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NetBSD/sys/dev/pci/sv.c
wiz 1ffa7b76c4 DMA, not dma nor Dma.
2003-05-03 18:10:37 +00:00

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/* $NetBSD: sv.c,v 1.22 2003/05/03 18:11:37 wiz Exp $ */
/* $OpenBSD: sv.c,v 1.2 1998/07/13 01:50:15 csapuntz Exp $ */
/*
* Copyright (c) 1999 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.
* 3. All advertising materials mentioning features or use of this software
* must display the following acknowledgement:
* This product includes software developed by the NetBSD
* Foundation, Inc. and its contributors.
* 4. Neither the name of The NetBSD Foundation nor the names of its
* contributors may be used to endorse or promote products derived
* from this software without specific prior written permission.
*
* 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) 1998 Constantine Paul Sapuntzakis
* All rights reserved
*
* Author: Constantine Paul Sapuntzakis (csapuntz@cvs.openbsd.org)
*
* 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. The author's name or those of the contributors may be used to
* endorse or promote products derived from this software without
* specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE AUTHOR(S) 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.
*/
/*
* S3 SonicVibes driver
* Heavily based on the eap driver by Lennart Augustsson
*/
#include <sys/cdefs.h>
__KERNEL_RCSID(0, "$NetBSD: sv.c,v 1.22 2003/05/03 18:11:37 wiz Exp $");
#include <sys/param.h>
#include <sys/systm.h>
#include <sys/kernel.h>
#include <sys/malloc.h>
#include <sys/device.h>
#include <dev/pci/pcireg.h>
#include <dev/pci/pcivar.h>
#include <dev/pci/pcidevs.h>
#include <sys/audioio.h>
#include <dev/audio_if.h>
#include <dev/mulaw.h>
#include <dev/auconv.h>
#include <dev/ic/i8237reg.h>
#include <dev/pci/svreg.h>
#include <dev/pci/svvar.h>
#include <machine/bus.h>
/* XXX
* The SonicVibes DMA is broken and only works on 24-bit addresses.
* As long as bus_dmamem_alloc_range() is missing we use the ISA
* DMA tag on i386.
*/
#if defined(i386)
#include "isa.h"
#if NISA > 0
#include <dev/isa/isavar.h>
#endif
#endif
#ifdef AUDIO_DEBUG
#define DPRINTF(x) if (svdebug) printf x
#define DPRINTFN(n,x) if (svdebug>(n)) printf x
int svdebug = 0;
#else
#define DPRINTF(x)
#define DPRINTFN(n,x)
#endif
int sv_match __P((struct device *, struct cfdata *, void *));
void sv_attach __P((struct device *, struct device *, void *));
int sv_intr __P((void *));
struct sv_dma {
bus_dmamap_t map;
caddr_t addr;
bus_dma_segment_t segs[1];
int nsegs;
size_t size;
struct sv_dma *next;
};
#define DMAADDR(p) ((p)->map->dm_segs[0].ds_addr)
#define KERNADDR(p) ((void *)((p)->addr))
CFATTACH_DECL(sv, sizeof(struct sv_softc),
sv_match, sv_attach, NULL, NULL);
struct audio_device sv_device = {
"S3 SonicVibes",
"",
"sv"
};
#define ARRAY_SIZE(foo) ((sizeof(foo)) / sizeof(foo[0]))
int sv_allocmem __P((struct sv_softc *, size_t, size_t, int, struct sv_dma *));
int sv_freemem __P((struct sv_softc *, struct sv_dma *));
int sv_open __P((void *, int));
void sv_close __P((void *));
int sv_query_encoding __P((void *, struct audio_encoding *));
int sv_set_params __P((void *, int, int, struct audio_params *, struct audio_params *));
int sv_round_blocksize __P((void *, int));
int sv_trigger_output __P((void *, void *, void *, int, void (*)(void *),
void *, struct audio_params *));
int sv_trigger_input __P((void *, void *, void *, int, void (*)(void *),
void *, struct audio_params *));
int sv_halt_output __P((void *));
int sv_halt_input __P((void *));
int sv_getdev __P((void *, struct audio_device *));
int sv_mixer_set_port __P((void *, mixer_ctrl_t *));
int sv_mixer_get_port __P((void *, mixer_ctrl_t *));
int sv_query_devinfo __P((void *, mixer_devinfo_t *));
void *sv_malloc __P((void *, int, size_t, struct malloc_type *, int));
void sv_free __P((void *, void *, struct malloc_type *));
size_t sv_round_buffersize __P((void *, int, size_t));
paddr_t sv_mappage __P((void *, void *, off_t, int));
int sv_get_props __P((void *));
#ifdef AUDIO_DEBUG
void sv_dumpregs __P((struct sv_softc *sc));
#endif
struct audio_hw_if sv_hw_if = {
sv_open,
sv_close,
NULL,
sv_query_encoding,
sv_set_params,
sv_round_blocksize,
NULL,
NULL,
NULL,
NULL,
NULL,
sv_halt_output,
sv_halt_input,
NULL,
sv_getdev,
NULL,
sv_mixer_set_port,
sv_mixer_get_port,
sv_query_devinfo,
sv_malloc,
sv_free,
sv_round_buffersize,
sv_mappage,
sv_get_props,
sv_trigger_output,
sv_trigger_input,
NULL,
};
static u_int8_t sv_read __P((struct sv_softc *, u_int8_t));
static u_int8_t sv_read_indirect __P((struct sv_softc *, u_int8_t));
static void sv_write __P((struct sv_softc *, u_int8_t, u_int8_t ));
static void sv_write_indirect __P((struct sv_softc *, u_int8_t, u_int8_t ));
static void sv_init_mixer __P((struct sv_softc *));
static void sv_defer __P((struct device *self));
static void
sv_write (sc, reg, val)
struct sv_softc *sc;
u_int8_t reg, val;
{
DPRINTFN(8,("sv_write(0x%x, 0x%x)\n", reg, val));
bus_space_write_1(sc->sc_iot, sc->sc_ioh, reg, val);
}
static u_int8_t
sv_read(sc, reg)
struct sv_softc *sc;
u_int8_t reg;
{
u_int8_t val;
val = bus_space_read_1(sc->sc_iot, sc->sc_ioh, reg);
DPRINTFN(8,("sv_read(0x%x) = 0x%x\n", reg, val));
return val;
}
static u_int8_t
sv_read_indirect(sc, reg)
struct sv_softc *sc;
u_int8_t reg;
{
u_int8_t val;
int s = splaudio();
sv_write(sc, SV_CODEC_IADDR, reg & SV_IADDR_MASK);
val = sv_read(sc, SV_CODEC_IDATA);
splx(s);
return (val);
}
static void
sv_write_indirect(sc, reg, val)
struct sv_softc *sc;
u_int8_t reg, val;
{
u_int8_t iaddr = reg & SV_IADDR_MASK;
int s = splaudio();
if (reg == SV_DMA_DATA_FORMAT)
iaddr |= SV_IADDR_MCE;
sv_write(sc, SV_CODEC_IADDR, iaddr);
sv_write(sc, SV_CODEC_IDATA, val);
splx(s);
}
int
sv_match(parent, match, aux)
struct device *parent;
struct cfdata *match;
void *aux;
{
struct pci_attach_args *pa = aux;
if (PCI_VENDOR(pa->pa_id) == PCI_VENDOR_S3 &&
PCI_PRODUCT(pa->pa_id) == PCI_PRODUCT_S3_SONICVIBES)
return (1);
return (0);
}
int pci_alloc_io __P((pci_chipset_tag_t pc, pcitag_t pt,
int pcioffs,
bus_space_tag_t iot, bus_size_t size,
bus_size_t align, bus_size_t bound, int flags,
bus_space_handle_t *ioh));
static pcireg_t pci_io_alloc_low, pci_io_alloc_high;
int
pci_alloc_io(pc, pt, pcioffs, iot, size, align, bound, flags, ioh)
pci_chipset_tag_t pc;
pcitag_t pt;
int pcioffs;
bus_space_tag_t iot;
bus_size_t size;
bus_size_t align;
bus_size_t bound;
int flags;
bus_space_handle_t *ioh;
{
bus_addr_t addr;
int error;
error = bus_space_alloc(iot, pci_io_alloc_low, pci_io_alloc_high,
size, align, bound, flags, &addr, ioh);
if (error)
return(error);
pci_conf_write(pc, pt, pcioffs, addr);
return (0);
}
/*
* Allocate IO addresses when all other configuration is done.
*/
void
sv_defer(self)
struct device *self;
{
struct sv_softc *sc = (struct sv_softc *)self;
pci_chipset_tag_t pc = sc->sc_pa.pa_pc;
pcitag_t pt = sc->sc_pa.pa_tag;
pcireg_t dmaio;
DPRINTF(("sv_defer: %p\n", sc));
/* XXX
* Get a reasonable default for the I/O range.
* Assume the range around SB_PORTBASE is valid on this PCI bus.
*/
pci_io_alloc_low = pci_conf_read(pc, pt, SV_SB_PORTBASE_SLOT);
pci_io_alloc_high = pci_io_alloc_low + 0x1000;
if (pci_alloc_io(pc, pt, SV_DMAA_CONFIG_OFF,
sc->sc_iot, SV_DMAA_SIZE, SV_DMAA_ALIGN, 0,
0, &sc->sc_dmaa_ioh)) {
printf("sv_attach: cannot allocate DMA A range\n");
return;
}
dmaio = pci_conf_read(pc, pt, SV_DMAA_CONFIG_OFF);
DPRINTF(("sv_attach: addr a dmaio=0x%lx\n", (u_long)dmaio));
pci_conf_write(pc, pt, SV_DMAA_CONFIG_OFF,
dmaio | SV_DMA_CHANNEL_ENABLE | SV_DMAA_EXTENDED_ADDR);
if (pci_alloc_io(pc, pt, SV_DMAC_CONFIG_OFF,
sc->sc_iot, SV_DMAC_SIZE, SV_DMAC_ALIGN, 0,
0, &sc->sc_dmac_ioh)) {
printf("sv_attach: cannot allocate DMA C range\n");
return;
}
dmaio = pci_conf_read(pc, pt, SV_DMAC_CONFIG_OFF);
DPRINTF(("sv_attach: addr c dmaio=0x%lx\n", (u_long)dmaio));
pci_conf_write(pc, pt, SV_DMAC_CONFIG_OFF,
dmaio | SV_DMA_CHANNEL_ENABLE);
sc->sc_dmaset = 1;
}
void
sv_attach(parent, self, aux)
struct device *parent, *self;
void *aux;
{
struct sv_softc *sc = (struct sv_softc *)self;
struct pci_attach_args *pa = aux;
pci_chipset_tag_t pc = pa->pa_pc;
pcitag_t pt = pa->pa_tag;
pci_intr_handle_t ih;
pcireg_t csr;
char const *intrstr;
u_int8_t reg;
struct audio_attach_args arg;
printf ("\n");
/* Map I/O registers */
if (pci_mapreg_map(pa, SV_ENHANCED_PORTBASE_SLOT,
PCI_MAPREG_TYPE_IO, 0,
&sc->sc_iot, &sc->sc_ioh, NULL, NULL)) {
printf("%s: can't map enhanced i/o space\n",
sc->sc_dev.dv_xname);
return;
}
if (pci_mapreg_map(pa, SV_FM_PORTBASE_SLOT,
PCI_MAPREG_TYPE_IO, 0,
&sc->sc_opliot, &sc->sc_oplioh, NULL, NULL)) {
printf("%s: can't map FM i/o space\n", sc->sc_dev.dv_xname);
return;
}
if (pci_mapreg_map(pa, SV_MIDI_PORTBASE_SLOT,
PCI_MAPREG_TYPE_IO, 0,
&sc->sc_midiiot, &sc->sc_midiioh, NULL, NULL)) {
printf("%s: can't map MIDI i/o space\n", sc->sc_dev.dv_xname);
return;
}
DPRINTF(("sv: IO ports: enhanced=0x%x, OPL=0x%x, MIDI=0x%x\n",
(int)sc->sc_ioh, (int)sc->sc_oplioh, (int)sc->sc_midiioh));
#if defined(alpha)
/* XXX Force allocation through the SGMAP. */
sc->sc_dmatag = alphabus_dma_get_tag(pa->pa_dmat, ALPHA_BUS_ISA);
#elif defined(i386) && NISA > 0
/* XXX
* The SonicVibes DMA is broken and only works on 24-bit addresses.
* As long as bus_dmamem_alloc_range() is missing we use the ISA
* DMA tag on i386.
*/
sc->sc_dmatag = &isa_bus_dma_tag;
#else
sc->sc_dmatag = pa->pa_dmat;
#endif
pci_conf_write(pc, pt, SV_DMAA_CONFIG_OFF, SV_DMAA_EXTENDED_ADDR);
pci_conf_write(pc, pt, SV_DMAC_CONFIG_OFF, 0);
/* Enable the device. */
csr = pci_conf_read(pc, pt, PCI_COMMAND_STATUS_REG);
pci_conf_write(pc, pt, PCI_COMMAND_STATUS_REG,
csr | PCI_COMMAND_MASTER_ENABLE);
sv_write_indirect(sc, SV_ANALOG_POWER_DOWN_CONTROL, 0);
sv_write_indirect(sc, SV_DIGITAL_POWER_DOWN_CONTROL, 0);
/* initialize codec registers */
reg = sv_read(sc, SV_CODEC_CONTROL);
reg |= SV_CTL_RESET;
sv_write(sc, SV_CODEC_CONTROL, reg);
delay(50);
reg = sv_read(sc, SV_CODEC_CONTROL);
reg &= ~SV_CTL_RESET;
reg |= SV_CTL_INTA | SV_CTL_ENHANCED;
/* This write clears the reset */
sv_write(sc, SV_CODEC_CONTROL, reg);
delay(50);
/* This write actually shoves the new values in */
sv_write(sc, SV_CODEC_CONTROL, reg);
DPRINTF(("sv_attach: control=0x%x\n", sv_read(sc, SV_CODEC_CONTROL)));
/* Enable DMA interrupts */
reg = sv_read(sc, SV_CODEC_INTMASK);
reg &= ~(SV_INTMASK_DMAA | SV_INTMASK_DMAC);
reg |= SV_INTMASK_UD | SV_INTMASK_SINT | SV_INTMASK_MIDI;
sv_write(sc, SV_CODEC_INTMASK, reg);
sv_read(sc, SV_CODEC_STATUS);
/* Map and establish the interrupt. */
if (pci_intr_map(pa, &ih)) {
printf("%s: couldn't map interrupt\n", sc->sc_dev.dv_xname);
return;
}
intrstr = pci_intr_string(pc, ih);
sc->sc_ih = pci_intr_establish(pc, ih, IPL_AUDIO, sv_intr, sc);
if (sc->sc_ih == NULL) {
printf("%s: couldn't establish interrupt",
sc->sc_dev.dv_xname);
if (intrstr != NULL)
printf(" at %s", intrstr);
printf("\n");
return;
}
printf("%s: interrupting at %s\n", sc->sc_dev.dv_xname, intrstr);
printf("%s: rev %d", sc->sc_dev.dv_xname,
sv_read_indirect(sc, SV_REVISION_LEVEL));
if (sv_read(sc, SV_CODEC_CONTROL) & SV_CTL_MD1)
printf(", reverb SRAM present");
if (!(sv_read_indirect(sc, SV_WAVETABLE_SOURCE_SELECT) & SV_WSS_WT0))
printf(", wavetable ROM present");
printf("\n");
sv_init_mixer(sc);
audio_attach_mi(&sv_hw_if, sc, &sc->sc_dev);
arg.type = AUDIODEV_TYPE_OPL;
arg.hwif = 0;
arg.hdl = 0;
(void)config_found(&sc->sc_dev, &arg, audioprint);
sc->sc_pa = *pa; /* for deferred setup */
config_defer(self, sv_defer);
}
#ifdef AUDIO_DEBUG
void
sv_dumpregs(sc)
struct sv_softc *sc;
{
int idx;
#if 0
for (idx = 0; idx < 0x50; idx += 4)
printf ("%02x = %x\n", idx,
pci_conf_read(pa->pa_pc, pa->pa_tag, idx));
#endif
for (idx = 0; idx < 6; idx++)
printf ("REG %02x = %02x\n", idx, sv_read(sc, idx));
for (idx = 0; idx < 0x32; idx++)
printf ("IREG %02x = %02x\n", idx, sv_read_indirect(sc, idx));
for (idx = 0; idx < 0x10; idx++)
printf ("DMA %02x = %02x\n", idx,
bus_space_read_1(sc->sc_iot, sc->sc_dmaa_ioh, idx));
}
#endif
int
sv_intr(p)
void *p;
{
struct sv_softc *sc = p;
u_int8_t intr;
intr = sv_read(sc, SV_CODEC_STATUS);
DPRINTFN(5,("sv_intr: intr=0x%x\n", intr));
if (!(intr & (SV_INTSTATUS_DMAA | SV_INTSTATUS_DMAC)))
return (0);
if (intr & SV_INTSTATUS_DMAA) {
if (sc->sc_pintr)
sc->sc_pintr(sc->sc_parg);
}
if (intr & SV_INTSTATUS_DMAC) {
if (sc->sc_rintr)
sc->sc_rintr(sc->sc_rarg);
}
return (1);
}
int
sv_allocmem(sc, size, align, direction, p)
struct sv_softc *sc;
size_t size;
size_t align;
int direction;
struct sv_dma *p;
{
int error;
p->size = size;
error = bus_dmamem_alloc(sc->sc_dmatag, p->size, align, 0,
p->segs, ARRAY_SIZE(p->segs),
&p->nsegs, BUS_DMA_NOWAIT);
if (error)
return (error);
error = bus_dmamem_map(sc->sc_dmatag, p->segs, p->nsegs, p->size,
&p->addr, BUS_DMA_NOWAIT|BUS_DMA_COHERENT);
if (error)
goto free;
error = bus_dmamap_create(sc->sc_dmatag, p->size, 1, p->size,
0, BUS_DMA_NOWAIT, &p->map);
if (error)
goto unmap;
error = bus_dmamap_load(sc->sc_dmatag, p->map, p->addr, p->size, NULL,
BUS_DMA_NOWAIT |
(direction == AUMODE_RECORD) ? BUS_DMA_READ : BUS_DMA_WRITE);
if (error)
goto destroy;
DPRINTF(("sv_allocmem: pa=%lx va=%lx pba=%lx\n",
(long)p->segs[0].ds_addr, (long)KERNADDR(p), (long)DMAADDR(p)));
return (0);
destroy:
bus_dmamap_destroy(sc->sc_dmatag, p->map);
unmap:
bus_dmamem_unmap(sc->sc_dmatag, p->addr, p->size);
free:
bus_dmamem_free(sc->sc_dmatag, p->segs, p->nsegs);
return (error);
}
int
sv_freemem(sc, p)
struct sv_softc *sc;
struct sv_dma *p;
{
bus_dmamap_unload(sc->sc_dmatag, p->map);
bus_dmamap_destroy(sc->sc_dmatag, p->map);
bus_dmamem_unmap(sc->sc_dmatag, p->addr, p->size);
bus_dmamem_free(sc->sc_dmatag, p->segs, p->nsegs);
return (0);
}
int
sv_open(addr, flags)
void *addr;
int flags;
{
struct sv_softc *sc = addr;
DPRINTF(("sv_open\n"));
if (!sc->sc_dmaset)
return (ENXIO);
sc->sc_pintr = 0;
sc->sc_rintr = 0;
return (0);
}
/*
* Close function is called at splaudio().
*/
void
sv_close(addr)
void *addr;
{
struct sv_softc *sc = addr;
DPRINTF(("sv_close\n"));
sv_halt_output(sc);
sv_halt_input(sc);
sc->sc_pintr = 0;
sc->sc_rintr = 0;
}
int
sv_query_encoding(addr, fp)
void *addr;
struct audio_encoding *fp;
{
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 = AUDIO_ENCODINGFLAG_EMULATED;
return (0);
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 = AUDIO_ENCODINGFLAG_EMULATED;
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);
}
}
int
sv_set_params(addr, setmode, usemode, play, rec)
void *addr;
int setmode, usemode;
struct audio_params *play, *rec;
{
struct sv_softc *sc = addr;
struct audio_params *p = NULL;
int mode;
u_int32_t val;
/*
* This device only has one clock, so make the sample rates match.
*/
if (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);
}
for (mode = AUMODE_RECORD; mode != -1;
mode = mode == AUMODE_RECORD ? AUMODE_PLAY : -1) {
if ((setmode & mode) == 0)
continue;
p = mode == AUMODE_PLAY ? play : rec;
if (p->sample_rate < 2000 || p->sample_rate > 48000 ||
(p->precision != 8 && p->precision != 16) ||
(p->channels != 1 && p->channels != 2))
return (EINVAL);
p->factor = 1;
p->sw_code = 0;
switch (p->encoding) {
case AUDIO_ENCODING_SLINEAR_BE:
if (p->precision == 16)
p->sw_code = swap_bytes;
else
p->sw_code = change_sign8;
break;
case AUDIO_ENCODING_SLINEAR_LE:
if (p->precision != 16)
p->sw_code = change_sign8;
break;
case AUDIO_ENCODING_ULINEAR_BE:
if (p->precision == 16) {
if (mode == AUMODE_PLAY)
p->sw_code = swap_bytes_change_sign16_le;
else
p->sw_code = change_sign16_swap_bytes_le;
}
break;
case AUDIO_ENCODING_ULINEAR_LE:
if (p->precision == 16)
p->sw_code = change_sign16_le;
break;
case AUDIO_ENCODING_ULAW:
if (mode == AUMODE_PLAY) {
p->factor = 2;
p->sw_code = mulaw_to_slinear16_le;
} else
p->sw_code = ulinear8_to_mulaw;
break;
case AUDIO_ENCODING_ALAW:
if (mode == AUMODE_PLAY) {
p->factor = 2;
p->sw_code = alaw_to_slinear16_le;
} else
p->sw_code = ulinear8_to_alaw;
break;
default:
return (EINVAL);
}
}
val = p->sample_rate * 65536 / 48000;
/*
* If the sample rate is exactly 48KHz, the fraction would overflow the
* register, so we have to bias it. This causes a little clock drift.
* The drift is below normal crystal tolerance (.0001%), so although
* this seems a little silly, we can pretty much ignore it.
* (I tested the output speed with values of 1-20, just to be sure this
* register isn't *supposed* to have a bias. It isn't.)
* - mycroft
*/
if (val > 65535)
val = 65535;
sv_write_indirect(sc, SV_PCM_SAMPLE_RATE_0, val & 0xff);
sv_write_indirect(sc, SV_PCM_SAMPLE_RATE_1, val >> 8);
#define F_REF 24576000
#define ABS(x) (((x) < 0) ? (-x) : (x))
if (setmode & AUMODE_RECORD) {
/* The ADC reference frequency (f_out) is 512 * sample rate */
/* f_out is dervied from the 24.576MHz crystal by three values:
M & N & R. The equation is as follows:
f_out = (m + 2) * f_ref / ((n + 2) * (2 ^ a))
with the constraint that:
80 MHz < (m + 2) / (n + 2) * f_ref <= 150MHz
and n, m >= 1
*/
int goal_f_out = 512 * rec->sample_rate;
int a, n, m, best_n = 0, best_m = 0, best_error = 10000000;
int pll_sample;
int error;
for (a = 0; a < 8; a++) {
if ((goal_f_out * (1 << a)) >= 80000000)
break;
}
/* a != 8 because sample_rate >= 2000 */
for (n = 33; n > 2; n--) {
m = (goal_f_out * n * (1 << a)) / F_REF;
if ((m > 257) || (m < 3))
continue;
pll_sample = (m * F_REF) / (n * (1 << a));
pll_sample /= 512;
/* Threshold might be good here */
error = pll_sample - rec->sample_rate;
error = ABS(error);
if (error < best_error) {
best_error = error;
best_n = n;
best_m = m;
if (error == 0) break;
}
}
best_n -= 2;
best_m -= 2;
sv_write_indirect(sc, SV_ADC_PLL_M, best_m);
sv_write_indirect(sc, SV_ADC_PLL_N,
best_n | (a << SV_PLL_R_SHIFT));
}
return (0);
}
int
sv_round_blocksize(addr, blk)
void *addr;
int blk;
{
return (blk & -32); /* keep good alignment */
}
int
sv_trigger_output(addr, start, end, blksize, intr, arg, param)
void *addr;
void *start, *end;
int blksize;
void (*intr) __P((void *));
void *arg;
struct audio_params *param;
{
struct sv_softc *sc = addr;
struct sv_dma *p;
u_int8_t mode;
int dma_count;
DPRINTFN(1, ("sv_trigger_output: sc=%p start=%p end=%p blksize=%d intr=%p(%p)\n",
addr, start, end, blksize, intr, arg));
sc->sc_pintr = intr;
sc->sc_parg = arg;
mode = sv_read_indirect(sc, SV_DMA_DATA_FORMAT);
mode &= ~(SV_DMAA_FORMAT16 | SV_DMAA_STEREO);
if (param->precision * param->factor == 16)
mode |= SV_DMAA_FORMAT16;
if (param->channels == 2)
mode |= SV_DMAA_STEREO;
sv_write_indirect(sc, SV_DMA_DATA_FORMAT, mode);
for (p = sc->sc_dmas; p && KERNADDR(p) != start; p = p->next)
;
if (!p) {
printf("sv_trigger_output: bad addr %p\n", start);
return (EINVAL);
}
dma_count = ((char *)end - (char *)start) - 1;
DPRINTF(("sv_trigger_output: DMA start loop input addr=%x cc=%d\n",
(int)DMAADDR(p), dma_count));
bus_space_write_4(sc->sc_iot, sc->sc_dmaa_ioh, SV_DMA_ADDR0,
DMAADDR(p));
bus_space_write_4(sc->sc_iot, sc->sc_dmaa_ioh, SV_DMA_COUNT0,
dma_count);
bus_space_write_1(sc->sc_iot, sc->sc_dmaa_ioh, SV_DMA_MODE,
DMA37MD_READ | DMA37MD_LOOP);
DPRINTF(("sv_trigger_output: current addr=%x\n",
bus_space_read_4(sc->sc_iot, sc->sc_dmaa_ioh, SV_DMA_ADDR0)));
dma_count = blksize - 1;
sv_write_indirect(sc, SV_DMAA_COUNT1, dma_count >> 8);
sv_write_indirect(sc, SV_DMAA_COUNT0, dma_count & 0xFF);
mode = sv_read_indirect(sc, SV_PLAY_RECORD_ENABLE);
sv_write_indirect(sc, SV_PLAY_RECORD_ENABLE, mode | SV_PLAY_ENABLE);
return (0);
}
int
sv_trigger_input(addr, start, end, blksize, intr, arg, param)
void *addr;
void *start, *end;
int blksize;
void (*intr) __P((void *));
void *arg;
struct audio_params *param;
{
struct sv_softc *sc = addr;
struct sv_dma *p;
u_int8_t mode;
int dma_count;
DPRINTFN(1, ("sv_trigger_input: sc=%p start=%p end=%p blksize=%d intr=%p(%p)\n",
addr, start, end, blksize, intr, arg));
sc->sc_rintr = intr;
sc->sc_rarg = arg;
mode = sv_read_indirect(sc, SV_DMA_DATA_FORMAT);
mode &= ~(SV_DMAC_FORMAT16 | SV_DMAC_STEREO);
if (param->precision * param->factor == 16)
mode |= SV_DMAC_FORMAT16;
if (param->channels == 2)
mode |= SV_DMAC_STEREO;
sv_write_indirect(sc, SV_DMA_DATA_FORMAT, mode);
for (p = sc->sc_dmas; p && KERNADDR(p) != start; p = p->next)
;
if (!p) {
printf("sv_trigger_input: bad addr %p\n", start);
return (EINVAL);
}
dma_count = (((char *)end - (char *)start) >> 1) - 1;
DPRINTF(("sv_trigger_input: DMA start loop input addr=%x cc=%d\n",
(int)DMAADDR(p), dma_count));
bus_space_write_4(sc->sc_iot, sc->sc_dmac_ioh, SV_DMA_ADDR0,
DMAADDR(p));
bus_space_write_4(sc->sc_iot, sc->sc_dmac_ioh, SV_DMA_COUNT0,
dma_count);
bus_space_write_1(sc->sc_iot, sc->sc_dmac_ioh, SV_DMA_MODE,
DMA37MD_WRITE | DMA37MD_LOOP);
DPRINTF(("sv_trigger_input: current addr=%x\n",
bus_space_read_4(sc->sc_iot, sc->sc_dmac_ioh, SV_DMA_ADDR0)));
dma_count = (blksize >> 1) - 1;
sv_write_indirect(sc, SV_DMAC_COUNT1, dma_count >> 8);
sv_write_indirect(sc, SV_DMAC_COUNT0, dma_count & 0xFF);
mode = sv_read_indirect(sc, SV_PLAY_RECORD_ENABLE);
sv_write_indirect(sc, SV_PLAY_RECORD_ENABLE, mode | SV_RECORD_ENABLE);
return (0);
}
int
sv_halt_output(addr)
void *addr;
{
struct sv_softc *sc = addr;
u_int8_t mode;
DPRINTF(("sv: sv_halt_output\n"));
mode = sv_read_indirect(sc, SV_PLAY_RECORD_ENABLE);
sv_write_indirect(sc, SV_PLAY_RECORD_ENABLE, mode & ~SV_PLAY_ENABLE);
return (0);
}
int
sv_halt_input(addr)
void *addr;
{
struct sv_softc *sc = addr;
u_int8_t mode;
DPRINTF(("sv: sv_halt_input\n"));
mode = sv_read_indirect(sc, SV_PLAY_RECORD_ENABLE);
sv_write_indirect(sc, SV_PLAY_RECORD_ENABLE, mode & ~SV_RECORD_ENABLE);
return (0);
}
int
sv_getdev(addr, retp)
void *addr;
struct audio_device *retp;
{
*retp = sv_device;
return (0);
}
/*
* Mixer related code is here
*
*/
#define SV_INPUT_CLASS 0
#define SV_OUTPUT_CLASS 1
#define SV_RECORD_CLASS 2
#define SV_LAST_CLASS 2
static const char *mixer_classes[] =
{ AudioCinputs, AudioCoutputs, AudioCrecord };
static const struct {
u_int8_t l_port;
u_int8_t r_port;
u_int8_t mask;
u_int8_t class;
const char *audio;
} ports[] = {
{ SV_LEFT_AUX1_INPUT_CONTROL, SV_RIGHT_AUX1_INPUT_CONTROL, SV_AUX1_MASK,
SV_INPUT_CLASS, "aux1" },
{ SV_LEFT_CD_INPUT_CONTROL, SV_RIGHT_CD_INPUT_CONTROL, SV_CD_MASK,
SV_INPUT_CLASS, AudioNcd },
{ SV_LEFT_LINE_IN_INPUT_CONTROL, SV_RIGHT_LINE_IN_INPUT_CONTROL, SV_LINE_IN_MASK,
SV_INPUT_CLASS, AudioNline },
{ SV_MIC_INPUT_CONTROL, 0, SV_MIC_MASK, SV_INPUT_CLASS, AudioNmicrophone },
{ SV_LEFT_SYNTH_INPUT_CONTROL, SV_RIGHT_SYNTH_INPUT_CONTROL,
SV_SYNTH_MASK, SV_INPUT_CLASS, AudioNfmsynth },
{ SV_LEFT_AUX2_INPUT_CONTROL, SV_RIGHT_AUX2_INPUT_CONTROL, SV_AUX2_MASK,
SV_INPUT_CLASS, "aux2" },
{ SV_LEFT_PCM_INPUT_CONTROL, SV_RIGHT_PCM_INPUT_CONTROL, SV_PCM_MASK,
SV_INPUT_CLASS, AudioNdac },
{ SV_LEFT_MIXER_OUTPUT_CONTROL, SV_RIGHT_MIXER_OUTPUT_CONTROL,
SV_MIXER_OUT_MASK, SV_OUTPUT_CLASS, AudioNmaster }
};
static const struct {
int idx;
const char *name;
} record_sources[] = {
{ SV_REC_CD, AudioNcd },
{ SV_REC_DAC, AudioNdac },
{ SV_REC_AUX2, "aux2" },
{ SV_REC_LINE, AudioNline },
{ SV_REC_AUX1, "aux1" },
{ SV_REC_MIC, AudioNmicrophone },
{ SV_REC_MIXER, AudioNmixerout }
};
#define SV_DEVICES_PER_PORT 2
#define SV_FIRST_MIXER (SV_LAST_CLASS + 1)
#define SV_LAST_MIXER (SV_DEVICES_PER_PORT * (ARRAY_SIZE(ports)) + SV_LAST_CLASS)
#define SV_RECORD_SOURCE (SV_LAST_MIXER + 1)
#define SV_MIC_BOOST (SV_LAST_MIXER + 2)
#define SV_RECORD_GAIN (SV_LAST_MIXER + 3)
#define SV_SRS_MODE (SV_LAST_MIXER + 4)
int
sv_query_devinfo(addr, dip)
void *addr;
mixer_devinfo_t *dip;
{
int i;
/* It's a class */
if (dip->index <= SV_LAST_CLASS) {
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = dip->index;
dip->next = dip->prev = AUDIO_MIXER_LAST;
strcpy(dip->label.name,
mixer_classes[dip->index]);
return (0);
}
if (dip->index >= SV_FIRST_MIXER &&
dip->index <= SV_LAST_MIXER) {
int off = dip->index - SV_FIRST_MIXER;
int mute = (off % SV_DEVICES_PER_PORT);
int idx = off / SV_DEVICES_PER_PORT;
dip->mixer_class = ports[idx].class;
strcpy(dip->label.name, ports[idx].audio);
if (!mute) {
dip->type = AUDIO_MIXER_VALUE;
dip->prev = AUDIO_MIXER_LAST;
dip->next = dip->index + 1;
if (ports[idx].r_port != 0)
dip->un.v.num_channels = 2;
else
dip->un.v.num_channels = 1;
strcpy(dip->un.v.units.name, AudioNvolume);
} else {
dip->type = AUDIO_MIXER_ENUM;
dip->prev = dip->index - 1;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNmute);
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);
}
switch (dip->index) {
case SV_RECORD_SOURCE:
dip->mixer_class = SV_RECORD_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = SV_RECORD_GAIN;
strcpy(dip->label.name, AudioNsource);
dip->type = AUDIO_MIXER_ENUM;
dip->un.e.num_mem = ARRAY_SIZE(record_sources);
for (i = 0; i < ARRAY_SIZE(record_sources); i++) {
strcpy(dip->un.e.member[i].label.name,
record_sources[i].name);
dip->un.e.member[i].ord = record_sources[i].idx;
}
return (0);
case SV_RECORD_GAIN:
dip->mixer_class = SV_RECORD_CLASS;
dip->prev = SV_RECORD_SOURCE;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, "gain");
dip->type = AUDIO_MIXER_VALUE;
dip->un.v.num_channels = 1;
strcpy(dip->un.v.units.name, AudioNvolume);
return (0);
case SV_MIC_BOOST:
dip->mixer_class = SV_RECORD_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, "micboost");
goto on_off;
case SV_SRS_MODE:
dip->mixer_class = SV_OUTPUT_CLASS;
dip->prev = dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNspatial);
on_off:
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);
}
int
sv_mixer_set_port(addr, cp)
void *addr;
mixer_ctrl_t *cp;
{
struct sv_softc *sc = addr;
u_int8_t reg;
int idx;
if (cp->dev >= SV_FIRST_MIXER &&
cp->dev <= SV_LAST_MIXER) {
int off = cp->dev - SV_FIRST_MIXER;
int mute = (off % SV_DEVICES_PER_PORT);
idx = off / SV_DEVICES_PER_PORT;
if (mute) {
if (cp->type != AUDIO_MIXER_ENUM)
return (EINVAL);
reg = sv_read_indirect(sc, ports[idx].l_port);
if (cp->un.ord)
reg |= SV_MUTE_BIT;
else
reg &= ~SV_MUTE_BIT;
sv_write_indirect(sc, ports[idx].l_port, reg);
if (ports[idx].r_port) {
reg = sv_read_indirect(sc, ports[idx].r_port);
if (cp->un.ord)
reg |= SV_MUTE_BIT;
else
reg &= ~SV_MUTE_BIT;
sv_write_indirect(sc, ports[idx].r_port, reg);
}
} else {
int lval, rval;
if (cp->type != AUDIO_MIXER_VALUE)
return (EINVAL);
if (cp->un.value.num_channels != 1 &&
cp->un.value.num_channels != 2)
return (EINVAL);
if (ports[idx].r_port == 0) {
if (cp->un.value.num_channels != 1)
return (EINVAL);
lval = cp->un.value.level[AUDIO_MIXER_LEVEL_MONO];
rval = 0; /* shut up GCC */
} else {
if (cp->un.value.num_channels != 2)
return (EINVAL);
lval = cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT];
rval = cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT];
}
reg = sv_read_indirect(sc, ports[idx].l_port);
reg &= ~(ports[idx].mask);
lval = (AUDIO_MAX_GAIN - lval) * ports[idx].mask /
AUDIO_MAX_GAIN;
reg |= lval;
sv_write_indirect(sc, ports[idx].l_port, reg);
if (ports[idx].r_port != 0) {
reg = sv_read_indirect(sc, ports[idx].r_port);
reg &= ~(ports[idx].mask);
rval = (AUDIO_MAX_GAIN - rval) * ports[idx].mask /
AUDIO_MAX_GAIN;
reg |= rval;
sv_write_indirect(sc, ports[idx].r_port, reg);
}
sv_read_indirect(sc, ports[idx].l_port);
}
return (0);
}
switch (cp->dev) {
case SV_RECORD_SOURCE:
if (cp->type != AUDIO_MIXER_ENUM)
return (EINVAL);
for (idx = 0; idx < ARRAY_SIZE(record_sources); idx++) {
if (record_sources[idx].idx == cp->un.ord)
goto found;
}
return (EINVAL);
found:
reg = sv_read_indirect(sc, SV_LEFT_ADC_INPUT_CONTROL);
reg &= ~SV_REC_SOURCE_MASK;
reg |= (((cp->un.ord) << SV_REC_SOURCE_SHIFT) & SV_REC_SOURCE_MASK);
sv_write_indirect(sc, SV_LEFT_ADC_INPUT_CONTROL, reg);
reg = sv_read_indirect(sc, SV_RIGHT_ADC_INPUT_CONTROL);
reg &= ~SV_REC_SOURCE_MASK;
reg |= (((cp->un.ord) << SV_REC_SOURCE_SHIFT) & SV_REC_SOURCE_MASK);
sv_write_indirect(sc, SV_RIGHT_ADC_INPUT_CONTROL, reg);
return (0);
case SV_RECORD_GAIN:
{
int val;
if (cp->type != AUDIO_MIXER_VALUE)
return (EINVAL);
if (cp->un.value.num_channels != 1)
return (EINVAL);
val = (cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] * SV_REC_GAIN_MASK)
/ AUDIO_MAX_GAIN;
reg = sv_read_indirect(sc, SV_LEFT_ADC_INPUT_CONTROL);
reg &= ~SV_REC_GAIN_MASK;
reg |= val;
sv_write_indirect(sc, SV_LEFT_ADC_INPUT_CONTROL, reg);
reg = sv_read_indirect(sc, SV_RIGHT_ADC_INPUT_CONTROL);
reg &= ~SV_REC_GAIN_MASK;
reg |= val;
sv_write_indirect(sc, SV_RIGHT_ADC_INPUT_CONTROL, reg);
}
return (0);
case SV_MIC_BOOST:
if (cp->type != AUDIO_MIXER_ENUM)
return (EINVAL);
reg = sv_read_indirect(sc, SV_LEFT_ADC_INPUT_CONTROL);
if (cp->un.ord) {
reg |= SV_MIC_BOOST_BIT;
} else {
reg &= ~SV_MIC_BOOST_BIT;
}
sv_write_indirect(sc, SV_LEFT_ADC_INPUT_CONTROL, reg);
return (0);
case SV_SRS_MODE:
if (cp->type != AUDIO_MIXER_ENUM)
return (EINVAL);
reg = sv_read_indirect(sc, SV_SRS_SPACE_CONTROL);
if (cp->un.ord) {
reg &= ~SV_SRS_SPACE_ONOFF;
} else {
reg |= SV_SRS_SPACE_ONOFF;
}
sv_write_indirect(sc, SV_SRS_SPACE_CONTROL, reg);
return (0);
}
return (EINVAL);
}
int
sv_mixer_get_port(addr, cp)
void *addr;
mixer_ctrl_t *cp;
{
struct sv_softc *sc = addr;
int val;
u_int8_t reg;
if (cp->dev >= SV_FIRST_MIXER &&
cp->dev <= SV_LAST_MIXER) {
int off = cp->dev - SV_FIRST_MIXER;
int mute = (off % 2);
int idx = off / 2;
if (mute) {
if (cp->type != AUDIO_MIXER_ENUM)
return (EINVAL);
reg = sv_read_indirect(sc, ports[idx].l_port);
cp->un.ord = ((reg & SV_MUTE_BIT) ? 1 : 0);
} else {
if (cp->type != AUDIO_MIXER_VALUE)
return (EINVAL);
if (cp->un.value.num_channels != 1 &&
cp->un.value.num_channels != 2)
return (EINVAL);
if ((ports[idx].r_port == 0 &&
cp->un.value.num_channels != 1) ||
(ports[idx].r_port != 0 &&
cp->un.value.num_channels != 2))
return (EINVAL);
reg = sv_read_indirect(sc, ports[idx].l_port);
reg &= ports[idx].mask;
val = AUDIO_MAX_GAIN - ((reg * AUDIO_MAX_GAIN) / ports[idx].mask);
if (ports[idx].r_port != 0) {
cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = val;
reg = sv_read_indirect(sc, ports[idx].r_port);
reg &= ports[idx].mask;
val = AUDIO_MAX_GAIN - ((reg * AUDIO_MAX_GAIN) / ports[idx].mask);
cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = val;
} else
cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = val;
}
return (0);
}
switch (cp->dev) {
case SV_RECORD_SOURCE:
if (cp->type != AUDIO_MIXER_ENUM)
return (EINVAL);
reg = sv_read_indirect(sc, SV_LEFT_ADC_INPUT_CONTROL);
cp->un.ord = ((reg & SV_REC_SOURCE_MASK) >> SV_REC_SOURCE_SHIFT);
return (0);
case SV_RECORD_GAIN:
if (cp->type != AUDIO_MIXER_VALUE)
return (EINVAL);
if (cp->un.value.num_channels != 1)
return (EINVAL);
reg = sv_read_indirect(sc, SV_LEFT_ADC_INPUT_CONTROL) & SV_REC_GAIN_MASK;
cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] =
(((unsigned int)reg) * AUDIO_MAX_GAIN) / SV_REC_GAIN_MASK;
return (0);
case SV_MIC_BOOST:
if (cp->type != AUDIO_MIXER_ENUM)
return (EINVAL);
reg = sv_read_indirect(sc, SV_LEFT_ADC_INPUT_CONTROL);
cp->un.ord = ((reg & SV_MIC_BOOST_BIT) ? 1 : 0);
return (0);
case SV_SRS_MODE:
if (cp->type != AUDIO_MIXER_ENUM)
return (EINVAL);
reg = sv_read_indirect(sc, SV_SRS_SPACE_CONTROL);
cp->un.ord = ((reg & SV_SRS_SPACE_ONOFF) ? 0 : 1);
return (0);
}
return (EINVAL);
}
static void
sv_init_mixer(sc)
struct sv_softc *sc;
{
mixer_ctrl_t cp;
int i;
cp.type = AUDIO_MIXER_ENUM;
cp.dev = SV_SRS_MODE;
cp.un.ord = 0;
sv_mixer_set_port(sc, &cp);
for (i = 0; i < ARRAY_SIZE(ports); i++) {
if (ports[i].audio == AudioNdac) {
cp.type = AUDIO_MIXER_ENUM;
cp.dev = SV_FIRST_MIXER + i * SV_DEVICES_PER_PORT + 1;
cp.un.ord = 0;
sv_mixer_set_port(sc, &cp);
break;
}
}
}
void *
sv_malloc(addr, direction, size, pool, flags)
void *addr;
int direction;
size_t size;
struct malloc_type *pool;
int flags;
{
struct sv_softc *sc = addr;
struct sv_dma *p;
int error;
p = malloc(sizeof(*p), pool, flags);
if (!p)
return (0);
error = sv_allocmem(sc, size, 16, direction, p);
if (error) {
free(p, pool);
return (0);
}
p->next = sc->sc_dmas;
sc->sc_dmas = p;
return (KERNADDR(p));
}
void
sv_free(addr, ptr, pool)
void *addr;
void *ptr;
struct malloc_type *pool;
{
struct sv_softc *sc = addr;
struct sv_dma **pp, *p;
for (pp = &sc->sc_dmas; (p = *pp) != NULL; pp = &p->next) {
if (KERNADDR(p) == ptr) {
sv_freemem(sc, p);
*pp = p->next;
free(p, pool);
return;
}
}
}
size_t
sv_round_buffersize(addr, direction, size)
void *addr;
int direction;
size_t size;
{
return (size);
}
paddr_t
sv_mappage(addr, mem, off, prot)
void *addr;
void *mem;
off_t off;
int prot;
{
struct sv_softc *sc = addr;
struct sv_dma *p;
if (off < 0)
return (-1);
for (p = sc->sc_dmas; p && KERNADDR(p) != mem; p = p->next)
;
if (!p)
return (-1);
return (bus_dmamem_mmap(sc->sc_dmatag, p->segs, p->nsegs,
off, prot, BUS_DMA_WAITOK));
}
int
sv_get_props(addr)
void *addr;
{
return (AUDIO_PROP_MMAP | AUDIO_PROP_INDEPENDENT | AUDIO_PROP_FULLDUPLEX);
}
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