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
|
/*
* SPDX-License-Identifier: BSD-2-Clause
*
* Copyright (c) 2024 The FreeBSD Foundation
* Copyright (c) 2025 Goran Mekić
*
* Portions of this software were developed by Christos Margiolis
* <christos@FreeBSD.org> under sponsorship from the FreeBSD Foundation.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND
* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
* ARE DISCLAIMED. IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE
* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
* SUCH DAMAGE.
*/
#include <sys/soundcard.h>
#include "oss.h"
/*
* Split input buffer into channels. The input buffer is in interleaved format,
* which means if we have 2 channels (L and R), this is what the buffer of 8
* samples would contain: L,R,L,R,L,R,L,R. The result of this function is a
* buffer containing: L,L,L,L,R,R,R,R.
*/
static void
to_channels(struct config *config, void *output)
{
uint8_t *in = config->buf;
uint8_t *out = output;
int i, channel, index, offset, byte;
/* Iterate over bytes in the input buffer */
for (byte = 0; byte < config->buffer_info.bytes;
byte += config->sample_size) {
/*
* Get index of a sample in the input buffer measured in
* samples
*/
i = byte / config->sample_size;
/* Get which channel is being processed */
channel = i % config->audio_info.max_channels;
/* Get offset of the sample inside a single channel */
offset = i / config->audio_info.max_channels;
/* Get index of a sample in the output buffer */
index = (channel * config->chsamples + offset) *
config->sample_size;
/* Copy singe sample from input to output */
memcpy(out+index, in+byte, config->sample_size);
}
}
/*
* Convert channels into interleaved format and put into output buffer
*/
static void
to_interleaved(struct config *config, void *input)
{
uint8_t *out = config->buf;
uint8_t *in = input;
int i, index, offset, channel, byte;
/* Iterate over bytes in the input buffer */
for (byte = 0; byte < config->buffer_info.bytes;
byte += config->sample_size) {
/*
* Get index of a sample in the input buffer measured in
* samples
*/
index = byte / config->sample_size;
/* Get which channel is being processed */
channel = index / config->chsamples;
/* Get offset of the sample inside a single channel */
offset = index % config->chsamples;
/* Get index of a sample in the output buffer */
i = (config->audio_info.max_channels * offset + channel) *
config->sample_size;
/* Copy singe sample from input to output */
memcpy(out+i, in+byte, config->sample_size);
}
}
int
main(int argc, char *argv[])
{
struct config config = {
.device = "/dev/dsp",
.mode = O_RDWR,
.format = AFMT_S32_NE,
.sample_rate = 48000,
};
int32_t *channels;
int rc, bytes;
oss_init(&config);
if (config.format != AFMT_S32_NE)
errx(1, "Device doesn't support signed 32bit samples. "
"Check with 'sndctl' if it can be configured for 's32le' format.");
bytes = config.buffer_info.bytes;
channels = malloc(bytes);
for (;;) {
if ((rc = read(config.fd, config.buf, bytes)) < bytes) {
warn("Requested %d bytes, but read %d!\n", bytes, rc);
break;
}
/*
* Strictly speaking, we could omit "channels" and operate only
* using config->buf, but this example tries to show the real
* world application usage. The problem is that the buffer is
* in interleaved format, and if you'd like to do any
* processing and/or mixing, it is easier to do that if samples
* are grouped per channel.
*/
to_channels(&config, channels);
to_interleaved(&config, channels);
if ((rc = write(config.fd, config.buf, bytes)) < bytes) {
warn("Requested %d bytes, but wrote %d!\n", bytes, rc);
break;
}
}
free(channels);
free(config.buf);
close(config.fd);
return (0);
}
|
