diff options
Diffstat (limited to 'apps')
-rw-r--r-- | apps/SOURCES | 1 | ||||
-rw-r--r-- | apps/codecs/SOURCES | 1 | ||||
-rw-r--r-- | apps/codecs/au.c | 319 | ||||
-rw-r--r-- | apps/codecs/codecs.make | 1 | ||||
-rw-r--r-- | apps/filetypes.c | 2 | ||||
-rw-r--r-- | apps/metadata.c | 11 | ||||
-rw-r--r-- | apps/metadata.h | 1 | ||||
-rw-r--r-- | apps/metadata/au.c | 122 | ||||
-rw-r--r-- | apps/metadata/metadata_parsers.h | 1 |
9 files changed, 459 insertions, 0 deletions
diff --git a/apps/SOURCES b/apps/SOURCES index 069e619451..47937fab70 100644 --- a/apps/SOURCES +++ b/apps/SOURCES | |||
@@ -188,6 +188,7 @@ metadata/rm.c | |||
188 | metadata/nsf.c | 188 | metadata/nsf.c |
189 | metadata/oma.c | 189 | metadata/oma.c |
190 | metadata/smaf.c | 190 | metadata/smaf.c |
191 | metadata/au.c | ||
191 | #endif | 192 | #endif |
192 | #ifdef HAVE_TAGCACHE | 193 | #ifdef HAVE_TAGCACHE |
193 | tagcache.c | 194 | tagcache.c |
diff --git a/apps/codecs/SOURCES b/apps/codecs/SOURCES index 9787caa122..b8a86aed8a 100644 --- a/apps/codecs/SOURCES +++ b/apps/codecs/SOURCES | |||
@@ -28,6 +28,7 @@ aiff.c | |||
28 | speex.c | 28 | speex.c |
29 | adx.c | 29 | adx.c |
30 | smaf.c | 30 | smaf.c |
31 | au.c | ||
31 | #if defined(HAVE_RECORDING) && !defined(SIMULATOR) | 32 | #if defined(HAVE_RECORDING) && !defined(SIMULATOR) |
32 | /* encoders */ | 33 | /* encoders */ |
33 | aiff_enc.c | 34 | aiff_enc.c |
diff --git a/apps/codecs/au.c b/apps/codecs/au.c new file mode 100644 index 0000000000..5632bf7a9b --- /dev/null +++ b/apps/codecs/au.c | |||
@@ -0,0 +1,319 @@ | |||
1 | /*************************************************************************** | ||
2 | * __________ __ ___. | ||
3 | * Open \______ \ ____ ____ | | _\_ |__ _______ ___ | ||
4 | * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / | ||
5 | * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < | ||
6 | * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ | ||
7 | * \/ \/ \/ \/ \/ | ||
8 | * $Id$ | ||
9 | * | ||
10 | * Copyright (C) 2010 Yoshihisa Uchida | ||
11 | * | ||
12 | * This program is free software; you can redistribute it and/or | ||
13 | * modify it under the terms of the GNU General Public License | ||
14 | * as published by the Free Software Foundation; either version 2 | ||
15 | * of the License, or (at your option) any later version. | ||
16 | * | ||
17 | * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY | ||
18 | * KIND, either express or implied. | ||
19 | * | ||
20 | ****************************************************************************/ | ||
21 | |||
22 | #include "codeclib.h" | ||
23 | #include "codecs/libpcm/support_formats.h" | ||
24 | |||
25 | CODEC_HEADER | ||
26 | |||
27 | /* Sun Audio file (Au file format) codec | ||
28 | * | ||
29 | * References | ||
30 | * [1] Sun Microsystems, Inc., Header file for Audio, .au, 1992 | ||
31 | * URL http://www.opengroup.org/public/pubs/external/auformat.html | ||
32 | * [2] Wikipedia, Au file format, URL: http://en.wikipedia.org/wiki/Sun_Audio | ||
33 | */ | ||
34 | |||
35 | #define PCM_SAMPLE_SIZE (1024*2) | ||
36 | |||
37 | static int32_t samples[PCM_SAMPLE_SIZE] IBSS_ATTR; | ||
38 | |||
39 | enum | ||
40 | { | ||
41 | AU_FORMAT_UNSUPPORT = 0, /* unsupported format */ | ||
42 | AU_FORMAT_MULAW, /* G.711 MULAW */ | ||
43 | AU_FORMAT_PCM, /* Linear PCM */ | ||
44 | AU_FORMAT_IEEE_FLOAT, /* IEEE float */ | ||
45 | AU_FORMAT_ALAW, /* G.711 ALAW */ | ||
46 | }; | ||
47 | |||
48 | static int support_formats[28][2] = { | ||
49 | { AU_FORMAT_UNSUPPORT, 0 }, | ||
50 | { AU_FORMAT_MULAW, 8 }, /* G.711 MULAW */ | ||
51 | { AU_FORMAT_PCM, 8 }, /* Linear PCM 8bit (signed) */ | ||
52 | { AU_FORMAT_PCM, 16 }, /* Linear PCM 16bit (signed, big endian) */ | ||
53 | { AU_FORMAT_PCM, 24 }, /* Linear PCM 24bit (signed, big endian) */ | ||
54 | { AU_FORMAT_PCM, 32 }, /* Linear PCM 32bit (signed, big endian) */ | ||
55 | { AU_FORMAT_IEEE_FLOAT, 32 }, /* Linear PCM float 32bit (signed, big endian) */ | ||
56 | { AU_FORMAT_IEEE_FLOAT, 64 }, /* Linear PCM float 64bit (signed, big endian) */ | ||
57 | { AU_FORMAT_UNSUPPORT, 0 }, /* Fragmented sample data */ | ||
58 | { AU_FORMAT_UNSUPPORT, 0 }, /* DSP program */ | ||
59 | { AU_FORMAT_UNSUPPORT, 0 }, /* 8bit fixed point */ | ||
60 | { AU_FORMAT_UNSUPPORT, 0 }, /* 16bit fixed point */ | ||
61 | { AU_FORMAT_UNSUPPORT, 0 }, /* 24bit fixed point */ | ||
62 | { AU_FORMAT_UNSUPPORT, 0 }, /* 32bit fixed point */ | ||
63 | { AU_FORMAT_UNSUPPORT, 0 }, | ||
64 | { AU_FORMAT_UNSUPPORT, 0 }, | ||
65 | { AU_FORMAT_UNSUPPORT, 0 }, | ||
66 | { AU_FORMAT_UNSUPPORT, 0 }, | ||
67 | { AU_FORMAT_UNSUPPORT, 0 }, /* 16bit linear with emphasis */ | ||
68 | { AU_FORMAT_UNSUPPORT, 0 }, /* 16bit linear compressed */ | ||
69 | { AU_FORMAT_UNSUPPORT, 0 }, /* 16bit linear with emphasis and compression */ | ||
70 | { AU_FORMAT_UNSUPPORT, 0 }, /* Music kit DSP commands */ | ||
71 | { AU_FORMAT_UNSUPPORT, 0 }, | ||
72 | { AU_FORMAT_UNSUPPORT, 0 }, /* G.721 MULAW */ | ||
73 | { AU_FORMAT_UNSUPPORT, 0 }, /* G.722 */ | ||
74 | { AU_FORMAT_UNSUPPORT, 0 }, /* G.723 3bit */ | ||
75 | { AU_FORMAT_UNSUPPORT, 0 }, /* G.723 5bit */ | ||
76 | { AU_FORMAT_ALAW, 8 }, /* G.711 ALAW */ | ||
77 | }; | ||
78 | |||
79 | const struct pcm_entry au_codecs[] = { | ||
80 | { AU_FORMAT_MULAW, get_itut_g711_mulaw_codec }, | ||
81 | { AU_FORMAT_PCM, get_linear_pcm_codec }, | ||
82 | { AU_FORMAT_IEEE_FLOAT, get_ieee_float_codec }, | ||
83 | { AU_FORMAT_ALAW, get_itut_g711_alaw_codec }, | ||
84 | }; | ||
85 | |||
86 | #define NUM_FORMATS 4 | ||
87 | |||
88 | static const struct pcm_codec *get_au_codec(uint32_t formattag) | ||
89 | { | ||
90 | int i; | ||
91 | |||
92 | for (i = 0; i < NUM_FORMATS; i++) | ||
93 | { | ||
94 | if (au_codecs[i].format_tag == formattag) | ||
95 | { | ||
96 | if (au_codecs[i].get_codec) | ||
97 | return au_codecs[i].get_codec(); | ||
98 | return 0; | ||
99 | } | ||
100 | } | ||
101 | return 0; | ||
102 | } | ||
103 | |||
104 | static unsigned int get_be32(uint8_t *buf) | ||
105 | { | ||
106 | return (buf[0] << 24) | (buf[1] << 16) | (buf[2] << 8) | buf[3]; | ||
107 | } | ||
108 | |||
109 | static int convert_au_format(unsigned int encoding, struct pcm_format *fmt) | ||
110 | { | ||
111 | if (encoding > 27) | ||
112 | { | ||
113 | fmt->formattag = AU_FORMAT_UNSUPPORT; | ||
114 | fmt->bitspersample = 0; | ||
115 | } | ||
116 | else | ||
117 | { | ||
118 | fmt->formattag = support_formats[encoding][0]; | ||
119 | fmt->bitspersample = support_formats[encoding][1]; | ||
120 | } | ||
121 | |||
122 | return fmt->formattag; | ||
123 | } | ||
124 | |||
125 | /* this is the codec entry point */ | ||
126 | enum codec_status codec_main(void) | ||
127 | { | ||
128 | int status = CODEC_OK; | ||
129 | struct pcm_format format; | ||
130 | uint32_t bytesdone, decodedsamples; | ||
131 | size_t n; | ||
132 | int bufcount; | ||
133 | int endofstream; | ||
134 | unsigned char *buf; | ||
135 | uint8_t *aubuf; | ||
136 | off_t firstblockposn; /* position of the first block in file */ | ||
137 | const struct pcm_codec *codec; | ||
138 | int offset = 0; | ||
139 | |||
140 | /* Generic codec initialisation */ | ||
141 | ci->configure(DSP_SET_SAMPLE_DEPTH, 28); | ||
142 | |||
143 | next_track: | ||
144 | if (codec_init()) { | ||
145 | DEBUGF("codec_init() error\n"); | ||
146 | status = CODEC_ERROR; | ||
147 | goto exit; | ||
148 | } | ||
149 | |||
150 | while (!*ci->taginfo_ready && !ci->stop_codec) | ||
151 | ci->sleep(1); | ||
152 | |||
153 | codec_set_replaygain(ci->id3); | ||
154 | |||
155 | ci->memset(&format, 0, sizeof(struct pcm_format)); | ||
156 | format.is_signed = true; | ||
157 | format.is_little_endian = false; | ||
158 | |||
159 | /* set format */ | ||
160 | buf = ci->request_buffer(&n, 24); | ||
161 | if (n < 24 || (memcmp(buf, ".snd", 4) != 0)) | ||
162 | { | ||
163 | /* | ||
164 | * headerless sun audio file | ||
165 | * It is decoded under conditions. | ||
166 | * format: G.711 mu-law | ||
167 | * channel: mono | ||
168 | * frequency: 8000 kHz | ||
169 | */ | ||
170 | offset = 0; | ||
171 | format.formattag = AU_FORMAT_MULAW; | ||
172 | format.channels = 1; | ||
173 | format.bitspersample = 8; | ||
174 | format.numbytes = ci->id3->filesize; | ||
175 | } | ||
176 | else | ||
177 | { | ||
178 | /* parse header */ | ||
179 | |||
180 | /* data offset */ | ||
181 | offset = get_be32(buf + 4); | ||
182 | if (offset < 24) | ||
183 | { | ||
184 | DEBUGF("CODEC_ERROR: sun audio offset size is small: %d\n", offset); | ||
185 | status = CODEC_ERROR; | ||
186 | goto done; | ||
187 | } | ||
188 | /* data size */ | ||
189 | format.numbytes = get_be32(buf + 8); | ||
190 | if (format.numbytes == (uint32_t)0xffffffff) | ||
191 | format.numbytes = ci->id3->filesize - offset; | ||
192 | /* encoding */ | ||
193 | format.formattag = convert_au_format(get_be32(buf + 12), &format); | ||
194 | if (format.formattag == AU_FORMAT_UNSUPPORT) | ||
195 | { | ||
196 | DEBUGF("CODEC_ERROR: sun audio unsupport format: %d\n", get_be32(buf + 12)); | ||
197 | status = CODEC_ERROR; | ||
198 | goto done; | ||
199 | } | ||
200 | /* skip sample rate */ | ||
201 | format.channels = get_be32(buf + 20); | ||
202 | if (format.channels == 0) { | ||
203 | DEBUGF("CODEC_ERROR: sun audio 0-channels file\n"); | ||
204 | status = CODEC_ERROR; | ||
205 | goto done; | ||
206 | } | ||
207 | } | ||
208 | |||
209 | /* advance to first WAVE chunk */ | ||
210 | ci->advance_buffer(offset); | ||
211 | |||
212 | firstblockposn = offset; | ||
213 | |||
214 | decodedsamples = 0; | ||
215 | codec = 0; | ||
216 | bytesdone = 0; | ||
217 | |||
218 | /* blockalign = 1 sample */ | ||
219 | format.blockalign = format.bitspersample * format.channels >> 3; | ||
220 | |||
221 | /* get codec */ | ||
222 | codec = get_au_codec(format.formattag); | ||
223 | if (!codec) | ||
224 | { | ||
225 | DEBUGF("CODEC_ERROR: unsupport sun audio format: %lx\n", format.formattag); | ||
226 | status = CODEC_ERROR; | ||
227 | goto done; | ||
228 | } | ||
229 | |||
230 | if (!codec->set_format(&format)) | ||
231 | { | ||
232 | status = CODEC_ERROR; | ||
233 | goto done; | ||
234 | } | ||
235 | |||
236 | if (format.numbytes == 0) { | ||
237 | DEBUGF("CODEC_ERROR: data size is 0\n"); | ||
238 | status = CODEC_ERROR; | ||
239 | goto done; | ||
240 | } | ||
241 | |||
242 | /* check chunksize */ | ||
243 | if ((format.chunksize / format.blockalign) * format.samplesperblock * format.channels | ||
244 | > PCM_SAMPLE_SIZE) | ||
245 | format.chunksize = (PCM_SAMPLE_SIZE / format.blockalign) * format.blockalign; | ||
246 | if (format.chunksize == 0) | ||
247 | { | ||
248 | DEBUGF("CODEC_ERROR: chunksize is 0\n"); | ||
249 | status = CODEC_ERROR; | ||
250 | goto done; | ||
251 | } | ||
252 | |||
253 | ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency); | ||
254 | if (format.channels == 2) { | ||
255 | ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED); | ||
256 | } else if (format.channels == 1) { | ||
257 | ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO); | ||
258 | } else { | ||
259 | DEBUGF("CODEC_ERROR: more than 2 channels\n"); | ||
260 | status = CODEC_ERROR; | ||
261 | goto done; | ||
262 | } | ||
263 | |||
264 | /* The main decoder loop */ | ||
265 | endofstream = 0; | ||
266 | |||
267 | while (!endofstream) { | ||
268 | ci->yield(); | ||
269 | if (ci->stop_codec || ci->new_track) { | ||
270 | break; | ||
271 | } | ||
272 | |||
273 | if (ci->seek_time) { | ||
274 | /* 2nd args(read_buffer) is unnecessary in the format which Sun Audio supports. */ | ||
275 | struct pcm_pos *newpos = codec->get_seek_pos(ci->seek_time, NULL); | ||
276 | |||
277 | decodedsamples = newpos->samples; | ||
278 | if (newpos->pos > format.numbytes) | ||
279 | break; | ||
280 | if (ci->seek_buffer(firstblockposn + newpos->pos)) | ||
281 | { | ||
282 | bytesdone = newpos->pos; | ||
283 | } | ||
284 | ci->seek_complete(); | ||
285 | } | ||
286 | |||
287 | aubuf = (uint8_t *)ci->request_buffer(&n, format.chunksize); | ||
288 | if (n == 0) | ||
289 | break; /* End of stream */ | ||
290 | if (bytesdone + n > format.numbytes) { | ||
291 | n = format.numbytes - bytesdone; | ||
292 | endofstream = 1; | ||
293 | } | ||
294 | |||
295 | status = codec->decode(aubuf, n, samples, &bufcount); | ||
296 | if (status == CODEC_ERROR) | ||
297 | { | ||
298 | DEBUGF("codec error\n"); | ||
299 | goto done; | ||
300 | } | ||
301 | |||
302 | ci->pcmbuf_insert(samples, NULL, bufcount); | ||
303 | ci->advance_buffer(n); | ||
304 | bytesdone += n; | ||
305 | decodedsamples += bufcount; | ||
306 | |||
307 | if (bytesdone >= format.numbytes) | ||
308 | endofstream = 1; | ||
309 | ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency); | ||
310 | } | ||
311 | status = CODEC_OK; | ||
312 | |||
313 | done: | ||
314 | if (ci->request_next_track()) | ||
315 | goto next_track; | ||
316 | |||
317 | exit: | ||
318 | return status; | ||
319 | } | ||
diff --git a/apps/codecs/codecs.make b/apps/codecs/codecs.make index 6a86517119..cf1a057d8c 100644 --- a/apps/codecs/codecs.make +++ b/apps/codecs/codecs.make | |||
@@ -91,6 +91,7 @@ $(CODECDIR)/atrac3_oma.codec : $(CODECDIR)/libatrac.a | |||
91 | $(CODECDIR)/aiff.codec : $(CODECDIR)/libpcm.a | 91 | $(CODECDIR)/aiff.codec : $(CODECDIR)/libpcm.a |
92 | $(CODECDIR)/wav.codec : $(CODECDIR)/libpcm.a | 92 | $(CODECDIR)/wav.codec : $(CODECDIR)/libpcm.a |
93 | $(CODECDIR)/smaf.codec : $(CODECDIR)/libpcm.a | 93 | $(CODECDIR)/smaf.codec : $(CODECDIR)/libpcm.a |
94 | $(CODECDIR)/au.codec : $(CODECDIR)/libpcm.a | ||
94 | 95 | ||
95 | $(CODECS): $(CODECLIB) # this must be last in codec dependency list | 96 | $(CODECS): $(CODECLIB) # this must be last in codec dependency list |
96 | 97 | ||
diff --git a/apps/filetypes.c b/apps/filetypes.c index 17a1f0345f..bbb52c8418 100644 --- a/apps/filetypes.c +++ b/apps/filetypes.c | |||
@@ -102,6 +102,8 @@ static const struct filetype inbuilt_filetypes[] = { | |||
102 | { "aa3", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA }, | 102 | { "aa3", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA }, |
103 | { "at3", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA }, | 103 | { "at3", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA }, |
104 | { "mmf", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA }, | 104 | { "mmf", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA }, |
105 | { "au", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA }, | ||
106 | { "snd", FILE_ATTR_AUDIO, Icon_Audio, VOICE_EXT_MPA }, | ||
105 | #endif | 107 | #endif |
106 | { "m3u", FILE_ATTR_M3U, Icon_Playlist, LANG_PLAYLIST }, | 108 | { "m3u", FILE_ATTR_M3U, Icon_Playlist, LANG_PLAYLIST }, |
107 | { "m3u8",FILE_ATTR_M3U, Icon_Playlist, LANG_PLAYLIST }, | 109 | { "m3u8",FILE_ATTR_M3U, Icon_Playlist, LANG_PLAYLIST }, |
diff --git a/apps/metadata.c b/apps/metadata.c index 36e97268ea..fe25bbc074 100644 --- a/apps/metadata.c +++ b/apps/metadata.c | |||
@@ -168,6 +168,9 @@ const struct afmt_entry audio_formats[AFMT_NUM_CODECS] = | |||
168 | /* SMAF (Synthetic music Mobile Application Format) */ | 168 | /* SMAF (Synthetic music Mobile Application Format) */ |
169 | [AFMT_SMAF] = | 169 | [AFMT_SMAF] = |
170 | AFMT_ENTRY("SMAF", "smaf", NULL, "mmf\0" ), | 170 | AFMT_ENTRY("SMAF", "smaf", NULL, "mmf\0" ), |
171 | /* Sun Audio file */ | ||
172 | [AFMT_AU] = | ||
173 | AFMT_ENTRY("AU", "au", NULL, "au\0snd\0" ), | ||
171 | #endif | 174 | #endif |
172 | }; | 175 | }; |
173 | 176 | ||
@@ -458,6 +461,14 @@ bool get_metadata(struct mp3entry* id3, int fd, const char* trackname) | |||
458 | return false; | 461 | return false; |
459 | } | 462 | } |
460 | break; | 463 | break; |
464 | |||
465 | case AFMT_AU: | ||
466 | if (!get_au_metadata(fd, id3)) | ||
467 | { | ||
468 | DEBUGF("get_au_metadata error\n"); | ||
469 | return false; | ||
470 | } | ||
471 | break; | ||
461 | 472 | ||
462 | #endif /* CONFIG_CODEC == SWCODEC */ | 473 | #endif /* CONFIG_CODEC == SWCODEC */ |
463 | 474 | ||
diff --git a/apps/metadata.h b/apps/metadata.h index 7f6843f6d9..f5ca99eeff 100644 --- a/apps/metadata.h +++ b/apps/metadata.h | |||
@@ -79,6 +79,7 @@ enum | |||
79 | AFMT_TM2, /* Atari 8bit tm2 format */ | 79 | AFMT_TM2, /* Atari 8bit tm2 format */ |
80 | AFMT_OMA_ATRAC3, /* Atrac3 in Sony OMA container */ | 80 | AFMT_OMA_ATRAC3, /* Atrac3 in Sony OMA container */ |
81 | AFMT_SMAF, /* SMAF */ | 81 | AFMT_SMAF, /* SMAF */ |
82 | AFMT_AU, /* Sun Audio file */ | ||
82 | #endif | 83 | #endif |
83 | 84 | ||
84 | /* add new formats at any index above this line to have a sensible order - | 85 | /* add new formats at any index above this line to have a sensible order - |
diff --git a/apps/metadata/au.c b/apps/metadata/au.c new file mode 100644 index 0000000000..1a7eef03c9 --- /dev/null +++ b/apps/metadata/au.c | |||
@@ -0,0 +1,122 @@ | |||
1 | /*************************************************************************** | ||
2 | * __________ __ ___. | ||
3 | * Open \______ \ ____ ____ | | _\_ |__ _______ ___ | ||
4 | * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / | ||
5 | * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < | ||
6 | * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ | ||
7 | * \/ \/ \/ \/ \/ | ||
8 | * $Id$ | ||
9 | * | ||
10 | * Copyright (C) 2010 Yoshihisa Uchida | ||
11 | * | ||
12 | * This program is free software; you can redistribute it and/or | ||
13 | * modify it under the terms of the GNU General Public License | ||
14 | * as published by the Free Software Foundation; either version 2 | ||
15 | * of the License, or (at your option) any later version. | ||
16 | * | ||
17 | * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY | ||
18 | * KIND, either express or implied. | ||
19 | * | ||
20 | ****************************************************************************/ | ||
21 | #include <stdio.h> | ||
22 | #include <string.h> | ||
23 | #include <stdlib.h> | ||
24 | #include <ctype.h> | ||
25 | #include <inttypes.h> | ||
26 | |||
27 | #include "system.h" | ||
28 | #include "metadata.h" | ||
29 | #include "metadata_common.h" | ||
30 | #include "metadata_parsers.h" | ||
31 | #include "logf.h" | ||
32 | |||
33 | static int bitspersamples[28] = { | ||
34 | 0, | ||
35 | 8, /* G.711 MULAW */ | ||
36 | 8, /* 8bit */ | ||
37 | 16, /* 16bit */ | ||
38 | 24, /* 24bit */ | ||
39 | 32, /* 32bit */ | ||
40 | 32, /* 32bit */ | ||
41 | 64, /* 64bit */ | ||
42 | 0, /* Fragmented sample data */ | ||
43 | 0, /* DSP program */ | ||
44 | 0, /* 8bit fixed point */ | ||
45 | 0, /* 16bit fixed point */ | ||
46 | 0, /* 24bit fixed point */ | ||
47 | 0, /* 32bit fixed point */ | ||
48 | 0, | ||
49 | 0, | ||
50 | 0, | ||
51 | 0, | ||
52 | 0, /* 16bit linear with emphasis */ | ||
53 | 0, /* 16bit linear compressed */ | ||
54 | 0, /* 16bit linear with emphasis and compression */ | ||
55 | 0, /* Music kit DSP commands */ | ||
56 | 0, | ||
57 | 0, /* G.721 MULAW */ | ||
58 | 0, /* G.722 */ | ||
59 | 0, /* G.723 3bit */ | ||
60 | 0, /* G.723 5bit */ | ||
61 | 8, /* G.711 ALAW */ | ||
62 | }; | ||
63 | |||
64 | static int get_au_bitspersample(unsigned int encoding) | ||
65 | { | ||
66 | if (encoding > 27) | ||
67 | return 0; | ||
68 | return bitspersamples[encoding]; | ||
69 | } | ||
70 | |||
71 | bool get_au_metadata(int fd, struct mp3entry* id3) | ||
72 | { | ||
73 | /* Use the trackname part of the id3 structure as a temporary buffer */ | ||
74 | unsigned char* buf = (unsigned char *)id3->path; | ||
75 | unsigned long totalsamples = 0; | ||
76 | unsigned long channels = 0; | ||
77 | unsigned long bitspersample = 0; | ||
78 | unsigned long numbytes = 0; | ||
79 | int read_bytes; | ||
80 | int offset; | ||
81 | |||
82 | id3->vbr = false; /* All Sun audio files are CBR */ | ||
83 | id3->filesize = filesize(fd); | ||
84 | |||
85 | if ((lseek(fd, 0, SEEK_SET) < 0) || ((read_bytes = read(fd, buf, 24)) < 0)) | ||
86 | return false; | ||
87 | |||
88 | if (read_bytes < 24 || (memcmp(buf, ".snd", 4) != 0)) | ||
89 | { | ||
90 | /* no header */ | ||
91 | numbytes = id3->filesize; | ||
92 | bitspersample = 8; | ||
93 | id3->frequency = 8000; | ||
94 | channels = 1; | ||
95 | } | ||
96 | else | ||
97 | { | ||
98 | /* data offset */ | ||
99 | offset = get_long_be(buf + 4); | ||
100 | if (offset < 24) | ||
101 | { | ||
102 | DEBUGF("CODEC_ERROR: sun audio offset size is small: %d\n", offset); | ||
103 | return false; | ||
104 | } | ||
105 | /* data size */ | ||
106 | numbytes = get_long_be(buf + 8); | ||
107 | if (numbytes == (uint32_t)0xffffffff) | ||
108 | numbytes = id3->filesize - offset; | ||
109 | /* bitspersample */ | ||
110 | bitspersample = get_au_bitspersample(get_long_be(buf + 12)); | ||
111 | /* sample rate */ | ||
112 | id3->frequency = get_long_be(buf + 16); | ||
113 | channels = get_long_be(buf + 20); | ||
114 | } | ||
115 | |||
116 | totalsamples = numbytes / ((((bitspersample - 1) / 8) + 1) * channels); | ||
117 | |||
118 | /* Calculate track length (in ms) and estimate the bitrate (in kbit/s) */ | ||
119 | id3->length = ((int64_t) totalsamples * 1000) / id3->frequency; | ||
120 | |||
121 | return true; | ||
122 | } | ||
diff --git a/apps/metadata/metadata_parsers.h b/apps/metadata/metadata_parsers.h index 1e439c807b..392f16a8d9 100644 --- a/apps/metadata/metadata_parsers.h +++ b/apps/metadata/metadata_parsers.h | |||
@@ -43,3 +43,4 @@ bool get_rm_metadata(int fd, struct mp3entry* id3); | |||
43 | bool get_nsf_metadata(int fd, struct mp3entry* id3); | 43 | bool get_nsf_metadata(int fd, struct mp3entry* id3); |
44 | bool get_oma_metadata(int fd, struct mp3entry* id3); | 44 | bool get_oma_metadata(int fd, struct mp3entry* id3); |
45 | bool get_smaf_metadata(int fd, struct mp3entry* id3); | 45 | bool get_smaf_metadata(int fd, struct mp3entry* id3); |
46 | bool get_au_metadata(int fd, struct mp3entry* id3); | ||