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NetBSD/sys/dev/isa/sbdsp.c
jmcneill 8a962f23f2 Merge jmcneill-audiomp3 branch, which is derived from ad-audiomp2. From 
the original ad-audiomp branch notes:

  Add MP locking to the audio drivers.

  Making the audio drivers MP safe is necessary before efforts
  can be made to make the VM system MP safe.

  The are two locks per device instance, an ISR lock and
  a character device lock. The ISR lock replaces calls to
  splaudio()/splx(), and will be held across calls to device
  methods which were called at splaudio() before (e.g.
  trigger_output). The character device lock is held across
  calls to nearly all of the methods, excluding some only
  used for initialization, e.g. get_locks.

Welcome to 5.99.57.
2011-11-23 23:07:28 +00:00

2498 lines
61 KiB
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/* $NetBSD: sbdsp.c,v 1.135 2011/11/23 23:07:32 jmcneill Exp $ */
/*-
* Copyright (c) 1999, 2008 The NetBSD Foundation, Inc.
* All rights reserved.
*
* This code is derived from software contributed to The NetBSD Foundation
* by Charles M. Hannum.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
*
* THIS SOFTWARE IS PROVIDED BY THE NETBSD FOUNDATION, INC. AND CONTRIBUTORS
* ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED
* TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
* PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION OR CONTRIBUTORS
* BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
* CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
* SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
* INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
* CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
* ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
* POSSIBILITY OF SUCH DAMAGE.
*/
/*
* Copyright (c) 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/cdefs.h>
__KERNEL_RCSID(0, "$NetBSD: sbdsp.c,v 1.135 2011/11/23 23:07:32 jmcneill Exp $");
#include "midi.h"
#include "mpu.h"
#include <sys/param.h>
#include <sys/systm.h>
#include <sys/kernel.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 <sys/malloc.h>
#include <sys/cpu.h>
#include <sys/intr.h>
#include <sys/bus.h>
#include <sys/audioio.h>
#include <dev/audio_if.h>
#include <dev/midi_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
#define DPRINTFN(n,x) if (sbdspdebug >= (n)) printf x
int sbdspdebug = 0;
#else
#define DPRINTF(x)
#define DPRINTFN(n,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 halt, cont;
u_char cmdchan;
};
static struct sbmode sbpmodes[] = {
{ SB_1, 1, 8, 4000,22727,SB_DSP_WDMA ,SB_DSP_HALT ,SB_DSP_CONT, 0, },
{ SB_20, 1, 8, 4000,22727,SB_DSP_WDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT, 0, },
{ SB_2x, 1, 8,22727,45454,SB_DSP_HS_OUTPUT,SB_DSP_HALT ,SB_DSP_CONT, 0, },
{ SB_2x, 1, 8, 4000,22727,SB_DSP_WDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT, 0, },
{ SB_PRO, 1, 8,22727,45454,SB_DSP_HS_OUTPUT,SB_DSP_HALT ,SB_DSP_CONT, 0, },
{ SB_PRO, 1, 8, 4000,22727,SB_DSP_WDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT, 0, },
{ SB_PRO, 2, 8,11025,22727,SB_DSP_HS_OUTPUT,SB_DSP_HALT ,SB_DSP_CONT, 0, },
/* Yes, we write the record mode to set 16-bit playback mode. weird, huh? */
{ SB_JAZZ,1, 8,22727,45454,SB_DSP_HS_OUTPUT,SB_DSP_HALT ,SB_DSP_CONT ,SB_DSP_RECORD_MONO },
{ SB_JAZZ,1, 8, 4000,22727,SB_DSP_WDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT ,SB_DSP_RECORD_MONO },
{ SB_JAZZ,2, 8,11025,22727,SB_DSP_HS_OUTPUT,SB_DSP_HALT ,SB_DSP_CONT ,SB_DSP_RECORD_STEREO },
{ SB_JAZZ,1,16,22727,45454,SB_DSP_HS_OUTPUT,SB_DSP_HALT ,SB_DSP_CONT ,JAZZ16_RECORD_MONO },
{ SB_JAZZ,1,16, 4000,22727,SB_DSP_WDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT ,JAZZ16_RECORD_MONO },
{ SB_JAZZ,2,16,11025,22727,SB_DSP_HS_OUTPUT,SB_DSP_HALT ,SB_DSP_CONT ,JAZZ16_RECORD_STEREO },
{ SB_16, 1, 8, 5000,49000,SB_DSP16_WDMA_8 ,SB_DSP_HALT ,SB_DSP_CONT, 0, },
{ SB_16, 2, 8, 5000,49000,SB_DSP16_WDMA_8 ,SB_DSP_HALT ,SB_DSP_CONT, 0, },
#define PLAY16 15 /* must be the index of the next entry in the table */
{ SB_16, 1,16, 5000,49000,SB_DSP16_WDMA_16,SB_DSP16_HALT,SB_DSP16_CONT, 0, },
{ SB_16, 2,16, 5000,49000,SB_DSP16_WDMA_16,SB_DSP16_HALT,SB_DSP16_CONT, 0, },
{ .model = -1 }
};
static struct sbmode sbrmodes[] = {
{ SB_1, 1, 8, 4000,12987,SB_DSP_RDMA ,SB_DSP_HALT ,SB_DSP_CONT, 0, },
{ SB_20, 1, 8, 4000,12987,SB_DSP_RDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT, 0, },
{ SB_2x, 1, 8,12987,14925,SB_DSP_HS_INPUT ,SB_DSP_HALT ,SB_DSP_CONT, 0, },
{ SB_2x, 1, 8, 4000,12987,SB_DSP_RDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT, 0, },
{ SB_PRO, 1, 8,22727,45454,SB_DSP_HS_INPUT ,SB_DSP_HALT ,SB_DSP_CONT ,SB_DSP_RECORD_MONO },
{ SB_PRO, 1, 8, 4000,22727,SB_DSP_RDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT ,SB_DSP_RECORD_MONO },
{ SB_PRO, 2, 8,11025,22727,SB_DSP_HS_INPUT ,SB_DSP_HALT ,SB_DSP_CONT ,SB_DSP_RECORD_STEREO },
{ SB_JAZZ,1, 8,22727,45454,SB_DSP_HS_INPUT ,SB_DSP_HALT ,SB_DSP_CONT ,SB_DSP_RECORD_MONO },
{ SB_JAZZ,1, 8, 4000,22727,SB_DSP_RDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT ,SB_DSP_RECORD_MONO },
{ SB_JAZZ,2, 8,11025,22727,SB_DSP_HS_INPUT ,SB_DSP_HALT ,SB_DSP_CONT ,SB_DSP_RECORD_STEREO },
{ SB_JAZZ,1,16,22727,45454,SB_DSP_HS_INPUT ,SB_DSP_HALT ,SB_DSP_CONT ,JAZZ16_RECORD_MONO },
{ SB_JAZZ,1,16, 4000,22727,SB_DSP_RDMA_LOOP,SB_DSP_HALT ,SB_DSP_CONT ,JAZZ16_RECORD_MONO },
{ SB_JAZZ,2,16,11025,22727,SB_DSP_HS_INPUT ,SB_DSP_HALT ,SB_DSP_CONT ,JAZZ16_RECORD_STEREO },
{ SB_16, 1, 8, 5000,49000,SB_DSP16_RDMA_8 ,SB_DSP_HALT ,SB_DSP_CONT, 0, },
{ SB_16, 2, 8, 5000,49000,SB_DSP16_RDMA_8 ,SB_DSP_HALT ,SB_DSP_CONT, 0, },
{ SB_16, 1,16, 5000,49000,SB_DSP16_RDMA_16,SB_DSP16_HALT,SB_DSP16_CONT, 0, },
{ SB_16, 2,16, 5000,49000,SB_DSP16_RDMA_16,SB_DSP16_HALT,SB_DSP16_CONT, 0, },
{ .model = -1 }
};
void sbversion(struct sbdsp_softc *);
void sbdsp_jazz16_probe(struct sbdsp_softc *);
void sbdsp_set_mixer_gain(struct sbdsp_softc *, int);
void sbdsp_pause(struct sbdsp_softc *);
int sbdsp_set_timeconst(struct sbdsp_softc *, int);
int sbdsp16_set_rate(struct sbdsp_softc *, int, int);
int sbdsp_set_in_ports(struct sbdsp_softc *, int);
void sbdsp_set_ifilter(void *, int);
int sbdsp_get_ifilter(void *);
int sbdsp_block_output(void *);
int sbdsp_block_input(void *);
static int sbdsp_adjust(int, int);
int sbdsp_midi_intr(void *);
static bool sbdsp_resume(device_t, const pmf_qual_t *);
#ifdef AUDIO_DEBUG
void sb_printsc(struct sbdsp_softc *);
void
sb_printsc(struct sbdsp_softc *sc)
{
int i;
printf("open %d DMA chan %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 intr16 %p\n",
sc->sc_intr8, sc->sc_intr16);
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(struct sbdsp_softc *sc, cfdata_t match)
{
if (sbdsp_reset(sc) < 0) {
DPRINTF(("sbdsp: couldn't reset card\n"));
return 0;
}
/* if flags set, go and probe the jazz16 stuff */
if (match->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(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;
bus_space_handle_t ioh;
iot = sc->sc_iot;
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) ||
(sc->sc_drq16 >= 0 &&
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(struct sbdsp_softc *sc)
{
int i, error;
u_int v;
mutex_enter(&sc->sc_lock);
mutex_spin_enter(&sc->sc_intr_lock);
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 / 2);
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 */
}
mutex_spin_exit(&sc->sc_intr_lock);
mutex_exit(&sc->sc_lock);
aprint_naive("\n");
aprint_normal(": 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;
if (sc->sc_drq8 != -1) {
sc->sc_drq8_maxsize = isa_dmamaxsize(sc->sc_ic,
sc->sc_drq8);
error = isa_dmamap_create(sc->sc_ic, sc->sc_drq8,
sc->sc_drq8_maxsize, BUS_DMA_WAITOK|BUS_DMA_ALLOCNOW);
if (error) {
aprint_error_dev(sc->sc_dev,
"can't create map for drq %d\n", sc->sc_drq8);
return;
}
}
if (sc->sc_drq16 != -1 && sc->sc_drq16 != sc->sc_drq8) {
sc->sc_drq16_maxsize = isa_dmamaxsize(sc->sc_ic,
sc->sc_drq16);
error = isa_dmamap_create(sc->sc_ic, sc->sc_drq16,
sc->sc_drq16_maxsize, BUS_DMA_WAITOK|BUS_DMA_ALLOCNOW);
if (error) {
aprint_error_dev(sc->sc_dev,
"can't create map for drq %d\n", sc->sc_drq16);
isa_dmamap_destroy(sc->sc_ic, sc->sc_drq8);
return;
}
}
if (!pmf_device_register(sc->sc_dev, NULL, sbdsp_resume))
aprint_error_dev(sc->sc_dev, "couldn't establish power handler\n");
}
static bool
sbdsp_resume(device_t dv, const pmf_qual_t *qual)
{
struct sbdsp_softc *sc = device_private(dv);
/* Reset the mixer. */
mutex_enter(&sc->sc_lock);
mutex_spin_enter(&sc->sc_intr_lock);
sbdsp_mix_write(sc, SBP_MIX_RESET, SBP_MIX_RESET);
mutex_spin_exit(&sc->sc_intr_lock);
mutex_exit(&sc->sc_lock);
return true;
}
void
sbdsp_mix_write(struct sbdsp_softc *sc, int mixerport, int val)
{
bus_space_tag_t iot;
bus_space_handle_t ioh;
iot = sc->sc_iot;
ioh = sc->sc_ioh;
bus_space_write_1(iot, ioh, SBP_MIXER_ADDR, mixerport);
delay(20);
bus_space_write_1(iot, ioh, SBP_MIXER_DATA, val);
delay(30);
}
int
sbdsp_mix_read(struct sbdsp_softc *sc, int mixerport)
{
bus_space_tag_t iot;
bus_space_handle_t ioh;
int val;
iot = sc->sc_iot;
ioh = sc->sc_ioh;
bus_space_write_1(iot, ioh, SBP_MIXER_ADDR, mixerport);
delay(20);
val = bus_space_read_1(iot, ioh, SBP_MIXER_DATA);
delay(30);
return val;
}
/*
* Various routines to interface to higher level audio driver
*/
int
sbdsp_query_encoding(void *addr, struct audio_encoding *fp)
{
struct sbdsp_softc *sc;
int emul;
sc = addr;
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(
void *addr,
int setmode, int usemode,
audio_params_t *play, audio_params_t *rec,
stream_filter_list_t *pfil, stream_filter_list_t *rfil)
{
struct sbdsp_softc *sc;
struct sbmode *m;
u_int rate, tc, bmode;
stream_filter_factory_t *swcode;
int model;
int chan;
struct audio_params *p;
audio_params_t hw;
stream_filter_list_t *fil;
int mode;
sc = addr;
if (sc->sc_open == SB_OPEN_MIDI)
return EBUSY;
/* Later models work like SB16. */
model = min(sc->sc_model, SB_16);
/*
* Prior to the SB16, we have only one clock, so make the sample
* rates match.
*/
if (!ISSB16CLASS(sc) &&
play->sample_rate != rec->sample_rate &&
usemode == (AUMODE_PLAY | AUMODE_RECORD)) {
if (setmode == AUMODE_PLAY) {
rec->sample_rate = play->sample_rate;
setmode |= AUMODE_RECORD;
} else if (setmode == AUMODE_RECORD) {
play->sample_rate = rec->sample_rate;
setmode |= AUMODE_PLAY;
} else
return EINVAL;
}
/* 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 = NULL;
fil = mode == AUMODE_PLAY ? pfil : rfil;
hw = *p;
tc = 1;
bmode = -1;
if (model == SB_16) {
switch (p->encoding) {
case AUDIO_ENCODING_SLINEAR_BE:
if (p->precision == 16) {
hw.encoding = AUDIO_ENCODING_SLINEAR_LE;
swcode = swap_bytes;
}
/* fall into */
case AUDIO_ENCODING_SLINEAR_LE:
bmode = SB_BMODE_SIGNED;
break;
case AUDIO_ENCODING_ULINEAR_BE:
if (p->precision == 16) {
hw.encoding = AUDIO_ENCODING_ULINEAR_LE;
swcode = swap_bytes;
}
/* fall into */
case AUDIO_ENCODING_ULINEAR_LE:
bmode = SB_BMODE_UNSIGNED;
break;
case AUDIO_ENCODING_ULAW:
hw.encoding = AUDIO_ENCODING_ULINEAR_LE;
if (mode == AUMODE_PLAY) {
hw.precision = hw.validbits = 16;
swcode = mulaw_to_linear16;
m = &sbpmodes[PLAY16];
} else
swcode = linear8_to_mulaw;
bmode = SB_BMODE_UNSIGNED;
break;
case AUDIO_ENCODING_ALAW:
hw.encoding = AUDIO_ENCODING_ULINEAR_LE;
if (mode == AUMODE_PLAY) {
hw.precision = hw.validbits = 16;
swcode = alaw_to_linear16;
m = &sbpmodes[PLAY16];
} else
swcode = linear8_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:
hw.encoding = AUDIO_ENCODING_SLINEAR_LE;
swcode = change_sign16;
break;
case AUDIO_ENCODING_SLINEAR_BE:
hw.encoding = AUDIO_ENCODING_SLINEAR_LE;
swcode = swap_bytes;
break;
case AUDIO_ENCODING_ULINEAR_BE:
hw.encoding = AUDIO_ENCODING_SLINEAR_LE;
swcode = swap_bytes_change_sign16;
break;
case AUDIO_ENCODING_ULAW:
hw.encoding = AUDIO_ENCODING_ULINEAR_LE;
swcode = mode == AUMODE_PLAY ?
mulaw_to_linear8 : linear8_to_mulaw;
break;
case AUDIO_ENCODING_ALAW:
hw.encoding = AUDIO_ENCODING_ULINEAR_LE;
swcode = mode == AUMODE_PLAY ?
alaw_to_linear8 : linear8_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;
hw.sample_rate = p->sample_rate;
} else {
switch (p->encoding) {
case AUDIO_ENCODING_SLINEAR_BE:
case AUDIO_ENCODING_SLINEAR_LE:
hw.encoding = AUDIO_ENCODING_ULINEAR_LE;
swcode = change_sign8;
break;
case AUDIO_ENCODING_ULINEAR_BE:
case AUDIO_ENCODING_ULINEAR_LE:
break;
case AUDIO_ENCODING_ULAW:
hw.encoding = AUDIO_ENCODING_ULINEAR_LE;
swcode = mode == AUMODE_PLAY ?
mulaw_to_linear8 : linear8_to_mulaw;
break;
case AUDIO_ENCODING_ALAW:
hw.encoding = AUDIO_ENCODING_ULINEAR_LE;
swcode = mode == AUMODE_PLAY ?
alaw_to_linear8 : linear8_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;
hw.sample_rate = p->sample_rate;
}
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;
}
if (swcode != NULL)
fil->append(fil, swcode, &hw);
DPRINTF(("sbdsp_set_params: model=%d, mode=%d, rate=%u, "
"prec=%d, chan=%d, enc=%d -> tc=%02x, cmd=%02x, "
"bmode=%02x, cmdchan=%02x\n", sc->sc_model, mode,
p->sample_rate, p->precision, p->channels,
p->encoding, tc, m->cmd, bmode, m->cmdchan));
}
if (sc->sc_fullduplex &&
usemode == (AUMODE_PLAY | AUMODE_RECORD) &&
sc->sc_i.dmachan == sc->sc_o.dmachan) {
DPRINTF(("sbdsp_set_params: 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. */
hw.precision = hw.validbits = 16;
pfil->append(pfil, linear8_to_linear16, &hw);
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(void *addr, int which)
{
struct sbdsp_softc *sc;
int mixval;
sc = addr;
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(void *addr)
{
struct sbdsp_softc *sc;
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(struct sbdsp_softc *sc, int mask)
{
int bitsl, bitsr;
int sbport;
KASSERT(mutex_owned(&sc->sc_lock));
KASSERT(mutex_owned(&sc->sc_intr_lock));
if (sc->sc_open == SB_OPEN_MIDI)
return EBUSY;
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(void *addr, int newstate)
{
struct sbdsp_softc *sc;
sc = addr;
if (sc->sc_open == SB_OPEN_MIDI)
return EBUSY;
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(void *addr, int blk, int mode,
const audio_params_t *param)
{
return blk & -4; /* round to biggest sample size */
}
int
sbdsp_open(void *addr, int flags)
{
struct sbdsp_softc *sc;
int error, state;
sc = addr;
DPRINTF(("sbdsp_open: sc=%p\n", sc));
if (sc->sc_open != SB_CLOSED)
return EBUSY;
sc->sc_open = SB_OPEN_AUDIO;
state = 0;
if (sc->sc_drq8 != -1) {
error = isa_drq_alloc(sc->sc_ic, sc->sc_drq8);
if (error != 0)
goto bad;
state |= 1;
}
if (sc->sc_drq16 != -1 && sc->sc_drq16 != sc->sc_drq8) {
error = isa_drq_alloc(sc->sc_ic, sc->sc_drq16);
if (error != 0)
goto bad;
state |= 2;
}
if (sbdsp_reset(sc) != 0) {
error = EIO;
goto bad;
}
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;
bad:
if (state & 1)
isa_drq_free(sc->sc_ic, sc->sc_drq8);
if (state & 2)
isa_drq_free(sc->sc_ic, sc->sc_drq16);
sc->sc_open = SB_CLOSED;
return error;
}
void
sbdsp_close(void *addr)
{
struct sbdsp_softc *sc;
sc = addr;
DPRINTF(("sbdsp_close: sc=%p\n", sc));
sbdsp_spkroff(sc);
sc->spkr_state = SPKR_OFF;
sc->sc_intr8 = 0;
sc->sc_intr16 = 0;
if (sc->sc_drq8 != -1)
isa_drq_free(sc->sc_ic, sc->sc_drq8);
if (sc->sc_drq16 != -1 && sc->sc_drq16 != sc->sc_drq8)
isa_drq_free(sc->sc_ic, sc->sc_drq16);
sc->sc_open = SB_CLOSED;
DPRINTF(("sbdsp_close: closed\n"));
}
/*
* Lower-level routines
*/
/*
* Reset the card.
* Return non-zero if the card isn't detected.
*/
int
sbdsp_reset(struct sbdsp_softc *sc)
{
bus_space_tag_t iot;
bus_space_handle_t ioh;
iot = sc->sc_iot;
ioh = sc->sc_ioh;
sc->sc_intr8 = 0;
sc->sc_intr16 = 0;
sc->sc_intrm = 0;
/*
* 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(struct sbdsp_softc *sc, int v)
{
bus_space_tag_t iot;
bus_space_handle_t ioh;
int i;
u_char x;
iot = sc->sc_iot;
ioh = sc->sc_ioh;
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(struct sbdsp_softc *sc)
{
bus_space_tag_t iot;
bus_space_handle_t ioh;
int i;
u_char x;
iot = sc->sc_iot;
ioh = sc->sc_ioh;
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;
}
void
sbdsp_pause(struct sbdsp_softc *sc)
{
KASSERT(mutex_owned(&sc->sc_intr_lock));
mutex_spin_exit(&sc->sc_intr_lock);
(void)kpause("sbpause", false, hz/8, &sc->sc_lock);
mutex_spin_enter(&sc->sc_intr_lock);
}
/*
* 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(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(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(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
#if 0
/* XXX figure out a good way of determining the model */
/* XXX what about SB_32 */
if (SBVER_MINOR(v) == 16)
sc->sc_model = SB_64;
else
#endif
sc->sc_model = SB_16;
break;
}
}
int
sbdsp_set_timeconst(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(struct sbdsp_softc *sc, int cmd, int 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_trigger_input(
void *addr,
void *start, void *end,
int blksize,
void (*intr)(void *),
void *arg,
const audio_params_t *param)
{
struct sbdsp_softc *sc;
int stereo;
int width;
int filter;
sc = addr;
stereo = param->channels == 2;
width = param->precision;
#ifdef DIAGNOSTIC
if (stereo && (blksize & 1)) {
DPRINTF(("stereo record odd bytes (%d)\n", blksize));
return EIO;
}
if (sc->sc_i.run != SB_NOTRUNNING)
printf("sbdsp_trigger_input: already running\n");
#endif
sc->sc_intrr = intr;
sc->sc_argr = arg;
if (width == 8) {
#ifdef DIAGNOSTIC
if (sc->sc_i.dmachan != sc->sc_drq8) {
printf("sbdsp_trigger_input: width=%d bad chan %d\n",
width, sc->sc_i.dmachan);
return EIO;
}
#endif
sc->sc_intr8 = sbdsp_block_input;
} else {
#ifdef DIAGNOSTIC
if (sc->sc_i.dmachan != sc->sc_drq16) {
printf("sbdsp_trigger_input: width=%d bad chan %d\n",
width, sc->sc_i.dmachan);
return EIO;
}
#endif
sc->sc_intr16 = sbdsp_block_input;
}
if ((sc->sc_model == SB_JAZZ) ? (sc->sc_i.dmachan > 3) : (width == 16))
blksize >>= 1;
--blksize;
sc->sc_i.blksize = blksize;
if (ISSBPRO(sc)) {
if (sbdsp_wdsp(sc, sc->sc_i.modep->cmdchan) < 0)
return EIO;
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_trigger_input: rate=%d set failed\n",
sc->sc_i.rate));
return EIO;
}
} else {
if (sbdsp_set_timeconst(sc, sc->sc_i.tc)) {
DPRINTF(("sbdsp_trigger_input: tc=%d set failed\n",
sc->sc_i.rate));
return EIO;
}
}
DPRINTF(("sbdsp: DMA start loop input start=%p end=%p chan=%d\n",
start, end, sc->sc_i.dmachan));
isa_dmastart(sc->sc_ic, sc->sc_i.dmachan, start,
(char *)end - (char *)start, NULL,
DMAMODE_READ | DMAMODE_LOOPDEMAND, BUS_DMA_NOWAIT);
return sbdsp_block_input(addr);
}
int
sbdsp_block_input(void *addr)
{
struct sbdsp_softc *sc;
int cc;
sc = addr;
cc = sc->sc_i.blksize;
DPRINTFN(2, ("sbdsp_block_input: sc=%p cc=%d\n", addr, cc));
if (sc->sc_i.run != SB_NOTRUNNING)
sc->sc_intrr(sc->sc_argr);
if (sc->sc_model == SB_1) {
/* Non-looping mode, start DMA */
if (sbdsp_wdsp(sc, sc->sc_i.modep->cmd) < 0 ||
sbdsp_wdsp(sc, cc) < 0 ||
sbdsp_wdsp(sc, cc >> 8) < 0) {
DPRINTF(("sbdsp_block_input: SB1 DMA start failed\n"));
return EIO;
}
sc->sc_i.run = SB_RUNNING;
} else if (sc->sc_i.run == SB_NOTRUNNING) {
/* Initialize looping PCM */
if (ISSB16CLASS(sc)) {
DPRINTFN(3, ("sbdsp16 input command cmd=0x%02x bmode=0x%02x cc=%d\n",
sc->sc_i.modep->cmd, sc->sc_i.bmode, cc));
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_block_input: SB16 DMA start failed\n"));
return EIO;
}
} else {
DPRINTF(("sbdsp_block_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_block_input: SB2 DMA blocksize failed\n"));
return EIO;
}
if (sbdsp_wdsp(sc, sc->sc_i.modep->cmd) < 0) {
DPRINTF(("sbdsp_block_input: SB2 DMA start failed\n"));
return EIO;
}
}
sc->sc_i.run = SB_LOOPING;
}
return 0;
}
int
sbdsp_trigger_output(
void *addr,
void *start, void *end,
int blksize,
void (*intr)(void *),
void *arg,
const audio_params_t *param)
{
struct sbdsp_softc *sc;
int stereo;
int width;
int cmd;
sc = addr;
stereo = param->channels == 2;
width = param->precision;
#ifdef DIAGNOSTIC
if (stereo && (blksize & 1)) {
DPRINTF(("stereo playback odd bytes (%d)\n", blksize));
return EIO;
}
if (sc->sc_o.run != SB_NOTRUNNING)
printf("sbdsp_trigger_output: already running\n");
#endif
sc->sc_intrp = intr;
sc->sc_argp = arg;
if (width == 8) {
#ifdef DIAGNOSTIC
if (sc->sc_o.dmachan != sc->sc_drq8) {
printf("sbdsp_trigger_output: width=%d bad chan %d\n",
width, sc->sc_o.dmachan);
return EIO;
}
#endif
sc->sc_intr8 = sbdsp_block_output;
} else {
#ifdef DIAGNOSTIC
if (sc->sc_o.dmachan != sc->sc_drq16) {
printf("sbdsp_trigger_output: width=%d bad chan %d\n",
width, sc->sc_o.dmachan);
return EIO;
}
#endif
sc->sc_intr16 = sbdsp_block_output;
}
if ((sc->sc_model == SB_JAZZ) ? (sc->sc_o.dmachan > 3) : (width == 16))
blksize >>= 1;
--blksize;
sc->sc_o.blksize = blksize;
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 EIO;
}
if (ISSB16CLASS(sc)) {
if (sbdsp16_set_rate(sc, SB_DSP16_OUTPUTRATE, sc->sc_o.rate)) {
DPRINTF(("sbdsp_trigger_output: rate=%d set failed\n",
sc->sc_o.rate));
return EIO;
}
} else {
if (sbdsp_set_timeconst(sc, sc->sc_o.tc)) {
DPRINTF(("sbdsp_trigger_output: tc=%d set failed\n",
sc->sc_o.rate));
return EIO;
}
}
DPRINTF(("sbdsp: DMA start loop output start=%p end=%p chan=%d\n",
start, end, sc->sc_o.dmachan));
isa_dmastart(sc->sc_ic, sc->sc_o.dmachan, start,
(char *)end - (char *)start, NULL,
DMAMODE_WRITE | DMAMODE_LOOPDEMAND, BUS_DMA_NOWAIT);
return sbdsp_block_output(addr);
}
int
sbdsp_block_output(void *addr)
{
struct sbdsp_softc *sc;
int cc;
sc = addr;
cc = sc->sc_o.blksize;
DPRINTFN(2, ("sbdsp_block_output: sc=%p cc=%d\n", addr, cc));
if (sc->sc_o.run != SB_NOTRUNNING)
sc->sc_intrp(sc->sc_argp);
if (sc->sc_model == SB_1) {
/* Non-looping mode, initialized. Start DMA and PCM */
if (sbdsp_wdsp(sc, sc->sc_o.modep->cmd) < 0 ||
sbdsp_wdsp(sc, cc) < 0 ||
sbdsp_wdsp(sc, cc >> 8) < 0) {
DPRINTF(("sbdsp_block_output: SB1 DMA start failed\n"));
return EIO;
}
sc->sc_o.run = SB_RUNNING;
} else if (sc->sc_o.run == SB_NOTRUNNING) {
/* Initialize looping PCM */
if (ISSB16CLASS(sc)) {
DPRINTF(("sbdsp_block_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_block_output: SB16 DMA start failed\n"));
return EIO;
}
} else {
DPRINTF(("sbdsp_block_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_block_output: SB2 DMA blocksize failed\n"));
return EIO;
}
if (sbdsp_wdsp(sc, sc->sc_o.modep->cmd) < 0) {
DPRINTF(("sbdsp_block_output: SB2 DMA start failed\n"));
return EIO;
}
}
sc->sc_o.run = SB_LOOPING;
}
return 0;
}
int
sbdsp_halt_output(void *addr)
{
struct sbdsp_softc *sc;
sc = addr;
if (sc->sc_o.run != SB_NOTRUNNING) {
if (sbdsp_wdsp(sc, sc->sc_o.modep->halt) < 0)
printf("sbdsp_halt_output: failed to halt\n");
isa_dmaabort(sc->sc_ic, sc->sc_o.dmachan);
sc->sc_o.run = SB_NOTRUNNING;
}
return 0;
}
int
sbdsp_halt_input(void *addr)
{
struct sbdsp_softc *sc;
sc = addr;
if (sc->sc_i.run != SB_NOTRUNNING) {
if (sbdsp_wdsp(sc, sc->sc_i.modep->halt) < 0)
printf("sbdsp_halt_input: failed to halt\n");
isa_dmaabort(sc->sc_ic, sc->sc_i.dmachan);
sc->sc_i.run = SB_NOTRUNNING;
}
return 0;
}
/*
* 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(void *arg)
{
struct sbdsp_softc *sc = arg;
#if NMPU > 0
struct mpu_softc *sc_mpu = device_private(sc->sc_mpudev);
#endif
u_char irq;
DPRINTFN(2, ("sbdsp_intr: intr8=%p, intr16=%p\n",
sc->sc_intr8, sc->sc_intr16));
mutex_spin_enter(&sc->sc_intr_lock);
if (ISSB16CLASS(sc)) {
irq = sbdsp_mix_read(sc, SBP_IRQ_STATUS);
if ((irq & (SBP_IRQ_DMA8 | SBP_IRQ_DMA16 | SBP_IRQ_MPU401)) == 0) {
mutex_spin_exit(&sc->sc_intr_lock);
DPRINTF(("sbdsp_intr: Spurious interrupt 0x%x\n", irq));
return 0;
}
} else {
/* XXXX CHECK FOR INTERRUPT */
irq = SBP_IRQ_DMA8;
}
sc->sc_interrupts++;
delay(10); /* XXX why? */
/* clear interrupt */
if (irq & SBP_IRQ_DMA8) {
bus_space_read_1(sc->sc_iot, sc->sc_ioh, SBP_DSP_IRQACK8);
if (sc->sc_intr8)
sc->sc_intr8(arg);
}
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(arg);
}
#if NMPU > 0
if ((irq & SBP_IRQ_MPU401) && sc_mpu) {
mpu_intr(sc_mpu);
}
#endif
mutex_spin_exit(&sc->sc_intr_lock);
return 1;
}
/* Like val & mask, but make sure the result is correctly rounded. */
#define MAXVAL 256
static int
sbdsp_adjust(int val, int mask)
{
val += (MAXVAL - mask) >> 1;
if (val >= MAXVAL)
val = MAXVAL-1;
return val & mask;
}
void
sbdsp_set_mixer_gain(struct sbdsp_softc *sc, int port)
{
int src, gain;
KASSERT(mutex_owned(&sc->sc_lock));
KASSERT(mutex_owned(&sc->sc_intr_lock));
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(void *addr, mixer_ctrl_t *cp)
{
struct sbdsp_softc *sc;
int lgain, rgain;
int mask, bits;
int lmask, rmask, lbits, rbits;
int mute, swap;
int error;
sc = addr;
KASSERT(mutex_owned(&sc->sc_lock));
if (sc->sc_open == SB_OPEN_MIDI)
return EBUSY;
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;
mutex_spin_enter(&sc->sc_intr_lock);
error = 0;
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) {
mutex_spin_exit(&sc->sc_intr_lock);
return EINVAL;
}
switch (cp->dev) {
case SB_TREBLE:
sbdsp_set_ifilter(addr, cp->un.ord ? SB_TREBLE : 0);
mutex_spin_exit(&sc->sc_intr_lock);
return 0;
case SB_BASS:
sbdsp_set_ifilter(addr, cp->un.ord ? SB_BASS : 0);
mutex_spin_exit(&sc->sc_intr_lock);
return 0;
}
}
case SB_PCSPEAKER:
case SB_INPUT_GAIN:
case SB_OUTPUT_GAIN:
if (!ISSBM1745(sc)) {
error = EINVAL;
break;
}
case SB_MIC_VOL:
case SB_LINE_IN_VOL:
if (sc->sc_mixer_model == SBM_CT1335) {
error = EINVAL;
break;
}
case SB_VOICE_VOL:
case SB_MIDI_VOL:
case SB_CD_VOL:
case SB_MASTER_VOL:
if (cp->type != AUDIO_MIXER_VALUE) {
error = EINVAL;
break;
}
/*
* 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) {
error = EINVAL;
break;
}
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) {
error = EINVAL;
break;
}
/* 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) {
error = EINVAL;
break;
}
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:
error = EINVAL;
break;
}
break;
}
if (error == 0) {
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)
error = EINVAL;
else
error = sbdsp_set_in_ports(sc, cp->un.mask);
} else {
if (cp->type != AUDIO_MIXER_ENUM)
error = EINVAL;
else {
sc->in_port = cp->un.ord;
error = sbdsp_set_in_ports(sc, 1 << cp->un.ord);
}
}
break;
case SB_AGC:
if (!ISSBM1745(sc) || cp->type != AUDIO_MIXER_ENUM)
error = EINVAL;
else
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) {
error = EINVAL;
break;
}
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) {
error = EINVAL;
break;
}
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:
error = EINVAL;
break;
}
mutex_spin_exit(&sc->sc_intr_lock);
return error;
}
int
sbdsp_mixer_get_port(void *addr, mixer_ctrl_t *cp)
{
struct sbdsp_softc *sc;
sc = addr;
KASSERT(mutex_owned(&sc->sc_lock));
if (sc->sc_open == SB_OPEN_MIDI)
return EBUSY;
DPRINTF(("sbdsp_mixer_get_port: port=%d\n", cp->dev));
if (sc->sc_mixer_model == SBM_NONE)
return EINVAL;
mutex_spin_enter(&sc->sc_intr_lock);
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;
mutex_spin_exit(&sc->sc_intr_lock);
return 0;
case SB_BASS:
cp->un.ord = sbdsp_get_ifilter(addr) == SB_BASS;
mutex_spin_exit(&sc->sc_intr_lock);
return 0;
}
}
case SB_PCSPEAKER:
case SB_INPUT_GAIN:
case SB_OUTPUT_GAIN:
if (!ISSBM1745(sc)) {
mutex_spin_exit(&sc->sc_intr_lock);
return EINVAL;
}
case SB_MIC_VOL:
case SB_LINE_IN_VOL:
if (sc->sc_mixer_model == SBM_CT1335) {
mutex_spin_exit(&sc->sc_intr_lock);
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) {
mutex_spin_exit(&sc->sc_intr_lock);
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:
mutex_spin_exit(&sc->sc_intr_lock);
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)) {
mutex_spin_exit(&sc->sc_intr_lock);
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:
mutex_spin_exit(&sc->sc_intr_lock);
return EINVAL;
}
mutex_spin_exit(&sc->sc_intr_lock);
return 0;
}
int
sbdsp_mixer_query_devinfo(void *addr, mixer_devinfo_t *dip)
{
struct sbdsp_softc *sc = addr;
int chan, class, is1745;
sc = addr;
DPRINTF(("sbdsp_mixer_query_devinfo: model=%d index=%d\n",
sc->sc_mixer_model, dip->index));
KASSERT(mutex_owned(&sc->sc_lock));
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(void *addr, int direction, size_t size)
{
struct sbdsp_softc *sc;
int drq;
sc = addr;
if (sc->sc_drq8 != -1)
drq = sc->sc_drq8;
else
drq = sc->sc_drq16;
return isa_malloc(sc->sc_ic, drq, size, M_DEVBUF, M_WAITOK);
}
void
sb_free(void *addr, void *ptr, size_t size)
{
isa_free(ptr, M_DEVBUF);
}
size_t
sb_round_buffersize(void *addr, int direction, size_t size)
{
struct sbdsp_softc *sc;
bus_size_t maxsize;
sc = addr;
if (sc->sc_drq8 != -1)
maxsize = sc->sc_drq8_maxsize;
else
maxsize = sc->sc_drq16_maxsize;
if (size > maxsize)
size = maxsize;
return size;
}
paddr_t
sb_mappage(void *addr, void *mem, off_t off, int prot)
{
return isa_mappage(mem, off, prot);
}
int
sbdsp_get_props(void *addr)
{
struct sbdsp_softc *sc;
sc = addr;
return AUDIO_PROP_MMAP | AUDIO_PROP_INDEPENDENT |
(sc->sc_fullduplex ? AUDIO_PROP_FULLDUPLEX : 0);
}
void
sbdsp_get_locks(void *addr, kmutex_t **intr, kmutex_t **proc)
{
struct sbdsp_softc *sc;
sc = addr;
*intr = &sc->sc_intr_lock;
*proc = &sc->sc_lock;
}
#if NMPU > 0
/*
* MIDI related routines.
*/
int
sbdsp_midi_open(void *addr, int flags, void (*iintr)(void *, int),
void (*ointr)(void *), void *arg)
{
struct sbdsp_softc *sc;
sc = addr;
DPRINTF(("sbdsp_midi_open: sc=%p\n", sc));
if (sc->sc_open != SB_CLOSED)
return EBUSY;
if (sbdsp_reset(sc) != 0)
return EIO;
sc->sc_open = SB_OPEN_MIDI;
if (sc->sc_model >= SB_20)
if (sbdsp_wdsp(sc, SB_MIDI_UART_INTR)) /* enter UART mode */
return EIO;
sc->sc_intr8 = sbdsp_midi_intr;
sc->sc_intrm = iintr;
sc->sc_argm = arg;
return 0;
}
void
sbdsp_midi_close(void *addr)
{
struct sbdsp_softc *sc;
sc = addr;
DPRINTF(("sbdsp_midi_close: sc=%p\n", sc));
if (sc->sc_model >= SB_20)
sbdsp_reset(sc); /* exit UART mode */
sc->sc_intrm = 0;
sc->sc_open = SB_CLOSED;
}
int
sbdsp_midi_output(void *addr, int d)
{
struct sbdsp_softc *sc;
sc = addr;
if (sc->sc_model < SB_20 && sbdsp_wdsp(sc, SB_MIDI_WRITE))
return EIO;
if (sbdsp_wdsp(sc, d))
return EIO;
return 0;
}
void
sbdsp_midi_getinfo(void *addr, struct midi_info *mi)
{
struct sbdsp_softc *sc;
sc = addr;
mi->name = sc->sc_model < SB_20 ? "SB MIDI cmd" : "SB MIDI UART";
mi->props = MIDI_PROP_CAN_INPUT;
}
int
sbdsp_midi_intr(void *addr)
{
struct sbdsp_softc *sc;
sc = addr;
KASSERT(mutex_owned(&sc->sc_intr_lock));
sc->sc_intrm(sc->sc_argm, sbdsp_rdsp(sc));
return (0);
}
#endif
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