File: [cvs.NetBSD.org] / src / sys / dev / ic / ad1848.c (download)
Revision 1.9, Mon Nov 1 18:12:19 1999 UTC (24 years, 5 months ago) by augustss
Branch: MAIN
CVS Tags: wrstuden-devbsize-base, wrstuden-devbsize-19991221, netbsd-1-5-base, netbsd-1-5-RELEASE, netbsd-1-5-PATCH002, netbsd-1-5-PATCH001, netbsd-1-5-BETA2, netbsd-1-5-BETA, netbsd-1-5-ALPHA2, minoura-xpg4dl-base, minoura-xpg4dl, fvdl-softdep-base, comdex-fall-1999-base, comdex-fall-1999, chs-ubc2-newbase
Branch point for: netbsd-1-5
Changes since 1.8: +5 -5
lines
Make little endian and big endian version of all functions
that deal with 16 bit samples. Let the drivers decide which ones
to use.
|
/* $NetBSD: ad1848.c,v 1.9 1999/11/01 18:12:19 augustss 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.
* 3. All advertising materials mentioning features or use of this software
* must display the following acknowledgement:
* This product includes software developed by the NetBSD
* Foundation, Inc. and its contributors.
* 4. Neither the name of The NetBSD Foundation nor the names of its
* contributors may be used to endorse or promote products derived
* from this software without specific prior written permission.
*
* 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/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 <machine/cpu.h>
#include <machine/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
#ifdef AUDIO_DEBUG
#define DPRINTF(x) if (ad1848debug) printf x
int ad1848debug = 0;
#else
#define DPRINTF(x)
#endif
/*
* Initial values for the indirect registers of CS4248/AD1848.
*/
static 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 */
};
__inline int ad_read __P((struct ad1848_softc *, int));
__inline void ad_write __P((struct ad1848_softc *, int, int));
static void ad_set_MCE __P((struct ad1848_softc *, int));
static void wait_for_calibration __P((struct ad1848_softc *));
int
ad1848_to_vol(cp, 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(cp, 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(sc, reg)
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(sc, reg, data)
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(sc, reg)
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(sc, reg, val)
struct ad1848_softc *sc;
int reg, 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(sc, state)
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(sc)
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(sc)
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
/*
* Attach hardware to driver, attach hardware driver to audio
* pseudo-device driver .
*/
void
ad1848_attach(sc)
struct ad1848_softc *sc;
{
int i;
static struct ad1848_volume vol_mid = {220, 220};
static struct ad1848_volume vol_0 = {0, 0};
struct audio_params pparams, rparams;
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);
pparams = audio_default;
rparams = audio_default;
ad1848_set_params(sc, AUMODE_RECORD|AUMODE_PLAY, 0, &pparams, &rparams);
/* 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) {
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
*/
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 },
{ 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(sc, device, mute)
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
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
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(sc, device, gp)
struct ad1848_softc *sc;
int device;
struct ad1848_volume *gp;
{
struct ad1848_mixerinfo *info = &mixer_channel_info[device];
u_char reg;
u_int atten;
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(sc, device, gp)
struct ad1848_softc *sc;
int device;
struct ad1848_volume *gp;
{
*gp = sc->gains[device];
return(0);
}
int
ad1848_get_rec_gain(sc, gp)
struct ad1848_softc *sc;
struct ad1848_volume *gp;
{
*gp = sc->rec_gain;
return(0);
}
int
ad1848_set_rec_gain(sc, gp)
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(sc, mute, set)
struct ad1848_softc *sc;
int mute, 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(sc, gp)
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(sc, gp)
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 ad1848_devmap_t *
ad1848_mixer_find_dev __P((ad1848_devmap_t *, int, mixer_ctrl_t *));
static ad1848_devmap_t *
ad1848_mixer_find_dev(map, cnt, cp)
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(ac, map, cnt, cp)
struct ad1848_softc *ac;
struct ad1848_devmap *map;
int cnt;
mixer_ctrl_t *cp;
{
ad1848_devmap_t *entry;
struct ad1848_volume vol;
int error = EINVAL;
int dev;
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(ac, map, cnt, cp)
struct ad1848_softc *ac;
struct ad1848_devmap *map;
int cnt;
mixer_ctrl_t *cp;
{
ad1848_devmap_t *entry;
struct ad1848_volume vol;
int error = EINVAL;
int dev;
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(addr, fp)
void *addr;
struct audio_encoding *fp;
{
struct ad1848_softc *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 ? 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)
return EINVAL;
strcpy(fp->name, AudioEadpcm);
fp->encoding = AUDIO_ENCODING_ADPCM;
fp->precision = 8;
fp->flags = 0;
break;
default:
return EINVAL;
/*NOTREACHED*/
}
return (0);
}
int
ad1848_set_params(addr, setmode, usemode, p, r)
void *addr;
int setmode, usemode;
struct audio_params *p, *r;
{
struct ad1848_softc *sc = addr;
int error, bits, enc;
void (*pswcode) __P((void *, u_char *buf, int cnt));
void (*rswcode) __P((void *, u_char *buf, int cnt));
DPRINTF(("ad1848_set_params: %d %d %d %ld\n",
p->encoding, p->precision, p->channels, p->sample_rate));
enc = p->encoding;
pswcode = rswcode = 0;
switch (enc) {
case AUDIO_ENCODING_SLINEAR_LE:
if (p->precision == 8) {
enc = AUDIO_ENCODING_ULINEAR_LE;
pswcode = rswcode = change_sign8;
}
break;
case AUDIO_ENCODING_SLINEAR_BE:
if (p->precision == 16 && sc->mode == 1) {
enc = AUDIO_ENCODING_SLINEAR_LE;
pswcode = rswcode = swap_bytes;
}
break;
case AUDIO_ENCODING_ULINEAR_LE:
if (p->precision == 16) {
enc = AUDIO_ENCODING_SLINEAR_LE;
pswcode = rswcode = change_sign16_le;
}
break;
case AUDIO_ENCODING_ULINEAR_BE:
if (p->precision == 16) {
if (sc->mode == 1) {
enc = AUDIO_ENCODING_SLINEAR_LE;
pswcode = swap_bytes_change_sign16_le;
rswcode = change_sign16_swap_bytes_le;
} else {
enc = AUDIO_ENCODING_SLINEAR_BE;
pswcode = rswcode = change_sign16_be;
}
}
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;
p->sw_code = pswcode;
r->sw_code = rswcode;
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(sc, 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(sc)
struct ad1848_softc *sc;
{
return (sc->rec_port);
}
int
ad1848_round_blocksize(addr, blk)
void *addr;
int blk;
{
/* Round to a multiple of the biggest sample size. */
return (blk &= -4);
}
int
ad1848_open(addr, flags)
void *addr;
int flags;
{
struct ad1848_softc *sc = addr;
u_char reg;
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(addr)
void *addr;
{
struct ad1848_softc *sc = addr;
u_char reg;
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(addr)
void *addr;
{
struct ad1848_softc *sc = addr;
int timeout;
u_char fs;
int s;
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;
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 resyncronization. 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 resyncronization. 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(sc)
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(sc, argp)
struct ad1848_softc *sc;
u_long *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 = *argp;
static 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 = -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(addr)
void *addr;
{
struct ad1848_softc *sc = addr;
u_char reg;
DPRINTF(("ad1848: ad1848_halt_output\n"));
reg = ad_read(sc, SP_INTERFACE_CONFIG);
ad_write(sc, SP_INTERFACE_CONFIG, reg & ~PLAYBACK_ENABLE);
return(0);
}
int
ad1848_halt_input(addr)
void *addr;
{
struct ad1848_softc *sc = addr;
u_char reg;
DPRINTF(("ad1848: ad1848_halt_input\n"));
reg = ad_read(sc, SP_INTERFACE_CONFIG);
ad_write(sc, SP_INTERFACE_CONFIG, reg & ~CAPTURE_ENABLE);
return(0);
}
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