Aren
/
NetBSD
1
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity
NetBSD/sys/dev/isa/sbdsp.c

1781 lines
40 KiB
C
Raw Blame History

/* $NetBSD: sbdsp.c,v 1.26 1996/05/12 23:53:38 mycroft 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>.
*/
#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/pio.h>
#include <sys/audioio.h>
#include <dev/audio_if.h>
#include <dev/isa/isavar.h>
#include <dev/isa/isadmavar.h>
#include <i386/isa/icu.h> /* XXX BROKEN; WHY? */
#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;
int sbdsp_srtotc __P((struct sbdsp_softc *sc, int sr, int isdac,
int *tcp, int *modep));
u_int sbdsp_jazz16_probe __P((struct sbdsp_softc *));
/*
* 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
*/
#define SB_LS_MIN 0x06 /* 4000 Hz */
#define SB_8K 0x83 /* 8000 Hz */
#define SBPRO_ADC_LS_MAX 0xd4 /* 22727 Hz */
#define SBPRO_ADC_HS_MAX 0xea /* 45454 Hz */
#define SBCLA_ADC_LS_MAX 0xb3 /* 12987 Hz */
#define SBCLA_ADC_HS_MAX 0xbd /* 14925 Hz */
#define SB_DAC_LS_MAX 0xd4 /* 22727 Hz */
#define SB_DAC_HS_MAX 0xea /* 45454 Hz */
int sbdsp16_wait __P((int));
void sbdsp_to __P((void *));
void sbdsp_pause __P((struct sbdsp_softc *));
int sbdsp_setrate __P((struct sbdsp_softc *, int, int, int *));
int sbdsp_tctosr __P((struct sbdsp_softc *, int));
int sbdsp_set_timeconst __P((struct sbdsp_softc *, int));
#ifdef AUDIO_DEBUG
void sb_printsc __P((struct sbdsp_softc *));
#endif
#ifdef AUDIO_DEBUG
void
sb_printsc(sc)
struct sbdsp_softc *sc;
{
int i;
printf("open %d dmachan %d iobase %x\n",
sc->sc_open, sc->sc_drq, sc->sc_iobase);
printf("irate %d itc %d imode %d orate %d otc %d omode %d encoding %x\n",
sc->sc_irate, sc->sc_itc, sc->sc_imode,
sc->sc_orate, sc->sc_otc, sc->sc_omode, sc->encoding);
printf("outport %d inport %d spkron %d nintr %d\n",
sc->out_port, sc->in_port, sc->spkr_state, sc->sc_interrupts);
printf("precision %d channels %d intr %x arg %x\n",
sc->sc_precision, sc->sc_channels, sc->sc_intr, sc->sc_arg);
printf("gain: ");
for (i = 0; i < SB_NDEVS; i++)
printf("%d ", sc->gain[i]);
printf("\n");
}
#endif
/*
* 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 != 0)
sc->sc_model = sbdsp_jazz16_probe(sc);
else
sc->sc_model = sbversion(sc);
return 1;
}
/*
* Try add-on stuff for Jazz16.
*/
u_int
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};
u_int rval = sbversion(sc);
register int iobase = sc->sc_iobase;
if (jazz16_drq_conf[sc->sc_drq] == (u_char)-1 ||
jazz16_irq_conf[sc->sc_irq] == (u_char)-1)
return rval; /* give up, we can't do it. */
outb(JAZZ16_CONFIG_PORT, JAZZ16_WAKEUP);
delay(10000); /* delay 10 ms */
outb(JAZZ16_CONFIG_PORT, JAZZ16_SETBASE);
outb(JAZZ16_CONFIG_PORT, iobase & 0x70);
if (sbdsp_reset(sc) < 0)
return rval; /* XXX? what else could we do? */
if (sbdsp_wdsp(iobase, JAZZ16_READ_VER))
return rval;
if (sbdsp_rdsp(iobase) != JAZZ16_VER_JAZZ)
return rval;
if (sbdsp_wdsp(iobase, JAZZ16_SET_DMAINTR) ||
/* set both 8 & 16-bit drq to same channel, it works fine. */
sbdsp_wdsp(iobase,
(jazz16_drq_conf[sc->sc_drq] << 4) |
jazz16_drq_conf[sc->sc_drq]) ||
sbdsp_wdsp(iobase, jazz16_irq_conf[sc->sc_irq])) {
DPRINTF(("sbdsp: can't write jazz16 probe stuff"));
return rval;
}
return (rval | MODEL_JAZZ16);
}
/*
* Attach hardware to driver, attach hardware driver to audio
* pseudo-device driver .
*/
void
sbdsp_attach(sc)
struct sbdsp_softc *sc;
{
/* Set defaults */
if (ISSB16CLASS(sc))
sc->sc_irate = sc->sc_orate = 8000;
else if (ISSBPROCLASS(sc))
sc->sc_itc = sc->sc_otc = SB_8K;
else
sc->sc_itc = sc->sc_otc = SB_8K;
sc->sc_precision = 8;
sc->sc_channels = 1;
sc->encoding = AUDIO_ENCODING_ULAW;
(void) sbdsp_set_in_port(sc, SB_MIC_PORT);
(void) sbdsp_set_out_port(sc, SB_SPEAKER);
if (ISSBPROCLASS(sc)) {
int i;
/* set mixer to default levels, by sending a mixer
reset command. */
sbdsp_mix_write(sc, SBP_MIX_RESET, SBP_MIX_RESET);
/* then some adjustments :) */
sbdsp_mix_write(sc, SBP_CD_VOL,
sbdsp_stereo_vol(SBP_MAXVOL, SBP_MAXVOL));
sbdsp_mix_write(sc, SBP_DAC_VOL,
sbdsp_stereo_vol(SBP_MAXVOL, SBP_MAXVOL));
sbdsp_mix_write(sc, SBP_MASTER_VOL,
sbdsp_stereo_vol(SBP_MAXVOL/2, SBP_MAXVOL/2));
sbdsp_mix_write(sc, SBP_LINE_VOL,
sbdsp_stereo_vol(SBP_MAXVOL, SBP_MAXVOL));
for (i = 0; i < SB_NDEVS; i++)
sc->gain[i] = sbdsp_stereo_vol(SBP_MAXVOL, SBP_MAXVOL);
sc->in_filter = 0; /* no filters turned on, please */
}
printf(": dsp v%d.%02d%s\n",
SBVER_MAJOR(sc->sc_model), SBVER_MINOR(sc->sc_model),
ISJAZZ16(sc) ? ": <Jazz16>" : "");
#ifdef notyet
sbdsp_mix_write(sc, SBP_SET_IRQ, 0x04);
sbdsp_mix_write(sc, SBP_SET_DRQ, 0x22);
printf("sbdsp_attach: irq=%02x, drq=%02x\n",
sbdsp_mix_read(sc, SBP_SET_IRQ),
sbdsp_mix_read(sc, SBP_SET_DRQ));
#else
if (ISSB16CLASS(sc))
sc->sc_model = 0x0300;
#endif
}
/*
* Various routines to interface to higher level audio driver
*/
void
sbdsp_mix_write(sc, mixerport, val)
struct sbdsp_softc *sc;
int mixerport;
int val;
{
int iobase = sc->sc_iobase;
outb(iobase + SBP_MIXER_ADDR, mixerport);
delay(10);
outb(iobase + SBP_MIXER_DATA, val);
delay(30);
}
int
sbdsp_mix_read(sc, mixerport)
struct sbdsp_softc *sc;
int mixerport;
{
int iobase = sc->sc_iobase;
outb(iobase + SBP_MIXER_ADDR, mixerport);
delay(10);
return inb(iobase + SBP_MIXER_DATA);
}
int
sbdsp_set_in_sr(addr, sr)
void *addr;
u_long sr;
{
register struct sbdsp_softc *sc = addr;
if (ISSB16CLASS(sc))
return (sbdsp_setrate(sc, sr, SB_INPUT_RATE, &sc->sc_irate));
else
return (sbdsp_srtotc(sc, sr, SB_INPUT_RATE, &sc->sc_itc, &sc->sc_imode));
}
u_long
sbdsp_get_in_sr(addr)
void *addr;
{
register struct sbdsp_softc *sc = addr;
if (ISSB16CLASS(sc))
return (sc->sc_irate);
else
return (sbdsp_tctosr(sc, sc->sc_itc));
}
int
sbdsp_set_out_sr(addr, sr)
void *addr;
u_long sr;
{
register struct sbdsp_softc *sc = addr;
if (ISSB16CLASS(sc))
return (sbdsp_setrate(sc, sr, SB_OUTPUT_RATE, &sc->sc_orate));
else
return (sbdsp_srtotc(sc, sr, SB_OUTPUT_RATE, &sc->sc_otc, &sc->sc_omode));
}
u_long
sbdsp_get_out_sr(addr)
void *addr;
{
register struct sbdsp_softc *sc = addr;
if (ISSB16CLASS(sc))
return (sc->sc_orate);
else
return (sbdsp_tctosr(sc, sc->sc_otc));
}
int
sbdsp_query_encoding(addr, fp)
void *addr;
struct audio_encoding *fp;
{
switch (fp->index) {
case 0:
strcpy(fp->name, AudioEmulaw);
fp->format_id = AUDIO_ENCODING_ULAW;
break;
case 1:
strcpy(fp->name, AudioEpcm16);
fp->format_id = AUDIO_ENCODING_PCM16;
break;
default:
return (EINVAL);
}
return (0);
}
int
sbdsp_set_encoding(addr, encoding)
void *addr;
u_int encoding;
{
register struct sbdsp_softc *sc = addr;
switch (encoding) {
case AUDIO_ENCODING_ULAW:
sc->encoding = AUDIO_ENCODING_ULAW;
break;
case AUDIO_ENCODING_LINEAR:
sc->encoding = AUDIO_ENCODING_LINEAR;
break;
default:
return (EINVAL);
}
return (0);
}
int
sbdsp_get_encoding(addr)
void *addr;
{
register struct sbdsp_softc *sc = addr;
return (sc->encoding);
}
int
sbdsp_set_precision(addr, precision)
void *addr;
u_int precision;
{
register struct sbdsp_softc *sc = addr;
if (ISSB16CLASS(sc) || ISJAZZ16(sc)) {
if (precision != 16 && precision != 8)
return (EINVAL);
sc->sc_precision = precision;
} else {
if (precision != 8)
return (EINVAL);
sc->sc_precision = precision;
}
return (0);
}
int
sbdsp_get_precision(addr)
void *addr;
{
register struct sbdsp_softc *sc = addr;
return (sc->sc_precision);
}
int
sbdsp_set_channels(addr, channels)
void *addr;
int channels;
{
register struct sbdsp_softc *sc = addr;
if (ISSBPROCLASS(sc)) {
if (channels != 1 && channels != 2)
return (EINVAL);
sc->sc_channels = channels;
sc->sc_dmadir = SB_DMA_NONE;
/*
* XXXX
* With 2 channels, SBPro can't do more than 22kHz.
* No framework to check this.
*/
} else {
if (channels != 1)
return (EINVAL);
sc->sc_channels = channels;
}
return (0);
}
int
sbdsp_get_channels(addr)
void *addr;
{
register struct sbdsp_softc *sc = addr;
return (sc->sc_channels);
}
int
sbdsp_set_ifilter(addr, which)
void *addr;
int which;
{
register struct sbdsp_softc *sc = addr;
int mixval;
if (ISSBPROCLASS(sc)) {
mixval = sbdsp_mix_read(sc, SBP_INFILTER) & ~SBP_IFILTER_MASK;
switch (which) {
case 0:
mixval |= SBP_FILTER_OFF;
break;
case SBP_TREBLE_EQ:
mixval |= SBP_FILTER_ON | SBP_IFILTER_HIGH;
break;
case SBP_BASS_EQ:
mixval |= SBP_FILTER_ON | SBP_IFILTER_LOW;
break;
default:
return (EINVAL);
}
sc->in_filter = mixval & SBP_IFILTER_MASK;
sbdsp_mix_write(sc, SBP_INFILTER, mixval);
return (0);
} else
return (EINVAL);
}
int
sbdsp_get_ifilter(addr)
void *addr;
{
register struct sbdsp_softc *sc = addr;
if (ISSBPROCLASS(sc)) {
sc->in_filter =
sbdsp_mix_read(sc, SBP_INFILTER) & SBP_IFILTER_MASK;
switch (sc->in_filter) {
case SBP_FILTER_ON|SBP_IFILTER_HIGH:
return (SBP_TREBLE_EQ);
case SBP_FILTER_ON|SBP_IFILTER_LOW:
return (SBP_BASS_EQ);
case SBP_FILTER_OFF:
default:
return (0);
}
} else
return (0);
}
int
sbdsp_set_out_port(addr, port)
void *addr;
int port;
{
register struct sbdsp_softc *sc = addr;
sc->out_port = port; /* Just record it */
return (0);
}
int
sbdsp_get_out_port(addr)
void *addr;
{
register struct sbdsp_softc *sc = addr;
return (sc->out_port);
}
int
sbdsp_set_in_port(addr, port)
void *addr;
int port;
{
register struct sbdsp_softc *sc = addr;
int mixport, sbport;
if (ISSBPROCLASS(sc)) {
switch (port) {
case SB_MIC_PORT:
sbport = SBP_FROM_MIC;
mixport = SBP_MIC_VOL;
break;
case SB_LINE_IN_PORT:
sbport = SBP_FROM_LINE;
mixport = SBP_LINE_VOL;
break;
case SB_CD_PORT:
sbport = SBP_FROM_CD;
mixport = SBP_CD_VOL;
break;
case SB_DAC_PORT:
case SB_FM_PORT:
default:
return (EINVAL);
}
} else {
switch (port) {
case SB_MIC_PORT:
sbport = SBP_FROM_MIC;
mixport = SBP_MIC_VOL;
break;
default:
return (EINVAL);
}
}
sc->in_port = port; /* Just record it */
if (ISSBPROCLASS(sc)) {
/* record from that port */
sbdsp_mix_write(sc, SBP_RECORD_SOURCE,
SBP_RECORD_FROM(sbport, SBP_FILTER_OFF, SBP_IFILTER_HIGH));
/* fetch gain from that port */
sc->gain[port] = sbdsp_mix_read(sc, mixport);
}
return (0);
}
int
sbdsp_get_in_port(addr)
void *addr;
{
register struct sbdsp_softc *sc = addr;
return (sc->in_port);
}
int
sbdsp_speaker_ctl(addr, newstate)
void *addr;
int newstate;
{
register 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;
{
register struct sbdsp_softc *sc = addr;
sc->sc_last_hs_size = 0;
/* Higher speeds need bigger blocks to avoid popping and silence gaps. */
if (blk < NBPG/4 || blk > NBPG/2) {
if (ISSB16CLASS(sc)) {
if (sc->sc_orate > 8000 || sc->sc_irate > 8000)
blk = NBPG/2;
} else {
if (sc->sc_otc > SB_8K || sc->sc_itc < SB_8K)
blk = NBPG/2;
}
}
/* don't try to DMA too much at once, though. */
if (blk > NBPG)
blk = NBPG;
if (sc->sc_channels == 2)
return (blk & ~1); /* must be even to preserve stereo separation */
else
return (blk); /* Anything goes :-) */
}
int
sbdsp_commit_settings(addr)
void *addr;
{
register struct sbdsp_softc *sc = addr;
/* due to potentially unfortunate ordering in the above layers,
re-do a few sets which may be important--input gains
(adjust the proper channels), number of input channels (hit the
record rate and set mode) */
if (ISSBPRO(sc)) {
/*
* With 2 channels, SBPro can't do more than 22kHz.
* Whack the rates down to speed if necessary.
* Reset the time constant anyway
* because it may have been adjusted with a different number
* of channels, which means it might have computed the wrong
* mode (low/high speed).
*/
if (sc->sc_channels == 2 &&
sbdsp_tctosr(sc, sc->sc_itc) > 22727) {
sbdsp_srtotc(sc, 22727, SB_INPUT_RATE,
&sc->sc_itc, &sc->sc_imode);
} else
sbdsp_srtotc(sc, sbdsp_tctosr(sc, sc->sc_itc),
SB_INPUT_RATE, &sc->sc_itc,
&sc->sc_imode);
if (sc->sc_channels == 2 &&
sbdsp_tctosr(sc, sc->sc_otc) > 22727) {
sbdsp_srtotc(sc, 22727, SB_OUTPUT_RATE,
&sc->sc_otc, &sc->sc_omode);
} else
sbdsp_srtotc(sc, sbdsp_tctosr(sc, sc->sc_otc),
SB_OUTPUT_RATE, &sc->sc_otc,
&sc->sc_omode);
}
if (ISSB16CLASS(sc) || ISJAZZ16(sc)) {
if (sc->encoding == AUDIO_ENCODING_ULAW &&
sc->sc_precision == 16) {
sc->sc_precision = 8;
return EINVAL; /* XXX what should we really do? */
}
}
/*
* XXX
* Should wait for chip to be idle.
*/
sc->sc_dmadir = SB_DMA_NONE;
return 0;
}
int
sbdsp_open(sc, dev, flags)
register struct sbdsp_softc *sc;
dev_t dev;
int flags;
{
DPRINTF(("sbdsp_open: sc=0x%x\n", sc));
if (sc->sc_open != 0 || sbdsp_reset(sc) != 0)
return ENXIO;
sc->sc_open = 1;
sc->sc_mintr = 0;
if (ISSBPROCLASS(sc) &&
sbdsp_wdsp(sc->sc_iobase, 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=0x%x\n", sc));
sc->sc_open = 0;
sbdsp_spkroff(sc);
sc->spkr_state = SPKR_OFF;
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)
register struct sbdsp_softc *sc;
{
register int iobase = sc->sc_iobase;
sc->sc_intr = 0;
if (sc->sc_dmadir != SB_DMA_NONE) {
isa_dmaabort(sc->sc_drq);
sc->sc_dmadir = SB_DMA_NONE;
}
sc->sc_last_hs_size = 0;
/*
* See SBK, section 11.3.
* We pulse a reset signal into the card.
* Gee, what a brilliant hardware design.
*/
outb(iobase + SBP_DSP_RESET, 1);
delay(10);
outb(iobase + SBP_DSP_RESET, 0);
delay(30);
if (sbdsp_rdsp(iobase) != SB_MAGIC)
return -1;
return 0;
}
int
sbdsp16_wait(iobase)
int iobase;
{
register int i;
for (i = SBDSP_NPOLL; --i >= 0; ) {
register u_char x;
x = inb(iobase + 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(int iobase, int v)
{
register int i;
for (i = SBDSP_NPOLL; --i >= 0; ) {
register u_char x;
x = inb(iobase + SBP_DSP_WSTAT);
delay(10);
if ((x & SB_DSP_BUSY) != 0)
continue;
outb(iobase + SBP_DSP_WRITE, v);
delay(10);
return 0;
}
++sberr.wdsp;
return -1;
}
/*
* Read a byte from the DSP, using polling.
*/
int
sbdsp_rdsp(int iobase)
{
register int i;
for (i = SBDSP_NPOLL; --i >= 0; ) {
register u_char x;
x = inb(iobase + SBP_DSP_RSTAT);
delay(10);
if ((x & SB_DSP_READY) == 0)
continue;
x = inb(iobase + 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->sc_iobase, 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->sc_iobase, 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.
*/
short
sbversion(sc)
struct sbdsp_softc *sc;
{
register int iobase = sc->sc_iobase;
short v;
if (sbdsp_wdsp(iobase, SB_DSP_VERSION) < 0)
return 0;
v = sbdsp_rdsp(iobase) << 8;
v |= sbdsp_rdsp(iobase);
return ((v >= 0) ? v : 0);
}
/*
* Halt a DMA in progress. A low-speed transfer can be
* resumed with sbdsp_contdma().
*/
int
sbdsp_haltdma(addr)
void *addr;
{
register struct sbdsp_softc *sc = addr;
DPRINTF(("sbdsp_haltdma: sc=0x%x\n", sc));
sbdsp_reset(sc);
return 0;
}
int
sbdsp_contdma(addr)
void *addr;
{
register struct sbdsp_softc *sc = addr;
DPRINTF(("sbdsp_contdma: sc=0x%x\n", sc));
/* XXX how do we reinitialize the DMA controller state? do we care? */
(void)sbdsp_wdsp(sc->sc_iobase, SB_DSP_CONT);
return(0);
}
int
sbdsp_setrate(sc, sr, isdac, ratep)
register struct sbdsp_softc *sc;
int sr;
int isdac;
int *ratep;
{
/*
* XXXX
* More checks here?
*/
if (sr < 5000 || sr > 44100)
return (EINVAL);
*ratep = sr;
return (0);
}
/*
* Convert a linear sampling rate into the DAC time constant.
* Set *mode to indicate the high/low-speed DMA operation.
* Because of limitations of the card, not all rates are possible.
* We return the time constant of the closest possible rate.
* The sampling rate limits are different for the DAC and ADC,
* so isdac indicates output, and !isdac indicates input.
*/
int
sbdsp_srtotc(sc, sr, isdac, tcp, modep)
register struct sbdsp_softc *sc;
int sr;
int isdac;
int *tcp, *modep;
{
int tc, realtc, mode;
/*
* Don't forget to compute which mode we'll be in based on whether
* we need to double the rate for stereo on SBPRO.
*/
if (sr == 0) {
tc = SB_LS_MIN;
mode = SB_ADAC_LS;
goto out;
}
tc = 256 - (1000000 / sr);
if (sc->sc_channels == 2 && ISSBPRO(sc))
/* compute based on 2x sample rate when needed */
realtc = 256 - ( 500000 / sr);
else
realtc = tc;
if (tc < SB_LS_MIN) {
tc = SB_LS_MIN;
mode = SB_ADAC_LS; /* NB: 2x minimum speed is still low
* speed mode. */
goto out;
} else if (isdac) {
if (realtc <= SB_DAC_LS_MAX)
mode = SB_ADAC_LS;
else {
mode = SB_ADAC_HS;
if (tc > SB_DAC_HS_MAX)
tc = SB_DAC_HS_MAX;
}
} else {
int adc_ls_max, adc_hs_max;
/* XXX use better rounding--compare distance to nearest tc on both
sides of requested speed */
if (ISSBPROCLASS(sc)) {
adc_ls_max = SBPRO_ADC_LS_MAX;
adc_hs_max = SBPRO_ADC_HS_MAX;
} else {
adc_ls_max = SBCLA_ADC_LS_MAX;
adc_hs_max = SBCLA_ADC_HS_MAX;
}
if (realtc <= adc_ls_max)
mode = SB_ADAC_LS;
else {
mode = SB_ADAC_HS;
if (tc > adc_hs_max)
tc = adc_hs_max;
}
}
out:
*tcp = tc;
*modep = mode;
return (0);
}
/*
* Convert a DAC time constant to a sampling rate.
* See SBK, section 12.
*/
int
sbdsp_tctosr(sc, tc)
register struct sbdsp_softc *sc;
int tc;
{
int adc;
if (ISSBPROCLASS(sc))
adc = SBPRO_ADC_HS_MAX;
else
adc = SBCLA_ADC_HS_MAX;
if (tc > adc)
tc = adc;
return (1000000 / (256 - tc));
}
int
sbdsp_set_timeconst(sc, tc)
register struct sbdsp_softc *sc;
int tc;
{
register int iobase;
/*
* A SBPro in stereo mode uses time constants at double the
* actual rate.
*/
if (ISSBPRO(sc) && sc->sc_channels == 2)
tc = 256 - ((256 - tc) / 2);
DPRINTF(("sbdsp_set_timeconst: sc=%p tc=%d\n", sc, tc));
iobase = sc->sc_iobase;
if (sbdsp_wdsp(iobase, SB_DSP_TIMECONST) < 0 ||
sbdsp_wdsp(iobase, tc) < 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;
{
register struct sbdsp_softc *sc = addr;
register int iobase;
#ifdef AUDIO_DEBUG
if (sbdspdebug > 1)
Dprintf("sbdsp_dma_input: cc=%d 0x%x (0x%x)\n", cc, intr, arg);
#endif
if (sc->sc_channels == 2 && (cc & 1)) {
DPRINTF(("sbdsp_dma_input: stereo input, odd bytecnt\n"));
return EIO;
}
iobase = sc->sc_iobase;
if (sc->sc_dmadir != SB_DMA_IN) {
if (ISSBPRO(sc)) {
if (sc->sc_channels == 2) {
if (ISJAZZ16(sc) && sc->sc_precision == 16) {
if (sbdsp_wdsp(iobase,
JAZZ16_RECORD_STEREO) < 0) {
goto badmode;
}
} else if (sbdsp_wdsp(iobase,
SB_DSP_RECORD_STEREO) < 0)
goto badmode;
sbdsp_mix_write(sc, SBP_INFILTER,
(sbdsp_mix_read(sc, SBP_INFILTER) &
~SBP_IFILTER_MASK) | SBP_FILTER_OFF);
} else {
if (ISJAZZ16(sc) && sc->sc_precision == 16) {
if (sbdsp_wdsp(iobase,
JAZZ16_RECORD_MONO) < 0)
{
goto badmode;
}
} else if (sbdsp_wdsp(iobase, SB_DSP_RECORD_MONO) < 0)
goto badmode;
sbdsp_mix_write(sc, SBP_INFILTER,
(sbdsp_mix_read(sc, SBP_INFILTER) &
~SBP_IFILTER_MASK) | sc->in_filter);
}
}
if (ISSB16CLASS(sc)) {
if (sbdsp_wdsp(iobase, SB_DSP16_INPUTRATE) < 0 ||
sbdsp_wdsp(iobase, sc->sc_irate >> 8) < 0 ||
sbdsp_wdsp(iobase, sc->sc_irate) < 0)
goto giveup;
} else
sbdsp_set_timeconst(sc, sc->sc_itc);
sc->sc_dmadir = SB_DMA_IN;
}
isa_dmastart(DMAMODE_READ, p, cc, sc->sc_drq);
sc->sc_intr = intr;
sc->sc_arg = arg;
sc->dmaflags = DMAMODE_READ;
sc->dmaaddr = p;
sc->dmacnt = cc; /* DMA controller is strange...? */
if ((ISSB16CLASS(sc) && sc->sc_precision == 16) ||
(ISJAZZ16(sc) && sc->sc_drq > 3))
cc >>= 1;
--cc;
if (ISSB16CLASS(sc)) {
if (sbdsp_wdsp(iobase, sc->sc_precision == 16 ? SB_DSP16_RDMA_16 :
SB_DSP16_RDMA_8) < 0 ||
sbdsp_wdsp(iobase, (sc->sc_precision == 16 ? 0x10 : 0x00) |
(sc->sc_channels == 2 ? 0x20 : 0x00)) < 0 ||
sbdsp16_wait(iobase) ||
sbdsp_wdsp(iobase, cc) < 0 ||
sbdsp_wdsp(iobase, cc >> 8) < 0) {
DPRINTF(("sbdsp_dma_input: SB16 DMA start failed\n"));
goto giveup;
}
} else if (sc->sc_imode == SB_ADAC_LS) {
if (sbdsp_wdsp(iobase, SB_DSP_RDMA) < 0 ||
sbdsp_wdsp(iobase, cc) < 0 ||
sbdsp_wdsp(iobase, cc >> 8) < 0) {
DPRINTF(("sbdsp_dma_input: LS DMA start failed\n"));
goto giveup;
}
} else {
if (cc != sc->sc_last_hs_size) {
if (sbdsp_wdsp(iobase, SB_DSP_BLOCKSIZE) < 0 ||
sbdsp_wdsp(iobase, cc) < 0 ||
sbdsp_wdsp(iobase, cc >> 8) < 0) {
DPRINTF(("sbdsp_dma_input: HS DMA start failed\n"));
goto giveup;
}
sc->sc_last_hs_size = cc;
}
if (sbdsp_wdsp(iobase, SB_DSP_HS_INPUT) < 0) {
DPRINTF(("sbdsp_dma_input: HS DMA restart failed\n"));
goto giveup;
}
}
return 0;
giveup:
sbdsp_reset(sc);
return EIO;
badmode:
DPRINTF(("sbdsp_dma_input: can't set %s mode\n",
sc->sc_channels == 2 ? "stereo" : "mono"));
return EIO;
}
int
sbdsp_dma_output(addr, p, cc, intr, arg)
void *addr;
void *p;
int cc;
void (*intr) __P((void *));
void *arg;
{
register struct sbdsp_softc *sc = addr;
register int iobase;
#ifdef AUDIO_DEBUG
if (sbdspdebug > 1)
Dprintf("sbdsp_dma_output: cc=%d 0x%x (0x%x)\n", cc, intr, arg);
#endif
if (sc->sc_channels == 2 && (cc & 1)) {
DPRINTF(("stereo playback odd bytes (%d)\n", cc));
return EIO;
}
iobase = sc->sc_iobase;
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) |
(sc->sc_channels == 2 ? SBP_PLAYMODE_STEREO : SBP_PLAYMODE_MONO));
if (ISJAZZ16(sc)) {
/* Yes, we write the record mode to set
16-bit playback mode. weird, huh? */
if (sc->sc_precision == 16) {
sbdsp_wdsp(iobase,
sc->sc_channels == 2 ?
JAZZ16_RECORD_STEREO :
JAZZ16_RECORD_MONO);
} else {
sbdsp_wdsp(iobase,
sc->sc_channels == 2 ?
SB_DSP_RECORD_STEREO :
SB_DSP_RECORD_MONO);
}
}
}
if (ISSB16CLASS(sc)) {
if (sbdsp_wdsp(iobase, SB_DSP16_OUTPUTRATE) < 0 ||
sbdsp_wdsp(iobase, sc->sc_orate >> 8) < 0 ||
sbdsp_wdsp(iobase, sc->sc_orate) < 0)
goto giveup;
} else
sbdsp_set_timeconst(sc, sc->sc_otc);
sc->sc_dmadir = SB_DMA_OUT;
}
isa_dmastart(DMAMODE_WRITE, p, cc, sc->sc_drq);
sc->sc_intr = intr;
sc->sc_arg = arg;
sc->dmaflags = DMAMODE_WRITE;
sc->dmaaddr = p;
sc->dmacnt = cc; /* a vagary of how DMA works, apparently. */
if ((ISSB16CLASS(sc) && sc->sc_precision == 16) ||
(ISJAZZ16(sc) && sc->sc_drq > 3))
cc >>= 1;
--cc;
if (ISSB16CLASS(sc)) {
if (sbdsp_wdsp(iobase, sc->sc_precision == 16 ? SB_DSP16_WDMA_16 :
SB_DSP16_WDMA_8) < 0 ||
sbdsp_wdsp(iobase, (sc->sc_precision == 16 ? 0x10 : 0x00) |
(sc->sc_channels == 2 ? 0x20 : 0x00)) < 0 ||
sbdsp16_wait(iobase) ||
sbdsp_wdsp(iobase, cc) < 0 ||
sbdsp_wdsp(iobase, cc >> 8) < 0) {
DPRINTF(("sbdsp_dma_output: SB16 DMA start failed\n"));
goto giveup;
}
} else if (sc->sc_omode == SB_ADAC_LS) {
if (sbdsp_wdsp(iobase, SB_DSP_WDMA) < 0 ||
sbdsp_wdsp(iobase, cc) < 0 ||
sbdsp_wdsp(iobase, cc >> 8) < 0) {
DPRINTF(("sbdsp_dma_output: LS DMA start failed\n"));
goto giveup;
}
} else {
if (cc != sc->sc_last_hs_size) {
if (sbdsp_wdsp(iobase, SB_DSP_BLOCKSIZE) < 0 ||
sbdsp_wdsp(iobase, cc) < 0 ||
sbdsp_wdsp(iobase, cc >> 8) < 0) {
DPRINTF(("sbdsp_dma_output: HS DMA start failed\n"));
goto giveup;
}
sc->sc_last_hs_size = cc;
}
if (sbdsp_wdsp(iobase, SB_DSP_HS_OUTPUT) < 0) {
DPRINTF(("sbdsp_dma_output: HS DMA restart failed\n"));
goto giveup;
}
}
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. The three modes are mutually
* exclusive so we know a priori which event has occurred.
*/
int
sbdsp_intr(arg)
void *arg;
{
register struct sbdsp_softc *sc = arg;
u_char x;
#ifdef AUDIO_DEBUG
if (sbdspdebug > 1)
Dprintf("sbdsp_intr: intr=0x%x\n", sc->sc_intr);
#endif
if (!isa_dmafinished(sc->sc_drq)) {
printf("sbdsp_intr: not finished\n");
return 0;
}
sc->sc_interrupts++;
/* clear interrupt */
#ifdef notyet
x = sbdsp_mix_read(sc, 0x82);
x = inb(sc->sc_iobase + 15);
#endif
x = inb(sc->sc_iobase + SBP_DSP_RSTAT);
delay(10);
#if 0
if (sc->sc_mintr != 0) {
x = sbdsp_rdsp(sc->sc_iobase);
(*sc->sc_mintr)(sc->sc_arg, x);
} else
#endif
if (sc->sc_intr != 0) {
isa_dmadone(sc->dmaflags, sc->dmaaddr, sc->dmacnt, sc->sc_drq);
(*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->sc_iobase, 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->sc_iobase, v) < 0)
++sberr.wmidi;
}
#endif
u_int
sbdsp_get_silence(encoding)
int encoding;
{
#define ULAW_SILENCE 0x7f
#define LINEAR_SILENCE 0
u_int auzero;
switch (encoding) {
case AUDIO_ENCODING_ULAW:
auzero = ULAW_SILENCE;
break;
case AUDIO_ENCODING_PCM16:
default:
auzero = LINEAR_SILENCE;
break;
}
return (auzero);
}
int
sbdsp_setfd(addr, flag)
void *addr;
int flag;
{
/* Can't do full-duplex */
return(ENOTTY);
}
int
sbdsp_mixer_set_port(addr, cp)
void *addr;
mixer_ctrl_t *cp;
{
register struct sbdsp_softc *sc = addr;
int src, gain;
DPRINTF(("sbdsp_mixer_set_port: port=%d num_channels=%d\n", cp->dev,
cp->un.value.num_channels));
if (!ISSBPROCLASS(sc))
return EINVAL;
/*
* Everything is a value except for SBPro BASS/TREBLE and
* RECORD_SOURCE
*/
switch (cp->dev) {
case SB_SPEAKER:
cp->dev = SB_MASTER_VOL;
case SB_MIC_PORT:
case SB_LINE_IN_PORT:
case SB_DAC_PORT:
case SB_FM_PORT:
case SB_CD_PORT:
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_PORT:
if (cp->un.value.num_channels != 1)
return EINVAL;
/* handle funny microphone gain */
gain = SBP_AGAIN_TO_MICGAIN(cp->un.value.level[AUDIO_MIXER_LEVEL_MONO]);
break;
case SB_LINE_IN_PORT:
case SB_DAC_PORT:
case SB_FM_PORT:
case SB_CD_PORT:
case SB_MASTER_VOL:
switch (cp->un.value.num_channels) {
case 1:
gain = sbdsp_mono_vol(SBP_AGAIN_TO_SBGAIN(cp->un.value.level[AUDIO_MIXER_LEVEL_MONO]));
break;
case 2:
gain = sbdsp_stereo_vol(SBP_AGAIN_TO_SBGAIN(cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT]),
SBP_AGAIN_TO_SBGAIN(cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT]));
break;
default:
return EINVAL;
}
break;
default:
return EINVAL;
}
switch (cp->dev) {
case SB_MIC_PORT:
src = SBP_MIC_VOL;
break;
case SB_MASTER_VOL:
src = SBP_MASTER_VOL;
break;
case SB_LINE_IN_PORT:
src = SBP_LINE_VOL;
break;
case SB_DAC_PORT:
src = SBP_DAC_VOL;
break;
case SB_FM_PORT:
src = SBP_FM_VOL;
break;
case SB_CD_PORT:
src = SBP_CD_VOL;
break;
default:
return EINVAL;
}
sbdsp_mix_write(sc, src, gain);
sc->gain[cp->dev] = gain;
break;
case SB_TREBLE:
case SB_BASS:
case SB_RECORD_SOURCE:
if (cp->type != AUDIO_MIXER_ENUM)
return EINVAL;
switch (cp->dev) {
case SB_TREBLE:
return sbdsp_set_ifilter(addr, cp->un.ord ? SBP_TREBLE_EQ : 0);
case SB_BASS:
return sbdsp_set_ifilter(addr, cp->un.ord ? SBP_BASS_EQ : 0);
case SB_RECORD_SOURCE:
return sbdsp_set_in_port(addr, cp->un.ord);
}
break;
default:
return EINVAL;
}
return (0);
}
int
sbdsp_mixer_get_port(addr, cp)
void *addr;
mixer_ctrl_t *cp;
{
register struct sbdsp_softc *sc = addr;
int gain;
DPRINTF(("sbdsp_mixer_get_port: port=%d", cp->dev));
if (!ISSBPROCLASS(sc))
return EINVAL;
switch (cp->dev) {
case SB_SPEAKER:
cp->dev = SB_MASTER_VOL;
case SB_MIC_PORT:
case SB_LINE_IN_PORT:
case SB_DAC_PORT:
case SB_FM_PORT:
case SB_CD_PORT:
case SB_MASTER_VOL:
gain = sc->gain[cp->dev];
switch (cp->dev) {
case SB_MIC_PORT:
if (cp->un.value.num_channels != 1)
return EINVAL;
cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = SBP_MICGAIN_TO_AGAIN(gain);
break;
case SB_LINE_IN_PORT:
case SB_DAC_PORT:
case SB_FM_PORT:
case SB_CD_PORT:
case SB_MASTER_VOL:
switch (cp->un.value.num_channels) {
case 1:
cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = SBP_SBGAIN_TO_AGAIN(gain);
break;
case 2:
cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = SBP_LEFTGAIN(gain);
cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = SBP_RIGHTGAIN(gain);
break;
default:
return EINVAL;
}
break;
}
break;
case SB_TREBLE:
case SB_BASS:
case SB_RECORD_SOURCE:
switch (cp->dev) {
case SB_TREBLE:
cp->un.ord = sbdsp_get_ifilter(addr) == SBP_TREBLE_EQ;
return 0;
case SB_BASS:
cp->un.ord = sbdsp_get_ifilter(addr) == SBP_BASS_EQ;
return 0;
case SB_RECORD_SOURCE:
cp->un.ord = sbdsp_get_in_port(addr);
return 0;
}
break;
default:
return EINVAL;
}
return (0);
}
int
sbdsp_mixer_query_devinfo(addr, dip)
void *addr;
register mixer_devinfo_t *dip;
{
register struct sbdsp_softc *sc = addr;
DPRINTF(("sbdsp_mixer_query_devinfo: index=%d\n", dip->index));
switch (dip->index) {
case SB_MIC_PORT:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = SB_INPUT_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_SPEAKER:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = SB_OUTPUT_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNspeaker);
dip->un.v.num_channels = 1;
strcpy(dip->un.v.units.name, AudioNvolume);
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_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 (ISSBPROCLASS(sc)) {
switch (dip->index) {
case SB_LINE_IN_PORT:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = SB_INPUT_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_DAC_PORT:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = SB_INPUT_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNdac);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
return 0;
case SB_CD_PORT:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = SB_INPUT_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNcd);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
return 0;
case SB_FM_PORT:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = SB_INPUT_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNfmsynth);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
return 0;
case SB_MASTER_VOL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = SB_OUTPUT_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNvolume);
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->type = AUDIO_MIXER_ENUM;
dip->prev = AUDIO_MIXER_LAST;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNsource);
dip->un.e.num_mem = 3;
strcpy(dip->un.e.member[0].label.name, AudioNmicrophone);
dip->un.e.member[0].ord = SB_MIC_PORT;
strcpy(dip->un.e.member[1].label.name, AudioNcd);
dip->un.e.member[1].ord = SB_CD_PORT;
strcpy(dip->un.e.member[2].label.name, AudioNline);
dip->un.e.member[2].ord = SB_LINE_IN_PORT;
return 0;
case SB_BASS:
dip->type = AUDIO_MIXER_ENUM;
dip->mixer_class = SB_INPUT_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNbass);
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->type = AUDIO_MIXER_ENUM;
dip->mixer_class = SB_INPUT_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNtreble);
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;
}
}
return ENXIO;
}
Reference in New Issue View Git Blame Copy Permalink