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NetBSD/sys/dev/isa/aria.c

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/*-
* Copyright (c) 1995, 1996, 1998 Roland C. Dowdeswell. 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 Roland C. Dowdeswell.
* 4. The name of the authors may not be used to endorse or promote products
* derived from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHORS ``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 AUTHORS 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.
*/
/*-
* TODO:
* o Test the driver on cards other than a single
* Prometheus Aria 16.
* o Look into where aria_prometheus_kludge() belongs.
* o Add some dma code. It accomplishes its goal by
* direct IO at the moment.
* o Different programs should be able to open the device
* with O_RDONLY and O_WRONLY at the same time. But I
* do not see support for this in /sys/dev/audio.c, so
* I cannot effectively code it.
* o We should nicely deal with the cards that can do mulaw
* and alaw output.
* o Rework the mixer interface.
* o Deal with the lvls better. We need to do better mapping
* between logarithmic scales and the one byte that
* we are passed.
* o Deal better with cards that have no mixer.
*/
#include "aria.h"
#if NARIA > 0
#include <sys/param.h>
#include <sys/systm.h>
#include <sys/errno.h>
#include <sys/ioctl.h>
#include <sys/syslog.h>
#include <sys/device.h>
#include <sys/proc.h>
#include <sys/buf.h>
#include <vm/vm.h>
#include <machine/cpu.h>
#include <machine/pio.h>
#include <machine/bus.h>
#include <sys/audioio.h>
#include <dev/audio_if.h>
#include <dev/auconv.h>
#include <dev/mulaw.h>
#include <dev/isa/isavar.h>
#include <i386/isa/icu.h>
#include <dev/isa/ariareg.h>
#define FREAD 1
#define FWRITE 2
#ifdef AUDIO_DEBUG
#define DPRINTF(x) printf x
int ariadebug = 0;
#else
#define DPRINTF(x)
#endif
struct aria_mixdev_info {
u_char num_channels;
u_char level[2];
u_char mute;
};
struct aria_mixmaster {
u_char num_channels;
u_char level[2];
u_char treble[2];
u_char bass[2];
};
struct aria_softc {
struct device sc_dev; /* base device */
struct device *sc_isa; /* pointer to ISA parent */
void *sc_ih; /* interrupt vectoring */
bus_space_tag_t sc_iot; /* Tag on 'da bus. */
bus_space_handle_t sc_ioh; /* Handle of iospace */
u_short sc_open; /* reference count of open calls */
u_short sc_play; /* non-paused play chans 2**chan */
u_short sc_record; /* non-paused record chans 2**chan */
/* XXX -- keep this? */
u_short sc_gain[2]; /* left/right gain (play) */
u_long sc_rate; /* Sample rate for input and output */
u_int sc_encoding; /* audio encoding -- ulaw/linear */
int sc_chans; /* # of channels */
int sc_precision; /* # bits per sample */
u_long sc_interrupts; /* number of interrupts taken */
void (*sc_rintr)(void*); /* record transfer completion intr handler */
void (*sc_pintr)(void*); /* play transfer completion intr handler */
void *sc_rarg; /* arg for sc_rintr() */
void *sc_parg; /* arg for sc_pintr() */
int sc_blocksize; /* literal dio block size */
void *sc_rdiobuffer; /* record: where the next samples should be */
void *sc_pdiobuffer; /* play: where the next samples are */
u_short sc_hardware; /* bit field of hardware present */
#define ARIA_TELEPHONE 0x0001 /* has telephone input */
#define ARIA_MIXER 0x0002 /* has SC18075 digital mixer */
#define ARIA_MODEL 0x0004 /* is SC18025 (=0) or SC18026 (=1) */
struct aria_mixdev_info aria_mix[6];
struct aria_mixmaster ariamix_master;
u_char aria_mix_source;
int sc_sendcmd_err;
};
#ifdef __BROKEN_INDIRECT_CONFIG
int ariaprobe __P((struct device *, void *, void *));
#else
int ariaprobe __P((struct device *, struct cfdata *, void *));
#endif
void ariaattach __P((struct device *, struct device *, void *));
void ariaclose __P((void *));
int ariaopen __P((void *, int));
int ariareset __P((bus_space_tag_t, bus_space_handle_t));
int aria_reset __P((struct aria_softc *));
int aria_getdev __P((void *, struct audio_device *));
void aria_do_kludge __P((bus_space_tag_t, bus_space_handle_t,
bus_space_handle_t,
u_short, u_short, u_short, u_short));
void aria_prometheus_kludge __P((struct isa_attach_args *,
bus_space_handle_t));
int aria_query_encoding __P((void *, struct audio_encoding *));
int aria_round_blocksize __P((void *, int));
int aria_speaker_ctl __P((void *, int));
int aria_commit_settings __P((void *));
int aria_set_params __P((void *, int, int,
struct audio_params *, struct audio_params *));
int aria_get_props __P((void *));
int aria_start_output __P((void *, void *, int,
void (*) __P((void *)), void*));
int aria_start_input __P((void *, void *, int,
void (*) __P((void *)), void*));
int aria_halt_input __P((void *));
int aria_halt_output __P((void *));
int aria_sendcmd __P((struct aria_softc *, u_short, int, int, int));
u_short aria_getdspmem __P((struct aria_softc *, u_short));
void aria_putdspmem __P((struct aria_softc *, u_short, u_short));
int aria_intr __P((void *));
short ariaversion __P((struct aria_softc *));
void aria_set_mixer __P((struct aria_softc *, int));
void aria_mix_write __P((struct aria_softc *, int, int));
int aria_mix_read __P((struct aria_softc *, int));
int aria_mixer_set_port __P((void *, mixer_ctrl_t *));
int aria_mixer_get_port __P((void *, mixer_ctrl_t *));
int aria_mixer_query_devinfo __P((void *, mixer_devinfo_t *));
struct cfattach aria_ca = {
sizeof(struct aria_softc), ariaprobe, ariaattach
};
/* XXX temporary test for 1.3 */
#ifndef AudioNaux
/* 1.3 */
struct cfdriver aria_cd = {
NULL, "aria", DV_DULL
};
#endif
struct audio_device aria_device = {
"Aria 16(se)",
"x",
"aria"
};
/*
* Define our interface to the higher level audio driver.
*/
struct audio_hw_if aria_hw_if = {
ariaopen,
ariaclose,
NULL,
aria_query_encoding,
aria_set_params,
aria_round_blocksize,
aria_commit_settings,
NULL,
NULL,
aria_start_output,
aria_start_input,
aria_halt_input,
aria_halt_output,
NULL,
aria_getdev,
NULL,
aria_mixer_set_port,
aria_mixer_get_port,
aria_mixer_query_devinfo,
NULL,
NULL,
NULL,
NULL,
aria_get_props,
};
/*
* Probe / attach routines.
*/
/*
* Probe for the aria hardware.
*/
int
#ifdef __BROKEN_INDIRECT_CONFIG
ariaprobe(parent, match, aux)
struct device *parent;
void *match;
#else
ariaprobe(parent, cf, aux)
struct device *parent;
struct cfdata *cf;
#endif
void *aux;
{
bus_space_handle_t ioh;
struct isa_attach_args *ia = aux;
#ifdef __BROKEN_INDIRECT_CONFIG
struct cfdata *cf = ((struct device *)match)->dv_cfdata;
#endif
if (!ARIA_BASE_VALID(ia->ia_iobase)) {
printf("aria: configured iobase %d invalid\n", ia->ia_iobase);
return 0;
}
if (!ARIA_IRQ_VALID(ia->ia_irq)) {
printf("aria: configured irq %d invalid\n", ia->ia_irq);
return 0;
}
if (bus_space_map(ia->ia_iot, ia->ia_iobase, ARIADSP_NPORT, 0, &ioh)) {
DPRINTF(("aria: aria probe failed\n"));
return 0;
}
if (cf->cf_flags & 1)
aria_prometheus_kludge(ia, ioh);
if (ariareset(ia->ia_iot, ioh) != 0) {
DPRINTF(("aria: aria probe failed\n"));
bus_space_unmap(ia->ia_iot, ioh, ARIADSP_NPORT);
return 0;
}
bus_space_unmap(ia->ia_iot, ioh, ARIADSP_NPORT);
ia->ia_iosize = ARIADSP_NPORT;
DPRINTF(("aria: aria probe succeeded\n"));
return 1;
}
/*
* I didn't call this a kludge for
* nothing. This is cribbed from
* ariainit, the author of that
* disassembled some code to discover
* how to set up the initial values of
* the card. Without this, the card
* is dead. (It will not respond to _any_
* input at all.)
*
* ariainit can be found (ftp) at:
* ftp://ftp.wi.leidenuniv.nl/pub/audio/aria/programming/contrib/ariainit.zip
* currently.
*/
void
aria_prometheus_kludge(ia, ioh1)
struct isa_attach_args *ia;
bus_space_handle_t ioh1;
{
bus_space_tag_t iot;
bus_space_handle_t ioh;
u_short end;
DPRINTF(("aria: begin aria_prometheus_kludge\n"));
/* Begin Config Sequence */
iot = ia->ia_iot;
bus_space_map(iot, 0x200, 8, 0, &ioh);
bus_space_write_1(iot, ioh, 4, 0x4c);
bus_space_write_1(iot, ioh, 5, 0x42);
bus_space_write_1(iot, ioh, 6, 0x00);
bus_space_write_2(iot, ioh, 0, 0x0f);
bus_space_write_1(iot, ioh, 1, 0x00);
bus_space_write_2(iot, ioh, 0, 0x02);
bus_space_write_1(iot, ioh, 1, ia->ia_iobase>>2);
/*
* These next three lines set up the iobase
* and the irq; and disable the drq.
*/
aria_do_kludge(iot, ioh, ioh1, 0x111, ((ia->ia_iobase-0x280)>>2)+0xA0,
0xbf, 0xa0);
aria_do_kludge(iot, ioh, ioh1, 0x011, ia->ia_irq-6, 0xf8, 0x00);
aria_do_kludge(iot, ioh, ioh1, 0x011, 0x00, 0xef, 0x00);
/* The rest of these lines just disable everything else */
aria_do_kludge(iot, ioh, ioh1, 0x113, 0x00, 0x88, 0x00);
aria_do_kludge(iot, ioh, ioh1, 0x013, 0x00, 0xf8, 0x00);
aria_do_kludge(iot, ioh, ioh1, 0x013, 0x00, 0xef, 0x00);
aria_do_kludge(iot, ioh, ioh1, 0x117, 0x00, 0x88, 0x00);
aria_do_kludge(iot, ioh, ioh1, 0x017, 0x00, 0xff, 0x00);
/* End Sequence */
bus_space_write_1(iot, ioh, 0, 0x0f);
end = bus_space_read_1(iot, ioh1, 0);
bus_space_write_2(iot, ioh, 0, 0x0f);
bus_space_write_1(iot, ioh, 1, end|0x80);
bus_space_read_1(iot, ioh, 0);
bus_space_unmap(iot, ioh, 8);
/*
* This delay is necessary for some reason,
* at least it would crash, and sometimes not
* probe properly if it did not exist.
*/
delay(1000000);
}
void
aria_do_kludge(iot, ioh, ioh1, func, bits, and, or)
bus_space_tag_t iot;
bus_space_handle_t ioh;
bus_space_handle_t ioh1;
u_short func;
u_short bits;
u_short and;
u_short or;
{
u_int i;
if (func & 0x100) {
func &= ~0x100;
if (bits) {
bus_space_write_2(iot, ioh, 0, func-1);
bus_space_write_1(iot, ioh, 1, bits);
}
} else
or |= bits;
bus_space_write_1(iot, ioh, 0, func);
i = bus_space_read_1(iot, ioh1, 0);
bus_space_write_2(iot, ioh, 0, func);
bus_space_write_1(iot, ioh, 1, (i&and) | or);
}
/*
* Attach hardware to driver, attach hardware driver to audio
* pseudo-device driver.
*/
void
ariaattach(parent, self, aux)
struct device *parent, *self;
void *aux;
{
bus_space_handle_t ioh;
struct aria_softc *sc = (void *)self;
struct isa_attach_args *ia = aux;
u_short i;
if (bus_space_map(ia->ia_iot, ia->ia_iobase, ARIADSP_NPORT, 0, &ioh))
panic("%s: can map io port range", self->dv_xname);
sc->sc_iot = ia->ia_iot;
sc->sc_ioh = ioh;
sc->sc_isa = parent;
sc->sc_ih = isa_intr_establish(ia->ia_ic, ia->ia_irq, IST_EDGE,
IPL_AUDIO, aria_intr, sc);
DPRINTF(("isa_intr_establish() returns (%x)\n", (unsigned) sc->sc_ih));
i = aria_getdspmem(sc, ARIAA_HARDWARE_A);
sc->sc_hardware = 0;
sc->sc_hardware |= ((i>>13)&0x01)==1 ? ARIA_TELEPHONE:0;
sc->sc_hardware |= (((i>>5)&0x07))==0x04 ? ARIA_MIXER:0;
sc->sc_hardware |= (aria_getdspmem(sc, ARIAA_MODEL_A)>=1)?ARIA_MODEL:0;
sc->sc_open = 0;
sc->sc_play = 0;
sc->sc_record = 0;
sc->sc_rate = 7875;
sc->sc_chans = 1;
sc->sc_blocksize = 1024;
sc->sc_precision = 8;
sc->sc_rintr = 0;
sc->sc_rarg = 0;
sc->sc_pintr = 0;
sc->sc_parg = 0;
sc->sc_gain[0] = 127;
sc->sc_gain[1] = 127;
for (i=0; i<6; i++) {
if (i == ARIAMIX_TEL_LVL)
sc->aria_mix[i].num_channels = 1;
else
sc->aria_mix[i].num_channels = 2;
sc->aria_mix[i].level[0] = 127;
sc->aria_mix[i].level[1] = 127;
}
sc->ariamix_master.num_channels = 2;
sc->ariamix_master.level[0] = 222;
sc->ariamix_master.level[1] = 222;
sc->ariamix_master.bass[0] = 127;
sc->ariamix_master.bass[1] = 127;
sc->ariamix_master.treble[0] = 127;
sc->ariamix_master.treble[1] = 127;
sc->aria_mix_source = 0;
aria_commit_settings(sc);
printf(": dsp %s", (ARIA_MODEL&sc->sc_hardware)?"SC18026":"SC18025");
if (ARIA_TELEPHONE&sc->sc_hardware)
printf(", tel");
if (ARIA_MIXER&sc->sc_hardware)
printf(", SC18075 mixer");
printf("\n");
sprintf(aria_device.version, "%s",
ARIA_MODEL & sc->sc_hardware ? "SC18026" : "SC18025");
audio_attach_mi(&aria_hw_if, 0, (void *)sc, &sc->sc_dev);
}
/*
* Various routines to interface to higher level audio driver
*/
int
ariaopen(addr, flags)
void *addr;
int flags;
{
struct aria_softc *sc = addr;
DPRINTF(("ariaopen() called\n"));
if (!sc)
return ENXIO;
if ((flags&FREAD) && (sc->sc_open & ARIAR_OPEN_RECORD))
return ENXIO;
if ((flags&FWRITE) && (sc->sc_open & ARIAR_OPEN_PLAY))
return ENXIO;
if (flags&FREAD)
sc->sc_open |= ARIAR_OPEN_RECORD;
if (flags&FWRITE)
sc->sc_open |= ARIAR_OPEN_PLAY;
sc->sc_play = 0;
sc->sc_record= 0;
sc->sc_rintr = 0;
sc->sc_rarg = 0;
sc->sc_pintr = 0;
sc->sc_parg = 0;
return 0;
}
int
aria_getdev(addr, retp)
void *addr;
struct audio_device *retp;
{
*retp = aria_device;
return 0;
}
/*
* Various routines to interface to higher level audio driver
*/
int
aria_query_encoding(addr, fp)
void *addr;
struct audio_encoding *fp;
{
struct aria_softc *sc = addr;
switch (fp->index) {
case 0:
strcpy(fp->name, AudioEmulaw);
fp->encoding = AUDIO_ENCODING_ULAW;
fp->precision = 8;
if ((ARIA_MODEL&sc->sc_hardware) == 0)
fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
break;
case 1:
strcpy(fp->name, AudioEalaw);
fp->encoding = AUDIO_ENCODING_ALAW;
fp->precision = 8;
if ((ARIA_MODEL&sc->sc_hardware) == 0)
fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
break;
case 2:
strcpy(fp->name, AudioEslinear);
fp->encoding = AUDIO_ENCODING_SLINEAR;
fp->precision = 8;
fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
break;
case 3:
strcpy(fp->name, AudioEslinear_le);
fp->encoding = AUDIO_ENCODING_SLINEAR_LE;
fp->precision = 16;
fp->flags = 0;
break;
case 4:
strcpy(fp->name, AudioEslinear_be);
fp->encoding = AUDIO_ENCODING_SLINEAR_BE;
fp->precision = 16;
fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
break;
case 5:
strcpy(fp->name, AudioEulinear);
fp->encoding = AUDIO_ENCODING_ULINEAR;
fp->precision = 8;
fp->flags = 0;
break;
case 6:
strcpy(fp->name, AudioEulinear_le);
fp->encoding = AUDIO_ENCODING_ULINEAR_LE;
fp->precision = 16;
fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
break;
case 7:
strcpy(fp->name, AudioEulinear_be);
fp->encoding = AUDIO_ENCODING_ULINEAR_BE;
fp->precision = 16;
fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
break;
default:
return(EINVAL);
/*NOTREACHED*/
}
return (0);
}
/*
* Store blocksize in bytes.
*/
int
aria_round_blocksize(addr, blk)
void *addr;
int blk;
{
int i;
#if 0 /* XXX -- this is being a tad bit of a problem... */
for (i=64; i<1024; i*=2)
if (blk <= i)
break;
#else
i = 1024;
#endif
return(i);
}
int
aria_get_props(addr)
void *addr;
{
return AUDIO_PROP_FULLDUPLEX;
}
int
aria_set_params(addr, setmode, usemode, p, r)
void *addr;
int setmode, usemode;
struct audio_params *p, *r;
{
struct aria_softc *sc = addr;
switch(p->encoding) {
case AUDIO_ENCODING_ULAW:
case AUDIO_ENCODING_ALAW:
case AUDIO_ENCODING_SLINEAR:
case AUDIO_ENCODING_SLINEAR_LE:
case AUDIO_ENCODING_SLINEAR_BE:
case AUDIO_ENCODING_ULINEAR:
case AUDIO_ENCODING_ULINEAR_LE:
case AUDIO_ENCODING_ULINEAR_BE:
break;
default:
return (EINVAL);
}
if (p->sample_rate <= 9450)
p->sample_rate = 7875;
else if (p->sample_rate <= 13387)
p->sample_rate = 11025;
else if (p->sample_rate <= 18900)
p->sample_rate = 15750;
else if (p->sample_rate <= 26775)
p->sample_rate = 22050;
else if (p->sample_rate <= 37800)
p->sample_rate = 31500;
else
p->sample_rate = 44100;
sc->sc_encoding = p->encoding;
sc->sc_precision = p->precision;
sc->sc_chans = p->channels;
sc->sc_rate = p->sample_rate;
switch(p->encoding) {
case AUDIO_ENCODING_ULAW:
if ((ARIA_MODEL&sc->sc_hardware) == 0) {
p->sw_code = mulaw_to_ulinear8;
r->sw_code = ulinear8_to_mulaw;
}
break;
case AUDIO_ENCODING_ALAW:
if ((ARIA_MODEL&sc->sc_hardware) == 0) {
p->sw_code = alaw_to_ulinear8;
r->sw_code = ulinear8_to_alaw;
}
break;
case AUDIO_ENCODING_SLINEAR:
p->sw_code = r->sw_code = change_sign8;
break;
case AUDIO_ENCODING_ULINEAR_LE:
p->sw_code = r->sw_code = change_sign16;
break;
case AUDIO_ENCODING_SLINEAR_BE:
p->sw_code = r->sw_code = swap_bytes;
break;
case AUDIO_ENCODING_ULINEAR_BE:
p->sw_code = r->sw_code = swap_bytes_change_sign16;
break;
}
return 0;
}
/*
* This is where all of the twiddling goes on.
*/
int
aria_commit_settings(addr)
void *addr;
{
struct aria_softc *sc = addr;
bus_space_tag_t iot = sc->sc_iot;
bus_space_handle_t ioh = sc->sc_ioh;
static u_char tones[16] =
{ 7, 6, 5, 4, 3, 2, 1, 0, 8, 9, 10, 11, 12, 13, 14, 15 };
u_short format;
u_short left, right;
u_short samp;
u_char i;
DPRINTF(("aria_commit_settings\n"));
switch (sc->sc_rate) {
case 7875: format = 0x00; samp = 0x60; break;
case 11025: format = 0x00; samp = 0x40; break;
case 15750: format = 0x10; samp = 0x60; break;
case 22050: format = 0x10; samp = 0x40; break;
case 31500: format = 0x10; samp = 0x20; break;
case 44100: format = 0x20; samp = 0x00; break;
default: format = 0x00; samp = 0x40; break;/* XXX can we get here? */
}
if ((ARIA_MODEL&sc->sc_hardware) != 0) {
format |= (sc->sc_encoding==AUDIO_ENCODING_ULAW)?0x06:0x00;
format |= (sc->sc_encoding==AUDIO_ENCODING_ALAW)?0x08:0x00;
}
format |= (sc->sc_precision==16)?0x02:0x00;
format |= (sc->sc_chans==2)?1:0;
samp |= bus_space_read_2(iot, ioh, ARIADSP_STATUS) & ~0x60;
aria_sendcmd(sc, ARIADSPC_FORMAT, format, -1, -1);
bus_space_write_2(iot, ioh, ARIADSP_CONTROL, samp);
if (sc->sc_hardware&ARIA_MIXER) {
for (i = 0; i < 6; i++)
aria_set_mixer(sc, i);
if (sc->sc_chans==2) {
aria_sendcmd(sc, ARIADSPC_CHAN_VOL, ARIAR_PLAY_CHAN,
((sc->sc_gain[0]+sc->sc_gain[1])/2)<<7,
-1);
aria_sendcmd(sc, ARIADSPC_CHAN_PAN, ARIAR_PLAY_CHAN,
(sc->sc_gain[0]-sc->sc_gain[1])/4+0x40,
-1);
} else {
aria_sendcmd(sc, ARIADSPC_CHAN_VOL, ARIAR_PLAY_CHAN,
sc->sc_gain[0]<<7, -1);
aria_sendcmd(sc, ARIADSPC_CHAN_PAN, ARIAR_PLAY_CHAN,
0x40, -1);
}
aria_sendcmd(sc, ARIADSPC_MASMONMODE,
sc->ariamix_master.num_channels != 2, -1, -1);
aria_sendcmd(sc, ARIADSPC_MIXERVOL, 0x0004,
sc->ariamix_master.level[0] << 7,
sc->ariamix_master.level[1] << 7);
/* Convert treble/bass from byte to soundcard style */
left = (tones[(sc->ariamix_master.treble[0]>>4)&0x0f]<<8) |
tones[(sc->ariamix_master.bass[0]>>4)&0x0f];
right = (tones[(sc->ariamix_master.treble[1]>>4)&0x0f]<<8) |
tones[(sc->ariamix_master.bass[1]>>4)&0x0f];
aria_sendcmd(sc, ARIADSPC_TONE, left, right, -1);
}
aria_sendcmd(sc, ARIADSPC_BLOCKSIZE, sc->sc_blocksize/2, -1, -1);
/*
* If we think that the card is recording or playing, start it up again here.
* Some of the previous commands turn the channels off.
*/
if (sc->sc_record&(1<<ARIAR_RECORD_CHAN))
aria_sendcmd(sc, ARIADSPC_START_REC, ARIAR_RECORD_CHAN, -1,-1);
if (sc->sc_play&(1<<ARIAR_PLAY_CHAN))
aria_sendcmd(sc, ARIADSPC_START_PLAY, ARIAR_PLAY_CHAN, -1, -1);
return(0);
}
void
aria_set_mixer(sc, i)
struct aria_softc *sc;
int i;
{
u_char source;
switch(i) {
case ARIAMIX_MIC_LVL: source = 0x0001; break;
case ARIAMIX_CD_LVL: source = 0x0002; break;
case ARIAMIX_LINE_IN_LVL: source = 0x0008; break;
case ARIAMIX_TEL_LVL: source = 0x0020; break;
case ARIAMIX_AUX_LVL: source = 0x0010; break;
case ARIAMIX_DAC_LVL: source = 0x0004; break;
default: source = 0x0000; break;
}
if (source != 0x0000 && source != 0x0004) {
if (sc->aria_mix[i].mute == 1)
aria_sendcmd(sc, ARIADSPC_INPMONMODE, source, 3, -1);
else
aria_sendcmd(sc, ARIADSPC_INPMONMODE, source,
sc->aria_mix[i].num_channels != 2, -1);
aria_sendcmd(sc, ARIADSPC_INPMONMODE, 0x8000|source,
sc->aria_mix[i].num_channels != 2, -1);
aria_sendcmd(sc, ARIADSPC_MIXERVOL, source,
sc->aria_mix[i].level[0] << 7,
sc->aria_mix[i].level[1] << 7);
}
if (sc->aria_mix_source == i) {
aria_sendcmd(sc, ARIADSPC_ADCSOURCE, source, -1, -1);
if (sc->sc_open & ARIAR_OPEN_RECORD)
aria_sendcmd(sc, ARIADSPC_ADCCONTROL, 1, -1, -1);
else
aria_sendcmd(sc, ARIADSPC_ADCCONTROL, 0, -1, -1);
}
}
void
ariaclose(addr)
void *addr;
{
struct aria_softc *sc = addr;
DPRINTF(("aria_close sc=0x%x\n", (unsigned) sc));
sc->sc_rintr = 0;
sc->sc_pintr = 0;
sc->sc_rdiobuffer = 0;
sc->sc_pdiobuffer = 0;
if (sc->sc_play&(1<<ARIAR_PLAY_CHAN) &&
sc->sc_open & ARIAR_OPEN_PLAY) {
aria_sendcmd(sc, ARIADSPC_STOP_PLAY, ARIAR_PLAY_CHAN, -1, -1);
sc->sc_play &= ~(1<<ARIAR_PLAY_CHAN);
}
if (sc->sc_record&(1<<ARIAR_RECORD_CHAN) &&
sc->sc_open & ARIAR_OPEN_RECORD) {
aria_sendcmd(sc, ARIADSPC_STOP_REC, ARIAR_RECORD_CHAN, -1, -1);
sc->sc_record &= ~(1<<ARIAR_RECORD_CHAN);
}
sc->sc_open = 0;
if (aria_reset(sc) != 0) {
delay(500);
aria_reset(sc);
}
}
/*
* Reset the hardware.
*/
int ariareset(iot, ioh)
bus_space_tag_t iot;
bus_space_handle_t ioh;
{
struct aria_softc tmp, *sc = &tmp;
sc->sc_iot = iot;
sc->sc_ioh = ioh;
return aria_reset(sc);
}
int
aria_reset(sc)
struct aria_softc *sc;
{
bus_space_tag_t iot = sc->sc_iot;
bus_space_handle_t ioh = sc->sc_ioh;
int fail=0;
int i;
bus_space_write_2(iot, ioh, ARIADSP_CONTROL,
ARIAR_ARIA_SYNTH | ARIAR_SR22K|ARIAR_DSPINTWR);
aria_putdspmem(sc, 0x6102, 0);
fail |= aria_sendcmd(sc, ARIADSPC_SYSINIT, 0x0000, 0x0000, 0x0000);
for (i=0; i < ARIAR_NPOLL; i++)
if (aria_getdspmem(sc, ARIAA_TASK_A) == 1)
break;
bus_space_write_2(iot, ioh, ARIADSP_CONTROL,
ARIAR_ARIA_SYNTH|ARIAR_SR22K | ARIAR_DSPINTWR |
ARIAR_PCINTWR);
fail |= aria_sendcmd(sc, ARIADSPC_MODE, ARIAV_MODE_NO_SYNTH,-1,-1);
return (fail);
}
/*
* Lower-level routines
*/
void
aria_putdspmem(sc, loc, val)
struct aria_softc *sc;
u_short loc;
u_short val;
{
bus_space_tag_t iot = sc->sc_iot;
bus_space_handle_t ioh = sc->sc_ioh;
bus_space_write_2(iot, ioh, ARIADSP_DMAADDRESS, loc);
bus_space_write_2(iot, ioh, ARIADSP_DMADATA, val);
}
u_short
aria_getdspmem(sc, loc)
struct aria_softc *sc;
u_short loc;
{
bus_space_tag_t iot = sc->sc_iot;
bus_space_handle_t ioh = sc->sc_ioh;
bus_space_write_2(iot, ioh, ARIADSP_DMAADDRESS, loc);
return bus_space_read_2(iot, ioh, ARIADSP_DMADATA);
}
/*
* aria_sendcmd()
* each full DSP command is unified into this
* function.
*/
#define ARIASEND(data, flag) \
for (i = ARIAR_NPOLL; \
(bus_space_read_2(iot, ioh, ARIADSP_STATUS) & ARIAR_BUSY) && i>0; \
i--) \
; \
if (bus_space_read_2(iot, ioh, ARIADSP_STATUS) & ARIAR_BUSY) \
fail |= flag; \
bus_space_write_2(iot, ioh, ARIADSP_WRITE, (u_short)data)
int
aria_sendcmd(sc, command, arg1, arg2, arg3)
struct aria_softc *sc;
u_short command;
int arg1;
int arg2;
int arg3;
{
bus_space_tag_t iot = sc->sc_iot;
bus_space_handle_t ioh = sc->sc_ioh;
int i, fail = 0;
ARIASEND(command, 1);
if (arg1 != -1) {
ARIASEND(arg1, 2);
}
if (arg2 != -1) {
ARIASEND(arg2, 4);
}
if (arg3 != -1) {
ARIASEND(arg3, 8);
}
ARIASEND(ARIADSPC_TERM, 16);
if (fail) {
sc->sc_sendcmd_err++;
#ifdef AUDIO_DEBUG
DPRINTF(("aria_sendcmd: failure=(%d) cmd=(0x%x) fail=(0x%x)\n",
sc->sc_sendcmd_err, command, fail));
#endif
return -1;
}
return 0;
}
#undef ARIASEND
int
aria_halt_input(addr)
void *addr;
{
struct aria_softc *sc = addr;
DPRINTF(("aria_halt_input\n"));
if (sc->sc_record & (1<<0)) {
aria_sendcmd(sc, ARIADSPC_STOP_REC, 0, -1, -1);
sc->sc_record &= ~(1<<0);
}
return(0);
}
int
aria_halt_output(addr)
void *addr;
{
struct aria_softc *sc = addr;
DPRINTF(("aria_halt_output\n"));
if (sc->sc_play & (1<<1)) {
aria_sendcmd(sc, ARIADSPC_STOP_PLAY, 1, -1, -1);
sc->sc_play &= ~(1<<1);
}
return(0);
}
/*
* Here we just set up the buffers. If we receive
* an interrupt without these set, it is ignored.
*/
int
aria_start_input(addr, p, cc, intr, arg)
void *addr;
void *p;
int cc;
void (*intr) __P((void *));
void *arg;
{
struct aria_softc *sc = addr;
DPRINTF(("aria_start_input %d @ %x\n", cc, (unsigned) p));
if (cc != sc->sc_blocksize) {
DPRINTF(("aria_start_input reqsize %d not sc_blocksize %d\n",
cc, sc->sc_blocksize));
return EINVAL;
}
sc->sc_rarg = arg;
sc->sc_rintr = intr;
sc->sc_rdiobuffer = p;
if (!(sc->sc_record&(1<<ARIAR_RECORD_CHAN))) {
aria_sendcmd(sc, ARIADSPC_START_REC, ARIAR_RECORD_CHAN, -1,-1);
sc->sc_record |= (1<<ARIAR_RECORD_CHAN);
}
return 0;
}
int
aria_start_output(addr, p, cc, intr, arg)
void *addr;
void *p;
int cc;
void (*intr) __P((void *));
void *arg;
{
struct aria_softc *sc = addr;
DPRINTF(("aria_start_output %d @ %x\n", cc, (unsigned) p));
if (cc != sc->sc_blocksize) {
DPRINTF(("aria_start_output reqsize %d not sc_blocksize %d\n",
cc, sc->sc_blocksize));
return EINVAL;
}
sc->sc_parg = arg;
sc->sc_pintr = intr;
sc->sc_pdiobuffer = p;
if (!(sc->sc_play&(1<<ARIAR_PLAY_CHAN))) {
aria_sendcmd(sc, ARIADSPC_START_PLAY, ARIAR_PLAY_CHAN, -1, -1);
sc->sc_play |= (1<<ARIAR_PLAY_CHAN);
}
return 0;
}
/*
* Process an interrupt. This should be a
* request (from the card) to write or read
* samples.
*/
int
aria_intr(arg)
void *arg;
{
struct aria_softc *sc = arg;
bus_space_tag_t iot = sc->sc_iot;
bus_space_handle_t ioh = sc->sc_ioh;
u_short *pdata = sc->sc_pdiobuffer;
u_short *rdata = sc->sc_rdiobuffer;
u_short address;
#if 0 /* XXX -- BAD BAD BAD (That this is #define'd out */
DPRINTF(("Checking to see if this is our intr\n"));
if ((inw(iobase) & 1) != 0x1)
return 0; /* not for us */
#endif
sc->sc_interrupts++;
DPRINTF(("aria_intr\n"));
if ((sc->sc_open & ARIAR_OPEN_PLAY) && (pdata!=NULL)) {
DPRINTF(("aria_intr play=(%x)\n", (unsigned) pdata));
address = 0x8000 - 2*(sc->sc_blocksize);
address+= aria_getdspmem(sc, ARIAA_PLAY_FIFO_A);
bus_space_write_2(iot, ioh, ARIADSP_DMAADDRESS, address);
bus_space_write_multi_2(iot, ioh, ARIADSP_DMADATA, pdata,
sc->sc_blocksize / 2);
if (sc->sc_pintr != NULL)
(*sc->sc_pintr)(sc->sc_parg);
}
if ((sc->sc_open & ARIAR_OPEN_RECORD) && (rdata!=NULL)) {
DPRINTF(("aria_intr record=(%x)\n", (unsigned) rdata));
address = 0x8000 - (sc->sc_blocksize);
address+= aria_getdspmem(sc, ARIAA_REC_FIFO_A);
bus_space_write_2(iot, ioh, ARIADSP_DMAADDRESS, address);
bus_space_read_multi_2(iot, ioh, ARIADSP_DMADATA, rdata,
sc->sc_blocksize / 2);
if (sc->sc_rintr != NULL)
(*sc->sc_rintr)(sc->sc_rarg);
}
aria_sendcmd(sc, ARIADSPC_TRANSCOMPLETE, -1, -1, -1);
return 1;
}
int
aria_mixer_set_port(addr, cp)
void *addr;
mixer_ctrl_t *cp;
{
struct aria_softc *sc = addr;
int error = EINVAL;
DPRINTF(("aria_mixer_set_port\n"));
/* This could be done better, no mixer still has some controls. */
if (!(ARIA_MIXER & sc->sc_hardware))
return ENXIO;
if (cp->type == AUDIO_MIXER_VALUE) {
mixer_level_t *mv = &cp->un.value;
switch (cp->dev) {
case ARIAMIX_MIC_LVL:
if (mv->num_channels == 1 || mv->num_channels == 2) {
sc->aria_mix[ARIAMIX_MIC_LVL].num_channels =
mv->num_channels;
sc->aria_mix[ARIAMIX_MIC_LVL].level[0] =
mv->level[0];
sc->aria_mix[ARIAMIX_MIC_LVL].level[1] =
mv->level[1];
error = 0;
}
break;
case ARIAMIX_LINE_IN_LVL:
if (mv->num_channels == 1 || mv->num_channels == 2) {
sc->aria_mix[ARIAMIX_LINE_IN_LVL].num_channels=
mv->num_channels;
sc->aria_mix[ARIAMIX_LINE_IN_LVL].level[0] =
mv->level[0];
sc->aria_mix[ARIAMIX_LINE_IN_LVL].level[1] =
mv->level[1];
error = 0;
}
break;
case ARIAMIX_CD_LVL:
if (mv->num_channels == 1 || mv->num_channels == 2) {
sc->aria_mix[ARIAMIX_CD_LVL].num_channels =
mv->num_channels;
sc->aria_mix[ARIAMIX_CD_LVL].level[0] =
mv->level[0];
sc->aria_mix[ARIAMIX_CD_LVL].level[1] =
mv->level[1];
error = 0;
}
break;
case ARIAMIX_TEL_LVL:
if (mv->num_channels == 1) {
sc->aria_mix[ARIAMIX_TEL_LVL].num_channels =
mv->num_channels;
sc->aria_mix[ARIAMIX_TEL_LVL].level[0] =
mv->level[0];
error = 0;
}
break;
case ARIAMIX_DAC_LVL:
if (mv->num_channels == 1 || mv->num_channels == 2) {
sc->aria_mix[ARIAMIX_DAC_LVL].num_channels =
mv->num_channels;
sc->aria_mix[ARIAMIX_DAC_LVL].level[0] =
mv->level[0];
sc->aria_mix[ARIAMIX_DAC_LVL].level[1] =
mv->level[1];
error = 0;
}
break;
case ARIAMIX_AUX_LVL:
if (mv->num_channels == 1 || mv->num_channels == 2) {
sc->aria_mix[ARIAMIX_AUX_LVL].num_channels =
mv->num_channels;
sc->aria_mix[ARIAMIX_AUX_LVL].level[0] =
mv->level[0];
sc->aria_mix[ARIAMIX_AUX_LVL].level[1] =
mv->level[1];
error = 0;
}
break;
case ARIAMIX_MASTER_LVL:
if (mv->num_channels == 1 || mv->num_channels == 2) {
sc->ariamix_master.num_channels =
mv->num_channels;
sc->ariamix_master.level[0] = mv->level[0];
sc->ariamix_master.level[1] = mv->level[1];
error = 0;
}
break;
case ARIAMIX_MASTER_TREBLE:
if (mv->num_channels == 2) {
sc->ariamix_master.treble[0] =
mv->level[0] == 0 ? 1 : mv->level[0];
sc->ariamix_master.treble[1] =
mv->level[1] == 0 ? 1 : mv->level[1];
error = 0;
}
break;
case ARIAMIX_MASTER_BASS:
if (mv->num_channels == 2) {
sc->ariamix_master.bass[0] =
mv->level[0] == 0 ? 1 : mv->level[0];
sc->ariamix_master.bass[1] =
mv->level[1] == 0 ? 1 : mv->level[1];
error = 0;
}
break;
case ARIAMIX_OUT_LVL:
if (mv->num_channels == 1 || mv->num_channels == 2) {
sc->sc_gain[0] = mv->level[0];
sc->sc_gain[1] = mv->level[1];
error = 0;
}
break;
default:
}
}
if (cp->type == AUDIO_MIXER_ENUM)
switch(cp->dev) {
case ARIAMIX_RECORD_SOURCE:
if (cp->un.ord>=0 && cp->un.ord<=6) {
sc->aria_mix_source = cp->un.ord;
error = 0;
}
break;
case ARIAMIX_MIC_MUTE:
if (cp->un.ord == 0 || cp->un.ord == 1) {
sc->aria_mix[ARIAMIX_MIC_LVL].mute =cp->un.ord;
error = 0;
}
break;
case ARIAMIX_LINE_IN_MUTE:
if (cp->un.ord == 0 || cp->un.ord == 1) {
sc->aria_mix[ARIAMIX_LINE_IN_LVL].mute =
cp->un.ord;
error = 0;
}
break;
case ARIAMIX_CD_MUTE:
if (cp->un.ord == 0 || cp->un.ord == 1) {
sc->aria_mix[ARIAMIX_CD_LVL].mute = cp->un.ord;
error = 0;
}
break;
case ARIAMIX_DAC_MUTE:
if (cp->un.ord == 0 || cp->un.ord == 1) {
sc->aria_mix[ARIAMIX_DAC_LVL].mute =cp->un.ord;
error = 0;
}
break;
case ARIAMIX_AUX_MUTE:
if (cp->un.ord == 0 || cp->un.ord == 1) {
sc->aria_mix[ARIAMIX_AUX_LVL].mute =cp->un.ord;
error = 0;
}
break;
case ARIAMIX_TEL_MUTE:
if (cp->un.ord == 0 || cp->un.ord == 1) {
sc->aria_mix[ARIAMIX_TEL_LVL].mute =cp->un.ord;
error = 0;
}
break;
default:
/* NOTREACHED */
return ENXIO;
}
return(error);
}
int
aria_mixer_get_port(addr, cp)
void *addr;
mixer_ctrl_t *cp;
{
struct aria_softc *sc = addr;
int error = EINVAL;
DPRINTF(("aria_mixer_get_port\n"));
/* This could be done better, no mixer still has some controls. */
if (!(ARIA_MIXER&sc->sc_hardware))
return ENXIO;
switch (cp->dev) {
case ARIAMIX_MIC_LVL:
if (cp->type == AUDIO_MIXER_VALUE) {
cp->un.value.num_channels =
sc->aria_mix[ARIAMIX_MIC_LVL].num_channels;
cp->un.value.level[0] =
sc->aria_mix[ARIAMIX_MIC_LVL].level[0];
cp->un.value.level[1] =
sc->aria_mix[ARIAMIX_MIC_LVL].level[1];
error = 0;
}
break;
case ARIAMIX_LINE_IN_LVL:
if (cp->type == AUDIO_MIXER_VALUE) {
cp->un.value.num_channels =
sc->aria_mix[ARIAMIX_LINE_IN_LVL].num_channels;
cp->un.value.level[0] =
sc->aria_mix[ARIAMIX_LINE_IN_LVL].level[0];
cp->un.value.level[1] =
sc->aria_mix[ARIAMIX_LINE_IN_LVL].level[1];
error = 0;
}
break;
case ARIAMIX_CD_LVL:
if (cp->type == AUDIO_MIXER_VALUE) {
cp->un.value.num_channels =
sc->aria_mix[ARIAMIX_CD_LVL].num_channels;
cp->un.value.level[0] =
sc->aria_mix[ARIAMIX_CD_LVL].level[0];
cp->un.value.level[1] =
sc->aria_mix[ARIAMIX_CD_LVL].level[1];
error = 0;
}
break;
case ARIAMIX_TEL_LVL:
if (cp->type == AUDIO_MIXER_VALUE) {
cp->un.value.num_channels =
sc->aria_mix[ARIAMIX_TEL_LVL].num_channels;
cp->un.value.level[0] =
sc->aria_mix[ARIAMIX_TEL_LVL].level[0];
error = 0;
}
break;
case ARIAMIX_DAC_LVL:
if (cp->type == AUDIO_MIXER_VALUE) {
cp->un.value.num_channels =
sc->aria_mix[ARIAMIX_DAC_LVL].num_channels;
cp->un.value.level[0] =
sc->aria_mix[ARIAMIX_DAC_LVL].level[0];
cp->un.value.level[1] =
sc->aria_mix[ARIAMIX_DAC_LVL].level[1];
error = 0;
}
break;
case ARIAMIX_AUX_LVL:
if (cp->type == AUDIO_MIXER_VALUE) {
cp->un.value.num_channels =
sc->aria_mix[ARIAMIX_AUX_LVL].num_channels;
cp->un.value.level[0] =
sc->aria_mix[ARIAMIX_AUX_LVL].level[0];
cp->un.value.level[1] =
sc->aria_mix[ARIAMIX_AUX_LVL].level[1];
error = 0;
}
break;
case ARIAMIX_MIC_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
cp->un.ord = sc->aria_mix[ARIAMIX_MIC_LVL].mute;
error = 0;
}
break;
case ARIAMIX_LINE_IN_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
cp->un.ord = sc->aria_mix[ARIAMIX_LINE_IN_LVL].mute;
error = 0;
}
break;
case ARIAMIX_CD_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
cp->un.ord = sc->aria_mix[ARIAMIX_CD_LVL].mute;
error = 0;
}
break;
case ARIAMIX_DAC_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
cp->un.ord = sc->aria_mix[ARIAMIX_DAC_LVL].mute;
error = 0;
}
break;
case ARIAMIX_AUX_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
cp->un.ord = sc->aria_mix[ARIAMIX_AUX_LVL].mute;
error = 0;
}
break;
case ARIAMIX_TEL_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
cp->un.ord = sc->aria_mix[ARIAMIX_TEL_LVL].mute;
error = 0;
}
break;
case ARIAMIX_MASTER_LVL:
if (cp->type == AUDIO_MIXER_VALUE) {
cp->un.value.num_channels =
sc->ariamix_master.num_channels;
cp->un.value.level[0] = sc->ariamix_master.level[0];
cp->un.value.level[1] = sc->ariamix_master.level[1];
error = 0;
}
break;
case ARIAMIX_MASTER_TREBLE:
if (cp->type == AUDIO_MIXER_VALUE) {
cp->un.value.num_channels = 2;
cp->un.value.level[0] = sc->ariamix_master.treble[0];
cp->un.value.level[1] = sc->ariamix_master.treble[1];
error = 0;
}
break;
case ARIAMIX_MASTER_BASS:
if (cp->type == AUDIO_MIXER_VALUE) {
cp->un.value.num_channels = 2;
cp->un.value.level[0] = sc->ariamix_master.bass[0];
cp->un.value.level[1] = sc->ariamix_master.bass[1];
error = 0;
}
break;
case ARIAMIX_OUT_LVL:
if (cp->type == AUDIO_MIXER_VALUE) {
cp->un.value.num_channels = sc->sc_chans;
cp->un.value.level[0] = sc->sc_gain[0];
cp->un.value.level[1] = sc->sc_gain[1];
error = 0;
}
break;
case ARIAMIX_RECORD_SOURCE:
if (cp->type == AUDIO_MIXER_ENUM) {
cp->un.ord = sc->aria_mix_source;
error = 0;
}
break;
default:
return ENXIO;
/* NOT REACHED */
}
return(error);
}
int
aria_mixer_query_devinfo(addr, dip)
void *addr;
mixer_devinfo_t *dip;
{
struct aria_softc *sc = addr;
DPRINTF(("aria_mixer_query_devinfo\n"));
/* This could be done better, no mixer still has some controls. */
if (!(ARIA_MIXER & sc->sc_hardware))
return ENXIO;
dip->prev = dip->next = AUDIO_MIXER_LAST;
switch(dip->index) {
case ARIAMIX_MIC_LVL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = ARIAMIX_INPUT_CLASS;
dip->next = ARIAMIX_MIC_MUTE;
strcpy(dip->label.name, AudioNmicrophone);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case ARIAMIX_LINE_IN_LVL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = ARIAMIX_INPUT_CLASS;
dip->next = ARIAMIX_LINE_IN_MUTE;
strcpy(dip->label.name, AudioNline);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case ARIAMIX_CD_LVL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = ARIAMIX_INPUT_CLASS;
dip->next = ARIAMIX_CD_MUTE;
strcpy(dip->label.name, AudioNcd);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case ARIAMIX_TEL_LVL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = ARIAMIX_INPUT_CLASS;
dip->next = ARIAMIX_TEL_MUTE;
strcpy(dip->label.name, "telephone");
dip->un.v.num_channels = 1;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case ARIAMIX_DAC_LVL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = ARIAMIX_INPUT_CLASS;
dip->next = ARIAMIX_DAC_MUTE;
strcpy(dip->label.name, AudioNdac);
dip->un.v.num_channels = 1;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case ARIAMIX_AUX_LVL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = ARIAMIX_INPUT_CLASS;
dip->next = ARIAMIX_AUX_MUTE;
strcpy(dip->label.name, AudioNoutput);
dip->un.v.num_channels = 1;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case ARIAMIX_MIC_MUTE:
dip->prev = ARIAMIX_MIC_LVL;
goto mute;
case ARIAMIX_LINE_IN_MUTE:
dip->prev = ARIAMIX_LINE_IN_LVL;
goto mute;
case ARIAMIX_CD_MUTE:
dip->prev = ARIAMIX_CD_LVL;
goto mute;
case ARIAMIX_DAC_MUTE:
dip->prev = ARIAMIX_DAC_LVL;
goto mute;
case ARIAMIX_AUX_MUTE:
dip->prev = ARIAMIX_AUX_LVL;
goto mute;
case ARIAMIX_TEL_MUTE:
dip->prev = ARIAMIX_TEL_LVL;
goto mute;
mute:
dip->mixer_class = ARIAMIX_INPUT_CLASS;
dip->type = AUDIO_MIXER_ENUM;
strcpy(dip->label.name, AudioNmute);
dip->un.e.num_mem = 2;
strcpy(dip->un.e.member[0].label.name, AudioNoff);
dip->un.e.member[0].ord = 0;
strcpy(dip->un.e.member[1].label.name, AudioNon);
dip->un.e.member[1].ord = 1;
break;
case ARIAMIX_MASTER_LVL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = ARIAMIX_OUTPUT_CLASS;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNvolume);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case ARIAMIX_MASTER_TREBLE:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = ARIAMIX_EQ_CLASS;
strcpy(dip->label.name, AudioNtreble);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNtreble);
break;
case ARIAMIX_MASTER_BASS:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = ARIAMIX_EQ_CLASS;
strcpy(dip->label.name, AudioNbass);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNbass);
break;
case ARIAMIX_OUT_LVL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = ARIAMIX_OUTPUT_CLASS;
strcpy(dip->label.name, AudioNoutput);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case ARIAMIX_RECORD_SOURCE:
dip->mixer_class = ARIAMIX_RECORD_CLASS;
dip->type = AUDIO_MIXER_ENUM;
strcpy(dip->label.name, AudioNsource);
dip->un.e.num_mem = 6;
strcpy(dip->un.e.member[0].label.name, AudioNoutput);
dip->un.e.member[0].ord = ARIAMIX_AUX_LVL;
strcpy(dip->un.e.member[1].label.name, AudioNmicrophone);
dip->un.e.member[1].ord = ARIAMIX_MIC_LVL;
strcpy(dip->un.e.member[2].label.name, AudioNdac);
dip->un.e.member[2].ord = ARIAMIX_DAC_LVL;
strcpy(dip->un.e.member[3].label.name, AudioNline);
dip->un.e.member[3].ord = ARIAMIX_LINE_IN_LVL;
strcpy(dip->un.e.member[4].label.name, AudioNcd);
dip->un.e.member[4].ord = ARIAMIX_CD_LVL;
strcpy(dip->un.e.member[5].label.name, "telephone");
dip->un.e.member[5].ord = ARIAMIX_TEL_LVL;
break;
case ARIAMIX_INPUT_CLASS:
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = ARIAMIX_INPUT_CLASS;
strcpy(dip->label.name, AudioCinputs);
break;
case ARIAMIX_OUTPUT_CLASS:
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = ARIAMIX_OUTPUT_CLASS;
strcpy(dip->label.name, AudioCoutputs);
break;
case ARIAMIX_RECORD_CLASS:
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = ARIAMIX_RECORD_CLASS;
strcpy(dip->label.name, AudioCrecord);
break;
case ARIAMIX_EQ_CLASS:
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = ARIAMIX_EQ_CLASS;
strcpy(dip->label.name, AudioCequalization);
break;
default:
return ENXIO;
/*NOTREACHED*/
}
return 0;
}
#endif /* NARIA */
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