_generator'
+// for kNumIterations. Returns the number of frames classified as noise.
+int RunClassifier(std::function<float()> sample_generator, int rate) {
+  SignalClassifier classifier(&data_dumper);
+  std::array<float, 480> signal;
+  classifier.Initialize(rate);
+  const size_t samples_per_channel = rtc::CheckedDivExact(rate, 100);
+  int number_of_noise_frames = 0;
+  for (int i = 0; i < kNumIterations; ++i) {
+    for (size_t j = 0; j < samples_per_channel; ++j) {
+      signal[j] = sample_generator();
+    }
+    number_of_noise_frames +=
+        classifier.Analyze({&signal[0], samples_per_channel}) ==
+        SignalClassifier::SignalType::kStationary;
+  }
+  return number_of_noise_frames;
+}
+
+float WhiteNoiseGenerator() {
+  return static_cast<float>(rand_gen.Rand(std::numeric_limits<int16_t>::min(),
+                                          std::numeric_limits<int16_t>::max()));
+}
+}  // namespace
+
+// White random noise is stationary, but does not trigger the detector
+// every frame due to the randomness.
+TEST(AutomaticGainController2SignalClassifier, WhiteNoise) {
+  for (const auto rate : {8000, 16000, 32000, 48000}) {
+    const int number_of_noise_frames = RunClassifier(WhiteNoiseGenerator, rate);
+    EXPECT_GT(number_of_noise_frames, kNumIterations / 2);
+  }
+}
+
+// Sine curves are (very) stationary. They trigger the detector all
+// the time. Except for a few initial frames.
+TEST(AutomaticGainController2SignalClassifier, SineTone) {
+  for (const auto rate : {8000, 16000, 32000, 48000}) {
+    test::SineGenerator gen(600.f, rate);
+    const int number_of_noise_frames = RunClassifier(gen, rate);
+    EXPECT_GE(number_of_noise_frames, kNumIterations - 5);
+  }
+}
+
+// Pulses are transient if they are far enough apart. They shouldn't
+// trigger the noise detector.
+TEST(AutomaticGainController2SignalClassifier, PulseTone) {
+  for (const auto rate : {8000, 16000, 32000, 48000}) {
+    test::PulseGenerator gen(30.f, rate);
+    const int number_of_noise_frames = RunClassifier(gen, rate);
+    EXPECT_EQ(number_of_noise_frames, 0);
+  }
+}
+}  // namespace webrtc