wav.c revision 1.14.6.2 1 1.14.6.2 martin /* $NetBSD: wav.c,v 1.14.6.2 2024/03/25 15:11:33 martin Exp $ */
2 1.3 mrg
3 1.1 mrg /*
4 1.14.6.1 martin * Copyright (c) 2002, 2009, 2013, 2015, 2019, 2024 Matthew R. Green
5 1.1 mrg * All rights reserved.
6 1.1 mrg *
7 1.1 mrg * Redistribution and use in source and binary forms, with or without
8 1.1 mrg * modification, are permitted provided that the following conditions
9 1.1 mrg * are met:
10 1.1 mrg * 1. Redistributions of source code must retain the above copyright
11 1.1 mrg * notice, this list of conditions and the following disclaimer.
12 1.1 mrg * 2. Redistributions in binary form must reproduce the above copyright
13 1.1 mrg * notice, this list of conditions and the following disclaimer in the
14 1.1 mrg * documentation and/or other materials provided with the distribution.
15 1.1 mrg *
16 1.1 mrg * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
17 1.1 mrg * IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
18 1.1 mrg * OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
19 1.1 mrg * IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
20 1.1 mrg * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING,
21 1.1 mrg * BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
22 1.1 mrg * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED
23 1.1 mrg * AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
24 1.1 mrg * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
25 1.1 mrg * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
26 1.1 mrg * SUCH DAMAGE.
27 1.1 mrg */
28 1.1 mrg
29 1.1 mrg /*
30 1.1 mrg * WAV support for the audio tools; thanks go to the sox utility for
31 1.1 mrg * clearing up issues with WAV files.
32 1.1 mrg */
33 1.6 agc #include <sys/cdefs.h>
34 1.6 agc
35 1.6 agc #ifndef lint
36 1.14.6.2 martin __RCSID("$NetBSD: wav.c,v 1.14.6.2 2024/03/25 15:11:33 martin Exp $");
37 1.6 agc #endif
38 1.6 agc
39 1.1 mrg
40 1.1 mrg #include <sys/types.h>
41 1.1 mrg #include <sys/audioio.h>
42 1.1 mrg #include <sys/ioctl.h>
43 1.1 mrg #include <sys/time.h>
44 1.1 mrg
45 1.1 mrg #include <ctype.h>
46 1.1 mrg #include <err.h>
47 1.1 mrg #include <stdio.h>
48 1.1 mrg #include <stdlib.h>
49 1.1 mrg #include <string.h>
50 1.9 mrg #include <stdint.h>
51 1.11 mrg #include <unistd.h>
52 1.14.6.1 martin #include <stdbool.h>
53 1.1 mrg
54 1.1 mrg #include "libaudio.h"
55 1.11 mrg #include "auconv.h"
56 1.1 mrg
57 1.10 joerg static const struct {
58 1.1 mrg int wenc;
59 1.2 mrg const char *wname;
60 1.1 mrg } wavencs[] = {
61 1.1 mrg { WAVE_FORMAT_UNKNOWN, "Microsoft Official Unknown" },
62 1.1 mrg { WAVE_FORMAT_PCM, "Microsoft PCM" },
63 1.1 mrg { WAVE_FORMAT_ADPCM, "Microsoft ADPCM" },
64 1.14.6.1 martin { WAVE_FORMAT_IEEE_FLOAT,"Microsoft IEEE Floating-Point" },
65 1.1 mrg { WAVE_FORMAT_ALAW, "Microsoft A-law" },
66 1.5 wiz { WAVE_FORMAT_MULAW, "Microsoft mu-law" },
67 1.1 mrg { WAVE_FORMAT_OKI_ADPCM,"OKI ADPCM" },
68 1.1 mrg { WAVE_FORMAT_DIGISTD, "Digistd format" },
69 1.1 mrg { WAVE_FORMAT_DIGIFIX, "Digifix format" },
70 1.1 mrg { -1, "?Unknown?" },
71 1.1 mrg };
72 1.1 mrg
73 1.2 mrg const char *
74 1.2 mrg wav_enc_from_val(int encoding)
75 1.1 mrg {
76 1.1 mrg int i;
77 1.1 mrg
78 1.1 mrg for (i = 0; wavencs[i].wenc != -1; i++)
79 1.1 mrg if (wavencs[i].wenc == encoding)
80 1.1 mrg break;
81 1.1 mrg return (wavencs[i].wname);
82 1.1 mrg }
83 1.1 mrg
84 1.1 mrg /*
85 1.1 mrg * sample header is:
86 1.1 mrg *
87 1.1 mrg * RIFF\^@^C^@WAVEfmt ^P^@^@^@^A^@^B^@D<AC>^@^@^P<B1>^B^@^D^@^P^@data^@^@^C^@^@^@^@^@^@^@^@^@^@
88 1.1 mrg *
89 1.1 mrg */
90 1.1 mrg /*
91 1.1 mrg * WAV format helpers
92 1.1 mrg */
93 1.14.6.1 martin
94 1.14.6.1 martin static bool
95 1.14.6.1 martin find_riff_chunk(const char search[4], size_t *remainp, char **wherep, uint32_t *partlen)
96 1.14.6.1 martin {
97 1.14.6.1 martin wav_audioheaderpart part;
98 1.14.6.1 martin
99 1.14.6.1 martin *partlen = 0;
100 1.14.6.1 martin
101 1.14.6.1 martin #define ADJUST(l) do { \
102 1.14.6.2 martin if (l > *(remainp)) \
103 1.14.6.1 martin return false; \
104 1.14.6.1 martin *(wherep) += (l); \
105 1.14.6.1 martin *(remainp) -= (l); \
106 1.14.6.1 martin } while (0)
107 1.14.6.1 martin
108 1.14.6.1 martin while (*remainp >= sizeof part) {
109 1.14.6.1 martin const char *emsg = "";
110 1.14.6.1 martin uint32_t len;
111 1.14.6.1 martin
112 1.14.6.1 martin memcpy(&part, *wherep, sizeof part);
113 1.14.6.1 martin ADJUST(sizeof part);
114 1.14.6.1 martin len = getle32(part.len);
115 1.14.6.1 martin if (len % 2) {
116 1.14.6.1 martin emsg = " (odd length, adjusted)";
117 1.14.6.1 martin len += 1;
118 1.14.6.1 martin }
119 1.14.6.1 martin if (strncmp(part.name, search, sizeof *search) == 0) {
120 1.14.6.1 martin *partlen = len;
121 1.14.6.1 martin if (verbose > 1)
122 1.14.6.1 martin fprintf(stderr, "Found part %.04s length %d%s\n",
123 1.14.6.1 martin part.name, len, emsg);
124 1.14.6.1 martin return true;
125 1.14.6.1 martin }
126 1.14.6.1 martin ADJUST(len);
127 1.14.6.1 martin if (verbose > 1)
128 1.14.6.1 martin fprintf(stderr, "Skipping part %.04s length %d%s\n",
129 1.14.6.1 martin part.name, len, emsg);
130 1.14.6.1 martin }
131 1.14.6.1 martin #undef ADJUST
132 1.14.6.1 martin
133 1.14.6.1 martin return false;
134 1.14.6.1 martin }
135 1.14.6.1 martin
136 1.1 mrg /*
137 1.1 mrg * find a .wav header, etc. returns header length on success
138 1.1 mrg */
139 1.1 mrg ssize_t
140 1.10 joerg audio_wav_parse_hdr(void *hdr, size_t sz, u_int *enc, u_int *prec,
141 1.13 mrg u_int *sample, u_int *channels, off_t *datasize)
142 1.1 mrg {
143 1.14.6.1 martin char *where = hdr;
144 1.1 mrg wav_audioheaderfmt fmt;
145 1.9 mrg wav_audiohdrextensible ext;
146 1.14.6.1 martin size_t remain = sz;
147 1.7 mrg u_int newenc, newprec;
148 1.14.6.1 martin uint32_t len = 0;
149 1.9 mrg u_int16_t fmttag;
150 1.2 mrg static const char
151 1.2 mrg strfmt[4] = "fmt ",
152 1.2 mrg strRIFF[4] = "RIFF",
153 1.2 mrg strWAVE[4] = "WAVE",
154 1.2 mrg strdata[4] = "data";
155 1.14.6.1 martin bool found;
156 1.14.6.1 martin
157 1.1 mrg if (sz < 32)
158 1.1 mrg return (AUDIO_ENOENT);
159 1.1 mrg
160 1.14.6.1 martin #define ADJUST(l) do { \
161 1.14.6.2 martin if ((l) > remain) \
162 1.14.6.1 martin return (AUDIO_ESHORTHDR); \
163 1.14.6.1 martin where += (l); \
164 1.14.6.1 martin remain -= (l); \
165 1.14.6.1 martin } while (0)
166 1.14.6.1 martin
167 1.14.6.1 martin if (memcmp(where, strRIFF, sizeof strRIFF) != 0)
168 1.1 mrg return (AUDIO_ENOENT);
169 1.14.6.1 martin ADJUST(sizeof strRIFF);
170 1.14.6.1 martin /* XXX we ignore the RIFF length here */
171 1.14.6.1 martin ADJUST(4);
172 1.14.6.1 martin if (memcmp(where, strWAVE, sizeof strWAVE) != 0)
173 1.1 mrg return (AUDIO_ENOENT);
174 1.14.6.1 martin ADJUST(sizeof strWAVE);
175 1.1 mrg
176 1.14.6.1 martin found = find_riff_chunk(strfmt, &remain, &where, &len);
177 1.1 mrg
178 1.1 mrg /* too short ? */
179 1.14.6.1 martin if (!found || remain <= sizeof fmt)
180 1.1 mrg return (AUDIO_ESHORTHDR);
181 1.1 mrg
182 1.14.6.1 martin memcpy(&fmt, where, sizeof fmt);
183 1.9 mrg fmttag = getle16(fmt.tag);
184 1.9 mrg if (verbose)
185 1.14.6.1 martin printf("WAVE format tag/len: %04x/%u\n", fmttag, len);
186 1.9 mrg
187 1.9 mrg if (fmttag == WAVE_FORMAT_EXTENSIBLE) {
188 1.14.6.1 martin if (len < sizeof(fmt) + sizeof(ext)) {
189 1.14.6.1 martin if (verbose)
190 1.14.6.1 martin fprintf(stderr, "short WAVE ext fmt\n");
191 1.9 mrg return (AUDIO_ESHORTHDR);
192 1.14.6.1 martin }
193 1.14.6.1 martin if (remain <= sizeof ext + sizeof fmt) {
194 1.14.6.1 martin if (verbose)
195 1.14.6.1 martin fprintf(stderr, "WAVE ext truncated\n");
196 1.9 mrg return (AUDIO_ESHORTHDR);
197 1.14.6.1 martin }
198 1.14.6.1 martin memcpy(&ext, where + sizeof fmt, sizeof ext);
199 1.14 jdolecek fmttag = getle16(ext.sub_tag);
200 1.14.6.1 martin uint16_t sublen = getle16(ext.len);
201 1.9 mrg if (verbose)
202 1.14.6.1 martin printf("WAVE extensible tag/len: %04x/%u\n", fmttag, sublen);
203 1.14.6.1 martin
204 1.14.6.1 martin /*
205 1.14.6.1 martin * XXXMRG: it may be that part.len (aka sizeof fmt + sizeof ext)
206 1.14.6.1 martin * should equal sizeof fmt + sizeof ext.len + sublen? this block
207 1.14.6.1 martin * is only entered for part.len == 40, where ext.len is expected
208 1.14.6.1 martin * to be 22 (sizeof ext.len = 2, sizeof fmt = 16).
209 1.14.6.1 martin *
210 1.14.6.1 martin * warn about this, but don't consider it an error.
211 1.14.6.1 martin */
212 1.14.6.1 martin if (getle16(ext.len) != 22 && verbose) {
213 1.14.6.1 martin fprintf(stderr, "warning: WAVE ext.len %u not 22\n",
214 1.14.6.1 martin getle16(ext.len));
215 1.14.6.1 martin }
216 1.14.6.1 martin } else if (len < sizeof(fmt)) {
217 1.14.6.1 martin if (verbose)
218 1.14.6.1 martin fprintf(stderr, "WAVE fmt unsupported size %u\n", len);
219 1.14.6.1 martin return (AUDIO_EWAVUNSUPP);
220 1.9 mrg }
221 1.14.6.1 martin ADJUST(len);
222 1.9 mrg
223 1.9 mrg switch (fmttag) {
224 1.1 mrg default:
225 1.1 mrg return (AUDIO_EWAVUNSUPP);
226 1.1 mrg
227 1.1 mrg case WAVE_FORMAT_PCM:
228 1.9 mrg case WAVE_FORMAT_ADPCM:
229 1.9 mrg case WAVE_FORMAT_OKI_ADPCM:
230 1.9 mrg case WAVE_FORMAT_IMA_ADPCM:
231 1.9 mrg case WAVE_FORMAT_DIGIFIX:
232 1.9 mrg case WAVE_FORMAT_DIGISTD:
233 1.1 mrg switch (getle16(fmt.bits_per_sample)) {
234 1.1 mrg case 8:
235 1.1 mrg newprec = 8;
236 1.1 mrg break;
237 1.1 mrg case 16:
238 1.1 mrg newprec = 16;
239 1.1 mrg break;
240 1.1 mrg case 24:
241 1.1 mrg newprec = 24;
242 1.1 mrg break;
243 1.1 mrg case 32:
244 1.1 mrg newprec = 32;
245 1.1 mrg break;
246 1.1 mrg default:
247 1.1 mrg return (AUDIO_EWAVBADPCM);
248 1.1 mrg }
249 1.1 mrg if (newprec == 8)
250 1.1 mrg newenc = AUDIO_ENCODING_ULINEAR_LE;
251 1.1 mrg else
252 1.1 mrg newenc = AUDIO_ENCODING_SLINEAR_LE;
253 1.1 mrg break;
254 1.1 mrg case WAVE_FORMAT_ALAW:
255 1.1 mrg newenc = AUDIO_ENCODING_ALAW;
256 1.1 mrg newprec = 8;
257 1.1 mrg break;
258 1.1 mrg case WAVE_FORMAT_MULAW:
259 1.1 mrg newenc = AUDIO_ENCODING_ULAW;
260 1.1 mrg newprec = 8;
261 1.1 mrg break;
262 1.14.6.1 martin case WAVE_FORMAT_IEEE_FLOAT:
263 1.14.6.1 martin switch (getle16(fmt.bits_per_sample)) {
264 1.14.6.1 martin case 32:
265 1.14.6.1 martin newenc = AUDIO_ENCODING_LIBAUDIO_FLOAT32;
266 1.14.6.1 martin newprec = 32;
267 1.14.6.1 martin break;
268 1.14.6.1 martin case 64:
269 1.14.6.1 martin newenc = AUDIO_ENCODING_LIBAUDIO_FLOAT64;
270 1.14.6.1 martin newprec = 32;
271 1.14.6.1 martin break;
272 1.14.6.1 martin default:
273 1.14.6.1 martin return (AUDIO_EWAVBADPCM);
274 1.14.6.1 martin }
275 1.14.6.1 martin break;
276 1.1 mrg }
277 1.1 mrg
278 1.14.6.1 martin found = find_riff_chunk(strdata, &remain, &where, &len);
279 1.14.6.1 martin if (!found)
280 1.14.6.1 martin return (AUDIO_EWAVNODATA);
281 1.1 mrg
282 1.14.6.2 martin if (channels)
283 1.14.6.2 martin *channels = (u_int)getle16(fmt.channels);
284 1.14.6.2 martin if (sample)
285 1.14.6.2 martin *sample = getle32(fmt.sample_rate);
286 1.14.6.2 martin if (enc)
287 1.14.6.2 martin *enc = newenc;
288 1.14.6.2 martin if (prec)
289 1.14.6.2 martin *prec = newprec;
290 1.14.6.2 martin if (datasize)
291 1.14.6.2 martin *datasize = (off_t)len;
292 1.14.6.2 martin return (where - (char *)hdr);
293 1.14.6.1 martin
294 1.14.6.1 martin #undef ADJUST
295 1.1 mrg }
296 1.11 mrg
297 1.11 mrg
298 1.11 mrg /*
299 1.11 mrg * prepare a WAV header for writing; we fill in hdrp, lenp and leftp,
300 1.11 mrg * and expect our caller (wav_write_header()) to use them.
301 1.11 mrg */
302 1.11 mrg int
303 1.13 mrg wav_prepare_header(struct track_info *ti, void **hdrp, size_t *lenp, int *leftp)
304 1.11 mrg {
305 1.11 mrg /*
306 1.11 mrg * WAV header we write looks like this:
307 1.11 mrg *
308 1.11 mrg * bytes purpose
309 1.11 mrg * 0-3 "RIFF"
310 1.14.6.1 martin * 4-7 RIFF chunk length (file length minus 8)
311 1.11 mrg * 8-15 "WAVEfmt "
312 1.11 mrg * 16-19 format size
313 1.11 mrg * 20-21 format tag
314 1.11 mrg * 22-23 number of channels
315 1.11 mrg * 24-27 sample rate
316 1.11 mrg * 28-31 average bytes per second
317 1.11 mrg * 32-33 block alignment
318 1.11 mrg * 34-35 bits per sample
319 1.11 mrg *
320 1.11 mrg * then for ULAW and ALAW outputs, we have an extended chunk size
321 1.11 mrg * and a WAV "fact" to add:
322 1.11 mrg *
323 1.11 mrg * 36-37 length of extension (== 0)
324 1.11 mrg * 38-41 "fact"
325 1.11 mrg * 42-45 fact size
326 1.11 mrg * 46-49 number of samples written
327 1.11 mrg * 50-53 "data"
328 1.11 mrg * 54-57 data length
329 1.11 mrg * 58- raw audio data
330 1.11 mrg *
331 1.11 mrg * for PCM outputs we have just the data remaining:
332 1.11 mrg *
333 1.11 mrg * 36-39 "data"
334 1.11 mrg * 40-43 data length
335 1.11 mrg * 44- raw audio data
336 1.11 mrg *
337 1.11 mrg * RIFF\^@^C^@WAVEfmt ^P^@^@^@^A^@^B^@D<AC>^@^@^P<B1>^B^@^D^@^P^@data^@^@^C^@^@^@^@^@^@^@^@^@^@
338 1.11 mrg */
339 1.11 mrg static char wavheaderbuf[64];
340 1.11 mrg char *p = wavheaderbuf;
341 1.11 mrg const char *riff = "RIFF",
342 1.11 mrg *wavefmt = "WAVEfmt ",
343 1.11 mrg *fact = "fact",
344 1.11 mrg *data = "data";
345 1.11 mrg u_int32_t filelen, fmtsz, sps, abps, factsz = 4, nsample, datalen;
346 1.12 christos u_int16_t fmttag, nchan, align, extln = 0;
347 1.11 mrg
348 1.13 mrg if (ti->header_info)
349 1.11 mrg warnx("header information not supported for WAV");
350 1.11 mrg *leftp = 0;
351 1.11 mrg
352 1.13 mrg switch (ti->precision) {
353 1.11 mrg case 8:
354 1.11 mrg break;
355 1.11 mrg case 16:
356 1.11 mrg break;
357 1.14.6.1 martin case 24:
358 1.14.6.1 martin break;
359 1.11 mrg case 32:
360 1.11 mrg break;
361 1.11 mrg default:
362 1.11 mrg {
363 1.11 mrg static int warned = 0;
364 1.11 mrg
365 1.11 mrg if (warned == 0) {
366 1.13 mrg warnx("can not support precision of %d", ti->precision);
367 1.11 mrg warned = 1;
368 1.11 mrg }
369 1.11 mrg }
370 1.11 mrg return (-1);
371 1.11 mrg }
372 1.11 mrg
373 1.13 mrg switch (ti->encoding) {
374 1.11 mrg case AUDIO_ENCODING_ULAW:
375 1.11 mrg fmttag = WAVE_FORMAT_MULAW;
376 1.11 mrg fmtsz = 18;
377 1.13 mrg align = ti->channels;
378 1.11 mrg break;
379 1.11 mrg
380 1.11 mrg case AUDIO_ENCODING_ALAW:
381 1.11 mrg fmttag = WAVE_FORMAT_ALAW;
382 1.11 mrg fmtsz = 18;
383 1.13 mrg align = ti->channels;
384 1.11 mrg break;
385 1.11 mrg
386 1.11 mrg /*
387 1.11 mrg * we could try to support RIFX but it seems to be more portable
388 1.11 mrg * to output little-endian data for WAV files.
389 1.11 mrg */
390 1.11 mrg case AUDIO_ENCODING_ULINEAR_BE:
391 1.11 mrg case AUDIO_ENCODING_SLINEAR_BE:
392 1.11 mrg case AUDIO_ENCODING_ULINEAR_LE:
393 1.11 mrg case AUDIO_ENCODING_SLINEAR_LE:
394 1.11 mrg case AUDIO_ENCODING_PCM16:
395 1.11 mrg
396 1.11 mrg #if BYTE_ORDER == LITTLE_ENDIAN
397 1.11 mrg case AUDIO_ENCODING_ULINEAR:
398 1.11 mrg case AUDIO_ENCODING_SLINEAR:
399 1.11 mrg #endif
400 1.11 mrg fmttag = WAVE_FORMAT_PCM;
401 1.11 mrg fmtsz = 16;
402 1.13 mrg align = ti->channels * (ti->precision / 8);
403 1.11 mrg break;
404 1.11 mrg
405 1.11 mrg default:
406 1.11 mrg #if 0 // move into record.c, and maybe merge.c
407 1.11 mrg {
408 1.11 mrg static int warned = 0;
409 1.11 mrg
410 1.11 mrg if (warned == 0) {
411 1.13 mrg const char *s = wav_enc_from_val(ti->encoding);
412 1.11 mrg
413 1.11 mrg if (s == NULL)
414 1.11 mrg warnx("can not support encoding of %s", s);
415 1.11 mrg else
416 1.13 mrg warnx("can not support encoding of %d", ti->encoding);
417 1.11 mrg warned = 1;
418 1.11 mrg }
419 1.11 mrg }
420 1.11 mrg #endif
421 1.13 mrg ti->format = AUDIO_FORMAT_NONE;
422 1.11 mrg return (-1);
423 1.11 mrg }
424 1.11 mrg
425 1.13 mrg nchan = ti->channels;
426 1.13 mrg sps = ti->sample_rate;
427 1.11 mrg
428 1.11 mrg /* data length */
429 1.13 mrg if (ti->outfd == STDOUT_FILENO)
430 1.11 mrg datalen = 0;
431 1.13 mrg else if (ti->total_size != -1)
432 1.13 mrg datalen = ti->total_size;
433 1.11 mrg else
434 1.11 mrg datalen = 0;
435 1.11 mrg
436 1.11 mrg /* file length */
437 1.11 mrg filelen = 4 + (8 + fmtsz) + (8 + datalen);
438 1.11 mrg if (fmttag != WAVE_FORMAT_PCM)
439 1.11 mrg filelen += 8 + factsz;
440 1.11 mrg
441 1.13 mrg abps = (double)align*ti->sample_rate / (double)1 + 0.5;
442 1.11 mrg
443 1.13 mrg nsample = (datalen / ti->precision) / ti->sample_rate;
444 1.14.6.1 martin
445 1.11 mrg /*
446 1.11 mrg * now we've calculated the info, write it out!
447 1.11 mrg */
448 1.11 mrg #define put32(x) do { \
449 1.11 mrg u_int32_t _f; \
450 1.11 mrg putle32(_f, (x)); \
451 1.11 mrg memcpy(p, &_f, 4); \
452 1.11 mrg } while (0)
453 1.11 mrg #define put16(x) do { \
454 1.11 mrg u_int16_t _f; \
455 1.11 mrg putle16(_f, (x)); \
456 1.11 mrg memcpy(p, &_f, 2); \
457 1.11 mrg } while (0)
458 1.11 mrg memcpy(p, riff, 4);
459 1.11 mrg p += 4; /* 4 */
460 1.11 mrg put32(filelen);
461 1.11 mrg p += 4; /* 8 */
462 1.11 mrg memcpy(p, wavefmt, 8);
463 1.11 mrg p += 8; /* 16 */
464 1.11 mrg put32(fmtsz);
465 1.11 mrg p += 4; /* 20 */
466 1.11 mrg put16(fmttag);
467 1.11 mrg p += 2; /* 22 */
468 1.11 mrg put16(nchan);
469 1.11 mrg p += 2; /* 24 */
470 1.11 mrg put32(sps);
471 1.11 mrg p += 4; /* 28 */
472 1.11 mrg put32(abps);
473 1.11 mrg p += 4; /* 32 */
474 1.11 mrg put16(align);
475 1.11 mrg p += 2; /* 34 */
476 1.13 mrg put16(ti->precision);
477 1.11 mrg p += 2; /* 36 */
478 1.11 mrg /* NON PCM formats have an extended chunk; write it */
479 1.11 mrg if (fmttag != WAVE_FORMAT_PCM) {
480 1.11 mrg put16(extln);
481 1.11 mrg p += 2; /* 38 */
482 1.11 mrg memcpy(p, fact, 4);
483 1.11 mrg p += 4; /* 42 */
484 1.11 mrg put32(factsz);
485 1.11 mrg p += 4; /* 46 */
486 1.11 mrg put32(nsample);
487 1.11 mrg p += 4; /* 50 */
488 1.11 mrg }
489 1.11 mrg memcpy(p, data, 4);
490 1.11 mrg p += 4; /* 40/54 */
491 1.11 mrg put32(datalen);
492 1.11 mrg p += 4; /* 44/58 */
493 1.11 mrg #undef put32
494 1.11 mrg #undef put16
495 1.11 mrg
496 1.11 mrg *hdrp = wavheaderbuf;
497 1.11 mrg *lenp = (p - wavheaderbuf);
498 1.11 mrg
499 1.11 mrg return 0;
500 1.11 mrg }
501 1.11 mrg
502 1.11 mrg write_conv_func
503 1.13 mrg wav_write_get_conv_func(struct track_info *ti)
504 1.11 mrg {
505 1.11 mrg write_conv_func conv_func = NULL;
506 1.11 mrg
507 1.13 mrg switch (ti->encoding) {
508 1.11 mrg
509 1.11 mrg /*
510 1.11 mrg * we could try to support RIFX but it seems to be more portable
511 1.11 mrg * to output little-endian data for WAV files.
512 1.11 mrg */
513 1.11 mrg case AUDIO_ENCODING_ULINEAR_BE:
514 1.11 mrg #if BYTE_ORDER == BIG_ENDIAN
515 1.11 mrg case AUDIO_ENCODING_ULINEAR:
516 1.11 mrg #endif
517 1.13 mrg if (ti->precision == 16)
518 1.11 mrg conv_func = change_sign16_swap_bytes_be;
519 1.13 mrg else if (ti->precision == 32)
520 1.11 mrg conv_func = change_sign32_swap_bytes_be;
521 1.11 mrg break;
522 1.11 mrg
523 1.11 mrg case AUDIO_ENCODING_SLINEAR_BE:
524 1.11 mrg #if BYTE_ORDER == BIG_ENDIAN
525 1.11 mrg case AUDIO_ENCODING_SLINEAR:
526 1.11 mrg #endif
527 1.13 mrg if (ti->precision == 8)
528 1.11 mrg conv_func = change_sign8;
529 1.13 mrg else if (ti->precision == 16)
530 1.11 mrg conv_func = swap_bytes;
531 1.13 mrg else if (ti->precision == 32)
532 1.11 mrg conv_func = swap_bytes32;
533 1.11 mrg break;
534 1.11 mrg
535 1.11 mrg case AUDIO_ENCODING_ULINEAR_LE:
536 1.11 mrg #if BYTE_ORDER == LITTLE_ENDIAN
537 1.11 mrg case AUDIO_ENCODING_ULINEAR:
538 1.11 mrg #endif
539 1.13 mrg if (ti->precision == 16)
540 1.11 mrg conv_func = change_sign16_le;
541 1.13 mrg else if (ti->precision == 32)
542 1.11 mrg conv_func = change_sign32_le;
543 1.11 mrg break;
544 1.11 mrg
545 1.11 mrg case AUDIO_ENCODING_SLINEAR_LE:
546 1.11 mrg case AUDIO_ENCODING_PCM16:
547 1.11 mrg #if BYTE_ORDER == LITTLE_ENDIAN
548 1.11 mrg case AUDIO_ENCODING_SLINEAR:
549 1.11 mrg #endif
550 1.13 mrg if (ti->precision == 8)
551 1.11 mrg conv_func = change_sign8;
552 1.11 mrg break;
553 1.11 mrg
554 1.11 mrg default:
555 1.13 mrg ti->format = AUDIO_FORMAT_NONE;
556 1.11 mrg }
557 1.11 mrg
558 1.11 mrg return conv_func;
559 1.11 mrg }
560