1 /*
2 * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "modules/audio_processing/agc2/gain_applier.h"
12
13 #include "api/array_view.h"
14 #include "modules/audio_processing/agc2/agc2_common.h"
15 #include "rtc_base/numerics/safe_minmax.h"
16
17 namespace webrtc {
18 namespace {
19
20 // Returns true when the gain factor is so close to 1 that it would
21 // not affect int16 samples.
GainCloseToOne(float gain_factor)22 bool GainCloseToOne(float gain_factor) {
23 return 1.f - 1.f / kMaxFloatS16Value <= gain_factor &&
24 gain_factor <= 1.f + 1.f / kMaxFloatS16Value;
25 }
26
ClipSignal(AudioFrameView<float> signal)27 void ClipSignal(AudioFrameView<float> signal) {
28 for (size_t k = 0; k < signal.num_channels(); ++k) {
29 rtc::ArrayView<float> channel_view = signal.channel(k);
30 for (auto& sample : channel_view) {
31 sample = rtc::SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
32 }
33 }
34 }
35
ApplyGainWithRamping(float last_gain_linear,float gain_at_end_of_frame_linear,float inverse_samples_per_channel,AudioFrameView<float> float_frame)36 void ApplyGainWithRamping(float last_gain_linear,
37 float gain_at_end_of_frame_linear,
38 float inverse_samples_per_channel,
39 AudioFrameView<float> float_frame) {
40 // Do not modify the signal.
41 if (last_gain_linear == gain_at_end_of_frame_linear &&
42 GainCloseToOne(gain_at_end_of_frame_linear)) {
43 return;
44 }
45
46 // Gain is constant and different from 1.
47 if (last_gain_linear == gain_at_end_of_frame_linear) {
48 for (size_t k = 0; k < float_frame.num_channels(); ++k) {
49 rtc::ArrayView<float> channel_view = float_frame.channel(k);
50 for (auto& sample : channel_view) {
51 sample *= gain_at_end_of_frame_linear;
52 }
53 }
54 return;
55 }
56
57 // The gain changes. We have to change slowly to avoid discontinuities.
58 const float increment = (gain_at_end_of_frame_linear - last_gain_linear) *
59 inverse_samples_per_channel;
60 float gain = last_gain_linear;
61 for (size_t i = 0; i < float_frame.samples_per_channel(); ++i) {
62 for (size_t ch = 0; ch < float_frame.num_channels(); ++ch) {
63 float_frame.channel(ch)[i] *= gain;
64 }
65 gain += increment;
66 }
67 }
68
69 } // namespace
70
GainApplier(bool hard_clip_samples,float initial_gain_factor)71 GainApplier::GainApplier(bool hard_clip_samples, float initial_gain_factor)
72 : hard_clip_samples_(hard_clip_samples),
73 last_gain_factor_(initial_gain_factor),
74 current_gain_factor_(initial_gain_factor) {}
75
ApplyGain(AudioFrameView<float> signal)76 void GainApplier::ApplyGain(AudioFrameView<float> signal) {
77 if (static_cast<int>(signal.samples_per_channel()) != samples_per_channel_) {
78 Initialize(signal.samples_per_channel());
79 }
80
81 ApplyGainWithRamping(last_gain_factor_, current_gain_factor_,
82 inverse_samples_per_channel_, signal);
83
84 last_gain_factor_ = current_gain_factor_;
85
86 if (hard_clip_samples_) {
87 ClipSignal(signal);
88 }
89 }
90
SetGainFactor(float gain_factor)91 void GainApplier::SetGainFactor(float gain_factor) {
92 RTC_DCHECK_GT(gain_factor, 0.f);
93 current_gain_factor_ = gain_factor;
94 }
95
Initialize(size_t samples_per_channel)96 void GainApplier::Initialize(size_t samples_per_channel) {
97 RTC_DCHECK_GT(samples_per_channel, 0);
98 samples_per_channel_ = static_cast<int>(samples_per_channel);
99 inverse_samples_per_channel_ = 1.f / samples_per_channel_;
100 }
101
102 } // namespace webrtc
103