1 /*
2 * Copyright (c) 2021 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 #include "modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.h"
11
12 #include <algorithm>
13
14 #include "api/array_view.h"
15 #include "modules/audio_processing/audio_buffer.h"
16 #include "rtc_base/checks.h"
17 #include "rtc_base/numerics/safe_minmax.h"
18
19 namespace webrtc {
20
AudioSamplesScaler(float initial_gain)21 AudioSamplesScaler::AudioSamplesScaler(float initial_gain)
22 : previous_gain_(initial_gain), target_gain_(initial_gain) {}
23
Process(AudioBuffer & audio_buffer)24 void AudioSamplesScaler::Process(AudioBuffer& audio_buffer) {
25 if (static_cast<int>(audio_buffer.num_frames()) != samples_per_channel_) {
26 // Update the members depending on audio-buffer length if needed.
27 RTC_DCHECK_GT(audio_buffer.num_frames(), 0);
28 samples_per_channel_ = static_cast<int>(audio_buffer.num_frames());
29 one_by_samples_per_channel_ = 1.f / samples_per_channel_;
30 }
31
32 if (target_gain_ == 1.f && previous_gain_ == target_gain_) {
33 // If only a gain of 1 is to be applied, do an early return without applying
34 // any gain.
35 return;
36 }
37
38 float gain = previous_gain_;
39 if (previous_gain_ == target_gain_) {
40 // Apply a non-changing gain.
41 for (size_t channel = 0; channel < audio_buffer.num_channels(); ++channel) {
42 rtc::ArrayView<float> channel_view(audio_buffer.channels()[channel],
43 samples_per_channel_);
44 for (float& sample : channel_view) {
45 sample *= gain;
46 }
47 }
48 } else {
49 const float increment =
50 (target_gain_ - previous_gain_) * one_by_samples_per_channel_;
51
52 if (increment > 0.f) {
53 // Apply an increasing gain.
54 for (size_t channel = 0; channel < audio_buffer.num_channels();
55 ++channel) {
56 gain = previous_gain_;
57 rtc::ArrayView<float> channel_view(audio_buffer.channels()[channel],
58 samples_per_channel_);
59 for (float& sample : channel_view) {
60 gain = std::min(gain + increment, target_gain_);
61 sample *= gain;
62 }
63 }
64 } else {
65 // Apply a decreasing gain.
66 for (size_t channel = 0; channel < audio_buffer.num_channels();
67 ++channel) {
68 gain = previous_gain_;
69 rtc::ArrayView<float> channel_view(audio_buffer.channels()[channel],
70 samples_per_channel_);
71 for (float& sample : channel_view) {
72 gain = std::max(gain + increment, target_gain_);
73 sample *= gain;
74 }
75 }
76 }
77 }
78 previous_gain_ = target_gain_;
79
80 // Saturate the samples to be in the S16 range.
81 for (size_t channel = 0; channel < audio_buffer.num_channels(); ++channel) {
82 rtc::ArrayView<float> channel_view(audio_buffer.channels()[channel],
83 samples_per_channel_);
84 for (float& sample : channel_view) {
85 constexpr float kMinFloatS16Value = -32768.f;
86 constexpr float kMaxFloatS16Value = 32767.f;
87 sample = rtc::SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
88 }
89 }
90 }
91
92 } // namespace webrtc
93