diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn
index 4c6cfab576..6dd8babd78 100644
--- a/modules/audio_processing/agc2/BUILD.gn
+++ b/modules/audio_processing/agc2/BUILD.gn
@@ -110,24 +110,15 @@ rtc_library("gain_applier") {
 
 rtc_library("noise_level_estimator") {
   sources = [
-    "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 = [
     ":biquad_filter",
     "..:apm_logging",
     "..:audio_frame_view",
     "../../../api:array_view",
-    "../../../common_audio",
-    "../../../common_audio/third_party/ooura:fft_size_128",
     "../../../rtc_base:checks",
-    "../../../rtc_base:macromagic",
     "../../../system_wrappers",
   ]
 
@@ -241,10 +232,7 @@ rtc_library("noise_estimator_unittests") {
   testonly = true
   configs += [ "..:apm_debug_dump" ]
 
-  sources = [
-    "noise_level_estimator_unittest.cc",
-    "signal_classifier_unittest.cc",
-  ]
+  sources = [ "noise_level_estimator_unittest.cc" ]
   deps = [
     ":noise_level_estimator",
     ":test_utils",
diff --git a/modules/audio_processing/agc2/adaptive_agc.cc b/modules/audio_processing/agc2/adaptive_agc.cc
index 3fc9008db1..a9f9622d00 100644
--- a/modules/audio_processing/agc2/adaptive_agc.cc
+++ b/modules/audio_processing/agc2/adaptive_agc.cc
@@ -41,17 +41,6 @@ AvailableCpuFeatures GetAllowedCpuFeatures(
   return features;
 }
 
-std::unique_ptr<NoiseLevelEstimator> CreateNoiseLevelEstimator(
-    NoiseEstimatorType estimator_type,
-    ApmDataDumper* apm_data_dumper) {
-  switch (estimator_type) {
-    case NoiseEstimatorType::kStationaryNoise:
-      return CreateStationaryNoiseEstimator(apm_data_dumper);
-    case NoiseEstimatorType::kNoiseFloor:
-      return CreateNoiseFloorEstimator(apm_data_dumper);
-  }
-}
-
 }  // namespace
 
 AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper,
@@ -65,8 +54,7 @@ AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper,
                        config.max_output_noise_level_dbfs,
                        config.dry_run),
       apm_data_dumper_(apm_data_dumper),
-      noise_level_estimator_(
-          CreateNoiseLevelEstimator(config.noise_estimator, apm_data_dumper)),
+      noise_level_estimator_(CreateNoiseFloorEstimator(apm_data_dumper)),
       saturation_protector_(
           CreateSaturationProtector(kSaturationProtectorInitialHeadroomDb,
                                     kSaturationProtectorExtraHeadroomDb,
diff --git a/modules/audio_processing/agc2/down_sampler.cc b/modules/audio_processing/agc2/down_sampler.cc
deleted file mode 100644
index fd1a2c3a46..0000000000
--- a/modules/audio_processing/agc2/down_sampler.cc
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
- *  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("agc2_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("agc2_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
deleted file mode 100644
index a44f96fa2d..0000000000
--- a/modules/audio_processing/agc2/down_sampler.h
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- *  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"
-
-namespace webrtc {
-
-class ApmDataDumper;
-
-class DownSampler {
- public:
-  explicit DownSampler(ApmDataDumper* data_dumper);
-
-  DownSampler() = delete;
-  DownSampler(const DownSampler&) = delete;
-  DownSampler& operator=(const DownSampler&) = delete;
-
-  void Initialize(int sample_rate_hz);
-
-  void DownSample(rtc::ArrayView<const float> in, rtc::ArrayView<float> out);
-
- private:
-  ApmDataDumper* const data_dumper_;
-  int sample_rate_hz_;
-  int down_sampling_factor_;
-  BiQuadFilter low_pass_filter_;
-};
-
-}  // 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 10e8437d3f..3d9aa1a4ae 100644
--- a/modules/audio_processing/agc2/noise_level_estimator.cc
+++ b/modules/audio_processing/agc2/noise_level_estimator.cc
@@ -17,13 +17,12 @@
 #include <numeric>
 
 #include "api/array_view.h"
-#include "common_audio/include/audio_util.h"
-#include "modules/audio_processing/agc2/signal_classifier.h"
 #include "modules/audio_processing/logging/apm_data_dumper.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
 namespace {
+
 constexpr int kFramesPerSecond = 100;
 
 float FrameEnergy(const AudioFrameView<const float>& audio) {
@@ -37,108 +36,16 @@ float FrameEnergy(const AudioFrameView<const float>& audio) {
   return energy;
 }
 
-float EnergyToDbfs(float signal_energy, size_t num_samples) {
-  const float rms = std::sqrt(signal_energy / num_samples);
-  return FloatS16ToDbfs(rms);
+float EnergyToDbfs(float signal_energy, int num_samples) {
+  RTC_DCHECK_GE(signal_energy, 0.0f);
+  const float rms_square = signal_energy / num_samples;
+  constexpr float kMinDbfs = -90.30899869919436f;
+  if (rms_square <= 1.0f) {
+    return kMinDbfs;
+  }
+  return 10.0f * std::log10(rms_square) + kMinDbfs;
 }
 
-class NoiseLevelEstimatorImpl : public NoiseLevelEstimator {
- public:
-  NoiseLevelEstimatorImpl(ApmDataDumper* data_dumper)
-      : data_dumper_(data_dumper), signal_classifier_(data_dumper) {
-    // Initially assume that 48 kHz will be used. `Analyze()` will detect the
-    // used sample rate and call `Initialize()` again if needed.
-    Initialize(/*sample_rate_hz=*/48000);
-  }
-  NoiseLevelEstimatorImpl(const NoiseLevelEstimatorImpl&) = delete;
-  NoiseLevelEstimatorImpl& operator=(const NoiseLevelEstimatorImpl&) = delete;
-  ~NoiseLevelEstimatorImpl() = default;
-
-  float Analyze(const AudioFrameView<const float>& frame) override {
-    data_dumper_->DumpRaw("agc2_noise_level_estimator_hold_counter",
-                          noise_energy_hold_counter_);
-    const int sample_rate_hz =
-        static_cast<int>(frame.samples_per_channel() * kFramesPerSecond);
-    if (sample_rate_hz != sample_rate_hz_) {
-      Initialize(sample_rate_hz);
-    }
-    const float frame_energy = FrameEnergy(frame);
-    if (frame_energy <= 0.f) {
-      RTC_DCHECK_GE(frame_energy, 0.f);
-      data_dumper_->DumpRaw("agc2_noise_level_estimator_signal_type", -1);
-      return EnergyToDbfs(noise_energy_, frame.samples_per_channel());
-    }
-
-    if (first_update_) {
-      // Initialize the noise energy to the frame energy.
-      first_update_ = false;
-      data_dumper_->DumpRaw("agc2_noise_level_estimator_signal_type", -1);
-      noise_energy_ = std::max(frame_energy, min_noise_energy_);
-      return EnergyToDbfs(noise_energy_, frame.samples_per_channel());
-    }
-
-    const SignalClassifier::SignalType signal_type =
-        signal_classifier_.Analyze(frame.channel(0));
-    data_dumper_->DumpRaw("agc2_noise_level_estimator_signal_type",
-                          static_cast<int>(signal_type));
-
-    // 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) {
-          constexpr float kMaxNoiseEnergyFactor = 1.01f;
-          noise_energy_ =
-              std::min(noise_energy_ * kMaxNoiseEnergyFactor, frame_energy);
-        }
-      } else {
-        // Update smoothly downwards with a limited maximum update magnitude.
-        constexpr float kMinNoiseEnergyFactor = 0.9f;
-        constexpr float kNoiseEnergyDeltaFactor = 0.05f;
-        noise_energy_ =
-            std::max(noise_energy_ * kMinNoiseEnergyFactor,
-                     noise_energy_ - kNoiseEnergyDeltaFactor *
-                                         (noise_energy_ - frame_energy));
-        // Prevent an energy increase for the next 10 seconds.
-        constexpr int kNumFramesToEnergyIncreaseAllowed = 1000;
-        noise_energy_hold_counter_ = kNumFramesToEnergyIncreaseAllowed;
-      }
-    } else {
-      // TODO(bugs.webrtc.org/7494): Remove to not forget the estimated level.
-      // 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.
-    noise_energy_ = std::max(noise_energy_, min_noise_energy_);
-    return EnergyToDbfs(noise_energy_, frame.samples_per_channel());
-  }
-
- private:
-  void Initialize(int sample_rate_hz) {
-    sample_rate_hz_ = sample_rate_hz;
-    noise_energy_ = 1.0f;
-    first_update_ = true;
-    // Initialize the minimum noise energy to -84 dBFS.
-    min_noise_energy_ = sample_rate_hz * 2.0f * 2.0f / kFramesPerSecond;
-    noise_energy_hold_counter_ = 0;
-    signal_classifier_.Initialize(sample_rate_hz);
-  }
-
-  ApmDataDumper* const data_dumper_;
-  int sample_rate_hz_;
-  float min_noise_energy_;
-  bool first_update_;
-  float noise_energy_;
-  int noise_energy_hold_counter_;
-  SignalClassifier signal_classifier_;
-};
-
 // Updates the noise floor with instant decay and slow attack. This tuning is
 // specific for AGC2, so that (i) it can promptly increase the gain if the noise
 // floor drops (instant decay) and (ii) in case of music or fast speech, due to
@@ -186,7 +93,8 @@ class NoiseFloorEstimator : public NoiseLevelEstimator {
       // Ignore frames when muted or below the minimum measurable energy.
       data_dumper_->DumpRaw("agc2_noise_floor_estimator_preliminary_level",
                             noise_energy_);
-      return EnergyToDbfs(noise_energy_, frame.samples_per_channel());
+      return EnergyToDbfs(noise_energy_,
+                          static_cast<int>(frame.samples_per_channel()));
     }
 
     if (preliminary_noise_energy_set_) {
@@ -220,7 +128,8 @@ class NoiseFloorEstimator : public NoiseLevelEstimator {
       noise_energy_ = std::min(noise_energy_, preliminary_noise_energy_);
       counter_--;
     }
-    return EnergyToDbfs(noise_energy_, frame.samples_per_channel());
+    return EnergyToDbfs(noise_energy_,
+                        static_cast<int>(frame.samples_per_channel()));
   }
 
  private:
@@ -247,11 +156,6 @@ class NoiseFloorEstimator : public NoiseLevelEstimator {
 
 }  // namespace
 
-std::unique_ptr<NoiseLevelEstimator> CreateStationaryNoiseEstimator(
-    ApmDataDumper* data_dumper) {
-  return std::make_unique<NoiseLevelEstimatorImpl>(data_dumper);
-}
-
 std::unique_ptr<NoiseLevelEstimator> CreateNoiseFloorEstimator(
     ApmDataDumper* data_dumper) {
   return std::make_unique<NoiseFloorEstimator>(data_dumper);
diff --git a/modules/audio_processing/agc2/noise_level_estimator.h b/modules/audio_processing/agc2/noise_level_estimator.h
index 94aecda7fc..9f3b957486 100644
--- a/modules/audio_processing/agc2/noise_level_estimator.h
+++ b/modules/audio_processing/agc2/noise_level_estimator.h
@@ -27,10 +27,6 @@ class NoiseLevelEstimator {
   virtual float Analyze(const AudioFrameView<const float>& frame) = 0;
 };
 
-// Creates a noise level estimator based on stationarity detection.
-std::unique_ptr<NoiseLevelEstimator> CreateStationaryNoiseEstimator(
-    ApmDataDumper* data_dumper);
-
 // Creates a noise level estimator based on noise floor detection.
 std::unique_ptr<NoiseLevelEstimator> CreateNoiseFloorEstimator(
     ApmDataDumper* data_dumper);
diff --git a/modules/audio_processing/agc2/noise_level_estimator_unittest.cc b/modules/audio_processing/agc2/noise_level_estimator_unittest.cc
index 51ad1ba00a..8168c5a229 100644
--- a/modules/audio_processing/agc2/noise_level_estimator_unittest.cc
+++ b/modules/audio_processing/agc2/noise_level_estimator_unittest.cc
@@ -50,45 +50,6 @@ class NoiseEstimatorParametrization : public ::testing::TestWithParam<int> {
   int sample_rate_hz() const { return GetParam(); }
 };
 
-// White random noise is stationary, but does not trigger the detector
-// every frame due to the randomness.
-TEST_P(NoiseEstimatorParametrization, StationaryNoiseEstimatorWithRandomNoise) {
-  ApmDataDumper data_dumper(0);
-  auto estimator = CreateStationaryNoiseEstimator(&data_dumper);
-
-  test::WhiteNoiseGenerator gen(/*min_amplitude=*/test::kMinS16,
-                                /*max_amplitude=*/test::kMaxS16);
-  const float noise_level_dbfs =
-      RunEstimator(gen, *estimator, sample_rate_hz());
-  EXPECT_NEAR(noise_level_dbfs, -5.5f, 1.0f);
-}
-
-// Sine curves are (very) stationary. They trigger the detector all
-// the time. Except for a few initial frames.
-TEST_P(NoiseEstimatorParametrization, StationaryNoiseEstimatorWithSineTone) {
-  ApmDataDumper data_dumper(0);
-  auto estimator = CreateStationaryNoiseEstimator(&data_dumper);
-
-  test::SineGenerator gen(/*amplitude=*/test::kMaxS16, /*frequency_hz=*/600.0f,
-                          sample_rate_hz());
-  const float noise_level_dbfs =
-      RunEstimator(gen, *estimator, sample_rate_hz());
-  EXPECT_NEAR(noise_level_dbfs, -3.0f, 1.0f);
-}
-
-// Pulses are transient if they are far enough apart. They shouldn't
-// trigger the noise detector.
-TEST_P(NoiseEstimatorParametrization, StationaryNoiseEstimatorWithPulseTone) {
-  ApmDataDumper data_dumper(0);
-  auto estimator = CreateStationaryNoiseEstimator(&data_dumper);
-
-  test::PulseGenerator gen(/*pulse_amplitude=*/test::kMaxS16,
-                           /*no_pulse_amplitude=*/10.0f, /*frequency_hz=*/20.0f,
-                           sample_rate_hz());
-  const int noise_level_dbfs = RunEstimator(gen, *estimator, sample_rate_hz());
-  EXPECT_NEAR(noise_level_dbfs, -79.0f, 1.0f);
-}
-
 // Checks that full scale white noise maps to about -5.5 dBFS.
 TEST_P(NoiseEstimatorParametrization, NoiseFloorEstimatorWithRandomNoise) {
   ApmDataDumper data_dumper(0);
@@ -122,7 +83,8 @@ TEST_P(NoiseEstimatorParametrization, NoiseFloorEstimatorWithPulseTone) {
   constexpr float kNoPulseAmplitude = 10.0f;
   test::PulseGenerator gen(/*pulse_amplitude=*/test::kMaxS16, kNoPulseAmplitude,
                            /*frequency_hz=*/20.0f, sample_rate_hz());
-  const int noise_level_dbfs = RunEstimator(gen, *estimator, sample_rate_hz());
+  const float noise_level_dbfs =
+      RunEstimator(gen, *estimator, sample_rate_hz());
   const float expected_noise_floor_dbfs =
       20.0f * std::log10f(kNoPulseAmplitude / test::kMaxS16);
   EXPECT_NEAR(noise_level_dbfs, expected_noise_floor_dbfs, 0.5f);
diff --git a/modules/audio_processing/agc2/noise_spectrum_estimator.cc b/modules/audio_processing/agc2/noise_spectrum_estimator.cc
deleted file mode 100644
index f283f4e27f..0000000000
--- a/modules/audio_processing/agc2/noise_spectrum_estimator.cc
+++ /dev/null
@@ -1,70 +0,0 @@
-/*
- *  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"
-#include "rtc_base/checks.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("agc2_noise_spectrum", 65, noise_spectrum_);
-  data_dumper_->DumpRaw("agc2_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
deleted file mode 100644
index e9895f0b1e..0000000000
--- a/modules/audio_processing/agc2/noise_spectrum_estimator.h
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- *  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"
-
-namespace webrtc {
-
-class ApmDataDumper;
-
-class NoiseSpectrumEstimator {
- public:
-  explicit NoiseSpectrumEstimator(ApmDataDumper* data_dumper);
-
-  NoiseSpectrumEstimator() = delete;
-  NoiseSpectrumEstimator(const NoiseSpectrumEstimator&) = delete;
-  NoiseSpectrumEstimator& operator=(const NoiseSpectrumEstimator&) = delete;
-
-  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];
-};
-
-}  // 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
deleted file mode 100644
index 3ef8dd775b..0000000000
--- a/modules/audio_processing/agc2/signal_classifier.cc
+++ /dev/null
@@ -1,177 +0,0 @@
-/*
- *  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/checks.h"
-#include "system_wrappers/include/cpu_features_wrapper.h"
-
-namespace webrtc {
-namespace {
-
-bool IsSse2Available() {
-#if defined(WEBRTC_ARCH_X86_FAMILY)
-  return GetCPUInfo(kSSE2) != 0;
-#else
-  return false;
-#endif
-}
-
-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("agc2_num_stationary_bands", 1, &num_stationary_bands);
-  data_dumper->DumpRaw("agc2_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_),
-      ooura_fft_(IsSse2Available()) {
-  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
deleted file mode 100644
index 20cce920f0..0000000000
--- a/modules/audio_processing/agc2/signal_classifier.h
+++ /dev/null
@@ -1,73 +0,0 @@
-/*
- *  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 "common_audio/third_party/ooura/fft_size_128/ooura_fft.h"
-#include "modules/audio_processing/agc2/down_sampler.h"
-#include "modules/audio_processing/agc2/noise_spectrum_estimator.h"
-
-namespace webrtc {
-
-class ApmDataDumper;
-class AudioBuffer;
-
-class SignalClassifier {
- public:
-  enum class SignalType { kNonStationary, kStationary };
-
-  explicit SignalClassifier(ApmDataDumper* data_dumper);
-
-  SignalClassifier() = delete;
-  SignalClassifier(const SignalClassifier&) = delete;
-  SignalClassifier& operator=(const SignalClassifier&) = delete;
-
-  ~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() = delete;
-    FrameExtender(const FrameExtender&) = delete;
-    FrameExtender& operator=(const FrameExtender&) = delete;
-
-    ~FrameExtender();
-
-    void ExtendFrame(rtc::ArrayView<const float> x,
-                     rtc::ArrayView<float> x_extended);
-
-   private:
-    std::vector<float> x_old_;
-  };
-
-  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_;
-};
-
-}  // 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
deleted file mode 100644
index f1a3a664f0..0000000000
--- a/modules/audio_processing/agc2/signal_classifier_unittest.cc
+++ /dev/null
@@ -1,86 +0,0 @@
-/*
- *  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 "api/function_view.h"
-#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 {
-constexpr int kNumIterations = 100;
-
-// Runs the signal classifier on audio generated by 'sample_generator'
-// for kNumIterations. Returns the number of frames classified as noise.
-float RunClassifier(rtc::FunctionView<float()> sample_generator,
-                    int sample_rate_hz) {
-  ApmDataDumper data_dumper(0);
-  SignalClassifier classifier(&data_dumper);
-  std::array<float, 480> signal;
-  classifier.Initialize(sample_rate_hz);
-  const size_t samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 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;
-}
-
-class SignalClassifierParametrization : public ::testing::TestWithParam<int> {
- protected:
-  int sample_rate_hz() const { return GetParam(); }
-};
-
-// White random noise is stationary, but does not trigger the detector
-// every frame due to the randomness.
-TEST_P(SignalClassifierParametrization, WhiteNoise) {
-  test::WhiteNoiseGenerator gen(/*min_amplitude=*/test::kMinS16,
-                                /*max_amplitude=*/test::kMaxS16);
-  const int number_of_noise_frames = RunClassifier(gen, sample_rate_hz());
-  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_P(SignalClassifierParametrization, SineTone) {
-  test::SineGenerator gen(/*amplitude=*/test::kMaxS16, /*frequency_hz=*/600.0f,
-                          sample_rate_hz());
-  const int number_of_noise_frames = RunClassifier(gen, sample_rate_hz());
-  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_P(SignalClassifierParametrization, PulseTone) {
-  test::PulseGenerator gen(/*pulse_amplitude=*/test::kMaxS16,
-                           /*no_pulse_amplitude=*/10.0f, /*frequency_hz=*/20.0f,
-                           sample_rate_hz());
-  const int number_of_noise_frames = RunClassifier(gen, sample_rate_hz());
-  EXPECT_EQ(number_of_noise_frames, 0);
-}
-
-INSTANTIATE_TEST_SUITE_P(GainController2SignalClassifier,
-                         SignalClassifierParametrization,
-                         ::testing::Values(8000, 16000, 32000, 48000));
-
-}  // namespace
-}  // namespace webrtc
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index 675a956203..5af8aa9ddb 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -3109,11 +3109,6 @@ TEST(AudioProcessing, GainController2ConfigEqual) {
   b_adaptive.dry_run = a_adaptive.dry_run;
   EXPECT_EQ(a, b);
 
-  a_adaptive.noise_estimator = AudioProcessing::Config::GainController2::
-      NoiseEstimator::kStationaryNoise;
-  b_adaptive.noise_estimator = a_adaptive.noise_estimator;
-  EXPECT_EQ(a, b);
-
   a_adaptive.vad_reset_period_ms++;
   b_adaptive.vad_reset_period_ms = a_adaptive.vad_reset_period_ms;
   EXPECT_EQ(a, b);
@@ -3171,11 +3166,6 @@ TEST(AudioProcessing, GainController2ConfigNotEqual) {
   EXPECT_NE(a, b);
   a_adaptive = b_adaptive;
 
-  a_adaptive.noise_estimator = AudioProcessing::Config::GainController2::
-      NoiseEstimator::kStationaryNoise;
-  EXPECT_NE(a, b);
-  a_adaptive = b_adaptive;
-
   a_adaptive.vad_reset_period_ms++;
   EXPECT_NE(a, b);
   a_adaptive = b_adaptive;
diff --git a/modules/audio_processing/include/audio_processing.cc b/modules/audio_processing/include/audio_processing.cc
index 44a90d6e76..21270a76af 100644
--- a/modules/audio_processing/include/audio_processing.cc
+++ b/modules/audio_processing/include/audio_processing.cc
@@ -46,17 +46,6 @@ std::string GainController1ModeToString(const Agc1Config::Mode& mode) {
   RTC_CHECK_NOTREACHED();
 }
 
-std::string GainController2NoiseEstimatorToString(
-    const Agc2Config::NoiseEstimator& type) {
-  switch (type) {
-    case Agc2Config::NoiseEstimator::kStationaryNoise:
-      return "StationaryNoise";
-    case Agc2Config::NoiseEstimator::kNoiseFloor:
-      return "NoiseFloor";
-  }
-  RTC_CHECK_NOTREACHED();
-}
-
 }  // namespace
 
 constexpr int AudioProcessing::kNativeSampleRatesHz[];
@@ -99,7 +88,6 @@ bool Agc1Config::operator==(const Agc1Config& rhs) const {
 bool Agc2Config::AdaptiveDigital::operator==(
     const Agc2Config::AdaptiveDigital& rhs) const {
   return enabled == rhs.enabled && dry_run == rhs.dry_run &&
-         noise_estimator == rhs.noise_estimator &&
          vad_reset_period_ms == rhs.vad_reset_period_ms &&
          adjacent_speech_frames_threshold ==
              rhs.adjacent_speech_frames_threshold &&
@@ -204,9 +192,6 @@ std::string AudioProcessing::Config::ToString() const {
       << " }, adaptive_digital: { enabled: "
       << gain_controller2.adaptive_digital.enabled
       << ", dry_run: " << gain_controller2.adaptive_digital.dry_run
-      << ", noise_estimator: "
-      << GainController2NoiseEstimatorToString(
-             gain_controller2.adaptive_digital.noise_estimator)
       << ", vad_reset_period_ms: "
       << gain_controller2.adaptive_digital.vad_reset_period_ms
       << ", adjacent_speech_frames_threshold: "
diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h
index 02a961ad7e..8887ca2ebd 100644
--- a/modules/audio_processing/include/audio_processing.h
+++ b/modules/audio_processing/include/audio_processing.h
@@ -402,7 +402,6 @@ class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
         bool enabled = false;
         // Run the adaptive digital controller but the signal is not modified.
         bool dry_run = false;
-        NoiseEstimator noise_estimator = kNoiseFloor;
         int vad_reset_period_ms = 1500;
         int adjacent_speech_frames_threshold = 12;
         float max_gain_change_db_per_second = 3.0f;
@@ -411,6 +410,7 @@ class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
         bool avx2_allowed = true;
         bool neon_allowed = true;
         // TODO(crbug.com/webrtc/7494): Remove deprecated settings below.
+        NoiseEstimator noise_estimator = kNoiseFloor;
         float vad_probability_attack = 1.0f;
         LevelEstimator level_estimator = kRms;
         int level_estimator_adjacent_speech_frames_threshold = 12;