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
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
|
/* Normalizer plugin
*
* Limitations:
* - only AFMT_S16_LE supported
* - no parameters yet => tweak the values by editing the #defines
*
* License: GPLv2
* Author: pl <p_l@gmx.fr> (c) 2002 and beyond...
*
* Sources: some ideas from volnorm for xmms
*
* */
#define PLUGIN
#include <stdio.h>
#include <stdlib.h>
#include <inttypes.h>
#include <math.h> // for sqrt()
#include "audio_out.h"
#include "audio_plugin.h"
#include "audio_plugin_internal.h"
#include "afmt.h"
static ao_info_t info = {
"Volume normalizer",
"volnorm",
"pl <p_l@gmx.fr>",
""
};
LIBAO_PLUGIN_EXTERN(volnorm)
// mul is the value by which the samples are scaled
// and has to be in [MUL_MIN, MUL_MAX]
#define MUL_INIT 1.0
#define MUL_MIN 0.1
#define MUL_MAX 15.0
static float mul;
// "history" value of the filter
static float lastavg;
// SMOOTH_* must be in ]0.0, 1.0[
// The new value accounts for SMOOTH_MUL in the value and history
#define SMOOTH_MUL 0.06
#define SMOOTH_LASTAVG 0.06
// Some limits
#define MIN_S16 -32768
#define MAX_S16 32767
// ideal average level
#define MID_S16 (MAX_S16 * 0.25)
// silence level
#define SIL_S16 (MAX_S16 * 0.02)
// local data
static struct {
int inuse; // This plugin is in use TRUE, FALSE
int format; // sample fomat
} pl_volnorm = {0, 0};
// minimal interface
static int control(int cmd,int arg){
switch(cmd){
case AOCONTROL_PLUGIN_SET_LEN:
return CONTROL_OK;
}
return CONTROL_UNKNOWN;
}
// minimal interface
// open & setup audio device
// return: 1=success 0=fail
static int init(){
switch(ao_plugin_data.format){
case(AFMT_S16_LE):
break;
default:
fprintf(stderr,"[pl_volnorm] Audio format not yet supported.\n");
return 0;
}
pl_volnorm.format = ao_plugin_data.format;
pl_volnorm.inuse = 1;
reset();
printf("[pl_volnorm] Normalizer plugin in use.\n");
return 1;
}
// close plugin
static void uninit(){
pl_volnorm.inuse=0;
}
// empty buffers
static void reset(){
mul = MUL_INIT;
switch(ao_plugin_data.format) {
case(AFMT_S16_LE):
lastavg = MID_S16;
break;
default:
fprintf(stderr,"[pl_volnorm] internal inconsistency - please bugreport.\n");
*(char *) 0 = 0;
}
}
// processes 'ao_plugin_data.len' bytes of 'data'
// called for every block of data
static int play(){
switch(pl_volnorm.format){
case(AFMT_S16_LE): {
#define CLAMP(x,m,M) do { if ((x)<(m)) (x) = (m); else if ((x)>(M)) (x) = (M); } while(0)
int16_t* data=(int16_t*)ao_plugin_data.data;
int len=ao_plugin_data.len / 2; // 16 bits samples
int32_t i;
register int32_t tmp;
register float curavg;
float newavg;
float neededmul;
// average of the current samples
curavg = 0.0;
for (i = 0; i < len ; ++i) {
tmp = data[i];
curavg += tmp * tmp;
}
curavg = sqrt(curavg / (float) len);
if (curavg > SIL_S16) {
neededmul = MID_S16 / ( curavg * mul);
mul = (1.0 - SMOOTH_MUL) * mul + SMOOTH_MUL * neededmul;
// Clamp the mul coefficient
CLAMP(mul, MUL_MIN, MUL_MAX);
}
// Scale & clamp the samples
for (i=0; i < len ; ++i) {
tmp = data[i] * mul;
CLAMP(tmp, MIN_S16, MAX_S16);
data[i] = tmp;
}
// Evaluation of newavg (not 100% accurate because of values clamping)
newavg = mul * curavg;
#if 0
printf("time = %d len = %d curavg = %6.0f lastavg = %6.0f newavg = %6.0f\n"
" needed_m = %2.2f m = %2.2f\n\n",
time(NULL), len, curavg, lastavg, newavg, neededmul, mul);
#endif
lastavg = (1.0 - SMOOTH_LASTAVG) * lastavg + SMOOTH_LASTAVG * newavg;
break;
}
default:
return 0;
}
return 1;
}
|