diff --git a/webrtc/media/engine/fakewebrtcvoiceengine.h b/webrtc/media/engine/fakewebrtcvoiceengine.h
index f27810cefe..c8fb9cfe02 100644
--- a/webrtc/media/engine/fakewebrtcvoiceengine.h
+++ b/webrtc/media/engine/fakewebrtcvoiceengine.h
@@ -93,7 +93,6 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
   WEBRTC_STUB_CONST(stream_delay_ms, ());
   WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ());
   WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed));
-  WEBRTC_VOID_STUB(SetLevelControllerInitialLevel, (float level));
   WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset));
   WEBRTC_STUB_CONST(delay_offset_ms, ());
   WEBRTC_STUB(StartDebugRecording,
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc
index 750bcb6fd3..3b3a9518b1 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc
@@ -992,11 +992,6 @@ void AudioProcessingImpl::set_stream_key_pressed(bool key_pressed) {
   capture_.key_pressed = key_pressed;
 }
 
-void AudioProcessingImpl::SetLevelControllerInitialLevel(float level) {
-  rtc::CritScope cs(&crit_capture_);
-  private_submodules_->level_controller->SetInitialLevel(level);
-}
-
 void AudioProcessingImpl::set_delay_offset_ms(int offset) {
   rtc::CritScope cs(&crit_capture_);
   capture_.delay_offset_ms = offset;
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index 5ba3638e21..4b9011dc88 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -82,7 +82,6 @@ class AudioProcessingImpl : public AudioProcessing {
   void set_delay_offset_ms(int offset) override;
   int delay_offset_ms() const override;
   void set_stream_key_pressed(bool key_pressed) override;
-  void SetLevelControllerInitialLevel(float level) override;
 
   // Render-side exclusive methods possibly running APM in a
   // multi-threaded manner. Acquire the render lock.
diff --git a/webrtc/modules/audio_processing/include/audio_processing.h b/webrtc/modules/audio_processing/include/audio_processing.h
index 2f1ae99b93..09e5d5be1c 100644
--- a/webrtc/modules/audio_processing/include/audio_processing.h
+++ b/webrtc/modules/audio_processing/include/audio_processing.h
@@ -410,11 +410,6 @@ class AudioProcessing {
   // with this chunk of audio.
   virtual void set_stream_key_pressed(bool key_pressed) = 0;
 
-  // Sets the initial peak level to use inside the level controller in order
-  // to compute the signal gain. The unit for the peak level is dBFS and
-  // the allowed range is [-100, 0].
-  virtual void SetLevelControllerInitialLevel(float level) = 0;
-
   // Sets a delay |offset| in ms to add to the values passed in through
   // set_stream_delay_ms(). May be positive or negative.
   //
diff --git a/webrtc/modules/audio_processing/level_controller/gain_selector.cc b/webrtc/modules/audio_processing/level_controller/gain_selector.cc
index 80d9c0f1c5..2accd7180c 100644
--- a/webrtc/modules/audio_processing/level_controller/gain_selector.cc
+++ b/webrtc/modules/audio_processing/level_controller/gain_selector.cc
@@ -42,12 +42,10 @@ void GainSelector::Initialize(int sample_rate_hz) {
 float GainSelector::GetNewGain(float peak_level,
                                float noise_energy,
                                float saturating_gain,
-                               bool gain_jumpstart,
                                SignalClassifier::SignalType signal_type) {
   RTC_DCHECK_LT(0.f, peak_level);
 
-  if (signal_type == SignalClassifier::SignalType::kHighlyNonStationary ||
-      gain_jumpstart) {
+  if (signal_type == SignalClassifier::SignalType::kHighlyNonStationary) {
     highly_nonstationary_signal_hold_counter_ = 100;
   } else {
     highly_nonstationary_signal_hold_counter_ =
diff --git a/webrtc/modules/audio_processing/level_controller/gain_selector.h b/webrtc/modules/audio_processing/level_controller/gain_selector.h
index 78b9101500..3d00499652 100644
--- a/webrtc/modules/audio_processing/level_controller/gain_selector.h
+++ b/webrtc/modules/audio_processing/level_controller/gain_selector.h
@@ -24,7 +24,6 @@ class GainSelector {
   float GetNewGain(float peak_level,
                    float noise_energy,
                    float saturating_gain,
-                   bool gain_jumpstart,
                    SignalClassifier::SignalType signal_type);
 
  private:
diff --git a/webrtc/modules/audio_processing/level_controller/level_controller.cc b/webrtc/modules/audio_processing/level_controller/level_controller.cc
index 07618e3ba7..a9fed9bf95 100644
--- a/webrtc/modules/audio_processing/level_controller/level_controller.cc
+++ b/webrtc/modules/audio_processing/level_controller/level_controller.cc
@@ -155,11 +155,6 @@ LevelController::LevelController()
 
 LevelController::~LevelController() {}
 
-void LevelController::SetInitialLevel(float level) {
-  peak_level_estimator_.SetInitialPeakLevel(level);
-  gain_jumpstart_ = true;
-}
-
 void LevelController::Initialize(int sample_rate_hz) {
   RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
              sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
@@ -211,11 +206,8 @@ void LevelController::Process(AudioBuffer* audio) {
   float saturating_gain = saturating_gain_estimator_.GetGain();
 
   // Compute the new gain to apply.
-  last_gain_ = gain_selector_.GetNewGain(
-      peak_level, noise_energy, saturating_gain, gain_jumpstart_, signal_type);
-
-  // Unflag the jumpstart of the gain as it should only happen once.
-  gain_jumpstart_ = false;
+  last_gain_ = gain_selector_.GetNewGain(peak_level, noise_energy,
+                                         saturating_gain, signal_type);
 
   // Apply the gain to the signal.
   int num_saturations = gain_applier_.Process(last_gain_, audio);
diff --git a/webrtc/modules/audio_processing/level_controller/level_controller.h b/webrtc/modules/audio_processing/level_controller/level_controller.h
index 1d7f174562..3d203f908d 100644
--- a/webrtc/modules/audio_processing/level_controller/level_controller.h
+++ b/webrtc/modules/audio_processing/level_controller/level_controller.h
@@ -38,11 +38,6 @@ class LevelController {
   void Process(AudioBuffer* audio);
   float GetLastGain() { return last_gain_; }
 
-  // Sets the initial peak level to use inside the level controller in order
-  // to compute the signal gain. The unit for the peak level is dBFS and
-  // the allowed range is [-100, 0].
-  void SetInitialLevel(float level);
-
  private:
   class Metrics {
    public:
@@ -76,7 +71,6 @@ class LevelController {
   float dc_level_[2];
   float dc_forgetting_factor_;
   float last_gain_;
-  bool gain_jumpstart_ = false;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(LevelController);
 };
diff --git a/webrtc/modules/audio_processing/level_controller/level_controller_unittest.cc b/webrtc/modules/audio_processing/level_controller/level_controller_unittest.cc
index 74d129f323..c470c2fad7 100644
--- a/webrtc/modules/audio_processing/level_controller/level_controller_unittest.cc
+++ b/webrtc/modules/audio_processing/level_controller/level_controller_unittest.cc
@@ -12,7 +12,6 @@
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/array_view.h"
-#include "webrtc/base/optional.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/level_controller/level_controller.h"
@@ -28,13 +27,9 @@ const int kNumFramesToProcess = 1000;
 // any errors.
 void RunBitexactnessTest(int sample_rate_hz,
                          size_t num_channels,
-                         rtc::Optional<float> initial_level,
                          rtc::ArrayView<const float> output_reference) {
   LevelController level_controller;
   level_controller.Initialize(sample_rate_hz);
-  if (initial_level) {
-    level_controller.SetInitialLevel(*initial_level);
-  }
 
   int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
   const StreamConfig capture_config(sample_rate_hz, num_channels, false);
@@ -76,19 +71,19 @@ void RunBitexactnessTest(int sample_rate_hz,
 TEST(LevelControlBitExactnessTest, DISABLED_Mono8kHz) {
   const float kOutputReference[] = {-0.013939f, -0.012154f, -0.009054f};
   RunBitexactnessTest(AudioProcessing::kSampleRate8kHz, 1,
-                      rtc::Optional<float>(), kOutputReference);
+                      kOutputReference);
 }
 
 TEST(LevelControlBitExactnessTest, DISABLED_Mono16kHz) {
   const float kOutputReference[] = {-0.013706f, -0.013215f, -0.013018f};
   RunBitexactnessTest(AudioProcessing::kSampleRate16kHz, 1,
-                      rtc::Optional<float>(), kOutputReference);
+                      kOutputReference);
 }
 
 TEST(LevelControlBitExactnessTest, DISABLED_Mono32kHz) {
   const float kOutputReference[] = {-0.014495f, -0.016425f, -0.016085f};
   RunBitexactnessTest(AudioProcessing::kSampleRate32kHz, 1,
-                      rtc::Optional<float>(), kOutputReference);
+                      kOutputReference);
 }
 
 // TODO(peah): Investigate why this particular testcase differ between Android
@@ -101,41 +96,35 @@ TEST(LevelControlBitExactnessTest, DISABLED_Mono48kHz) {
   const float kOutputReference[] = {-0.015949f, -0.016957f, -0.019478f};
 #endif
   RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 1,
-                      rtc::Optional<float>(), kOutputReference);
+                      kOutputReference);
 }
 
 TEST(LevelControlBitExactnessTest, DISABLED_Stereo8kHz) {
   const float kOutputReference[] = {-0.014063f, -0.008450f, -0.012159f,
                                     -0.051967f, -0.023202f, -0.047858f};
   RunBitexactnessTest(AudioProcessing::kSampleRate8kHz, 2,
-                      rtc::Optional<float>(), kOutputReference);
+                      kOutputReference);
 }
 
 TEST(LevelControlBitExactnessTest, DISABLED_Stereo16kHz) {
   const float kOutputReference[] = {-0.012714f, -0.005896f, -0.012220f,
                                     -0.053306f, -0.024549f, -0.051527f};
   RunBitexactnessTest(AudioProcessing::kSampleRate16kHz, 2,
-                      rtc::Optional<float>(), kOutputReference);
+                      kOutputReference);
 }
 
 TEST(LevelControlBitExactnessTest, DISABLED_Stereo32kHz) {
   const float kOutputReference[] = {-0.011737f, -0.007018f, -0.013446f,
                                     -0.053505f, -0.026292f, -0.056221f};
   RunBitexactnessTest(AudioProcessing::kSampleRate32kHz, 2,
-                      rtc::Optional<float>(), kOutputReference);
+                      kOutputReference);
 }
 
 TEST(LevelControlBitExactnessTest, DISABLED_Stereo48kHz) {
   const float kOutputReference[] = {-0.010643f, -0.006334f, -0.011377f,
                                     -0.049088f, -0.023600f, -0.050465f};
   RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 2,
-                      rtc::Optional<float>(), kOutputReference);
-}
-
-TEST(LevelControlBitExactnessTest, DISABLED_MonoInitial48kHz) {
-  const float kOutputReference[] = {-0.013753f, -0.014623f, -0.016797f};
-  RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 1,
-                      rtc::Optional<float>(2000), kOutputReference);
+                      kOutputReference);
 }
 
 
diff --git a/webrtc/modules/audio_processing/level_controller/peak_level_estimator.cc b/webrtc/modules/audio_processing/level_controller/peak_level_estimator.cc
index 9d4fe3312c..2ba806c8ee 100644
--- a/webrtc/modules/audio_processing/level_controller/peak_level_estimator.cc
+++ b/webrtc/modules/audio_processing/level_controller/peak_level_estimator.cc
@@ -13,14 +13,10 @@
 #include <algorithm>
 
 #include "webrtc/modules/audio_processing/audio_buffer.h"
+#include "webrtc/modules/audio_processing/level_controller/lc_constants.h"
 #include "webrtc/modules/audio_processing/logging/apm_data_dumper.h"
 
 namespace webrtc {
-namespace {
-
-const float kMinLevel = 30.f;
-
-}  // namespace
 
 PeakLevelEstimator::PeakLevelEstimator() {
   Initialize();
@@ -29,26 +25,15 @@ PeakLevelEstimator::PeakLevelEstimator() {
 PeakLevelEstimator::~PeakLevelEstimator() {}
 
 void PeakLevelEstimator::Initialize() {
-  peak_level_ = initial_peak_level_;
+  peak_level_ = kTargetLcPeakLevel;
   hold_counter_ = 0;
   initialization_phase_ = true;
 }
 
-void PeakLevelEstimator::SetInitialPeakLevel(float level) {
-  RTC_DCHECK_LE(-100.f, level);
-  RTC_DCHECK_GE(0.f, level);
-
-  float linear_level = std::pow(10.f, level / 20.f) * 32768.f;
-
-  // Limit the supplied level to the level range used internally.
-  initial_peak_level_ = std::max(linear_level, kMinLevel);
-  Initialize();
-}
-
 float PeakLevelEstimator::Analyze(SignalClassifier::SignalType signal_type,
                                   float frame_peak_level) {
   if (frame_peak_level == 0) {
-    RTC_DCHECK_LE(kMinLevel, peak_level_);
+    RTC_DCHECK_LE(30.f, peak_level_);
     return peak_level_;
   }
 
@@ -72,7 +57,7 @@ float PeakLevelEstimator::Analyze(SignalClassifier::SignalType signal_type,
     }
   }
 
-  peak_level_ = std::max(peak_level_, kMinLevel);
+  peak_level_ = std::max(peak_level_, 30.f);
 
   return peak_level_;
 }
diff --git a/webrtc/modules/audio_processing/level_controller/peak_level_estimator.h b/webrtc/modules/audio_processing/level_controller/peak_level_estimator.h
index 941dd36be3..270bbc3ad0 100644
--- a/webrtc/modules/audio_processing/level_controller/peak_level_estimator.h
+++ b/webrtc/modules/audio_processing/level_controller/peak_level_estimator.h
@@ -12,7 +12,6 @@
 #define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_PEAK_LEVEL_ESTIMATOR_H_
 
 #include "webrtc/base/constructormagic.h"
-#include "webrtc/modules/audio_processing/level_controller/lc_constants.h"
 #include "webrtc/modules/audio_processing/level_controller/signal_classifier.h"
 
 namespace webrtc {
@@ -24,11 +23,9 @@ class PeakLevelEstimator {
   void Initialize();
   float Analyze(SignalClassifier::SignalType signal_type,
                 float frame_peak_level);
-  void SetInitialPeakLevel(float level);
 
  private:
   float peak_level_;
-  float initial_peak_level_ = kTargetLcPeakLevel;
   int hold_counter_;
   bool initialization_phase_;