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
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
|
/*
* Copyright (C) 2003-2011 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.
*/
/*
* Parser functions for audio related objects.
*
*/
#include "config.h"
#include "audio_parser.h"
#include "audio_format.h"
#include "audio_check.h"
#include <assert.h>
#include <string.h>
#include <stdlib.h>
/**
* The GLib quark used for errors reported by this library.
*/
static inline GQuark
audio_parser_quark(void)
{
return g_quark_from_static_string("audio_parser");
}
static bool
parse_sample_rate(const char *src, bool mask, uint32_t *sample_rate_r,
const char **endptr_r, GError **error_r)
{
unsigned long value;
char *endptr;
if (mask && *src == '*') {
*sample_rate_r = 0;
*endptr_r = src + 1;
return true;
}
value = strtoul(src, &endptr, 10);
if (endptr == src) {
g_set_error(error_r, audio_parser_quark(), 0,
"Failed to parse the sample rate");
return false;
} else if (!audio_check_sample_rate(value, error_r))
return false;
*sample_rate_r = value;
*endptr_r = endptr;
return true;
}
static bool
parse_sample_format(const char *src, bool mask,
enum sample_format *sample_format_r,
const char **endptr_r, GError **error_r)
{
unsigned long value;
char *endptr;
enum sample_format sample_format;
if (mask && *src == '*') {
*sample_format_r = SAMPLE_FORMAT_UNDEFINED;
*endptr_r = src + 1;
return true;
}
if (*src == 'f') {
*sample_format_r = SAMPLE_FORMAT_FLOAT;
*endptr_r = src + 1;
return true;
}
if (memcmp(src, "dsd", 3) == 0) {
*sample_format_r = SAMPLE_FORMAT_DSD;
*endptr_r = src + 3;
return true;
}
value = strtoul(src, &endptr, 10);
if (endptr == src) {
g_set_error(error_r, audio_parser_quark(), 0,
"Failed to parse the sample format");
return false;
}
switch (value) {
case 8:
sample_format = SAMPLE_FORMAT_S8;
break;
case 16:
sample_format = SAMPLE_FORMAT_S16;
break;
case 24:
if (memcmp(endptr, "_3", 2) == 0)
/* for backwards compatibility */
endptr += 2;
sample_format = SAMPLE_FORMAT_S24_P32;
break;
case 32:
sample_format = SAMPLE_FORMAT_S32;
break;
default:
g_set_error(error_r, audio_parser_quark(), 0,
"Invalid sample format: %lu", value);
return false;
}
assert(audio_valid_sample_format(sample_format));
*sample_format_r = sample_format;
*endptr_r = endptr;
return true;
}
static bool
parse_channel_count(const char *src, bool mask, uint8_t *channels_r,
const char **endptr_r, GError **error_r)
{
unsigned long value;
char *endptr;
if (mask && *src == '*') {
*channels_r = 0;
*endptr_r = src + 1;
return true;
}
value = strtoul(src, &endptr, 10);
if (endptr == src) {
g_set_error(error_r, audio_parser_quark(), 0,
"Failed to parse the channel count");
return false;
} else if (!audio_check_channel_count(value, error_r))
return false;
*channels_r = value;
*endptr_r = endptr;
return true;
}
bool
audio_format_parse(struct audio_format *dest, const char *src,
bool mask, GError **error_r)
{
uint32_t rate;
enum sample_format sample_format;
uint8_t channels;
audio_format_clear(dest);
/* parse sample rate */
if (!parse_sample_rate(src, mask, &rate, &src, error_r))
return false;
if (*src++ != ':') {
g_set_error(error_r, audio_parser_quark(), 0,
"Sample format missing");
return false;
}
/* parse sample format */
if (!parse_sample_format(src, mask, &sample_format, &src, error_r))
return false;
if (*src++ != ':') {
g_set_error(error_r, audio_parser_quark(), 0,
"Channel count missing");
return false;
}
/* parse channel count */
if (!parse_channel_count(src, mask, &channels, &src, error_r))
return false;
if (*src != 0) {
g_set_error(error_r, audio_parser_quark(), 0,
"Extra data after channel count: %s", src);
return false;
}
audio_format_init(dest, rate, sample_format, channels);
assert(mask ? audio_format_mask_valid(dest)
: audio_format_valid(dest));
return true;
}
|
