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
150
151
|
/*
Instru2Light - Illumine un instrument de musique en temps réel
Copyright (C) 2012-2013 Ludovic Pouzenc <lpouzenc@gmail.com>
This program is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 3 of the License, or
(at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program. If not, see <http://www.gnu.org/licenses/>.
*/
#include "compute.h"
#include "fft.h"
#include <math.h>
#define MIN_SAMPLES 256
#define MAX_SAMPLES 2048
void compute_spectrum(float *data, int width, float output[PSHalf]);
float compute_level(const float *data, size_t nsamples, int rate) {
size_t i;
float input[MAX_SAMPLES], pwrspec[PSHalf];
float value;
int f, min_f_index, max_f_index;
if (nsamples >= MAX_SAMPLES) {
printf("WARN : nsamples >= MAX_SAMPLES : %zu >= %i\n", nsamples, MAX_SAMPLES);
nsamples=MAX_SAMPLES;
}
if (nsamples < MIN_SAMPLES) {
printf("WARN : nsamples < MIN_SAMPLES : %zu >= %i\n", nsamples, MIN_SAMPLES);
return -120.f;
}
/* Replicate with symmetry the sound to obtain an input buffer of the minimal len
for (i=0;i<MIN_SAMPLES;i++) {
if ( (i/nsamples)%2==1 )
input[i]=data[i]; // First channel only
else
input[i]=data[nsamples-i-1];
}
nsamples=MIN_SAMPLES;
} else {*/
for (i=0;i<nsamples;i++) {
input[i]=data[i]; // First channel only
}
//}
compute_spectrum(input, nsamples, pwrspec);
// Compute the mean power for 200Hz to 2000Hz band
min_f_index=((float)PSHalf)*200.f/(((float)rate)/2.f);
max_f_index=((float)PSHalf)*2000.f/(((float)rate)/2.f);
value=0.f;
for (f=min_f_index;f<=max_f_index;f++) {
value+=pwrspec[f];
}
// Mean value
value=value/(max_f_index-min_f_index+1);
return value;
}
/*
This function was adapted from Audacity 1.3.13
(orignally in C++ and with more genericity and functionnality)
Original Author : Dominic Mazzoni
Licenced under GPL 2.0
*/
void compute_spectrum(float *data, int width, float output[PSHalf]) {
int i, start, windows;
float temp;
float in[PSNumS];
float out[PSHalf];
float processed[PSHalf]={0.0f};
start = 0;
windows = 0;
while (start + PSNumS <= width) {
// Windowing : Hanning
for (i=0; i<PSNumS; i++)
in[i] = data[start+i] *(0.50-0.50*cos(2*M_PI*i/(PSNumS-1)));
// Returns only the real part of the result
PowerSpectrum(in, out);
for (i=0; i<PSHalf; i++)
processed[i] += out[i];
start += PSHalf;
windows++;
}
// Convert to decibels
// But do it safely; -Inf is nobody's friend
for (i = 0; i < PSHalf; i++){
temp=(processed[i] / PSNumS / windows);
if (temp > 0.0)
output[i] = 10*log10(temp);
else
output[i] = 0;
}
}
void audio2hsv_1(float audio_level, float *light_h, float *light_s, float *light_v) {
static float hue=0;
static float value_decay[8]={0.f};
static int decay_idx=0;
int i;
float level_norm, temp;
// Testé avec micro externe sur lud-msi
//[-38dB;0dB] -> [0.0;1.0]
level_norm=(38.f+audio_level)/38.f;
if (level_norm<0.0f) level_norm=0.f;
//if (level_norm<0.1f) level_norm=0.f; //FIXME : ici cache misere pour le tremblement sur plancher bruit
if (level_norm>1.f) level_norm=1.f;
if (decay_idx>=8) decay_idx=0;
value_decay[decay_idx++]=level_norm;
hue=(hue+0.0002f);
if (hue>1.f) hue-=1.f;
// printf("%+3.1f %+1.3f\n", audio_level, level_norm);
// Dummy code
// *light_h=hue;
temp=hue+level_norm/3.f;
*light_h=temp-(int)temp; // Fractionnal part
*light_s=1.f;
// *light_v=level_norm;
temp=0.f;
for(i=0;i<8;i++) temp+=value_decay[i];
temp/=8.f;
*light_v=temp;
}
|