diff options
Diffstat (limited to 'sound')
40 files changed, 359 insertions, 710 deletions
diff --git a/sound/Kconfig b/sound/Kconfig index b3e53e616ec9..fcad760f5691 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 02fe81ca88fd..194af3b01e13 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 b906b9a1a81e..c73d30c4f462 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 5fd52e4d7079..20becc89f6f6 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 93fa6720d197..cc15d1d65a22 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 80f5b1af9be8..000000000000 --- 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 6123c7531110..b865e45a8f9b 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 d8eac3f28947..a4af53b5c1cf 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 318ff0c823e7..8cfc41fbe368 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 5b9d6c18bc45..9191b32d9130 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 135a2b77cc4a..a513651fa149 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 06e9e88e4c05..bb14e4c67e89 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 15523e60351c..0470461cc03e 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 09d24c76f034..a65e1193c89a 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 6b8ae7b5cd54..1d369ff73805 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 2439e84dcb21..4b200da1bd18 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 85c81feb10cf..a45c1169762b 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 deecdd2d5d37..888b6313eeca 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 0c9413d5341b..98bc3b7681b5 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 0dce331a2a3b..a1b10d1a384d 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 64b859925c0b..7717e01fc071 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 1492744ad67f..7be3b3357045 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 aa40d985138f..3e99fe5131dd 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 b8cae1758642..ce5515e3f2b0 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 5e32f2ed5fc2..2981afae842c 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 ae0fc9b135d4..b0f618e44840 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 0a505938e42b..08e09d72790f 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 45be94201c89..6bbbd2ab0ee7 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 6a829eef2a4f..9db2770e9640 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 027e1a40f8a1..c7adea38274c 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 b0cff9f33b7e..d88ad5ca526c 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 f484dcd63408..dfcb344092e4 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 a4e149b7f0eb..498ca2e03519 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 97a4d6308bd6..c25f5276ad6f 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 4a3f62d1f295..c071f9603a38 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 f90b45f56220..f90a2ac888cf 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 507b2ed5d58b..d441c3b64631 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 0eb1722f6581..1d61109e09fa 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 49c998186592..dbca7c909a31 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 63c8f45c0c22..67c91230c197 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, |