diff --git a/webrtc/modules/audio_processing/aec3/main_filter_update_gain.cc b/webrtc/modules/audio_processing/aec3/main_filter_update_gain.cc
index 9cfb08bdb0..3c4959c11c 100644
--- a/webrtc/modules/audio_processing/aec3/main_filter_update_gain.cc
+++ b/webrtc/modules/audio_processing/aec3/main_filter_update_gain.cc
@@ -22,6 +22,7 @@ namespace webrtc {
 namespace {
 
 constexpr float kHErrorInitial = 10000.f;
+constexpr int kPoorExcitationCounterInitial = 1000;
 
 }  // namespace
 
@@ -30,7 +31,7 @@ int MainFilterUpdateGain::instance_count_ = 0;
 MainFilterUpdateGain::MainFilterUpdateGain()
     : data_dumper_(
           new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
-      poor_excitation_counter_(1000) {
+      poor_excitation_counter_(kPoorExcitationCounterInitial) {
   H_error_.fill(kHErrorInitial);
 }
 
@@ -38,6 +39,8 @@ MainFilterUpdateGain::~MainFilterUpdateGain() {}
 
 void MainFilterUpdateGain::HandleEchoPathChange() {
   H_error_.fill(kHErrorInitial);
+  poor_excitation_counter_ = kPoorExcitationCounterInitial;
+  call_counter_ = 0;
 }
 
 void MainFilterUpdateGain::Compute(
diff --git a/webrtc/modules/audio_processing/aec3/render_delay_buffer.cc b/webrtc/modules/audio_processing/aec3/render_delay_buffer.cc
index 6e6766ab4d..12dae1b8b4 100644
--- a/webrtc/modules/audio_processing/aec3/render_delay_buffer.cc
+++ b/webrtc/modules/audio_processing/aec3/render_delay_buffer.cc
@@ -97,6 +97,7 @@ class RenderDelayBufferImpl final : public RenderDelayBuffer {
   DownsampledRenderBuffer downsampled_render_buffer_;
   DecimatorBy4 render_decimator_;
   ApiCallJitterBuffer api_call_jitter_buffer_;
+  const std::vector<std::vector<float>> zero_block_;
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderDelayBufferImpl);
 };
 
@@ -107,7 +108,8 @@ RenderDelayBufferImpl::RenderDelayBufferImpl(size_t num_bands)
           num_bands,
           std::max(kResidualEchoPowerRenderWindowSize, kAdaptiveFilterLength),
           std::vector<size_t>(1, kAdaptiveFilterLength)),
-      api_call_jitter_buffer_(num_bands) {
+      api_call_jitter_buffer_(num_bands),
+      zero_block_(num_bands, std::vector<float>(kBlockSize, 0.f)) {
   buffer_.fill(std::vector<std::vector<float>>(
       num_bands, std::vector<float>(kBlockSize, 0.f)));
 
@@ -147,12 +149,11 @@ bool RenderDelayBufferImpl::Insert(
 
 bool RenderDelayBufferImpl::UpdateBuffers() {
   bool underrun = true;
-  // Update the buffers with a new block if such is available, otherwise repeat
-  // the previous block.
+  // Update the buffers with a new block if such is available, otherwise insert
+  // a block of silence.
   if (api_call_jitter_buffer_.Size() > 0) {
     last_insert_index_ = (last_insert_index_ + 1) % buffer_.size();
     api_call_jitter_buffer_.Remove(&buffer_[last_insert_index_]);
-
     underrun = false;
   }
 
@@ -162,14 +163,22 @@ bool RenderDelayBufferImpl::UpdateBuffers() {
       downsampled_render_buffer_.buffer.size();
 
   std::array<float, kSubBlockSize> render_downsampled;
-  render_decimator_.Decimate(buffer_[last_insert_index_][0],
-                             render_downsampled);
+  if (underrun) {
+    render_decimator_.Decimate(zero_block_[0], render_downsampled);
+  } else {
+    render_decimator_.Decimate(buffer_[last_insert_index_][0],
+                               render_downsampled);
+  }
   std::copy(render_downsampled.rbegin(), render_downsampled.rend(),
             downsampled_render_buffer_.buffer.begin() +
                 downsampled_render_buffer_.position);
 
-  fft_buffer_.Insert(
-      buffer_[(last_insert_index_ - delay_ + buffer_.size()) % buffer_.size()]);
+  if (underrun) {
+    fft_buffer_.Insert(zero_block_);
+  } else {
+    fft_buffer_.Insert(buffer_[(last_insert_index_ - delay_ + buffer_.size()) %
+                               buffer_.size()]);
+  }
   return !underrun;
 }
 
diff --git a/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.cc b/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.cc
index 85bc11fb0d..61a1f575b1 100644
--- a/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.cc
+++ b/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.cc
@@ -17,6 +17,13 @@
 
 namespace webrtc {
 
+void ShadowFilterUpdateGain::HandleEchoPathChange() {
+  // TODO(peah): Check whether this counter should instead be initialized to a
+  // large value.
+  poor_signal_excitation_counter_ = 0;
+  call_counter_ = 0;
+}
+
 void ShadowFilterUpdateGain::Compute(
     const RenderBuffer& render_buffer,
     const RenderSignalAnalyzer& render_signal_analyzer,
diff --git a/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.h b/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.h
index a67b8fb636..72218b6608 100644
--- a/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.h
+++ b/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.h
@@ -21,6 +21,9 @@ namespace webrtc {
 // Provides functionality for computing the fixed gain for the shadow filter.
 class ShadowFilterUpdateGain {
  public:
+  // Takes action in the case of a known echo path change.
+  void HandleEchoPathChange();
+
   // Computes the gain.
   void Compute(const RenderBuffer& render_buffer,
                const RenderSignalAnalyzer& render_signal_analyzer,
@@ -30,6 +33,8 @@ class ShadowFilterUpdateGain {
                FftData* G);
 
  private:
+  // TODO(peah): Check whether this counter should instead be initialized to a
+  // large value.
   size_t poor_signal_excitation_counter_ = 0;
   size_t call_counter_ = 0;
 };
diff --git a/webrtc/modules/audio_processing/aec3/subtractor.cc b/webrtc/modules/audio_processing/aec3/subtractor.cc
index 4b462238e2..3897d947e1 100644
--- a/webrtc/modules/audio_processing/aec3/subtractor.cc
+++ b/webrtc/modules/audio_processing/aec3/subtractor.cc
@@ -55,6 +55,7 @@ void Subtractor::HandleEchoPathChange(
     main_filter_.HandleEchoPathChange();
     shadow_filter_.HandleEchoPathChange();
     G_main_.HandleEchoPathChange();
+    G_shadow_.HandleEchoPathChange();
   }
 }