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NetBSD/sys/dev/audio.c

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/* $NetBSD: audio.c,v 1.169 2002/11/26 18:49:40 christos Exp $ */
/*
* 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.
*/
/*
* This is a (partially) SunOS-compatible /dev/audio driver for NetBSD.
*
* This code tries to do something half-way sensible with
* half-duplex hardware, such as with the SoundBlaster hardware. With
* half-duplex hardware allowing O_RDWR access doesn't really make
* sense. However, closing and opening the device to "turn around the
* line" is relatively expensive and costs a card reset (which can
* take some time, at least for the SoundBlaster hardware). Instead
* we allow O_RDWR access, and provide an ioctl to set the "mode",
* i.e. playing or recording.
*
* If you write to a half-duplex device in record mode, the data is
* tossed. If you read from the device in play mode, you get silence
* filled buffers at the rate at which samples are naturally
* generated.
*
* If you try to set both play and record mode on a half-duplex
* device, playing takes precedence.
*/
/*
* Todo:
* - Add softaudio() isr processing for wakeup, poll, signals,
* and silence fill.
*/
#include <sys/cdefs.h>
__KERNEL_RCSID(0, "$NetBSD: audio.c,v 1.169 2002/11/26 18:49:40 christos Exp $");
#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/poll.h>
#include <sys/malloc.h>
#include <sys/proc.h>
#include <sys/systm.h>
#include <sys/syslog.h>
#include <sys/kernel.h>
#include <sys/signalvar.h>
#include <sys/conf.h>
#include <sys/audioio.h>
#include <sys/device.h>
#include <dev/audio_if.h>
#include <dev/audiovar.h>
#include <machine/endian.h>
#ifdef AUDIO_DEBUG
#define DPRINTF(x) if (audiodebug) printf x
#define DPRINTFN(n,x) if (audiodebug>(n)) printf x
int audiodebug = AUDIO_DEBUG;
#else
#define DPRINTF(x)
#define DPRINTFN(n,x)
#endif
#define ROUNDSIZE(x) x &= -16 /* round to nice boundary */
int audio_blk_ms = AUDIO_BLK_MS;
int audiosetinfo(struct audio_softc *, struct audio_info *);
int audiogetinfo(struct audio_softc *, struct audio_info *);
int audio_open(dev_t, struct audio_softc *, int, int, struct proc *);
int audio_close(struct audio_softc *, int, int, struct proc *);
int audio_read(struct audio_softc *, struct uio *, int);
int audio_write(struct audio_softc *, struct uio *, int);
int audio_ioctl(struct audio_softc *, u_long, caddr_t, int, struct proc *);
int audio_poll(struct audio_softc *, int, struct proc *);
int audio_kqfilter(struct audio_softc *, struct knote *);
paddr_t audio_mmap(struct audio_softc *, off_t, int);
int mixer_open(dev_t, struct audio_softc *, int, int, struct proc *);
int mixer_close(struct audio_softc *, int, int, struct proc *);
int mixer_ioctl(struct audio_softc *, u_long, caddr_t, int, struct proc *);
static void mixer_remove(struct audio_softc *, struct proc *p);
static void mixer_signal(struct audio_softc *);
void audio_init_record(struct audio_softc *);
void audio_init_play(struct audio_softc *);
int audiostartr(struct audio_softc *);
int audiostartp(struct audio_softc *);
void audio_rint(void *);
void audio_pint(void *);
int audio_check_params(struct audio_params *);
void audio_calc_blksize(struct audio_softc *, int);
void audio_fill_silence(struct audio_params *, u_char *, int);
int audio_silence_copyout(struct audio_softc *, int, struct uio *);
void audio_init_ringbuffer(struct audio_ringbuffer *);
int audio_initbufs(struct audio_softc *);
void audio_calcwater(struct audio_softc *);
static __inline int audio_sleep_timo(int *, char *, int);
static __inline int audio_sleep(int *, char *);
static __inline void audio_wakeup(int *);
int audio_drain(struct audio_softc *);
void audio_clear(struct audio_softc *);
static __inline void audio_pint_silence
(struct audio_softc *, struct audio_ringbuffer *, u_char *, int);
int audio_alloc_ring
(struct audio_softc *, struct audio_ringbuffer *, int, size_t);
void audio_free_ring(struct audio_softc *, struct audio_ringbuffer *);
int audioprobe(struct device *, struct cfdata *, void *);
void audioattach(struct device *, struct device *, void *);
int audiodetach(struct device *, int);
int audioactivate(struct device *, enum devact);
struct portname {
char *name;
int mask;
};
static struct portname itable[] = {
{ AudioNmicrophone, AUDIO_MICROPHONE },
{ AudioNline, AUDIO_LINE_IN },
{ AudioNcd, AUDIO_CD },
{ 0 }
};
static struct portname otable[] = {
{ AudioNspeaker, AUDIO_SPEAKER },
{ AudioNheadphone, AUDIO_HEADPHONE },
{ AudioNline, AUDIO_LINE_OUT },
{ 0 }
};
void au_check_ports(struct audio_softc *, struct au_mixer_ports *,
mixer_devinfo_t *, int, char *, char *,
struct portname *);
int au_set_gain(struct audio_softc *, struct au_mixer_ports *,
int, int);
void au_get_gain(struct audio_softc *, struct au_mixer_ports *,
u_int *, u_char *);
int au_set_port(struct audio_softc *, struct au_mixer_ports *,
u_int);
int au_get_port(struct audio_softc *, struct au_mixer_ports *);
int au_get_lr_value(struct audio_softc *, mixer_ctrl_t *,
int *, int *r);
int au_set_lr_value(struct audio_softc *, mixer_ctrl_t *,
int, int);
int au_portof(struct audio_softc *, char *);
dev_type_open(audioopen);
dev_type_close(audioclose);
dev_type_read(audioread);
dev_type_write(audiowrite);
dev_type_ioctl(audioioctl);
dev_type_poll(audiopoll);
dev_type_mmap(audiommap);
dev_type_kqfilter(audiokqfilter);
const struct cdevsw audio_cdevsw = {
audioopen, audioclose, audioread, audiowrite, audioioctl,
nostop, notty, audiopoll, audiommap, audiokqfilter,
};
/* The default audio mode: 8 kHz mono ulaw */
struct audio_params audio_default =
{ 8000, AUDIO_ENCODING_ULAW, 8, 1, 0, 1, 1 };
CFATTACH_DECL(audio, sizeof(struct audio_softc),
audioprobe, audioattach, audiodetach, audioactivate);
extern struct cfdriver audio_cd;
int
audioprobe(struct device *parent, struct cfdata *match, void *aux)
{
struct audio_attach_args *sa = aux;
DPRINTF(("audioprobe: type=%d sa=%p hw=%p\n",
sa->type, sa, sa->hwif));
return (sa->type == AUDIODEV_TYPE_AUDIO) ? 1 : 0;
}
void
audioattach(struct device *parent, struct device *self, void *aux)
{
struct audio_softc *sc = (void *)self;
struct audio_attach_args *sa = aux;
struct audio_hw_if *hwp = sa->hwif;
void *hdlp = sa->hdl;
int error;
mixer_devinfo_t mi;
int iclass, oclass, props;
#ifdef DIAGNOSTIC
if (hwp == 0 ||
hwp->open == 0 ||
hwp->close == 0 ||
hwp->query_encoding == 0 ||
hwp->set_params == 0 ||
(hwp->start_output == 0 && hwp->trigger_output == 0) ||
(hwp->start_input == 0 && hwp->trigger_input == 0) ||
hwp->halt_output == 0 ||
hwp->halt_input == 0 ||
hwp->getdev == 0 ||
hwp->set_port == 0 ||
hwp->get_port == 0 ||
hwp->query_devinfo == 0 ||
hwp->get_props == 0) {
printf(": missing method\n");
sc->hw_if = 0;
return;
}
#endif
props = hwp->get_props(hdlp);
if (props & AUDIO_PROP_FULLDUPLEX)
printf(": full duplex");
else
printf(": half duplex");
if (props & AUDIO_PROP_MMAP)
printf(", mmap");
if (props & AUDIO_PROP_INDEPENDENT)
printf(", independent");
printf("\n");
sc->hw_if = hwp;
sc->hw_hdl = hdlp;
sc->sc_dev = parent;
error = audio_alloc_ring(sc, &sc->sc_pr, AUMODE_PLAY, AU_RING_SIZE);
if (error) {
sc->hw_if = 0;
printf("audio: could not allocate play buffer\n");
return;
}
error = audio_alloc_ring(sc, &sc->sc_rr, AUMODE_RECORD, AU_RING_SIZE);
if (error) {
audio_free_ring(sc, &sc->sc_pr);
sc->hw_if = 0;
printf("audio: could not allocate record buffer\n");
return;
}
sc->sc_pconvbuffer = malloc(AU_RING_SIZE, M_DEVBUF, M_WAITOK);
sc->sc_pconvbuffer_size = AU_RING_SIZE;
sc->sc_rconvbuffer = malloc(AU_RING_SIZE, M_DEVBUF, M_WAITOK);
sc->sc_rconvbuffer_size = AU_RING_SIZE;
/*
* Set default softc params
*/
sc->sc_pparams = audio_default;
sc->sc_rparams = audio_default;
/* Set up some default values */
sc->sc_blkset = 0;
audio_calc_blksize(sc, AUMODE_RECORD);
audio_calc_blksize(sc, AUMODE_PLAY);
audio_init_ringbuffer(&sc->sc_rr);
audio_init_ringbuffer(&sc->sc_pr);
audio_calcwater(sc);
sc->sc_input_fragment_length = 0;
iclass = oclass = -1;
sc->sc_inports.index = -1;
sc->sc_inports.master = -1;
sc->sc_inports.nports = 0;
sc->sc_inports.isenum = 0;
sc->sc_inports.allports = 0;
sc->sc_outports.index = -1;
sc->sc_outports.master = -1;
sc->sc_outports.nports = 0;
sc->sc_outports.isenum = 0;
sc->sc_outports.allports = 0;
sc->sc_monitor_port = -1;
for(mi.index = 0; ; mi.index++) {
if (hwp->query_devinfo(hdlp, &mi) != 0)
break;
if (mi.type == AUDIO_MIXER_CLASS &&
strcmp(mi.label.name, AudioCrecord) == 0)
iclass = mi.index;
if (mi.type == AUDIO_MIXER_CLASS &&
strcmp(mi.label.name, AudioCmonitor) == 0)
oclass = mi.index;
}
for(mi.index = 0; ; mi.index++) {
if (hwp->query_devinfo(hdlp, &mi) != 0)
break;
if (mi.type == AUDIO_MIXER_CLASS)
continue;
au_check_ports(sc, &sc->sc_inports, &mi, iclass,
AudioNsource, AudioNrecord, itable);
au_check_ports(sc, &sc->sc_outports, &mi, oclass,
AudioNoutput, AudioNmaster, otable);
if (mi.mixer_class == oclass &&
(strcmp(mi.label.name, AudioNmonitor) == 0))
sc->sc_monitor_port = mi.index;
}
DPRINTF(("audio_attach: inputs ports=0x%x, input master=%d, "
"output ports=0x%x, output master=%d\n",
sc->sc_inports.allports, sc->sc_inports.master,
sc->sc_outports.allports, sc->sc_outports.master));
}
int
audioactivate(struct device *self, enum devact act)
{
struct audio_softc *sc = (struct audio_softc *)self;
switch (act) {
case DVACT_ACTIVATE:
return (EOPNOTSUPP);
case DVACT_DEACTIVATE:
sc->sc_dying = 1;
break;
}
return (0);
}
int
audiodetach(struct device *self, int flags)
{
struct audio_softc *sc = (struct audio_softc *)self;
int maj, mn;
int s;
DPRINTF(("audio_detach: sc=%p flags=%d\n", sc, flags));
sc->sc_dying = 1;
wakeup(&sc->sc_wchan);
wakeup(&sc->sc_rchan);
s = splaudio();
if (--sc->sc_refcnt >= 0) {
if (tsleep(&sc->sc_refcnt, PZERO, "auddet", hz * 120))
printf("audiodetach: %s didn't detach\n",
sc->dev.dv_xname);
}
splx(s);
/* free resources */
audio_free_ring(sc, &sc->sc_pr);
audio_free_ring(sc, &sc->sc_rr);
free(sc->sc_pconvbuffer, M_DEVBUF);
free(sc->sc_rconvbuffer, M_DEVBUF);
/* locate the major number */
maj = cdevsw_lookup_major(&audio_cdevsw);
/* Nuke the vnodes for any open instances (calls close). */
mn = self->dv_unit;
vdevgone(maj, mn | SOUND_DEVICE, mn | SOUND_DEVICE, VCHR);
vdevgone(maj, mn | AUDIO_DEVICE, mn | AUDIO_DEVICE, VCHR);
vdevgone(maj, mn | AUDIOCTL_DEVICE, mn | AUDIOCTL_DEVICE, VCHR);
vdevgone(maj, mn | MIXER_DEVICE, mn | MIXER_DEVICE, VCHR);
return (0);
}
int
au_portof(struct audio_softc *sc, char *name)
{
mixer_devinfo_t mi;
for(mi.index = 0;
sc->hw_if->query_devinfo(sc->hw_hdl, &mi) == 0;
mi.index++)
if (strcmp(mi.label.name, name) == 0)
return mi.index;
return -1;
}
void
au_check_ports(struct audio_softc *sc, struct au_mixer_ports *ports,
mixer_devinfo_t *mi, int cls, char *name, char *mname,
struct portname *tbl)
{
int i, j;
if (mi->mixer_class != cls)
return;
if (strcmp(mi->label.name, mname) == 0) {
ports->master = mi->index;
return;
}
if (strcmp(mi->label.name, name) != 0)
return;
if (mi->type == AUDIO_MIXER_ENUM) {
ports->index = mi->index;
for(i = 0; tbl[i].name; i++) {
for(j = 0; j < mi->un.e.num_mem; j++) {
if (strcmp(mi->un.e.member[j].label.name,
tbl[i].name) == 0) {
ports->aumask[ports->nports] = tbl[i].mask;
ports->misel [ports->nports] = mi->un.e.member[j].ord;
ports->miport[ports->nports++] =
au_portof(sc, mi->un.e.member[j].label.name);
ports->allports |= tbl[i].mask;
}
}
}
ports->isenum = 1;
} else if (mi->type == AUDIO_MIXER_SET) {
ports->index = mi->index;
for(i = 0; tbl[i].name; i++) {
for(j = 0; j < mi->un.s.num_mem; j++) {
if (strcmp(mi->un.s.member[j].label.name,
tbl[i].name) == 0) {
ports->aumask[ports->nports] = tbl[i].mask;
ports->misel [ports->nports] = mi->un.s.member[j].mask;
ports->miport[ports->nports++] =
au_portof(sc, mi->un.s.member[j].label.name);
ports->allports |= tbl[i].mask;
}
}
}
}
}
/*
* Called from hardware driver. This is where the MI audio driver gets
* probed/attached to the hardware driver.
*/
struct device *
audio_attach_mi(struct audio_hw_if *ahwp, void *hdlp, struct device *dev)
{
struct audio_attach_args arg;
#ifdef DIAGNOSTIC
if (ahwp == NULL) {
printf("audio_attach_mi: NULL\n");
return (0);
}
#endif
arg.type = AUDIODEV_TYPE_AUDIO;
arg.hwif = ahwp;
arg.hdl = hdlp;
return (config_found(dev, &arg, audioprint));
}
#ifdef AUDIO_DEBUG
void audio_printsc(struct audio_softc *);
void audio_print_params(char *, struct audio_params *);
void
audio_printsc(struct audio_softc *sc)
{
printf("hwhandle %p hw_if %p ", sc->hw_hdl, sc->hw_if);
printf("open 0x%x mode 0x%x\n", sc->sc_open, sc->sc_mode);
printf("rchan 0x%x wchan 0x%x ", sc->sc_rchan, sc->sc_wchan);
printf("rring used 0x%x pring used=%d\n",
sc->sc_rr.used, sc->sc_pr.used);
printf("rbus 0x%x pbus 0x%x ", sc->sc_rbus, sc->sc_pbus);
printf("blksize %d", sc->sc_pr.blksize);
printf("hiwat %d lowat %d\n", sc->sc_pr.usedhigh, sc->sc_pr.usedlow);
}
void
audio_print_params(char *s, struct audio_params *p)
{
printf("audio: %s sr=%ld, enc=%d, chan=%d, prec=%d\n", s,
p->sample_rate, p->encoding, p->channels, p->precision);
}
#endif
int
audio_alloc_ring(struct audio_softc *sc, struct audio_ringbuffer *r,
int direction, size_t bufsize)
{
struct audio_hw_if *hw = sc->hw_if;
void *hdl = sc->hw_hdl;
/*
* Alloc DMA play and record buffers
*/
if (bufsize < AUMINBUF)
bufsize = AUMINBUF;
ROUNDSIZE(bufsize);
if (hw->round_buffersize)
bufsize = hw->round_buffersize(hdl, direction, bufsize);
if (hw->allocm)
r->start = hw->allocm(hdl, direction, bufsize,
M_DEVBUF, M_WAITOK);
else
r->start = malloc(bufsize, M_DEVBUF, M_WAITOK);
if (r->start == 0)
return ENOMEM;
r->bufsize = bufsize;
return 0;
}
void
audio_free_ring(struct audio_softc *sc, struct audio_ringbuffer *r)
{
if (sc->hw_if->freem)
sc->hw_if->freem(sc->hw_hdl, r->start, M_DEVBUF);
else
free(r->start, M_DEVBUF);
r->start = 0;
}
int
audioopen(dev_t dev, int flags, int ifmt, struct proc *p)
{
struct audio_softc *sc;
int error;
sc = device_lookup(&audio_cd, AUDIOUNIT(dev));
if (sc == NULL)
return (ENXIO);
if (sc->sc_dying)
return (EIO);
sc->sc_refcnt++;
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_open(dev, sc, flags, ifmt, p);
break;
case AUDIOCTL_DEVICE:
error = 0;
break;
case MIXER_DEVICE:
error = mixer_open(dev, sc, flags, ifmt, p);
break;
default:
error = ENXIO;
break;
}
if (--sc->sc_refcnt < 0)
wakeup(&sc->sc_refcnt);
return (error);
}
int
audioclose(dev_t dev, int flags, int ifmt, struct proc *p)
{
int unit = AUDIOUNIT(dev);
struct audio_softc *sc = audio_cd.cd_devs[unit];
int error;
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_close(sc, flags, ifmt, p);
break;
case MIXER_DEVICE:
error = mixer_close(sc, flags, ifmt, p);
break;
case AUDIOCTL_DEVICE:
error = 0;
break;
default:
error = ENXIO;
break;
}
return (error);
}
int
audioread(dev_t dev, struct uio *uio, int ioflag)
{
struct audio_softc *sc;
int error;
sc = device_lookup(&audio_cd, AUDIOUNIT(dev));
if (sc == NULL)
return (ENXIO);
if (sc->sc_dying)
return (EIO);
sc->sc_refcnt++;
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_read(sc, uio, ioflag);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
error = ENODEV;
break;
default:
error = ENXIO;
break;
}
if (--sc->sc_refcnt < 0)
wakeup(&sc->sc_refcnt);
return (error);
}
int
audiowrite(dev_t dev, struct uio *uio, int ioflag)
{
struct audio_softc *sc;
int error;
sc = device_lookup(&audio_cd, AUDIOUNIT(dev));
if (sc == NULL)
return (ENXIO);
if (sc->sc_dying)
return (EIO);
sc->sc_refcnt++;
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_write(sc, uio, ioflag);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
error = ENODEV;
break;
default:
error = ENXIO;
break;
}
if (--sc->sc_refcnt < 0)
wakeup(&sc->sc_refcnt);
return (error);
}
int
audioioctl(dev_t dev, u_long cmd, caddr_t addr, int flag, struct proc *p)
{
int unit = AUDIOUNIT(dev);
struct audio_softc *sc = audio_cd.cd_devs[unit];
int error;
if (sc->sc_dying)
return (EIO);
sc->sc_refcnt++;
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
case AUDIOCTL_DEVICE:
error = audio_ioctl(sc, cmd, addr, flag, p);
break;
case MIXER_DEVICE:
error = mixer_ioctl(sc, cmd, addr, flag, p);
break;
default:
error = ENXIO;
break;
}
if (--sc->sc_refcnt < 0)
wakeup(&sc->sc_refcnt);
return (error);
}
int
audiopoll(dev_t dev, int events, struct proc *p)
{
int unit = AUDIOUNIT(dev);
struct audio_softc *sc = audio_cd.cd_devs[unit];
int error;
if (sc->sc_dying)
return (EIO);
sc->sc_refcnt++;
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_poll(sc, events, p);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
error = ENODEV;
break;
default:
error = ENXIO;
break;
}
if (--sc->sc_refcnt < 0)
wakeup(&sc->sc_refcnt);
return (error);
}
int
audiokqfilter(dev_t dev, struct knote *kn)
{
int unit = AUDIOUNIT(dev);
struct audio_softc *sc = audio_cd.cd_devs[unit];
int rv;
if (sc->sc_dying)
return (1);
sc->sc_refcnt++;
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
rv = audio_kqfilter(sc, kn);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
rv = 1;
break;
default:
rv = 1;
}
if (--sc->sc_refcnt < 0)
wakeup(&sc->sc_refcnt);
return (rv);
}
paddr_t
audiommap(dev_t dev, off_t off, int prot)
{
int unit = AUDIOUNIT(dev);
struct audio_softc *sc = audio_cd.cd_devs[unit];
paddr_t error;
if (sc->sc_dying)
return (-1);
sc->sc_refcnt++;
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
error = audio_mmap(sc, off, prot);
break;
case AUDIOCTL_DEVICE:
case MIXER_DEVICE:
error = -1;
break;
default:
error = -1;
break;
}
if (--sc->sc_refcnt < 0)
wakeup(&sc->sc_refcnt);
return (error);
}
/*
* Audio driver
*/
void
audio_init_ringbuffer(struct audio_ringbuffer *rp)
{
int nblks;
int blksize = rp->blksize;
if (blksize < AUMINBLK)
blksize = AUMINBLK;
nblks = rp->bufsize / blksize;
if (nblks < AUMINNOBLK) {
nblks = AUMINNOBLK;
blksize = rp->bufsize / nblks;
ROUNDSIZE(blksize);
}
DPRINTF(("audio_init_ringbuffer: blksize=%d\n", blksize));
rp->blksize = blksize;
rp->maxblks = nblks;
rp->used = 0;
rp->end = rp->start + nblks * blksize;
rp->inp = rp->outp = rp->start;
rp->stamp = 0;
rp->drops = 0;
rp->pause = 0;
rp->copying = 0;
rp->needfill = 0;
rp->mmapped = 0;
}
int
audio_initbufs(struct audio_softc *sc)
{
struct audio_hw_if *hw = sc->hw_if;
int error;
DPRINTF(("audio_initbufs: mode=0x%x\n", sc->sc_mode));
audio_init_ringbuffer(&sc->sc_rr);
#if NAURATECONV > 0
auconv_init_context(&sc->sc_rconv, sc->sc_rparams.hw_sample_rate,
sc->sc_rparams.sample_rate,
sc->sc_rr.start, sc->sc_rr.end);
#endif
if (hw->init_input && (sc->sc_mode & AUMODE_RECORD)) {
error = hw->init_input(sc->hw_hdl, sc->sc_rr.start,
sc->sc_rr.end - sc->sc_rr.start);
if (error)
return error;
}
audio_init_ringbuffer(&sc->sc_pr);
#if NAURATECONV > 0
auconv_init_context(&sc->sc_pconv, sc->sc_pparams.sample_rate,
sc->sc_pparams.hw_sample_rate,
sc->sc_pr.start, sc->sc_pr.end);
#endif
sc->sc_sil_count = 0;
if (hw->init_output && (sc->sc_mode & AUMODE_PLAY)) {
error = hw->init_output(sc->hw_hdl, sc->sc_pr.start,
sc->sc_pr.end - sc->sc_pr.start);
if (error)
return error;
}
#ifdef AUDIO_INTR_TIME
#define double u_long
sc->sc_pnintr = 0;
sc->sc_pblktime = (u_long)(
(double)sc->sc_pr.blksize * 100000 /
(double)(sc->sc_pparams.precision / NBBY *
sc->sc_pparams.channels *
sc->sc_pparams.sample_rate)) * 10;
DPRINTF(("audio: play blktime = %lu for %d\n",
sc->sc_pblktime, sc->sc_pr.blksize));
sc->sc_rnintr = 0;
sc->sc_rblktime = (u_long)(
(double)sc->sc_rr.blksize * 100000 /
(double)(sc->sc_rparams.precision / NBBY *
sc->sc_rparams.channels *
sc->sc_rparams.sample_rate)) * 10;
DPRINTF(("audio: record blktime = %lu for %d\n",
sc->sc_rblktime, sc->sc_rr.blksize));
#undef double
#endif
return 0;
}
void
audio_calcwater(struct audio_softc *sc)
{
sc->sc_pr.usedhigh = sc->sc_pr.end - sc->sc_pr.start;
sc->sc_pr.usedlow = sc->sc_pr.usedhigh * 3 / 4; /* set low at 75% */
if (sc->sc_pr.usedlow == sc->sc_pr.usedhigh)
sc->sc_pr.usedlow -= sc->sc_pr.blksize;
sc->sc_rr.usedhigh =
sc->sc_pr.end - sc->sc_pr.start - sc->sc_pr.blksize;
sc->sc_rr.usedlow = 0;
}
static __inline int
audio_sleep_timo(int *chan, char *label, int timo)
{
int st;
if (!label)
label = "audio";
DPRINTFN(3, ("audio_sleep_timo: chan=%p, label=%s, timo=%d\n",
chan, label, timo));
*chan = 1;
st = tsleep(chan, PWAIT | PCATCH, label, timo);
*chan = 0;
#ifdef AUDIO_DEBUG
if (st != 0 && st != EINTR)
DPRINTF(("audio_sleep: woke up st=%d\n", st));
#endif
return (st);
}
static __inline int
audio_sleep(int *chan, char *label)
{
return audio_sleep_timo(chan, label, 0);
}
/* call at splaudio() */
static __inline void
audio_wakeup(int *chan)
{
DPRINTFN(3, ("audio_wakeup: chan=%p, *chan=%d\n", chan, *chan));
if (*chan) {
wakeup(chan);
*chan = 0;
}
}
int
audio_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt,
struct proc *p)
{
int error;
int mode;
struct audio_hw_if *hw;
struct audio_info ai;
hw = sc->hw_if;
if (!hw)
return ENXIO;
DPRINTF(("audio_open: flags=0x%x sc=%p hdl=%p\n",
flags, sc, sc->hw_hdl));
if ((sc->sc_open & (AUOPEN_READ|AUOPEN_WRITE)) != 0)
return (EBUSY);
error = hw->open(sc->hw_hdl, flags);
if (error)
return (error);
sc->sc_async_audio = 0;
sc->sc_rchan = 0;
sc->sc_wchan = 0;
sc->sc_blkset = 0; /* Block sizes not set yet */
sc->sc_sil_count = 0;
sc->sc_rbus = 0;
sc->sc_pbus = 0;
sc->sc_eof = 0;
sc->sc_playdrop = 0;
sc->sc_full_duplex = 0;
/* doesn't always work right on SB.
(flags & (FWRITE|FREAD)) == (FWRITE|FREAD) &&
(hw->get_props(sc->hw_hdl) & AUDIO_PROP_FULLDUPLEX);
*/
mode = 0;
if (flags & FREAD) {
sc->sc_open |= AUOPEN_READ;
mode |= AUMODE_RECORD;
}
if (flags & FWRITE) {
sc->sc_open |= AUOPEN_WRITE;
mode |= AUMODE_PLAY | AUMODE_PLAY_ALL;
}
/*
* Multiplex device: /dev/audio (MU-Law) and /dev/sound (linear)
* The /dev/audio is always (re)set to 8-bit MU-Law mono
* For the other devices, you get what they were last set to.
*/
if (ISDEVAUDIO(dev)) {
/* /dev/audio */
sc->sc_rparams = audio_default;
sc->sc_pparams = audio_default;
}
#ifdef DIAGNOSTIC
/*
* Sample rate and precision are supposed to be set to proper
* default values by the hardware driver, so that it may give
* us these values.
*/
if (sc->sc_rparams.precision == 0 || sc->sc_pparams.precision == 0) {
printf("audio_open: 0 precision\n");
return EINVAL;
}
#endif
AUDIO_INITINFO(&ai);
ai.record.sample_rate = sc->sc_rparams.sample_rate;
ai.record.encoding = sc->sc_rparams.encoding;
ai.record.channels = sc->sc_rparams.channels;
ai.record.precision = sc->sc_rparams.precision;
ai.play.sample_rate = sc->sc_pparams.sample_rate;
ai.play.encoding = sc->sc_pparams.encoding;
ai.play.channels = sc->sc_pparams.channels;
ai.play.precision = sc->sc_pparams.precision;
ai.mode = mode;
error = audiosetinfo(sc, &ai);
if (error)
goto bad;
/* audio_close() decreases sc_pr.usedlow, recalculate here */
audio_calcwater(sc);
DPRINTF(("audio_open: done sc_mode = 0x%x\n", sc->sc_mode));
return 0;
bad:
hw->close(sc->hw_hdl);
sc->sc_open = 0;
sc->sc_mode = 0;
sc->sc_full_duplex = 0;
return error;
}
/*
* Must be called from task context.
*/
void
audio_init_record(struct audio_softc *sc)
{
int s = splaudio();
if (sc->hw_if->speaker_ctl &&
(!sc->sc_full_duplex || (sc->sc_mode & AUMODE_PLAY) == 0))
sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_OFF);
sc->sc_rconvbuffer_begin = 0;
sc->sc_rconvbuffer_end = 0;
splx(s);
}
/*
* Must be called from task context.
*/
void
audio_init_play(struct audio_softc *sc)
{
int s = splaudio();
sc->sc_wstamp = sc->sc_pr.stamp;
if (sc->hw_if->speaker_ctl)
sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_ON);
sc->sc_input_fragment_length = 0;
splx(s);
}
int
audio_drain(struct audio_softc *sc)
{
int error, drops;
struct audio_ringbuffer *cb = &sc->sc_pr;
int s;
DPRINTF(("audio_drain: enter busy=%d used=%d\n",
sc->sc_pbus, sc->sc_pr.used));
if (sc->sc_pr.mmapped || sc->sc_pr.used <= 0)
return 0;
if (!sc->sc_pbus) {
/* We've never started playing, probably because the
* block was too short. Pad it and start now.
*/
int cc;
u_char *inp = cb->inp;
cc = cb->blksize - (inp - cb->start) % cb->blksize;
audio_fill_silence(&sc->sc_pparams, inp, cc);
inp += cc;
if (inp >= cb->end)
inp = cb->start;
s = splaudio();
cb->used += cc;
cb->inp = inp;
error = audiostartp(sc);
splx(s);
if (error)
return error;
}
/*
* Play until a silence block has been played, then we
* know all has been drained.
* XXX This should be done some other way to avoid
* playing silence.
*/
#ifdef DIAGNOSTIC
if (cb->copying) {
printf("audio_drain: copying in progress!?!\n");
cb->copying = 0;
}
#endif
drops = cb->drops;
error = 0;
s = splaudio();
while (cb->drops == drops && !error) {
DPRINTF(("audio_drain: used=%d, drops=%ld\n",
sc->sc_pr.used, cb->drops));
/*
* When the process is exiting, it ignores all signals and
* we can't interrupt this sleep, so we set a timeout
* just in case.
*/
error = audio_sleep_timo(&sc->sc_wchan, "aud_dr", 30*hz);
if (sc->sc_dying)
error = EIO;
}
splx(s);
return error;
}
/*
* Close an audio chip.
*/
/* ARGSUSED */
int
audio_close(struct audio_softc *sc, int flags, int ifmt, struct proc *p)
{
struct audio_hw_if *hw = sc->hw_if;
int s;
DPRINTF(("audio_close: sc=%p\n", sc));
s = splaudio();
/* Stop recording. */
if ((flags & FREAD) && sc->sc_rbus) {
/*
* XXX Some drivers (e.g. SB) use the same routine
* to halt input and output so don't halt input if
* in full duplex mode. These drivers should be fixed.
*/
if (!sc->sc_full_duplex ||
sc->hw_if->halt_input != sc->hw_if->halt_output)
sc->hw_if->halt_input(sc->hw_hdl);
sc->sc_rbus = 0;
}
/*
* Block until output drains, but allow ^C interrupt.
*/
sc->sc_pr.usedlow = sc->sc_pr.blksize; /* avoid excessive wakeups */
/*
* If there is pending output, let it drain (unless
* the output is paused).
*/
if ((flags & FWRITE) && sc->sc_pbus) {
if (!sc->sc_pr.pause && !audio_drain(sc) && hw->drain)
(void)hw->drain(sc->hw_hdl);
sc->hw_if->halt_output(sc->hw_hdl);
sc->sc_pbus = 0;
}
hw->close(sc->hw_hdl);
if (flags & FREAD) {
sc->sc_open &= ~AUOPEN_READ;
sc->sc_mode &= ~AUMODE_RECORD;
}
if (flags & FWRITE) {
sc->sc_open &= ~AUOPEN_WRITE;
sc->sc_mode &= ~(AUMODE_PLAY|AUMODE_PLAY_ALL);
}
sc->sc_async_audio = 0;
sc->sc_full_duplex = 0;
splx(s);
DPRINTF(("audio_close: done\n"));
return (0);
}
int
audio_read(struct audio_softc *sc, struct uio *uio, int ioflag)
{
struct audio_ringbuffer *cb = &sc->sc_rr;
u_char *outp;
int error, s, used, cc, n;
const struct audio_params *params;
int hw_bits_per_sample;
if (cb->mmapped)
return EINVAL;
DPRINTFN(1,("audio_read: cc=%lu mode=%d\n",
(unsigned long)uio->uio_resid, sc->sc_mode));
params = &sc->sc_rparams;
switch (params->hw_encoding) {
case AUDIO_ENCODING_SLINEAR_LE:
case AUDIO_ENCODING_SLINEAR_BE:
case AUDIO_ENCODING_ULINEAR_LE:
case AUDIO_ENCODING_ULINEAR_BE:
hw_bits_per_sample = params->hw_channels * params->precision
* params->factor;
break;
default:
hw_bits_per_sample = 8 * params->factor / params->factor_denom;
}
error = 0;
/*
* If hardware is half-duplex and currently playing, return
* silence blocks based on the number of blocks we have output.
*/
if (!sc->sc_full_duplex &&
(sc->sc_mode & AUMODE_PLAY)) {
while (uio->uio_resid > 0 && !error) {
s = splaudio();
for(;;) {
cc = sc->sc_pr.stamp - sc->sc_wstamp;
if (cc > 0)
break;
DPRINTF(("audio_read: stamp=%lu, wstamp=%lu\n",
sc->sc_pr.stamp, sc->sc_wstamp));
if (ioflag & IO_NDELAY) {
splx(s);
return (EWOULDBLOCK);
}
error = audio_sleep(&sc->sc_rchan, "aud_hr");
if (sc->sc_dying)
error = EIO;
if (error) {
splx(s);
return (error);
}
}
splx(s);
if (uio->uio_resid < cc)
cc = uio->uio_resid;
DPRINTFN(1,("audio_read: reading in write mode, "
"cc=%d\n", cc));
error = audio_silence_copyout(sc, cc, uio);
sc->sc_wstamp += cc;
}
return (error);
}
while (uio->uio_resid > 0 && !error) {
if (sc->sc_rconvbuffer_end - sc->sc_rconvbuffer_begin <= 0) {
s = splaudio();
while (cb->used * 8 < hw_bits_per_sample) {
if (!sc->sc_rbus) {
error = audiostartr(sc);
if (error) {
splx(s);
return (error);
}
}
if (ioflag & IO_NDELAY) {
splx(s);
return (EWOULDBLOCK);
}
DPRINTFN(1, ("audio_read: sleep used=%d\n",
cb->used));
error = audio_sleep(&sc->sc_rchan, "aud_rd");
if (sc->sc_dying)
error = EIO;
if (error) {
splx(s);
return error;
}
}
/*
* Move data in the ring buffer to sc_rconvbuffer as
* possible with/without rate conversion.
*/
used = cb->used;
outp = cb->outp;
cb->copying = 1;
splx(s);
cc = used - cb->usedlow; /* maximum to read */
if (cc > sc->sc_rconvbuffer_size)
cc = sc->sc_rconvbuffer_size;
n = cc * params->factor / params->factor_denom;
if (n < cc)
cc = n;
/* cc must be aligned by the sample size */
cc = (cc * 8 / hw_bits_per_sample) * hw_bits_per_sample / 8;
#ifdef DIAGNOSTIC
if (cc == 0)
printf("audio_read: cc=0 hw_bits_per_sample=%d\n",
hw_bits_per_sample);
#endif
/*
* The format of data in the ring buffer is
* [hw_sample_rate, hw_encoding, hw_precision, hw_channels]
*/
#if NAURATECONV > 0
sc->sc_rconvbuffer_end =
auconv_record(&sc->sc_rconv, params,
sc->sc_rconvbuffer, outp, cc);
#else
n = cb->end - outp;
if (cc <= n) {
memcpy(sc->sc_rconvbuffer, outp, cc);
} else {
memcpy(sc->sc_rconvbuffer, outp, n);
memcpy(sc->sc_rconvbuffer + n, cb->start,
cc - n);
}
sc->sc_rconvbuffer_end = cc;
#endif /* !NAURATECONV */
/*
* The format of data in sc_rconvbuffer is
* [sample_rate, hw_encoding, hw_precision, channels]
*/
outp += cc;
if (outp >= cb->end)
outp -= cb->end - cb->start;
s = splaudio();
cb->outp = outp;
cb->used -= cc;
cb->copying = 0;
splx(s);
if (params->sw_code) {
cc = sc->sc_rconvbuffer_end;
#ifdef DIAGNOSTIC
if (cc % params->factor != 0)
printf("audio_read: cc is not aligned"
": cc=%d factor=%d\n", cc,
params->factor);
#endif
cc = cc * params->factor_denom / params->factor;
#ifdef DIAGNOSTIC
if (cc == 0)
printf("audio_read: cc=0 "
"factor=%d/%d\n",
params->factor,
params->factor_denom);
#endif
params->sw_code(sc->hw_hdl, sc->sc_rconvbuffer,
cc);
sc->sc_rconvbuffer_end = cc;
}
sc->sc_rconvbuffer_begin = 0;
/*
* The format of data in sc_rconvbuffer is
* [sample_rate, encoding, precision, channels]
*/
}
cc = sc->sc_rconvbuffer_end - sc->sc_rconvbuffer_begin;
if (uio->uio_resid < cc)
cc = uio->uio_resid; /* and no more than we want */
DPRINTFN(0,("audio_read: buffer=%p[%d] (~ %d), cc=%d\n",
sc->sc_rconvbuffer, sc->sc_rconvbuffer_begin,
sc->sc_rconvbuffer_end, cc));
n = uio->uio_resid;
error = uiomove(sc->sc_rconvbuffer + sc->sc_rconvbuffer_begin,
cc, uio);
cc = n - uio->uio_resid; /* number of bytes actually moved */
sc->sc_rconvbuffer_begin += cc;
}
return (error);
}
void
audio_clear(struct audio_softc *sc)
{
int s = splaudio();
if (sc->sc_rbus) {
audio_wakeup(&sc->sc_rchan);
sc->hw_if->halt_input(sc->hw_hdl);
sc->sc_rbus = 0;
}
if (sc->sc_pbus) {
audio_wakeup(&sc->sc_wchan);
sc->hw_if->halt_output(sc->hw_hdl);
sc->sc_pbus = 0;
}
splx(s);
}
void
audio_calc_blksize(struct audio_softc *sc, int mode)
{
struct audio_hw_if *hw = sc->hw_if;
struct audio_params *parm;
struct audio_ringbuffer *rb;
int bs;
if (sc->sc_blkset)
return;
if (mode == AUMODE_PLAY) {
parm = &sc->sc_pparams;
rb = &sc->sc_pr;
} else {
parm = &sc->sc_rparams;
rb = &sc->sc_rr;
}
bs = parm->hw_sample_rate * audio_blk_ms / 1000 *
parm->hw_channels * parm->precision / NBBY *
parm->factor;
ROUNDSIZE(bs);
if (hw->round_blocksize)
bs = hw->round_blocksize(sc->hw_hdl, bs);
/*
* The blocksize should never be 0, but a faulty
* driver might set it wrong. Just use something.
*/
if (bs <= 0)
bs = 512;
rb->blksize = bs;
DPRINTF(("audio_calc_blksize: %s blksize=%d\n",
mode == AUMODE_PLAY ? "play" : "record", bs));
}
void
audio_fill_silence(struct audio_params *params, u_char *p, int n)
{
u_char auzero0, auzero1 = 0; /* initialize to please gcc */
int nfill = 1;
switch (params->hw_encoding) {
case AUDIO_ENCODING_ULAW:
auzero0 = 0x7f;
break;
case AUDIO_ENCODING_ALAW:
auzero0 = 0x55;
break;
case AUDIO_ENCODING_MPEG_L1_STREAM:
case AUDIO_ENCODING_MPEG_L1_PACKETS:
case AUDIO_ENCODING_MPEG_L1_SYSTEM:
case AUDIO_ENCODING_MPEG_L2_STREAM:
case AUDIO_ENCODING_MPEG_L2_PACKETS:
case AUDIO_ENCODING_MPEG_L2_SYSTEM:
case AUDIO_ENCODING_ADPCM: /* is this right XXX */
case AUDIO_ENCODING_SLINEAR_LE:
case AUDIO_ENCODING_SLINEAR_BE:
auzero0 = 0;/* fortunately this works for any number of bits */
break;
case AUDIO_ENCODING_ULINEAR_LE:
case AUDIO_ENCODING_ULINEAR_BE:
if (params->hw_precision > 8) {
nfill = (params->hw_precision + NBBY - 1)/ NBBY;
auzero0 = 0x80;
auzero1 = 0;
} else
auzero0 = 0x80;
break;
default:
DPRINTF(("audio: bad encoding %d\n", params->encoding));
auzero0 = 0;
break;
}
if (nfill == 1) {
while (--n >= 0)
*p++ = auzero0; /* XXX memset */
} else /* nfill must no longer be 2 */ {
if (params->hw_encoding == AUDIO_ENCODING_ULINEAR_LE) {
int k = nfill;
while (--k > 0)
*p++ = auzero1;
n -= nfill - 1;
}
while (n >= nfill) {
int k = nfill;
*p++ = auzero0;
while (--k > 0)
*p++ = auzero1;
n -= nfill;
}
if (n-- > 0) /* XXX must be 1 - DIAGNOSTIC check? */
*p++ = auzero0;
}
}
int
audio_silence_copyout(struct audio_softc *sc, int n, struct uio *uio)
{
int error;
int k;
u_char zerobuf[128];
audio_fill_silence(&sc->sc_rparams, zerobuf, sizeof zerobuf);
error = 0;
while (n > 0 && uio->uio_resid > 0 && !error) {
k = min(n, min(uio->uio_resid, sizeof zerobuf));
error = uiomove(zerobuf, k, uio);
n -= k;
}
return (error);
}
int
audio_write(struct audio_softc *sc, struct uio *uio, int ioflag)
{
struct audio_ringbuffer *cb = &sc->sc_pr;
u_char *inp, *einp;
int saveerror, error, s, n, cc, used;
struct audio_params *params;
int samples, hw_bits_per_sample, user_bits_per_sample;
int input_remain, space;
DPRINTFN(2,("audio_write: sc=%p count=%lu used=%d(hi=%d)\n",
sc, (unsigned long)uio->uio_resid, sc->sc_pr.used,
sc->sc_pr.usedhigh));
if (cb->mmapped)
return EINVAL;
if (uio->uio_resid == 0) {
sc->sc_eof++;
return 0;
}
/*
* If half-duplex and currently recording, throw away data.
*/
if (!sc->sc_full_duplex &&
(sc->sc_mode & AUMODE_RECORD)) {
uio->uio_offset += uio->uio_resid;
uio->uio_resid = 0;
DPRINTF(("audio_write: half-dpx read busy\n"));
return (0);
}
if (!(sc->sc_mode & AUMODE_PLAY_ALL) && sc->sc_playdrop > 0) {
n = min(sc->sc_playdrop, uio->uio_resid);
DPRINTF(("audio_write: playdrop %d\n", n));
uio->uio_offset += n;
uio->uio_resid -= n;
sc->sc_playdrop -= n;
if (uio->uio_resid == 0)
return 0;
}
params = &sc->sc_pparams;
DPRINTFN(1, ("audio_write: sr=%ld, enc=%d, prec=%d, chan=%d, sw=%p, "
"fact=%d\n",
sc->sc_pparams.sample_rate, sc->sc_pparams.encoding,
sc->sc_pparams.precision, sc->sc_pparams.channels,
sc->sc_pparams.sw_code, sc->sc_pparams.factor));
/*
* For some encodings, handle data in sample unit.
*/
switch (params->hw_encoding) {
case AUDIO_ENCODING_SLINEAR_LE:
case AUDIO_ENCODING_SLINEAR_BE:
case AUDIO_ENCODING_ULINEAR_LE:
case AUDIO_ENCODING_ULINEAR_BE:
hw_bits_per_sample = params->hw_channels * params->precision
* params->factor;
user_bits_per_sample = params->channels * params->precision;
break;
default:
hw_bits_per_sample = 8 * params->factor / params->factor_denom;
user_bits_per_sample = 8;
}
#ifdef DIAGNOSTIC
if (hw_bits_per_sample > MAX_SAMPLE_SIZE * 8) {
printf("audio_write(): Invalid sample size: cur=%d max=%d\n",
hw_bits_per_sample / 8, MAX_SAMPLE_SIZE);
}
#endif
space = ((params->hw_sample_rate / params->sample_rate) + 1)
* hw_bits_per_sample / 8;
error = 0;
while ((input_remain = uio->uio_resid + sc->sc_input_fragment_length) > 0
&& !error) {
s = splaudio();
if (input_remain < user_bits_per_sample / 8) {
n = uio->uio_resid;
DPRINTF(("audio_write: fragment uiomove length=%d\n", n));
error = uiomove(sc->sc_input_fragment
+ sc->sc_input_fragment_length,
n, uio);
if (!error)
sc->sc_input_fragment_length += n;
splx(s);
return (error);
}
while (cb->used + space >= cb->usedhigh) {
DPRINTFN(2, ("audio_write: sleep used=%d lowat=%d "
"hiwat=%d\n",
cb->used, cb->usedlow, cb->usedhigh));
if (ioflag & IO_NDELAY) {
splx(s);
return (EWOULDBLOCK);
}
error = audio_sleep(&sc->sc_wchan, "aud_wr");
if (sc->sc_dying)
error = EIO;
if (error) {
splx(s);
return (error);
}
}
used = cb->used;
inp = cb->inp;
cb->copying = 1;
splx(s);
cc = cb->usedhigh - used; /* maximum to write */
/* cc may be greater than the size of the ring buffer */
if (cc > cb->end - cb->start)
cc = cb->end - cb->start;
/* cc: # of bytes we can write to the ring buffer */
samples = cc * 8 / hw_bits_per_sample;
#ifdef DIAGNOSTIC
if (samples == 0)
printf("audio_write: samples (cc/hw_bps) == 0\n");
#endif
/* samples: # of samples we can write to the ring buffer */
samples = samples * params->sample_rate / params->hw_sample_rate;
#ifdef DIAGNOSTIC
if (samples == 0)
printf("audio_write: samples (rate/hw_rate) == 0 "
"usedhigh-used=%d cc/hw_bps=%d/%d "
"rate/hw_rate=%ld/%ld space=%d\n",
cb->usedhigh - cb->used, cc,
hw_bits_per_sample / 8, params->sample_rate,
params->hw_sample_rate, space);
#endif
/* samples: # of samples in source data */
cc = samples * user_bits_per_sample / 8;
/* cc: # of bytes in source data */
if (input_remain < cc) /* and no more than we have */
cc = (input_remain * 8 / user_bits_per_sample)
* user_bits_per_sample / 8;
#ifdef DIAGNOSTIC
if (cc == 0)
printf("audio_write: cc == 0\n");
#endif
if (cc * params->factor / params->factor_denom
> sc->sc_pconvbuffer_size) {
/*
* cc = (pconv / factor / user_bps ) * user_bps
*/
cc = (sc->sc_pconvbuffer_size * params->factor_denom
* 8 / params->factor / user_bits_per_sample)
* user_bits_per_sample / 8;
}
#ifdef DIAGNOSTIC
/*
* This should never happen since the block size and and
* block pointers are always nicely aligned.
*/
if (cc == 0) {
printf("audio_write: cc == 0, swcode=%p, factor=%d "
"remain=%d u_bps=%d hw_bps=%d\n",
sc->sc_pparams.sw_code, sc->sc_pparams.factor,
input_remain, user_bits_per_sample,
hw_bits_per_sample);
cb->copying = 0;
return EINVAL;
}
#endif
DPRINTFN(1, ("audio_write: uiomove cc=%d inp=%p, left=%lu\n",
cc, inp, (unsigned long)uio->uio_resid));
memcpy(sc->sc_pconvbuffer, sc->sc_input_fragment,
sc->sc_input_fragment_length);
cc -= sc->sc_input_fragment_length;
n = uio->uio_resid;
error = uiomove(sc->sc_pconvbuffer + sc->sc_input_fragment_length,
cc, uio);
if (cc != n - uio->uio_resid) {
printf("audio_write: uiomove didn't move requested "
"amount: requested=%d, actual=%ld\n",
cc, (long)n - uio->uio_resid);
}
/* number of bytes actually moved */
cc = sc->sc_input_fragment_length + n - uio->uio_resid;
sc->sc_input_fragment_length = 0;
#ifdef AUDIO_DEBUG
if (error)
printf("audio_write:(1) uiomove failed %d; cc=%d "
"inp=%p\n", error, cc, inp);
#endif
/*
* Continue even if uiomove() failed because we may have
* gotten a partial block.
*/
/*
* The format of data in sc_pconvbuffer is:
* [sample_rate, encoding, precision, channels]
*/
if (sc->sc_pparams.sw_code) {
sc->sc_pparams.sw_code(sc->hw_hdl,
sc->sc_pconvbuffer, cc);
/* Adjust count after the expansion. */
cc = cc * sc->sc_pparams.factor
/ sc->sc_pparams.factor_denom;
DPRINTFN(1, ("audio_write: expanded cc=%d\n", cc));
}
/*
* The format of data in sc_pconvbuffer is:
* [sample_rate, hw_encoding, hw_precision, channels]
*/
#if NAURATECONV > 0
cc = auconv_play(&sc->sc_pconv, params, inp,
sc->sc_pconvbuffer, cc);
#else
n = cb->end - inp;
if (cc <= n) {
memcpy(inp, sc->sc_pconvbuffer, cc);
} else {
memcpy(inp, sc->sc_pconvbuffer, n);
memcpy(cb->start, sc->sc_pconvbuffer + n, cc - n);
}
#endif /* !NAURATECONV */
/*
* The format of data in inp is:
* [hw_sample_rate, hw_encoding, hw_precision, hw_channels]
* cc is the size of data actually written to inp.
*/
einp = cb->inp + cc;
if (einp >= cb->end)
einp -= cb->end - cb->start; /* not cb->bufsize */
s = splaudio();
/*
* This is a very suboptimal way of keeping track of
* silence in the buffer, but it is simple.
*/
sc->sc_sil_count = 0;
cb->inp = einp;
cb->used += cc;
/*
* If the interrupt routine wants the last block filled AND
* the copy did not fill the last block completely it needs to
* be padded.
*/
if (cb->needfill &&
(inp - cb->start) / cb->blksize ==
(einp - cb->start) / cb->blksize) {
/* Figure out how many bytes to a block boundary. */
cc = cb->blksize - (einp - cb->start) % cb->blksize;
DPRINTF(("audio_write: partial fill %d\n", cc));
} else
cc = 0;
cb->needfill = 0;
cb->copying = 0;
if (!sc->sc_pbus && !cb->pause) {
saveerror = error;
error = audiostartp(sc);
if (saveerror != 0) {
/* Report the first error that occurred. */
error = saveerror;
}
}
splx(s);
if (cc != 0) {
DPRINTFN(1, ("audio_write: fill %d\n", cc));
audio_fill_silence(&sc->sc_pparams, einp, cc);
}
}
return (error);
}
int
audio_ioctl(struct audio_softc *sc, u_long cmd, caddr_t addr, int flag,
struct proc *p)
{
struct audio_hw_if *hw = sc->hw_if;
struct audio_offset *ao;
int error = 0, s, offs, fd;
int rbus, pbus;
DPRINTF(("audio_ioctl(%lu,'%c',%lu)\n",
IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff));
switch (cmd) {
case FIONBIO:
/* All handled in the upper FS layer. */
break;
case FIOASYNC:
if (*(int *)addr) {
if (sc->sc_async_audio)
return (EBUSY);
sc->sc_async_audio = p;
DPRINTF(("audio_ioctl: FIOASYNC %p\n", p));
} else
sc->sc_async_audio = 0;
break;
case AUDIO_FLUSH:
DPRINTF(("AUDIO_FLUSH\n"));
rbus = sc->sc_rbus;
pbus = sc->sc_pbus;
audio_clear(sc);
s = splaudio();
error = audio_initbufs(sc);
if (error) {
splx(s);
return error;
}
if ((sc->sc_mode & AUMODE_PLAY) && !sc->sc_pbus && pbus)
error = audiostartp(sc);
if (!error &&
(sc->sc_mode & AUMODE_RECORD) && !sc->sc_rbus && rbus)
error = audiostartr(sc);
splx(s);
break;
/*
* Number of read (write) samples dropped. We don't know where or
* when they were dropped.
*/
case AUDIO_RERROR:
*(int *)addr = sc->sc_rr.drops;
break;
case AUDIO_PERROR:
*(int *)addr = sc->sc_pr.drops;
break;
/*
* Offsets into buffer.
*/
case AUDIO_GETIOFFS:
s = splaudio();
/* figure out where next DMA will start */
ao = (struct audio_offset *)addr;
ao->samples = sc->sc_rr.stamp;
ao->deltablks =
(sc->sc_rr.stamp - sc->sc_rr.stamp_last) / sc->sc_rr.blksize;
sc->sc_rr.stamp_last = sc->sc_rr.stamp;
ao->offset = sc->sc_rr.inp - sc->sc_rr.start;
splx(s);
break;
case AUDIO_GETOOFFS:
s = splaudio();
/* figure out where next DMA will start */
ao = (struct audio_offset *)addr;
offs = sc->sc_pr.outp - sc->sc_pr.start + sc->sc_pr.blksize;
if (sc->sc_pr.start + offs >= sc->sc_pr.end)
offs = 0;
ao->samples = sc->sc_pr.stamp;
ao->deltablks =
(sc->sc_pr.stamp - sc->sc_pr.stamp_last) / sc->sc_pr.blksize;
sc->sc_pr.stamp_last = sc->sc_pr.stamp;
ao->offset = offs;
splx(s);
break;
/*
* How many bytes will elapse until mike hears the first
* sample of what we write next?
*/
case AUDIO_WSEEK:
*(u_long *)addr = sc->sc_rr.used;
break;
case AUDIO_SETINFO:
{
struct audio_info *info = (struct audio_info *)addr;
DPRINTF(("AUDIO_SETINFO mode=0x%x\n", sc->sc_mode));
/* Ensure PLAY/RECORD mode is set correctly */
if (info->mode != ~0) {
info->mode &= ~(AUMODE_PLAY|AUMODE_RECORD);
info->mode |= sc->sc_mode;
}
error = audiosetinfo(sc, info);
break;
}
case AUDIO_GETINFO:
DPRINTF(("AUDIO_GETINFO\n"));
error = audiogetinfo(sc, (struct audio_info *)addr);
break;
case AUDIO_DRAIN:
DPRINTF(("AUDIO_DRAIN\n"));
error = audio_drain(sc);
if (!error && hw->drain)
error = hw->drain(sc->hw_hdl);
break;
case AUDIO_GETDEV:
DPRINTF(("AUDIO_GETDEV\n"));
error = hw->getdev(sc->hw_hdl, (audio_device_t *)addr);
break;
case AUDIO_GETENC:
DPRINTF(("AUDIO_GETENC\n"));
error =
hw->query_encoding(sc->hw_hdl, (struct audio_encoding *)addr);
break;
case AUDIO_GETFD:
DPRINTF(("AUDIO_GETFD\n"));
*(int *)addr = sc->sc_full_duplex;
break;
case AUDIO_SETFD:
DPRINTF(("AUDIO_SETFD\n"));
fd = *(int *)addr;
if (hw->get_props(sc->hw_hdl) & AUDIO_PROP_FULLDUPLEX) {
if (hw->setfd)
error = hw->setfd(sc->hw_hdl, fd);
else
error = 0;
if (!error)
sc->sc_full_duplex = fd;
} else {
if (fd)
error = ENOTTY;
else
error = 0;
}
break;
case AUDIO_GETPROPS:
DPRINTF(("AUDIO_GETPROPS\n"));
*(int *)addr = hw->get_props(sc->hw_hdl);
break;
default:
if (hw->dev_ioctl) {
error = hw->dev_ioctl(sc->hw_hdl, cmd, addr, flag, p);
} else {
DPRINTF(("audio_ioctl: unknown ioctl\n"));
error = EINVAL;
}
break;
}
DPRINTF(("audio_ioctl(%lu,'%c',%lu) result %d\n",
IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, error));
return (error);
}
int
audio_poll(struct audio_softc *sc, int events, struct proc *p)
{
int revents = 0;
int s = splaudio();
DPRINTF(("audio_poll: events=0x%x mode=%d\n", events, sc->sc_mode));
if (events & (POLLIN | POLLRDNORM))
/*
* If half duplex and playing, audio_read() will generate
* silence at the play rate; poll for silence being
* available. Otherwise, poll for recorded sound.
*/
if ((!sc->sc_full_duplex && (sc->sc_mode & AUMODE_PLAY)) ?
sc->sc_pr.stamp > sc->sc_wstamp :
sc->sc_rr.used > sc->sc_rr.usedlow)
revents |= events & (POLLIN | POLLRDNORM);
if (events & (POLLOUT | POLLWRNORM))
/*
* If half duplex and recording, audio_write() will throw
* away play data, which means we are always ready to write.
* Otherwise, poll for play buffer being below its low water
* mark.
*/
if ((!sc->sc_full_duplex && (sc->sc_mode & AUMODE_RECORD)) ||
sc->sc_pr.used <= sc->sc_pr.usedlow)
revents |= events & (POLLOUT | POLLWRNORM);
if (revents == 0) {
if (events & (POLLIN | POLLRDNORM))
selrecord(p, &sc->sc_rsel);
if (events & (POLLOUT | POLLWRNORM))
selrecord(p, &sc->sc_wsel);
}
splx(s);
return (revents);
}
static void
filt_audiordetach(struct knote *kn)
{
struct audio_softc *sc = kn->kn_hook;
int s;
s = splaudio();
SLIST_REMOVE(&sc->sc_rsel.sel_klist, kn, knote, kn_selnext);
splx(s);
}
static int
filt_audioread(struct knote *kn, long hint)
{
struct audio_softc *sc = kn->kn_hook;
int s;
/* XXXLUKEM (thorpej): please make sure this is right */
s = splaudio();
if (sc->sc_mode & AUMODE_PLAY)
kn->kn_data = sc->sc_pr.stamp - sc->sc_wstamp;
else
kn->kn_data = sc->sc_rr.used - sc->sc_rr.usedlow;
splx(s);
return (kn->kn_data > 0);
}
static const struct filterops audioread_filtops =
{ 1, NULL, filt_audiordetach, filt_audioread };
static void
filt_audiowdetach(struct knote *kn)
{
struct audio_softc *sc = kn->kn_hook;
int s;
s = splaudio();
SLIST_REMOVE(&sc->sc_wsel.sel_klist, kn, knote, kn_selnext);
splx(s);
}
static int
filt_audiowrite(struct knote *kn, long hint)
{
struct audio_softc *sc = kn->kn_hook;
int s;
/* XXXLUKEM (thorpej): please make sure this is right */
s = splaudio();
kn->kn_data = sc->sc_pr.usedlow - sc->sc_pr.used;
splx(s);
return (kn->kn_data > 0);
}
static const struct filterops audiowrite_filtops =
{ 1, NULL, filt_audiowdetach, filt_audiowrite };
int
audio_kqfilter(struct audio_softc *sc, struct knote *kn)
{
struct klist *klist;
int s;
switch (kn->kn_filter) {
case EVFILT_READ:
klist = &sc->sc_rsel.sel_klist;
kn->kn_fop = &audioread_filtops;
break;
case EVFILT_WRITE:
klist = &sc->sc_wsel.sel_klist;
kn->kn_fop = &audiowrite_filtops;
break;
default:
return (1);
}
kn->kn_hook = sc;
s = splaudio();
SLIST_INSERT_HEAD(klist, kn, kn_selnext);
splx(s);
return (0);
}
paddr_t
audio_mmap(struct audio_softc *sc, off_t off, int prot)
{
struct audio_hw_if *hw = sc->hw_if;
struct audio_ringbuffer *cb;
int s;
DPRINTF(("audio_mmap: off=%lld, prot=%d\n", (long long)off, prot));
if (!(hw->get_props(sc->hw_hdl) & AUDIO_PROP_MMAP) || !hw->mappage)
return -1;
#if 0
/* XXX
* The idea here was to use the protection to determine if
* we are mapping the read or write buffer, but it fails.
* The VM system is broken in (at least) two ways.
* 1) If you map memory VM_PROT_WRITE you SIGSEGV
* when writing to it, so VM_PROT_READ|VM_PROT_WRITE
* has to be used for mmapping the play buffer.
* 2) Even if calling mmap() with VM_PROT_READ|VM_PROT_WRITE
* audio_mmap will get called at some point with VM_PROT_READ
* only.
* So, alas, we always map the play buffer for now.
*/
if (prot == (VM_PROT_READ|VM_PROT_WRITE) ||
prot == VM_PROT_WRITE)
cb = &sc->sc_pr;
else if (prot == VM_PROT_READ)
cb = &sc->sc_rr;
else
return -1;
#else
cb = &sc->sc_pr;
#endif
if ((u_int)off >= cb->bufsize)
return -1;
if (!cb->mmapped) {
cb->mmapped = 1;
if (cb == &sc->sc_pr) {
audio_fill_silence(&sc->sc_pparams, cb->start,
cb->bufsize);
s = splaudio();
if (!sc->sc_pbus)
(void)audiostartp(sc);
splx(s);
} else {
s = splaudio();
if (!sc->sc_rbus)
(void)audiostartr(sc);
splx(s);
}
}
return hw->mappage(sc->hw_hdl, cb->start, off, prot);
}
int
audiostartr(struct audio_softc *sc)
{
int error;
DPRINTF(("audiostartr: start=%p used=%d(hi=%d) mmapped=%d\n",
sc->sc_rr.start, sc->sc_rr.used, sc->sc_rr.usedhigh,
sc->sc_rr.mmapped));
if (sc->hw_if->trigger_input)
error = sc->hw_if->trigger_input(sc->hw_hdl, sc->sc_rr.start,
sc->sc_rr.end, sc->sc_rr.blksize,
audio_rint, (void *)sc, &sc->sc_rparams);
else
error = sc->hw_if->start_input(sc->hw_hdl, sc->sc_rr.start,
sc->sc_rr.blksize, audio_rint, (void *)sc);
if (error) {
DPRINTF(("audiostartr failed: %d\n", error));
return error;
}
sc->sc_rbus = 1;
return 0;
}
int
audiostartp(struct audio_softc *sc)
{
int error;
DPRINTF(("audiostartp: start=%p used=%d(hi=%d) mmapped=%d\n",
sc->sc_pr.start, sc->sc_pr.used, sc->sc_pr.usedhigh,
sc->sc_pr.mmapped));
if (!sc->sc_pr.mmapped && sc->sc_pr.used < sc->sc_pr.blksize)
return 0;
if (sc->hw_if->trigger_output)
error = sc->hw_if->trigger_output(sc->hw_hdl, sc->sc_pr.start,
sc->sc_pr.end, sc->sc_pr.blksize,
audio_pint, (void *)sc, &sc->sc_pparams);
else
error = sc->hw_if->start_output(sc->hw_hdl, sc->sc_pr.outp,
sc->sc_pr.blksize, audio_pint, (void *)sc);
if (error) {
DPRINTF(("audiostartp failed: %d\n", error));
return error;
}
sc->sc_pbus = 1;
return 0;
}
/*
* When the play interrupt routine finds that the write isn't keeping
* the buffer filled it will insert silence in the buffer to make up
* for this. The part of the buffer that is filled with silence
* is kept track of in a very approximate way: it starts at sc_sil_start
* and extends sc_sil_count bytes. If there is already silence in
* the requested area nothing is done; so when the whole buffer is
* silent nothing happens. When the writer starts again sc_sil_count
* is set to 0.
*/
/* XXX
* Putting silence into the output buffer should not really be done
* at splaudio, but there is no softaudio level to do it at yet.
*/
static __inline void
audio_pint_silence(struct audio_softc *sc, struct audio_ringbuffer *cb,
u_char *inp, int cc)
{
u_char *s, *e, *p, *q;
if (sc->sc_sil_count > 0) {
s = sc->sc_sil_start; /* start of silence */
e = s + sc->sc_sil_count; /* end of sil., may be beyond end */
p = inp; /* adjusted pointer to area to fill */
if (p < s)
p += cb->end - cb->start;
q = p+cc;
/* Check if there is already silence. */
if (!(s <= p && p < e &&
s <= q && q <= e)) {
if (s <= p)
sc->sc_sil_count = max(sc->sc_sil_count, q-s);
DPRINTFN(5,("audio_pint_silence: fill cc=%d inp=%p, "
"count=%d size=%d\n",
cc, inp, sc->sc_sil_count,
(int)(cb->end - cb->start)));
audio_fill_silence(&sc->sc_pparams, inp, cc);
} else {
DPRINTFN(5,("audio_pint_silence: already silent "
"cc=%d inp=%p\n", cc, inp));
}
} else {
sc->sc_sil_start = inp;
sc->sc_sil_count = cc;
DPRINTFN(5, ("audio_pint_silence: start fill %p %d\n",
inp, cc));
audio_fill_silence(&sc->sc_pparams, inp, cc);
}
}
/*
* Called from HW driver module on completion of dma output.
* Start output of new block, wrap in ring buffer if needed.
* If no more buffers to play, output zero instead.
* Do a wakeup if necessary.
*/
void
audio_pint(void *v)
{
struct audio_softc *sc = v;
struct audio_hw_if *hw = sc->hw_if;
struct audio_ringbuffer *cb = &sc->sc_pr;
u_char *inp;
int cc, ccr;
int blksize;
int error;
if (!sc->sc_open)
return; /* ignore interrupt if not open */
blksize = cb->blksize;
cb->outp += blksize;
if (cb->outp >= cb->end)
cb->outp = cb->start;
cb->stamp += blksize / sc->sc_pparams.factor;
if (cb->mmapped) {
DPRINTFN(5, ("audio_pint: mmapped outp=%p cc=%d inp=%p\n",
cb->outp, blksize, cb->inp));
if (!hw->trigger_output)
(void)hw->start_output(sc->hw_hdl, cb->outp,
blksize, audio_pint, (void *)sc);
return;
}
#ifdef AUDIO_INTR_TIME
{
struct timeval tv;
u_long t;
microtime(&tv);
t = tv.tv_usec + 1000000 * tv.tv_sec;
if (sc->sc_pnintr) {
long lastdelta, totdelta;
lastdelta = t - sc->sc_plastintr - sc->sc_pblktime;
if (lastdelta > sc->sc_pblktime / 3) {
printf("audio: play interrupt(%d) off "
"relative by %ld us (%lu)\n",
sc->sc_pnintr, lastdelta,
sc->sc_pblktime);
}
totdelta = t - sc->sc_pfirstintr -
sc->sc_pblktime * sc->sc_pnintr;
if (totdelta > sc->sc_pblktime) {
printf("audio: play interrupt(%d) off "
"absolute by %ld us (%lu) (LOST)\n",
sc->sc_pnintr, totdelta,
sc->sc_pblktime);
sc->sc_pnintr++; /* avoid repeated messages */
}
} else
sc->sc_pfirstintr = t;
sc->sc_plastintr = t;
sc->sc_pnintr++;
}
#endif
cb->used -= blksize;
if (cb->used < blksize) {
/* we don't have a full block to use */
if (cb->copying) {
/* writer is in progress, don't disturb */
cb->needfill = 1;
DPRINTFN(1, ("audio_pint: copying in progress\n"));
} else {
inp = cb->inp;
cc = blksize - (inp - cb->start) % blksize;
ccr = cc / sc->sc_pparams.factor;
if (cb->pause)
cb->pdrops += ccr;
else {
cb->drops += ccr;
sc->sc_playdrop += ccr;
}
audio_pint_silence(sc, cb, inp, cc);
inp += cc;
if (inp >= cb->end)
inp = cb->start;
cb->inp = inp;
cb->used += cc;
/* Clear next block so we keep ahead of the DMA. */
if (cb->used + cc < cb->usedhigh)
audio_pint_silence(sc, cb, inp, blksize);
}
}
DPRINTFN(5, ("audio_pint: outp=%p cc=%d\n", cb->outp, blksize));
if (!hw->trigger_output) {
error = hw->start_output(sc->hw_hdl, cb->outp, blksize,
audio_pint, (void *)sc);
if (error) {
/* XXX does this really help? */
DPRINTF(("audio_pint restart failed: %d\n", error));
audio_clear(sc);
}
}
DPRINTFN(2, ("audio_pint: mode=%d pause=%d used=%d lowat=%d\n",
sc->sc_mode, cb->pause, cb->used, cb->usedlow));
if ((sc->sc_mode & AUMODE_PLAY) && !cb->pause) {
if (cb->used <= cb->usedlow) {
audio_wakeup(&sc->sc_wchan);
selnotify(&sc->sc_wsel, 0);
if (sc->sc_async_audio) {
DPRINTFN(3, ("audio_pint: sending SIGIO %p\n",
sc->sc_async_audio));
psignal(sc->sc_async_audio, SIGIO);
}
}
}
/* Possible to return one or more "phantom blocks" now. */
if (!sc->sc_full_duplex && sc->sc_rchan) {
audio_wakeup(&sc->sc_rchan);
selnotify(&sc->sc_rsel, 0);
if (sc->sc_async_audio)
psignal(sc->sc_async_audio, SIGIO);
}
}
/*
* Called from HW driver module on completion of dma input.
* Mark it as input in the ring buffer (fiddle pointers).
* Do a wakeup if necessary.
*/
void
audio_rint(void *v)
{
struct audio_softc *sc = v;
struct audio_hw_if *hw = sc->hw_if;
struct audio_ringbuffer *cb = &sc->sc_rr;
int blksize;
int error;
if (!sc->sc_open)
return; /* ignore interrupt if not open */
blksize = cb->blksize;
cb->inp += blksize;
if (cb->inp >= cb->end)
cb->inp = cb->start;
cb->stamp += blksize;
if (cb->mmapped) {
DPRINTFN(2, ("audio_rint: mmapped inp=%p cc=%d\n",
cb->inp, blksize));
if (!hw->trigger_input)
(void)hw->start_input(sc->hw_hdl, cb->inp, blksize,
audio_rint, (void *)sc);
return;
}
#ifdef AUDIO_INTR_TIME
{
struct timeval tv;
u_long t;
microtime(&tv);
t = tv.tv_usec + 1000000 * tv.tv_sec;
if (sc->sc_rnintr) {
long lastdelta, totdelta;
lastdelta = t - sc->sc_rlastintr - sc->sc_rblktime;
if (lastdelta > sc->sc_rblktime / 5) {
printf("audio: record interrupt(%d) off "
"relative by %ld us (%lu)\n",
sc->sc_rnintr, lastdelta,
sc->sc_rblktime);
}
totdelta = t - sc->sc_rfirstintr -
sc->sc_rblktime * sc->sc_rnintr;
if (totdelta > sc->sc_rblktime / 2) {
sc->sc_rnintr++;
printf("audio: record interrupt(%d) off "
"absolute by %ld us (%lu)\n",
sc->sc_rnintr, totdelta,
sc->sc_rblktime);
sc->sc_rnintr++; /* avoid repeated messages */
}
} else
sc->sc_rfirstintr = t;
sc->sc_rlastintr = t;
sc->sc_rnintr++;
}
#endif
cb->used += blksize;
if (cb->pause) {
DPRINTFN(1, ("audio_rint: pdrops %lu\n", cb->pdrops));
cb->pdrops += blksize;
cb->outp += blksize;
if (cb->outp >= cb->end)
cb->outp = cb->start;
cb->used -= blksize;
} else if (cb->used + blksize >= cb->usedhigh && !cb->copying) {
DPRINTFN(1, ("audio_rint: drops %lu\n", cb->drops));
cb->drops += blksize;
cb->outp += blksize;
if (cb->outp >= cb->end)
cb->outp = cb->start;
cb->used -= blksize;
}
DPRINTFN(2, ("audio_rint: inp=%p cc=%d used=%d\n",
cb->inp, blksize, cb->used));
if (!hw->trigger_input) {
error = hw->start_input(sc->hw_hdl, cb->inp, blksize,
audio_rint, (void *)sc);
if (error) {
/* XXX does this really help? */
DPRINTF(("audio_rint: restart failed: %d\n", error));
audio_clear(sc);
}
}
audio_wakeup(&sc->sc_rchan);
selnotify(&sc->sc_rsel, 0);
if (sc->sc_async_audio)
psignal(sc->sc_async_audio, SIGIO);
}
int
audio_check_params(struct audio_params *p)
{
if (p->encoding == AUDIO_ENCODING_PCM16) {
if (p->precision == 8)
p->encoding = AUDIO_ENCODING_ULINEAR;
else
p->encoding = AUDIO_ENCODING_SLINEAR;
} else if (p->encoding == AUDIO_ENCODING_PCM8) {
if (p->precision == 8)
p->encoding = AUDIO_ENCODING_ULINEAR;
else
return EINVAL;
}
if (p->encoding == AUDIO_ENCODING_SLINEAR)
#if BYTE_ORDER == LITTLE_ENDIAN
p->encoding = AUDIO_ENCODING_SLINEAR_LE;
#else
p->encoding = AUDIO_ENCODING_SLINEAR_BE;
#endif
if (p->encoding == AUDIO_ENCODING_ULINEAR)
#if BYTE_ORDER == LITTLE_ENDIAN
p->encoding = AUDIO_ENCODING_ULINEAR_LE;
#else
p->encoding = AUDIO_ENCODING_ULINEAR_BE;
#endif
switch (p->encoding) {
case AUDIO_ENCODING_ULAW:
case AUDIO_ENCODING_ALAW:
if (p->precision != 8)
return (EINVAL);
break;
case AUDIO_ENCODING_ADPCM:
if (p->precision != 4 && p->precision != 8)
return (EINVAL);
break;
case AUDIO_ENCODING_SLINEAR_LE:
case AUDIO_ENCODING_SLINEAR_BE:
case AUDIO_ENCODING_ULINEAR_LE:
case AUDIO_ENCODING_ULINEAR_BE:
/* XXX is: our zero-fill can handle any multiple of 8 */
if (p->precision != 8 && p->precision != 16 &&
p->precision != 24 && p->precision != 32)
return (EINVAL);
break;
case AUDIO_ENCODING_MPEG_L1_STREAM:
case AUDIO_ENCODING_MPEG_L1_PACKETS:
case AUDIO_ENCODING_MPEG_L1_SYSTEM:
case AUDIO_ENCODING_MPEG_L2_STREAM:
case AUDIO_ENCODING_MPEG_L2_PACKETS:
case AUDIO_ENCODING_MPEG_L2_SYSTEM:
break;
default:
return (EINVAL);
}
if (p->channels < 1 || p->channels > 8) /* sanity check # of channels*/
return (EINVAL);
return (0);
}
int
au_set_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int l, int r)
{
ct->type = AUDIO_MIXER_VALUE;
ct->un.value.num_channels = 2;
ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = l;
ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = r;
if (sc->hw_if->set_port(sc->hw_hdl, ct) == 0)
return 0;
ct->un.value.num_channels = 1;
ct->un.value.level[AUDIO_MIXER_LEVEL_MONO] = (l+r)/2;
return sc->hw_if->set_port(sc->hw_hdl, ct);
}
int
au_set_gain(struct audio_softc *sc, struct au_mixer_ports *ports,
int gain, int balance)
{
mixer_ctrl_t ct;
int i, error;
int l, r;
u_int mask;
int nset;
if (balance == AUDIO_MID_BALANCE) {
l = r = gain;
} else if (balance < AUDIO_MID_BALANCE) {
l = gain;
r = (balance * gain) / AUDIO_MID_BALANCE;
} else {
r = gain;
l = ((AUDIO_RIGHT_BALANCE - balance) * gain)
/ AUDIO_MID_BALANCE;
}
DPRINTF(("au_set_gain: gain=%d balance=%d, l=%d r=%d\n",
gain, balance, l, r));
if (ports->index == -1) {
usemaster:
if (ports->master == -1)
return 0; /* just ignore it silently */
ct.dev = ports->master;
error = au_set_lr_value(sc, &ct, l, r);
} else {
ct.dev = ports->index;
if (ports->isenum) {
ct.type = AUDIO_MIXER_ENUM;
error = sc->hw_if->get_port(sc->hw_hdl, &ct);
if (error)
return error;
for(i = 0; i < ports->nports; i++) {
if (ports->misel[i] == ct.un.ord) {
ct.dev = ports->miport[i];
if (ct.dev == -1 ||
au_set_lr_value(sc, &ct, l, r))
goto usemaster;
else
break;
}
}
} else {
ct.type = AUDIO_MIXER_SET;
error = sc->hw_if->get_port(sc->hw_hdl, &ct);
if (error)
return error;
mask = ct.un.mask;
nset = 0;
for(i = 0; i < ports->nports; i++) {
if (ports->misel[i] & mask) {
ct.dev = ports->miport[i];
if (ct.dev != -1 &&
au_set_lr_value(sc, &ct, l, r) == 0)
nset++;
}
}
if (nset == 0)
goto usemaster;
}
}
if (!error)
mixer_signal(sc);
return error;
}
int
au_get_lr_value(struct audio_softc *sc, mixer_ctrl_t *ct, int *l, int *r)
{
int error;
ct->un.value.num_channels = 2;
if (sc->hw_if->get_port(sc->hw_hdl, ct) == 0) {
*l = ct->un.value.level[AUDIO_MIXER_LEVEL_LEFT];
*r = ct->un.value.level[AUDIO_MIXER_LEVEL_RIGHT];
} else {
ct->un.value.num_channels = 1;
error = sc->hw_if->get_port(sc->hw_hdl, ct);
if (error)
return error;
*r = *l = ct->un.value.level[AUDIO_MIXER_LEVEL_MONO];
}
return 0;
}
void
au_get_gain(struct audio_softc *sc, struct au_mixer_ports *ports,
u_int *pgain, u_char *pbalance)
{
mixer_ctrl_t ct;
int i, l, r, n;
int lgain = AUDIO_MAX_GAIN/2, rgain = AUDIO_MAX_GAIN/2;
if (ports->index == -1) {
usemaster:
if (ports->master == -1)
goto bad;
ct.dev = ports->master;
ct.type = AUDIO_MIXER_VALUE;
if (au_get_lr_value(sc, &ct, &lgain, &rgain))
goto bad;
} else {
ct.dev = ports->index;
if (ports->isenum) {
ct.type = AUDIO_MIXER_ENUM;
if (sc->hw_if->get_port(sc->hw_hdl, &ct))
goto bad;
ct.type = AUDIO_MIXER_VALUE;
for(i = 0; i < ports->nports; i++) {
if (ports->misel[i] == ct.un.ord) {
ct.dev = ports->miport[i];
if (ct.dev == -1 ||
au_get_lr_value(sc, &ct,
&lgain, &rgain))
goto usemaster;
else
break;
}
}
} else {
ct.type = AUDIO_MIXER_SET;
if (sc->hw_if->get_port(sc->hw_hdl, &ct))
goto bad;
ct.type = AUDIO_MIXER_VALUE;
lgain = rgain = n = 0;
for(i = 0; i < ports->nports; i++) {
if (ports->misel[i] & ct.un.mask) {
ct.dev = ports->miport[i];
if (ct.dev == -1 ||
au_get_lr_value(sc, &ct, &l, &r))
goto usemaster;
else {
lgain += l;
rgain += r;
n++;
}
}
}
if (n != 0) {
lgain /= n;
rgain /= n;
}
}
}
bad:
if (lgain == rgain) { /* handles lgain==rgain==0 */
*pgain = lgain;
*pbalance = AUDIO_MID_BALANCE;
} else if (lgain < rgain) {
*pgain = rgain;
/* balance should be > AUDIO_MID_BALANCE */
*pbalance = AUDIO_RIGHT_BALANCE -
(AUDIO_MID_BALANCE * lgain) / rgain;
} else /* lgain > rgain */ {
*pgain = lgain;
/* balance should be < AUDIO_MID_BALANCE */
*pbalance = (AUDIO_MID_BALANCE * rgain) / lgain;
}
}
int
au_set_port(struct audio_softc *sc, struct au_mixer_ports *ports, u_int port)
{
mixer_ctrl_t ct;
int i, error;
if (port == 0 && ports->allports == 0)
return 0; /* allow this special case */
if (ports->index == -1)
return EINVAL;
ct.dev = ports->index;
if (ports->isenum) {
if (port & (port-1))
return EINVAL; /* Only one port allowed */
ct.type = AUDIO_MIXER_ENUM;
error = EINVAL;
for(i = 0; i < ports->nports; i++)
if (ports->aumask[i] == port) {
ct.un.ord = ports->misel[i];
error = sc->hw_if->set_port(sc->hw_hdl, &ct);
break;
}
} else {
ct.type = AUDIO_MIXER_SET;
ct.un.mask = 0;
for(i = 0; i < ports->nports; i++)
if (ports->aumask[i] & port)
ct.un.mask |= ports->misel[i];
if (port != 0 && ct.un.mask == 0)
error = EINVAL;
else
error = sc->hw_if->set_port(sc->hw_hdl, &ct);
}
if (!error)
mixer_signal(sc);
return error;
}
int
au_get_port(struct audio_softc *sc, struct au_mixer_ports *ports)
{
mixer_ctrl_t ct;
int i, aumask;
if (ports->index == -1)
return 0;
ct.dev = ports->index;
ct.type = ports->isenum ? AUDIO_MIXER_ENUM : AUDIO_MIXER_SET;
if (sc->hw_if->get_port(sc->hw_hdl, &ct))
return 0;
aumask = 0;
if (ports->isenum) {
for(i = 0; i < ports->nports; i++)
if (ct.un.ord == ports->misel[i])
aumask = ports->aumask[i];
} else {
for(i = 0; i < ports->nports; i++)
if (ct.un.mask & ports->misel[i])
aumask |= ports->aumask[i];
}
return aumask;
}
#if NAURATECONV <= 0
/* dummy function for the case that aurateconv is not linked */
int
auconv_check_params(const struct audio_params *params)
{
if (params->hw_channels == params->channels
&& params->hw_sample_rate == params->sample_rate)
return 0; /* No conversion */
return (EINVAL);
}
#endif /* !NAURATECONV */
int
audiosetinfo(struct audio_softc *sc, struct audio_info *ai)
{
struct audio_prinfo *r = &ai->record, *p = &ai->play;
int cleared;
int s, setmode, modechange = 0;
int error;
struct audio_hw_if *hw = sc->hw_if;
struct audio_params pp, rp;
int np, nr;
unsigned int blks;
int oldpblksize, oldrblksize;
int rbus, pbus;
u_int gain;
u_char balance;
if (hw == 0) /* HW has not attached */
return(ENXIO);
rbus = sc->sc_rbus;
pbus = sc->sc_pbus;
error = 0;
cleared = 0;
pp = sc->sc_pparams; /* Temporary encoding storage in */
rp = sc->sc_rparams; /* case setting the modes fails. */
nr = np = 0;
if (p->sample_rate != ~0) {
pp.sample_rate = p->sample_rate;
np++;
}
if (r->sample_rate != ~0) {
rp.sample_rate = r->sample_rate;
nr++;
}
if (p->encoding != ~0) {
pp.encoding = p->encoding;
np++;
}
if (r->encoding != ~0) {
rp.encoding = r->encoding;
nr++;
}
if (p->precision != ~0) {
pp.precision = p->precision;
np++;
}
if (r->precision != ~0) {
rp.precision = r->precision;
nr++;
}
if (p->channels != ~0) {
pp.channels = p->channels;
np++;
}
if (r->channels != ~0) {
rp.channels = r->channels;
nr++;
}
#ifdef AUDIO_DEBUG
if (audiodebug && nr)
audio_print_params("Setting record params", &rp);
if (audiodebug && np)
audio_print_params("Setting play params", &pp);
#endif
if (nr && (error = audio_check_params(&rp)))
return error;
if (np && (error = audio_check_params(&pp)))
return error;
setmode = 0;
if (nr) {
if (!cleared)
audio_clear(sc);
modechange = cleared = 1;
rp.sw_code = 0;
rp.factor = 1;
rp.factor_denom = 1;
rp.hw_sample_rate = rp.sample_rate;
rp.hw_encoding = rp.encoding;
rp.hw_precision = rp.precision;
rp.hw_channels = rp.channels;
setmode |= AUMODE_RECORD;
}
if (np) {
if (!cleared)
audio_clear(sc);
modechange = cleared = 1;
pp.sw_code = 0;
pp.factor = 1;
pp.factor_denom = 1;
pp.hw_sample_rate = pp.sample_rate;
pp.hw_encoding = pp.encoding;
pp.hw_precision = pp.precision;
pp.hw_channels = pp.channels;
setmode |= AUMODE_PLAY;
}
if (ai->mode != ~0) {
if (!cleared)
audio_clear(sc);
modechange = cleared = 1;
sc->sc_mode = ai->mode;
if (sc->sc_mode & AUMODE_PLAY_ALL)
sc->sc_mode |= AUMODE_PLAY;
if ((sc->sc_mode & AUMODE_PLAY) && !sc->sc_full_duplex)
/* Play takes precedence */
sc->sc_mode &= ~AUMODE_RECORD;
}
if (modechange) {
int orig_p_channels, orig_p_rate;
int orig_r_channels, orig_r_rate;
int indep;
indep = hw->get_props(sc->hw_hdl) & AUDIO_PROP_INDEPENDENT;
if (!indep) {
if (setmode == AUMODE_RECORD)
pp = rp;
else if (setmode == AUMODE_PLAY)
rp = pp;
}
/* Some device drivers change channels/sample_rate and change
* no channels/sample_rate. */
orig_p_channels = pp.channels;
orig_p_rate = pp.sample_rate;
orig_r_channels = rp.channels;
orig_r_rate = rp.sample_rate;
error = hw->set_params(sc->hw_hdl, setmode,
sc->sc_mode & (AUMODE_PLAY | AUMODE_RECORD), &pp, &rp);
if (error)
return (error);
if (np) {
if (orig_p_channels != pp.channels)
pp.hw_channels = pp.channels;
if (orig_p_rate != pp.sample_rate)
pp.hw_sample_rate = pp.sample_rate;
error = auconv_check_params(&pp);
if (error)
return (error);
}
if (nr) {
if (orig_r_channels != rp.channels)
rp.hw_channels = rp.channels;
if (orig_r_rate != rp.sample_rate)
rp.hw_sample_rate = rp.sample_rate;
error = auconv_check_params(&rp);
if (error)
return (error);
}
if (!indep) {
if (setmode == AUMODE_RECORD) {
pp.sample_rate = rp.sample_rate;
pp.encoding = rp.encoding;
pp.channels = rp.channels;
pp.precision = rp.precision;
} else if (setmode == AUMODE_PLAY) {
rp.sample_rate = pp.sample_rate;
rp.encoding = pp.encoding;
rp.channels = pp.channels;
rp.precision = pp.precision;
}
}
sc->sc_rparams = rp;
sc->sc_pparams = pp;
}
oldpblksize = sc->sc_pr.blksize;
oldrblksize = sc->sc_rr.blksize;
/* Play params can affect the record params, so recalculate blksize. */
if (nr || np) {
audio_calc_blksize(sc, AUMODE_RECORD);
audio_calc_blksize(sc, AUMODE_PLAY);
}
#ifdef AUDIO_DEBUG
if (audiodebug > 1 && nr)
audio_print_params("After setting record params", &sc->sc_rparams);
if (audiodebug > 1 && np)
audio_print_params("After setting play params", &sc->sc_pparams);
#endif
if (p->port != ~0) {
if (!cleared)
audio_clear(sc);
cleared = 1;
error = au_set_port(sc, &sc->sc_outports, p->port);
if (error)
return(error);
}
if (r->port != ~0) {
if (!cleared)
audio_clear(sc);
cleared = 1;
error = au_set_port(sc, &sc->sc_inports, r->port);
if (error)
return(error);
}
if (p->gain != ~0) {
au_get_gain(sc, &sc->sc_outports, &gain, &balance);
error = au_set_gain(sc, &sc->sc_outports, p->gain, balance);
if (error)
return(error);
}
if (r->gain != ~0) {
au_get_gain(sc, &sc->sc_inports, &gain, &balance);
error = au_set_gain(sc, &sc->sc_inports, r->gain, balance);
if (error)
return(error);
}
if (p->balance != (u_char)~0) {
au_get_gain(sc, &sc->sc_outports, &gain, &balance);
error = au_set_gain(sc, &sc->sc_outports, gain, p->balance);
if (error)
return(error);
}
if (r->balance != (u_char)~0) {
au_get_gain(sc, &sc->sc_inports, &gain, &balance);
error = au_set_gain(sc, &sc->sc_inports, gain, r->balance);
if (error)
return(error);
}
if (ai->monitor_gain != ~0 &&
sc->sc_monitor_port != -1) {
mixer_ctrl_t ct;
ct.dev = sc->sc_monitor_port;
ct.type = AUDIO_MIXER_VALUE;
ct.un.value.num_channels = 1;
ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = ai->monitor_gain;
error = sc->hw_if->set_port(sc->hw_hdl, &ct);
if (error)
return(error);
}
if (p->pause != (u_char)~0) {
sc->sc_pr.pause = p->pause;
if (!p->pause && !sc->sc_pbus && (sc->sc_mode & AUMODE_PLAY)) {
s = splaudio();
error = audiostartp(sc);
splx(s);
if (error)
return error;
}
}
if (r->pause != (u_char)~0) {
sc->sc_rr.pause = r->pause;
if (!r->pause && !sc->sc_rbus &&
(sc->sc_mode & AUMODE_RECORD)) {
s = splaudio();
error = audiostartr(sc);
splx(s);
if (error)
return error;
}
}
if (ai->blocksize != ~0) {
/* Block size specified explicitly. */
if (!cleared)
audio_clear(sc);
cleared = 1;
if (ai->blocksize == 0) {
audio_calc_blksize(sc, AUMODE_RECORD);
audio_calc_blksize(sc, AUMODE_PLAY);
sc->sc_blkset = 0;
} else {
int bs = ai->blocksize;
if (hw->round_blocksize)
bs = hw->round_blocksize(sc->hw_hdl, bs);
/*
* The blocksize should never be 0, but a faulty
* driver might set it wrong. Just use something.
*/
if (bs <= 0)
bs = 512;
sc->sc_pr.blksize = sc->sc_rr.blksize = bs;
sc->sc_blkset = 1;
}
}
if (ai->mode != ~0) {
if (sc->sc_mode & AUMODE_PLAY)
audio_init_play(sc);
if (sc->sc_mode & AUMODE_RECORD)
audio_init_record(sc);
}
if (hw->commit_settings) {
error = hw->commit_settings(sc->hw_hdl);
if (error)
return (error);
}
if (cleared) {
s = splaudio();
error = audio_initbufs(sc);
if (error) goto err;
if (sc->sc_pr.blksize != oldpblksize ||
sc->sc_rr.blksize != oldrblksize)
audio_calcwater(sc);
if ((sc->sc_mode & AUMODE_PLAY) &&
pbus && !sc->sc_pbus)
error = audiostartp(sc);
if (!error &&
(sc->sc_mode & AUMODE_RECORD) &&
rbus && !sc->sc_rbus)
error = audiostartr(sc);
err:
splx(s);
if (error)
return error;
}
/* Change water marks after initializing the buffers. */
if (ai->hiwat != ~0) {
blks = ai->hiwat;
if (blks > sc->sc_pr.maxblks)
blks = sc->sc_pr.maxblks;
if (blks < 2)
blks = 2;
sc->sc_pr.usedhigh = blks * sc->sc_pr.blksize;
}
if (ai->lowat != ~0) {
blks = ai->lowat;
if (blks > sc->sc_pr.maxblks - 1)
blks = sc->sc_pr.maxblks - 1;
sc->sc_pr.usedlow = blks * sc->sc_pr.blksize;
}
if (ai->hiwat != ~0 || ai->lowat != ~0) {
if (sc->sc_pr.usedlow > sc->sc_pr.usedhigh - sc->sc_pr.blksize)
sc->sc_pr.usedlow =
sc->sc_pr.usedhigh - sc->sc_pr.blksize;
}
return (0);
}
int
audiogetinfo(struct audio_softc *sc, struct audio_info *ai)
{
struct audio_prinfo *r = &ai->record, *p = &ai->play;
struct audio_hw_if *hw = sc->hw_if;
if (hw == 0) /* HW has not attached */
return(ENXIO);
p->sample_rate = sc->sc_pparams.sample_rate;
r->sample_rate = sc->sc_rparams.sample_rate;
p->channels = sc->sc_pparams.channels;
r->channels = sc->sc_rparams.channels;
p->precision = sc->sc_pparams.precision;
r->precision = sc->sc_rparams.precision;
p->encoding = sc->sc_pparams.encoding;
r->encoding = sc->sc_rparams.encoding;
r->port = au_get_port(sc, &sc->sc_inports);
p->port = au_get_port(sc, &sc->sc_outports);
r->avail_ports = sc->sc_inports.allports;
p->avail_ports = sc->sc_outports.allports;
au_get_gain(sc, &sc->sc_inports, &r->gain, &r->balance);
au_get_gain(sc, &sc->sc_outports, &p->gain, &p->balance);
if (sc->sc_monitor_port != -1) {
mixer_ctrl_t ct;
ct.dev = sc->sc_monitor_port;
ct.type = AUDIO_MIXER_VALUE;
ct.un.value.num_channels = 1;
if (sc->hw_if->get_port(sc->hw_hdl, &ct))
ai->monitor_gain = 0;
else
ai->monitor_gain =
ct.un.value.level[AUDIO_MIXER_LEVEL_MONO];
} else
ai->monitor_gain = 0;
p->seek = sc->sc_pr.used;
r->seek = sc->sc_rr.used;
p->samples = sc->sc_pr.stamp - sc->sc_pr.drops;
r->samples = sc->sc_rr.stamp - sc->sc_rr.drops;
p->eof = sc->sc_eof;
r->eof = 0;
p->pause = sc->sc_pr.pause;
r->pause = sc->sc_rr.pause;
p->error = sc->sc_pr.drops != 0;
r->error = sc->sc_rr.drops != 0;
p->waiting = r->waiting = 0; /* open never hangs */
p->open = (sc->sc_open & AUOPEN_WRITE) != 0;
r->open = (sc->sc_open & AUOPEN_READ) != 0;
p->active = sc->sc_pbus;
r->active = sc->sc_rbus;
p->buffer_size = sc->sc_pr.bufsize;
r->buffer_size = sc->sc_rr.bufsize;
ai->blocksize = sc->sc_pr.blksize;
ai->hiwat = sc->sc_pr.usedhigh / sc->sc_pr.blksize;
ai->lowat = sc->sc_pr.usedlow / sc->sc_pr.blksize;
ai->mode = sc->sc_mode;
return (0);
}
/*
* Mixer driver
*/
int
mixer_open(dev_t dev, struct audio_softc *sc, int flags, int ifmt,
struct proc *p)
{
if (!sc->hw_if)
return (ENXIO);
DPRINTF(("mixer_open: flags=0x%x sc=%p\n", flags, sc));
return (0);
}
/*
* Remove a process from those to be signalled on mixer activity.
*/
static void
mixer_remove(struct audio_softc *sc, struct proc *p)
{
struct mixer_asyncs **pm, *m;
for(pm = &sc->sc_async_mixer; *pm; pm = &(*pm)->next) {
if ((*pm)->proc == p) {
m = *pm;
*pm = m->next;
free(m, M_DEVBUF);
return;
}
}
}
/*
* Signal all processes waiting for the mixer.
*/
static void
mixer_signal(struct audio_softc *sc)
{
struct mixer_asyncs *m;
for(m = sc->sc_async_mixer; m; m = m->next)
psignal(m->proc, SIGIO);
}
/*
* Close a mixer device
*/
/* ARGSUSED */
int
mixer_close(struct audio_softc *sc, int flags, int ifmt, struct proc *p)
{
DPRINTF(("mixer_close: sc %p\n", sc));
mixer_remove(sc, p);
return (0);
}
int
mixer_ioctl(struct audio_softc *sc, u_long cmd, caddr_t addr, int flag,
struct proc *p)
{
struct audio_hw_if *hw = sc->hw_if;
int error = EINVAL;
DPRINTF(("mixer_ioctl(%lu,'%c',%lu)\n",
IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff));
switch (cmd) {
case FIOASYNC:
mixer_remove(sc, p); /* remove old entry */
if (*(int *)addr) {
struct mixer_asyncs *ma;
ma = malloc(sizeof (struct mixer_asyncs),
M_DEVBUF, M_WAITOK);
ma->next = sc->sc_async_mixer;
ma->proc = p;
sc->sc_async_mixer = ma;
}
error = 0;
break;
case AUDIO_GETDEV:
DPRINTF(("AUDIO_GETDEV\n"));
error = hw->getdev(sc->hw_hdl, (audio_device_t *)addr);
break;
case AUDIO_MIXER_DEVINFO:
DPRINTF(("AUDIO_MIXER_DEVINFO\n"));
((mixer_devinfo_t *)addr)->un.v.delta = 0; /* default */
error = hw->query_devinfo(sc->hw_hdl, (mixer_devinfo_t *)addr);
break;
case AUDIO_MIXER_READ:
DPRINTF(("AUDIO_MIXER_READ\n"));
error = hw->get_port(sc->hw_hdl, (mixer_ctrl_t *)addr);
break;
case AUDIO_MIXER_WRITE:
DPRINTF(("AUDIO_MIXER_WRITE\n"));
error = hw->set_port(sc->hw_hdl, (mixer_ctrl_t *)addr);
if (!error && hw->commit_settings)
error = hw->commit_settings(sc->hw_hdl);
if (!error)
mixer_signal(sc);
break;
default:
if (hw->dev_ioctl)
error = hw->dev_ioctl(sc->hw_hdl, cmd, addr, flag, p);
else
error = EINVAL;
break;
}
DPRINTF(("mixer_ioctl(%lu,'%c',%lu) result %d\n",
IOCPARM_LEN(cmd), (char)IOCGROUP(cmd), cmd&0xff, error));
return (error);
}
#endif /* NAUDIO > 0 */
#include "midi.h"
#if NAUDIO == 0 && (NMIDI > 0 || NMIDIBUS > 0)
#include <sys/param.h>
#include <sys/systm.h>
#include <sys/device.h>
#include <sys/audioio.h>
#include <dev/audio_if.h>
#endif
#if NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0)
int
audioprint(void *aux, const char *pnp)
{
struct audio_attach_args *arg = aux;
const char *type;
if (pnp != NULL) {
switch (arg->type) {
case AUDIODEV_TYPE_AUDIO:
type = "audio";
break;
case AUDIODEV_TYPE_MIDI:
type = "midi";
break;
case AUDIODEV_TYPE_OPL:
type = "opl";
break;
case AUDIODEV_TYPE_MPU:
type = "mpu";
break;
default:
panic("audioprint: unknown type %d", arg->type);
}
printf("%s at %s", type, pnp);
}
return (UNCONF);
}
#endif /* NAUDIO > 0 || (NMIDI > 0 || NMIDIBUS > 0) */
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