Aren/NetBSD
1
0
Fork 0
Code Issues Pull Requests Packages Projects Releases Wiki Activity
121f6570d7
NetBSD/sys/dev/isa/sbdsp.c
mycroft 7847c9efee Various: 
* Snap the sample rate when setting it, and remember only the time constant.
* Set the time constant when changing between play/record.
* Always return the actual sample rate with AUDIO_GETINFO.
1996-02-16 10:10:21 +00:00

1452 lines
32 KiB
C
Raw Blame History

/* $NetBSD: sbdsp.c,v 1.16 1996/02/16 10:10:21 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/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;
/*
* 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 */
#ifdef AUDIO_DEBUG
void
sb_printsc(struct sbdsp_softc *sc)
{
int i;
printf("open %d dmachan %d iobase %x\n",
sc->sc_open, sc->sc_drq, sc->sc_iobase);
printf("itc %d imode %d otc %d omode %d encoding %x\n",
sc->sc_itc, sc->sc_imode, 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("chans %x intr %x arg %x\n",
sc->sc_chans, 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;
{
register int iobase = sc->sc_iobase;
if (sbdsp_reset(sc) < 0) {
DPRINTF(("sbdsp: couldn't reset card\n"));
return 0;
}
sc->sc_model = sbversion(sc);
return 1;
}
/*
* Attach hardware to driver, attach hardware driver to audio
* pseudo-device driver .
*/
void
sbdsp_attach(sc)
struct sbdsp_softc *sc;
{
register int iobase = sc->sc_iobase;
/* Set defaults */
if (ISSBPROCLASS(sc))
sc->sc_itc = sc->sc_otc = SBPRO_ADC_HS_MAX;
else
sc->sc_itc = sc->sc_otc = SBCLA_ADC_HS_MAX;
sc->sc_chans = 1;
sc->encoding = AUDIO_ENCODING_LINEAR;
(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, SBP_MAXVOL));
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);
}
printf(": dsp v%d.%02d\n",
SBVER_MAJOR(sc->sc_model), SBVER_MINOR(sc->sc_model));
}
/*
* 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;
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;
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;
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;
return (sbdsp_tctosr(sc, sc->sc_otc));
}
int
sbdsp_query_encoding(addr, fp)
void *addr;
struct audio_encoding *fp;
{
register struct sbdsp_softc *sc = addr;
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, enc)
void *addr;
u_int enc;
{
register struct sbdsp_softc *sc = addr;
switch(enc){
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, prec)
void *addr;
u_int prec;
{
if (prec != 8)
return (EINVAL);
return (0);
}
int
sbdsp_get_precision(addr)
void *addr;
{
return (8);
}
int
sbdsp_set_channels(addr, chans)
void *addr;
int chans;
{
register struct sbdsp_softc *sc = addr;
if (ISSBPROCLASS(sc)) {
if (chans != 1 && chans != 2)
return (EINVAL);
sc->sc_chans = chans;
#if 0
if (rval = sbdsp_set_in_sr_real(addr, sc->sc_irate))
return rval;
#endif
sbdsp_mix_write(sc, SBP_STEREO,
(sbdsp_mix_read(sc, SBP_STEREO) & ~SBP_PLAYMODE_MASK) |
(chans == 2 ? SBP_PLAYMODE_STEREO : SBP_PLAYMODE_MONO));
/* recording channels needs to be done right when we start
DMA recording. Just record number of channels for now
and set stereo when ready. */
} else {
if (chans != 1)
return (EINVAL);
sc->sc_chans = chans;
}
return (0);
}
int
sbdsp_get_channels(addr)
void *addr;
{
register struct sbdsp_softc *sc = addr;
#if 0
/* recording stereo may frob the mixer output */
if (ISSBPROCLASS(sc)) {
if ((sbdsp_mix_read(sc, SBP_STEREO) & SBP_PLAYMODE_MASK) == SBP_PLAYMODE_STEREO)
sc->sc_chans = 2;
else
sc->sc_chans = 1;
} else
sc->sc_chans = 1;
#endif
return (sc->sc_chans);
}
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_FILTER_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 ((sc->sc_otc > SB_8K || sc->sc_itc > SB_8K) &&
(blk > NBPG/2 || blk < NBPG/4))
blk = NBPG/2;
/* don't try to DMA too much at once, though. */
if (blk > NBPG)
blk = NBPG;
if (sc->sc_chans == 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) */
/*
* 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;
}
/*
* 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);
}
/*
* 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, mode;
if (sr == 0) {
tc = SB_LS_MIN;
mode = SB_ADAC_LS;
goto out;
}
tc = 256 - (1000000 / sr);
if (tc < SB_LS_MIN) {
tc = SB_LS_MIN;
mode = SB_ADAC_LS;
goto out;
} else if (isdac) {
if (tc <= 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 (tc <= 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_tc(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_chans == 2)
tc = 256 - ((256 - tc) / 2);
DPRINTF(("sbdsp_set_tc: 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)();
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_chans == 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 (ISSBPROCLASS(sc)) {
if (sc->sc_chans == 2) {
if (sbdsp_wdsp(iobase, SB_DSP_RECORD_STEREO) < 0)
goto badmode;
sbdsp_mix_write(sc, SBP_INFILTER,
sbdsp_mix_read(sc, SBP_INFILTER) | SBP_FILTER_OFF);
} else {
if (sbdsp_wdsp(iobase, SB_DSP_RECORD_MONO) < 0)
goto badmode;
sbdsp_mix_write(sc, SBP_INFILTER, sc->sc_itc > SB_8K ?
sbdsp_mix_read(sc, SBP_INFILTER) | SBP_FILTER_OFF :
sbdsp_mix_read(sc, SBP_INFILTER) & ~SBP_FILTER_MASK);
}
}
sbdsp_set_tc(sc, sc->sc_itc);
sc->sc_dmadir = SB_DMA_IN;
}
isa_dmastart(B_READ, p, cc, sc->sc_drq);
sc->sc_intr = intr;
sc->sc_arg = arg;
sc->dmaflags = B_READ;
sc->dmaaddr = p;
sc->dmacnt = --cc; /* DMA controller is strange...? */
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_chans == 2 ? "stereo" : "mono"));
return EIO;
}
int
sbdsp_dma_output(addr, p, cc, intr, arg)
void *addr;
void *p;
int cc;
void (*intr)();
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_chans == 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 (ISSBPROCLASS(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_chans == 2 ? SBP_PLAYMODE_STEREO : SBP_PLAYMODE_MONO));
}
sbdsp_set_tc(sc, sc->sc_otc);
sc->sc_dmadir = SB_DMA_OUT;
}
isa_dmastart(B_WRITE, p, cc, sc->sc_drq);
sc->sc_intr = intr;
sc->sc_arg = arg;
sc->dmaflags = B_WRITE;
sc->dmaaddr = p;
sc->dmacnt = --cc; /* a vagary of how DMA works, apparently. */
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
sc->sc_interrupts++;
/* clear interrupt */
x = inb(sc->sc_iobase + SBP_DSP_RSTAT);
delay(10);
#if 0
if ((x & SB_DSP_READY) == 0) {
printf("sbdsp_intr: still busy\n");
return 0;
}
#endif
#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) {
/*
* The SBPro used to develop and test this driver often
* generated dma underruns--it interrupted to signal
* completion of the DMA input recording block, but the
* ISA DMA controller didn't think the channel was
* finished. Maybe this is just a bus speed issue, I dunno,
* but it seems strange and leads to channel-flipping with
* stereo recording. Sigh.
*/
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(enc)
int enc;
{
#define ULAW_SILENCE 0x7f
#define LINEAR_SILENCE 0
u_int auzero;
switch (enc) {
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 error = 0;
int src, gain;
int left, right;
DPRINTF(("sbdsp_mixer_set_port: port=%d num_channels=%d\n", cp->dev, cp->un.value.num_channels));
/*
* Everything is a value except for SBPro special OUTPUT_MODE and
* RECORD_SOURCE
*/
if (cp->type != AUDIO_MIXER_VALUE) {
if (!ISSBPROCLASS(sc) || (cp->dev != SB_OUTPUT_MODE &&
cp->dev != SB_RECORD_SOURCE))
return EINVAL;
}
else {
/*
* 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.
*/
if (cp->un.value.num_channels == 2) {
left = cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT];
right = cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT];
}
else
left = right = cp->un.value.level[AUDIO_MIXER_LEVEL_MONO];
}
if (ISSBPROCLASS(sc)) {
/* The _PORT things are all signal inputs to the mixer.
* Here we are tweaking their mixing level.
*
* We can also tweak the output stage volume (MASTER_VOL)
*/
gain = sbdsp_stereo_vol(SBP_AGAIN_TO_SBGAIN(left),
SBP_AGAIN_TO_SBGAIN(right));
switch(cp->dev) {
case SB_MIC_PORT:
src = SBP_MIC_VOL;
if (cp->un.value.num_channels != 1)
error = EINVAL;
else
/* handle funny microphone gain */
gain = SBP_AGAIN_TO_MICGAIN(left);
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;
case SB_SPEAKER:
cp->dev = SB_MASTER_VOL;
case SB_MASTER_VOL:
src = SBP_MASTER_VOL;
break;
#if 0
case SB_OUTPUT_MODE:
if (cp->type == AUDIO_MIXER_ENUM)
return sbdsp_set_channels(addr, cp->un.ord);
/* fall through...carefully! */
#endif
case SB_RECORD_SOURCE:
if (cp->type == AUDIO_MIXER_ENUM)
return sbdsp_set_in_port(addr, cp->un.ord);
/* else fall through: bad input */
case SB_TREBLE:
case SB_BASS:
default:
error = EINVAL;
break;
}
if (!error)
sbdsp_mix_write(sc, src, gain);
}
else if (cp->dev != SB_MIC_PORT &&
cp->dev != SB_SPEAKER)
error = EINVAL;
if (!error)
sc->gain[cp->dev] = gain;
return(error);
}
int
sbdsp_mixer_get_port(addr, cp)
void *addr;
mixer_ctrl_t *cp;
{
register struct sbdsp_softc *sc = addr;
int error = 0;
int done = 0;
DPRINTF(("sbdsp_mixer_get_port: port=%d", cp->dev));
if (ISSBPROCLASS(sc))
switch(cp->dev) {
case SB_MIC_PORT:
if (cp->un.value.num_channels == 1) {
cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] =
SBP_MICGAIN_TO_AGAIN(sc->gain[cp->dev]);
return 0;
}
else
return EINVAL;
break;
case SB_LINE_IN_PORT:
case SB_DAC_PORT:
case SB_FM_PORT:
case SB_CD_PORT:
case SB_MASTER_VOL:
break;
case SB_SPEAKER:
cp->dev = SB_MASTER_VOL;
break;
default:
error = EINVAL;
break;
}
else {
if (cp->un.value.num_channels != 1) /* no stereo on SB classic */
error = EINVAL;
else
switch(cp->dev) {
case SB_MIC_PORT:
break;
case SB_SPEAKER:
break;
default:
error = EINVAL;
break;
}
}
if (error == 0) {
if (cp->un.value.num_channels == 1) {
cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] =
SBP_SBGAIN_TO_AGAIN(sc->gain[cp->dev]);
}
else if (cp->un.value.num_channels == 2) {
cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT] =
SBP_LEFTGAIN(sc->gain[cp->dev]);
cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] =
SBP_RIGHTGAIN(sc->gain[cp->dev]);
} else
return EINVAL;
}
return(error);
}
int
sbdsp_mixer_query_devinfo(addr, dip)
void *addr;
register mixer_devinfo_t *dip;
{
register struct sbdsp_softc *sc = addr;
int done = 0;
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);
done = 1;
break;
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);
done = 1;
break;
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);
done = 1;
break;
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);
done = 1;
break;
}
if (!done) {
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);
break;
case SB_DAC_PORT:
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, AudioNdac);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
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);
break;
case SB_FM_PORT:
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, AudioNfmsynth);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case SB_MASTER_VOL:
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = SB_OUTPUT_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = /*TREBLE, BASS not handled, nor is SB_OUTPUT_MODE*/SB_RECORD_SOURCE;
strcpy(dip->label.name, AudioNvolume);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
#if 0
case SB_OUTPUT_MODE:
dip->mixer_class = SB_OUTPUT_CLASS;
dip->type = AUDIO_MIXER_ENUM;
dip->prev = SB_MASTER_VOL;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNmode);
dip->un.e.num_mem = 2;
strcpy(dip->un.e.member[0].label.name, AudioNmono);
dip->un.e.member[0].ord = 1; /* nchans */
strcpy(dip->un.e.member[1].label.name, AudioNstereo);
dip->un.e.member[1].ord = 2; /* nchans */
break;
#endif
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;
break;
case SB_BASS:
case SB_TREBLE:
default:
return ENXIO;
/*NOTREACHED*/
}
else
return ENXIO;
}
DPRINTF(("AUDIO_MIXER_DEVINFO: name=%s\n", dip->label.name));
return 0;
}
Reference in New Issue View Git Blame Copy Permalink