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
|
/* the Music Player Daemon (MPD)
* Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
* This project's homepage is: http://www.musicpd.org
*
* libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net>
*
* 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#include "../decoder_api.h"
#ifdef HAVE_AUDIOFILE
#include "../log.h"
#include <sys/stat.h>
#include <audiofile.h>
/* pick 1020 since its devisible for 8,16,24, and 32-bit audio */
#define CHUNK_SIZE 1020
static int getAudiofileTotalTime(char *file)
{
int total_time;
AFfilehandle af_fp = afOpenFile(file, "r", NULL);
if (af_fp == AF_NULL_FILEHANDLE) {
return -1;
}
total_time = (int)
((double)afGetFrameCount(af_fp, AF_DEFAULT_TRACK)
/ afGetRate(af_fp, AF_DEFAULT_TRACK));
afCloseFile(af_fp);
return total_time;
}
static int audiofile_decode(struct decoder * decoder, char *path)
{
int fs, frame_count;
AFfilehandle af_fp;
int bits;
struct audio_format audio_format;
float total_time;
uint16_t bitRate;
struct stat st;
int ret, current = 0;
char chunk[CHUNK_SIZE];
if (stat(path, &st) < 0) {
ERROR("failed to stat: %s\n", path);
return -1;
}
af_fp = afOpenFile(path, "r", NULL);
if (af_fp == AF_NULL_FILEHANDLE) {
ERROR("failed to open: %s\n", path);
return -1;
}
afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
AF_SAMPFMT_TWOSCOMP, 16);
afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
audio_format.bits = (uint8_t)bits;
audio_format.sampleRate =
(unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
audio_format.channels =
(uint8_t)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
total_time = ((float)frame_count / (float)audio_format.sampleRate);
bitRate = (uint16_t)(st.st_size * 8.0 / total_time / 1000.0 + 0.5);
if (audio_format.bits != 8 && audio_format.bits != 16) {
ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n",
path, audio_format.bits);
afCloseFile(af_fp);
return -1;
}
fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1);
decoder_initialized(decoder, &audio_format, total_time);
do {
if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
decoder_clear(decoder);
current = decoder_seek_where(decoder) *
audio_format.sampleRate;
afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
decoder_command_finished(decoder);
}
ret = afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk,
CHUNK_SIZE / fs);
if (ret <= 0)
break;
current += ret;
decoder_data(decoder, NULL, 1,
chunk, ret * fs,
(float)current / (float)audio_format.sampleRate,
bitRate, NULL);
} while (decoder_get_command(decoder) != DECODE_COMMAND_STOP);
decoder_flush(decoder);
afCloseFile(af_fp);
return 0;
}
static struct tag *audiofileTagDup(char *file)
{
struct tag *ret = NULL;
int total_time = getAudiofileTotalTime(file);
if (total_time >= 0) {
if (!ret)
ret = tag_new();
ret->time = total_time;
} else {
DEBUG
("audiofileTagDup: Failed to get total song time from: %s\n",
file);
}
return ret;
}
static const char *audiofileSuffixes[] = { "wav", "au", "aiff", "aif", NULL };
struct decoder_plugin audiofilePlugin = {
.name = "audiofile",
.file_decode = audiofile_decode,
.tag_dup = audiofileTagDup,
.stream_types = INPUT_PLUGIN_STREAM_FILE,
.suffixes = audiofileSuffixes,
};
#else
struct decoder_plugin audiofilePlugin;
#endif /* HAVE_AUDIOFILE */
|
