summaryrefslogtreecommitdiffstats
path: root/sound
diff options
context:
space:
mode:
authorTakashi Iwai <tiwai@suse.de>2009-12-14 18:01:56 +0100
committerTakashi Iwai <tiwai@suse.de>2009-12-14 18:01:56 +0100
commitb89371621e5bedc84498ced2c5c33976bd1b2f64 (patch)
treeb309919239586e25617a17785b827577b1abb6b5 /sound
parent63978ab3e3e963db28093b53bb4598f2702e1ad7 (diff)
parent74c2b45b714e49b427584b4bd8f44f1a24d82d9c (diff)
downloadop-kernel-dev-b89371621e5bedc84498ced2c5c33976bd1b2f64.zip
op-kernel-dev-b89371621e5bedc84498ced2c5c33976bd1b2f64.tar.gz
Merge branch 'next/isa' into topic/misc
Diffstat (limited to 'sound')
-rw-r--r--sound/Kconfig4
-rw-r--r--sound/isa/Kconfig21
-rw-r--r--sound/isa/Makefile2
-rw-r--r--sound/isa/als100.c121
-rw-r--r--sound/isa/cs423x/cs4236.c2
-rw-r--r--sound/isa/dt019x.c321
-rw-r--r--sound/isa/opti9xx/miro.c2
-rw-r--r--sound/isa/opti9xx/opti92x-ad1848.c95
-rw-r--r--sound/isa/sb/sb_mixer.c330
-rw-r--r--sound/isa/wss/wss_lib.c80
-rw-r--r--sound/oss/Kconfig2
-rw-r--r--sound/oss/dmasound/dmasound_paula.c2
-rw-r--r--sound/pci/ca0106/ca0106_proc.c2
-rw-r--r--sound/pci/cs46xx/imgs/cwcdma.asp9
-rw-r--r--sound/pci/emu10k1/emu10k1x.c2
-rw-r--r--sound/pci/hda/patch_cirrus.c2
-rw-r--r--sound/pci/hda/patch_cmedia.c2
-rw-r--r--sound/pci/hda/patch_realtek.c2
-rw-r--r--sound/pci/ice1712/juli.c2
-rw-r--r--sound/pci/rme9652/hdspm.c4
-rw-r--r--sound/pcmcia/pdaudiocf/pdaudiocf.c6
-rw-r--r--sound/pcmcia/vx/vxpocket.c6
-rw-r--r--sound/soc/codecs/uda134x.c4
-rw-r--r--sound/soc/codecs/wm8903.c6
-rw-r--r--sound/soc/codecs/wm8993.c4
-rw-r--r--sound/soc/omap/ams-delta.c4
-rw-r--r--sound/soc/omap/n810.c2
-rw-r--r--sound/soc/omap/omap-mcbsp.c6
-rw-r--r--sound/soc/omap/omap-pcm.c2
-rw-r--r--sound/soc/omap/omap2evm.c2
-rw-r--r--sound/soc/omap/omap3beagle.c2
-rw-r--r--sound/soc/omap/omap3evm.c2
-rw-r--r--sound/soc/omap/osk5912.c2
-rw-r--r--sound/soc/omap/overo.c2
-rw-r--r--sound/soc/omap/sdp3430.c2
-rw-r--r--sound/soc/omap/zoom2.c2
-rw-r--r--sound/soc/s3c24xx/s3c24xx_simtec.c2
-rw-r--r--sound/soc/s6000/s6000-pcm.c2
-rw-r--r--sound/sound_core.c2
-rw-r--r--sound/synth/emux/soundfont.c2
40 files changed, 359 insertions, 710 deletions
diff --git a/sound/Kconfig b/sound/Kconfig
index b3e53e6..fcad760f 100644
--- a/sound/Kconfig
+++ b/sound/Kconfig
@@ -1,6 +1,3 @@
-# sound/Config.in
-#
-
menuconfig SOUND
tristate "Sound card support"
depends on HAS_IOMEM
@@ -136,4 +133,3 @@ config AC97_BUS
sound subsystem and other function drivers completely unrelated to
sound although they're sharing the AC97 bus. Concerned drivers
should "select" this.
-
diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig
index 02fe81c..194af3b0 100644
--- a/sound/isa/Kconfig
+++ b/sound/isa/Kconfig
@@ -63,15 +63,16 @@ config SND_AD1848
will be called snd-ad1848.
config SND_ALS100
- tristate "Avance Logic ALS100/ALS120"
+ tristate "Diamond Tech. DT-019x and Avance Logic ALSxxx"
depends on PNP
select ISAPNP
select SND_OPL3_LIB
select SND_MPU401_UART
select SND_SB16_DSP
help
- Say Y here to include support for soundcards based on Avance
- Logic ALS100, ALS110, ALS120 and ALS200 chips.
+ Say Y here to include support for soundcards based on the
+ Diamond Technologies DT-019X or Avance Logic chips: ALS007,
+ ALS100, ALS110, ALS120 and ALS200 chips.
To compile this driver as a module, choose M here: the module
will be called snd-als100.
@@ -127,20 +128,6 @@ config SND_CS4236
To compile this driver as a module, choose M here: the module
will be called snd-cs4236.
-config SND_DT019X
- tristate "Diamond Technologies DT-019X, Avance Logic ALS-007"
- depends on PNP
- select ISAPNP
- select SND_OPL3_LIB
- select SND_MPU401_UART
- select SND_SB16_DSP
- help
- Say Y here to include support for soundcards based on the
- Diamond Technologies DT-019X or Avance Logic ALS-007 chips.
-
- To compile this driver as a module, choose M here: the module
- will be called snd-dt019x.
-
config SND_ES968
tristate "Generic ESS ES968 driver"
depends on PNP
diff --git a/sound/isa/Makefile b/sound/isa/Makefile
index b906b9a..c73d30c 100644
--- a/sound/isa/Makefile
+++ b/sound/isa/Makefile
@@ -7,7 +7,6 @@ snd-adlib-objs := adlib.o
snd-als100-objs := als100.o
snd-azt2320-objs := azt2320.o
snd-cmi8330-objs := cmi8330.o
-snd-dt019x-objs := dt019x.o
snd-es18xx-objs := es18xx.o
snd-opl3sa2-objs := opl3sa2.o
snd-sc6000-objs := sc6000.o
@@ -19,7 +18,6 @@ obj-$(CONFIG_SND_ADLIB) += snd-adlib.o
obj-$(CONFIG_SND_ALS100) += snd-als100.o
obj-$(CONFIG_SND_AZT2320) += snd-azt2320.o
obj-$(CONFIG_SND_CMI8330) += snd-cmi8330.o
-obj-$(CONFIG_SND_DT019X) += snd-dt019x.o
obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o
obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o
obj-$(CONFIG_SND_SC6000) += snd-sc6000.o
diff --git a/sound/isa/als100.c b/sound/isa/als100.c
index 5fd52e4..20becc8 100644
--- a/sound/isa/als100.c
+++ b/sound/isa/als100.c
@@ -2,9 +2,13 @@
/*
card-als100.c - driver for Avance Logic ALS100 based soundcards.
Copyright (C) 1999-2000 by Massimo Piccioni <dafastidio@libero.it>
+ Copyright (C) 1999-2002 by Massimo Piccioni <dafastidio@libero.it>
Thanks to Pierfrancesco 'qM2' Passerini.
+ Generalised for soundcards based on DT-0196 and ALS-007 chips
+ by Jonathan Woithe <jwoithe@physics.adelaide.edu.au>: June 2002.
+
This program is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2 of the License, or
@@ -33,10 +37,10 @@
#define PFX "als100: "
-MODULE_AUTHOR("Massimo Piccioni <dafastidio@libero.it>");
-MODULE_DESCRIPTION("Avance Logic ALS1X0");
-MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS100 - PRO16PNP},"
+MODULE_DESCRIPTION("Avance Logic ALS007/ALS1X0");
+MODULE_SUPPORTED_DEVICE("{{Diamond Technologies DT-019X},"
+ "{Avance Logic ALS-007}}"
+ "{{Avance Logic,ALS100 - PRO16PNP},"
"{Avance Logic,ALS110},"
"{Avance Logic,ALS120},"
"{Avance Logic,ALS200},"
@@ -45,9 +49,12 @@ MODULE_SUPPORTED_DEVICE("{{Avance Logic,ALS100 - PRO16PNP},"
"{Avance Logic,ALS120},"
"{RTL,RTL3000}}");
+MODULE_AUTHOR("Massimo Piccioni <dafastidio@libero.it>");
+MODULE_LICENSE("GPL");
+
static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
-static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_ISAPNP; /* Enable this card */
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */
static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
@@ -57,14 +64,15 @@ static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */
static int dma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */
module_param_array(index, int, NULL, 0444);
-MODULE_PARM_DESC(index, "Index value for als100 based soundcard.");
+MODULE_PARM_DESC(index, "Index value for Avance Logic based soundcard.");
module_param_array(id, charp, NULL, 0444);
-MODULE_PARM_DESC(id, "ID string for als100 based soundcard.");
+MODULE_PARM_DESC(id, "ID string for Avance Logic based soundcard.");
module_param_array(enable, bool, NULL, 0444);
-MODULE_PARM_DESC(enable, "Enable als100 based soundcard.");
+MODULE_PARM_DESC(enable, "Enable Avance Logic based soundcard.");
+
+MODULE_ALIAS("snd-dt019x");
struct snd_card_als100 {
- int dev_no;
struct pnp_dev *dev;
struct pnp_dev *devmpu;
struct pnp_dev *devopl;
@@ -72,25 +80,43 @@ struct snd_card_als100 {
};
static struct pnp_card_device_id snd_als100_pnpids[] = {
+ /* DT197A30 */
+ { .id = "RWB1688",
+ .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } },
+ .driver_data = SB_HW_DT019X },
+ /* DT0196 / ALS-007 */
+ { .id = "ALS0007",
+ .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } },
+ .driver_data = SB_HW_DT019X },
/* ALS100 - PRO16PNP */
- { .id = "ALS0001", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } } },
+ { .id = "ALS0001",
+ .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } },
+ .driver_data = SB_HW_ALS100 },
/* ALS110 - MF1000 - Digimate 3D Sound */
- { .id = "ALS0110", .devs = { { "@@@1001" }, { "@X@1001" }, { "@H@1001" } } },
+ { .id = "ALS0110",
+ .devs = { { "@@@1001" }, { "@X@1001" }, { "@H@1001" } },
+ .driver_data = SB_HW_ALS100 },
/* ALS120 */
- { .id = "ALS0120", .devs = { { "@@@2001" }, { "@X@2001" }, { "@H@2001" } } },
+ { .id = "ALS0120",
+ .devs = { { "@@@2001" }, { "@X@2001" }, { "@H@2001" } },
+ .driver_data = SB_HW_ALS100 },
/* ALS200 */
- { .id = "ALS0200", .devs = { { "@@@0020" }, { "@X@0020" }, { "@H@0001" } } },
+ { .id = "ALS0200",
+ .devs = { { "@@@0020" }, { "@X@0020" }, { "@H@0001" } },
+ .driver_data = SB_HW_ALS100 },
/* ALS200 OEM */
- { .id = "ALS0200", .devs = { { "@@@0020" }, { "@X@0020" }, { "@H@0020" } } },
+ { .id = "ALS0200",
+ .devs = { { "@@@0020" }, { "@X@0020" }, { "@H@0020" } },
+ .driver_data = SB_HW_ALS100 },
/* RTL3000 */
- { .id = "RTL3000", .devs = { { "@@@2001" }, { "@X@2001" }, { "@H@2001" } } },
- { .id = "", } /* end */
+ { .id = "RTL3000",
+ .devs = { { "@@@2001" }, { "@X@2001" }, { "@H@2001" } },
+ .driver_data = SB_HW_ALS100 },
+ { .id = "" } /* end */
};
MODULE_DEVICE_TABLE(pnp_card, snd_als100_pnpids);
-#define DRIVER_NAME "snd-card-als100"
-
static int __devinit snd_card_als100_pnp(int dev, struct snd_card_als100 *acard,
struct pnp_card_link *card,
const struct pnp_card_device_id *id)
@@ -113,8 +139,12 @@ static int __devinit snd_card_als100_pnp(int dev, struct snd_card_als100 *acard,
return err;
}
port[dev] = pnp_port_start(pdev, 0);
- dma8[dev] = pnp_dma(pdev, 1);
- dma16[dev] = pnp_dma(pdev, 0);
+ if (id->driver_data == SB_HW_DT019X)
+ dma8[dev] = pnp_dma(pdev, 0);
+ else {
+ dma8[dev] = pnp_dma(pdev, 1);
+ dma16[dev] = pnp_dma(pdev, 0);
+ }
irq[dev] = pnp_irq(pdev, 0);
pdev = acard->devmpu;
@@ -175,22 +205,33 @@ static int __devinit snd_card_als100_probe(int dev,
}
snd_card_set_dev(card, &pcard->card->dev);
- if ((error = snd_sbdsp_create(card, port[dev],
- irq[dev],
- snd_sb16dsp_interrupt,
- dma8[dev],
- dma16[dev],
- SB_HW_ALS100, &chip)) < 0) {
+ if (pid->driver_data == SB_HW_DT019X)
+ dma16[dev] = -1;
+
+ error = snd_sbdsp_create(card, port[dev], irq[dev],
+ snd_sb16dsp_interrupt,
+ dma8[dev], dma16[dev],
+ pid->driver_data,
+ &chip);
+ if (error < 0) {
snd_card_free(card);
return error;
}
acard->chip = chip;
- strcpy(card->driver, "ALS100");
- strcpy(card->shortname, "Avance Logic ALS100");
- sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d",
- card->shortname, chip->name, chip->port,
- irq[dev], dma8[dev], dma16[dev]);
+ if (pid->driver_data == SB_HW_DT019X) {
+ strcpy(card->driver, "DT-019X");
+ strcpy(card->shortname, "Diamond Tech. DT-019X");
+ sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d",
+ card->shortname, chip->name, chip->port,
+ irq[dev], dma8[dev]);
+ } else {
+ strcpy(card->driver, "ALS100");
+ strcpy(card->shortname, "Avance Logic ALS100");
+ sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d",
+ card->shortname, chip->name, chip->port,
+ irq[dev], dma8[dev], dma16[dev]);
+ }
if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) {
snd_card_free(card);
@@ -203,9 +244,19 @@ static int __devinit snd_card_als100_probe(int dev,
}
if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) {
- if (snd_mpu401_uart_new(card, 0, MPU401_HW_ALS100,
+ int mpu_type = MPU401_HW_ALS100;
+
+ if (mpu_irq[dev] == SNDRV_AUTO_IRQ)
+ mpu_irq[dev] = -1;
+
+ if (pid->driver_data == SB_HW_DT019X)
+ mpu_type = MPU401_HW_MPU401;
+
+ if (snd_mpu401_uart_new(card, 0,
+ mpu_type,
mpu_port[dev], 0,
- mpu_irq[dev], IRQF_DISABLED,
+ mpu_irq[dev],
+ mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0,
NULL) < 0)
snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx\n", mpu_port[dev]);
}
@@ -291,7 +342,7 @@ static int snd_als100_pnp_resume(struct pnp_card_link *pcard)
static struct pnp_card_driver als100_pnpc_driver = {
.flags = PNP_DRIVER_RES_DISABLE,
- .name = "als100",
+ .name = "als100",
.id_table = snd_als100_pnpids,
.probe = snd_als100_pnp_detect,
.remove = __devexit_p(snd_als100_pnp_remove),
@@ -312,7 +363,7 @@ static int __init alsa_card_als100_init(void)
if (!als100_devices) {
pnp_unregister_card_driver(&als100_pnpc_driver);
#ifdef MODULE
- snd_printk(KERN_ERR "no ALS100 based soundcards found\n");
+ snd_printk(KERN_ERR "no Avance Logic based soundcards found\n");
#endif
return -ENODEV;
}
diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c
index 93fa672..cc15d1d 100644
--- a/sound/isa/cs423x/cs4236.c
+++ b/sound/isa/cs423x/cs4236.c
@@ -177,7 +177,7 @@ static struct pnp_card_device_id snd_cs423x_pnpids[] = {
{ .id = "CSC0437", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
/* Digital PC 5000 Onboard - CS4236B */
{ .id = "CSC0735", .devs = { { "CSC0000" }, { "CSC0010" } } },
- /* some uknown CS4236B */
+ /* some unknown CS4236B */
{ .id = "CSC0b35", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
/* Intel PR440FX Onboard sound */
{ .id = "CSC0b36", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
diff --git a/sound/isa/dt019x.c b/sound/isa/dt019x.c
deleted file mode 100644
index 80f5b1a..0000000
--- a/sound/isa/dt019x.c
+++ /dev/null
@@ -1,321 +0,0 @@
-
-/*
- dt019x.c - driver for Diamond Technologies DT-0197H based soundcards.
- Copyright (C) 1999, 2002 by Massimo Piccioni <dafastidio@libero.it>
-
- Generalised for soundcards based on DT-0196 and ALS-007 chips
- by Jonathan Woithe <jwoithe@physics.adelaide.edu.au>: June 2002.
-
- This program is free software; you can redistribute it and/or modify
- it under the terms of the GNU General Public License as published by
- the Free Software Foundation; either version 2 of the License, or
- (at your option) any later version.
-
- This program is distributed in the hope that it will be useful,
- but WITHOUT ANY WARRANTY; without even the implied warranty of
- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- GNU General Public License for more details.
-
- You should have received a copy of the GNU General Public License
- along with this program; if not, write to the Free Software
- Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
-*/
-
-#include <linux/init.h>
-#include <linux/wait.h>
-#include <linux/pnp.h>
-#include <linux/moduleparam.h>
-#include <sound/core.h>
-#include <sound/initval.h>
-#include <sound/mpu401.h>
-#include <sound/opl3.h>
-#include <sound/sb.h>
-
-#define PFX "dt019x: "
-
-MODULE_AUTHOR("Massimo Piccioni <dafastidio@libero.it>");
-MODULE_DESCRIPTION("Diamond Technologies DT-019X / Avance Logic ALS-007");
-MODULE_LICENSE("GPL");
-MODULE_SUPPORTED_DEVICE("{{Diamond Technologies DT-019X},"
- "{Avance Logic ALS-007}}");
-
-static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
-static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */
-static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */
-static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
-static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
-static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* PnP setup */
-static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* PnP setup */
-static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* PnP setup */
-static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* PnP setup */
-
-module_param_array(index, int, NULL, 0444);
-MODULE_PARM_DESC(index, "Index value for DT-019X based soundcard.");
-module_param_array(id, charp, NULL, 0444);
-MODULE_PARM_DESC(id, "ID string for DT-019X based soundcard.");
-module_param_array(enable, bool, NULL, 0444);
-MODULE_PARM_DESC(enable, "Enable DT-019X based soundcard.");
-
-struct snd_card_dt019x {
- struct pnp_dev *dev;
- struct pnp_dev *devmpu;
- struct pnp_dev *devopl;
- struct snd_sb *chip;
-};
-
-static struct pnp_card_device_id snd_dt019x_pnpids[] = {
- /* DT197A30 */
- { .id = "RWB1688", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" }, } },
- /* DT0196 / ALS-007 */
- { .id = "ALS0007", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" }, } },
- { .id = "", }
-};
-
-MODULE_DEVICE_TABLE(pnp_card, snd_dt019x_pnpids);
-
-
-#define DRIVER_NAME "snd-card-dt019x"
-
-
-static int __devinit snd_card_dt019x_pnp(int dev, struct snd_card_dt019x *acard,
- struct pnp_card_link *card,
- const struct pnp_card_device_id *pid)
-{
- struct pnp_dev *pdev;
- int err;
-
- acard->dev = pnp_request_card_device(card, pid->devs[0].id, NULL);
- if (acard->dev == NULL)
- return -ENODEV;
-
- acard->devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL);
- acard->devopl = pnp_request_card_device(card, pid->devs[2].id, NULL);
-
- pdev = acard->dev;
-
- err = pnp_activate_dev(pdev);
- if (err < 0) {
- snd_printk(KERN_ERR PFX "DT-019X AUDIO pnp configure failure\n");
- return err;
- }
-
- port[dev] = pnp_port_start(pdev, 0);
- dma8[dev] = pnp_dma(pdev, 0);
- irq[dev] = pnp_irq(pdev, 0);
- snd_printdd("dt019x: found audio interface: port=0x%lx, irq=0x%x, dma=0x%x\n",
- port[dev],irq[dev],dma8[dev]);
-
- pdev = acard->devmpu;
- if (pdev != NULL) {
- err = pnp_activate_dev(pdev);
- if (err < 0) {
- pnp_release_card_device(pdev);
- snd_printk(KERN_ERR PFX "DT-019X MPU401 pnp configure failure, skipping\n");
- goto __mpu_error;
- }
- mpu_port[dev] = pnp_port_start(pdev, 0);
- mpu_irq[dev] = pnp_irq(pdev, 0);
- snd_printdd("dt019x: found MPU-401: port=0x%lx, irq=0x%x\n",
- mpu_port[dev],mpu_irq[dev]);
- } else {
- __mpu_error:
- acard->devmpu = NULL;
- mpu_port[dev] = -1;
- }
-
- pdev = acard->devopl;
- if (pdev != NULL) {
- err = pnp_activate_dev(pdev);
- if (err < 0) {
- pnp_release_card_device(pdev);
- snd_printk(KERN_ERR PFX "DT-019X OPL3 pnp configure failure, skipping\n");
- goto __fm_error;
- }
- fm_port[dev] = pnp_port_start(pdev, 0);
- snd_printdd("dt019x: found OPL3 synth: port=0x%lx\n",fm_port[dev]);
- } else {
- __fm_error:
- acard->devopl = NULL;
- fm_port[dev] = -1;
- }
-
- return 0;
-}
-
-static int __devinit snd_card_dt019x_probe(int dev, struct pnp_card_link *pcard, const struct pnp_card_device_id *pid)
-{
- int error;
- struct snd_sb *chip;
- struct snd_card *card;
- struct snd_card_dt019x *acard;
- struct snd_opl3 *opl3;
-
- error = snd_card_create(index[dev], id[dev], THIS_MODULE,
- sizeof(struct snd_card_dt019x), &card);
- if (error < 0)
- return error;
- acard = card->private_data;
-
- snd_card_set_dev(card, &pcard->card->dev);
- if ((error = snd_card_dt019x_pnp(dev, acard, pcard, pid))) {
- snd_card_free(card);
- return error;
- }
-
- if ((error = snd_sbdsp_create(card, port[dev],
- irq[dev],
- snd_sb16dsp_interrupt,
- dma8[dev],
- -1,
- SB_HW_DT019X,
- &chip)) < 0) {
- snd_card_free(card);
- return error;
- }
- acard->chip = chip;
-
- strcpy(card->driver, "DT-019X");
- strcpy(card->shortname, "Diamond Tech. DT-019X");
- sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d",
- card->shortname, chip->name, chip->port,
- irq[dev], dma8[dev]);
-
- if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) {
- snd_card_free(card);
- return error;
- }
- if ((error = snd_sbmixer_new(chip)) < 0) {
- snd_card_free(card);
- return error;
- }
-
- if (mpu_port[dev] > 0 && mpu_port[dev] != SNDRV_AUTO_PORT) {
- if (mpu_irq[dev] == SNDRV_AUTO_IRQ)
- mpu_irq[dev] = -1;
- if (snd_mpu401_uart_new(card, 0,
-/* MPU401_HW_SB,*/
- MPU401_HW_MPU401,
- mpu_port[dev], 0,
- mpu_irq[dev],
- mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0,
- NULL) < 0)
- snd_printk(KERN_ERR PFX "no MPU-401 device at 0x%lx ?\n", mpu_port[dev]);
- }
-
- if (fm_port[dev] > 0 && fm_port[dev] != SNDRV_AUTO_PORT) {
- if (snd_opl3_create(card,
- fm_port[dev],
- fm_port[dev] + 2,
- OPL3_HW_AUTO, 0, &opl3) < 0) {
- snd_printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx ?\n",
- fm_port[dev], fm_port[dev] + 2);
- } else {
- if ((error = snd_opl3_timer_new(opl3, 0, 1)) < 0) {
- snd_card_free(card);
- return error;
- }
- if ((error = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) {
- snd_card_free(card);
- return error;
- }
- }
- }
-
- if ((error = snd_card_register(card)) < 0) {
- snd_card_free(card);
- return error;
- }
- pnp_set_card_drvdata(pcard, card);
- return 0;
-}
-
-static unsigned int __devinitdata dt019x_devices;
-
-static int __devinit snd_dt019x_pnp_probe(struct pnp_card_link *card,
- const struct pnp_card_device_id *pid)
-{
- static int dev;
- int res;
-
- for ( ; dev < SNDRV_CARDS; dev++) {
- if (!enable[dev])
- continue;
- res = snd_card_dt019x_probe(dev, card, pid);
- if (res < 0)
- return res;
- dev++;
- dt019x_devices++;
- return 0;
- }
- return -ENODEV;
-}
-
-static void __devexit snd_dt019x_pnp_remove(struct pnp_card_link * pcard)
-{
- snd_card_free(pnp_get_card_drvdata(pcard));
- pnp_set_card_drvdata(pcard, NULL);
-}
-
-#ifdef CONFIG_PM
-static int snd_dt019x_pnp_suspend(struct pnp_card_link *pcard, pm_message_t state)
-{
- struct snd_card *card = pnp_get_card_drvdata(pcard);
- struct snd_card_dt019x *acard = card->private_data;
- struct snd_sb *chip = acard->chip;
-
- snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
- snd_pcm_suspend_all(chip->pcm);
- snd_sbmixer_suspend(chip);
- return 0;
-}
-
-static int snd_dt019x_pnp_resume(struct pnp_card_link *pcard)
-{
- struct snd_card *card = pnp_get_card_drvdata(pcard);
- struct snd_card_dt019x *acard = card->private_data;
- struct snd_sb *chip = acard->chip;
-
- snd_sbdsp_reset(chip);
- snd_sbmixer_resume(chip);
- snd_power_change_state(card, SNDRV_CTL_POWER_D0);
- return 0;
-}
-#endif
-
-static struct pnp_card_driver dt019x_pnpc_driver = {
- .flags = PNP_DRIVER_RES_DISABLE,
- .name = "dt019x",
- .id_table = snd_dt019x_pnpids,
- .probe = snd_dt019x_pnp_probe,
- .remove = __devexit_p(snd_dt019x_pnp_remove),
-#ifdef CONFIG_PM
- .suspend = snd_dt019x_pnp_suspend,
- .resume = snd_dt019x_pnp_resume,
-#endif
-};
-
-static int __init alsa_card_dt019x_init(void)
-{
- int err;
-
- err = pnp_register_card_driver(&dt019x_pnpc_driver);
- if (err)
- return err;
-
- if (!dt019x_devices) {
- pnp_unregister_card_driver(&dt019x_pnpc_driver);
-#ifdef MODULE
- snd_printk(KERN_ERR "no DT-019X / ALS-007 based soundcards found\n");
-#endif
- return -ENODEV;
- }
- return 0;
-}
-
-static void __exit alsa_card_dt019x_exit(void)
-{
- pnp_unregister_card_driver(&dt019x_pnpc_driver);
-}
-
-module_init(alsa_card_dt019x_init)
-module_exit(alsa_card_dt019x_exit)
diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c
index 6123c75..b865e45 100644
--- a/sound/isa/opti9xx/miro.c
+++ b/sound/isa/opti9xx/miro.c
@@ -133,7 +133,7 @@ struct snd_miro {
static struct snd_miro_aci aci_device;
static char * snd_opti9xx_names[] = {
- "unkown",
+ "unknown",
"82C928", "82C929",
"82C924", "82C925",
"82C930", "82C931", "82C933"
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index d8eac3f..a4af53b 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -33,6 +33,7 @@
#include <asm/io.h>
#include <asm/dma.h>
#include <sound/core.h>
+#include <sound/tlv.h>
#include <sound/wss.h>
#include <sound/mpu401.h>
#include <sound/opl3.h>
@@ -179,7 +180,7 @@ MODULE_DEVICE_TABLE(pnp_card, snd_opti9xx_pnpids);
#endif
static char * snd_opti9xx_names[] = {
- "unkown",
+ "unknown",
"82C928", "82C929",
"82C924", "82C925",
"82C930", "82C931", "82C933"
@@ -546,6 +547,93 @@ __skip_mpu:
#ifdef OPTi93X
+static const DECLARE_TLV_DB_SCALE(db_scale_5bit_3db_step, -9300, 300, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_5bit, -4650, 150, 0);
+static const DECLARE_TLV_DB_SCALE(db_scale_4bit_12db_max, -3300, 300, 0);
+
+static struct snd_kcontrol_new snd_opti93x_controls[] = {
+WSS_DOUBLE("Master Playback Switch", 0,
+ OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1),
+WSS_DOUBLE_TLV("Master Playback Volume", 0,
+ OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1,
+ db_scale_5bit_3db_step),
+WSS_DOUBLE_TLV("PCM Playback Volume", 0,
+ CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1,
+ db_scale_5bit),
+WSS_DOUBLE_TLV("FM Playback Volume", 0,
+ CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1,
+ db_scale_4bit_12db_max),
+WSS_DOUBLE("Line Playback Switch", 0,
+ CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1),
+WSS_DOUBLE_TLV("Line Playback Volume", 0,
+ CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1,
+ db_scale_4bit_12db_max),
+WSS_DOUBLE("Mic Playback Switch", 0,
+ OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1),
+WSS_DOUBLE_TLV("Mic Playback Volume", 0,
+ OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1,
+ db_scale_4bit_12db_max),
+WSS_DOUBLE_TLV("CD Playback Volume", 0,
+ CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1,
+ db_scale_4bit_12db_max),
+WSS_DOUBLE("Aux Playback Switch", 0,
+ OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1),
+WSS_DOUBLE_TLV("Aux Playback Volume", 0,
+ OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1,
+ db_scale_4bit_12db_max),
+};
+
+static int __devinit snd_opti93x_mixer(struct snd_wss *chip)
+{
+ struct snd_card *card;
+ unsigned int idx;
+ struct snd_ctl_elem_id id1, id2;
+ int err;
+
+ if (snd_BUG_ON(!chip || !chip->pcm))
+ return -EINVAL;
+
+ card = chip->card;
+
+ strcpy(card->mixername, chip->pcm->name);
+
+ memset(&id1, 0, sizeof(id1));
+ memset(&id2, 0, sizeof(id2));
+ id1.iface = id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+ /* reassign AUX0 switch to CD */
+ strcpy(id1.name, "Aux Playback Switch");
+ strcpy(id2.name, "CD Playback Switch");
+ err = snd_ctl_rename_id(card, &id1, &id2);
+ if (err < 0) {
+ snd_printk(KERN_ERR "Cannot rename opti93x control\n");
+ return err;
+ }
+ /* reassign AUX1 switch to FM */
+ strcpy(id1.name, "Aux Playback Switch"); id1.index = 1;
+ strcpy(id2.name, "FM Playback Switch");
+ err = snd_ctl_rename_id(card, &id1, &id2);
+ if (err < 0) {
+ snd_printk(KERN_ERR "Cannot rename opti93x control\n");
+ return err;
+ }
+ /* remove AUX1 volume */
+ strcpy(id1.name, "Aux Playback Volume"); id1.index = 1;
+ snd_ctl_remove_id(card, &id1);
+
+ /* Replace WSS volume controls with OPTi93x volume controls */
+ id1.index = 0;
+ for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) {
+ strcpy(id1.name, snd_opti93x_controls[idx].name);
+ snd_ctl_remove_id(card, &id1);
+
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_opti93x_controls[idx], chip));
+ if (err < 0)
+ return err;
+ }
+ return 0;
+}
+
static irqreturn_t snd_opti93x_interrupt(int irq, void *dev_id)
{
struct snd_opti9xx *chip = dev_id;
@@ -754,6 +842,11 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card)
error = snd_wss_mixer(codec);
if (error < 0)
return error;
+#ifdef OPTi93X
+ error = snd_opti93x_mixer(codec);
+ if (error < 0)
+ return error;
+#endif
#ifdef CS4231
error = snd_wss_timer(codec, 0, &timer);
if (error < 0)
diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c
index 318ff0c..8cfc41f 100644
--- a/sound/isa/sb/sb_mixer.c
+++ b/sound/isa/sb/sb_mixer.c
@@ -528,20 +528,11 @@ int snd_sbmixer_add_ctl(struct snd_sb *chip, const char *name, int index, int ty
* SB 2.0 specific mixer elements
*/
-static struct sbmix_elem snd_sb20_ctl_master_play_vol =
- SB_SINGLE("Master Playback Volume", SB_DSP20_MASTER_DEV, 1, 7);
-static struct sbmix_elem snd_sb20_ctl_pcm_play_vol =
- SB_SINGLE("PCM Playback Volume", SB_DSP20_PCM_DEV, 1, 3);
-static struct sbmix_elem snd_sb20_ctl_synth_play_vol =
- SB_SINGLE("Synth Playback Volume", SB_DSP20_FM_DEV, 1, 7);
-static struct sbmix_elem snd_sb20_ctl_cd_play_vol =
- SB_SINGLE("CD Playback Volume", SB_DSP20_CD_DEV, 1, 7);
-
-static struct sbmix_elem *snd_sb20_controls[] = {
- &snd_sb20_ctl_master_play_vol,
- &snd_sb20_ctl_pcm_play_vol,
- &snd_sb20_ctl_synth_play_vol,
- &snd_sb20_ctl_cd_play_vol
+static struct sbmix_elem snd_sb20_controls[] = {
+ SB_SINGLE("Master Playback Volume", SB_DSP20_MASTER_DEV, 1, 7),
+ SB_SINGLE("PCM Playback Volume", SB_DSP20_PCM_DEV, 1, 3),
+ SB_SINGLE("Synth Playback Volume", SB_DSP20_FM_DEV, 1, 7),
+ SB_SINGLE("CD Playback Volume", SB_DSP20_CD_DEV, 1, 7)
};
static unsigned char snd_sb20_init_values[][2] = {
@@ -552,41 +543,24 @@ static unsigned char snd_sb20_init_values[][2] = {
/*
* SB Pro specific mixer elements
*/
-static struct sbmix_elem snd_sbpro_ctl_master_play_vol =
- SB_DOUBLE("Master Playback Volume", SB_DSP_MASTER_DEV, SB_DSP_MASTER_DEV, 5, 1, 7);
-static struct sbmix_elem snd_sbpro_ctl_pcm_play_vol =
- SB_DOUBLE("PCM Playback Volume", SB_DSP_PCM_DEV, SB_DSP_PCM_DEV, 5, 1, 7);
-static struct sbmix_elem snd_sbpro_ctl_pcm_play_filter =
- SB_SINGLE("PCM Playback Filter", SB_DSP_PLAYBACK_FILT, 5, 1);
-static struct sbmix_elem snd_sbpro_ctl_synth_play_vol =
- SB_DOUBLE("Synth Playback Volume", SB_DSP_FM_DEV, SB_DSP_FM_DEV, 5, 1, 7);
-static struct sbmix_elem snd_sbpro_ctl_cd_play_vol =
- SB_DOUBLE("CD Playback Volume", SB_DSP_CD_DEV, SB_DSP_CD_DEV, 5, 1, 7);
-static struct sbmix_elem snd_sbpro_ctl_line_play_vol =
- SB_DOUBLE("Line Playback Volume", SB_DSP_LINE_DEV, SB_DSP_LINE_DEV, 5, 1, 7);
-static struct sbmix_elem snd_sbpro_ctl_mic_play_vol =
- SB_SINGLE("Mic Playback Volume", SB_DSP_MIC_DEV, 1, 3);
-static struct sbmix_elem snd_sbpro_ctl_capture_source =
+static struct sbmix_elem snd_sbpro_controls[] = {
+ SB_DOUBLE("Master Playback Volume",
+ SB_DSP_MASTER_DEV, SB_DSP_MASTER_DEV, 5, 1, 7),
+ SB_DOUBLE("PCM Playback Volume",
+ SB_DSP_PCM_DEV, SB_DSP_PCM_DEV, 5, 1, 7),
+ SB_SINGLE("PCM Playback Filter", SB_DSP_PLAYBACK_FILT, 5, 1),
+ SB_DOUBLE("Synth Playback Volume",
+ SB_DSP_FM_DEV, SB_DSP_FM_DEV, 5, 1, 7),
+ SB_DOUBLE("CD Playback Volume", SB_DSP_CD_DEV, SB_DSP_CD_DEV, 5, 1, 7),
+ SB_DOUBLE("Line Playback Volume",
+ SB_DSP_LINE_DEV, SB_DSP_LINE_DEV, 5, 1, 7),
+ SB_SINGLE("Mic Playback Volume", SB_DSP_MIC_DEV, 1, 3),
{
.name = "Capture Source",
.type = SB_MIX_CAPTURE_PRO
- };
-static struct sbmix_elem snd_sbpro_ctl_capture_filter =
- SB_SINGLE("Capture Filter", SB_DSP_CAPTURE_FILT, 5, 1);
-static struct sbmix_elem snd_sbpro_ctl_capture_low_filter =
- SB_SINGLE("Capture Low-Pass Filter", SB_DSP_CAPTURE_FILT, 3, 1);
-
-static struct sbmix_elem *snd_sbpro_controls[] = {
- &snd_sbpro_ctl_master_play_vol,
- &snd_sbpro_ctl_pcm_play_vol,
- &snd_sbpro_ctl_pcm_play_filter,
- &snd_sbpro_ctl_synth_play_vol,
- &snd_sbpro_ctl_cd_play_vol,
- &snd_sbpro_ctl_line_play_vol,
- &snd_sbpro_ctl_mic_play_vol,
- &snd_sbpro_ctl_capture_source,
- &snd_sbpro_ctl_capture_filter,
- &snd_sbpro_ctl_capture_low_filter
+ },
+ SB_SINGLE("Capture Filter", SB_DSP_CAPTURE_FILT, 5, 1),
+ SB_SINGLE("Capture Low-Pass Filter", SB_DSP_CAPTURE_FILT, 3, 1)
};
static unsigned char snd_sbpro_init_values[][2] = {
@@ -598,68 +572,42 @@ static unsigned char snd_sbpro_init_values[][2] = {
/*
* SB16 specific mixer elements
*/
-static struct sbmix_elem snd_sb16_ctl_master_play_vol =
- SB_DOUBLE("Master Playback Volume", SB_DSP4_MASTER_DEV, (SB_DSP4_MASTER_DEV + 1), 3, 3, 31);
-static struct sbmix_elem snd_sb16_ctl_3d_enhance_switch =
- SB_SINGLE("3D Enhancement Switch", SB_DSP4_3DSE, 0, 1);
-static struct sbmix_elem snd_sb16_ctl_tone_bass =
- SB_DOUBLE("Tone Control - Bass", SB_DSP4_BASS_DEV, (SB_DSP4_BASS_DEV + 1), 4, 4, 15);
-static struct sbmix_elem snd_sb16_ctl_tone_treble =
- SB_DOUBLE("Tone Control - Treble", SB_DSP4_TREBLE_DEV, (SB_DSP4_TREBLE_DEV + 1), 4, 4, 15);
-static struct sbmix_elem snd_sb16_ctl_pcm_play_vol =
- SB_DOUBLE("PCM Playback Volume", SB_DSP4_PCM_DEV, (SB_DSP4_PCM_DEV + 1), 3, 3, 31);
-static struct sbmix_elem snd_sb16_ctl_synth_capture_route =
- SB16_INPUT_SW("Synth Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 6, 5);
-static struct sbmix_elem snd_sb16_ctl_synth_play_vol =
- SB_DOUBLE("Synth Playback Volume", SB_DSP4_SYNTH_DEV, (SB_DSP4_SYNTH_DEV + 1), 3, 3, 31);
-static struct sbmix_elem snd_sb16_ctl_cd_capture_route =
- SB16_INPUT_SW("CD Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 2, 1);
-static struct sbmix_elem snd_sb16_ctl_cd_play_switch =
- SB_DOUBLE("CD Playback Switch", SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 2, 1, 1);
-static struct sbmix_elem snd_sb16_ctl_cd_play_vol =
- SB_DOUBLE("CD Playback Volume", SB_DSP4_CD_DEV, (SB_DSP4_CD_DEV + 1), 3, 3, 31);
-static struct sbmix_elem snd_sb16_ctl_line_capture_route =
- SB16_INPUT_SW("Line Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 4, 3);
-static struct sbmix_elem snd_sb16_ctl_line_play_switch =
- SB_DOUBLE("Line Playback Switch", SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, 1);
-static struct sbmix_elem snd_sb16_ctl_line_play_vol =
- SB_DOUBLE("Line Playback Volume", SB_DSP4_LINE_DEV, (SB_DSP4_LINE_DEV + 1), 3, 3, 31);
-static struct sbmix_elem snd_sb16_ctl_mic_capture_route =
- SB16_INPUT_SW("Mic Capture Route", SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 0, 0);
-static struct sbmix_elem snd_sb16_ctl_mic_play_switch =
- SB_SINGLE("Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1);
-static struct sbmix_elem snd_sb16_ctl_mic_play_vol =
- SB_SINGLE("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31);
-static struct sbmix_elem snd_sb16_ctl_pc_speaker_vol =
- SB_SINGLE("Beep Volume", SB_DSP4_SPEAKER_DEV, 6, 3);
-static struct sbmix_elem snd_sb16_ctl_capture_vol =
- SB_DOUBLE("Capture Volume", SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3);
-static struct sbmix_elem snd_sb16_ctl_play_vol =
- SB_DOUBLE("Playback Volume", SB_DSP4_OGAIN_DEV, (SB_DSP4_OGAIN_DEV + 1), 6, 6, 3);
-static struct sbmix_elem snd_sb16_ctl_auto_mic_gain =
- SB_SINGLE("Mic Auto Gain", SB_DSP4_MIC_AGC, 0, 1);
-
-static struct sbmix_elem *snd_sb16_controls[] = {
- &snd_sb16_ctl_master_play_vol,
- &snd_sb16_ctl_3d_enhance_switch,
- &snd_sb16_ctl_tone_bass,
- &snd_sb16_ctl_tone_treble,
- &snd_sb16_ctl_pcm_play_vol,
- &snd_sb16_ctl_synth_capture_route,
- &snd_sb16_ctl_synth_play_vol,
- &snd_sb16_ctl_cd_capture_route,
- &snd_sb16_ctl_cd_play_switch,
- &snd_sb16_ctl_cd_play_vol,
- &snd_sb16_ctl_line_capture_route,
- &snd_sb16_ctl_line_play_switch,
- &snd_sb16_ctl_line_play_vol,
- &snd_sb16_ctl_mic_capture_route,
- &snd_sb16_ctl_mic_play_switch,
- &snd_sb16_ctl_mic_play_vol,
- &snd_sb16_ctl_pc_speaker_vol,
- &snd_sb16_ctl_capture_vol,
- &snd_sb16_ctl_play_vol,
- &snd_sb16_ctl_auto_mic_gain
+static struct sbmix_elem snd_sb16_controls[] = {
+ SB_DOUBLE("Master Playback Volume",
+ SB_DSP4_MASTER_DEV, (SB_DSP4_MASTER_DEV + 1), 3, 3, 31),
+ SB_DOUBLE("PCM Playback Volume",
+ SB_DSP4_PCM_DEV, (SB_DSP4_PCM_DEV + 1), 3, 3, 31),
+ SB16_INPUT_SW("Synth Capture Route",
+ SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 6, 5),
+ SB_DOUBLE("Synth Playback Volume",
+ SB_DSP4_SYNTH_DEV, (SB_DSP4_SYNTH_DEV + 1), 3, 3, 31),
+ SB16_INPUT_SW("CD Capture Route",
+ SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 2, 1),
+ SB_DOUBLE("CD Playback Switch",
+ SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 2, 1, 1),
+ SB_DOUBLE("CD Playback Volume",
+ SB_DSP4_CD_DEV, (SB_DSP4_CD_DEV + 1), 3, 3, 31),
+ SB16_INPUT_SW("Mic Capture Route",
+ SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 0, 0),
+ SB_SINGLE("Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1),
+ SB_SINGLE("Mic Playback Volume", SB_DSP4_MIC_DEV, 3, 31),
+ SB_SINGLE("Beep Volume", SB_DSP4_SPEAKER_DEV, 6, 3),
+ SB_DOUBLE("Capture Volume",
+ SB_DSP4_IGAIN_DEV, (SB_DSP4_IGAIN_DEV + 1), 6, 6, 3),
+ SB_DOUBLE("Playback Volume",
+ SB_DSP4_OGAIN_DEV, (SB_DSP4_OGAIN_DEV + 1), 6, 6, 3),
+ SB16_INPUT_SW("Line Capture Route",
+ SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, 4, 3),
+ SB_DOUBLE("Line Playback Switch",
+ SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, 1),
+ SB_DOUBLE("Line Playback Volume",
+ SB_DSP4_LINE_DEV, (SB_DSP4_LINE_DEV + 1), 3, 3, 31),
+ SB_SINGLE("Mic Auto Gain", SB_DSP4_MIC_AGC, 0, 1),
+ SB_SINGLE("3D Enhancement Switch", SB_DSP4_3DSE, 0, 1),
+ SB_DOUBLE("Tone Control - Bass",
+ SB_DSP4_BASS_DEV, (SB_DSP4_BASS_DEV + 1), 4, 4, 15),
+ SB_DOUBLE("Tone Control - Treble",
+ SB_DSP4_TREBLE_DEV, (SB_DSP4_TREBLE_DEV + 1), 4, 4, 15)
};
static unsigned char snd_sb16_init_values[][2] = {
@@ -678,46 +626,34 @@ static unsigned char snd_sb16_init_values[][2] = {
/*
* DT019x specific mixer elements
*/
-static struct sbmix_elem snd_dt019x_ctl_master_play_vol =
- SB_DOUBLE("Master Playback Volume", SB_DT019X_MASTER_DEV, SB_DT019X_MASTER_DEV, 4,0, 15);
-static struct sbmix_elem snd_dt019x_ctl_pcm_play_vol =
- SB_DOUBLE("PCM Playback Volume", SB_DT019X_PCM_DEV, SB_DT019X_PCM_DEV, 4,0, 15);
-static struct sbmix_elem snd_dt019x_ctl_synth_play_vol =
- SB_DOUBLE("Synth Playback Volume", SB_DT019X_SYNTH_DEV, SB_DT019X_SYNTH_DEV, 4,0, 15);
-static struct sbmix_elem snd_dt019x_ctl_cd_play_vol =
- SB_DOUBLE("CD Playback Volume", SB_DT019X_CD_DEV, SB_DT019X_CD_DEV, 4,0, 15);
-static struct sbmix_elem snd_dt019x_ctl_mic_play_vol =
- SB_SINGLE("Mic Playback Volume", SB_DT019X_MIC_DEV, 4, 7);
-static struct sbmix_elem snd_dt019x_ctl_pc_speaker_vol =
- SB_SINGLE("Beep Volume", SB_DT019X_SPKR_DEV, 0, 7);
-static struct sbmix_elem snd_dt019x_ctl_line_play_vol =
- SB_DOUBLE("Line Playback Volume", SB_DT019X_LINE_DEV, SB_DT019X_LINE_DEV, 4,0, 15);
-static struct sbmix_elem snd_dt019x_ctl_pcm_play_switch =
- SB_DOUBLE("PCM Playback Switch", SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 2,1, 1);
-static struct sbmix_elem snd_dt019x_ctl_synth_play_switch =
- SB_DOUBLE("Synth Playback Switch", SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 4,3, 1);
-static struct sbmix_elem snd_dt019x_ctl_capture_source =
+static struct sbmix_elem snd_dt019x_controls[] = {
+ /* ALS4000 below has some parts which we might be lacking,
+ * e.g. snd_als4000_ctl_mono_playback_switch - check it! */
+ SB_DOUBLE("Master Playback Volume",
+ SB_DT019X_MASTER_DEV, SB_DT019X_MASTER_DEV, 4, 0, 15),
+ SB_DOUBLE("PCM Playback Switch",
+ SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 2, 1, 1),
+ SB_DOUBLE("PCM Playback Volume",
+ SB_DT019X_PCM_DEV, SB_DT019X_PCM_DEV, 4, 0, 15),
+ SB_DOUBLE("Synth Playback Switch",
+ SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 4, 3, 1),
+ SB_DOUBLE("Synth Playback Volume",
+ SB_DT019X_SYNTH_DEV, SB_DT019X_SYNTH_DEV, 4, 0, 15),
+ SB_DOUBLE("CD Playback Switch",
+ SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 2, 1, 1),
+ SB_DOUBLE("CD Playback Volume",
+ SB_DT019X_CD_DEV, SB_DT019X_CD_DEV, 4, 0, 15),
+ SB_SINGLE("Mic Playback Switch", SB_DSP4_OUTPUT_SW, 0, 1),
+ SB_SINGLE("Mic Playback Volume", SB_DT019X_MIC_DEV, 4, 7),
+ SB_SINGLE("Beep Volume", SB_DT019X_SPKR_DEV, 0, 7),
+ SB_DOUBLE("Line Playback Switch",
+ SB_DSP4_OUTPUT_SW, SB_DSP4_OUTPUT_SW, 4, 3, 1),
+ SB_DOUBLE("Line Playback Volume",
+ SB_DT019X_LINE_DEV, SB_DT019X_LINE_DEV, 4, 0, 15),
{
.name = "Capture Source",
.type = SB_MIX_CAPTURE_DT019X
- };
-
-static struct sbmix_elem *snd_dt019x_controls[] = {
- /* ALS4000 below has some parts which we might be lacking,
- * e.g. snd_als4000_ctl_mono_playback_switch - check it! */
- &snd_dt019x_ctl_master_play_vol,
- &snd_dt019x_ctl_pcm_play_vol,
- &snd_dt019x_ctl_synth_play_vol,
- &snd_dt019x_ctl_cd_play_vol,
- &snd_dt019x_ctl_mic_play_vol,
- &snd_dt019x_ctl_pc_speaker_vol,
- &snd_dt019x_ctl_line_play_vol,
- &snd_sb16_ctl_mic_play_switch,
- &snd_sb16_ctl_cd_play_switch,
- &snd_sb16_ctl_line_play_switch,
- &snd_dt019x_ctl_pcm_play_switch,
- &snd_dt019x_ctl_synth_play_switch,
- &snd_dt019x_ctl_capture_source
+ }
};
static unsigned char snd_dt019x_init_values[][2] = {
@@ -735,82 +671,37 @@ static unsigned char snd_dt019x_init_values[][2] = {
/*
* ALS4000 specific mixer elements
*/
-static struct sbmix_elem snd_als4000_ctl_master_mono_playback_switch =
- SB_SINGLE("Master Mono Playback Switch", SB_ALS4000_MONO_IO_CTRL, 5, 1);
-static struct sbmix_elem snd_als4k_ctl_master_mono_capture_route = {
+static struct sbmix_elem snd_als4000_controls[] = {
+ SB_DOUBLE("PCM Playback Switch",
+ SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 2, 1, 1),
+ SB_DOUBLE("Synth Playback Switch",
+ SB_DT019X_OUTPUT_SW2, SB_DT019X_OUTPUT_SW2, 4, 3, 1),
+ SB_SINGLE("Mic Boost (+20dB)", SB_ALS4000_MIC_IN_GAIN, 0, 0x03),
+ SB_SINGLE("Master Mono Playback Switch", SB_ALS4000_MONO_IO_CTRL, 5, 1),
+ {
.name = "Master Mono Capture Route",
.type = SB_MIX_MONO_CAPTURE_ALS4K
- };
-static struct sbmix_elem snd_als4000_ctl_mono_playback_switch =
- SB_SINGLE("Mono Playback Switch", SB_DT019X_OUTPUT_SW2, 0, 1);
-static struct sbmix_elem snd_als4000_ctl_mic_20db_boost =
- SB_SINGLE("Mic Boost (+20dB)", SB_ALS4000_MIC_IN_GAIN, 0, 0x03);
-static struct sbmix_elem snd_als4000_ctl_mixer_analog_loopback =
- SB_SINGLE("Analog Loopback Switch", SB_ALS4000_MIC_IN_GAIN, 7, 0x01);
-static struct sbmix_elem snd_als4000_ctl_mixer_digital_loopback =
+ },
+ SB_SINGLE("Mono Playback Switch", SB_DT019X_OUTPUT_SW2, 0, 1),
+ SB_SINGLE("Analog Loopback Switch", SB_ALS4000_MIC_IN_GAIN, 7, 0x01),
+ SB_SINGLE("3D Control - Switch", SB_ALS4000_3D_SND_FX, 6, 0x01),
SB_SINGLE("Digital Loopback Switch",
- SB_ALS4000_CR3_CONFIGURATION, 7, 0x01);
-/* FIXME: functionality of 3D controls might be swapped, I didn't find
- * a description of how to identify what is supposed to be what */
-static struct sbmix_elem snd_als4000_3d_control_switch =
- SB_SINGLE("3D Control - Switch", SB_ALS4000_3D_SND_FX, 6, 0x01);
-static struct sbmix_elem snd_als4000_3d_control_ratio =
- SB_SINGLE("3D Control - Level", SB_ALS4000_3D_SND_FX, 0, 0x07);
-static struct sbmix_elem snd_als4000_3d_control_freq =
+ SB_ALS4000_CR3_CONFIGURATION, 7, 0x01),
+ /* FIXME: functionality of 3D controls might be swapped, I didn't find
+ * a description of how to identify what is supposed to be what */
+ SB_SINGLE("3D Control - Level", SB_ALS4000_3D_SND_FX, 0, 0x07),
/* FIXME: maybe there's actually some standard 3D ctrl name for it?? */
- SB_SINGLE("3D Control - Freq", SB_ALS4000_3D_SND_FX, 4, 0x03);
-static struct sbmix_elem snd_als4000_3d_control_delay =
+ SB_SINGLE("3D Control - Freq", SB_ALS4000_3D_SND_FX, 4, 0x03),
/* FIXME: ALS4000a.pdf mentions BBD (Bucket Brigade Device) time delay,
* but what ALSA 3D attribute is that actually? "Center", "Depth",
* "Wide" or "Space" or even "Level"? Assuming "Wide" for now... */
- SB_SINGLE("3D Control - Wide", SB_ALS4000_3D_TIME_DELAY, 0, 0x0f);
-static struct sbmix_elem snd_als4000_3d_control_poweroff_switch =
- SB_SINGLE("3D PowerOff Switch", SB_ALS4000_3D_TIME_DELAY, 4, 0x01);
-static struct sbmix_elem snd_als4000_ctl_3db_freq_control_switch =
+ SB_SINGLE("3D Control - Wide", SB_ALS4000_3D_TIME_DELAY, 0, 0x0f),
+ SB_SINGLE("3D PowerOff Switch", SB_ALS4000_3D_TIME_DELAY, 4, 0x01),
SB_SINGLE("Master Playback 8kHz / 20kHz LPF Switch",
- SB_ALS4000_FMDAC, 5, 0x01);
+ SB_ALS4000_FMDAC, 5, 0x01),
#ifdef NOT_AVAILABLE
-static struct sbmix_elem snd_als4000_ctl_fmdac =
- SB_SINGLE("FMDAC Switch (Option ?)", SB_ALS4000_FMDAC, 0, 0x01);
-static struct sbmix_elem snd_als4000_ctl_qsound =
- SB_SINGLE("QSound Mode", SB_ALS4000_QSOUND, 1, 0x1f);
-#endif
-
-static struct sbmix_elem *snd_als4000_controls[] = {
- /* ALS4000a.PDF regs page */
- &snd_sb16_ctl_master_play_vol, /* MX30/31 12 */
- &snd_dt019x_ctl_pcm_play_switch, /* MX4C 16 */
- &snd_sb16_ctl_pcm_play_vol, /* MX32/33 12 */
- &snd_sb16_ctl_synth_capture_route, /* MX3D/3E 14 */
- &snd_dt019x_ctl_synth_play_switch, /* MX4C 16 */
- &snd_sb16_ctl_synth_play_vol, /* MX34/35 12/13 */
- &snd_sb16_ctl_cd_capture_route, /* MX3D/3E 14 */
- &snd_sb16_ctl_cd_play_switch, /* MX3C 14 */
- &snd_sb16_ctl_cd_play_vol, /* MX36/37 13 */
- &snd_sb16_ctl_line_capture_route, /* MX3D/3E 14 */
- &snd_sb16_ctl_line_play_switch, /* MX3C 14 */
- &snd_sb16_ctl_line_play_vol, /* MX38/39 13 */
- &snd_sb16_ctl_mic_capture_route, /* MX3D/3E 14 */
- &snd_als4000_ctl_mic_20db_boost, /* MX4D 16 */
- &snd_sb16_ctl_mic_play_switch, /* MX3C 14 */
- &snd_sb16_ctl_mic_play_vol, /* MX3A 13 */
- &snd_sb16_ctl_pc_speaker_vol, /* MX3B 14 */
- &snd_sb16_ctl_capture_vol, /* MX3F/40 15 */
- &snd_sb16_ctl_play_vol, /* MX41/42 15 */
- &snd_als4000_ctl_master_mono_playback_switch, /* MX4C 16 */
- &snd_als4k_ctl_master_mono_capture_route, /* MX4B 16 */
- &snd_als4000_ctl_mono_playback_switch, /* MX4C 16 */
- &snd_als4000_ctl_mixer_analog_loopback, /* MX4D 16 */
- &snd_als4000_ctl_mixer_digital_loopback, /* CR3 21 */
- &snd_als4000_3d_control_switch, /* MX50 17 */
- &snd_als4000_3d_control_ratio, /* MX50 17 */
- &snd_als4000_3d_control_freq, /* MX50 17 */
- &snd_als4000_3d_control_delay, /* MX51 18 */
- &snd_als4000_3d_control_poweroff_switch, /* MX51 18 */
- &snd_als4000_ctl_3db_freq_control_switch, /* MX4F 17 */
-#ifdef NOT_AVAILABLE
- &snd_als4000_ctl_fmdac,
- &snd_als4000_ctl_qsound,
+ SB_SINGLE("FMDAC Switch (Option ?)", SB_ALS4000_FMDAC, 0, 0x01),
+ SB_SINGLE("QSound Mode", SB_ALS4000_QSOUND, 1, 0x1f),
#endif
};
@@ -829,11 +720,10 @@ static unsigned char snd_als4000_init_values[][2] = {
{ SB_ALS4000_MIC_IN_GAIN, 0 },
};
-
/*
*/
static int snd_sbmixer_init(struct snd_sb *chip,
- struct sbmix_elem **controls,
+ struct sbmix_elem *controls,
int controls_count,
unsigned char map[][2],
int map_count,
@@ -856,7 +746,8 @@ static int snd_sbmixer_init(struct snd_sb *chip,
}
for (idx = 0; idx < controls_count; idx++) {
- if ((err = snd_sbmixer_add_ctl_elem(chip, controls[idx])) < 0)
+ err = snd_sbmixer_add_ctl_elem(chip, &controls[idx]);
+ if (err < 0)
return err;
}
snd_component_add(card, name);
@@ -908,6 +799,15 @@ int snd_sbmixer_new(struct snd_sb *chip)
return err;
break;
case SB_HW_ALS4000:
+ /* use only the first 16 controls from SB16 */
+ err = snd_sbmixer_init(chip,
+ snd_sb16_controls,
+ 16,
+ snd_sb16_init_values,
+ ARRAY_SIZE(snd_sb16_init_values),
+ "ALS4000");
+ if (err < 0)
+ return err;
if ((err = snd_sbmixer_init(chip,
snd_als4000_controls,
ARRAY_SIZE(snd_als4000_controls),
diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c
index 5b9d6c1..9191b32 100644
--- a/sound/isa/wss/wss_lib.c
+++ b/sound/isa/wss/wss_lib.c
@@ -2014,6 +2014,7 @@ static int snd_wss_info_mux(struct snd_kcontrol *kcontrol,
case WSS_HW_INTERWAVE:
ptexts = gusmax_texts;
break;
+ case WSS_HW_OPTI93X:
case WSS_HW_OPL3SA2:
ptexts = opl3sa_texts;
break;
@@ -2246,54 +2247,12 @@ WSS_SINGLE("Beep Bypass Playback Switch", 0,
CS4231_MONO_CTRL, 5, 1, 0),
};
-static struct snd_kcontrol_new snd_opti93x_controls[] = {
-WSS_DOUBLE("Master Playback Switch", 0,
- OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1),
-WSS_DOUBLE_TLV("Master Playback Volume", 0,
- OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1,
- db_scale_6bit),
-WSS_DOUBLE("PCM Playback Switch", 0,
- CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1),
-WSS_DOUBLE("PCM Playback Volume", 0,
- CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1),
-WSS_DOUBLE("FM Playback Switch", 0,
- CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1),
-WSS_DOUBLE("FM Playback Volume", 0,
- CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1),
-WSS_DOUBLE("Line Playback Switch", 0,
- CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1),
-WSS_DOUBLE("Line Playback Volume", 0,
- CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1),
-WSS_DOUBLE("Mic Playback Switch", 0,
- OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1),
-WSS_DOUBLE("Mic Playback Volume", 0,
- OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1),
-WSS_DOUBLE("Mic Boost", 0,
- CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0),
-WSS_DOUBLE("CD Playback Switch", 0,
- CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1),
-WSS_DOUBLE("CD Playback Volume", 0,
- CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1),
-WSS_DOUBLE("Aux Playback Switch", 0,
- OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1),
-WSS_DOUBLE("Aux Playback Volume", 0,
- OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1),
-WSS_DOUBLE("Capture Volume", 0,
- CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0),
-{
- .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
- .name = "Capture Source",
- .info = snd_wss_info_mux,
- .get = snd_wss_get_mux,
- .put = snd_wss_put_mux,
-}
-};
-
int snd_wss_mixer(struct snd_wss *chip)
{
struct snd_card *card;
unsigned int idx;
int err;
+ int count = ARRAY_SIZE(snd_wss_controls);
if (snd_BUG_ON(!chip || !chip->pcm))
return -EINVAL;
@@ -2302,28 +2261,19 @@ int snd_wss_mixer(struct snd_wss *chip)
strcpy(card->mixername, chip->pcm->name);
- if (chip->hardware == WSS_HW_OPTI93X)
- for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) {
- err = snd_ctl_add(card,
- snd_ctl_new1(&snd_opti93x_controls[idx],
- chip));
- if (err < 0)
- return err;
- }
- else {
- int count = ARRAY_SIZE(snd_wss_controls);
-
- /* Use only the first 11 entries on AD1848 */
- if (chip->hardware & WSS_HW_AD1848_MASK)
- count = 11;
-
- for (idx = 0; idx < count; idx++) {
- err = snd_ctl_add(card,
- snd_ctl_new1(&snd_wss_controls[idx],
- chip));
- if (err < 0)
- return err;
- }
+ /* Use only the first 11 entries on AD1848 */
+ if (chip->hardware & WSS_HW_AD1848_MASK)
+ count = 11;
+ /* There is no loopback on OPTI93X */
+ else if (chip->hardware == WSS_HW_OPTI93X)
+ count = 9;
+
+ for (idx = 0; idx < count; idx++) {
+ err = snd_ctl_add(card,
+ snd_ctl_new1(&snd_wss_controls[idx],
+ chip));
+ if (err < 0)
+ return err;
}
return 0;
}
diff --git a/sound/oss/Kconfig b/sound/oss/Kconfig
index 135a2b7..a513651 100644
--- a/sound/oss/Kconfig
+++ b/sound/oss/Kconfig
@@ -1,5 +1,3 @@
-# drivers/sound/Config.in
-#
# 18 Apr 1998, Michael Elizabeth Chastain, <mailto:mec@shout.net>
# More hacking for modularisation.
#
diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c
index 06e9e88..bb14e4c 100644
--- a/sound/oss/dmasound/dmasound_paula.c
+++ b/sound/oss/dmasound/dmasound_paula.c
@@ -657,7 +657,7 @@ static int AmiStateInfo(char *buffer, size_t space)
len += sprintf(buffer+len, "\tsound.volume_right = %d [0...64]\n",
dmasound.volume_right);
if (len >= space) {
- printk(KERN_ERR "dmasound_paula: overlowed state buffer alloc.\n") ;
+ printk(KERN_ERR "dmasound_paula: overflowed state buffer alloc.\n") ;
len = space ;
}
return len;
diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c
index 15523e6..0470461 100644
--- a/sound/pci/ca0106/ca0106_proc.c
+++ b/sound/pci/ca0106/ca0106_proc.c
@@ -233,7 +233,7 @@ static void snd_ca0106_proc_dump_iec958( struct snd_info_buffer *buffer, u32 val
snd_iprintf(buffer, "user-defined\n");
break;
default:
- snd_iprintf(buffer, "unkown\n");
+ snd_iprintf(buffer, "unknown\n");
break;
}
snd_iprintf(buffer, "Sample Bits: ");
diff --git a/sound/pci/cs46xx/imgs/cwcdma.asp b/sound/pci/cs46xx/imgs/cwcdma.asp
index 09d24c7..a65e119 100644
--- a/sound/pci/cs46xx/imgs/cwcdma.asp
+++ b/sound/pci/cs46xx/imgs/cwcdma.asp
@@ -26,10 +26,11 @@
//
//
// The purpose of this code is very simple: make it possible to tranfser
-// the samples 'as they are' with no alteration from a PCMreader SCB (DMA from host)
-// to any other SCB. This is useful for AC3 throug SPDIF. SRC (source rate converters)
-// task always alters the samples in some how, however it's from 48khz -> 48khz. The
-// alterations are not audible, but AC3 wont work.
+// the samples 'as they are' with no alteration from a PCMreader
+// SCB (DMA from host) to any other SCB. This is useful for AC3 through SPDIF.
+// SRC (source rate converters) task always alters the samples in somehow,
+// however it's from 48khz -> 48khz.
+// The alterations are not audible, but AC3 wont work.
//
// ...
// |
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index 6b8ae7b..1d369ff 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -184,7 +184,7 @@ MODULE_PARM_DESC(enable, "Enable the EMU10K1X soundcard.");
* The hardware has 3 channels for playback and 1 for capture.
* - channel 0 is the front channel
* - channel 1 is the rear channel
- * - channel 2 is the center/lfe chanel
+ * - channel 2 is the center/lfe channel
* Volume is controlled by the AC97 for the front and rear channels by
* the PCM Playback Volume, Sigmatel Surround Playback Volume and
* Surround Playback Volume. The Sigmatel 4-Speaker Stereo switch affects
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 2439e84..4b200da 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -938,7 +938,7 @@ static void init_input(struct hda_codec *codec)
coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */
if (is_active_pin(codec, CS_DMIC1_PIN_NID))
coef |= 0x1800; /* DMIC1 enable 2 channels, disable GPIO0
- * No effect if SPDIF_OUT2 is slected in
+ * No effect if SPDIF_OUT2 is selected in
* IDX_SPDIF_CTL.
*/
cs_vendor_coef_set(codec, IDX_ADC_CFG, coef);
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index 85c81fe..a45c116 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -66,7 +66,7 @@ struct cmi_spec {
struct hda_pcm pcm_rec[2]; /* PCM information */
- /* pin deafault configuration */
+ /* pin default configuration */
hda_nid_t pin_nid[NUM_PINS];
unsigned int def_conf[NUM_PINS];
unsigned int pin_def_confs;
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index deecdd2..888b631 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6621,7 +6621,7 @@ static struct hda_input_mux alc889A_mb31_capture_source = {
/* Front Mic (0x01) unused */
{ "Line", 0x2 },
/* Line 2 (0x03) unused */
- /* CD (0x04) unsused? */
+ /* CD (0x04) unused? */
},
};
diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c
index 0c9413d..98bc3b7 100644
--- a/sound/pci/ice1712/juli.c
+++ b/sound/pci/ice1712/juli.c
@@ -380,7 +380,7 @@ static struct snd_kcontrol_new juli_mute_controls[] __devinitdata = {
* inputs) are fed from Xilinx.
*
* I even checked traces on board and coded a support in driver for
- * an alternative possiblity - the unused I2S ICE output channels
+ * an alternative possibility - the unused I2S ICE output channels
* switched to HW-IN/SPDIF-IN and providing the monitoring signal to
* the DAC - to no avail. The I2S outputs seem to be unconnected.
*
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 0dce331..a1b10d1 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -3017,7 +3017,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
insel = "Coaxial";
break;
default:
- insel = "Unkown";
+ insel = "Unknown";
}
switch (hdspm->control_register & HDSPM_SyncRefMask) {
@@ -3028,7 +3028,7 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
syncref = "MADI";
break;
default:
- syncref = "Unkown";
+ syncref = "Unknown";
}
snd_iprintf(buffer, "Inputsel = %s, SyncRef = %s\n", insel,
syncref);
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c
index 64b8599..7717e01 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c
@@ -131,7 +131,7 @@ static int snd_pdacf_probe(struct pcmcia_device *link)
return err;
}
- snd_card_set_dev(card, &handle_to_dev(link));
+ snd_card_set_dev(card, &link->dev);
pdacf->index = i;
card_list[i] = card;
@@ -142,12 +142,10 @@ static int snd_pdacf_probe(struct pcmcia_device *link)
link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO;
link->io.NumPorts1 = 16;
- link->irq.Attributes = IRQ_TYPE_EXCLUSIVE | IRQ_HANDLE_PRESENT | IRQ_FORCED_PULSE;
+ link->irq.Attributes = IRQ_TYPE_EXCLUSIVE | IRQ_FORCED_PULSE;
// link->irq.Attributes = IRQ_TYPE_DYNAMIC_SHARING|IRQ_FIRST_SHARED;
- link->irq.IRQInfo1 = 0 /* | IRQ_LEVEL_ID */;
link->irq.Handler = pdacf_interrupt;
- link->irq.Instance = pdacf;
link->conf.Attributes = CONF_ENABLE_IRQ;
link->conf.IntType = INT_MEMORY_AND_IO;
link->conf.ConfigIndex = 1;
diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c
index 1492744..7be3b33 100644
--- a/sound/pcmcia/vx/vxpocket.c
+++ b/sound/pcmcia/vx/vxpocket.c
@@ -161,11 +161,9 @@ static int snd_vxpocket_new(struct snd_card *card, int ibl,
link->io.Attributes1 = IO_DATA_PATH_WIDTH_AUTO;
link->io.NumPorts1 = 16;
- link->irq.Attributes = IRQ_TYPE_EXCLUSIVE | IRQ_HANDLE_PRESENT;
+ link->irq.Attributes = IRQ_TYPE_EXCLUSIVE;
- link->irq.IRQInfo1 = IRQ_LEVEL_ID;
link->irq.Handler = &snd_vx_irq_handler;
- link->irq.Instance = chip;
link->conf.Attributes = CONF_ENABLE_IRQ;
link->conf.IntType = INT_MEMORY_AND_IO;
@@ -244,7 +242,7 @@ static int vxpocket_config(struct pcmcia_device *link)
if (ret)
goto failed;
- chip->dev = &handle_to_dev(link);
+ chip->dev = &link->dev;
snd_card_set_dev(chip->card, chip->dev);
if (snd_vxpocket_assign_resources(chip, link->io.BasePort1, link->irq.AssignedIRQ) < 0)
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index aa40d98..3e99fe5 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -101,7 +101,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg,
pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value);
if (reg >= UDA134X_REGS_NUM) {
- printk(KERN_ERR "%s unkown register: reg: %u",
+ printk(KERN_ERR "%s unknown register: reg: %u",
__func__, reg);
return -EINVAL;
}
@@ -552,7 +552,7 @@ static int uda134x_soc_probe(struct platform_device *pdev)
ARRAY_SIZE(uda1341_snd_controls));
break;
default:
- printk(KERN_ERR "%s unkown codec type: %d",
+ printk(KERN_ERR "%s unknown codec type: %d",
__func__, pd->model);
return -EINVAL;
}
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index b8cae17..ce5515e 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -607,7 +607,7 @@ SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1,
SOC_SINGLE("DRC Switch", WM8903_DRC_0, 15, 1, 0),
SOC_ENUM("DRC Compressor Slope R0", drc_slope_r0),
SOC_ENUM("DRC Compressor Slope R1", drc_slope_r1),
-SOC_SINGLE_TLV("DRC Compressor Threashold Volume", WM8903_DRC_3, 5, 124, 1,
+SOC_SINGLE_TLV("DRC Compressor Threshold Volume", WM8903_DRC_3, 5, 124, 1,
drc_tlv_thresh),
SOC_SINGLE_TLV("DRC Volume", WM8903_DRC_3, 0, 30, 1, drc_tlv_amp),
SOC_SINGLE_TLV("DRC Minimum Gain Volume", WM8903_DRC_1, 2, 3, 1, drc_tlv_min),
@@ -617,11 +617,11 @@ SOC_ENUM("DRC Decay Rate", drc_decay),
SOC_ENUM("DRC FF Delay", drc_ff_delay),
SOC_SINGLE("DRC Anticlip Switch", WM8903_DRC_0, 1, 1, 0),
SOC_SINGLE("DRC QR Switch", WM8903_DRC_0, 2, 1, 0),
-SOC_SINGLE_TLV("DRC QR Threashold Volume", WM8903_DRC_0, 6, 3, 0, drc_tlv_max),
+SOC_SINGLE_TLV("DRC QR Threshold Volume", WM8903_DRC_0, 6, 3, 0, drc_tlv_max),
SOC_ENUM("DRC QR Decay Rate", drc_qr_decay),
SOC_SINGLE("DRC Smoothing Switch", WM8903_DRC_0, 3, 1, 0),
SOC_SINGLE("DRC Smoothing Hysteresis Switch", WM8903_DRC_0, 0, 1, 0),
-SOC_ENUM("DRC Smoothing Threashold", drc_smoothing),
+SOC_ENUM("DRC Smoothing Threshold", drc_smoothing),
SOC_SINGLE_TLV("DRC Startup Volume", WM8903_DRC_0, 6, 18, 0, drc_tlv_startup),
SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8903_ADC_DIGITAL_VOLUME_LEFT,
diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c
index 5e32f2e..2981afa 100644
--- a/sound/soc/codecs/wm8993.c
+++ b/sound/soc/codecs/wm8993.c
@@ -689,7 +689,7 @@ SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8993_DIGITAL_SIDE_TONE,
SOC_SINGLE("DRC Switch", WM8993_DRC_CONTROL_1, 15, 1, 0),
SOC_ENUM("DRC Path", drc_path),
-SOC_SINGLE_TLV("DRC Compressor Threashold Volume", WM8993_DRC_CONTROL_2,
+SOC_SINGLE_TLV("DRC Compressor Threshold Volume", WM8993_DRC_CONTROL_2,
2, 60, 1, drc_comp_threash),
SOC_SINGLE_TLV("DRC Compressor Amplitude Volume", WM8993_DRC_CONTROL_3,
11, 30, 1, drc_comp_amp),
@@ -709,7 +709,7 @@ SOC_SINGLE_TLV("DRC Quick Release Volume", WM8993_DRC_CONTROL_3, 2, 3, 0,
SOC_ENUM("DRC Quick Release Rate", drc_qr_rate),
SOC_SINGLE("DRC Smoothing Switch", WM8993_DRC_CONTROL_1, 11, 1, 0),
SOC_SINGLE("DRC Smoothing Hysteresis Switch", WM8993_DRC_CONTROL_1, 8, 1, 0),
-SOC_ENUM("DRC Smoothing Hysteresis Threashold", drc_smooth),
+SOC_ENUM("DRC Smoothing Hysteresis Threshold", drc_smooth),
SOC_SINGLE_TLV("DRC Startup Volume", WM8993_DRC_CONTROL_4, 8, 18, 0,
drc_startup_tlv),
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index ae0fc9b..b0f618e 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -31,8 +31,8 @@
#include <asm/mach-types.h>
-#include <mach/board-ams-delta.h>
-#include <mach/mcbsp.h>
+#include <plat/board-ams-delta.h>
+#include <plat/mcbsp.h>
#include "omap-mcbsp.h"
#include "omap-pcm.h"
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 0a50593..08e09d7 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -32,7 +32,7 @@
#include <asm/mach-types.h>
#include <mach/hardware.h>
#include <linux/gpio.h>
-#include <mach/mcbsp.h>
+#include <plat/mcbsp.h>
#include "omap-mcbsp.h"
#include "omap-pcm.h"
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 45be942..6bbbd2a 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -31,9 +31,9 @@
#include <sound/initval.h>
#include <sound/soc.h>
-#include <mach/control.h>
-#include <mach/dma.h>
-#include <mach/mcbsp.h>
+#include <plat/control.h>
+#include <plat/dma.h>
+#include <plat/mcbsp.h>
#include "omap-mcbsp.h"
#include "omap-pcm.h"
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 6a829ee..9db2770 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -28,7 +28,7 @@
#include <sound/pcm_params.h>
#include <sound/soc.h>
-#include <mach/dma.h>
+#include <plat/dma.h>
#include "omap-pcm.h"
static const struct snd_pcm_hardware omap_pcm_hardware = {
diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c
index 027e1a4..c7adea3 100644
--- a/sound/soc/omap/omap2evm.c
+++ b/sound/soc/omap/omap2evm.c
@@ -31,7 +31,7 @@
#include <asm/mach-types.h>
#include <mach/hardware.h>
#include <mach/gpio.h>
-#include <mach/mcbsp.h>
+#include <plat/mcbsp.h>
#include "omap-mcbsp.h"
#include "omap-pcm.h"
diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c
index b0cff9f..d88ad5c 100644
--- a/sound/soc/omap/omap3beagle.c
+++ b/sound/soc/omap/omap3beagle.c
@@ -29,7 +29,7 @@
#include <asm/mach-types.h>
#include <mach/hardware.h>
#include <mach/gpio.h>
-#include <mach/mcbsp.h>
+#include <plat/mcbsp.h>
#include "omap-mcbsp.h"
#include "omap-pcm.h"
diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c
index f484dcd..dfcb344 100644
--- a/sound/soc/omap/omap3evm.c
+++ b/sound/soc/omap/omap3evm.c
@@ -27,7 +27,7 @@
#include <asm/mach-types.h>
#include <mach/hardware.h>
#include <mach/gpio.h>
-#include <mach/mcbsp.h>
+#include <plat/mcbsp.h>
#include "omap-mcbsp.h"
#include "omap-pcm.h"
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index a4e149b7..498ca2e 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -31,7 +31,7 @@
#include <asm/mach-types.h>
#include <mach/hardware.h>
#include <linux/gpio.h>
-#include <mach/mcbsp.h>
+#include <plat/mcbsp.h>
#include "omap-mcbsp.h"
#include "omap-pcm.h"
diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c
index 97a4d630..c25f527 100644
--- a/sound/soc/omap/overo.c
+++ b/sound/soc/omap/overo.c
@@ -29,7 +29,7 @@
#include <asm/mach-types.h>
#include <mach/hardware.h>
#include <mach/gpio.h>
-#include <mach/mcbsp.h>
+#include <plat/mcbsp.h>
#include "omap-mcbsp.h"
#include "omap-pcm.h"
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index 4a3f62d..c071f96 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -34,7 +34,7 @@
#include <asm/mach-types.h>
#include <mach/hardware.h>
#include <mach/gpio.h>
-#include <mach/mcbsp.h>
+#include <plat/mcbsp.h>
#include "omap-mcbsp.h"
#include "omap-pcm.h"
diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c
index f90b45f..f90a2ac 100644
--- a/sound/soc/omap/zoom2.c
+++ b/sound/soc/omap/zoom2.c
@@ -29,7 +29,7 @@
#include <asm/mach-types.h>
#include <mach/hardware.h>
#include <mach/gpio.h>
-#include <mach/mcbsp.h>
+#include <plat/mcbsp.h>
#include "omap-mcbsp.h"
#include "omap-pcm.h"
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c
index 507b2ed..d441c3b 100644
--- a/sound/soc/s3c24xx/s3c24xx_simtec.c
+++ b/sound/soc/s3c24xx/s3c24xx_simtec.c
@@ -270,7 +270,7 @@ static int attach_gpio_amp(struct device *dev,
gpio_direction_output(pd->amp_gain[1], 0);
}
- /* note, curently we assume GPA0 isn't valid amp */
+ /* note, currently we assume GPA0 isn't valid amp */
if (pdata->amp_gpio > 0) {
ret = gpio_request(pd->amp_gpio, "gpio-amp");
if (ret) {
diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c
index 0eb1722..1d61109 100644
--- a/sound/soc/s6000/s6000-pcm.c
+++ b/sound/soc/s6000/s6000-pcm.c
@@ -196,7 +196,7 @@ static int s6000_pcm_start(struct snd_pcm_substream *substream)
0 /* destination skip after chunk (impossible) */,
4 /* 16 byte burst size */,
-1 /* don't conserve bandwidth */,
- 0 /* low watermark irq descriptor theshold */,
+ 0 /* low watermark irq descriptor threshold */,
0 /* disable hardware timestamps */,
1 /* enable channel */);
diff --git a/sound/sound_core.c b/sound/sound_core.c
index 49c9981..dbca7c9 100644
--- a/sound/sound_core.c
+++ b/sound/sound_core.c
@@ -353,7 +353,7 @@ static struct sound_unit *chains[SOUND_STEP];
* @dev: device pointer
*
* Allocate a special sound device by minor number from the sound
- * subsystem. The allocated number is returned on succes. On failure
+ * subsystem. The allocated number is returned on success. On failure
* a negative error code is returned.
*/
diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c
index 63c8f45..67c9123 100644
--- a/sound/synth/emux/soundfont.c
+++ b/sound/synth/emux/soundfont.c
@@ -374,7 +374,7 @@ sf_zone_new(struct snd_sf_list *sflist, struct snd_soundfont *sf)
/*
- * increment sample couter
+ * increment sample counter
*/
static void
set_sample_counter(struct snd_sf_list *sflist, struct snd_soundfont *sf,
OpenPOWER on IntegriCloud