diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc b/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
index 6f1635f..69673c0 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
+++ b/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
@@ -346,11 +346,14 @@
         config.filter.main.length_blocks, config.filter.main.length_blocks,
         config.filter.config_change_duration_blocks, num_render_channels,
         DetectOptimization(), &data_dumper);
-    std::vector<std::array<float, kFftLengthBy2Plus1>> H2(
-        filter.max_filter_size_partitions(),
-        std::array<float, kFftLengthBy2Plus1>());
-    std::vector<float> h(
-        GetTimeDomainLength(filter.max_filter_size_partitions()), 0.f);
+    std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>> H2(
+        kNumCaptureChannels, std::vector<std::array<float, kFftLengthBy2Plus1>>(
+                                 filter.max_filter_size_partitions(),
+                                 std::array<float, kFftLengthBy2Plus1>()));
+    std::vector<std::vector<float>> h(
+        kNumCaptureChannels,
+        std::vector<float>(
+            GetTimeDomainLength(filter.max_filter_size_partitions()), 0.f));
     Aec3Fft fft;
     config.delay.default_delay = 1;
     std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
@@ -454,11 +457,11 @@
         render_buffer->SpectralSum(filter.SizePartitions(), &render_power);
         gain.Compute(render_power, render_signal_analyzer, E,
                      filter.SizePartitions(), false, &G);
-        filter.Adapt(*render_buffer, G, &h);
+        filter.Adapt(*render_buffer, G, &h[0]);
         aec_state.HandleEchoPathChange(EchoPathVariability(
             false, EchoPathVariability::DelayAdjustment::kNone, false));
 
-        filter.ComputeFrequencyResponse(&H2);
+        filter.ComputeFrequencyResponse(&H2[0]);
         aec_state.Update(delay_estimate, H2, h, *render_buffer, E2_main, Y2,
                          output);
       }
diff --git a/modules/audio_processing/aec3/aec_state.cc b/modules/audio_processing/aec3/aec_state.cc
index 4b30d30..803a598 100644
--- a/modules/audio_processing/aec3/aec_state.cc
+++ b/modules/audio_processing/aec3/aec_state.cc
@@ -66,18 +66,26 @@
       filter_quality_state_(config_),
       erl_estimator_(2 * kNumBlocksPerSecond),
       erle_estimator_(2 * kNumBlocksPerSecond, config_, num_capture_channels),
-      filter_analyzer_(config_),
+      max_echo_path_gain_(config_.ep_strength.default_gain),
+      filter_analyzers_(num_capture_channels),
       echo_audibility_(
           config_.echo_audibility.use_stationarity_properties_at_init),
       reverb_model_estimator_(config_),
-      subtractor_output_analyzers_(num_capture_channels) {}
+      subtractor_output_analyzers_(num_capture_channels) {
+  for (size_t ch = 0; ch < num_capture_channels; ++ch) {
+    filter_analyzers_[ch] = std::make_unique<FilterAnalyzer>(config_);
+  }
+}
 
 AecState::~AecState() = default;
 
 void AecState::HandleEchoPathChange(
     const EchoPathVariability& echo_path_variability) {
   const auto full_reset = [&]() {
-    filter_analyzer_.Reset();
+    for (auto& filter_analyzer : filter_analyzers_) {
+      filter_analyzer->Reset();
+    }
+    max_echo_path_gain_ = config_.ep_strength.default_gain;
     capture_signal_saturation_ = false;
     strong_not_saturated_render_blocks_ = 0;
     blocks_with_active_render_ = 0;
@@ -104,26 +112,43 @@
 
 void AecState::Update(
     const absl::optional<DelayEstimate>& external_delay,
-    const std::vector<std::array<float, kFftLengthBy2Plus1>>&
+    rtc::ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
         adaptive_filter_frequency_response,
-    const std::vector<float>& adaptive_filter_impulse_response,
+    rtc::ArrayView<const std::vector<float>> adaptive_filter_impulse_response,
     const RenderBuffer& render_buffer,
     const std::array<float, kFftLengthBy2Plus1>& E2_main,
     const std::array<float, kFftLengthBy2Plus1>& Y2,
     rtc::ArrayView<const SubtractorOutput> subtractor_output) {
-  RTC_DCHECK_EQ(subtractor_output.size(), subtractor_output_analyzers_.size());
+  const size_t num_capture_channels = filter_analyzers_.size();
+  RTC_DCHECK_EQ(num_capture_channels, subtractor_output.size());
+  RTC_DCHECK_EQ(num_capture_channels, subtractor_output_analyzers_.size());
+  RTC_DCHECK_EQ(num_capture_channels,
+                adaptive_filter_frequency_response.size());
+  RTC_DCHECK_EQ(num_capture_channels, adaptive_filter_impulse_response.size());
 
-  // Analyze the filter output.
+  // Analyze the filter outputs and filters.
+  bool any_filter_converged = false;
+  bool all_filters_diverged = true;
+  bool any_filter_consistent = false;
+  max_echo_path_gain_ = 0.f;
   for (size_t ch = 0; ch < subtractor_output.size(); ++ch) {
     subtractor_output_analyzers_[ch].Update(subtractor_output[ch]);
-  }
+    any_filter_converged = any_filter_converged ||
+                           subtractor_output_analyzers_[ch].ConvergedFilter();
+    all_filters_diverged = all_filters_diverged &&
+                           subtractor_output_analyzers_[ch].DivergedFilter();
 
-  // Analyze the properties of the filter.
-  filter_analyzer_.Update(adaptive_filter_impulse_response, render_buffer);
+    filter_analyzers_[ch]->Update(adaptive_filter_impulse_response[ch],
+                                  render_buffer);
+    any_filter_consistent =
+        any_filter_consistent || filter_analyzers_[ch]->Consistent();
+    max_echo_path_gain_ =
+        std::max(max_echo_path_gain_, filter_analyzers_[ch]->Gain());
+  }
 
   // Estimate the direct path delay of the filter.
   if (config_.filter.use_linear_filter) {
-    delay_state_.Update(filter_analyzer_, external_delay,
+    delay_state_.Update(filter_analyzers_, external_delay,
                         strong_not_saturated_render_blocks_);
   }
 
@@ -170,7 +195,7 @@
                                           /*channel=*/0);
   const auto& X2_input_erle = X2_reverb;
 
-  erle_estimator_.Update(render_buffer, adaptive_filter_frequency_response,
+  erle_estimator_.Update(render_buffer, adaptive_filter_frequency_response[0],
                          X2_input_erle, Y2, E2_main,
                          subtractor_output_analyzers_[0].ConvergedFilter(),
                          config_.erle.onset_detection);
@@ -188,24 +213,22 @@
 
   // Detect whether the transparent mode should be activated.
   transparent_state_.Update(delay_state_.DirectPathFilterDelay(),
-                            filter_analyzer_.Consistent(),
-                            subtractor_output_analyzers_[0].ConvergedFilter(),
-                            subtractor_output_analyzers_[0].DivergedFilter(),
-                            active_render, SaturatedCapture());
+                            any_filter_consistent, any_filter_converged,
+                            all_filters_diverged, active_render,
+                            SaturatedCapture());
 
   // Analyze the quality of the filter.
-  filter_quality_state_.Update(
-      active_render, TransparentMode(), SaturatedCapture(),
-      filter_analyzer_.Consistent(), external_delay,
-      subtractor_output_analyzers_[0].ConvergedFilter());
+  filter_quality_state_.Update(active_render, TransparentMode(),
+                               SaturatedCapture(), external_delay,
+                               any_filter_converged);
 
   // Update the reverb estimate.
   const bool stationary_block =
       config_.echo_audibility.use_stationarity_properties &&
       echo_audibility_.IsBlockStationary();
 
-  reverb_model_estimator_.Update(filter_analyzer_.GetAdjustedFilter(),
-                                 adaptive_filter_frequency_response,
+  reverb_model_estimator_.Update(filter_analyzers_[0]->GetAdjustedFilter(),
+                                 adaptive_filter_frequency_response[0],
                                  erle_estimator_.GetInstLinearQualityEstimate(),
                                  delay_state_.DirectPathFilterDelay(),
                                  UsableLinearEstimate(), stationary_block);
@@ -217,18 +240,16 @@
   data_dumper_->DumpRaw("aec3_erle", Erle()[0]);
   data_dumper_->DumpRaw("aec3_usable_linear_estimate", UsableLinearEstimate());
   data_dumper_->DumpRaw("aec3_transparent_mode", TransparentMode());
-  data_dumper_->DumpRaw("aec3_filter_delay", filter_analyzer_.DelayBlocks());
+  data_dumper_->DumpRaw("aec3_filter_delay",
+                        filter_analyzers_[0]->DelayBlocks());
 
-  data_dumper_->DumpRaw("aec3_consistent_filter",
-                        filter_analyzer_.Consistent());
+  data_dumper_->DumpRaw("aec3_any_filter_consistent", any_filter_consistent);
   data_dumper_->DumpRaw("aec3_initial_state",
                         initial_state_.InitialStateActive());
   data_dumper_->DumpRaw("aec3_capture_saturation", SaturatedCapture());
   data_dumper_->DumpRaw("aec3_echo_saturation", SaturatedEcho());
-  data_dumper_->DumpRaw("aec3_converged_filter",
-                        subtractor_output_analyzers_[0].ConvergedFilter());
-  data_dumper_->DumpRaw("aec3_diverged_filter",
-                        subtractor_output_analyzers_[0].DivergedFilter());
+  data_dumper_->DumpRaw("aec3_any_filter_converged", any_filter_converged);
+  data_dumper_->DumpRaw("aec3_all_filters_diverged", all_filters_diverged);
 
   data_dumper_->DumpRaw("aec3_external_delay_avaliable",
                         external_delay ? 1 : 0);
@@ -268,7 +289,7 @@
     : delay_headroom_samples_(config.delay.delay_headroom_samples) {}
 
 void AecState::FilterDelay::Update(
-    const FilterAnalyzer& filter_analyzer,
+    const std::vector<std::unique_ptr<FilterAnalyzer>>& filter_analyzers,
     const absl::optional<DelayEstimate>& external_delay,
     size_t blocks_with_proper_filter_adaptation) {
   // Update the delay based on the external delay.
@@ -285,7 +306,12 @@
   if (delay_estimator_may_not_have_converged && external_delay_) {
     filter_delay_blocks_ = delay_headroom_samples_ / kBlockSize;
   } else {
-    filter_delay_blocks_ = filter_analyzer.DelayBlocks();
+    // Conservatively use the min delay among the filters.
+    filter_delay_blocks_ = filter_analyzers[0]->DelayBlocks();
+    for (size_t ch = 1; ch < filter_analyzers.size(); ++ch) {
+      filter_delay_blocks_ =
+          std::min(filter_delay_blocks_, filter_analyzers[ch]->DelayBlocks());
+    }
   }
 }
 
@@ -306,16 +332,16 @@
 }
 
 void AecState::TransparentMode::Update(int filter_delay_blocks,
-                                       bool consistent_filter,
-                                       bool converged_filter,
-                                       bool diverged_filter,
+                                       bool any_filter_consistent,
+                                       bool any_filter_converged,
+                                       bool all_filters_diverged,
                                        bool active_render,
                                        bool saturated_capture) {
   ++capture_block_counter_;
   strong_not_saturated_render_blocks_ +=
       active_render && !saturated_capture ? 1 : 0;
 
-  if (consistent_filter && filter_delay_blocks < 5) {
+  if (any_filter_consistent && filter_delay_blocks < 5) {
     sane_filter_observed_ = true;
     active_blocks_since_sane_filter_ = 0;
   } else if (active_render) {
@@ -331,7 +357,7 @@
         active_blocks_since_sane_filter_ <= 30 * kNumBlocksPerSecond;
   }
 
-  if (converged_filter) {
+  if (any_filter_converged) {
     recent_convergence_during_activity_ = true;
     active_non_converged_sequence_size_ = 0;
     non_converged_sequence_size_ = 0;
@@ -347,7 +373,7 @@
     }
   }
 
-  if (!diverged_filter) {
+  if (!all_filters_diverged) {
     diverged_sequence_size_ = 0;
   } else if (++diverged_sequence_size_ >= 60) {
     // TODO(peah): Change these lines to ensure proper triggering of usable
@@ -387,16 +413,15 @@
     bool active_render,
     bool transparent_mode,
     bool saturated_capture,
-    bool consistent_estimate_,
     const absl::optional<DelayEstimate>& external_delay,
-    bool converged_filter) {
+    bool any_filter_converged) {
   // Update blocks counter.
   const bool filter_update = active_render && !saturated_capture;
   filter_update_blocks_since_reset_ += filter_update ? 1 : 0;
   filter_update_blocks_since_start_ += filter_update ? 1 : 0;
 
   // Store convergence flag when observed.
-  convergence_seen_ = convergence_seen_ || converged_filter;
+  convergence_seen_ = convergence_seen_ || any_filter_converged;
 
   // Verify requirements for achieving a decent filter. The requirements for
   // filter adaptation at call startup are more restrictive than after an
diff --git a/modules/audio_processing/aec3/aec_state.h b/modules/audio_processing/aec3/aec_state.h
index f860987..f6a31d8 100644
--- a/modules/audio_processing/aec3/aec_state.h
+++ b/modules/audio_processing/aec3/aec_state.h
@@ -57,7 +57,7 @@
   }
 
   // Returns the estimated echo path gain.
-  float EchoPathGain() const { return filter_analyzer_.Gain(); }
+  float EchoPathGain() const { return max_echo_path_gain_; }
 
   // Returns whether the render signal is currently active.
   bool ActiveRender() const { return blocks_with_active_render_ > 200; }
@@ -131,18 +131,20 @@
   // Updates the aec state.
   // TODO(bugs.webrtc.org/10913): Handle multi-channel adaptive filter response.
   // TODO(bugs.webrtc.org/10913): Compute multi-channel ERL, ERLE, and reverb.
-  void Update(const absl::optional<DelayEstimate>& external_delay,
-              const std::vector<std::array<float, kFftLengthBy2Plus1>>&
-                  adaptive_filter_frequency_response,
-              const std::vector<float>& adaptive_filter_impulse_response,
-              const RenderBuffer& render_buffer,
-              const std::array<float, kFftLengthBy2Plus1>& E2_main,
-              const std::array<float, kFftLengthBy2Plus1>& Y2,
-              rtc::ArrayView<const SubtractorOutput> subtractor_output);
+  void Update(
+      const absl::optional<DelayEstimate>& external_delay,
+      rtc::ArrayView<const std::vector<std::array<float, kFftLengthBy2Plus1>>>
+          adaptive_filter_frequency_response,
+      rtc::ArrayView<const std::vector<float>> adaptive_filter_impulse_response,
+      const RenderBuffer& render_buffer,
+      const std::array<float, kFftLengthBy2Plus1>& E2_main,
+      const std::array<float, kFftLengthBy2Plus1>& Y2,
+      rtc::ArrayView<const SubtractorOutput> subtractor_output);
 
   // Returns filter length in blocks.
   int FilterLengthBlocks() const {
-    return filter_analyzer_.FilterLengthBlocks();
+    // All filters have the same length, so arbitrarily return channel 0 length.
+    return filter_analyzers_[/*channel=*/0]->FilterLengthBlocks();
   }
 
  private:
@@ -191,9 +193,10 @@
     int DirectPathFilterDelay() const { return filter_delay_blocks_; }
 
     // Updates the delay estimates based on new data.
-    void Update(const FilterAnalyzer& filter_analyzer,
-                const absl::optional<DelayEstimate>& external_delay,
-                size_t blocks_with_proper_filter_adaptation);
+    void Update(
+        const std::vector<std::unique_ptr<FilterAnalyzer>>& filter_analyzer,
+        const absl::optional<DelayEstimate>& external_delay,
+        size_t blocks_with_proper_filter_adaptation);
 
    private:
     const int delay_headroom_samples_;
@@ -216,9 +219,9 @@
 
     // Updates the detection deciscion based on new data.
     void Update(int filter_delay_blocks,
-                bool consistent_filter,
-                bool converged_filter,
-                bool diverged_filter,
+                bool any_filter_consistent,
+                bool any_filter_converged,
+                bool all_filters_diverged,
                 bool active_render,
                 bool saturated_capture);
 
@@ -257,9 +260,8 @@
     void Update(bool active_render,
                 bool transparent_mode,
                 bool saturated_capture,
-                bool consistent_estimate_,
                 const absl::optional<DelayEstimate>& external_delay,
-                bool converged_filter);
+                bool any_filter_converged);
 
    private:
     bool usable_linear_estimate_ = false;
@@ -290,8 +292,9 @@
   ErleEstimator erle_estimator_;
   size_t strong_not_saturated_render_blocks_ = 0;
   size_t blocks_with_active_render_ = 0;
+  float max_echo_path_gain_;
   bool capture_signal_saturation_ = false;
-  FilterAnalyzer filter_analyzer_;
+  std::vector<std::unique_ptr<FilterAnalyzer>> filter_analyzers_;
   absl::optional<DelayEstimate> external_delay_;
   EchoAudibility echo_audibility_;
   ReverbModelEstimator reverb_model_estimator_;
diff --git a/modules/audio_processing/aec3/aec_state_unittest.cc b/modules/audio_processing/aec3/aec_state_unittest.cc
index 5997ab1..95a2134 100644
--- a/modules/audio_processing/aec3/aec_state_unittest.cc
+++ b/modules/audio_processing/aec3/aec_state_unittest.cc
@@ -55,17 +55,23 @@
     y[ch].fill(1000.f);
   }
   Aec3Fft fft;
-  std::vector<std::array<float, kFftLengthBy2Plus1>>
-      converged_filter_frequency_response(10);
-  for (auto& v : converged_filter_frequency_response) {
-    v.fill(0.01f);
+  std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>>
+  converged_filter_frequency_response(
+      num_capture_channels,
+      std::vector<std::array<float, kFftLengthBy2Plus1>>(10));
+  for (auto& v_ch : converged_filter_frequency_response) {
+    for (auto& v : v_ch) {
+      v.fill(0.01f);
+    }
   }
-  std::vector<std::array<float, kFftLengthBy2Plus1>>
+  std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>>
       diverged_filter_frequency_response = converged_filter_frequency_response;
-  converged_filter_frequency_response[2].fill(100.f);
-  converged_filter_frequency_response[2][0] = 1.f;
-  std::vector<float> impulse_response(
-      GetTimeDomainLength(config.filter.main.length_blocks), 0.f);
+  converged_filter_frequency_response[0][2].fill(100.f);
+  converged_filter_frequency_response[0][2][0] = 1.f;
+  std::vector<std::vector<float>> impulse_response(
+      num_capture_channels,
+      std::vector<float>(GetTimeDomainLength(config.filter.main.length_blocks),
+                         0.f));
 
   // Verify that linear AEC usability is true when the filter is converged
   for (size_t band = 0; band < kNumBands; ++band) {
@@ -243,20 +249,28 @@
   x.fill(0.f);
   y.fill(0.f);
 
-  std::vector<std::array<float, kFftLengthBy2Plus1>> frequency_response(
-      kFilterLengthBlocks);
-  for (auto& v : frequency_response) {
-    v.fill(0.01f);
+  std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>>
+  frequency_response(
+      kNumCaptureChannels,
+      std::vector<std::array<float, kFftLengthBy2Plus1>>(kFilterLengthBlocks));
+  for (auto& v_ch : frequency_response) {
+    for (auto& v : v_ch) {
+      v.fill(0.01f);
+    }
   }
 
-  std::vector<float> impulse_response(
-      GetTimeDomainLength(config.filter.main.length_blocks), 0.f);
+  std::vector<std::vector<float>> impulse_response(
+      kNumCaptureChannels,
+      std::vector<float>(GetTimeDomainLength(config.filter.main.length_blocks),
+                         0.f));
 
   // Verify that the filter delay for a converged filter is properly
   // identified.
   for (int k = 0; k < kFilterLengthBlocks; ++k) {
-    std::fill(impulse_response.begin(), impulse_response.end(), 0.f);
-    impulse_response[k * kBlockSize + 1] = 1.f;
+    for (auto& ir : impulse_response) {
+      std::fill(ir.begin(), ir.end(), 0.f);
+      ir[k * kBlockSize + 1] = 1.f;
+    }
 
     state.HandleEchoPathChange(echo_path_variability);
     subtractor_output[0].ComputeMetrics(y);
diff --git a/modules/audio_processing/aec3/echo_remover.cc b/modules/audio_processing/aec3/echo_remover.cc
index 31736bf..0127df1 100644
--- a/modules/audio_processing/aec3/echo_remover.cc
+++ b/modules/audio_processing/aec3/echo_remover.cc
@@ -384,9 +384,9 @@
 
   // Update the AEC state information.
   // TODO(bugs.webrtc.org/10913): Take all subtractors into account.
-  aec_state_.Update(external_delay, subtractor_.FilterFrequencyResponse()[0],
-                    subtractor_.FilterImpulseResponse()[0], *render_buffer,
-                    E2[0], Y2[0], subtractor_output);
+  aec_state_.Update(external_delay, subtractor_.FilterFrequencyResponse(),
+                    subtractor_.FilterImpulseResponse(), *render_buffer, E2[0],
+                    Y2[0], subtractor_output);
 
   // Choose the linear output.
   const auto& Y_fft = aec_state_.UseLinearFilterOutput() ? E : Y;
diff --git a/modules/audio_processing/aec3/filter_analyzer.cc b/modules/audio_processing/aec3/filter_analyzer.cc
index 138c188..313460f 100644
--- a/modules/audio_processing/aec3/filter_analyzer.cc
+++ b/modules/audio_processing/aec3/filter_analyzer.cc
@@ -96,8 +96,8 @@
   filter_length_blocks_ = filter_time_domain.size() * (1.f / kBlockSize);
 
   consistent_estimate_ = consistent_filter_detector_.Detect(
-      h_highpass_, region_, render_buffer.Block(-delay_blocks_)[0][0],
-      peak_index_, delay_blocks_);
+      h_highpass_, region_, render_buffer.Block(-delay_blocks_)[0], peak_index_,
+      delay_blocks_);
 }
 
 void FilterAnalyzer::UpdateFilterGain(
@@ -176,7 +176,7 @@
 bool FilterAnalyzer::ConsistentFilterDetector::Detect(
     rtc::ArrayView<const float> filter_to_analyze,
     const FilterRegion& region,
-    rtc::ArrayView<const float> x_block,
+    rtc::ArrayView<const std::vector<float>> x_block,
     size_t peak_index,
     int delay_blocks) {
   if (region.start_sample_ == 0) {
@@ -212,9 +212,15 @@
   }
 
   if (significant_peak_) {
-    const float x_energy = std::inner_product(x_block.begin(), x_block.end(),
-                                              x_block.begin(), 0.f);
-    const bool active_render_block = x_energy > active_render_threshold_;
+    bool active_render_block = false;
+    for (auto& x_channel : x_block) {
+      const float x_energy = std::inner_product(
+          x_channel.begin(), x_channel.end(), x_channel.begin(), 0.f);
+      if (x_energy > active_render_threshold_) {
+        active_render_block = true;
+        break;
+      }
+    }
 
     if (consistent_delay_reference_ == delay_blocks) {
       if (active_render_block) {
diff --git a/modules/audio_processing/aec3/filter_analyzer.h b/modules/audio_processing/aec3/filter_analyzer.h
index bcce528..de6c8a7 100644
--- a/modules/audio_processing/aec3/filter_analyzer.h
+++ b/modules/audio_processing/aec3/filter_analyzer.h
@@ -33,6 +33,9 @@
   explicit FilterAnalyzer(const EchoCanceller3Config& config);
   ~FilterAnalyzer();
 
+  FilterAnalyzer(const FilterAnalyzer&) = delete;
+  FilterAnalyzer& operator=(const FilterAnalyzer&) = delete;
+
   // Resets the analysis.
   void Reset();
 
@@ -82,7 +85,7 @@
     void Reset();
     bool Detect(rtc::ArrayView<const float> filter_to_analyze,
                 const FilterRegion& region,
-                rtc::ArrayView<const float> x_block,
+                rtc::ArrayView<const std::vector<float>> x_block,
                 size_t peak_index,
                 int delay_blocks);
 
@@ -110,8 +113,6 @@
   int filter_length_blocks_;
   FilterRegion region_;
   ConsistentFilterDetector consistent_filter_detector_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(FilterAnalyzer);
 };
 
 }  // namespace webrtc
diff --git a/modules/audio_processing/aec3/main_filter_update_gain_unittest.cc b/modules/audio_processing/aec3/main_filter_update_gain_unittest.cc
index 1a9e792..4725af9 100644
--- a/modules/audio_processing/aec3/main_filter_update_gain_unittest.cc
+++ b/modules/audio_processing/aec3/main_filter_update_gain_unittest.cc
@@ -59,14 +59,19 @@
                                   config.filter.shadow.length_blocks,
                                   config.filter.config_change_duration_blocks,
                                   1, optimization, &data_dumper);
-  std::vector<std::array<float, kFftLengthBy2Plus1>> H2(
-      main_filter.max_filter_size_partitions(),
-      std::array<float, kFftLengthBy2Plus1>());
-  for (auto& H2_k : H2) {
-    H2_k.fill(0.f);
+  std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>> H2(
+      kNumChannels, std::vector<std::array<float, kFftLengthBy2Plus1>>(
+                        main_filter.max_filter_size_partitions(),
+                        std::array<float, kFftLengthBy2Plus1>()));
+  for (auto& H2_ch : H2) {
+    for (auto& H2_k : H2_ch) {
+      H2_k.fill(0.f);
+    }
   }
-  std::vector<float> h(
-      GetTimeDomainLength(main_filter.max_filter_size_partitions()), 0.f);
+  std::vector<std::vector<float>> h(
+      kNumChannels,
+      std::vector<float>(
+          GetTimeDomainLength(main_filter.max_filter_size_partitions()), 0.f));
 
   Aec3Fft fft;
   std::array<float, kBlockSize> x_old;
@@ -183,15 +188,15 @@
         main_filter.SizePartitions(), &render_power);
 
     std::array<float, kFftLengthBy2Plus1> erl;
-    ComputeErl(optimization, H2, erl);
+    ComputeErl(optimization, H2[0], erl);
     main_gain.Compute(render_power, render_signal_analyzer, output[0], erl,
                       main_filter.SizePartitions(), saturation, &G);
-    main_filter.Adapt(*render_delay_buffer->GetRenderBuffer(), G, &h);
+    main_filter.Adapt(*render_delay_buffer->GetRenderBuffer(), G, &h[0]);
 
     // Update the delay.
     aec_state.HandleEchoPathChange(EchoPathVariability(
         false, EchoPathVariability::DelayAdjustment::kNone, false));
-    main_filter.ComputeFrequencyResponse(&H2);
+    main_filter.ComputeFrequencyResponse(&H2[0]);
     aec_state.Update(delay_estimate, H2, h,
                      *render_delay_buffer->GetRenderBuffer(), E2_main, Y2,
                      output);
diff --git a/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc b/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc
index 55f634b..7dbdbbe 100644
--- a/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc
+++ b/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc
@@ -28,7 +28,7 @@
 
       EchoCanceller3Config config;
       ResidualEchoEstimator estimator(config, num_render_channels);
-      AecState aec_state(config, num_render_channels);
+      AecState aec_state(config, num_capture_channels);
       std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
           RenderDelayBuffer::Create(config, kSampleRateHz,
                                     num_render_channels));
@@ -44,20 +44,26 @@
           kNumBands,
           std::vector<std::vector<float>>(num_render_channels,
                                           std::vector<float>(kBlockSize, 0.f)));
-      std::vector<std::array<float, kFftLengthBy2Plus1>> H2(10);
+      std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>> H2(
+          num_capture_channels,
+          std::vector<std::array<float, kFftLengthBy2Plus1>>(10));
       Random random_generator(42U);
-      std::vector<SubtractorOutput> output(num_render_channels);
+      std::vector<SubtractorOutput> output(num_capture_channels);
       std::array<float, kBlockSize> y;
       absl::optional<DelayEstimate> delay_estimate;
 
-      for (auto& H2_k : H2) {
-        H2_k.fill(0.01f);
+      for (auto& H2_ch : H2) {
+        for (auto& H2_k : H2_ch) {
+          H2_k.fill(0.01f);
+        }
+        H2_ch[2].fill(10.f);
+        H2_ch[2][0] = 0.1f;
       }
-      H2[2].fill(10.f);
-      H2[2][0] = 0.1f;
 
-      std::vector<float> h(
-          GetTimeDomainLength(config.filter.main.length_blocks), 0.f);
+      std::vector<std::vector<float>> h(
+          num_capture_channels,
+          std::vector<float>(
+              GetTimeDomainLength(config.filter.main.length_blocks), 0.f));
 
       for (auto& subtractor_output : output) {
         subtractor_output.Reset();
diff --git a/modules/audio_processing/aec3/subtractor_unittest.cc b/modules/audio_processing/aec3/subtractor_unittest.cc
index 23e7ead..717b481 100644
--- a/modules/audio_processing/aec3/subtractor_unittest.cc
+++ b/modules/audio_processing/aec3/subtractor_unittest.cc
@@ -145,8 +145,8 @@
 
     aec_state.HandleEchoPathChange(EchoPathVariability(
         false, EchoPathVariability::DelayAdjustment::kNone, false));
-    aec_state.Update(delay_estimate, subtractor.FilterFrequencyResponse()[0],
-                     subtractor.FilterImpulseResponse()[0],
+    aec_state.Update(delay_estimate, subtractor.FilterFrequencyResponse(),
+                     subtractor.FilterImpulseResponse(),
                      *render_delay_buffer->GetRenderBuffer(), E2_main, Y2,
                      output);
   }
diff --git a/modules/audio_processing/aec3/suppression_gain_unittest.cc b/modules/audio_processing/aec3/suppression_gain_unittest.cc
index 465227c..490c7ec 100644
--- a/modules/audio_processing/aec3/suppression_gain_unittest.cc
+++ b/modules/audio_processing/aec3/suppression_gain_unittest.cc
@@ -97,14 +97,14 @@
 
   // Ensure that the gain is no longer forced to zero.
   for (int k = 0; k <= kNumBlocksPerSecond / 5 + 1; ++k) {
-    aec_state.Update(delay_estimate, subtractor.FilterFrequencyResponse()[0],
-                     subtractor.FilterImpulseResponse()[0],
+    aec_state.Update(delay_estimate, subtractor.FilterFrequencyResponse(),
+                     subtractor.FilterImpulseResponse(),
                      *render_delay_buffer->GetRenderBuffer(), E2, Y2, output);
   }
 
   for (int k = 0; k < 100; ++k) {
-    aec_state.Update(delay_estimate, subtractor.FilterFrequencyResponse()[0],
-                     subtractor.FilterImpulseResponse()[0],
+    aec_state.Update(delay_estimate, subtractor.FilterFrequencyResponse(),
+                     subtractor.FilterImpulseResponse(),
                      *render_delay_buffer->GetRenderBuffer(), E2, Y2, output);
     suppression_gain.GetGain(E2, S2, R2, N2, analyzer, aec_state, x,
                              &high_bands_gain, &g);
@@ -120,8 +120,8 @@
   N2.fill(0.f);
 
   for (int k = 0; k < 100; ++k) {
-    aec_state.Update(delay_estimate, subtractor.FilterFrequencyResponse()[0],
-                     subtractor.FilterImpulseResponse()[0],
+    aec_state.Update(delay_estimate, subtractor.FilterFrequencyResponse(),
+                     subtractor.FilterImpulseResponse(),
                      *render_delay_buffer->GetRenderBuffer(), E2, Y2, output);
     suppression_gain.GetGain(E2, S2, R2, N2, analyzer, aec_state, x,
                              &high_bands_gain, &g);