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86ad297c12
NetBSD/sys/dev/isa/sbdsp.c
augustss 86ad297c12 Fix a braino in record source selection and a pasto in detecting 
16 bit mode recording.
1997-06-13 19:21:59 +00:00

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/* $NetBSD: sbdsp.c,v 1.58 1997/06/13 19:21:59 augustss Exp $ */
/*
* 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.
*
*/
/*
* SoundBlaster Pro code provided by John Kohl, based on lots of
* information he gleaned from Steve Haehnichen <steve@vigra.com>'s
* SBlast driver for 386BSD and DOS driver code from Daniel Sachs
* <sachs@meibm15.cen.uiuc.edu>.
* Lots of rewrites by Lennart Augustsson <augustss@cs.chalmers.se>
* with information from SB "Hardware Programming Guide" and the
* Linux drivers.
*/
#include <sys/param.h>
#include <sys/systm.h>
#include <sys/errno.h>
#include <sys/ioctl.h>
#include <sys/syslog.h>
#include <sys/device.h>
#include <sys/proc.h>
#include <sys/buf.h>
#include <vm/vm.h>
#include <machine/cpu.h>
#include <machine/intr.h>
#include <machine/bus.h>
#include <sys/audioio.h>
#include <dev/audio_if.h>
#include <dev/mulaw.h>
#include <dev/isa/isavar.h>
#include <dev/isa/isadmavar.h>
#include <dev/isa/sbreg.h>
#include <dev/isa/sbdspvar.h>
#ifdef AUDIO_DEBUG
extern void Dprintf __P((const char *, ...));
#define DPRINTF(x) if (sbdspdebug) Dprintf x
int sbdspdebug = 0;
#else
#define DPRINTF(x)
#endif
#ifndef SBDSP_NPOLL
#define SBDSP_NPOLL 3000
#endif
struct {
int wdsp;
int rdsp;
int wmidi;
} sberr;
/*
* Time constant routines follow. See SBK, section 12.
* Although they don't come out and say it (in the docs),
* the card clearly uses a 1MHz countdown timer, as the
* low-speed formula (p. 12-4) is:
* tc = 256 - 10^6 / sr
* In high-speed mode, the constant is the upper byte of a 16-bit counter,
* and a 256MHz clock is used:
* tc = 65536 - 256 * 10^ 6 / sr
* Since we can only use the upper byte of the HS TC, the two formulae
* are equivalent. (Why didn't they say so?) E.g.,
* (65536 - 256 * 10 ^ 6 / x) >> 8 = 256 - 10^6 / x
*
* The crossover point (from low- to high-speed modes) is different
* for the SBPRO and SB20. The table on p. 12-5 gives the following data:
*
* SBPRO SB20
* ----- --------
* input ls min 4 KHz 4 KHz
* input ls max 23 KHz 13 KHz
* input hs max 44.1 KHz 15 KHz
* output ls min 4 KHz 4 KHz
* output ls max 23 KHz 23 KHz
* output hs max 44.1 KHz 44.1 KHz
*/
/* XXX Should we round the tc?
#define SB_RATE_TO_TC(x) (((65536 - 256 * 1000000 / (x)) + 128) >> 8)
*/
#define SB_RATE_TO_TC(x) (256 - 1000000 / (x))
#define SB_TC_TO_RATE(tc) (1000000 / (256 - (tc)))
struct sbmode {
short model;
u_char channels;
u_char precision;
u_short lowrate, highrate;
u_char cmd;
u_char cmdchan;
};
static struct sbmode sbpmodes[] = {
{ SB_1, 1, 8, 4000, 22727, SB_DSP_WDMA },
{ SB_20, 1, 8, 4000, 22727, SB_DSP_WDMA_LOOP },
{ SB_2x, 1, 8, 4000, 22727, SB_DSP_WDMA_LOOP },
{ SB_2x, 1, 8, 22727, 45454, SB_DSP_HS_OUTPUT },
{ SB_PRO, 1, 8, 4000, 22727, SB_DSP_WDMA_LOOP },
{ SB_PRO, 1, 8, 22727, 45454, SB_DSP_HS_OUTPUT },
{ SB_PRO, 2, 8, 11025, 22727, SB_DSP_HS_OUTPUT },
/* Yes, we write the record mode to set 16-bit playback mode. weird, huh? */
{ SB_JAZZ, 1, 8, 4000, 22727, SB_DSP_WDMA_LOOP, SB_DSP_RECORD_MONO },
{ SB_JAZZ, 1, 8, 22727, 45454, SB_DSP_HS_OUTPUT, SB_DSP_RECORD_MONO },
{ SB_JAZZ, 2, 8, 11025, 22727, SB_DSP_HS_OUTPUT, SB_DSP_RECORD_STEREO },
{ SB_JAZZ, 1, 16, 4000, 22727, SB_DSP_WDMA_LOOP, JAZZ16_RECORD_MONO },
{ SB_JAZZ, 1, 16, 22727, 45454, SB_DSP_HS_OUTPUT, JAZZ16_RECORD_MONO },
{ SB_JAZZ, 2, 16, 11025, 22727, SB_DSP_HS_OUTPUT, JAZZ16_RECORD_STEREO },
{ SB_16, 1, 8, 5000, 45000, SB_DSP16_WDMA_8 },
{ SB_16, 2, 8, 5000, 45000, SB_DSP16_WDMA_8 },
{ SB_16, 1, 16, 5000, 45000, SB_DSP16_WDMA_16 },
{ SB_16, 2, 16, 5000, 45000, SB_DSP16_WDMA_16 },
{ -1 }
};
static struct sbmode sbrmodes[] = {
{ SB_1, 1, 8, 4000, 12987, SB_DSP_RDMA },
{ SB_20, 1, 8, 4000, 12987, SB_DSP_RDMA_LOOP },
{ SB_2x, 1, 8, 4000, 12987, SB_DSP_RDMA_LOOP },
{ SB_2x, 1, 8, 12987, 14925, SB_DSP_HS_INPUT },
{ SB_PRO, 1, 8, 4000, 22727, SB_DSP_RDMA_LOOP, SB_DSP_RECORD_MONO },
{ SB_PRO, 1, 8, 22727, 45454, SB_DSP_HS_INPUT, SB_DSP_RECORD_MONO },
{ SB_PRO, 2, 8, 11025, 22727, SB_DSP_HS_INPUT, SB_DSP_RECORD_STEREO },
{ SB_JAZZ, 1, 8, 4000, 22727, SB_DSP_RDMA_LOOP, SB_DSP_RECORD_MONO },
{ SB_JAZZ, 1, 8, 22727, 45454, SB_DSP_HS_INPUT, SB_DSP_RECORD_MONO },
{ SB_JAZZ, 2, 8, 11025, 22727, SB_DSP_HS_INPUT, SB_DSP_RECORD_STEREO },
{ SB_JAZZ, 1, 16, 4000, 22727, SB_DSP_RDMA_LOOP, JAZZ16_RECORD_MONO },
{ SB_JAZZ, 1, 16, 22727, 45454, SB_DSP_HS_INPUT, JAZZ16_RECORD_MONO },
{ SB_JAZZ, 2, 16, 11025, 22727, SB_DSP_HS_INPUT, JAZZ16_RECORD_STEREO },
{ SB_16, 1, 8, 5000, 45000, SB_DSP16_RDMA_8 },
{ SB_16, 2, 8, 5000, 45000, SB_DSP16_RDMA_8 },
{ SB_16, 1, 16, 5000, 45000, SB_DSP16_RDMA_16 },
{ SB_16, 2, 16, 5000, 45000, SB_DSP16_RDMA_16 },
{ -1 }
};
void sbversion __P((struct sbdsp_softc *));
void sbdsp_jazz16_probe __P((struct sbdsp_softc *));
void sbdsp_set_mixer_gain __P((struct sbdsp_softc *sc, int port));
int sbdsp16_wait __P((struct sbdsp_softc *));
void sbdsp_to __P((void *));
void sbdsp_pause __P((struct sbdsp_softc *));
int sbdsp_set_timeconst __P((struct sbdsp_softc *, int));
int sbdsp16_set_rate __P((struct sbdsp_softc *, int, int));
int sbdsp_set_in_ports __P((struct sbdsp_softc *, int));
void sbdsp_set_ifilter __P((void *, int));
int sbdsp_get_ifilter __P((void *));
#ifdef AUDIO_DEBUG
void sb_printsc __P((struct sbdsp_softc *));
void
sb_printsc(sc)
struct sbdsp_softc *sc;
{
int i;
printf("open %d dmachan %d/%d/%d iobase 0x%x irq %d\n",
(int)sc->sc_open, sc->dmachan, sc->sc_drq8, sc->sc_drq16,
sc->sc_iobase, sc->sc_irq);
printf("irate %d itc %x orate %d otc %x\n",
sc->sc_irate, sc->sc_itc,
sc->sc_orate, sc->sc_otc);
printf("outport %u inport %u spkron %u nintr %lu\n",
sc->out_port, sc->in_port, sc->spkr_state, sc->sc_interrupts);
printf("intr %p arg %p\n",
sc->sc_intr, sc->sc_arg);
printf("gain:");
for (i = 0; i < SB_NDEVS; i++)
printf(" %u,%u", sc->gain[i][SB_LEFT], sc->gain[i][SB_RIGHT]);
printf("\n");
}
#endif /* AUDIO_DEBUG */
/*
* Probe / attach routines.
*/
/*
* Probe for the soundblaster hardware.
*/
int
sbdsp_probe(sc)
struct sbdsp_softc *sc;
{
if (sbdsp_reset(sc) < 0) {
DPRINTF(("sbdsp: couldn't reset card\n"));
return 0;
}
/* if flags set, go and probe the jazz16 stuff */
if (sc->sc_dev.dv_cfdata->cf_flags & 1)
sbdsp_jazz16_probe(sc);
else
sbversion(sc);
if (sc->sc_model == SB_UNK) {
/* Unknown SB model found. */
DPRINTF(("sbdsp: unknown SB model found\n"));
return 0;
}
return 1;
}
/*
* Try add-on stuff for Jazz16.
*/
void
sbdsp_jazz16_probe(sc)
struct sbdsp_softc *sc;
{
static u_char jazz16_irq_conf[16] = {
-1, -1, 0x02, 0x03,
-1, 0x01, -1, 0x04,
-1, 0x02, 0x05, -1,
-1, -1, -1, 0x06};
static u_char jazz16_drq_conf[8] = {
-1, 0x01, -1, 0x02,
-1, 0x03, -1, 0x04};
bus_space_tag_t iot = sc->sc_iot;
bus_space_handle_t ioh;
sbversion(sc);
DPRINTF(("jazz16 probe\n"));
if (bus_space_map(iot, JAZZ16_CONFIG_PORT, 1, 0, &ioh)) {
DPRINTF(("bus map failed\n"));
return;
}
if (jazz16_drq_conf[sc->sc_drq8] == (u_char)-1 ||
jazz16_irq_conf[sc->sc_irq] == (u_char)-1) {
DPRINTF(("drq/irq check failed\n"));
goto done; /* give up, we can't do it. */
}
bus_space_write_1(iot, ioh, 0, JAZZ16_WAKEUP);
delay(10000); /* delay 10 ms */
bus_space_write_1(iot, ioh, 0, JAZZ16_SETBASE);
bus_space_write_1(iot, ioh, 0, sc->sc_iobase & 0x70);
if (sbdsp_reset(sc) < 0) {
DPRINTF(("sbdsp_reset check failed\n"));
goto done; /* XXX? what else could we do? */
}
if (sbdsp_wdsp(sc, JAZZ16_READ_VER)) {
DPRINTF(("read16 setup failed\n"));
goto done;
}
if (sbdsp_rdsp(sc) != JAZZ16_VER_JAZZ) {
DPRINTF(("read16 failed\n"));
goto done;
}
/* XXX set both 8 & 16-bit drq to same channel, it works fine. */
sc->sc_drq16 = sc->sc_drq8;
if (sbdsp_wdsp(sc, JAZZ16_SET_DMAINTR) ||
sbdsp_wdsp(sc, (jazz16_drq_conf[sc->sc_drq16] << 4) |
jazz16_drq_conf[sc->sc_drq8]) ||
sbdsp_wdsp(sc, jazz16_irq_conf[sc->sc_irq])) {
DPRINTF(("sbdsp: can't write jazz16 probe stuff\n"));
} else {
DPRINTF(("jazz16 detected!\n"));
sc->sc_model = SB_JAZZ;
sc->sc_mixer_model = SBM_CT1345; /* XXX really? */
}
done:
bus_space_unmap(iot, ioh, 1);
}
/*
* Attach hardware to driver, attach hardware driver to audio
* pseudo-device driver .
*/
void
sbdsp_attach(sc)
struct sbdsp_softc *sc;
{
struct audio_params xparams;
int i;
u_int v;
/*
* Create our DMA maps.
*/
if (sc->sc_drq8 != -1) {
if (isa_dmamap_create(sc->sc_isa, sc->sc_drq8,
MAXPHYS /* XXX */, BUS_DMA_NOWAIT|BUS_DMA_ALLOCNOW)) {
printf("%s: can't create map for drq %d\n",
sc->sc_dev.dv_xname, sc->sc_drq8);
return;
}
}
if (sc->sc_drq16 != -1 && sc->sc_drq16 != sc->sc_drq8) {
if (isa_dmamap_create(sc->sc_isa, sc->sc_drq16,
MAXPHYS /* XXX */, BUS_DMA_NOWAIT|BUS_DMA_ALLOCNOW)) {
printf("%s: can't create map for drq %d\n",
sc->sc_dev.dv_xname, sc->sc_drq16);
return;
}
}
sbdsp_set_params(sc, AUMODE_RECORD, &audio_default, &xparams);
sbdsp_set_params(sc, AUMODE_PLAY, &audio_default, &xparams);
sbdsp_set_in_port(sc, SB_MIC_VOL);
sbdsp_set_out_port(sc, SB_MASTER_VOL);
if (sc->sc_mixer_model != SBM_NONE) {
/* Reset the mixer.*/
sbdsp_mix_write(sc, SBP_MIX_RESET, SBP_MIX_RESET);
/* And set our own default values */
for (i = 0; i < SB_NDEVS; i++) {
switch(i) {
case SB_MIC_VOL:
case SB_LINE_IN_VOL:
v = 0;
break;
case SB_BASS:
case SB_TREBLE:
v = SB_ADJUST_GAIN(sc, AUDIO_MAX_GAIN/2);
break;
default:
v = SB_ADJUST_GAIN(sc, AUDIO_MAX_GAIN * 3 / 4);
break;
}
sc->gain[i][SB_LEFT] = sc->gain[i][SB_RIGHT] = v;
sbdsp_set_mixer_gain(sc, i);
}
sc->in_filter = 0; /* no filters turned on, please */
}
printf(": dsp v%d.%02d%s\n",
SBVER_MAJOR(sc->sc_version), SBVER_MINOR(sc->sc_version),
sc->sc_model == SB_JAZZ ? ": <Jazz16>" : "");
}
void
sbdsp_mix_write(sc, mixerport, val)
struct sbdsp_softc *sc;
int mixerport;
int val;
{
bus_space_tag_t iot = sc->sc_iot;
bus_space_handle_t ioh = sc->sc_ioh;
int s;
s = splaudio();
bus_space_write_1(iot, ioh, SBP_MIXER_ADDR, mixerport);
delay(20);
bus_space_write_1(iot, ioh, SBP_MIXER_DATA, val);
delay(30);
splx(s);
}
int
sbdsp_mix_read(sc, mixerport)
struct sbdsp_softc *sc;
int mixerport;
{
bus_space_tag_t iot = sc->sc_iot;
bus_space_handle_t ioh = sc->sc_ioh;
int val;
int s;
s = splaudio();
bus_space_write_1(iot, ioh, SBP_MIXER_ADDR, mixerport);
delay(20);
val = bus_space_read_1(iot, ioh, SBP_MIXER_DATA);
delay(30);
splx(s);
return val;
}
/*
* Various routines to interface to higher level audio driver
*/
int
sbdsp_query_encoding(addr, fp)
void *addr;
struct audio_encoding *fp;
{
struct sbdsp_softc *sc = addr;
int emul;
emul = ISSB16CLASS(sc) ? 0 : AUDIO_ENCODINGFLAG_EMULATED;
switch (fp->index) {
case 0:
strcpy(fp->name, AudioEulinear);
fp->encoding = AUDIO_ENCODING_ULINEAR;
fp->precision = 8;
fp->flags = 0;
return 0;
case 1:
strcpy(fp->name, AudioEmulaw);
fp->encoding = AUDIO_ENCODING_ULAW;
fp->precision = 8;
fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
return 0;
case 2:
strcpy(fp->name, AudioEalaw);
fp->encoding = AUDIO_ENCODING_ALAW;
fp->precision = 8;
fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
return 0;
case 3:
strcpy(fp->name, AudioElinear);
fp->encoding = AUDIO_ENCODING_LINEAR;
fp->precision = 8;
fp->flags = emul;
return 0;
}
if (!ISSB16CLASS(sc) && sc->sc_model != SB_JAZZ)
return EINVAL;
switch(fp->index) {
case 4:
strcpy(fp->name, AudioElinear_le);
fp->encoding = AUDIO_ENCODING_LINEAR_LE;
fp->precision = 16;
fp->flags = 0;
return 0;
case 5:
strcpy(fp->name, AudioEulinear_le);
fp->encoding = AUDIO_ENCODING_ULINEAR_LE;
fp->precision = 16;
fp->flags = emul;
return 0;
case 6:
strcpy(fp->name, AudioElinear_be);
fp->encoding = AUDIO_ENCODING_LINEAR_BE;
fp->precision = 16;
fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
return 0;
case 7:
strcpy(fp->name, AudioEulinear_be);
fp->encoding = AUDIO_ENCODING_ULINEAR_BE;
fp->precision = 16;
fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
return 0;
default:
return EINVAL;
}
return 0;
}
int
sbdsp_set_params(addr, mode, p, q)
void *addr;
int mode;
struct audio_params *p, *q;
{
struct sbdsp_softc *sc = addr;
struct sbmode *m;
u_int rate, tc = 1, bmode = -1;
void (*swcode) __P((void *, u_char *buf, int cnt));
for(m = mode == AUMODE_PLAY ? sbpmodes : sbrmodes;
m->model != -1; m++) {
if (sc->sc_model == m->model &&
p->channels == m->channels &&
p->precision == m->precision &&
p->sample_rate >= m->lowrate &&
p->sample_rate < m->highrate)
break;
}
if (m->model == -1)
return EINVAL;
rate = p->sample_rate;
swcode = 0;
if (m->model == SB_16) {
switch (p->encoding) {
case AUDIO_ENCODING_LINEAR_BE:
if (p->precision == 16)
swcode = swap_bytes;
/* fall into */
case AUDIO_ENCODING_LINEAR_LE:
bmode = 0x10;
break;
case AUDIO_ENCODING_ULINEAR_BE:
if (p->precision == 16)
swcode = swap_bytes;
/* fall into */
case AUDIO_ENCODING_ULINEAR_LE:
bmode = 0;
break;
case AUDIO_ENCODING_ULAW:
swcode = mode == AUMODE_PLAY ?
mulaw_to_ulinear8 : ulinear8_to_mulaw;
bmode = 0;
break;
case AUDIO_ENCODING_ALAW:
swcode = mode == AUMODE_PLAY ?
alaw_to_ulinear8 : ulinear8_to_alaw;
bmode = 0;
break;
default:
return EINVAL;
}
if (p->channels == 2)
bmode |= 0x20;
} else if (m->model == SB_JAZZ && m->precision == 16) {
switch (p->encoding) {
case AUDIO_ENCODING_LINEAR_LE:
break;
case AUDIO_ENCODING_ULINEAR_LE:
swcode = change_sign16;
break;
case AUDIO_ENCODING_LINEAR_BE:
swcode = swap_bytes;
break;
case AUDIO_ENCODING_ULINEAR_BE:
swcode = mode == AUMODE_PLAY ?
swap_bytes_change_sign16 : change_sign16_swap_bytes;
break;
case AUDIO_ENCODING_ULAW:
swcode = mode == AUMODE_PLAY ?
mulaw_to_ulinear8 : ulinear8_to_mulaw;
break;
case AUDIO_ENCODING_ALAW:
swcode = mode == AUMODE_PLAY ?
alaw_to_ulinear8 : ulinear8_to_alaw;
break;
default:
return EINVAL;
}
tc = SB_RATE_TO_TC(p->sample_rate * p->channels);
p->sample_rate = SB_TC_TO_RATE(tc) / p->channels;
} else {
switch (p->encoding) {
case AUDIO_ENCODING_LINEAR_BE:
case AUDIO_ENCODING_LINEAR_LE:
swcode = change_sign8;
break;
case AUDIO_ENCODING_ULINEAR_BE:
case AUDIO_ENCODING_ULINEAR_LE:
break;
case AUDIO_ENCODING_ULAW:
swcode = mode == AUMODE_PLAY ?
mulaw_to_ulinear8 : ulinear8_to_mulaw;
break;
case AUDIO_ENCODING_ALAW:
swcode = mode == AUMODE_PLAY ?
alaw_to_ulinear8 : ulinear8_to_alaw;
break;
default:
return EINVAL;
}
tc = SB_RATE_TO_TC(p->sample_rate * p->channels);
p->sample_rate = SB_TC_TO_RATE(tc) / p->channels;
}
if (mode == AUMODE_PLAY) {
sc->sc_orate = rate;
sc->sc_otc = tc;
sc->sc_omodep = m;
sc->sc_obmode = bmode;
} else {
sc->sc_irate = rate;
sc->sc_itc = tc;
sc->sc_imodep = m;
sc->sc_ibmode = bmode;
}
p->sw_code = swcode;
/* Update setting for the other mode. */
q->encoding = p->encoding;
q->channels = p->channels;
q->precision = p->precision;
/*
* XXX
* Should wait for chip to be idle.
*/
sc->sc_dmadir = SB_DMA_NONE;
DPRINTF(("set_params: model=%d, rate=%ld, prec=%d, chan=%d, enc=%d -> tc=%02x, cmd=%02x, bmode=%02x, cmdchan=%02x, swcode=%p\n",
sc->sc_model, p->sample_rate, p->precision, p->channels,
p->encoding, tc, m->cmd, bmode, m->cmdchan, swcode));
return 0;
}
void
sbdsp_set_ifilter(addr, which)
void *addr;
int which;
{
struct sbdsp_softc *sc = addr;
int mixval;
mixval = sbdsp_mix_read(sc, SBP_INFILTER) & ~SBP_IFILTER_MASK;
switch (which) {
case 0:
mixval |= SBP_FILTER_OFF;
break;
case SB_TREBLE:
mixval |= SBP_FILTER_ON | SBP_IFILTER_HIGH;
break;
case SB_BASS:
mixval |= SBP_FILTER_ON | SBP_IFILTER_LOW;
break;
default:
return;
}
sc->in_filter = mixval & SBP_IFILTER_MASK;
sbdsp_mix_write(sc, SBP_INFILTER, mixval);
}
int
sbdsp_get_ifilter(addr)
void *addr;
{
struct sbdsp_softc *sc = addr;
sc->in_filter =
sbdsp_mix_read(sc, SBP_INFILTER) & SBP_IFILTER_MASK;
switch (sc->in_filter) {
case SBP_FILTER_ON|SBP_IFILTER_HIGH:
return SB_TREBLE;
case SBP_FILTER_ON|SBP_IFILTER_LOW:
return SB_BASS;
default:
return 0;
}
}
int
sbdsp_set_out_port(addr, port)
void *addr;
int port;
{
struct sbdsp_softc *sc = addr;
sc->out_port = port; /* Just record it */
return 0;
}
int
sbdsp_get_out_port(addr)
void *addr;
{
struct sbdsp_softc *sc = addr;
return sc->out_port;
}
int
sbdsp_set_in_port(addr, port)
void *addr;
int port;
{
return sbdsp_set_in_ports(addr, 1 << port);
}
int
sbdsp_set_in_ports(sc, mask)
struct sbdsp_softc *sc;
int mask;
{
int bitsl, bitsr;
int sbport;
int i;
DPRINTF(("sbdsp_set_in_ports: model=%d, mask=%x\n",
sc->sc_mixer_model, mask));
switch(sc->sc_mixer_model) {
case SBM_NONE:
return EINVAL;
case SBM_CT1335:
if (mask != (1 << SB_MIC_VOL))
return EINVAL;
break;
case SBM_CT1345:
switch (mask) {
case 1 << SB_MIC_VOL:
sbport = SBP_FROM_MIC;
break;
case 1 << SB_LINE_IN_VOL:
sbport = SBP_FROM_LINE;
break;
case 1 << SB_CD_VOL:
sbport = SBP_FROM_CD;
break;
default:
return EINVAL;
}
sbdsp_mix_write(sc, SBP_RECORD_SOURCE,
SBP_RECORD_FROM(sbport, SBP_FILTER_OFF, SBP_IFILTER_HIGH));
break;
case SBM_CT1745:
if (mask & ~((1<<SB_MIDI_VOL) | (1<<SB_LINE_IN_VOL) |
(1<<SB_CD_VOL) | (1<<SB_MIC_VOL)))
return EINVAL;
bitsr = 0;
if (mask & (1<<SB_MIDI_VOL)) bitsr |= SBP_MIDI_SRC_R;
if (mask & (1<<SB_LINE_IN_VOL)) bitsr |= SBP_LINE_SRC_R;
if (mask & (1<<SB_CD_VOL)) bitsr |= SBP_CD_SRC_R;
bitsl = SB_SRC_R_TO_L(bitsr);
if (mask & (1<<SB_MIC_VOL)) {
bitsl |= SBP_MIC_SRC;
bitsr |= SBP_MIC_SRC;
}
sbdsp_mix_write(sc, SBP_RECORD_SOURCE_L, bitsl);
sbdsp_mix_write(sc, SBP_RECORD_SOURCE_R, bitsr);
break;
}
sc->in_mask = mask;
/* XXX
* We have to fake a single port since the upper layer
* expects one.
*/
for(i = 0; i < SB_NPORT; i++) {
if (mask & (1 << i)) {
sc->in_port = i;
break;
}
}
return 0;
}
int
sbdsp_get_in_port(addr)
void *addr;
{
struct sbdsp_softc *sc = addr;
return sc->in_port;
}
int
sbdsp_speaker_ctl(addr, newstate)
void *addr;
int newstate;
{
struct sbdsp_softc *sc = addr;
if ((newstate == SPKR_ON) &&
(sc->spkr_state == SPKR_OFF)) {
sbdsp_spkron(sc);
sc->spkr_state = SPKR_ON;
}
if ((newstate == SPKR_OFF) &&
(sc->spkr_state == SPKR_ON)) {
sbdsp_spkroff(sc);
sc->spkr_state = SPKR_OFF;
}
return 0;
}
int
sbdsp_round_blocksize(addr, blk)
void *addr;
int blk;
{
if (blk > NBPG/3)
blk = NBPG/3; /* XXX allow at least 3 blocks */
/* Round to a multiple of the biggest sample size. */
blk &= -4;
return blk;
}
int
sbdsp_commit_settings(addr)
void *addr;
{
return 0;
}
int
sbdsp_open(sc, dev, flags)
struct sbdsp_softc *sc;
dev_t dev;
int flags;
{
DPRINTF(("sbdsp_open: sc=%p\n", sc));
if (sc->sc_open != 0 || sbdsp_reset(sc) != 0)
return ENXIO;
sc->sc_open = 1;
sc->sc_mintr = 0;
if (ISSBPRO(sc) &&
sbdsp_wdsp(sc, SB_DSP_RECORD_MONO) < 0) {
DPRINTF(("sbdsp_open: can't set mono mode\n"));
/* we'll readjust when it's time for DMA. */
}
/*
* Leave most things as they were; users must change things if
* the previous process didn't leave it they way they wanted.
* Looked at another way, it's easy to set up a configuration
* in one program and leave it for another to inherit.
*/
DPRINTF(("sbdsp_open: opened\n"));
return 0;
}
void
sbdsp_close(addr)
void *addr;
{
struct sbdsp_softc *sc = addr;
DPRINTF(("sbdsp_close: sc=%p\n", sc));
sc->sc_open = 0;
sbdsp_spkroff(sc);
sc->spkr_state = SPKR_OFF;
sc->sc_intr = 0;
sc->sc_mintr = 0;
sbdsp_haltdma(sc);
DPRINTF(("sbdsp_close: closed\n"));
}
/*
* Lower-level routines
*/
/*
* Reset the card.
* Return non-zero if the card isn't detected.
*/
int
sbdsp_reset(sc)
struct sbdsp_softc *sc;
{
bus_space_tag_t iot = sc->sc_iot;
bus_space_handle_t ioh = sc->sc_ioh;
sc->sc_intr = 0;
if (sc->sc_dmadir != SB_DMA_NONE) {
isa_dmaabort(sc->sc_isa, sc->dmachan);
sc->sc_dmadir = SB_DMA_NONE;
}
/*
* See SBK, section 11.3.
* We pulse a reset signal into the card.
* Gee, what a brilliant hardware design.
*/
bus_space_write_1(iot, ioh, SBP_DSP_RESET, 1);
delay(10);
bus_space_write_1(iot, ioh, SBP_DSP_RESET, 0);
delay(30);
if (sbdsp_rdsp(sc) != SB_MAGIC)
return -1;
return 0;
}
int
sbdsp16_wait(sc)
struct sbdsp_softc *sc;
{
bus_space_tag_t iot = sc->sc_iot;
bus_space_handle_t ioh = sc->sc_ioh;
int i;
u_char x;
for (i = SBDSP_NPOLL; --i >= 0; ) {
x = bus_space_read_1(iot, ioh, SBP_DSP_WSTAT);
delay(10);
if ((x & SB_DSP_BUSY) == 0)
continue;
return 0;
}
++sberr.wdsp;
return -1;
}
/*
* Write a byte to the dsp.
* XXX We are at the mercy of the card as we use a
* polling loop and wait until it can take the byte.
*/
int
sbdsp_wdsp(sc, v)
struct sbdsp_softc *sc;
int v;
{
bus_space_tag_t iot = sc->sc_iot;
bus_space_handle_t ioh = sc->sc_ioh;
int i;
u_char x;
for (i = SBDSP_NPOLL; --i >= 0; ) {
x = bus_space_read_1(iot, ioh, SBP_DSP_WSTAT);
delay(10);
if ((x & SB_DSP_BUSY) != 0)
continue;
bus_space_write_1(iot, ioh, SBP_DSP_WRITE, v);
delay(10);
return 0;
}
++sberr.wdsp;
return -1;
}
/*
* Read a byte from the DSP, using polling.
*/
int
sbdsp_rdsp(sc)
struct sbdsp_softc *sc;
{
bus_space_tag_t iot = sc->sc_iot;
bus_space_handle_t ioh = sc->sc_ioh;
int i;
u_char x;
for (i = SBDSP_NPOLL; --i >= 0; ) {
x = bus_space_read_1(iot, ioh, SBP_DSP_RSTAT);
delay(10);
if ((x & SB_DSP_READY) == 0)
continue;
x = bus_space_read_1(iot, ioh, SBP_DSP_READ);
delay(10);
return x;
}
++sberr.rdsp;
return -1;
}
/*
* Doing certain things (like toggling the speaker) make
* the SB hardware go away for a while, so pause a little.
*/
void
sbdsp_to(arg)
void *arg;
{
wakeup(arg);
}
void
sbdsp_pause(sc)
struct sbdsp_softc *sc;
{
extern int hz;
timeout(sbdsp_to, sbdsp_to, hz/8);
(void)tsleep(sbdsp_to, PWAIT, "sbpause", 0);
}
/*
* Turn on the speaker. The SBK documention says this operation
* can take up to 1/10 of a second. Higher level layers should
* probably let the task sleep for this amount of time after
* calling here. Otherwise, things might not work (because
* sbdsp_wdsp() and sbdsp_rdsp() will probably timeout.)
*
* These engineers had their heads up their ass when
* they designed this card.
*/
void
sbdsp_spkron(sc)
struct sbdsp_softc *sc;
{
(void)sbdsp_wdsp(sc, SB_DSP_SPKR_ON);
sbdsp_pause(sc);
}
/*
* Turn off the speaker; see comment above.
*/
void
sbdsp_spkroff(sc)
struct sbdsp_softc *sc;
{
(void)sbdsp_wdsp(sc, SB_DSP_SPKR_OFF);
sbdsp_pause(sc);
}
/*
* Read the version number out of the card. Return major code
* in high byte, and minor code in low byte.
*/
void
sbversion(sc)
struct sbdsp_softc *sc;
{
int v;
sc->sc_model = SB_UNK;
sc->sc_version = 0;
if (sbdsp_wdsp(sc, SB_DSP_VERSION) < 0)
return;
v = sbdsp_rdsp(sc) << 8;
v |= sbdsp_rdsp(sc);
if (v < 0)
return;
sc->sc_version = v;
switch(SBVER_MAJOR(v)) {
case 1:
sc->sc_mixer_model = SBM_NONE;
sc->sc_model = SB_1;
break;
case 2:
/* Some SB2 have a mixer, some don't. */
sbdsp_mix_write(sc, SBP_1335_MASTER_VOL, 0x04);
sbdsp_mix_write(sc, SBP_1335_MIDI_VOL, 0x06);
/* Check if we can read back the mixer values. */
if ((sbdsp_mix_read(sc, SBP_1335_MASTER_VOL) & 0x0e) == 0x04 &&
(sbdsp_mix_read(sc, SBP_1335_MIDI_VOL) & 0x0e) == 0x06)
sc->sc_mixer_model = SBM_CT1335;
else
sc->sc_mixer_model = SBM_NONE;
if (SBVER_MINOR(v) == 0)
sc->sc_model = SB_20;
else
sc->sc_model = SB_2x;
break;
case 3:
sc->sc_mixer_model = SBM_CT1345;
sc->sc_model = SB_PRO;
break;
case 4:
sc->sc_mixer_model = SBM_CT1745;
sc->sc_model = SB_16;
break;
}
}
/*
* Halt a DMA in progress. A low-speed transfer can be
* resumed with sbdsp_contdma().
*/
int
sbdsp_haltdma(addr)
void *addr;
{
struct sbdsp_softc *sc = addr;
DPRINTF(("sbdsp_haltdma: sc=%p\n", sc));
sbdsp_reset(sc);
return 0;
}
int
sbdsp_contdma(addr)
void *addr;
{
struct sbdsp_softc *sc = addr;
DPRINTF(("sbdsp_contdma: sc=%p\n", sc));
/* XXX how do we reinitialize the DMA controller state? do we care? */
(void)sbdsp_wdsp(sc, SB_DSP_CONT);
return 0;
}
int
sbdsp_set_timeconst(sc, tc)
struct sbdsp_softc *sc;
int tc;
{
DPRINTF(("sbdsp_set_timeconst: sc=%p tc=%d\n", sc, tc));
if (sbdsp_wdsp(sc, SB_DSP_TIMECONST) < 0 ||
sbdsp_wdsp(sc, tc) < 0)
return EIO;
return 0;
}
int
sbdsp16_set_rate(sc, cmd, rate)
struct sbdsp_softc *sc;
int cmd, rate;
{
DPRINTF(("sbdsp16_set_rate: sc=%p rate=%d\n", sc, rate));
if (sbdsp_wdsp(sc, cmd) < 0 ||
sbdsp_wdsp(sc, rate >> 8) < 0 ||
sbdsp_wdsp(sc, rate) < 0)
return EIO;
return 0;
}
int
sbdsp_dma_input(addr, p, cc, intr, arg)
void *addr;
void *p;
int cc;
void (*intr) __P((void *));
void *arg;
{
struct sbdsp_softc *sc = addr;
int loop = sc->sc_model != SB_1;
int stereo = sc->sc_imodep->channels == 2;
int filter;
#ifdef AUDIO_DEBUG
if (sbdspdebug > 1)
Dprintf("sbdsp_dma_input: cc=%d 0x%x (0x%x)\n", cc, intr, arg);
#endif
#ifdef DIAGNOSTIC
if (sc->sc_imodep->channels == 2 && (cc & 1)) {
DPRINTF(("stereo record odd bytes (%d)\n", cc));
return EIO;
}
#endif
if (sc->sc_dmadir != SB_DMA_IN) {
if (ISSBPRO(sc)) {
if (sbdsp_wdsp(sc, sc->sc_imodep->cmdchan) < 0)
goto badmode;
filter = stereo ? SBP_FILTER_OFF : sc->in_filter;
sbdsp_mix_write(sc, SBP_INFILTER,
(sbdsp_mix_read(sc, SBP_INFILTER) &
~SBP_IFILTER_MASK) | filter);
}
if (ISSB16CLASS(sc)) {
if (sbdsp16_set_rate(sc, SB_DSP16_INPUTRATE,
sc->sc_irate))
goto giveup;
} else {
if (sbdsp_set_timeconst(sc, sc->sc_itc))
goto giveup;
}
sc->sc_dmadir = SB_DMA_IN;
sc->dmaflags = DMAMODE_READ;
if (loop)
sc->dmaflags |= DMAMODE_LOOP;
} else {
/* If already started; just return. */
if (loop)
return 0;
}
sc->dmaaddr = p;
sc->dmacnt = loop ? (NBPG/cc)*cc : cc;
sc->dmachan = sc->sc_imodep->precision == 16 ? sc->sc_drq16 : sc->sc_drq8;
#ifdef AUDIO_DEBUG
if (sbdspdebug > 1)
Dprintf("sbdsp_dma_input: dmastart %x %p %d %d\n",
sc->dmaflags, sc->dmaaddr, sc->dmacnt, sc->dmachan);
#endif
isa_dmastart(sc->sc_isa, sc->dmachan, sc->dmaaddr,
sc->dmacnt, NULL, sc->dmaflags, BUS_DMA_NOWAIT);
sc->sc_intr = intr;
sc->sc_arg = arg;
if ((sc->sc_model == SB_JAZZ && sc->dmachan > 3) ||
(sc->sc_model != SB_JAZZ && sc->sc_imodep->precision == 16))
cc >>= 1;
--cc;
if (ISSB16CLASS(sc)) {
if (sbdsp_wdsp(sc, sc->sc_imodep->cmd) < 0 ||
sbdsp_wdsp(sc, sc->sc_ibmode) < 0 ||
sbdsp16_wait(sc) ||
sbdsp_wdsp(sc, cc) < 0 ||
sbdsp_wdsp(sc, cc >> 8) < 0) {
DPRINTF(("sbdsp_dma_input: SB16 DMA start failed\n"));
goto giveup;
}
} else {
if (loop) {
DPRINTF(("sbdsp_dma_input: set blocksize=%d\n", cc));
if (sbdsp_wdsp(sc, SB_DSP_BLOCKSIZE) < 0 ||
sbdsp_wdsp(sc, cc) < 0 ||
sbdsp_wdsp(sc, cc >> 8) < 0) {
DPRINTF(("sbdsp_dma_input: SB2 DMA start failed\n"));
goto giveup;
}
if (sbdsp_wdsp(sc, sc->sc_imodep->cmd) < 0) {
DPRINTF(("sbdsp_dma_input: SB2 DMA restart failed\n"));
goto giveup;
}
} else {
if (sbdsp_wdsp(sc, sc->sc_imodep->cmd) < 0 ||
sbdsp_wdsp(sc, cc) < 0 ||
sbdsp_wdsp(sc, cc >> 8) < 0) {
DPRINTF(("sbdsp_dma_input: SB1 DMA start failed\n"));
goto giveup;
}
}
}
return 0;
giveup:
sbdsp_reset(sc);
return EIO;
badmode:
DPRINTF(("sbdsp_dma_input: can't set mode\n"));
return EIO;
}
int
sbdsp_dma_output(addr, p, cc, intr, arg)
void *addr;
void *p;
int cc;
void (*intr) __P((void *));
void *arg;
{
struct sbdsp_softc *sc = addr;
int loop = sc->sc_model != SB_1;
int stereo = sc->sc_omodep->channels == 2;
int cmd;
#ifdef AUDIO_DEBUG
if (sbdspdebug > 1)
Dprintf("sbdsp_dma_output: cc=%d 0x%x (0x%x)\n", cc, intr, arg);
#endif
#ifdef DIAGNOSTIC
if (stereo && (cc & 1)) {
DPRINTF(("stereo playback odd bytes (%d)\n", cc));
return EIO;
}
#endif
if (sc->sc_dmadir != SB_DMA_OUT) {
if (ISSBPRO(sc)) {
/* make sure we re-set stereo mixer bit when we start
output. */
sbdsp_mix_write(sc, SBP_STEREO,
(sbdsp_mix_read(sc, SBP_STEREO) & ~SBP_PLAYMODE_MASK) |
(stereo ? SBP_PLAYMODE_STEREO : SBP_PLAYMODE_MONO));
cmd = sc->sc_omodep->cmdchan;
if (cmd && sbdsp_wdsp(sc, cmd) < 0)
goto badmode;
}
if (ISSB16CLASS(sc)) {
if (sbdsp16_set_rate(sc, SB_DSP16_OUTPUTRATE,
sc->sc_orate))
goto giveup;
} else {
if (sbdsp_set_timeconst(sc, sc->sc_otc))
goto giveup;
}
sc->sc_dmadir = SB_DMA_OUT;
sc->dmaflags = DMAMODE_WRITE;
if (loop)
sc->dmaflags |= DMAMODE_LOOP;
} else {
/* Already started; just return. */
if (loop)
return 0;
}
sc->dmaaddr = p;
sc->dmacnt = loop ? (NBPG/cc)*cc : cc;
sc->dmachan = sc->sc_omodep->precision == 16 ? sc->sc_drq16 : sc->sc_drq8;
#ifdef AUDIO_DEBUG
if (sbdspdebug > 1)
Dprintf("sbdsp_dma_output: dmastart %x %p %d %d\n",
sc->dmaflags, sc->dmaaddr, sc->dmacnt, sc->dmachan);
#endif
isa_dmastart(sc->sc_isa, sc->dmachan, sc->dmaaddr,
sc->dmacnt, NULL, sc->dmaflags, BUS_DMA_NOWAIT);
sc->sc_intr = intr;
sc->sc_arg = arg;
if ((sc->sc_model == SB_JAZZ && sc->dmachan > 3) ||
(sc->sc_model != SB_JAZZ && sc->sc_omodep->precision == 16))
cc >>= 1;
--cc;
if (ISSB16CLASS(sc)) {
if (sbdsp_wdsp(sc, sc->sc_omodep->cmd) < 0 ||
sbdsp_wdsp(sc, sc->sc_obmode) < 0 ||
sbdsp16_wait(sc) ||
sbdsp_wdsp(sc, cc) < 0 ||
sbdsp_wdsp(sc, cc >> 8) < 0) {
DPRINTF(("sbdsp_dma_output: SB16 DMA start failed\n"));
goto giveup;
}
} else {
if (loop) {
DPRINTF(("sbdsp_dma_output: set blocksize=%d\n", cc));
if (sbdsp_wdsp(sc, SB_DSP_BLOCKSIZE) < 0 ||
sbdsp_wdsp(sc, cc) < 0 ||
sbdsp_wdsp(sc, cc >> 8) < 0) {
DPRINTF(("sbdsp_dma_output: SB2 DMA blocksize failed\n"));
goto giveup;
}
if (sbdsp_wdsp(sc, sc->sc_omodep->cmd) < 0) {
DPRINTF(("sbdsp_dma_output: SB2 DMA start failed\n"));
goto giveup;
}
} else {
if (sbdsp_wdsp(sc, sc->sc_omodep->cmd) < 0 ||
sbdsp_wdsp(sc, cc) < 0 ||
sbdsp_wdsp(sc, cc >> 8) < 0) {
DPRINTF(("sbdsp_dma_output: SB1 DMA start failed\n"));
goto giveup;
}
}
}
return 0;
giveup:
sbdsp_reset(sc);
return EIO;
badmode:
DPRINTF(("sbdsp_dma_output: can't set mode\n"));
return EIO;
}
/*
* Only the DSP unit on the sound blaster generates interrupts.
* There are three cases of interrupt: reception of a midi byte
* (when mode is enabled), completion of dma transmission, or
* completion of a dma reception. The three modes are mutually
* exclusive so we know a priori which event has occurred.
*/
int
sbdsp_intr(arg)
void *arg;
{
struct sbdsp_softc *sc = arg;
int loop = sc->sc_model != SB_1;
u_char irq;
#ifdef AUDIO_DEBUG
if (sbdspdebug > 1)
Dprintf("sbdsp_intr: intr=0x%x\n", sc->sc_intr);
#endif
if (ISSB16CLASS(sc)) {
irq = sbdsp_mix_read(sc, SBP_IRQ_STATUS);
if ((irq & (SBP_IRQ_DMA8 | SBP_IRQ_DMA16)) == 0)
return 0;
} else {
if (!loop && !isa_dmafinished(sc->sc_isa, sc->dmachan))
return 0;
irq = SBP_IRQ_DMA8;
}
sc->sc_interrupts++;
delay(10); /* XXX why? */
#if 0
if (sc->sc_mintr != 0) {
x = sbdsp_rdsp(sc);
(*sc->sc_mintr)(sc->sc_arg, x);
} else
#endif
if (sc->sc_intr != 0) {
/* clear interrupt */
if (irq & SBP_IRQ_DMA8)
bus_space_read_1(sc->sc_iot, sc->sc_ioh, SBP_DSP_IRQACK8);
if (irq & SBP_IRQ_DMA16)
bus_space_read_1(sc->sc_iot, sc->sc_ioh, SBP_DSP_IRQACK16);
if (!loop)
isa_dmadone(sc->sc_isa, sc->dmachan);
(*sc->sc_intr)(sc->sc_arg);
} else {
return 0;
}
return 1;
}
#if 0
/*
* Enter midi uart mode and arrange for read interrupts
* to vector to `intr'. This puts the card in a mode
* which allows only midi I/O; the card must be reset
* to leave this mode. Unfortunately, the card does not
* use transmit interrupts, so bytes must be output
* using polling. To keep the polling overhead to a
* minimum, output should be driven off a timer.
* This is a little tricky since only 320us separate
* consecutive midi bytes.
*/
void
sbdsp_set_midi_mode(sc, intr, arg)
struct sbdsp_softc *sc;
void (*intr)();
void *arg;
{
sbdsp_wdsp(sc, SB_MIDI_UART_INTR);
sc->sc_mintr = intr;
sc->sc_intr = 0;
sc->sc_arg = arg;
}
/*
* Write a byte to the midi port, when in midi uart mode.
*/
void
sbdsp_midi_output(sc, v)
struct sbdsp_softc *sc;
int v;
{
if (sbdsp_wdsp(sc, v) < 0)
++sberr.wmidi;
}
#endif
int
sbdsp_setfd(addr, flag)
void *addr;
int flag;
{
/* Can't do full-duplex */
return ENOTTY;
}
void
sbdsp_set_mixer_gain(sc, port)
struct sbdsp_softc *sc;
int port;
{
int src, gain;
switch(sc->sc_mixer_model) {
case SBM_NONE:
return;
case SBM_CT1335:
gain = SB_1335_GAIN(sc->gain[port][SB_LEFT]);
switch(port) {
case SB_MASTER_VOL:
src = SBP_1335_MASTER_VOL;
break;
case SB_MIDI_VOL:
src = SBP_1335_MIDI_VOL;
break;
case SB_CD_VOL:
src = SBP_1335_CD_VOL;
break;
case SB_VOICE_VOL:
src = SBP_1335_VOICE_VOL;
gain = SB_1335_MASTER_GAIN(sc->gain[port][SB_LEFT]);
break;
default:
return;
}
sbdsp_mix_write(sc, src, gain);
break;
case SBM_CT1345:
gain = SB_STEREO_GAIN(sc->gain[port][SB_LEFT],
sc->gain[port][SB_RIGHT]);
switch (port) {
case SB_MIC_VOL:
src = SBP_MIC_VOL;
gain = SB_MIC_GAIN(sc->gain[port][SB_LEFT]);
break;
case SB_MASTER_VOL:
src = SBP_MASTER_VOL;
break;
case SB_LINE_IN_VOL:
src = SBP_LINE_VOL;
break;
case SB_VOICE_VOL:
src = SBP_VOICE_VOL;
break;
case SB_MIDI_VOL:
src = SBP_MIDI_VOL;
break;
case SB_CD_VOL:
src = SBP_CD_VOL;
break;
default:
return;
}
sbdsp_mix_write(sc, src, gain);
break;
case SBM_CT1745:
switch (port) {
case SB_MIC_VOL:
src = SB16P_MIC_L;
break;
case SB_MASTER_VOL:
src = SB16P_MASTER_L;
break;
case SB_LINE_IN_VOL:
src = SB16P_LINE_L;
break;
case SB_VOICE_VOL:
src = SB16P_VOICE_L;
break;
case SB_MIDI_VOL:
src = SB16P_MIDI_L;
break;
case SB_CD_VOL:
src = SB16P_CD_L;
break;
case SB_INPUT_GAIN:
src = SB16P_INPUT_GAIN_L;
break;
case SB_OUTPUT_GAIN:
src = SB16P_OUTPUT_GAIN_L;
break;
case SB_TREBLE:
src = SB16P_TREBLE_L;
break;
case SB_BASS:
src = SB16P_BASS_L;
break;
case SB_PCSPEAKER:
sbdsp_mix_write(sc, SB16P_PCSPEAKER, sc->gain[port][SB_LEFT]);
return;
default:
return;
}
sbdsp_mix_write(sc, src, sc->gain[port][SB_LEFT]);
sbdsp_mix_write(sc, SB16P_L_TO_R(src), sc->gain[port][SB_RIGHT]);
break;
}
}
int
sbdsp_mixer_set_port(addr, cp)
void *addr;
mixer_ctrl_t *cp;
{
struct sbdsp_softc *sc = addr;
int lgain, rgain;
DPRINTF(("sbdsp_mixer_set_port: port=%d num_channels=%d\n", cp->dev,
cp->un.value.num_channels));
if (sc->sc_mixer_model == SBM_NONE)
return EINVAL;
switch (cp->dev) {
case SB_TREBLE:
case SB_BASS:
if (sc->sc_mixer_model == SBM_CT1345) {
if (cp->type != AUDIO_MIXER_ENUM)
return EINVAL;
switch (cp->dev) {
case SB_TREBLE:
sbdsp_set_ifilter(addr, cp->un.ord ? SB_TREBLE : 0);
return 0;
case SB_BASS:
sbdsp_set_ifilter(addr, cp->un.ord ? SB_BASS : 0);
return 0;
}
}
case SB_PCSPEAKER:
case SB_INPUT_GAIN:
case SB_OUTPUT_GAIN:
if (sc->sc_mixer_model != SBM_CT1745)
return EINVAL;
case SB_MIC_VOL:
case SB_LINE_IN_VOL:
if (sc->sc_mixer_model == SBM_CT1335)
return EINVAL;
case SB_VOICE_VOL:
case SB_MIDI_VOL:
case SB_CD_VOL:
case SB_MASTER_VOL:
if (cp->type != AUDIO_MIXER_VALUE)
return EINVAL;
/*
* All the mixer ports are stereo except for the microphone.
* If we get a single-channel gain value passed in, then we
* duplicate it to both left and right channels.
*/
switch (cp->dev) {
case SB_MIC_VOL:
if (cp->un.value.num_channels != 1)
return EINVAL;
lgain = rgain = SB_ADJUST_MIC_GAIN(sc,
cp->un.value.level[AUDIO_MIXER_LEVEL_MONO]);
break;
case SB_PCSPEAKER:
if (cp->un.value.num_channels != 1)
return EINVAL;
/* fall into */
case SB_INPUT_GAIN:
case SB_OUTPUT_GAIN:
lgain = rgain = SB_ADJUST_2_GAIN(sc,
cp->un.value.level[AUDIO_MIXER_LEVEL_MONO]);
break;
default:
switch (cp->un.value.num_channels) {
case 1:
lgain = rgain = SB_ADJUST_GAIN(sc,
cp->un.value.level[AUDIO_MIXER_LEVEL_MONO]);
break;
case 2:
if (sc->sc_mixer_model == SBM_CT1335)
return EINVAL;
lgain = SB_ADJUST_GAIN(sc,
cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT]);
rgain = SB_ADJUST_GAIN(sc,
cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT]);
break;
default:
return EINVAL;
}
break;
}
sc->gain[cp->dev][SB_LEFT] = lgain;
sc->gain[cp->dev][SB_RIGHT] = rgain;
sbdsp_set_mixer_gain(sc, cp->dev);
break;
case SB_RECORD_SOURCE:
if (sc->sc_mixer_model == SBM_CT1745) {
if (cp->type != AUDIO_MIXER_SET)
return EINVAL;
return sbdsp_set_in_ports(sc, cp->un.mask);
} else {
if (cp->type != AUDIO_MIXER_ENUM)
return EINVAL;
return sbdsp_set_in_port(sc, cp->un.ord);
}
break;
case SB_AGC:
if (sc->sc_mixer_model != SBM_CT1745 || cp->type != AUDIO_MIXER_ENUM)
return EINVAL;
sbdsp_mix_write(sc, SB16P_AGC, cp->un.ord & 1);
break;
default:
return EINVAL;
}
return 0;
}
int
sbdsp_mixer_get_port(addr, cp)
void *addr;
mixer_ctrl_t *cp;
{
struct sbdsp_softc *sc = addr;
DPRINTF(("sbdsp_mixer_get_port: port=%d\n", cp->dev));
if (sc->sc_mixer_model == SBM_NONE)
return EINVAL;
switch (cp->dev) {
case SB_TREBLE:
case SB_BASS:
if (sc->sc_mixer_model == SBM_CT1345) {
switch (cp->dev) {
case SB_TREBLE:
cp->un.ord = sbdsp_get_ifilter(addr) == SB_TREBLE;
return 0;
case SB_BASS:
cp->un.ord = sbdsp_get_ifilter(addr) == SB_BASS;
return 0;
}
}
case SB_PCSPEAKER:
case SB_INPUT_GAIN:
case SB_OUTPUT_GAIN:
if (sc->sc_mixer_model != SBM_CT1745)
return EINVAL;
case SB_MIC_VOL:
case SB_LINE_IN_VOL:
if (sc->sc_mixer_model == SBM_CT1335)
return EINVAL;
case SB_VOICE_VOL:
case SB_MIDI_VOL:
case SB_CD_VOL:
case SB_MASTER_VOL:
switch (cp->dev) {
case SB_MIC_VOL:
case SB_PCSPEAKER:
if (cp->un.value.num_channels != 1)
return EINVAL;
/* fall into */
default:
switch (cp->un.value.num_channels) {
case 1:
cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] =
sc->gain[cp->dev][SB_LEFT];
break;
case 2:
cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT] =
sc->gain[cp->dev][SB_LEFT];
cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] =
sc->gain[cp->dev][SB_RIGHT];
break;
default:
return EINVAL;
}
break;
}
break;
case SB_RECORD_SOURCE:
if (sc->sc_mixer_model == SBM_CT1745)
cp->un.mask = sc->in_mask;
else
cp->un.ord = sc->in_port;
break;
case SB_AGC:
if (sc->sc_mixer_model != SBM_CT1745)
return EINVAL;
cp->un.ord = sbdsp_mix_read(sc, SB16P_AGC);
break;
default:
return EINVAL;
}
return 0;
}
int
sbdsp_mixer_query_devinfo(addr, dip)
void *addr;
mixer_devinfo_t *dip;
{
struct sbdsp_softc *sc = addr;
int chan, class;
DPRINTF(("sbdsp_mixer_query_devinfo: model=%d index=%d\n",
sc->sc_mixer_model, dip->index));
if (sc->sc_mixer_model == SBM_NONE)
return ENXIO;
chan = sc->sc_mixer_model == SBM_CT1335 ? 1 : 2;
class = sc->sc_mixer_model == SBM_CT1745 ? SB_INPUT_CLASS : SB_OUTPUT_CLASS;
switch (dip->index) {
case SB_MASTER_VOL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = SB_OUTPUT_CLASS;
dip->prev = dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNmaster);
dip->un.v.num_channels = chan;
strcpy(dip->un.v.units.name, AudioNvolume);
return 0;
case SB_MIDI_VOL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = class;
dip->prev = AUDIO_MIXER_LAST;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNfmsynth);
dip->un.v.num_channels = chan;
strcpy(dip->un.v.units.name, AudioNvolume);
return 0;
case SB_CD_VOL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = class;
dip->prev = AUDIO_MIXER_LAST;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNcd);
dip->un.v.num_channels = chan;
strcpy(dip->un.v.units.name, AudioNvolume);
return 0;
case SB_VOICE_VOL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = class;
dip->prev = AUDIO_MIXER_LAST;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNdac);
dip->un.v.num_channels = chan;
strcpy(dip->un.v.units.name, AudioNvolume);
return 0;
case SB_OUTPUT_CLASS:
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = SB_OUTPUT_CLASS;
dip->next = dip->prev = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioCOutputs);
return 0;
}
if (sc->sc_mixer_model == SBM_CT1335)
return ENXIO;
switch (dip->index) {
case SB_MIC_VOL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = class;
dip->prev = AUDIO_MIXER_LAST;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNmicrophone);
dip->un.v.num_channels = 1;
strcpy(dip->un.v.units.name, AudioNvolume);
return 0;
case SB_LINE_IN_VOL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = class;
dip->prev = AUDIO_MIXER_LAST;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNline);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
return 0;
case SB_RECORD_SOURCE:
dip->mixer_class = SB_RECORD_CLASS;
dip->prev = dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNsource);
if (sc->sc_mixer_model == SBM_CT1745) {
dip->type = AUDIO_MIXER_SET;
dip->un.s.num_mem = 4;
strcpy(dip->un.s.member[0].label.name, AudioNmicrophone);
dip->un.s.member[0].mask = 1 << SB_MIC_VOL;
strcpy(dip->un.s.member[1].label.name, AudioNcd);
dip->un.s.member[1].mask = 1 << SB_CD_VOL;
strcpy(dip->un.s.member[2].label.name, AudioNline);
dip->un.s.member[2].mask = 1 << SB_LINE_IN_VOL;
strcpy(dip->un.s.member[3].label.name, AudioNfmsynth);
dip->un.s.member[3].mask = 1 << SB_MIDI_VOL;
} else {
dip->type = AUDIO_MIXER_ENUM;
dip->un.e.num_mem = 3;
strcpy(dip->un.e.member[0].label.name, AudioNmicrophone);
dip->un.e.member[0].ord = SB_MIC_VOL;
strcpy(dip->un.e.member[1].label.name, AudioNcd);
dip->un.e.member[1].ord = SB_CD_VOL;
strcpy(dip->un.e.member[2].label.name, AudioNline);
dip->un.e.member[2].ord = SB_LINE_IN_VOL;
}
return 0;
case SB_BASS:
dip->prev = dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNbass);
if (sc->sc_mixer_model == SBM_CT1745) {
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = SB_EQUALIZATION_CLASS;
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNbass);
} else {
dip->type = AUDIO_MIXER_ENUM;
dip->mixer_class = SB_INPUT_CLASS;
dip->un.e.num_mem = 2;
strcpy(dip->un.e.member[0].label.name, AudioNoff);
dip->un.e.member[0].ord = 0;
strcpy(dip->un.e.member[1].label.name, AudioNon);
dip->un.e.member[1].ord = 1;
}
return 0;
case SB_TREBLE:
dip->prev = dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNtreble);
if (sc->sc_mixer_model == SBM_CT1745) {
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = SB_EQUALIZATION_CLASS;
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNtreble);
} else {
dip->type = AUDIO_MIXER_ENUM;
dip->mixer_class = SB_INPUT_CLASS;
dip->un.e.num_mem = 2;
strcpy(dip->un.e.member[0].label.name, AudioNoff);
dip->un.e.member[0].ord = 0;
strcpy(dip->un.e.member[1].label.name, AudioNon);
dip->un.e.member[1].ord = 1;
}
return 0;
case SB_RECORD_CLASS: /* record source class */
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = SB_RECORD_CLASS;
dip->next = dip->prev = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioCRecord);
return 0;
}
if (sc->sc_mixer_model == SBM_CT1345)
return ENXIO;
switch(dip->index) {
case SB_PCSPEAKER:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = SB_INPUT_CLASS;
dip->prev = dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, "pc_speaker");
dip->un.v.num_channels = 1;
strcpy(dip->un.v.units.name, AudioNvolume);
return 0;
case SB_INPUT_GAIN:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = SB_INPUT_CLASS;
dip->prev = dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNinput);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
return 0;
case SB_OUTPUT_GAIN:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = SB_OUTPUT_CLASS;
dip->prev = dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNoutput);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
return 0;
case SB_AGC:
dip->type = AUDIO_MIXER_ENUM;
dip->mixer_class = SB_INPUT_CLASS;
dip->prev = dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, "AGC");
dip->un.e.num_mem = 2;
strcpy(dip->un.e.member[0].label.name, AudioNoff);
dip->un.e.member[0].ord = 0;
strcpy(dip->un.e.member[1].label.name, AudioNon);
dip->un.e.member[1].ord = 1;
return 0;
case SB_INPUT_CLASS:
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = SB_INPUT_CLASS;
dip->next = dip->prev = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioCInputs);
return 0;
case SB_EQUALIZATION_CLASS:
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = SB_EQUALIZATION_CLASS;
dip->next = dip->prev = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioCEqualization);
return 0;
}
return ENXIO;
}
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