Update to current webrtc library

This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.

webrtc/modules/audio_processing/aec3/subtractor.h

@@ -0,0 +1,137 @@
+/*
+ *  Copyright (c) 2017 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_AEC3_SUBTRACTOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_H_
+
+#include <math.h>
+#include <stddef.h>
+
+#include <array>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/audio/echo_canceller3_config.h"
+#include "modules/audio_processing/aec3/adaptive_fir_filter.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/aec3/aec3_fft.h"
+#include "modules/audio_processing/aec3/aec_state.h"
+#include "modules/audio_processing/aec3/coarse_filter_update_gain.h"
+#include "modules/audio_processing/aec3/echo_path_variability.h"
+#include "modules/audio_processing/aec3/refined_filter_update_gain.h"
+#include "modules/audio_processing/aec3/render_buffer.h"
+#include "modules/audio_processing/aec3/render_signal_analyzer.h"
+#include "modules/audio_processing/aec3/subtractor_output.h"
+#include "modules/audio_processing/logging/apm_data_dumper.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Proves linear echo cancellation functionality
+class Subtractor {
+ public:
+  Subtractor(const EchoCanceller3Config& config,
+             size_t num_render_channels,
+             size_t num_capture_channels,
+             ApmDataDumper* data_dumper,
+             Aec3Optimization optimization);
+  ~Subtractor();
+  Subtractor(const Subtractor&) = delete;
+  Subtractor& operator=(const Subtractor&) = delete;
+
+  // Performs the echo subtraction.
+  void Process(const RenderBuffer& render_buffer,
+               const std::vector<std::vector<float>>& capture,
+               const RenderSignalAnalyzer& render_signal_analyzer,
+               const AecState& aec_state,
+               rtc::ArrayView<SubtractorOutput> outputs);
+
+  void HandleEchoPathChange(const EchoPathVariability& echo_path_variability);
+
+  // Exits the initial state.
+  void ExitInitialState();
+
+  // Returns the block-wise frequency responses for the refined adaptive
+  // filters.
+  const std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>>&
+  FilterFrequencyResponses() const {
+    return refined_frequency_responses_;
+  }
+
+  // Returns the estimates of the impulse responses for the refined adaptive
+  // filters.
+  const std::vector<std::vector<float>>& FilterImpulseResponses() const {
+    return refined_impulse_responses_;
+  }
+
+  void DumpFilters() {
+    data_dumper_->DumpRaw(
+        "aec3_subtractor_h_refined",
+        rtc::ArrayView<const float>(
+            refined_impulse_responses_[0].data(),
+            GetTimeDomainLength(
+                refined_filters_[0]->max_filter_size_partitions())));
+
+    refined_filters_[0]->DumpFilter("aec3_subtractor_H_refined");
+    coarse_filter_[0]->DumpFilter("aec3_subtractor_H_coarse");
+  }
+
+ private:
+  class FilterMisadjustmentEstimator {
+   public:
+    FilterMisadjustmentEstimator() = default;
+    ~FilterMisadjustmentEstimator() = default;
+    // Update the misadjustment estimator.
+    void Update(const SubtractorOutput& output);
+    // GetMisadjustment() Returns a recommended scale for the filter so the
+    // prediction error energy gets closer to the energy that is seen at the
+    // microphone input.
+    float GetMisadjustment() const {
+      RTC_DCHECK_GT(inv_misadjustment_, 0.0f);
+      // It is not aiming to adjust all the estimated mismatch. Instead,
+      // it adjusts half of that estimated mismatch.
+      return 2.f / sqrtf(inv_misadjustment_);
+    }
+    // Returns true if the prediciton error energy is significantly larger
+    // than the microphone signal energy and, therefore, an adjustment is
+    // recommended.
+    bool IsAdjustmentNeeded() const { return inv_misadjustment_ > 10.f; }
+    void Reset();
+    void Dump(ApmDataDumper* data_dumper) const;
+
+   private:
+    const int n_blocks_ = 4;
+    int n_blocks_acum_ = 0;
+    float e2_acum_ = 0.f;
+    float y2_acum_ = 0.f;
+    float inv_misadjustment_ = 0.f;
+    int overhang_ = 0.f;
+  };
+
+  const Aec3Fft fft_;
+  ApmDataDumper* data_dumper_;
+  const Aec3Optimization optimization_;
+  const EchoCanceller3Config config_;
+  const size_t num_capture_channels_;
+
+  std::vector<std::unique_ptr<AdaptiveFirFilter>> refined_filters_;
+  std::vector<std::unique_ptr<AdaptiveFirFilter>> coarse_filter_;
+  std::vector<std::unique_ptr<RefinedFilterUpdateGain>> refined_gains_;
+  std::vector<std::unique_ptr<CoarseFilterUpdateGain>> coarse_gains_;
+  std::vector<FilterMisadjustmentEstimator> filter_misadjustment_estimators_;
+  std::vector<size_t> poor_coarse_filter_counters_;
+  std::vector<std::vector<std::array<float, kFftLengthBy2Plus1>>>
+      refined_frequency_responses_;
+  std::vector<std::vector<float>> refined_impulse_responses_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_H_