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
|
/*
* Copyright (C) 2003-2009 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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 2 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, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "config.h"
#include "pcm_mix.h"
#include "pcm_volume.h"
#include "pcm_utils.h"
#include "audio_format.h"
#include <glib.h>
#include <math.h>
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "pcm"
static void
pcm_add_8(int8_t *buffer1, const int8_t *buffer2,
unsigned num_samples, int volume1, int volume2)
{
while (num_samples > 0) {
int32_t sample1 = *buffer1;
int32_t sample2 = *buffer2++;
sample1 = ((sample1 * volume1 + sample2 * volume2) +
pcm_volume_dither() + PCM_VOLUME_1 / 2)
/ PCM_VOLUME_1;
*buffer1++ = pcm_range(sample1, 8);
--num_samples;
}
}
static void
pcm_add_16(int16_t *buffer1, const int16_t *buffer2,
unsigned num_samples, int volume1, int volume2)
{
while (num_samples > 0) {
int32_t sample1 = *buffer1;
int32_t sample2 = *buffer2++;
sample1 = ((sample1 * volume1 + sample2 * volume2) +
pcm_volume_dither() + PCM_VOLUME_1 / 2)
/ PCM_VOLUME_1;
*buffer1++ = pcm_range(sample1, 16);
--num_samples;
}
}
static void
pcm_add_24(int32_t *buffer1, const int32_t *buffer2,
unsigned num_samples, unsigned volume1, unsigned volume2)
{
while (num_samples > 0) {
int64_t sample1 = *buffer1;
int64_t sample2 = *buffer2++;
sample1 = ((sample1 * volume1 + sample2 * volume2) +
pcm_volume_dither() + PCM_VOLUME_1 / 2)
/ PCM_VOLUME_1;
*buffer1++ = pcm_range(sample1, 24);
--num_samples;
}
}
static void
pcm_add_32(int32_t *buffer1, const int32_t *buffer2,
unsigned num_samples, unsigned volume1, unsigned volume2)
{
while (num_samples > 0) {
int64_t sample1 = *buffer1;
int64_t sample2 = *buffer2++;
sample1 = ((sample1 * volume1 + sample2 * volume2) +
pcm_volume_dither() + PCM_VOLUME_1 / 2)
/ PCM_VOLUME_1;
*buffer1++ = pcm_range_64(sample1, 32);
--num_samples;
}
}
static void
pcm_add(void *buffer1, const void *buffer2, size_t size,
int vol1, int vol2,
const struct audio_format *format)
{
switch (format->format) {
case SAMPLE_FORMAT_S8:
pcm_add_8((int8_t *)buffer1, (const int8_t *)buffer2,
size, vol1, vol2);
break;
case SAMPLE_FORMAT_S16:
pcm_add_16((int16_t *)buffer1, (const int16_t *)buffer2,
size / 2, vol1, vol2);
break;
case SAMPLE_FORMAT_S24_P32:
pcm_add_24((int32_t*)buffer1,
(const int32_t*)buffer2,
size / 4, vol1, vol2);
break;
case SAMPLE_FORMAT_S32:
pcm_add_32((int32_t*)buffer1,
(const int32_t*)buffer2,
size / 4, vol1, vol2);
break;
default:
g_error("format %s not supported by pcm_add",
sample_format_to_string(format->format));
}
}
void
pcm_mix(void *buffer1, const void *buffer2, size_t size,
const struct audio_format *format, float portion1)
{
int vol1;
float s = sin(M_PI_2 * portion1);
s *= s;
vol1 = s * PCM_VOLUME_1 + 0.5;
vol1 = vol1 > PCM_VOLUME_1 ? PCM_VOLUME_1 : (vol1 < 0 ? 0 : vol1);
pcm_add(buffer1, buffer2, size, vol1, PCM_VOLUME_1 - vol1, format);
}
|
