* Copyright (c) 1999, 2008 The NetBSD Foundation, Inc.

/* $NetBSD: ad1848.c,v 1.34 2024/02/09 22:08:34 andvar Exp $ */

/*-
* Copyright (c) 1999, 2008 The NetBSD Foundation, Inc.
* All rights reserved.
*
* This code is derived from software contributed to The NetBSD Foundation
* by Ken Hornstein and John Kohl.
*
* 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.
*
* THIS SOFTWARE IS PROVIDED BY THE NETBSD FOUNDATION, INC. 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 FOUNDATION 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.
*/
/*
* Copyright (c) 1994 John Brezak
* 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.
*
*/

/*
* Copyright by Hannu Savolainen 1994
*
* 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.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR 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 AUTHOR 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.
*
*/
/*
* Portions of this code are from the VOXware support for the ad1848
* by Hannu Savolainen <[email protected]>
*
* Portions also ripped from the SoundBlaster driver for NetBSD.
*/

#include <sys/cdefs.h>
__KERNEL_RCSID(0, "$NetBSD: ad1848.c,v 1.34 2024/02/09 22:08:34 andvar Exp $");

#include <sys/param.h>
#include <sys/systm.h>
#include <sys/errno.h>
#include <sys/ioctl.h>
#include <sys/device.h>
#include <sys/fcntl.h>
/*#include <sys/syslog.h>*/
/*#include <sys/proc.h>*/

#include <sys/cpu.h>
#include <sys/bus.h>

#include <sys/audioio.h>
#include <dev/audio/audio_if.h>

#include <dev/ic/ad1848reg.h>
#include <dev/ic/cs4231reg.h>
#include <dev/ic/cs4237reg.h>
#include <dev/ic/ad1848var.h>
#if 0
#include <dev/isa/cs4231var.h>
#endif

/*
* AD1845 on some machines don't match the AD1845 doc
* and defining AD1845_HACK to 1 works around the problems.
* options AD1845_HACK=0 should work if you have ``correct'' one.
*/
#ifndef AD1845_HACK
#define AD1845_HACK 1 /* weird mixer, can't play slinear_be */
#endif

#ifdef AUDIO_DEBUG
#define DPRINTF(x) if (ad1848debug) printf x
int ad1848debug = 0;
void ad1848_dump_regs(struct ad1848_softc *);
#else
#define DPRINTF(x)
#endif

/* The HW supports more frequencies but I chose several major ones. */
static const struct audio_format ad1848_formats[] = {
{
.mode = AUMODE_PLAY | AUMODE_RECORD,
.encoding = AUDIO_ENCODING_SLINEAR_LE,
.validbits = 16,
.precision = 16,
.channels = 2,
.channel_mask = AUFMT_STEREO,
.frequency_type = 6,
.frequency = { 8000, 11025, 16000, 22050, 44100, 48000 },
},
};
#define AD1848_NFORMATS __arraycount(ad1848_formats)

/*
* Initial values for the indirect registers of CS4248/AD1848.
*/
static const int ad1848_init_values[] = {
GAIN_12|INPUT_MIC_GAIN_ENABLE, /* Left Input Control */
GAIN_12|INPUT_MIC_GAIN_ENABLE, /* Right Input Control */
ATTEN_12, /* Left Aux #1 Input Control */
ATTEN_12, /* Right Aux #1 Input Control */
ATTEN_12, /* Left Aux #2 Input Control */
ATTEN_12, /* Right Aux #2 Input Control */
/* bits 5-0 are attenuation select */
ATTEN_12, /* Left DAC output Control */
ATTEN_12, /* Right DAC output Control */
CLOCK_XTAL1|FMT_PCM8, /* Clock and Data Format */
SINGLE_DMA|AUTO_CAL_ENABLE, /* Interface Config */
INTERRUPT_ENABLE, /* Pin control */
0x00, /* Test and Init */
MODE2, /* Misc control */
ATTEN_0<<2, /* Digital Mix Control */
0, /* Upper base Count */
0, /* Lower base Count */

/* These are for CS4231 &c. only (additional registers): */
0, /* Alt feature 1 */
0, /* Alt feature 2 */
ATTEN_12, /* Left line in */
ATTEN_12, /* Right line in */
0, /* Timer low */
0, /* Timer high */
0, /* unused */
0, /* unused */
0, /* IRQ status */
0, /* unused */
/* Mono input (a.k.a speaker) (mic) Control */
MONO_INPUT_MUTE|ATTEN_6, /* mute speaker by default */
0, /* unused */
0, /* record format */
0, /* Crystal Clock Select */
0, /* upper record count */
0 /* lower record count */
};

int
ad1848_to_vol(mixer_ctrl_t *cp, struct ad1848_volume *vol)
{

if (cp->un.value.num_channels == 1) {
vol->left =
vol->right = cp->un.value.level[AUDIO_MIXER_LEVEL_MONO];
return 1;
}
else if (cp->un.value.num_channels == 2) {
vol->left = cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT];
vol->right = cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT];
return 1;
}
return 0;
}

int
ad1848_from_vol(mixer_ctrl_t *cp, struct ad1848_volume *vol)
{

if (cp->un.value.num_channels == 1) {
cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = vol->left;
return 1;
}
else if (cp->un.value.num_channels == 2) {
cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = vol->left;
cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = vol->right;
return 1;
}
return 0;
}

int
ad_read(struct ad1848_softc *sc, int reg)
{
int x;

ADWRITE(sc, AD1848_IADDR, (reg & 0xff) | sc->MCE_bit);
x = ADREAD(sc, AD1848_IDATA);
/* printf("(%02x<-%02x) ", reg|sc->MCE_bit, x); */
return x;
}

void
ad_write(struct ad1848_softc *sc, int reg, int data)
{

ADWRITE(sc, AD1848_IADDR, (reg & 0xff) | sc->MCE_bit);
ADWRITE(sc, AD1848_IDATA, data & 0xff);
/* printf("(%02x->%02x) ", reg|sc->MCE_bit, data); */
}

/*
* extended registers (mode 3) require an additional level of
* indirection through CS_XREG (I23).
*/

int
ad_xread(struct ad1848_softc *sc, int reg)
{
int x;

ADWRITE(sc, AD1848_IADDR, CS_XREG | sc->MCE_bit);
ADWRITE(sc, AD1848_IDATA, (reg | ALT_F3_XRAE) & 0xff);
x = ADREAD(sc, AD1848_IDATA);

return x;
}

void
ad_xwrite(struct ad1848_softc *sc, int reg, int val)
{

ADWRITE(sc, AD1848_IADDR, CS_XREG | sc->MCE_bit);
ADWRITE(sc, AD1848_IDATA, (reg | ALT_F3_XRAE) & 0xff);
ADWRITE(sc, AD1848_IDATA, val & 0xff);
}

static void
ad_set_MCE(struct ad1848_softc *sc, int state)
{

if (state)
sc->MCE_bit = MODE_CHANGE_ENABLE;
else
sc->MCE_bit = 0;
ADWRITE(sc, AD1848_IADDR, sc->MCE_bit);
}

static void
wait_for_calibration(struct ad1848_softc *sc)
{
int timeout;

DPRINTF(("ad1848: Auto calibration started.\n"));
/*
* Wait until the auto calibration process has finished.
*
* 1) Wait until the chip becomes ready (reads don't return 0x80).
* 2) Wait until the ACI bit of I11 gets on and then off.
* Because newer chips are fast we may never see the ACI
* bit go on. Just delay a little instead.
*/
timeout = 10000;
while (timeout > 0 && ADREAD(sc, AD1848_IADDR) == SP_IN_INIT) {
delay(10);
timeout--;
}
if (timeout <= 0) {
DPRINTF(("ad1848: Auto calibration timed out(1).\n"));
}

/* Set register addr */
ADWRITE(sc, AD1848_IADDR, SP_TEST_AND_INIT);
/* Wait for address to appear when read back. */
timeout = 100000;
while (timeout > 0 &&
(ADREAD(sc, AD1848_IADDR)&SP_IADDR_MASK) != SP_TEST_AND_INIT) {
delay(10);
timeout--;
}
if (timeout <= 0) {
DPRINTF(("ad1848: Auto calibration timed out(1.5).\n"));
}

if (!(ad_read(sc, SP_TEST_AND_INIT) & AUTO_CAL_IN_PROG)) {
if (sc->mode > 1) {
/* A new chip, just delay a little. */
delay(100); /* XXX what should it be? */
} else {
timeout = 10000;
while (timeout > 0 &&
!(ad_read(sc, SP_TEST_AND_INIT) &
AUTO_CAL_IN_PROG)) {
delay(10);
timeout--;
}
if (timeout <= 0) {
DPRINTF(("ad1848: Auto calibration timed out(2).\n"));
}
}
}

timeout = 10000;
while (timeout > 0 &&
ad_read(sc, SP_TEST_AND_INIT) & AUTO_CAL_IN_PROG) {
delay(10);
timeout--;
}
if (timeout <= 0) {
DPRINTF(("ad1848: Auto calibration timed out(3).\n"));
}
}

#ifdef AUDIO_DEBUG
void
ad1848_dump_regs(struct ad1848_softc *sc)
{
int i;
u_char r;

printf("ad1848 status=%02x", ADREAD(sc, AD1848_STATUS));
printf(" regs: ");
for (i = 0; i < 16; i++) {
r = ad_read(sc, i);
printf("%02x ", r);
}
if (sc->mode >= 2) {
for (i = 16; i < 32; i++) {
r = ad_read(sc, i);
printf("%02x ", r);
}
}
printf("\n");
}
#endif /* AUDIO_DEBUG */

/*
* Attach hardware to driver, attach hardware driver to audio
* pseudo-device driver .
*/
void
ad1848_attach(struct ad1848_softc *sc)
{
static struct ad1848_volume vol_mid = {220, 220};
static struct ad1848_volume vol_0 = {0, 0};
int i;
int timeout;

/* Initialize the ad1848... */
for (i = 0; i < 0x10; i++) {
ad_write(sc, i, ad1848_init_values[i]);
timeout = 100000;
while (timeout > 0 && ADREAD(sc, AD1848_IADDR) & SP_IN_INIT)
timeout--;
}
/* ...and additional CS4231 stuff too */
if (sc->mode >= 2) {
ad_write(sc, SP_INTERFACE_CONFIG, 0); /* disable SINGLE_DMA */
for (i = 0x10; i < 0x20; i++)
if (ad1848_init_values[i] != 0) {
ad_write(sc, i, ad1848_init_values[i]);
timeout = 100000;
while (timeout > 0 &&
ADREAD(sc, AD1848_IADDR) & SP_IN_INIT)
timeout--;
}
}
ad1848_reset(sc);

/* Set default gains */
ad1848_set_rec_gain(sc, &vol_mid);
ad1848_set_channel_gain(sc, AD1848_DAC_CHANNEL, &vol_mid);
ad1848_set_channel_gain(sc, AD1848_MONITOR_CHANNEL, &vol_0);
ad1848_set_channel_gain(sc, AD1848_AUX1_CHANNEL, &vol_mid); /* CD volume */
sc->mute[AD1848_MONITOR_CHANNEL] = MUTE_ALL;
if (sc->mode >= 2
#if AD1845_HACK
&& sc->is_ad1845 == 0
#endif
) {
ad1848_set_channel_gain(sc, AD1848_AUX2_CHANNEL, &vol_mid); /* CD volume */
ad1848_set_channel_gain(sc, AD1848_LINE_CHANNEL, &vol_mid);
ad1848_set_channel_gain(sc, AD1848_MONO_CHANNEL, &vol_0);
sc->mute[AD1848_MONO_CHANNEL] = MUTE_ALL;
} else
ad1848_set_channel_gain(sc, AD1848_AUX2_CHANNEL, &vol_0);

/* Set default port */
ad1848_set_rec_port(sc, MIC_IN_PORT);

printf(": %s", sc->chip_name);
}

/*
* Various routines to interface to higher level audio driver
*/
static const struct ad1848_mixerinfo {
int left_reg;
int right_reg;
int atten_bits;
int atten_mask;
} mixer_channel_info[] =
{
{ SP_LEFT_AUX2_CONTROL, SP_RIGHT_AUX2_CONTROL, AUX_INPUT_ATTEN_BITS,
AUX_INPUT_ATTEN_MASK },
{ SP_LEFT_AUX1_CONTROL, SP_RIGHT_AUX1_CONTROL, AUX_INPUT_ATTEN_BITS,
AUX_INPUT_ATTEN_MASK },
{ SP_LEFT_OUTPUT_CONTROL, SP_RIGHT_OUTPUT_CONTROL,
OUTPUT_ATTEN_BITS, OUTPUT_ATTEN_MASK },
{ CS_LEFT_LINE_CONTROL, CS_RIGHT_LINE_CONTROL, LINE_INPUT_ATTEN_BITS,
LINE_INPUT_ATTEN_MASK },
{ CS_MONO_IO_CONTROL, 0, MONO_INPUT_ATTEN_BITS, MONO_INPUT_ATTEN_MASK },
{ CS_MONO_IO_CONTROL, 0, 0, 0 },
{ SP_DIGITAL_MIX, 0, OUTPUT_ATTEN_BITS, MIX_ATTEN_MASK }
};

/*
* This function doesn't set the mute flags but does use them.
* The mute flags reflect the mutes that have been applied by the user.
* However, the driver occasionally wants to mute devices (e.g. when changing
* sampling rate). These operations should not affect the mute flags.
*/

void
ad1848_mute_channel(struct ad1848_softc *sc, int device, int mute)
{
u_char reg;

reg = ad_read(sc, mixer_channel_info[device].left_reg);

if (mute & MUTE_LEFT) {
if (device == AD1848_MONITOR_CHANNEL) {
if (sc->open_mode & FREAD)
ad1848_mute_wave_output(sc, WAVE_UNMUTE1, 0);
ad_write(sc, mixer_channel_info[device].left_reg,
reg & ~DIGITAL_MIX1_ENABLE);
} else if (device == AD1848_OUT_CHANNEL)
ad_write(sc, mixer_channel_info[device].left_reg,
reg | MONO_OUTPUT_MUTE);
else
ad_write(sc, mixer_channel_info[device].left_reg,
reg | 0x80);
} else if (!(sc->mute[device] & MUTE_LEFT)) {
if (device == AD1848_MONITOR_CHANNEL) {
ad_write(sc, mixer_channel_info[device].left_reg,
reg | DIGITAL_MIX1_ENABLE);
if (sc->open_mode & FREAD)
ad1848_mute_wave_output(sc, WAVE_UNMUTE1, 1);
} else if (device == AD1848_OUT_CHANNEL)
ad_write(sc, mixer_channel_info[device].left_reg,
reg & ~MONO_OUTPUT_MUTE);
else
ad_write(sc, mixer_channel_info[device].left_reg,
reg & ~0x80);
}

if (!mixer_channel_info[device].right_reg)
return;

reg = ad_read(sc, mixer_channel_info[device].right_reg);

if (mute & MUTE_RIGHT) {
ad_write(sc, mixer_channel_info[device].right_reg, reg | 0x80);
} else if (!(sc->mute[device] & MUTE_RIGHT)) {
ad_write(sc, mixer_channel_info[device].right_reg, reg &~0x80);
}
}

int
ad1848_set_channel_gain(struct ad1848_softc *sc, int device,
struct ad1848_volume *gp)
{
const struct ad1848_mixerinfo *info;
u_char reg;
u_int atten;

info = &mixer_channel_info[device];
sc->gains[device] = *gp;

atten = (AUDIO_MAX_GAIN - gp->left) * (info->atten_bits + 1) /
(AUDIO_MAX_GAIN + 1);

reg = ad_read(sc, info->left_reg) & (info->atten_mask);
if (device == AD1848_MONITOR_CHANNEL)
reg |= ((atten & info->atten_bits) << 2);
else
reg |= ((atten & info->atten_bits));

ad_write(sc, info->left_reg, reg);

if (!info->right_reg)
return 0;

atten = (AUDIO_MAX_GAIN - gp->right) * (info->atten_bits + 1) /
(AUDIO_MAX_GAIN + 1);
reg = ad_read(sc, info->right_reg);
reg &= info->atten_mask;
ad_write(sc, info->right_reg, (atten & info->atten_bits) | reg);

return 0;
}

int
ad1848_get_device_gain(struct ad1848_softc *sc, int device,
struct ad1848_volume *gp)
{

*gp = sc->gains[device];
return 0;
}

int
ad1848_get_rec_gain(struct ad1848_softc *sc, struct ad1848_volume *gp)
{

*gp = sc->rec_gain;
return 0;
}

int
ad1848_set_rec_gain(struct ad1848_softc *sc, struct ad1848_volume *gp)
{
u_char reg, gain;

DPRINTF(("ad1848_set_rec_gain: %d:%d\n", gp->left, gp->right));

sc->rec_gain = *gp;

gain = (gp->left * (GAIN_22_5 + 1)) / (AUDIO_MAX_GAIN + 1);
reg = ad_read(sc, SP_LEFT_INPUT_CONTROL);
reg &= INPUT_GAIN_MASK;
ad_write(sc, SP_LEFT_INPUT_CONTROL, (gain & 0x0f) | reg);

gain = (gp->right * (GAIN_22_5 + 1)) / (AUDIO_MAX_GAIN + 1);
reg = ad_read(sc, SP_RIGHT_INPUT_CONTROL);
reg &= INPUT_GAIN_MASK;
ad_write(sc, SP_RIGHT_INPUT_CONTROL, (gain & 0x0f) | reg);

return 0;
}

void
ad1848_mute_wave_output(struct ad1848_softc *sc, int mute, int set)
{
int m;

DPRINTF(("ad1848_mute_wave_output: %d, %d\n", mute, set));

if (mute == WAVE_MUTE2_INIT) {
sc->wave_mute_status = 0;
mute = WAVE_MUTE2;
}
if (set)
m = sc->wave_mute_status |= mute;
else
m = sc->wave_mute_status &= ~mute;

if (m & WAVE_MUTE0 || ((m & WAVE_UNMUTE1) == 0 && m & WAVE_MUTE2))
ad1848_mute_channel(sc, AD1848_DAC_CHANNEL, MUTE_ALL);
else
ad1848_mute_channel(sc, AD1848_DAC_CHANNEL,
sc->mute[AD1848_DAC_CHANNEL]);
}

int
ad1848_set_mic_gain(struct ad1848_softc *sc, struct ad1848_volume *gp)
{
u_char reg;

DPRINTF(("cs4231_set_mic_gain: %d\n", gp->left));

if (gp->left > AUDIO_MAX_GAIN/2) {
sc->mic_gain_on = 1;
reg = ad_read(sc, SP_LEFT_INPUT_CONTROL);
ad_write(sc, SP_LEFT_INPUT_CONTROL,
reg | INPUT_MIC_GAIN_ENABLE);
} else {
sc->mic_gain_on = 0;
reg = ad_read(sc, SP_LEFT_INPUT_CONTROL);
ad_write(sc, SP_LEFT_INPUT_CONTROL,
reg & ~INPUT_MIC_GAIN_ENABLE);
}

return 0;
}

int
ad1848_get_mic_gain(struct ad1848_softc *sc, struct ad1848_volume *gp)
{
if (sc->mic_gain_on)
gp->left = gp->right = AUDIO_MAX_GAIN;
else
gp->left = gp->right = AUDIO_MIN_GAIN;
return 0;
}

static const ad1848_devmap_t *
ad1848_mixer_find_dev(const ad1848_devmap_t *map, int cnt, mixer_ctrl_t *cp)
{
int i;

for (i = 0; i < cnt; i++) {
if (map[i].id == cp->dev) {
return (&map[i]);
}
}
return 0;
}

int
ad1848_mixer_get_port(struct ad1848_softc *ac, const struct ad1848_devmap *map,
int cnt, mixer_ctrl_t *cp)
{
const ad1848_devmap_t *entry;
struct ad1848_volume vol;
int error;
int dev;

error = EINVAL;
if (!(entry = ad1848_mixer_find_dev(map, cnt, cp)))
return ENXIO;

dev = entry->dev;

switch (entry->kind) {
case AD1848_KIND_LVL:
if (cp->type != AUDIO_MIXER_VALUE)
break;

if (dev < AD1848_AUX2_CHANNEL ||
dev > AD1848_MONITOR_CHANNEL)
break;

if (cp->un.value.num_channels != 1 &&
mixer_channel_info[dev].right_reg == 0)
break;

error = ad1848_get_device_gain(ac, dev, &vol);
if (!error)
ad1848_from_vol(cp, &vol);

break;

case AD1848_KIND_MUTE:
if (cp->type != AUDIO_MIXER_ENUM) break;

cp->un.ord = ac->mute[dev] ? 1 : 0;
error = 0;
break;

case AD1848_KIND_RECORDGAIN:
if (cp->type != AUDIO_MIXER_VALUE) break;

error = ad1848_get_rec_gain(ac, &vol);
if (!error)
ad1848_from_vol(cp, &vol);

break;

case AD1848_KIND_MICGAIN:
if (cp->type != AUDIO_MIXER_VALUE) break;

error = ad1848_get_mic_gain(ac, &vol);
if (!error)
ad1848_from_vol(cp, &vol);

break;

case AD1848_KIND_RECORDSOURCE:
if (cp->type != AUDIO_MIXER_ENUM) break;
cp->un.ord = ad1848_get_rec_port(ac);
error = 0;
break;

default:
printf ("Invalid kind\n");
break;
}

return error;
}

int
ad1848_mixer_set_port(struct ad1848_softc *ac, const struct ad1848_devmap *map,
int cnt, mixer_ctrl_t *cp)
{
const ad1848_devmap_t *entry;
struct ad1848_volume vol;
int error;
int dev;

error = EINVAL;
if (!(entry = ad1848_mixer_find_dev(map, cnt, cp)))
return ENXIO;

dev = entry->dev;

switch (entry->kind) {
case AD1848_KIND_LVL:
if (cp->type != AUDIO_MIXER_VALUE)
break;

if (dev < AD1848_AUX2_CHANNEL ||
dev > AD1848_MONITOR_CHANNEL)
break;

if (cp->un.value.num_channels != 1 &&
mixer_channel_info[dev].right_reg == 0)
break;

ad1848_to_vol(cp, &vol);
error = ad1848_set_channel_gain(ac, dev, &vol);
break;

case AD1848_KIND_MUTE:
if (cp->type != AUDIO_MIXER_ENUM) break;

ac->mute[dev] = (cp->un.ord ? MUTE_ALL : 0);
ad1848_mute_channel(ac, dev, ac->mute[dev]);
error = 0;
break;

case AD1848_KIND_RECORDGAIN:
if (cp->type != AUDIO_MIXER_VALUE) break;

ad1848_to_vol(cp, &vol);
error = ad1848_set_rec_gain(ac, &vol);
break;

case AD1848_KIND_MICGAIN:
if (cp->type != AUDIO_MIXER_VALUE) break;

ad1848_to_vol(cp, &vol);
error = ad1848_set_mic_gain(ac, &vol);
break;

case AD1848_KIND_RECORDSOURCE:
if (cp->type != AUDIO_MIXER_ENUM) break;

error = ad1848_set_rec_port(ac, cp->un.ord);
break;

default:
printf ("Invalid kind\n");
break;
}

return error;
}

int
ad1848_query_format(void *addr, audio_format_query_t *afp)
{

return audio_query_format(ad1848_formats, AD1848_NFORMATS, afp);
}

int
ad1848_set_format(void *addr, int setmode,
const audio_params_t *p, const audio_params_t *r,
audio_filter_reg_t *pfil, audio_filter_reg_t *rfil)
{
struct ad1848_softc *sc;
int rate;
int error;

/* *p and *r are the identical because !AUDIO_PROP_INDEPENDENT. */
DPRINTF(("%s: %u %u %u %u\n", __func__,
p->encoding, p->precision, p->channels, p->sample_rate));

sc = addr;

rate = p->sample_rate;
error = ad1848_set_speed(sc, &rate);
if (error)
return error;

sc->format_bits = FMT_TWOS_COMP >> 5;
sc->channels = p->channels;
sc->precision = p->precision;
sc->need_commit = 1;

DPRINTF(("%s succeeded\n", __func__));
return 0;
}

int
ad1848_set_rec_port(struct ad1848_softc *sc, int port)
{
u_char inp, reg;

DPRINTF(("ad1848_set_rec_port: 0x%x\n", port));

if (port == MIC_IN_PORT)
inp = MIC_INPUT;
else if (port == LINE_IN_PORT)
inp = LINE_INPUT;
else if (port == DAC_IN_PORT)
inp = MIXED_DAC_INPUT;
else if (sc->mode >= 2 && port == AUX1_IN_PORT)
inp = AUX_INPUT;
else
return EINVAL;

reg = ad_read(sc, SP_LEFT_INPUT_CONTROL);
reg &= INPUT_SOURCE_MASK;
ad_write(sc, SP_LEFT_INPUT_CONTROL, (inp|reg));

reg = ad_read(sc, SP_RIGHT_INPUT_CONTROL);
reg &= INPUT_SOURCE_MASK;
ad_write(sc, SP_RIGHT_INPUT_CONTROL, (inp|reg));

sc->rec_port = port;

return 0;
}

int
ad1848_get_rec_port(struct ad1848_softc *sc)
{
return sc->rec_port;
}

int
ad1848_open(void *addr, int flags)
{
struct ad1848_softc *sc;
u_char reg;

sc = addr;
DPRINTF(("ad1848_open: sc=%p\n", sc));

sc->open_mode = flags;

/* Enable interrupts */
DPRINTF(("ad1848_open: enable intrs\n"));
reg = ad_read(sc, SP_PIN_CONTROL);
ad_write(sc, SP_PIN_CONTROL, reg | INTERRUPT_ENABLE);

/* If recording && monitoring, the playback part is also used. */
if (flags & FREAD && sc->mute[AD1848_MONITOR_CHANNEL] == 0)
ad1848_mute_wave_output(sc, WAVE_UNMUTE1, 1);

#ifdef AUDIO_DEBUG
if (ad1848debug)
ad1848_dump_regs(sc);
#endif

return 0;
}

void
ad1848_close(void *addr)
{
struct ad1848_softc *sc;
u_char reg;

sc = addr;
sc->open_mode = 0;

ad1848_mute_wave_output(sc, WAVE_UNMUTE1, 0);

/* Disable interrupts */
DPRINTF(("ad1848_close: disable intrs\n"));
reg = ad_read(sc, SP_PIN_CONTROL);
ad_write(sc, SP_PIN_CONTROL, reg & ~INTERRUPT_ENABLE);

#ifdef AUDIO_DEBUG
if (ad1848debug)
ad1848_dump_regs(sc);
#endif
}

/*
* Lower-level routines
*/
int
ad1848_commit_settings(void *addr)
{
struct ad1848_softc *sc;
int timeout;
u_char fs;

sc = addr;
if (!sc->need_commit)
return 0;

mutex_spin_enter(&sc->sc_intr_lock);

ad1848_mute_wave_output(sc, WAVE_MUTE0, 1);

ad_set_MCE(sc, 1); /* Enables changes to the format select reg */

fs = sc->speed_bits | (sc->format_bits << 5);

if (sc->channels == 2)
fs |= FMT_STEREO;

/*
* OPL3-SA2 (YMF711) is sometimes busy here.
* Wait until it becomes ready.
*/
for (timeout = 0;
timeout < 1000 && ADREAD(sc, AD1848_IADDR) & SP_IN_INIT; timeout++)
delay(10);

ad_write(sc, SP_CLOCK_DATA_FORMAT, fs);

/*
* If mode >= 2 (CS4231), set I28 also.
* It's the capture format register.
*/
if (sc->mode >= 2) {
/*
* Gravis Ultrasound MAX SDK sources says something about
* errata sheets, with the implication that these inb()s
* are necessary.
*/
(void)ADREAD(sc, AD1848_IDATA);
(void)ADREAD(sc, AD1848_IDATA);
/* Write to I8 starts resynchronization. Wait for completion. */
timeout = 100000;
while (timeout > 0 && ADREAD(sc, AD1848_IADDR) == SP_IN_INIT)
timeout--;

ad_write(sc, CS_REC_FORMAT, fs);
(void)ADREAD(sc, AD1848_IDATA);
(void)ADREAD(sc, AD1848_IDATA);
/* Now wait for resync for capture side of the house */
}
/*
* Write to I8 starts resynchronization. Wait until it completes.
*/
timeout = 100000;
while (timeout > 0 && ADREAD(sc, AD1848_IADDR) == SP_IN_INIT) {
delay(10);
timeout--;
}

if (ADREAD(sc, AD1848_IADDR) == SP_IN_INIT)
printf("ad1848_commit: Auto calibration timed out\n");

/*
* Starts the calibration process and
* enters playback mode after it.
*/
ad_set_MCE(sc, 0);
wait_for_calibration(sc);

ad1848_mute_wave_output(sc, WAVE_MUTE0, 0);

mutex_spin_exit(&sc->sc_intr_lock);

sc->need_commit = 0;

return 0;
}

void
ad1848_reset(struct ad1848_softc *sc)
{
u_char r;

DPRINTF(("ad1848_reset\n"));

/* Clear the PEN and CEN bits */
r = ad_read(sc, SP_INTERFACE_CONFIG);
r &= ~(CAPTURE_ENABLE | PLAYBACK_ENABLE);
ad_write(sc, SP_INTERFACE_CONFIG, r);

if (sc->mode >= 2) {
ADWRITE(sc, AD1848_IADDR, CS_IRQ_STATUS);
ADWRITE(sc, AD1848_IDATA, 0);
}
/* Clear interrupt status */
ADWRITE(sc, AD1848_STATUS, 0);
#ifdef AUDIO_DEBUG
if (ad1848debug)
ad1848_dump_regs(sc);
#endif
}

int
ad1848_set_speed(struct ad1848_softc *sc, u_int *argp)
{
/*
* The sampling speed is encoded in the least significant nible of I8.
* The LSB selects the clock source (0=24.576 MHz, 1=16.9344 MHz) and
* other three bits select the divisor (indirectly):
*
* The available speeds are in the following table. Keep the speeds in
* the increasing order.
*/
typedef struct {
int speed;
u_char bits;
} speed_struct;
u_long arg;

static const speed_struct speed_table[] = {
{5510, (0 << 1) | 1},
{5510, (0 << 1) | 1},
{6620, (7 << 1) | 1},
{8000, (0 << 1) | 0},
{9600, (7 << 1) | 0},
{11025, (1 << 1) | 1},
{16000, (1 << 1) | 0},
{18900, (2 << 1) | 1},
{22050, (3 << 1) | 1},
{27420, (2 << 1) | 0},
{32000, (3 << 1) | 0},
{33075, (6 << 1) | 1},
{37800, (4 << 1) | 1},
{44100, (5 << 1) | 1},
{48000, (6 << 1) | 0}
};

int i, n, selected;

arg = *argp;
selected = -1;
n = sizeof(speed_table) / sizeof(speed_struct);

if (arg < speed_table[0].speed)
selected = 0;
if (arg > speed_table[n - 1].speed)
selected = n - 1;

for (i = 1 /*really*/ ; selected == -1 && i < n; i++)
if (speed_table[i].speed == arg)
selected = i;
else if (speed_table[i].speed > arg) {
int diff1, diff2;

diff1 = arg - speed_table[i - 1].speed;
diff2 = speed_table[i].speed - arg;

if (diff1 < diff2)
selected = i - 1;
else
selected = i;
}

if (selected == -1) {
printf("ad1848: Can't find speed???\n");
selected = 3;
}

sc->speed_bits = speed_table[selected].bits;
sc->need_commit = 1;
*argp = speed_table[selected].speed;

return 0;
}

/*
* Halt I/O
*/
int
ad1848_halt_output(void *addr)
{
struct ad1848_softc *sc;
u_char reg;

DPRINTF(("ad1848: ad1848_halt_output\n"));
sc = addr;
reg = ad_read(sc, SP_INTERFACE_CONFIG);
ad_write(sc, SP_INTERFACE_CONFIG, reg & ~PLAYBACK_ENABLE);

return 0;
}

int
ad1848_halt_input(void *addr)
{
struct ad1848_softc *sc;
u_char reg;

DPRINTF(("ad1848: ad1848_halt_input\n"));
sc = addr;
reg = ad_read(sc, SP_INTERFACE_CONFIG);
ad_write(sc, SP_INTERFACE_CONFIG, reg & ~CAPTURE_ENABLE);

return 0;
}

void
ad1848_get_locks(void *addr, kmutex_t **intr, kmutex_t **thread)
{
struct ad1848_softc *sc;

sc = addr;
*intr = &sc->sc_intr_lock;
*thread = &sc->sc_lock;
}

void
ad1848_init_locks(struct ad1848_softc *sc, int ipl)
{

mutex_init(&sc->sc_lock, MUTEX_DEFAULT, IPL_NONE);
mutex_init(&sc->sc_intr_lock, MUTEX_DEFAULT, ipl);
}

void
ad1848_destroy_locks(struct ad1848_softc *sc)
{

mutex_destroy(&sc->sc_lock);
mutex_destroy(&sc->sc_intr_lock);
}