diff --git a/api/audio/echo_canceller3_config.cc b/api/audio/echo_canceller3_config.cc
index aeb809efa9..5f1923e90f 100644
--- a/api/audio/echo_canceller3_config.cc
+++ b/api/audio/echo_canceller3_config.cc
@@ -153,6 +153,7 @@ bool EchoCanceller3Config::Validate(EchoCanceller3Config* config) {
 
   res = res & Limit(&c->filter.config_change_duration_blocks, 0, 100000);
   res = res & Limit(&c->filter.initial_state_seconds, 0.f, 100.f);
+  res = res & Limit(&c->filter.coarse_reset_hangover_blocks, 0, 2500);
 
   res = res & Limit(&c->erle.min, 1.f, 100000.f);
   res = res & Limit(&c->erle.max_l, 1.f, 100000.f);
diff --git a/api/audio/echo_canceller3_config.h b/api/audio/echo_canceller3_config.h
index 3ed11ff8b3..55281af93b 100644
--- a/api/audio/echo_canceller3_config.h
+++ b/api/audio/echo_canceller3_config.h
@@ -86,6 +86,7 @@ struct RTC_EXPORT EchoCanceller3Config {
 
     size_t config_change_duration_blocks = 250;
     float initial_state_seconds = 2.5f;
+    int coarse_reset_hangover_blocks = 25;
     bool conservative_initial_phase = false;
     bool enable_coarse_filter_output_usage = true;
     bool use_linear_filter = true;
diff --git a/api/audio/echo_canceller3_config_json.cc b/api/audio/echo_canceller3_config_json.cc
index 907b472714..9d10da9949 100644
--- a/api/audio/echo_canceller3_config_json.cc
+++ b/api/audio/echo_canceller3_config_json.cc
@@ -223,6 +223,8 @@ void Aec3ConfigFromJsonString(absl::string_view json_string,
               &cfg.filter.config_change_duration_blocks);
     ReadParam(section, "initial_state_seconds",
               &cfg.filter.initial_state_seconds);
+    ReadParam(section, "coarse_reset_hangover_blocks",
+              &cfg.filter.coarse_reset_hangover_blocks);
     ReadParam(section, "conservative_initial_phase",
               &cfg.filter.conservative_initial_phase);
     ReadParam(section, "enable_coarse_filter_output_usage",
@@ -502,6 +504,8 @@ std::string Aec3ConfigToJsonString(const EchoCanceller3Config& config) {
       << config.filter.config_change_duration_blocks << ",";
   ost << "\"initial_state_seconds\": " << config.filter.initial_state_seconds
       << ",";
+  ost << "\"coarse_reset_hangover_blocks\": "
+      << config.filter.coarse_reset_hangover_blocks << ",";
   ost << "\"conservative_initial_phase\": "
       << (config.filter.conservative_initial_phase ? "true" : "false") << ",";
   ost << "\"enable_coarse_filter_output_usage\": "
diff --git a/modules/audio_processing/aec3/aec_state.cc b/modules/audio_processing/aec3/aec_state.cc
index c7361093f3..5b31e3cb9f 100644
--- a/modules/audio_processing/aec3/aec_state.cc
+++ b/modules/audio_processing/aec3/aec_state.cc
@@ -322,6 +322,11 @@ void AecState::Update(
                         external_delay ? 1 : 0);
   data_dumper_->DumpRaw("aec3_filter_tail_freq_resp_est",
                         GetReverbFrequencyResponse());
+  data_dumper_->DumpRaw("aec3_subtractor_y2", subtractor_output[0].y2);
+  data_dumper_->DumpRaw("aec3_subtractor_e2_coarse",
+                        subtractor_output[0].e2_coarse);
+  data_dumper_->DumpRaw("aec3_subtractor_e2_refined",
+                        subtractor_output[0].e2_refined);
 }
 
 AecState::InitialState::InitialState(const EchoCanceller3Config& config)
diff --git a/modules/audio_processing/aec3/refined_filter_update_gain.cc b/modules/audio_processing/aec3/refined_filter_update_gain.cc
index 138329ad38..db5203d542 100644
--- a/modules/audio_processing/aec3/refined_filter_update_gain.cc
+++ b/modules/audio_processing/aec3/refined_filter_update_gain.cc
@@ -73,6 +73,7 @@ void RefinedFilterUpdateGain::Compute(
     rtc::ArrayView<const float> erl,
     size_t size_partitions,
     bool saturated_capture_signal,
+    bool disallow_leakage_diverged,
     FftData* gain_fft) {
   RTC_DCHECK(gain_fft);
   // Introducing shorter notation to improve readability.
@@ -125,7 +126,7 @@ void RefinedFilterUpdateGain::Compute(
 
   // H_error = H_error + factor * erl.
   for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
-    if (E2_coarse[k] >= E2_refined[k]) {
+    if (E2_refined[k] <= E2_coarse[k] || disallow_leakage_diverged) {
       H_error_[k] += current_config_.leakage_converged * erl[k];
     } else {
       H_error_[k] += current_config_.leakage_diverged * erl[k];
diff --git a/modules/audio_processing/aec3/refined_filter_update_gain.h b/modules/audio_processing/aec3/refined_filter_update_gain.h
index 5730979567..ae4fe84df5 100644
--- a/modules/audio_processing/aec3/refined_filter_update_gain.h
+++ b/modules/audio_processing/aec3/refined_filter_update_gain.h
@@ -51,6 +51,7 @@ class RefinedFilterUpdateGain {
                rtc::ArrayView<const float> erl,
                size_t size_partitions,
                bool saturated_capture_signal,
+               bool disallow_leakage_diverged,
                FftData* gain_fft);
 
   // Sets a new config.
diff --git a/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc b/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc
index 2393fddd6f..6fce858f16 100644
--- a/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc
+++ b/modules/audio_processing/aec3/refined_filter_update_gain_unittest.cc
@@ -196,7 +196,8 @@ void RunFilterUpdateTest(int num_blocks_to_process,
     std::array<float, kFftLengthBy2Plus1> erl;
     ComputeErl(optimization, H2[0], erl);
     refined_gain.Compute(render_power, render_signal_analyzer, output[0], erl,
-                         refined_filter.SizePartitions(), saturation, &G);
+                         refined_filter.SizePartitions(), saturation, false,
+                         &G);
     refined_filter.Adapt(*render_delay_buffer->GetRenderBuffer(), G, &h[0]);
 
     // Update the delay.
@@ -247,7 +248,7 @@ TEST(RefinedFilterUpdateGainDeathTest, NullDataOutputGain) {
   erl.fill(0.f);
   EXPECT_DEATH(
       gain.Compute(render_power, analyzer, output, erl,
-                   config.filter.refined.length_blocks, false, nullptr),
+                   config.filter.refined.length_blocks, false, false, nullptr),
       "");
 }
 
diff --git a/modules/audio_processing/aec3/subtractor.cc b/modules/audio_processing/aec3/subtractor.cc
index d15229934f..d10e4ffc52 100644
--- a/modules/audio_processing/aec3/subtractor.cc
+++ b/modules/audio_processing/aec3/subtractor.cc
@@ -19,11 +19,17 @@
 #include "modules/audio_processing/logging/apm_data_dumper.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/numerics/safe_minmax.h"
+#include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
 namespace {
 
+bool UseCoarseFilterResetHangover() {
+  return !field_trial::IsEnabled(
+      "WebRTC-Aec3CoarseFilterResetHangoverKillSwitch");
+}
+
 void PredictionError(const Aec3Fft& fft,
                      const FftData& S,
                      rtc::ArrayView<const float> y,
@@ -66,12 +72,14 @@ Subtractor::Subtractor(const EchoCanceller3Config& config,
       optimization_(optimization),
       config_(config),
       num_capture_channels_(num_capture_channels),
+      use_coarse_filter_reset_hangover_(UseCoarseFilterResetHangover()),
       refined_filters_(num_capture_channels_),
       coarse_filter_(num_capture_channels_),
       refined_gains_(num_capture_channels_),
       coarse_gains_(num_capture_channels_),
       filter_misadjustment_estimators_(num_capture_channels_),
       poor_coarse_filter_counters_(num_capture_channels_, 0),
+      coarse_filter_reset_hangover_(num_capture_channels_, 0),
       refined_frequency_responses_(
           num_capture_channels_,
           std::vector<std::array<float, kFftLengthBy2Plus1>>(
@@ -228,11 +236,19 @@ void Subtractor::Process(const RenderBuffer& render_buffer,
 
     // Update the refined filter.
     if (!refined_filters_adjusted) {
+      // Do not allow the performance of the coarse filter to affect the
+      // adaptation speed of the refined filter just after the coarse filter has
+      // been reset.
+      const bool disallow_leakage_diverged =
+          coarse_filter_reset_hangover_[ch] > 0 &&
+          use_coarse_filter_reset_hangover_;
+
       std::array<float, kFftLengthBy2Plus1> erl;
       ComputeErl(optimization_, refined_frequency_responses_[ch], erl);
       refined_gains_[ch]->Compute(X2_refined, render_signal_analyzer, output,
                                   erl, refined_filters_[ch]->SizePartitions(),
-                                  aec_state.SaturatedCapture(), &G);
+                                  aec_state.SaturatedCapture(),
+                                  disallow_leakage_diverged, &G);
     } else {
       G.re.fill(0.f);
       G.im.fill(0.f);
@@ -256,6 +272,8 @@ void Subtractor::Process(const RenderBuffer& render_buffer,
       coarse_gains_[ch]->Compute(X2_coarse, render_signal_analyzer, E_coarse,
                                  coarse_filter_[ch]->SizePartitions(),
                                  aec_state.SaturatedCapture(), &G);
+      coarse_filter_reset_hangover_[ch] =
+          std::max(coarse_filter_reset_hangover_[ch] - 1, 0);
     } else {
       poor_coarse_filter_counters_[ch] = 0;
       coarse_filter_[ch]->SetFilter(refined_filters_[ch]->SizePartitions(),
@@ -263,6 +281,8 @@ void Subtractor::Process(const RenderBuffer& render_buffer,
       coarse_gains_[ch]->Compute(X2_coarse, render_signal_analyzer, E_refined,
                                  coarse_filter_[ch]->SizePartitions(),
                                  aec_state.SaturatedCapture(), &G);
+      coarse_filter_reset_hangover_[ch] =
+          config_.filter.coarse_reset_hangover_blocks;
     }
 
     coarse_filter_[ch]->Adapt(render_buffer, G);
diff --git a/modules/audio_processing/aec3/subtractor.h b/modules/audio_processing/aec3/subtractor.h
index 42ca3729ca..560f6568eb 100644
--- a/modules/audio_processing/aec3/subtractor.h
+++ b/modules/audio_processing/aec3/subtractor.h
@@ -120,6 +120,7 @@ class Subtractor {
   const Aec3Optimization optimization_;
   const EchoCanceller3Config config_;
   const size_t num_capture_channels_;
+  const bool use_coarse_filter_reset_hangover_;
 
   std::vector<std::unique_ptr<AdaptiveFirFilter>> refined_filters_;
   std::vector<std::unique_ptr<AdaptiveFirFilter>> coarse_filter_;
@@ -127,6 +128,7 @@ class Subtractor {
   std::vector<std::unique_ptr<CoarseFilterUpdateGain>> coarse_gains_;
   std::vector<FilterMisadjustmentEstimator> filter_misadjustment_estimators_;
   std::vector<size_t> poor_coarse_filter_counters_;
+  std::vector<int> coarse_filter_reset_hangover_;
   std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>>
       refined_frequency_responses_;
   std::vector<std::vector<float>> refined_impulse_responses_;
diff --git a/resources/audio_processing/output_data_float.pb.sha1 b/resources/audio_processing/output_data_float.pb.sha1
index 6c3ab91815..c895b963f4 100644
--- a/resources/audio_processing/output_data_float.pb.sha1
+++ b/resources/audio_processing/output_data_float.pb.sha1
@@ -1 +1 @@
-ed1172c80a1a001a8aa7ac0680a99018cbb7d278
\ No newline at end of file
+365a02046fdb30357d649e73766d2f6eb2b33677
\ No newline at end of file
diff --git a/resources/audio_processing/output_data_float_avx2.pb.sha1 b/resources/audio_processing/output_data_float_avx2.pb.sha1
index 2d4ad0c141..12ec621bba 100644
--- a/resources/audio_processing/output_data_float_avx2.pb.sha1
+++ b/resources/audio_processing/output_data_float_avx2.pb.sha1
@@ -1 +1 @@
-a1dd718a6882bf8033a934e5beec73086cc91240
\ No newline at end of file
+847035cbe0bc7ee0620c32fa5ac857cc5b2c7ec4
\ No newline at end of file