Get audio framework from MI-land, scrap the local variant. Some NetBSD merging

7 files changed, 29 insertions, 2778 deletions

diff --git a/sys/arch/amiga/amiga/conf.c b/sys/arch/amiga/amiga/conf.c
index 6d2d7408ef9..196e24b7750 100644
--- a/sys/arch/amiga/amiga/conf.c
+++ b/sys/arch/amiga/amiga/conf.c
@@ -1,4 +1,4 @@
-/*	$OpenBSD: conf.c,v 1.19 1997/10/16 06:48:12 niklas Exp $	*/
+/*	$OpenBSD: conf.c,v 1.20 1998/04/26 22:30:57 niklas Exp $	*/
 /*	$NetBSD: conf.c,v 1.42 1997/01/07 11:35:03 mrg Exp $	*/
 
 /*-
@@ -101,7 +101,7 @@ dev_decl(filedesc,open);
 #include "com.h"
 #include "lpt.h"
 #include "uk.h"
-#include "new_audio.h"
+#include "audio.h"
 cdev_decl(audio);
 
 struct cdevsw	cdevsw[] =
@@ -145,7 +145,7 @@ struct cdevsw	cdevsw[] =
 	cdev_uk_init(NUK,uk),		/* 36: unknown SCSI */
 	cdev_disk_init(NWD,wd),		/* 37: ST506/ESDI/IDE disk */
 	cdev_disk_init(NACD,acd),	/* 38: ATAPI CD-ROM */
-	cdev_audio_init(NNEW_AUDIO,audio),	/* 39: cc audio interface */
+	cdev_audio_init(NAUDIO,audio),	/* 39: cc audio interface */
 	cdev_ch_init(NCH,ch),		/* 40: SCSI autochanger */
 	cdev_disk_init(NRD,rd),		/* 41: RAM disk */
 };
diff --git a/sys/arch/amiga/conf/GENERIC b/sys/arch/amiga/conf/GENERIC
index 72c11a46a3a..95d0fbabfc2 100644
--- a/sys/arch/amiga/conf/GENERIC
+++ b/sys/arch/amiga/conf/GENERIC
@@ -1,4 +1,4 @@
-#	$OpenBSD: GENERIC,v 1.23 1997/10/07 11:04:52 niklas Exp $
+#	$OpenBSD: GENERIC,v 1.24 1998/04/26 22:30:59 niklas Exp $
 #	$NetBSD: GENERIC,v 1.85 1997/08/27 19:32:49 is Exp $
 
 #
@@ -148,7 +148,7 @@ com*		at drsupio? port ?	# DraCo serial
 a2kbbc0 	at mainbus0		# A2000 battery backed clock
 a34kbbc0	at mainbus0		# A3000/A4000 battery backed clock
 aucc*		at mainbus0		# Amiga CC audio
-new_audio*	at aucc?
+audio*		at aucc?
 
 # Ethernet cards
 le*		at zbus0		# Lance ethernet.
diff --git a/sys/arch/amiga/conf/files.amiga b/sys/arch/amiga/conf/files.amiga
index 1a403a7ca14..9ba612d74d0 100644
--- a/sys/arch/amiga/conf/files.amiga
+++ b/sys/arch/amiga/conf/files.amiga
@@ -1,4 +1,4 @@
-#	$OpenBSD: files.amiga,v 1.27 1998/04/05 09:09:09 niklas Exp $
+#	$OpenBSD: files.amiga,v 1.28 1998/04/26 22:31:01 niklas Exp $
 #	$NetBSD: files.amiga,v 1.62 1997/08/27 19:32:47 is Exp $
 
 
@@ -7,17 +7,6 @@ maxpartitions 16
 
 maxusers 2 8 64
 
-# Transient audio stuff that will end up in the MI parts
-define	new_audio {}
-define	auconv
-
-# audio device, attaches to audio hardware driver
-device	new_audio
-attach	new_audio at new_audio
-
-file dev/auconv.c			auconv
-file arch/amiga/dev/audio.c		new_audio		needs-flag
-	
 device	mainbus {}
 attach	mainbus at root
 
@@ -62,8 +51,8 @@ device	par
 attach	par at mainbus
 file	arch/amiga/dev/par.c		par needs-count
 
-# audio		XXX s/new_audio/audio/ after the new audio is generally used.
-device aucc: new_audio
+# audio
+device aucc: audio
 attach aucc at mainbus
 file   arch/amiga/dev/aucc.c		aucc needs-flag 
 
diff --git a/sys/arch/amiga/dev/aucc.c b/sys/arch/amiga/dev/aucc.c
index 391e710a192..38593edb6f1 100644
--- a/sys/arch/amiga/dev/aucc.c
+++ b/sys/arch/amiga/dev/aucc.c
@@ -1,6 +1,6 @@
-/*	$OpenBSD: aucc.c,v 1.2 1997/10/07 11:04:56 niklas Exp $	*/
-/*	$NetBSD: aucc.c,v 1.18 1997/08/24 22:31:23 augustss Exp $	*/
-#undef AUDIO_DEBUG
+/*	$OpenBSD: aucc.c,v 1.3 1998/04/26 22:31:03 niklas Exp $	*/
+/*	$NetBSD: aucc.c,v 1.22 1998/01/12 10:39:10 thorpej Exp $	*/
+
 /*
  * Copyright (c) 1997 Stephan Thesing
  * All rights reserved.
@@ -40,15 +40,16 @@
 #include <sys/ioctl.h>
 #include <sys/device.h>
 #include <sys/proc.h>
+#include <sys/audioio.h>
+
+#include <dev/audio_if.h>
 
 #include <machine/cpu.h>
-#include <machine/audioio.h>
 
 #include <amiga/amiga/cc.h>
 #include <amiga/amiga/custom.h>
 #include <amiga/amiga/device.h>
 
-#include <amiga/dev/audio_if.h>
 #include <amiga/dev/auccvar.h>
 
 #ifdef LEV6_DEFER
@@ -169,10 +170,6 @@ void	aucc_close __P((void *));
 int	aucc_set_out_sr __P((void *, u_long));
 int	aucc_query_encoding __P((void *, struct audio_encoding *));
 int	aucc_round_blocksize __P((void *, int));
-int	aucc_set_out_port __P((void *, int));
-int	aucc_get_out_port __P((void *));
-int	aucc_set_in_port __P((void *, int));
-int	aucc_get_in_port __P((void *));
 int	aucc_commit_settings __P((void *));
 int	aucc_start_output __P((void *, void *, int, void (*)(void *),
 	    void *));
@@ -180,8 +177,6 @@ int	aucc_start_input __P((void *, void *, int, void (*)(void *),
 	    void *));
 int	aucc_halt_output __P((void *));
 int	aucc_halt_input __P((void *));
-int	aucc_cont_output __P((void *));
-int	aucc_cont_input __P((void *));
 int	aucc_getdev __P((void *, struct audio_device *));
 int	aucc_set_port __P((void *, mixer_ctrl_t *));
 int	aucc_get_port __P((void *, mixer_ctrl_t *));
@@ -198,10 +193,6 @@ struct audio_hw_if sa_hw_if = {
 	aucc_query_encoding,
 	aucc_set_params,
 	aucc_round_blocksize,
-	aucc_set_out_port,
-	aucc_get_out_port,
-	aucc_set_in_port,
-	aucc_get_in_port,
 	aucc_commit_settings,
 	NULL,
 	NULL,
@@ -209,8 +200,6 @@ struct audio_hw_if sa_hw_if = {
 	aucc_start_input,
 	aucc_halt_output,
 	aucc_halt_input,
-	aucc_cont_output,
-	aucc_cont_input,
 	NULL,
 	aucc_getdev,
 	NULL,
@@ -378,7 +367,7 @@ aucc_query_encoding(addr, fp)
 {
 	switch (fp->index) {	
 	case 0:
-		strcpy(fp->name, AudioElinear);
+		strcpy(fp->name, AudioEslinear);
 		fp->encoding = AUDIO_ENCODING_SLINEAR;
 		fp->precision = 8;
 		fp->flags = 0;
@@ -460,46 +449,6 @@ aucc_round_blocksize(addr, blk)
 }
 
 int
-aucc_set_out_port(addr, port) /* can set channels  */ 
-	void *addr;
-	int port;
-{
-	struct aucc_softc *sc = addr;
-
-	/* port is mask for channels 0..3 */
-	if ((port < 0) || (port > 15))
-		return (EINVAL);
-
-	sc->sc_channelmask = port;
-
-	return (0);
-}
-
-int
-aucc_get_out_port(addr) 
-	void *addr;
-{
-	struct aucc_softc *sc = addr;
-
-	return (sc->sc_channelmask);
-}
-
-int
-aucc_set_in_port(addr, port)
-	void *addr;
-	int port;
-{
-	return (EINVAL); /* no input possible */
-}
-
-int
-aucc_get_in_port(addr)
-	void *addr;
-{
-	return (0);
-}
-
-int
 aucc_commit_settings(addr)
 	void *addr;
 {
@@ -679,23 +628,6 @@ aucc_halt_input(addr)
 }
 
 int
-aucc_cont_output(addr)
-	void *addr;
-{
-	DPRINTF(("aucc_cont_output: never called, what should it do?!\n"));
-	/* reenable DMA XXX */
-	return (ENXIO);
-}
-
-int
-aucc_cont_input(addr)
-	void *addr;
-{
-	DPRINTF(("aucc_cont_input: never called, what should it do?!\n"));
-	return (0);
-}
-
-int
 aucc_getdev(addr, retp)
         void *addr;
         struct audio_device *retp;
@@ -735,11 +667,20 @@ aucc_set_port(addr, cp)
 #endif
 
 		/* set volume for channel 0..i-1 */
-		if (i > 1)
+
+		/* evil workaround for xanim bug, IMO */
+		if ((sc->sc_channels == 1) && (i == 2)) {
+			sc->sc_channel[0].nd_volume = 
+			    sc->sc_channel[3].nd_volume = 
+			    cp->un.value.level[0] >> 2;
+			sc->sc_channel[1].nd_volume = 
+			    sc->sc_channel[2].nd_volume = 
+			    cp->un.value.level[1] >> 2;
+		} else if (i > 1) {
 			for (j = 0; j < i; j++)
 	 			sc->sc_channel[j].nd_volume =
 				    cp->un.value.level[j] >> 2;
-		else if (sc->sc_channels > 1)
+		} else if (sc->sc_channels > 1)
 			for (j = 0; j < sc->sc_channels; j++)
 	 			sc->sc_channel[j].nd_volume =
 				    cp->un.value.level[0] >> 2;
@@ -809,7 +750,7 @@ aucc_query_devinfo(addr, dip)
 		dip->type = AUDIO_MIXER_SET;
 		dip->mixer_class = AUCC_OUTPUT_CLASS;
 		dip->prev = dip->next = AUDIO_MIXER_LAST;
-                strcpy(dip->label.name, AudioNspeaker);
+                strcpy(dip->label.name, AudioNmaster);
 		for (i = 0; i < 16; i++) {
 			sprintf(dip->un.s.member[i].label.name,
 			    "channelmask%d", i);
@@ -831,11 +772,11 @@ aucc_query_devinfo(addr, dip)
 		dip->type = AUDIO_MIXER_CLASS;
 		dip->mixer_class = AUCC_OUTPUT_CLASS;
 		dip->next = dip->prev = AUDIO_MIXER_LAST;
-		strcpy(dip->label.name, AudioCOutputs);
+		strcpy(dip->label.name, AudioCoutputs);
 		break;
+
 	default:
 		return (ENXIO);
-		/*NOTREACHED*/
 	}
 
 	DPRINTF(("AUDIO_MIXER_DEVINFO: name=%s\n", dip->label.name));
diff --git a/sys/arch/amiga/dev/audio.c b/sys/arch/amiga/dev/audio.c
deleted file mode 100644
index 538615a72c0..00000000000
--- a/sys/arch/amiga/dev/audio.c
+++ /dev/null
@@ -1,2377 +0,0 @@
-/*	$OpenBSD: audio.c,v 1.1 1997/10/07 11:04:57 niklas Exp $	*/
-/*	$NetBSD: audio.c,v 1.71 1997/09/06 01:14:48 augustss Exp $	*/
-
-/*
- * Copyright (c) 1991-1993 Regents of the University of California.
- * All rights reserved.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions
- * are met:
- * 1. Redistributions of source code must retain the above copyright
- *    notice, this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright
- *    notice, this list of conditions and the following disclaimer in the
- *    documentation and/or other materials provided with the distribution.
- * 3. All advertising materials mentioning features or use of this software
- *    must display the following acknowledgement:
- *	This product includes software developed by the Computer Systems
- *	Engineering Group at Lawrence Berkeley Laboratory.
- * 4. Neither the name of the University nor of the Laboratory may be used
- *    to endorse or promote products derived from this software without
- *    specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND
- * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
- * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
- * ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
- * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
- * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
- * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
- * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
- * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
- * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
- * SUCH DAMAGE.
- */
-
-/*
- * This is a (partially) SunOS-compatible /dev/audio driver for Net- & OpenBSD.
- * 
- * This code tries to do something half-way sensible with
- * half-duplex hardware, such as with the SoundBlaster hardware.  With
- * half-duplex hardware allowing O_RDWR access doesn't really make
- * sense.  However, closing and opening the device to "turn around the
- * line" is relatively expensive and costs a card reset (which can
- * take some time, at least for the SoundBlaster hardware).  Instead
- * we allow O_RDWR access, and provide an ioctl to set the "mode",
- * i.e. playing or recording.
- *
- * If you write to a half-duplex device in record mode, the data is
- * tossed.  If you read from the device in play mode, you get silence
- * filled buffers at the rate at which samples are naturally
- * generated.
- *
- * If you try to set both play and record mode on a half-duplex
- * device, playing takes precedence.
- */
-
-/*
- * Todo:
- * - Add softaudio() isr processing for wakeup, poll, signals, 
- *   and silence fill.
- */
-
-#include "new_audio.h"
-#if NNEW_AUDIO > 0
-
-#include <sys/param.h>
-#include <sys/ioctl.h>
-#include <sys/fcntl.h>
-#include <sys/vnode.h>
-#include <sys/select.h>
-#include <sys/poll.h>
-#include <sys/malloc.h>
-#include <sys/proc.h>
-#include <sys/systm.h>
-#include <sys/syslog.h>
-#include <sys/kernel.h>
-#include <sys/signalvar.h>
-#include <sys/conf.h>
-#include <sys/device.h>
-
-#include <machine/audioio.h>
-#include <machine/endian.h>
-
-#include <amiga/dev/audio_if.h>
-#include <amiga/dev/audiovar.h>
-
-#include <vm/vm.h>
-#include <vm/vm_prot.h>
-
-#ifdef AUDIO_DEBUG
-#define DPRINTF(x)	if (audiodebug) printf x
-int	audiodebug = 0;
-#else
-#define DPRINTF(x)
-#endif
-
-#define ROUNDSIZE(x) x &= -16	/* round to nice boundary */
-
-int	audio_blk_ms = AUDIO_BLK_MS;
-
-int	audiosetinfo __P((struct audio_softc *, struct audio_info *));
-int	audiogetinfo __P((struct audio_softc *, struct audio_info *));
-
-int	audio_open __P((dev_t, int, int, struct proc *));
-int	audio_close __P((dev_t, int, int, struct proc *));
-int	audio_read __P((dev_t, struct uio *, int));
-int	audio_write __P((dev_t, struct uio *, int));
-int	audio_ioctl __P((dev_t, int, caddr_t, int, struct proc *));
-int	audio_select __P((dev_t, int, struct proc *));
-int	audio_mmap __P((dev_t, int, int));
-
-int	mixer_open __P((dev_t, int, int, struct proc *));
-int	mixer_close __P((dev_t, int, int, struct proc *));
-int	mixer_ioctl __P((dev_t, int, caddr_t, int, struct proc *));
-static	void mixer_remove __P((struct audio_softc *, struct proc *p));
-static	void mixer_signal __P((struct audio_softc *));
-    
-void	audio_init_record __P((struct audio_softc *));
-void	audio_init_play __P((struct audio_softc *));
-int	audiostartr __P((struct audio_softc *));
-int	audiostartp __P((struct audio_softc *));
-void	audio_rint __P((void *));
-void	audio_pint __P((void *));
-int	audio_check_params __P((struct audio_params *));
-
-void	audio_calc_blksize __P((struct audio_softc *, int));
-void	audio_fill_silence __P((struct audio_params *, u_char *, int));
-int	audio_silence_copyout __P((struct audio_softc *, int, struct uio *));
-
-void	audio_init_ringbuffer __P((struct audio_ringbuffer *));
-int	audio_initbufs __P((struct audio_softc *));
-void	audio_calcwater __P((struct audio_softc *));
-static __inline int audio_sleep_timo __P((int *, char *, int));
-static __inline int audio_sleep __P((int *, char *));
-static __inline void audio_wakeup __P((int *));
-int	audio_drain __P((struct audio_softc *));
-void	audio_clear __P((struct audio_softc *));
-static __inline void audio_pint_silence __P((struct audio_softc *, struct audio_ringbuffer *, u_char *, int));
-
-int	audio_alloc_ring __P((struct audio_softc *, struct audio_ringbuffer *, int));
-void	audio_free_ring __P((struct audio_softc *, struct audio_ringbuffer *));
-
-int	audioprint __P((void *, const char *));
-
-int	audioprobe __P((struct device *, void *, void *));
-void	audioattach __P((struct device *, struct device *, void *));
-
-/* The default audio mode: 8 kHz mono ulaw */
-struct audio_params audio_default = 
-	{ 8000, AUDIO_ENCODING_ULAW, 8, 1, 0, 1 };
-
-struct cfattach new_audio_ca = {
-	sizeof(struct audio_softc), audioprobe, audioattach
-};
-
-struct cfdriver new_audio_cd = {
-	NULL, "audio", DV_DULL
-};
-
-int
-audioprobe(parent, match, aux)
-	struct device *parent;
-	void *match;
-	void *aux;
-{
-	struct audio_attach_args *sa = aux;
-
-	DPRINTF(("audioprobe: done=%d sa=%p hw=%p\n", 
-		 sa->audiodone, sa, sa->ahw));
-	if (sa->audiodone)
-		return 0;
-	return sa->ahw != 0;
-}
-
-void
-audioattach(parent, self, aux)
-	struct device *parent, *self;
-	void *aux;
-{
-	struct audio_softc *sc = (void *)self;
-	struct audio_attach_args *sa = aux;
-	struct audio_hw_if *hwp = sa->ahw;
-	void *hdlp = sa->hdl;
-	int error;
-
-	printf("\n");
-
-	sa->audiodone++;
-
-#ifdef DIAGNOSTIC
-	if (hwp == 0 ||
-	    hwp->open == 0 ||
-	    hwp->close == 0 ||
-	    hwp->query_encoding == 0 ||
-	    hwp->set_params == 0 ||
-	    hwp->set_out_port == 0 ||
-	    hwp->get_out_port == 0 ||
-	    hwp->set_in_port == 0 ||
-	    hwp->get_in_port == 0 ||
-	    hwp->start_output == 0 ||
-	    hwp->start_input == 0 ||
-	    hwp->halt_output == 0 ||
-	    hwp->halt_input == 0 ||
-	    hwp->cont_output == 0 ||
-	    hwp->cont_input == 0 ||
-	    hwp->getdev == 0 ||
-	    hwp->set_port == 0 ||
-	    hwp->get_port == 0 ||
-	    hwp->query_devinfo == 0 ||
-	    hwp->get_props == 0) {
-		printf("audio: missing method\n");
-		sc->hw_if = 0;
-		return;
-        }
-#endif
-
-	sc->hw_if = hwp;
-	sc->hw_hdl = hdlp;
-	sc->sc_dev = parent;
-
-	error = audio_alloc_ring(sc, &sc->sc_pr, AU_RING_SIZE);
-	if (error) {
-		sc->hw_if = 0;
-		return;
-	}
-	error = audio_alloc_ring(sc, &sc->sc_rr, AU_RING_SIZE);
-	if (error) {
-		audio_free_ring(sc, &sc->sc_pr);
-		sc->hw_if = 0;
-		return;
-	}
-	
-	/*
-	 * Set default softc params
-	 */
-	sc->sc_pparams = audio_default;
-	sc->sc_rparams = audio_default;
-
-	/* Set up some default values */
-	sc->sc_blkset = 0;
-	audio_calc_blksize(sc, AUMODE_RECORD);
-	audio_calc_blksize(sc, AUMODE_PLAY);
-	audio_init_ringbuffer(&sc->sc_rr);
-	audio_init_ringbuffer(&sc->sc_pr);
-	audio_calcwater(sc);
-}
-
-/*
- * Called from hardware driver.  This is where the MI audio driver gets
- * probed/attached to the hardware driver.
- */
-void
-audio_attach_mi(ahwp, mhwp, hdlp, dev)
-	struct audio_hw_if *ahwp;
-	struct midi_hw_if *mhwp;
-	void *hdlp;
-	struct device *dev;
-{
-	struct audio_attach_args arg;
-
-	arg.ahw = ahwp;
-	arg.mhw = mhwp;
-	arg.hdl = hdlp;
-	arg.audiodone = arg.mididone = 0;
-	while(config_found(dev, &arg, 0))
-		;
-}
-
-#ifdef AUDIO_DEBUG
-void	audio_printsc __P((struct audio_softc *));
-void	audio_print_params __P((char *, struct audio_params *));
-
-void
-audio_printsc(sc)
-	struct audio_softc *sc;
-{
-	printf("hwhandle %p hw_if %p ", sc->hw_hdl, sc->hw_if);
-	printf("open 0x%x mode 0x%x\n", sc->sc_open, sc->sc_mode);
-	printf("rchan 0x%x wchan 0x%x ", sc->sc_rchan, sc->sc_wchan);
-	printf("rring used 0x%x pring used=%d\n", sc->sc_rr.used, sc->sc_pr.used);
-	printf("rbus 0x%x pbus 0x%x ", sc->sc_rbus, sc->sc_pbus);
-	printf("blksize %d", sc->sc_pr.blksize);
-	printf("hiwat %d lowat %d\n", sc->sc_pr.usedhigh, sc->sc_pr.usedlow);
-}
-
-void
-audio_print_params(s, p)
-	char *s;
-	struct audio_params *p;
-{
-	printf("audio: %s sr=%ld, enc=%d, chan=%d, prec=%d\n", s,
-	       p->sample_rate, p->encoding, p->channels, p->precision);
-}
-#endif
-
-int
-audio_alloc_ring(sc, r, bufsize)
-	struct audio_softc *sc;
-	struct audio_ringbuffer *r;
-	int bufsize;
-{
-	struct audio_hw_if *hw = sc->hw_if;
-	void *hdl = sc->hw_hdl;
-	/*
-	 * Alloc DMA play and record buffers
-	 */
-	ROUNDSIZE(bufsize);
-	if (bufsize < AUMINBUF)
-		bufsize = AUMINBUF;
-	if (hw->round_buffersize)
-		bufsize = hw->round_buffersize(hdl, bufsize);
-	r->bufsize = bufsize;
-	if (hw->alloc)
-	    r->start = hw->alloc(hdl, r->bufsize, M_DEVBUF, M_WAITOK);
-	else
-	    r->start = malloc(bufsize, M_DEVBUF, M_WAITOK);
-	if (r->start == 0)
-		return ENOMEM;
-	return 0;
-}
-
-void
-audio_free_ring(sc, r)
-	struct audio_softc *sc;
-	struct audio_ringbuffer *r;
-{
-	if (sc->hw_if->free) {
-	    sc->hw_if->free(sc->hw_hdl, r->start, M_DEVBUF);
-	} else {
-	    free(r->start, M_DEVBUF);
-	}
-}
-
-int
-audioopen(dev, flags, ifmt, p)
-	dev_t dev;
-	int flags, ifmt;
-	struct proc *p;
-{
-
-	switch (AUDIODEV(dev)) {
-	case SOUND_DEVICE:
-	case AUDIO_DEVICE:
-	case AUDIOCTL_DEVICE:
-		return (audio_open(dev, flags, ifmt, p));
-	case MIXER_DEVICE:
-		return (mixer_open(dev, flags, ifmt, p));
-	default:
-		return (ENXIO);
-	}
-}
-
-int
-audioclose(dev, flags, ifmt, p)
-	dev_t dev;
-	int flags, ifmt;
-	struct proc *p;
-{
-
-	switch (AUDIODEV(dev)) {
-	case SOUND_DEVICE:
-	case AUDIO_DEVICE:
-		return (audio_close(dev, flags, ifmt, p));
-	case MIXER_DEVICE:
-		return (mixer_close(dev, flags, ifmt, p));
-	case AUDIOCTL_DEVICE:
-		return 0;
-	default:
-		return (ENXIO);
-	}
-}
-
-int
-audioread(dev, uio, ioflag)
-	dev_t dev;
-	struct uio *uio;
-	int ioflag;
-{
-
-	switch (AUDIODEV(dev)) {
-	case SOUND_DEVICE:
-	case AUDIO_DEVICE:
-		return (audio_read(dev, uio, ioflag));
-	case AUDIOCTL_DEVICE:
-	case MIXER_DEVICE:
-		return (ENODEV);
-	default:
-		return (ENXIO);
-	}
-}
-
-int
-audiowrite(dev, uio, ioflag)
-	dev_t dev;
-	struct uio *uio;
-	int ioflag;
-{
-
-	switch (AUDIODEV(dev)) {
-	case SOUND_DEVICE:
-	case AUDIO_DEVICE:
-		return (audio_write(dev, uio, ioflag));
-	case AUDIOCTL_DEVICE:
-	case MIXER_DEVICE:
-		return (ENODEV);
-	default:
-		return (ENXIO);
-	}
-}
-
-int
-audioioctl(dev, cmd, addr, flag, p)
-	dev_t dev;
-	u_long cmd;
-	caddr_t addr;
-	int flag;
-	struct proc *p;
-{
-
-	switch (AUDIODEV(dev)) {
-	case SOUND_DEVICE:
-	case AUDIO_DEVICE:
-	case AUDIOCTL_DEVICE:
-		return (audio_ioctl(dev, cmd, addr, flag, p));
-	case MIXER_DEVICE:
-		return (mixer_ioctl(dev, cmd, addr, flag, p));
-	default:
-		return (ENXIO);
-	}
-}
-
-int
-audioselect(dev, events, p)
-	dev_t dev;
-	int events;
-	struct proc *p;
-{
-
-	switch (AUDIODEV(dev)) {
-	case SOUND_DEVICE:
-	case AUDIO_DEVICE:
-		return (audio_select(dev, events, p));
-	case AUDIOCTL_DEVICE:
-	case MIXER_DEVICE:
-		return (0);
-	default:
-		return (0);
-	}
-}
-
-int
-audiommap(dev, off, prot)
-	dev_t dev;
-	int off, prot;
-{
-
-	switch (AUDIODEV(dev)) {
-	case SOUND_DEVICE:
-	case AUDIO_DEVICE:
-		return (audio_mmap(dev, off, prot));
-	case AUDIOCTL_DEVICE:
-	case MIXER_DEVICE:
-		return -1;
-	default:
-		return -1;
-	}
-}
-
-/*
- * Audio driver
- */
-void
-audio_init_ringbuffer(rp)
-	struct audio_ringbuffer *rp;
-{
-	int nblks;
-	int blksize = rp->blksize;
-
-	if (blksize < AUMINBLK)
-		blksize = AUMINBLK;
-	nblks = rp->bufsize / blksize;
-	if (nblks < AUMINNOBLK) {
-		nblks = AUMINNOBLK;
-		blksize = rp->bufsize / nblks;
-		ROUNDSIZE(blksize);
-	}
-	DPRINTF(("audio_init_ringbuffer: blksize=%d\n", blksize));
-	rp->blksize = blksize;
-	rp->maxblks = nblks;
-	rp->used = 0;
-	rp->end = rp->start + nblks * blksize;
-	rp->inp = rp->outp = rp->start;
-	rp->stamp = 0;
-	rp->drops = 0;
-	rp->pause = 0;
-	rp->copying = 0;
-	rp->needfill = 0;
-	rp->mmapped = 0;
-}
-
-int
-audio_initbufs(sc)
-	struct audio_softc *sc;
-{
-	struct audio_hw_if *hw = sc->hw_if;
-	int error;
-
-	DPRINTF(("audio_initbufs: mode=0x%x\n", sc->sc_mode));
-	audio_init_ringbuffer(&sc->sc_rr);
-	if (hw->init_input && (sc->sc_mode & AUMODE_RECORD)) {
-		error = hw->init_input(sc->hw_hdl, sc->sc_rr.start,
-				       sc->sc_rr.end - sc->sc_rr.start);
-		if (error)
-			return error;
-	}
-
-	audio_init_ringbuffer(&sc->sc_pr);
-	sc->sc_sil_count = 0;
-	if (hw->init_output && (sc->sc_mode & AUMODE_PLAY)) {
-		error = hw->init_output(sc->hw_hdl, sc->sc_pr.start,
-					sc->sc_pr.end - sc->sc_pr.start);
-		if (error)
-			return error;
-	}
-
-#ifdef AUDIO_INTR_TIME
-	sc->sc_pnintr = 0;
-	sc->sc_pblktime = (u_long)(
-	    (double)sc->sc_pr.blksize * 1e6 / 
-	    (double)(sc->sc_pparams.precision / NBBY * 
-                     sc->sc_pparams.channels * 
-		     sc->sc_pparams.sample_rate));
-	DPRINTF(("audio: play blktime = %lu for %d\n", 
-		 sc->sc_pblktime, sc->sc_pr.blksize));
-	sc->sc_rnintr = 0;
-	sc->sc_rblktime = (u_long)(
-	    (double)sc->sc_rr.blksize * 1e6 / 
-	    (double)(sc->sc_rparams.precision / NBBY * 
-                     sc->sc_rparams.channels * 
-		     sc->sc_rparams.sample_rate));
-	DPRINTF(("audio: record blktime = %lu for %d\n", 
-		 sc->sc_rblktime, sc->sc_rr.blksize));
-#endif
-
-	return 0;
-}
-
-void
-audio_calcwater(sc)
-	struct audio_softc *sc;
-{
-	sc->sc_pr.usedhigh = sc->sc_pr.end - sc->sc_pr.start;
-	sc->sc_pr.usedlow = sc->sc_pr.usedhigh * 3 / 4;	/* set lowater at 75% */
-	if (sc->sc_pr.usedlow == sc->sc_pr.usedhigh)
-		sc->sc_pr.usedlow -= sc->sc_pr.blksize;
-	sc->sc_rr.usedhigh = sc->sc_pr.end - sc->sc_pr.start - sc->sc_pr.blksize;
-	sc->sc_rr.usedlow = 0;
-}
-
-static __inline int
-audio_sleep_timo(chan, label, timo)
-	int *chan;
-	char *label;
-	int timo;
-{
-	int st;
-
-	if (!label)
-		label = "audio";
-
-	*chan = 1;
-	st = tsleep(chan, PWAIT | PCATCH, label, timo);
-	*chan = 0;
-#ifdef AUDIO_DEBUG
-	if (st != 0)
-	    printf("audio_sleep: %d\n", st);
-#endif
-	return (st);
-}
-
-static __inline int
-audio_sleep(chan, label)
-	int *chan;
-	char *label;
-{
-	return audio_sleep_timo(chan, label, 0);
-}
-
-/* call at splaudio() */
-static __inline void
-audio_wakeup(chan)
-	int *chan;
-{
-	if (*chan) {
-		wakeup(chan);
-		*chan = 0;
-	}
-}
-
-int
-audio_open(dev, flags, ifmt, p)
-	dev_t dev;
-	int flags, ifmt;
-	struct proc *p;
-{
-	int unit = AUDIOUNIT(dev);
-	struct audio_softc *sc;
-	int error;
-	int mode;
-	struct audio_hw_if *hw;
-	struct audio_info ai;
-
-	if (unit >= new_audio_cd.cd_ndevs ||
-	    (sc = new_audio_cd.cd_devs[unit]) == NULL)
-		return ENXIO;
-
-	hw = sc->hw_if;
-	if (!hw)
-		return ENXIO;
-
-	DPRINTF(("audio_open: dev=0x%x flags=0x%x sc=%p hdl=%p\n", dev, flags, sc, sc->hw_hdl));
-
-	if (ISDEVAUDIOCTL(dev))
-		return 0;
-
-	if ((sc->sc_open & (AUOPEN_READ|AUOPEN_WRITE)) != 0)
-		return (EBUSY);
-
-	error = hw->open(sc->hw_hdl, flags);
-	if (error)
-		return (error);
-
-	sc->sc_async_audio = 0;
-	sc->sc_rchan = 0;
-	sc->sc_wchan = 0;
-	sc->sc_blkset = 0; /* Block sizes not set yet */
-	sc->sc_sil_count = 0;
-	sc->sc_rbus = 0;
-	sc->sc_pbus = 0;
-	sc->sc_eof = 0;
-	sc->sc_playdrop = 0;
-
-	sc->sc_full_duplex = 0;
-/* doesn't always work right on SB.
-		(flags & (FWRITE|FREAD)) == (FWRITE|FREAD) &&
-		(hw->get_props(sc->hw_hdl) & AUDIO_PROP_FULLDUPLEX);
-*/
-
-	mode = 0;
-	if (flags & FREAD) {
-		sc->sc_open |= AUOPEN_READ;
-		mode |= AUMODE_RECORD;
-	}
-	if (flags & FWRITE) {
-		sc->sc_open |= AUOPEN_WRITE;
-		mode |= AUMODE_PLAY | AUMODE_PLAY_ALL;
-	}
-
-	/*
-	 * Multiplex device: /dev/audio (MU-Law) and /dev/sound (linear)
-	 * The /dev/audio is always (re)set to 8-bit MU-Law mono
-	 * For the other devices, you get what they were last set to.
-	 */
-	if (ISDEVAUDIO(dev)) {
-		/* /dev/audio */
-		sc->sc_rparams = audio_default;
-		sc->sc_pparams = audio_default;
-	}
-#ifdef DIAGNOSTIC
-	/*
-	 * Sample rate and precision are supposed to be set to proper
-	 * default values by the hardware driver, so that it may give
-	 * us these values.
-	 */
-	if (sc->sc_rparams.precision == 0 || sc->sc_pparams.precision == 0) {
-		printf("audio_open: 0 precision\n");
-		return EINVAL;
-	}
-#endif
-
-	AUDIO_INITINFO(&ai);
-	ai.record.sample_rate = sc->sc_rparams.sample_rate;
-	ai.record.encoding    = sc->sc_rparams.encoding;
-	ai.record.channels    = sc->sc_rparams.channels;
-	ai.record.precision   = sc->sc_rparams.precision;
-	ai.play.sample_rate   = sc->sc_pparams.sample_rate;
-	ai.play.encoding      = sc->sc_pparams.encoding;
-	ai.play.channels      = sc->sc_pparams.channels;
-	ai.play.precision     = sc->sc_pparams.precision;
-	ai.mode		      = mode;
-	sc->sc_pr.blksize = sc->sc_rr.blksize = 0; /* force recalculation */
-	error = audiosetinfo(sc, &ai);
-	if (error)
-		goto bad;
-
-	DPRINTF(("audio_open: done sc_mode = 0x%x\n", sc->sc_mode));
-	
-	return 0;
-
-bad:
-	hw->close(sc->hw_hdl);
-	sc->sc_open = 0;
-	sc->sc_mode = 0;
-	sc->sc_full_duplex = 0;
-	return error;
-}
-
-/*
- * Must be called from task context.
- */
-void
-audio_init_record(sc)
-	struct audio_softc *sc;
-{
-	int s = splaudio();
-
-	if (sc->hw_if->speaker_ctl &&
-	    (!sc->sc_full_duplex || (sc->sc_mode & AUMODE_PLAY) == 0))
-		sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_OFF);
-	splx(s);
-}
-
-/*
- * Must be called from task context.
- */
-void
-audio_init_play(sc)
-	struct audio_softc *sc;
-{
-	int s = splaudio();
-
-	sc->sc_wstamp = sc->sc_pr.stamp;
-	if (sc->hw_if->speaker_ctl)
-		sc->hw_if->speaker_ctl(sc->hw_hdl, SPKR_ON);
-	splx(s);
-}
-
-int
-audio_drain(sc)
-	struct audio_softc *sc;
-{
-	int error, drops;
-	struct audio_ringbuffer *cb = &sc->sc_pr;
-	int s;
-
-	if (sc->sc_pr.mmapped || sc->sc_pr.used <= 0)
-		return 0;
-	if (!sc->sc_pbus) {
-		/* We've never started playing, probably because the
-		 * block was too short.  Pad it and start now.
-		 */
-		int cc;
-		u_char *inp = cb->inp;
-
-		cc = cb->blksize - (inp - cb->start) % cb->blksize;
-		audio_fill_silence(&sc->sc_pparams, inp, cc);
-		inp += cc;
-		if (inp >= cb->end)
-			inp = cb->start;
-		s = splaudio();
-		cb->used += cc;
-		cb->inp = inp;
-		error = audiostartp(sc);
-		splx(s);
-		if (error)
-			return error;
-	}
-	/* 
-	 * Play until a silence block has been played, then we
-	 * know all has been drained.
-	 * XXX This should be done some other way to avoid
-	 * playing silence.
-	 */
-#ifdef DIAGNOSTIC
-	if (cb->copying) {
-		printf("audio_drain: copying in progress!?!\n");
-		cb->copying = 0;
-	}
-#endif
-	drops = cb->drops;
-	error = 0;
-	s = splaudio();
-	while (cb->drops == drops && !error) {
-		DPRINTF(("audio_drain: used=%d, drops=%ld\n", sc->sc_pr.used, cb->drops));
-		/*
-		 * When the process is exiting, it ignores all signals and
-		 * we can't interrupt this sleep, so we set a timeout just in case.
-		 */
-		error = audio_sleep_timo(&sc->sc_wchan, "aud_dr", 30*hz);
-	}
-	splx(s);
-	return error;
-}
-
-/*
- * Close an audio chip.
- */
-/* ARGSUSED */
-int
-audio_close(dev, flags, ifmt, p)
-	dev_t dev;
-	int flags, ifmt;
-	struct proc *p;
-{
-	int unit = AUDIOUNIT(dev);
-	struct audio_softc *sc = new_audio_cd.cd_devs[unit];
-	struct audio_hw_if *hw = sc->hw_if;
-	int s;
-
-	DPRINTF(("audio_close: unit=%d\n", unit));
-
-	/*
-	 * Block until output drains, but allow ^C interrupt.
-	 */
-	sc->sc_pr.usedlow = sc->sc_pr.blksize;	/* avoid excessive wakeups */
-	s = splaudio();
-	/*
-	 * If there is pending output, let it drain (unless
-	 * the output is paused).
-	 */
-	if (!sc->sc_pr.pause) {
-		if (!audio_drain(sc) && hw->drain)
-			(void)hw->drain(sc->hw_hdl);
-	}
-	
-	hw->close(sc->hw_hdl);
-	
-	if (flags & FREAD)
-		sc->sc_open &= ~AUOPEN_READ;
-	if (flags & FWRITE)
-		sc->sc_open &= ~AUOPEN_WRITE;
-
-	sc->sc_async_audio = 0;
-	sc->sc_mode = 0;
-	sc->sc_full_duplex = 0;
-	splx(s);
-	DPRINTF(("audio_close: done\n"));
-
-	return (0);
-}
-
-int
-audio_read(dev, uio, ioflag)
-	dev_t dev;
-	struct uio *uio;
-	int ioflag;
-{
-	int unit = AUDIOUNIT(dev);
-	struct audio_softc *sc = new_audio_cd.cd_devs[unit];
-	struct audio_ringbuffer *cb = &sc->sc_rr;
-	u_char *outp;
-	int error, s, used, cc, n;
-
-	if (cb->mmapped)
-		return EINVAL;
-
-#ifdef AUDIO_DEBUG
-	if (audiodebug > 1)
-		printf("audio_read: cc=%d mode=%d\n", uio->uio_resid, sc->sc_mode);
-#endif
-
-	error = 0;
-	/*
-	 * If hardware is half-duplex and currently playing, return
-	 * silence blocks based on the number of blocks we have output.
-	 */
-	if (!sc->sc_full_duplex &&
-	    (sc->sc_mode & AUMODE_PLAY)) {
-		while (uio->uio_resid > 0 && !error) {
-			s = splaudio();
-			for(;;) {
-				cc = sc->sc_pr.stamp - sc->sc_wstamp;
-				if (cc > 0)
-					break;
-				DPRINTF(("audio_read: stamp=%lu, wstamp=%lu\n", 
-					 sc->sc_pr.stamp, sc->sc_wstamp));
-				if (ioflag & IO_NDELAY) {
-					splx(s);
-					return EWOULDBLOCK;
-				}
-				error = audio_sleep(&sc->sc_rchan, "aud_hr");
-				if (error) {
-					splx(s);
-					return error;
-				}
-			}
-			splx(s);
-
-			if (uio->uio_resid < cc)
-				cc = uio->uio_resid;
-#ifdef AUDIO_DEBUG
-			if (audiodebug > 1)
-				printf("audio_read: reading in write mode, cc=%d\n", cc);
-#endif	
-			error = audio_silence_copyout(sc, cc, uio);
-			sc->sc_wstamp += cc;
-		} 
-		return (error);
-	}
-	while (uio->uio_resid > 0 && !error) {
-		s = splaudio();
-		while (cb->used <= 0) {
-			if (ioflag & IO_NDELAY) {
-				splx(s);
-				return EWOULDBLOCK;
-			}
-			if (!sc->sc_rbus) {
-				error = audiostartr(sc);
-				if (error) {
-					splx(s);
-					return error;
-				}
-			}
-#ifdef AUDIO_DEBUG
-			if (audiodebug > 2)
-				printf("audio_read: sleep used=%d\n", cb->used);
-#endif
-			error = audio_sleep(&sc->sc_rchan, "aud_rd");
-			if (error) {
-				splx(s);
-				return error;
-			}
-		}
-		used = cb->used;
-		outp = cb->outp;
-		cb->copying = 1;
-		splx(s);
-		cc = used - cb->usedlow; /* maximum to read */
-		n = cb->end - outp;
-		if (n < cc)
-			cc = n;	/* don't read beyond end of buffer */
-		
-		if (uio->uio_resid < cc)
-			cc = uio->uio_resid; /* and no more than we want */
-
-		if (sc->sc_rparams.sw_code)
-			sc->sc_rparams.sw_code(sc->hw_hdl, outp, cc);
-#ifdef AUDIO_DEBUG
-		if (audiodebug > 1)
-			printf("audio_read: outp=%p, cc=%d\n", outp, cc);
-#endif	
-		error = uiomove(outp, cc, uio);
-		used -= cc;
-		outp += cc;
-		if (outp >= cb->end)
-			outp = cb->start;
-		s = splaudio();
-		cb->outp = outp;
-		cb->used = used;
-		cb->copying = 0;
-		splx(s);
-	}
-	return (error);
-}
-
-void
-audio_clear(sc)
-	struct audio_softc *sc;
-{
-	int s = splaudio();
-
-	if (sc->sc_rbus) {
-		audio_wakeup(&sc->sc_rchan);
-		sc->hw_if->halt_input(sc->hw_hdl);
-		sc->sc_rbus = 0;
-	}
-	if (sc->sc_pbus) {
-		audio_wakeup(&sc->sc_wchan);
-		sc->hw_if->halt_output(sc->hw_hdl);
-		sc->sc_pbus = 0;
-	}
-	splx(s);
-}
-
-void
-audio_calc_blksize(sc, mode)
-	struct audio_softc *sc;
-	int mode;
-{
-	struct audio_hw_if *hw = sc->hw_if;
-	struct audio_params *parm;
-	struct audio_ringbuffer *rb;
-    	int bs;
-
-	if (sc->sc_blkset)
-		return;
-
-	if (mode == AUMODE_PLAY) {
-		parm = &sc->sc_pparams;
-		rb = &sc->sc_pr;
-	} else {
-		parm = &sc->sc_rparams;
-		rb = &sc->sc_rr;
-	}
-	
-	bs = parm->sample_rate * audio_blk_ms / 1000 *
-	     parm->channels * parm->precision / NBBY *
-	     parm->factor;
-	ROUNDSIZE(bs);
-	if (hw->round_blocksize)
-		bs = hw->round_blocksize(sc->hw_hdl, bs);
-	rb->blksize = bs;
-
-	DPRINTF(("audio_calc_blksize: %s blksize=%d\n", 
-		 mode == AUMODE_PLAY ? "play" : "record", bs));
-}
-
-void
-audio_fill_silence(params, p, n)
-	struct audio_params *params;
-        u_char *p;
-        int n;
-{
-	u_char auzero0, auzero1 = 0; /* initialize to please gcc */
-	int nfill = 1;
-
-	switch (params->encoding) {
-	case AUDIO_ENCODING_ULAW:
-	    	auzero0 = 0x7f; 
-		break;
-	case AUDIO_ENCODING_ALAW:
-		auzero0 = 0x55;
-		break;
-	case AUDIO_ENCODING_ADPCM: /* is this right XXX */
-	case AUDIO_ENCODING_SLINEAR_LE:
-	case AUDIO_ENCODING_SLINEAR_BE:
-		auzero0 = 0;	/* fortunately this works for both 8 and 16 bits */
-		break;
-	case AUDIO_ENCODING_ULINEAR_LE:
-	case AUDIO_ENCODING_ULINEAR_BE:
-		if (params->precision == 16) {
-			nfill = 2;
-			if (params->encoding == AUDIO_ENCODING_ULINEAR_LE) {
-				auzero0 = 0;
-				auzero1 = 0x80;
-			} else {
-				auzero0 = 0x80;
-				auzero1 = 0;
-			}
-		} else
-			auzero0 = 0x80;
-		break;
-	default:
-		printf("audio: bad encoding %d\n", params->encoding);
-		auzero0 = 0;
-		break;
-	}
-	if (nfill == 1) {
-		while (--n >= 0)
-			*p++ = auzero0; /* XXX memset */
-	} else /* nfill must be 2 */ {
-		while (n > 1) {
-			*p++ = auzero0;
-			*p++ = auzero1;
-			n -= 2;
-		}
-	}
-}
-
-int
-audio_silence_copyout(sc, n, uio)
-	struct audio_softc *sc;
-	int n;
-	struct uio *uio;
-{
-	int error;
-	int k;
-	u_char zerobuf[128];
-
-	audio_fill_silence(&sc->sc_rparams, zerobuf, sizeof zerobuf);
-
-	error = 0;
-        while (n > 0 && uio->uio_resid > 0 && !error) {
-		k = min(n, min(uio->uio_resid, sizeof zerobuf));
-		error = uiomove(zerobuf, k, uio);
-		n -= k;
-	}
-        return (error);
-}
-
-int
-audio_write(dev, uio, ioflag)
-	dev_t dev;
-	struct uio *uio;
-	int ioflag;
-{
-	int unit = AUDIOUNIT(dev);
-	struct audio_softc *sc = new_audio_cd.cd_devs[unit];
-	struct audio_ringbuffer *cb = &sc->sc_pr;
-	u_char *inp, *einp;
-	int error, s, n, cc, used;
-
-	DPRINTF(("audio_write: sc=%p(unit=%d) count=%d used=%d(hi=%d)\n", sc, unit,
-		 uio->uio_resid, sc->sc_pr.used, sc->sc_pr.usedhigh));
-
-	if (cb->mmapped)
-		return EINVAL;
-
-	if (uio->uio_resid == 0) {
-		sc->sc_eof++;
-		return 0;
-	}
-
-	/*
-	 * If half-duplex and currently recording, throw away data.
-	 */
-	if (!sc->sc_full_duplex &&
-	    (sc->sc_mode & AUMODE_RECORD)) {
-		uio->uio_offset += uio->uio_resid;
-		uio->uio_resid = 0;
-		DPRINTF(("audio_write: half-dpx read busy\n"));
-		return (0);
-	}
-
-	if (!(sc->sc_mode & AUMODE_PLAY_ALL) && sc->sc_playdrop > 0) {
-		n = min(sc->sc_playdrop, uio->uio_resid);
-		DPRINTF(("audio_write: playdrop %d\n", n));
-		uio->uio_offset += n;
-		uio->uio_resid -= n;
-		sc->sc_playdrop -= n;
-		if (uio->uio_resid == 0)
-			return 0;
-	}
-
-#ifdef AUDIO_DEBUG
-	if (audiodebug > 1)
-		printf("audio_write: sr=%ld, enc=%d, prec=%d, chan=%d, sw=%p, fact=%d\n",
-		       sc->sc_pparams.sample_rate, sc->sc_pparams.encoding,
-		       sc->sc_pparams.precision, sc->sc_pparams.channels,
-		       sc->sc_pparams.sw_code, sc->sc_pparams.factor);
-#endif
-
-	error = 0;
-	while (uio->uio_resid > 0 && !error) {
-		s = splaudio();
-		while (cb->used >= cb->usedhigh) {
-			DPRINTF(("audio_write: sleep used=%d lowat=%d hiwat=%d\n", 
-				 cb->used, cb->usedlow, cb->usedhigh));
-			if (ioflag & IO_NDELAY) {
-				splx(s);
-				return (EWOULDBLOCK);
-			}
-			error = audio_sleep(&sc->sc_wchan, "aud_wr");
-			if (error) {
-				splx(s);
-				return error;
-			}
-		}
-		used = cb->used;
-		inp = cb->inp;
-		cb->copying = 1;
-		splx(s);
-		cc = cb->usedhigh - used; 	/* maximum to write */
-		n = cb->end - inp;
-		if (sc->sc_pparams.factor != 1) {
-			/* Compensate for software coding expansion factor. */
-			n /= sc->sc_pparams.factor;
-			cc /= sc->sc_pparams.factor;
-		}
-		if (n < cc)
-			cc = n;			/* don't write beyond end of buffer */
-		if (uio->uio_resid < cc)
-			cc = uio->uio_resid; 	/* and no more than we have */
-
-#ifdef DIAGNOSTIC
-		/* 
-		 * This should never happen since the block size and and
-		 * block pointers are always nicely aligned. 
-		 */
-		if (cc == 0) {
-			printf("audio_write: cc == 0, swcode=%p, factor=%d\n",
-			       sc->sc_pparams.sw_code, sc->sc_pparams.factor);
-			cb->copying = 0;
-			return EINVAL;
-		}
-#endif
-#ifdef AUDIO_DEBUG
-		if (audiodebug > 1)
-		    printf("audio_write: uiomove cc=%d inp=%p, left=%d\n", cc, inp, uio->uio_resid);
-#endif
-		n = uio->uio_resid;
-		error = uiomove(inp, cc, uio);
-		cc = n - uio->uio_resid; /* number of bytes actually moved */
-#ifdef AUDIO_DEBUG
-		if (error)
-		        printf("audio_write:(1) uiomove failed %d; cc=%d inp=%p\n",
-			       error, cc, inp);
-#endif
-		/* 
-		 * Continue even if uiomove() failed because we may have
-		 * gotten a partial block.
-		 */
-
-		if (sc->sc_pparams.sw_code) {
-			sc->sc_pparams.sw_code(sc->hw_hdl, inp, cc);
-			/* Adjust count after the expansion. */
-			cc *= sc->sc_pparams.factor;
-#ifdef AUDIO_DEBUG
-			if (audiodebug > 1)
-				printf("audio_write: expanded cc=%d\n", cc);
-#endif
-		}
-
-		einp = cb->inp + cc;
-		if (einp >= cb->end)
-			einp = cb->start;
-
-		s = splaudio();
-		/*
-		 * This is a very suboptimal way of keeping track of
-		 * silence in the buffer, but it is simple.
-		 */
-		sc->sc_sil_count = 0;
-
-		cb->inp = einp;
-		cb->used += cc;
-		/* If the interrupt routine wants the last block filled AND
-		 * the copy did not fill the last block completely it needs to
-		 * be padded.
-		 */
-		if (cb->needfill &&
-		    (inp  - cb->start) / cb->blksize == 
-		    (einp - cb->start) / cb->blksize) {
-			/* Figure out how many bytes there is to a block boundary. */
-			cc = cb->blksize - (einp - cb->start) % cb->blksize;
-			DPRINTF(("audio_write: partial fill %d\n", cc));
-		} else
-			cc = 0;
-		cb->needfill = 0;
-		cb->copying = 0;
-		if (!sc->sc_pbus && !cb->pause)
-			error = audiostartp(sc); /* XXX should not clobber error */
-		splx(s);
-		if (cc) {
-#ifdef AUDIO_DEBUG
-			if (audiodebug > 1)
-				printf("audio_write: fill %d\n", cc);
-#endif
-			audio_fill_silence(&sc->sc_pparams, einp, cc);
-		}
-	}
-	return (error);
-}
-
-int
-audio_ioctl(dev, cmd, addr, flag, p)
-	dev_t dev;
-	int cmd;
-	caddr_t addr;
-	int flag;
-	struct proc *p;
-{
-	int unit = AUDIOUNIT(dev);
-	struct audio_softc *sc = new_audio_cd.cd_devs[unit];
-	struct audio_hw_if *hw = sc->hw_if;
-	struct audio_offset *ao;
-	int error = 0, s, offs, fd;
-
-	DPRINTF(("audio_ioctl(%d,'%c',%d)\n",
-	          IOCPARM_LEN(cmd), IOCGROUP(cmd), cmd&0xff));
-	switch (cmd) {
-	case FIONBIO:
-		/* All handled in the upper FS layer. */
-		break;
-
-	case FIOASYNC:
-		if (*(int *)addr) {
-			if (sc->sc_async_audio)
-				return (EBUSY);
-			sc->sc_async_audio = p;
-			DPRINTF(("audio_ioctl: FIOASYNC %p\n", p));
-		} else
-			sc->sc_async_audio = 0;
-		break;
-
-	case AUDIO_FLUSH:
-		DPRINTF(("AUDIO_FLUSH\n"));
-		audio_clear(sc);
-		s = splaudio();
-		error = audio_initbufs(sc);
-		if (error) {
-			splx(s);
-			return error;
-		}
-		if ((sc->sc_mode & AUMODE_PLAY) && !sc->sc_pbus)
-			error = audiostartp(sc);
-		if (!error &&
-		    (sc->sc_mode & AUMODE_RECORD) && !sc->sc_rbus)
-			error = audiostartr(sc);
-		splx(s);
-		break;
-
-	/*
-	 * Number of read (write) samples dropped.  We don't know where or
-	 * when they were dropped.
-	 */
-	case AUDIO_RERROR:
-		*(int *)addr = sc->sc_rr.drops;
-		break;
-
-	case AUDIO_PERROR:
-		*(int *)addr = sc->sc_pr.drops;
-		break;
-
-	/*
-	 * Offsets into buffer.
-	 */
-	case AUDIO_GETIOFFS:
-		s = splaudio();
-		/* figure out where next DMA will start */
-		ao = (struct audio_offset *)addr;
-		ao->samples = sc->sc_rr.stamp;
-		ao->deltablks = (sc->sc_rr.stamp - sc->sc_rr.stamp_last) / sc->sc_rr.blksize;
-		sc->sc_rr.stamp_last = sc->sc_rr.stamp;
-		ao->offset = sc->sc_rr.inp - sc->sc_rr.start;
-		splx(s);
-		break;
-
-	case AUDIO_GETOOFFS:
-		s = splaudio();
-		/* figure out where next DMA will start */
-		ao = (struct audio_offset *)addr;
-		offs = sc->sc_pr.outp - sc->sc_pr.start + sc->sc_pr.blksize;
-		if (sc->sc_pr.start + offs >= sc->sc_pr.end)
-			offs = 0;
-		ao->samples = sc->sc_pr.stamp;
-		ao->deltablks = (sc->sc_pr.stamp - sc->sc_pr.stamp_last) / sc->sc_pr.blksize;
-		sc->sc_pr.stamp_last = sc->sc_pr.stamp;
-		ao->offset = offs;
-		splx(s);
-		break;
-
-	/*
-	 * How many bytes will elapse until mike hears the first
-	 * sample of what we write next?
-	 */
-	case AUDIO_WSEEK:
-		*(u_long *)addr = sc->sc_rr.used;
-		break;
-
-	case AUDIO_SETINFO:
-		DPRINTF(("AUDIO_SETINFO mode=0x%x\n", sc->sc_mode));
-		error = audiosetinfo(sc, (struct audio_info *)addr);
-		break;
-
-	case AUDIO_GETINFO:
-		DPRINTF(("AUDIO_GETINFO\n"));
-		error = audiogetinfo(sc, (struct audio_info *)addr);
-		break;
-
-	case AUDIO_DRAIN:
-		DPRINTF(("AUDIO_DRAIN\n"));
-		error = audio_drain(sc);
-		if (!error && hw->drain)
-		    error = hw->drain(sc->hw_hdl);
-		break;
-
-	case AUDIO_GETDEV:
-		DPRINTF(("AUDIO_GETDEV\n"));
-		error = hw->getdev(sc->hw_hdl, (audio_device_t *)addr);
-		break;
-		
-	case AUDIO_GETENC:
-		DPRINTF(("AUDIO_GETENC\n"));
-		error = hw->query_encoding(sc->hw_hdl, (struct audio_encoding *)addr);
-		break;
-
-#ifdef COMPAT_12
-	/* GETPROPS contains the same info (and more) */
-	case AUDIO_GETFD:
-		DPRINTF(("AUDIO_GETFD\n"));
-		*(int *)addr = 
-		  (hw->get_props(sc->hw_hdl) & AUDIO_PROP_FULLDUPLEX) != 0;
-		break;
-#endif
-	case AUDIO_SETFD:
-		DPRINTF(("AUDIO_SETFD\n"));
-		fd = *(int *)addr;
-		if (hw->get_props(sc->hw_hdl) & AUDIO_PROP_FULLDUPLEX) {
-			if (hw->setfd)
-				error = hw->setfd(sc->hw_hdl, fd);
-			else
-				error = 0;
-			if (!error)
-				sc->sc_full_duplex = fd;
-		} else {
-			if (fd)
-				error = ENOTTY;
-			else
-				error = 0;
-		}
-		break;
-
-	case AUDIO_GETPROPS:
-		DPRINTF(("AUDIO_GETPROPS\n"));
-		*(int *)addr = hw->get_props(sc->hw_hdl);
-		break;
-
-	default:
-		DPRINTF(("audio_ioctl: unknown ioctl\n"));
-		error = EINVAL;
-		break;
-	}
-	DPRINTF(("audio_ioctl(%d,'%c',%d) result %d\n",
-	          IOCPARM_LEN(cmd), IOCGROUP(cmd), cmd&0xff, error));
-	return (error);
-}
-
-int
-audio_select(dev, rw, p)
-	dev_t dev;
-	int rw;
-	struct proc *p;
-{
-	int unit = AUDIOUNIT(dev);
-	struct audio_softc *sc = new_audio_cd.cd_devs[unit];
-	int s = splaudio();
-
-#if 0
-	DPRINTF(("audio_select: rw=%d mode=%d rblks=%d rr.nblk=%d\n",
-	         rw, sc->sc_mode, sc->sc_rblks, sc->rr.nblk));
-#endif
-	switch (rw) {
-
-	case FREAD:
-		if ((sc->sc_mode & AUMODE_PLAY) ?
-		    sc->sc_pr.stamp > sc->sc_wstamp : 
-		    sc->sc_rr.used > sc->sc_rr.usedlow) {
-			splx(s);
-			return (1);
-		}
-		selrecord(p, &sc->sc_rsel);
-		break;
-
-	case FWRITE:
-		if (sc->sc_mode & AUMODE_RECORD ||
-		    sc->sc_pr.used <= sc->sc_pr.usedlow) {
-			splx(s);
-			return (1);
-		}
-		selrecord(p, &sc->sc_wsel);
-		break;
-	}
-	splx(s);
-	return (0);
-}
-
-int
-audio_mmap(dev, off, prot)
-	dev_t dev;
-	int off, prot;
-{
-	int s;
-	int unit = AUDIOUNIT(dev);
-	struct audio_softc *sc = new_audio_cd.cd_devs[unit];
-	struct audio_hw_if *hw = sc->hw_if;
-	struct audio_ringbuffer *cb;
-
-	DPRINTF(("audio_mmap: off=%d, prot=%d\n", off, prot));
-
-	if (!(hw->get_props(sc->hw_hdl) & AUDIO_PROP_MMAP) || !hw->mappage)
-		return -1;
-#if 0
-/* XXX
- * The idea here was to use the protection to determine if
- * we are mapping the read or write buffer, but it fails.
- * The VM system is broken in (at least) two ways.
- * 1) If you map memory VM_PROT_WRITE you SIGSEGV
- *    when writing to it, so VM_PROT_READ|VM_PROT_WRITE
- *    has to be used for mmapping the play buffer.
- * 2) Even if calling mmap() with VM_PROT_READ|VM_PROT_WRITE
- *    audio_mmap will get called at some point with VM_PROT_READ
- *    only.
- * So, alas, we always map the play buffer for now.
- */
-	if (prot == (VM_PROT_READ|VM_PROT_WRITE) ||
-	    prot == VM_PROT_WRITE)
-		cb = &sc->sc_pr;
-	else if (prot == VM_PROT_READ)
-		cb = &sc->sc_rr;
-	else
-		return -1;
-#else
-	cb = &sc->sc_pr;
-#endif
-
-	if (off >= cb->bufsize)
-		return -1;
-	if (!cb->mmapped) {
-		cb->mmapped = 1;
-		if (cb == &sc->sc_pr) {
-			audio_fill_silence(&sc->sc_pparams, cb->start, cb->bufsize);
-			s = splaudio();
-			if (!sc->sc_pbus)
-				(void)audiostartp(sc);
-			splx(s);
-		} else {
-			s = splaudio();
-			if (!sc->sc_rbus)
-				(void)audiostartr(sc);
-			splx(s);
-		}
-	}
-
-	return hw->mappage(sc->hw_hdl, cb->start, off, prot);
-}
-
-int
-audiostartr(sc)
-	struct audio_softc *sc;
-{
-	int error;
-    
-    	DPRINTF(("audiostartr: start=%p used=%d(hi=%d) mmapped=%d\n", 
-		 sc->sc_rr.start, sc->sc_rr.used, sc->sc_rr.usedhigh, 
-		 sc->sc_rr.mmapped));
-
-	error = sc->hw_if->start_input(sc->hw_hdl, sc->sc_rr.start, 
-				       sc->sc_rr.blksize, audio_rint, (void *)sc);
-	if (error) {
-		DPRINTF(("audiostartr failed: %d\n", error));
-		return error;
-	}
-	sc->sc_rbus = 1;
-	return 0;
-}
-
-int
-audiostartp(sc)
-	struct audio_softc *sc;
-{
-	int error;
-    
-    	DPRINTF(("audiostartp: start=%p used=%d(hi=%d) mmapped=%d\n", 
-		 sc->sc_pr.start, sc->sc_pr.used, sc->sc_pr.usedhigh,
-		 sc->sc_pr.mmapped));
-    
-	if (sc->sc_pr.used >= sc->sc_pr.blksize || sc->sc_pr.mmapped) {
-		error = sc->hw_if->start_output(sc->hw_hdl, sc->sc_pr.outp,
-					sc->sc_pr.blksize, audio_pint, (void *)sc);
-		if (error) {
-			DPRINTF(("audiostartp failed: %d\n", error));
-		    	return error;
-		}
-		sc->sc_pbus = 1;
-	}
-	return 0;
-}
-
-/*
- * When the play interrupt routine finds that the write isn't keeping
- * the buffer filled it will insert silence in the buffer to make up
- * for this.  The part of the buffer that is filled with silence
- * is kept track of in a very approcimate way: it starts at sc_sil_start
- * and extends sc_sil_count bytes.  If the writer doesn't write sc_sil_count
- * get to encompass the whole buffer after which no more filling needs
- * to be done.  When the writer starts again sc_sil_count is set to 0.
- */
-/* XXX
- * Putting silence into the output buffer should not really be done
- * at splaudio, but there is no softaudio level to do it at yet.
- */
-static __inline void
-audio_pint_silence(sc, cb, inp, cc)
-	struct audio_softc *sc;
-	struct audio_ringbuffer *cb;
-	u_char *inp;
-	int cc;
-{
-	u_char *s, *e, *p, *q;
-
-	if (sc->sc_sil_count > 0) {
-		s = sc->sc_sil_start; /* start of silence */
-		e = s + sc->sc_sil_count; /* end of silence, may be beyond end */
-		p = inp;	/* adjusted pointer to area to fill */
-		if (p < s)
-			p += cb->end - cb->start;
-		q = p+cc;
-		/* Check if there is already silence. */
-		if (!(s <= p && p <  e &&
-		      s <= q && q <= e)) {
-			if (s <= p)
-				sc->sc_sil_count = max(sc->sc_sil_count, q - s);
-#ifdef AUDIO_DEBUG
-			if (audiodebug > 2)
-				printf("audio_pint_silence: fill cc=%d inp=%p, count=%d size=%d\n", 
-				       cc, inp, sc->sc_sil_count, (int)(cb->end - cb->start));
-#endif
-			audio_fill_silence(&sc->sc_pparams, inp, cc);
-		} else {
-#ifdef AUDIO_DEBUG
-			if (audiodebug > 2)
-				printf("audio_pint_silence: already silent cc=%d inp=%p\n", cc, inp);
-#endif
-			
-		}
-	} else {
-		sc->sc_sil_start = inp;
-		sc->sc_sil_count = cc;
-#ifdef AUDIO_DEBUG
-		if (audiodebug > 2)
-			printf("audio_pint_silence: start fill %p %d\n", inp, cc);
-#endif
-		audio_fill_silence(&sc->sc_pparams, inp, cc);
-	}
-}
-
-/*
- * Called from HW driver module on completion of dma output.
- * Start output of new block, wrap in ring buffer if needed.
- * If no more buffers to play, output zero instead.
- * Do a wakeup if necessary.
- */
-void
-audio_pint(v)
-	void *v;
-{
-	struct audio_softc *sc = v;
-	struct audio_hw_if *hw = sc->hw_if;
-	struct audio_ringbuffer *cb = &sc->sc_pr;
-	u_char *inp;
-	int cc, ccr;
-	int error;
-
-	cb->outp += cb->blksize;
-	if (cb->outp >= cb->end)
-		cb->outp = cb->start;
-	cb->stamp += cb->blksize / sc->sc_pparams.factor;
-	if (cb->mmapped) {
-#ifdef AUDIO_DEBUG
-		if (audiodebug > 2)
-			printf("audio_pint: mmapped outp=%p cc=%d inp=%p\n", 
-			       cb->outp, cb->blksize, cb->inp);
-#endif
-		(void)hw->start_output(sc->hw_hdl, cb->outp, cb->blksize,
-				       audio_pint, (void *)sc);
-		return;
-	}
-		
-#ifdef AUDIO_INTR_TIME
-	{
-		struct timeval tv;
-		u_long t;
-		microtime(&tv);
-		t = tv.tv_usec + 1000000 * tv.tv_sec;
-		if (sc->sc_pnintr) {
-			long lastdelta, totdelta;
-			lastdelta = t - sc->sc_plastintr - sc->sc_pblktime;
-			if (lastdelta > sc->sc_pblktime / 5) {
-				printf("audio: play interrupt(%d) off relative by %ld us (%lu)\n", 
-				       sc->sc_pnintr, lastdelta, sc->sc_pblktime);
-			}
-			totdelta = t - sc->sc_pfirstintr - sc->sc_pblktime * sc->sc_pnintr;
-			if (totdelta > sc->sc_pblktime / 2) {
-				sc->sc_pnintr++;
-				printf("audio: play interrupt(%d) off absolute by %ld us (%lu)\n", 
-				       sc->sc_pnintr, totdelta, sc->sc_pblktime);
-				sc->sc_pnintr++; /* avoid repeated messages */
-			}
-		} else
-			sc->sc_pfirstintr = t;
-		sc->sc_plastintr = t;
-		sc->sc_pnintr++;
-	}
-#endif
-
-	cb->used -= cb->blksize;
-	if (cb->used < cb->blksize) {
-		/* we don't have a full block to use */
-		if (cb->copying) {
-			/* writer is in progress, don't disturb */
-			cb->needfill = 1;
-#ifdef AUDIO_DEBUG
-			if (audiodebug > 1)
-			    printf("audio_pint: copying in progress\n");
-#endif
-		} else {
-			inp = cb->inp;
-			cc = cb->blksize - (inp - cb->start) % cb->blksize;
-			ccr = cc / sc->sc_pparams.factor;
-			if (cb->pause)
-				cb->pdrops += ccr;
-			else {
-				cb->drops += ccr;
-				sc->sc_playdrop += ccr;
-			}
-			audio_pint_silence(sc, cb, inp, cc);
-			inp += cc;
-			if (inp >= cb->end)
-				inp = cb->start;
-			cb->inp = inp;
-			cb->used += cc;
-
-			/* Clear next block so we keep ahead of the DMA. */
-			if (cb->used + cc < cb->usedhigh)
-				audio_pint_silence(sc, cb, inp, cb->blksize);
-		}
-	}
-
-#ifdef AUDIO_DEBUG
-	if (audiodebug > 3)
-		printf("audio_pint: outp=%p cc=%d\n", cb->outp, cb->blksize);
-#endif
-	error = hw->start_output(sc->hw_hdl, cb->outp, cb->blksize,
-				 audio_pint, (void *)sc);
-	if (error) {
-		/* XXX does this really help? */
-		DPRINTF(("audio_pint restart failed: %d\n", error));
-		audio_clear(sc);
-	}
-
-#ifdef AUDIO_DEBUG
-	if (audiodebug > 3)
-		printf("audio_pint: mode=%d pause=%d used=%d lowat=%d\n",
-			 sc->sc_mode, cb->pause, cb->used, cb->usedlow);
-#endif
-	if ((sc->sc_mode & AUMODE_PLAY) && !cb->pause) {
-		if (cb->used <= cb->usedlow) {
-			audio_wakeup(&sc->sc_wchan);
-			selwakeup(&sc->sc_wsel);
-			if (sc->sc_async_audio) {
-#ifdef AUDIO_DEBUG
-				if (audiodebug > 3)
-					printf("audio_pint: sending SIGIO %p\n", 
-					       sc->sc_async_audio);
-#endif
-				psignal(sc->sc_async_audio, SIGIO);
-			}
-		}
-	}
-
-	/* Possible to return one or more "phantom blocks" now. */
-	if (!sc->sc_full_duplex && sc->sc_rchan) {
-		audio_wakeup(&sc->sc_rchan);
-		selwakeup(&sc->sc_rsel);
-		if (sc->sc_async_audio)
-			psignal(sc->sc_async_audio, SIGIO);
-	}
-}
-
-/*
- * Called from HW driver module on completion of dma input.
- * Mark it as input in the ring buffer (fiddle pointers).
- * Do a wakeup if necessary.
- */
-void
-audio_rint(v)
-	void *v;
-{
-	struct audio_softc *sc = v;
-	struct audio_hw_if *hw = sc->hw_if;
-	struct audio_ringbuffer *cb = &sc->sc_rr;
-	int error;
-
-	cb->inp += cb->blksize;
-	if (cb->inp >= cb->end)
-		cb->inp = cb->start;
-	cb->stamp += cb->blksize;
-	if (cb->mmapped) {
-#ifdef AUDIO_DEBUG
-		if (audiodebug > 2)
-			printf("audio_rint: mmapped inp=%p cc=%d\n", 
-			       cb->inp, cb->blksize);
-#endif
-		(void)hw->start_input(sc->hw_hdl, cb->inp, cb->blksize,
-				      audio_rint, (void *)sc);
-		return;
-	}
-
-#ifdef AUDIO_INTR_TIME
-	{
-		struct timeval tv;
-		u_long t;
-		microtime(&tv);
-		t = tv.tv_usec + 1000000 * tv.tv_sec;
-		if (sc->sc_rnintr) {
-			long lastdelta, totdelta;
-			lastdelta = t - sc->sc_rlastintr - sc->sc_rblktime;
-			if (lastdelta > sc->sc_rblktime / 5) {
-				printf("audio: record interrupt(%d) off relative by %ld us (%lu)\n", 
-				       sc->sc_rnintr, lastdelta, sc->sc_rblktime);
-			}
-			totdelta = t - sc->sc_rfirstintr - sc->sc_rblktime * sc->sc_rnintr;
-			if (totdelta > sc->sc_rblktime / 2) {
-				sc->sc_rnintr++;
-				printf("audio: record interrupt(%d) off absolute by %ld us (%lu)\n", 
-				       sc->sc_rnintr, totdelta, sc->sc_rblktime);
-				sc->sc_rnintr++; /* avoid repeated messages */
-			}
-		} else
-			sc->sc_rfirstintr = t;
-		sc->sc_rlastintr = t;
-		sc->sc_rnintr++;
-	}
-#endif
-
-	cb->used += cb->blksize;
-	if (cb->pause) {
-#ifdef AUDIO_DEBUG
-		if (audiodebug > 1)
-			printf("audio_rint: pdrops %lu\n", cb->pdrops);
-#endif
-		cb->pdrops += cb->blksize;
-		cb->outp += cb->blksize;
-		cb->used -= cb->blksize;
-	} else if (cb->used + cb->blksize >= cb->usedhigh && !cb->copying) {
-#ifdef AUDIO_DEBUG
-		if (audiodebug > 1)
-			printf("audio_rint: drops %lu\n", cb->drops);
-#endif
-		cb->drops += cb->blksize;
-		cb->outp += cb->blksize;
-		cb->used -= cb->blksize;
-	}
-
-#ifdef AUDIO_DEBUG
-	if (audiodebug > 2)
-		printf("audio_rint: inp=%p cc=%d used=%d\n", 
-		       cb->inp, cb->blksize, cb->used);
-#endif
-	error = hw->start_input(sc->hw_hdl, cb->inp, cb->blksize,
-				audio_rint, (void *)sc);
-	if (error) {
-		/* XXX does this really help? */
-		DPRINTF(("audio_rint: restart failed: %d\n", error));
-		audio_clear(sc);
-	}
-
-	audio_wakeup(&sc->sc_rchan);
-	selwakeup(&sc->sc_rsel);
-	if (sc->sc_async_audio)
-		psignal(sc->sc_async_audio, SIGIO);
-}
-
-int
-audio_check_params(p)
-	struct audio_params *p;
-{
-#if defined(COMPAT_12)
-	if (p->encoding == AUDIO_ENCODING_PCM16) {
-		if (p->precision == 8)
-			p->encoding = AUDIO_ENCODING_ULINEAR;
-		else
-			p->encoding = AUDIO_ENCODING_SLINEAR;
-	} else if (p->encoding == AUDIO_ENCODING_PCM8) {
-		if (p->precision == 8)
-			p->encoding = AUDIO_ENCODING_ULINEAR;
-		else
-			return EINVAL;
-	}
-#endif
-
-	if (p->encoding == AUDIO_ENCODING_SLINEAR)
-#if BYTE_ORDER == LITTLE_ENDIAN
-		p->encoding = AUDIO_ENCODING_SLINEAR_LE;
-#else
-		p->encoding = AUDIO_ENCODING_SLINEAR_BE;
-#endif
-	if (p->encoding == AUDIO_ENCODING_ULINEAR)
-#if BYTE_ORDER == LITTLE_ENDIAN
-		p->encoding = AUDIO_ENCODING_ULINEAR_LE;
-#else
-		p->encoding = AUDIO_ENCODING_ULINEAR_BE;
-#endif
-
-	switch (p->encoding) {
-	case AUDIO_ENCODING_ULAW:
-	case AUDIO_ENCODING_ALAW:
-	case AUDIO_ENCODING_ADPCM:
-		if (p->precision != 8)
-			return (EINVAL);
-		break;
-	case AUDIO_ENCODING_SLINEAR_LE:
-	case AUDIO_ENCODING_SLINEAR_BE:
-	case AUDIO_ENCODING_ULINEAR_LE:
-	case AUDIO_ENCODING_ULINEAR_BE:
-		if (p->precision != 8 && p->precision != 16)
-			return (EINVAL);
-		break;
-	default:
-		return (EINVAL);
-	}
-
-	if (p->channels < 1 || p->channels > 8)	/* sanity check # of channels */
-		return (EINVAL);
-
-	return (0);
-}
-
-int
-audiosetinfo(sc, ai)
-	struct audio_softc *sc;
-	struct audio_info *ai;
-{
-	struct audio_prinfo *r = &ai->record, *p = &ai->play;
-	int cleared;
-	int s, setmode;
-	int error;
-	struct audio_hw_if *hw = sc->hw_if;
-	mixer_ctrl_t ct;
-	struct audio_params pp, rp;
-	int np, nr;
-	unsigned int blks;
-	int oldpblksize, oldrblksize;
-	int rbus, pbus;
-	
-	if (hw == 0)		/* HW has not attached */
-		return(ENXIO);
-
-	rbus = sc->sc_rbus;
-	pbus = sc->sc_pbus;
-	error = 0;
-	cleared = 0;
-
-	pp = sc->sc_pparams;	/* Temporary encoding storage in */
-	rp = sc->sc_rparams;	/* case setting the modes fails. */
-	nr = np = 0;
-
-	if (p->sample_rate != ~0) {
-		pp.sample_rate = p->sample_rate;
-		np++;
-	}
-	if (r->sample_rate != ~0) {
-		rp.sample_rate = r->sample_rate;
-		nr++;
-	}
-	if (p->encoding != ~0) {
-		pp.encoding = p->encoding;
-		np++;
-	}	
-	if (r->encoding != ~0) {
-		rp.encoding = r->encoding;
-		nr++;
-	}
-	if (p->precision != ~0) {
-		pp.precision = p->precision;
-		np++;
-	}
-	if (r->precision != ~0) {
-		rp.precision = r->precision;
-		nr++;
-	}
-	if (p->channels != ~0) {
-		pp.channels = p->channels;
-		np++;
-	}
-	if (r->channels != ~0) {
-		rp.channels = r->channels;
-		nr++;
-	}
-#ifdef AUDIO_DEBUG
-	if (audiodebug && nr)
-	    audio_print_params("Setting record params", &rp);
-	if (audiodebug && np)
-	    audio_print_params("Setting play params", &pp);
-#endif
-	if (nr && (error = audio_check_params(&rp)))
-		return error;
-	if (np && (error = audio_check_params(&pp)))
-		return error;
-	setmode = 0;
-	if (nr) {
-		if (!cleared)
-			audio_clear(sc);
-		cleared = 1;
-		rp.sw_code = 0;
-		rp.factor = 1;
-		setmode |= AUMODE_RECORD;
-	}
-	if (np) {
-		if (!cleared)
-			audio_clear(sc);
-		cleared = 1;
-		pp.sw_code = 0;
-		pp.factor = 1;
-		setmode |= AUMODE_PLAY;
-	}
-
-	if (ai->mode != ~0) {
-		if (!cleared)
-			audio_clear(sc);
-		cleared = 1;
-		sc->sc_mode = ai->mode;
-		if (sc->sc_mode & AUMODE_PLAY_ALL)
-			sc->sc_mode |= AUMODE_PLAY;
-		if ((sc->sc_mode & AUMODE_PLAY) && !sc->sc_full_duplex)
-			/* Play takes precedence */
-			sc->sc_mode &= ~AUMODE_RECORD;
-	}
-
-	if (setmode) {
-		int indep = hw->get_props(sc->hw_hdl) & AUDIO_PROP_INDEPENDENT;
-		if (!indep) {
-			if (setmode == AUMODE_RECORD)
-				pp = rp;
-			else if (setmode == AUMODE_PLAY)
-				rp = pp;
-		}
-		error = hw->set_params(sc->hw_hdl, setmode, sc->sc_mode, &pp, &rp);
-		if (error)
-			return (error);
-		if (!indep) {
-			if (setmode == AUMODE_RECORD) {
-				pp.sample_rate = rp.sample_rate;
-				pp.encoding    = rp.encoding;
-				pp.channels    = rp.channels;
-				pp.precision   = rp.precision;
-			} else if (setmode == AUMODE_PLAY) {
-				rp.sample_rate = pp.sample_rate;
-				rp.encoding    = pp.encoding;
-				rp.channels    = pp.channels;
-				rp.precision   = pp.precision;
-			}
-		}
-		if (setmode & AUMODE_RECORD)
-			sc->sc_rparams = rp;
-		if (setmode & AUMODE_PLAY)
-			sc->sc_pparams = pp;
-	}
-
-	oldpblksize = sc->sc_pr.blksize;
-	oldrblksize = sc->sc_rr.blksize;
-	/* Play params can affect the record params, so recalculate blksize. */
-	if (nr || np) {
-		audio_calc_blksize(sc, AUMODE_RECORD);
-		audio_calc_blksize(sc, AUMODE_PLAY);
-	}
-#ifdef AUDIO_DEBUG
-	if (audiodebug > 1 && nr)
-	    audio_print_params("After setting record params", &sc->sc_rparams);
-	if (audiodebug > 1 && np)
-	    audio_print_params("After setting play params", &sc->sc_pparams);
-#endif
-
-	if (p->port != ~0) {
-		if (!cleared)
-			audio_clear(sc);
-		cleared = 1;
-
-		error = hw->set_out_port(sc->hw_hdl, p->port);
-		if (error)
-			return(error);
-	}
-	if (r->port != ~0) {
-		if (!cleared)
-			audio_clear(sc);
-		cleared = 1;
-
-		error = hw->set_in_port(sc->hw_hdl, r->port);
-		if (error)
-			return(error);
-	}
-	if (p->gain != ~0) {
-		ct.dev = hw->get_out_port(sc->hw_hdl);
-		ct.type = AUDIO_MIXER_VALUE;
-		ct.un.value.num_channels = 1;
-		ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = p->gain;
-		error = hw->set_port(sc->hw_hdl, &ct);
-		if (error)
-			return(error);
-	}
-	if (r->gain != ~0) {
-		ct.dev = hw->get_in_port(sc->hw_hdl);
-		ct.type = AUDIO_MIXER_VALUE;
-		ct.un.value.num_channels = 1;
-		ct.un.value.level[AUDIO_MIXER_LEVEL_MONO] = r->gain;
-		error = hw->set_port(sc->hw_hdl, &ct);
-		if (error)
-			return(error);
-	}
-	
-	if (p->pause != (u_char)~0) {
-		sc->sc_pr.pause = p->pause;
-		if (!p->pause && !sc->sc_pbus) {
-			s = splaudio();
-			error = audiostartp(sc);
-			splx(s);
-			if (error)
-				return error;
-		}
-	}
-	if (r->pause != (u_char)~0) {
-		sc->sc_rr.pause = r->pause;
-		if (!r->pause && !sc->sc_rbus) {
-			s = splaudio();
-			error = audiostartr(sc);
-			splx(s);
-			if (error)
-				return error;
-		}
-	}
-
-	if (ai->blocksize != ~0) {
-		/* Block size specified explicitly. */
-		if (!cleared)
-			audio_clear(sc);
-		cleared = 1;
-
-		if (ai->blocksize == 0) {
-			audio_calc_blksize(sc, AUMODE_RECORD);
-			audio_calc_blksize(sc, AUMODE_PLAY);
-			sc->sc_blkset = 0;
-		} else {
-			int bs = ai->blocksize;
-			if (hw->round_blocksize)
-				bs = hw->round_blocksize(sc->hw_hdl, bs);
-			sc->sc_pr.blksize = sc->sc_rr.blksize = bs;
-			sc->sc_blkset = 1;
-		}
-	}
-
-	if (ai->mode != ~0) {
-		if (sc->sc_mode & AUMODE_PLAY)
-			audio_init_play(sc);
-		if (sc->sc_mode & AUMODE_RECORD)
-			audio_init_record(sc);
-	}
-
-	if (hw->commit_settings) {
-		error = hw->commit_settings(sc->hw_hdl);
-		if (error)
-			return (error);
-		if (p->gain != ~0 || r->gain != ~0)
-			mixer_signal(sc);
-	}
-
-	if (cleared) {
-		s = splaudio();
-		error = audio_initbufs(sc);
-		if (error) goto err;
-		if (sc->sc_pr.blksize != oldpblksize ||
-		    sc->sc_rr.blksize != oldrblksize)
-			audio_calcwater(sc);
-		if ((sc->sc_mode & AUMODE_PLAY) &&
-		    pbus && !sc->sc_pbus)
-			error = audiostartp(sc);
-		if (!error && 
-		    (sc->sc_mode & AUMODE_RECORD) &&
-		    rbus && !sc->sc_rbus)
-			error = audiostartr(sc);
-	err:
-		splx(s);
-		if (error)
-			return error;
-	}
-	
-	/* Change water marks after initializing the buffers. */
-	if (ai->hiwat != ~0) {
-		blks = ai->hiwat;
-		if (blks > sc->sc_pr.maxblks)
-			blks = sc->sc_pr.maxblks;
-		if (blks < 1)
-			blks = 1;
-		sc->sc_pr.usedhigh = blks * sc->sc_pr.blksize;
-	}
-	if (ai->lowat != ~0) {
-		blks = ai->lowat;
-		if (blks > sc->sc_pr.maxblks - 1)
-			blks = sc->sc_pr.maxblks - 1;
-		sc->sc_pr.usedlow = blks * sc->sc_pr.blksize;
-	}
-
-	return (0);
-}
-
-int
-audiogetinfo(sc, ai)
-	struct audio_softc *sc;
-	struct audio_info *ai;
-{
-	struct audio_prinfo *r = &ai->record, *p = &ai->play;
-	struct audio_hw_if *hw = sc->hw_if;
-	mixer_ctrl_t ct;
-	
-	if (hw == 0)		/* HW has not attached */
-		return(ENXIO);
-	
-	p->sample_rate = sc->sc_pparams.sample_rate;
-	r->sample_rate = sc->sc_rparams.sample_rate;
-	p->channels = sc->sc_pparams.channels;
-	r->channels = sc->sc_rparams.channels;
-	p->precision = sc->sc_pparams.precision;
-	r->precision = sc->sc_rparams.precision;
-	p->encoding = sc->sc_pparams.encoding;
-	r->encoding = sc->sc_rparams.encoding;
-
-	r->port = hw->get_in_port(sc->hw_hdl);
-	p->port = hw->get_out_port(sc->hw_hdl);
-
-	ct.dev = r->port;
-	ct.type = AUDIO_MIXER_VALUE;
-	ct.un.value.num_channels = 1;
-	if (hw->get_port(sc->hw_hdl, &ct) == 0)
-	    r->gain = ct.un.value.level[AUDIO_MIXER_LEVEL_MONO];
-	else
-	    r->gain = AUDIO_MAX_GAIN/2;
-
-	ct.dev = p->port;
-	ct.un.value.num_channels = 1;
-	if (hw->get_port(sc->hw_hdl, &ct) == 0)
-	    p->gain = ct.un.value.level[AUDIO_MIXER_LEVEL_MONO];
-	else
-	    p->gain = AUDIO_MAX_GAIN/2;
-
-	p->seek = sc->sc_pr.used;
-	r->seek = sc->sc_rr.used;
-
-	p->samples = sc->sc_pr.stamp - sc->sc_pr.drops;
-	r->samples = sc->sc_rr.stamp - sc->sc_rr.drops;
-
-	p->eof = sc->sc_eof;
-	r->eof = 0;
-
-	p->pause = sc->sc_pr.pause;
-	r->pause = sc->sc_rr.pause;
-
-	p->error = sc->sc_pr.drops != 0;
-	r->error = sc->sc_rr.drops != 0;
-
-	p->waiting = r->waiting = 0;		/* open never hangs */
-
-	p->open = (sc->sc_open & AUOPEN_WRITE) != 0;
-	r->open = (sc->sc_open & AUOPEN_READ) != 0;
-
-	p->active = sc->sc_pbus;
-	r->active = sc->sc_rbus;
-
-	ai->buffersize = sc->sc_pr.bufsize;
-	ai->blocksize = sc->sc_pr.blksize;
-	ai->hiwat = sc->sc_pr.usedhigh / sc->sc_pr.blksize;
-	ai->lowat = sc->sc_pr.usedlow / sc->sc_pr.blksize;
-	ai->backlog = 0;	/* unused */
-	ai->mode = sc->sc_mode;
-
-	return (0);
-}
-
-/*
- * Mixer driver
- */
-int
-mixer_open(dev, flags, ifmt, p)
-	dev_t dev;
-	int flags, ifmt;
-	struct proc *p;
-{
-	int unit = AUDIOUNIT(dev);
-	struct audio_softc *sc;
-
-	if (unit >= new_audio_cd.cd_ndevs ||
-	    (sc = new_audio_cd.cd_devs[unit]) == NULL)
-		return ENXIO;
-
-	if (!sc->hw_if)
-		return (ENXIO);
-
-	DPRINTF(("mixer_open: dev=0x%x flags=0x%x sc=%p\n", dev, flags, sc));
-
-	return (0);
-}
-
-/*
- * Remove a process from those to be signalled on mixer activity.
- */
-static void
-mixer_remove(sc, p)
-	struct audio_softc *sc;
-	struct proc *p;
-{
-	struct mixer_asyncs **pm, *m;
-
-	for(pm = &sc->sc_async_mixer; *pm; pm = &(*pm)->next) {
-		if ((*pm)->proc == p) {
-			m = *pm;
-			*pm = m->next;
-			free(m, M_DEVBUF);
-			return;
-		}
-	}
-}
-
-/*
- * Signal all processes waitinf for the mixer.
- */
-static void
-mixer_signal(sc)
-	struct audio_softc *sc;
-{
-	struct mixer_asyncs *m;
-
-	for(m = sc->sc_async_mixer; m; m = m->next)
-		psignal(m->proc, SIGIO);
-}
-
-/*
- * Close a mixer device
- */
-/* ARGSUSED */
-int
-mixer_close(dev, flags, ifmt, p)
-	dev_t dev;
-	int flags, ifmt;
-	struct proc *p;
-{
-	int unit = AUDIOUNIT(dev);
-	struct audio_softc *sc = new_audio_cd.cd_devs[unit];
-
-	DPRINTF(("mixer_close: unit %d\n", AUDIOUNIT(dev)));
-	
-	mixer_remove(sc, p);
-
-	return (0);
-}
-
-int
-mixer_ioctl(dev, cmd, addr, flag, p)
-	dev_t dev;
-	int cmd;
-	caddr_t addr;
-	int flag;
-	struct proc *p;
-{
-	int unit = AUDIOUNIT(dev);
-	struct audio_softc *sc = new_audio_cd.cd_devs[unit];
-	struct audio_hw_if *hw = sc->hw_if;
-	int error = EINVAL;
-
-	DPRINTF(("mixer_ioctl(%d,'%c',%d)\n",
-		 IOCPARM_LEN(cmd), IOCGROUP(cmd), cmd&0xff));
-
-	switch (cmd) {
-	case FIOASYNC:
-		mixer_remove(sc, p); /* remove old entry */
-		if (*(int *)addr) {
-			struct mixer_asyncs *ma;
-			ma = malloc(sizeof (struct mixer_asyncs), M_DEVBUF, M_WAITOK);
-			ma->next = sc->sc_async_mixer;
-			ma->proc = p;
-			sc->sc_async_mixer = ma;
-		}
-		error = 0;
-		break;
-
-	case AUDIO_GETDEV:
-		DPRINTF(("AUDIO_GETDEV\n"));
-		error = hw->getdev(sc->hw_hdl, (audio_device_t *)addr);
-		break;
-		
-	case AUDIO_MIXER_DEVINFO:
-		DPRINTF(("AUDIO_MIXER_DEVINFO\n"));
-		error = hw->query_devinfo(sc->hw_hdl, (mixer_devinfo_t *)addr);
-		break;
-
-	case AUDIO_MIXER_READ:
-		DPRINTF(("AUDIO_MIXER_READ\n"));
-		error = hw->get_port(sc->hw_hdl, (mixer_ctrl_t *)addr);
-		break;
-
-	case AUDIO_MIXER_WRITE:
-		DPRINTF(("AUDIO_MIXER_WRITE\n"));
-		error = hw->set_port(sc->hw_hdl, (mixer_ctrl_t *)addr);
-		if (!error && hw->commit_settings)
-			error = hw->commit_settings(sc->hw_hdl);
-		if (!error)
-			mixer_signal(sc);
-		break;
-
-	default:
-		error = EINVAL;
-		break;
-	}
-	DPRINTF(("mixer_ioctl(%d,'%c',%d) result %d\n",
-		 IOCPARM_LEN(cmd), IOCGROUP(cmd), cmd&0xff, error));
-	return (error);
-}
-#endif
diff --git a/sys/arch/amiga/dev/audio_if.h b/sys/arch/amiga/dev/audio_if.h
deleted file mode 100644
index 7daecc79b34..00000000000
--- a/sys/arch/amiga/dev/audio_if.h
+++ /dev/null
@@ -1,172 +0,0 @@
-/*	$OpenBSD: audio_if.h,v 1.1 1997/10/07 11:04:59 niklas Exp $	*/
-/*	$NetBSD: audio_if.h,v 1.21 1997/09/06 01:14:49 augustss Exp $	*/
-
-/*
- * Copyright (c) 1994 Havard Eidnes.
- * 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.
- *
- */
-
-/*
- * Generic interface to hardware driver.
- */
-
-struct audio_softc;
-
-struct audio_params {
-	u_long	sample_rate;			/* sample rate */
-	u_int	encoding;			/* e.g. ulaw, linear, etc */
-	u_int	precision;			/* bits/sample */
-	u_int	channels;			/* mono(1), stereo(2) */
-	/* Software en/decode functions, set if SW coding required by HW */
-	void	(*sw_code)__P((void *, u_char *, int));
-	int	factor;				/* coding space change */
-};
-
-/* The default audio mode: 8 kHz mono ulaw */
-extern struct audio_params audio_default;
-
-struct audio_hw_if {
-	int	(*open)__P((void *, int));	/* open hardware */
-	void	(*close)__P((void *));		/* close hardware */
-	int	(*drain)__P((void *));		/* Optional: drain buffers */
-	
-	/* Encoding. */
-	/* XXX should we have separate in/out? */
-	int	(*query_encoding)__P((void *, struct audio_encoding *));
-
-	/* Set the audio encoding parameters (record and play).
-	 * Return 0 on success, or an error code if the 
-	 * requested parameters are impossible.
-	 * The values in the params struct may be changed (e.g. rounding
-	 * to the nearest sample rate.)
-	 */
-        int	(*set_params)__P((void *, int, int, struct audio_params *, struct audio_params *));
-  
-	/* Hardware may have some say in the blocksize to choose */
-	int	(*round_blocksize)__P((void *, int));
-
-	/* Ports (in/out ports) */
-	int	(*set_out_port)__P((void *, int));
-	int	(*get_out_port)__P((void *));
-	int	(*set_in_port)__P((void *, int));
-	int	(*get_in_port)__P((void *));
-
-	/*
-	 * Changing settings may require taking device out of "data mode",
-	 * which can be quite expensive.  Also, audiosetinfo() may
-	 * change several settings in quick succession.  To avoid
-	 * having to take the device in/out of "data mode", we provide
-	 * this function which indicates completion of settings
-	 * adjustment.
-	 */
-	int	(*commit_settings)__P((void *));
-
-	/* Start input/output routines. These usually control DMA. */
-	int	(*init_output)__P((void *, void *, int));
-	int	(*init_input)__P((void *, void *, int));
-	int	(*start_output)__P((void *, void *, int,
-				    void (*)(void *), void *));
-	int	(*start_input)__P((void *, void *, int,
-				   void (*)(void *), void *));
-	int	(*halt_output)__P((void *));
-	int	(*halt_input)__P((void *));
-	int	(*cont_output)__P((void *));
-	int	(*cont_input)__P((void *));
-
-	int	(*speaker_ctl)__P((void *, int));
-#define SPKR_ON		1
-#define SPKR_OFF	0
-
-	int	(*getdev)__P((void *, struct audio_device *));
-	int	(*setfd)__P((void *, int));
-	
-	/* Mixer (in/out ports) */
-	int	(*set_port)__P((void *, mixer_ctrl_t *));
-	int	(*get_port)__P((void *, mixer_ctrl_t *));
-
-	int	(*query_devinfo)__P((void *, mixer_devinfo_t *));
-	
-	/* Allocate/free memory for the ring buffer. Usually malloc/free. */
-	void	*(*alloc)__P((void *, unsigned long, int, int));
-	void	(*free)__P((void *, void *, int));
-	unsigned long (*round_buffersize)__P((void *, unsigned long));
-	int	(*mappage)__P((void *, void *, int, int));
-
-	int 	(*get_props)__P((void *)); /* device properties */
-};
-
-struct midi_info {
-	char	*name;		/* Name of MIDI hardware */
-	int	props;
-};
-#define MIDI_PROP_OUT_INTR 1
-
-struct midi_hw_if {
-	int	(*open)__P((void *, int, 	/* open hardware */
-			    void (*)__P((void *, int)),
-			    void (*)__P((void *)),
-			    void *));
-	void	(*close)__P((void *));		/* close hardware */
-	int	(*output)__P((void *, int));	/* output a byte */
-	void	(*getinfo)__P((void *, struct midi_info *));
-	int	(*ioctl)__P((u_long, caddr_t, int, struct proc *));
-};
-
-struct audio_attach_args {
-	struct audio_hw_if *ahw;
-	struct midi_hw_if *mhw;
-	void *hdl;
-	char audiodone, mididone;
-};
-
-/* Attach the MI driver(s) to the MD driver. */
-extern void	audio_attach_mi __P((struct audio_hw_if *, struct midi_hw_if *, void *, struct device *));
-
-/* Device identity flags */
-#define SOUND_DEVICE		0
-#define AUDIO_DEVICE		0x80
-#define AUDIOCTL_DEVICE		0xc0
-#define MIXER_DEVICE		0x10
-
-#define AUDIOUNIT(x)		(minor(x)&0x0f)
-#define AUDIODEV(x)		(minor(x)&0xf0)
-
-#define ISDEVSOUND(x)		(AUDIODEV(minor(x)) == SOUND_DEVICE)
-#define ISDEVAUDIO(x)		(AUDIODEV(minor(x)) == AUDIO_DEVICE)
-#define ISDEVAUDIOCTL(x)	(AUDIODEV(minor(x)) == AUDIOCTL_DEVICE)
-#define ISDEVMIXER(x)		(AUDIODEV(minor(x)) == MIXER_DEVICE)
-
-#ifndef __i386__
-#define splaudio splbio		/* XXX */
-#define IPL_AUDIO IPL_BIO	/* XXX */
-#endif
-
diff --git a/sys/arch/amiga/dev/audiovar.h b/sys/arch/amiga/dev/audiovar.h
deleted file mode 100644
index 6e91e1c3b82..00000000000
--- a/sys/arch/amiga/dev/audiovar.h
+++ /dev/null
@@ -1,130 +0,0 @@
-/*	$OpenBSD: audiovar.h,v 1.1 1997/10/07 11:04:59 niklas Exp $	*/
-/*	$NetBSD: audiovar.h,v 1.16 1997/08/19 23:49:58 augustss Exp $	*/
-
-/*
- * Copyright (c) 1991-1993 Regents of the University of California.
- * All rights reserved.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions
- * are met:
- * 1. Redistributions of source code must retain the above copyright
- *    notice, this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright
- *    notice, this list of conditions and the following disclaimer in the
- *    documentation and/or other materials provided with the distribution.
- * 3. All advertising materials mentioning features or use of this software
- *    must display the following acknowledgement:
- *	This product includes software developed by the Computer Systems
- *	Engineering Group at Lawrence Berkeley Laboratory.
- * 4. Neither the name of the University nor of the Laboratory may be used
- *    to endorse or promote products derived from this software without
- *    specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND
- * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
- * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
- * ARE DISCLAIMED.  IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
- * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
- * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
- * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
- * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
- * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
- * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
- * SUCH DAMAGE.
- *
- *	From: Header: audiovar.h,v 1.3 93/07/18 14:07:25 mccanne Exp  (LBL)
- */
-
-/*
- * Initial/default block duration is both configurable and patchable.
- */
-#ifndef AUDIO_BLK_MS
-#define AUDIO_BLK_MS	50	/* 50 ms */
-#endif
-
-#ifndef AU_RING_SIZE
-#define AU_RING_SIZE		65536
-#endif
-
-#define AUMINBUF 512
-#define AUMINBLK 32
-#define AUMINNOBLK 3
-struct audio_ringbuffer {
-	int	bufsize;	/* allocated memory */
-	int	blksize;	/* I/O block size */
-	int	maxblks;	/* no of blocks in ring */
-	u_char	*start;		/* start of buffer area */
-	u_char	*end;		/* end of buffer area */
-	u_char	*inp;		/* input pointer (to buffer) */
-	u_char	*outp;		/* output pointer (from buffer) */
-	int	used;		/* no of used bytes */
-	int	usedlow;	/* start writer when used falls below this */
-	int	usedhigh;	/* stop writer when used goes above this */
-	u_long	stamp;		/* bytes transferred */
-	u_long	stamp_last;	/* old value of bytes transferred */
-	u_long	drops;		/* missed samples from over/underrun */
-	u_long	pdrops;		/* paused samples */
-	char	pause;		/* transfer is paused */
-	char	copying;	/* data is being copied */
-	char	needfill;	/* buffer needs filling when copying is done */
-	char	mmapped;	/* device is mmap()-ed */
-};
-
-/*
- * Software state, per audio device.
- */
-struct audio_softc {
-	struct	device dev;
-	void	*hw_hdl;	/* Hardware driver handle */
-	struct	audio_hw_if *hw_if; /* Hardware interface */
-	struct	device *sc_dev;	/* Hardware device struct */
-	u_char	sc_open;	/* single use device */
-#define AUOPEN_READ	0x01
-#define AUOPEN_WRITE	0x02
-	u_char	sc_mode;	/* bitmask for RECORD/PLAY */
-
-	struct	selinfo sc_wsel; /* write selector */
-	struct	selinfo sc_rsel; /* read selector */
-	struct	proc *sc_async_audio;	/* process who wants audio SIGIO */
-	struct	mixer_asyncs {
-		struct mixer_asyncs *next;
-		struct proc *proc;
-	} *sc_async_mixer;  /* processes who want mixer SIGIO */
-
-	/* Sleep channels for reading and writing. */
-	int	sc_rchan;
-	int	sc_wchan;
-
-	/* Ring buffers, separate for record and play. */
-	struct	audio_ringbuffer sc_rr; /* Record ring */
-	struct	audio_ringbuffer sc_pr; /* Play ring */
-
-	u_char	sc_blkset;	/* Blocksize has been set */
-
-	u_char	*sc_sil_start;	/* start of silence in buffer */
-	int	sc_sil_count;	/* # of silence bytes */
-
-	u_char	sc_rbus;	/* input dma in progress */
-	u_char	sc_pbus;	/* output dma in progress */
-
-	struct	audio_params sc_pparams;	/* play encoding parameters */
-	struct	audio_params sc_rparams;	/* record encoding parameters */
-
-	int	sc_eof;		/* EOF, i.e. zero sixed write, counter */
-	u_long	sc_wstamp;
-	u_long	sc_playdrop;
-
-	int	sc_full_duplex;	/* device in full duplex mode */
-
-#ifdef AUDIO_INTR_TIME
-	u_long	sc_pfirstintr;	/* first time we saw a xmit interrupt */
-	int	sc_pnintr;	/* number of interrupts */
-	u_long	sc_plastintr;	/* last time we saw a xmit interrupt */
-	long	sc_pblktime;	/* nominal time between interrupts */
-	u_long	sc_rfirstintr;	/* first time we saw a rec interrupt */
-	int	sc_rnintr;	/* number of interrupts */
-	u_long	sc_rlastintr;	/* last time we saw a xrec interrupt */
-	long	sc_rblktime;	/* nominal time between interrupts */
-#endif
-};