1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
|
/*
* 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_;
const bool use_coarse_filter_reset_hangover_;
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<int> coarse_filter_reset_hangover_;
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_
|
