dmesg@ says pss has never been seen in the wild and the manpage says it

doesn't work.  goodbye.  ok miod

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