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
176
177
178
179
180
|
/*
* Copyright (c) 2010 Clemens Ladisch <clemens@ladisch.de>
*
* Permission to use, copy, modify, and/or distribute this software for any
* purpose with or without fee is hereby granted, provided that the above
* copyright notice and this permission notice appear in all copies.
*
* THE SOFTWARE IS PROVIDED "AS IS" AND THE AUTHOR DISCLAIMS ALL WARRANTIES
* WITH REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF
* MERCHANTABILITY AND FITNESS. IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR
* ANY SPECIAL, DIRECT, INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
* WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
* ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT OF
* OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.
*/
/*
* The functions in this file map the value ranges of ALSA mixer controls onto
* the interval 0..1.
*
* The mapping is designed so that the position in the interval is proportional
* to the volume as a human ear would perceive it (i.e., the position is the
* cubic root of the linear sample multiplication factor). For controls with
* a small range (24 dB or less), the mapping is linear in the dB values so
* that each step has the same size visually. Only for controls without dB
* information, a linear mapping of the hardware volume register values is used
* (this is the same algorithm as used in the old alsamixer).
*
* When setting the volume, 'dir' is the rounding direction:
* -1/0/1 = down/nearest/up.
*/
#include <math.h>
#include <stdbool.h>
#include "volume_mapping.h"
#ifdef __UCLIBC__
/* 10^x = 10^(log e^x) = (e^x)^log10 = e^(x * log 10) */
#define exp10(x) (exp((x) * log(10)))
#endif /* __UCLIBC__ */
#define MAX_LINEAR_DB_SCALE 24
static inline bool use_linear_dB_scale(long dBmin, long dBmax)
{
return dBmax - dBmin <= MAX_LINEAR_DB_SCALE * 100;
}
static long lrint_dir(double x, int dir)
{
if (dir > 0)
return lrint(ceil(x));
else if (dir < 0)
return lrint(floor(x));
else
return lrint(x);
}
enum ctl_dir { PLAYBACK, CAPTURE };
static int (* const get_dB_range[2])(snd_mixer_elem_t *, long *, long *) = {
snd_mixer_selem_get_playback_dB_range,
snd_mixer_selem_get_capture_dB_range,
};
static int (* const get_raw_range[2])(snd_mixer_elem_t *, long *, long *) = {
snd_mixer_selem_get_playback_volume_range,
snd_mixer_selem_get_capture_volume_range,
};
static int (* const get_dB[2])(snd_mixer_elem_t *, snd_mixer_selem_channel_id_t, long *) = {
snd_mixer_selem_get_playback_dB,
snd_mixer_selem_get_capture_dB,
};
static int (* const get_raw[2])(snd_mixer_elem_t *, snd_mixer_selem_channel_id_t, long *) = {
snd_mixer_selem_get_playback_volume,
snd_mixer_selem_get_capture_volume,
};
static int (* const set_dB[2])(snd_mixer_elem_t *, long, int) = {
snd_mixer_selem_set_playback_dB_all,
snd_mixer_selem_set_capture_dB_all,
};
static int (* const set_raw[2])(snd_mixer_elem_t *, long) = {
snd_mixer_selem_set_playback_volume_all,
snd_mixer_selem_set_capture_volume_all,
};
static double get_normalized_volume(snd_mixer_elem_t *elem,
snd_mixer_selem_channel_id_t channel,
enum ctl_dir ctl_dir)
{
long min, max, value;
double normalized, min_norm;
int err;
err = get_dB_range[ctl_dir](elem, &min, &max);
if (err < 0 || min >= max) {
err = get_raw_range[ctl_dir](elem, &min, &max);
if (err < 0 || min == max)
return 0;
err = get_raw[ctl_dir](elem, channel, &value);
if (err < 0)
return 0;
return (value - min) / (double)(max - min);
}
err = get_dB[ctl_dir](elem, channel, &value);
if (err < 0)
return 0;
if (use_linear_dB_scale(min, max))
return (value - min) / (double)(max - min);
normalized = exp10((value - max) / 6000.0);
if (min != SND_CTL_TLV_DB_GAIN_MUTE) {
min_norm = exp10((min - max) / 6000.0);
normalized = (normalized - min_norm) / (1 - min_norm);
}
return normalized;
}
static int set_normalized_volume(snd_mixer_elem_t *elem,
double volume,
int dir,
enum ctl_dir ctl_dir)
{
long min, max, value;
double min_norm;
int err;
err = get_dB_range[ctl_dir](elem, &min, &max);
if (err < 0 || min >= max) {
err = get_raw_range[ctl_dir](elem, &min, &max);
if (err < 0)
return err;
value = lrint_dir(volume * (max - min), dir) + min;
return set_raw[ctl_dir](elem, value);
}
if (use_linear_dB_scale(min, max)) {
value = lrint_dir(volume * (max - min), dir) + min;
return set_dB[ctl_dir](elem, value, dir);
}
if (min != SND_CTL_TLV_DB_GAIN_MUTE) {
min_norm = exp10((min - max) / 6000.0);
volume = volume * (1 - min_norm) + min_norm;
}
value = lrint_dir(6000.0 * log10(volume), dir) + max;
return set_dB[ctl_dir](elem, value, dir);
}
double get_normalized_playback_volume(snd_mixer_elem_t *elem,
snd_mixer_selem_channel_id_t channel)
{
return get_normalized_volume(elem, channel, PLAYBACK);
}
double get_normalized_capture_volume(snd_mixer_elem_t *elem,
snd_mixer_selem_channel_id_t channel)
{
return get_normalized_volume(elem, channel, CAPTURE);
}
int set_normalized_playback_volume(snd_mixer_elem_t *elem,
double volume,
int dir)
{
return set_normalized_volume(elem, volume, dir, PLAYBACK);
}
int set_normalized_capture_volume(snd_mixer_elem_t *elem,
double volume,
int dir)
{
return set_normalized_volume(elem, volume, dir, CAPTURE);
}
|
