CVS commit: [isaki-audio2] src/sys/dev/isa
Module Name:src
Committed By: isaki
Date: Sun May 5 01:57:06 UTC 2019
Modified Files:
src/sys/dev/isa [isaki-audio2]: sbdsp.c
Log Message:
Fix typos (made in this branch).
To generate a diff of this commit:
cvs rdiff -u -r1.139.2.3 -r1.139.2.4 src/sys/dev/isa/sbdsp.c
Please note that diffs are not public domain; they are subject to the
copyright notices on the relevant files.
Modified files:
Index: src/sys/dev/isa/sbdsp.c
diff -u src/sys/dev/isa/sbdsp.c:1.139.2.3 src/sys/dev/isa/sbdsp.c:1.139.2.4
--- src/sys/dev/isa/sbdsp.c:1.139.2.3 Sat May 4 07:20:10 2019
+++ src/sys/dev/isa/sbdsp.c Sun May 5 01:57:06 2019
@@ -1,4 +1,4 @@
-/* $NetBSD: sbdsp.c,v 1.139.2.3 2019/05/04 07:20:10 isaki Exp $ */
+/* $NetBSD: sbdsp.c,v 1.139.2.4 2019/05/05 01:57:06 isaki Exp $ */
/*-
* Copyright (c) 1999, 2008 The NetBSD Foundation, Inc.
@@ -74,7 +74,7 @@
*/
#include
-__KERNEL_RCSID(0, "$NetBSD: sbdsp.c,v 1.139.2.3 2019/05/04 07:20:10 isaki Exp $");
+__KERNEL_RCSID(0, "$NetBSD: sbdsp.c,v 1.139.2.4 2019/05/05 01:57:06 isaki Exp $");
#include "midi.h"
#include "mpu.h"
@@ -462,7 +462,7 @@ sbdsp_attach(struct sbdsp_softc *sc)
}
}
- /* Construct sc_format from model */
+ /* Construct sc_formats from model */
sbdsp_init_format(sc);
if (sc->sc_nformats == 0) {
aprint_error_dev(sc->sc_dev,
@@ -597,7 +597,7 @@ sbdsp_init_format(struct sbdsp_softc *sc
idx = (m->precision / 16) * 2 + (m->channels - 1);
d = [idx];
if (d->mode == 0) {
-/* The first elements of this room */
+/* The first element of this room */
*d = tmp;
continue;
}
CVS commit: [isaki-audio2] src/sys/dev/isa
Module Name:src
Committed By: isaki
Date: Fri May 3 03:00:33 UTC 2019
Modified Files:
src/sys/dev/isa [isaki-audio2]: pas.c sb.c sbdsp.c sbdspvar.h
Log Message:
Adapt sbdsp to audio2.
- Use new query_format/set_format interfaces.
The formats are created from sb[pr]modes tables.
- Drop INDEPENDENT property for models proir to SB_16.
To generate a diff of this commit:
cvs rdiff -u -r1.71 -r1.71.2.1 src/sys/dev/isa/pas.c
cvs rdiff -u -r1.90 -r1.90.2.1 src/sys/dev/isa/sb.c
cvs rdiff -u -r1.139.2.1 -r1.139.2.2 src/sys/dev/isa/sbdsp.c
cvs rdiff -u -r1.61 -r1.61.54.1 src/sys/dev/isa/sbdspvar.h
Please note that diffs are not public domain; they are subject to the
copyright notices on the relevant files.
Modified files:
Index: src/sys/dev/isa/pas.c
diff -u src/sys/dev/isa/pas.c:1.71 src/sys/dev/isa/pas.c:1.71.2.1
--- src/sys/dev/isa/pas.c:1.71 Sat Mar 16 12:09:58 2019
+++ src/sys/dev/isa/pas.c Fri May 3 03:00:33 2019
@@ -1,4 +1,4 @@
-/* $NetBSD: pas.c,v 1.71 2019/03/16 12:09:58 isaki Exp $ */
+/* $NetBSD: pas.c,v 1.71.2.1 2019/05/03 03:00:33 isaki Exp $ */
/*
* Copyright (c) 1991-1993 Regents of the University of California.
@@ -57,7 +57,7 @@
#include
-__KERNEL_RCSID(0, "$NetBSD: pas.c,v 1.71 2019/03/16 12:09:58 isaki Exp $");
+__KERNEL_RCSID(0, "$NetBSD: pas.c,v 1.71.2.1 2019/05/03 03:00:33 isaki Exp $");
#include
#include
@@ -125,8 +125,8 @@ void pasconf(int, int, int, int);
const struct audio_hw_if pas_hw_if = {
.open = sbdsp_open,
.close = sbdsp_close,
- .query_encoding = sbdsp_query_encoding,
- .set_params = sbdsp_set_params,
+ .query_format = sbdsp_query_format,
+ .set_format = sbdsp_set_format,
.round_blocksize = sbdsp_round_blocksize,
.halt_output = sbdsp_halt_output,
.halt_input = sbdsp_halt_input,
@@ -138,7 +138,6 @@ const struct audio_hw_if pas_hw_if = {
.allocm = sb_malloc,
.freem = sb_free,
.round_buffersize = sb_round_buffersize,
- .mappage = sb_mappage,
.get_props = sbdsp_get_props,
.trigger_output = sbdsp_trigger_output,
.trigger_input = sbdsp_trigger_input,
Index: src/sys/dev/isa/sb.c
diff -u src/sys/dev/isa/sb.c:1.90 src/sys/dev/isa/sb.c:1.90.2.1
--- src/sys/dev/isa/sb.c:1.90 Sat Mar 16 12:09:58 2019
+++ src/sys/dev/isa/sb.c Fri May 3 03:00:33 2019
@@ -1,4 +1,4 @@
-/* $NetBSD: sb.c,v 1.90 2019/03/16 12:09:58 isaki Exp $ */
+/* $NetBSD: sb.c,v 1.90.2.1 2019/05/03 03:00:33 isaki Exp $ */
/*
* Copyright (c) 1991-1993 Regents of the University of California.
@@ -35,7 +35,7 @@
*/
#include
-__KERNEL_RCSID(0, "$NetBSD: sb.c,v 1.90 2019/03/16 12:09:58 isaki Exp $");
+__KERNEL_RCSID(0, "$NetBSD: sb.c,v 1.90.2.1 2019/05/03 03:00:33 isaki Exp $");
#include "midi.h"
@@ -82,8 +82,8 @@ int sb_getdev(void *, struct audio_devic
const struct audio_hw_if sb_hw_if = {
.open = sbdsp_open,
.close = sbdsp_close,
- .query_encoding = sbdsp_query_encoding,
- .set_params = sbdsp_set_params,
+ .query_format = sbdsp_query_format,
+ .set_format = sbdsp_set_format,
.round_blocksize = sbdsp_round_blocksize,
.halt_output = sbdsp_halt_output,
.halt_input = sbdsp_halt_input,
@@ -95,7 +95,6 @@ const struct audio_hw_if sb_hw_if = {
.allocm = sb_malloc,
.freem = sb_free,
.round_buffersize = sb_round_buffersize,
- .mappage = sb_mappage,
.get_props = sbdsp_get_props,
.trigger_output = sbdsp_trigger_output,
.trigger_input = sbdsp_trigger_input,
Index: src/sys/dev/isa/sbdsp.c
diff -u src/sys/dev/isa/sbdsp.c:1.139.2.1 src/sys/dev/isa/sbdsp.c:1.139.2.2
--- src/sys/dev/isa/sbdsp.c:1.139.2.1 Sun Apr 21 06:17:02 2019
+++ src/sys/dev/isa/sbdsp.c Fri May 3 03:00:33 2019
@@ -1,4 +1,4 @@
-/* $NetBSD: sbdsp.c,v 1.139.2.1 2019/04/21 06:17:02 isaki Exp $ */
+/* $NetBSD: sbdsp.c,v 1.139.2.2 2019/05/03 03:00:33 isaki Exp $ */
/*-
* Copyright (c) 1999, 2008 The NetBSD Foundation, Inc.
@@ -74,7 +74,7 @@
*/
#include
-__KERNEL_RCSID(0, "$NetBSD: sbdsp.c,v 1.139.2.1 2019/04/21 06:17:02 isaki Exp $");
+__KERNEL_RCSID(0, "$NetBSD: sbdsp.c,v 1.139.2.2 2019/05/03 03:00:33 isaki Exp $");
#include "midi.h"
#include "mpu.h"
@@ -95,6 +95,7 @@ __KERNEL_RCSID(0, "$NetBSD: sbdsp.c,v 1.
#include
#include
+#include
#include
#include
@@ -206,8 +207,34 @@ static struct sbmode sbrmodes[] = {
{ .model = -1 }
};
+/*
+ * We actually can specify any value within the frequency range defined
+ * above. But according to definition of SB_RATE_TO_TC macro, only some
+ * of them are dividable (it's preferable, not mandatory). There are 9
+ * values in the range that satisfy this condition but it's too much.
+ */
+static const int sbdsp_rates[] = {
+ 4000,
+ /* 5000, */
+ /* 6250, */
+ /* 1, */
+ 12500,
+ /* 15625, */
+ 2,
+ /* 25000, */
+ 31250,
+};
+
void sbversion(struct sbdsp_softc *);
void sbdsp_jazz16_probe(struct sbdsp_softc *);
+void sbdsp_sbmode2format(struct audio_format *, const struct sbmode *, int);
+int sbdsp_set_format16(struct sbdsp_softc *, int,
+ const
CVS commit: [isaki-audio2] src/sys/dev/isa
Module Name:src
Committed By: isaki
Date: Thu Apr 25 14:00:20 UTC 2019
Modified Files:
src/sys/dev/isa [isaki-audio2]: ess.c
Log Message:
Adapt to audio2.
- Drop INDEPENDENT property. Both play and rec share the sample rate.
To generate a diff of this commit:
cvs rdiff -u -r1.84.2.1 -r1.84.2.2 src/sys/dev/isa/ess.c
Please note that diffs are not public domain; they are subject to the
copyright notices on the relevant files.
Modified files:
Index: src/sys/dev/isa/ess.c
diff -u src/sys/dev/isa/ess.c:1.84.2.1 src/sys/dev/isa/ess.c:1.84.2.2
--- src/sys/dev/isa/ess.c:1.84.2.1 Sun Apr 21 05:11:22 2019
+++ src/sys/dev/isa/ess.c Thu Apr 25 14:00:20 2019
@@ -1,4 +1,4 @@
-/* $NetBSD: ess.c,v 1.84.2.1 2019/04/21 05:11:22 isaki Exp $ */
+/* $NetBSD: ess.c,v 1.84.2.2 2019/04/25 14:00:20 isaki Exp $ */
/*
* Copyright 1997
@@ -66,7 +66,7 @@
*/
#include
-__KERNEL_RCSID(0, "$NetBSD: ess.c,v 1.84.2.1 2019/04/21 05:11:22 isaki Exp $");
+__KERNEL_RCSID(0, "$NetBSD: ess.c,v 1.84.2.2 2019/04/25 14:00:20 isaki Exp $");
#include
#include
@@ -83,8 +83,6 @@ __KERNEL_RCSID(0, "$NetBSD: ess.c,v 1.84
#include
#include
-#include
-#include
#include
#include
@@ -115,15 +113,14 @@ unsigned uuu;
int ess_setup_sc(struct ess_softc *, int);
-int ess_open(void *, int);
void ess_close(void *);
int ess_getdev(void *, struct audio_device *);
-int ess_drain(void *);
-int ess_query_encoding(void *, struct audio_encoding *);
+int ess_query_format(void *, audio_format_query_t *);
-int ess_set_params(void *, int, int, audio_params_t *,
- audio_params_t *, stream_filter_list_t *, stream_filter_list_t *);
+int ess_set_format(void *, int,
+ const audio_params_t *, const audio_params_t *,
+ audio_filter_reg_t *, audio_filter_reg_t *);
int ess_round_blocksize(void *, int, int, const audio_params_t *);
@@ -215,11 +212,9 @@ struct audio_device ess_device = {
*/
const struct audio_hw_if ess_1788_hw_if = {
- .open = ess_open,
.close = ess_close,
- .drain = ess_drain,
- .query_encoding = ess_query_encoding,
- .set_params = ess_set_params,
+ .query_format = ess_query_format,
+ .set_format = ess_set_format,
.round_blocksize = ess_round_blocksize,
.halt_output = ess_audio1_halt,
.halt_input = ess_audio1_halt,
@@ -231,7 +226,6 @@ const struct audio_hw_if ess_1788_hw_if
.allocm = ess_malloc,
.freem = ess_free,
.round_buffersize = ess_round_buffersize,
- .mappage = ess_mappage,
.get_props = ess_1788_get_props,
.trigger_output = ess_audio1_trigger_output,
.trigger_input = ess_audio1_trigger_input,
@@ -239,11 +233,9 @@ const struct audio_hw_if ess_1788_hw_if
};
const struct audio_hw_if ess_1888_hw_if = {
- .open = ess_open,
.close = ess_close,
- .drain = ess_drain,
- .query_encoding = ess_query_encoding,
- .set_params = ess_set_params,
+ .query_format = ess_query_format,
+ .set_format = ess_set_format,
.round_blocksize = ess_round_blocksize,
.halt_output = ess_audio2_halt,
.halt_input = ess_audio1_halt,
@@ -255,35 +247,25 @@ const struct audio_hw_if ess_1888_hw_if
.allocm = ess_malloc,
.freem = ess_free,
.round_buffersize = ess_round_buffersize,
- .mappage = ess_mappage,
.get_props = ess_1888_get_props,
.trigger_output = ess_audio2_trigger_output,
.trigger_input = ess_audio1_trigger_input,
.get_locks = ess_get_locks,
};
-#define ESS_NFORMATS 8
-#define ESS_FORMAT(enc, prec, ch, chmask) \
- { \
- .mode = AUMODE_PLAY | AUMODE_RECORD, \
- .encoding = (enc), \
- .validbits = (prec), \
- .precision = (prec), \
- .channels = (ch), \
- .channel_mask = (chmask), \
- .frequency_type = 0, \
- .frequency = { ESS_MINRATE, ESS_MAXRATE }, \
- }
-static const struct audio_format ess_formats[ESS_NFORMATS] = {
- ESS_FORMAT(AUDIO_ENCODING_SLINEAR_LE, 16, 2, AUFMT_STEREO),
- ESS_FORMAT(AUDIO_ENCODING_SLINEAR_LE, 16, 1, AUFMT_STEREO),
- ESS_FORMAT(AUDIO_ENCODING_ULINEAR_LE, 16, 2, AUFMT_STEREO),
- ESS_FORMAT(AUDIO_ENCODING_ULINEAR_LE, 16, 1, AUFMT_STEREO),
- ESS_FORMAT(AUDIO_ENCODING_ULINEAR_LE, 8, 2, AUFMT_STEREO),
- ESS_FORMAT(AUDIO_ENCODING_ULINEAR_LE, 8, 1, AUFMT_STEREO),
- ESS_FORMAT(AUDIO_ENCODING_SLINEAR_LE, 8, 2, AUFMT_STEREO),
- ESS_FORMAT(AUDIO_ENCODING_SLINEAR_LE, 8, 1, AUFMT_STEREO),
+static const struct audio_format ess_formats[] = {
+ {
+ .mode = AUMODE_PLAY | AUMODE_RECORD,
+ .encoding = AUDIO_ENCODING_SLINEAR_LE,
+ .validbits = 16,
+ .precision = 16,
+ .channels = 2,
+ .channel_mask = AUFMT_STEREO,
+ .frequency_type = 0,
+ .frequency = { ESS_MINRATE, ESS_MAXRATE },
+ },
};
+#define ESS_NFORMATS __arraycount(ess_formats)
#ifdef AUDIO_DEBUG
void ess_printsc(struct ess_softc *);
@@ -1097,13 +1079,6 @@ skip:
* Various routines to interface to higher level audio driver
*/
-int
-ess_open(void *addr, int flags)
-{
-
- return 0;
-}
-
void
ess_close(void *addr)
{
@@ -1118,27 +1093,6 @@ ess_close(void *addr)
DPRINTF(("ess_close: closed\n"));
}
-/*
- * Wait
CVS commit: [isaki-audio2] src/sys/dev/isa
Module Name:src
Committed By: isaki
Date: Sun Apr 21 06:17:03 UTC 2019
Modified Files:
src/sys/dev/isa [isaki-audio2]: sbdsp.c
Log Message:
Not yet fully adapted to audio2 but make it compilable.
- Drop FULLDUPLEX property. It is done in a tricky way.
To generate a diff of this commit:
cvs rdiff -u -r1.139 -r1.139.2.1 src/sys/dev/isa/sbdsp.c
Please note that diffs are not public domain; they are subject to the
copyright notices on the relevant files.
Modified files:
Index: src/sys/dev/isa/sbdsp.c
diff -u src/sys/dev/isa/sbdsp.c:1.139 src/sys/dev/isa/sbdsp.c:1.139.2.1
--- src/sys/dev/isa/sbdsp.c:1.139 Sun Feb 3 03:19:27 2019
+++ src/sys/dev/isa/sbdsp.c Sun Apr 21 06:17:02 2019
@@ -1,4 +1,4 @@
-/* $NetBSD: sbdsp.c,v 1.139 2019/02/03 03:19:27 mrg Exp $ */
+/* $NetBSD: sbdsp.c,v 1.139.2.1 2019/04/21 06:17:02 isaki Exp $ */
/*-
* Copyright (c) 1999, 2008 The NetBSD Foundation, Inc.
@@ -74,7 +74,7 @@
*/
#include
-__KERNEL_RCSID(0, "$NetBSD: sbdsp.c,v 1.139 2019/02/03 03:19:27 mrg Exp $");
+__KERNEL_RCSID(0, "$NetBSD: sbdsp.c,v 1.139.2.1 2019/04/21 06:17:02 isaki Exp $");
#include "midi.h"
#include "mpu.h"
@@ -96,8 +96,6 @@ __KERNEL_RCSID(0, "$NetBSD: sbdsp.c,v 1.
#include
#include
#include
-#include
-#include
#include
#include
@@ -412,9 +410,8 @@ sbdsp_attach(struct sbdsp_softc *sc)
SBVER_MAJOR(sc->sc_version), SBVER_MINOR(sc->sc_version),
sc->sc_model == SB_JAZZ ? ": " : "");
- sc->sc_fullduplex = ISSB16CLASS(sc) &&
- sc->sc_drq8 != -1 && sc->sc_drq16 != -1 &&
- sc->sc_drq8 != sc->sc_drq16;
+ /* XXX It's not true full duplex. */
+ sc->sc_fullduplex = 0;
if (sc->sc_drq8 != -1) {
sc->sc_drq8_maxsize = isa_dmamaxsize(sc->sc_ic,
@@ -574,12 +571,10 @@ sbdsp_set_params(
struct sbdsp_softc *sc;
struct sbmode *m;
u_int rate, tc, bmode;
- stream_filter_factory_t *swcode;
int model;
int chan;
struct audio_params *p;
- audio_params_t hw;
- stream_filter_list_t *fil;
+ audio_params_t hw __unused;
int mode;
sc = addr;
@@ -627,55 +622,17 @@ sbdsp_set_params(
if (m->model == -1)
return EINVAL;
rate = p->sample_rate;
- swcode = NULL;
- fil = mode == AUMODE_PLAY ? pfil : rfil;
hw = *p;
tc = 1;
bmode = -1;
if (model == SB_16) {
switch (p->encoding) {
- case AUDIO_ENCODING_SLINEAR_BE:
-if (p->precision == 16) {
- hw.encoding = AUDIO_ENCODING_SLINEAR_LE;
- swcode = swap_bytes;
-}
/* FALLTHROUGH */
case AUDIO_ENCODING_SLINEAR:
case AUDIO_ENCODING_SLINEAR_LE:
bmode = SB_BMODE_SIGNED;
break;
- case AUDIO_ENCODING_ULINEAR_BE:
-if (p->precision == 16) {
- hw.encoding = AUDIO_ENCODING_ULINEAR_LE;
- swcode = swap_bytes;
-}
-/* FALLTHROUGH */
- case AUDIO_ENCODING_ULINEAR_LE:
-bmode = SB_BMODE_UNSIGNED;
-break;
-
- case AUDIO_ENCODING_ULAW:
-hw.encoding = AUDIO_ENCODING_ULINEAR_LE;
-if (mode == AUMODE_PLAY) {
- hw.precision = hw.validbits = 16;
- swcode = mulaw_to_linear16;
- m = [PLAY16];
-} else
- swcode = linear8_to_mulaw;
-bmode = SB_BMODE_UNSIGNED;
-break;
-
- case AUDIO_ENCODING_ALAW:
-hw.encoding = AUDIO_ENCODING_ULINEAR_LE;
-if (mode == AUMODE_PLAY) {
- hw.precision = hw.validbits = 16;
- swcode = alaw_to_linear16;
- m = [PLAY16];
-} else
- swcode = linear8_to_alaw;
-bmode = SB_BMODE_UNSIGNED;
-break;
default:
return EINVAL;
}
@@ -686,28 +643,6 @@ sbdsp_set_params(
case AUDIO_ENCODING_SLINEAR:
case AUDIO_ENCODING_SLINEAR_LE:
break;
- case AUDIO_ENCODING_ULINEAR_LE:
-hw.encoding = AUDIO_ENCODING_SLINEAR_LE;
-swcode = change_sign16;
-break;
- case AUDIO_ENCODING_SLINEAR_BE:
-hw.encoding = AUDIO_ENCODING_SLINEAR_LE;
-swcode = swap_bytes;
-break;
- case AUDIO_ENCODING_ULINEAR_BE:
-hw.encoding = AUDIO_ENCODING_SLINEAR_LE;
-swcode = swap_bytes_change_sign16;
-break;
- case AUDIO_ENCODING_ULAW:
-hw.encoding = AUDIO_ENCODING_ULINEAR_LE;
-swcode = mode == AUMODE_PLAY ?
- mulaw_to_linear8 : linear8_to_mulaw;
-break;
- case AUDIO_ENCODING_ALAW:
-hw.encoding = AUDIO_ENCODING_ULINEAR_LE;
-swcode = mode == AUMODE_PLAY ?
- alaw_to_linear8 : linear8_to_alaw;
-break;
default:
return EINVAL;
}
@@ -715,32 +650,7 @@ sbdsp_set_params(
p->sample_rate = SB_TC_TO_RATE(tc) / p->channels;
hw.sample_rate = p->sample_rate;
} else {
- switch (p->encoding) {
- case AUDIO_ENCODING_SLINEAR_BE:
- case AUDIO_ENCODING_SLINEAR_LE:
- case AUDIO_ENCODING_SLINEAR:
-hw.encoding = AUDIO_ENCODING_ULINEAR_LE;
-swcode = change_sign8;
-break;
- case AUDIO_ENCODING_ULINEAR_BE:
- case AUDIO_ENCODING_ULINEAR_LE:
-break;
- case AUDIO_ENCODING_ULAW:
-hw.encoding = AUDIO_ENCODING_ULINEAR_LE;
-swcode = mode == AUMODE_PLAY ?
- mulaw_to_linear8 : linear8_to_mulaw;
-break;
- case AUDIO_ENCODING_ALAW:
-hw.encoding =
CVS commit: [isaki-audio2] src/sys/dev/isa
Module Name:src
Committed By: isaki
Date: Sun Apr 21 06:09:01 UTC 2019
Modified Files:
src/sys/dev/isa [isaki-audio2]: aria.c
Log Message:
Adapt to audio2.
- Drop FULLDUPLEX property. The driver doesn't seem to be written
as full duplex.
To generate a diff of this commit:
cvs rdiff -u -r1.39 -r1.39.2.1 src/sys/dev/isa/aria.c
Please note that diffs are not public domain; they are subject to the
copyright notices on the relevant files.
Modified files:
Index: src/sys/dev/isa/aria.c
diff -u src/sys/dev/isa/aria.c:1.39 src/sys/dev/isa/aria.c:1.39.2.1
--- src/sys/dev/isa/aria.c:1.39 Sat Mar 16 12:09:58 2019
+++ src/sys/dev/isa/aria.c Sun Apr 21 06:09:01 2019
@@ -1,4 +1,4 @@
-/* $NetBSD: aria.c,v 1.39 2019/03/16 12:09:58 isaki Exp $ */
+/* $NetBSD: aria.c,v 1.39.2.1 2019/04/21 06:09:01 isaki Exp $ */
/*-
* Copyright (c) 1995, 1996, 1998 The NetBSD Foundation, Inc.
@@ -36,12 +36,7 @@
* o Look into where aria_prometheus_kludge() belongs.
* o Add some DMA code. It accomplishes its goal by
* direct IO at the moment.
- * o Different programs should be able to open the device
- * with O_RDONLY and O_WRONLY at the same time. But I
- * do not see support for this in /sys/dev/audio.c, so
* I cannot effectively code it.
- * o We should nicely deal with the cards that can do mu-law
- * and A-law output.
* o Rework the mixer interface.
* o Deal with the lvls better. We need to do better mapping
* between logarithmic scales and the one byte that
@@ -50,7 +45,7 @@
*/
#include
-__KERNEL_RCSID(0, "$NetBSD: aria.c,v 1.39 2019/03/16 12:09:58 isaki Exp $");
+__KERNEL_RCSID(0, "$NetBSD: aria.c,v 1.39.2.1 2019/04/21 06:09:01 isaki Exp $");
#include
#include
@@ -66,8 +61,6 @@ __KERNEL_RCSID(0, "$NetBSD: aria.c,v 1.3
#include
#include
-#include
-#include
#include
#include
@@ -147,12 +140,13 @@ void aria_do_kludge(bus_space_tag_t, bus
u_short, u_short, u_short, u_short);
void aria_prometheus_kludge(struct isa_attach_args *, bus_space_handle_t);
-int aria_query_encoding(void *, struct audio_encoding *);
+int aria_query_format(void *, audio_format_query_t *);
int aria_round_blocksize(void *, int, int, const audio_params_t *);
int aria_speaker_ctl(void *, int);
int aria_commit_settings(void *);
-int aria_set_params(void *, int, int, audio_params_t *, audio_params_t *,
- stream_filter_list_t *, stream_filter_list_t *);
+int aria_set_format(void *, int,
+ const audio_params_t *, const audio_params_t *,
+ audio_filter_reg_t *, audio_filter_reg_t *);
int aria_get_props(void *);
void aria_get_locks(void *, kmutex_t **, kmutex_t **);
@@ -196,6 +190,24 @@ struct audio_device aria_device = {
"aria"
};
+#define ARIA_FORMAT(enc, prec) \
+ { \
+ .mode = AUMODE_PLAY | AUMODE_RECORD, \
+ .encoding = (enc), \
+ .validbits = (prec), \
+ .precision = (prec), \
+ .channels = 2, \
+ .channel_mask = AUFMT_STEREO, \
+ .frequency_type = 6, \
+ .frequency = { 7875, 11025, 15750, 22050, 31500, 44100 }, \
+ }
+/* XXX Some models seem to support mulaw/alaw. */
+const struct audio_format aria_formats[] = {
+ ARIA_FORMAT(AUDIO_ENCODING_ULINEAR, 8),
+ ARIA_FORMAT(AUDIO_ENCODING_SLINEAR_LE, 16),
+};
+#define ARIA_NFORMATS __arraycount(aria_formats)
+
/*
* Define our interface to the higher level audio driver.
*/
@@ -203,8 +215,8 @@ struct audio_device aria_device = {
const struct audio_hw_if aria_hw_if = {
.open = ariaopen,
.close = ariaclose,
- .query_encoding = aria_query_encoding,
- .set_params = aria_set_params,
+ .query_format = aria_query_format,
+ .set_format = aria_set_format,
.round_blocksize = aria_round_blocksize,
.commit_settings = aria_commit_settings,
.start_output = aria_start_output,
@@ -500,68 +512,10 @@ aria_getdev(void *addr, struct audio_dev
*/
int
-aria_query_encoding(void *addr, struct audio_encoding *fp)
+aria_query_format(void *addr, audio_format_query_t *afp)
{
- struct aria_softc *sc;
- sc = addr;
- switch (fp->index) {
- case 0:
- strcpy(fp->name, AudioEmulaw);
- fp->encoding = AUDIO_ENCODING_ULAW;
- fp->precision = 8;
- if ((ARIA_MODEL>sc_hardware) == 0)
-fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
- break;
- case 1:
- strcpy(fp->name, AudioEalaw);
- fp->encoding = AUDIO_ENCODING_ALAW;
- fp->precision = 8;
- if ((ARIA_MODEL>sc_hardware) == 0)
-fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
- break;
- case 2:
- strcpy(fp->name, AudioEslinear);
- fp->encoding = AUDIO_ENCODING_SLINEAR;
- fp->precision = 8;
- fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
- break;
- case 3:
- strcpy(fp->name, AudioEslinear_le);
- fp->encoding = AUDIO_ENCODING_SLINEAR_LE;
- fp->precision = 16;
- fp->flags = 0;
- break;
- case 4:
- strcpy(fp->name, AudioEslinear_be);
- fp->encoding = AUDIO_ENCODING_SLINEAR_BE;
- fp->precision = 16;
- fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
- break;
- case 5:
-
