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
140
141
142
143
144
145
146
147
148
149
|
/*
* 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.
*/
#include "modules/audio_processing/agc2/limiter.h"
#include <algorithm>
#include <array>
#include <cmath>
#include <cstddef>
#include "absl/strings/string_view.h"
#include "api/array_view.h"
#include "api/audio/audio_view.h"
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/numerics/safe_minmax.h"
namespace webrtc {
namespace {
// This constant affects the way scaling factors are interpolated for the first
// sub-frame of a frame. Only in the case in which the first sub-frame has an
// estimated level which is greater than the that of the previous analyzed
// sub-frame, linear interpolation is replaced with a power function which
// reduces the chances of over-shooting (and hence saturation), however reducing
// the fixed gain effectiveness.
constexpr float kAttackFirstSubframeInterpolationPower = 8.0f;
void InterpolateFirstSubframe(float last_factor,
float current_factor,
ArrayView<float> subframe) {
const int n = dchecked_cast<int>(subframe.size());
constexpr float p = kAttackFirstSubframeInterpolationPower;
for (int i = 0; i < n; ++i) {
subframe[i] = std::pow(1.f - i / n, p) * (last_factor - current_factor) +
current_factor;
}
}
void ComputePerSampleSubframeFactors(
const std::array<float, kSubFramesInFrame + 1>& scaling_factors,
MonoView<float> per_sample_scaling_factors) {
const size_t num_subframes = scaling_factors.size() - 1;
const int subframe_size = CheckedDivExact(
SamplesPerChannel(per_sample_scaling_factors), num_subframes);
// Handle first sub-frame differently in case of attack.
const bool is_attack = scaling_factors[0] > scaling_factors[1];
if (is_attack) {
InterpolateFirstSubframe(
scaling_factors[0], scaling_factors[1],
per_sample_scaling_factors.subview(0, subframe_size));
}
for (size_t i = is_attack ? 1 : 0; i < num_subframes; ++i) {
const int subframe_start = i * subframe_size;
const float scaling_start = scaling_factors[i];
const float scaling_end = scaling_factors[i + 1];
const float scaling_diff = (scaling_end - scaling_start) / subframe_size;
for (int j = 0; j < subframe_size; ++j) {
per_sample_scaling_factors[subframe_start + j] =
scaling_start + scaling_diff * j;
}
}
}
void ScaleSamples(MonoView<const float> per_sample_scaling_factors,
DeinterleavedView<float> signal) {
const int samples_per_channel = signal.samples_per_channel();
RTC_DCHECK_EQ(samples_per_channel,
SamplesPerChannel(per_sample_scaling_factors));
for (size_t i = 0; i < signal.num_channels(); ++i) {
MonoView<float> channel = signal[i];
for (int j = 0; j < samples_per_channel; ++j) {
channel[j] = SafeClamp(channel[j] * per_sample_scaling_factors[j],
kMinFloatS16Value, kMaxFloatS16Value);
}
}
}
} // namespace
Limiter::Limiter(ApmDataDumper* apm_data_dumper,
size_t samples_per_channel,
absl::string_view histogram_name)
: interp_gain_curve_(apm_data_dumper, histogram_name),
level_estimator_(samples_per_channel, apm_data_dumper),
apm_data_dumper_(apm_data_dumper) {
RTC_DCHECK_LE(samples_per_channel, kMaximalNumberOfSamplesPerChannel);
}
Limiter::~Limiter() = default;
void Limiter::Process(DeinterleavedView<float> signal) {
RTC_DCHECK_LE(signal.samples_per_channel(),
kMaximalNumberOfSamplesPerChannel);
const std::array<float, kSubFramesInFrame> level_estimate =
level_estimator_.ComputeLevel(signal);
RTC_DCHECK_EQ(level_estimate.size() + 1, scaling_factors_.size());
scaling_factors_[0] = last_scaling_factor_;
std::transform(level_estimate.begin(), level_estimate.end(),
scaling_factors_.begin() + 1, [this](float x) {
return interp_gain_curve_.LookUpGainToApply(x);
});
MonoView<float> per_sample_scaling_factors(&per_sample_scaling_factors_[0],
signal.samples_per_channel());
ComputePerSampleSubframeFactors(scaling_factors_, per_sample_scaling_factors);
ScaleSamples(per_sample_scaling_factors, signal);
last_scaling_factor_ = scaling_factors_.back();
// Dump data for debug.
apm_data_dumper_->DumpRaw("agc2_limiter_last_scaling_factor",
last_scaling_factor_);
apm_data_dumper_->DumpRaw(
"agc2_limiter_region",
static_cast<int>(interp_gain_curve_.get_stats().region));
}
InterpolatedGainCurve::Stats Limiter::GetGainCurveStats() const {
return interp_gain_curve_.get_stats();
}
void Limiter::SetSamplesPerChannel(size_t samples_per_channel) {
RTC_DCHECK_LE(samples_per_channel, kMaximalNumberOfSamplesPerChannel);
level_estimator_.SetSamplesPerChannel(samples_per_channel);
}
void Limiter::Reset() {
level_estimator_.Reset();
}
float Limiter::LastAudioLevel() const {
return level_estimator_.LastAudioLevel();
}
} // namespace webrtc
|
