Add Sun Audio codec. (FS#10433)
[kugel-rb.git] / apps / codecs / au.c
blob5632bf7a9b763ec4e1fa582282d64d6f37647928
1 /***************************************************************************
2 * __________ __ ___.
3 * Open \______ \ ____ ____ | | _\_ |__ _______ ___
4 * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
5 * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
6 * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
7 * \/ \/ \/ \/ \/
8 * $Id$
10 * Copyright (C) 2010 Yoshihisa Uchida
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.
17 * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
18 * KIND, either express or implied.
20 ****************************************************************************/
22 #include "codeclib.h"
23 #include "codecs/libpcm/support_formats.h"
25 CODEC_HEADER
27 /* Sun Audio file (Au file format) codec
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
35 #define PCM_SAMPLE_SIZE (1024*2)
37 static int32_t samples[PCM_SAMPLE_SIZE] IBSS_ATTR;
39 enum
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 */
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 */
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 },
86 #define NUM_FORMATS 4
88 static const struct pcm_codec *get_au_codec(uint32_t formattag)
90 int i;
92 for (i = 0; i < NUM_FORMATS; i++)
94 if (au_codecs[i].format_tag == formattag)
96 if (au_codecs[i].get_codec)
97 return au_codecs[i].get_codec();
98 return 0;
101 return 0;
104 static unsigned int get_be32(uint8_t *buf)
106 return (buf[0] << 24) | (buf[1] << 16) | (buf[2] << 8) | buf[3];
109 static int convert_au_format(unsigned int encoding, struct pcm_format *fmt)
111 if (encoding > 27)
113 fmt->formattag = AU_FORMAT_UNSUPPORT;
114 fmt->bitspersample = 0;
116 else
118 fmt->formattag = support_formats[encoding][0];
119 fmt->bitspersample = support_formats[encoding][1];
122 return fmt->formattag;
125 /* this is the codec entry point */
126 enum codec_status codec_main(void)
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;
140 /* Generic codec initialisation */
141 ci->configure(DSP_SET_SAMPLE_DEPTH, 28);
143 next_track:
144 if (codec_init()) {
145 DEBUGF("codec_init() error\n");
146 status = CODEC_ERROR;
147 goto exit;
150 while (!*ci->taginfo_ready && !ci->stop_codec)
151 ci->sleep(1);
153 codec_set_replaygain(ci->id3);
155 ci->memset(&format, 0, sizeof(struct pcm_format));
156 format.is_signed = true;
157 format.is_little_endian = false;
159 /* set format */
160 buf = ci->request_buffer(&n, 24);
161 if (n < 24 || (memcmp(buf, ".snd", 4) != 0))
164 * headerless sun audio file
165 * It is decoded under conditions.
166 * format: G.711 mu-law
167 * channel: mono
168 * frequency: 8000 kHz
170 offset = 0;
171 format.formattag = AU_FORMAT_MULAW;
172 format.channels = 1;
173 format.bitspersample = 8;
174 format.numbytes = ci->id3->filesize;
176 else
178 /* parse header */
180 /* data offset */
181 offset = get_be32(buf + 4);
182 if (offset < 24)
184 DEBUGF("CODEC_ERROR: sun audio offset size is small: %d\n", offset);
185 status = CODEC_ERROR;
186 goto done;
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)
196 DEBUGF("CODEC_ERROR: sun audio unsupport format: %d\n", get_be32(buf + 12));
197 status = CODEC_ERROR;
198 goto done;
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;
209 /* advance to first WAVE chunk */
210 ci->advance_buffer(offset);
212 firstblockposn = offset;
214 decodedsamples = 0;
215 codec = 0;
216 bytesdone = 0;
218 /* blockalign = 1 sample */
219 format.blockalign = format.bitspersample * format.channels >> 3;
221 /* get codec */
222 codec = get_au_codec(format.formattag);
223 if (!codec)
225 DEBUGF("CODEC_ERROR: unsupport sun audio format: %lx\n", format.formattag);
226 status = CODEC_ERROR;
227 goto done;
230 if (!codec->set_format(&format))
232 status = CODEC_ERROR;
233 goto done;
236 if (format.numbytes == 0) {
237 DEBUGF("CODEC_ERROR: data size is 0\n");
238 status = CODEC_ERROR;
239 goto done;
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)
248 DEBUGF("CODEC_ERROR: chunksize is 0\n");
249 status = CODEC_ERROR;
250 goto done;
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;
264 /* The main decoder loop */
265 endofstream = 0;
267 while (!endofstream) {
268 ci->yield();
269 if (ci->stop_codec || ci->new_track) {
270 break;
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);
277 decodedsamples = newpos->samples;
278 if (newpos->pos > format.numbytes)
279 break;
280 if (ci->seek_buffer(firstblockposn + newpos->pos))
282 bytesdone = newpos->pos;
284 ci->seek_complete();
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;
295 status = codec->decode(aubuf, n, samples, &bufcount);
296 if (status == CODEC_ERROR)
298 DEBUGF("codec error\n");
299 goto done;
302 ci->pcmbuf_insert(samples, NULL, bufcount);
303 ci->advance_buffer(n);
304 bytesdone += n;
305 decodedsamples += bufcount;
307 if (bytesdone >= format.numbytes)
308 endofstream = 1;
309 ci->set_elapsed(decodedsamples*1000LL/ci->id3->frequency);
311 status = CODEC_OK;
313 done:
314 if (ci->request_next_track())
315 goto next_track;
317 exit:
318 return status;