Home | History | Annotate | Line # | Download | only in ic
ad1848.c revision 1.22
      1 /*	$NetBSD: ad1848.c,v 1.22 2005/12/24 20:27:29 perry Exp $	*/
      2 
      3 /*-
      4  * Copyright (c) 1999 The NetBSD Foundation, Inc.
      5  * All rights reserved.
      6  *
      7  * This code is derived from software contributed to The NetBSD Foundation
      8  * by Ken Hornstein and John Kohl.
      9  *
     10  * Redistribution and use in source and binary forms, with or without
     11  * modification, are permitted provided that the following conditions
     12  * are met:
     13  * 1. Redistributions of source code must retain the above copyright
     14  *    notice, this list of conditions and the following disclaimer.
     15  * 2. Redistributions in binary form must reproduce the above copyright
     16  *    notice, this list of conditions and the following disclaimer in the
     17  *    documentation and/or other materials provided with the distribution.
     18  * 3. All advertising materials mentioning features or use of this software
     19  *    must display the following acknowledgement:
     20  *	This product includes software developed by the NetBSD
     21  *	Foundation, Inc. and its contributors.
     22  * 4. Neither the name of The NetBSD Foundation nor the names of its
     23  *    contributors may be used to endorse or promote products derived
     24  *    from this software without specific prior written permission.
     25  *
     26  * THIS SOFTWARE IS PROVIDED BY THE NETBSD FOUNDATION, INC. AND CONTRIBUTORS
     27  * ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED
     28  * TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
     29  * PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR CONTRIBUTORS
     30  * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
     31  * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
     32  * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
     33  * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
     34  * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
     35  * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
     36  * POSSIBILITY OF SUCH DAMAGE.
     37  */
     38 /*
     39  * Copyright (c) 1994 John Brezak
     40  * Copyright (c) 1991-1993 Regents of the University of California.
     41  * All rights reserved.
     42  *
     43  * Redistribution and use in source and binary forms, with or without
     44  * modification, are permitted provided that the following conditions
     45  * are met:
     46  * 1. Redistributions of source code must retain the above copyright
     47  *    notice, this list of conditions and the following disclaimer.
     48  * 2. Redistributions in binary form must reproduce the above copyright
     49  *    notice, this list of conditions and the following disclaimer in the
     50  *    documentation and/or other materials provided with the distribution.
     51  * 3. All advertising materials mentioning features or use of this software
     52  *    must display the following acknowledgement:
     53  *	This product includes software developed by the Computer Systems
     54  *	Engineering Group at Lawrence Berkeley Laboratory.
     55  * 4. Neither the name of the University nor of the Laboratory may be used
     56  *    to endorse or promote products derived from this software without
     57  *    specific prior written permission.
     58  *
     59  * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND
     60  * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
     61  * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
     62  * ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
     63  * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
     64  * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
     65  * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
     66  * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
     67  * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
     68  * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
     69  * SUCH DAMAGE.
     70  *
     71  */
     72 
     73 /*
     74  * Copyright by Hannu Savolainen 1994
     75  *
     76  * Redistribution and use in source and binary forms, with or without
     77  * modification, are permitted provided that the following conditions are
     78  * met: 1. Redistributions of source code must retain the above copyright
     79  * notice, this list of conditions and the following disclaimer. 2.
     80  * Redistributions in binary form must reproduce the above copyright notice,
     81  * this list of conditions and the following disclaimer in the documentation
     82  * and/or other materials provided with the distribution.
     83  *
     84  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR AND CONTRIBUTORS ``AS IS'' AND ANY
     85  * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
     86  * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
     87  * DISCLAIMED.  IN NO EVENT SHALL THE AUTHOR OR CONTRIBUTORS BE LIABLE FOR
     88  * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
     89  * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
     90  * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
     91  * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
     92  * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
     93  * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
     94  * SUCH DAMAGE.
     95  *
     96  */
     97 /*
     98  * Portions of this code are from the VOXware support for the ad1848
     99  * by Hannu Savolainen <hannu (at) voxware.pp.fi>
    100  *
    101  * Portions also supplied from the SoundBlaster driver for NetBSD.
    102  */
    103 
    104 #include <sys/cdefs.h>
    105 __KERNEL_RCSID(0, "$NetBSD: ad1848.c,v 1.22 2005/12/24 20:27:29 perry Exp $");
    106 
    107 #include <sys/param.h>
    108 #include <sys/systm.h>
    109 #include <sys/errno.h>
    110 #include <sys/ioctl.h>
    111 #include <sys/device.h>
    112 #include <sys/fcntl.h>
    113 /*#include <sys/syslog.h>*/
    114 /*#include <sys/proc.h>*/
    115 
    116 #include <machine/cpu.h>
    117 #include <machine/bus.h>
    118 
    119 #include <sys/audioio.h>
    120 
    121 #include <dev/audio_if.h>
    122 #include <dev/auconv.h>
    123 
    124 #include <dev/ic/ad1848reg.h>
    125 #include <dev/ic/cs4231reg.h>
    126 #include <dev/ic/cs4237reg.h>
    127 #include <dev/ic/ad1848var.h>
    128 #if 0
    129 #include <dev/isa/cs4231var.h>
    130 #endif
    131 
    132 /*
    133  * AD1845 on some machines don't match the AD1845 doc
    134  * and defining AD1845_HACK to 1 works around the problems.
    135  * options AD1845_HACK=0  should work if you have ``correct'' one.
    136  */
    137 #ifndef AD1845_HACK
    138 #define AD1845_HACK	1	/* weird mixer, can't play slinear_be */
    139 #endif
    140 
    141 #ifdef AUDIO_DEBUG
    142 #define DPRINTF(x)	if (ad1848debug) printf x
    143 int	ad1848debug = 0;
    144 #else
    145 #define DPRINTF(x)
    146 #endif
    147 
    148 /*
    149  * Initial values for the indirect registers of CS4248/AD1848.
    150  */
    151 static const int ad1848_init_values[] = {
    152     GAIN_12|INPUT_MIC_GAIN_ENABLE,	/* Left Input Control */
    153     GAIN_12|INPUT_MIC_GAIN_ENABLE,	/* Right Input Control */
    154     ATTEN_12,				/* Left Aux #1 Input Control */
    155     ATTEN_12,				/* Right Aux #1 Input Control */
    156     ATTEN_12,				/* Left Aux #2 Input Control */
    157     ATTEN_12,				/* Right Aux #2 Input Control */
    158     /* bits 5-0 are attenuation select */
    159     ATTEN_12,				/* Left DAC output Control */
    160     ATTEN_12,				/* Right DAC output Control */
    161     CLOCK_XTAL1|FMT_PCM8,		/* Clock and Data Format */
    162     SINGLE_DMA|AUTO_CAL_ENABLE,		/* Interface Config */
    163     INTERRUPT_ENABLE,			/* Pin control */
    164     0x00,				/* Test and Init */
    165     MODE2,				/* Misc control */
    166     ATTEN_0<<2,				/* Digital Mix Control */
    167     0,					/* Upper base Count */
    168     0,					/* Lower base Count */
    169 
    170     /* These are for CS4231 &c. only (additional registers): */
    171     0,					/* Alt feature 1 */
    172     0,					/* Alt feature 2 */
    173     ATTEN_12,				/* Left line in */
    174     ATTEN_12,				/* Right line in */
    175     0,					/* Timer low */
    176     0,					/* Timer high */
    177     0,					/* unused */
    178     0,					/* unused */
    179     0,					/* IRQ status */
    180     0,					/* unused */
    181 			/* Mono input (a.k.a speaker) (mic) Control */
    182     MONO_INPUT_MUTE|ATTEN_6,		/* mute speaker by default */
    183     0,					/* unused */
    184     0,					/* record format */
    185     0,					/* Crystal Clock Select */
    186     0,					/* upper record count */
    187     0					/* lower record count */
    188 };
    189 
    190 
    191 int
    192 ad1848_to_vol(mixer_ctrl_t *cp, struct ad1848_volume *vol)
    193 {
    194 
    195 	if (cp->un.value.num_channels == 1) {
    196 		vol->left =
    197 		vol->right = cp->un.value.level[AUDIO_MIXER_LEVEL_MONO];
    198 		return 1;
    199 	}
    200 	else if (cp->un.value.num_channels == 2) {
    201 		vol->left  = cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT];
    202 		vol->right = cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT];
    203 		return 1;
    204 	}
    205 	return 0;
    206 }
    207 
    208 int
    209 ad1848_from_vol(mixer_ctrl_t *cp, struct ad1848_volume *vol)
    210 {
    211 
    212 	if (cp->un.value.num_channels == 1) {
    213 		cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = vol->left;
    214 		return 1;
    215 	}
    216 	else if (cp->un.value.num_channels == 2) {
    217 		cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = vol->left;
    218 		cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = vol->right;
    219 		return 1;
    220 	}
    221 	return 0;
    222 }
    223 
    224 
    225 inline int
    226 ad_read(struct ad1848_softc *sc, int reg)
    227 {
    228 	int x;
    229 
    230 	ADWRITE(sc, AD1848_IADDR, (reg & 0xff) | sc->MCE_bit);
    231 	x = ADREAD(sc, AD1848_IDATA);
    232 	/*  printf("(%02x<-%02x) ", reg|sc->MCE_bit, x); */
    233 	return x;
    234 }
    235 
    236 inline void
    237 ad_write(struct ad1848_softc *sc, int reg, int data)
    238 {
    239 
    240 	ADWRITE(sc, AD1848_IADDR, (reg & 0xff) | sc->MCE_bit);
    241 	ADWRITE(sc, AD1848_IDATA, data & 0xff);
    242 	/* printf("(%02x->%02x) ", reg|sc->MCE_bit, data); */
    243 }
    244 
    245 /*
    246  * extended registers (mode 3) require an additional level of
    247  * indirection through CS_XREG (I23).
    248  */
    249 
    250 inline int
    251 ad_xread(struct ad1848_softc *sc, int reg)
    252 {
    253 	int x;
    254 
    255 	ADWRITE(sc, AD1848_IADDR, CS_XREG | sc->MCE_bit);
    256 	ADWRITE(sc, AD1848_IDATA, (reg | ALT_F3_XRAE) & 0xff);
    257 	x = ADREAD(sc, AD1848_IDATA);
    258 
    259 	return x;
    260 }
    261 
    262 inline void
    263 ad_xwrite(struct ad1848_softc *sc, int reg, int val)
    264 {
    265 
    266 	ADWRITE(sc, AD1848_IADDR, CS_XREG | sc->MCE_bit);
    267 	ADWRITE(sc, AD1848_IDATA, (reg | ALT_F3_XRAE) & 0xff);
    268 	ADWRITE(sc, AD1848_IDATA, val & 0xff);
    269 }
    270 
    271 static void
    272 ad_set_MCE(struct ad1848_softc *sc, int state)
    273 {
    274 
    275 	if (state)
    276 		sc->MCE_bit = MODE_CHANGE_ENABLE;
    277 	else
    278 		sc->MCE_bit = 0;
    279 	ADWRITE(sc, AD1848_IADDR, sc->MCE_bit);
    280 }
    281 
    282 static void
    283 wait_for_calibration(struct ad1848_softc *sc)
    284 {
    285 	int timeout;
    286 
    287 	DPRINTF(("ad1848: Auto calibration started.\n"));
    288 	/*
    289 	 * Wait until the auto calibration process has finished.
    290 	 *
    291 	 * 1) Wait until the chip becomes ready (reads don't return 0x80).
    292 	 * 2) Wait until the ACI bit of I11 gets on and then off.
    293 	 *    Because newer chips are fast we may never see the ACI
    294 	 *    bit go on.  Just delay a little instead.
    295 	 */
    296 	timeout = 10000;
    297 	while (timeout > 0 && ADREAD(sc, AD1848_IADDR) == SP_IN_INIT) {
    298 		delay(10);
    299 		timeout--;
    300 	}
    301 	if (timeout <= 0)
    302 		DPRINTF(("ad1848: Auto calibration timed out(1).\n"));
    303 
    304 	/* Set register addr */
    305 	ADWRITE(sc, AD1848_IADDR, SP_TEST_AND_INIT);
    306 	/* Wait for address to appear when read back. */
    307 	timeout = 100000;
    308 	while (timeout > 0 &&
    309 	       (ADREAD(sc, AD1848_IADDR)&SP_IADDR_MASK) != SP_TEST_AND_INIT) {
    310 		delay(10);
    311 		timeout--;
    312 	}
    313 	if (timeout <= 0)
    314 		DPRINTF(("ad1848: Auto calibration timed out(1.5).\n"));
    315 
    316 	if (!(ad_read(sc, SP_TEST_AND_INIT) & AUTO_CAL_IN_PROG)) {
    317 		if (sc->mode > 1) {
    318 			/* A new chip, just delay a little. */
    319 			delay(100);	/* XXX what should it be? */
    320 		} else {
    321 			timeout = 10000;
    322 			while (timeout > 0 &&
    323 			       !(ad_read(sc, SP_TEST_AND_INIT) &
    324 				 AUTO_CAL_IN_PROG)) {
    325 				delay(10);
    326 				timeout--;
    327 			}
    328 			if (timeout <= 0)
    329 				DPRINTF(("ad1848: Auto calibration timed out(2).\n"));
    330 		}
    331 	}
    332 
    333 	timeout = 10000;
    334 	while (timeout > 0 &&
    335 	       ad_read(sc, SP_TEST_AND_INIT) & AUTO_CAL_IN_PROG) {
    336 		delay(10);
    337 		timeout--;
    338 	}
    339 	if (timeout <= 0)
    340 		DPRINTF(("ad1848: Auto calibration timed out(3).\n"));
    341 }
    342 
    343 #ifdef AUDIO_DEBUG
    344 void
    345 ad1848_dump_regs(struct ad1848_softc *sc)
    346 {
    347 	int i;
    348 	u_char r;
    349 
    350 	printf("ad1848 status=%02x", ADREAD(sc, AD1848_STATUS));
    351 	printf(" regs: ");
    352 	for (i = 0; i < 16; i++) {
    353 		r = ad_read(sc, i);
    354 		printf("%02x ", r);
    355 	}
    356 	if (sc->mode >= 2) {
    357 		for (i = 16; i < 32; i++) {
    358 			r = ad_read(sc, i);
    359 			printf("%02x ", r);
    360 		}
    361 	}
    362 	printf("\n");
    363 }
    364 #endif /* AUDIO_DEBUG */
    365 
    366 
    367 /*
    368  * Attach hardware to driver, attach hardware driver to audio
    369  * pseudo-device driver .
    370  */
    371 void
    372 ad1848_attach(struct ad1848_softc *sc)
    373 {
    374 	static struct ad1848_volume vol_mid = {220, 220};
    375 	static struct ad1848_volume vol_0   = {0, 0};
    376 	int i;
    377 	int timeout;
    378 
    379 	/* Initialize the ad1848... */
    380 	for (i = 0; i < 0x10; i++) {
    381 		ad_write(sc, i, ad1848_init_values[i]);
    382 		timeout = 100000;
    383 		while (timeout > 0 && ADREAD(sc, AD1848_IADDR) & SP_IN_INIT)
    384 			timeout--;
    385 	}
    386 	/* ...and additional CS4231 stuff too */
    387 	if (sc->mode >= 2) {
    388 		ad_write(sc, SP_INTERFACE_CONFIG, 0); /* disable SINGLE_DMA */
    389 		for (i = 0x10; i < 0x20; i++)
    390 			if (ad1848_init_values[i] != 0) {
    391 				ad_write(sc, i, ad1848_init_values[i]);
    392 				timeout = 100000;
    393 				while (timeout > 0 &&
    394 				       ADREAD(sc, AD1848_IADDR) & SP_IN_INIT)
    395 					timeout--;
    396 			}
    397 	}
    398 	ad1848_reset(sc);
    399 
    400 	/* Set default gains */
    401 	ad1848_set_rec_gain(sc, &vol_mid);
    402 	ad1848_set_channel_gain(sc, AD1848_DAC_CHANNEL, &vol_mid);
    403 	ad1848_set_channel_gain(sc, AD1848_MONITOR_CHANNEL, &vol_0);
    404 	ad1848_set_channel_gain(sc, AD1848_AUX1_CHANNEL, &vol_mid);	/* CD volume */
    405 	sc->mute[AD1848_MONITOR_CHANNEL] = MUTE_ALL;
    406 	if (sc->mode >= 2
    407 #if AD1845_HACK
    408 	    && sc->is_ad1845 == 0
    409 #endif
    410 		) {
    411 		ad1848_set_channel_gain(sc, AD1848_AUX2_CHANNEL, &vol_mid); /* CD volume */
    412 		ad1848_set_channel_gain(sc, AD1848_LINE_CHANNEL, &vol_mid);
    413 		ad1848_set_channel_gain(sc, AD1848_MONO_CHANNEL, &vol_0);
    414 		sc->mute[AD1848_MONO_CHANNEL] = MUTE_ALL;
    415 	} else
    416 		ad1848_set_channel_gain(sc, AD1848_AUX2_CHANNEL, &vol_0);
    417 
    418 	/* Set default port */
    419 	ad1848_set_rec_port(sc, MIC_IN_PORT);
    420 
    421 	printf(": %s", sc->chip_name);
    422 }
    423 
    424 /*
    425  * Various routines to interface to higher level audio driver
    426  */
    427 static const struct ad1848_mixerinfo {
    428 	int  left_reg;
    429 	int  right_reg;
    430 	int  atten_bits;
    431 	int  atten_mask;
    432 } mixer_channel_info[] =
    433 {
    434   { SP_LEFT_AUX2_CONTROL, SP_RIGHT_AUX2_CONTROL, AUX_INPUT_ATTEN_BITS,
    435     AUX_INPUT_ATTEN_MASK },
    436   { SP_LEFT_AUX1_CONTROL, SP_RIGHT_AUX1_CONTROL, AUX_INPUT_ATTEN_BITS,
    437     AUX_INPUT_ATTEN_MASK },
    438   { SP_LEFT_OUTPUT_CONTROL, SP_RIGHT_OUTPUT_CONTROL,
    439     OUTPUT_ATTEN_BITS, OUTPUT_ATTEN_MASK },
    440   { CS_LEFT_LINE_CONTROL, CS_RIGHT_LINE_CONTROL, LINE_INPUT_ATTEN_BITS,
    441     LINE_INPUT_ATTEN_MASK },
    442   { CS_MONO_IO_CONTROL, 0, MONO_INPUT_ATTEN_BITS, MONO_INPUT_ATTEN_MASK },
    443   { CS_MONO_IO_CONTROL, 0, 0, 0 },
    444   { SP_DIGITAL_MIX, 0, OUTPUT_ATTEN_BITS, MIX_ATTEN_MASK }
    445 };
    446 
    447 /*
    448  *  This function doesn't set the mute flags but does use them.
    449  *  The mute flags reflect the mutes that have been applied by the user.
    450  *  However, the driver occasionally wants to mute devices (e.g. when chaing
    451  *  sampling rate). These operations should not affect the mute flags.
    452  */
    453 
    454 void
    455 ad1848_mute_channel(struct ad1848_softc *sc, int device, int mute)
    456 {
    457 	u_char reg;
    458 
    459 	reg = ad_read(sc, mixer_channel_info[device].left_reg);
    460 
    461 	if (mute & MUTE_LEFT) {
    462 		if (device == AD1848_MONITOR_CHANNEL) {
    463 			if (sc->open_mode & FREAD)
    464 				ad1848_mute_wave_output(sc, WAVE_UNMUTE1, 0);
    465 			ad_write(sc, mixer_channel_info[device].left_reg,
    466 				 reg & ~DIGITAL_MIX1_ENABLE);
    467 		} else if (device == AD1848_OUT_CHANNEL)
    468 			ad_write(sc, mixer_channel_info[device].left_reg,
    469 				 reg | MONO_OUTPUT_MUTE);
    470 		else
    471 			ad_write(sc, mixer_channel_info[device].left_reg,
    472 				 reg | 0x80);
    473 	} else if (!(sc->mute[device] & MUTE_LEFT)) {
    474 		if (device == AD1848_MONITOR_CHANNEL) {
    475 			ad_write(sc, mixer_channel_info[device].left_reg,
    476 				 reg | DIGITAL_MIX1_ENABLE);
    477 			if (sc->open_mode & FREAD)
    478 				ad1848_mute_wave_output(sc, WAVE_UNMUTE1, 1);
    479 		} else if (device == AD1848_OUT_CHANNEL)
    480 			ad_write(sc, mixer_channel_info[device].left_reg,
    481 				 reg & ~MONO_OUTPUT_MUTE);
    482 		else
    483 			ad_write(sc, mixer_channel_info[device].left_reg,
    484 				 reg & ~0x80);
    485 	}
    486 
    487 	if (!mixer_channel_info[device].right_reg)
    488 		return;
    489 
    490 	reg = ad_read(sc, mixer_channel_info[device].right_reg);
    491 
    492 	if (mute & MUTE_RIGHT) {
    493 		ad_write(sc, mixer_channel_info[device].right_reg, reg | 0x80);
    494 	} else if (!(sc->mute[device] & MUTE_RIGHT)) {
    495 		ad_write(sc, mixer_channel_info[device].right_reg, reg &~0x80);
    496 	}
    497 }
    498 
    499 int
    500 ad1848_set_channel_gain(struct ad1848_softc *sc, int device,
    501     struct ad1848_volume *gp)
    502 {
    503 	const struct ad1848_mixerinfo *info;
    504 	u_char reg;
    505 	u_int atten;
    506 
    507 	info = &mixer_channel_info[device];
    508 	sc->gains[device] = *gp;
    509 
    510 	atten = (AUDIO_MAX_GAIN - gp->left) * (info->atten_bits + 1) /
    511 		(AUDIO_MAX_GAIN + 1);
    512 
    513 	reg = ad_read(sc, info->left_reg) & (info->atten_mask);
    514 	if (device == AD1848_MONITOR_CHANNEL)
    515 		reg |= ((atten & info->atten_bits) << 2);
    516 	else
    517 		reg |= ((atten & info->atten_bits));
    518 
    519 	ad_write(sc, info->left_reg, reg);
    520 
    521 	if (!info->right_reg)
    522 		return 0;
    523 
    524 	atten = (AUDIO_MAX_GAIN - gp->right) * (info->atten_bits + 1) /
    525 		(AUDIO_MAX_GAIN + 1);
    526 	reg = ad_read(sc, info->right_reg);
    527 	reg &= info->atten_mask;
    528 	ad_write(sc, info->right_reg, (atten & info->atten_bits) | reg);
    529 
    530 	return 0;
    531 }
    532 
    533 int
    534 ad1848_get_device_gain(struct ad1848_softc *sc, int device,
    535     struct ad1848_volume *gp)
    536 {
    537 
    538 	*gp = sc->gains[device];
    539 	return 0;
    540 }
    541 
    542 int
    543 ad1848_get_rec_gain(struct ad1848_softc *sc, struct ad1848_volume *gp)
    544 {
    545 
    546 	*gp = sc->rec_gain;
    547 	return 0;
    548 }
    549 
    550 int
    551 ad1848_set_rec_gain(struct ad1848_softc *sc, struct ad1848_volume *gp)
    552 {
    553 	u_char reg, gain;
    554 
    555 	DPRINTF(("ad1848_set_rec_gain: %d:%d\n", gp->left, gp->right));
    556 
    557 	sc->rec_gain = *gp;
    558 
    559 	gain = (gp->left * (GAIN_22_5 + 1)) / (AUDIO_MAX_GAIN + 1);
    560 	reg = ad_read(sc, SP_LEFT_INPUT_CONTROL);
    561 	reg &= INPUT_GAIN_MASK;
    562 	ad_write(sc, SP_LEFT_INPUT_CONTROL, (gain & 0x0f) | reg);
    563 
    564 	gain = (gp->right * (GAIN_22_5 + 1)) / (AUDIO_MAX_GAIN + 1);
    565 	reg = ad_read(sc, SP_RIGHT_INPUT_CONTROL);
    566 	reg &= INPUT_GAIN_MASK;
    567 	ad_write(sc, SP_RIGHT_INPUT_CONTROL, (gain & 0x0f) | reg);
    568 
    569 	return 0;
    570 }
    571 
    572 void
    573 ad1848_mute_wave_output(struct ad1848_softc *sc, int mute, int set)
    574 {
    575 	int m;
    576 
    577 	DPRINTF(("ad1848_mute_wave_output: %d, %d\n", mute, set));
    578 
    579 	if (mute == WAVE_MUTE2_INIT) {
    580 		sc->wave_mute_status = 0;
    581 		mute = WAVE_MUTE2;
    582 	}
    583 	if (set)
    584 		m = sc->wave_mute_status |= mute;
    585 	else
    586 		m = sc->wave_mute_status &= ~mute;
    587 
    588 	if (m & WAVE_MUTE0 || ((m & WAVE_UNMUTE1) == 0 && m & WAVE_MUTE2))
    589 		ad1848_mute_channel(sc, AD1848_DAC_CHANNEL, MUTE_ALL);
    590 	else
    591 		ad1848_mute_channel(sc, AD1848_DAC_CHANNEL,
    592 					    sc->mute[AD1848_DAC_CHANNEL]);
    593 }
    594 
    595 int
    596 ad1848_set_mic_gain(struct ad1848_softc *sc, struct ad1848_volume *gp)
    597 {
    598 	u_char reg;
    599 
    600 	DPRINTF(("cs4231_set_mic_gain: %d\n", gp->left));
    601 
    602 	if (gp->left > AUDIO_MAX_GAIN/2) {
    603 		sc->mic_gain_on = 1;
    604 		reg = ad_read(sc, SP_LEFT_INPUT_CONTROL);
    605 		ad_write(sc, SP_LEFT_INPUT_CONTROL,
    606 			 reg | INPUT_MIC_GAIN_ENABLE);
    607 	} else {
    608 		sc->mic_gain_on = 0;
    609 		reg = ad_read(sc, SP_LEFT_INPUT_CONTROL);
    610 		ad_write(sc, SP_LEFT_INPUT_CONTROL,
    611 			 reg & ~INPUT_MIC_GAIN_ENABLE);
    612 	}
    613 
    614 	return 0;
    615 }
    616 
    617 int
    618 ad1848_get_mic_gain(struct ad1848_softc *sc, struct ad1848_volume *gp)
    619 {
    620 	if (sc->mic_gain_on)
    621 		gp->left = gp->right = AUDIO_MAX_GAIN;
    622 	else
    623 		gp->left = gp->right = AUDIO_MIN_GAIN;
    624 	return 0;
    625 }
    626 
    627 static ad1848_devmap_t *
    628 ad1848_mixer_find_dev(ad1848_devmap_t *map, int cnt, mixer_ctrl_t *cp)
    629 {
    630 	int i;
    631 
    632 	for (i = 0; i < cnt; i++) {
    633 		if (map[i].id == cp->dev) {
    634 			return (&map[i]);
    635 		}
    636 	}
    637 	return 0;
    638 }
    639 
    640 int
    641 ad1848_mixer_get_port(struct ad1848_softc *ac, struct ad1848_devmap *map,
    642     int cnt, mixer_ctrl_t *cp)
    643 {
    644 	ad1848_devmap_t *entry;
    645 	struct ad1848_volume vol;
    646 	int error;
    647 	int dev;
    648 
    649 	error = EINVAL;
    650 	if (!(entry = ad1848_mixer_find_dev(map, cnt, cp)))
    651 		return ENXIO;
    652 
    653 	dev = entry->dev;
    654 
    655 	switch (entry->kind) {
    656 	case AD1848_KIND_LVL:
    657 		if (cp->type != AUDIO_MIXER_VALUE)
    658 			break;
    659 
    660 		if (dev < AD1848_AUX2_CHANNEL ||
    661 		    dev > AD1848_MONITOR_CHANNEL)
    662 			break;
    663 
    664 		if (cp->un.value.num_channels != 1 &&
    665 		    mixer_channel_info[dev].right_reg == 0)
    666 			break;
    667 
    668 		error = ad1848_get_device_gain(ac, dev, &vol);
    669 		if (!error)
    670 			ad1848_from_vol(cp, &vol);
    671 
    672 		break;
    673 
    674 	case AD1848_KIND_MUTE:
    675 		if (cp->type != AUDIO_MIXER_ENUM) break;
    676 
    677 		cp->un.ord = ac->mute[dev] ? 1 : 0;
    678 		error = 0;
    679 		break;
    680 
    681 	case AD1848_KIND_RECORDGAIN:
    682 		if (cp->type != AUDIO_MIXER_VALUE) break;
    683 
    684 		error = ad1848_get_rec_gain(ac, &vol);
    685 		if (!error)
    686 			ad1848_from_vol(cp, &vol);
    687 
    688 		break;
    689 
    690 	case AD1848_KIND_MICGAIN:
    691 		if (cp->type != AUDIO_MIXER_VALUE) break;
    692 
    693 		error = ad1848_get_mic_gain(ac, &vol);
    694 		if (!error)
    695 			ad1848_from_vol(cp, &vol);
    696 
    697 		break;
    698 
    699 	case AD1848_KIND_RECORDSOURCE:
    700 		if (cp->type != AUDIO_MIXER_ENUM) break;
    701 		cp->un.ord = ad1848_get_rec_port(ac);
    702 		error = 0;
    703 		break;
    704 
    705 	default:
    706 		printf ("Invalid kind\n");
    707 		break;
    708 	}
    709 
    710 	return error;
    711 }
    712 
    713 int
    714 ad1848_mixer_set_port(struct ad1848_softc *ac, struct ad1848_devmap *map,
    715     int cnt, mixer_ctrl_t *cp)
    716 {
    717 	ad1848_devmap_t *entry;
    718 	struct ad1848_volume vol;
    719 	int error;
    720 	int dev;
    721 
    722 	error = EINVAL;
    723 	if (!(entry = ad1848_mixer_find_dev(map, cnt, cp)))
    724 		return ENXIO;
    725 
    726 	dev = entry->dev;
    727 
    728 	switch (entry->kind) {
    729 	case AD1848_KIND_LVL:
    730 		if (cp->type != AUDIO_MIXER_VALUE)
    731 			break;
    732 
    733 		if (dev < AD1848_AUX2_CHANNEL ||
    734 		    dev > AD1848_MONITOR_CHANNEL)
    735 			break;
    736 
    737 		if (cp->un.value.num_channels != 1 &&
    738 		    mixer_channel_info[dev].right_reg == 0)
    739 			break;
    740 
    741 		ad1848_to_vol(cp, &vol);
    742 		error = ad1848_set_channel_gain(ac, dev, &vol);
    743 		break;
    744 
    745 	case AD1848_KIND_MUTE:
    746 		if (cp->type != AUDIO_MIXER_ENUM) break;
    747 
    748 		ac->mute[dev] = (cp->un.ord ? MUTE_ALL : 0);
    749 		ad1848_mute_channel(ac, dev, ac->mute[dev]);
    750 		error = 0;
    751 		break;
    752 
    753 	case AD1848_KIND_RECORDGAIN:
    754 		if (cp->type != AUDIO_MIXER_VALUE) break;
    755 
    756 		ad1848_to_vol(cp, &vol);
    757 		error = ad1848_set_rec_gain(ac, &vol);
    758 		break;
    759 
    760 	case AD1848_KIND_MICGAIN:
    761 		if (cp->type != AUDIO_MIXER_VALUE) break;
    762 
    763 		ad1848_to_vol(cp, &vol);
    764 		error = ad1848_set_mic_gain(ac, &vol);
    765 		break;
    766 
    767 	case AD1848_KIND_RECORDSOURCE:
    768 		if (cp->type != AUDIO_MIXER_ENUM) break;
    769 
    770 		error = ad1848_set_rec_port(ac,  cp->un.ord);
    771 		break;
    772 
    773 	default:
    774 		printf ("Invalid kind\n");
    775 		break;
    776 	}
    777 
    778 	return error;
    779 }
    780 
    781 int
    782 ad1848_query_encoding(void *addr, struct audio_encoding *fp)
    783 {
    784 	struct ad1848_softc *sc;
    785 
    786 	sc = addr;
    787 	switch (fp->index) {
    788 	case 0:
    789 		strcpy(fp->name, AudioEmulaw);
    790 		fp->encoding = AUDIO_ENCODING_ULAW;
    791 		fp->precision = 8;
    792 		fp->flags = 0;
    793 		break;
    794 	case 1:
    795 		strcpy(fp->name, AudioEalaw);
    796 		fp->encoding = AUDIO_ENCODING_ALAW;
    797 		fp->precision = 8;
    798 		fp->flags = 0;
    799 		break;
    800 	case 2:
    801 		strcpy(fp->name, AudioEslinear_le);
    802 		fp->encoding = AUDIO_ENCODING_SLINEAR_LE;
    803 		fp->precision = 16;
    804 		fp->flags = 0;
    805 		break;
    806 	case 3:
    807 		strcpy(fp->name, AudioEulinear);
    808 		fp->encoding = AUDIO_ENCODING_ULINEAR;
    809 		fp->precision = 8;
    810 		fp->flags = 0;
    811 		break;
    812 
    813 	case 4: /* only on CS4231 */
    814 		strcpy(fp->name, AudioEslinear_be);
    815 		fp->encoding = AUDIO_ENCODING_SLINEAR_BE;
    816 		fp->precision = 16;
    817 		fp->flags = sc->mode == 1
    818 #if AD1845_HACK
    819 		    || sc->is_ad1845
    820 #endif
    821 			? AUDIO_ENCODINGFLAG_EMULATED : 0;
    822 		break;
    823 
    824 		/* emulate some modes */
    825 	case 5:
    826 		strcpy(fp->name, AudioEslinear);
    827 		fp->encoding = AUDIO_ENCODING_SLINEAR;
    828 		fp->precision = 8;
    829 		fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
    830 		break;
    831 	case 6:
    832 		strcpy(fp->name, AudioEulinear_le);
    833 		fp->encoding = AUDIO_ENCODING_ULINEAR_LE;
    834 		fp->precision = 16;
    835 		fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
    836 		break;
    837 	case 7:
    838 		strcpy(fp->name, AudioEulinear_be);
    839 		fp->encoding = AUDIO_ENCODING_ULINEAR_BE;
    840 		fp->precision = 16;
    841 		fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
    842 		break;
    843 
    844 	case 8: /* only on CS4231 */
    845 		if (sc->mode == 1 || sc->is_ad1845)
    846 			return EINVAL;
    847 		strcpy(fp->name, AudioEadpcm);
    848 		fp->encoding = AUDIO_ENCODING_ADPCM;
    849 		fp->precision = 4;
    850 		fp->flags = 0;
    851 		break;
    852 	default:
    853 		return EINVAL;
    854 		/*NOTREACHED*/
    855 	}
    856 	return 0;
    857 }
    858 
    859 int
    860 ad1848_set_params(void *addr, int setmode, int usemode, audio_params_t *p,
    861     audio_params_t *r, stream_filter_list_t *pfil, stream_filter_list_t *rfil)
    862 {
    863 	audio_params_t phw, rhw;
    864 	struct ad1848_softc *sc;
    865 	int error, bits, enc;
    866 	stream_filter_factory_t *pswcode;
    867 	stream_filter_factory_t *rswcode;
    868 
    869 	DPRINTF(("ad1848_set_params: %u %u %u %u\n",
    870 		 p->encoding, p->precision, p->channels, p->sample_rate));
    871 
    872 	sc = addr;
    873 	enc = p->encoding;
    874 	pswcode = rswcode = 0;
    875 	phw = *p;
    876 	rhw = *r;
    877 	switch (enc) {
    878 	case AUDIO_ENCODING_SLINEAR_LE:
    879 		if (p->precision == 8) {
    880 			enc = AUDIO_ENCODING_ULINEAR_LE;
    881 			phw.encoding = AUDIO_ENCODING_ULINEAR_LE;
    882 			rhw.encoding = AUDIO_ENCODING_ULINEAR_LE;
    883 			pswcode = rswcode = change_sign8;
    884 		}
    885 		break;
    886 	case AUDIO_ENCODING_SLINEAR_BE:
    887 		if (p->precision == 16 && (sc->mode == 1
    888 #if AD1845_HACK
    889 		    || sc->is_ad1845
    890 #endif
    891 			)) {
    892 			enc = AUDIO_ENCODING_SLINEAR_LE;
    893 			phw.encoding = AUDIO_ENCODING_SLINEAR_LE;
    894 			rhw.encoding = AUDIO_ENCODING_SLINEAR_LE;
    895 			pswcode = rswcode = swap_bytes;
    896 		}
    897 		break;
    898 	case AUDIO_ENCODING_ULINEAR_LE:
    899 		if (p->precision == 16) {
    900 			enc = AUDIO_ENCODING_SLINEAR_LE;
    901 			phw.encoding = AUDIO_ENCODING_SLINEAR_LE;
    902 			rhw.encoding = AUDIO_ENCODING_SLINEAR_LE;
    903 			pswcode = rswcode = change_sign16;
    904 		}
    905 		break;
    906 	case AUDIO_ENCODING_ULINEAR_BE:
    907 		if (p->precision == 16) {
    908 			if (sc->mode == 1
    909 #if AD1845_HACK
    910 			    || sc->is_ad1845
    911 #endif
    912 				) {
    913 				enc = AUDIO_ENCODING_SLINEAR_LE;
    914 				phw.encoding = AUDIO_ENCODING_SLINEAR_LE;
    915 				rhw.encoding = AUDIO_ENCODING_SLINEAR_LE;
    916 				pswcode = swap_bytes_change_sign16;
    917 				rswcode = swap_bytes_change_sign16;
    918 			} else {
    919 				enc = AUDIO_ENCODING_SLINEAR_BE;
    920 				phw.encoding = AUDIO_ENCODING_SLINEAR_BE;
    921 				rhw.encoding = AUDIO_ENCODING_SLINEAR_BE;
    922 				pswcode = rswcode = change_sign16;
    923 			}
    924 		}
    925 		break;
    926 	}
    927 	switch (enc) {
    928 	case AUDIO_ENCODING_ULAW:
    929 		bits = FMT_ULAW >> 5;
    930 		break;
    931 	case AUDIO_ENCODING_ALAW:
    932 		bits = FMT_ALAW >> 5;
    933 		break;
    934 	case AUDIO_ENCODING_ADPCM:
    935 		bits = FMT_ADPCM >> 5;
    936 		break;
    937 	case AUDIO_ENCODING_SLINEAR_LE:
    938 		if (p->precision == 16)
    939 			bits = FMT_TWOS_COMP >> 5;
    940 		else
    941 			return EINVAL;
    942 		break;
    943 	case AUDIO_ENCODING_SLINEAR_BE:
    944 		if (p->precision == 16)
    945 			bits = FMT_TWOS_COMP_BE >> 5;
    946 		else
    947 			return EINVAL;
    948 		break;
    949 	case AUDIO_ENCODING_ULINEAR_LE:
    950 		if (p->precision == 8)
    951 			bits = FMT_PCM8 >> 5;
    952 		else
    953 			return EINVAL;
    954 		break;
    955 	default:
    956 		return EINVAL;
    957 	}
    958 
    959 	if (p->channels < 1 || p->channels > 2)
    960 		return EINVAL;
    961 
    962 	error = ad1848_set_speed(sc, &p->sample_rate);
    963 	if (error)
    964 		return error;
    965 	phw.sample_rate = p->sample_rate;
    966 
    967 	if (pswcode != NULL)
    968 		pfil->append(pfil, pswcode, &phw);
    969 	if (rswcode != NULL)
    970 		rfil->append(rfil, rswcode, &rhw);
    971 
    972 	sc->format_bits = bits;
    973 	sc->channels = p->channels;
    974 	sc->precision = p->precision;
    975 	sc->need_commit = 1;
    976 
    977 	DPRINTF(("ad1848_set_params succeeded, bits=%x\n", bits));
    978 	return 0;
    979 }
    980 
    981 int
    982 ad1848_set_rec_port(struct ad1848_softc *sc, int port)
    983 {
    984 	u_char inp, reg;
    985 
    986 	DPRINTF(("ad1848_set_rec_port: 0x%x\n", port));
    987 
    988 	if (port == MIC_IN_PORT)
    989 		inp = MIC_INPUT;
    990 	else if (port == LINE_IN_PORT)
    991 		inp = LINE_INPUT;
    992 	else if (port == DAC_IN_PORT)
    993 		inp = MIXED_DAC_INPUT;
    994 	else if (sc->mode >= 2 && port == AUX1_IN_PORT)
    995 		inp = AUX_INPUT;
    996 	else
    997 		return EINVAL;
    998 
    999 	reg = ad_read(sc, SP_LEFT_INPUT_CONTROL);
   1000 	reg &= INPUT_SOURCE_MASK;
   1001 	ad_write(sc, SP_LEFT_INPUT_CONTROL, (inp|reg));
   1002 
   1003 	reg = ad_read(sc, SP_RIGHT_INPUT_CONTROL);
   1004 	reg &= INPUT_SOURCE_MASK;
   1005 	ad_write(sc, SP_RIGHT_INPUT_CONTROL, (inp|reg));
   1006 
   1007 	sc->rec_port = port;
   1008 
   1009 	return 0;
   1010 }
   1011 
   1012 int
   1013 ad1848_get_rec_port(struct ad1848_softc *sc)
   1014 {
   1015 	return sc->rec_port;
   1016 }
   1017 
   1018 int
   1019 ad1848_round_blocksize(void *addr, int blk,
   1020     int mode, const audio_params_t *param)
   1021 {
   1022 
   1023 	/* Round to a multiple of the biggest sample size. */
   1024 	return blk &= -4;
   1025 }
   1026 
   1027 int
   1028 ad1848_open(void *addr, int flags)
   1029 {
   1030 	struct ad1848_softc *sc;
   1031 	u_char reg;
   1032 
   1033 	sc = addr;
   1034 	DPRINTF(("ad1848_open: sc=%p\n", sc));
   1035 
   1036 	sc->open_mode = flags;
   1037 
   1038 	/* Enable interrupts */
   1039 	DPRINTF(("ad1848_open: enable intrs\n"));
   1040 	reg = ad_read(sc, SP_PIN_CONTROL);
   1041 	ad_write(sc, SP_PIN_CONTROL, reg | INTERRUPT_ENABLE);
   1042 
   1043 	/* If recording && monitoring, the playback part is also used. */
   1044 	if (flags & FREAD && sc->mute[AD1848_MONITOR_CHANNEL] == 0)
   1045 		ad1848_mute_wave_output(sc, WAVE_UNMUTE1, 1);
   1046 
   1047 #ifdef AUDIO_DEBUG
   1048 	if (ad1848debug)
   1049 		ad1848_dump_regs(sc);
   1050 #endif
   1051 
   1052 	return 0;
   1053 }
   1054 
   1055 /*
   1056  * Close function is called at splaudio().
   1057  */
   1058 void
   1059 ad1848_close(void *addr)
   1060 {
   1061 	struct ad1848_softc *sc;
   1062 	u_char reg;
   1063 
   1064 	sc = addr;
   1065 	sc->open_mode = 0;
   1066 
   1067 	ad1848_mute_wave_output(sc, WAVE_UNMUTE1, 0);
   1068 
   1069 	/* Disable interrupts */
   1070 	DPRINTF(("ad1848_close: disable intrs\n"));
   1071 	reg = ad_read(sc, SP_PIN_CONTROL);
   1072 	ad_write(sc, SP_PIN_CONTROL, reg & ~INTERRUPT_ENABLE);
   1073 
   1074 #ifdef AUDIO_DEBUG
   1075 	if (ad1848debug)
   1076 		ad1848_dump_regs(sc);
   1077 #endif
   1078 }
   1079 
   1080 /*
   1081  * Lower-level routines
   1082  */
   1083 int
   1084 ad1848_commit_settings(void *addr)
   1085 {
   1086 	struct ad1848_softc *sc;
   1087 	int timeout;
   1088 	u_char fs;
   1089 	int s;
   1090 
   1091 	sc = addr;
   1092 	if (!sc->need_commit)
   1093 		return 0;
   1094 
   1095 	s = splaudio();
   1096 
   1097 	ad1848_mute_wave_output(sc, WAVE_MUTE0, 1);
   1098 
   1099 	ad_set_MCE(sc, 1);	/* Enables changes to the format select reg */
   1100 
   1101 	fs = sc->speed_bits | (sc->format_bits << 5);
   1102 
   1103 	if (sc->channels == 2)
   1104 		fs |= FMT_STEREO;
   1105 
   1106 	/*
   1107 	 * OPL3-SA2 (YMF711) is sometimes busy here.
   1108 	 * Wait until it becomes ready.
   1109 	 */
   1110 	for (timeout = 0;
   1111 	    timeout < 1000 && ADREAD(sc, AD1848_IADDR) & SP_IN_INIT; timeout++)
   1112 		delay(10);
   1113 
   1114 	ad_write(sc, SP_CLOCK_DATA_FORMAT, fs);
   1115 
   1116 	/*
   1117 	 * If mode >= 2 (CS4231), set I28 also.
   1118 	 * It's the capture format register.
   1119 	 */
   1120 	if (sc->mode >= 2) {
   1121 		/*
   1122 		 * Gravis Ultrasound MAX SDK sources says something about
   1123 		 * errata sheets, with the implication that these inb()s
   1124 		 * are necessary.
   1125 		 */
   1126 		(void)ADREAD(sc, AD1848_IDATA);
   1127 		(void)ADREAD(sc, AD1848_IDATA);
   1128 		/* Write to I8 starts resynchronization. Wait for completion. */
   1129 		timeout = 100000;
   1130 		while (timeout > 0 && ADREAD(sc, AD1848_IADDR) == SP_IN_INIT)
   1131 			timeout--;
   1132 
   1133 		ad_write(sc, CS_REC_FORMAT, fs);
   1134 		(void)ADREAD(sc, AD1848_IDATA);
   1135 		(void)ADREAD(sc, AD1848_IDATA);
   1136 		/* Now wait for resync for capture side of the house */
   1137 	}
   1138 	/*
   1139 	 * Write to I8 starts resynchronization. Wait until it completes.
   1140 	 */
   1141 	timeout = 100000;
   1142 	while (timeout > 0 && ADREAD(sc, AD1848_IADDR) == SP_IN_INIT) {
   1143 		delay(10);
   1144 		timeout--;
   1145 	}
   1146 
   1147 	if (ADREAD(sc, AD1848_IADDR) == SP_IN_INIT)
   1148 		printf("ad1848_commit: Auto calibration timed out\n");
   1149 
   1150 	/*
   1151 	 * Starts the calibration process and
   1152 	 * enters playback mode after it.
   1153 	 */
   1154 	ad_set_MCE(sc, 0);
   1155 	wait_for_calibration(sc);
   1156 
   1157 	ad1848_mute_wave_output(sc, WAVE_MUTE0, 0);
   1158 
   1159 	splx(s);
   1160 
   1161 	sc->need_commit = 0;
   1162 	return 0;
   1163 }
   1164 
   1165 void
   1166 ad1848_reset(struct ad1848_softc *sc)
   1167 {
   1168 	u_char r;
   1169 
   1170 	DPRINTF(("ad1848_reset\n"));
   1171 
   1172 	/* Clear the PEN and CEN bits */
   1173 	r = ad_read(sc, SP_INTERFACE_CONFIG);
   1174 	r &= ~(CAPTURE_ENABLE | PLAYBACK_ENABLE);
   1175 	ad_write(sc, SP_INTERFACE_CONFIG, r);
   1176 
   1177 	if (sc->mode >= 2) {
   1178 		ADWRITE(sc, AD1848_IADDR, CS_IRQ_STATUS);
   1179 		ADWRITE(sc, AD1848_IDATA, 0);
   1180 	}
   1181 	/* Clear interrupt status */
   1182 	ADWRITE(sc, AD1848_STATUS, 0);
   1183 #ifdef AUDIO_DEBUG
   1184 	if (ad1848debug)
   1185 		ad1848_dump_regs(sc);
   1186 #endif
   1187 }
   1188 
   1189 int
   1190 ad1848_set_speed(struct ad1848_softc *sc, u_int *argp)
   1191 {
   1192 	/*
   1193 	 * The sampling speed is encoded in the least significant nible of I8.
   1194 	 * The LSB selects the clock source (0=24.576 MHz, 1=16.9344 MHz) and
   1195 	 * other three bits select the divisor (indirectly):
   1196 	 *
   1197 	 * The available speeds are in the following table. Keep the speeds in
   1198 	 * the increasing order.
   1199 	 */
   1200 	typedef struct {
   1201 		int	speed;
   1202 		u_char	bits;
   1203 	} speed_struct;
   1204 	u_long arg;
   1205 
   1206 	static const speed_struct speed_table[] =  {
   1207 		{5510, (0 << 1) | 1},
   1208 		{5510, (0 << 1) | 1},
   1209 		{6620, (7 << 1) | 1},
   1210 		{8000, (0 << 1) | 0},
   1211 		{9600, (7 << 1) | 0},
   1212 		{11025, (1 << 1) | 1},
   1213 		{16000, (1 << 1) | 0},
   1214 		{18900, (2 << 1) | 1},
   1215 		{22050, (3 << 1) | 1},
   1216 		{27420, (2 << 1) | 0},
   1217 		{32000, (3 << 1) | 0},
   1218 		{33075, (6 << 1) | 1},
   1219 		{37800, (4 << 1) | 1},
   1220 		{44100, (5 << 1) | 1},
   1221 		{48000, (6 << 1) | 0}
   1222 	};
   1223 
   1224 	int i, n, selected;
   1225 
   1226 	arg = *argp;
   1227 	selected = -1;
   1228 	n = sizeof(speed_table) / sizeof(speed_struct);
   1229 
   1230 	if (arg < speed_table[0].speed)
   1231 		selected = 0;
   1232 	if (arg > speed_table[n - 1].speed)
   1233 		selected = n - 1;
   1234 
   1235 	for (i = 1 /*really*/ ; selected == -1 && i < n; i++)
   1236 		if (speed_table[i].speed == arg)
   1237 			selected = i;
   1238 		else if (speed_table[i].speed > arg) {
   1239 			int diff1, diff2;
   1240 
   1241 			diff1 = arg - speed_table[i - 1].speed;
   1242 			diff2 = speed_table[i].speed - arg;
   1243 
   1244 			if (diff1 < diff2)
   1245 				selected = i - 1;
   1246 			else
   1247 				selected = i;
   1248 		}
   1249 
   1250 	if (selected == -1) {
   1251 		printf("ad1848: Can't find speed???\n");
   1252 		selected = 3;
   1253 	}
   1254 
   1255 	sc->speed_bits = speed_table[selected].bits;
   1256 	sc->need_commit = 1;
   1257 	*argp = speed_table[selected].speed;
   1258 
   1259 	return 0;
   1260 }
   1261 
   1262 /*
   1263  * Halt I/O
   1264  */
   1265 int
   1266 ad1848_halt_output(void *addr)
   1267 {
   1268 	struct ad1848_softc *sc;
   1269 	u_char reg;
   1270 
   1271 	DPRINTF(("ad1848: ad1848_halt_output\n"));
   1272 	sc = addr;
   1273 	reg = ad_read(sc, SP_INTERFACE_CONFIG);
   1274 	ad_write(sc, SP_INTERFACE_CONFIG, reg & ~PLAYBACK_ENABLE);
   1275 
   1276 	return 0;
   1277 }
   1278 
   1279 int
   1280 ad1848_halt_input(void *addr)
   1281 {
   1282 	struct ad1848_softc *sc;
   1283 	u_char reg;
   1284 
   1285 	DPRINTF(("ad1848: ad1848_halt_input\n"));
   1286 	sc = addr;
   1287 	reg = ad_read(sc, SP_INTERFACE_CONFIG);
   1288 	ad_write(sc, SP_INTERFACE_CONFIG, reg & ~CAPTURE_ENABLE);
   1289 
   1290 	return 0;
   1291 }
   1292