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NetBSD/sys/arch/sparc/dev/bsd_audio.c
deraadt 941871d6ab XXmatch change (and a few __P() as well)
1994-11-23 07:02:02 +00:00

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/* $NetBSD: bsd_audio.c,v 1.7 1994/11/23 07:02:02 deraadt Exp $ */
/*
* Copyright (c) 1991, 1992, 1993
* The Regents of the University of California. All rights reserved.
*
* This software was developed by the Computer Systems Engineering group
* at Lawrence Berkeley Laboratory under DARPA contract BG 91-66 and
* contributed to Berkeley.
*
* All advertising materials mentioning features or use of this software
* must display the following acknowledgement:
* This product includes software developed by the University of
* California, Lawrence Berkeley Laboratory.
*
* 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 University of
* California, Berkeley and its contributors.
* 4. Neither the name of the University nor the names of its contributors
* 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.
*
* @(#)bsd_audio.c 8.1 (Berkeley) 6/11/93
*/
#include "bsdaudio.h"
#if NBSDAUDIO > 0
#include <sys/param.h>
#include <sys/systm.h>
#if BSD < 199103
#ifndef SUNOS
#define SUNOS
#endif
#endif
#include <sys/errno.h>
#include <sys/file.h>
#include <sys/proc.h>
#include <sys/user.h>
#include <sys/vnode.h>
#include <sys/ioctl.h>
#include <sys/time.h>
#ifndef SUNOS
#include <sys/tty.h>
#endif
#include <sys/uio.h>
#ifdef SUNOS
#include <sundev/mbvar.h>
#include <sun4c/intreg.h>
#else
#include <sys/device.h>
#include <machine/autoconf.h>
#endif
#include <machine/cpu.h>
/*
* Avoid name clashes with SunOS so we can config either the bsd or sun
* streams driver in a SunOS kernel.
*/
#ifdef SUNOS
#include <sbusdev/bsd_audioreg.h>
#include <sbusdev/bsd_audiovar.h>
#include <sbusdev/bsd_audioio.h>
struct selinfo {
struct proc *si_proc;
int si_coll;
};
#else
#include <sparc/dev/bsd_audioreg.h>
#include <sparc/dev/bsd_audiovar.h>
#include <machine/bsd_audioio.h>
#endif
#ifdef SUNOS
#include "bsd_audiocompat.h"
#endif
/*
* Initial/default block size is patchable.
*/
int audio_blocksize = DEFBLKSIZE;
int audio_backlog = 400; /* 50ms in samples */
/*
* Software state, per AMD79C30 audio chip.
*/
struct audio_softc {
#ifndef SUNOS
struct device sc_dev; /* base device */
struct intrhand sc_hwih; /* hardware interrupt vector */
struct intrhand sc_swih; /* software interrupt vector */
#endif
int sc_interrupts; /* number of interrupts taken */
int sc_open; /* single use device */
u_long sc_wseek; /* timestamp of last frame written */
u_long sc_rseek; /* timestamp of last frame read */
struct mapreg sc_map; /* current contents of map registers */
struct selinfo sc_wsel; /* write selector */
struct selinfo sc_rsel; /* read selector */
/*
* keep track of levels so we don't have to convert back from
* MAP gain constants
*/
int sc_rlevel; /* record level */
int sc_plevel; /* play level */
int sc_mlevel; /* monitor level */
/* sc_au is special in that the hardware interrupt handler uses it */
struct auio sc_au; /* recv and xmit buffers, etc */
};
/* interrupt interfaces */
#ifndef AUDIO_C_HANDLER
int audiohwintr __P((void *));
#endif
int audioswintr __P((void *));
/* forward declarations */
int audio_sleep __P((struct aucb *, int));
void audio_setmap __P((volatile struct amd7930 *, struct mapreg *));
static void init_amd();
#if !defined(AUDIO_C_HANDLER) || defined(SUNOS)
struct auio *audio_au;
extern void audio_trap();
#endif
#ifdef SUNOS
struct audio_softc audio_softc;
#define SOFTC(dev) &audio_softc
#define UIOMOVE(cp, len, code, uio) uiomove(cp, len, code, uio)
#define AUDIOOPEN(d, f, i, p)\
audioopen(d, f, i)\
dev_t d; int f, i;
#define AUDIOCLOSE(d, f, i, p)\
audioclose(d, f, i)\
dev_t d; int f, i;
#define AUDIOREAD(d, u, f) \
audioread(d, u) dev_t d; struct uio *u;
#define AUDIOWRITE(d, u, f) \
audiowrite(d, u) dev_t d; struct uio *u;
#define AUDIOIOCTL(d, c, a, f, o)\
audioioctl(d, c, a, f)\
dev_t d; int c; caddr_t a; int f;
#define AUDIOSELECT(d, r, p)\
audio_select(d, r, p)\
dev_t d; int r; struct proc *p;
#define AUDIO_SET_SWINTR set_intreg(IR_SOFT_INT4, 1)
int
audioselect(dev, rw)
register dev_t dev;
int rw;
{
return (audio_select(dev, rw, u.u_procp));
}
static void
selrecord(p, si)
struct proc *p;
struct selinfo *si;
{
if (si->si_proc != 0)
si->si_coll = 1;
else
si->si_proc = p;
}
#define SELWAKEUP(si) \
{\
if ((si)->si_proc != 0) {\
selwakeup((si)->si_proc, (si)->si_coll); \
(si)->si_proc = 0;\
(si)->si_coll = 0;\
}\
}
static int audioattach();
static int audioidentify();
struct dev_ops bsdaudio_ops = {
0,
audioidentify,
audioattach,
};
static int
audioidentify(cp)
char *cp;
{
return (strcmp(cp, "audio") == 0);
}
static int
audioattach(dev)
struct dev_info *dev;
{
register struct audio_softc *sc;
register volatile struct amd7930 *amd;
struct dev_reg *reg;
sc = &audio_softc;
if (dev->devi_nreg != 1 || dev->devi_nintr != 1) {
printf("audio: bad config\n");
return (-1);
}
reg = dev->devi_reg;
amd = (struct amd7930 *)map_regs(reg->reg_addr, reg->reg_size,
reg->reg_bustype);
sc->sc_au.au_amd = amd;
init_amd(amd);
audio_au = &sc->sc_au;
#ifndef AUDIO_C_HANDLER
settrap(dev->devi_intr->int_pri, audio_trap);
#else
/* XXX */
addintr(dev->devi_intr->int_pri, audiohwintr, dev->devi_name,
dev->devi_unit);
#endif
addintr(4, audioswintr, dev->devi_name, dev->devi_unit);
report_dev(dev);
return (0);
}
#else
#define AUDIOOPEN(d, f, i, p) audioopen(dev_t d, int f, int i, struct proc *p)
#define AUDIOCLOSE(d, f, i, p) audioclose(dev_t d, int f, int i, \
struct proc *p)
#define AUDIOREAD(d, u, f) audioread(dev_t d, struct uio *u, int f)
#define AUDIOWRITE(d, u, f) audiowrite(dev_t d, struct uio *u, int f)
#define AUDIOIOCTL(d, c, a, f, o)\
audioioctl(dev_t dev, u_long c, caddr_t a, int f, struct proc *p)
#define AUDIOSELECT(d, r, p) audioselect(dev_t dev, int rw, struct proc *p)
#define SELWAKEUP selwakeup
#define AUDIO_SET_SWINTR ienab_bis(IE_L6)
/* autoconfiguration driver */
void audioattach __P((struct device *, struct device *, void *));
int audiomatch __P((struct device *, void *, void *));
struct cfdriver audiocd =
{ NULL, "audio", audiomatch, audioattach,
DV_DULL, sizeof(struct audio_softc) };
#define SOFTC(dev) audiocd.cd_devs[minor(dev)]
#define UIOMOVE(cp, len, code, uio) uiomove(cp, len, uio)
int
audiomatch(parent, vcf, aux)
struct device *parent;
void *vcf, *aux;
{
struct cfdata *cf = vcf;
register struct confargs *ca = aux;
register struct romaux *ra = &ca->ca_ra;
if (cputyp==CPU_SUN4)
return (0);
return (strcmp(cf->cf_driver->cd_name, ra->ra_name) == 0);
}
/*
* Audio chip found.
*/
void
audioattach(parent, self, args)
struct device *parent, *self;
void *args;
{
register struct audio_softc *sc = (struct audio_softc *)self;
register struct confargs *ca = args;
register struct romaux *ra = &ca->ca_ra;
register volatile struct amd7930 *amd;
register int pri;
if (ra->ra_nintr != 1) {
printf(": expected 1 interrupt, got %d\n", ra->ra_nintr);
return;
}
pri = ra->ra_intr[0].int_pri;
printf(" pri %d, softpri %d\n", pri, PIL_AUSOFT);
amd = (volatile struct amd7930 *)(ra->ra_vaddr ?
ra->ra_vaddr : mapiodev(ra->ra_paddr, sizeof (*amd), ca->ca_bustype));
sc->sc_au.au_amd = amd;
init_amd(amd);
#ifndef AUDIO_C_HANDLER
audio_au = &sc->sc_au;
intr_fasttrap(pri, audio_trap);
#else
sc->sc_hwih.ih_fun = audiohwintr;
sc->sc_hwih.ih_arg = &sc->sc_au;
intr_establish(pri, &sc->sc_hwih);
#endif
sc->sc_swih.ih_fun = audioswintr;
sc->sc_swih.ih_arg = sc;
intr_establish(PIL_AUSOFT, &sc->sc_swih);
}
#endif
static void
init_amd(amd)
register volatile struct amd7930 *amd;
{
/* disable interrupts */
amd->cr = AMDR_INIT;
amd->dr = AMD_INIT_PMS_ACTIVE | AMD_INIT_INT_DISABLE;
/*
* Initialize the mux unit. We use MCR3 to route audio (MAP)
* through channel Bb. MCR1 and MCR2 are unused.
* Setting the INT enable bit in MCR4 will generate an interrupt
* on each converted audio sample.
*/
amd->cr = AMDR_MUX_1_4;
amd->dr = 0;
amd->dr = 0;
amd->dr = (AMD_MCRCHAN_BB << 4) | AMD_MCRCHAN_BA;
amd->dr = AMD_MCR4_INT_ENABLE;
}
static int audio_default_level = 150;
static void ausetrgain __P((struct audio_softc *, int));
static void ausetpgain __P((struct audio_softc *, int));
static void ausetmgain __P((struct audio_softc *, int));
static int audiosetinfo __P((struct audio_softc *, struct audio_info *));
static int audiogetinfo __P((struct audio_softc *, struct audio_info *));
struct sun_audio_info;
static int sunaudiosetinfo __P((struct audio_softc *,
struct sun_audio_info *));
static int sunaudiogetinfo __P((struct audio_softc *,
struct sun_audio_info *));
static void audio_setmmr2 __P((volatile struct amd7930 *, int));
/* ARGSUSED */
int
AUDIOOPEN(dev, flags, ifmt, p)
{
register struct audio_softc *sc;
register volatile struct amd7930 *amd;
int unit = minor(dev);
#ifdef SUNOS
if (unit > 0)
return (ENXIO);
sc = &audio_softc;
#else
if (unit >= audiocd.cd_ndevs || (sc = audiocd.cd_devs[unit]) == NULL)
return (ENXIO);
#endif
if (sc->sc_open)
return (EBUSY);
sc->sc_open = 1;
sc->sc_au.au_lowat = audio_blocksize;
sc->sc_au.au_hiwat = AUCB_SIZE - sc->sc_au.au_lowat;
sc->sc_au.au_blksize = audio_blocksize;
sc->sc_au.au_backlog = audio_backlog;
/* set up read and write blocks and `dead sound' zero value. */
AUCB_INIT(&sc->sc_au.au_rb);
sc->sc_au.au_rb.cb_thresh = AUCB_SIZE;
AUCB_INIT(&sc->sc_au.au_wb);
sc->sc_au.au_wb.cb_thresh = -1;
/* nothing read or written yet */
sc->sc_rseek = 0;
sc->sc_wseek = 0;
bzero((char *)&sc->sc_map, sizeof sc->sc_map);
/* default to speaker */
sc->sc_map.mr_mmr2 = AMD_MMR2_AINB | AMD_MMR2_LS;
/* enable interrupts and set parameters established above */
amd = sc->sc_au.au_amd;
audio_setmmr2(amd, sc->sc_map.mr_mmr2);
ausetrgain(sc, audio_default_level);
ausetpgain(sc, audio_default_level);
ausetmgain(sc, 0);
amd->cr = AMDR_INIT;
amd->dr = AMD_INIT_PMS_ACTIVE;
return (0);
}
static int
audio_drain(sc)
register struct audio_softc *sc;
{
register int error;
while (!AUCB_EMPTY(&sc->sc_au.au_wb))
if ((error = audio_sleep(&sc->sc_au.au_wb, 0)) != 0)
return (error);
return (0);
}
/*
* Close an audio chip.
*/
/* ARGSUSED */
int
AUDIOCLOSE(dev, flags, ifmt, p)
{
register struct audio_softc *sc = SOFTC(dev);
register volatile struct amd7930 *amd;
register struct aucb *cb;
register int s;
/*
* Block until output drains, but allow ^C interrupt.
*/
sc->sc_au.au_lowat = 0; /* avoid excessive wakeups */
s = splaudio();
/*
* If there is pending output, let it drain (unless
* the output is paused).
*/
cb = &sc->sc_au.au_wb;
if (!AUCB_EMPTY(cb) && !cb->cb_pause)
(void)audio_drain(sc);
/*
* Disable interrupts, clear open flag, and done.
*/
amd = sc->sc_au.au_amd;
amd->cr = AMDR_INIT;
amd->dr = AMD_INIT_PMS_ACTIVE | AMD_INIT_INT_DISABLE;
splx(s);
sc->sc_open = 0;
return (0);
}
int
audio_sleep(cb, thresh)
register struct aucb *cb;
register int thresh;
{
register int error;
register int s = splaudio();
cb->cb_thresh = thresh;
error = tsleep((caddr_t)cb, (PZERO + 1) | PCATCH, "audio", 0);
splx(s);
return (error);
}
/* ARGSUSED */
int
AUDIOREAD(dev, uio, ioflag)
{
register struct audio_softc *sc = SOFTC(dev);
register struct aucb *cb;
register int n, head, taildata, error;
register int blocksize = sc->sc_au.au_blksize;
if (uio->uio_resid == 0)
return (0);
cb = &sc->sc_au.au_rb;
error = 0;
cb->cb_drops = 0;
sc->sc_rseek = sc->sc_au.au_stamp - AUCB_LEN(cb);
do {
while (AUCB_LEN(cb) < blocksize) {
#ifndef SUNOS
if (ioflag & IO_NDELAY) {
error = EWOULDBLOCK;
return (error);
}
#endif
if ((error = audio_sleep(cb, blocksize)) != 0)
return (error);
}
/*
* The space calculation can only err on the short
* side if an interrupt occurs during processing:
* only cb_tail is altered in the interrupt code.
*/
head = cb->cb_head;
if ((n = AUCB_LEN(cb)) > uio->uio_resid)
n = uio->uio_resid;
taildata = AUCB_SIZE - head;
if (n > taildata) {
error = UIOMOVE((caddr_t)cb->cb_data + head,
taildata, UIO_READ, uio);
if (error == 0)
error = UIOMOVE((caddr_t)cb->cb_data,
n - taildata, UIO_READ, uio);
} else
error = UIOMOVE((caddr_t)cb->cb_data + head, n,
UIO_READ, uio);
if (error)
break;
head = AUCB_MOD(head + n);
cb->cb_head = head;
} while (uio->uio_resid >= blocksize);
return (error);
}
/* ARGSUSED */
int
AUDIOWRITE(dev, uio, ioflag)
{
register struct audio_softc *sc = SOFTC(dev);
register struct aucb *cb = &sc->sc_au.au_wb;
register int n, tail, tailspace, error, first, watermark;
error = 0;
first = 1;
while (uio->uio_resid > 0) {
watermark = sc->sc_au.au_hiwat;
while (AUCB_LEN(cb) > watermark) {
#ifndef SUNOS
if (ioflag & IO_NDELAY) {
error = EWOULDBLOCK;
return (error);
}
#endif
if ((error = audio_sleep(cb, watermark)) != 0)
return (error);
watermark = sc->sc_au.au_lowat;
}
/*
* The only value that can change on an interrupt is
* cb->cb_head. We only pull that out once to decide
* how much to write into cb_data; if we lose a race
* and cb_head changes, we will merely be overly
* conservative. For a legitimate time stamp,
* however, we need to synchronize the accesses to
* au_stamp and cb_head at a high ipl below.
*/
tail = cb->cb_tail;
if ((n = (AUCB_SIZE - 1) - AUCB_LEN(cb)) > uio->uio_resid) {
n = uio->uio_resid;
if (cb->cb_head == tail &&
n <= sc->sc_au.au_blksize &&
sc->sc_au.au_stamp - sc->sc_wseek > 400) {
/*
* 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.).
*/
register u_int* ip;
register int muzero = 0x7f7f7f7f;
register int i = splaudio();
cb->cb_head = cb->cb_tail = 0;
splx(i);
tail = sc->sc_au.au_backlog;
ip = (u_int*)cb->cb_data;
for (i = tail >> 2; --i >= 0; )
*ip++ = muzero;
}
}
tailspace = AUCB_SIZE - tail;
if (n > tailspace) {
/* write first part at tail and rest at head */
error = UIOMOVE((caddr_t)cb->cb_data + tail,
tailspace, UIO_WRITE, uio);
if (error == 0)
error = UIOMOVE((caddr_t)cb->cb_data,
n - tailspace, UIO_WRITE, uio);
} else
error = UIOMOVE((caddr_t)cb->cb_data + tail, n,
UIO_WRITE, uio);
if (error)
break;
tail = AUCB_MOD(tail + n);
if (first) {
register int s = splaudio();
sc->sc_wseek = AUCB_LEN(cb) + sc->sc_au.au_stamp + 1;
/*
* To guarantee that a write is contiguous in the
* sample space, we clear the drop count the first
* time through. If we later get drops, we will
* break out of the loop below, before writing
* a new frame.
*/
cb->cb_drops = 0;
cb->cb_tail = tail;
splx(s);
first = 0;
} else {
if (cb->cb_drops != 0)
break;
cb->cb_tail = tail;
}
}
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];
};
#ifndef SUNOS
#define SUNAUDIO_GETINFO _IOR('A', 1, struct sun_audio_info)
#define SUNAUDIO_SETINFO _IOWR('A', 2, struct sun_audio_info)
#else
#define SUNAUDIO_GETINFO _IOR(A, 1, struct sun_audio_info)
#define SUNAUDIO_SETINFO _IOWR(A, 2, struct sun_audio_info)
#endif
/* ARGSUSED */
int
AUDIOIOCTL(dev, cmd, addr, flag, p)
{
register struct audio_softc *sc = SOFTC(dev);
int error = 0, s;
switch (cmd) {
case AUDIO_GETMAP:
bcopy((caddr_t)&sc->sc_map, addr, sizeof(sc->sc_map));
break;
case AUDIO_SETMAP:
bcopy(addr, (caddr_t)&sc->sc_map, sizeof(sc->sc_map));
sc->sc_map.mr_mmr2 &= 0x7f;
audio_setmap(sc->sc_au.au_amd, &sc->sc_map);
break;
case AUDIO_FLUSH:
s = splaudio();
AUCB_INIT(&sc->sc_au.au_rb);
AUCB_INIT(&sc->sc_au.au_wb);
sc->sc_au.au_stamp = 0;
splx(s);
sc->sc_wseek = 0;
sc->sc_rseek = 0;
break;
/*
* 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;
case AUDIO_SETINFO:
error = audiosetinfo(sc, (struct audio_info *)addr);
break;
case AUDIO_GETINFO:
error = audiogetinfo(sc, (struct audio_info *)addr);
break;
case SUNAUDIO_GETINFO:
error = sunaudiogetinfo(sc, (struct sun_audio_info *)addr);
break;
case SUNAUDIO_SETINFO:
error = sunaudiosetinfo(sc, (struct sun_audio_info *)addr);
break;
case AUDIO_DRAIN:
error = audio_drain(sc);
break;
default:
error = EINVAL;
break;
}
return (error);
}
/* ARGSUSED */
int
AUDIOSELECT(dev, rw, p)
{
register struct audio_softc *sc = SOFTC(dev);
register struct aucb *cb;
register int s = splaudio();
switch (rw) {
case FREAD:
cb = &sc->sc_au.au_rb;
if (AUCB_LEN(cb) >= sc->sc_au.au_blksize) {
splx(s);
return (1);
}
selrecord(p, &sc->sc_rsel);
cb->cb_thresh = sc->sc_au.au_blksize;
break;
case FWRITE:
cb = &sc->sc_au.au_wb;
if (AUCB_LEN(cb) <= sc->sc_au.au_lowat) {
splx(s);
return (1);
}
selrecord(p, &sc->sc_wsel);
cb->cb_thresh = sc->sc_au.au_lowat;
break;
}
splx(s);
return (0);
}
#ifdef AUDIO_C_HANDLER
int
audiohwintr(au0)
void *au0;
{
#ifdef SUNOS
register struct auio *au = audio_au;
#else
register struct auio *au = au0;
#endif
register volatile struct amd7930 *amd = au->au_amd;
register struct aucb *cb;
register int h, t, k;
k = amd->ir; /* clear interrupt */
++au->au_stamp;
/* receive incoming data */
cb = &au->au_rb;
h = cb->cb_head;
t = cb->cb_tail;
k = AUCB_MOD(t + 1);
if (h == k)
cb->cb_drops++;
else if (cb->cb_pause != 0)
cb->cb_pdrops++;
else {
cb->cb_data[t] = amd->bbrb;
cb->cb_tail = t = k;
}
if (AUCB_MOD(t - h) >= cb->cb_thresh) {
cb->cb_thresh = AUCB_SIZE;
cb->cb_waking = 1;
AUDIO_SET_SWINTR;
}
/* send outgoing data */
cb = &au->au_wb;
h = cb->cb_head;
t = cb->cb_tail;
if (h == t)
cb->cb_drops++;
else if (cb->cb_pause != 0)
cb->cb_pdrops++;
else {
cb->cb_head = h = AUCB_MOD(h + 1);
amd->bbtb = cb->cb_data[h];
}
if (AUCB_MOD(t - h) <= cb->cb_thresh) {
cb->cb_thresh = -1;
cb->cb_waking = 1;
AUDIO_SET_SWINTR;
}
return (1);
}
#endif
/* ARGSUSED */
int
audioswintr(sc0)
void *sc0;
{
register struct audio_softc *sc;
register int s, ret = 0;
#ifdef SUNOS
sc = &audio_softc;
#else
sc = sc0;
#endif
s = splaudio();
if (sc->sc_au.au_rb.cb_waking != 0) {
sc->sc_au.au_rb.cb_waking = 0;
splx(s);
ret = 1;
wakeup((caddr_t)&sc->sc_au.au_rb);
SELWAKEUP(&sc->sc_rsel);
}
if (sc->sc_au.au_wb.cb_waking != 0) {
sc->sc_au.au_wb.cb_waking = 0;
splx(s);
ret = 1;
wakeup((caddr_t)&sc->sc_au.au_wb);
SELWAKEUP(&sc->sc_wsel);
} else
splx(s);
return (ret);
}
/* Write 16 bits of data from variable v to the data port of the audio chip */
#define WAMD16(amd, v) ((amd)->dr = (v), (amd)->dr = (v) >> 8)
void
audio_setmap(amd, map)
register volatile struct amd7930 *amd;
register struct mapreg *map;
{
register int i, s, v;
s = splaudio();
amd->cr = AMDR_MAP_1_10;
for (i = 0; i < 8; i++) {
v = map->mr_x[i];
WAMD16(amd, v);
}
for (i = 0; i < 8; ++i) {
v = map->mr_r[i];
WAMD16(amd, v);
}
v = map->mr_gx; WAMD16(amd, v);
v = map->mr_gr; WAMD16(amd, v);
v = map->mr_ger; WAMD16(amd, v);
v = map->mr_stgr; WAMD16(amd, v);
v = map->mr_ftgr; WAMD16(amd, v);
v = map->mr_atgr; WAMD16(amd, v);
amd->dr = map->mr_mmr1;
amd->dr = map->mr_mmr2;
splx(s);
}
/*
* Set the mmr1 register and one other 16 bit register in the audio chip.
* The other register is indicated by op and val.
*/
void
audio_setmmr1(amd, mmr1, op, val)
register volatile struct amd7930 *amd;
register int mmr1;
register int op;
register int val;
{
register int s = splaudio();
amd->cr = AMDR_MAP_MMR1;
amd->dr = mmr1;
amd->cr = op;
WAMD16(amd, val);
splx(s);
}
/*
* Set the mmr2 register.
*/
static void
audio_setmmr2(amd, mmr2)
register volatile struct amd7930 *amd;
register int mmr2;
{
register int s = splaudio();
amd->cr = AMDR_MAP_MMR2;
amd->dr = mmr2;
splx(s);
}
/*
* gx, gr & stg gains. this table must contain 256 elements with
* the 0th being "infinity" (the magic value 9008). The remaining
* elements match sun's gain curve (but with higher resolution):
* -18 to 0dB in .16dB steps then 0 to 12dB in .08dB steps.
*/
static const u_short gx_coeff[256] = {
0x9008, 0x8b7c, 0x8b51, 0x8b45, 0x8b42, 0x8b3b, 0x8b36, 0x8b33,
0x8b32, 0x8b2a, 0x8b2b, 0x8b2c, 0x8b25, 0x8b23, 0x8b22, 0x8b22,
0x9122, 0x8b1a, 0x8aa3, 0x8aa3, 0x8b1c, 0x8aa6, 0x912d, 0x912b,
0x8aab, 0x8b12, 0x8aaa, 0x8ab2, 0x9132, 0x8ab4, 0x913c, 0x8abb,
0x9142, 0x9144, 0x9151, 0x8ad5, 0x8aeb, 0x8a79, 0x8a5a, 0x8a4a,
0x8b03, 0x91c2, 0x91bb, 0x8a3f, 0x8a33, 0x91b2, 0x9212, 0x9213,
0x8a2c, 0x921d, 0x8a23, 0x921a, 0x9222, 0x9223, 0x922d, 0x9231,
0x9234, 0x9242, 0x925b, 0x92dd, 0x92c1, 0x92b3, 0x92ab, 0x92a4,
0x92a2, 0x932b, 0x9341, 0x93d3, 0x93b2, 0x93a2, 0x943c, 0x94b2,
0x953a, 0x9653, 0x9782, 0x9e21, 0x9d23, 0x9cd2, 0x9c23, 0x9baa,
0x9bde, 0x9b33, 0x9b22, 0x9b1d, 0x9ab2, 0xa142, 0xa1e5, 0x9a3b,
0xa213, 0xa1a2, 0xa231, 0xa2eb, 0xa313, 0xa334, 0xa421, 0xa54b,
0xada4, 0xac23, 0xab3b, 0xaaab, 0xaa5c, 0xb1a3, 0xb2ca, 0xb3bd,
0xbe24, 0xbb2b, 0xba33, 0xc32b, 0xcb5a, 0xd2a2, 0xe31d, 0x0808,
0x72ba, 0x62c2, 0x5c32, 0x52db, 0x513e, 0x4cce, 0x43b2, 0x4243,
0x41b4, 0x3b12, 0x3bc3, 0x3df2, 0x34bd, 0x3334, 0x32c2, 0x3224,
0x31aa, 0x2a7b, 0x2aaa, 0x2b23, 0x2bba, 0x2c42, 0x2e23, 0x25bb,
0x242b, 0x240f, 0x231a, 0x22bb, 0x2241, 0x2223, 0x221f, 0x1a33,
0x1a4a, 0x1acd, 0x2132, 0x1b1b, 0x1b2c, 0x1b62, 0x1c12, 0x1c32,
0x1d1b, 0x1e71, 0x16b1, 0x1522, 0x1434, 0x1412, 0x1352, 0x1323,
0x1315, 0x12bc, 0x127a, 0x1235, 0x1226, 0x11a2, 0x1216, 0x0a2a,
0x11bc, 0x11d1, 0x1163, 0x0ac2, 0x0ab2, 0x0aab, 0x0b1b, 0x0b23,
0x0b33, 0x0c0f, 0x0bb3, 0x0c1b, 0x0c3e, 0x0cb1, 0x0d4c, 0x0ec1,
0x079a, 0x0614, 0x0521, 0x047c, 0x0422, 0x03b1, 0x03e3, 0x0333,
0x0322, 0x031c, 0x02aa, 0x02ba, 0x02f2, 0x0242, 0x0232, 0x0227,
0x0222, 0x021b, 0x01ad, 0x0212, 0x01b2, 0x01bb, 0x01cb, 0x01f6,
0x0152, 0x013a, 0x0133, 0x0131, 0x012c, 0x0123, 0x0122, 0x00a2,
0x011b, 0x011e, 0x0114, 0x00b1, 0x00aa, 0x00b3, 0x00bd, 0x00ba,
0x00c5, 0x00d3, 0x00f3, 0x0062, 0x0051, 0x0042, 0x003b, 0x0033,
0x0032, 0x002a, 0x002c, 0x0025, 0x0023, 0x0022, 0x001a, 0x0021,
0x001b, 0x001b, 0x001d, 0x0015, 0x0013, 0x0013, 0x0012, 0x0012,
0x000a, 0x000a, 0x0011, 0x0011, 0x000b, 0x000b, 0x000c, 0x000e,
};
/*
* second stage play gain.
*/
static const u_short ger_coeff[] = {
0x431f, /* 5. dB */
0x331f, /* 5.5 dB */
0x40dd, /* 6. dB */
0x11dd, /* 6.5 dB */
0x440f, /* 7. dB */
0x411f, /* 7.5 dB */
0x311f, /* 8. dB */
0x5520, /* 8.5 dB */
0x10dd, /* 9. dB */
0x4211, /* 9.5 dB */
0x410f, /* 10. dB */
0x111f, /* 10.5 dB */
0x600b, /* 11. dB */
0x00dd, /* 11.5 dB */
0x4210, /* 12. dB */
0x110f, /* 13. dB */
0x7200, /* 14. dB */
0x2110, /* 15. dB */
0x2200, /* 15.9 dB */
0x000b, /* 16.9 dB */
0x000f /* 18. dB */
#define NGER (sizeof(ger_coeff) / sizeof(ger_coeff[0]))
};
static void
ausetrgain(sc, level)
register struct audio_softc *sc;
register int level;
{
level &= 0xff;
sc->sc_rlevel = level;
sc->sc_map.mr_mmr1 |= AMD_MMR1_GX;
sc->sc_map.mr_gx = gx_coeff[level];
audio_setmmr1(sc->sc_au.au_amd, sc->sc_map.mr_mmr1,
AMDR_MAP_GX, sc->sc_map.mr_gx);
}
static void
ausetpgain(sc, level)
register struct audio_softc *sc;
register int level;
{
register int gi, s;
register volatile struct amd7930 *amd;
level &= 0xff;
sc->sc_plevel = level;
sc->sc_map.mr_mmr1 |= AMD_MMR1_GER|AMD_MMR1_GR;
level *= 256 + NGER;
level >>= 8;
if (level >= 256) {
gi = level - 256;
level = 255;
} else
gi = 0;
sc->sc_map.mr_ger = ger_coeff[gi];
sc->sc_map.mr_gr = gx_coeff[level];
amd = sc->sc_au.au_amd;
s = splaudio();
amd->cr = AMDR_MAP_MMR1;
amd->dr = sc->sc_map.mr_mmr1;
amd->cr = AMDR_MAP_GR;
gi = sc->sc_map.mr_gr;
WAMD16(amd, gi);
amd->cr = AMDR_MAP_GER;
gi = sc->sc_map.mr_ger;
WAMD16(amd, gi);
splx(s);
}
static void
ausetmgain(sc, level)
register struct audio_softc *sc;
register int level;
{
level &= 0xff;
sc->sc_mlevel = level;
sc->sc_map.mr_mmr1 |= AMD_MMR1_STG;
sc->sc_map.mr_stgr = gx_coeff[level];
audio_setmmr1(sc->sc_au.au_amd, sc->sc_map.mr_mmr1,
AMDR_MAP_STG, sc->sc_map.mr_stgr);
}
static int
audiosetinfo(sc, ai)
struct audio_softc *sc;
struct audio_info *ai;
{
struct audio_prinfo *r = &ai->record, *p = &ai->play;
register int s, bsize;
if (p->gain != ~0)
ausetpgain(sc, p->gain);
if (r->gain != ~0)
ausetrgain(sc, r->gain);
if (ai->monitor_gain != ~0)
ausetmgain(sc, ai->monitor_gain);
if (p->port == AUDIO_SPEAKER) {
sc->sc_map.mr_mmr2 |= AMD_MMR2_LS;
audio_setmmr2(sc->sc_au.au_amd, sc->sc_map.mr_mmr2);
} else if (p->port == AUDIO_HEADPHONE) {
sc->sc_map.mr_mmr2 &=~ AMD_MMR2_LS;
audio_setmmr2(sc->sc_au.au_amd, sc->sc_map.mr_mmr2);
}
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;
if (ai->blocksize != ~0) {
if (ai->blocksize == 0)
bsize = ai->blocksize = DEFBLKSIZE;
else if (ai->blocksize > MAXBLKSIZE)
bsize = ai->blocksize = MAXBLKSIZE;
else
bsize = ai->blocksize;
s = splaudio();
sc->sc_au.au_blksize = bsize;
/* AUDIO_FLUSH */
AUCB_INIT(&sc->sc_au.au_rb);
AUCB_INIT(&sc->sc_au.au_wb);
splx(s);
}
if (ai->hiwat != ~0 && (unsigned)ai->hiwat < AUCB_SIZE)
sc->sc_au.au_hiwat = ai->hiwat;
if (ai->lowat != ~0 && ai->lowat < AUCB_SIZE)
sc->sc_au.au_lowat = ai->lowat;
if (ai->backlog != ~0 && ai->backlog < (AUCB_SIZE/2))
sc->sc_au.au_backlog = ai->backlog;
return (0);
}
static int
sunaudiosetinfo(sc, ai)
struct audio_softc *sc;
struct sun_audio_info *ai;
{
struct sun_audio_prinfo *r = &ai->record, *p = &ai->play;
if (p->gain != ~0)
ausetpgain(sc, p->gain);
if (r->gain != ~0)
ausetrgain(sc, r->gain);
if (ai->monitor_gain != ~0)
ausetmgain(sc, ai->monitor_gain);
if (p->port == AUDIO_SPEAKER) {
sc->sc_map.mr_mmr2 |= AMD_MMR2_LS;
audio_setmmr2(sc->sc_au.au_amd, sc->sc_map.mr_mmr2);
} else if (p->port == AUDIO_HEADPHONE) {
sc->sc_map.mr_mmr2 &=~ AMD_MMR2_LS;
audio_setmmr2(sc->sc_au.au_amd, sc->sc_map.mr_mmr2);
}
/*
* The bsd driver does not distinguish between paused and active.
* (In the sun driver, not active means samples are not ouput
* at all, but paused means the last streams buffer is drained
* and then output stops.) If either are 0, then when stop output.
* Otherwise, if either are non-zero, we resume.
*/
if (p->pause == 0 || p->active == 0)
sc->sc_au.au_wb.cb_pause = 0;
else if (p->pause != (u_char)~0 || p->active != (u_char)~0)
sc->sc_au.au_wb.cb_pause = 1;
if (r->pause == 0 || r->active == 0)
sc->sc_au.au_rb.cb_pause = 0;
else if (r->pause != (u_char)~0 || r->active != (u_char)~0)
sc->sc_au.au_rb.cb_pause = 1;
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 = r->sample_rate = 8000;
p->channels = r->channels = 1;
p->precision = r->precision = 8;
p->encoding = r->encoding = AUDIO_ENCODING_ULAW;
ai->monitor_gain = sc->sc_mlevel;
r->gain = sc->sc_rlevel;
p->gain = sc->sc_plevel;
r->port = 1; p->port = (sc->sc_map.mr_mmr2 & AMD_MMR2_LS) ?
AUDIO_SPEAKER : AUDIO_HEADPHONE;
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;
p->open = sc->sc_open;
r->open = sc->sc_open;
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;
p->seek = sc->sc_wseek;
r->seek = sc->sc_rseek;
ai->blocksize = sc->sc_au.au_blksize;
ai->hiwat = sc->sc_au.au_hiwat;
ai->lowat = sc->sc_au.au_lowat;
ai->backlog = sc->sc_au.au_backlog;
return (0);
}
static int
sunaudiogetinfo(sc, ai)
struct audio_softc *sc;
struct sun_audio_info *ai;
{
struct sun_audio_prinfo *r = &ai->record, *p = &ai->play;
p->sample_rate = r->sample_rate = 8000;
p->channels = r->channels = 1;
p->precision = r->precision = 8;
p->encoding = r->encoding = AUDIO_ENCODING_ULAW;
ai->monitor_gain = sc->sc_mlevel;
r->gain = sc->sc_rlevel;
p->gain = sc->sc_plevel;
r->port = 1; p->port = (sc->sc_map.mr_mmr2 & AMD_MMR2_LS) ?
AUDIO_SPEAKER : AUDIO_HEADPHONE;
p->active = p->pause = sc->sc_au.au_wb.cb_pause;
r->active = 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;
p->waiting = 0;
r->waiting = 0;
p->eof = 0;
r->eof = 0;
p->open = sc->sc_open;
r->open = sc->sc_open;
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;
return (0);
}
#endif
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