xref: /netbsd/usr.bin/audio/record/record.c (revision 6550d01e)
1 /*	$NetBSD: record.c,v 1.50 2010/12/29 18:49:41 wiz Exp $	*/
2 
3 /*
4  * Copyright (c) 1999, 2002, 2003, 2005, 2010 Matthew R. Green
5  * All rights reserved.
6  *
7  * Redistribution and use in source and binary forms, with or without
8  * modification, are permitted provided that the following conditions
9  * are met:
10  * 1. Redistributions of source code must retain the above copyright
11  *    notice, this list of conditions and the following disclaimer.
12  * 2. Redistributions in binary form must reproduce the above copyright
13  *    notice, this list of conditions and the following disclaimer in the
14  *    documentation and/or other materials provided with the distribution.
15  *
16  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
17  * IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
18  * OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
19  * IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
20  * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING,
21  * BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
22  * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED
23  * AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
24  * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
25  * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
26  * SUCH DAMAGE.
27  */
28 
29 /*
30  * SunOS compatible audiorecord(1)
31  */
32 #include <sys/cdefs.h>
33 
34 #ifndef lint
35 __RCSID("$NetBSD: record.c,v 1.50 2010/12/29 18:49:41 wiz Exp $");
36 #endif
37 
38 
39 #include <sys/param.h>
40 #include <sys/audioio.h>
41 #include <sys/ioctl.h>
42 #include <sys/time.h>
43 #include <sys/uio.h>
44 
45 #include <err.h>
46 #include <fcntl.h>
47 #include <paths.h>
48 #include <signal.h>
49 #include <stdio.h>
50 #include <stdlib.h>
51 #include <string.h>
52 #include <unistd.h>
53 #include <util.h>
54 
55 #include "libaudio.h"
56 #include "auconv.h"
57 
58 audio_info_t info, oinfo;
59 ssize_t	total_size = -1;
60 const char *device;
61 int	format = AUDIO_FORMAT_DEFAULT;
62 char	*header_info;
63 char	default_info[8] = { '\0', '\0', '\0', '\0', '\0', '\0', '\0', '\0' };
64 int	audiofd, outfd;
65 int	qflag, aflag, fflag;
66 int	verbose;
67 int	monitor_gain, omonitor_gain;
68 int	gain;
69 int	balance;
70 int	port;
71 int	encoding;
72 char	*encoding_str;
73 int	precision;
74 int	sample_rate;
75 int	channels;
76 struct timeval record_time;
77 struct timeval start_time;
78 
79 void (*conv_func) (u_char *, int);
80 
81 void usage (void);
82 int main (int, char *[]);
83 int timeleft (struct timeval *, struct timeval *);
84 void cleanup (int) __dead;
85 int write_header_sun (void **, size_t *, int *);
86 int write_header_wav (void **, size_t *, int *);
87 void write_header (void);
88 void rewrite_header (void);
89 
90 int
91 main(argc, argv)
92 	int argc;
93 	char *argv[];
94 {
95 	u_char	*buffer;
96 	size_t	len, bufsize = 0;
97 	int	ch, no_time_limit = 1;
98 	const char *defdevice = _PATH_SOUND;
99 
100 	while ((ch = getopt(argc, argv, "ab:B:C:F:c:d:e:fhi:m:P:p:qt:s:Vv:")) != -1) {
101 		switch (ch) {
102 		case 'a':
103 			aflag++;
104 			break;
105 		case 'b':
106 			decode_int(optarg, &balance);
107 			if (balance < 0 || balance > 63)
108 				errx(1, "balance must be between 0 and 63");
109 			break;
110 		case 'B':
111 			bufsize = strsuftoll("read buffer size", optarg,
112 					     1, UINT_MAX);
113 			break;
114 		case 'C':
115 			/* Ignore, compatibility */
116 			break;
117 		case 'F':
118 			format = audio_format_from_str(optarg);
119 			if (format < 0)
120 				errx(1, "Unknown audio format; supported "
121 				    "formats: \"sun\", \"wav\", and \"none\"");
122 			break;
123 		case 'c':
124 			decode_int(optarg, &channels);
125 			if (channels < 0 || channels > 16)
126 				errx(1, "channels must be between 0 and 16");
127 			break;
128 		case 'd':
129 			device = optarg;
130 			break;
131 		case 'e':
132 			encoding_str = optarg;
133 			break;
134 		case 'f':
135 			fflag++;
136 			break;
137 		case 'i':
138 			header_info = optarg;
139 			break;
140 		case 'm':
141 			decode_int(optarg, &monitor_gain);
142 			if (monitor_gain < 0 || monitor_gain > 255)
143 				errx(1, "monitor volume must be between 0 and 255");
144 			break;
145 		case 'P':
146 			decode_int(optarg, &precision);
147 			if (precision != 4 && precision != 8 &&
148 			    precision != 16 && precision != 24 &&
149 			    precision != 32)
150 				errx(1, "precision must be between 4, 8, 16, 24 or 32");
151 			break;
152 		case 'p':
153 			len = strlen(optarg);
154 
155 			if (strncmp(optarg, "mic", len) == 0)
156 				port |= AUDIO_MICROPHONE;
157 			else if (strncmp(optarg, "cd", len) == 0 ||
158 			           strncmp(optarg, "internal-cd", len) == 0)
159 				port |= AUDIO_CD;
160 			else if (strncmp(optarg, "line", len) == 0)
161 				port |= AUDIO_LINE_IN;
162 			else
163 				errx(1,
164 			    "port must be `cd', `internal-cd', `mic', or `line'");
165 			break;
166 		case 'q':
167 			qflag++;
168 			break;
169 		case 's':
170 			decode_int(optarg, &sample_rate);
171 			if (sample_rate < 0 || sample_rate > 48000 * 2)	/* XXX */
172 				errx(1, "sample rate must be between 0 and 96000");
173 			break;
174 		case 't':
175 			no_time_limit = 0;
176 			decode_time(optarg, &record_time);
177 			break;
178 		case 'V':
179 			verbose++;
180 			break;
181 		case 'v':
182 			decode_int(optarg, &gain);
183 			if (gain < 0 || gain > 255)
184 				errx(1, "volume must be between 0 and 255");
185 			break;
186 		/* case 'h': */
187 		default:
188 			usage();
189 			/* NOTREACHED */
190 		}
191 	}
192 	argc -= optind;
193 	argv += optind;
194 
195 	if (argc != 1)
196 		usage();
197 
198 	/*
199 	 * convert the encoding string into a value.
200 	 */
201 	if (encoding_str) {
202 		encoding = audio_enc_to_val(encoding_str);
203 		if (encoding == -1)
204 			errx(1, "unknown encoding, bailing...");
205 	}
206 
207 	/*
208 	 * open the output file
209 	 */
210 	if (argv[0][0] != '-' || argv[0][1] != '\0') {
211 		/* intuit the file type from the name */
212 		if (format == AUDIO_FORMAT_DEFAULT)
213 		{
214 			size_t flen = strlen(*argv);
215 			const char *arg = *argv;
216 
217 			if (strcasecmp(arg + flen - 3, ".au") == 0)
218 				format = AUDIO_FORMAT_SUN;
219 			else if (strcasecmp(arg + flen - 4, ".wav") == 0)
220 				format = AUDIO_FORMAT_WAV;
221 		}
222 		outfd = open(*argv, O_CREAT|(aflag ? O_APPEND : O_TRUNC)|O_WRONLY, 0666);
223 		if (outfd < 0)
224 			err(1, "could not open %s", *argv);
225 	} else
226 		outfd = STDOUT_FILENO;
227 
228 	/*
229 	 * open the audio device
230 	 */
231 	if (device == NULL && (device = getenv("AUDIODEVICE")) == NULL &&
232 	    (device = getenv("AUDIODEV")) == NULL) /* Sun compatibility */
233 		device = defdevice;
234 
235 	audiofd = open(device, O_RDONLY);
236 	if (audiofd < 0 && device == defdevice) {
237 		device = _PATH_SOUND0;
238 		audiofd = open(device, O_RDONLY);
239 	}
240 	if (audiofd < 0)
241 		err(1, "failed to open %s", device);
242 
243 	/*
244 	 * work out the buffer size to use, and allocate it.  also work out
245 	 * what the old monitor gain value is, so that we can reset it later.
246 	 */
247 	if (ioctl(audiofd, AUDIO_GETINFO, &oinfo) < 0)
248 		err(1, "failed to get audio info");
249 	if (bufsize == 0) {
250 		bufsize = oinfo.record.buffer_size;
251 		if (bufsize < 32 * 1024)
252 			bufsize = 32 * 1024;
253 	}
254 	omonitor_gain = oinfo.monitor_gain;
255 
256 	buffer = malloc(bufsize);
257 	if (buffer == NULL)
258 		err(1, "couldn't malloc buffer of %d size", (int)bufsize);
259 
260 	/*
261 	 * set up audio device for recording with the speified parameters
262 	 */
263 	AUDIO_INITINFO(&info);
264 
265 	/*
266 	 * for these, get the current values for stuffing into the header
267 	 */
268 #define SETINFO(x)	if (x) \
269 				info.record.x = x; \
270 			else \
271 				info.record.x = x = oinfo.record.x;
272 	SETINFO (sample_rate)
273 	SETINFO (channels)
274 	SETINFO (precision)
275 	SETINFO (encoding)
276 	SETINFO (gain)
277 	SETINFO (port)
278 	SETINFO (balance)
279 #undef SETINFO
280 
281 	if (monitor_gain)
282 		info.monitor_gain = monitor_gain;
283 	else
284 		monitor_gain = oinfo.monitor_gain;
285 
286 	info.mode = AUMODE_RECORD;
287 	if (ioctl(audiofd, AUDIO_SETINFO, &info) < 0)
288 		err(1, "failed to set audio info");
289 
290 	signal(SIGINT, cleanup);
291 	write_header();
292 	total_size = 0;
293 
294 	if (verbose && conv_func) {
295 		const char *s = NULL;
296 
297 		if (conv_func == swap_bytes)
298 			s = "swap bytes (16 bit)";
299 		else if (conv_func == swap_bytes32)
300 			s = "swap bytes (32 bit)";
301 		else if (conv_func == change_sign16_be)
302 			s = "change sign (big-endian, 16 bit)";
303 		else if (conv_func == change_sign16_le)
304 			s = "change sign (little-endian, 16 bit)";
305 		else if (conv_func == change_sign32_be)
306 			s = "change sign (big-endian, 32 bit)";
307 		else if (conv_func == change_sign32_le)
308 			s = "change sign (little-endian, 32 bit)";
309 		else if (conv_func == change_sign16_swap_bytes_be)
310 			s = "change sign & swap bytes (big-endian, 16 bit)";
311 		else if (conv_func == change_sign16_swap_bytes_le)
312 			s = "change sign & swap bytes (little-endian, 16 bit)";
313 		else if (conv_func == change_sign32_swap_bytes_be)
314 			s = "change sign (big-endian, 32 bit)";
315 		else if (conv_func == change_sign32_swap_bytes_le)
316 			s = "change sign & swap bytes (little-endian, 32 bit)";
317 
318 		if (s)
319 			fprintf(stderr, "%s: converting, using function: %s\n",
320 			    getprogname(), s);
321 		else
322 			fprintf(stderr, "%s: using unnamed conversion "
323 					"function\n", getprogname());
324 	}
325 
326 	if (verbose)
327 		fprintf(stderr,
328 		   "sample_rate=%d channels=%d precision=%d encoding=%s\n",
329 		   info.record.sample_rate, info.record.channels,
330 		   info.record.precision,
331 		   audio_enc_from_val(info.record.encoding));
332 
333 	if (!no_time_limit && verbose)
334 		fprintf(stderr, "recording for %lu seconds, %lu microseconds\n",
335 		    (u_long)record_time.tv_sec, (u_long)record_time.tv_usec);
336 
337 	(void)gettimeofday(&start_time, NULL);
338 	while (no_time_limit || timeleft(&start_time, &record_time)) {
339 		if ((size_t)read(audiofd, buffer, bufsize) != bufsize)
340 			err(1, "read failed");
341 		if (conv_func)
342 			(*conv_func)(buffer, bufsize);
343 		if ((size_t)write(outfd, buffer, bufsize) != bufsize)
344 			err(1, "write failed");
345 		total_size += bufsize;
346 	}
347 	cleanup(0);
348 }
349 
350 int
351 timeleft(start_tvp, record_tvp)
352 	struct timeval *start_tvp;
353 	struct timeval *record_tvp;
354 {
355 	struct timeval now, diff;
356 
357 	(void)gettimeofday(&now, NULL);
358 	timersub(&now, start_tvp, &diff);
359 	timersub(record_tvp, &diff, &now);
360 
361 	return (now.tv_sec > 0 || (now.tv_sec == 0 && now.tv_usec > 0));
362 }
363 
364 void
365 cleanup(signo)
366 	int signo;
367 {
368 
369 	rewrite_header();
370 	close(outfd);
371 	if (omonitor_gain) {
372 		AUDIO_INITINFO(&info);
373 		info.monitor_gain = omonitor_gain;
374 		if (ioctl(audiofd, AUDIO_SETINFO, &info) < 0)
375 			err(1, "failed to reset audio info");
376 	}
377 	close(audiofd);
378 	if (signo != 0) {
379 		(void)raise_default_signal(signo);
380 	}
381 	exit(0);
382 }
383 
384 int
385 write_header_sun(hdrp, lenp, leftp)
386 	void **hdrp;
387 	size_t *lenp;
388 	int *leftp;
389 {
390 	static int warned = 0;
391 	static sun_audioheader auh;
392 	int sunenc, oencoding = encoding;
393 
394 	/* only perform conversions if we don't specify the encoding */
395 	switch (encoding) {
396 	case AUDIO_ENCODING_ULINEAR_LE:
397 #if BYTE_ORDER == LITTLE_ENDIAN
398 	case AUDIO_ENCODING_ULINEAR:
399 #endif
400 		if (precision == 16)
401 			conv_func = change_sign16_swap_bytes_le;
402 		else if (precision == 32)
403 			conv_func = change_sign32_swap_bytes_le;
404 		if (conv_func)
405 			encoding = AUDIO_ENCODING_SLINEAR_BE;
406 		break;
407 
408 	case AUDIO_ENCODING_ULINEAR_BE:
409 #if BYTE_ORDER == BIG_ENDIAN
410 	case AUDIO_ENCODING_ULINEAR:
411 #endif
412 		if (precision == 16)
413 			conv_func = change_sign16_be;
414 		else if (precision == 32)
415 			conv_func = change_sign32_be;
416 		if (conv_func)
417 			encoding = AUDIO_ENCODING_SLINEAR_BE;
418 		break;
419 
420 	case AUDIO_ENCODING_SLINEAR_LE:
421 #if BYTE_ORDER == LITTLE_ENDIAN
422 	case AUDIO_ENCODING_SLINEAR:
423 #endif
424 		if (precision == 16)
425 			conv_func = swap_bytes;
426 		else if (precision == 32)
427 			conv_func = swap_bytes32;
428 		if (conv_func)
429 			encoding = AUDIO_ENCODING_SLINEAR_BE;
430 		break;
431 
432 #if BYTE_ORDER == BIG_ENDIAN
433 	case AUDIO_ENCODING_SLINEAR:
434 		encoding = AUDIO_ENCODING_SLINEAR_BE;
435 		break;
436 #endif
437 	}
438 
439 	/* if we can't express this as a Sun header, don't write any */
440 	if (audio_encoding_to_sun(encoding, precision, &sunenc) != 0) {
441 		if (!qflag && !warned) {
442 			const char *s = audio_enc_from_val(oencoding);
443 
444 			if (s == NULL)
445 				s = "(unknown)";
446 			warnx("failed to convert to sun encoding from %s "
447 			      "(precision %d);\nSun audio header not written",
448 			      s, precision);
449 		}
450 		format = AUDIO_FORMAT_NONE;
451 		conv_func = 0;
452 		warned = 1;
453 		return -1;
454 	}
455 
456 	auh.magic = htonl(AUDIO_FILE_MAGIC);
457 	if (outfd == STDOUT_FILENO)
458 		auh.data_size = htonl(AUDIO_UNKNOWN_SIZE);
459 	else if (total_size != -1)
460 		auh.data_size = htonl(total_size);
461 	else
462 		auh.data_size = 0;
463 	auh.encoding = htonl(sunenc);
464 	auh.sample_rate = htonl(sample_rate);
465 	auh.channels = htonl(channels);
466 	if (header_info) {
467 		int 	len, infolen;
468 
469 		infolen = ((len = strlen(header_info)) + 7) & 0xfffffff8;
470 		*leftp = infolen - len;
471 		auh.hdr_size = htonl(sizeof(auh) + infolen);
472 	} else {
473 		*leftp = sizeof(default_info);
474 		auh.hdr_size = htonl(sizeof(auh) + *leftp);
475 	}
476 	*(sun_audioheader **)hdrp = &auh;
477 	*lenp = sizeof auh;
478 	return 0;
479 }
480 
481 int
482 write_header_wav(hdrp, lenp, leftp)
483 	void **hdrp;
484 	size_t *lenp;
485 	int *leftp;
486 {
487 	/*
488 	 * WAV header we write looks like this:
489 	 *
490 	 *      bytes   purpose
491 	 *      0-3     "RIFF"
492 	 *      4-7     file length (minus 8)
493 	 *      8-15    "WAVEfmt "
494 	 *      16-19   format size
495 	 *      20-21   format tag
496 	 *      22-23   number of channels
497 	 *      24-27   sample rate
498 	 *      28-31   average bytes per second
499 	 *      32-33   block alignment
500 	 *      34-35   bits per sample
501 	 *
502 	 * then for ULAW and ALAW outputs, we have an extended chunk size
503 	 * and a WAV "fact" to add:
504 	 *
505 	 *      36-37   length of extension (== 0)
506 	 *      38-41   "fact"
507 	 *      42-45   fact size
508 	 *      46-49   number of samples written
509 	 *      50-53   "data"
510 	 *      54-57   data length
511 	 *      58-     raw audio data
512 	 *
513 	 * for PCM outputs we have just the data remaining:
514 	 *
515 	 *      36-39   "data"
516 	 *      40-43   data length
517 	 *      44-     raw audio data
518 	 *
519 	 *	RIFF\^@^C^@WAVEfmt ^P^@^@^@^A^@^B^@D<AC>^@^@^P<B1>^B^@^D^@^P^@data^@^@^C^@^@^@^@^@^@^@^@^@^@
520 	 */
521 	char	wavheaderbuf[64], *p = wavheaderbuf;
522 	const char *riff = "RIFF",
523 	    *wavefmt = "WAVEfmt ",
524 	    *fact = "fact",
525 	    *data = "data";
526 	u_int32_t filelen, fmtsz, sps, abps, factsz = 4, nsample, datalen;
527 	u_int16_t fmttag, nchan, align, bps, extln = 0;
528 
529 	if (header_info)
530 		warnx("header information not supported for WAV");
531 	*leftp = 0;
532 
533 	switch (precision) {
534 	case 8:
535 		bps = 8;
536 		break;
537 	case 16:
538 		bps = 16;
539 		break;
540 	case 32:
541 		bps = 32;
542 		break;
543 	default:
544 		{
545 			static int warned = 0;
546 
547 			if (warned == 0) {
548 				warnx("can not support precision of %d", precision);
549 				warned = 1;
550 			}
551 		}
552 		return (-1);
553 	}
554 
555 	switch (encoding) {
556 	case AUDIO_ENCODING_ULAW:
557 		fmttag = WAVE_FORMAT_MULAW;
558 		fmtsz = 18;
559 		align = channels;
560 		break;
561 
562 	case AUDIO_ENCODING_ALAW:
563 		fmttag = WAVE_FORMAT_ALAW;
564 		fmtsz = 18;
565 		align = channels;
566 		break;
567 
568 	/*
569 	 * we could try to support RIFX but it seems to be more portable
570 	 * to output little-endian data for WAV files.
571 	 */
572 	case AUDIO_ENCODING_ULINEAR_BE:
573 #if BYTE_ORDER == BIG_ENDIAN
574 	case AUDIO_ENCODING_ULINEAR:
575 #endif
576 		if (bps == 16)
577 			conv_func = change_sign16_swap_bytes_be;
578 		else if (bps == 32)
579 			conv_func = change_sign32_swap_bytes_be;
580 		goto fmt_pcm;
581 
582 	case AUDIO_ENCODING_SLINEAR_BE:
583 #if BYTE_ORDER == BIG_ENDIAN
584 	case AUDIO_ENCODING_SLINEAR:
585 #endif
586 		if (bps == 8)
587 			conv_func = change_sign8;
588 		else if (bps == 16)
589 			conv_func = swap_bytes;
590 		else if (bps == 32)
591 			conv_func = swap_bytes32;
592 		goto fmt_pcm;
593 
594 	case AUDIO_ENCODING_ULINEAR_LE:
595 #if BYTE_ORDER == LITTLE_ENDIAN
596 	case AUDIO_ENCODING_ULINEAR:
597 #endif
598 		if (bps == 16)
599 			conv_func = change_sign16_le;
600 		else if (bps == 32)
601 			conv_func = change_sign32_le;
602 		/* FALLTHROUGH */
603 
604 	case AUDIO_ENCODING_SLINEAR_LE:
605 	case AUDIO_ENCODING_PCM16:
606 #if BYTE_ORDER == LITTLE_ENDIAN
607 	case AUDIO_ENCODING_SLINEAR:
608 #endif
609 		if (bps == 8)
610 			conv_func = change_sign8;
611 fmt_pcm:
612 		fmttag = WAVE_FORMAT_PCM;
613 		fmtsz = 16;
614 		align = channels * (bps / 8);
615 		break;
616 
617 	default:
618 		{
619 			static int warned = 0;
620 
621 			if (warned == 0) {
622 				const char *s = wav_enc_from_val(encoding);
623 
624 				if (s == NULL)
625 					warnx("can not support encoding of %s", s);
626 				else
627 					warnx("can not support encoding of %d", encoding);
628 				warned = 1;
629 			}
630 		}
631 		format = AUDIO_FORMAT_NONE;
632 		return (-1);
633 	}
634 
635 	nchan = channels;
636 	sps = sample_rate;
637 
638 	/* data length */
639 	if (outfd == STDOUT_FILENO)
640 		datalen = 0;
641 	else if (total_size != -1)
642 		datalen = total_size;
643 	else
644 		datalen = 0;
645 
646 	/* file length */
647 	filelen = 4 + (8 + fmtsz) + (8 + datalen);
648 	if (fmttag != WAVE_FORMAT_PCM)
649 		filelen += 8 + factsz;
650 
651 	abps = (double)align*sample_rate / (double)1 + 0.5;
652 
653 	nsample = (datalen / bps) / sample_rate;
654 
655 	/*
656 	 * now we've calculated the info, write it out!
657 	 */
658 #define put32(x) do { \
659 	u_int32_t _f; \
660 	putle32(_f, (x)); \
661 	memcpy(p, &_f, 4); \
662 } while (0)
663 #define put16(x) do { \
664 	u_int16_t _f; \
665 	putle16(_f, (x)); \
666 	memcpy(p, &_f, 2); \
667 } while (0)
668 	memcpy(p, riff, 4);
669 	p += 4;				/* 4 */
670 	put32(filelen);
671 	p += 4;				/* 8 */
672 	memcpy(p, wavefmt, 8);
673 	p += 8;				/* 16 */
674 	put32(fmtsz);
675 	p += 4;				/* 20 */
676 	put16(fmttag);
677 	p += 2;				/* 22 */
678 	put16(nchan);
679 	p += 2;				/* 24 */
680 	put32(sps);
681 	p += 4;				/* 28 */
682 	put32(abps);
683 	p += 4;				/* 32 */
684 	put16(align);
685 	p += 2;				/* 34 */
686 	put16(bps);
687 	p += 2;				/* 36 */
688 	/* NON PCM formats have an extended chunk; write it */
689 	if (fmttag != WAVE_FORMAT_PCM) {
690 		put16(extln);
691 		p += 2;			/* 38 */
692 		memcpy(p, fact, 4);
693 		p += 4;			/* 42 */
694 		put32(factsz);
695 		p += 4;			/* 46 */
696 		put32(nsample);
697 		p += 4;			/* 50 */
698 	}
699 	memcpy(p, data, 4);
700 	p += 4;				/* 40/54 */
701 	put32(datalen);
702 	p += 4;				/* 44/58 */
703 #undef put32
704 #undef put16
705 
706 	*hdrp = wavheaderbuf;
707 	*lenp = (p - wavheaderbuf);
708 
709 	return 0;
710 }
711 
712 void
713 write_header()
714 {
715 	struct iovec iv[3];
716 	int veclen, left, tlen;
717 	void *hdr;
718 	size_t hdrlen;
719 
720 	switch (format) {
721 	case AUDIO_FORMAT_DEFAULT:
722 	case AUDIO_FORMAT_SUN:
723 		if (write_header_sun(&hdr, &hdrlen, &left) != 0)
724 			return;
725 		break;
726 	case AUDIO_FORMAT_WAV:
727 		if (write_header_wav(&hdr, &hdrlen, &left) != 0)
728 			return;
729 		break;
730 	case AUDIO_FORMAT_NONE:
731 		return;
732 	default:
733 		errx(1, "unknown audio format");
734 	}
735 
736 	veclen = 0;
737 	tlen = 0;
738 
739 	if (hdrlen != 0) {
740 		iv[veclen].iov_base = hdr;
741 		iv[veclen].iov_len = hdrlen;
742 		tlen += iv[veclen++].iov_len;
743 	}
744 	if (header_info) {
745 		iv[veclen].iov_base = header_info;
746 		iv[veclen].iov_len = (int)strlen(header_info) + 1;
747 		tlen += iv[veclen++].iov_len;
748 	}
749 	if (left) {
750 		iv[veclen].iov_base = default_info;
751 		iv[veclen].iov_len = left;
752 		tlen += iv[veclen++].iov_len;
753 	}
754 
755 	if (tlen == 0)
756 		return;
757 
758 	if (writev(outfd, iv, veclen) != tlen)
759 		err(1, "could not write audio header");
760 }
761 
762 void
763 rewrite_header()
764 {
765 
766 	/* can't do this here! */
767 	if (outfd == STDOUT_FILENO)
768 		return;
769 
770 	if (lseek(outfd, SEEK_SET, 0) < 0)
771 		err(1, "could not seek to start of file for header rewrite");
772 	write_header();
773 }
774 
775 void
776 usage()
777 {
778 
779 	fprintf(stderr, "Usage: %s [-afhqV] [options] {files ...|-}\n",
780 	    getprogname());
781 	fprintf(stderr, "Options:\n\t"
782 	    "-B buffer size\n\t"
783 	    "-b balance (0-63)\n\t"
784 	    "-c channels\n\t"
785 	    "-d audio device\n\t"
786 	    "-e encoding\n\t"
787 	    "-F format\n\t"
788 	    "-i header information\n\t"
789 	    "-m monitor volume\n\t"
790 	    "-P precision (4, 8, 16, 24, or 32 bits)\n\t"
791 	    "-p input port\n\t"
792 	    "-s sample rate\n\t"
793 	    "-t recording time\n\t"
794 	    "-v volume\n");
795 	exit(EXIT_FAILURE);
796 }
797