NetBSD/sys/arch/i386/isa/bsd_audio.c

1022 lines
23 KiB
C

/*
* Copyright (c) 1991-1993 Regents of the University of California.
* 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. All advertising materials mentioning features or use of this software
* must display the following acknowledgement:
* This product includes software developed by the Computer Systems
* Engineering Group at Lawrence Berkeley Laboratory.
* 4. Neither the name of the University nor of the Laboratory may be used
* to endorse or promote products derived from this software without
* specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``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 REGENTS OR CONTRIBUTORS 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.
*
* $Id: bsd_audio.c,v 1.2 1994/04/24 01:29:56 mycroft Exp $
*/
/*
* This is a (partially) SunOS-compatible /dev/audio driver for NetBSD.
* This code assumes SoundBlaster type hardware, supported by the
* code in isa/sb.c. A major problem with this hardware is that it
* is half-duplex. E.g., you cannot simultaneously record and play
* samples. Thus, it doesn't really make sense to allow O_RDWR access.
* However, opening and closing the device to "turn around the line"
* is relatively expensive and costs a card reset (which can take
* some time). Instead, we allow O_RDWR access, and provide an
* ioctl to set the "mode", e.g., playing or recording. If you
* write to the device in record mode, the data is tossed. If you
* read from the device in play mode, you get zero filled buffers
* at the rate at which samples are naturally generated.
*/
#include "audio.h"
#if NAUDIO > 0
#include <sys/param.h>
#include <sys/ioctl.h>
#include <sys/fcntl.h>
#include <sys/vnode.h>
#include <sys/select.h>
#include <sys/malloc.h>
#include <machine/bsd_audioio.h>
#include <i386/isa/sbreg.h>
#define AUDIODEBUG if (audiodebug) printf
int audiodebug = 0;
/*
* Initial/default block size is patchable.
*/
#ifndef AUDIOBLKSIZE
#ifdef SBPRO
#define AUDIOBLKSIZE 1024 /* ~20ms at 43478 Hz */
#else
#define AUDIOBLKSIZE 160 /* 20ms at 8KHz */
#endif
#endif
int audio_blocksize = AUDIOBLKSIZE;
int audio_backlog = 3; /* 60ms in samples */
/* XXX */
#define splaudio splclock
/*
* Software state, per audio device.
*/
struct audio_softc {
struct sb_softc *sc_sb;
u_char sc_open; /* single use device */
u_char sc_mode; /* */
u_char sc_rbus; /* input dma in progress */
u_char sc_pbus; /* output dma in progress */
u_char sc_rulaw;
u_char sc_pulaw;
u_char sc_pad[2];
u_long sc_wseek; /* timestamp of last frame written */
u_long sc_rseek; /* timestamp of last frame read */
u_long sc_orate; /* input sampling rate */
u_long sc_irate; /* output sampling rate */
struct selinfo sc_wsel; /* write selector */
struct selinfo sc_rsel; /* read selector */
int sc_rlevel; /* record level */
int sc_plevel; /* play level */
/*
* Sleep channels for reading and writing.
*/
int sc_rchan;
int sc_wchan;
/*
* Buffer management.
*/
u_char *sc_hp; /* head */
u_char *sc_tp; /* tail */
u_char *sc_bp; /* start of buffer */
u_char *sc_ep; /* end of buffer */
u_char *sc_zp; /* block of silence */
int sc_nblk;
int sc_maxblk;
int sc_lowat; /* xmit low water mark (for wakeup) */
int sc_hiwat; /* xmit high water mark (for wakeup) */
int sc_blksize; /* recv block (chunk) size */
int sc_backlog; /* # blks of xmit backlog to gen. */
int sc_rblks; /* number of phantom record blocks */
};
/* XXX */
struct sb_softc *sbopen();
static int audio_default_level = 150;
static void ausetrgain(struct audio_softc *, int);
static void ausetpgain(struct audio_softc *, int);
static int audiosetinfo(struct audio_softc *, struct audio_info *);
static int audiogetinfo(struct audio_softc *, struct audio_info *);
struct audio_softc audio_softc;
void audio_init_record(struct audio_softc *);
void audio_init_play(struct audio_softc *);
void audiostartr(struct audio_softc *);
void audiostartp(struct audio_softc *);
void audio_rint(struct audio_softc *);
void audio_pint(struct audio_softc *);
void audio_tomulaw(register u_char *, register int);
void audio_frommulaw(register u_char *, register int);
audio_initbuf(struct audio_softc *sc)
{
register int nblk = NBPG / sc->sc_blksize;
sc->sc_ep = sc->sc_bp + nblk * sc->sc_blksize;
sc->sc_hp = sc->sc_tp = sc->sc_bp;
sc->sc_maxblk = nblk;
sc->sc_nblk = 0;
sc->sc_lowat = 1;
sc->sc_hiwat = nblk - sc->sc_lowat;
}
static inline int
audio_sleep(int *chan)
{
int st;
*chan = 1;
st = (tsleep((caddr_t)chan, PWAIT | PCATCH, "audio", 0));
*chan = 0;
return (st);
}
static inline void
audio_wakeup(int *chan)
{
if (*chan) {
wakeup((caddr_t)chan);
*chan = 0;
}
}
/*XXX*/
int auzero[1024];
void
audioattach(int unused)
{
AUDIODEBUG("audio: attach\n");
}
int
audioopen(dev_t dev, int flags, int ifmt, struct proc *p)
{
register struct audio_softc *sc = &audio_softc;
int s;
AUDIODEBUG("audio: open\n");
if (sc->sc_open != 0 || (sc->sc_sb = sbopen()) == 0)
return (EBUSY);
sc->sc_open = 1;
/*
* Allocate a single page so it won't cross a page boundary.
* This way the dma carried out in the sb module will be efficient
* (i.e., at_dma() won't have to make a copy)
*/
sc->sc_zp = malloc(NBPG, M_DEVBUF, M_WAITOK);
if (sc->sc_zp == 0)
goto nobufs;
sc->sc_bp = malloc(NBPG, M_DEVBUF, M_WAITOK);
if (sc->sc_bp == 0) {
free(sc->sc_zp, M_DEVBUF);
goto nobufs;
}
sc->sc_blksize = audio_blocksize;
sc->sc_backlog = audio_backlog;
audio_initbuf(sc);
/* nothing read or written yet */
sc->sc_rseek = 0;
sc->sc_wseek = 0;
sc->sc_rchan = 0;
sc->sc_wchan = 0;
/*
* Here's a hack: do ulaw conversion if high bit of
* minor device is set. That way, we can have /dev/audio
* (minor 0x80) do ulaw conversion, and /dev/sound or
* whatever, do linear.
*/
if (minor(dev) & 0x80) {
/* /dev/audio */
int i;
sc->sc_pulaw = sc->sc_rulaw = 1;
sc->sc_orate = 8000;
sc->sc_irate = 8000;
for (i = NBPG / 4; --i >= 0; ) {
((u_long *)sc->sc_zp)[i] = 0x7f7f7f7f;
auzero[i] = 0x80808080;
}
} else {
/* /dev/sound */
sc->sc_pulaw = sc->sc_rulaw = 0;
#ifdef SBPRO
sc->sc_orate = 43478;
sc->sc_irate = 43478;
#else
#ifdef notdef
sc->sc_orate = 14925;
sc->sc_irate = 14925;
#endif
sc->sc_orate = 8000;
sc->sc_irate = 8000;
#endif
bzero(sc->sc_zp, NBPG);
}
ausetrgain(sc, audio_default_level);
ausetpgain(sc, audio_default_level);
/* XXX */
s = splaudio();
sc->sc_rbus = 0;
sc->sc_pbus = 0;
if ((flags & FREAD) != 0) {
audio_init_record(sc);
audiostartr(sc);
} else {
audio_init_play(sc);
audio_pint(sc);
}
splx(s);
return (0);
nobufs:
sbclose(sc->sc_sb);
sc->sc_open = 0;
return (ENOBUFS);
}
audio_to(caddr_t arg)
{
wakeup(arg);
}
/*
* Wait a little while because doing certain things to
* the soundblaster (like toggling the speaker) make
* it go away for a while.
*/
void
audio_pause(struct audio_softc *sc)
{
extern int hz;
timeout(audio_to, audio_to, hz / 8);
(void)tsleep((caddr_t)audio_to, PWAIT, "audio", 0);
}
/*
* Must be called from task context.
*/
void
audio_init_record(struct audio_softc *sc)
{
register int s = splaudio();
sc->sc_mode = AUMODE_RECORD;
(void)sb_set_sr(sc->sc_sb, &sc->sc_irate, SB_INPUT_RATE);
sb_spkroff(sc->sc_sb);
audio_pause(sc);
splx(s);
}
/*
* Must be called from task context.
*/
void
audio_init_play(struct audio_softc *sc)
{
register int s = splaudio();
sc->sc_mode = AUMODE_PLAY;
sc->sc_rblks = 0;
(void)sb_set_sr(sc->sc_sb, &sc->sc_orate, SB_OUTPUT_RATE);
sb_spkron(sc->sc_sb);
audio_pause(sc);
splx(s);
}
static int
audio_drain(sc)
register struct audio_softc *sc;
{
register int error;
while (sc->sc_nblk > 0) {
error = audio_sleep(&sc->sc_wchan);
if (error != 0)
return (error);
}
return (0);
}
/*
* Close an audio chip.
*/
/* ARGSUSED */
int
audioclose(dev_t dev, int flags, int ifmt, struct proc *p)
{
register struct audio_softc *sc = &audio_softc;
register struct aucb *cb;
register int s;
AUDIODEBUG("audio: close\n");
/*
* Block until output drains, but allow ^C interrupt.
*/
sc->sc_lowat = 0; /* avoid excessive wakeups */
s = splaudio();
/*
* If there is pending output, let it drain (unless
* the output is paused).
*/
if (sc->sc_pbus && sc->sc_nblk > 0)
(void)audio_drain(sc);
sbclose(sc->sc_sb);
splx(s);
free(sc->sc_bp, M_DEVBUF);
free(sc->sc_zp, M_DEVBUF);
sc->sc_open = 0;
return (0);
}
int
audioread(dev_t dev, struct uio *uio, int ioflag)
{
register struct audio_softc *sc = &audio_softc;
register u_char *hp;
register int blocksize = sc->sc_blksize;
register int error, s;
if (uio->uio_resid == 0)
return (0);
if (uio->uio_resid < blocksize)
return (EINVAL);
if (sc->sc_mode == AUMODE_PLAY) {
/*
* If we're in play mode, return silence blocks
* based on the number of blocks we have output.
*/
do {
s = splaudio();
while (sc->sc_rblks <= 0) {
if (ioflag & IO_NDELAY) {
splx(s);
return (EWOULDBLOCK);
}
error = audio_sleep(&sc->sc_rchan);
if (error != 0) {
splx(s);
return (error);
}
}
splx(s);
/*XXX handle ulaw 0 */
error = uiomove(sc->sc_zp, blocksize, uio);
if (error)
break;
--sc->sc_rblks;
} while (uio->uio_resid >= blocksize);
return (error);
}
error = 0;
do {
while (sc->sc_nblk <= 0) {
if (ioflag & IO_NDELAY) {
error = EWOULDBLOCK;
return (error);
}
s = splaudio();
if (!sc->sc_rbus)
audiostartr(sc);
error = audio_sleep(&sc->sc_rchan);
splx(s);
if (error != 0)
return (error);
}
hp = sc->sc_hp;
if (sc->sc_rulaw)
audio_tomulaw(hp, blocksize);
error = uiomove(hp, blocksize, uio);
if (error)
break;
hp += blocksize;
if (hp >= sc->sc_ep)
hp = sc->sc_bp;
sc->sc_hp = hp;
--sc->sc_nblk;
} while (uio->uio_resid >= blocksize);
return (error);
}
void
audio_clear(struct audio_softc *sc)
{
register int s = splaudio();
if (sc->sc_rbus || sc->sc_pbus) {
sb_haltdma(sc->sc_sb);
sc->sc_rbus = 0;
sc->sc_pbus = 0;
}
sc->sc_nblk = 0;
sc->sc_hp = sc->sc_tp = sc->sc_bp;
splx(s);
}
int
audiowrite(dev_t dev, struct uio *uio, int ioflag)
{
register struct audio_softc *sc = &audio_softc;
register u_char *tp;
register int error, s, cc;
register int blocksize = sc->sc_blksize;
/*
* If currently recording, throw away data.
*/
if (sc->sc_mode != AUMODE_PLAY) {
uio->uio_offset += uio->uio_resid;
uio->uio_resid = 0;
return (0);
}
error = 0;
while (uio->uio_resid > 0) {
register int watermark = sc->sc_hiwat;
s = splaudio();
while (sc->sc_nblk > watermark) {
if (ioflag & IO_NDELAY) {
splx(s);
error = EWOULDBLOCK;
return (error);
}
error = audio_sleep(&sc->sc_wchan);
if (error != 0) {
splx(s);
return (error);
}
watermark = sc->sc_lowat;
}
splx(s);
if (sc->sc_nblk == 0 && uio->uio_resid <= blocksize) {
/*
* the write is 'small', the buffer is empty
* and we have been silent for at least 50ms
* so we might be dealing with an application
* that writes frames synchronously with
* reading them. If so, we need an output
* backlog to cover scheduling delays or
* there will be gaps in the sound output.
* Also take this opportunity to reset the
* buffer pointers in case we ended up on
* a bad boundary (odd byte, blksize bytes
* from end, etc.).
*/
s = splaudio();/*XXX*/
sc->sc_hp = sc->sc_bp;
bzero(sc->sc_hp, 3 * blocksize);
sc->sc_nblk = 3;
sc->sc_tp = sc->sc_hp + 3 * blocksize;
splx(s);
}
tp = sc->sc_tp;
cc = uio->uio_resid;
if (cc < blocksize) {
error = uiomove(tp, cc, uio);
if (error)
break;
tp += cc;
cc = blocksize - cc;
while (--cc >= 0)
*tp++ = 0x7f;
} else {
error = uiomove(tp, blocksize, uio);
if (error)
break;
tp += blocksize;
}
if (sc->sc_pulaw)
audio_frommulaw(sc->sc_tp, blocksize);
if (tp >= sc->sc_ep)
tp = sc->sc_bp;
sc->sc_tp = tp;
++sc->sc_nblk;
/*
* If output isn't active, start it up.
*/
s = splaudio();
if (sc->sc_pbus == 0)
audiostartp(sc);
splx(s);
}
return (error);
}
/* Sun audio compatibility */
struct sun_audio_prinfo {
u_int sample_rate;
u_int channels;
u_int precision;
u_int encoding;
u_int gain;
u_int port;
u_int reserved0[4];
u_int samples;
u_int eof;
u_char pause;
u_char error;
u_char waiting;
u_char reserved1[3];
u_char open;
u_char active;
};
struct sun_audio_info {
struct sun_audio_prinfo play;
struct sun_audio_prinfo record;
u_int monitor_gain;
u_int reserved[4];
};
int
audioioctl(dev_t dev, int cmd, caddr_t addr, int flag, struct proc *p)
{
register struct audio_softc *sc = &audio_softc;
int error = 0, i, s;
AUDIODEBUG("audio: ioctl(0x%x)\n", cmd);
switch (cmd) {
case AUDIO_FLUSH:
AUDIODEBUG("AUDIO_FLUSH\n");
audio_clear(sc);
if (sc->sc_mode != AUMODE_PLAY)
audiostartr(sc);
break;
#ifdef notdef
/*
* Number of read samples dropped. We don't know where or
* when they were dropped.
*/
case AUDIO_RERROR:
*(int *)addr = sc->sc_au.au_rb.cb_drops != 0;
break;
/*
* How many samples will elapse until mike hears the first
* sample of what we last wrote?
*/
case AUDIO_WSEEK:
s = splaudio();
*(u_long *)addr = sc->sc_wseek - sc->sc_au.au_stamp
+ AUCB_LEN(&sc->sc_au.au_rb);
splx(s);
break;
#endif
case AUDIO_SETINFO:
AUDIODEBUG("AUDIO_SETINFO\n");
error = audiosetinfo(sc, (struct audio_info *)addr);
break;
case AUDIO_GETINFO:
AUDIODEBUG("AUDIO_GETINFO\n");
error = audiogetinfo(sc, (struct audio_info *)addr);
break;
case AUDIO_DRAIN:
AUDIODEBUG("AUDIO_DRAIN\n");
error = audio_drain(sc);
break;
default:
AUDIODEBUG("audio: unknown ioctl\n");
error = EINVAL;
break;
}
AUDIODEBUG("audio: ioctl(%d) result %d\n", cmd, error);
return (error);
}
int
audioselect(dev_t dev, int rw, struct proc *p)
{
register struct audio_softc *sc = &audio_softc;
register int s = splaudio();
switch (rw) {
case FREAD:
if (sc->sc_mode == AUMODE_PLAY) {
if (sc->sc_rblks > 0) {
splx(s);
return (1);
}
} else if (sc->sc_nblk > 0) {
splx(s);
return (1);
}
selrecord(p, &sc->sc_rsel);
break;
case FWRITE:
/*
* Can write if we're recording because it gets preempted.
* Otherwise, can write when below low water.
* XXX this won't work right if we're in
* record mode -- we need to note that a write
* select has happed and flip the speaker.
*/
if (sc->sc_mode != AUMODE_PLAY ||
sc->sc_nblk < sc->sc_lowat) {
splx(s);
return (1);
}
selrecord(p, &sc->sc_wsel);
break;
}
splx(s);
return (0);
}
void
audiostartr(struct audio_softc *sc)
{
sb_dma_input(sc->sc_sb, sc->sc_tp, sc->sc_blksize,
audio_rint, (void *)sc);
sc->sc_rbus = 1;
}
void
audiostartp(struct audio_softc *sc)
{
/*XXX check for nblk == 0 */
sb_dma_output(sc->sc_sb, sc->sc_hp, sc->sc_blksize,
audio_pint, (void *)sc);
sc->sc_pbus = 1;
}
void
audio_pint(struct audio_softc *sc)
{
register u_char *hp;
register int cc = sc->sc_blksize;
if (sc->sc_nblk > 0) {
--sc->sc_nblk;
hp = sc->sc_hp;
hp += cc;
if (hp >= sc->sc_ep)
hp = sc->sc_bp;
sc->sc_hp = hp;
sb_dma_output(sc->sc_sb, hp, cc, audio_pint, (void *)sc);
} else {
sb_dma_output(sc->sc_sb, auzero, cc,
audio_pint, (void *)sc);
}
++sc->sc_rblks;
if (sc->sc_mode == AUMODE_PLAY) {
if (sc->sc_nblk <= sc->sc_lowat) {
audio_wakeup(&sc->sc_wchan);
selwakeup(&sc->sc_wsel);
}
}
/*XXX*/
audio_wakeup(&sc->sc_rchan);
selwakeup(&sc->sc_rsel);
}
/*
* Called from sb module on completion of dma input.
* Copy the input frame into the ring buffer at the
* current position. Do a wakeup if necessary.
*/
void
audio_rint(struct audio_softc *sc)
{
register u_char *tp;
register int cc = sc->sc_blksize;
tp = sc->sc_tp;
tp += cc;
if (tp >= sc->sc_ep)
tp = sc->sc_bp;
if (++sc->sc_nblk < sc->sc_maxblk)
sb_dma_input(sc->sc_sb, tp, cc, audio_rint, (void *)sc);
else
sc->sc_rbus = 0;
sc->sc_tp = tp;
audio_wakeup(&sc->sc_rchan);
selwakeup(&sc->sc_rsel);
}
static void
ausetrgain(sc, level)
register struct audio_softc *sc;
register int level;
{
#ifdef SBPRO
/* XXX */
#endif
}
/*
* XXX Looks like we need a pro to do volume control...
*/
static void
ausetpgain(sc, level)
register struct audio_softc *sc;
register int level;
{
#ifdef SBPRO
/* XXX */
#endif
}
static int
audiosetinfo(sc, ai)
struct audio_softc *sc;
struct audio_info *ai;
{
struct audio_prinfo *r = &ai->record, *p = &ai->play;
register int cleared = 0;
register int s, bsize;
if (p->gain != ~0)
ausetpgain(sc, p->gain);
if (r->gain != ~0)
ausetrgain(sc, r->gain);
if (p->sample_rate != ~0) {
if (!cleared)
audio_clear(sc);
cleared = 1;
p->sample_rate = sb_round_sr(p->sample_rate, SB_OUTPUT_RATE);
sc->sc_orate = p->sample_rate;
if (sc->sc_mode == AUMODE_PLAY)
(void)sb_set_sr(sc->sc_sb, &sc->sc_orate,
SB_OUTPUT_RATE);
}
if (r->sample_rate != ~0) {
if (!cleared)
audio_clear(sc);
cleared = 1;
r->sample_rate = sb_round_sr(r->sample_rate, SB_INPUT_RATE);
sc->sc_irate = r->sample_rate;
if (sc->sc_mode != AUMODE_PLAY)
(void)sb_set_sr(sc->sc_sb, &sc->sc_irate,
SB_INPUT_RATE);
}
if (p->encoding != ~0) {
if (!cleared)
audio_clear(sc);
cleared = 1;
if (p->encoding == AUDIO_ENCODING_ULAW)
sc->sc_pulaw = 1;
else {
sc->sc_pulaw = 0;
p->encoding = AUDIO_ENCODING_LINEAR;
}
}
if (r->encoding != ~0) {
if (r->encoding == AUDIO_ENCODING_ULAW)
sc->sc_rulaw = 1;
else {
r->encoding = AUDIO_ENCODING_LINEAR;
sc->sc_rulaw = 0;
}
}
#ifdef notdef
if (p->pause != (u_char)~0)
sc->sc_au.au_wb.cb_pause = p->pause;
if (r->pause != (u_char)~0)
sc->sc_au.au_rb.cb_pause = r->pause;
#endif
if (ai->blocksize != ~0) {
if (!cleared)
audio_clear(sc);
cleared = 1;
if (ai->blocksize == 0)
bsize = audio_blocksize;
else if (ai->blocksize > NBPG/2)
bsize = NBPG/2;
else
bsize = ai->blocksize;
ai->blocksize = sc->sc_blksize = bsize;
audio_initbuf(sc);
}
if (ai->hiwat != ~0) {
if ((unsigned)ai->hiwat > sc->sc_maxblk)
ai->hiwat = sc->sc_maxblk;
sc->sc_hiwat = ai->hiwat;
}
if (ai->lowat != ~0) {
if ((unsigned)ai->lowat > sc->sc_maxblk)
ai->lowat = sc->sc_maxblk;
sc->sc_lowat = ai->lowat;
}
if (ai->backlog != ~0) {
if ((unsigned)ai->backlog > (sc->sc_maxblk/2))
ai->backlog = sc->sc_maxblk/2;
sc->sc_backlog = ai->backlog;
}
if (ai->mode != ~0) {
if (!cleared)
audio_clear(sc);
cleared = 1;
sc->sc_mode = ai->mode;
if (sc->sc_mode == AUMODE_PLAY)
audio_init_play(sc);
else
audio_init_record(sc);
}
if (cleared) {
if (sc->sc_mode != AUMODE_PLAY)
audiostartr(sc);
else
audiostartp(sc);
}
return (0);
}
static int
audiogetinfo(sc, ai)
struct audio_softc *sc;
struct audio_info *ai;
{
struct audio_prinfo *r = &ai->record, *p = &ai->play;
p->sample_rate = sc->sc_orate;
r->sample_rate = sc->sc_irate;
p->channels = r->channels = 1;
p->precision = r->precision = 8;
p->encoding = sc->sc_pulaw ? AUDIO_ENCODING_ULAW :
AUDIO_ENCODING_LINEAR;
r->encoding = sc->sc_rulaw ? AUDIO_ENCODING_ULAW :
AUDIO_ENCODING_LINEAR;
ai->monitor_gain = 0;
r->gain = sc->sc_rlevel;
p->gain = sc->sc_plevel;
r->port = 1; p->port = AUDIO_SPEAKER;
#ifdef notdef
p->pause = sc->sc_au.au_wb.cb_pause;
r->pause = sc->sc_au.au_rb.cb_pause;
p->error = sc->sc_au.au_wb.cb_drops != 0;
r->error = sc->sc_au.au_rb.cb_drops != 0;
#endif
p->open = sc->sc_open;
r->open = sc->sc_open;
#ifdef notdef
p->samples = sc->sc_au.au_stamp - sc->sc_au.au_wb.cb_pdrops;
r->samples = sc->sc_au.au_stamp - sc->sc_au.au_rb.cb_pdrops;
#endif
p->seek = sc->sc_wseek;
r->seek = sc->sc_rseek;
ai->blocksize = sc->sc_blksize;
ai->hiwat = sc->sc_hiwat;
ai->lowat = sc->sc_lowat;
ai->backlog = sc->sc_backlog;
ai->mode = sc->sc_mode;
return (0);
}
u_char mulawtolin[256] = {
128, 4, 8, 12, 16, 20, 24, 28,
32, 36, 40, 44, 48, 52, 56, 60,
64, 66, 68, 70, 72, 74, 76, 78,
80, 82, 84, 86, 88, 90, 92, 94,
96, 97, 98, 99, 100, 101, 102, 103,
104, 105, 106, 107, 108, 109, 110, 111,
112, 112, 113, 113, 114, 114, 115, 115,
116, 116, 117, 117, 118, 118, 119, 119,
120, 120, 120, 121, 121, 121, 121, 122,
122, 122, 122, 123, 123, 123, 123, 124,
124, 124, 124, 124, 125, 125, 125, 125,
125, 125, 125, 125, 126, 126, 126, 126,
126, 126, 126, 126, 126, 126, 126, 126,
127, 127, 127, 127, 127, 127, 127, 127,
127, 127, 127, 127, 127, 127, 127, 127,
127, 127, 127, 127, 127, 127, 127, 127,
255, 251, 247, 243, 239, 235, 231, 227,
223, 219, 215, 211, 207, 203, 199, 195,
191, 189, 187, 185, 183, 181, 179, 177,
175, 173, 171, 169, 167, 165, 163, 161,
159, 158, 157, 156, 155, 154, 153, 152,
151, 150, 149, 148, 147, 146, 145, 144,
143, 143, 142, 142, 141, 141, 140, 140,
139, 139, 138, 138, 137, 137, 136, 136,
135, 135, 135, 134, 134, 134, 134, 133,
133, 133, 133, 132, 132, 132, 132, 131,
131, 131, 131, 131, 130, 130, 130, 130,
130, 130, 130, 130, 129, 129, 129, 129,
129, 129, 129, 129, 129, 129, 129, 129,
128, 128, 128, 128, 128, 128, 128, 128,
128, 128, 128, 128, 128, 128, 128, 128,
128, 128, 128, 128, 128, 128, 128, 128,
};
u_char lintomulaw[256] = {
0, 0, 0, 0, 0, 1, 1, 1,
1, 2, 2, 2, 2, 3, 3, 3,
3, 4, 4, 4, 4, 5, 5, 5,
5, 6, 6, 6, 6, 7, 7, 7,
7, 8, 8, 8, 8, 9, 9, 9,
9, 10, 10, 10, 10, 11, 11, 11,
11, 12, 12, 12, 12, 13, 13, 13,
13, 14, 14, 14, 14, 15, 15, 15,
15, 16, 16, 17, 17, 18, 18, 19,
19, 20, 20, 21, 21, 22, 22, 23,
23, 24, 24, 25, 25, 26, 26, 27,
27, 28, 28, 29, 29, 30, 30, 31,
31, 32, 33, 34, 35, 36, 37, 38,
39, 40, 41, 42, 43, 44, 45, 46,
47, 48, 50, 52, 54, 56, 58, 60,
62, 65, 69, 73, 77, 83, 91, 103,
255, 231, 219, 211, 205, 201, 197, 193,
190, 188, 186, 184, 182, 180, 178, 176,
175, 174, 173, 172, 171, 170, 169, 168,
167, 166, 165, 164, 163, 162, 161, 160,
159, 159, 158, 158, 157, 157, 156, 156,
155, 155, 154, 154, 153, 153, 152, 152,
151, 151, 150, 150, 149, 149, 148, 148,
147, 147, 146, 146, 145, 145, 144, 144,
143, 143, 143, 143, 142, 142, 142, 142,
141, 141, 141, 141, 140, 140, 140, 140,
139, 139, 139, 139, 138, 138, 138, 138,
137, 137, 137, 137, 136, 136, 136, 136,
135, 135, 135, 135, 134, 134, 134, 134,
133, 133, 133, 133, 132, 132, 132, 132,
131, 131, 131, 131, 130, 130, 130, 130,
129, 129, 129, 129, 128, 128, 128, 128,
};
void
audio_tomulaw(register u_char *p, register int cc)
{
register u_char *utab = lintomulaw;
while (--cc >= 0) {
*p = utab[*p];
++p;
}
}
void
audio_frommulaw(register u_char *p, register int cc)
{
register u_char *utab = mulawtolin;
while (--cc >= 0) {
*p = utab[*p];
++p;
}
}
#endif