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/*
vplay.c - plays and records
CREATIVE LABS VOICE-files, Microsoft WAVE-files and raw data
Autor: Michael Beck - beck@informatik.hu-berlin.de
*/
#include <stdio.h>
#include <malloc.h>
#include <unistd.h>
#include <stdlib.h>
#include <string.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/soundcard.h>
#include "fmtheaders.h"
#include <netinet/in.h>
#define DEFAULT_DSP_SPEED 8000
#define RECORD 0
#define PLAY 1
#define AUDIO "/dev/dsp"
#define min(a,b) ((a) <= (b) ? (a) : (b))
#define d_printf(x) if (verbose_mode) fprintf x
#define VOC_FMT 0
#define WAVE_FMT 1
#define RAW_DATA 2
#define SND_FMT 3
/* global data */
int timelimit = 0, dsp_speed = DEFAULT_DSP_SPEED, dsp_stereo = 0;
int samplesize = 8;
int quiet_mode = 0;
int verbose_mode = 0;
int convert = 0;
int record_type = VOC_FMT;
u_long count;
int audio, abuf_size, zbuf_size;
int direction, omode;
u_char *audiobuf, *zerobuf;
char *command;
int vocminor, vocmajor; /* .VOC version */
/* defaults for playing raw data */
struct
{
int timelimit, dsp_speed, dsp_stereo, samplesize;
}
raw_info =
{
0, DEFAULT_DSP_SPEED, 0, 8};
/* needed prototypes */
void record_play (char *name);
void start_voc (int fd, u_long count);
void start_wave (int fd, u_long count);
void start_snd (int fd, u_long count);
void end_voc (int fd);
void end_wave_raw (int fd);
void end_snd (int fd);
u_long calc_count ();
struct fmt_record
{
void (*start) (int fd, u_long count);
void (*end) (int fd);
char *what;
}
fmt_rec_table[] =
{
{
start_voc, end_voc, "VOC"}
,
{
start_wave, end_wave_raw, "WAVE"}
,
{
NULL, end_wave_raw, "raw data"}
,
{
start_snd, end_snd, "SND"}
};
int
main (int argc, char *argv[])
{
int c, n = 0;
command = argv[0];
if (strstr (argv[0], "vrec"))
{
direction = RECORD;
omode = O_RDONLY;
}
else if (strstr (argv[0], "vplay"))
{
direction = PLAY;
omode = O_WRONLY;
}
else
{
fprintf (stderr,
"Error: command should be named either vrec or vplay\n");
exit (1);
}
while ((c = getopt (argc, argv, "aqs:St:b:vrwd")) != EOF)
switch (c)
{
case 'S':
dsp_stereo = raw_info.dsp_stereo = 1;
break;
case 'q':
quiet_mode = 1;
break;
case 'r':
record_type = RAW_DATA;
break;
case 'v':
record_type = VOC_FMT;
break;
case 'w':
record_type = WAVE_FMT;
break;
case 'a':
record_type = SND_FMT;
break;
case 's':
dsp_speed = atoi (optarg);
if (dsp_speed < 300)
dsp_speed *= 1000;
raw_info.dsp_speed = dsp_speed;
break;
case 't':
timelimit = raw_info.timelimit = atoi (optarg);
break;
case 'b':
samplesize = raw_info.samplesize = atoi (optarg);
break;
case 'd':
verbose_mode = 1;
quiet_mode = 0;
break;
default:
fprintf (stderr,
"Usage: %s [-qvwraS] [-t secs] [-s Hz] [-b 8|12|16] [filename]\n",
command);
exit (-1);
}
audio = open (AUDIO, omode, 0);
if (audio == -1)
{
perror (AUDIO);
exit (-1);
}
if (optind > argc - 1)
record_play (NULL);
else
while (optind <= argc - 1)
{
if (n++ > 0)
if (ioctl (audio, SNDCTL_DSP_SYNC, NULL) < 0)
{
perror (AUDIO);
exit (-1);
}
record_play (argv[optind++]);
}
close (audio);
return 0;
}
/*
test, if it is a .VOC file and return >=0 if ok (this is the length of rest)
< 0 if not
*/
int
test_vocfile (void *buffer)
{
VocHeader *vp = buffer;
if (!strcmp ((const char *) vp->magic, MAGIC_STRING))
{
vocminor = vp->version & 0xFF;
vocmajor = vp->version / 256;
if (vp->version != (0x1233 - vp->coded_ver))
return -2; /* coded version mismatch */
return vp->headerlen - sizeof (VocHeader); /* 0 mostly */
}
return -1; /* magic string fail */
}
/*
test, if it's a .WAV file, 0 if ok (and set the speed, stereo etc.)
< 0 if not
*/
int
test_wavefile (void *buffer)
{
WaveHeader *wp = buffer;
if (wp->main_chunk == RIFF && wp->chunk_type == WAVE &&
wp->sub_chunk == FMT && wp->data_chunk == DATA)
{
if (wp->format != PCM_CODE)
{
fprintf (stderr, "%s: can't play not PCM-coded WAVE-files\n",
command);
exit (-1);
}
if (wp->modus > 2)
{
fprintf (stderr, "%s: can't play WAVE-files with %d tracks\n",
command, wp->modus);
exit (-1);
}
dsp_stereo = (wp->modus == WAVE_STEREO) ? 1 : 0;
samplesize = wp->bit_p_spl;
dsp_speed = wp->sample_fq;
count = wp->data_length;
return 0;
}
return -1;
}
/*
* test, if it's a .SND file, 0 if ok (and set the speed, stereo etc.)
* < 0 if not
*/
int
test_sndfile (void *buffer, int fd)
{
long infolen;
char *info;
SndHeader *snd = buffer;
if (snd->magic == SND_MAGIC)
convert = 0;
else
{
if (htonl (snd->magic) == SND_MAGIC)
{
convert = 1;
snd->dataLocation = htonl (snd->dataLocation);
snd->dataSize = htonl (snd->dataSize);
snd->dataFormat = htonl (snd->dataFormat);
snd->samplingRate = htonl (snd->samplingRate);
snd->channelCount = htonl (snd->channelCount);
}
else
{
/* No SoundFile */
return (-1);
}
}
switch (snd->dataFormat)
{
case SND_FORMAT_LINEAR_8:
samplesize = 8;
break;
case SND_FORMAT_LINEAR_16:
samplesize = 16;
break;
default:
fprintf (stderr, "%s: Unsupported SND_FORMAT\n", command);
exit (-1);
}
dsp_stereo = (snd->channelCount == 2) ? 1 : 0;
dsp_speed = snd->samplingRate;
count = snd->dataSize;
/* read Info-Strings */
infolen = snd->dataLocation - sizeof (SndHeader);
info = (char *) malloc (infolen);
read (fd, info, infolen);
if (!quiet_mode)
fprintf (stderr, "SoundFile Info: %s\n", info);
free (info);
return 0;
}
/*
writing zeros from the zerobuf to simulate silence,
perhaps it's enough to use a long var instead of zerobuf ?
*/
void
write_zeros (unsigned x)
{
unsigned l;
while (x)
{
l = min (x, zbuf_size);
if (write (audio, (char *) zerobuf, l) != l)
{
perror (AUDIO);
exit (-1);
}
x -= l;
}
}
/* if need a SYNC,
(is needed if we plan to change speed, stereo ... during output)
*/
void
sync_dsp (void)
{
#if 0
if (ioctl (audio, SNDCTL_DSP_SYNC, NULL) < 0)
{
perror (AUDIO);
exit (-1);
}
#endif
}
/* setting the speed for output */
void
set_dsp_speed (int *dsp_speed)
{
if (ioctl (audio, SNDCTL_DSP_SPEED, dsp_speed) < 0)
{
fprintf (stderr, "%s: unable to set audio speed\n", command);
perror (AUDIO);
exit (-1);
}
}
/* if to_mono:
compress 8 bit stereo data 2:1, needed if we want play 'one track';
this is usefull, if you habe SB 1.0 - 2.0 (I have 1.0) and want
hear the sample (in Mono)
if to_8:
compress 16 bit to 8 by using hi-byte; wave-files use signed words,
so we need to convert it in "unsigned" sample (0x80 is now zero)
WARNING: this procedure can't compress 16 bit stereo to 16 bit mono,
because if you have a 16 (or 12) bit card you should have
stereo (or I'm wrong ?? )
*/
u_long
one_channel (u_char * buf, u_long l, char to_mono, char to_8)
{
register u_char *w = buf;
register u_char *w2 = buf;
char ofs = 0;
u_long incr = 0;
u_long c, ret;
if (to_mono)
++incr;
if (to_8)
{
++incr;
++w2;
ofs = 128;
}
ret = c = l >> incr;
incr = incr << 1;
while (c--)
{
*w++ = *w2 + ofs;
w2 += incr;
}
return ret;
}
/*
ok, let's play a .voc file
*/
void
vplay (int fd, int ofs, char *name)
{
int l, real_l;
BlockType *bp;
Voice_data *vd;
Ext_Block *eb;
u_long nextblock, in_buffer;
u_char *data = audiobuf;
char was_extended = 0, output = 0;
u_short *sp, repeat = 0;
u_long silence;
int filepos = 0;
char one_chn = 0;
#define COUNT(x) nextblock -= x; in_buffer -=x ;data += x
/* first SYNC the dsp */
sync_dsp ();
if (!quiet_mode)
fprintf (stderr, "Playing Creative Labs Voice file ...\n");
/* first we waste the rest of header, ugly but we don't need seek */
while (ofs > abuf_size)
{
read (fd, (char *) audiobuf, abuf_size);
ofs -= abuf_size;
}
if (ofs)
read (fd, audiobuf, ofs);
/* .voc files are 8 bit (now) */
samplesize = VOC_SAMPLESIZE;
ioctl (audio, SNDCTL_DSP_SETFMT, &samplesize);
if (samplesize != VOC_SAMPLESIZE)
{
fprintf (stderr, "%s: unable to set 8 bit sample size!\n", command);
exit (-1);
}
/* and there are MONO by default */
dsp_stereo = MODE_MONO;
ioctl (audio, SNDCTL_DSP_STEREO, &dsp_stereo);
in_buffer = nextblock = 0;
while (1)
{
Fill_the_buffer: /* need this for repeat */
if (in_buffer < 32)
{
/* move the rest of buffer to pos 0 and fill the audiobuf up */
if (in_buffer)
memcpy ((char *) audiobuf, data, in_buffer);
data = audiobuf;
if ((l =
read (fd, (char *) audiobuf + in_buffer,
abuf_size - in_buffer)) > 0)
in_buffer += l;
else if (!in_buffer)
{
/* the file is truncated, so simulate 'Terminator'
and reduce the datablock for save landing */
nextblock = audiobuf[0] = 0;
if (l == -1)
{
perror (name);
exit (-1);
}
}
}
while (!nextblock)
{ /* this is a new block */
bp = (BlockType *) data;
COUNT (sizeof (BlockType));
nextblock = DATALEN (bp);
if (output && !quiet_mode)
fprintf (stderr, "\n"); /* write /n after ASCII-out */
output = 0;
switch (bp->type)
{
case 0:
d_printf ((stderr, "Terminator\n"));
return; /* VOC-file stop */
case 1:
vd = (Voice_data *) data;
COUNT (sizeof (Voice_data));
/* we need a SYNC, before we can set new SPEED, STEREO ... */
sync_dsp ();
if (!was_extended)
{
dsp_speed = (int) (vd->tc);
dsp_speed = 1000000 / (256 - dsp_speed);
d_printf ((stderr, "Voice data %d Hz\n", dsp_speed));
if (vd->pack)
{ /* /dev/dsp can't it */
fprintf (stderr, "%s: can't play packed .voc files\n",
command);
return;
}
if (dsp_stereo)
{ /* if we are in Stereo-Mode, switch back */
dsp_stereo = MODE_MONO;
ioctl (audio, SNDCTL_DSP_STEREO, &dsp_stereo);
}
}
else
{ /* there was extended block */
if (one_chn) /* if one Stereo fails, why test another ? */
dsp_stereo = MODE_MONO;
else if (dsp_stereo)
{ /* want Stereo */
/* shit, my MACRO dosn't work here */
#ifdef OSS_VERSION
if (ioctl (audio, SNDCTL_DSP_STEREO, &dsp_stereo) < 0)
{
#else
if (dsp_stereo !=
ioctl (audio, SNDCTL_DSP_STEREO, dsp_stereo))
{
#endif
dsp_stereo = MODE_MONO;
fprintf (stderr,
"%s: can't play in Stereo; playing only one channel\n",
command);
one_chn = 1;
}
}
was_extended = 0;
}
set_dsp_speed (&dsp_speed);
break;
case 2: /* nothing to do, pure data */
d_printf ((stderr, "Voice continuation\n"));
break;
case 3: /* a silence block, no data, only a count */
sp = (u_short *) data;
COUNT (sizeof (u_short));
dsp_speed = (int) (*data);
COUNT (1);
dsp_speed = 1000000 / (256 - dsp_speed);
sync_dsp ();
set_dsp_speed (&dsp_speed);
silence = (((u_long) * sp) * 1000) / dsp_speed;
d_printf ((stderr, "Silence for %ld ms\n", silence));
write_zeros (*sp);
break;
case 4: /* a marker for syncronisation, no effect */
sp = (u_short *) data;
COUNT (sizeof (u_short));
d_printf ((stderr, "Marker %d\n", *sp));
break;
case 5: /* ASCII text, we copy to stderr */
output = 1;
d_printf ((stderr, "ASCII - text :\n"));
break;
case 6: /* repeat marker, says repeatcount */
/* my specs don't say it: maybe this can be recursive, but
I don't think somebody use it */
repeat = *(u_short *) data;
COUNT (sizeof (u_short));
d_printf ((stderr, "Repeat loop %d times\n", repeat));
if (filepos >= 0) /* if < 0, one seek fails, why test another */
if ((filepos = lseek (fd, 0, 1)) < 0)
{
fprintf (stderr,
"%s: can't play loops; %s isn't seekable\n",
command, name);
repeat = 0;
}
else
filepos -= in_buffer; /* set filepos after repeat */
else
repeat = 0;
break;
case 7: /* ok, lets repeat that be rewinding tape */
if (repeat)
{
if (repeat != 0xFFFF)
{
d_printf ((stderr, "Repeat loop %d\n", repeat));
--repeat;
}
else
d_printf ((stderr, "Neverending loop\n"));
lseek (fd, filepos, 0);
in_buffer = 0; /* clear the buffer */
goto Fill_the_buffer;
}
else
d_printf ((stderr, "End repeat loop\n"));
break;
case 8: /* the extension to play Stereo, I have SB 1.0 :-( */
was_extended = 1;
eb = (Ext_Block *) data;
COUNT (sizeof (Ext_Block));
dsp_speed = (int) (eb->tc);
dsp_speed = 256000000L / (65536 - dsp_speed);
dsp_stereo = eb->mode;
if (dsp_stereo == MODE_STEREO)
dsp_speed = dsp_speed >> 1;
if (eb->pack)
{ /* /dev/dsp can't it */
fprintf (stderr, "%s: can't play packed .voc files\n",
command);
return;
}
d_printf ((stderr, "Extended block %s %d Hz\n",
(eb->mode ? "Stereo" : "Mono"), dsp_speed));
break;
default:
fprintf (stderr, "%s: unknown blocktype %d. terminate.\n",
command, bp->type);
return;
} /* switch (bp->type) */
} /* while (! nextblock) */
/* put nextblock data bytes to dsp */
l = min (in_buffer, nextblock);
if (l)
{
if (output && !quiet_mode)
write (2, data, l); /* to stderr */
else
{
real_l = one_chn ? one_channel (data, l, one_chn, 0) : l;
if (write (audio, data, real_l) != real_l)
{
perror (AUDIO);
exit (-1);
}
}
COUNT (l);
}
} /* while(1) */
}
/* that was a big one, perhaps somebody split it :-) */
/* setting the globals for playing raw data */
void
init_raw_data (void)
{
timelimit = raw_info.timelimit;
dsp_speed = raw_info.dsp_speed;
dsp_stereo = raw_info.dsp_stereo;
samplesize = raw_info.samplesize;
}
/* calculate the data count to read from/to dsp */
u_long
calc_count (void)
{
u_long count;
if (!timelimit)
count = 0x7fffffff;
else
{
count = timelimit * dsp_speed;
if (dsp_stereo)
count *= 2;
if (samplesize != 8)
count *= 2;
}
return count;
}
/* write a .VOC-header */
void
start_voc (int fd, u_long cnt)
{
VocHeader vh;
BlockType bt;
Voice_data vd;
Ext_Block eb;
strcpy ((char *) vh.magic, MAGIC_STRING);
vh.headerlen = sizeof (VocHeader);
vh.version = ACTUAL_VERSION;
vh.coded_ver = 0x1233 - ACTUAL_VERSION;
write (fd, &vh, sizeof (VocHeader));
if (dsp_stereo)
{
/* write a extended block */
bt.type = 8;
bt.datalen = 4;
bt.datalen_m = bt.datalen_h = 0;
write (fd, &bt, sizeof (BlockType));
eb.tc = (u_short) (65536 - 256000000L / (dsp_speed << 1));
eb.pack = 0;
eb.mode = 1;
write (fd, &eb, sizeof (Ext_Block));
}
bt.type = 1;
cnt += sizeof (Voice_data); /* Voice_data block follows */
bt.datalen = (u_char) (cnt & 0xFF);
bt.datalen_m = (u_char) ((cnt & 0xFF00) >> 8);
bt.datalen_h = (u_char) ((cnt & 0xFF0000) >> 16);
write (fd, &bt, sizeof (BlockType));
vd.tc = (u_char) (256 - (1000000 / dsp_speed));
vd.pack = 0;
write (fd, &vd, sizeof (Voice_data));
}
/* write a WAVE-header */
void
start_wave (int fd, u_long cnt)
{
WaveHeader wh;
wh.main_chunk = RIFF;
wh.length = cnt + sizeof (WaveHeader) - 8;
wh.chunk_type = WAVE;
wh.sub_chunk = FMT;
wh.sc_len = 16;
wh.format = PCM_CODE;
wh.modus = dsp_stereo ? 2 : 1;
wh.sample_fq = dsp_speed;
wh.byte_p_spl = ((samplesize == 8) ? 1 : 2) * (dsp_stereo ? 2 : 1);
wh.byte_p_sec = dsp_speed * wh.modus * wh.byte_p_spl;
wh.bit_p_spl = samplesize;
wh.data_chunk = DATA;
wh.data_length = cnt;
write (fd, &wh, sizeof (WaveHeader));
}
/* closing .VOC */
void
end_voc (int fd)
{
char dummy = 0; /* Write a Terminator */
write (fd, &dummy, 1);
if (fd != 1)
close (fd);
}
void
end_wave_raw (int fd)
{ /* only close output */
if (fd != 1)
close (fd);
}
void
start_snd (int fd, u_long count)
{
SndHeader snd;
char *sndinfo = "Recorded by vrec\000";
snd.magic = SND_MAGIC;
snd.dataLocation = sizeof (SndHeader) + strlen (sndinfo);
snd.dataSize = count;
switch (samplesize)
{
case 8:
snd.dataFormat = SND_FORMAT_LINEAR_8;
break;
case 16:
snd.dataFormat = SND_FORMAT_LINEAR_16;
break;
default:
fprintf (stderr,
"%d bit: unsupported sample size for NeXt sound file!\n",
samplesize);
exit (-1);
}
snd.samplingRate = dsp_speed;
snd.channelCount = dsp_stereo ? 2 : 1;
write (fd, &snd, sizeof (SndHeader));
write (fd, sndinfo, strlen (sndinfo));
}
void
end_snd (int fd)
{
if (fd != 1)
close (fd);
}
/* playing/recording raw data, this proc handels WAVE files and
recording .VOCs (as one block) */
void
recplay (int fd, int loaded, u_long count, int rtype, char *name)
{
int l, real_l;
u_long c;
char one_chn = 0;
char to_8 = 0;
int tmps;
sync_dsp ();
tmps = samplesize;
ioctl (audio, SNDCTL_DSP_SETFMT, &tmps);
if (tmps != samplesize)
{
fprintf (stderr, "%s: unable to set %d bit sample size",
command, samplesize);
if (samplesize == 16)
{
samplesize = 8;
ioctl (audio, SNDCTL_DSP_SETFMT, &samplesize);
if (samplesize != 8)
{
fprintf (stderr, "%s: unable to set 8 bit sample size!\n",
command);
exit (-1);
}
fprintf (stderr, "; playing 8 bit\n");
to_8 = 1;
}
else
{
fprintf (stderr, "\n");
exit (-1);
}
}
#ifdef OSS_VERSION
if (ioctl (audio, SNDCTL_DSP_STEREO, &dsp_stereo) < 0)
{
#else
if (dsp_stereo != ioctl (audio, SNDCTL_DSP_STEREO, dsp_stereo))
{
#endif
if (direction == PLAY)
{
fprintf (stderr,
"%s: can't play in Stereo; playing only one channel\n",
command);
dsp_stereo = MODE_MONO;
one_chn = 1;
}
else
{
fprintf (stderr, "%s: can't record in Stereo\n", command);
exit (-1);
}
}
set_dsp_speed (&dsp_speed);
if (!quiet_mode)
{
fprintf (stderr, "%s %s : ",
(direction == PLAY) ? "Playing" : "Recording",
fmt_rec_table[rtype].what);
if (samplesize != 8)
fprintf (stderr, "%d bit, ", samplesize);
fprintf (stderr, "Speed %d Hz ", dsp_speed);
fprintf (stderr, "%d bits ", samplesize);
fprintf (stderr, "%s ...\n", dsp_stereo ? "Stereo" : "Mono");
}
abuf_size = -1;
#if 0
/*
* There is no idea in using SNDCTL_DSP_GETBLKSIZE in applications like this.
* Using some fixed local buffer size will work equally well.
*/
ioctl (audio, SNDCTL_DSP_GETBLKSIZE, &abuf_size);
if (abuf_size == -1)
{
perror (AUDIO);
exit (-1);
}
#endif
zbuf_size = 256;
if ((audiobuf = (u_char *) malloc (abuf_size)) == NULL ||
(zerobuf = (u_char *) malloc (zbuf_size)) == NULL)
{
fprintf (stderr, "%s: unable to allocate input/output buffer\n",
command);
exit (-1);
}
memset ((char *) zerobuf, 128, zbuf_size);
if (direction == PLAY)
{
while (count)
{
c = count;
if (c > abuf_size)
c = abuf_size;
if ((l = read (fd, (char *) audiobuf + loaded, c - loaded)) > 0)
{
l += loaded;
loaded = 0; /* correct the count; ugly but ... */
real_l = (one_chn
|| to_8) ? one_channel (audiobuf, l, one_chn,
to_8) : l;
/* change byte order if necessary */
if (convert && (samplesize == 16))
{
long i;
for (i = 0; i < real_l; i += 2)
*((short *) (audiobuf + i)) =
htons (*((short *) (audiobuf + i)));
}
if (write (audio, (char *) audiobuf, real_l) != real_l)
{
perror (AUDIO);
exit (-1);
}
count -= l;
}
else
{
if (l == -1)
{
perror (name);
exit (-1);
}
count = 0; /* Stop */
}
} /* while (count) */
}
else
{ /* we are recording */
while (count)
{
c = count;
if (c > abuf_size)
c = abuf_size;
if ((l = read (audio, (char *) audiobuf, c)) > 0)
{
if (write (fd, (char *) audiobuf, l) != l)
{
perror (name);
exit (-1);
}
count -= l;
}
if (l == -1)
{
perror (AUDIO);
exit (-1);
}
} /* While count */
}
}
/*
let's play or record it (record_type says VOC/WAVE/raw)
*/
void
record_play (char *name)
{
int fd, ofs;
if (direction == PLAY)
{
if (!name)
{
fd = 0;
name = "stdin";
}
else if ((fd = open (name, O_RDONLY, 0)) == -1)
{
perror (name);
exit (-1);
}
/* Read the smallest header first, then the
missing bytes for the next, etc. */
/* read SND-header */
read (fd, (char *) audiobuf, sizeof (SndHeader));
if (test_sndfile (audiobuf, fd) >= 0)
recplay (fd, 0, count, SND_FMT, name);
else
{
/* read VOC-Header */
read (fd, (char *) audiobuf + sizeof (SndHeader),
sizeof (VocHeader) - sizeof (SndHeader));
if ((ofs = test_vocfile (audiobuf)) >= 0)
vplay (fd, ofs, name);
else
{
/* read bytes for WAVE-header */
read (fd, (char *) audiobuf + sizeof (VocHeader),
sizeof (WaveHeader) - sizeof (VocHeader));
if (test_wavefile (audiobuf) >= 0)
recplay (fd, 0, count, WAVE_FMT, name);
else
{
/* should be raw data */
init_raw_data ();
count = calc_count ();
recplay (fd, sizeof (WaveHeader), count, RAW_DATA, name);
}
}
}
if (fd != 0)
close (fd);
}
else
{ /* recording ... */
if (!name)
{
fd = 1;
name = "stdout";
}
else
{
if ((fd = open (name, O_WRONLY | O_CREAT | O_TRUNC, 0666)) == -1)
{
perror (name);
exit (-1);
}
}
count = calc_count () & 0xFFFFFFFE;
/* WAVE-file should be even (I'm not sure), but wasting one byte
isn't a problem (this can only be in 8 bit mono) */
if (fmt_rec_table[record_type].start)
fmt_rec_table[record_type].start (fd, count);
recplay (fd, 0, count, record_type, name);
fmt_rec_table[record_type].end (fd);
}
}
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