Expand PMF_FN_* macros.

[netbsd-mini2440.git] / sys / dev / ic / ad1848.c

/*      $NetBSD: ad1848.c,v 1.28 2008/04/28 18:49:27 garbled Exp $      */

/*-
 * Copyright (c) 1999 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 <hannu@voxware.pp.fi>
 *
 * Portions also supplied from the SoundBlaster driver for NetBSD.
 */

#include <sys/cdefs.h>
__KERNEL_RCSID(0, "$NetBSD: ad1848.c,v 1.28 2008/04/28 18:49:27 garbled 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_if.h>
#include <dev/auconv.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

/*
 * 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;
}


inline 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;
}

inline 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).
 */

inline 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;
}

inline 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 chaing
 *  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_encoding(void *addr, struct audio_encoding *fp)
{
        struct ad1848_softc *sc;

        sc = addr;
        switch (fp->index) {
        case 0:
                strcpy(fp->name, AudioEmulaw);
                fp->encoding = AUDIO_ENCODING_ULAW;
                fp->precision = 8;
                fp->flags = 0;
                break;
        case 1:
                strcpy(fp->name, AudioEalaw);
                fp->encoding = AUDIO_ENCODING_ALAW;
                fp->precision = 8;
                fp->flags = 0;
                break;
        case 2:
                strcpy(fp->name, AudioEslinear_le);
                fp->encoding = AUDIO_ENCODING_SLINEAR_LE;
                fp->precision = 16;
                fp->flags = 0;
                break;
        case 3:
                strcpy(fp->name, AudioEulinear);
                fp->encoding = AUDIO_ENCODING_ULINEAR;
                fp->precision = 8;
                fp->flags = 0;
                break;

        case 4: /* only on CS4231 */
                strcpy(fp->name, AudioEslinear_be);
                fp->encoding = AUDIO_ENCODING_SLINEAR_BE;
                fp->precision = 16;
                fp->flags = sc->mode == 1
#if AD1845_HACK
                    || sc->is_ad1845
#endif
                        ? AUDIO_ENCODINGFLAG_EMULATED : 0;
                break;

                /* emulate some modes */
        case 5:
                strcpy(fp->name, AudioEslinear);
                fp->encoding = AUDIO_ENCODING_SLINEAR;
                fp->precision = 8;
                fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
                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;

        case 8: /* only on CS4231 */
                if (sc->mode == 1 || sc->is_ad1845)
                        return EINVAL;
                strcpy(fp->name, AudioEadpcm);
                fp->encoding = AUDIO_ENCODING_ADPCM;
                fp->precision = 4;
                fp->flags = 0;
                break;
        default:
                return EINVAL;
                /*NOTREACHED*/
        }
        return 0;
}

int
ad1848_set_params(void *addr, int setmode, int usemode,
    audio_params_t *p, audio_params_t *r, stream_filter_list_t *pfil,
    stream_filter_list_t *rfil)
{
        audio_params_t phw, rhw;
        struct ad1848_softc *sc;
        int error, bits, enc;
        stream_filter_factory_t *pswcode;
        stream_filter_factory_t *rswcode;

        DPRINTF(("ad1848_set_params: %u %u %u %u\n",
                 p->encoding, p->precision, p->channels, p->sample_rate));

        sc = addr;
        enc = p->encoding;
        pswcode = rswcode = 0;
        phw = *p;
        rhw = *r;
        switch (enc) {
        case AUDIO_ENCODING_SLINEAR_LE:
                if (p->precision == 8) {
                        enc = AUDIO_ENCODING_ULINEAR_LE;
                        phw.encoding = AUDIO_ENCODING_ULINEAR_LE;
                        rhw.encoding = AUDIO_ENCODING_ULINEAR_LE;
                        pswcode = rswcode = change_sign8;
                }
                break;
        case AUDIO_ENCODING_SLINEAR_BE:
                if (p->precision == 16 && (sc->mode == 1
#if AD1845_HACK
                    || sc->is_ad1845
#endif
                        )) {
                        enc = AUDIO_ENCODING_SLINEAR_LE;
                        phw.encoding = AUDIO_ENCODING_SLINEAR_LE;
                        rhw.encoding = AUDIO_ENCODING_SLINEAR_LE;
                        pswcode = rswcode = swap_bytes;
                }
                break;
        case AUDIO_ENCODING_ULINEAR_LE:
                if (p->precision == 16) {
                        enc = AUDIO_ENCODING_SLINEAR_LE;
                        phw.encoding = AUDIO_ENCODING_SLINEAR_LE;
                        rhw.encoding = AUDIO_ENCODING_SLINEAR_LE;
                        pswcode = rswcode = change_sign16;
                }
                break;
        case AUDIO_ENCODING_ULINEAR_BE:
                if (p->precision == 16) {
                        if (sc->mode == 1
#if AD1845_HACK
                            || sc->is_ad1845
#endif
                                ) {
                                enc = AUDIO_ENCODING_SLINEAR_LE;
                                phw.encoding = AUDIO_ENCODING_SLINEAR_LE;
                                rhw.encoding = AUDIO_ENCODING_SLINEAR_LE;
                                pswcode = swap_bytes_change_sign16;
                                rswcode = swap_bytes_change_sign16;
                        } else {
                                enc = AUDIO_ENCODING_SLINEAR_BE;
                                phw.encoding = AUDIO_ENCODING_SLINEAR_BE;
                                rhw.encoding = AUDIO_ENCODING_SLINEAR_BE;
                                pswcode = rswcode = change_sign16;
                        }
                }
                break;
        }
        switch (enc) {
        case AUDIO_ENCODING_ULAW:
                bits = FMT_ULAW >> 5;
                break;
        case AUDIO_ENCODING_ALAW:
                bits = FMT_ALAW >> 5;
                break;
        case AUDIO_ENCODING_ADPCM:
                bits = FMT_ADPCM >> 5;
                break;
        case AUDIO_ENCODING_SLINEAR_LE:
                if (p->precision == 16)
                        bits = FMT_TWOS_COMP >> 5;
                else
                        return EINVAL;
                break;
        case AUDIO_ENCODING_SLINEAR_BE:
                if (p->precision == 16)
                        bits = FMT_TWOS_COMP_BE >> 5;
                else
                        return EINVAL;
                break;
        case AUDIO_ENCODING_ULINEAR_LE:
                if (p->precision == 8)
                        bits = FMT_PCM8 >> 5;
                else
                        return EINVAL;
                break;
        default:
                return EINVAL;
        }

        if (p->channels < 1 || p->channels > 2)
                return EINVAL;

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

        if (pswcode != NULL)
                pfil->append(pfil, pswcode, &phw);
        if (rswcode != NULL)
                rfil->append(rfil, rswcode, &rhw);

        sc->format_bits = bits;
        sc->channels = p->channels;
        sc->precision = p->precision;
        sc->need_commit = 1;

        DPRINTF(("ad1848_set_params succeeded, bits=%x\n", bits));
        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_round_blocksize(void *addr, int blk,
    int mode, const audio_params_t *param)
{

        /* Round to a multiple of the biggest sample size. */
        return blk &= -4;
}

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;
}

/*
 * Close function is called at splaudio().
 */
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;
        int s;

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

        s = splaudio();

        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);

        splx(s);

        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;
}