summaryrefslogtreecommitdiff
path: root/sys
diff options
context:
space:
mode:
Diffstat (limited to 'sys')
-rw-r--r--sys/arch/i386/conf/GENERIC6
-rw-r--r--sys/dev/isa/ad1848var.h3
-rw-r--r--sys/dev/isa/files.isa11
-rw-r--r--sys/dev/isa/pss.c1286
-rw-r--r--sys/dev/isa/pssreg.h165
5 files changed, 3 insertions, 1468 deletions
diff --git a/sys/arch/i386/conf/GENERIC b/sys/arch/i386/conf/GENERIC
index a258198f689..a261e82a444 100644
--- a/sys/arch/i386/conf/GENERIC
+++ b/sys/arch/i386/conf/GENERIC
@@ -1,4 +1,4 @@
-# $OpenBSD: GENERIC,v 1.719 2011/06/26 23:19:10 tedu Exp $
+# $OpenBSD: GENERIC,v 1.720 2011/06/29 17:48:22 tedu Exp $
#
# For further information on compiling OpenBSD kernels, see the config(8)
# man page.
@@ -706,9 +706,6 @@ ipgphy* at mii? # IC Plus IP1000A PHYs
mlphy* at mii? # Micro Linear 6692 PHY
ukphy* at mii? # "unknown" PHYs
-pss0 at isa? port 0x220 irq 7 drq 6 # Personal Sound System
-sp0 at pss0 port 0x530 irq 10 drq 0 # sound port driver
-
eap* at pci? # Ensoniq AudioPCI S5016
eso* at pci? # ESS Solo-1 PCI AudioDrive
sv* at pci? # S3 SonicVibes (S3 617)
@@ -754,7 +751,6 @@ spkr0 at pcppi? # PC speaker
audio* at sb?
audio* at gus?
audio* at pas?
-audio* at sp?
audio* at ess?
audio* at wss?
audio* at ym?
diff --git a/sys/dev/isa/ad1848var.h b/sys/dev/isa/ad1848var.h
index b5d1e73bed9..0e193f77465 100644
--- a/sys/dev/isa/ad1848var.h
+++ b/sys/dev/isa/ad1848var.h
@@ -1,4 +1,4 @@
-/* $OpenBSD: ad1848var.h,v 1.13 2010/06/30 11:21:35 jakemsr Exp $ */
+/* $OpenBSD: ad1848var.h,v 1.14 2011/06/29 17:48:22 tedu Exp $ */
/* $NetBSD: ad1848var.h,v 1.22 1998/01/19 22:18:26 augustss Exp $ */
/*
@@ -92,7 +92,6 @@ struct ad1848_softc {
void *sc_parg; /* play arg for sc_intr() */
void *sc_rarg; /* rec arg for sc_intr() */
- /* Only used by pss XXX */
int sc_iobase;
};
diff --git a/sys/dev/isa/files.isa b/sys/dev/isa/files.isa
index 1b407e9d74a..bbc475d949d 100644
--- a/sys/dev/isa/files.isa
+++ b/sys/dev/isa/files.isa
@@ -1,4 +1,4 @@
-# $OpenBSD: files.isa,v 1.110 2011/06/28 20:19:19 matthieu Exp $
+# $OpenBSD: files.isa,v 1.111 2011/06/29 17:48:22 tedu Exp $
# $NetBSD: files.isa,v 1.21 1996/05/16 03:45:55 mycroft Exp $
#
# Config file and device description for machine-independent ISA code.
@@ -241,15 +241,6 @@ define ics2101
file dev/isa/ics2101.c ics2101
-# Audio systems based on Echo Speech Corp. ESC61[45] ASICs
-device pss {[port = -1], [size = 0],
- [iomem = -1], [iosiz = 0],
- [irq = -1], [drq = -1]}
-attach pss at isa
-device sp: audio, isa_dma, ad1848, auconv
-attach sp at pss
-file dev/isa/pss.c pss needs-flag
-
# Microsoft Windows Sound System
device wss: audio, isa_dma, ad1848, auconv
file dev/isa/wss.c wss needs-flag
diff --git a/sys/dev/isa/pss.c b/sys/dev/isa/pss.c
deleted file mode 100644
index 57d0a2d8641..00000000000
--- a/sys/dev/isa/pss.c
+++ /dev/null
@@ -1,1286 +0,0 @@
-/* $OpenBSD: pss.c,v 1.25 2011/06/29 12:17:40 tedu Exp $ */
-/* $NetBSD: pss.c,v 1.38 1998/01/12 09:43:44 thorpej Exp $ */
-
-/*
- * Copyright (c) 1994 John Brezak
- * 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.
- *
- */
-
-/*
- * Copyright (c) 1993 Analog Devices Inc. All rights reserved
- *
- * Portions provided by Marc.Hoffman@analog.com and
- * Greg.Yukna@analog.com .
- *
- */
-
-/*
- * Todo:
- * - Provide PSS driver to access DSP
- * - Provide MIDI driver to access MPU
- * - Finish support for CD drive (Sony and SCSI)
- */
-
-#include <sys/param.h>
-#include <sys/systm.h>
-#include <sys/errno.h>
-#include <sys/ioctl.h>
-#include <sys/syslog.h>
-#include <sys/device.h>
-#include <sys/proc.h>
-#include <sys/buf.h>
-
-#include <machine/cpu.h>
-#include <machine/intr.h>
-#include <machine/bus.h>
-
-#include <sys/audioio.h>
-#include <dev/audio_if.h>
-
-#include <dev/isa/isavar.h>
-#include <dev/isa/isadmavar.h>
-
-#include <dev/isa/ad1848var.h>
-#include <dev/isa/wssreg.h>
-#include <dev/isa/pssreg.h>
-
-/* XXX Default WSS base */
-#define WSS_BASE_ADDRESS 0x0530
-
-/*
- * Mixer devices
- */
-#define PSS_MIC_IN_LVL 0
-#define PSS_LINE_IN_LVL 1
-#define PSS_DAC_LVL 2
-#define PSS_REC_LVL 3
-#define PSS_MON_LVL 4
-#define PSS_MASTER_VOL 5
-#define PSS_MASTER_TREBLE 6
-#define PSS_MASTER_BASS 7
-#define PSS_MIC_IN_MUTE 8
-#define PSS_LINE_IN_MUTE 9
-#define PSS_DAC_MUTE 10
-
-#define PSS_OUTPUT_MODE 11
-#define PSS_SPKR_MONO 0
-#define PSS_SPKR_STEREO 1
-#define PSS_SPKR_PSEUDO 2
-#define PSS_SPKR_SPATIAL 3
-
-#define PSS_RECORD_SOURCE 12
-
-/* Classes */
-#define PSS_INPUT_CLASS 13
-#define PSS_RECORD_CLASS 14
-#define PSS_MONITOR_CLASS 15
-#define PSS_OUTPUT_CLASS 16
-
-
-struct pss_softc {
- struct device sc_dev; /* base device */
- void *sc_ih; /* interrupt vectoring */
-
- int sc_iobase; /* I/O port base address */
- int sc_drq; /* dma channel */
-
- struct ad1848_softc *ad1848_sc;
-
- int out_port;
-
- struct ad1848_volume master_volume;
- int master_mode;
-
- int monitor_treble;
- int monitor_bass;
-
- int mic_mute, cd_mute, dac_mute;
-};
-
-#ifdef AUDIO_DEBUG
-#define DPRINTF(x) if (pssdebug) printf x
-int pssdebug = 0;
-#else
-#define DPRINTF(x)
-#endif
-
-int pssprobe(struct device *, void *, void *);
-void pssattach(struct device *, struct device *, void *);
-
-int spprobe(struct device *, void *, void *);
-void spattach(struct device *, struct device *, void *);
-
-int pssintr(void *);
-
-int pss_speaker_ctl(void *, int);
-
-int pss_getdev(void *, struct audio_device *);
-
-int pss_mixer_set_port(void *, mixer_ctrl_t *);
-int pss_mixer_get_port(void *, mixer_ctrl_t *);
-int pss_query_devinfo(void *, mixer_devinfo_t *);
-
-#ifdef PSS_DSP
-void pss_dspwrite(struct pss_softc *, int);
-#endif
-void pss_setaddr(int, int);
-int pss_setint(int, int);
-int pss_setdma(int, int);
-int pss_testirq(struct pss_softc *, int);
-int pss_testdma(struct pss_softc *, int);
-#ifdef AUDIO_DEBUG
-void pss_dump_regs(struct pss_softc *);
-#endif
-int pss_set_master_gain(struct pss_softc *, struct ad1848_volume *);
-int pss_set_master_mode(struct pss_softc *, int);
-int pss_set_treble(struct pss_softc *, u_int);
-int pss_set_bass(struct pss_softc *, u_int);
-int pss_get_master_gain(struct pss_softc *, struct ad1848_volume *);
-int pss_get_master_mode(struct pss_softc *, u_int *);
-int pss_get_treble(struct pss_softc *, u_char *);
-int pss_get_bass(struct pss_softc *, u_char *);
-
-#ifdef AUDIO_DEBUG
-void wss_dump_regs(struct ad1848_softc *);
-#endif
-
-/*
- * Define our interface to the higher level audio driver.
- */
-
-struct audio_hw_if pss_audio_if = {
- ad1848_open,
- ad1848_close,
- NULL,
- ad1848_query_encoding,
- ad1848_set_params,
- ad1848_round_blocksize,
- ad1848_commit_settings,
- NULL,
- NULL,
- NULL,
- NULL,
- ad1848_halt_output,
- ad1848_halt_input,
- pss_speaker_ctl,
- pss_getdev,
- NULL,
- pss_mixer_set_port,
- pss_mixer_get_port,
- pss_query_devinfo,
- ad1848_malloc,
- ad1848_free,
- ad1848_round,
- ad1848_mappage,
- ad1848_get_props,
- ad1848_trigger_output,
- ad1848_trigger_input,
- NULL
-};
-
-
-/* Interrupt translation for WSS config */
-static u_char wss_interrupt_bits[16] = {
- 0xff, 0xff, 0xff, 0xff,
- 0xff, 0xff, 0xff, 0x08,
- 0xff, 0x10, 0x18, 0x20,
- 0xff, 0xff, 0xff, 0xff
-};
-/* ditto for WSS DMA channel */
-static u_char wss_dma_bits[4] = {1, 2, 0, 3};
-
-struct cfattach pss_ca = {
- sizeof(struct pss_softc), pssprobe, pssattach
-};
-
-struct cfdriver pss_cd = {
- NULL, "pss", DV_DULL, 1
-};
-
-struct cfattach sp_ca = {
- sizeof(struct ad1848_softc), spprobe, spattach
-};
-
-struct cfdriver sp_cd = {
- NULL, "sp", DV_DULL
-};
-
-struct audio_device pss_device = {
- "pss,ad1848",
- "",
- "PSS"
-};
-
-#ifdef PSS_DSP
-void
-pss_dspwrite(sc, data)
- struct pss_softc *sc;
- int data;
-{
- int i;
- int pss_base = sc->sc_iobase;
-
- /*
- * Note! the i<5000000 is an emergency exit. The dsp_command() is sometimes
- * called while interrupts are disabled. This means that the timer is
- * disabled also. However the timeout situation is a abnormal condition.
- * Normally the DSP should be ready to accept commands after just couple of
- * loops.
- */
- for (i = 0; i < 5000000; i++) {
- if (inw(pss_base+PSS_STATUS) & PSS_WRITE_EMPTY) {
- outw(pss_base+PSS_DATA, data);
- return;
- }
- }
- printf ("pss: DSP Command (%04x) Timeout.\n", data);
-}
-#endif /* PSS_DSP */
-
-void
-pss_setaddr(addr, configAddr)
- int addr;
- int configAddr;
-{
- int val;
-
- val = inw(configAddr);
- val &= ADDR_MASK;
- val |= (addr << 4);
- outw(configAddr,val);
-}
-
-/* pss_setint
- * This function sets the correct bits in the
- * configuration register to
- * enable the chosen interrupt.
- */
-int
-pss_setint(intNum, configAddress)
- int intNum;
- int configAddress;
-{
- int val;
-
- switch(intNum) {
- case 3:
- val = inw(configAddress);
- val &= INT_MASK;
- val |= INT_3_BITS;
- break;
- case 5:
- val = inw(configAddress);
- val &= INT_MASK;
- val |= INT_5_BITS;
- break;
- case 7:
- val = inw(configAddress);
- val &= INT_MASK;
- val |= INT_7_BITS;
- break;
- case 9:
- val = inw(configAddress);
- val &= INT_MASK;
- val |= INT_9_BITS;
- break;
- case 10:
- val = inw(configAddress);
- val &= INT_MASK;
- val |= INT_10_BITS;
- break;
- case 11:
- val = inw(configAddress);
- val &= INT_MASK;
- val |= INT_11_BITS;
- break;
- case 12:
- val = inw(configAddress);
- val &= INT_MASK;
- val |= INT_12_BITS;
- break;
- default:
- DPRINTF(("pss_setint: invalid irq (%d)\n", intNum));
- return 1;
- }
- outw(configAddress,val);
- return 0;
-}
-
-int
-pss_setdma(dmaNum, configAddress)
- int dmaNum;
- int configAddress;
-{
- int val;
-
- switch(dmaNum) {
- case 0:
- val = inw(configAddress);
- val &= DMA_MASK;
- val |= DMA_0_BITS;
- break;
- case 1:
- val = inw(configAddress);
- val &= DMA_MASK;
- val |= DMA_1_BITS;
- break;
- case 3:
- val = inw(configAddress);
- val &= DMA_MASK;
- val |= DMA_3_BITS;
- break;
- case 5:
- val = inw(configAddress);
- val &= DMA_MASK;
- val |= DMA_5_BITS;
- break;
- case 6:
- val = inw(configAddress);
- val &= DMA_MASK;
- val |= DMA_6_BITS;
- break;
- case 7:
- val = inw(configAddress);
- val &= DMA_MASK;
- val |= DMA_7_BITS;
- break;
- default:
- DPRINTF(("pss_setdma: invalid drq (%d)\n", dmaNum));
- return 1;
- }
- outw(configAddress, val);
- return 0;
-}
-
-/*
- * This function tests an interrupt number to see if
- * it is available. It takes the interrupt button
- * as its argument and returns TRUE if the interrupt
- * is ok.
-*/
-int
-pss_testirq(struct pss_softc *sc, int intNum)
-{
- int config = sc->sc_iobase + PSS_CONFIG;
- int val;
- int ret;
- int i;
-
- /* Set the interrupt bits */
- switch(intNum) {
- case 3:
- val = inw(config);
- val &= INT_MASK; /* Special: 0 */
- break;
- case 5:
- val = inw(config);
- val &= INT_MASK;
- val |= INT_TEST_BIT | INT_5_BITS;
- break;
- case 7:
- val = inw(config);
- val &= INT_MASK;
- val |= INT_TEST_BIT | INT_7_BITS;
- break;
- case 9:
- val = inw(config);
- val &= INT_MASK;
- val |= INT_TEST_BIT | INT_9_BITS;
- break;
- case 10:
- val = inw(config);
- val &= INT_MASK;
- val |= INT_TEST_BIT | INT_10_BITS;
- break;
- case 11:
- val = inw(config);
- val &= INT_MASK;
- val |= INT_TEST_BIT | INT_11_BITS;
- break;
- case 12:
- val = inw(config);
- val &= INT_MASK;
- val |= INT_TEST_BIT | INT_12_BITS;
- break;
- default:
- DPRINTF(("pss_testirq: invalid irq (%d)\n", intNum));
- return 0;
- }
- outw(config, val);
-
- /* Check if the interrupt is in use */
- /* Do it a few times in case there is a delay */
- ret = 0;
- for (i = 0; i < 5; i++) {
- val = inw(config);
- if (val & INT_TEST_PASS) {
- ret = 1;
- break;
- }
- }
-
- /* Clear the Test bit and the interrupt bits */
- val = inw(config);
- val &= INT_TEST_BIT_MASK & INT_MASK;
- outw(config, val);
- return(ret);
-}
-
-/*
- * This function tests a dma channel to see if
- * it is available. It takes the DMA channel button
- * as its argument and returns TRUE if the channel
- * is ok.
- */
-int
-pss_testdma(sc, dmaNum)
- struct pss_softc *sc;
- int dmaNum;
-{
- int config = sc->sc_iobase + PSS_CONFIG;
- int val;
- int i, ret;
-
- switch (dmaNum) {
- case 0:
- val = inw(config);
- val &= DMA_MASK;
- val |= DMA_TEST_BIT | DMA_0_BITS;
- break;
- case 1:
- val = inw(config);
- val &= DMA_MASK;
- val |= DMA_TEST_BIT | DMA_1_BITS;
- break;
- case 3:
- val = inw(config);
- val &= DMA_MASK;
- val |= DMA_TEST_BIT | DMA_3_BITS;
- break;
- case 5:
- val = inw(config);
- val &= DMA_MASK;
- val |= DMA_TEST_BIT | DMA_5_BITS;
- break;
- case 6:
- val = inw(config);
- val &= DMA_MASK;
- val |= DMA_TEST_BIT | DMA_6_BITS;
- break;
- case 7:
- val = inw(config);
- val &= DMA_MASK;
- val |= DMA_TEST_BIT | DMA_7_BITS;
- break;
- default:
- DPRINTF(("pss_testdma: invalid drq (%d)\n", dmaNum));
- return 0;
- }
- outw(config, val);
-
- /* Check if the DMA channel is in use */
- /* Do it a few times in case there is a delay */
- ret = 0;
- for (i = 0; i < 3; i++) {
- val = inw(config);
- if (val & DMA_TEST_PASS) {
- ret = 1;
- break;
- }
- }
-
- /* Clear the Test bit and the DMA bits */
- val = inw(config);
- val &= DMA_TEST_BIT_MASK & DMA_MASK;
- outw(config, val);
- return(ret);
-}
-
-#ifdef AUDIO_DEBUG
-void
-wss_dump_regs(sc)
- struct ad1848_softc *sc;
-{
-
- printf("WSS reg: status=%02x\n",
- (u_char)inb(sc->sc_iobase-WSS_CODEC+WSS_STATUS));
-}
-
-void
-pss_dump_regs(sc)
- struct pss_softc *sc;
-{
-
- printf("PSS regs: status=%04x vers=%04x ",
- (u_short)inw(sc->sc_iobase+PSS_STATUS),
- (u_short)inw(sc->sc_iobase+PSS_ID_VERS));
-
- printf("config=%04x wss_config=%04x\n",
- (u_short)inw(sc->sc_iobase+PSS_CONFIG),
- (u_short)inw(sc->sc_iobase+PSS_WSS_CONFIG));
-}
-#endif
-
-/*
- * Probe for the PSS hardware.
- */
-int
-pssprobe(parent, self, aux)
- struct device *parent;
- void *self;
- void *aux;
-{
- struct pss_softc *sc = self;
- struct isa_attach_args *ia = aux;
- int iobase = ia->ia_iobase;
-
- if (!PSS_BASE_VALID(iobase)) {
- DPRINTF(("pss: configured iobase %x invalid\n", iobase));
- return 0;
- }
-
- /* Need to probe for iobase when IOBASEUNK {0x220 0x240} */
- if (iobase == IOBASEUNK) {
-
- iobase = 0x220;
- if ((inw(iobase+PSS_ID_VERS) & 0xff00) == 0x4500)
- goto pss_found;
-
- iobase = 0x240;
- if ((inw(iobase+PSS_ID_VERS) & 0xff00) == 0x4500)
- goto pss_found;
-
- DPRINTF(("pss: no PSS found (at 0x220 or 0x240)\n"));
- return 0;
- }
- else if ((inw(iobase+PSS_ID_VERS) & 0xff00) != 0x4500) {
- DPRINTF(("pss: not a PSS - %x\n", inw(iobase+PSS_ID_VERS)));
- return 0;
- }
-
-pss_found:
- sc->sc_iobase = iobase;
-
- /* Clear WSS config */
- pss_setaddr(WSS_BASE_ADDRESS, sc->sc_iobase+PSS_WSS_CONFIG); /* XXX! */
- outb(WSS_BASE_ADDRESS+WSS_CONFIG, 0);
-
- /* Clear config registers (POR reset state) */
- outw(sc->sc_iobase+PSS_CONFIG, 0);
- outw(sc->sc_iobase+PSS_WSS_CONFIG, 0);
- outw(sc->sc_iobase+SB_CONFIG, 0);
- outw(sc->sc_iobase+MIDI_CONFIG, 0);
- outw(sc->sc_iobase+CD_CONFIG, 0);
-
- if (ia->ia_irq == IRQUNK) {
- int i;
- for (i = 0; i < 16; i++) {
- if (pss_testirq(sc, i) != 0)
- break;
- }
- if (i == 16) {
- DPRINTF(("pss: unable to locate free IRQ channel\n"));
- return 0;
- }
- else {
- ia->ia_irq = i;
- DPRINTF(("pss: found IRQ %d free\n", i));
- }
- }
- else {
- if (pss_testirq(sc, ia->ia_irq) == 0) {
- DPRINTF(("pss: configured IRQ unavailable (%d)\n", ia->ia_irq));
- return 0;
- }
- }
-
- /* XXX Need to deal with DRQUNK */
- if (pss_testdma(sc, ia->ia_drq) == 0) {
- DPRINTF(("pss: configured DMA channel unavailable (%d)\n", ia->ia_drq));
- return 0;
- }
-
- ia->ia_iosize = PSS_NPORT;
-
- /* Initialize PSS irq and dma */
- pss_setint(ia->ia_irq, sc->sc_iobase+PSS_CONFIG);
- pss_setdma(sc->sc_drq, sc->sc_iobase+PSS_CONFIG);
-
- return 1;
-}
-
-/*
- * Probe for the Soundport (ad1848)
- */
-int
-spprobe(parent, match, aux)
- struct device *parent;
- void *match, *aux;
-{
- struct ad1848_softc *sc = match;
- struct pss_softc *pc = (void *) parent;
- struct cfdata *cf = (void *)sc->sc_dev.dv_cfdata;
- struct isa_attach_args *ia = aux;
- u_char bits;
- int i;
-
- sc->sc_iot = ia->ia_iot;
- sc->sc_iobase = cf->cf_iobase + WSS_CODEC;
-
- /* Set WSS io address */
- pss_setaddr(cf->cf_iobase, pc->sc_iobase+PSS_WSS_CONFIG);
-
- /* Is there an ad1848 chip at the WSS iobase ? */
- if (ad1848_probe(sc) == 0) {
- DPRINTF(("sp: no ad1848 ? iobase=%x\n", sc->sc_iobase));
- return 0;
- }
-
- /* Setup WSS interrupt and DMA if auto */
- if (cf->cf_irq == IRQUNK) {
-
- /* Find unused IRQ for WSS */
- for (i = 0; i < 12; i++) {
- if (wss_interrupt_bits[i] != 0xff) {
- if (pss_testirq(pc, i))
- break;
- }
- }
- if (i == 12) {
- DPRINTF(("sp: unable to locate free IRQ for WSS\n"));
- return 0;
- }
- else {
- cf->cf_irq = i;
- sc->sc_irq = i;
- DPRINTF(("sp: found IRQ %d free\n", i));
- }
- }
- else {
- sc->sc_irq = cf->cf_irq;
- if (pss_testirq(pc, sc->sc_irq) == 0) {
- DPRINTF(("sp: configured IRQ unavailable (%d)\n", sc->sc_irq));
- return 0;
- }
- }
-
- if (cf->cf_drq == DRQUNK) {
- /* Find unused DMA channel for WSS */
- for (i = 0; i < 4; i++) {
- if (wss_dma_bits[i]) {
- if (pss_testdma(pc, i))
- break;
- }
- }
- if (i == 4) {
- DPRINTF(("sp: unable to locate free DMA channel for WSS\n"));
- return 0;
- }
- else {
- sc->sc_drq = cf->cf_drq = i;
- DPRINTF(("sp: found DMA %d free\n", i));
- }
- }
- else {
- if (pss_testdma(pc, sc->sc_drq) == 0) {
- DPRINTF(("sp: configured DMA channel unavailable (%d)\n", sc->sc_drq));
- return 0;
- }
- sc->sc_drq = cf->cf_drq;
- }
- sc->sc_recdrq = sc->sc_drq;
-
- /* Set WSS config registers */
- if ((bits = wss_interrupt_bits[sc->sc_irq]) == 0xff) {
- DPRINTF(("sp: invalid interrupt configuration (irq=%d)\n", sc->sc_irq));
- return 0;
- }
-
- outb(sc->sc_iobase+WSS_CONFIG, (bits | 0x40));
- if ((inb(sc->sc_iobase+WSS_STATUS) & 0x40) == 0) /* XXX What do these bits mean ? */
- DPRINTF(("sp: IRQ %x\n", inb(sc->sc_iobase+WSS_STATUS)));
-
- outb(sc->sc_iobase+WSS_CONFIG, (bits | wss_dma_bits[sc->sc_drq]));
-
- pc->ad1848_sc = sc;
- sc->parent = pc;
-
- return 1;
-}
-
-/*
- * Attach hardware to driver, attach hardware driver to audio
- * pseudo-device driver .
- */
-void
-pssattach(parent, self, aux)
- struct device *parent, *self;
- void *aux;
-{
- struct pss_softc *sc = (struct pss_softc *)self;
- struct isa_attach_args *ia = (struct isa_attach_args *)aux;
- int iobase = ia->ia_iobase;
- u_char vers;
- struct ad1848_volume vol = {150, 150};
-
- sc->sc_iobase = iobase;
- sc->sc_drq = ia->ia_drq;
-
- /* Setup interrupt handler for PSS */
- sc->sc_ih = isa_intr_establish(ia->ia_ic, ia->ia_irq, IST_EDGE, IPL_AUDIO,
- pssintr, sc, sc->sc_dev.dv_xname);
-
- vers = (inw(sc->sc_iobase+PSS_ID_VERS)&0xff) - 1;
- printf(": ESC614%c\n", (vers > 0)?'A'+vers:' ');
-
- (void)config_found(self, ia->ia_ic, NULL); /* XXX */
-
- sc->out_port = PSS_MASTER_VOL;
-
- (void)pss_set_master_mode(sc, PSS_SPKR_STEREO);
- (void)pss_set_master_gain(sc, &vol);
- (void)pss_set_treble(sc, AUDIO_MAX_GAIN/2);
- (void)pss_set_bass(sc, AUDIO_MAX_GAIN/2);
-
- audio_attach_mi(&pss_audio_if, sc->ad1848_sc, &sc->ad1848_sc->sc_dev);
-}
-
-void
-spattach(parent, self, aux)
- struct device *parent, *self;
- void *aux;
-{
- struct ad1848_softc *sc = (struct ad1848_softc *)self;
- struct cfdata *cf = (void *)sc->sc_dev.dv_cfdata;
- isa_chipset_tag_t ic = aux; /* XXX */
- int iobase = cf->cf_iobase;
-
- sc->sc_iobase = iobase;
- sc->sc_drq = cf->cf_drq;
-
- sc->sc_ih = isa_intr_establish(ic, cf->cf_irq, IST_EDGE, IPL_AUDIO,
- ad1848_intr, sc, sc->sc_dev.dv_xname);
-
- sc->sc_isa = parent->dv_parent;
-
- ad1848_attach(sc);
-
- printf("\n");
-}
-
-int
-pss_set_master_gain(sc, gp)
- struct pss_softc *sc;
- struct ad1848_volume *gp;
-{
- DPRINTF(("pss_set_master_gain: %d:%d\n", gp->left, gp->right));
-
-#ifdef PSS_DSP
- if (gp->left > PHILLIPS_VOL_MAX)
- gp->left = PHILLIPS_VOL_MAX;
- if (gp->left < PHILLIPS_VOL_MIN)
- gp->left = PHILLIPS_VOL_MIN;
- if (gp->right > PHILLIPS_VOL_MAX)
- gp->right = PHILLIPS_VOL_MAX;
- if (gp->right < PHILLIPS_VOL_MIN)
- gp->right = PHILLIPS_VOL_MIN;
-
- pss_dspwrite(sc, SET_MASTER_COMMAND);
- pss_dspwrite(sc, MASTER_VOLUME_LEFT|(PHILLIPS_VOL_CONSTANT + gp->left / PHILLIPS_VOL_STEP));
- pss_dspwrite(sc, SET_MASTER_COMMAND);
- pss_dspwrite(sc, MASTER_VOLUME_RIGHT|(PHILLIPS_VOL_CONSTANT + gp->right / PHILLIPS_VOL_STEP));
-#endif
-
- sc->master_volume = *gp;
- return(0);
-}
-
-int
-pss_set_master_mode(sc, mode)
- struct pss_softc *sc;
- int mode;
-{
- short phillips_mode;
-
- DPRINTF(("pss_set_master_mode: %d\n", mode));
-
- if (mode == PSS_SPKR_STEREO)
- phillips_mode = PSS_STEREO;
- else if (mode == PSS_SPKR_PSEUDO)
- phillips_mode = PSS_PSEUDO;
- else if (mode == PSS_SPKR_SPATIAL)
- phillips_mode = PSS_SPATIAL;
- else if (mode == PSS_SPKR_MONO)
- phillips_mode = PSS_MONO;
- else
- return (EINVAL);
-
-#ifdef PSS_DSP
- pss_dspwrite(sc, SET_MASTER_COMMAND);
- pss_dspwrite(sc, MASTER_SWITCH | mode);
-#endif
-
- sc->master_mode = mode;
-
- return(0);
-}
-
-int
-pss_set_treble(sc, treb)
- struct pss_softc *sc;
- u_int treb;
-{
- DPRINTF(("pss_set_treble: %d\n", treb));
-
-#ifdef PSS_DSP
- if (treb > PHILLIPS_TREBLE_MAX)
- treb = PHILLIPS_TREBLE_MAX;
- if (treb < PHILLIPS_TREBLE_MIN)
- treb = PHILLIPS_TREBLE_MIN;
- pss_dspwrite(sc, SET_MASTER_COMMAND);
- pss_dspwrite(sc, MASTER_TREBLE|(PHILLIPS_TREBLE_CONSTANT + treb / PHILLIPS_TREBLE_STEP));
-#endif
-
- sc->monitor_treble = treb;
-
- return(0);
-}
-
-int
-pss_set_bass(sc, bass)
- struct pss_softc *sc;
- u_int bass;
-{
- DPRINTF(("pss_set_bass: %d\n", bass));
-
-#ifdef PSS_DSP
- if (bass > PHILLIPS_BASS_MAX)
- bass = PHILLIPS_BASS_MAX;
- if (bass < PHILLIPS_BASS_MIN)
- bass = PHILLIPS_BASS_MIN;
- pss_dspwrite(sc, SET_MASTER_COMMAND);
- pss_dspwrite(sc, MASTER_BASS|(PHILLIPS_BASS_CONSTANT + bass / PHILLIPS_BASS_STEP));
-#endif
-
- sc->monitor_bass = bass;
-
- return(0);
-}
-
-int
-pss_get_master_gain(sc, gp)
- struct pss_softc *sc;
- struct ad1848_volume *gp;
-{
- *gp = sc->master_volume;
- return(0);
-}
-
-int
-pss_get_master_mode(sc, mode)
- struct pss_softc *sc;
- u_int *mode;
-{
- *mode = sc->master_mode;
- return(0);
-}
-
-int
-pss_get_treble(sc, tp)
- struct pss_softc *sc;
- u_char *tp;
-{
- *tp = sc->monitor_treble;
- return(0);
-}
-
-int
-pss_get_bass(sc, bp)
- struct pss_softc *sc;
- u_char *bp;
-{
- *bp = sc->monitor_bass;
- return(0);
-}
-
-int
-pss_speaker_ctl(addr, newstate)
- void *addr;
- int newstate;
-{
- return(0);
-}
-
-int
-pssintr(arg)
- void *arg;
-{
- struct pss_softc *sc = arg;
- u_short sr;
-
- sr = inw(sc->sc_iobase+PSS_STATUS);
-
- DPRINTF(("pssintr: sc=%p st=%x\n", sc, sr));
-
- /* Acknowledge intr */
- outw(sc->sc_iobase+PSS_IRQ_ACK, 0);
-
- /* Is it one of ours ? */
- if (sr & (PSS_WRITE_EMPTY|PSS_READ_FULL|PSS_IRQ|PSS_DMQ_TC)) {
- /* XXX do something */
- return 1;
- }
-
- return 0;
-}
-
-int
-pss_getdev(addr, retp)
- void *addr;
- struct audio_device *retp;
-{
- DPRINTF(("pss_getdev: retp=%p\n", retp));
-
- *retp = pss_device;
- return 0;
-}
-
-static ad1848_devmap_t mappings[] = {
-{ PSS_MIC_IN_LVL, AD1848_KIND_LVL, AD1848_AUX2_CHANNEL },
-{ PSS_LINE_IN_LVL, AD1848_KIND_LVL, AD1848_AUX1_CHANNEL },
-{ PSS_DAC_LVL, AD1848_KIND_LVL, AD1848_DAC_CHANNEL },
-{ PSS_MON_LVL, AD1848_KIND_LVL, AD1848_MONO_CHANNEL },
-{ PSS_MIC_IN_MUTE, AD1848_KIND_MUTE, AD1848_AUX2_CHANNEL },
-{ PSS_LINE_IN_MUTE, AD1848_KIND_MUTE, AD1848_AUX1_CHANNEL },
-{ PSS_DAC_MUTE, AD1848_KIND_MUTE, AD1848_DAC_CHANNEL },
-{ PSS_REC_LVL, AD1848_KIND_RECORDGAIN, -1 },
-{ PSS_RECORD_SOURCE, AD1848_KIND_RECORDSOURCE, -1}
-};
-
-static int nummap = sizeof(mappings) / sizeof(mappings[0]);
-
-int
-pss_mixer_set_port(addr, cp)
- void *addr;
- mixer_ctrl_t *cp;
-{
- struct ad1848_softc *ac = addr;
- struct pss_softc *sc = ac->parent;
- struct ad1848_volume vol;
- int error = ad1848_mixer_set_port(ac, mappings, nummap, cp);
-
- if (error != ENXIO)
- return (error);
-
- switch (cp->dev) {
- case PSS_MASTER_VOL: /* master volume */
- if (cp->type == AUDIO_MIXER_VALUE) {
- if (ad1848_to_vol(cp, &vol))
- error = pss_set_master_gain(sc, &vol);
- }
- break;
-
- case PSS_OUTPUT_MODE:
- if (cp->type == AUDIO_MIXER_ENUM)
- error = pss_set_master_mode(sc, cp->un.ord);
- break;
-
- case PSS_MASTER_TREBLE: /* master treble */
- if (cp->type == AUDIO_MIXER_VALUE && cp->un.value.num_channels == 1)
- error = pss_set_treble(sc, (u_char)cp->un.value.level[AUDIO_MIXER_LEVEL_MONO]);
- break;
-
- case PSS_MASTER_BASS: /* master bass */
- if (cp->type == AUDIO_MIXER_VALUE && cp->un.value.num_channels == 1)
- error = pss_set_bass(sc, (u_char)cp->un.value.level[AUDIO_MIXER_LEVEL_MONO]);
- break;
-
- default:
- return ENXIO;
- /*NOTREACHED*/
- }
-
- return 0;
-}
-
-int
-pss_mixer_get_port(addr, cp)
- void *addr;
- mixer_ctrl_t *cp;
-{
- struct ad1848_softc *ac = addr;
- struct pss_softc *sc = ac->parent;
- struct ad1848_volume vol;
- u_char eq;
- int error = ad1848_mixer_get_port(ac, mappings, nummap, cp);
-
- if (error != ENXIO)
- return (error);
-
- error = EINVAL;
-
- switch (cp->dev) {
- case PSS_MASTER_VOL: /* master volume */
- if (cp->type == AUDIO_MIXER_VALUE) {
- error = pss_get_master_gain(sc, &vol);
- if (!error)
- ad1848_from_vol(cp, &vol);
- }
- break;
-
- case PSS_MASTER_TREBLE: /* master treble */
- if (cp->type == AUDIO_MIXER_VALUE && cp->un.value.num_channels == 1) {
- error = pss_get_treble(sc, &eq);
- if (!error)
- cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = eq;
- }
- break;
-
- case PSS_MASTER_BASS: /* master bass */
- if (cp->type == AUDIO_MIXER_VALUE && cp->un.value.num_channels == 1) {
- error = pss_get_bass(sc, &eq);
- if (!error)
- cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = eq;
- }
- break;
-
- case PSS_OUTPUT_MODE:
- if (cp->type == AUDIO_MIXER_ENUM)
- error = pss_get_master_mode(sc, &cp->un.ord);
- break;
-
- default:
- error = ENXIO;
- break;
- }
-
- return(error);
-}
-
-int
-pss_query_devinfo(addr, dip)
- void *addr;
- mixer_devinfo_t *dip;
-{
- DPRINTF(("pss_query_devinfo: index=%d\n", dip->index));
-
- switch(dip->index) {
- case PSS_MIC_IN_LVL: /* Microphone */
- dip->type = AUDIO_MIXER_VALUE;
- dip->mixer_class = PSS_INPUT_CLASS;
- dip->prev = AUDIO_MIXER_LAST;
- dip->next = PSS_MIC_IN_MUTE;
- strlcpy(dip->label.name, AudioNmicrophone, sizeof dip->label.name);
- dip->un.v.num_channels = 2;
- strlcpy(dip->un.v.units.name, AudioNvolume,
- sizeof dip->un.v.units.name);
- break;
-
- case PSS_LINE_IN_LVL: /* line/CD */
- dip->type = AUDIO_MIXER_VALUE;
- dip->mixer_class = PSS_INPUT_CLASS;
- dip->prev = AUDIO_MIXER_LAST;
- dip->next = PSS_LINE_IN_MUTE;
- strlcpy(dip->label.name, AudioNcd, sizeof dip->label.name);
- dip->un.v.num_channels = 2;
- strlcpy(dip->un.v.units.name, AudioNvolume,
- sizeof dip->un.v.units.name);
- break;
-
- case PSS_DAC_LVL: /* dacout */
- dip->type = AUDIO_MIXER_VALUE;
- dip->mixer_class = PSS_INPUT_CLASS;
- dip->prev = AUDIO_MIXER_LAST;
- dip->next = PSS_DAC_MUTE;
- strlcpy(dip->label.name, AudioNdac, sizeof dip->label.name);
- dip->un.v.num_channels = 2;
- strlcpy(dip->un.v.units.name, AudioNvolume,
- sizeof dip->un.v.units.name);
- break;
-
- case PSS_REC_LVL: /* record level */
- dip->type = AUDIO_MIXER_VALUE;
- dip->mixer_class = PSS_RECORD_CLASS;
- dip->prev = AUDIO_MIXER_LAST;
- dip->next = PSS_RECORD_SOURCE;
- strlcpy(dip->label.name, AudioNrecord, sizeof dip->label.name);
- dip->un.v.num_channels = 2;
- strlcpy(dip->un.v.units.name, AudioNvolume,
- sizeof dip->un.v.units.name);
- break;
-
- case PSS_MON_LVL: /* monitor level */
- dip->type = AUDIO_MIXER_VALUE;
- dip->mixer_class = PSS_MONITOR_CLASS;
- dip->next = dip->prev = AUDIO_MIXER_LAST;
- strlcpy(dip->label.name, AudioNmonitor, sizeof dip->label.name);
- dip->un.v.num_channels = 1;
- strlcpy(dip->un.v.units.name, AudioNvolume,
- sizeof dip->un.v.units.name);
- break;
-
- case PSS_MASTER_VOL: /* master volume */
- dip->type = AUDIO_MIXER_VALUE;
- dip->mixer_class = PSS_OUTPUT_CLASS;
- dip->prev = AUDIO_MIXER_LAST;
- dip->next = PSS_OUTPUT_MODE;
- strlcpy(dip->label.name, AudioNmaster, sizeof dip->label.name);
- dip->un.v.num_channels = 2;
- strlcpy(dip->un.v.units.name, AudioNvolume,
- sizeof dip->un.v.units.name);
- break;
-
- case PSS_MASTER_TREBLE: /* master treble */
- dip->type = AUDIO_MIXER_VALUE;
- dip->mixer_class = PSS_OUTPUT_CLASS;
- dip->next = dip->prev = AUDIO_MIXER_LAST;
- strlcpy(dip->label.name, AudioNtreble, sizeof dip->label.name);
- dip->un.v.num_channels = 1;
- strlcpy(dip->un.v.units.name, AudioNtreble,
- sizeof dip->un.v.units.name);
- break;
-
- case PSS_MASTER_BASS: /* master bass */
- dip->type = AUDIO_MIXER_VALUE;
- dip->mixer_class = PSS_OUTPUT_CLASS;
- dip->next = dip->prev = AUDIO_MIXER_LAST;
- strlcpy(dip->label.name, AudioNbass, sizeof dip->label.name);
- dip->un.v.num_channels = 1;
- strlcpy(dip->un.v.units.name, AudioNbass, sizeof dip->un.v.units.name);
- break;
-
- case PSS_OUTPUT_CLASS: /* output class descriptor */
- dip->type = AUDIO_MIXER_CLASS;
- dip->mixer_class = PSS_OUTPUT_CLASS;
- dip->next = dip->prev = AUDIO_MIXER_LAST;
- strlcpy(dip->label.name, AudioCoutputs, sizeof dip->label.name);
- break;
-
- case PSS_INPUT_CLASS: /* input class descriptor */
- dip->type = AUDIO_MIXER_CLASS;
- dip->mixer_class = PSS_INPUT_CLASS;
- dip->next = dip->prev = AUDIO_MIXER_LAST;
- strlcpy(dip->label.name, AudioCinputs, sizeof dip->label.name);
- break;
-
- case PSS_MONITOR_CLASS: /* monitor class descriptor */
- dip->type = AUDIO_MIXER_CLASS;
- dip->mixer_class = PSS_MONITOR_CLASS;
- dip->next = dip->prev = AUDIO_MIXER_LAST;
- strlcpy(dip->label.name, AudioCmonitor, sizeof dip->label.name);
- break;
-
- case PSS_RECORD_CLASS: /* record source class */
- dip->type = AUDIO_MIXER_CLASS;
- dip->mixer_class = PSS_RECORD_CLASS;
- dip->next = dip->prev = AUDIO_MIXER_LAST;
- strlcpy(dip->label.name, AudioCrecord, sizeof dip->label.name);
- break;
-
- case PSS_MIC_IN_MUTE:
- dip->mixer_class = PSS_INPUT_CLASS;
- dip->type = AUDIO_MIXER_ENUM;
- dip->prev = PSS_MIC_IN_LVL;
- dip->next = AUDIO_MIXER_LAST;
- goto mute;
-
- case PSS_LINE_IN_MUTE:
- dip->mixer_class = PSS_INPUT_CLASS;
- dip->type = AUDIO_MIXER_ENUM;
- dip->prev = PSS_LINE_IN_LVL;
- dip->next = AUDIO_MIXER_LAST;
- goto mute;
-
- case PSS_DAC_MUTE:
- dip->mixer_class = PSS_INPUT_CLASS;
- dip->type = AUDIO_MIXER_ENUM;
- dip->prev = PSS_DAC_LVL;
- dip->next = AUDIO_MIXER_LAST;
- mute:
- strlcpy(dip->label.name, AudioNmute, sizeof dip->label.name);
- dip->un.e.num_mem = 2;
- strlcpy(dip->un.e.member[0].label.name, AudioNoff,
- sizeof dip->un.e.member[0].label.name);
- dip->un.e.member[0].ord = 0;
- strlcpy(dip->un.e.member[1].label.name, AudioNon,
- sizeof dip->un.e.member[1].label.name);
- dip->un.e.member[1].ord = 1;
- break;
-
- case PSS_OUTPUT_MODE:
- dip->mixer_class = PSS_OUTPUT_CLASS;
- dip->type = AUDIO_MIXER_ENUM;
- dip->prev = PSS_MASTER_VOL;
- dip->next = AUDIO_MIXER_LAST;
- strlcpy(dip->label.name, AudioNmode, sizeof dip->label.name);
- dip->un.e.num_mem = 4;
- strlcpy(dip->un.e.member[0].label.name, AudioNmono,
- sizeof dip->un.e.member[0].label.name);
- dip->un.e.member[0].ord = PSS_SPKR_MONO;
- strlcpy(dip->un.e.member[1].label.name, AudioNstereo,
- sizeof dip->un.e.member[1].label.name);
- dip->un.e.member[1].ord = PSS_SPKR_STEREO;
- strlcpy(dip->un.e.member[2].label.name, AudioNpseudo,
- sizeof dip->un.e.member[2].label.name);
- dip->un.e.member[2].ord = PSS_SPKR_PSEUDO;
- strlcpy(dip->un.e.member[3].label.name, AudioNspatial,
- sizeof dip->un.e.member[3].label.name);
- dip->un.e.member[3].ord = PSS_SPKR_SPATIAL;
- break;
-
- case PSS_RECORD_SOURCE:
- dip->mixer_class = PSS_RECORD_CLASS;
- dip->type = AUDIO_MIXER_ENUM;
- dip->prev = PSS_REC_LVL;
- dip->next = AUDIO_MIXER_LAST;
- strlcpy(dip->label.name, AudioNsource, sizeof dip->label.name);
- dip->un.e.num_mem = 3;
- strlcpy(dip->un.e.member[0].label.name, AudioNmicrophone,
- sizeof dip->un.e.member[0].label.name);
- dip->un.e.member[0].ord = PSS_MIC_IN_LVL;
- strlcpy(dip->un.e.member[1].label.name, AudioNcd,
- sizeof dip->un.e.member[1].label.name);
- dip->un.e.member[1].ord = PSS_LINE_IN_LVL;
- strlcpy(dip->un.e.member[2].label.name, AudioNdac,
- sizeof dip->un.e.member[2].label.name);
- dip->un.e.member[2].ord = PSS_DAC_LVL;
- break;
-
- default:
- return ENXIO;
- /*NOTREACHED*/
- }
- DPRINTF(("AUDIO_MIXER_DEVINFO: name=%s\n", dip->label.name));
-
- return 0;
-}
diff --git a/sys/dev/isa/pssreg.h b/sys/dev/isa/pssreg.h
deleted file mode 100644
index 4899c184ff6..00000000000
--- a/sys/dev/isa/pssreg.h
+++ /dev/null
@@ -1,165 +0,0 @@
-/* $OpenBSD: pssreg.h,v 1.3 2007/10/26 15:00:49 martin Exp $ */
-/* $NetBSD: pssreg.h,v 1.2 1995/05/08 22:02:09 brezak Exp $ */
-
-/*
- * Copyright (c) 1994 John Brezak
- * 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.
- *
- */
-/*
- * Copyright (c) 1993 Analog Devices Inc. All rights reserved
- */
-
-/*
- * Macros to detect valid hardware configuration data.
- */
-#define PSS_BASE_VALID(base) ((base) == 0x220 || (base) == 0x240)
-
-/*
- * ESC614 Interface chip
- */
-#define ADDR_MASK 0x003f
-
-#define INT_MASK 0xffc7
-#define INT_3_BITS 0x0008
-#define INT_5_BITS 0x0010
-#define INT_7_BITS 0x0018
-#define INT_9_BITS 0x0020
-#define INT_10_BITS 0x0028
-#define INT_11_BITS 0x0030
-#define INT_12_BITS 0x0038
-
-#define INT_TEST_BIT 0x0200
-#define INT_TEST_PASS 0x0100
-#define INT_TEST_BIT_MASK 0xFDFF
-
-#define DMA_MASK 0xfff8
-#define DMA_0_BITS 0x0001
-#define DMA_1_BITS 0x0002
-#define DMA_3_BITS 0x0003
-#define DMA_5_BITS 0x0004
-#define DMA_6_BITS 0x0005
-#define DMA_7_BITS 0x0006
-
-#define DMA_TEST_BIT 0x0080
-#define DMA_TEST_PASS 0x0040
-#define DMA_TEST_BIT_MASK 0xFF7F
-
-/* Echo DSP Flags */
-#define DSP_FLAG3 0x10
-#define DSP_FLAG2 0x08
-#define DSP_FLAG1 0x80
-#define DSP_FLAG0 0x40
-
-/* ESC614 register offsets */
-#define PSS_NPORT 32
-
-#define PSS_DATA 0x00
-#define PSS_STATUS 0x02
-#define PSS_CONTROL 0x02
-#define PSS_ID_VERS 0x04
-#define PSS_IRQ_ACK 0x04
-
-#define PSS_CONFIG 0x10
-#define PSS_WSS_CONFIG 0x12
-#define SB_CONFIG 0x14
-#define CD_CONFIG 0x16
-#define MIDI_CONFIG 0x18
-#define UART_CONFIG 0x1a
-
-/* PSS control register */
-#define PSS_WEIE 0x8000
-#define PSS_RFIE 0x4000
-#define PSS_RESET 0x2000
-#define PSS_FLAG1 0x1000
-#define PSS_FLAG0 0x0800
-
-/* PSS status register */
-#define PSS_WRITE_EMPTY 0x8000
-#define PSS_READ_FULL 0x4000
-#define PSS_IRQ 0x2000
-#define PSS_DMQ_TC 0x1000
-#define PSS_FLAG3 0x0800
-#define PSS_FLAG2 0x0400
-
-/* Game control register */
-#define GAME_BIT 0x0400
-#define GAME_BIT_MASK 0xfbff
-
-/* MPU registers */
-#define MIDI_NPORT 8
-
-#define MIDI_DATA_REG 0x00
-#define MIDI_STATUS_REG 0x01
-#define MIDI_COMMAND_REG 0x01
-
-#define MIDI_SR_RF 0x80
-#define MIDI_SR_TE 0x40
-
-/* CD Interface registers */
-#define CD_NPORT 16
-
-#define CD_POL_MASK 0xFFBF
-#define CD_POL_BIT 0x0040
-
-/* Philips amplifier controls: only via DSP */
-/* DSP commands */
-#define SET_MASTER_COMMAND 0x0010
-#define MASTER_VOLUME_LEFT 0x0000
-#define MASTER_VOLUME_RIGHT 0x0100
-#define MASTER_BASS 0x0200
-#define MASTER_TREBLE 0x0300
-#define MASTER_SWITCH 0x0800
-
-#define PSS_STEREO 0x00ce
-#define PSS_PSEUDO 0x00d6
-#define PSS_SPATIAL 0x00de
-#define PSS_MONO 0x00c6
-
-#define PHILLIPS_VOL_MIN -64
-#define PHILLIPS_VOL_MAX 6
-#define PHILLIPS_VOL_DELTA 70
-#define PHILLIPS_VOL_INITIAL -20
-#define PHILLIPS_VOL_CONSTANT 252
-#define PHILLIPS_VOL_STEP 2
-#define PHILLIPS_BASS_MIN -12
-#define PHILLIPS_BASS_MAX 15
-#define PHILLIPS_BASS_DELTA 27
-#define PHILLIPS_BASS_INITIAL 0
-#define PHILLIPS_BASS_CONSTANT 246
-#define PHILLIPS_BASS_STEP 2
-#define PHILLIPS_TREBLE_MIN -12
-#define PHILLIPS_TREBLE_MAX 12
-#define PHILLIPS_TREBLE_DELTA 24
-#define PHILLIPS_TREBLE_INITIAL 0
-#define PHILLIPS_TREBLE_CONSTANT 246
-#define PHILLIPS_TREBLE_STEP 2