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NetBSD/sys/dev/audio.c

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/* $NetBSD: audio.c,v 1.63 1997/08/08 00:03:26 augustss 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 and signals.
* - Add SIGIO generation for changes in the mixer device.
*/
#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 <vm/vm.h>
#include <vm/vm_prot.h>
#include <machine/endian.h>
#ifdef AUDIO_DEBUG
#define DPRINTF(x) if (audiodebug) printf x
int audiodebug = 0;
#else
#define DPRINTF(x)
#endif
#define ROUNDSIZE(x) x &= -16 /* round to nice boundary */
int audio_blk_ms = AUDIO_BLK_MS;
struct audio_softc **audio_softc;
int naudio = 0; /* Current size of audio_softc */
int audiosetinfo __P((struct audio_softc *, struct audio_info *));
int audiogetinfo __P((struct audio_softc *, struct audio_info *));
int audio_open __P((dev_t, int, int, struct proc *));
int audio_close __P((dev_t, int, int, struct proc *));
int audio_read __P((dev_t, struct uio *, int));
int audio_write __P((dev_t, struct uio *, int));
int audio_ioctl __P((dev_t, int, caddr_t, int, struct proc *));
int audio_poll __P((dev_t, int, struct proc *));
int audio_mmap __P((dev_t, int, int));
int mixer_open __P((dev_t, int, int, struct proc *));
int mixer_close __P((dev_t, int, int, struct proc *));
int mixer_ioctl __P((dev_t, int, caddr_t, int, struct proc *));
void audio_init_record __P((struct audio_softc *));
void audio_init_play __P((struct audio_softc *));
void audiostartr __P((struct audio_softc *));
void audiostartp __P((struct audio_softc *));
void audio_rint __P((void *));
void audio_pint __P((void *));
int audio_check_params __P((struct audio_params *));
void audio_calc_blksize __P((struct audio_softc *, int));
void audio_fill_silence __P((struct audio_params *, u_char *, int));
int audio_silence_copyout __P((struct audio_softc *, int, struct uio *));
int audio_hardware_attach __P((struct audio_hw_if *, void *, struct device *));
void audio_init_ringbuffer __P((struct audio_ringbuffer *));
void audio_initbufs __P((struct audio_softc *));
void audio_calcwater __P((struct audio_softc *));
static __inline int audio_sleep_timo __P((int *, char *, int));
static __inline int audio_sleep __P((int *, char *));
static __inline void audio_wakeup __P((int *));
int audio_drain __P((struct audio_softc *));
void audio_clear __P((struct audio_softc *));
static __inline void audio_pint_silence __P((struct audio_softc *, struct audio_ringbuffer *, u_char *, int));
int audio_alloc_ring __P((struct audio_softc *, struct audio_ringbuffer *, int));
void audio_free_ring __P((struct audio_softc *, struct audio_ringbuffer *));
/* The default audio mode: 8 kHz mono ulaw */
struct audio_params audio_default =
{ 8000, AUDIO_ENCODING_ULAW, 8, 1, 0, 1 };
#ifdef AUDIO_DEBUG
void audio_printsc __P((struct audio_softc *));
void audio_print_params __P((char *, struct audio_params *));
void
audio_printsc(sc)
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(s, p)
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(sc, r, bufsize)
struct audio_softc *sc;
struct audio_ringbuffer *r;
int bufsize;
{
struct audio_hw_if *hw = sc->hw_if;
void *hdl = sc->hw_hdl;
/*
* Alloc DMA play and record buffers
*/
ROUNDSIZE(bufsize);
if (bufsize < AUMINBUF)
bufsize = AUMINBUF;
if (hw->round_buffersize)
bufsize = hw->round_buffersize(hdl, bufsize);
r->bufsize = bufsize;
if (hw->alloc)
r->start = hw->alloc(hdl, r->bufsize, M_DEVBUF, M_WAITOK);
else
r->start = malloc(bufsize, M_DEVBUF, M_WAITOK);
if (r->start == 0)
return ENOMEM;
return 0;
}
void
audio_free_ring(sc, r)
struct audio_softc *sc;
struct audio_ringbuffer *r;
{
if (sc->hw_if->free) {
sc->hw_if->free(sc->hw_hdl, r->start, M_DEVBUF);
} else {
free(r->start, M_DEVBUF);
}
}
/*
* Called from hardware driver.
*/
int
audio_hardware_attach(hwp, hdlp, dev)
struct audio_hw_if *hwp;
void *hdlp;
struct device *dev;
{
struct audio_softc *sc;
int error;
int n;
/* Find a free slot. */
for(n = 0; n < naudio && audio_softc[n]; n++)
;
if (n >= naudio) {
/* No free slots, allocate one */
struct audio_softc **new;
naudio++;
new = malloc(naudio * sizeof(struct audio_softc *),
M_DEVBUF, M_WAITOK);
if (audio_softc) {
bcopy(audio_softc, new, n * sizeof(struct audio_softc *));
free(audio_softc, M_DEVBUF);
}
audio_softc = new;
audio_softc[n] = 0;
}
/* Malloc a softc for the device. */
sc = malloc(sizeof(struct audio_softc), M_DEVBUF, M_WAITOK);
bzero(sc, sizeof(struct audio_softc));
#ifdef DIAGNOSTIC
if (hwp == 0 ||
hwp->open == 0 ||
hwp->close == 0 ||
hwp->query_encoding == 0 ||
hwp->set_params == 0 ||
hwp->set_out_port == 0 ||
hwp->get_out_port == 0 ||
hwp->set_in_port == 0 ||
hwp->get_in_port == 0 ||
hwp->start_output == 0 ||
hwp->start_input == 0 ||
hwp->halt_output == 0 ||
hwp->halt_input == 0 ||
hwp->cont_output == 0 ||
hwp->cont_input == 0 ||
hwp->getdev == 0 ||
hwp->set_port == 0 ||
hwp->get_port == 0 ||
hwp->query_devinfo == 0 ||
hwp->get_props == 0) {
printf("audio: missing method\n");
free(sc, M_DEVBUF);
return(EINVAL);
}
#endif
sc->hw_if = hwp;
sc->hw_hdl = hdlp;
sc->sc_dev = dev;
error = audio_alloc_ring(sc, &sc->sc_pr, AU_RING_SIZE);
if (error)
return error;
error = audio_alloc_ring(sc, &sc->sc_rr, AU_RING_SIZE);
if (error) {
audio_free_ring(sc, &sc->sc_pr);
return error;
}
audio_softc[n] = sc;
/*
* Set default softc params
*/
sc->sc_pparams = audio_default;
sc->sc_rparams = audio_default;
printf("audio%d at %s\n", n, dev->dv_xname);
return(0);
}
int
audio_hardware_detach(hwp, hdlp)
struct audio_hw_if *hwp;
void *hdlp;
{
struct audio_softc *sc;
int n;
#ifdef DIAGNOSTIC
if (!hwp)
panic("audio_hardware_detach: null hwp");
#endif
for(sc = 0, n = 0; n < naudio; n++) {
sc = audio_softc[n];
if (sc && sc->hw_if == hwp && sc->hw_hdl == hdlp)
break;
}
if (n >= naudio)
return(EINVAL);
if (sc->sc_open)
return(EBUSY);
/* Free audio buffers */
audio_free_ring(sc, &sc->sc_rr);
audio_free_ring(sc, &sc->sc_pr);
free(sc, M_DEVBUF);
audio_softc[n] = 0;
return(0);
}
int
audioopen(dev, flags, ifmt, p)
dev_t dev;
int flags, ifmt;
struct proc *p;
{
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
return (audio_open(dev, flags, ifmt, p));
case MIXER_DEVICE:
return (mixer_open(dev, flags, ifmt, p));
default:
return (ENXIO);
}
}
int
audioclose(dev, flags, ifmt, p)
dev_t dev;
int flags, ifmt;
struct proc *p;
{
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
return (audio_close(dev, flags, ifmt, p));
case MIXER_DEVICE:
return (mixer_close(dev, flags, ifmt, p));
default:
return (ENXIO);
}
}
int
audioread(dev, uio, ioflag)
dev_t dev;
struct uio *uio;
int ioflag;
{
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
return (audio_read(dev, uio, ioflag));
case MIXER_DEVICE:
return (ENODEV);
default:
return (ENXIO);
}
}
int
audiowrite(dev, uio, ioflag)
dev_t dev;
struct uio *uio;
int ioflag;
{
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
return (audio_write(dev, uio, ioflag));
case MIXER_DEVICE:
return (ENODEV);
default:
return (ENXIO);
}
}
int
audioioctl(dev, cmd, addr, flag, p)
dev_t dev;
u_long cmd;
caddr_t addr;
int flag;
struct proc *p;
{
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
return (audio_ioctl(dev, cmd, addr, flag, p));
case MIXER_DEVICE:
return (mixer_ioctl(dev, cmd, addr, flag, p));
default:
return (ENXIO);
}
}
int
audiopoll(dev, events, p)
dev_t dev;
int events;
struct proc *p;
{
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
return (audio_poll(dev, events, p));
case MIXER_DEVICE:
return (0);
default:
return (0);
}
}
int
audiommap(dev, off, prot)
dev_t dev;
int off, prot;
{
switch (AUDIODEV(dev)) {
case SOUND_DEVICE:
case AUDIO_DEVICE:
return (audio_mmap(dev, off, prot));
case MIXER_DEVICE:
return -1;
default:
return -1;
}
}
/*
* Audio driver
*/
void
audio_init_ringbuffer(rp)
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;
}
void
audio_initbufs(sc)
struct audio_softc *sc;
{
struct audio_hw_if *hw = sc->hw_if;
DPRINTF(("audio_initbufs: mode=0x%x\n", sc->sc_mode));
audio_init_ringbuffer(&sc->sc_rr);
if (hw->init_input && (sc->sc_mode & AUMODE_RECORD))
hw->init_input(sc->hw_hdl, sc->sc_rr.start,
sc->sc_rr.end - sc->sc_rr.start);
audio_init_ringbuffer(&sc->sc_pr);
sc->sc_sil_count = 0;
if (hw->init_output && (sc->sc_mode & AUMODE_PLAY))
hw->init_output(sc->hw_hdl, sc->sc_pr.start,
sc->sc_pr.end - sc->sc_pr.start);
#ifdef AUDIO_INTR_TIME
sc->sc_pnintr = 0;
sc->sc_pblktime = (u_long)(
(double)sc->sc_pr.blksize * 1e6 /
(double)(sc->sc_pparams.precision / NBBY *
sc->sc_pparams.channels *
sc->sc_pparams.sample_rate));
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 * 1e6 /
(double)(sc->sc_rparams.precision / NBBY *
sc->sc_rparams.channels *
sc->sc_rparams.sample_rate));
DPRINTF(("audio: record blktime = %lu for %d\n",
sc->sc_rblktime, sc->sc_rr.blksize));
#endif
}
void
audio_calcwater(sc)
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 lowater 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(chan, label, timo)
int *chan;
char *label;
int timo;
{
int st;
if (!label)
label = "audio";
*chan = 1;
st = tsleep(chan, PWAIT | PCATCH, label, timo);
*chan = 0;
#ifdef AUDIO_DEBUG
if (st != 0)
printf("audio_sleep: %d\n", st);
#endif
return (st);
}
static __inline int
audio_sleep(chan, label)
int *chan;
char *label;
{
return audio_sleep_timo(chan, label, 0);
}
static __inline void
audio_wakeup(chan)
int *chan;
{
if (*chan) {
wakeup(chan);
*chan = 0;
}
}
int
audio_open(dev, flags, ifmt, p)
dev_t dev;
int flags, ifmt;
struct proc *p;
{
int unit = AUDIOUNIT(dev);
struct audio_softc *sc;
int error;
struct audio_hw_if *hw;
if (unit >= naudio || !audio_softc[unit]) {
DPRINTF(("audio_open: invalid device unit - %d\n", unit));
return (ENODEV);
}
sc = audio_softc[unit];
hw = sc->hw_if;
DPRINTF(("audio_open: dev=0x%x flags=0x%x sc=%p hdl=%p\n", dev, flags, sc, sc->hw_hdl));
#ifdef DIAGNOSTIC
if (hw == 0) { /* Hardware has not attached to us... */
printf("audio_open: hw==0\n");
return (ENXIO);
}
#endif
if ((sc->sc_open & (AUOPEN_READ|AUOPEN_WRITE)) != 0)
return (EBUSY);
error = hw->open(sc->hw_hdl, flags);
if (error)
return (error);
if (flags & FREAD)
sc->sc_open |= AUOPEN_READ;
if (flags & FWRITE)
sc->sc_open |= AUOPEN_WRITE;
/*
* 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;
if (flags & FREAD) {
error = hw->set_params(sc->hw_hdl, AUMODE_RECORD,
&sc->sc_rparams, &sc->sc_pparams);
if (error)
return (error);
}
if (flags & FWRITE) {
error = hw->set_params(sc->hw_hdl, AUMODE_PLAY,
&sc->sc_pparams, &sc->sc_rparams);
if (error)
return (error);
}
}
#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
sc->sc_blkset = 0; /* Block sizes not set yet */
if (flags & FREAD) {
audio_calc_blksize(sc, AUMODE_RECORD);
sc->sc_mode = AUMODE_RECORD;
}
if (flags & FWRITE) {
audio_calc_blksize(sc, AUMODE_PLAY);
sc->sc_mode = AUMODE_PLAY | AUMODE_PLAY_ALL;
}
audio_initbufs(sc);
audio_calcwater(sc);
sc->sc_playdrop = 0;
DPRINTF(("audio_open: rr.buf=%p-%p pr.buf=%p-%p\n",
sc->sc_rr.start, sc->sc_rr.end, sc->sc_pr.start, sc->sc_pr.end));
if (hw->commit_settings)
hw->commit_settings(sc->hw_hdl);
sc->sc_rchan = 0;
sc->sc_wchan = 0;
sc->sc_rbus = 0;
sc->sc_pbus = 0;
sc->sc_eof = 0;
if ((flags & (FWRITE|FREAD)) == (FWRITE|FREAD))
sc->sc_full_duplex =
(hw->get_props(sc->hw_hdl) & AUDIO_PROP_FULLDUPLEX) != 0;
if (flags & FWRITE)
audio_init_play(sc);
if (flags & FREAD) {
/* Play takes precedence if HW is half-duplex */
if (sc->sc_full_duplex || (flags & FWRITE) == 0)
audio_init_record(sc);
}
return (0);
}
/*
* Must be called from task context.
*/
void
audio_init_record(sc)
struct audio_softc *sc;
{
int s = splaudio();
sc->sc_mode |= AUMODE_RECORD;
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);
splx(s);
}
/*
* Must be called from task context.
*/
void
audio_init_play(sc)
struct audio_softc *sc;
{
int s = splaudio();
sc->sc_wstamp = sc->sc_pr.stamp;
sc->sc_mode |= AUMODE_PLAY;
if (sc->hw_if->speaker_ctl)
sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_ON);
splx(s);
}
int
audio_drain(sc)
struct audio_softc *sc;
{
int error, drops;
struct audio_ringbuffer *cb = &sc->sc_pr;
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 s, 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;
audiostartp(sc);
splx(s);
}
/*
* 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.
*/
drops = cb->drops;
while (cb->drops == drops) {
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 (error)
return (error);
}
return (0);
}
/*
* Close an audio chip.
*/
/* ARGSUSED */
int
audio_close(dev, flags, ifmt, p)
dev_t dev;
int flags, ifmt;
struct proc *p;
{
int unit = AUDIOUNIT(dev);
struct audio_softc *sc = audio_softc[unit];
struct audio_hw_if *hw = sc->hw_if;
int s;
DPRINTF(("audio_close: unit=%d\n", unit));
/*
* Block until output drains, but allow ^C interrupt.
*/
sc->sc_pr.usedlow = sc->sc_pr.blksize; /* avoid excessive wakeups */
s = splaudio();
/*
* If there is pending output, let it drain (unless
* the output is paused).
*/
if (!sc->sc_pr.pause) {
if (!audio_drain(sc) && hw->drain)
(void)hw->drain(sc->hw_hdl);
}
hw->close(sc->hw_hdl);
if (flags & FREAD)
sc->sc_open &= ~AUOPEN_READ;
if (flags & FWRITE)
sc->sc_open &= ~AUOPEN_WRITE;
sc->sc_async = 0;
splx(s);
DPRINTF(("audio_close: done\n"));
return (0);
}
int
audio_read(dev, uio, ioflag)
dev_t dev;
struct uio *uio;
int ioflag;
{
int unit = AUDIOUNIT(dev);
struct audio_softc *sc = audio_softc[unit];
struct audio_ringbuffer *cb = &sc->sc_rr;
u_char *outp;
int error, s, used, cc, n;
if (cb->mmapped)
return EINVAL;
DPRINTF(("audio_read: cc=%d mode=%d\n", uio->uio_resid, sc->sc_mode));
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 (error) {
splx(s);
return (error);
}
}
splx(s);
if (uio->uio_resid < cc)
cc = uio->uio_resid;
#ifdef AUDIO_DEBUG
if (audiodebug > 1)
printf("audio_read: reading in write mode, cc=%d\n", cc);
#endif
error = audio_silence_copyout(sc, cc, uio);
sc->sc_wstamp += cc;
}
return (error);
}
while (uio->uio_resid > 0 && !error) {
while (cb->used <= 0) {
if (ioflag & IO_NDELAY) {
error = EWOULDBLOCK;
return (error);
}
s = splaudio();
if (!sc->sc_rbus)
audiostartr(sc);
#ifdef AUDIO_DEBUG
if (audiodebug > 2)
printf("audio_read: sleep used=%d\n", cb->used);
#endif
error = audio_sleep(&sc->sc_rchan, "aud rd");
splx(s);
if (error)
return (error);
}
s = splaudio();
used = cb->used;
outp = cb->outp;
cb->copying = 1;
splx(s);
cc = used - cb->usedlow; /* maximum to read */
n = cb->end - outp;
if (n < cc)
cc = n; /* don't read beyond end of buffer */
if (uio->uio_resid < cc)
cc = uio->uio_resid; /* and no more than we want */
if (sc->sc_rparams.sw_code)
sc->sc_rparams.sw_code(sc->hw_hdl, outp, cc);
#ifdef AUDIO_DEBUG
if (audiodebug > 1)
printf("audio_read: outp=%p, cc=%d\n", outp, cc);
#endif
error = uiomove(outp, cc, uio);
used -= cc;
outp += cc;
if (outp >= cb->end)
outp = cb->start;
s = splaudio();
cb->outp = outp;
cb->used = used;
cb->copying = 0;
splx(s);
}
return (error);
}
void
audio_clear(sc)
struct audio_softc *sc;
{
int s = splaudio();
if (sc->sc_rbus) {
sc->hw_if->halt_input(sc->hw_hdl);
sc->sc_rbus = 0;
}
if (sc->sc_pbus) {
sc->hw_if->halt_output(sc->hw_hdl);
sc->sc_pbus = 0;
}
splx(s);
}
void
audio_calc_blksize(sc, mode)
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->sample_rate * audio_blk_ms / 1000 *
parm->channels * parm->precision / NBBY *
parm->factor;
ROUNDSIZE(bs);
if (hw->round_blocksize)
bs = hw->round_blocksize(sc->hw_hdl, bs);
rb->blksize = bs;
DPRINTF(("audio_calc_blksize: %s blksize=%d\n",
mode == AUMODE_PLAY ? "play" : "record", bs));
}
void
audio_fill_silence(params, p, n)
struct audio_params *params;
u_char *p;
int n;
{
u_char auzero0, auzero1 = 0; /* initialize to please gcc */
int nfill = 1;
switch (params->encoding) {
case AUDIO_ENCODING_ULAW:
auzero0 = 0x7f;
break;
case AUDIO_ENCODING_ALAW:
auzero0 = 0x55;
break;
case AUDIO_ENCODING_ADPCM: /* is this right XXX */
case AUDIO_ENCODING_SLINEAR_LE:
case AUDIO_ENCODING_SLINEAR_BE:
auzero0 = 0; /* fortunately this works for both 8 and 16 bits */
break;
case AUDIO_ENCODING_ULINEAR_LE:
case AUDIO_ENCODING_ULINEAR_BE:
if (params->precision == 16) {
nfill = 2;
if (params->encoding == AUDIO_ENCODING_ULINEAR_LE) {
auzero0 = 0;
auzero1 = 0x80;
} else {
auzero0 = 0x80;
auzero1 = 0;
}
} else
auzero0 = 0x80;
break;
default:
printf("audio: bad encoding %d\n", params->encoding);
auzero0 = 0;
break;
}
if (nfill == 1) {
while (--n >= 0)
*p++ = auzero0; /* XXX memset */
} else /* nfill must be 2 */ {
while (n > 1) {
*p++ = auzero0;
*p++ = auzero1;
n -= 2;
}
}
}
int
audio_silence_copyout(sc, n, uio)
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(dev, uio, ioflag)
dev_t dev;
struct uio *uio;
int ioflag;
{
int unit = AUDIOUNIT(dev);
struct audio_softc *sc = audio_softc[unit];
struct audio_ringbuffer *cb = &sc->sc_pr;
u_char *inp, *einp;
int error, s, n, cc, used;
DPRINTF(("audio_write: count=%d used=%d(hi=%d)\n", 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);
uio->uio_offset += n;
uio->uio_resid -= n;
sc->sc_playdrop -= n;
if (uio->uio_resid == 0)
return 0;
}
error = 0;
while (uio->uio_resid > 0 && !error) {
while (cb->used >= cb->usedhigh) {
DPRINTF(("audio_write: sleep used=%d lowat=%d hiwat=%d\n",
cb->used, cb->usedlow, cb->usedhigh));
if (ioflag & IO_NDELAY)
return (EWOULDBLOCK);
error = audio_sleep(&sc->sc_wchan, "aud wr");
if (error)
return (error);
}
s = splaudio();
used = cb->used;
inp = cb->inp;
cb->copying = 1;
splx(s);
cc = cb->usedhigh - used; /* maximum to write */
n = cb->end - inp;
if (sc->sc_pparams.factor != 1) {
/* Compensate for software coding expansion factor. */
n /= sc->sc_pparams.factor;
cc /= sc->sc_pparams.factor;
}
if (n < cc)
cc = n; /* don't write beyond end of buffer */
if (uio->uio_resid < cc)
cc = uio->uio_resid; /* and no more than we have */
#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, factor=%d\n",
sc->sc_pparams.factor);
return EINVAL;
}
#endif
#ifdef AUDIO_DEBUG
if (audiodebug > 1)
printf("audio_write: uiomove cc=%d inp=%p, left=%d\n", cc, inp, uio->uio_resid);
#endif
n = uio->uio_resid;
error = uiomove(inp, cc, uio);
cc = n - uio->uio_resid; /* number of bytes actually moved */
#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.
*/
if (sc->sc_pparams.sw_code) {
sc->sc_pparams.sw_code(sc->hw_hdl, inp, cc);
/* Adjust count after the expansion. */
cc *= sc->sc_pparams.factor;
#ifdef AUDIO_DEBUG
if (audiodebug > 1)
printf("audio_write: expanded cc=%d\n", cc);
#endif
}
einp = cb->inp + cc;
if (einp >= cb->end)
einp = cb->start;
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 there is 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)
audiostartp(sc);
splx(s);
if (cc) {
#ifdef AUDIO_DEBUG
if (audiodebug > 1)
printf("audio_write: fill %d\n", cc);
#endif
audio_fill_silence(&sc->sc_pparams, einp, cc);
}
}
return (error);
}
int
audio_ioctl(dev, cmd, addr, flag, p)
dev_t dev;
int cmd;
caddr_t addr;
int flag;
struct proc *p;
{
int unit = AUDIOUNIT(dev);
struct audio_softc *sc = audio_softc[unit];
struct audio_hw_if *hw = sc->hw_if;
struct audio_offset *ao;
int error = 0, s, offs;
DPRINTF(("audio_ioctl(%d,'%c',%d)\n",
IOCPARM_LEN(cmd), 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)
return (EBUSY);
sc->sc_async = p;
} else
sc->sc_async = 0;
break;
case AUDIO_FLUSH:
DPRINTF(("AUDIO_FLUSH\n"));
audio_clear(sc);
sc->sc_blkset = 0;
s = splaudio();
audio_initbufs(sc);
if ((sc->sc_mode & AUMODE_PLAY) && !sc->sc_pbus)
audiostartp(sc);
/* Again, play takes precedence on half-duplex hardware */
if ((sc->sc_mode & AUMODE_RECORD) &&
(sc->sc_full_duplex ||
((sc->sc_mode & AUMODE_PLAY) == 0)))
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:
DPRINTF(("AUDIO_SETINFO mode=0x%x\n", sc->sc_mode));
error = audiosetinfo(sc, (struct audio_info *)addr);
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;
#ifdef COMPAT_12
/* GETPROPS contains the same info (and more) */
case AUDIO_GETFD:
DPRINTF(("AUDIO_GETFD\n"));
*(int *)addr =
(hw->get_props(sc->hw_hdl) & AUDIO_PROP_FULLDUPLEX) != 0;
break;
#endif
case AUDIO_SETFD:
DPRINTF(("AUDIO_SETFD\n"));
if (hw->get_props(sc->hw_hdl) & AUDIO_PROP_FULLDUPLEX) {
if (hw->setfd)
error = hw->setfd(sc->hw_hdl, *(int *)addr);
else
error = 0;
if (!error)
sc->sc_full_duplex = *(int *)addr;
} else {
if (*(int *)addr)
error = ENOTTY;
else
error = 0;
}
break;
case AUDIO_GETPROPS:
DPRINTF(("AUDIO_GETPROPS\n"));
*(int *)addr = hw->get_props(sc->hw_hdl);
break;
default:
DPRINTF(("audio_ioctl: unknown ioctl\n"));
error = EINVAL;
break;
}
DPRINTF(("audio_ioctl(%d,'%c',%d) result %d\n",
IOCPARM_LEN(cmd), IOCGROUP(cmd), cmd&0xff, error));
return (error);
}
int
audio_poll(dev, events, p)
dev_t dev;
int events;
struct proc *p;
{
int unit = AUDIOUNIT(dev);
struct audio_softc *sc = audio_softc[unit];
int revents = 0;
int s = splaudio();
#if 0
DPRINTF(("audio_poll: events=%d mode=%d rr.nblk=%d\n",
events, sc->sc_mode, sc->sc_rr.nblk));
#endif
if (events & (POLLIN | POLLRDNORM))
if ((sc->sc_mode & AUMODE_PLAY) ?
0/*XXX*/ : sc->sc_rr.used > sc->sc_rr.usedlow)
revents |= events & (POLLIN | POLLRDNORM);
if (events & (POLLOUT | POLLWRNORM))
if (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);
}
int
audio_mmap(dev, off, prot)
dev_t dev;
int off, prot;
{
int s;
int unit = AUDIOUNIT(dev);
struct audio_softc *sc = audio_softc[unit];
struct audio_hw_if *hw = sc->hw_if;
struct audio_ringbuffer *cb;
DPRINTF(("audio_mmap: off=%d, prot=%d\n", 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 (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)
audiostartp(sc);
splx(s);
} else {
s = splaudio();
if (!sc->sc_rbus)
audiostartr(sc);
splx(s);
}
}
return hw->mappage(sc->hw_hdl, cb->start, off, prot);
}
void
audiostartr(sc)
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));
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));
audio_clear(sc);
} else
sc->sc_rbus = 1;
}
void
audiostartp(sc)
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.used >= sc->sc_pr.blksize || sc->sc_pr.mmapped) {
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));
} else {
sc->sc_pbus = 1;
}
}
}
/*
* 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 approcimate way: it starts at sc_sil_start
* and extends sc_sil_count bytes. If the writer doesn't write sc_sil_count
* get to encompass the whole buffer after which no more filling needs
* to be done. 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(sc, cb, inp, cc)
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 silence, 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);
#ifdef AUDIO_DEBUG
if (audiodebug > 2)
printf("audio_pint_silence: fill cc=%d inp=%p, count=%d size=%d\n",
cc, inp, sc->sc_sil_count, (int)(cb->end - cb->start));
#endif
audio_fill_silence(&sc->sc_pparams, inp, cc);
} else {
#ifdef AUDIO_DEBUG
if (audiodebug > 2)
printf("audio_pint_silence: already silent cc=%d inp=%p\n", cc, inp);
#endif
}
} else {
sc->sc_sil_start = inp;
sc->sc_sil_count = cc;
#ifdef AUDIO_DEBUG
if (audiodebug > 2)
printf("audio_pint_silence: start fill %p %d\n", inp, cc);
#endif
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(v)
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;
int error;
cb->outp += cb->blksize;
if (cb->outp >= cb->end)
cb->outp = cb->start;
cb->stamp += cb->blksize;
if (cb->mmapped) {
#ifdef AUDIO_DEBUG
if (audiodebug > 2)
printf("audio_pint: mmapped outp=%p cc=%d inp=%p\n",
cb->outp, cb->blksize, cb->inp);
#endif
hw->start_output(sc->hw_hdl, cb->outp, cb->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 / 5) {
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 / 2) {
sc->sc_pnintr++;
printf("audio: play interrupt(%d) off absolute by %ld us (%lu)\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 -= cb->blksize;
if (cb->used < cb->blksize) {
/* we don't have a full block to use */
if (cb->copying) {
/* writer is in progress, don't disturb */
cb->needfill = 1;
#ifdef AUDIO_DEBUG
if (audiodebug > 1)
printf("audio_pint: copying in progress\n");
#endif
} else {
inp = cb->inp;
cc = cb->blksize - (inp - cb->start) % cb->blksize;
if (cb->pause)
cb->pdrops += cc;
else {
cb->drops += cc;
sc->sc_playdrop += cc;
}
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, cb->blksize);
}
}
#ifdef AUDIO_DEBUG
if (audiodebug > 3)
printf("audio_pint: outp=%p cc=%d\n", cb->outp, cb->blksize);
#endif
error = hw->start_output(sc->hw_hdl, cb->outp, cb->blksize,
audio_pint, (void *)sc);
if (error) {
/* XXX does this really help? */
DPRINTF(("audio_pint restart failed: %d\n", error));
audio_clear(sc);
}
#ifdef AUDIO_DEBUG
if (audiodebug > 3)
printf("audio_pint: mode=%d pause=%d used=%d lowat=%d\n",
sc->sc_mode, cb->pause, cb->used, cb->usedlow);
#endif
if ((sc->sc_mode & AUMODE_PLAY) && !cb->pause) {
if (cb->used <= cb->usedlow) {
audio_wakeup(&sc->sc_wchan);
selwakeup(&sc->sc_wsel);
if (sc->sc_async)
psignal(sc->sc_async, SIGIO);
}
}
/* Possible to return one or more "phantom blocks" now. */
if (!sc->sc_full_duplex && sc->sc_rchan) {
audio_wakeup(&sc->sc_rchan);
selwakeup(&sc->sc_rsel);
if (sc->sc_async)
psignal(sc->sc_async, 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(v)
void *v;
{
struct audio_softc *sc = v;
struct audio_hw_if *hw = sc->hw_if;
struct audio_ringbuffer *cb = &sc->sc_rr;
int error;
cb->inp += cb->blksize;
if (cb->inp >= cb->end)
cb->inp = cb->start;
cb->stamp += cb->blksize;
if (cb->mmapped) {
#ifdef AUDIO_DEBUG
if (audiodebug > 2)
printf("audio_rint: mmapped inp=%p cc=%d\n",
cb->inp, cb->blksize);
#endif
hw->start_output(sc->hw_hdl, cb->inp, cb->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 += cb->blksize;
if (cb->pause) {
DPRINTF(("audio_rint: pdrops %lu\n", cb->pdrops));
cb->pdrops += cb->blksize;
cb->outp += cb->blksize;
cb->used -= cb->blksize;
} else if (cb->used + cb->blksize >= cb->usedhigh && !cb->copying) {
DPRINTF(("audio_rint: drops %lu\n", cb->drops));
cb->drops += cb->blksize;
cb->outp += cb->blksize;
cb->used -= cb->blksize;
}
#ifdef AUDIO_DEBUG
if (audiodebug > 2)
printf("audio_rint: inp=%p cc=%d used=%d\n",
cb->inp, cb->blksize, cb->used);
#endif
error = hw->start_input(sc->hw_hdl, cb->inp, cb->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);
selwakeup(&sc->sc_rsel);
if (sc->sc_async)
psignal(sc->sc_async, SIGIO);
}
int
audio_check_params(p)
struct audio_params *p;
{
#if defined(COMPAT_12)
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;
}
#endif
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:
case AUDIO_ENCODING_ADPCM:
if (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:
if (p->precision != 8 && p->precision != 16)
return (EINVAL);
break;
default:
return (EINVAL);
}
if (p->channels < 1 || p->channels > 8) /* sanity check # of channels */
return (EINVAL);
return (0);
}
int
audiosetinfo(sc, ai)
struct audio_softc *sc;
struct audio_info *ai;
{
struct audio_prinfo *r = &ai->record, *p = &ai->play;
int cleared = 0;
int s, error = 0;
struct audio_hw_if *hw = sc->hw_if;
mixer_ctrl_t ct;
struct audio_params pp, rp;
int np, nr;
unsigned int blks;
if (hw == 0) /* HW has not attached */
return(ENXIO);
pp = sc->sc_pparams;
rp = sc->sc_rparams;
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;
if (nr) {
if (!cleared)
audio_clear(sc);
cleared = 1;
rp.sw_code = 0;
rp.factor = 1;
error = hw->set_params(sc->hw_hdl, AUMODE_RECORD,
&rp, &sc->sc_pparams);
if (error)
return (error);
sc->sc_rparams = rp;
}
if (np) {
if (!cleared)
audio_clear(sc);
cleared = 1;
pp.sw_code = 0;
pp.factor = 1;
error = hw->set_params(sc->hw_hdl, AUMODE_PLAY,
&pp, &sc->sc_rparams);
if (error)
return (error);
sc->sc_pparams = pp;
}
/* 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 = hw->set_out_port(sc->hw_hdl, p->port);
if (error)
return(error);
}
if (r->port != ~0) {
if (!cleared)
audio_clear(sc);
cleared = 1;
error = hw->set_in_port(sc->hw_hdl, r->port);
if (error)
return(error);
}
if (p->gain != ~0) {
ct.dev = hw->get_out_port(sc->hw_hdl);
ct.type = AUDIO_MIXER_VALUE;
ct.un.value.num_channels = 1;
ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = p->gain;
error = hw->set_port(sc->hw_hdl, &ct);
if (error)
return(error);
}
if (r->gain != ~0) {
ct.dev = hw->get_in_port(sc->hw_hdl);
ct.type = AUDIO_MIXER_VALUE;
ct.un.value.num_channels = 1;
ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = r->gain;
error = hw->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) {
s = splaudio();
audiostartp(sc);
splx(s);
}
}
if (r->pause != (u_char)~0) {
sc->sc_rr.pause = r->pause;
if (!r->pause) {
s = splaudio();
audiostartr(sc);
splx(s);
}
}
if (ai->blocksize != ~0) {
/* Block size specified explicitly. */
if (!cleared)
audio_clear(sc);
cleared = 1;
/* No need to check the blocksize, audio_initbufs() does that. */
sc->sc_pr.blksize = ai->blocksize;
sc->sc_rr.blksize = ai->blocksize;
sc->sc_blkset = 1;
}
if (ai->mode != ~0) {
if (!cleared)
audio_clear(sc);
cleared = 1;
sc->sc_mode = ai->mode;
if (sc->sc_mode & AUMODE_PLAY) {
audio_init_play(sc);
if (!sc->sc_full_duplex) /* Play takes precedence */
sc->sc_mode &= ~AUMODE_RECORD;
}
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();
audio_initbufs(sc);
audio_calcwater(sc);
if (sc->sc_mode & AUMODE_PLAY)
audiostartp(sc);
if ((sc->sc_mode & AUMODE_RECORD) &&
(sc->sc_full_duplex ||
((sc->sc_mode & AUMODE_PLAY) == 0)))
audiostartr(sc);
splx(s);
}
/* 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 < 1)
blks = 1;
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;
}
return (0);
}
int
audiogetinfo(sc, ai)
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;
mixer_ctrl_t ct;
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 = hw->get_in_port(sc->hw_hdl);
p->port = hw->get_out_port(sc->hw_hdl);
ct.dev = r->port;
ct.type = AUDIO_MIXER_VALUE;
ct.un.value.num_channels = 1;
if (hw->get_port(sc->hw_hdl, &ct) == 0)
r->gain = ct.un.value.level[AUDIO_MIXER_LEVEL_MONO];
else
r->gain = AUDIO_MAX_GAIN/2;
ct.dev = p->port;
ct.un.value.num_channels = 1;
if (hw->get_port(sc->hw_hdl, &ct) == 0)
p->gain = ct.un.value.level[AUDIO_MIXER_LEVEL_MONO];
else
p->gain = AUDIO_MAX_GAIN/2;
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;
ai->buffersize = sc->sc_pr.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->backlog = 0; /* unused */
ai->mode = sc->sc_mode;
return (0);
}
/*
* Mixer driver
*/
int
mixer_open(dev, flags, ifmt, p)
dev_t dev;
int flags, ifmt;
struct proc *p;
{
int unit = AUDIOUNIT(dev);
struct audio_softc *sc;
if (unit >= naudio || !audio_softc[unit]) {
DPRINTF(("mixer_open: invalid device unit - %d\n", unit));
return (ENODEV);
}
sc = audio_softc[unit];
DPRINTF(("mixer_open: dev=0x%x flags=0x%x sc=%p\n", dev, flags, sc));
if (sc->hw_if == 0) /* Hardware has not attached to us... */
return (ENXIO);
return (0);
}
/*
* Close a mixer device
*/
/* ARGSUSED */
int
mixer_close(dev, flags, ifmt, p)
dev_t dev;
int flags, ifmt;
struct proc *p;
{
DPRINTF(("mixer_close: unit %d\n", AUDIOUNIT(dev)));
return (0);
}
int
mixer_ioctl(dev, cmd, addr, flag, p)
dev_t dev;
int cmd;
caddr_t addr;
int flag;
struct proc *p;
{
int unit = AUDIOUNIT(dev);
struct audio_softc *sc = audio_softc[unit];
struct audio_hw_if *hw = sc->hw_if;
int error = EINVAL;
DPRINTF(("mixer_ioctl(%d,'%c',%d)\n",
IOCPARM_LEN(cmd), IOCGROUP(cmd), cmd&0xff));
switch (cmd) {
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"));
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);
break;
default:
error = EINVAL;
break;
}
DPRINTF(("mixer_ioctl(%d,'%c',%d) result %d\n",
IOCPARM_LEN(cmd), IOCGROUP(cmd), cmd&0xff, error));
return (error);
}
#endif
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