diff --git a/sys/arch/arm32/mainbus/lmcaudio.c b/sys/arch/arm32/mainbus/lmcaudio.c
index 4c05d0d0e1dd..6180697cbe3a 100644
--- a/sys/arch/arm32/mainbus/lmcaudio.c
+++ b/sys/arch/arm32/mainbus/lmcaudio.c
@@ -1,4 +1,4 @@
-/* $NetBSD: lmcaudio.c,v 1.6 1997/05/07 18:51:31 augustss Exp $ */
+/* $NetBSD: lmcaudio.c,v 1.7 1997/05/09 22:16:27 augustss Exp $ */
 
 /*
  * Copyright (c) 1996, Danny C Tsen.
@@ -329,17 +329,14 @@ lmcaudio_drain(addr)
  | Interface to the generic audio driver                                     |
  * ************************************************************************* */
 
-int    lmcaudio_query_encoding  __P((void *, struct audio_encoding *));
-int    lmcaudio_set_out_params  __P((void *, struct audio_params *));
-int    lmcaudio_set_in_params   __P((void *, struct audio_params *));
-int    lmcaudio_round_blocksize __P((void *, int));
+int    lmcaudio_query_encoding   __P((void *, struct audio_encoding *));
+int    lmcaudio_set_params       __P((void *, int, struct audio_params *, struct audio_params *));
+int    lmcaudio_round_blocksize  __P((void *, int));
 int    lmcaudio_set_out_port	 __P((void *, int));
 int    lmcaudio_get_out_port	 __P((void *));
 int    lmcaudio_set_in_port	 __P((void *, int));
 int    lmcaudio_get_in_port  	 __P((void *));
-int    lmcaudio_commit_settings __P((void *));
-void   lmcaudio_sw_encode	 __P((void *, int, u_char *, int));
-void   lmcaudio_sw_decode	 __P((void *, int, u_char *, int));
+int    lmcaudio_commit_settings  __P((void *));
 int    lmcaudio_start_output	 __P((void *, void *, int, void (*)(), void *));
 int    lmcaudio_start_input	 __P((void *, void *, int, void (*)(), void *));
 int    lmcaudio_halt_output	 __P((void *));
@@ -360,31 +357,21 @@ struct audio_device lmcaudio_device = {
 };
 
 int
-lmcaudio_set_in_params(addr, p)
+lmcaudio_set_params(addr, mode, p, q)
 	void *addr;
-	struct audio_params *p;
+	int mode;
+	struct audio_params *p, *q;
 {
-	struct vidcaudio_softc *sc = addr;
-
 	if (p->encoding != AUDIO_ENCODING_LINEAR_LE ||
 	    p->precision != 16 ||
 	    p->channels != 2)
 		return EINVAL;
-	return 0;
-}
 
-int
-lmcaudio_set_out_params(addr, p)
-	void *addr;
-	struct audio_params *p;
-{
-	struct vidcaudio_softc *sc = addr;
-
-	if (p->encoding != AUDIO_ENCODING_LINEAR_LE ||
-	    p->precision != 16 ||
-	    p->channels != 2)
-		return EINVAL;
-	lmcaudio_rate(p->sample_rate);
+	/* Update setting for the other mode. */
+	q->sample_rate = p->sample_rate;
+	q->encoding = p->encoding;
+	q->channels = p->channels;
+	q->precision = p->precision;
 	return 0;
 }
 
@@ -466,31 +453,6 @@ printf ( "DEBUG: committ_settings\n" );
 	return(0);
 }
 
-void
-lmcaudio_sw_encode(addr, e, p, cc)
-	void *addr;
-	int e;
-	u_char *p;
-	int cc;
-{
-#ifdef DEBUG
-	printf ( "DEBUG: sw_encode\n" );    
-#endif
-	return;
-}
-
-void
-lmcaudio_sw_decode(addr, e, p, cc)
-	void *addr;
-	int e;
-	u_char *p;
-	int cc;
-{
-#ifdef DEBUG
-	printf ( "DEBUG: sw_decode\n" );    
-#endif
-}
-
 #define ROUND(s)  ( ((int)s) & (~(NBPG-1)) )
 
 extern char *auzero_block;
@@ -644,16 +606,13 @@ struct audio_hw_if lmcaudio_hw_if = {
 	lmcaudio_close,
 	lmcaudio_drain,
 	lmcaudio_query_encoding,
-	lmcaudio_set_out_params,
-	lmcaudio_set_in_params,
+	lmcaudio_set_params,
 	lmcaudio_round_blocksize,
 	lmcaudio_set_out_port,
 	lmcaudio_get_out_port,
 	lmcaudio_set_in_port,
 	lmcaudio_get_in_port,
 	lmcaudio_commit_settings,
-	lmcaudio_sw_encode,
-	lmcaudio_sw_decode,
 	lmcaudio_start_output,
 	lmcaudio_start_input,
 	lmcaudio_halt_output,
diff --git a/sys/arch/arm32/mainbus/vidcaudio.c b/sys/arch/arm32/mainbus/vidcaudio.c
index 217a0704fbe3..0f8d888bb4be 100644
--- a/sys/arch/arm32/mainbus/vidcaudio.c
+++ b/sys/arch/arm32/mainbus/vidcaudio.c
@@ -1,4 +1,4 @@
-/* $NetBSD: vidcaudio.c,v 1.12 1997/05/07 18:51:33 augustss Exp $ */
+/* $NetBSD: vidcaudio.c,v 1.13 1997/05/09 22:16:28 augustss Exp $ */
 
 /*
  * Copyright (c) 1995 Melvin Tang-Richardson
@@ -293,15 +293,13 @@ vidcaudio_close(addr)
  * ************************************************************************* */
 
 int    vidcaudio_query_encoding  __P((void *, struct audio_encoding *));
-int    vidcaudio_set_params	 __P((void *, struct audio_params *));
+int    vidcaudio_set_params	 __P((void *, int, struct audio_params *, struct audio_params *));
 int    vidcaudio_round_blocksize __P((void *, int));
 int    vidcaudio_set_out_port	 __P((void *, int));
 int    vidcaudio_get_out_port	 __P((void *));
 int    vidcaudio_set_in_port	 __P((void *, int));
 int    vidcaudio_get_in_port  	 __P((void *));
 int    vidcaudio_commit_settings __P((void *));
-void   vidcaudio_sw_encode	 __P((void *, int, u_char *, int));
-void   vidcaudio_sw_decode	 __P((void *, int, u_char *, int));
 int    vidcaudio_start_output	 __P((void *, void *, int, void (*)(), void *));
 int    vidcaudio_start_input	 __P((void *, void *, int, void (*)(), void *));
 int    vidcaudio_halt_output	 __P((void *));
@@ -339,16 +337,21 @@ int vidcaudio_query_encoding ( void *addr, struct audio_encoding *fp )
 }
 
 int
-vidcaudio_set_params(addr, p)
+vidcaudio_set_params(addr, mode, p, q)
 	void *addr;
-	struct audio_params *p;
+	struct audio_params *p, *q;
 {
-	struct vidcaudio_softc *sc = addr;
-
 	if (p->encoding != AUDIO_ENCODING_ULAW ||
 	    p->channels != 8)
 		return EINVAL;
 	vidcaudio_rate(4 * p->sample_rate / (3 * 1024)); /* XXX probably wrong */
+	p->sw_code = 0;
+
+	/* Update setting for the other mode. */
+	q->sample_rate = p->sample_rate;
+	q->encoding = p->encoding;
+	q->channels = p->channels;
+	q->precision = p->precision;
 	return 0;
 }
 
@@ -394,21 +397,6 @@ printf ( "DEBUG: committ_settings\n" );
     return 0;
 }
 
-void vidcaudio_sw_encode ( void *addr, int e, u_char *p, int cc )
-{
-#ifdef DEBUG
-    printf ( "DEBUG: sw_encode\n" );    
-#endif
-    return;
-}
-
-void vidcaudio_sw_decode ( void *addr, int e, u_char *p, int cc )
-{
-#ifdef DEBUG
-    printf ( "DEBUG: sw_decode\n" );    
-#endif
-}
-
 #define ROUND(s)  ( ((int)s) & (~(NBPG-1)) )
 
 int vidcaudio_start_output ( void *addr, void *p, int cc,
@@ -525,15 +513,12 @@ struct audio_hw_if vidcaudio_hw_if = {
     NULL,
     vidcaudio_query_encoding,
     vidcaudio_set_params,
-    vidcaudio_set_params,
     vidcaudio_round_blocksize,
     vidcaudio_set_out_port,
     vidcaudio_get_out_port,
     vidcaudio_set_in_port,
     vidcaudio_get_in_port,
     vidcaudio_commit_settings,
-    vidcaudio_sw_encode,
-    vidcaudio_sw_decode,
     vidcaudio_start_output,
     vidcaudio_start_input,
     vidcaudio_halt_output,
diff --git a/sys/arch/arm32/vidc/lmcaudio.c b/sys/arch/arm32/vidc/lmcaudio.c
index 4c05d0d0e1dd..6180697cbe3a 100644
--- a/sys/arch/arm32/vidc/lmcaudio.c
+++ b/sys/arch/arm32/vidc/lmcaudio.c
@@ -1,4 +1,4 @@
-/* $NetBSD: lmcaudio.c,v 1.6 1997/05/07 18:51:31 augustss Exp $ */
+/* $NetBSD: lmcaudio.c,v 1.7 1997/05/09 22:16:27 augustss Exp $ */
 
 /*
  * Copyright (c) 1996, Danny C Tsen.
@@ -329,17 +329,14 @@ lmcaudio_drain(addr)
  | Interface to the generic audio driver                                     |
  * ************************************************************************* */
 
-int    lmcaudio_query_encoding  __P((void *, struct audio_encoding *));
-int    lmcaudio_set_out_params  __P((void *, struct audio_params *));
-int    lmcaudio_set_in_params   __P((void *, struct audio_params *));
-int    lmcaudio_round_blocksize __P((void *, int));
+int    lmcaudio_query_encoding   __P((void *, struct audio_encoding *));
+int    lmcaudio_set_params       __P((void *, int, struct audio_params *, struct audio_params *));
+int    lmcaudio_round_blocksize  __P((void *, int));
 int    lmcaudio_set_out_port	 __P((void *, int));
 int    lmcaudio_get_out_port	 __P((void *));
 int    lmcaudio_set_in_port	 __P((void *, int));
 int    lmcaudio_get_in_port  	 __P((void *));
-int    lmcaudio_commit_settings __P((void *));
-void   lmcaudio_sw_encode	 __P((void *, int, u_char *, int));
-void   lmcaudio_sw_decode	 __P((void *, int, u_char *, int));
+int    lmcaudio_commit_settings  __P((void *));
 int    lmcaudio_start_output	 __P((void *, void *, int, void (*)(), void *));
 int    lmcaudio_start_input	 __P((void *, void *, int, void (*)(), void *));
 int    lmcaudio_halt_output	 __P((void *));
@@ -360,31 +357,21 @@ struct audio_device lmcaudio_device = {
 };
 
 int
-lmcaudio_set_in_params(addr, p)
+lmcaudio_set_params(addr, mode, p, q)
 	void *addr;
-	struct audio_params *p;
+	int mode;
+	struct audio_params *p, *q;
 {
-	struct vidcaudio_softc *sc = addr;
-
 	if (p->encoding != AUDIO_ENCODING_LINEAR_LE ||
 	    p->precision != 16 ||
 	    p->channels != 2)
 		return EINVAL;
-	return 0;
-}
 
-int
-lmcaudio_set_out_params(addr, p)
-	void *addr;
-	struct audio_params *p;
-{
-	struct vidcaudio_softc *sc = addr;
-
-	if (p->encoding != AUDIO_ENCODING_LINEAR_LE ||
-	    p->precision != 16 ||
-	    p->channels != 2)
-		return EINVAL;
-	lmcaudio_rate(p->sample_rate);
+	/* Update setting for the other mode. */
+	q->sample_rate = p->sample_rate;
+	q->encoding = p->encoding;
+	q->channels = p->channels;
+	q->precision = p->precision;
 	return 0;
 }
 
@@ -466,31 +453,6 @@ printf ( "DEBUG: committ_settings\n" );
 	return(0);
 }
 
-void
-lmcaudio_sw_encode(addr, e, p, cc)
-	void *addr;
-	int e;
-	u_char *p;
-	int cc;
-{
-#ifdef DEBUG
-	printf ( "DEBUG: sw_encode\n" );    
-#endif
-	return;
-}
-
-void
-lmcaudio_sw_decode(addr, e, p, cc)
-	void *addr;
-	int e;
-	u_char *p;
-	int cc;
-{
-#ifdef DEBUG
-	printf ( "DEBUG: sw_decode\n" );    
-#endif
-}
-
 #define ROUND(s)  ( ((int)s) & (~(NBPG-1)) )
 
 extern char *auzero_block;
@@ -644,16 +606,13 @@ struct audio_hw_if lmcaudio_hw_if = {
 	lmcaudio_close,
 	lmcaudio_drain,
 	lmcaudio_query_encoding,
-	lmcaudio_set_out_params,
-	lmcaudio_set_in_params,
+	lmcaudio_set_params,
 	lmcaudio_round_blocksize,
 	lmcaudio_set_out_port,
 	lmcaudio_get_out_port,
 	lmcaudio_set_in_port,
 	lmcaudio_get_in_port,
 	lmcaudio_commit_settings,
-	lmcaudio_sw_encode,
-	lmcaudio_sw_decode,
 	lmcaudio_start_output,
 	lmcaudio_start_input,
 	lmcaudio_halt_output,
diff --git a/sys/arch/arm32/vidc/vidcaudio.c b/sys/arch/arm32/vidc/vidcaudio.c
index 217a0704fbe3..0f8d888bb4be 100644
--- a/sys/arch/arm32/vidc/vidcaudio.c
+++ b/sys/arch/arm32/vidc/vidcaudio.c
@@ -1,4 +1,4 @@
-/* $NetBSD: vidcaudio.c,v 1.12 1997/05/07 18:51:33 augustss Exp $ */
+/* $NetBSD: vidcaudio.c,v 1.13 1997/05/09 22:16:28 augustss Exp $ */
 
 /*
  * Copyright (c) 1995 Melvin Tang-Richardson
@@ -293,15 +293,13 @@ vidcaudio_close(addr)
  * ************************************************************************* */
 
 int    vidcaudio_query_encoding  __P((void *, struct audio_encoding *));
-int    vidcaudio_set_params	 __P((void *, struct audio_params *));
+int    vidcaudio_set_params	 __P((void *, int, struct audio_params *, struct audio_params *));
 int    vidcaudio_round_blocksize __P((void *, int));
 int    vidcaudio_set_out_port	 __P((void *, int));
 int    vidcaudio_get_out_port	 __P((void *));
 int    vidcaudio_set_in_port	 __P((void *, int));
 int    vidcaudio_get_in_port  	 __P((void *));
 int    vidcaudio_commit_settings __P((void *));
-void   vidcaudio_sw_encode	 __P((void *, int, u_char *, int));
-void   vidcaudio_sw_decode	 __P((void *, int, u_char *, int));
 int    vidcaudio_start_output	 __P((void *, void *, int, void (*)(), void *));
 int    vidcaudio_start_input	 __P((void *, void *, int, void (*)(), void *));
 int    vidcaudio_halt_output	 __P((void *));
@@ -339,16 +337,21 @@ int vidcaudio_query_encoding ( void *addr, struct audio_encoding *fp )
 }
 
 int
-vidcaudio_set_params(addr, p)
+vidcaudio_set_params(addr, mode, p, q)
 	void *addr;
-	struct audio_params *p;
+	struct audio_params *p, *q;
 {
-	struct vidcaudio_softc *sc = addr;
-
 	if (p->encoding != AUDIO_ENCODING_ULAW ||
 	    p->channels != 8)
 		return EINVAL;
 	vidcaudio_rate(4 * p->sample_rate / (3 * 1024)); /* XXX probably wrong */
+	p->sw_code = 0;
+
+	/* Update setting for the other mode. */
+	q->sample_rate = p->sample_rate;
+	q->encoding = p->encoding;
+	q->channels = p->channels;
+	q->precision = p->precision;
 	return 0;
 }
 
@@ -394,21 +397,6 @@ printf ( "DEBUG: committ_settings\n" );
     return 0;
 }
 
-void vidcaudio_sw_encode ( void *addr, int e, u_char *p, int cc )
-{
-#ifdef DEBUG
-    printf ( "DEBUG: sw_encode\n" );    
-#endif
-    return;
-}
-
-void vidcaudio_sw_decode ( void *addr, int e, u_char *p, int cc )
-{
-#ifdef DEBUG
-    printf ( "DEBUG: sw_decode\n" );    
-#endif
-}
-
 #define ROUND(s)  ( ((int)s) & (~(NBPG-1)) )
 
 int vidcaudio_start_output ( void *addr, void *p, int cc,
@@ -525,15 +513,12 @@ struct audio_hw_if vidcaudio_hw_if = {
     NULL,
     vidcaudio_query_encoding,
     vidcaudio_set_params,
-    vidcaudio_set_params,
     vidcaudio_round_blocksize,
     vidcaudio_set_out_port,
     vidcaudio_get_out_port,
     vidcaudio_set_in_port,
     vidcaudio_get_in_port,
     vidcaudio_commit_settings,
-    vidcaudio_sw_encode,
-    vidcaudio_sw_decode,
     vidcaudio_start_output,
     vidcaudio_start_input,
     vidcaudio_halt_output,
diff --git a/sys/arch/sparc/dev/am7930_sparc.c b/sys/arch/sparc/dev/am7930_sparc.c
index e1cb76075fac..29b9fae3e5be 100644
--- a/sys/arch/sparc/dev/am7930_sparc.c
+++ b/sys/arch/sparc/dev/am7930_sparc.c
@@ -1,4 +1,4 @@
-/*	$NetBSD: am7930_sparc.c,v 1.19 1997/05/07 18:51:35 augustss Exp $	*/
+/*	$NetBSD: am7930_sparc.c,v 1.20 1997/05/09 22:16:29 augustss Exp $	*/
 
 /*
  * Copyright (c) 1995 Rolf Grossmann
@@ -206,7 +206,7 @@ static const u_short ger_coeff[] = {
 int	amd7930_open __P((dev_t, int));
 void	amd7930_close __P((void *));
 int	amd7930_query_encoding __P((void *, struct audio_encoding *));
-int	amd7930_set_params __P((void *, struct audio_params *));
+int	amd7930_set_params __P((void *, int, struct audio_params *, struct audio_params *));
 int	amd7930_round_blocksize __P((void *, int));
 int	amd7930_set_out_port __P((void *, int));
 int	amd7930_get_out_port __P((void *));
@@ -234,15 +234,12 @@ struct audio_hw_if sa_hw_if = {
 	NULL,
 	amd7930_query_encoding,
 	amd7930_set_params,
-	amd7930_set_params,
 	amd7930_round_blocksize,
 	amd7930_set_out_port,
 	amd7930_get_out_port,
 	amd7930_set_in_port,
 	amd7930_get_in_port,
 	amd7930_commit_settings,
-	NULL,
-	NULL,
 	amd7930_start_output,
 	amd7930_start_input,
 	amd7930_halt_output,
@@ -398,17 +395,25 @@ amd7930_close(addr)
 }
 
 int
-amd7930_set_params(addr, p)
+amd7930_set_params(addr, mode, p, q)
 	void *addr;
-	struct audio_params *p;
+	int mode;
+	struct audio_params *p, *q;
 {
 	if (p->sample_rate < 7500 || p->sample_rate > 8500 ||
 	    p->encoding != AUDIO_ENCODING_ULAW ||
 	    p->precision != 8 ||
 	    p->channels != 1)
 		return EINVAL;
-	p->sample_rate = 8000;
-	return 0;	/* no other sampling rates supported by amd chip */
+	p->sample_rate = 8000;	/* no other sampling rates supported by amd chip */
+
+	/* Update setting for the other mode. */
+	q->sample_rate = p->sample_rate;
+	q->encoding = p->encoding;
+	q->channels = p->channels;
+	q->precision = p->precision;
+
+	return 0;
 }
 
 int
diff --git a/sys/arch/sparc/dev/amd7930.c b/sys/arch/sparc/dev/amd7930.c
index 25159ecb2ac1..7a3fdb48839f 100644
--- a/sys/arch/sparc/dev/amd7930.c
+++ b/sys/arch/sparc/dev/amd7930.c
@@ -1,4 +1,4 @@
-/*	$NetBSD: amd7930.c,v 1.19 1997/05/07 18:51:35 augustss Exp $	*/
+/*	$NetBSD: amd7930.c,v 1.20 1997/05/09 22:16:29 augustss Exp $	*/
 
 /*
  * Copyright (c) 1995 Rolf Grossmann
@@ -206,7 +206,7 @@ static const u_short ger_coeff[] = {
 int	amd7930_open __P((dev_t, int));
 void	amd7930_close __P((void *));
 int	amd7930_query_encoding __P((void *, struct audio_encoding *));
-int	amd7930_set_params __P((void *, struct audio_params *));
+int	amd7930_set_params __P((void *, int, struct audio_params *, struct audio_params *));
 int	amd7930_round_blocksize __P((void *, int));
 int	amd7930_set_out_port __P((void *, int));
 int	amd7930_get_out_port __P((void *));
@@ -234,15 +234,12 @@ struct audio_hw_if sa_hw_if = {
 	NULL,
 	amd7930_query_encoding,
 	amd7930_set_params,
-	amd7930_set_params,
 	amd7930_round_blocksize,
 	amd7930_set_out_port,
 	amd7930_get_out_port,
 	amd7930_set_in_port,
 	amd7930_get_in_port,
 	amd7930_commit_settings,
-	NULL,
-	NULL,
 	amd7930_start_output,
 	amd7930_start_input,
 	amd7930_halt_output,
@@ -398,17 +395,25 @@ amd7930_close(addr)
 }
 
 int
-amd7930_set_params(addr, p)
+amd7930_set_params(addr, mode, p, q)
 	void *addr;
-	struct audio_params *p;
+	int mode;
+	struct audio_params *p, *q;
 {
 	if (p->sample_rate < 7500 || p->sample_rate > 8500 ||
 	    p->encoding != AUDIO_ENCODING_ULAW ||
 	    p->precision != 8 ||
 	    p->channels != 1)
 		return EINVAL;
-	p->sample_rate = 8000;
-	return 0;	/* no other sampling rates supported by amd chip */
+	p->sample_rate = 8000;	/* no other sampling rates supported by amd chip */
+
+	/* Update setting for the other mode. */
+	q->sample_rate = p->sample_rate;
+	q->encoding = p->encoding;
+	q->channels = p->channels;
+	q->precision = p->precision;
+
+	return 0;
 }
 
 int
diff --git a/sys/dev/audio.c b/sys/dev/audio.c
index 9ce13f583a26..ab8ec05a7f66 100644
--- a/sys/dev/audio.c
+++ b/sys/dev/audio.c
@@ -1,4 +1,4 @@
-/*	$NetBSD: audio.c,v 1.44 1997/05/07 19:24:34 augustss Exp $	*/
+/*	$NetBSD: audio.c,v 1.45 1997/05/09 22:16:30 augustss Exp $	*/
 
 /*
  * Copyright (c) 1991-1993 Regents of the University of California.
@@ -160,7 +160,8 @@ int	audio_drain __P((struct audio_softc *));
 void	audio_clear __P((struct audio_softc *));
 
 /* The default audio mode: 8 kHz mono ulaw */
-static struct audio_params audio_default = { 8000, AUDIO_ENCODING_ULAW, 8, 1 };
+static struct audio_params audio_default = 
+	{ 8000, AUDIO_ENCODING_ULAW, 8, 1, 0 };
 
 #ifdef AUDIO_DEBUG
 void
@@ -213,8 +214,7 @@ audio_hardware_attach(hwp, hdlp)
 	if (hwp->open == 0 ||
 	    hwp->close == 0 ||
 	    hwp->query_encoding == 0 ||
-	    hwp->set_in_params == 0 ||
-	    hwp->set_out_params == 0 ||
+	    hwp->set_params == 0 ||
 	    hwp->round_blocksize == 0 ||
 	    hwp->set_out_port == 0 ||
 	    hwp->get_out_port == 0 ||
@@ -540,10 +540,12 @@ audio_open(dev, flags, ifmt, p)
 		sc->sc_rparams = audio_default;
 		sc->sc_pparams = audio_default;
 		/** XXX should we abort on error? */
-		error = hw->set_in_params(sc->hw_hdl, &sc->sc_rparams);
+		error = hw->set_params(sc->hw_hdl, AUMODE_RECORD,
+				       &sc->sc_rparams, &sc->sc_pparams);
 		if (error)
 			return (error);
-		error = hw->set_out_params(sc->hw_hdl, &sc->sc_pparams);
+		error = hw->set_params(sc->hw_hdl, AUMODE_PLAY,
+				       &sc->sc_pparams, &sc->sc_rparams);
 		if (error)
 			return (error);
 	}
@@ -786,9 +788,8 @@ audio_read(dev, uio, ioflag)
 				return (error);
 		}
 		hp = cb->hp;
-		if (hw->sw_decode)
-			hw->sw_decode(sc->hw_hdl, sc->sc_rparams.encoding,
-				      hp, blocksize);
+		if (sc->sc_rparams.sw_code)
+			sc->sc_rparams.sw_code(sc->hw_hdl, hp, blocksize);
 		error = uiomove(hp, blocksize, uio);
 		if (error)
 			break;
@@ -941,8 +942,6 @@ audio_alloc_auzero(sc, bs)
 	struct audio_softc *sc;
 	int bs;
 {
-	struct audio_hw_if *hw = sc->hw_if;
-
 	if (sc->auzero_block)
 		free(sc->auzero_block, M_DEVBUF);
 
@@ -953,9 +952,8 @@ audio_alloc_auzero(sc, bs)
 	}
 #endif
 	audio_fill_silence(&sc->sc_pparams, sc->auzero_block, bs);
-	if (hw->sw_encode)
-		hw->sw_encode(sc->hw_hdl, sc->sc_pparams.encoding,
-			      sc->auzero_block, bs);
+	if (sc->sc_pparams.sw_code)
+		sc->sc_pparams.sw_code(sc->hw_hdl, sc->auzero_block, bs);
 }
 
     
@@ -1003,10 +1001,9 @@ audio_write(dev, uio, ioflag)
 				cc = min(cb->fill, uio->uio_resid);
 				error = uiomove(cb->otp, cc, uio);
 				if (error == 0) {
-					if (hw->sw_encode)
-						hw->sw_encode(sc->hw_hdl,
-						    sc->sc_pparams.encoding, cb->otp,
-						    cc);
+					if (sc->sc_pparams.sw_code)
+						sc->sc_pparams.sw_code(sc->hw_hdl,
+						    cb->otp, cc);
 					cb->fill -= cc;
 					cb->otp += cc;
 				}
@@ -1104,9 +1101,8 @@ audio_write(dev, uio, ioflag)
 			break;
 		}		    
 
-		if (hw->sw_encode)
-			hw->sw_encode(sc->hw_hdl, sc->sc_pparams.encoding, 
-				      cb->tp, blocksize);
+		if (sc->sc_pparams.sw_code)
+			sc->sc_pparams.sw_code(sc->hw_hdl, cb->tp, blocksize);
 
 		/* wrap the ring buffer if at end */
 		s = splaudio();
@@ -1604,7 +1600,7 @@ audiosetinfo(sc, ai)
 		return error;
 	if (nr) {
 		audio_clear(sc);
-		error = hw->set_in_params(sc->hw_hdl, &rp);
+		error = hw->set_params(sc->hw_hdl, AUMODE_RECORD, &rp, &sc->sc_pparams);
 		if (error)
 			return (error);
 		sc->sc_rparams = rp;
@@ -1614,7 +1610,7 @@ audiosetinfo(sc, ai)
 		if (!cleared)
 			audio_clear(sc);
 		cleared = 1;
-		error = hw->set_out_params(sc->hw_hdl, &pp);
+		error = hw->set_params(sc->hw_hdl, AUMODE_PLAY, &pp, &sc->sc_rparams);
 		if (error)
 			return (error);
 		sc->sc_pparams = pp;
diff --git a/sys/dev/audio_if.h b/sys/dev/audio_if.h
index ec33ec38e7ee..00d6c7534cb2 100644
--- a/sys/dev/audio_if.h
+++ b/sys/dev/audio_if.h
@@ -1,4 +1,4 @@
-/*	$NetBSD: audio_if.h,v 1.11 1997/04/29 21:01:45 augustss Exp $	*/
+/*	$NetBSD: audio_if.h,v 1.12 1997/05/09 22:16:31 augustss Exp $	*/
 
 /*
  * Copyright (c) 1994 Havard Eidnes.
@@ -45,6 +45,8 @@ struct audio_params {
 	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));
 };
 
 struct audio_hw_if {
@@ -62,8 +64,7 @@ struct audio_hw_if {
 	 * The values in the params struct may be changed (e.g. rounding
 	 * to the nearest sample rate.)
 	 */
-        int	(*set_out_params)__P((void *, struct audio_params *));
-        int	(*set_in_params)__P((void *, struct audio_params *));
+        int	(*set_params)__P((void *, int, struct audio_params *, struct audio_params *));
   
 	/* Hardware may have some say in the blocksize to choose */
 	int	(*round_blocksize)__P((void *, int));
@@ -84,10 +85,6 @@ struct audio_hw_if {
 	 */
 	int	(*commit_settings)__P((void *));
 
-	/* Software en/decode functions, set if SW coding required by HW */
-	void	(*sw_encode)__P((void *, int, u_char *, int));
-	void	(*sw_decode)__P((void *, int, u_char *, int));
-
 	/* Start input/output routines. These usually control DMA. */
 	int	(*start_output)__P((void *, void *, int,
 				    void (*)(void *), void *));
diff --git a/sys/dev/ic/am7930.c b/sys/dev/ic/am7930.c
index 5ccf0e1e66be..16f820f8d455 100644
--- a/sys/dev/ic/am7930.c
+++ b/sys/dev/ic/am7930.c
@@ -1,4 +1,4 @@
-/*	$NetBSD: am7930.c,v 1.19 1997/05/07 18:51:35 augustss Exp $	*/
+/*	$NetBSD: am7930.c,v 1.20 1997/05/09 22:16:29 augustss Exp $	*/
 
 /*
  * Copyright (c) 1995 Rolf Grossmann
@@ -206,7 +206,7 @@ static const u_short ger_coeff[] = {
 int	amd7930_open __P((dev_t, int));
 void	amd7930_close __P((void *));
 int	amd7930_query_encoding __P((void *, struct audio_encoding *));
-int	amd7930_set_params __P((void *, struct audio_params *));
+int	amd7930_set_params __P((void *, int, struct audio_params *, struct audio_params *));
 int	amd7930_round_blocksize __P((void *, int));
 int	amd7930_set_out_port __P((void *, int));
 int	amd7930_get_out_port __P((void *));
@@ -234,15 +234,12 @@ struct audio_hw_if sa_hw_if = {
 	NULL,
 	amd7930_query_encoding,
 	amd7930_set_params,
-	amd7930_set_params,
 	amd7930_round_blocksize,
 	amd7930_set_out_port,
 	amd7930_get_out_port,
 	amd7930_set_in_port,
 	amd7930_get_in_port,
 	amd7930_commit_settings,
-	NULL,
-	NULL,
 	amd7930_start_output,
 	amd7930_start_input,
 	amd7930_halt_output,
@@ -398,17 +395,25 @@ amd7930_close(addr)
 }
 
 int
-amd7930_set_params(addr, p)
+amd7930_set_params(addr, mode, p, q)
 	void *addr;
-	struct audio_params *p;
+	int mode;
+	struct audio_params *p, *q;
 {
 	if (p->sample_rate < 7500 || p->sample_rate > 8500 ||
 	    p->encoding != AUDIO_ENCODING_ULAW ||
 	    p->precision != 8 ||
 	    p->channels != 1)
 		return EINVAL;
-	p->sample_rate = 8000;
-	return 0;	/* no other sampling rates supported by amd chip */
+	p->sample_rate = 8000;	/* no other sampling rates supported by amd chip */
+
+	/* Update setting for the other mode. */
+	q->sample_rate = p->sample_rate;
+	q->encoding = p->encoding;
+	q->channels = p->channels;
+	q->precision = p->precision;
+
+	return 0;
 }
 
 int
diff --git a/sys/dev/isa/ad1848.c b/sys/dev/isa/ad1848.c
index 075a3ea6e317..e9d21849e711 100644
--- a/sys/dev/isa/ad1848.c
+++ b/sys/dev/isa/ad1848.c
@@ -1,4 +1,4 @@
-/*	$NetBSD: ad1848.c,v 1.26 1997/05/07 18:51:41 augustss Exp $	*/
+/*	$NetBSD: ad1848.c,v 1.27 1997/05/09 22:16:34 augustss Exp $	*/
 
 /*
  * Copyright (c) 1994 John Brezak
@@ -85,6 +85,7 @@
 
 #include <sys/audioio.h>
 #include <dev/audio_if.h>
+#include <dev/mulaw.h>
 
 #include <dev/isa/isavar.h>
 #include <dev/isa/isadmavar.h>
@@ -150,13 +151,15 @@ static int ad1848_init_values[] = {
 void	ad1848_reset __P((struct ad1848_softc *));
 int	ad1848_set_speed __P((struct ad1848_softc *, u_long *));
 void	ad1848_mute_monitor __P((void *, int));
-int	ad1848_set_params __P((void *, struct audio_params *));
 
 static int ad_read __P((struct ad1848_softc *, int));
 static __inline void ad_write __P((struct ad1848_softc *, int, int));
 static void ad_set_MCE __P((struct ad1848_softc *, int));
 static void wait_for_calibration __P((struct ad1848_softc *));
 
+void ad1848_sign8(void *, u_char *, int);
+void ad1848_sign16(void *, u_char *, int);
+void ad1848_swap(void *, u_char *, int);
 
 static int
 ad_read(sc, reg)
@@ -492,7 +495,7 @@ ad1848_attach(sc)
     int i;
     struct ad1848_volume vol_mid = {220, 220};
     struct ad1848_volume vol_0   = {0, 0};
-    struct audio_params params;
+    struct audio_params params, xparams;
     
     sc->sc_locked = 0;
 
@@ -513,8 +516,8 @@ ad1848_attach(sc)
     params.precision = 8;
     params.channels = 1;
     params.encoding = AUDIO_ENCODING_ULAW;
-    (void) ad1848_set_in_params(sc, &params);
-    (void) ad1848_set_out_params(sc, &params);
+    (void) ad1848_set_params(sc, AUMODE_RECORD, &params, &xparams);
+    (void) ad1848_set_params(sc, AUMODE_PLAY,   &params, &xparams);
 
     /* Set default gains */
     (void) ad1848_set_rec_gain(sc, &vol_mid);
@@ -923,6 +926,8 @@ ad1848_query_encoding(addr, fp)
     void *addr;
     struct audio_encoding *fp;
 {
+    struct ad1848_softc *sc = addr;
+
     switch (fp->index) {
     case 0:
 	strcpy(fp->name, AudioEmulaw);
@@ -937,48 +942,117 @@ ad1848_query_encoding(addr, fp)
 	fp->flags = 0;
 	break;
     case 2:
-	strcpy(fp->name, AudioEadpcm);
-	fp->encoding = AUDIO_ENCODING_ADPCM;
-	fp->precision = 8;
-	fp->flags = 0;
-	break;
-    case 3:
 	strcpy(fp->name, AudioElinear_le);
 	fp->encoding = AUDIO_ENCODING_LINEAR_LE;
 	fp->precision = 16;
 	fp->flags = 0;
 	break;
-    case 4:
-	strcpy(fp->name, AudioElinear_be);
-	fp->encoding = AUDIO_ENCODING_LINEAR_BE;
-	fp->precision = 16;
-	fp->flags = 0;
-	break;
-    case 5:
+    case 3:
 	strcpy(fp->name, AudioEulinear);
 	fp->encoding = AUDIO_ENCODING_ULINEAR;
 	fp->precision = 8;
 	fp->flags = 0;
 	break;
+
+    case 4: /* only on CS4231 */
+	strcpy(fp->name, AudioElinear_be);
+	fp->encoding = AUDIO_ENCODING_LINEAR_BE;
+	fp->precision = 16;
+	fp->flags = sc->mode == 1;
+	break;
+
+    /* emulate some modes */
+    case 5:
+	strcpy(fp->name, AudioElinear);
+	fp->encoding = AUDIO_ENCODING_LINEAR;
+	fp->precision = 8;
+	fp->flags = 1;
+	break;
+    case 6:
+	strcpy(fp->name, AudioEulinear_le);
+	fp->encoding = AUDIO_ENCODING_ULINEAR_LE;
+	fp->precision = 16;
+	fp->flags = 1;
+	break;
+
+    case 7: /* only on CS4231 */
+	if (sc->mode == 1)
+	    return EINVAL;
+	strcpy(fp->name, AudioEadpcm);
+	fp->encoding = AUDIO_ENCODING_ADPCM;
+	fp->precision = 8;
+	fp->flags = 0;
+	break;
     default:
-	return(EINVAL);
+	return EINVAL;
 	/*NOTREACHED*/
     }
     return (0);
 }
 
-int
-ad1848_set_params(addr, p)
+void
+ad1848_sign8(addr, p, cc)
     void *addr;
-    struct audio_params *p;
+    u_char *p;
+    int cc;
 {
-    register struct ad1848_softc *sc = addr;
-    int error, bits;
+    change_sign8(p, cc);
+}
+
+void
+ad1848_sign16(addr, p, cc)
+    void *addr;
+    u_char *p;
+    int cc;
+{
+    change_sign16(p, cc);
+}
+
+void
+ad1848_swap(addr, p, cc)
+    void *addr;
+    u_char *p;
+    int cc;
+{
+    swap_bytes(p, cc);
+}
+
+int
+ad1848_set_params(addr, mode, p, q)
+    void *addr;
+    int mode;
+    struct audio_params *p, *q;
+{
+    struct ad1848_softc *sc = addr;
+    int error, bits, enc;
 
     DPRINTF(("ad1848_set_params: %d %d %d %d\n", 
 	     p->encoding, p->precision, p->channels, p->sample_rate));
 
-    switch (p->encoding) {
+    p->sw_code = 0;
+
+    enc = p->encoding;
+    switch (enc) {
+    case AUDIO_ENCODING_LINEAR_LE:
+	if (p->precision == 8) {
+	    enc = AUDIO_ENCODING_ULINEAR_LE;
+	    p->sw_code = ad1848_sign8;
+	}
+	break;
+    case AUDIO_ENCODING_LINEAR_BE:
+	if (p->precision == 16 && sc->mode == 1) {
+	    enc = AUDIO_ENCODING_LINEAR_LE;
+	    p->sw_code = ad1848_swap;
+	}
+	break;
+    case AUDIO_ENCODING_ULINEAR_LE:
+	if (p->precision == 16) {
+	    enc = AUDIO_ENCODING_LINEAR_LE;
+	    p->sw_code = ad1848_sign16;
+	}
+	break;
+    }
+    switch (enc) {
     case AUDIO_ENCODING_ULAW:
 	bits = FMT_ULAW >> 5;
 	break;
@@ -1001,7 +1075,6 @@ ad1848_set_params(addr, p)
 	    return EINVAL;
 	break;
     case AUDIO_ENCODING_ULINEAR_LE:
-    case AUDIO_ENCODING_ULINEAR_BE:
 	if (p->precision == 8)
 	    bits = FMT_PCM8 >> 5;
 	else
@@ -1023,28 +1096,16 @@ ad1848_set_params(addr, p)
     sc->precision = p->precision;
     sc->need_commit = 1;
 
+    /* Update setting for the other mode. */
+    q->sample_rate = p->sample_rate;
+    q->encoding = p->encoding;
+    q->channels = p->channels;
+    q->precision = p->precision;
+
     DPRINTF(("ad1848_set_params succeeded\n"));
     return (0);
 }
 
-/* We always set both play and record parameters */
-int
-ad1848_set_in_params(addr, p)
-    void *addr;
-    struct audio_params *p;
-{
-    return ad1848_set_params(addr, p);
-}
-
-/* We always set both play and record parameters */
-int
-ad1848_set_out_params(addr, p)
-    void *addr;
-    struct audio_params *p;
-{
-    return ad1848_set_params(addr, p);
-}
-  
 int
 ad1848_set_rec_port(sc, port)
     register struct ad1848_softc *sc;
diff --git a/sys/dev/isa/ad1848var.h b/sys/dev/isa/ad1848var.h
index 840eb7a29955..5fcaf986cb4a 100644
--- a/sys/dev/isa/ad1848var.h
+++ b/sys/dev/isa/ad1848var.h
@@ -1,4 +1,4 @@
-/*	$NetBSD: ad1848var.h,v 1.13 1997/04/29 21:01:35 augustss Exp $	*/
+/*	$NetBSD: ad1848var.h,v 1.14 1997/05/09 22:16:35 augustss Exp $	*/
 
 /*
  * Copyright (c) 1994 John Brezak
@@ -108,8 +108,7 @@ void	ad1848_close __P((void *));
 void	ad1848_forceintr __P((struct ad1848_softc *));
 
 int	ad1848_query_encoding __P((void *, struct audio_encoding *));
-int	ad1848_set_in_params __P((void *, struct audio_params *));
-int	ad1848_set_out_params __P((void *, struct audio_params *));
+int	ad1848_set_params __P((void *, int, struct audio_params *, struct audio_params *));
 
 int	ad1848_round_blocksize __P((void *, int));
 
diff --git a/sys/dev/isa/files.isa b/sys/dev/isa/files.isa
index 9f294b32db01..d62cac16015a 100644
--- a/sys/dev/isa/files.isa
+++ b/sys/dev/isa/files.isa
@@ -1,4 +1,4 @@
-#	$NetBSD: files.isa,v 1.28 1997/04/04 20:56:41 mycroft Exp $
+#	$NetBSD: files.isa,v 1.29 1997/05/09 22:16:35 augustss Exp $
 #
 # Config.new file and device description for machine-independent ISA code.
 # Included by ports that need it.  Requires that the SCSI files be
@@ -221,12 +221,12 @@ device	pss {[port = -1], [size = 0],
 	     [iomem = -1], [iosiz = 0],
 	     [irq = -1], [drq = -1]}
 attach	pss at isa
-device	sp: audio, isadma, ad1848
+device	sp: audio, isadma, ad1848, mulaw
 attach	sp at pss
 file	dev/isa/pss.c			pss needs-flag
 
 # Microsoft Windows Sound System
-device	wss: audio, isadma, ad1848
+device	wss: audio, isadma, ad1848, mulaw
 attach	wss at isa
 file	dev/isa/wss.c			wss needs-flag
 
diff --git a/sys/dev/isa/gus.c b/sys/dev/isa/gus.c
index 695f8a32e355..21f9d0dc5eca 100644
--- a/sys/dev/isa/gus.c
+++ b/sys/dev/isa/gus.c
@@ -1,4 +1,4 @@
-/*	$NetBSD: gus.c,v 1.24 1997/05/07 18:51:43 augustss Exp $	*/
+/*	$NetBSD: gus.c,v 1.25 1997/05/09 22:16:37 augustss Exp $	*/
 
 /*-
  * Copyright (c) 1996 The NetBSD Foundation, Inc.
@@ -357,10 +357,8 @@ int	gus_set_in_gain __P((caddr_t, u_int, u_char));
 int	gus_get_in_gain __P((caddr_t));
 int	gus_set_out_gain __P((caddr_t, u_int, u_char));
 int	gus_get_out_gain __P((caddr_t));
-int 	gus_set_in_params __P((void *, struct audio_params *));
-int 	gus_set_out_params __P((void *, struct audio_params *));
-int 	gusmax_set_in_params __P((void *, struct audio_params *));
-int 	gusmax_set_out_params __P((void *, struct audio_params *));
+int 	gus_set_params __P((void *, int, struct audio_params *, struct audio_params *));
+int 	gusmax_set_params __P((void *, int, struct audio_params *, struct audio_params *));
 int	gus_round_blocksize __P((void *, int));
 int	gus_set_out_port __P((void *, int));
 int	gus_get_out_port __P((void *));
@@ -391,8 +389,9 @@ int	gus_getdev __P((void *, struct audio_device *));
 int	gus_set_io_params __P((struct gus_softc *, struct audio_params *));
 
 STATIC void	gus_deinterleave __P((struct gus_softc *, void *, int));
-STATIC void	gus_expand __P((void *, int, u_char *, int));
-STATIC void	gusmax_expand __P((void *, int, u_char *, int));
+STATIC void	gus_expand __P((void *, u_char *, int));
+STATIC void	gusmax_expand __P((void *, u_char *, int));
+STATIC void	gus_compress __P((void *, u_char *, int));
 
 STATIC int	gus_mic_ctl __P((void *, int));
 STATIC int	gus_linein_ctl __P((void *, int));
@@ -605,8 +604,7 @@ struct audio_hw_if gus_hw_if = {
 
 	gus_query_encoding,
 
-	gus_set_out_params,
-	gus_set_in_params,
+	gus_set_params,
 
 	gus_round_blocksize,
 
@@ -617,9 +615,6 @@ struct audio_hw_if gus_hw_if = {
 
 	gus_commit_settings,
 
-	gus_expand,
-	mulaw_compress,
-
 	gus_dma_output,
 	gus_dma_input,
 	gus_halt_out_dma,
@@ -998,27 +993,26 @@ gusopen(dev, flags)
 }
 
 STATIC void
-gusmax_expand(hdl, encoding, buf, count)
+gusmax_expand(hdl, buf, count)
 	void *hdl;
-	int encoding;
 	u_char *buf;
 	int count;
 {
 	register struct ad1848_softc *ac = hdl;
 
-	gus_expand(ac->parent, encoding, buf, count);
+	gus_expand(ac->parent, buf, count);
 }
 
 STATIC void
-gus_expand(hdl, encoding, buf, count)
+gus_expand(hdl, buf, count)
 	void *hdl;
-	int encoding;
 	u_char *buf;
 	int count;
 {
 	struct gus_softc *sc = hdl;
 
-	mulaw_expand(NULL, encoding, buf, count);
+	if (sc->sc_encoding == AUDIO_ENCODING_ULAW)
+		mulaw_to_ulinear8(buf, count);
 	/*
 	 * If we need stereo deinterleaving, do it now.
 	 */
@@ -1026,6 +1020,16 @@ gus_expand(hdl, encoding, buf, count)
 		gus_deinterleave(sc, (void *)buf, count);
 }
 
+STATIC void
+gus_compress(hdl, buf, count)
+	void *hdl;
+	u_char *buf;
+	int count;
+{
+	
+	ulinear8_to_mulaw(buf, count);
+}
+
 STATIC void
 gus_deinterleave(sc, buf, size)
 	register struct gus_softc *sc;
@@ -2099,39 +2103,28 @@ gus_set_volume(sc, voice, volume)
  */
 
 int
-gusmax_set_in_params(addr, p)
+gusmax_set_params(addr, mode, p, q)
 	void *addr;
-	struct audio_params *p;
+	int mode;
+	struct audio_params *p, *q;
 {
 	register struct ad1848_softc *ac = addr;
 	register struct gus_softc *sc = ac->parent;
 	int error;
 
-	error = ad1848_set_in_params(ac, p);
+	error = ad1848_set_params(ac, mode, p, q);
 	if (error)
 		return error;
-	return gus_set_in_params(sc, p);
+	error = gus_set_params(sc, mode, p, q);
+	p->sw_code = mode == AUMODE_RECORD ? gus_compress : gusmax_expand;
+	return error;
 }
 
 int
-gusmax_set_out_params(addr, p)
+gus_set_params(addr, mode, p, q)
 	void *addr;
-	struct audio_params *p;
-{
-	register struct ad1848_softc *ac = addr;
-	register struct gus_softc *sc = ac->parent;
-	int error;
-
-	error = ad1848_set_in_params(ac, p);
-	if (error)
-		return error;
-	return gus_set_in_params(sc, p);
-}
-
-int
-gus_set_in_params(addr, p)
-	void *addr;
-	struct audio_params *p;
+	int mode;
+	struct audio_params *p, *q;
 {
 	register struct gus_softc *sc = addr;
 	int error;
@@ -2139,28 +2132,19 @@ gus_set_in_params(addr, p)
 	error = gus_set_io_params(sc, p);
 	if (error)
 		return error;
-	sc->sc_irate = p->sample_rate;
-	return 0;
-}
+	if (p->sample_rate > gus_max_frequency[sc->sc_voices - GUS_MIN_VOICES])
+		p->sample_rate = gus_max_frequency[sc->sc_voices - GUS_MIN_VOICES];
+	if (mode == AUMODE_RECORD)
+		sc->sc_irate = p->sample_rate;
+	else
+		sc->sc_orate = p->sample_rate;
 
-int
-gus_set_out_params(addr, p)
-	void *addr;
-	struct audio_params *p;
-{
-	register struct gus_softc *sc = addr;
-	int rate = p->sample_rate;
-	int error;
-	
-	error = gus_set_io_params(sc, p);
-	if (error)
-		return error;
-
-	if (rate > gus_max_frequency[sc->sc_voices - GUS_MIN_VOICES])
-		rate = gus_max_frequency[sc->sc_voices - GUS_MIN_VOICES];
-
-	p->sample_rate = sc->sc_orate = rate;
+	p->sw_code = mode == AUMODE_RECORD ? gus_compress : gus_expand;
 
+	/* Update setting for the other mode. */
+	q->encoding = p->encoding;
+	q->channels = p->channels;
+	q->precision = p->precision;
 	return 0;
 }
 
@@ -2803,8 +2787,7 @@ gus_init_cs4231(sc)
 
 			ad1848_query_encoding, /* query encoding */
 
-			gusmax_set_out_params,
-			gusmax_set_in_params,
+			gusmax_set_params,
 
 			gusmax_round_blocksize,
 
@@ -2815,9 +2798,6 @@ gus_init_cs4231(sc)
 
 			gusmax_commit_settings,
 
-			gusmax_expand,	/* XXX use codec */
-			mulaw_compress,
-
 			gusmax_dma_output,
 			gusmax_dma_input,
 			gusmax_halt_out_dma,
diff --git a/sys/dev/isa/pas.c b/sys/dev/isa/pas.c
index ff6adeeacbb9..27b961e604ad 100644
--- a/sys/dev/isa/pas.c
+++ b/sys/dev/isa/pas.c
@@ -1,4 +1,4 @@
-/*	$NetBSD: pas.c,v 1.25 1997/04/29 21:01:37 augustss Exp $	*/
+/*	$NetBSD: pas.c,v 1.26 1997/05/09 22:16:38 augustss Exp $	*/
 
 /*
  * Copyright (c) 1991-1993 Regents of the University of California.
@@ -53,7 +53,6 @@
 
 #include <sys/audioio.h>
 #include <dev/audio_if.h>
-#include <dev/mulaw.h>
 
 #include <dev/isa/isavar.h>
 #include <dev/isa/isadmavar.h>
@@ -106,16 +105,13 @@ struct audio_hw_if pas_hw_if = {
 	sbdsp_close,
 	NULL,
 	sbdsp_query_encoding,
-	sbdsp_set_out_params,
-	sbdsp_set_in_params,
+	sbdsp_set_params,
 	sbdsp_round_blocksize,
 	sbdsp_set_out_port,
 	sbdsp_get_out_port,
 	sbdsp_set_in_port,
 	sbdsp_get_in_port,
 	sbdsp_commit_settings,
-	mulaw_expand,
-	mulaw_compress,
 	sbdsp_dma_output,
 	sbdsp_dma_input,
 	sbdsp_haltdma,
diff --git a/sys/dev/isa/pss.c b/sys/dev/isa/pss.c
index 6a584861bc4e..38104601b7d8 100644
--- a/sys/dev/isa/pss.c
+++ b/sys/dev/isa/pss.c
@@ -1,4 +1,4 @@
-/*	$NetBSD: pss.c,v 1.26 1997/05/07 18:51:45 augustss Exp $	*/
+/*	$NetBSD: pss.c,v 1.27 1997/05/09 22:16:39 augustss Exp $	*/
 
 /*
  * Copyright (c) 1994 John Brezak
@@ -235,16 +235,13 @@ struct audio_hw_if pss_audio_if = {
 	ad1848_close,
 	NULL,
 	ad1848_query_encoding,
-	ad1848_set_out_params,
-	ad1848_set_in_params,
+	ad1848_set_params,
 	ad1848_round_blocksize,
 	pss_set_out_port,
 	pss_get_out_port,
 	pss_set_in_port,
 	pss_get_in_port,
 	ad1848_commit_settings,
-	NULL,
-	NULL,
 	ad1848_dma_output,
 	ad1848_dma_input,
 	ad1848_halt_out_dma,
diff --git a/sys/dev/isa/sb.c b/sys/dev/isa/sb.c
index 8e3aaa4ee01f..bd8c66b922f1 100644
--- a/sys/dev/isa/sb.c
+++ b/sys/dev/isa/sb.c
@@ -1,4 +1,4 @@
-/*	$NetBSD: sb.c,v 1.45 1997/04/29 21:01:39 augustss Exp $	*/
+/*	$NetBSD: sb.c,v 1.46 1997/05/09 22:16:40 augustss Exp $	*/
 
 /*
  * Copyright (c) 1991-1993 Regents of the University of California.
@@ -48,7 +48,6 @@
 
 #include <sys/audioio.h>
 #include <dev/audio_if.h>
-#include <dev/mulaw.h>
 
 #include <dev/isa/isavar.h>
 #include <dev/isa/isadmavar.h>
@@ -87,16 +86,13 @@ struct audio_hw_if sb_hw_if = {
 	sbdsp_close,
 	NULL,
 	sbdsp_query_encoding,
-	sbdsp_set_out_params,
-	sbdsp_set_in_params,
+	sbdsp_set_params,
 	sbdsp_round_blocksize,
 	sbdsp_set_out_port,
 	sbdsp_get_out_port,
 	sbdsp_set_in_port,
 	sbdsp_get_in_port,
 	sbdsp_commit_settings,
-	mulaw_expand,
-	mulaw_compress,
 	sbdsp_dma_output,
 	sbdsp_dma_input,
 	sbdsp_haltdma,
diff --git a/sys/dev/isa/sbdsp.c b/sys/dev/isa/sbdsp.c
index dd0e03c93d65..7a0ade5e0a8d 100644
--- a/sys/dev/isa/sbdsp.c
+++ b/sys/dev/isa/sbdsp.c
@@ -1,4 +1,4 @@
-/*	$NetBSD: sbdsp.c,v 1.44 1997/05/07 18:51:47 augustss Exp $	*/
+/*	$NetBSD: sbdsp.c,v 1.45 1997/05/09 22:16:41 augustss Exp $	*/
 
 /*
  * Copyright (c) 1991-1993 Regents of the University of California.
@@ -57,6 +57,7 @@
 
 #include <sys/audioio.h>
 #include <dev/audio_if.h>
+#include <dev/mulaw.h>
 
 #include <dev/isa/isavar.h>
 #include <dev/isa/isadmavar.h>
@@ -128,6 +129,8 @@ int	sbdsp16_setrate __P((struct sbdsp_softc *, int, int, int *));
 int	sbdsp_tctosr __P((struct sbdsp_softc *, int));
 int	sbdsp_set_timeconst __P((struct sbdsp_softc *, int));
 int	sbdsp_set_io_params __P((struct sbdsp_softc *, struct audio_params *));
+void	sbdsp_mulaw_expand __P((void *, u_char *, int));
+void	sbdsp_mulaw_compress __P((void *, u_char *, int));
 
 #ifdef AUDIO_DEBUG
 void sb_printsc __P((struct sbdsp_softc *));
@@ -397,62 +400,78 @@ sbdsp_set_io_params(sc, p)
 	return (0);
 }
 
+void
+sbdsp_mulaw_expand(v, p, cc)
+	void *v;
+	u_char *p;
+	int cc;
+{
+	mulaw_to_ulinear8(p, cc);
+}
+
+void
+sbdsp_mulaw_compress(v, p, cc)
+	void *v;
+	u_char *p;
+	int cc;
+{
+	ulinear8_to_mulaw(p, cc);
+}
+
 int
-sbdsp_set_in_params(addr, p)
+sbdsp_set_params(addr, mode, p, q)
 	void *addr;
-	struct audio_params *p;
+	int mode;
+	struct audio_params *p, *q;
 {
 	register struct sbdsp_softc *sc = addr;
-	int error;
+	int error, rate;
 
 	error = sbdsp_set_io_params(sc, p);
 	if (error)
 		return error;
 
-	if (ISSB16CLASS(sc))
-		error = sbdsp16_setrate(sc, p->sample_rate, SB_INPUT_RATE, &sc->sc_irate);
-	else
-		error = sbdsp_srtotc(sc, p->sample_rate, SB_INPUT_RATE, &sc->sc_itc, &sc->sc_imode);
+	if (mode == AUMODE_RECORD) {
+		if (ISSB16CLASS(sc)) {
+			error = sbdsp16_setrate(sc, p->sample_rate, SB_INPUT_RATE, 
+						&sc->sc_irate);
+			rate = sc->sc_irate;
+		} else {
+			error = sbdsp_srtotc(sc, p->sample_rate, SB_INPUT_RATE, 
+					     &sc->sc_itc, &sc->sc_imode);
+			rate = sbdsp_tctosr(sc, sc->sc_itc);
+		}
+	} else {
+		if (ISSB16CLASS(sc)) {
+			error = sbdsp16_setrate(sc, p->sample_rate, SB_OUTPUT_RATE, 
+						&sc->sc_orate);
+			rate = sc->sc_orate;
+		} else {
+			error = sbdsp_srtotc(sc, p->sample_rate, SB_OUTPUT_RATE, 
+					     &sc->sc_otc, &sc->sc_omode);
+			rate = sbdsp_tctosr(sc, sc->sc_otc);
+		}
+	}
 	if (error)
 		return error;
 
 	sc->sc_precision = p->precision;
 	sc->sc_channels = p->channels;
-	if (ISSB16CLASS(sc))
-		p->sample_rate = sc->sc_irate;
-	else
-		p->sample_rate = sbdsp_tctosr(sc, sc->sc_itc);
+
+	p->sample_rate = rate;
+	if (p->encoding == AUDIO_ENCODING_ULAW) {
+		p->sw_code = mode == AUMODE_PLAY ? 
+			sbdsp_mulaw_expand : sbdsp_mulaw_compress;
+	} else
+		p->sw_code = 0;
+
+	/* Update setting for the other mode. */
+	q->encoding = p->encoding;
+	q->channels = p->channels;
+	q->precision = p->precision;
 	return 0;
 }
 
-int
-sbdsp_set_out_params(addr, p)
-	void *addr;
-	struct audio_params *p;
-{
-	register struct sbdsp_softc *sc = addr;
-	int error;
-
-	error = sbdsp_set_io_params(sc, p);
-	if (error)
-		return error;
-
-	if (ISSB16CLASS(sc))
-		error = sbdsp16_setrate(sc, p->sample_rate, SB_OUTPUT_RATE, &sc->sc_orate);
-	else
-		error = sbdsp_srtotc(sc, p->sample_rate, SB_OUTPUT_RATE, &sc->sc_otc, &sc->sc_omode);
-	if (error)
-		return error;
-
-	sc->sc_precision = p->precision;
-	sc->sc_channels = p->channels;
-	if (ISSB16CLASS(sc))
-		p->sample_rate = sc->sc_orate;
-	else
-		p->sample_rate = sbdsp_tctosr(sc, sc->sc_otc);
-	return 0;
-}
-  
 int
 sbdsp_set_ifilter(addr, which)
 	void *addr;
diff --git a/sys/dev/isa/sbdspvar.h b/sys/dev/isa/sbdspvar.h
index 48b46562f4d6..abb0a4d933e2 100644
--- a/sys/dev/isa/sbdspvar.h
+++ b/sys/dev/isa/sbdspvar.h
@@ -1,4 +1,4 @@
-/*	$NetBSD: sbdspvar.h,v 1.19 1997/04/29 21:01:40 augustss Exp $	*/
+/*	$NetBSD: sbdspvar.h,v 1.20 1997/05/09 22:16:42 augustss Exp $	*/
 
 /*
  * Copyright (c) 1991-1993 Regents of the University of California.
@@ -162,8 +162,7 @@ int	sbdsp_get_out_gain __P((void *));
 int	sbdsp_set_monitor_gain __P((void *, u_int));
 int	sbdsp_get_monitor_gain __P((void *));
 int	sbdsp_query_encoding __P((void *, struct audio_encoding *));
-int	sbdsp_set_out_params __P((void *, struct audio_params *));
-int	sbdsp_set_in_params __P((void *, struct audio_params *));
+int	sbdsp_set_params __P((void *, int, struct audio_params *, struct audio_params *));
 int	sbdsp_set_ifilter __P((void *, int));
 int	sbdsp_get_ifilter __P((void *));
 int	sbdsp_round_blocksize __P((void *, int));
diff --git a/sys/dev/isa/wss.c b/sys/dev/isa/wss.c
index 88ae5ecc2b19..d9ebbcbf4f13 100644
--- a/sys/dev/isa/wss.c
+++ b/sys/dev/isa/wss.c
@@ -1,4 +1,4 @@
-/*	$NetBSD: wss.c,v 1.24 1997/05/07 18:51:49 augustss Exp $	*/
+/*	$NetBSD: wss.c,v 1.25 1997/05/09 22:16:43 augustss Exp $	*/
 
 /*
  * Copyright (c) 1994 John Brezak
@@ -163,16 +163,13 @@ struct audio_hw_if wss_hw_if = {
 	ad1848_close,
 	NULL,
 	ad1848_query_encoding,
-	ad1848_set_out_params,
-	ad1848_set_in_params,
+	ad1848_set_params,
 	ad1848_round_blocksize,
 	wss_set_out_port,
 	wss_get_out_port,
 	wss_set_in_port,
 	wss_get_in_port,
 	ad1848_commit_settings,
-	NULL,
-	NULL,
 	ad1848_dma_output,
 	ad1848_dma_input,
 	ad1848_halt_out_dma,
diff --git a/sys/dev/mulaw.c b/sys/dev/mulaw.c
index c20a173ef406..ee8edcfba0dc 100644
--- a/sys/dev/mulaw.c
+++ b/sys/dev/mulaw.c
@@ -34,6 +34,7 @@
 
 #include <sys/types.h>
 #include <sys/audioio.h>
+#include <machine/endian.h>
 #include <dev/mulaw.h>
 
 static u_char mulawtolin[256] = {
@@ -107,48 +108,67 @@ static u_char lintomulaw[256] = {
 };
 
 void
-mulaw_compress(hdl, e, p, cc)
-	void *hdl;
-	int e;
+mulaw_to_ulinear8(p, cc)
 	u_char *p;
 	int cc;
 {
-	u_char *tab;
-
-	switch (e) {
-	case AUDIO_ENCODING_ULAW:
-		tab = lintomulaw;
-		break;
-	default:
-		return;
-	}
-
 	while (--cc >= 0) {
-		*p = tab[*p];
+		*p = mulawtolin[*p];
 		++p;
 	}
 }
 
 void
-mulaw_expand(hdl, e, p, cc)
-	void *hdl;
-	int e;
+ulinear8_to_mulaw(p, cc)
 	u_char *p;
 	int cc;
 {
-	u_char *tab;
-
-	switch (e) {
-	case AUDIO_ENCODING_ULAW:
-		tab = mulawtolin;
-		break;
-	default:
-		return;
-	}
-	
 	while (--cc >= 0) {
-		*p = tab[*p];
+		*p = lintomulaw[*p];
 		++p;
 	}
 }
 
+void
+change_sign8(p, cc)
+	u_char *p;
+	int cc;
+{
+	while (--cc >= 0) {
+		*p = *p ^ 0x80;
+		++p;
+	}
+}
+
+void
+change_sign16(p, cc)
+	u_char *p;
+	int cc;
+{
+#if BYTE_ORDER == LITTLE_ENDIAN
+	while ((cc -=2) >= 0) {
+		p[1] = p[1] ^ 0x80;
+		p += 2;
+	}
+#else
+	while ((cc -=2) >= 0) {
+		*p = *p ^ 0x80;
+		p += 2;
+	}
+#endif
+}
+
+void
+swap_bytes(p, cc)
+	u_char *p;
+	int cc;
+{
+	u_char t;
+
+	while ((cc -=2) >= 0) {
+		t = p[0];
+		p[0] = p[1];
+		p[1] = t;
+		p += 2;
+	}
+}
diff --git a/sys/dev/mulaw.h b/sys/dev/mulaw.h
index ec93ea3d17b3..21c848630554 100644
--- a/sys/dev/mulaw.h
+++ b/sys/dev/mulaw.h
@@ -1,4 +1,4 @@
-/*	$NetBSD: mulaw.h,v 1.2 1996/02/27 22:29:42 jtc Exp $	*/
+/*	$NetBSD: mulaw.h,v 1.3 1997/05/09 22:16:33 augustss Exp $	*/
 
 /*-
  * Copyright (c) 1996 The NetBSD Foundation, Inc.
@@ -36,8 +36,11 @@
  * POSSIBILITY OF SUCH DAMAGE.
  */
 
-/*
- * Convert 8-bit mu-law to/from 8 bit unsigned linear (PCM8)
- */
-extern void mulaw_compress __P((void *hw_hdl, int encoding, unsigned char *buf, int cnt));
-extern void mulaw_expand __P((void *hw_hdl, int encoding, unsigned char *buf, int cnt));
+/* Convert 8-bit mu-law to/from 8 bit unsigned linear. */
+extern void mulaw_to_ulinear8 __P((u_char *buf, int cnt));
+extern void ulinear8_to_mulaw __P((u_char *buf, int cnt));
+/* Convert between signed and unsigned. */
+extern void change_sign8 __P((u_char *buf, int cnt));
+extern void change_sign16 __P((u_char *buf, int cnt));
+/* Convert between little and big endian. */
+extern void swap_bytes __P((u_char *buf, int cnt));