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/*
* 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. 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_
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