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/reverb_decay_estimator.h

@@ -0,0 +1,112 @@
+/*
+ *  Copyright (c) 2018 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_REVERB_DECAY_ESTIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_REVERB_DECAY_ESTIMATOR_H_
+
+#include <array>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"  // kMaxAdaptiveFilter...
+
+namespace webrtc {
+
+class ApmDataDumper;
+struct EchoCanceller3Config;
+
+// Class for estimating the decay of the late reverb.
+class ReverbDecayEstimator {
+ public:
+  explicit ReverbDecayEstimator(const EchoCanceller3Config& config);
+  ~ReverbDecayEstimator();
+  // Updates the decay estimate.
+  void Update(rtc::ArrayView<const float> filter,
+              const absl::optional<float>& filter_quality,
+              int filter_delay_blocks,
+              bool usable_linear_filter,
+              bool stationary_signal);
+  // Returns the decay for the exponential model.
+  float Decay() const { return decay_; }
+  // Dumps debug data.
+  void Dump(ApmDataDumper* data_dumper) const;
+
+ private:
+  void EstimateDecay(rtc::ArrayView<const float> filter, int peak_block);
+  void AnalyzeFilter(rtc::ArrayView<const float> filter);
+
+  void ResetDecayEstimation();
+
+  // Class for estimating the decay of the late reverb from the linear filter.
+  class LateReverbLinearRegressor {
+   public:
+    // Resets the estimator to receive a specified number of data points.
+    void Reset(int num_data_points);
+    // Accumulates estimation data.
+    void Accumulate(float z);
+    // Estimates the decay.
+    float Estimate();
+    // Returns whether an estimate is available.
+    bool EstimateAvailable() const { return n_ == N_ && N_ != 0; }
+
+   public:
+    float nz_ = 0.f;
+    float nn_ = 0.f;
+    float count_ = 0.f;
+    int N_ = 0;
+    int n_ = 0;
+  };
+
+  // Class for identifying the length of the early reverb from the linear
+  // filter. For identifying the early reverberations, the impulse response is
+  // divided in sections and the tilt of each section is computed by a linear
+  // regressor.
+  class EarlyReverbLengthEstimator {
+   public:
+    explicit EarlyReverbLengthEstimator(int max_blocks);
+    ~EarlyReverbLengthEstimator();
+
+    // Resets the estimator.
+    void Reset();
+    // Accumulates estimation data.
+    void Accumulate(float value, float smoothing);
+    // Estimates the size in blocks of the early reverb.
+    int Estimate();
+    // Dumps debug data.
+    void Dump(ApmDataDumper* data_dumper) const;
+
+   private:
+    std::vector<float> numerators_smooth_;
+    std::vector<float> numerators_;
+    int coefficients_counter_;
+    int block_counter_ = 0;
+    int n_sections_ = 0;
+  };
+
+  const int filter_length_blocks_;
+  const int filter_length_coefficients_;
+  const bool use_adaptive_echo_decay_;
+  LateReverbLinearRegressor late_reverb_decay_estimator_;
+  EarlyReverbLengthEstimator early_reverb_estimator_;
+  int late_reverb_start_;
+  int late_reverb_end_;
+  int block_to_analyze_ = 0;
+  int estimation_region_candidate_size_ = 0;
+  bool estimation_region_identified_ = false;
+  std::vector<float> previous_gains_;
+  float decay_;
+  float tail_gain_ = 0.f;
+  float smoothing_constant_ = 0.f;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_REVERB_DECAY_ESTIMATOR_H_