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NetBSD/sys/dev/isa/sbdsp.c
augustss e7029fc0da Make the audio API (almost) SunOS compatible. 
The changes is to allow some limited mixer manipulation through
the audio device (instead of the mixer device).
This rendered 4 methods in audio_hw_if unused so garbage collect these.
1997-10-19 07:41:33 +00:00

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/* $NetBSD: sbdsp.c,v 1.76 1997/10/19 07:42:39 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/auconv.h>
#include <dev/isa/isavar.h>
#include <dev/isa/isadmavar.h>
#include <dev/isa/sbreg.h>
#include <dev/isa/sbdspvar.h>
#ifdef AUDIO_DEBUG
#define DPRINTF(x) if (sbdspdebug) printf 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 },
#define PLAY16 15 /* must be the index of the next entry in the table */
{ 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));
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 *));
static int sbdsp_dma_setup_input __P((struct sbdsp_softc *sc));
static int sbdsp_dma_setup_output __P((struct sbdsp_softc *sc));
static int sbdsp_adjust __P((int, int));
#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/%d iobase 0x%x irq %d\n",
(int)sc->sc_open, sc->sc_i.run, sc->sc_o.run,
sc->sc_drq8, sc->sc_drq16,
sc->sc_iobase, sc->sc_irq);
printf("irate %d itc %x orate %d otc %x\n",
sc->sc_i.rate, sc->sc_i.tc,
sc->sc_o.rate, sc->sc_o.tc);
printf("spkron %u nintr %lu\n",
sc->spkr_state, sc->sc_interrupts);
printf("intr8 %p arg8 %p\n",
sc->sc_intr8, sc->sc_arg16);
printf("intr16 %p arg16 %p\n",
sc->sc_intr8, sc->sc_arg16);
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 pparams, rparams;
int i;
u_int v;
/*
* Create our DMA maps.
*/
if (sc->sc_drq8 != -1) {
if (isa_dmamap_create(sc->sc_isa, sc->sc_drq8,
MAX_ISADMA, 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,
MAX_ISADMA, 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;
}
}
pparams = audio_default;
rparams = audio_default;
sbdsp_set_params(sc, AUMODE_RECORD|AUMODE_PLAY, 0, &pparams, &rparams);
sbdsp_set_in_ports(sc, 1 << SB_MIC_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;
case SB_CD_IN_MUTE:
case SB_MIC_IN_MUTE:
case SB_LINE_IN_MUTE:
case SB_MIDI_IN_MUTE:
case SB_CD_SWAP:
case SB_MIC_SWAP:
case SB_LINE_SWAP:
case SB_MIDI_SWAP:
case SB_CD_OUT_MUTE:
case SB_MIC_OUT_MUTE:
case SB_LINE_OUT_MUTE:
v = 0;
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>" : "");
sc->sc_fullduplex = ISSB16CLASS(sc) &&
sc->sc_drq8 != -1 && sc->sc_drq16 != -1 &&
sc->sc_drq8 != sc->sc_drq16;
}
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, AudioEslinear);
fp->encoding = AUDIO_ENCODING_SLINEAR;
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, AudioEslinear_le);
fp->encoding = AUDIO_ENCODING_SLINEAR_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, AudioEslinear_be);
fp->encoding = AUDIO_ENCODING_SLINEAR_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, setmode, usemode, play, rec)
void *addr;
int setmode, usemode;
struct audio_params *play, *rec;
{
struct sbdsp_softc *sc = addr;
struct sbmode *m;
u_int rate, tc, bmode;
void (*swcode) __P((void *, u_char *buf, int cnt));
int factor;
int model;
int chan;
struct audio_params *p;
int mode;
model = sc->sc_model;
if (model > SB_16)
model = SB_16; /* later models work like SB16 */
/* Set first record info, then play info */
for(mode = AUMODE_RECORD; mode != -1;
mode = mode == AUMODE_RECORD ? AUMODE_PLAY : -1) {
if ((setmode & mode) == 0)
continue;
p = mode == AUMODE_PLAY ? play : rec;
/* Locate proper commands */
for(m = mode == AUMODE_PLAY ? sbpmodes : sbrmodes;
m->model != -1; m++) {
if (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;
factor = 1;
tc = 1;
bmode = -1;
if (model == SB_16) {
switch (p->encoding) {
case AUDIO_ENCODING_SLINEAR_BE:
if (p->precision == 16)
swcode = swap_bytes;
/* fall into */
case AUDIO_ENCODING_SLINEAR_LE:
bmode = SB_BMODE_SIGNED;
break;
case AUDIO_ENCODING_ULINEAR_BE:
if (p->precision == 16)
swcode = swap_bytes;
/* fall into */
case AUDIO_ENCODING_ULINEAR_LE:
bmode = SB_BMODE_UNSIGNED;
break;
case AUDIO_ENCODING_ULAW:
if (mode == AUMODE_PLAY) {
swcode = mulaw_to_ulinear16;
factor = 2;
m = &sbpmodes[PLAY16];
} else
swcode = ulinear8_to_mulaw;
bmode = SB_BMODE_UNSIGNED;
break;
case AUDIO_ENCODING_ALAW:
if (mode == AUMODE_PLAY) {
swcode = alaw_to_ulinear16;
factor = 2;
m = &sbpmodes[PLAY16];
} else
swcode = ulinear8_to_alaw;
bmode = SB_BMODE_UNSIGNED;
break;
default:
return EINVAL;
}
if (p->channels == 2)
bmode |= SB_BMODE_STEREO;
} else if (m->model == SB_JAZZ && m->precision == 16) {
switch (p->encoding) {
case AUDIO_ENCODING_SLINEAR_LE:
break;
case AUDIO_ENCODING_ULINEAR_LE:
swcode = change_sign16;
break;
case AUDIO_ENCODING_SLINEAR_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_SLINEAR_BE:
case AUDIO_ENCODING_SLINEAR_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;
}
chan = m->precision == 16 ? sc->sc_drq16 : sc->sc_drq8;
if (mode == AUMODE_PLAY) {
sc->sc_o.rate = rate;
sc->sc_o.tc = tc;
sc->sc_o.modep = m;
sc->sc_o.bmode = bmode;
sc->sc_o.dmachan = chan;
} else {
sc->sc_i.rate = rate;
sc->sc_i.tc = tc;
sc->sc_i.modep = m;
sc->sc_i.bmode = bmode;
sc->sc_i.dmachan = chan;
}
p->sw_code = swcode;
p->factor = factor;
DPRINTF(("sbdsp_set_params: model=%d, mode=%d, rate=%ld, prec=%d, chan=%d, enc=%d -> tc=%02x, cmd=%02x, bmode=%02x, cmdchan=%02x, swcode=%p, factor=%d\n",
sc->sc_model, mode, p->sample_rate, p->precision, p->channels,
p->encoding, tc, m->cmd, bmode, m->cmdchan, swcode, factor));
}
/*
* XXX
* Should wait for chip to be idle.
*/
sc->sc_i.run = SB_NOTRUNNING;
sc->sc_o.run = SB_NOTRUNNING;
if (sc->sc_fullduplex &&
(usemode & (AUMODE_PLAY | AUMODE_RECORD)) == (AUMODE_PLAY | AUMODE_RECORD) &&
sc->sc_i.dmachan == sc->sc_o.dmachan) {
DPRINTF(("sbdsp_commit: fd=%d, usemode=%d, idma=%d, odma=%d\n", sc->sc_fullduplex, usemode, sc->sc_i.dmachan, sc->sc_o.dmachan));
if (sc->sc_o.dmachan == sc->sc_drq8) {
/* Use 16 bit DMA for playing by expanding the samples. */
play->sw_code = linear8_to_linear16;
play->factor = 2;
sc->sc_o.modep = &sbpmodes[PLAY16];
sc->sc_o.dmachan = sc->sc_drq16;
} else {
return EINVAL;
}
}
DPRINTF(("sbdsp_set_params ichan=%d, ochan=%d\n", sc->sc_i.dmachan, sc->sc_o.dmachan));
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_in_ports(sc, mask)
struct sbdsp_softc *sc;
int mask;
{
int bitsl, bitsr;
int sbport;
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, sbport | sc->in_filter);
break;
case SBM_CT1XX5:
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;
return 0;
}
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;
{
blk &= -4; /* round to biggest sample size */
return blk;
}
int
sbdsp_open(addr, flags)
void *addr;
int flags;
{
struct sbdsp_softc *sc = addr;
DPRINTF(("sbdsp_open: sc=%p\n", sc));
if (sc->sc_open != 0 || sbdsp_reset(sc) != 0)
return ENXIO;
sc->sc_open = 1;
sc->sc_openflags = flags;
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_intr8 = 0;
sc->sc_intr16 = 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_intr8 = 0;
sc->sc_intr16 = 0;
if (sc->sc_i.run != SB_NOTRUNNING) {
isa_dmaabort(sc->sc_isa, sc->sc_i.dmachan);
sc->sc_i.run = SB_NOTRUNNING;
}
if (sc->sc_o.run != SB_NOTRUNNING) {
isa_dmaabort(sc->sc_isa, sc->sc_o.dmachan);
sc->sc_o.run = SB_NOTRUNNING;
}
/*
* 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;
}
/*
* Write a byte to the dsp.
* 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) {
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) {
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.
* Store version information in the softc.
*/
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:
#if 0
/* XXX This does not work */
/* Most SB16 have a tone controls, but some don't. */
sbdsp_mix_write(sc, SB16P_TREBLE_L, 0x80);
/* Check if we can read back the mixer value. */
if ((sbdsp_mix_read(sc, SB16P_TREBLE_L) & 0xf0) == 0x80)
sc->sc_mixer_model = SBM_CT1745;
else
sc->sc_mixer_model = SBM_CT1XX5;
#else
sc->sc_mixer_model = SBM_CT1745;
#endif
/* XXX what about SB_32 */
if (SBVER_MINOR(v) == 16)
sc->sc_model = SB_64;
else
sc->sc_model = SB_16;
break;
}
}
/*
* Halt a DMA in progress.
*/
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_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 cmd=0x%02x rate=%d\n", sc, cmd, 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_init_input(addr, buf, cc)
void *addr;
void *buf;
int cc;
{
struct sbdsp_softc *sc = addr;
if (sc->sc_model == SB_1)
return 0;
sc->sc_i.run = SB_DMARUNNING;
DPRINTF(("sbdsp: dma start loop input addr=%p cc=%d chan=%d\n",
buf, cc, sc->sc_i.dmachan));
isa_dmastart(sc->sc_isa, sc->sc_i.dmachan, buf,
cc, NULL, DMAMODE_READ | DMAMODE_LOOP, BUS_DMA_NOWAIT);
return 0;
}
static int
sbdsp_dma_setup_input(sc)
struct sbdsp_softc *sc;
{
int stereo = sc->sc_i.modep->channels == 2;
int filter;
/* Initialize the PCM */
if (ISSBPRO(sc)) {
if (sbdsp_wdsp(sc, sc->sc_i.modep->cmdchan) < 0)
return 0;
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_i.rate)) {
DPRINTF(("sbdsp_dma_setup_input: rate=%d set failed\n",
sc->sc_i.rate));
return 0;
}
} else {
if (sbdsp_set_timeconst(sc, sc->sc_i.tc)) {
DPRINTF(("sbdsp_dma_setup_input: tc=%d set failed\n",
sc->sc_i.rate));
return 0;
}
}
return 1;
}
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;
#ifdef AUDIO_DEBUG
if (sbdspdebug > 1)
printf("sbdsp_dma_input: sc=%p buf=%p cc=%d intr=%p(%p)\n",
addr, p, cc, intr, arg);
#endif
#ifdef DIAGNOSTIC
if (sc->sc_i.modep->channels == 2 && (cc & 1)) {
DPRINTF(("stereo record odd bytes (%d)\n", cc));
return EIO;
}
#endif
if (sc->sc_i.modep->precision == 8) {
#ifdef DIAGNOSTIC
if (sc->sc_i.dmachan != sc->sc_drq8) {
printf("sbdsp_dma_input: prec=%d bad chan %d\n",
sc->sc_i.modep->precision, sc->sc_i.dmachan);
return EIO;
}
#endif
sc->sc_intr8 = intr;
sc->sc_arg8 = arg;
} else {
#ifdef DIAGNOSTIC
if (sc->sc_i.dmachan != sc->sc_drq16) {
printf("sbdsp_dma_input: prec=%d bad chan %d\n",
sc->sc_i.modep->precision, sc->sc_i.dmachan);
return EIO;
}
#endif
sc->sc_intr16 = intr;
sc->sc_arg16 = arg;
}
switch(sc->sc_i.run) {
case SB_NOTRUNNING:
/* Non-looping mode, not initialized */
sc->sc_i.run = SB_RUNNING;
if (!sbdsp_dma_setup_input(sc))
goto giveup;
/* fall into */
case SB_RUNNING:
/* Non-looping mode, start DMA */
#ifdef AUDIO_DEBUG
if (sbdspdebug > 2)
printf("sbdsp_dma_input: dmastart buf=%p cc=%d chan=%d\n",
p, cc, sc->sc_i.dmachan);
#endif
isa_dmastart(sc->sc_isa, sc->sc_i.dmachan, p,
cc, NULL, DMAMODE_READ, BUS_DMA_NOWAIT);
/* Start PCM in non-looping mode */
if ((sc->sc_model == SB_JAZZ && sc->sc_i.dmachan > 3) ||
(sc->sc_model != SB_JAZZ && sc->sc_i.modep->precision == 16))
cc >>= 1;
--cc;
if (sbdsp_wdsp(sc, sc->sc_i.modep->cmd) < 0 ||
sbdsp_wdsp(sc, cc) < 0 ||
sbdsp_wdsp(sc, cc >> 8) < 0) {
DPRINTF(("sbdsp_dma_input: SB1 DMA start failed\n"));
goto giveup;
}
break;
case SB_DMARUNNING:
/* Looping mode, not initialized */
sc->sc_i.run = SB_PCMRUNNING;
if (!sbdsp_dma_setup_input(sc))
goto giveup;
if ((sc->sc_model == SB_JAZZ && sc->sc_i.dmachan > 3) ||
(sc->sc_model != SB_JAZZ && sc->sc_i.modep->precision == 16))
cc >>= 1;
--cc;
/* Initialize looping PCM */
if (ISSB16CLASS(sc)) {
#ifdef AUDIO_DEBUG
if (sbdspdebug > 2)
printf("sbdsp16 input command cmd=0x%02x bmode=0x%02x cc=%d\n",
sc->sc_i.modep->cmd, sc->sc_i.bmode, cc);
#endif
if (sbdsp_wdsp(sc, sc->sc_i.modep->cmd) < 0 ||
sbdsp_wdsp(sc, sc->sc_i.bmode) < 0 ||
sbdsp_wdsp(sc, cc) < 0 ||
sbdsp_wdsp(sc, cc >> 8) < 0) {
DPRINTF(("sbdsp_dma_input: SB16 DMA start failed\n"));
DPRINTF(("sbdsp16 input command cmd=0x%02x bmode=0x%02x cc=%d\n",
sc->sc_i.modep->cmd, sc->sc_i.bmode, cc));
goto giveup;
}
} else {
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 blocksize failed\n"));
goto giveup;
}
if (sbdsp_wdsp(sc, sc->sc_i.modep->cmd) < 0) {
DPRINTF(("sbdsp_dma_input: SB2 DMA start failed\n"));
goto giveup;
}
}
break;
case SB_PCMRUNNING:
/* Looping mode, nothing to do */
break;
}
return 0;
giveup:
sbdsp_reset(sc);
return EIO;
}
int
sbdsp_dma_init_output(addr, buf, cc)
void *addr;
void *buf;
int cc;
{
struct sbdsp_softc *sc = addr;
if (sc->sc_model == SB_1)
return 0;
sc->sc_o.run = SB_DMARUNNING;
DPRINTF(("sbdsp: dma start loop output buf=%p cc=%d chan=%d\n",
buf, cc, sc->sc_o.dmachan));
isa_dmastart(sc->sc_isa, sc->sc_o.dmachan, buf,
cc, NULL, DMAMODE_WRITE | DMAMODE_LOOP, BUS_DMA_NOWAIT);
return 0;
}
static int
sbdsp_dma_setup_output(sc)
struct sbdsp_softc *sc;
{
int stereo = sc->sc_o.modep->channels == 2;
int cmd;
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_o.modep->cmdchan;
if (cmd && sbdsp_wdsp(sc, cmd) < 0)
return 0;
}
if (ISSB16CLASS(sc)) {
if (sbdsp16_set_rate(sc, SB_DSP16_OUTPUTRATE,
sc->sc_o.rate)) {
DPRINTF(("sbdsp_dma_setup_output: rate=%d set failed\n",
sc->sc_o.rate));
return 0;
}
} else {
if (sbdsp_set_timeconst(sc, sc->sc_o.tc)) {
DPRINTF(("sbdsp_dma_setup_output: tc=%d set failed\n",
sc->sc_o.rate));
return 0;
}
}
return 1;
}
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;
#ifdef AUDIO_DEBUG
if (sbdspdebug > 1)
printf("sbdsp_dma_output: sc=%p buf=%p cc=%d intr=%p(%p)\n", addr, p, cc, intr, arg);
#endif
#ifdef DIAGNOSTIC
if (sc->sc_o.modep->channels == 2 && (cc & 1)) {
DPRINTF(("stereo playback odd bytes (%d)\n", cc));
return EIO;
}
#endif
if (sc->sc_o.modep->precision == 8) {
#ifdef DIAGNOSTIC
if (sc->sc_o.dmachan != sc->sc_drq8) {
printf("sbdsp_dma_output: prec=%d bad chan %d\n",
sc->sc_o.modep->precision, sc->sc_o.dmachan);
return EIO;
}
#endif
sc->sc_intr8 = intr;
sc->sc_arg8 = arg;
} else {
#ifdef DIAGNOSTIC
if (sc->sc_o.dmachan != sc->sc_drq16) {
printf("sbdsp_dma_output: prec=%d bad chan %d\n",
sc->sc_o.modep->precision, sc->sc_o.dmachan);
return EIO;
}
#endif
sc->sc_intr16 = intr;
sc->sc_arg16 = arg;
}
switch(sc->sc_o.run) {
case SB_NOTRUNNING:
/* Non-looping mode, not initialized */
sc->sc_o.run = SB_RUNNING;
if (!sbdsp_dma_setup_output(sc))
goto giveup;
/* fall into */
case SB_RUNNING:
/* Non-looping mode, initialized. Start DMA and PCM */
#ifdef AUDIO_DEBUG
if (sbdspdebug > 2)
printf("sbdsp: start dma out addr=%p, cc=%d, chan=%d\n",
p, cc, sc->sc_o.dmachan);
#endif
isa_dmastart(sc->sc_isa, sc->sc_o.dmachan, p,
cc, NULL, DMAMODE_WRITE, BUS_DMA_NOWAIT);
if ((sc->sc_model == SB_JAZZ && sc->sc_o.dmachan > 3) ||
(sc->sc_model != SB_JAZZ && sc->sc_o.modep->precision == 16))
cc >>= 1;
--cc;
if (sbdsp_wdsp(sc, sc->sc_o.modep->cmd) < 0 ||
sbdsp_wdsp(sc, cc) < 0 ||
sbdsp_wdsp(sc, cc >> 8) < 0) {
DPRINTF(("sbdsp_dma_output: SB1 DMA start failed\n"));
goto giveup;
}
break;
case SB_DMARUNNING:
/* Looping mode, not initialized */
sc->sc_o.run = SB_PCMRUNNING;
if (!sbdsp_dma_setup_output(sc))
goto giveup;
if ((sc->sc_model == SB_JAZZ && sc->sc_o.dmachan > 3) ||
(sc->sc_model != SB_JAZZ && sc->sc_o.modep->precision == 16))
cc >>= 1;
--cc;
/* Initialize looping PCM */
if (ISSB16CLASS(sc)) {
DPRINTF(("sbdsp_dma_output: SB16 cmd=0x%02x bmode=0x%02x cc=%d\n",
sc->sc_o.modep->cmd,sc->sc_o.bmode, cc));
if (sbdsp_wdsp(sc, sc->sc_o.modep->cmd) < 0 ||
sbdsp_wdsp(sc, sc->sc_o.bmode) < 0 ||
sbdsp_wdsp(sc, cc) < 0 ||
sbdsp_wdsp(sc, cc >> 8) < 0) {
DPRINTF(("sbdsp_dma_output: SB16 DMA start failed\n"));
goto giveup;
}
} else {
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_o.modep->cmd) < 0) {
DPRINTF(("sbdsp_dma_output: SB2 DMA start failed\n"));
goto giveup;
}
}
break;
case SB_PCMRUNNING:
/* Looping mode, nothing to do */
break;
}
return 0;
giveup:
sbdsp_reset(sc);
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.
*
* If there is interrupt sharing or a spurious interrupt occurs
* there is no way to distinguish this on an SB2. So if you have
* an SB2 and experience problems, buy an SB16 (it's only $40).
*/
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)
printf("sbdsp_intr: intr8=%p, intr16=%p\n",
sc->sc_intr8, sc->sc_intr16);
#endif
if (ISSB16CLASS(sc)) {
irq = sbdsp_mix_read(sc, SBP_IRQ_STATUS);
if ((irq & (SBP_IRQ_DMA8 | SBP_IRQ_DMA16)) == 0) {
DPRINTF(("sbdsp_intr: Spurious interrupt 0x%x\n", irq));
return 0;
}
} else {
if (!loop && !isa_dmafinished(sc->sc_isa, sc->sc_drq8))
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_intr8 == 0 && sc->sc_intr16 == 0) {
DPRINTF(("sbdsp_intr: Unexpected interrupt 0x%x\n", irq));
/* XXX return 0;*/ /* Did not expect an interrupt */
}
/* clear interrupt */
if (irq & SBP_IRQ_DMA8) {
bus_space_read_1(sc->sc_iot, sc->sc_ioh, SBP_DSP_IRQACK8);
if (!loop)
isa_dmadone(sc->sc_isa, sc->sc_drq8);
if (sc->sc_intr8)
(*sc->sc_intr8)(sc->sc_arg8);
}
if (irq & SBP_IRQ_DMA16) {
bus_space_read_1(sc->sc_iot, sc->sc_ioh, SBP_DSP_IRQACK16);
if (sc->sc_intr16)
(*sc->sc_intr16)(sc->sc_arg16);
}
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
/* Mask a value 0-255, but round it first */
#define MAXVAL 256
static int
sbdsp_adjust(val, mask)
int val, mask;
{
val += (MAXVAL - mask) >> 1;
if (val >= MAXVAL)
val = MAXVAL-1;
return val & mask;
}
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_CT1XX5:
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;
int mask, bits;
int lmask, rmask, lbits, rbits;
int mute, swap;
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 ||
sc->sc_mixer_model == SBM_CT1XX5) {
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 (!ISSBM1745(sc))
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 (ISSBM1745(sc)) {
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;
sc->in_port = cp->un.ord;
return sbdsp_set_in_ports(sc, 1 << cp->un.ord);
}
break;
case SB_AGC:
if (!ISSBM1745(sc) || cp->type != AUDIO_MIXER_ENUM)
return EINVAL;
sbdsp_mix_write(sc, SB16P_AGC, cp->un.ord & 1);
break;
case SB_CD_OUT_MUTE:
mask = SB16P_SW_CD;
goto omute;
case SB_MIC_OUT_MUTE:
mask = SB16P_SW_MIC;
goto omute;
case SB_LINE_OUT_MUTE:
mask = SB16P_SW_LINE;
omute:
if (cp->type != AUDIO_MIXER_ENUM)
return EINVAL;
bits = sbdsp_mix_read(sc, SB16P_OSWITCH);
sc->gain[cp->dev][SB_LR] = cp->un.ord != 0;
if (cp->un.ord)
bits = bits & ~mask;
else
bits = bits | mask;
sbdsp_mix_write(sc, SB16P_OSWITCH, bits);
break;
case SB_MIC_IN_MUTE:
case SB_MIC_SWAP:
lmask = rmask = SB16P_SW_MIC;
goto imute;
case SB_CD_IN_MUTE:
case SB_CD_SWAP:
lmask = SB16P_SW_CD_L;
rmask = SB16P_SW_CD_R;
goto imute;
case SB_LINE_IN_MUTE:
case SB_LINE_SWAP:
lmask = SB16P_SW_LINE_L;
rmask = SB16P_SW_LINE_R;
goto imute;
case SB_MIDI_IN_MUTE:
case SB_MIDI_SWAP:
lmask = SB16P_SW_MIDI_L;
rmask = SB16P_SW_MIDI_R;
imute:
if (cp->type != AUDIO_MIXER_ENUM)
return EINVAL;
mask = lmask | rmask;
lbits = sbdsp_mix_read(sc, SB16P_ISWITCH_L) & ~mask;
rbits = sbdsp_mix_read(sc, SB16P_ISWITCH_R) & ~mask;
sc->gain[cp->dev][SB_LR] = cp->un.ord != 0;
if (SB_IS_IN_MUTE(cp->dev)) {
mute = cp->dev;
swap = mute - SB_CD_IN_MUTE + SB_CD_SWAP;
} else {
swap = cp->dev;
mute = swap + SB_CD_IN_MUTE - SB_CD_SWAP;
}
if (sc->gain[swap][SB_LR]) {
mask = lmask;
lmask = rmask;
rmask = mask;
}
if (!sc->gain[mute][SB_LR]) {
lbits = lbits | lmask;
rbits = rbits | rmask;
}
sbdsp_mix_write(sc, SB16P_ISWITCH_L, lbits);
sbdsp_mix_write(sc, SB16P_ISWITCH_L, rbits);
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 ||
sc->sc_mixer_model == SBM_CT1XX5) {
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 (!ISSBM1745(sc))
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 (ISSBM1745(sc))
cp->un.mask = sc->in_mask;
else
cp->un.ord = sc->in_port;
break;
case SB_AGC:
if (!ISSBM1745(sc))
return EINVAL;
cp->un.ord = sbdsp_mix_read(sc, SB16P_AGC);
break;
case SB_CD_IN_MUTE:
case SB_MIC_IN_MUTE:
case SB_LINE_IN_MUTE:
case SB_MIDI_IN_MUTE:
case SB_CD_SWAP:
case SB_MIC_SWAP:
case SB_LINE_SWAP:
case SB_MIDI_SWAP:
case SB_CD_OUT_MUTE:
case SB_MIC_OUT_MUTE:
case SB_LINE_OUT_MUTE:
cp->un.ord = sc->gain[cp->dev][SB_LR];
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, is1745;
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;
is1745 = ISSBM1745(sc);
class = is1745 ? 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 = is1745 ? SB_MIDI_IN_MUTE : 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 = is1745 ? SB_CD_IN_MUTE : 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 = is1745 ? SB_MIC_IN_MUTE : 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 = is1745 ? SB_LINE_IN_MUTE : 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 (ISSBM1745(sc)) {
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;
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;
}
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_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;
case SB_CD_IN_MUTE:
dip->prev = SB_CD_VOL;
dip->next = SB_CD_SWAP;
dip->mixer_class = SB_INPUT_CLASS;
goto mute;
case SB_MIC_IN_MUTE:
dip->prev = SB_MIC_VOL;
dip->next = SB_MIC_SWAP;
dip->mixer_class = SB_INPUT_CLASS;
goto mute;
case SB_LINE_IN_MUTE:
dip->prev = SB_LINE_IN_VOL;
dip->next = SB_LINE_SWAP;
dip->mixer_class = SB_INPUT_CLASS;
goto mute;
case SB_MIDI_IN_MUTE:
dip->prev = SB_MIDI_VOL;
dip->next = SB_MIDI_SWAP;
dip->mixer_class = SB_INPUT_CLASS;
goto mute;
case SB_CD_SWAP:
dip->prev = SB_CD_IN_MUTE;
dip->next = SB_CD_OUT_MUTE;
goto swap;
case SB_MIC_SWAP:
dip->prev = SB_MIC_IN_MUTE;
dip->next = SB_MIC_OUT_MUTE;
goto swap;
case SB_LINE_SWAP:
dip->prev = SB_LINE_IN_MUTE;
dip->next = SB_LINE_OUT_MUTE;
goto swap;
case SB_MIDI_SWAP:
dip->prev = SB_MIDI_IN_MUTE;
dip->next = AUDIO_MIXER_LAST;
swap:
dip->mixer_class = SB_INPUT_CLASS;
strcpy(dip->label.name, AudioNswap);
goto mute1;
case SB_CD_OUT_MUTE:
dip->prev = SB_CD_SWAP;
dip->next = AUDIO_MIXER_LAST;
dip->mixer_class = SB_OUTPUT_CLASS;
goto mute;
case SB_MIC_OUT_MUTE:
dip->prev = SB_MIC_SWAP;
dip->next = AUDIO_MIXER_LAST;
dip->mixer_class = SB_OUTPUT_CLASS;
goto mute;
case SB_LINE_OUT_MUTE:
dip->prev = SB_LINE_SWAP;
dip->next = AUDIO_MIXER_LAST;
dip->mixer_class = SB_OUTPUT_CLASS;
mute:
strcpy(dip->label.name, AudioNmute);
mute1:
dip->type = AUDIO_MIXER_ENUM;
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;
}
return ENXIO;
}
void *
sb_malloc(addr, size, pool, flags)
void *addr;
unsigned long size;
int pool;
int flags;
{
struct sbdsp_softc *sc = addr;
return isa_malloc(sc->sc_isa, 4, size, pool, flags);
}
void
sb_free(addr, ptr, pool)
void *addr;
void *ptr;
int pool;
{
isa_free(ptr, pool);
}
unsigned long
sb_round(addr, size)
void *addr;
unsigned long size;
{
if (size > MAX_ISADMA)
size = MAX_ISADMA;
return size;
}
int
sb_mappage(addr, mem, off, prot)
void *addr;
void *mem;
int off;
int prot;
{
return isa_mappage(mem, off, prot);
}
int
sbdsp_get_props(addr)
void *addr;
{
struct sbdsp_softc *sc = addr;
return AUDIO_PROP_MMAP |
(sc->sc_fullduplex ? AUDIO_PROP_FULLDUPLEX : 0);
}
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