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
|
/*
* Copyright (c) 2012 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 "voice_engine/audio_level.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "modules/include/module_common_types.h"
namespace webrtc {
namespace voe {
// Number of bars on the indicator.
// Note that the number of elements is specified because we are indexing it
// in the range of 0-32
constexpr int8_t kPermutation[33] = {0, 1, 2, 3, 4, 4, 5, 5, 5, 5, 6,
6, 6, 6, 6, 7, 7, 7, 7, 8, 8, 8,
9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9};
AudioLevel::AudioLevel()
: abs_max_(0), count_(0), current_level_(0), current_level_full_range_(0) {
WebRtcSpl_Init();
}
AudioLevel::~AudioLevel() {}
int8_t AudioLevel::Level() const {
rtc::CritScope cs(&crit_sect_);
return current_level_;
}
int16_t AudioLevel::LevelFullRange() const {
rtc::CritScope cs(&crit_sect_);
return current_level_full_range_;
}
void AudioLevel::Clear() {
rtc::CritScope cs(&crit_sect_);
abs_max_ = 0;
count_ = 0;
current_level_ = 0;
current_level_full_range_ = 0;
}
double AudioLevel::TotalEnergy() const {
rtc::CritScope cs(&crit_sect_);
return total_energy_;
}
double AudioLevel::TotalDuration() const {
rtc::CritScope cs(&crit_sect_);
return total_duration_;
}
void AudioLevel::ComputeLevel(const AudioFrame& audioFrame, double duration) {
// Check speech level (works for 2 channels as well)
int16_t abs_value = audioFrame.muted() ? 0 :
WebRtcSpl_MaxAbsValueW16(
audioFrame.data(),
audioFrame.samples_per_channel_ * audioFrame.num_channels_);
// Protect member access using a lock since this method is called on a
// dedicated audio thread in the RecordedDataIsAvailable() callback.
rtc::CritScope cs(&crit_sect_);
if (abs_value > abs_max_)
abs_max_ = abs_value;
// Update level approximately 10 times per second
if (count_++ == kUpdateFrequency) {
current_level_full_range_ = abs_max_;
count_ = 0;
// Highest value for a int16_t is 0x7fff = 32767
// Divide with 1000 to get in the range of 0-32 which is the range of the
// permutation vector
int32_t position = abs_max_ / 1000;
// Make it less likely that the bar stays at position 0. I.e. only if it's
// in the range 0-250 (instead of 0-1000)
if ((position == 0) && (abs_max_ > 250)) {
position = 1;
}
current_level_ = kPermutation[position];
// Decay the absolute maximum (divide by 4)
abs_max_ >>= 2;
}
// See the description for "totalAudioEnergy" in the WebRTC stats spec
// (https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy)
// for an explanation of these formulas. In short, we need a value that can
// be used to compute RMS audio levels over different time intervals, by
// taking the difference between the results from two getStats calls. To do
// this, the value needs to be of units "squared sample value * time".
double additional_energy =
static_cast<double>(current_level_full_range_) / INT16_MAX;
additional_energy *= additional_energy;
total_energy_ += additional_energy * duration;
total_duration_ += duration;
}
} // namespace voe
} // namespace webrtc
|
