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NetBSD/usr.bin/audio/record/record.c

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/* $NetBSD: record.c,v 1.43 2006/05/11 01:19:10 mrg Exp $ */
/*
* Copyright (c) 1999, 2002 Matthew R. Green
* All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
* 3. The name of the author may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
* IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
* OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
* IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
* INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING,
* BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED
* AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
* OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
* SUCH DAMAGE.
*/
/*
* SunOS compatible audiorecord(1)
*/
#include <sys/cdefs.h>
#ifndef lint
__RCSID("$NetBSD: record.c,v 1.43 2006/05/11 01:19:10 mrg Exp $");
#endif
#include <sys/types.h>
#include <sys/audioio.h>
#include <sys/ioctl.h>
#include <sys/time.h>
#include <sys/uio.h>
#include <err.h>
#include <fcntl.h>
#include <paths.h>
#include <signal.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <unistd.h>
#include "libaudio.h"
#include "auconv.h"
audio_info_t info, oinfo;
ssize_t total_size = -1;
const char *device;
int format = AUDIO_FORMAT_DEFAULT;
char *header_info;
char default_info[8] = { '\0', '\0', '\0', '\0', '\0', '\0', '\0', '\0' };
int audiofd, outfd;
int qflag, aflag, fflag;
int verbose;
int monitor_gain, omonitor_gain;
int gain;
int balance;
int port;
int encoding;
char *encoding_str;
int precision;
int sample_rate;
int channels;
struct timeval record_time;
struct timeval start_time;
void (*conv_func) (u_char *, int);
void usage (void);
int main (int, char *[]);
int timeleft (struct timeval *, struct timeval *);
void cleanup (int) __attribute__((__noreturn__));
int write_header_sun (void **, size_t *, int *);
int write_header_wav (void **, size_t *, int *);
void write_header (void);
void rewrite_header (void);
int
main(argc, argv)
int argc;
char *argv[];
{
u_char *buffer;
size_t len, bufsize;
int ch, no_time_limit = 1;
const char *defdevice = _PATH_SOUND;
while ((ch = getopt(argc, argv, "ab:C:F:c:d:e:fhi:m:P:p:qt:s:Vv:")) != -1) {
switch (ch) {
case 'a':
aflag++;
break;
case 'b':
decode_int(optarg, &balance);
if (balance < 0 || balance > 63)
errx(1, "balance must be between 0 and 63");
break;
case 'C':
/* Ignore, compatibility */
break;
case 'F':
format = audio_format_from_str(optarg);
if (format < 0)
errx(1, "Unknown audio format; supported "
"formats: \"sun\", \"wav\", and \"none\"");
break;
case 'c':
decode_int(optarg, &channels);
if (channels < 0 || channels > 16)
errx(1, "channels must be between 0 and 16");
break;
case 'd':
device = optarg;
break;
case 'e':
encoding_str = optarg;
break;
case 'f':
fflag++;
break;
case 'i':
header_info = optarg;
break;
case 'm':
decode_int(optarg, &monitor_gain);
if (monitor_gain < 0 || monitor_gain > 255)
errx(1, "monitor volume must be between 0 and 255");
break;
case 'P':
decode_int(optarg, &precision);
if (precision != 4 && precision != 8 &&
precision != 16 && precision != 24 &&
precision != 32)
errx(1, "precision must be between 4, 8, 16, 24 or 32");
break;
case 'p':
len = strlen(optarg);
if (strncmp(optarg, "mic", len) == 0)
port |= AUDIO_MICROPHONE;
else if (strncmp(optarg, "cd", len) == 0 ||
strncmp(optarg, "internal-cd", len) == 0)
port |= AUDIO_CD;
else if (strncmp(optarg, "line", len) == 0)
port |= AUDIO_LINE_IN;
else
errx(1,
"port must be `cd', `internal-cd', `mic', or `line'");
break;
case 'q':
qflag++;
break;
case 's':
decode_int(optarg, &sample_rate);
if (sample_rate < 0 || sample_rate > 48000 * 2) /* XXX */
errx(1, "sample rate must be between 0 and 96000");
break;
case 't':
no_time_limit = 0;
decode_time(optarg, &record_time);
break;
case 'V':
verbose++;
break;
case 'v':
decode_int(optarg, &gain);
if (gain < 0 || gain > 255)
errx(1, "volume must be between 0 and 255");
break;
/* case 'h': */
default:
usage();
/* NOTREACHED */
}
}
argc -= optind;
argv += optind;
if (argc != 1)
usage();
/*
* convert the encoding string into a value.
*/
if (encoding_str) {
encoding = audio_enc_to_val(encoding_str);
if (encoding == -1)
errx(1, "unknown encoding, bailing...");
}
#if 0
else
encoding = AUDIO_ENCODING_ULAW;
#endif
/*
* open the output file
*/
if (argv[0][0] != '-' || argv[0][1] != '\0') {
/* intuit the file type from the name */
if (format == AUDIO_FORMAT_DEFAULT)
{
size_t flen = strlen(*argv);
const char *arg = *argv;
if (strcasecmp(arg + flen - 3, ".au") == 0)
format = AUDIO_FORMAT_SUN;
else if (strcasecmp(arg + flen - 4, ".wav") == 0)
format = AUDIO_FORMAT_WAV;
}
outfd = open(*argv, O_CREAT|(aflag ? O_APPEND : O_TRUNC)|O_WRONLY, 0666);
if (outfd < 0)
err(1, "could not open %s", *argv);
} else
outfd = STDOUT_FILENO;
/*
* open the audio device
*/
if (device == NULL && (device = getenv("AUDIODEVICE")) == NULL &&
(device = getenv("AUDIODEV")) == NULL) /* Sun compatibility */
device = defdevice;
audiofd = open(device, O_RDONLY);
if (audiofd < 0 && device == defdevice) {
device = _PATH_SOUND0;
audiofd = open(device, O_RDONLY);
}
if (audiofd < 0)
err(1, "failed to open %s", device);
/*
* work out the buffer size to use, and allocate it. also work out
* what the old monitor gain value is, so that we can reset it later.
*/
if (ioctl(audiofd, AUDIO_GETINFO, &oinfo) < 0)
err(1, "failed to get audio info");
bufsize = oinfo.record.buffer_size;
if (bufsize < 32 * 1024)
bufsize = 32 * 1024;
omonitor_gain = oinfo.monitor_gain;
buffer = malloc(bufsize);
if (buffer == NULL)
err(1, "couldn't malloc buffer of %d size", (int)bufsize);
/*
* set up audio device for recording with the speified parameters
*/
AUDIO_INITINFO(&info);
/*
* for these, get the current values for stuffing into the header
*/
#define SETINFO(x) if (x) \
info.record.x = x; \
else \
info.record.x = x = oinfo.record.x;
SETINFO (sample_rate)
SETINFO (channels)
SETINFO (precision)
SETINFO (encoding)
SETINFO (gain)
SETINFO (port)
SETINFO (balance)
#undef SETINFO
if (monitor_gain)
info.monitor_gain = monitor_gain;
else
monitor_gain = oinfo.monitor_gain;
info.mode = AUMODE_RECORD;
if (ioctl(audiofd, AUDIO_SETINFO, &info) < 0)
err(1, "failed to set audio info");
signal(SIGINT, cleanup);
write_header();
total_size = 0;
if (verbose && conv_func) {
const char *s = NULL;
if (conv_func == swap_bytes)
s = "swap bytes (16 bit)";
else if (conv_func == swap_bytes32)
s = "swap bytes (32 bit)";
else if (conv_func == change_sign16_be)
s = "change sign (big-endian, 16 bit)";
else if (conv_func == change_sign16_le)
s = "change sign (little-endian, 16 bit)";
else if (conv_func == change_sign32_be)
s = "change sign (big-endian, 32 bit)";
else if (conv_func == change_sign32_le)
s = "change sign (little-endian, 32 bit)";
else if (conv_func == change_sign16_swap_bytes_be)
s = "change sign & swap bytes (big-endian, 16 bit)";
else if (conv_func == change_sign16_swap_bytes_le)
s = "change sign & swap bytes (little-endian, 16 bit)";
else if (conv_func == change_sign32_swap_bytes_be)
s = "change sign (big-endian, 32 bit)";
else if (conv_func == change_sign32_swap_bytes_le)
s = "change sign & swap bytes (little-endian, 32 bit)";
if (s)
fprintf(stderr, "%s: converting, using function: %s\n",
getprogname(), s);
else
fprintf(stderr, "%s: using unnamed conversion "
"function\n", getprogname());
}
if (verbose)
fprintf(stderr,
"sample_rate=%d channels=%d precision=%d encoding=%s\n",
info.record.sample_rate, info.record.channels,
info.record.precision,
audio_enc_from_val(info.record.encoding));
if (!no_time_limit && verbose)
fprintf(stderr, "recording for %lu seconds, %lu microseconds\n",
(u_long)record_time.tv_sec, (u_long)record_time.tv_usec);
(void)gettimeofday(&start_time, NULL);
while (no_time_limit || timeleft(&start_time, &record_time)) {
if (read(audiofd, buffer, bufsize) != bufsize)
err(1, "read failed");
if (conv_func)
(*conv_func)(buffer, bufsize);
if (write(outfd, buffer, bufsize) != bufsize)
err(1, "write failed");
total_size += bufsize;
}
cleanup(0);
}
int
timeleft(start_tvp, record_tvp)
struct timeval *start_tvp;
struct timeval *record_tvp;
{
struct timeval now, diff;
(void)gettimeofday(&now, NULL);
timersub(&now, start_tvp, &diff);
timersub(record_tvp, &diff, &now);
return (now.tv_sec > 0 || (now.tv_sec == 0 && now.tv_usec > 0));
}
void
cleanup(signo)
int signo;
{
rewrite_header();
close(outfd);
if (omonitor_gain) {
AUDIO_INITINFO(&info);
info.monitor_gain = omonitor_gain;
if (ioctl(audiofd, AUDIO_SETINFO, &info) < 0)
err(1, "failed to reset audio info");
}
close(audiofd);
exit(0);
}
int
write_header_sun(hdrp, lenp, leftp)
void **hdrp;
size_t *lenp;
int *leftp;
{
static int warned = 0;
static sun_audioheader auh;
int sunenc, oencoding = encoding;
/* only perform conversions if we don't specify the encoding */
switch (encoding) {
case AUDIO_ENCODING_ULINEAR_LE:
#if BYTE_ORDER == LITTLE_ENDIAN
case AUDIO_ENCODING_ULINEAR:
#endif
if (precision == 16)
conv_func = change_sign16_swap_bytes_le;
else if (precision == 32)
conv_func = change_sign32_swap_bytes_le;
if (conv_func)
encoding = AUDIO_ENCODING_SLINEAR_BE;
break;
case AUDIO_ENCODING_ULINEAR_BE:
#if BYTE_ORDER == BIG_ENDIAN
case AUDIO_ENCODING_ULINEAR:
#endif
if (precision == 16)
conv_func = change_sign16_be;
else if (precision == 32)
conv_func = change_sign32_be;
if (conv_func)
encoding = AUDIO_ENCODING_SLINEAR_BE;
break;
case AUDIO_ENCODING_SLINEAR_LE:
#if BYTE_ORDER == LITTLE_ENDIAN
case AUDIO_ENCODING_SLINEAR:
#endif
if (precision == 16)
conv_func = swap_bytes;
else if (precision == 32)
conv_func = swap_bytes32;
if (conv_func)
encoding = AUDIO_ENCODING_SLINEAR_BE;
break;
#if BYTE_ORDER == BIG_ENDIAN
case AUDIO_ENCODING_SLINEAR:
encoding = AUDIO_ENCODING_SLINEAR_BE;
break;
#endif
}
/* if we can't express this as a Sun header, don't write any */
if (audio_encoding_to_sun(encoding, precision, &sunenc) != 0) {
if (!qflag && !warned) {
const char *s = audio_enc_from_val(oencoding);
if (s == NULL)
s = "(unknown)";
warnx("failed to convert to sun encoding from %s "
"(precision %d);\nSun audio header not written",
s, precision);
}
format = AUDIO_FORMAT_NONE;
conv_func = 0;
warned = 1;
return -1;
}
auh.magic = htonl(AUDIO_FILE_MAGIC);
if (outfd == STDOUT_FILENO)
auh.data_size = htonl(AUDIO_UNKNOWN_SIZE);
else if (total_size != -1)
auh.data_size = htonl(total_size);
else
auh.data_size = 0;
auh.encoding = htonl(sunenc);
auh.sample_rate = htonl(sample_rate);
auh.channels = htonl(channels);
if (header_info) {
int len, infolen;
infolen = ((len = strlen(header_info)) + 7) & 0xfffffff8;
*leftp = infolen - len;
auh.hdr_size = htonl(sizeof(auh) + infolen);
} else {
*leftp = sizeof(default_info);
auh.hdr_size = htonl(sizeof(auh) + *leftp);
}
*(sun_audioheader **)hdrp = &auh;
*lenp = sizeof auh;
return 0;
}
int
write_header_wav(hdrp, lenp, leftp)
void **hdrp;
size_t *lenp;
int *leftp;
{
/*
* WAV header we write looks like this:
*
* bytes purpose
* 0-3 "RIFF"
* 4-7 file length (minus 8)
* 8-15 "WAVEfmt "
* 16-19 format size
* 20-21 format tag
* 22-23 number of channels
* 24-27 sample rate
* 28-31 average bytes per second
* 32-33 block alignment
* 34-35 bits per sample
*
* then for ULAW and ALAW outputs, we have an extended chunk size
* and a WAV "fact" to add:
*
* 36-37 length of extension (== 0)
* 38-41 "fact"
* 42-45 fact size
* 46-49 number of samples written
* 50-53 "data"
* 54-57 data length
* 58- raw audio data
*
* for PCM outputs we have just the data remaining:
*
* 36-39 "data"
* 40-43 data length
* 44- raw audio data
*
* RIFF\^@^C^@WAVEfmt ^P^@^@^@^A^@^B^@D<AC>^@^@^P<B1>^B^@^D^@^P^@data^@^@^C^@^@^@^@^@^@^@^@^@^@
*/
char wavheaderbuf[64], *p = wavheaderbuf;
const char *riff = "RIFF",
*wavefmt = "WAVEfmt ",
*fact = "fact",
*data = "data";
u_int32_t filelen, fmtsz, sps, abps, factsz = 4, nsample, datalen;
u_int16_t fmttag, nchan, align, bps, extln = 0;
if (header_info)
warnx("header information not supported for WAV");
*leftp = 0;
switch (precision) {
case 8:
bps = 8;
break;
case 16:
bps = 16;
break;
case 32:
bps = 32;
break;
default:
{
static int warned = 0;
if (warned == 0) {
warnx("can not support precision of %d", precision);
warned = 1;
}
}
return (-1);
}
switch (encoding) {
case AUDIO_ENCODING_ULAW:
fmttag = WAVE_FORMAT_MULAW;
fmtsz = 18;
align = channels;
break;
case AUDIO_ENCODING_ALAW:
fmttag = WAVE_FORMAT_ALAW;
fmtsz = 18;
align = channels;
break;
/*
* we could try to support RIFX but it seems to be more portable
* to output little-endian data for WAV files.
*/
case AUDIO_ENCODING_ULINEAR_BE:
#if BYTE_ORDER == BIG_ENDIAN
case AUDIO_ENCODING_ULINEAR:
#endif
if (bps == 16)
conv_func = change_sign16_swap_bytes_be;
else if (bps == 32)
conv_func = change_sign32_swap_bytes_be;
goto fmt_pcm;
case AUDIO_ENCODING_SLINEAR_BE:
#if BYTE_ORDER == BIG_ENDIAN
case AUDIO_ENCODING_SLINEAR:
#endif
if (bps == 8)
conv_func = change_sign8;
else if (bps == 16)
conv_func = swap_bytes;
else if (bps == 32)
conv_func = swap_bytes32;
goto fmt_pcm;
case AUDIO_ENCODING_ULINEAR_LE:
#if BYTE_ORDER == LITTLE_ENDIAN
case AUDIO_ENCODING_ULINEAR:
#endif
if (bps == 16)
conv_func = change_sign16_le;
else if (bps == 32)
conv_func = change_sign32_le;
/* FALLTHROUGH */
case AUDIO_ENCODING_SLINEAR_LE:
case AUDIO_ENCODING_PCM16:
#if BYTE_ORDER == LITTLE_ENDIAN
case AUDIO_ENCODING_SLINEAR:
#endif
if (bps == 8)
conv_func = change_sign8;
fmt_pcm:
fmttag = WAVE_FORMAT_PCM;
fmtsz = 16;
align = channels * (bps / 8);
break;
default:
{
static int warned = 0;
if (warned == 0) {
const char *s = wav_enc_from_val(encoding);
if (s == NULL)
warnx("can not support encoding of %s", s);
else
warnx("can not support encoding of %d", encoding);
warned = 1;
}
}
format = AUDIO_FORMAT_NONE;
return (-1);
}
nchan = channels;
sps = sample_rate;
/* data length */
if (outfd == STDOUT_FILENO)
datalen = 0;
else if (total_size != -1)
datalen = total_size;
else
datalen = 0;
/* file length */
filelen = 4 + (8 + fmtsz) + (8 + datalen);
if (fmttag != WAVE_FORMAT_PCM)
filelen += 8 + factsz;
abps = (double)align*sample_rate / (double)1 + 0.5;
nsample = (datalen / bps) / sample_rate;
/*
* now we've calculated the info, write it out!
*/
#define put32(x) do { \
u_int32_t _f; \
putle32(_f, (x)); \
memcpy(p, &_f, 4); \
} while (0)
#define put16(x) do { \
u_int16_t _f; \
putle16(_f, (x)); \
memcpy(p, &_f, 2); \
} while (0)
memcpy(p, riff, 4);
p += 4; /* 4 */
put32(filelen);
p += 4; /* 8 */
memcpy(p, wavefmt, 8);
p += 8; /* 16 */
put32(fmtsz);
p += 4; /* 20 */
put16(fmttag);
p += 2; /* 22 */
put16(nchan);
p += 2; /* 24 */
put32(sps);
p += 4; /* 28 */
put32(abps);
p += 4; /* 32 */
put16(align);
p += 2; /* 34 */
put16(bps);
p += 2; /* 36 */
/* NON PCM formats have an extended chunk; write it */
if (fmttag != WAVE_FORMAT_PCM) {
put16(extln);
p += 2; /* 38 */
memcpy(p, fact, 4);
p += 4; /* 42 */
put32(factsz);
p += 4; /* 46 */
put32(nsample);
p += 4; /* 50 */
}
memcpy(p, data, 4);
p += 4; /* 40/54 */
put32(datalen);
p += 4; /* 44/58 */
#undef put32
#undef put16
*hdrp = wavheaderbuf;
*lenp = (p - wavheaderbuf);
return 0;
}
void
write_header()
{
struct iovec iv[3];
int veclen, left, tlen;
void *hdr;
size_t hdrlen;
switch (format) {
case AUDIO_FORMAT_DEFAULT:
case AUDIO_FORMAT_SUN:
if (write_header_sun(&hdr, &hdrlen, &left) != 0)
return;
break;
case AUDIO_FORMAT_WAV:
if (write_header_wav(&hdr, &hdrlen, &left) != 0)
return;
break;
case AUDIO_FORMAT_NONE:
return;
default:
errx(1, "unknown audio format");
}
veclen = 0;
tlen = 0;
if (hdrlen != 0) {
iv[veclen].iov_base = hdr;
iv[veclen].iov_len = hdrlen;
tlen += iv[veclen++].iov_len;
}
if (header_info) {
iv[veclen].iov_base = header_info;
iv[veclen].iov_len = (int)strlen(header_info) + 1;
tlen += iv[veclen++].iov_len;
}
if (left) {
iv[veclen].iov_base = default_info;
iv[veclen].iov_len = left;
tlen += iv[veclen++].iov_len;
}
if (tlen == 0)
return;
if (writev(outfd, iv, veclen) != tlen)
err(1, "could not write audio header");
}
void
rewrite_header()
{
/* can't do this here! */
if (outfd == STDOUT_FILENO)
return;
if (lseek(outfd, SEEK_SET, 0) < 0)
err(1, "could not seek to start of file for header rewrite");
write_header();
}
void
usage()
{
fprintf(stderr, "Usage: %s [-afhqV] [options] {files ...|-}\n",
getprogname());
fprintf(stderr, "Options:\n\t"
"-b balance (0-63)\n\t"
"-c channels\n\t"
"-d audio device\n\t"
"-e encoding\n\t"
"-F format\n\t"
"-i header information\n\t"
"-m monitor volume\n\t"
"-P precision (4, 8, 16, 24, or 32 bits)\n\t"
"-p input port\n\t"
"-s sample rate\n\t"
"-t recording time\n\t"
"-v volume\n");
exit(EXIT_FAILURE);
}
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