| /* |
| * 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 <optional> |
| #include <vector> |
| |
| #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 std::optional<float>& filter_quality, |
| int filter_delay_blocks, |
| bool usable_linear_filter, |
| bool stationary_signal); |
| // Returns the decay for the exponential model. The parameter `mild` indicates |
| // which exponential decay to return, the default one or a milder one. |
| float Decay(bool mild) const { |
| if (use_adaptive_echo_decay_) { |
| return decay_; |
| } else { |
| return mild ? mild_decay_ : 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 mild_decay_; |
| float tail_gain_ = 0.f; |
| float smoothing_constant_ = 0.f; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_AEC3_REVERB_DECAY_ESTIMATOR_H_ |