From 14ff3e78304e3f7fe18f950c3aa0686e6800b3fb Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Thu, 10 Dec 2009 20:39:28 +0100 Subject: ALSA: dt019x: merge into the als100 driver The als100 driver is so similar to the dt019x/als007 driver that one driver's source can be used for both drivers with only few changes. Merge the dt019x driver into the als100. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/Kconfig | 21 +--- sound/isa/Makefile | 2 - sound/isa/als100.c | 121 ++++++++++++++------ sound/isa/dt019x.c | 321 ----------------------------------------------------- 4 files changed, 90 insertions(+), 375 deletions(-) delete mode 100644 sound/isa/dt019x.c diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 02fe81c..194af3b 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 + Copyright (C) 1999-2002 by Massimo Piccioni Thanks to Pierfrancesco 'qM2' Passerini. + Generalised for soundcards based on DT-0196 and ALS-007 chips + by Jonathan Woithe : 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 "); -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 "); +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/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 - - Generalised for soundcards based on DT-0196 and ALS-007 chips - by Jonathan Woithe : 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 -#include -#include -#include -#include -#include -#include -#include -#include - -#define PFX "dt019x: " - -MODULE_AUTHOR("Massimo Piccioni "); -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) -- cgit v1.1 From b2e8d7dab9d82be3851b8cbcc1ab64b1b2575844 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Thu, 10 Dec 2009 20:40:18 +0100 Subject: ALSA: opti93x: move controls definitions to opti93x driver Move OPTi93x controls definitions to the opti93x driver from the common wss-lib library module. These controls are used only by the opti93x driver. Also, fix capture source names. They are the same as opl3sa2 names. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/opti92x-ad1848.c | 85 ++++++++++++++++++++++++++++++++++++++ sound/isa/wss/wss_lib.c | 80 +++++++---------------------------- 2 files changed, 100 insertions(+), 65 deletions(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 106be6e..ea4a671 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -33,6 +33,7 @@ #include #include #include +#include #include #include #include @@ -546,6 +547,85 @@ __skip_mpu: #ifdef OPTi93X +static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 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 Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 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("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), +}; + +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_wss *codec = dev_id; @@ -752,6 +832,11 @@ 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/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; } -- cgit v1.1 From e9d0a803c127e2e30afb0df780ccb3af4e2adb28 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sat, 12 Dec 2009 09:51:03 +0100 Subject: ALSA: opti93x: use dB scale for mixer controls Add dB scale for mixer controls. Fix dB scale for Master Volume control. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/opti92x-ad1848.c | 36 ++++++++++++++++++++++-------------- 1 file changed, 22 insertions(+), 14 deletions(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index ea4a671..b0ea310 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -547,32 +547,40 @@ __skip_mpu: #ifdef OPTi93X -static const DECLARE_TLV_DB_SCALE(db_scale_6bit, -9450, 150, 0); +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_6bit), -WSS_DOUBLE("PCM Playback Volume", 0, - CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), -WSS_DOUBLE("FM Playback Volume", 0, - CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 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("Line Playback Volume", 0, - CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 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("Mic Playback Volume", 0, - OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), -WSS_DOUBLE("CD Playback Volume", 0, - CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 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("Aux Playback Volume", 0, - OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 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) -- cgit v1.1 From 74c2b45b714e49b427584b4bd8f44f1a24d82d9c Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 13 Dec 2009 21:13:44 +0100 Subject: ALSA: sb_mixer: convert pointer tables to mixer control tables Convert table of pointers to mixer controls into tables of the mixer controls. It saves about 20% of the snd-sb-common module size reported by lsmod. The als4000 uses part of sb16's control table. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/sb/sb_mixer.c | 330 +++++++++++++++++------------------------------- 1 file changed, 115 insertions(+), 215 deletions(-) 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), -- cgit v1.1 From 63978ab3e3e963db28093b53bb4598f2702e1ad7 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 14 Dec 2009 12:48:35 +0100 Subject: sound: add Edirol UA-101 support Add experimental support for the Edirol UA-101 audio/MIDI interface. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 7 + sound/usb/Kconfig | 12 + sound/usb/Makefile | 2 + sound/usb/ua101.c | 1457 +++++++++++++++++++++++ sound/usb/usbaudio.c | 54 - sound/usb/usbaudio.h | 1 - sound/usb/usbquirks.h | 31 - 7 files changed, 1478 insertions(+), 86 deletions(-) create mode 100644 sound/usb/ua101.c diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 8923597..7a0a4a9 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1791,6 +1791,13 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. The power-management is supported. + Module snd-ua101 + ---------------- + + Module for the Edirol UA-101 audio/MIDI interface. + + This module supports multiple devices, autoprobe and hotplugging. + Module snd-usb-audio -------------------- diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 73525c0..8c29258 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -21,6 +21,18 @@ config SND_USB_AUDIO To compile this driver as a module, choose M here: the module will be called snd-usb-audio. +config SND_USB_UA101 + tristate "Edirol UA-101 driver (EXPERIMENTAL)" + depends on EXPERIMENTAL + select SND_PCM + select SND_RAWMIDI + help + Say Y here to include support for the Edirol UA-101 audio/MIDI + interface. + + To compile this driver as a module, choose M here: the module + will be called snd-ua101. + config SND_USB_USX2Y tristate "Tascam US-122, US-224 and US-428 USB driver" depends on X86 || PPC || ALPHA diff --git a/sound/usb/Makefile b/sound/usb/Makefile index abb288b..5bf64ae 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -4,9 +4,11 @@ snd-usb-audio-objs := usbaudio.o usbmixer.o snd-usb-lib-objs := usbmidi.o +snd-ua101-objs := ua101.o # Toplevel Module Dependency obj-$(CONFIG_SND_USB_AUDIO) += snd-usb-audio.o snd-usb-lib.o +obj-$(CONFIG_SND_USB_UA101) += snd-ua101.o snd-usb-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usb-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usb-lib.o diff --git a/sound/usb/ua101.c b/sound/usb/ua101.c new file mode 100644 index 0000000..ab9f8a2 --- /dev/null +++ b/sound/usb/ua101.c @@ -0,0 +1,1457 @@ +/* + * Edirol UA-101 driver + * Copyright (c) Clemens Ladisch + * + * This driver is free software: you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver 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 driver. If not, see . + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "usbaudio.h" + +MODULE_DESCRIPTION("Edirol UA-101 driver"); +MODULE_AUTHOR("Clemens Ladisch "); +MODULE_LICENSE("GPL v2"); +MODULE_SUPPORTED_DEVICE("{{Edirol,UA-101}}"); + +/* I use my UA-1A for testing because I don't have a UA-101 ... */ +#define UA1A_HACK + +/* + * Should not be lower than the minimum scheduling delay of the host + * controller. Some Intel controllers need more than one frame; as long as + * that driver doesn't tell us about this, use 1.5 frames just to be sure. + */ +#define MIN_QUEUE_LENGTH 12 +/* Somewhat random. */ +#define MAX_QUEUE_LENGTH 30 +/* + * This magic value optimizes memory usage efficiency for the UA-101's packet + * sizes at all sample rates, taking into account the stupid cache pool sizes + * that usb_buffer_alloc() uses. + */ +#define DEFAULT_QUEUE_LENGTH 21 + +#define MAX_PACKET_SIZE 672 /* hardware specific */ +#define MAX_MEMORY_BUFFERS DIV_ROUND_UP(MAX_QUEUE_LENGTH, \ + PAGE_SIZE / MAX_PACKET_SIZE) + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static unsigned int queue_length = 21; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "card index"); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string"); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "enable card"); +module_param(queue_length, uint, 0644); +MODULE_PARM_DESC(queue_length, "USB queue length in microframes, " + __stringify(MIN_QUEUE_LENGTH)"-"__stringify(MAX_QUEUE_LENGTH)); + +enum { + INTF_PLAYBACK, + INTF_CAPTURE, + INTF_MIDI, + + INTF_COUNT +}; + +/* bits in struct ua101::states */ +enum { + USB_CAPTURE_RUNNING, + USB_PLAYBACK_RUNNING, + ALSA_CAPTURE_OPEN, + ALSA_PLAYBACK_OPEN, + ALSA_CAPTURE_RUNNING, + ALSA_PLAYBACK_RUNNING, + CAPTURE_URB_COMPLETED, + PLAYBACK_URB_COMPLETED, + DISCONNECTED, +}; + +struct ua101 { + struct usb_device *dev; + struct snd_card *card; + struct usb_interface *intf[INTF_COUNT]; + int card_index; + struct snd_pcm *pcm; + struct list_head midi_list; + u64 format_bit; + unsigned int rate; + unsigned int packets_per_second; + spinlock_t lock; + struct mutex mutex; + unsigned long states; + + /* FIFO to synchronize playback rate to capture rate */ + unsigned int rate_feedback_start; + unsigned int rate_feedback_count; + u8 rate_feedback[MAX_QUEUE_LENGTH]; + + struct list_head ready_playback_urbs; + struct tasklet_struct playback_tasklet; + wait_queue_head_t alsa_capture_wait; + wait_queue_head_t rate_feedback_wait; + wait_queue_head_t alsa_playback_wait; + struct ua101_stream { + struct snd_pcm_substream *substream; + unsigned int usb_pipe; + unsigned int channels; + unsigned int frame_bytes; + unsigned int max_packet_bytes; + unsigned int period_pos; + unsigned int buffer_pos; + unsigned int queue_length; + struct ua101_urb { + struct urb urb; + struct usb_iso_packet_descriptor iso_frame_desc[1]; + struct list_head ready_list; + } *urbs[MAX_QUEUE_LENGTH]; + struct { + unsigned int size; + void *addr; + dma_addr_t dma; + } buffers[MAX_MEMORY_BUFFERS]; + } capture, playback; + + unsigned int fps[10]; + unsigned int frame_counter; +}; + +static DEFINE_MUTEX(devices_mutex); +static unsigned int devices_used; +static struct usb_driver ua101_driver; + +static void abort_alsa_playback(struct ua101 *ua); +static void abort_alsa_capture(struct ua101 *ua); + +/* allocate virtual buffer; may be called more than once */ +static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, + size_t size) +{ + struct snd_pcm_runtime *runtime = subs->runtime; + + if (runtime->dma_area) { + if (runtime->dma_bytes >= size) + return 0; /* already large enough */ + vfree(runtime->dma_area); + } + runtime->dma_area = vmalloc_user(size); + if (!runtime->dma_area) + return -ENOMEM; + runtime->dma_bytes = size; + return 0; +} + +/* free virtual buffer; may be called more than once */ +static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs) +{ + struct snd_pcm_runtime *runtime = subs->runtime; + + vfree(runtime->dma_area); + runtime->dma_area = NULL; + return 0; +} + +/* get the physical page pointer at the given offset */ +static struct page *snd_pcm_get_vmalloc_page(struct snd_pcm_substream *subs, + unsigned long offset) +{ + void *pageptr = subs->runtime->dma_area + offset; + return vmalloc_to_page(pageptr); +} + +static const char *usb_error_string(int err) +{ + switch (err) { + case -ENODEV: + return "no device"; + case -ENOENT: + return "endpoint not enabled"; + case -EPIPE: + return "endpoint stalled"; + case -ENOSPC: + return "not enough bandwidth"; + case -ESHUTDOWN: + return "device disabled"; + case -EHOSTUNREACH: + return "device suspended"; + case -EINVAL: + case -EAGAIN: + case -EFBIG: + case -EMSGSIZE: + return "internal error"; + default: + return "unknown error"; + } +} + +static void abort_usb_capture(struct ua101 *ua) +{ + if (test_and_clear_bit(USB_CAPTURE_RUNNING, &ua->states)) { + wake_up(&ua->alsa_capture_wait); + wake_up(&ua->rate_feedback_wait); + } +} + +static void abort_usb_playback(struct ua101 *ua) +{ + if (test_and_clear_bit(USB_PLAYBACK_RUNNING, &ua->states)) + wake_up(&ua->alsa_playback_wait); +} + +static void playback_urb_complete(struct urb *usb_urb) +{ + struct ua101_urb *urb = (struct ua101_urb *)usb_urb; + struct ua101 *ua = urb->urb.context; + unsigned long flags; + + if (unlikely(urb->urb.status == -ENOENT || /* unlinked */ + urb->urb.status == -ENODEV || /* device removed */ + urb->urb.status == -ECONNRESET || /* unlinked */ + urb->urb.status == -ESHUTDOWN)) { /* device disabled */ + abort_usb_playback(ua); + abort_alsa_playback(ua); + return; + } + + if (test_bit(USB_PLAYBACK_RUNNING, &ua->states)) { + /* append URB to FIFO */ + spin_lock_irqsave(&ua->lock, flags); + list_add_tail(&urb->ready_list, &ua->ready_playback_urbs); + if (ua->rate_feedback_count > 0) + tasklet_schedule(&ua->playback_tasklet); + ua->playback.substream->runtime->delay -= + urb->urb.iso_frame_desc[0].length / + ua->playback.frame_bytes; + spin_unlock_irqrestore(&ua->lock, flags); + } +} + +static void first_playback_urb_complete(struct urb *urb) +{ + struct ua101 *ua = urb->context; + + urb->complete = playback_urb_complete; + playback_urb_complete(urb); + + set_bit(PLAYBACK_URB_COMPLETED, &ua->states); + wake_up(&ua->alsa_playback_wait); +} + +/* copy data from the ALSA ring buffer into the URB buffer */ +static bool copy_playback_data(struct ua101_stream *stream, struct urb *urb, + unsigned int frames) +{ + struct snd_pcm_runtime *runtime; + unsigned int frame_bytes, frames1; + const u8 *source; + + runtime = stream->substream->runtime; + frame_bytes = stream->frame_bytes; + source = runtime->dma_area + stream->buffer_pos * frame_bytes; + if (stream->buffer_pos + frames <= runtime->buffer_size) { + memcpy(urb->transfer_buffer, source, frames * frame_bytes); + } else { + /* wrap around at end of ring buffer */ + frames1 = runtime->buffer_size - stream->buffer_pos; + memcpy(urb->transfer_buffer, source, frames1 * frame_bytes); + memcpy(urb->transfer_buffer + frames1 * frame_bytes, + runtime->dma_area, (frames - frames1) * frame_bytes); + } + + stream->buffer_pos += frames; + if (stream->buffer_pos >= runtime->buffer_size) + stream->buffer_pos -= runtime->buffer_size; + stream->period_pos += frames; + if (stream->period_pos >= runtime->period_size) { + stream->period_pos -= runtime->period_size; + return true; + } + return false; +} + +static inline void add_with_wraparound(struct ua101 *ua, + unsigned int *value, unsigned int add) +{ + *value += add; + if (*value >= ua->playback.queue_length) + *value -= ua->playback.queue_length; +} + +static void playback_tasklet(unsigned long data) +{ + struct ua101 *ua = (void *)data; + unsigned long flags; + unsigned int frames; + struct ua101_urb *urb; + bool do_period_elapsed = false; + int err; + + if (unlikely(!test_bit(USB_PLAYBACK_RUNNING, &ua->states))) + return; + + /* + * Synchronizing the playback rate to the capture rate is done by using + * the same sequence of packet sizes for both streams. + * Submitting a playback URB therefore requires both a ready URB and + * the size of the corresponding capture packet, i.e., both playback + * and capture URBs must have been completed. Since the USB core does + * not guarantee that playback and capture complete callbacks are + * called alternately, we use two FIFOs for packet sizes and read URBs; + * submitting playback URBs is possible as long as both FIFOs are + * nonempty. + */ + spin_lock_irqsave(&ua->lock, flags); + while (ua->rate_feedback_count > 0 && + !list_empty(&ua->ready_playback_urbs)) { + /* take packet size out of FIFO */ + frames = ua->rate_feedback[ua->rate_feedback_start]; + add_with_wraparound(ua, &ua->rate_feedback_start, 1); + ua->rate_feedback_count--; + + /* take URB out of FIFO */ + urb = list_first_entry(&ua->ready_playback_urbs, + struct ua101_urb, ready_list); + list_del(&urb->ready_list); + + /* fill packet with data or silence */ + urb->urb.iso_frame_desc[0].length = + frames * ua->playback.frame_bytes; + if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) + do_period_elapsed |= copy_playback_data(&ua->playback, + &urb->urb, + frames); + else + memset(urb->urb.transfer_buffer, 0, + urb->urb.iso_frame_desc[0].length); + + /* and off you go ... */ + err = usb_submit_urb(&urb->urb, GFP_ATOMIC); + if (unlikely(err < 0)) { + spin_unlock_irqrestore(&ua->lock, flags); + abort_usb_playback(ua); + abort_alsa_playback(ua); + dev_err(&ua->dev->dev, "USB request error %d: %s\n", + err, usb_error_string(err)); + return; + } + ua->playback.substream->runtime->delay += frames; + } + spin_unlock_irqrestore(&ua->lock, flags); + if (do_period_elapsed) + snd_pcm_period_elapsed(ua->playback.substream); +} + +/* copy data from the URB buffer into the ALSA ring buffer */ +static bool copy_capture_data(struct ua101_stream *stream, struct urb *urb, + unsigned int frames) +{ + struct snd_pcm_runtime *runtime; + unsigned int frame_bytes, frames1; + u8 *dest; + + runtime = stream->substream->runtime; + frame_bytes = stream->frame_bytes; + dest = runtime->dma_area + stream->buffer_pos * frame_bytes; + if (stream->buffer_pos + frames <= runtime->buffer_size) { + memcpy(dest, urb->transfer_buffer, frames * frame_bytes); + } else { + /* wrap around at end of ring buffer */ + frames1 = runtime->buffer_size - stream->buffer_pos; + memcpy(dest, urb->transfer_buffer, frames1 * frame_bytes); + memcpy(runtime->dma_area, + urb->transfer_buffer + frames1 * frame_bytes, + (frames - frames1) * frame_bytes); + } + + stream->buffer_pos += frames; + if (stream->buffer_pos >= runtime->buffer_size) + stream->buffer_pos -= runtime->buffer_size; + stream->period_pos += frames; + if (stream->period_pos >= runtime->period_size) { + stream->period_pos -= runtime->period_size; + return true; + } + return false; +} + +static void capture_urb_complete(struct urb *urb) +{ + struct ua101 *ua = urb->context; + struct ua101_stream *stream = &ua->capture; + unsigned long flags; + unsigned int frames, write_ptr; + bool do_period_elapsed; + int err; + + if (unlikely(urb->status == -ENOENT || /* unlinked */ + urb->status == -ENODEV || /* device removed */ + urb->status == -ECONNRESET || /* unlinked */ + urb->status == -ESHUTDOWN)) /* device disabled */ + goto stream_stopped; + + if (urb->status >= 0 && urb->iso_frame_desc[0].status >= 0) + frames = urb->iso_frame_desc[0].actual_length / + stream->frame_bytes; + else + frames = 0; + + spin_lock_irqsave(&ua->lock, flags); + + if (frames > 0 && test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) + do_period_elapsed = copy_capture_data(stream, urb, frames); + else + do_period_elapsed = false; + + if (test_bit(USB_CAPTURE_RUNNING, &ua->states)) { + err = usb_submit_urb(urb, GFP_ATOMIC); + if (unlikely(err < 0)) { + spin_unlock_irqrestore(&ua->lock, flags); + dev_err(&ua->dev->dev, "USB request error %d: %s\n", + err, usb_error_string(err)); + goto stream_stopped; + } + + /* append packet size to FIFO */ + write_ptr = ua->rate_feedback_start; + add_with_wraparound(ua, &write_ptr, ua->rate_feedback_count); + ua->rate_feedback[write_ptr] = frames; + if (ua->rate_feedback_count < ua->playback.queue_length) { + ua->rate_feedback_count++; + if (ua->rate_feedback_count == + ua->playback.queue_length) + wake_up(&ua->rate_feedback_wait); + } else { + /* + * Ring buffer overflow; this happens when the playback + * stream is not running. Throw away the oldest entry, + * so that the playback stream, when it starts, sees + * the most recent packet sizes. + */ + add_with_wraparound(ua, &ua->rate_feedback_start, 1); + } + if (test_bit(USB_PLAYBACK_RUNNING, &ua->states) && + !list_empty(&ua->ready_playback_urbs)) + tasklet_schedule(&ua->playback_tasklet); + } + + spin_unlock_irqrestore(&ua->lock, flags); + + if (do_period_elapsed) + snd_pcm_period_elapsed(stream->substream); + + /* for debugging: measure the sample rate relative to the USB clock */ + ua->fps[ua->frame_counter++ / ua->packets_per_second] += frames; + if (ua->frame_counter >= ARRAY_SIZE(ua->fps) * ua->packets_per_second) { + printk(KERN_DEBUG "capture rate:"); + for (frames = 0; frames < ARRAY_SIZE(ua->fps); ++frames) + printk(KERN_CONT " %u", ua->fps[frames]); + printk(KERN_CONT "\n"); + memset(ua->fps, 0, sizeof(ua->fps)); + ua->frame_counter = 0; + } + return; + +stream_stopped: + abort_usb_playback(ua); + abort_usb_capture(ua); + abort_alsa_playback(ua); + abort_alsa_capture(ua); +} + +static void first_capture_urb_complete(struct urb *urb) +{ + struct ua101 *ua = urb->context; + + urb->complete = capture_urb_complete; + capture_urb_complete(urb); + + set_bit(CAPTURE_URB_COMPLETED, &ua->states); + wake_up(&ua->alsa_capture_wait); +} + +static int submit_stream_urbs(struct ua101 *ua, struct ua101_stream *stream) +{ + unsigned int i; + + for (i = 0; i < stream->queue_length; ++i) { + int err = usb_submit_urb(&stream->urbs[i]->urb, GFP_KERNEL); + if (err < 0) { + dev_err(&ua->dev->dev, "USB request error %d: %s\n", + err, usb_error_string(err)); + return err; + } + } + return 0; +} + +static void kill_stream_urbs(struct ua101_stream *stream) +{ + unsigned int i; + + for (i = 0; i < stream->queue_length; ++i) + usb_kill_urb(&stream->urbs[i]->urb); +} + +static int enable_iso_interface(struct ua101 *ua, unsigned int intf_index) +{ + struct usb_host_interface *alts; + + alts = ua->intf[intf_index]->cur_altsetting; + if (alts->desc.bAlternateSetting != 1) { + int err = usb_set_interface(ua->dev, + alts->desc.bInterfaceNumber, 1); + if (err < 0) { + dev_err(&ua->dev->dev, + "cannot initialize interface; error %d: %s\n", + err, usb_error_string(err)); + return err; + } + } + return 0; +} + +static void disable_iso_interface(struct ua101 *ua, unsigned int intf_index) +{ + struct usb_host_interface *alts; + + alts = ua->intf[intf_index]->cur_altsetting; + if (alts->desc.bAlternateSetting != 0) { + int err = usb_set_interface(ua->dev, + alts->desc.bInterfaceNumber, 0); + if (err < 0 && !test_bit(DISCONNECTED, &ua->states)) + dev_warn(&ua->dev->dev, + "interface reset failed; error %d: %s\n", + err, usb_error_string(err)); + } +} + +static void stop_usb_capture(struct ua101 *ua) +{ + clear_bit(USB_CAPTURE_RUNNING, &ua->states); + + kill_stream_urbs(&ua->capture); + + disable_iso_interface(ua, INTF_CAPTURE); +} + +static int start_usb_capture(struct ua101 *ua) +{ + int err; + + if (test_bit(DISCONNECTED, &ua->states)) + return -ENODEV; + + if (test_bit(USB_CAPTURE_RUNNING, &ua->states)) + return 0; + + kill_stream_urbs(&ua->capture); + + err = enable_iso_interface(ua, INTF_CAPTURE); + if (err < 0) + return err; + + clear_bit(CAPTURE_URB_COMPLETED, &ua->states); + ua->capture.urbs[0]->urb.complete = first_capture_urb_complete; + ua->rate_feedback_start = 0; + ua->rate_feedback_count = 0; + + set_bit(USB_CAPTURE_RUNNING, &ua->states); + err = submit_stream_urbs(ua, &ua->capture); + if (err < 0) + stop_usb_capture(ua); + return err; +} + +static void stop_usb_playback(struct ua101 *ua) +{ + clear_bit(USB_PLAYBACK_RUNNING, &ua->states); + + kill_stream_urbs(&ua->playback); + + tasklet_kill(&ua->playback_tasklet); + + disable_iso_interface(ua, INTF_PLAYBACK); +} + +static int start_usb_playback(struct ua101 *ua) +{ + unsigned int i, frames; + struct urb *urb; + int err = 0; + + if (test_bit(DISCONNECTED, &ua->states)) + return -ENODEV; + + if (test_bit(USB_PLAYBACK_RUNNING, &ua->states)) + return 0; + + kill_stream_urbs(&ua->playback); + tasklet_kill(&ua->playback_tasklet); + + err = enable_iso_interface(ua, INTF_PLAYBACK); + if (err < 0) + return err; + + clear_bit(PLAYBACK_URB_COMPLETED, &ua->states); + ua->playback.urbs[0]->urb.complete = + first_playback_urb_complete; + spin_lock_irq(&ua->lock); + INIT_LIST_HEAD(&ua->ready_playback_urbs); + spin_unlock_irq(&ua->lock); + + /* + * We submit the initial URBs all at once, so we have to wait for the + * packet size FIFO to be full. + */ + wait_event(ua->rate_feedback_wait, + ua->rate_feedback_count >= ua->playback.queue_length || + !test_bit(USB_CAPTURE_RUNNING, &ua->states) || + test_bit(DISCONNECTED, &ua->states)); + if (test_bit(DISCONNECTED, &ua->states)) { + stop_usb_playback(ua); + return -ENODEV; + } + if (!test_bit(USB_CAPTURE_RUNNING, &ua->states)) { + stop_usb_playback(ua); + return -EIO; + } + + for (i = 0; i < ua->playback.queue_length; ++i) { + /* all initial URBs contain silence */ + spin_lock_irq(&ua->lock); + frames = ua->rate_feedback[ua->rate_feedback_start]; + add_with_wraparound(ua, &ua->rate_feedback_start, 1); + ua->rate_feedback_count--; + spin_unlock_irq(&ua->lock); + urb = &ua->playback.urbs[i]->urb; + urb->iso_frame_desc[0].length = + frames * ua->playback.frame_bytes; + memset(urb->transfer_buffer, 0, + urb->iso_frame_desc[0].length); + } + + set_bit(USB_PLAYBACK_RUNNING, &ua->states); + err = submit_stream_urbs(ua, &ua->playback); + if (err < 0) + stop_usb_playback(ua); + return err; +} + +static void abort_alsa_capture(struct ua101 *ua) +{ + if (test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) + snd_pcm_stop(ua->capture.substream, SNDRV_PCM_STATE_XRUN); +} + +static void abort_alsa_playback(struct ua101 *ua) +{ + if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) + snd_pcm_stop(ua->playback.substream, SNDRV_PCM_STATE_XRUN); +} + +static int set_stream_hw(struct ua101 *ua, struct snd_pcm_substream *substream, + unsigned int channels) +{ + int err; + + substream->runtime->hw.info = + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_FIFO_IN_FRAMES; + substream->runtime->hw.formats = ua->format_bit; + substream->runtime->hw.rates = snd_pcm_rate_to_rate_bit(ua->rate); + substream->runtime->hw.rate_min = ua->rate; + substream->runtime->hw.rate_max = ua->rate; + substream->runtime->hw.channels_min = channels; + substream->runtime->hw.channels_max = channels; + substream->runtime->hw.buffer_bytes_max = 45000 * 1024; + substream->runtime->hw.period_bytes_min = 1; + substream->runtime->hw.period_bytes_max = UINT_MAX; + substream->runtime->hw.periods_min = 2; + substream->runtime->hw.periods_max = UINT_MAX; + err = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIOD_TIME, + 1500000 / ua->packets_per_second, + 8192000); + if (err < 0) + return err; + err = snd_pcm_hw_constraint_msbits(substream->runtime, 0, 32, 24); + return err; +} + +static int capture_pcm_open(struct snd_pcm_substream *substream) +{ + struct ua101 *ua = substream->private_data; + int err; + + ua->capture.substream = substream; + err = set_stream_hw(ua, substream, ua->capture.channels); + if (err < 0) + return err; + substream->runtime->hw.fifo_size = + DIV_ROUND_CLOSEST(ua->rate, ua->packets_per_second); + substream->runtime->delay = substream->runtime->hw.fifo_size; + + mutex_lock(&ua->mutex); + err = start_usb_capture(ua); + if (err >= 0) + set_bit(ALSA_CAPTURE_OPEN, &ua->states); + mutex_unlock(&ua->mutex); + return err; +} + +static int playback_pcm_open(struct snd_pcm_substream *substream) +{ + struct ua101 *ua = substream->private_data; + int err; + + ua->playback.substream = substream; + err = set_stream_hw(ua, substream, ua->playback.channels); + if (err < 0) + return err; + substream->runtime->hw.fifo_size = + DIV_ROUND_CLOSEST(ua->rate * ua->playback.queue_length, + ua->packets_per_second); + + mutex_lock(&ua->mutex); + err = start_usb_capture(ua); + if (err < 0) + goto error; + err = start_usb_playback(ua); + if (err < 0) { + if (!test_bit(ALSA_CAPTURE_OPEN, &ua->states)) + stop_usb_capture(ua); + goto error; + } + set_bit(ALSA_PLAYBACK_OPEN, &ua->states); +error: + mutex_unlock(&ua->mutex); + return err; +} + +static int capture_pcm_close(struct snd_pcm_substream *substream) +{ + struct ua101 *ua = substream->private_data; + + mutex_lock(&ua->mutex); + clear_bit(ALSA_CAPTURE_OPEN, &ua->states); + if (!test_bit(ALSA_PLAYBACK_OPEN, &ua->states)) + stop_usb_capture(ua); + mutex_unlock(&ua->mutex); + return 0; +} + +static int playback_pcm_close(struct snd_pcm_substream *substream) +{ + struct ua101 *ua = substream->private_data; + + mutex_lock(&ua->mutex); + stop_usb_playback(ua); + clear_bit(ALSA_PLAYBACK_OPEN, &ua->states); + if (!test_bit(ALSA_CAPTURE_OPEN, &ua->states)) + stop_usb_capture(ua); + mutex_unlock(&ua->mutex); + return 0; +} + +static int capture_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct ua101 *ua = substream->private_data; + int err; + + mutex_lock(&ua->mutex); + err = start_usb_capture(ua); + mutex_unlock(&ua->mutex); + if (err < 0) + return err; + + return snd_pcm_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); +} + +static int playback_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct ua101 *ua = substream->private_data; + int err; + + mutex_lock(&ua->mutex); + err = start_usb_capture(ua); + if (err >= 0) + err = start_usb_playback(ua); + mutex_unlock(&ua->mutex); + if (err < 0) + return err; + + return snd_pcm_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); +} + +static int ua101_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_free_vmalloc_buffer(substream); + return 0; +} + +static int capture_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct ua101 *ua = substream->private_data; + int err; + + mutex_lock(&ua->mutex); + err = start_usb_capture(ua); + mutex_unlock(&ua->mutex); + if (err < 0) + return err; + + /* + * The EHCI driver schedules the first packet of an iso stream at 10 ms + * in the future, i.e., no data is actually captured for that long. + * Take the wait here so that the stream is known to be actually + * running when the start trigger has been called. + */ + wait_event(ua->alsa_capture_wait, + test_bit(CAPTURE_URB_COMPLETED, &ua->states) || + !test_bit(USB_CAPTURE_RUNNING, &ua->states)); + if (test_bit(DISCONNECTED, &ua->states)) + return -ENODEV; + if (!test_bit(USB_CAPTURE_RUNNING, &ua->states)) + return -EIO; + + ua->capture.period_pos = 0; + ua->capture.buffer_pos = 0; + return 0; +} + +static int playback_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct ua101 *ua = substream->private_data; + int err; + + mutex_lock(&ua->mutex); + err = start_usb_capture(ua); + if (err >= 0) + err = start_usb_playback(ua); + mutex_unlock(&ua->mutex); + if (err < 0) + return err; + + /* see the comment in capture_pcm_prepare() */ + wait_event(ua->alsa_playback_wait, + test_bit(PLAYBACK_URB_COMPLETED, &ua->states) || + !test_bit(USB_PLAYBACK_RUNNING, &ua->states)); + if (test_bit(DISCONNECTED, &ua->states)) + return -ENODEV; + if (!test_bit(USB_PLAYBACK_RUNNING, &ua->states)) + return -EIO; + + substream->runtime->delay = 0; + ua->playback.period_pos = 0; + ua->playback.buffer_pos = 0; + return 0; +} + +static int capture_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct ua101 *ua = substream->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (!test_bit(USB_CAPTURE_RUNNING, &ua->states)) + return -EIO; + set_bit(ALSA_CAPTURE_RUNNING, &ua->states); + return 0; + case SNDRV_PCM_TRIGGER_STOP: + clear_bit(ALSA_CAPTURE_RUNNING, &ua->states); + return 0; + default: + return -EINVAL; + } +} + +static int playback_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct ua101 *ua = substream->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (!test_bit(USB_PLAYBACK_RUNNING, &ua->states)) + return -EIO; + set_bit(ALSA_PLAYBACK_RUNNING, &ua->states); + return 0; + case SNDRV_PCM_TRIGGER_STOP: + clear_bit(ALSA_PLAYBACK_RUNNING, &ua->states); + return 0; + default: + return -EINVAL; + } +} + +static inline snd_pcm_uframes_t ua101_pcm_pointer(struct ua101 *ua, + struct ua101_stream *stream) +{ + unsigned long flags; + unsigned int pos; + + spin_lock_irqsave(&ua->lock, flags); + pos = stream->buffer_pos; + spin_unlock_irqrestore(&ua->lock, flags); + return pos; +} + +static snd_pcm_uframes_t capture_pcm_pointer(struct snd_pcm_substream *subs) +{ + struct ua101 *ua = subs->private_data; + + return ua101_pcm_pointer(ua, &ua->capture); +} + +static snd_pcm_uframes_t playback_pcm_pointer(struct snd_pcm_substream *subs) +{ + struct ua101 *ua = subs->private_data; + + return ua101_pcm_pointer(ua, &ua->playback); +} + +static struct snd_pcm_ops capture_pcm_ops = { + .open = capture_pcm_open, + .close = capture_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = capture_pcm_hw_params, + .hw_free = ua101_pcm_hw_free, + .prepare = capture_pcm_prepare, + .trigger = capture_pcm_trigger, + .pointer = capture_pcm_pointer, + .page = snd_pcm_get_vmalloc_page, +}; + +static struct snd_pcm_ops playback_pcm_ops = { + .open = playback_pcm_open, + .close = playback_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = playback_pcm_hw_params, + .hw_free = ua101_pcm_hw_free, + .prepare = playback_pcm_prepare, + .trigger = playback_pcm_trigger, + .pointer = playback_pcm_pointer, + .page = snd_pcm_get_vmalloc_page, +}; + +static const struct uac_format_type_i_discrete_descriptor * +find_format_descriptor(struct usb_interface *interface) +{ + struct usb_host_interface *alt; + u8 *extra; + int extralen; + + if (interface->num_altsetting != 2) { + dev_err(&interface->dev, "invalid num_altsetting\n"); + return NULL; + } + + alt = &interface->altsetting[0]; + if (alt->desc.bNumEndpoints != 0) { + dev_err(&interface->dev, "invalid bNumEndpoints\n"); + return NULL; + } + + alt = &interface->altsetting[1]; + if (alt->desc.bNumEndpoints != 1) { + dev_err(&interface->dev, "invalid bNumEndpoints\n"); + return NULL; + } + + extra = alt->extra; + extralen = alt->extralen; + while (extralen >= sizeof(struct usb_descriptor_header)) { + struct uac_format_type_i_discrete_descriptor *desc; + + desc = (struct uac_format_type_i_discrete_descriptor *)extra; + if (desc->bLength > extralen) { + dev_err(&interface->dev, "descriptor overflow\n"); + return NULL; + } + if (desc->bLength == UAC_FORMAT_TYPE_I_DISCRETE_DESC_SIZE(1) && + desc->bDescriptorType == USB_DT_CS_INTERFACE && + desc->bDescriptorSubtype == UAC_FORMAT_TYPE) { + if (desc->bFormatType != UAC_FORMAT_TYPE_I_PCM || + desc->bSamFreqType != 1) { + dev_err(&interface->dev, + "invalid format type\n"); + return NULL; + } + return desc; + } + extralen -= desc->bLength; + extra += desc->bLength; + } + dev_err(&interface->dev, "sample format descriptor not found\n"); + return NULL; +} + +static int detect_usb_format(struct ua101 *ua) +{ + const struct uac_format_type_i_discrete_descriptor *fmt_capture; + const struct uac_format_type_i_discrete_descriptor *fmt_playback; + const struct usb_endpoint_descriptor *epd; + unsigned int rate2; + + fmt_capture = find_format_descriptor(ua->intf[INTF_CAPTURE]); + fmt_playback = find_format_descriptor(ua->intf[INTF_PLAYBACK]); + if (!fmt_capture || !fmt_playback) + return -ENXIO; + + switch (fmt_capture->bSubframeSize) { + case 3: + ua->format_bit = SNDRV_PCM_FMTBIT_S24_3LE; + break; + case 4: + ua->format_bit = SNDRV_PCM_FMTBIT_S32_LE; + break; + default: + dev_err(&ua->dev->dev, "sample width is not 24 or 32 bits\n"); + return -ENXIO; + } + if (fmt_capture->bSubframeSize != fmt_playback->bSubframeSize) { + dev_err(&ua->dev->dev, + "playback/capture sample widths do not match\n"); + return -ENXIO; + } + + if (fmt_capture->bBitResolution != 24 || + fmt_playback->bBitResolution != 24) { + dev_err(&ua->dev->dev, "sample width is not 24 bits\n"); + return -ENXIO; + } + + ua->rate = combine_triple(fmt_capture->tSamFreq[0]); + rate2 = combine_triple(fmt_playback->tSamFreq[0]); + if (ua->rate != rate2) { + dev_err(&ua->dev->dev, + "playback/capture rates do not match: %u/%u\n", + rate2, ua->rate); + return -ENXIO; + } + + switch (ua->dev->speed) { + case USB_SPEED_FULL: + ua->packets_per_second = 1000; + break; + case USB_SPEED_HIGH: + ua->packets_per_second = 8000; + break; + default: + dev_err(&ua->dev->dev, "unknown device speed\n"); + return -ENXIO; + } + + ua->capture.channels = fmt_capture->bNrChannels; + ua->playback.channels = fmt_playback->bNrChannels; + ua->capture.frame_bytes = + fmt_capture->bSubframeSize * ua->capture.channels; + ua->playback.frame_bytes = + fmt_playback->bSubframeSize * ua->playback.channels; + + epd = &ua->intf[INTF_CAPTURE]->altsetting[1].endpoint[0].desc; + if (!usb_endpoint_is_isoc_in(epd)) { + dev_err(&ua->dev->dev, "invalid capture endpoint\n"); + return -ENXIO; + } + ua->capture.usb_pipe = usb_rcvisocpipe(ua->dev, usb_endpoint_num(epd)); + ua->capture.max_packet_bytes = le16_to_cpu(epd->wMaxPacketSize); + + epd = &ua->intf[INTF_PLAYBACK]->altsetting[1].endpoint[0].desc; + if (!usb_endpoint_is_isoc_out(epd)) { + dev_err(&ua->dev->dev, "invalid playback endpoint\n"); + return -ENXIO; + } + ua->playback.usb_pipe = usb_sndisocpipe(ua->dev, usb_endpoint_num(epd)); + ua->playback.max_packet_bytes = le16_to_cpu(epd->wMaxPacketSize); + return 0; +} + +static int alloc_stream_buffers(struct ua101 *ua, struct ua101_stream *stream) +{ + unsigned int remaining_packets, packets, packets_per_page, i; + size_t size; + + stream->queue_length = queue_length; + stream->queue_length = max(stream->queue_length, + (unsigned int)MIN_QUEUE_LENGTH); + stream->queue_length = min(stream->queue_length, + (unsigned int)MAX_QUEUE_LENGTH); + + /* + * The cache pool sizes used by usb_buffer_alloc() (128, 512, 2048) are + * quite bad when used with the packet sizes of this device (e.g. 280, + * 520, 624). Therefore, we allocate and subdivide entire pages, using + * a smaller buffer only for the last chunk. + */ + remaining_packets = stream->queue_length; + packets_per_page = PAGE_SIZE / stream->max_packet_bytes; + for (i = 0; i < ARRAY_SIZE(stream->buffers); ++i) { + packets = min(remaining_packets, packets_per_page); + size = packets * stream->max_packet_bytes; + stream->buffers[i].addr = + usb_buffer_alloc(ua->dev, size, GFP_KERNEL, + &stream->buffers[i].dma); + if (!stream->buffers[i].addr) + return -ENOMEM; + stream->buffers[i].size = size; + remaining_packets -= packets; + if (!remaining_packets) + break; + } + if (remaining_packets) { + dev_err(&ua->dev->dev, "too many packets\n"); + return -ENXIO; + } + return 0; +} + +static void free_stream_buffers(struct ua101 *ua, struct ua101_stream *stream) +{ + unsigned int i; + + for (i = 0; i < ARRAY_SIZE(stream->buffers); ++i) + usb_buffer_free(ua->dev, + stream->buffers[i].size, + stream->buffers[i].addr, + stream->buffers[i].dma); +} + +static int alloc_stream_urbs(struct ua101 *ua, struct ua101_stream *stream, + void (*urb_complete)(struct urb *)) +{ + unsigned max_packet_size = stream->max_packet_bytes; + struct ua101_urb *urb; + unsigned int b, u = 0; + + for (b = 0; b < ARRAY_SIZE(stream->buffers); ++b) { + unsigned int size = stream->buffers[b].size; + u8 *addr = stream->buffers[b].addr; + dma_addr_t dma = stream->buffers[b].dma; + + while (size >= max_packet_size) { + if (u >= stream->queue_length) + goto bufsize_error; + urb = kmalloc(sizeof(*urb), GFP_KERNEL); + if (!urb) + return -ENOMEM; + usb_init_urb(&urb->urb); + urb->urb.dev = ua->dev; + urb->urb.pipe = stream->usb_pipe; + urb->urb.transfer_flags = URB_ISO_ASAP | + URB_NO_TRANSFER_DMA_MAP; + urb->urb.transfer_buffer = addr; + urb->urb.transfer_dma = dma; + urb->urb.transfer_buffer_length = max_packet_size; + urb->urb.number_of_packets = 1; + urb->urb.interval = 1; + urb->urb.context = ua; + urb->urb.complete = urb_complete; + urb->urb.iso_frame_desc[0].offset = 0; + urb->urb.iso_frame_desc[0].length = max_packet_size; + stream->urbs[u++] = urb; + size -= max_packet_size; + addr += max_packet_size; + dma += max_packet_size; + } + } + if (u == stream->queue_length) + return 0; +bufsize_error: + dev_err(&ua->dev->dev, "internal buffer size error\n"); + return -ENXIO; +} + +static void free_stream_urbs(struct ua101_stream *stream) +{ + unsigned int i; + + for (i = 0; i < stream->queue_length; ++i) + kfree(stream->urbs[i]); +} + +static void free_usb_related_resources(struct ua101 *ua, + struct usb_interface *interface) +{ + unsigned int i; + + free_stream_urbs(&ua->capture); + free_stream_urbs(&ua->playback); + free_stream_buffers(ua, &ua->capture); + free_stream_buffers(ua, &ua->playback); + + for (i = 0; i < ARRAY_SIZE(ua->intf); ++i) + if (ua->intf[i]) { + usb_set_intfdata(ua->intf[i], NULL); + if (ua->intf[i] != interface) + usb_driver_release_interface(&ua101_driver, + ua->intf[i]); + } +} + +static void ua101_card_free(struct snd_card *card) +{ + struct ua101 *ua = card->private_data; + + mutex_destroy(&ua->mutex); +} + +static int ua101_probe(struct usb_interface *interface, + const struct usb_device_id *usb_id) +{ + static const struct snd_usb_midi_endpoint_info midi_ep = { + .out_cables = 0x0001, + .in_cables = 0x0001 + }; + static const struct snd_usb_audio_quirk midi_quirk = { + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &midi_ep + }; + struct snd_card *card; + struct ua101 *ua; + unsigned int card_index, i; + char usb_path[32]; + int err; + + if (interface->altsetting->desc.bInterfaceNumber != 0) + return -ENODEV; + + mutex_lock(&devices_mutex); + + for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) + if (enable[card_index] && !(devices_used & (1 << card_index))) + break; + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + err = snd_card_create(index[card_index], id[card_index], THIS_MODULE, + sizeof(*ua), &card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return err; + } + card->private_free = ua101_card_free; + ua = card->private_data; + ua->dev = interface_to_usbdev(interface); + ua->card = card; + ua->card_index = card_index; + INIT_LIST_HEAD(&ua->midi_list); + spin_lock_init(&ua->lock); + mutex_init(&ua->mutex); + INIT_LIST_HEAD(&ua->ready_playback_urbs); + tasklet_init(&ua->playback_tasklet, + playback_tasklet, (unsigned long)ua); + init_waitqueue_head(&ua->alsa_capture_wait); + init_waitqueue_head(&ua->rate_feedback_wait); + init_waitqueue_head(&ua->alsa_playback_wait); + +#ifdef UA1A_HACK + if (ua->dev->descriptor.idProduct == cpu_to_le16(0x0018)) { + ua->intf[2] = interface; + ua->intf[0] = usb_ifnum_to_if(ua->dev, 1); + ua->intf[1] = usb_ifnum_to_if(ua->dev, 2); + usb_driver_claim_interface(&ua101_driver, ua->intf[0], ua); + usb_driver_claim_interface(&ua101_driver, ua->intf[1], ua); + } else { +#endif + ua->intf[0] = interface; + for (i = 1; i < ARRAY_SIZE(ua->intf); ++i) { + ua->intf[i] = usb_ifnum_to_if(ua->dev, i); + if (!ua->intf[i]) { + dev_err(&ua->dev->dev, "interface %u not found\n", i); + err = -ENXIO; + goto probe_error; + } + err = usb_driver_claim_interface(&ua101_driver, + ua->intf[i], ua); + if (err < 0) { + ua->intf[i] = NULL; + err = -EBUSY; + goto probe_error; + } + } +#ifdef UA1A_HACK + } +#endif + + snd_card_set_dev(card, &interface->dev); + +#ifdef UA1A_HACK + if (ua->dev->descriptor.idProduct == cpu_to_le16(0x0018)) { + ua->format_bit = SNDRV_PCM_FMTBIT_S16_LE; + ua->rate = 44100; + ua->packets_per_second = 1000; + ua->capture.channels = 2; + ua->playback.channels = 2; + ua->capture.frame_bytes = 4; + ua->playback.frame_bytes = 4; + ua->capture.usb_pipe = usb_rcvisocpipe(ua->dev, 2); + ua->playback.usb_pipe = usb_sndisocpipe(ua->dev, 1); + ua->capture.max_packet_bytes = 192; + ua->playback.max_packet_bytes = 192; + } else { +#endif + err = detect_usb_format(ua); + if (err < 0) + goto probe_error; +#ifdef UA1A_HACK + } +#endif + + strcpy(card->driver, "UA-101"); + strcpy(card->shortname, "UA-101"); + usb_make_path(ua->dev, usb_path, sizeof(usb_path)); + snprintf(ua->card->longname, sizeof(ua->card->longname), + "EDIROL UA-101 (serial %s), %u Hz at %s, %s speed", + ua->dev->serial ? ua->dev->serial : "?", ua->rate, usb_path, + ua->dev->speed == USB_SPEED_HIGH ? "high" : "full"); + + err = alloc_stream_buffers(ua, &ua->capture); + if (err < 0) + goto probe_error; + err = alloc_stream_buffers(ua, &ua->playback); + if (err < 0) + goto probe_error; + + err = alloc_stream_urbs(ua, &ua->capture, capture_urb_complete); + if (err < 0) + goto probe_error; + err = alloc_stream_urbs(ua, &ua->playback, playback_urb_complete); + if (err < 0) + goto probe_error; + + err = snd_pcm_new(card, "UA-101", 0, 1, 1, &ua->pcm); + if (err < 0) + goto probe_error; + ua->pcm->private_data = ua; + strcpy(ua->pcm->name, "UA-101"); + snd_pcm_set_ops(ua->pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_pcm_ops); + snd_pcm_set_ops(ua->pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_pcm_ops); + +#ifdef UA1A_HACK + if (ua->dev->descriptor.idProduct != cpu_to_le16(0x0018)) { +#endif + err = snd_usbmidi_create(card, ua->intf[INTF_MIDI], + &ua->midi_list, &midi_quirk); + if (err < 0) + goto probe_error; +#ifdef UA1A_HACK + } +#endif + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + usb_set_intfdata(interface, ua); + devices_used |= 1 << card_index; + + mutex_unlock(&devices_mutex); + return 0; + +probe_error: + free_usb_related_resources(ua, interface); + snd_card_free(card); + mutex_unlock(&devices_mutex); + return err; +} + +static void ua101_disconnect(struct usb_interface *interface) +{ + struct ua101 *ua = usb_get_intfdata(interface); + struct list_head *midi; + + if (!ua) + return; + + mutex_lock(&devices_mutex); + + set_bit(DISCONNECTED, &ua->states); + wake_up(&ua->rate_feedback_wait); + + /* make sure that userspace cannot create new requests */ + snd_card_disconnect(ua->card); + + /* make sure that there are no pending USB requests */ + __list_for_each(midi, &ua->midi_list) + snd_usbmidi_disconnect(midi); + abort_alsa_playback(ua); + abort_alsa_capture(ua); + mutex_lock(&ua->mutex); + stop_usb_playback(ua); + stop_usb_capture(ua); + mutex_unlock(&ua->mutex); + + free_usb_related_resources(ua, interface); + + devices_used &= ~(1 << ua->card_index); + + snd_card_free_when_closed(ua->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_device_id ua101_ids[] = { +#ifdef UA1A_HACK + { USB_DEVICE(0x0582, 0x0018) }, +#endif + { USB_DEVICE(0x0582, 0x007d) }, + { USB_DEVICE(0x0582, 0x008d) }, + { } +}; +MODULE_DEVICE_TABLE(usb, ua101_ids); + +static struct usb_driver ua101_driver = { + .name = "snd-ua101", + .id_table = ua101_ids, + .probe = ua101_probe, + .disconnect = ua101_disconnect, +#if 0 + .suspend = ua101_suspend, + .resume = ua101_resume, +#endif +}; + +static int __init alsa_card_ua101_init(void) +{ + return usb_register(&ua101_driver); +} + +static void __exit alsa_card_ua101_exit(void) +{ + usb_deregister(&ua101_driver); + mutex_destroy(&devices_mutex); +} + +module_init(alsa_card_ua101_init); +module_exit(alsa_card_ua101_exit); diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index b074a59..f352141 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3142,59 +3142,6 @@ static int create_ua1000_quirk(struct snd_usb_audio *chip, return 0; } -/* - * Create a stream for an Edirol UA-101 interface. - * Copy, paste and modify from Edirol UA-1000 - */ -static int create_ua101_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) -{ - static const struct audioformat ua101_format = { - .format = SNDRV_PCM_FORMAT_S32_LE, - .fmt_type = USB_FORMAT_TYPE_I, - .altsetting = 1, - .altset_idx = 1, - .attributes = 0, - .rates = SNDRV_PCM_RATE_CONTINUOUS, - }; - struct usb_host_interface *alts; - struct usb_interface_descriptor *altsd; - struct audioformat *fp; - int stream, err; - - if (iface->num_altsetting != 2) - return -ENXIO; - alts = &iface->altsetting[1]; - altsd = get_iface_desc(alts); - if (alts->extralen != 18 || alts->extra[1] != USB_DT_CS_INTERFACE || - altsd->bNumEndpoints != 1) - return -ENXIO; - - fp = kmemdup(&ua101_format, sizeof(*fp), GFP_KERNEL); - if (!fp) - return -ENOMEM; - - fp->channels = alts->extra[11]; - fp->iface = altsd->bInterfaceNumber; - fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; - fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; - fp->datainterval = parse_datainterval(chip, alts); - fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); - fp->rate_max = fp->rate_min = combine_triple(&alts->extra[15]); - - stream = (fp->endpoint & USB_DIR_IN) - ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; - err = add_audio_endpoint(chip, stream, fp); - if (err < 0) { - kfree(fp); - return err; - } - /* FIXME: playback must be synchronized to capture */ - usb_set_interface(chip->dev, fp->iface, 0); - return 0; -} - static int snd_usb_create_quirk(struct snd_usb_audio *chip, struct usb_interface *iface, const struct snd_usb_audio_quirk *quirk); @@ -3406,7 +3353,6 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, - [QUIRK_AUDIO_EDIROL_UA101] = create_ua101_quirk, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk }; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 40ba811..9826337 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -159,7 +159,6 @@ enum quirk_type { QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, QUIRK_AUDIO_EDIROL_UA1000, - QUIRK_AUDIO_EDIROL_UA101, QUIRK_AUDIO_EDIROL_UAXX, QUIRK_TYPE_COUNT diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index a892bda..bd6706c 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1266,37 +1266,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, -/* Roland UA-101 in High-Speed Mode only */ -{ - USB_DEVICE(0x0582, 0x007d), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Roland", - .product_name = "UA-101", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 0, - .type = QUIRK_AUDIO_EDIROL_UA101 - }, - { - .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA101 - }, - { - .ifnum = 2, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0001, - .in_cables = 0x0001 - } - }, - { - .ifnum = -1 - } - } - } -}, { /* has ID 0x0081 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0080), -- cgit v1.1 From 5b0cb1d850c26893b1468b3a519433a1b7a176be Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 8 Dec 2009 16:13:32 +0100 Subject: ALSA: hda - add more NID->Control mapping This set of changes add missing NID values to some static control elemenents. Also, it handles all "Capture Source" or "Input Source" controls. Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_codec.c | 64 +++++++++- sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/hda_generic.c | 3 +- sound/pci/hda/hda_local.h | 5 + sound/pci/hda/hda_proc.c | 23 ++-- sound/pci/hda/patch_analog.c | 31 +++++ sound/pci/hda/patch_cirrus.c | 4 + sound/pci/hda/patch_cmedia.c | 12 +- sound/pci/hda/patch_realtek.c | 120 ++++++++++++++++++- sound/pci/hda/patch_si3054.c | 1 + sound/pci/hda/patch_via.c | 273 +++++++++++++++++++++++++----------------- 11 files changed, 415 insertions(+), 122 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 9cfdb77..20100b1 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -931,6 +931,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) #endif list_del(&codec->list); snd_array_free(&codec->mixers); + snd_array_free(&codec->nids); codec->bus->caddr_tbl[codec->addr] = NULL; if (codec->patch_ops.free) codec->patch_ops.free(codec); @@ -985,7 +986,8 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr mutex_init(&codec->control_mutex); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); - snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 60); + snd_array_init(&codec->mixers, sizeof(struct hda_nid_item), 32); + snd_array_init(&codec->nids, sizeof(struct hda_nid_item), 32); snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); if (codec->bus->modelname) { @@ -1706,7 +1708,7 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, EXPORT_SYMBOL_HDA(snd_hda_find_mixer_ctl); /** - * snd_hda_ctl-add - Add a control element and assign to the codec + * snd_hda_ctl_add - Add a control element and assign to the codec * @codec: HD-audio codec * @nid: corresponding NID (optional) * @kctl: the control element to assign @@ -1747,6 +1749,35 @@ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, EXPORT_SYMBOL_HDA(snd_hda_ctl_add); /** + * snd_hda_add_nid - Assign a NID to a control element + * @codec: HD-audio codec + * @nid: corresponding NID (optional) + * @kctl: the control element to assign + * @index: index to kctl + * + * Add the given control element to an array inside the codec instance. + * This function is used when #snd_hda_ctl_add cannot be used for 1:1 + * NID:KCTL mapping - for example "Capture Source" selector. + */ +int snd_hda_add_nid(struct hda_codec *codec, struct snd_kcontrol *kctl, + unsigned int index, hda_nid_t nid) +{ + struct hda_nid_item *item; + + if (nid > 0) { + item = snd_array_new(&codec->nids); + if (!item) + return -ENOMEM; + item->kctl = kctl; + item->index = index; + item->nid = nid; + return 0; + } + return -EINVAL; +} +EXPORT_SYMBOL_HDA(snd_hda_add_nid); + +/** * snd_hda_ctls_clear - Clear all controls assigned to the given codec * @codec: HD-audio codec */ @@ -1757,6 +1788,7 @@ void snd_hda_ctls_clear(struct hda_codec *codec) for (i = 0; i < codec->mixers.used; i++) snd_ctl_remove(codec->bus->card, items[i].kctl); snd_array_free(&codec->mixers); + snd_array_free(&codec->nids); } /* pseudo device locking @@ -3476,6 +3508,8 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) for (; knew->name; knew++) { struct snd_kcontrol *kctl; + if (knew->iface == -1) /* skip this codec private value */ + continue; kctl = snd_ctl_new1(knew, codec); if (!kctl) return -ENOMEM; @@ -3496,6 +3530,32 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) } EXPORT_SYMBOL_HDA(snd_hda_add_new_ctls); +/** + * snd_hda_add_nids - assign nids to controls from the array + * @codec: the HDA codec + * @kctl: struct snd_kcontrol + * @index: index to kctl + * @nids: the array of hda_nid_t + * @size: count of hda_nid_t items + * + * This helper function assigns NIDs in the given array to a control element. + * + * Returns 0 if successful, or a negative error code. + */ +int snd_hda_add_nids(struct hda_codec *codec, struct snd_kcontrol *kctl, + unsigned int index, hda_nid_t *nids, unsigned int size) +{ + int err; + + for ( ; size > 0; size--, nids++) { + err = snd_hda_add_nid(codec, kctl, index, *nids); + if (err < 0) + return err; + } + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_add_nids); + #ifdef CONFIG_SND_HDA_POWER_SAVE static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 1d541b7..0d08ad5 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -789,6 +789,7 @@ struct hda_codec { u32 *wcaps; struct snd_array mixers; /* list of assigned mixer elements */ + struct snd_array nids; /* list of mapped mixer elements */ struct hda_cache_rec amp_cache; /* cache for amp access */ struct hda_cache_rec cmd_cache; /* cache for other commands */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 092c6a7..5ea2128 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -861,7 +861,8 @@ static int build_input_controls(struct hda_codec *codec) } /* create input MUX if multiple sources are available */ - err = snd_hda_ctl_add(codec, 0, snd_ctl_new1(&cap_sel, codec)); + err = snd_hda_ctl_add(codec, spec->adc_node->nid, + snd_ctl_new1(&cap_sel, codec)); if (err < 0) return err; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 5778ae8..98cf3f4 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -342,6 +342,8 @@ int snd_hda_check_board_codec_sid_config(struct hda_codec *codec, const struct snd_pci_quirk *tbl); int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew); +int snd_hda_add_nids(struct hda_codec *codec, struct snd_kcontrol *kctl, + unsigned int index, hda_nid_t *nids, unsigned int size); /* * unsolicited event handler @@ -466,11 +468,14 @@ int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); struct hda_nid_item { struct snd_kcontrol *kctl; + unsigned int index; hda_nid_t nid; }; int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, struct snd_kcontrol *kctl); +int snd_hda_add_nid(struct hda_codec *codec, struct snd_kcontrol *kctl, + unsigned int index, hda_nid_t nid); void snd_hda_ctls_clear(struct hda_codec *codec); /* diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index c9afc04..2e27d6a 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -61,18 +61,21 @@ static const char *get_wid_type_name(unsigned int wid_value) return "UNKNOWN Widget"; } -static void print_nid_mixers(struct snd_info_buffer *buffer, - struct hda_codec *codec, hda_nid_t nid) +static void print_nid_array(struct snd_info_buffer *buffer, + struct hda_codec *codec, hda_nid_t nid, + struct snd_array *array) { int i; - struct hda_nid_item *items = codec->mixers.list; + struct hda_nid_item *items = array->list, *item; struct snd_kcontrol *kctl; - for (i = 0; i < codec->mixers.used; i++) { - if (items[i].nid == nid) { - kctl = items[i].kctl; + for (i = 0; i < array->used; i++) { + item = &items[i]; + if (item->nid == nid) { + kctl = item->kctl; snd_iprintf(buffer, " Control: name=\"%s\", index=%i, device=%i\n", - kctl->id.name, kctl->id.index, kctl->id.device); + kctl->id.name, kctl->id.index + item->index, + kctl->id.device); } } } @@ -528,7 +531,8 @@ static void print_gpio(struct snd_info_buffer *buffer, (data & (1<mixers); + print_nid_array(buffer, codec, nid, &codec->nids); } static void print_codec_info(struct snd_info_entry *entry, @@ -608,7 +612,8 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, " CP"); snd_iprintf(buffer, "\n"); - print_nid_mixers(buffer, codec, nid); + print_nid_array(buffer, codec, nid, &codec->mixers); + print_nid_array(buffer, codec, nid, &codec->nids); print_nid_pcms(buffer, codec, nid); /* volume knob is a special widget that always have connection diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 447eda1..d418842 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -174,6 +174,7 @@ static struct snd_kcontrol_new ad_beep_mixer[] = { static int ad198x_build_controls(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; + struct snd_kcontrol *kctl; unsigned int i; int err; @@ -239,6 +240,28 @@ static int ad198x_build_controls(struct hda_codec *codec) } ad198x_free_kctls(codec); /* no longer needed */ + + /* assign Capture Source enums to NID */ + kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); + if (!kctl) + kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); + for (i = 0; kctl && i < kctl->count; i++) { + err = snd_hda_add_nids(codec, kctl, i, spec->capsrc_nids, + spec->input_mux->num_items); + if (err < 0) + return err; + } + + /* assign IEC958 enums to NID */ + kctl = snd_hda_find_mixer_ctl(codec, + SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source"); + if (kctl) { + err = snd_hda_add_nid(codec, kctl, 0, + spec->multiout.dig_out_nid); + if (err < 0) + return err; + } + return 0; } @@ -701,6 +724,7 @@ static struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "External Amplifier", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b, .info = ad198x_eapd_info, .get = ad198x_eapd_get, .put = ad198x_eapd_put, @@ -808,6 +832,7 @@ static struct snd_kcontrol_new ad1986a_automute_master_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x1a, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1986a_hp_master_sw_put, @@ -1608,6 +1633,7 @@ static struct snd_kcontrol_new ad1981_hp_mixers[] = { HDA_BIND_VOL("Master Playback Volume", &ad1981_hp_bind_master_vol), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .subdevice = HDA_SUBDEV_NID_FLAG | 0x05, .name = "Master Playback Switch", .info = ad198x_eapd_info, .get = ad198x_eapd_get, @@ -2121,6 +2147,7 @@ static struct snd_kcontrol_new ad1988_laptop_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "External Amplifier", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x12, .info = ad198x_eapd_info, .get = ad198x_eapd_get, .put = ad198x_eapd_put, @@ -2242,6 +2269,7 @@ static struct snd_kcontrol_new ad1988_spdif_out_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "IEC958 Playback Source", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b, .info = ad1988_spdif_playback_source_info, .get = ad1988_spdif_playback_source_get, .put = ad1988_spdif_playback_source_put, @@ -3728,6 +3756,7 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x21, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1884a_mobile_master_sw_put, @@ -3756,6 +3785,7 @@ static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x21, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1884a_mobile_master_sw_put, @@ -4097,6 +4127,7 @@ static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { /* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .subdevice = HDA_SUBDEV_NID_FLAG | 0x21, .name = "Master Playback Switch", .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 4b200da..d0b8c6d 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -759,6 +759,10 @@ static int build_input(struct hda_codec *codec) err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; + err = snd_hda_add_nids(codec, kctl, 0, spec->adc_nid, + spec->num_inputs); + if (err < 0) + return err; } if (spec->num_inputs > 1 && !spec->mic_detect) { diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index a45c116..cc1c223 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -315,7 +315,8 @@ static struct hda_verb cmi9880_allout_init[] = { static int cmi9880_build_controls(struct hda_codec *codec) { struct cmi_spec *spec = codec->spec; - int err; + struct snd_kcontrol *kctl; + int i, err; err = snd_hda_add_new_ctls(codec, cmi9880_basic_mixer); if (err < 0) @@ -340,6 +341,15 @@ static int cmi9880_build_controls(struct hda_codec *codec) if (err < 0) return err; } + + /* assign Capture Source enums to NID */ + kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); + for (i = 0; kctl && i < kctl->count; i++) { + err = snd_hda_add_nids(codec, kctl, i, spec->adc_nids, + spec->input_mux->num_items); + if (err < 0) + return err; + } return 0; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 888b631..6b0b872 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -627,6 +627,7 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, #define ALC_PIN_MODE(xname, nid, dir) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ .info = alc_pin_mode_info, \ .get = alc_pin_mode_get, \ .put = alc_pin_mode_put, \ @@ -678,6 +679,7 @@ static int alc_gpio_data_put(struct snd_kcontrol *kcontrol, } #define ALC_GPIO_DATA_SWITCH(xname, nid, mask) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ .info = alc_gpio_data_info, \ .get = alc_gpio_data_get, \ .put = alc_gpio_data_put, \ @@ -732,6 +734,7 @@ static int alc_spdif_ctrl_put(struct snd_kcontrol *kcontrol, } #define ALC_SPDIF_CTRL_SWITCH(xname, nid, mask) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ .info = alc_spdif_ctrl_info, \ .get = alc_spdif_ctrl_get, \ .put = alc_spdif_ctrl_put, \ @@ -785,6 +788,7 @@ static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol, #define ALC_EAPD_CTRL_SWITCH(xname, nid, mask) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = 0, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ .info = alc_eapd_ctrl_info, \ .get = alc_eapd_ctrl_get, \ .put = alc_eapd_ctrl_put, \ @@ -2410,6 +2414,15 @@ static const char *alc_slave_sws[] = { * build control elements */ +#define NID_MAPPING (-1) + +#define SUBDEV_SPEAKER_ (0 << 6) +#define SUBDEV_HP_ (1 << 6) +#define SUBDEV_LINE_ (2 << 6) +#define SUBDEV_SPEAKER(x) (SUBDEV_SPEAKER_ | ((x) & 0x3f)) +#define SUBDEV_HP(x) (SUBDEV_HP_ | ((x) & 0x3f)) +#define SUBDEV_LINE(x) (SUBDEV_LINE_ | ((x) & 0x3f)) + static void alc_free_kctls(struct hda_codec *codec); #ifdef CONFIG_SND_HDA_INPUT_BEEP @@ -2424,8 +2437,11 @@ static struct snd_kcontrol_new alc_beep_mixer[] = { static int alc_build_controls(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int err; - int i; + struct snd_kcontrol *kctl; + struct snd_kcontrol_new *knew; + int i, j, err; + unsigned int u; + hda_nid_t nid; for (i = 0; i < spec->num_mixers; i++) { err = snd_hda_add_new_ctls(codec, spec->mixers[i]); @@ -2494,6 +2510,73 @@ static int alc_build_controls(struct hda_codec *codec) } alc_free_kctls(codec); /* no longer needed */ + + /* assign Capture Source enums to NID */ + kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); + if (!kctl) + kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); + for (i = 0; kctl && i < kctl->count; i++) { + err = snd_hda_add_nids(codec, kctl, i, spec->capsrc_nids, + spec->input_mux->num_items); + if (err < 0) + return err; + } + if (spec->cap_mixer) { + const char *kname = kctl ? kctl->id.name : NULL; + for (knew = spec->cap_mixer; knew->name; knew++) { + if (kname && strcmp(knew->name, kname) == 0) + continue; + kctl = snd_hda_find_mixer_ctl(codec, knew->name); + for (i = 0; kctl && i < kctl->count; i++) { + err = snd_hda_add_nid(codec, kctl, i, + spec->adc_nids[i]); + if (err < 0) + return err; + } + } + } + + /* other nid->control mapping */ + for (i = 0; i < spec->num_mixers; i++) { + for (knew = spec->mixers[i]; knew->name; knew++) { + if (knew->iface != NID_MAPPING) + continue; + kctl = snd_hda_find_mixer_ctl(codec, knew->name); + if (kctl == NULL) + continue; + u = knew->subdevice; + for (j = 0; j < 4; j++, u >>= 8) { + nid = u & 0x3f; + if (nid == 0) + continue; + switch (u & 0xc0) { + case SUBDEV_SPEAKER_: + nid = spec->autocfg.speaker_pins[nid]; + break; + case SUBDEV_LINE_: + nid = spec->autocfg.line_out_pins[nid]; + break; + case SUBDEV_HP_: + nid = spec->autocfg.hp_pins[nid]; + break; + default: + continue; + } + err = snd_hda_add_nid(codec, kctl, 0, nid); + if (err < 0) + return err; + } + u = knew->private_value; + for (j = 0; j < 4; j++, u >>= 8) { + nid = u & 0xff; + if (nid == 0) + continue; + err = snd_hda_add_nid(codec, kctl, 0, nid); + if (err < 0) + return err; + } + } + } return 0; } @@ -3781,6 +3864,7 @@ static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol, #define PIN_CTL_TEST(xname,nid) { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ .info = alc_test_pin_ctl_info, \ .get = alc_test_pin_ctl_get, \ .put = alc_test_pin_ctl_put, \ @@ -3790,6 +3874,7 @@ static int alc_test_pin_src_put(struct snd_kcontrol *kcontrol, #define PIN_SRC_TEST(xname,nid) { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = xname, \ + .subdevice = HDA_SUBDEV_NID_FLAG | nid, \ .info = alc_test_pin_src_info, \ .get = alc_test_pin_src_get, \ .put = alc_test_pin_src_put, \ @@ -5080,6 +5165,7 @@ static struct snd_kcontrol_new alc260_hp_output_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x11, .info = snd_ctl_boolean_mono_info, .get = alc260_hp_master_sw_get, .put = alc260_hp_master_sw_put, @@ -5118,6 +5204,7 @@ static struct snd_kcontrol_new alc260_hp_3013_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x11, .info = snd_ctl_boolean_mono_info, .get = alc260_hp_master_sw_get, .put = alc260_hp_master_sw_put, @@ -10188,8 +10275,14 @@ static int alc262_hp_master_sw_put(struct snd_kcontrol *kcontrol, .info = snd_ctl_boolean_mono_info, \ .get = alc262_hp_master_sw_get, \ .put = alc262_hp_master_sw_put, \ + }, \ + { \ + .iface = NID_MAPPING, \ + .name = "Master Playback Switch", \ + .private_value = 0x15 | (0x16 << 8) | (0x1b << 16), \ } + static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = { ALC262_HP_MASTER_SWITCH, HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -10347,6 +10440,12 @@ static int alc262_hippo_master_sw_put(struct snd_kcontrol *kcontrol, .info = snd_ctl_boolean_mono_info, \ .get = alc262_hippo_master_sw_get, \ .put = alc262_hippo_master_sw_put, \ + }, \ + { \ + .iface = NID_MAPPING, \ + .name = "Master Playback Switch", \ + .subdevice = SUBDEV_HP(0) | (SUBDEV_LINE(0) << 8) | \ + (SUBDEV_SPEAKER(0) << 16), \ } static struct snd_kcontrol_new alc262_hippo_mixer[] = { @@ -10820,11 +10919,17 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc262_fujitsu_master_sw_put, .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), }, + { + .iface = NID_MAPPING, + .name = "Master Playback Switch", + .private_value = 0x1b, + }, HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), @@ -10855,6 +10960,7 @@ static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc262_lenovo_3000_master_sw_put, @@ -11009,6 +11115,11 @@ static struct snd_kcontrol_new alc262_ultra_capture_mixer[] = { .get = alc_mux_enum_get, .put = alc262_ultra_mux_enum_put, }, + { + .iface = NID_MAPPING, + .name = "Capture Source", + .private_value = 0x15, + }, { } /* end */ }; @@ -12026,6 +12137,7 @@ static struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -12041,6 +12153,7 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -12058,6 +12171,7 @@ static struct snd_kcontrol_new alc268_acer_dmic_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -13010,6 +13124,7 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -13030,6 +13145,7 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 43b436c..f419ee8 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -122,6 +122,7 @@ static int si3054_switch_put(struct snd_kcontrol *kcontrol, #define SI3054_KCONTROL(kname,reg,mask) { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = kname, \ + .subdevice = HDA_SUBDEV_NID_FLAG | reg, \ .info = si3054_switch_info, \ .get = si3054_switch_get, \ .put = si3054_switch_put, \ diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index b70e26a..64995e8 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -54,6 +54,8 @@ #include "hda_codec.h" #include "hda_local.h" +#define NID_MAPPING (-1) + /* amp values */ #define AMP_VAL_IDX_SHIFT 19 #define AMP_VAL_IDX_MASK (0x0f<<19) @@ -157,6 +159,19 @@ struct via_spec { #endif }; +static struct via_spec * via_new_spec(struct hda_codec *codec) +{ + struct via_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return NULL; + + codec->spec = spec; + spec->codec = codec; + return spec; +} + static enum VIA_HDA_CODEC get_codec_type(struct hda_codec *codec) { u32 vendor_id = codec->vendor_id; @@ -448,6 +463,22 @@ static int via_add_control(struct via_spec *spec, int type, const char *name, return 0; } +static struct snd_kcontrol_new *via_clone_control(struct via_spec *spec, + struct snd_kcontrol_new *tmpl) +{ + struct snd_kcontrol_new *knew; + + snd_array_init(&spec->kctls, sizeof(*knew), 32); + knew = snd_array_new(&spec->kctls); + if (!knew) + return NULL; + *knew = *tmpl; + knew->name = kstrdup(tmpl->name, GFP_KERNEL); + if (!knew->name) + return NULL; + return 0; +} + static void via_free_kctls(struct hda_codec *codec) { struct via_spec *spec = codec->spec; @@ -1088,24 +1119,9 @@ static int via_independent_hp_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct via_spec *spec = codec->spec; - hda_nid_t nid; + hda_nid_t nid = kcontrol->private_value; unsigned int pinsel; - switch (spec->codec_type) { - case VT1718S: - nid = 0x34; - break; - case VT2002P: - nid = 0x35; - break; - case VT1812: - nid = 0x3d; - break; - default: - nid = spec->autocfg.hp_pins[0]; - break; - } /* use !! to translate conn sel 2 for VT1718S */ pinsel = !!snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, @@ -1127,29 +1143,24 @@ static void activate_ctl(struct hda_codec *codec, const char *name, int active) } } +static hda_nid_t side_mute_channel(struct via_spec *spec) +{ + switch (spec->codec_type) { + case VT1708: return 0x1b; + case VT1709_10CH: return 0x29; + case VT1708B_8CH: /* fall thru */ + case VT1708S: return 0x27; + default: return 0; + } +} + static int update_side_mute_status(struct hda_codec *codec) { /* mute side channel */ struct via_spec *spec = codec->spec; unsigned int parm = spec->hp_independent_mode ? AMP_OUT_MUTE : AMP_OUT_UNMUTE; - hda_nid_t sw3; - - switch (spec->codec_type) { - case VT1708: - sw3 = 0x1b; - break; - case VT1709_10CH: - sw3 = 0x29; - break; - case VT1708B_8CH: - case VT1708S: - sw3 = 0x27; - break; - default: - sw3 = 0; - break; - } + hda_nid_t sw3 = side_mute_channel(spec); if (sw3) snd_hda_codec_write(codec, sw3, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -1162,28 +1173,11 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct via_spec *spec = codec->spec; - hda_nid_t nid = spec->autocfg.hp_pins[0]; + hda_nid_t nid = kcontrol->private_value; unsigned int pinsel = ucontrol->value.enumerated.item[0]; /* Get Independent Mode index of headphone pin widget */ spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel ? 1 : 0; - - switch (spec->codec_type) { - case VT1718S: - nid = 0x34; - pinsel = pinsel ? 2 : 0; /* indep HP use AOW4 (index 2) */ - spec->multiout.num_dacs = 4; - break; - case VT2002P: - nid = 0x35; - break; - case VT1812: - nid = 0x3d; - break; - default: - nid = spec->autocfg.hp_pins[0]; - break; - } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); if (spec->multiout.hp_nid && spec->multiout.hp_nid @@ -1207,18 +1201,55 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, return 0; } -static struct snd_kcontrol_new via_hp_mixer[] = { +static struct snd_kcontrol_new via_hp_mixer[2] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Independent HP", - .count = 1, .info = via_independent_hp_info, .get = via_independent_hp_get, .put = via_independent_hp_put, }, - { } /* end */ + { + .iface = NID_MAPPING, + .name = "Independent HP", + }, }; +static int via_hp_build(struct via_spec *spec) +{ + struct snd_kcontrol_new *knew; + hda_nid_t nid; + + knew = via_clone_control(spec, &via_hp_mixer[0]); + if (knew == NULL) + return -ENOMEM; + + switch (spec->codec_type) { + case VT1718S: + nid = 0x34; + break; + case VT2002P: + nid = 0x35; + break; + case VT1812: + nid = 0x3d; + break; + default: + nid = spec->autocfg.hp_pins[0]; + break; + } + + knew->subdevice = HDA_SUBDEV_NID_FLAG | nid; + knew->private_value = nid; + + knew = via_clone_control(spec, &via_hp_mixer[1]); + if (knew == NULL) + return -ENOMEM; + knew->subdevice = side_mute_channel(spec); + + return 0; +} + static void notify_aa_path_ctls(struct hda_codec *codec) { int i; @@ -1376,7 +1407,7 @@ static int via_smart51_put(struct snd_kcontrol *kcontrol, return 1; } -static struct snd_kcontrol_new via_smart51_mixer[] = { +static struct snd_kcontrol_new via_smart51_mixer[2] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Smart 5.1", @@ -1385,9 +1416,36 @@ static struct snd_kcontrol_new via_smart51_mixer[] = { .get = via_smart51_get, .put = via_smart51_put, }, - {} /* end */ + { + .iface = NID_MAPPING, + .name = "Smart 5.1", + } }; +static int via_smart51_build(struct via_spec *spec) +{ + struct snd_kcontrol_new *knew; + int index[] = { AUTO_PIN_MIC, AUTO_PIN_FRONT_MIC, AUTO_PIN_LINE }; + hda_nid_t nid; + int i; + + knew = via_clone_control(spec, &via_smart51_mixer[0]); + if (knew == NULL) + return -ENOMEM; + + for (i = 0; i < ARRAY_SIZE(index); i++) { + nid = spec->autocfg.input_pins[index[i]]; + if (nid) { + knew = via_clone_control(spec, &via_smart51_mixer[1]); + if (knew == NULL) + return -ENOMEM; + knew->subdevice = nid; + } + } + + return 0; +} + /* capture mixer elements */ static struct snd_kcontrol_new vt1708_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_INPUT), @@ -1819,8 +1877,9 @@ static struct hda_pcm_stream vt1708_pcm_digital_capture = { static int via_build_controls(struct hda_codec *codec) { struct via_spec *spec = codec->spec; - int err; - int i; + struct snd_kcontrol *kctl; + struct snd_kcontrol_new *knew; + int err, i; for (i = 0; i < spec->num_mixers; i++) { err = snd_hda_add_new_ctls(codec, spec->mixers[i]); @@ -1845,6 +1904,28 @@ static int via_build_controls(struct hda_codec *codec) return err; } + /* assign Capture Source enums to NID */ + kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); + for (i = 0; kctl && i < kctl->count; i++) { + err = snd_hda_add_nids(codec, kctl, i, spec->mux_nids, + spec->input_mux->num_items); + if (err < 0) + return err; + } + + /* other nid->control mapping */ + for (i = 0; i < spec->num_mixers; i++) { + for (knew = spec->mixers[i]; knew->name; knew++) { + if (knew->iface != NID_MAPPING) + continue; + kctl = snd_hda_find_mixer_ctl(codec, knew->name); + if (kctl == NULL) + continue; + err = snd_hda_add_nid(codec, kctl, 0, + knew->subdevice); + } + } + /* init power states */ set_jack_power_state(codec); analog_low_current_mode(codec, 1); @@ -2481,9 +2562,9 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); - spec->mixers[spec->num_mixers++] = via_smart51_mixer; + via_smart51_build(spec); return 1; } @@ -2554,12 +2635,10 @@ static int patch_vt1708(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1708_parse_auto_config(codec); if (err < 0) { @@ -2597,7 +2676,6 @@ static int patch_vt1708(struct hda_codec *codec) #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt1708_loopbacks; #endif - spec->codec = codec; INIT_DELAYED_WORK(&spec->vt1708_hp_work, vt1708_update_hp_jack_state); return 0; } @@ -3010,9 +3088,9 @@ static int vt1709_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); - spec->mixers[spec->num_mixers++] = via_smart51_mixer; + via_smart51_build(spec); return 1; } @@ -3032,12 +3110,10 @@ static int patch_vt1709_10ch(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - err = vt1709_parse_auto_config(codec); if (err < 0) { via_free(codec); @@ -3126,12 +3202,10 @@ static int patch_vt1709_6ch(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - err = vt1709_parse_auto_config(codec); if (err < 0) { via_free(codec); @@ -3581,9 +3655,9 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); - spec->mixers[spec->num_mixers++] = via_smart51_mixer; + via_smart51_build(spec); return 1; } @@ -3605,12 +3679,10 @@ static int patch_vt1708B_8ch(struct hda_codec *codec) if (get_codec_type(codec) == VT1708BCE) return patch_vt1708S(codec); /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1708B_parse_auto_config(codec); if (err < 0) { @@ -3657,12 +3729,10 @@ static int patch_vt1708B_4ch(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1708B_parse_auto_config(codec); if (err < 0) { @@ -4071,9 +4141,9 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); - spec->mixers[spec->num_mixers++] = via_smart51_mixer; + via_smart51_build(spec); return 1; } @@ -4103,12 +4173,10 @@ static int patch_vt1708S(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1708S_parse_auto_config(codec); if (err < 0) { @@ -4443,7 +4511,7 @@ static int vt1702_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); return 1; } @@ -4464,12 +4532,10 @@ static int patch_vt1702(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1702_parse_auto_config(codec); if (err < 0) { @@ -4865,9 +4931,9 @@ static int vt1718S_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); - spec->mixers[spec->num_mixers++] = via_smart51_mixer; + via_smart51_build(spec); return 1; } @@ -4888,12 +4954,10 @@ static int patch_vt1718S(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1718S_parse_auto_config(codec); if (err < 0) { @@ -5014,6 +5078,7 @@ static struct snd_kcontrol_new vt1716s_dmic_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Digital Mic Capture Switch", + .subdevice = HDA_SUBDEV_NID_FLAG | 0x26, .count = 1, .info = vt1716s_dmic_info, .get = vt1716s_dmic_get, @@ -5361,9 +5426,9 @@ static int vt1716S_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); - spec->mixers[spec->num_mixers++] = via_smart51_mixer; + via_smart51_build(spec); return 1; } @@ -5384,12 +5449,10 @@ static int patch_vt1716S(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1716S_parse_auto_config(codec); if (err < 0) { @@ -5719,7 +5782,7 @@ static int vt2002P_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); return 1; } @@ -5741,12 +5804,10 @@ static int patch_vt2002P(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt2002P_parse_auto_config(codec); if (err < 0) { @@ -6070,7 +6131,7 @@ static int vt1812_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux[0]; if (spec->hp_mux) - spec->mixers[spec->num_mixers++] = via_hp_mixer; + via_hp_build(spec); return 1; } @@ -6092,12 +6153,10 @@ static int patch_vt1812(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kzalloc(sizeof(*spec), GFP_KERNEL); + spec = via_new_spec(codec); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - /* automatic parse from the BIOS config */ err = vt1812_parse_auto_config(codec); if (err < 0) { -- cgit v1.1 From 9e3fd8719f624a43575b56a4777b1552399a8be8 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 8 Dec 2009 17:45:25 +0100 Subject: ALSA: hda - introduce HDA_SUBDEV_AMP_FLAG (ControlAmp in proc) The purpose of this changeset is to show information about amplifier setting in the codec proc file. Something like: Control: name="Front Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Front Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_codec.c | 14 +++++++++----- sound/pci/hda/hda_local.h | 11 ++++++++--- sound/pci/hda/hda_proc.c | 8 ++++++++ sound/pci/hda/patch_analog.c | 12 +++++++----- sound/pci/hda/patch_cirrus.c | 2 ++ sound/pci/hda/patch_conexant.c | 1 + sound/pci/hda/patch_realtek.c | 18 ++++++++++-------- sound/pci/hda/patch_sigmatel.c | 3 ++- sound/pci/hda/patch_via.c | 4 +++- 9 files changed, 50 insertions(+), 23 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 20100b1..c9af15e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1723,19 +1723,22 @@ EXPORT_SYMBOL_HDA(snd_hda_find_mixer_ctl); * * snd_hda_ctl_add() checks the control subdev id field whether * #HDA_SUBDEV_NID_FLAG bit is set. If set (and @nid is zero), the lower - * bits value is taken as the NID to assign. + * bits value is taken as the NID to assign. The #HDA_NID_ITEM_AMP bit + * specifies if kctl->private_value is a HDA amplifier value. */ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, struct snd_kcontrol *kctl) { int err; + unsigned short flags = 0; struct hda_nid_item *item; - if (kctl->id.subdevice & HDA_SUBDEV_NID_FLAG) { - if (nid == 0) - nid = kctl->id.subdevice & 0xffff; + if (kctl->id.subdevice & HDA_SUBDEV_AMP_FLAG) + flags |= HDA_NID_ITEM_AMP; + if ((kctl->id.subdevice & HDA_SUBDEV_NID_FLAG) != 0 && nid == 0) + nid = kctl->id.subdevice & 0xffff; + if (kctl->id.subdevice & 0xf0000000) kctl->id.subdevice = 0; - } err = snd_ctl_add(codec->bus->card, kctl); if (err < 0) return err; @@ -1744,6 +1747,7 @@ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, return -ENOMEM; item->kctl = kctl; item->nid = nid; + item->flags = flags; return 0; } EXPORT_SYMBOL_HDA(snd_hda_ctl_add); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 98cf3f4..0a25647 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -31,6 +31,7 @@ * in snd_hda_ctl_add(), so that this value won't appear in the outside. */ #define HDA_SUBDEV_NID_FLAG (1U << 31) +#define HDA_SUBDEV_AMP_FLAG (1U << 30) /* * for mixer controls @@ -42,7 +43,7 @@ /* mono volume with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = HDA_SUBDEV_NID_FLAG | (nid), \ + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | (nid), \ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ @@ -63,7 +64,7 @@ /* mono mute switch with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = HDA_SUBDEV_NID_FLAG | (nid), \ + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | (nid), \ .info = snd_hda_mixer_amp_switch_info, \ .get = snd_hda_mixer_amp_switch_get, \ .put = snd_hda_mixer_amp_switch_put, \ @@ -81,7 +82,7 @@ /* special beep mono mute switch with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_MUTE_BEEP_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = HDA_SUBDEV_NID_FLAG | (nid), \ + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | (nid), \ .info = snd_hda_mixer_amp_switch_info, \ .get = snd_hda_mixer_amp_switch_get, \ .put = snd_hda_mixer_amp_switch_put_beep, \ @@ -466,10 +467,14 @@ u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid); u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid); int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid); +/* flags for hda_nid_item */ +#define HDA_NID_ITEM_AMP (1<<0) + struct hda_nid_item { struct snd_kcontrol *kctl; unsigned int index; hda_nid_t nid; + unsigned short flags; }; int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 2e27d6a..f97d35d 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -76,6 +76,14 @@ static void print_nid_array(struct snd_info_buffer *buffer, " Control: name=\"%s\", index=%i, device=%i\n", kctl->id.name, kctl->id.index + item->index, kctl->id.device); + if (item->flags & HDA_NID_ITEM_AMP) + snd_iprintf(buffer, + " ControlAmp: chs=%lu, dir=%s, " + "idx=%lu, ofs=%lu\n", + get_amp_channels(kctl), + get_amp_direction(kctl) ? "Out" : "In", + get_amp_index(kctl), + get_amp_offset(kctl)); } } } diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index d418842..5e2bb18 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -832,7 +832,7 @@ static struct snd_kcontrol_new ad1986a_automute_master_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x1a, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x1a, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1986a_hp_master_sw_put, @@ -2602,7 +2602,9 @@ static int add_control(struct ad198x_spec *spec, int type, const char *name, if (! knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = HDA_SUBDEV_NID_FLAG | get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_NID_FLAG | + HDA_SUBDEV_AMP_FLAG | + get_amp_nid_(val); knew->private_value = val; return 0; } @@ -3756,7 +3758,7 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x21, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x21, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1884a_mobile_master_sw_put, @@ -3785,7 +3787,7 @@ static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x21, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x21, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1884a_mobile_master_sw_put, @@ -4127,7 +4129,7 @@ static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { /* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .subdevice = HDA_SUBDEV_NID_FLAG | 0x21, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x21, .name = "Master Playback Switch", .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index d0b8c6d..e51f665 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -500,6 +500,7 @@ static int add_mute(struct hda_codec *codec, const char *name, int index, knew.private_value = pval; snprintf(tmp, sizeof(tmp), "%s %s Switch", name, dir_sfx[dir]); *kctlp = snd_ctl_new1(&knew, codec); + (*kctlp)->id.subdevice = HDA_SUBDEV_AMP_FLAG; return snd_hda_ctl_add(codec, get_amp_nid_(pval), *kctlp); } @@ -513,6 +514,7 @@ static int add_volume(struct hda_codec *codec, const char *name, knew.private_value = pval; snprintf(tmp, sizeof(tmp), "%s %s Volume", name, dir_sfx[dir]); *kctlp = snd_ctl_new1(&knew, codec); + (*kctlp)->id.subdevice = HDA_SUBDEV_AMP_FLAG; return snd_hda_ctl_add(codec, get_amp_nid_(pval), *kctlp); } diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index a09c03c..b68650a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2178,6 +2178,7 @@ static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x10, .info = snd_hda_mixer_amp_volume_info, .get = snd_hda_mixer_amp_volume_get, .put = snd_hda_mixer_amp_volume_put, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6b0b872..87bf7bd 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4414,7 +4414,9 @@ static int add_control(struct alc_spec *spec, int type, const char *name, if (!knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = HDA_SUBDEV_NID_FLAG | get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_NID_FLAG | + HDA_SUBDEV_AMP_FLAG | + get_amp_nid_(val); knew->private_value = val; return 0; } @@ -10919,7 +10921,7 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc262_fujitsu_master_sw_put, @@ -10960,7 +10962,7 @@ static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x1b, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x1b, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc262_lenovo_3000_master_sw_put, @@ -12137,7 +12139,7 @@ static struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -12153,7 +12155,7 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -12171,7 +12173,7 @@ static struct snd_kcontrol_new alc268_acer_dmic_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -13124,7 +13126,7 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -13145,7 +13147,7 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | 0x14, + .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 3d59f83..1ee586b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2702,7 +2702,8 @@ stac_control_new(struct sigmatel_spec *spec, return NULL; } if (nid) - knew->subdevice = HDA_SUBDEV_NID_FLAG | nid; + knew->subdevice = HDA_SUBDEV_NID_FLAG | + HDA_SUBDEV_AMP_FLAG | nid; return knew; } diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 64995e8..b94cdee 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -458,7 +458,9 @@ static int via_add_control(struct via_spec *spec, int type, const char *name, if (!knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = HDA_SUBDEV_NID_FLAG | get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_NID_FLAG | + HDA_SUBDEV_AMP_FLAG | + get_amp_nid_(val); knew->private_value = val; return 0; } -- cgit v1.1 From 5e26dfd0615868872cb44842f1e1428c7b414ab0 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 10 Dec 2009 13:57:01 +0100 Subject: ALSA: hda - simplify usage of HDA_SUBDEV_AMP_FLAG The HDA_SUBDEV_NID_FLAG is duplicate for amplifier control elements. Move get_amp_nid_() call to the snd_hda_ctl_add() function. Signed-off-by: Jaroslav Kysela --- sound/pci/hda/hda_codec.c | 7 +++++-- sound/pci/hda/hda_local.h | 6 +++--- sound/pci/hda/patch_analog.c | 16 ++++++---------- sound/pci/hda/patch_cirrus.c | 4 ++-- sound/pci/hda/patch_conexant.c | 2 +- sound/pci/hda/patch_realtek.c | 21 +++++++++------------ sound/pci/hda/patch_sigmatel.c | 8 +++----- sound/pci/hda/patch_via.c | 4 +--- 8 files changed, 30 insertions(+), 38 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c9af15e..c848ec0 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1733,11 +1733,14 @@ int snd_hda_ctl_add(struct hda_codec *codec, hda_nid_t nid, unsigned short flags = 0; struct hda_nid_item *item; - if (kctl->id.subdevice & HDA_SUBDEV_AMP_FLAG) + if (kctl->id.subdevice & HDA_SUBDEV_AMP_FLAG) { flags |= HDA_NID_ITEM_AMP; + if (nid == 0) + nid = get_amp_nid_(kctl->private_value); + } if ((kctl->id.subdevice & HDA_SUBDEV_NID_FLAG) != 0 && nid == 0) nid = kctl->id.subdevice & 0xffff; - if (kctl->id.subdevice & 0xf0000000) + if (kctl->id.subdevice & (HDA_SUBDEV_NID_FLAG|HDA_SUBDEV_AMP_FLAG)) kctl->id.subdevice = 0; err = snd_ctl_add(codec->bus->card, kctl); if (err < 0) diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 0a25647..d505d05 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -43,7 +43,7 @@ /* mono volume with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | (nid), \ + .subdevice = HDA_SUBDEV_AMP_FLAG, \ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ @@ -64,7 +64,7 @@ /* mono mute switch with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | (nid), \ + .subdevice = HDA_SUBDEV_AMP_FLAG, \ .info = snd_hda_mixer_amp_switch_info, \ .get = snd_hda_mixer_amp_switch_get, \ .put = snd_hda_mixer_amp_switch_put, \ @@ -82,7 +82,7 @@ /* special beep mono mute switch with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_MUTE_BEEP_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | (nid), \ + .subdevice = HDA_SUBDEV_AMP_FLAG, \ .info = snd_hda_mixer_amp_switch_info, \ .get = snd_hda_mixer_amp_switch_get, \ .put = snd_hda_mixer_amp_switch_put_beep, \ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 5e2bb18..e75b5e5 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -209,9 +209,7 @@ static int ad198x_build_controls(struct hda_codec *codec) if (!kctl) return -ENOMEM; kctl->private_value = spec->beep_amp; - err = snd_hda_ctl_add(codec, - get_amp_nid_(spec->beep_amp), - kctl); + err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; } @@ -832,7 +830,7 @@ static struct snd_kcontrol_new ad1986a_automute_master_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x1a, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1986a_hp_master_sw_put, @@ -2602,9 +2600,7 @@ static int add_control(struct ad198x_spec *spec, int type, const char *name, if (! knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = HDA_SUBDEV_NID_FLAG | - HDA_SUBDEV_AMP_FLAG | - get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_AMP_FLAG; knew->private_value = val; return 0; } @@ -3758,7 +3754,7 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x21, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1884a_mobile_master_sw_put, @@ -3787,7 +3783,7 @@ static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x21, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = ad1884a_mobile_master_sw_put, @@ -4129,7 +4125,7 @@ static struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = { /* HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x21, + .subdevice = HDA_SUBDEV_AMP_FLAG, .name = "Master Playback Switch", .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index e51f665..eeb91f6 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -501,7 +501,7 @@ static int add_mute(struct hda_codec *codec, const char *name, int index, snprintf(tmp, sizeof(tmp), "%s %s Switch", name, dir_sfx[dir]); *kctlp = snd_ctl_new1(&knew, codec); (*kctlp)->id.subdevice = HDA_SUBDEV_AMP_FLAG; - return snd_hda_ctl_add(codec, get_amp_nid_(pval), *kctlp); + return snd_hda_ctl_add(codec, 0, *kctlp); } static int add_volume(struct hda_codec *codec, const char *name, @@ -515,7 +515,7 @@ static int add_volume(struct hda_codec *codec, const char *name, snprintf(tmp, sizeof(tmp), "%s %s Volume", name, dir_sfx[dir]); *kctlp = snd_ctl_new1(&knew, codec); (*kctlp)->id.subdevice = HDA_SUBDEV_AMP_FLAG; - return snd_hda_ctl_add(codec, get_amp_nid_(pval), *kctlp); + return snd_hda_ctl_add(codec, 0, *kctlp); } static void fix_volume_caps(struct hda_codec *codec, hda_nid_t dac) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index b68650a..1ab2958 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2178,7 +2178,7 @@ static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = { .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x10, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_volume_info, .get = snd_hda_mixer_amp_volume_get, .put = snd_hda_mixer_amp_volume_put, diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 87bf7bd..cb76795 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2482,8 +2482,7 @@ static int alc_build_controls(struct hda_codec *codec) if (!kctl) return -ENOMEM; kctl->private_value = spec->beep_amp; - err = snd_hda_ctl_add(codec, - get_amp_nid_(spec->beep_amp), kctl); + err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; } @@ -4414,9 +4413,7 @@ static int add_control(struct alc_spec *spec, int type, const char *name, if (!knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = HDA_SUBDEV_NID_FLAG | - HDA_SUBDEV_AMP_FLAG | - get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_AMP_FLAG; knew->private_value = val; return 0; } @@ -10921,7 +10918,7 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc262_fujitsu_master_sw_put, @@ -10962,7 +10959,7 @@ static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x1b, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc262_lenovo_3000_master_sw_put, @@ -12139,7 +12136,7 @@ static struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -12155,7 +12152,7 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -12173,7 +12170,7 @@ static struct snd_kcontrol_new alc268_acer_dmic_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -13126,7 +13123,7 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, @@ -13147,7 +13144,7 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", - .subdevice = HDA_SUBDEV_NID_FLAG | HDA_SUBDEV_AMP_FLAG | 0x14, + .subdevice = HDA_SUBDEV_AMP_FLAG, .info = snd_hda_mixer_amp_switch_info, .get = snd_hda_mixer_amp_switch_get, .put = alc268_acer_master_sw_put, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 1ee586b..0bafea9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2685,7 +2685,7 @@ static struct snd_kcontrol_new * stac_control_new(struct sigmatel_spec *spec, struct snd_kcontrol_new *ktemp, const char *name, - hda_nid_t nid) + unsigned int subdev) { struct snd_kcontrol_new *knew; @@ -2701,9 +2701,7 @@ stac_control_new(struct sigmatel_spec *spec, spec->kctls.alloced--; return NULL; } - if (nid) - knew->subdevice = HDA_SUBDEV_NID_FLAG | - HDA_SUBDEV_AMP_FLAG | nid; + knew->subdevice = subdev; return knew; } @@ -2713,7 +2711,7 @@ static int stac92xx_add_control_temp(struct sigmatel_spec *spec, unsigned long val) { struct snd_kcontrol_new *knew = stac_control_new(spec, ktemp, name, - get_amp_nid_(val)); + HDA_SUBDEV_AMP_FLAG); if (!knew) return -ENOMEM; knew->index = idx; diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index b94cdee..de4839e 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -458,9 +458,7 @@ static int via_add_control(struct via_spec *spec, int type, const char *name, if (!knew->name) return -ENOMEM; if (get_amp_nid_(val)) - knew->subdevice = HDA_SUBDEV_NID_FLAG | - HDA_SUBDEV_AMP_FLAG | - get_amp_nid_(val); + knew->subdevice = HDA_SUBDEV_AMP_FLAG; knew->private_value = val; return 0; } -- cgit v1.1 From 926a01ce1ef5e27281af0270e4476979c0522954 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 16 Dec 2009 16:15:00 +0100 Subject: ALSA: Release v1.0.22 Signed-off-by: Jaroslav Kysela --- include/sound/version.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/version.h b/include/sound/version.h index 2293914..1f5d4872 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.21" +#define CONFIG_SND_VERSION "1.0.22" #define CONFIG_SND_DATE "" -- cgit v1.1 From 6c941c8556dd9269be621cd8159fc60e955a91b3 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 16 Dec 2009 16:15:00 +0100 Subject: ALSA: Release v1.0.22 Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- include/sound/version.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/version.h b/include/sound/version.h index 2293914..1f5d4872 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.21" +#define CONFIG_SND_VERSION "1.0.22" #define CONFIG_SND_DATE "" -- cgit v1.1 From d1409ae4cecb4af260759bdfdf88fafca23a9940 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 17 Dec 2009 15:01:31 +0100 Subject: ALSA: hda - Fix NULL dereference in kctl-NID mapping in patch_realtek.c capsrc_nids can be NULL, and adc_nids should be taken as fallback. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 36556b1..0124352 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2517,7 +2517,10 @@ static int alc_build_controls(struct hda_codec *codec) if (!kctl) kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); for (i = 0; kctl && i < kctl->count; i++) { - err = snd_hda_add_nids(codec, kctl, i, spec->capsrc_nids, + hda_nid_t *nids = spec->capsrc_nids; + if (!nids) + nids = spec->adc_nids; + err = snd_hda_add_nids(codec, kctl, i, nids, spec->input_mux->num_items); if (err < 0) return err; -- cgit v1.1 From 681b84e17747e1c208e8e1acc54cc5e612da84d1 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:29:00 +0100 Subject: sound: pcm: add vmalloc buffer helper functions There are now five copies of the code to allocate a PCM buffer using vmalloc(). Add a sixth in the core so that the others can be removed. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 38 ++++++++++++++++++++++++++++++++++ sound/core/pcm_memory.c | 54 +++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 92 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c83a4a7..0ad2d28 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -905,6 +905,44 @@ int snd_pcm_lib_preallocate_pages_for_all(struct snd_pcm *pcm, int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size); int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream); +int _snd_pcm_lib_alloc_vmalloc_buffer(struct snd_pcm_substream *substream, + size_t size, gfp_t gfp_flags); +int snd_pcm_lib_free_vmalloc_buffer(struct snd_pcm_substream *substream); +struct page *snd_pcm_lib_get_vmalloc_page(struct snd_pcm_substream *substream, + unsigned long offset); +#if 0 /* for kernel-doc */ +/** + * snd_pcm_lib_alloc_vmalloc_buffer - allocate virtual DMA buffer + * @substream: the substream to allocate the buffer to + * @size: the requested buffer size, in bytes + * + * Allocates the PCM substream buffer using vmalloc(), i.e., the memory is + * contiguous in kernel virtual space, but not in physical memory. Use this + * if the buffer is accessed by kernel code but not by device DMA. + * + * Returns 1 if the buffer was changed, 0 if not changed, or a negative error + * code. + */ +static int snd_pcm_lib_alloc_vmalloc_buffer + (struct snd_pcm_substream *substream, size_t size); +/** + * snd_pcm_lib_alloc_vmalloc_32_buffer - allocate 32-bit-addressable buffer + * @substream: the substream to allocate the buffer to + * @size: the requested buffer size, in bytes + * + * This function works like snd_pcm_lib_alloc_vmalloc_buffer(), but uses + * vmalloc_32(), i.e., the pages are allocated from 32-bit-addressable memory. + */ +static int snd_pcm_lib_alloc_vmalloc_32_buffer + (struct snd_pcm_substream *substream, size_t size); +#endif +#define snd_pcm_lib_alloc_vmalloc_buffer(subs, size) \ + _snd_pcm_lib_alloc_vmalloc_buffer \ + (subs, size, GFP_KERNEL | __GFP_HIGHMEM | __GFP_ZERO) +#define snd_pcm_lib_alloc_vmalloc_32_buffer(subs, size) \ + _snd_pcm_lib_alloc_vmalloc_buffer \ + (subs, size, GFP_KERNEL | GFP_DMA32 | __GFP_ZERO) + #ifdef CONFIG_SND_DMA_SGBUF /* * SG-buffer handling diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index caa7796..d9727c7 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -434,3 +434,57 @@ int snd_pcm_lib_free_pages(struct snd_pcm_substream *substream) } EXPORT_SYMBOL(snd_pcm_lib_free_pages); + +int _snd_pcm_lib_alloc_vmalloc_buffer(struct snd_pcm_substream *substream, + size_t size, gfp_t gfp_flags) +{ + struct snd_pcm_runtime *runtime; + + if (PCM_RUNTIME_CHECK(substream)) + return -EINVAL; + runtime = substream->runtime; + if (runtime->dma_area) { + if (runtime->dma_bytes >= size) + return 0; /* already large enough */ + vfree(runtime->dma_area); + } + runtime->dma_area = __vmalloc(size, gfp_flags, PAGE_KERNEL); + if (!runtime->dma_area) + return -ENOMEM; + runtime->dma_bytes = size; + return 1; +} +EXPORT_SYMBOL(_snd_pcm_lib_alloc_vmalloc_buffer); + +/** + * snd_pcm_lib_free_vmalloc_buffer - free vmalloc buffer + * @substream: the substream with a buffer allocated by + * snd_pcm_lib_alloc_vmalloc_buffer() + */ +int snd_pcm_lib_free_vmalloc_buffer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime; + + if (PCM_RUNTIME_CHECK(substream)) + return -EINVAL; + runtime = substream->runtime; + vfree(runtime->dma_area); + runtime->dma_area = NULL; + return 0; +} +EXPORT_SYMBOL(snd_pcm_lib_free_vmalloc_buffer); + +/** + * snd_pcm_lib_get_vmalloc_page - map vmalloc buffer offset to page struct + * @substream: the substream with a buffer allocated by + * snd_pcm_lib_alloc_vmalloc_buffer() + * @offset: offset in the buffer + * + * This function is to be used as the page callback in the PCM ops. + */ +struct page *snd_pcm_lib_get_vmalloc_page(struct snd_pcm_substream *substream, + unsigned long offset) +{ + return vmalloc_to_page(substream->runtime->dma_area + offset); +} +EXPORT_SYMBOL(snd_pcm_lib_get_vmalloc_page); -- cgit v1.1 From d20fb5dc076a4cf0fdd00ab5a4e752ea3611e484 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:29:49 +0100 Subject: sound: pdaudiocf: use vmalloc buffer helper functions Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the equivalent core functions instead. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 51 +++------------------------------- 1 file changed, 4 insertions(+), 47 deletions(-) diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 5cfa608..0afa683 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -21,7 +21,6 @@ */ #include -#include #include #include #include @@ -29,49 +28,6 @@ /* - * we use a vmalloc'ed (sg-)buffer - */ - -/* get the physical page pointer on the given offset */ -static struct page *snd_pcm_get_vmalloc_page(struct snd_pcm_substream *subs, unsigned long offset) -{ - void *pageptr = subs->runtime->dma_area + offset; - return vmalloc_to_page(pageptr); -} - -/* - * hw_params callback - * NOTE: this may be called not only once per pcm open! - */ -static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - if (runtime->dma_area) { - if (runtime->dma_bytes >= size) - return 0; /* already enough large */ - vfree(runtime->dma_area); - } - runtime->dma_area = vmalloc_32_user(size); - if (! runtime->dma_area) - return -ENOMEM; - runtime->dma_bytes = size; - return 0; -} - -/* - * hw_free callback - * NOTE: this may be called not only once per pcm open! - */ -static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - - vfree(runtime->dma_area); - runtime->dma_area = NULL; - return 0; -} - -/* * clear the SRAM contents */ static int pdacf_pcm_clear_sram(struct snd_pdacf *chip) @@ -147,7 +103,8 @@ static int pdacf_pcm_trigger(struct snd_pcm_substream *subs, int cmd) static int pdacf_pcm_hw_params(struct snd_pcm_substream *subs, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_alloc_vmalloc_buffer(subs, params_buffer_bytes(hw_params)); + return snd_pcm_lib_alloc_vmalloc_32_buffer + (subs, params_buffer_bytes(hw_params)); } /* @@ -155,7 +112,7 @@ static int pdacf_pcm_hw_params(struct snd_pcm_substream *subs, */ static int pdacf_pcm_hw_free(struct snd_pcm_substream *subs) { - return snd_pcm_free_vmalloc_buffer(subs); + return snd_pcm_lib_free_vmalloc_buffer(subs); } /* @@ -319,7 +276,7 @@ static struct snd_pcm_ops pdacf_pcm_capture_ops = { .prepare = pdacf_pcm_prepare, .trigger = pdacf_pcm_trigger, .pointer = pdacf_pcm_capture_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; -- cgit v1.1 From 6cedf8696d6a01bba391bdae06231243cfe2f48a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:30:24 +0100 Subject: sound: sgio2audio: use vmalloc buffer helper functions Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the equivalent core functions instead. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/mips/sgio2audio.c | 31 ++++++------------------------- 1 file changed, 6 insertions(+), 25 deletions(-) diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index f1d9d16..9b486be 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -26,7 +26,6 @@ #include #include #include -#include #include #include #include @@ -603,25 +602,14 @@ static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream) static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { - struct snd_pcm_runtime *runtime = substream->runtime; - int size = params_buffer_bytes(hw_params); - - /* alloc virtual 'dma' area */ - if (runtime->dma_area) - vfree(runtime->dma_area); - runtime->dma_area = vmalloc_user(size); - if (runtime->dma_area == NULL) - return -ENOMEM; - runtime->dma_bytes = size; - return 0; + return snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); } /* hw_free callback */ static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream) { - vfree(substream->runtime->dma_area); - substream->runtime->dma_area = NULL; - return 0; + return snd_pcm_lib_free_vmalloc_buffer(substream); } /* prepare callback */ @@ -692,13 +680,6 @@ snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream) chip->channel[chan->idx].pos); } -/* get the physical page pointer on the given offset */ -static struct page *snd_sgio2audio_page(struct snd_pcm_substream *substream, - unsigned long offset) -{ - return vmalloc_to_page(substream->runtime->dma_area + offset); -} - /* operators */ static struct snd_pcm_ops snd_sgio2audio_playback1_ops = { .open = snd_sgio2audio_playback1_open, @@ -709,7 +690,7 @@ static struct snd_pcm_ops snd_sgio2audio_playback1_ops = { .prepare = snd_sgio2audio_pcm_prepare, .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, - .page = snd_sgio2audio_page, + .page = snd_pcm_lib_get_vmalloc_page, }; static struct snd_pcm_ops snd_sgio2audio_playback2_ops = { @@ -721,7 +702,7 @@ static struct snd_pcm_ops snd_sgio2audio_playback2_ops = { .prepare = snd_sgio2audio_pcm_prepare, .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, - .page = snd_sgio2audio_page, + .page = snd_pcm_lib_get_vmalloc_page, }; static struct snd_pcm_ops snd_sgio2audio_capture_ops = { @@ -733,7 +714,7 @@ static struct snd_pcm_ops snd_sgio2audio_capture_ops = { .prepare = snd_sgio2audio_pcm_prepare, .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, - .page = snd_sgio2audio_page, + .page = snd_pcm_lib_get_vmalloc_page, }; /* -- cgit v1.1 From 149feef54bf543448dd4ec5820ef8ae178021c3a Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:30:55 +0100 Subject: sound: vx: use vmalloc buffer helper functions Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the equivalent core functions instead. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/drivers/vx/vx_pcm.c | 59 ++++------------------------------------------- 1 file changed, 5 insertions(+), 54 deletions(-) diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index 6644d00..c8385d2 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -46,7 +46,6 @@ */ #include -#include #include #include #include @@ -56,55 +55,6 @@ /* - * we use a vmalloc'ed (sg-)buffer - */ - -/* get the physical page pointer on the given offset */ -static struct page *snd_pcm_get_vmalloc_page(struct snd_pcm_substream *subs, - unsigned long offset) -{ - void *pageptr = subs->runtime->dma_area + offset; - return vmalloc_to_page(pageptr); -} - -/* - * allocate a buffer via vmalloc_32(). - * called from hw_params - * NOTE: this may be called not only once per pcm open! - */ -static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - if (runtime->dma_area) { - /* already allocated */ - if (runtime->dma_bytes >= size) - return 0; /* already enough large */ - vfree(runtime->dma_area); - } - runtime->dma_area = vmalloc_32(size); - if (! runtime->dma_area) - return -ENOMEM; - memset(runtime->dma_area, 0, size); - runtime->dma_bytes = size; - return 1; /* changed */ -} - -/* - * free the buffer. - * called from hw_free callback - * NOTE: this may be called not only once per pcm open! - */ -static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - - vfree(runtime->dma_area); - runtime->dma_area = NULL; - return 0; -} - - -/* * read three pending pcm bytes via inb() */ static void vx_pcm_read_per_bytes(struct vx_core *chip, struct snd_pcm_runtime *runtime, @@ -865,7 +815,8 @@ static snd_pcm_uframes_t vx_pcm_playback_pointer(struct snd_pcm_substream *subs) static int vx_pcm_hw_params(struct snd_pcm_substream *subs, struct snd_pcm_hw_params *hw_params) { - return snd_pcm_alloc_vmalloc_buffer(subs, params_buffer_bytes(hw_params)); + return snd_pcm_lib_alloc_vmalloc_32_buffer + (subs, params_buffer_bytes(hw_params)); } /* @@ -873,7 +824,7 @@ static int vx_pcm_hw_params(struct snd_pcm_substream *subs, */ static int vx_pcm_hw_free(struct snd_pcm_substream *subs) { - return snd_pcm_free_vmalloc_buffer(subs); + return snd_pcm_lib_free_vmalloc_buffer(subs); } /* @@ -953,7 +904,7 @@ static struct snd_pcm_ops vx_pcm_playback_ops = { .prepare = vx_pcm_prepare, .trigger = vx_pcm_trigger, .pointer = vx_pcm_playback_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; @@ -1173,7 +1124,7 @@ static struct snd_pcm_ops vx_pcm_capture_ops = { .prepare = vx_pcm_prepare, .trigger = vx_pcm_trigger, .pointer = vx_pcm_capture_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; -- cgit v1.1 From c55675e348d9630c1ca69a190529bed1108c649d Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:31:31 +0100 Subject: sound: usb-audio: use vmalloc buffer helper functions Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the equivalent core functions instead. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 46 +++++----------------------------------------- 1 file changed, 5 insertions(+), 41 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index af8869a..31b63ea 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -44,7 +44,6 @@ #include #include #include -#include #include #include #include @@ -735,41 +734,6 @@ static void snd_complete_sync_urb(struct urb *urb) } -/* get the physical page pointer at the given offset */ -static struct page *snd_pcm_get_vmalloc_page(struct snd_pcm_substream *subs, - unsigned long offset) -{ - void *pageptr = subs->runtime->dma_area + offset; - return vmalloc_to_page(pageptr); -} - -/* allocate virtual buffer; may be called more than once */ -static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - if (runtime->dma_area) { - if (runtime->dma_bytes >= size) - return 0; /* already large enough */ - vfree(runtime->dma_area); - } - runtime->dma_area = vmalloc_user(size); - if (!runtime->dma_area) - return -ENOMEM; - runtime->dma_bytes = size; - return 0; -} - -/* free virtual buffer; may be called more than once */ -static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - - vfree(runtime->dma_area); - runtime->dma_area = NULL; - return 0; -} - - /* * unlink active urbs. */ @@ -1449,8 +1413,8 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, unsigned int channels, rate, format; int ret, changed; - ret = snd_pcm_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + ret = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); if (ret < 0) return ret; @@ -1507,7 +1471,7 @@ static int snd_usb_hw_free(struct snd_pcm_substream *substream) subs->period_bytes = 0; if (!subs->stream->chip->shutdown) release_substream_urbs(subs, 0); - return snd_pcm_free_vmalloc_buffer(substream); + return snd_pcm_lib_free_vmalloc_buffer(substream); } /* @@ -1973,7 +1937,7 @@ static struct snd_pcm_ops snd_usb_playback_ops = { .prepare = snd_usb_pcm_prepare, .trigger = snd_usb_pcm_playback_trigger, .pointer = snd_usb_pcm_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; static struct snd_pcm_ops snd_usb_capture_ops = { @@ -1985,7 +1949,7 @@ static struct snd_pcm_ops snd_usb_capture_ops = { .prepare = snd_usb_pcm_prepare, .trigger = snd_usb_pcm_capture_trigger, .pointer = snd_usb_pcm_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; -- cgit v1.1 From 5b4b2a41a1a80f5560364b7ef001486cd8fb5230 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 18 Dec 2009 09:32:00 +0100 Subject: sound: ua101: use vmalloc buffer helper functions Remove this duplicate of snd_pcm_alloc_vmalloc_buffer and use the equivalent core functions instead. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/ua101.c | 52 +++++++--------------------------------------------- 1 file changed, 7 insertions(+), 45 deletions(-) diff --git a/sound/usb/ua101.c b/sound/usb/ua101.c index ab9f8a2..16dc7bd 100644 --- a/sound/usb/ua101.c +++ b/sound/usb/ua101.c @@ -19,7 +19,6 @@ #include #include #include -#include #include #include #include @@ -145,42 +144,6 @@ static struct usb_driver ua101_driver; static void abort_alsa_playback(struct ua101 *ua); static void abort_alsa_capture(struct ua101 *ua); -/* allocate virtual buffer; may be called more than once */ -static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, - size_t size) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - - if (runtime->dma_area) { - if (runtime->dma_bytes >= size) - return 0; /* already large enough */ - vfree(runtime->dma_area); - } - runtime->dma_area = vmalloc_user(size); - if (!runtime->dma_area) - return -ENOMEM; - runtime->dma_bytes = size; - return 0; -} - -/* free virtual buffer; may be called more than once */ -static int snd_pcm_free_vmalloc_buffer(struct snd_pcm_substream *subs) -{ - struct snd_pcm_runtime *runtime = subs->runtime; - - vfree(runtime->dma_area); - runtime->dma_area = NULL; - return 0; -} - -/* get the physical page pointer at the given offset */ -static struct page *snd_pcm_get_vmalloc_page(struct snd_pcm_substream *subs, - unsigned long offset) -{ - void *pageptr = subs->runtime->dma_area + offset; - return vmalloc_to_page(pageptr); -} - static const char *usb_error_string(int err) { switch (err) { @@ -791,8 +754,8 @@ static int capture_pcm_hw_params(struct snd_pcm_substream *substream, if (err < 0) return err; - return snd_pcm_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); } static int playback_pcm_hw_params(struct snd_pcm_substream *substream, @@ -809,14 +772,13 @@ static int playback_pcm_hw_params(struct snd_pcm_substream *substream, if (err < 0) return err; - return snd_pcm_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); } static int ua101_pcm_hw_free(struct snd_pcm_substream *substream) { - snd_pcm_free_vmalloc_buffer(substream); - return 0; + return snd_pcm_lib_free_vmalloc_buffer(substream); } static int capture_pcm_prepare(struct snd_pcm_substream *substream) @@ -948,7 +910,7 @@ static struct snd_pcm_ops capture_pcm_ops = { .prepare = capture_pcm_prepare, .trigger = capture_pcm_trigger, .pointer = capture_pcm_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; static struct snd_pcm_ops playback_pcm_ops = { @@ -960,7 +922,7 @@ static struct snd_pcm_ops playback_pcm_ops = { .prepare = playback_pcm_prepare, .trigger = playback_pcm_trigger, .pointer = playback_pcm_pointer, - .page = snd_pcm_get_vmalloc_page, + .page = snd_pcm_lib_get_vmalloc_page, }; static const struct uac_format_type_i_discrete_descriptor * -- cgit v1.1 From 0c2fd1bf4c6cb6095d8b3088d285167e66c12147 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 18 Dec 2009 16:41:39 +0100 Subject: ALSA: hda - Check class to identify Nvidia controller chips Instead of listing all individual PCI IDs, check the matching with the PCI class together with the vendor id for Nvidia. This simplifies the pci id entries. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 30 ++++-------------------------- 1 file changed, 4 insertions(+), 26 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 9b56f93..93eaf4f 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2694,32 +2694,10 @@ static struct pci_device_id azx_ids[] = { /* ULI M5461 */ { PCI_DEVICE(0x10b9, 0x5461), .driver_data = AZX_DRIVER_ULI }, /* NVIDIA MCP */ - { PCI_DEVICE(0x10de, 0x026c), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0371), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x03e4), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x03f0), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x044a), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x044b), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x055c), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x055d), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0590), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0774), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0775), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0776), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0777), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x07fc), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x07fd), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0ac0), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0ac1), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0ac2), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0ac3), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0be2), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0be3), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0be4), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0d94), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0d95), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0d96), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0d97), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(PCI_VENDOR_ID_NVIDIA, PCI_ANY_ID), + .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8, + .class_mask = 0xffffff, + .driver_data = AZX_DRIVER_NVIDIA }, /* Teradici */ { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA }, /* Creative X-Fi (CA0110-IBG) */ -- cgit v1.1 From ad8decb7f5dfd556e4a8400e37b127cd20d8e4c5 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Sun, 20 Dec 2009 19:01:50 +0100 Subject: ALSA: jazz16: Add support for Media Vision Jazz16 chipset This is one of Sound Blaster Pro compatible chipsets which is supported by Linux OSS driver and was missing native supoort for ALSA. The Jazz16 audio codec is Crystal CS4216 which is capable of playback and recording up to 48 kHz stereo. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- include/sound/sb.h | 1 + sound/isa/Kconfig | 16 ++ sound/isa/sb/Makefile | 2 + sound/isa/sb/jazz16.c | 385 +++++++++++++++++++++++++++++++++++++++++++++++ sound/isa/sb/sb8_main.c | 117 ++++++++++++-- sound/isa/sb/sb_common.c | 3 + sound/isa/sb/sb_mixer.c | 3 + 7 files changed, 511 insertions(+), 16 deletions(-) create mode 100644 sound/isa/sb/jazz16.c diff --git a/include/sound/sb.h b/include/sound/sb.h index 4e62ee1..9535354 100644 --- a/include/sound/sb.h +++ b/include/sound/sb.h @@ -33,6 +33,7 @@ enum sb_hw_type { SB_HW_20, SB_HW_201, SB_HW_PRO, + SB_HW_JAZZ16, /* Media Vision Jazz16 */ SB_HW_16, SB_HW_16CSP, /* SB16 with CSP chip */ SB_HW_ALS100, /* Avance Logic ALS100 chip */ diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 194af3b..755a0a5 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -239,6 +239,22 @@ config SND_INTERWAVE_STB To compile this driver as a module, choose M here: the module will be called snd-interwave-stb. +config SND_JAZZ16 + tristate "Media Vision Jazz16 card and compatibles" + select SND_OPL3_LIB + select SND_MPU401_UART + select SND_SB8_DSP + help + Say Y here to include support for soundcards based on the + Media Vision Jazz16 chipset: digital chip MVD1216 (Jazz16), + codec MVA416 (CS4216) and mixer MVA514 (ICS2514). + Media Vision's Jazz16 cards were sold under names Pro Sonic 16, + Premium 3-D and Pro 3-D. There were also OEMs cards with the + Jazz16 chipset. + + To compile this driver as a module, choose M here: the module + will be called snd-jazz16. + config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" select SND_OPL3_LIB diff --git a/sound/isa/sb/Makefile b/sound/isa/sb/Makefile index faeffceb..af36696 100644 --- a/sound/isa/sb/Makefile +++ b/sound/isa/sb/Makefile @@ -12,6 +12,7 @@ snd-sb16-objs := sb16.o snd-sbawe-objs := sbawe.o emu8000.o snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o snd-es968-objs := es968.o +snd-jazz16-objs := jazz16.o # Toplevel Module Dependency obj-$(CONFIG_SND_SB_COMMON) += snd-sb-common.o @@ -21,6 +22,7 @@ obj-$(CONFIG_SND_SB8) += snd-sb8.o obj-$(CONFIG_SND_SB16) += snd-sb16.o obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o obj-$(CONFIG_SND_ES968) += snd-es968.o +obj-$(CONFIG_SND_JAZZ16) += snd-jazz16.o ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c new file mode 100644 index 0000000..d52966b --- /dev/null +++ b/sound/isa/sb/jazz16.c @@ -0,0 +1,385 @@ + +/* + * jazz16.c - driver for Media Vision Jazz16 based soundcards. + * Copyright (C) 2009 Krzysztof Helt + * Based on patches posted by Rask Ingemann Lambertsen and Rene Herman. + * Based on OSS Sound Blaster driver. + * + * This file is subject to the terms and conditions of the GNU General Public + * License. See the file COPYING in the main directory of this archive for + * more details. + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#define SNDRV_LEGACY_FIND_FREE_IRQ +#define SNDRV_LEGACY_FIND_FREE_DMA +#include + +#define PFX "jazz16: " + +MODULE_DESCRIPTION("Media Vision Jazz16"); +MODULE_SUPPORTED_DEVICE("{{Media Vision ??? }," + "{RTL,RTL3000}}"); + +MODULE_AUTHOR("Krzysztof Helt "); +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; /* Enable this card */ +static unsigned long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static unsigned long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int dma8[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; +static int dma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Media Vision Jazz16 based soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for Media Vision Jazz16 based soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Media Vision Jazz16 based soundcard."); +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for jazz16 driver."); +module_param_array(mpu_port, long, NULL, 0444); +MODULE_PARM_DESC(mpu_port, "MPU-401 port # for jazz16 driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for jazz16 driver."); +module_param_array(mpu_irq, int, NULL, 0444); +MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for jazz16 driver."); +module_param_array(dma8, int, NULL, 0444); +MODULE_PARM_DESC(dma8, "DMA8 # for jazz16 driver."); +module_param_array(dma16, int, NULL, 0444); +MODULE_PARM_DESC(dma16, "DMA16 # for jazz16 driver."); + +#define SB_JAZZ16_WAKEUP 0xaf +#define SB_JAZZ16_SET_PORTS 0x50 +#define SB_DSP_GET_JAZZ_BRD_REV 0xfa +#define SB_JAZZ16_SET_DMAINTR 0xfb +#define SB_DSP_GET_JAZZ_MODEL 0xfe + +struct snd_card_jazz16 { + struct snd_sb *chip; +}; + +static irqreturn_t jazz16_interrupt(int irq, void *chip) +{ + return snd_sb8dsp_interrupt(chip); +} + +static int __devinit jazz16_configure_ports(unsigned long port, + unsigned long mpu_port, int idx) +{ + unsigned char val; + + if (!request_region(0x201, 1, "jazz16 config")) { + snd_printk(KERN_ERR "config port region is already in use.\n"); + return -EBUSY; + } + outb(SB_JAZZ16_WAKEUP - idx, 0x201); + udelay(100); + outb(SB_JAZZ16_SET_PORTS + idx, 0x201); + udelay(100); + val = port & 0x70; + val |= (mpu_port & 0x30) >> 4; + outb(val, 0x201); + + release_region(0x201, 1); + return 0; +} + +static int __devinit jazz16_detect_board(unsigned long port, + unsigned long mpu_port) +{ + int err; + int val; + struct snd_sb chip; + + if (!request_region(port, 0x10, "jazz16")) { + snd_printk(KERN_ERR "I/O port region is already in use.\n"); + return -EBUSY; + } + /* just to call snd_sbdsp_command/reset/get_byte() */ + chip.port = port; + + err = snd_sbdsp_reset(&chip); + if (err < 0) + for (val = 0; val < 4; val++) { + err = jazz16_configure_ports(port, mpu_port, val); + if (err < 0) + break; + + err = snd_sbdsp_reset(&chip); + if (!err) + break; + } + if (err < 0) { + err = -ENODEV; + goto err_unmap; + } + if (!snd_sbdsp_command(&chip, SB_DSP_GET_JAZZ_BRD_REV)) { + err = -EBUSY; + goto err_unmap; + } + val = snd_sbdsp_get_byte(&chip); + if (val >= 0x30) + snd_sbdsp_get_byte(&chip); + + if ((val & 0xf0) != 0x10) { + err = -ENODEV; + goto err_unmap; + } + if (!snd_sbdsp_command(&chip, SB_DSP_GET_JAZZ_MODEL)) { + err = -EBUSY; + goto err_unmap; + } + snd_sbdsp_get_byte(&chip); + err = snd_sbdsp_get_byte(&chip); + snd_printd("Media Vision Jazz16 board detected: rev 0x%x, model 0x%x\n", + val, err); + + err = 0; + +err_unmap: + release_region(port, 0x10); + return err; +} + +static int __devinit jazz16_configure_board(struct snd_sb *chip, int mpu_irq) +{ + static unsigned char jazz_irq_bits[] = { 0, 0, 2, 3, 0, 1, 0, 4, + 0, 2, 5, 0, 0, 0, 0, 6 }; + static unsigned char jazz_dma_bits[] = { 0, 1, 0, 2, 0, 3, 0, 4 }; + + if (jazz_dma_bits[chip->dma8] == 0 || + jazz_dma_bits[chip->dma16] == 0 || + jazz_irq_bits[chip->irq] == 0) + return -EINVAL; + + if (!snd_sbdsp_command(chip, SB_JAZZ16_SET_DMAINTR)) + return -EBUSY; + + if (!snd_sbdsp_command(chip, + jazz_dma_bits[chip->dma8] | + (jazz_dma_bits[chip->dma16] << 4))) + return -EBUSY; + + if (!snd_sbdsp_command(chip, + jazz_irq_bits[chip->irq] | + (jazz_irq_bits[mpu_irq] << 4))) + return -EBUSY; + + return 0; +} + +static int __devinit snd_jazz16_match(struct device *devptr, unsigned int dev) +{ + if (!enable[dev]) + return 0; + if (port[dev] == SNDRV_AUTO_PORT) { + snd_printk(KERN_ERR "please specify port\n"); + return 0; + } + if (dma16[dev] != SNDRV_AUTO_DMA && + dma16[dev] != 5 && dma16[dev] != 7) { + snd_printk(KERN_ERR "dma16 must be 5 or 7"); + return 0; + } + return 1; +} + +static int __devinit snd_jazz16_probe(struct device *devptr, unsigned int dev) +{ + struct snd_card *card; + struct snd_card_jazz16 *jazz16; + struct snd_sb *chip; + struct snd_opl3 *opl3; + static int possible_irqs[] = {2, 3, 5, 7, 9, 10, 15, -1}; + static int possible_dmas8[] = {1, 3, -1}; + static int possible_dmas16[] = {5, 7, -1}; + int err, xirq, xdma8, xdma16, xmpu_port, xmpu_irq; + + err = snd_card_create(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_card_jazz16), &card); + if (err < 0) + return err; + + jazz16 = card->private_data; + + xirq = irq[dev]; + if (xirq == SNDRV_AUTO_IRQ) { + xirq = snd_legacy_find_free_irq(possible_irqs); + if (xirq < 0) { + snd_printk(KERN_ERR "unable to find a free IRQ\n"); + err = -EBUSY; + goto err_free; + } + } + xdma8 = dma8[dev]; + if (xdma8 == SNDRV_AUTO_DMA) { + xdma8 = snd_legacy_find_free_dma(possible_dmas8); + if (xdma8 < 0) { + snd_printk(KERN_ERR "unable to find a free DMA8\n"); + err = -EBUSY; + goto err_free; + } + } + xdma16 = dma16[dev]; + if (xdma16 == SNDRV_AUTO_DMA) { + xdma16 = snd_legacy_find_free_dma(possible_dmas16); + if (xdma16 < 0) { + snd_printk(KERN_ERR "unable to find a free DMA16\n"); + err = -EBUSY; + goto err_free; + } + } + + xmpu_port = mpu_port[dev]; + if (xmpu_port == SNDRV_AUTO_PORT) + xmpu_port = 0; + err = jazz16_detect_board(port[dev], xmpu_port); + if (err < 0) { + printk(KERN_ERR "Media Vision Jazz16 board not detected\n"); + goto err_free; + } + err = snd_sbdsp_create(card, port[dev], irq[dev], + jazz16_interrupt, + dma8[dev], dma16[dev], + SB_HW_JAZZ16, + &chip); + if (err < 0) + goto err_free; + + xmpu_irq = mpu_irq[dev]; + if (xmpu_irq == SNDRV_AUTO_IRQ || mpu_port[dev] == SNDRV_AUTO_PORT) + xmpu_irq = 0; + err = jazz16_configure_board(chip, xmpu_irq); + if (err < 0) { + printk(KERN_ERR "Media Vision Jazz16 configuration failed\n"); + goto err_free; + } + + jazz16->chip = chip; + + strcpy(card->driver, "jazz16"); + strcpy(card->shortname, "Media Vision Jazz16"); + sprintf(card->longname, + "Media Vision Jazz16 at 0x%lx, irq %d, dma8 %d, dma16 %d", + port[dev], xirq, xdma8, xdma16); + + err = snd_sb8dsp_pcm(chip, 0, NULL); + if (err < 0) + goto err_free; + err = snd_sbmixer_new(chip); + if (err < 0) + goto err_free; + + err = snd_opl3_create(card, chip->port, chip->port + 2, + OPL3_HW_AUTO, 1, &opl3); + if (err < 0) + snd_printk(KERN_WARNING "no OPL device at 0x%lx-0x%lx\n", + chip->port, chip->port + 2); + else { + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + goto err_free; + } + 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_MPU401, + mpu_port[dev], 0, + mpu_irq[dev], + mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, + NULL) < 0) + snd_printk(KERN_ERR "no MPU-401 device at 0x%lx\n", + mpu_port[dev]); + } + + snd_card_set_dev(card, devptr); + + err = snd_card_register(card); + if (err < 0) + goto err_free; + + dev_set_drvdata(devptr, card); + return 0; + +err_free: + snd_card_free(card); + return err; +} + +static int __devexit snd_jazz16_remove(struct device *devptr, unsigned int dev) +{ + struct snd_card *card = dev_get_drvdata(devptr); + + dev_set_drvdata(devptr, NULL); + snd_card_free(card); + return 0; +} + +#ifdef CONFIG_PM +static int snd_jazz16_suspend(struct device *pdev, unsigned int n, + pm_message_t state) +{ + struct snd_card *card = dev_get_drvdata(pdev); + struct snd_card_jazz16 *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_jazz16_resume(struct device *pdev, unsigned int n) +{ + struct snd_card *card = dev_get_drvdata(pdev); + struct snd_card_jazz16 *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 isa_driver snd_jazz16_driver = { + .match = snd_jazz16_match, + .probe = snd_jazz16_probe, + .remove = __devexit_p(snd_jazz16_remove), +#ifdef CONFIG_PM + .suspend = snd_jazz16_suspend, + .resume = snd_jazz16_resume, +#endif + .driver = { + .name = "jazz16" + }, +}; + +static int __init alsa_card_jazz16_init(void) +{ + return isa_register_driver(&snd_jazz16_driver, SNDRV_CARDS); +} + +static void __exit alsa_card_jazz16_exit(void) +{ + isa_unregister_driver(&snd_jazz16_driver); +} + +module_init(alsa_card_jazz16_init) +module_exit(alsa_card_jazz16_exit) diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index 658d557..3222aed 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -106,9 +106,21 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int mixreg, rate, size, count; + unsigned char format; + unsigned char stereo = runtime->channels > 1; + int dma; rate = runtime->rate; switch (chip->hardware) { + case SB_HW_JAZZ16: + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) { + if (chip->mode & SB_MODE_CAPTURE_16) + return -EBUSY; + else + chip->mode |= SB_MODE_PLAYBACK_16; + } + chip->playback_format = SB_DSP_LO_OUTPUT_AUTO; + break; case SB_HW_PRO: if (runtime->channels > 1) { if (snd_BUG_ON(rate != SB8_RATE(11025) && @@ -133,11 +145,21 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) default: return -EINVAL; } + if (chip->mode & SB_MODE_PLAYBACK_16) { + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + dma = chip->dma16; + } else { + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + chip->mode |= SB_MODE_PLAYBACK_8; + dma = chip->dma8; + } size = chip->p_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->p_period_size = snd_pcm_lib_period_bytes(substream); spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_ON); - if (runtime->channels > 1) { + if (chip->hardware == SB_HW_JAZZ16) + snd_sbdsp_command(chip, format); + else if (stereo) { /* set playback stereo mode */ spin_lock(&chip->mixer_lock); mixreg = snd_sbmixer_read(chip, SB_DSP_STEREO_SW); @@ -147,15 +169,14 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) /* Soundblaster hardware programming reference guide, 3-23 */ snd_sbdsp_command(chip, SB_DSP_DMA8_EXIT); runtime->dma_area[0] = 0x80; - snd_dma_program(chip->dma8, runtime->dma_addr, 1, DMA_MODE_WRITE); + snd_dma_program(dma, runtime->dma_addr, 1, DMA_MODE_WRITE); /* force interrupt */ - chip->mode = SB_MODE_HALT; snd_sbdsp_command(chip, SB_DSP_OUTPUT); snd_sbdsp_command(chip, 0); snd_sbdsp_command(chip, 0); } snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); - if (runtime->channels > 1) { + if (stereo) { snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); spin_lock(&chip->mixer_lock); /* save output filter status and turn it off */ @@ -168,13 +189,15 @@ static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) snd_sbdsp_command(chip, 256 - runtime->rate_den); } if (chip->playback_format != SB_DSP_OUTPUT) { + if (chip->mode & SB_MODE_PLAYBACK_16) + count /= 2; count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); return 0; } @@ -212,7 +235,6 @@ static int snd_sb8_playback_trigger(struct snd_pcm_substream *substream, snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); } spin_unlock_irqrestore(&chip->reg_lock, flags); - chip->mode = (cmd == SNDRV_PCM_TRIGGER_START) ? SB_MODE_PLAYBACK_8 : SB_MODE_HALT; return 0; } @@ -234,9 +256,21 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int mixreg, rate, size, count; + unsigned char format; + unsigned char stereo = runtime->channels > 1; + int dma; rate = runtime->rate; switch (chip->hardware) { + case SB_HW_JAZZ16: + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) { + if (chip->mode & SB_MODE_PLAYBACK_16) + return -EBUSY; + else + chip->mode |= SB_MODE_CAPTURE_16; + } + chip->capture_format = SB_DSP_LO_INPUT_AUTO; + break; case SB_HW_PRO: if (runtime->channels > 1) { if (snd_BUG_ON(rate != SB8_RATE(11025) && @@ -262,14 +296,24 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) default: return -EINVAL; } + if (chip->mode & SB_MODE_CAPTURE_16) { + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + dma = chip->dma16; + } else { + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + chip->mode |= SB_MODE_CAPTURE_8; + dma = chip->dma8; + } size = chip->c_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->c_period_size = snd_pcm_lib_period_bytes(substream); spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); - if (runtime->channels > 1) + if (chip->hardware == SB_HW_JAZZ16) + snd_sbdsp_command(chip, format); + else if (stereo) snd_sbdsp_command(chip, SB_DSP_STEREO_8BIT); snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); - if (runtime->channels > 1) { + if (stereo) { snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); spin_lock(&chip->mixer_lock); /* save input filter status and turn it off */ @@ -282,13 +326,15 @@ static int snd_sb8_capture_prepare(struct snd_pcm_substream *substream) snd_sbdsp_command(chip, 256 - runtime->rate_den); } if (chip->capture_format != SB_DSP_INPUT) { + if (chip->mode & SB_MODE_PLAYBACK_16) + count /= 2; count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); return 0; } @@ -328,7 +374,6 @@ static int snd_sb8_capture_trigger(struct snd_pcm_substream *substream, snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); } spin_unlock_irqrestore(&chip->reg_lock, flags); - chip->mode = (cmd == SNDRV_PCM_TRIGGER_START) ? SB_MODE_CAPTURE_8 : SB_MODE_HALT; return 0; } @@ -339,13 +384,21 @@ irqreturn_t snd_sb8dsp_interrupt(struct snd_sb *chip) snd_sb_ack_8bit(chip); switch (chip->mode) { - case SB_MODE_PLAYBACK_8: /* ok.. playback is active */ + case SB_MODE_PLAYBACK_16: /* ok.. playback is active */ + if (chip->hardware != SB_HW_JAZZ16) + break; + /* fallthru */ + case SB_MODE_PLAYBACK_8: substream = chip->playback_substream; runtime = substream->runtime; if (chip->playback_format == SB_DSP_OUTPUT) snd_sb8_playback_trigger(substream, SNDRV_PCM_TRIGGER_START); snd_pcm_period_elapsed(substream); break; + case SB_MODE_CAPTURE_16: + if (chip->hardware != SB_HW_JAZZ16) + break; + /* fallthru */ case SB_MODE_CAPTURE_8: substream = chip->capture_substream; runtime = substream->runtime; @@ -361,10 +414,15 @@ static snd_pcm_uframes_t snd_sb8_playback_pointer(struct snd_pcm_substream *subs { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; - if (chip->mode != SB_MODE_PLAYBACK_8) + if (chip->mode & SB_MODE_PLAYBACK_8) + dma = chip->dma8; + else if (chip->mode & SB_MODE_PLAYBACK_16) + dma = chip->dma16; + else return 0; - ptr = snd_dma_pointer(chip->dma8, chip->p_dma_size); + ptr = snd_dma_pointer(dma, chip->p_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -372,10 +430,15 @@ static snd_pcm_uframes_t snd_sb8_capture_pointer(struct snd_pcm_substream *subst { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; - if (chip->mode != SB_MODE_CAPTURE_8) + if (chip->mode & SB_MODE_CAPTURE_8) + dma = chip->dma8; + else if (chip->mode & SB_MODE_CAPTURE_16) + dma = chip->dma16; + else return 0; - ptr = snd_dma_pointer(chip->dma8, chip->c_dma_size); + ptr = snd_dma_pointer(dma, chip->c_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -446,6 +509,13 @@ static int snd_sb8_open(struct snd_pcm_substream *substream) runtime->hw = snd_sb8_capture; } switch (chip->hardware) { + case SB_HW_JAZZ16: + runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; + runtime->hw.rates |= SNDRV_PCM_RATE_8000_48000; + runtime->hw.rate_min = 4000; + runtime->hw.rate_max = 50000; + runtime->hw.channels_max = 2; + break; case SB_HW_PRO: runtime->hw.rate_max = 44100; runtime->hw.channels_max = 2; @@ -468,6 +538,14 @@ static int snd_sb8_open(struct snd_pcm_substream *substream) } snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_clock); + if (chip->dma8 > 3 || chip->dma16 >= 0) { + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 2); + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 2); + runtime->hw.buffer_bytes_max = 128 * 1024 * 1024; + runtime->hw.period_bytes_max = 128 * 1024 * 1024; + } return 0; } @@ -480,6 +558,10 @@ static int snd_sb8_close(struct snd_pcm_substream *substream) chip->capture_substream = NULL; spin_lock_irqsave(&chip->open_lock, flags); chip->open &= ~SB_OPEN_PCM; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + chip->mode &= ~SB_MODE_PLAYBACK; + else + chip->mode &= ~SB_MODE_CAPTURE; spin_unlock_irqrestore(&chip->open_lock, flags); return 0; } @@ -515,6 +597,7 @@ int snd_sb8dsp_pcm(struct snd_sb *chip, int device, struct snd_pcm ** rpcm) struct snd_card *card = chip->card; struct snd_pcm *pcm; int err; + size_t max_prealloc = 64 * 1024; if (rpcm) *rpcm = NULL; @@ -527,9 +610,11 @@ int snd_sb8dsp_pcm(struct snd_sb *chip, int device, struct snd_pcm ** rpcm) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sb8_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sb8_capture_ops); + if (chip->dma8 > 3 || chip->dma16 >= 0) + max_prealloc = 128 * 1024; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), - 64*1024, 64*1024); + 64*1024, max_prealloc); if (rpcm) *rpcm = pcm; diff --git a/sound/isa/sb/sb_common.c b/sound/isa/sb/sb_common.c index 27a6515..eae6c1c 100644 --- a/sound/isa/sb/sb_common.c +++ b/sound/isa/sb/sb_common.c @@ -170,6 +170,9 @@ static int snd_sbdsp_probe(struct snd_sb * chip) case SB_HW_CS5530: str = "16 (CS5530)"; break; + case SB_HW_JAZZ16: + str = "Pro (Jazz16)"; + break; default: return -ENODEV; } diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 8cfc41f..6496822 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -779,6 +779,7 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_PRO: + case SB_HW_JAZZ16: if ((err = snd_sbmixer_init(chip, snd_sbpro_controls, ARRAY_SIZE(snd_sbpro_controls), @@ -929,6 +930,7 @@ void snd_sbmixer_suspend(struct snd_sb *chip) save_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: save_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: @@ -955,6 +957,7 @@ void snd_sbmixer_resume(struct snd_sb *chip) restore_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: restore_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: -- cgit v1.1 From ee7c343c0134bf126b4235e65c407711b77174da Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 21 Dec 2009 12:41:37 +0100 Subject: ALSA: pcm - Add missing inclusion of linux/vmalloc.h Signed-off-by: Takashi Iwai --- sound/core/pcm_memory.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index d9727c7..d6d49d6 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include #include -- cgit v1.1 From 8374e24c23448cabf6e78db2c83841c56c5df1e1 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Mon, 21 Dec 2009 17:07:08 +0100 Subject: ALSA: refine rate selection in snd_interval_ratnum() Refine the rate selection by choosing the rate closer to the requested one in case of selecting single frequency. Previously, the higher rate was always selected. Also, fix problem with the best_diff unsigned int value wrapping (turning negative). Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a27545b..b07cc36 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -745,10 +745,13 @@ int snd_interval_ratnum(struct snd_interval *i, unsigned int rats_count, struct snd_ratnum *rats, unsigned int *nump, unsigned int *denp) { - unsigned int best_num, best_diff, best_den; + unsigned int best_num, best_den; + int best_diff; unsigned int k; struct snd_interval t; int err; + unsigned int result_num, result_den; + int result_diff; best_num = best_den = best_diff = 0; for (k = 0; k < rats_count; ++k) { @@ -770,6 +773,8 @@ int snd_interval_ratnum(struct snd_interval *i, den -= r; } diff = num - q * den; + if (diff < 0) + diff = -diff; if (best_num == 0 || diff * best_den < best_diff * den) { best_diff = diff; @@ -784,6 +789,9 @@ int snd_interval_ratnum(struct snd_interval *i, t.min = div_down(best_num, best_den); t.openmin = !!(best_num % best_den); + result_num = best_num; + result_diff = best_diff; + result_den = best_den; best_num = best_den = best_diff = 0; for (k = 0; k < rats_count; ++k) { unsigned int num = rats[k].num; @@ -806,6 +814,8 @@ int snd_interval_ratnum(struct snd_interval *i, den += rats[k].den_step - r; } diff = q * den - num; + if (diff < 0) + diff = -diff; if (best_num == 0 || diff * best_den < best_diff * den) { best_diff = diff; @@ -825,10 +835,14 @@ int snd_interval_ratnum(struct snd_interval *i, return err; if (snd_interval_single(i)) { + if (best_diff * result_den < result_diff * best_den) { + result_num = best_num; + result_den = best_den; + } if (nump) - *nump = best_num; + *nump = result_num; if (denp) - *denp = best_den; + *denp = result_den; } return err; } -- cgit v1.1 From 41116e926cb92292fa4fcbe888ae8133fa0038e6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Dec 2009 09:00:14 +0100 Subject: ALSA: cs46xx - Fix suspend/resume with new DSP Fix the basic suspend/resume of snd-cs46xx drivers with new DSP. References: https://bugzilla.redhat.com/show_bug.cgi?id=498287 https://bugzilla.redhat.com/show_bug.cgi?id=160751 Tested-by: Florian Zumbiehl Signed-off-by: Takashi Iwai --- include/sound/cs46xx_dsp_spos.h | 6 ++++-- sound/pci/cs46xx/cs46xx_lib.c | 2 +- sound/pci/cs46xx/dsp_spos.c | 42 +++++++++++++++++++++++++++++++++---- sound/pci/cs46xx/dsp_spos.h | 4 ++++ sound/pci/cs46xx/dsp_spos_scb_lib.c | 33 +++++++++++++---------------- 5 files changed, 62 insertions(+), 25 deletions(-) diff --git a/include/sound/cs46xx_dsp_spos.h b/include/sound/cs46xx_dsp_spos.h index 7c44667..49b03c9 100644 --- a/include/sound/cs46xx_dsp_spos.h +++ b/include/sound/cs46xx_dsp_spos.h @@ -118,9 +118,11 @@ struct dsp_scb_descriptor { struct snd_info_entry *proc_info; int ref_count; - spinlock_t lock; - int deleted; + u16 volume[2]; + unsigned int deleted :1; + unsigned int updated :1; + unsigned int volume_set :1; }; struct dsp_task_descriptor { diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 1be96ea..e6b4a87 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -3597,7 +3597,7 @@ static struct cs_card_type __devinitdata cards[] = { #ifdef CONFIG_PM static unsigned int saved_regs[] = { BA0_ACOSV, - BA0_ASER_FADDR, + /*BA0_ASER_FADDR,*/ BA0_ASER_MASTER, BA1_PVOL, BA1_CVOL, diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index f4f0c8f..3e5ca8f 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -298,6 +298,9 @@ void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip) if (ins->scbs[i].deleted) continue; cs46xx_dsp_proc_free_scb_desc ( (ins->scbs + i) ); +#ifdef CONFIG_PM + kfree(ins->scbs[i].data); +#endif } kfree(ins->code.data); @@ -974,13 +977,11 @@ static struct dsp_scb_descriptor * _map_scb (struct snd_cs46xx *chip, char * nam index = find_free_scb_index (ins); + memset(&ins->scbs[index], 0, sizeof(ins->scbs[index])); strcpy(ins->scbs[index].scb_name, name); ins->scbs[index].address = dest; ins->scbs[index].index = index; - ins->scbs[index].proc_info = NULL; ins->scbs[index].ref_count = 1; - ins->scbs[index].deleted = 0; - spin_lock_init(&ins->scbs[index].lock); desc = (ins->scbs + index); ins->scbs[index].scb_symbol = add_symbol (chip, name, dest, SYMBOL_PARAMETER); @@ -1022,17 +1023,29 @@ _map_task_tree (struct snd_cs46xx *chip, char * name, u32 dest, u32 size) return desc; } +#define SCB_BYTES (0x10 * 4) + struct dsp_scb_descriptor * cs46xx_dsp_create_scb (struct snd_cs46xx *chip, char * name, u32 * scb_data, u32 dest) { struct dsp_scb_descriptor * desc; +#ifdef CONFIG_PM + /* copy the data for resume */ + scb_data = kmemdup(scb_data, SCB_BYTES, GFP_KERNEL); + if (!scb_data) + return NULL; +#endif + desc = _map_scb (chip,name,dest); if (desc) { desc->data = scb_data; _dsp_create_scb(chip,scb_data,dest); } else { snd_printk(KERN_ERR "dsp_spos: failed to map SCB\n"); +#ifdef CONFIG_PM + kfree(scb_data); +#endif } return desc; @@ -1988,7 +2001,28 @@ int cs46xx_dsp_resume(struct snd_cs46xx * chip) continue; _dsp_create_scb(chip, s->data, s->address); } - + for (i = 0; i < ins->nscb; i++) { + struct dsp_scb_descriptor *s = &ins->scbs[i]; + if (s->deleted) + continue; + if (s->updated) + cs46xx_dsp_spos_update_scb(chip, s); + if (s->volume_set) + cs46xx_dsp_scb_set_volume(chip, s, + s->volume[0], s->volume[1]); + } + if (ins->spdif_status_out & DSP_SPDIF_STATUS_HW_ENABLED) { + cs46xx_dsp_enable_spdif_hw(chip); + snd_cs46xx_poke(chip, (ins->ref_snoop_scb->address + 2) << 2, + (OUTPUT_SNOOP_BUFFER + 0x10) << 0x10); + if (ins->spdif_status_out & DSP_SPDIF_STATUS_PLAYBACK_OPEN) + cs46xx_poke_via_dsp(chip, SP_SPDOUT_CSUV, + ins->spdif_csuv_stream); + } + if (chip->dsp_spos_instance->spdif_status_in) { + cs46xx_poke_via_dsp(chip, SP_ASER_COUNTDOWN, 0x80000005); + cs46xx_poke_via_dsp(chip, SP_SPDIN_CONTROL, 0x800003ff); + } return 0; } #endif diff --git a/sound/pci/cs46xx/dsp_spos.h b/sound/pci/cs46xx/dsp_spos.h index f9e169d..ca47a81 100644 --- a/sound/pci/cs46xx/dsp_spos.h +++ b/sound/pci/cs46xx/dsp_spos.h @@ -212,6 +212,7 @@ static inline void cs46xx_dsp_spos_update_scb (struct snd_cs46xx * chip, (scb->address + SCBsubListPtr) << 2, (scb->sub_list_ptr->address << 0x10) | (scb->next_scb_ptr->address)); + scb->updated = 1; } static inline void cs46xx_dsp_scb_set_volume (struct snd_cs46xx * chip, @@ -222,6 +223,9 @@ static inline void cs46xx_dsp_scb_set_volume (struct snd_cs46xx * chip, snd_cs46xx_poke(chip, (scb->address + SCBVolumeCtrl) << 2, val); snd_cs46xx_poke(chip, (scb->address + SCBVolumeCtrl + 1) << 2, val); + scb->volume_set = 1; + scb->volume[0] = left; + scb->volume[1] = right; } #endif /* __DSP_SPOS_H__ */ #endif /* CONFIG_SND_CS46XX_NEW_DSP */ diff --git a/sound/pci/cs46xx/dsp_spos_scb_lib.c b/sound/pci/cs46xx/dsp_spos_scb_lib.c index dd7c41b..00b148a 100644 --- a/sound/pci/cs46xx/dsp_spos_scb_lib.c +++ b/sound/pci/cs46xx/dsp_spos_scb_lib.c @@ -115,7 +115,6 @@ static void cs46xx_dsp_proc_scb_info_read (struct snd_info_entry *entry, static void _dsp_unlink_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * scb) { struct dsp_spos_instance * ins = chip->dsp_spos_instance; - unsigned long flags; if ( scb->parent_scb_ptr ) { /* unlink parent SCB */ @@ -153,8 +152,6 @@ static void _dsp_unlink_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor scb->next_scb_ptr = ins->the_null_scb; } - spin_lock_irqsave(&chip->reg_lock, flags); - /* update parent first entry in DSP RAM */ cs46xx_dsp_spos_update_scb(chip,scb->parent_scb_ptr); @@ -162,7 +159,6 @@ static void _dsp_unlink_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor cs46xx_dsp_spos_update_scb(chip,scb); scb->parent_scb_ptr = NULL; - spin_unlock_irqrestore(&chip->reg_lock, flags); } } @@ -197,9 +193,9 @@ void cs46xx_dsp_remove_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * goto _end; #endif - spin_lock_irqsave(&scb->lock, flags); + spin_lock_irqsave(&chip->reg_lock, flags); _dsp_unlink_scb (chip,scb); - spin_unlock_irqrestore(&scb->lock, flags); + spin_unlock_irqrestore(&chip->reg_lock, flags); cs46xx_dsp_proc_free_scb_desc(scb); if (snd_BUG_ON(!scb->scb_symbol)) @@ -207,6 +203,10 @@ void cs46xx_dsp_remove_scb (struct snd_cs46xx *chip, struct dsp_scb_descriptor * remove_symbol (chip,scb->scb_symbol); ins->scbs[scb->index].deleted = 1; +#ifdef CONFIG_PM + kfree(ins->scbs[scb->index].data); + ins->scbs[scb->index].data = NULL; +#endif if (scb->index < ins->scb_highest_frag_index) ins->scb_highest_frag_index = scb->index; @@ -1508,20 +1508,17 @@ int cs46xx_dsp_pcm_unlink (struct snd_cs46xx * chip, chip->dsp_spos_instance->npcm_channels <= 0)) return -EIO; - spin_lock(&pcm_channel->src_scb->lock); - + spin_lock_irqsave(&chip->reg_lock, flags); if (pcm_channel->unlinked) { - spin_unlock(&pcm_channel->src_scb->lock); + spin_unlock_irqrestore(&chip->reg_lock, flags); return -EIO; } - spin_lock_irqsave(&chip->reg_lock, flags); pcm_channel->unlinked = 1; - spin_unlock_irqrestore(&chip->reg_lock, flags); _dsp_unlink_scb (chip,pcm_channel->pcm_reader_scb); + spin_unlock_irqrestore(&chip->reg_lock, flags); - spin_unlock(&pcm_channel->src_scb->lock); return 0; } @@ -1533,10 +1530,10 @@ int cs46xx_dsp_pcm_link (struct snd_cs46xx * chip, struct dsp_scb_descriptor * src_scb = pcm_channel->src_scb; unsigned long flags; - spin_lock(&pcm_channel->src_scb->lock); + spin_lock_irqsave(&chip->reg_lock, flags); if (pcm_channel->unlinked == 0) { - spin_unlock(&pcm_channel->src_scb->lock); + spin_unlock_irqrestore(&chip->reg_lock, flags); return -EIO; } @@ -1552,8 +1549,6 @@ int cs46xx_dsp_pcm_link (struct snd_cs46xx * chip, snd_BUG_ON(pcm_channel->pcm_reader_scb->parent_scb_ptr); pcm_channel->pcm_reader_scb->parent_scb_ptr = parent_scb; - spin_lock_irqsave(&chip->reg_lock, flags); - /* update SCB entry in DSP RAM */ cs46xx_dsp_spos_update_scb(chip,pcm_channel->pcm_reader_scb); @@ -1562,8 +1557,6 @@ int cs46xx_dsp_pcm_link (struct snd_cs46xx * chip, pcm_channel->unlinked = 0; spin_unlock_irqrestore(&chip->reg_lock, flags); - - spin_unlock(&pcm_channel->src_scb->lock); return 0; } @@ -1596,13 +1589,17 @@ cs46xx_add_record_source (struct snd_cs46xx *chip, struct dsp_scb_descriptor * s int cs46xx_src_unlink(struct snd_cs46xx *chip, struct dsp_scb_descriptor * src) { + unsigned long flags; + if (snd_BUG_ON(!src->parent_scb_ptr)) return -EINVAL; /* mute SCB */ cs46xx_dsp_scb_set_volume (chip,src,0,0); + spin_lock_irqsave(&chip->reg_lock, flags); _dsp_unlink_scb (chip,src); + spin_unlock_irqrestore(&chip->reg_lock, flags); return 0; } -- cgit v1.1 From 75d1aeb9d6899b10420d10284e8ea894b2794224 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 22 Dec 2009 11:56:32 +0100 Subject: ALSA: hda - Add Bass Speaker switch for HP dv7 The bass speaker is controlled via GPIO5. Tested-by: Wael Nasreddine Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 57 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 57 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 0bafea9..a4526d0 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -5402,6 +5402,54 @@ static int stac92hd71bxx_connected_smuxes(struct hda_codec *codec, return 0; } +/* HP dv7 bass switch - GPIO5 */ +#define stac_hp_bass_gpio_info snd_ctl_boolean_mono_info +static int stac_hp_bass_gpio_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + ucontrol->value.integer.value[0] = !!(spec->gpio_data & 0x20); + return 0; +} + +static int stac_hp_bass_gpio_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + unsigned int gpio_data; + + gpio_data = (spec->gpio_data & ~0x20) | + (ucontrol->value.integer.value[0] ? 0x20 : 0); + if (gpio_data == spec->gpio_data) + return 0; + spec->gpio_data = gpio_data; + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); + return 1; +} + +static struct snd_kcontrol_new stac_hp_bass_sw_ctrl = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = stac_hp_bass_gpio_info, + .get = stac_hp_bass_gpio_get, + .put = stac_hp_bass_gpio_put, +}; + +static int stac_add_hp_bass_switch(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + + if (!stac_control_new(spec, &stac_hp_bass_sw_ctrl, + "Bass Speaker Playback Switch", 0)) + return -ENOMEM; + + spec->gpio_mask |= 0x20; + spec->gpio_dir |= 0x20; + spec->gpio_data |= 0x20; + return 0; +} + static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -5642,6 +5690,15 @@ again: return err; } + /* enable bass on HP dv7 */ + if (spec->board_config == STAC_HP_DV5) { + unsigned int cap; + cap = snd_hda_param_read(codec, 0x1, AC_PAR_GPIO_CAP); + cap &= AC_GPIO_IO_COUNT; + if (cap >= 6) + stac_add_hp_bass_switch(codec); + } + codec->proc_widget_hook = stac92hd7x_proc_hook; return 0; -- cgit v1.1 From 21949f00a022e090a7e8bc9a01dfca88273c6146 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 23 Dec 2009 08:31:59 +0100 Subject: ALSA: hda - Fix NID association for capture mixers Fix the wrong implementation of NID <-> kctl mapping for capture mixers introduced by the ocmmit 5b0cb1d850c26893b1468b3a519433a1b7a176be. So far, the driver returns an error at probe. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 26 -------------------------- sound/pci/hda/hda_local.h | 2 -- sound/pci/hda/patch_analog.c | 3 +-- sound/pci/hda/patch_cirrus.c | 12 ++++++++---- sound/pci/hda/patch_cmedia.c | 3 +-- sound/pci/hda/patch_realtek.c | 3 +-- sound/pci/hda/patch_via.c | 3 +-- 7 files changed, 12 insertions(+), 40 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index c848ec0..29c90d7 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3537,32 +3537,6 @@ int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew) } EXPORT_SYMBOL_HDA(snd_hda_add_new_ctls); -/** - * snd_hda_add_nids - assign nids to controls from the array - * @codec: the HDA codec - * @kctl: struct snd_kcontrol - * @index: index to kctl - * @nids: the array of hda_nid_t - * @size: count of hda_nid_t items - * - * This helper function assigns NIDs in the given array to a control element. - * - * Returns 0 if successful, or a negative error code. - */ -int snd_hda_add_nids(struct hda_codec *codec, struct snd_kcontrol *kctl, - unsigned int index, hda_nid_t *nids, unsigned int size) -{ - int err; - - for ( ; size > 0; size--, nids++) { - err = snd_hda_add_nid(codec, kctl, index, *nids); - if (err < 0) - return err; - } - return 0; -} -EXPORT_SYMBOL_HDA(snd_hda_add_nids); - #ifdef CONFIG_SND_HDA_POWER_SAVE static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index d505d05..7cee364 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -343,8 +343,6 @@ int snd_hda_check_board_codec_sid_config(struct hda_codec *codec, const struct snd_pci_quirk *tbl); int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew); -int snd_hda_add_nids(struct hda_codec *codec, struct snd_kcontrol *kctl, - unsigned int index, hda_nid_t *nids, unsigned int size); /* * unsolicited event handler diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 92b72d4..45ee352 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -244,8 +244,7 @@ static int ad198x_build_controls(struct hda_codec *codec) if (!kctl) kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); for (i = 0; kctl && i < kctl->count; i++) { - err = snd_hda_add_nids(codec, kctl, i, spec->capsrc_nids, - spec->input_mux->num_items); + err = snd_hda_add_nid(codec, kctl, i, spec->capsrc_nids[i]); if (err < 0) return err; } diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 093cfbb..7de782a 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -753,6 +753,7 @@ static int build_input(struct hda_codec *codec) spec->capture_bind[1] = make_bind_capture(codec, &snd_hda_bind_vol); for (i = 0; i < 2; i++) { struct snd_kcontrol *kctl; + int n; if (!spec->capture_bind[i]) return -ENOMEM; kctl = snd_ctl_new1(&cs_capture_ctls[i], codec); @@ -762,10 +763,13 @@ static int build_input(struct hda_codec *codec) err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; - err = snd_hda_add_nids(codec, kctl, 0, spec->adc_nid, - spec->num_inputs); - if (err < 0) - return err; + for (n = 0; n < AUTO_PIN_LAST; n++) { + if (!spec->adc_nid[n]) + continue; + err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[i]); + if (err < 0) + return err; + } } if (spec->num_inputs > 1 && !spec->mic_detect) { diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index cc1c223..ff60908 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -345,8 +345,7 @@ static int cmi9880_build_controls(struct hda_codec *codec) /* assign Capture Source enums to NID */ kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); for (i = 0; kctl && i < kctl->count; i++) { - err = snd_hda_add_nids(codec, kctl, i, spec->adc_nids, - spec->input_mux->num_items); + err = snd_hda_add_nid(codec, kctl, i, spec->adc_nids[i]); if (err < 0) return err; } diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e7cdc6a..a451990 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2551,8 +2551,7 @@ static int alc_build_controls(struct hda_codec *codec) hda_nid_t *nids = spec->capsrc_nids; if (!nids) nids = spec->adc_nids; - err = snd_hda_add_nids(codec, kctl, i, nids, - spec->input_mux->num_items); + err = snd_hda_add_nid(codec, kctl, i, nids[i]); if (err < 0) return err; } diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index de4839e..9ddc373 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1907,8 +1907,7 @@ static int via_build_controls(struct hda_codec *codec) /* assign Capture Source enums to NID */ kctl = snd_hda_find_mixer_ctl(codec, "Input Source"); for (i = 0; kctl && i < kctl->count; i++) { - err = snd_hda_add_nids(codec, kctl, i, spec->mux_nids, - spec->input_mux->num_items); + err = snd_hda_add_nid(codec, kctl, i, spec->mux_nids[i]); if (err < 0) return err; } -- cgit v1.1 From 44eba3e82b35ae796826a65d8040001582adc10a Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Wed, 23 Dec 2009 18:02:41 +0100 Subject: ALSA: jazz16: refine dma and irq selection Narrow the dma and irq selection after the DOS driver. Add ALSA configuration description as well. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 15 +++++++++++++++ sound/isa/sb/jazz16.c | 21 ++++++++++++++++++++- sound/isa/sb/sb8_main.c | 3 ++- 3 files changed, 37 insertions(+), 2 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 7a0a4a9..c540637 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1123,6 +1123,21 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. This module supports multiple cards, autoprobe and ISA PnP. + Module snd-jazz16 + ------------------- + + Module for Media Vision Jazz16 chipset. The chipset consists of 3 chips: + MVD1216 + MVA416 + MVA514. + + port - port # for SB DSP chip (0x210,0x220,0x230,0x240,0x250,0x260) + irq - IRQ # for SB DSP chip (3,5,7,9,10,15) + dma8 - DMA # for SB DSP chip (1,3) + dma16 - DMA # for SB DSP chip (5,7) + mpu_port - MPU-401 port # (0x300,0x310,0x320,0x330) + mpu_irq - MPU-401 irq # (2,3,5,7) + + This module supports multiple cards. + Module snd-korg1212 ------------------- diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c index d52966b..8d21a3f 100644 --- a/sound/isa/sb/jazz16.c +++ b/sound/isa/sb/jazz16.c @@ -189,10 +189,29 @@ static int __devinit snd_jazz16_match(struct device *devptr, unsigned int dev) if (port[dev] == SNDRV_AUTO_PORT) { snd_printk(KERN_ERR "please specify port\n"); return 0; + } else if (port[dev] == 0x200 || (port[dev] & ~0x270)) { + snd_printk(KERN_ERR "incorrect port specified\n"); + return 0; + } + if (dma8[dev] != SNDRV_AUTO_DMA && + dma8[dev] != 1 && dma8[dev] != 3) { + snd_printk(KERN_ERR "dma8 must be 1 or 3\n"); + return 0; } if (dma16[dev] != SNDRV_AUTO_DMA && dma16[dev] != 5 && dma16[dev] != 7) { - snd_printk(KERN_ERR "dma16 must be 5 or 7"); + snd_printk(KERN_ERR "dma16 must be 5 or 7\n"); + return 0; + } + if (mpu_port[dev] != SNDRV_AUTO_PORT && + (mpu_port[dev] & ~0x030) != 0x300) { + snd_printk(KERN_ERR "incorrect mpu_port specified\n"); + return 0; + } + if (mpu_irq[dev] != SNDRV_AUTO_DMA && + mpu_irq[dev] != 2 && mpu_irq[dev] != 3 && + mpu_irq[dev] != 5 && mpu_irq[dev] != 7) { + snd_printk(KERN_ERR "mpu_irq must be 2, 3, 5 or 7\n"); return 0; } return 1; diff --git a/sound/isa/sb/sb8_main.c b/sound/isa/sb/sb8_main.c index 3222aed..7d84c9f 100644 --- a/sound/isa/sb/sb8_main.c +++ b/sound/isa/sb/sb8_main.c @@ -510,7 +510,8 @@ static int snd_sb8_open(struct snd_pcm_substream *substream) } switch (chip->hardware) { case SB_HW_JAZZ16: - runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; + if (chip->dma16 == 5 || chip->dma16 == 7) + runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; runtime->hw.rates |= SNDRV_PCM_RATE_8000_48000; runtime->hw.rate_min = 4000; runtime->hw.rate_max = 50000; -- cgit v1.1 From 043958e602ac2cbf918c0dab1e4e2a7f9751ebf6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 26 Dec 2009 10:36:12 +0100 Subject: ALSA: hda - Add more hints for GPIO setup of IDT/STAC codecs gpio_led, gpio_led_polarity and gpio_mute are added now. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 14 ++++++++++++++ 1 file changed, 14 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 247be19..69dd5a4 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4184,9 +4184,23 @@ static void stac_store_hints(struct hda_codec *codec) p = snd_hda_get_hint(codec, "eapd_mask"); if (p) spec->eapd_mask = simple_strtoul(p, NULL, 0) & spec->gpio_mask; + p = snd_hda_get_hint(codec, "gpio_mute"); + if (p) + spec->gpio_mute = simple_strtoul(p, NULL, 0) & spec->gpio_mask; val = snd_hda_get_bool_hint(codec, "eapd_switch"); if (val >= 0) spec->eapd_switch = val; + p = snd_hda_get_hint(codec, "gpio_led_polarity"); + if (p) + spec->gpio_led_polarity = simple_strtoul(p, NULL, 0); + p = snd_hda_get_hint(codec, "gpio_led"); + if (p) { + spec->gpio_led = simple_strtoul(p, NULL, 0); + spec->gpio_mask |= spec->gpio_led; + spec->gpio_dir |= spec->gpio_led; + if (spec->gpio_led_polarity) + spec->gpio_data |= spec->gpio_led; + } } static int stac92xx_init(struct hda_codec *codec) -- cgit v1.1 From 92ee6162c48fab24f0676969f0f147fc12f8f21c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 27 Dec 2009 11:18:59 +0100 Subject: ALSA: hda - Add snd_hda_shutup_pins() helper function Add a common helper function for clearing pin controls before suspend. Use the pincfg array instead of looking through all widget tree. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 19 +++++++++++++++++++ sound/pci/hda/hda_codec.h | 1 + sound/pci/hda/patch_sigmatel.c | 12 +----------- 3 files changed, 21 insertions(+), 11 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b3554df..94ae69f 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -899,6 +899,25 @@ static void restore_pincfgs(struct hda_codec *codec) } } +/** + * snd_hda_shutup_pins - Shut up all pins + * @codec: the HDA codec + * + * Clear all pin controls to shup up before suspend for avoiding click noise. + * The controls aren't cached so that they can be resumed properly. + */ +void snd_hda_shutup_pins(struct hda_codec *codec) +{ + int i; + for (i = 0; i < codec->init_pins.used; i++) { + struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + /* use read here for syncing after issuing each verb */ + snd_hda_codec_read(codec, pin->nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + } +} +EXPORT_SYMBOL_HDA(snd_hda_shutup_pins); + static void init_hda_cache(struct hda_cache_rec *cache, unsigned int record_size); static void free_hda_cache(struct hda_cache_rec *cache); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 0d08ad5..11c4aa8 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -898,6 +898,7 @@ int snd_hda_codec_set_pincfg(struct hda_codec *codec, hda_nid_t nid, unsigned int cfg); int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, hda_nid_t nid, unsigned int cfg); /* for hwdep */ +void snd_hda_shutup_pins(struct hda_codec *codec); /* * Mixer diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 69dd5a4..dc1d9f1 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4385,18 +4385,8 @@ static void stac92xx_free_kctls(struct hda_codec *codec) static void stac92xx_shutup(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - int i; - hda_nid_t nid; - /* reset each pin before powering down DAC/ADC to avoid click noise */ - nid = codec->start_nid; - for (i = 0; i < codec->num_nodes; i++, nid++) { - unsigned int wcaps = get_wcaps(codec, nid); - unsigned int wid_type = get_wcaps_type(wcaps); - if (wid_type == AC_WID_PIN) - snd_hda_codec_read(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0); - } + snd_hda_shutup_pins(codec); if (spec->eapd_mask) stac_gpio_set(codec, spec->gpio_mask, -- cgit v1.1 From a4e09aa3cf592d9f084ff4ceb216be40c4c265dc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 27 Dec 2009 11:22:24 +0100 Subject: ALSA: hda - Fix click noises at suspend/free with Realtek codecs Call snd_hda_shutup_pins() at suspend and free for avoiding click noises. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6361e6b..cd6d139 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3693,6 +3693,11 @@ static int alc_build_pcms(struct hda_codec *codec) return 0; } +static inline void alc_shutup(struct hda_codec *codec) +{ + snd_hda_shutup_pins(codec); +} + static void alc_free_kctls(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -3713,6 +3718,7 @@ static void alc_free(struct hda_codec *codec) if (!spec) return; + alc_shutup(codec); alc_free_kctls(codec); kfree(spec); snd_hda_detach_beep_device(codec); @@ -3722,6 +3728,7 @@ static void alc_free(struct hda_codec *codec) static int alc_suspend(struct hda_codec *codec, pm_message_t state) { struct alc_spec *spec = codec->spec; + alc_shutup(codec); if (spec && spec->power_hook) spec->power_hook(codec, 0); return 0; -- cgit v1.1 From b82855a0d76ebda1cc14c00040560d77bfa042ce Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 27 Dec 2009 11:24:56 +0100 Subject: ALSA: hda - Add sanity check for storing the user-defined pin configs Check whether the given NID is a pin widget before storing the user-defined pin configs. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_codec.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 94ae69f..d02ea89 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -824,6 +824,9 @@ int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, struct hda_pincfg *pin; unsigned int oldcfg; + if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN) + return -EINVAL; + oldcfg = snd_hda_codec_get_pincfg(codec, nid); pin = look_up_pincfg(codec, list, nid); if (!pin) { -- cgit v1.1 From 014c41fce1bd5cec381e70fc6f58fdfc96cdaf69 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 27 Dec 2009 13:53:24 +0100 Subject: ALSA: hda - Use strict_strtoul() Rewrite the codes to use strict_strtoul() instead of simple_strtoul(). Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_hwdep.c | 7 ++++-- sound/pci/hda/patch_sigmatel.c | 48 +++++++++++++++++++++++------------------- 2 files changed, 31 insertions(+), 24 deletions(-) diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 40ccb41..b36919c 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -293,8 +293,11 @@ static ssize_t type##_store(struct device *dev, \ { \ struct snd_hwdep *hwdep = dev_get_drvdata(dev); \ struct hda_codec *codec = hwdep->private_data; \ - char *after; \ - codec->type = simple_strtoul(buf, &after, 0); \ + unsigned long val; \ + int err = strict_strtoul(buf, 0, &val); \ + if (err < 0) \ + return err; \ + codec->type = val; \ return count; \ } diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index dc1d9f1..e28c810 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4159,43 +4159,47 @@ static void stac92xx_power_down(struct hda_codec *codec) static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid, int enable); +static inline int get_int_hint(struct hda_codec *codec, const char *key, + int *valp) +{ + const char *p; + p = snd_hda_get_hint(codec, key); + if (p) { + unsigned long val; + if (!strict_strtoul(p, 0, &val)) { + *valp = val; + return 1; + } + } + return 0; +} + /* override some hints from the hwdep entry */ static void stac_store_hints(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - const char *p; int val; val = snd_hda_get_bool_hint(codec, "hp_detect"); if (val >= 0) spec->hp_detect = val; - p = snd_hda_get_hint(codec, "gpio_mask"); - if (p) { - spec->gpio_mask = simple_strtoul(p, NULL, 0); + if (get_int_hint(codec, "gpio_mask", &spec->gpio_mask)) { spec->eapd_mask = spec->gpio_dir = spec->gpio_data = spec->gpio_mask; } - p = snd_hda_get_hint(codec, "gpio_dir"); - if (p) - spec->gpio_dir = simple_strtoul(p, NULL, 0) & spec->gpio_mask; - p = snd_hda_get_hint(codec, "gpio_data"); - if (p) - spec->gpio_data = simple_strtoul(p, NULL, 0) & spec->gpio_mask; - p = snd_hda_get_hint(codec, "eapd_mask"); - if (p) - spec->eapd_mask = simple_strtoul(p, NULL, 0) & spec->gpio_mask; - p = snd_hda_get_hint(codec, "gpio_mute"); - if (p) - spec->gpio_mute = simple_strtoul(p, NULL, 0) & spec->gpio_mask; + if (get_int_hint(codec, "gpio_dir", &spec->gpio_dir)) + spec->gpio_mask &= spec->gpio_mask; + if (get_int_hint(codec, "gpio_data", &spec->gpio_data)) + spec->gpio_dir &= spec->gpio_mask; + if (get_int_hint(codec, "eapd_mask", &spec->eapd_mask)) + spec->eapd_mask &= spec->gpio_mask; + if (get_int_hint(codec, "gpio_mute", &spec->gpio_mute)) + spec->gpio_mute &= spec->gpio_mask; val = snd_hda_get_bool_hint(codec, "eapd_switch"); if (val >= 0) spec->eapd_switch = val; - p = snd_hda_get_hint(codec, "gpio_led_polarity"); - if (p) - spec->gpio_led_polarity = simple_strtoul(p, NULL, 0); - p = snd_hda_get_hint(codec, "gpio_led"); - if (p) { - spec->gpio_led = simple_strtoul(p, NULL, 0); + get_int_hint(codec, "gpio_led_polarity", &spec->gpio_led_polarity); + if (get_int_hint(codec, "gpio_led", &spec->gpio_led)) { spec->gpio_mask |= spec->gpio_led; spec->gpio_dir |= spec->gpio_led; if (spec->gpio_led_polarity) -- cgit v1.1 From ea52bf260ecbb175339af3178c15788df21b7516 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 27 Dec 2009 18:48:29 -0500 Subject: ALSA: hda: Add powerdown for Analog Devices HDA codecs This patch ports powerdown fixes to AD198x. Currently we only turn off Front and HP for suspend, but this is easily extended for additional nids. Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 68 ++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 68 insertions(+) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 45ee352..cecd3c1 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -441,6 +441,11 @@ static int ad198x_build_pcms(struct hda_codec *codec) return 0; } +static inline void ad198x_shutup(struct hda_codec *codec) +{ + snd_hda_shutup_pins(codec); +} + static void ad198x_free_kctls(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; @@ -454,6 +459,46 @@ static void ad198x_free_kctls(struct hda_codec *codec) snd_array_free(&spec->kctls); } +static void ad198x_power_eapd_write(struct hda_codec *codec, hda_nid_t front, + hda_nid_t hp) +{ + struct ad198x_spec *spec = codec->spec; + snd_hda_codec_write(codec, front, 0, AC_VERB_SET_EAPD_BTLENABLE, + !spec->inv_eapd ? 0x00 : 0x02); + snd_hda_codec_write(codec, hp, 0, AC_VERB_SET_EAPD_BTLENABLE, + !spec->inv_eapd ? 0x00 : 0x02); +} + +static void ad198x_power_eapd(struct hda_codec *codec) +{ + /* We currently only handle front, HP */ + switch (codec->vendor_id) { + case 0x11d41882: + case 0x11d4882a: + case 0x11d41884: + case 0x11d41984: + case 0x11d41883: + case 0x11d4184a: + case 0x11d4194a: + case 0x11d4194b: + ad198x_power_eapd_write(codec, 0x12, 0x11); + break; + case 0x11d41981: + case 0x11d41983: + ad198x_power_eapd_write(codec, 0x05, 0x06); + break; + case 0x11d41986: + ad198x_power_eapd_write(codec, 0x1b, 0x1a); + break; + case 0x11d41988: + case 0x11d4198b: + case 0x11d4989a: + case 0x11d4989b: + ad198x_power_eapd_write(codec, 0x29, 0x22); + break; + } +} + static void ad198x_free(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; @@ -461,11 +506,29 @@ static void ad198x_free(struct hda_codec *codec) if (!spec) return; + ad198x_shutup(codec); ad198x_free_kctls(codec); kfree(spec); snd_hda_detach_beep_device(codec); } +#ifdef SND_HDA_NEEDS_RESUME +static int ad198x_suspend(struct hda_codec *codec, pm_message_t state) +{ + ad198x_shutup(codec); + ad198x_power_eapd(codec); + return 0; +} + +static int ad198x_resume(struct hda_codec *codec) +{ + ad198x_init(codec); + snd_hda_codec_resume_amp(codec); + snd_hda_codec_resume_cache(codec); + return 0; +} +#endif + static struct hda_codec_ops ad198x_patch_ops = { .build_controls = ad198x_build_controls, .build_pcms = ad198x_build_pcms, @@ -474,6 +537,11 @@ static struct hda_codec_ops ad198x_patch_ops = { #ifdef CONFIG_SND_HDA_POWER_SAVE .check_power_status = ad198x_check_power_status, #endif +#ifdef SND_HDA_NEEDS_RESUME + .suspend = ad198x_suspend, + .resume = ad198x_resume, +#endif + .reboot_notify = ad198x_shutup, }; -- cgit v1.1 From c97259df3f2e163c72f4d0685c61fb2e026dc989 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 27 Dec 2009 18:52:08 -0500 Subject: ALSA: hda: Refactor powerdown for Realtek HDA codecs This patch converts the alc889 Aspire-specific powerdown to a generic one. Like the previous effort, it currently only handles Front and PCM but can be easily extended to cover other nids. The existing hook for alc889 Aspire-specific remains enabled. Upon further testing, I've added its use for ALC861_AUTO as well. Following patches will enable them for other quirks. Tested-by: Dr. David Alan Gilbert Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 60 +++++++++++++++++++++++++++---------------- 1 file changed, 38 insertions(+), 22 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cd6d139..141ff44 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -338,7 +338,7 @@ struct alc_spec { void (*init_hook)(struct hda_codec *codec); void (*unsol_event)(struct hda_codec *codec, unsigned int res); #ifdef CONFIG_SND_HDA_POWER_SAVE - void (*power_hook)(struct hda_codec *codec, int power); + void (*power_hook)(struct hda_codec *codec); #endif /* for pin sensing */ @@ -391,7 +391,7 @@ struct alc_config_preset { void (*init_hook)(struct hda_codec *); #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_amp_list *loopbacks; - void (*power_hook)(struct hda_codec *codec, int power); + void (*power_hook)(struct hda_codec *codec); #endif }; @@ -1835,16 +1835,6 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x1b; } -#ifdef CONFIG_SND_HDA_POWER_SAVE -static void alc889_power_eapd(struct hda_codec *codec, int power) -{ - snd_hda_codec_write(codec, 0x14, 0, - AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0); - snd_hda_codec_write(codec, 0x15, 0, - AC_VERB_SET_EAPD_BTLENABLE, power ? 2 : 0); -} -#endif - /* * ALC880 3-stack model * @@ -3725,12 +3715,40 @@ static void alc_free(struct hda_codec *codec) } #ifdef CONFIG_SND_HDA_POWER_SAVE +static void alc_power_eapd(struct hda_codec *codec) +{ + /* We currently only handle front, HP */ + switch (codec->vendor_id) { + case 0x10ec0260: + snd_hda_codec_write(codec, 0x0f, 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x00); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x00); + break; + case 0x10ec0262: + case 0x10ec0267: + case 0x10ec0268: + case 0x10ec0269: + case 0x10ec0272: + case 0x10ec0660: + case 0x10ec0662: + case 0x10ec0663: + case 0x10ec0862: + case 0x10ec0889: + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x00); + snd_hda_codec_write(codec, 0x15, 0, + AC_VERB_SET_EAPD_BTLENABLE, 0x00); + break; + } +} + static int alc_suspend(struct hda_codec *codec, pm_message_t state) { struct alc_spec *spec = codec->spec; alc_shutup(codec); if (spec && spec->power_hook) - spec->power_hook(codec, 0); + spec->power_hook(codec); return 0; } #endif @@ -3738,16 +3756,9 @@ static int alc_suspend(struct hda_codec *codec, pm_message_t state) #ifdef SND_HDA_NEEDS_RESUME static int alc_resume(struct hda_codec *codec) { -#ifdef CONFIG_SND_HDA_POWER_SAVE - struct alc_spec *spec = codec->spec; -#endif codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); -#ifdef CONFIG_SND_HDA_POWER_SAVE - if (spec && spec->power_hook) - spec->power_hook(codec, 1); -#endif return 0; } #endif @@ -3767,6 +3778,7 @@ static struct hda_codec_ops alc_patch_ops = { .suspend = alc_suspend, .check_power_status = alc_check_power_status, #endif + .reboot_notify = alc_shutup, }; @@ -9547,7 +9559,7 @@ static struct alc_config_preset alc882_presets[] = { .setup = alc889_acer_aspire_8930g_setup, .init_hook = alc_automute_amp, #ifdef CONFIG_SND_HDA_POWER_SAVE - .power_hook = alc889_power_eapd, + .power_hook = alc_power_eapd, #endif }, [ALC888_ACER_ASPIRE_7730G] = { @@ -14984,9 +14996,13 @@ static int patch_alc861(struct hda_codec *codec) spec->vmaster_nid = 0x03; codec->patch_ops = alc_patch_ops; - if (board_config == ALC861_AUTO) + if (board_config == ALC861_AUTO) { spec->init_hook = alc861_auto_init; #ifdef CONFIG_SND_HDA_POWER_SAVE + spec->power_hook = alc_power_eapd; +#endif + } +#ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) spec->loopback.amplist = alc861_loopbacks; #endif -- cgit v1.1 From 7d2b451e65d255427c108e990507964ac39c13ee Mon Sep 17 00:00:00 2001 From: Sergiy Kovalchuk Date: Sun, 27 Dec 2009 09:13:41 -0800 Subject: ALSA: usb-audio - Added functionality for E-mu 0404USB/0202USB/TrackerPre Added functionality: 1) Extension Units support (all XU settings now available at alsamixer, kmix, etc): - "AnalogueIn soft limiter" switch; - "Sample rate" selector (values 0,1,2,3,4,5 corresponds to 44.1 48 ... 192 kHz); - "DigitalIn CLK source" selector (internal/external) (**); - "DigitalOut format SPDIF/AC3" switch (**); (**)E-mu-0404usb only. 2) Automatic device sample rate adjustment depending on substream samplerate for both capture and playback substream. [minor coding-style fixes by tiwai] Signed-off-by: Sergiy Kovalchuk Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 49 ++++++++++++++++++++++++++++++++++ sound/usb/usbaudio.h | 13 +++++++++ sound/usb/usbmixer.c | 75 +++++++++++++++++++++++++++++++++++++++++++++++++--- 3 files changed, 133 insertions(+), 4 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 31b63ea..286fa14 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1271,6 +1271,47 @@ static int init_usb_sample_rate(struct usb_device *dev, int iface, } /* + * For E-Mu 0404USB/0202USB/TrackerPre sample rate should be set for device, + * not for interface. + */ +static void set_format_emu_quirk(struct snd_usb_substream *subs, + struct audioformat *fmt) +{ + unsigned char emu_samplerate_id = 0; + + /* When capture is active + * sample rate shouldn't be changed + * by playback substream + */ + if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) { + if (subs->stream->substream[SNDRV_PCM_STREAM_CAPTURE].interface != -1) + return; + } + + switch (fmt->rate_min) { + case 48000: + emu_samplerate_id = EMU_QUIRK_SR_48000HZ; + break; + case 88200: + emu_samplerate_id = EMU_QUIRK_SR_88200HZ; + break; + case 96000: + emu_samplerate_id = EMU_QUIRK_SR_96000HZ; + break; + case 176400: + emu_samplerate_id = EMU_QUIRK_SR_176400HZ; + break; + case 192000: + emu_samplerate_id = EMU_QUIRK_SR_192000HZ; + break; + default: + emu_samplerate_id = EMU_QUIRK_SR_44100HZ; + break; + } + snd_emuusb_set_samplerate(subs->stream->chip, emu_samplerate_id); +} + +/* * find a matching format and set up the interface */ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) @@ -1383,6 +1424,14 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->cur_audiofmt = fmt; + switch (subs->stream->chip->usb_id) { + case USB_ID(0x041e, 0x3f02): /* E-Mu 0202 USB */ + case USB_ID(0x041e, 0x3f04): /* E-Mu 0404 USB */ + case USB_ID(0x041e, 0x3f0a): /* E-Mu Tracker Pre */ + set_format_emu_quirk(subs, fmt); + break; + } + #if 0 printk(KERN_DEBUG "setting done: format = %d, rate = %d..%d, channels = %d\n", diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 9826337..1522167 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -208,6 +208,16 @@ struct snd_usb_midi_endpoint_info { /* */ +/*E-mu USB samplerate control quirk*/ +enum { + EMU_QUIRK_SR_44100HZ = 0, + EMU_QUIRK_SR_48000HZ, + EMU_QUIRK_SR_88200HZ, + EMU_QUIRK_SR_96000HZ, + EMU_QUIRK_SR_176400HZ, + EMU_QUIRK_SR_192000HZ +}; + #define combine_word(s) ((*(s)) | ((unsigned int)(s)[1] << 8)) #define combine_triple(s) (combine_word(s) | ((unsigned int)(s)[2] << 16)) #define combine_quad(s) (combine_triple(s) | ((unsigned int)(s)[3] << 24)) @@ -233,6 +243,9 @@ void snd_usbmidi_input_stop(struct list_head* p); void snd_usbmidi_input_start(struct list_head* p); void snd_usbmidi_disconnect(struct list_head *p); +void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, + unsigned char samplerate_id); + /* * retrieve usb_interface descriptor from the host interface * (conditional for compatibility with the older API) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index c998220..f5596cf 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -186,6 +186,21 @@ enum { USB_PROC_DCR_RELEASE = 6, }; +/*E-mu 0202(0404) eXtension Unit(XU) control*/ +enum { + USB_XU_CLOCK_RATE = 0xe301, + USB_XU_CLOCK_SOURCE = 0xe302, + USB_XU_DIGITAL_IO_STATUS = 0xe303, + USB_XU_DEVICE_OPTIONS = 0xe304, + USB_XU_DIRECT_MONITORING = 0xe305, + USB_XU_METERING = 0xe306 +}; +enum { + USB_XU_CLOCK_SOURCE_SELECTOR = 0x02, /* clock source*/ + USB_XU_CLOCK_RATE_SELECTOR = 0x03, /* clock rate */ + USB_XU_DIGITAL_FORMAT_SELECTOR = 0x01, /* the spdif format */ + USB_XU_SOFT_LIMIT_SELECTOR = 0x03 /* soft limiter */ +}; /* * manual mapping of mixer names @@ -1330,7 +1345,32 @@ static struct procunit_info procunits[] = { { USB_PROC_DCR, "DCR", dcr_proc_info }, { 0 }, }; - +/* + * predefined data for extension units + */ +static struct procunit_value_info clock_rate_xu_info[] = { + { USB_XU_CLOCK_RATE_SELECTOR, "Selector", USB_MIXER_U8, 0 }, + { 0 } +}; +static struct procunit_value_info clock_source_xu_info[] = { + { USB_XU_CLOCK_SOURCE_SELECTOR, "External", USB_MIXER_BOOLEAN }, + { 0 } +}; +static struct procunit_value_info spdif_format_xu_info[] = { + { USB_XU_DIGITAL_FORMAT_SELECTOR, "SPDIF/AC3", USB_MIXER_BOOLEAN }, + { 0 } +}; +static struct procunit_value_info soft_limit_xu_info[] = { + { USB_XU_SOFT_LIMIT_SELECTOR, " ", USB_MIXER_BOOLEAN }, + { 0 } +}; +static struct procunit_info extunits[] = { + { USB_XU_CLOCK_RATE, "Clock rate", clock_rate_xu_info }, + { USB_XU_CLOCK_SOURCE, "DigitalIn CLK source", clock_source_xu_info }, + { USB_XU_DIGITAL_IO_STATUS, "DigitalOut format:", spdif_format_xu_info }, + { USB_XU_DEVICE_OPTIONS, "AnalogueIn Soft Limit", soft_limit_xu_info }, + { 0 } +}; /* * build a processing/extension unit */ @@ -1391,8 +1431,18 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned cval->max = dsc[15]; cval->res = 1; cval->initialized = 1; - } else - get_min_max(cval, valinfo->min_value); + } else { + if (type == USB_XU_CLOCK_RATE) { + /* E-Mu USB 0404/0202/TrackerPre + * samplerate control quirk + */ + cval->min = 0; + cval->max = 5; + cval->res = 1; + cval->initialized = 1; + } else + get_min_max(cval, valinfo->min_value); + } kctl = snd_ctl_new1(&mixer_procunit_ctl, cval); if (! kctl) { @@ -1433,7 +1483,7 @@ static int parse_audio_processing_unit(struct mixer_build *state, int unitid, un static int parse_audio_extension_unit(struct mixer_build *state, int unitid, unsigned char *desc) { - return build_audio_procunit(state, unitid, desc, NULL, "Extension Unit"); + return build_audio_procunit(state, unitid, desc, extunits, "Extension Unit"); } @@ -2109,6 +2159,23 @@ static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) return 0; } +void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, + unsigned char samplerate_id) +{ + struct usb_mixer_interface *mixer; + struct usb_mixer_elem_info *cval; + int unitid = 12; /* SamleRate ExtensionUnit ID */ + + list_for_each_entry(mixer, &chip->mixer_list, list) { + cval = mixer->id_elems[unitid]; + if (cval) { + set_cur_ctl_value(cval, cval->control << 8, samplerate_id); + snd_usb_mixer_notify_id(mixer, unitid); + } + break; + } +} + int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, int ignore_error) { -- cgit v1.1 From adc8d31326c32a2a1e145ab80accbc3c6570b117 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 27 Dec 2009 12:19:57 -0500 Subject: ALSA: usb-audio: make buffer pointer based on bytes instead on frames Since there are devices that do not align the size of their data packets to frame boundaries, the driver needs to be able to keep track of partial frames. This patch prepares for support for such devices by changing the hwptr_done variable from a frame counter to a byte counter. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 76 +++++++++++++++++++++++++--------------------------- 1 file changed, 37 insertions(+), 39 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 286fa14..8fcb5d5 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -173,7 +173,7 @@ struct snd_usb_substream { unsigned int running: 1; /* running status */ - unsigned int hwptr_done; /* processed frame position in the buffer */ + unsigned int hwptr_done; /* processed byte position in the buffer */ unsigned int transfer_done; /* processed frames since last period update */ unsigned long active_mask; /* bitmask of active urbs */ unsigned long unlink_mask; /* bitmask of unlinked urbs */ @@ -342,7 +342,7 @@ static int retire_capture_urb(struct snd_usb_substream *subs, unsigned long flags; unsigned char *cp; int i; - unsigned int stride, len, oldptr; + unsigned int stride, frames, bytes, oldptr; int period_elapsed = 0; stride = runtime->frame_bits >> 3; @@ -353,29 +353,28 @@ static int retire_capture_urb(struct snd_usb_substream *subs, snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); // continue; } - len = urb->iso_frame_desc[i].actual_length / stride; - if (! len) - continue; + frames = urb->iso_frame_desc[i].actual_length / stride; + bytes = frames * stride; /* update the current pointer */ spin_lock_irqsave(&subs->lock, flags); oldptr = subs->hwptr_done; - subs->hwptr_done += len; - if (subs->hwptr_done >= runtime->buffer_size) - subs->hwptr_done -= runtime->buffer_size; - subs->transfer_done += len; + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + subs->transfer_done += frames; if (subs->transfer_done >= runtime->period_size) { subs->transfer_done -= runtime->period_size; period_elapsed = 1; } spin_unlock_irqrestore(&subs->lock, flags); /* copy a data chunk */ - if (oldptr + len > runtime->buffer_size) { - unsigned int cnt = runtime->buffer_size - oldptr; - unsigned int blen = cnt * stride; - memcpy(runtime->dma_area + oldptr * stride, cp, blen); - memcpy(runtime->dma_area, cp + blen, len * stride - blen); + if (oldptr + bytes > runtime->buffer_size * stride) { + unsigned int bytes1 = + runtime->buffer_size * stride - oldptr; + memcpy(runtime->dma_area + oldptr, cp, bytes1); + memcpy(runtime->dma_area, cp + bytes1, bytes - bytes1); } else { - memcpy(runtime->dma_area + oldptr * stride, cp, len * stride); + memcpy(runtime->dma_area + oldptr, cp, bytes); } } if (period_elapsed) @@ -562,24 +561,24 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, struct urb *urb) { - int i, stride, offs; - unsigned int counts; + int i, stride; + unsigned int counts, frames, bytes; unsigned long flags; int period_elapsed = 0; struct snd_urb_ctx *ctx = urb->context; stride = runtime->frame_bits >> 3; - offs = 0; + frames = 0; urb->dev = ctx->subs->dev; /* we need to set this at each time */ urb->number_of_packets = 0; spin_lock_irqsave(&subs->lock, flags); for (i = 0; i < ctx->packets; i++) { counts = snd_usb_audio_next_packet_size(subs); /* set up descriptor */ - urb->iso_frame_desc[i].offset = offs * stride; + urb->iso_frame_desc[i].offset = frames * stride; urb->iso_frame_desc[i].length = counts * stride; - offs += counts; + frames += counts; urb->number_of_packets++; subs->transfer_done += counts; if (subs->transfer_done >= runtime->period_size) { @@ -589,7 +588,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, if (subs->transfer_done > 0) { /* FIXME: fill-max mode is not * supported yet */ - offs -= subs->transfer_done; + frames -= subs->transfer_done; counts -= subs->transfer_done; urb->iso_frame_desc[i].length = counts * stride; @@ -599,7 +598,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, if (i < ctx->packets) { /* add a transfer delimiter */ urb->iso_frame_desc[i].offset = - offs * stride; + frames * stride; urb->iso_frame_desc[i].length = 0; urb->number_of_packets++; } @@ -609,26 +608,25 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, if (period_elapsed) /* finish at the period boundary */ break; } - if (subs->hwptr_done + offs > runtime->buffer_size) { + bytes = frames * stride; + if (subs->hwptr_done + bytes > runtime->buffer_size * stride) { /* err, the transferred area goes over buffer boundary. */ - unsigned int len = runtime->buffer_size - subs->hwptr_done; + unsigned int bytes1 = + runtime->buffer_size * stride - subs->hwptr_done; memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done * stride, - len * stride); - memcpy(urb->transfer_buffer + len * stride, - runtime->dma_area, - (offs - len) * stride); + runtime->dma_area + subs->hwptr_done, bytes1); + memcpy(urb->transfer_buffer + bytes1, + runtime->dma_area, bytes - bytes1); } else { memcpy(urb->transfer_buffer, - runtime->dma_area + subs->hwptr_done * stride, - offs * stride); + runtime->dma_area + subs->hwptr_done, bytes); } - subs->hwptr_done += offs; - if (subs->hwptr_done >= runtime->buffer_size) - subs->hwptr_done -= runtime->buffer_size; - runtime->delay += offs; + subs->hwptr_done += bytes; + if (subs->hwptr_done >= runtime->buffer_size * stride) + subs->hwptr_done -= runtime->buffer_size * stride; + runtime->delay += frames; spin_unlock_irqrestore(&subs->lock, flags); - urb->transfer_buffer_length = offs * stride; + urb->transfer_buffer_length = bytes; if (period_elapsed) snd_pcm_period_elapsed(subs->pcm_substream); return 0; @@ -901,18 +899,18 @@ static int wait_clear_urbs(struct snd_usb_substream *subs) /* - * return the current pcm pointer. just return the hwptr_done value. + * return the current pcm pointer. just based on the hwptr_done value. */ static snd_pcm_uframes_t snd_usb_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_usb_substream *subs; - snd_pcm_uframes_t hwptr_done; + unsigned int hwptr_done; subs = (struct snd_usb_substream *)substream->runtime->private_data; spin_lock(&subs->lock); hwptr_done = subs->hwptr_done; spin_unlock(&subs->lock); - return hwptr_done; + return hwptr_done / (substream->runtime->frame_bits >> 3); } -- cgit v1.1 From 98e89f606c38a310a20342f90e0c453e6afadf18 Mon Sep 17 00:00:00 2001 From: "John S. Gruber" Date: Sun, 27 Dec 2009 12:19:58 -0500 Subject: ALSA: usb-audio: relax urb data align. restriction HVR-950Q and HVR-850 only Addressing audio quality problem. In sound/usb/usbaudio.c, for the Hauppage HVR-950Q and HVR-850 only, change retire_capture_urb to allow transfers on audio sub-slot boundaries rather than audio slots boundaries. With these devices the left and right channel samples can be split between two different urbs. Throwing away extra channel samples causes a sound quality problem for stereo streams as the left and right channels are swapped repeatedly, perhaps many times per second. Urbs unaligned on sub-slot boundaries are still truncated to the next lowest stride (audio slot) to retain synchronization on samples even though left/right channel synchronization may be lost in this case. Detect the quirk using a case statement in snd_usb_audio_probe. BugLink: https://bugs.launchpad.net/ubuntu/+bug/495745 Signed-off-by: John S. Gruber Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 31 +++++++++++++++++++++++++++++-- sound/usb/usbaudio.h | 1 + 2 files changed, 30 insertions(+), 2 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 8fcb5d5..617515f 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -169,6 +169,7 @@ struct snd_usb_substream { unsigned int curpacksize; /* current packet size in bytes (for capture) */ unsigned int curframesize; /* current packet size in frames (for capture) */ unsigned int fill_max: 1; /* fill max packet size always */ + unsigned int txfr_quirk:1; /* allow sub-frame alignment */ unsigned int fmt_type; /* USB audio format type (1-3) */ unsigned int running: 1; /* running status */ @@ -353,14 +354,25 @@ static int retire_capture_urb(struct snd_usb_substream *subs, snd_printd(KERN_ERR "frame %d active: %d\n", i, urb->iso_frame_desc[i].status); // continue; } - frames = urb->iso_frame_desc[i].actual_length / stride; - bytes = frames * stride; + bytes = urb->iso_frame_desc[i].actual_length; + frames = bytes / stride; + if (!subs->txfr_quirk) + bytes = frames * stride; + if (bytes % (runtime->sample_bits >> 3) != 0) { +#ifdef CONFIG_SND_DEBUG_VERBOSE + int oldbytes = bytes; +#endif + bytes = frames * stride; + snd_printdd(KERN_ERR "Corrected urb data len. %d->%d\n", + oldbytes, bytes); + } /* update the current pointer */ spin_lock_irqsave(&subs->lock, flags); oldptr = subs->hwptr_done; subs->hwptr_done += bytes; if (subs->hwptr_done >= runtime->buffer_size * stride) subs->hwptr_done -= runtime->buffer_size * stride; + frames = (bytes + (oldptr % stride)) / stride; subs->transfer_done += frames; if (subs->transfer_done >= runtime->period_size) { subs->transfer_done -= runtime->period_size; @@ -2238,6 +2250,7 @@ static void init_substream(struct snd_usb_stream *as, int stream, struct audiofo subs->stream = as; subs->direction = stream; subs->dev = as->chip->dev; + subs->txfr_quirk = as->chip->txfr_quirk; if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) { subs->ops = audio_urb_ops[stream]; } else { @@ -3618,6 +3631,20 @@ static void *snd_usb_audio_probe(struct usb_device *dev, } } + switch (chip->usb_id) { + case USB_ID(0x2040, 0x7200): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7221): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7222): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7223): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7224): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7225): /* Hauppage hvr950Q */ + case USB_ID(0x2040, 0x7230): /* Hauppage hvr850 */ + case USB_ID(0x2040, 0x7250): /* Hauppage hvr950Q */ + chip->txfr_quirk = 1; + break; + default: + chip->txfr_quirk = 0; + } err = 1; /* continue */ if (quirk && quirk->ifnum != QUIRK_NO_INTERFACE) { /* need some special handlings */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 1522167..d180554b8 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -125,6 +125,7 @@ struct snd_usb_audio { struct snd_card *card; u32 usb_id; int shutdown; + unsigned int txfr_quirk:1; /* Subframe boundaries on transfers */ int num_interfaces; int num_suspended_intf; -- cgit v1.1 From 52a7a5835173af61b9f6c3038212370d9717526f Mon Sep 17 00:00:00 2001 From: "John S. Gruber" Date: Sun, 27 Dec 2009 12:19:59 -0500 Subject: ALSA: usb-audio: use usbquirk.h for detection of HVR-950Q/850 Detect the HVR-950Q HVR-850 urb data alignment quirk using usbquirk.h rather than using a case statement in snd_usb_audio_probe. Signed-off-by: John S. Gruber Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 30 ++++++------- sound/usb/usbaudio.h | 1 + sound/usb/usbquirks.h | 114 ++++++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 130 insertions(+), 15 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 617515f..4ada98e 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3203,6 +3203,18 @@ static int ignore_interface_quirk(struct snd_usb_audio *chip, return 0; } +/* + * Allow alignment on audio sub-slot (channel samples) rather than + * on audio slots (audio frames) + */ +static int create_align_transfer_quirk(struct snd_usb_audio *chip, + struct usb_interface *iface, + const struct snd_usb_audio_quirk *quirk) +{ + chip->txfr_quirk = 1; + return 1; /* Continue with creating streams and mixer */ +} + /* * boot quirks @@ -3377,7 +3389,8 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, - [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk + [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, + [QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk }; if (quirk->type < QUIRK_TYPE_COUNT) { @@ -3631,20 +3644,7 @@ static void *snd_usb_audio_probe(struct usb_device *dev, } } - switch (chip->usb_id) { - case USB_ID(0x2040, 0x7200): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7221): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7222): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7223): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7224): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7225): /* Hauppage hvr950Q */ - case USB_ID(0x2040, 0x7230): /* Hauppage hvr850 */ - case USB_ID(0x2040, 0x7250): /* Hauppage hvr950Q */ - chip->txfr_quirk = 1; - break; - default: - chip->txfr_quirk = 0; - } + chip->txfr_quirk = 0; err = 1; /* continue */ if (quirk && quirk->ifnum != QUIRK_NO_INTERFACE) { /* need some special handlings */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index d180554b8..9d8cea4 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -161,6 +161,7 @@ enum quirk_type { QUIRK_AUDIO_FIXED_ENDPOINT, QUIRK_AUDIO_EDIROL_UA1000, QUIRK_AUDIO_EDIROL_UAXX, + QUIRK_AUDIO_ALIGN_TRANSFER, QUIRK_TYPE_COUNT }; diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index bd6706c..65bbd22 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -2074,6 +2074,120 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Hauppauge HVR-950Q and HVR-850 */ +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7200), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7201), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7202), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7203), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7204), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7205), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7250), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-950Q", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, +{ + USB_DEVICE_VENDOR_SPEC(0x2040, 0x7230), + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Hauppauge", + .product_name = "HVR-850", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_ALIGN_TRANSFER, + } +}, + { /* * Some USB MIDI devices don't have an audio control interface, -- cgit v1.1 From 4757968dbff3d43f373f08de973014a9bd41ef0a Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Mon, 28 Dec 2009 16:15:03 +0100 Subject: ALSA: Release v1.0.22.1 Signed-off-by: Jaroslav Kysela --- include/sound/version.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/version.h b/include/sound/version.h index 1f5d4872..7fed234 100644 --- a/include/sound/version.h +++ b/include/sound/version.h @@ -1,3 +1,3 @@ /* include/version.h */ -#define CONFIG_SND_VERSION "1.0.22" +#define CONFIG_SND_VERSION "1.0.22.1" #define CONFIG_SND_DATE "" -- cgit v1.1 From 741b20cfb9109760937f403d18d731bfde31f56f Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 17 Dec 2009 17:34:39 +0100 Subject: ALSA: pcm_lib.c - convert second xrun_debug() parameter to use defines To increase code readability, convert send xrun_debug() argument to use defines. Signed-off-by: Jaroslav Kysela --- sound/core/pcm_lib.c | 27 +++++++++++++++++---------- 1 file changed, 17 insertions(+), 10 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 30f4108..9621236 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -126,15 +126,22 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram } } +#define XRUN_DEBUG_BASIC (1<<0) +#define XRUN_DEBUG_STACK (1<<1) /* dump also stack */ +#define XRUN_DEBUG_JIFFIESCHECK (1<<2) /* do jiffies check */ +#define XRUN_DEBUG_PERIODUPDATE (1<<3) /* full period update info */ +#define XRUN_DEBUG_HWPTRUPDATE (1<<4) /* full hwptr update info */ + #ifdef CONFIG_SND_PCM_XRUN_DEBUG -#define xrun_debug(substream, mask) ((substream)->pstr->xrun_debug & (mask)) +#define xrun_debug(substream, mask) \ + ((substream)->pstr->xrun_debug & (mask)) #else #define xrun_debug(substream, mask) 0 #endif -#define dump_stack_on_xrun(substream) do { \ - if (xrun_debug(substream, 2)) \ - dump_stack(); \ +#define dump_stack_on_xrun(substream) do { \ + if (xrun_debug(substream, XRUN_DEBUG_STACK)) \ + dump_stack(); \ } while (0) static void pcm_debug_name(struct snd_pcm_substream *substream, @@ -154,7 +161,7 @@ static void xrun(struct snd_pcm_substream *substream) if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); - if (xrun_debug(substream, 1)) { + if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { char name[16]; pcm_debug_name(substream, name, sizeof(name)); snd_printd(KERN_DEBUG "XRUN: %s\n", name); @@ -215,7 +222,7 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, #define hw_ptr_error(substream, fmt, args...) \ do { \ - if (xrun_debug(substream, 1)) { \ + if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ if (printk_ratelimit()) { \ snd_printd("PCM: " fmt, ##args); \ } \ @@ -237,7 +244,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) xrun(substream); return -EPIPE; } - if (xrun_debug(substream, 8)) { + if (xrun_debug(substream, XRUN_DEBUG_PERIODUPDATE)) { char name[16]; pcm_debug_name(substream, name, sizeof(name)); snd_printd("period_update: %s: pos=0x%x/0x%x/0x%x, " @@ -290,7 +297,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) } /* Do jiffies check only in xrun_debug mode */ - if (!xrun_debug(substream, 4)) + if (!xrun_debug(substream, XRUN_DEBUG_JIFFIESCHECK)) goto no_jiffies_check; /* Skip the jiffies check for hardwares with BATCH flag. @@ -369,7 +376,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) xrun(substream); return -EPIPE; } - if (xrun_debug(substream, 16)) { + if (xrun_debug(substream, XRUN_DEBUG_HWPTRUPDATE)) { char name[16]; pcm_debug_name(substream, name, sizeof(name)); snd_printd("hw_update: %s: pos=0x%x/0x%x/0x%x, " @@ -403,7 +410,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) new_hw_ptr = hw_base + pos; } /* Do jiffies check only in xrun_debug mode */ - if (!xrun_debug(substream, 4)) + if (!xrun_debug(substream, XRUN_DEBUG_JIFFIESCHECK)) goto no_jiffies_check; if (delta < runtime->delay) goto no_jiffies_check; -- cgit v1.1 From 4d96eb255c53ab5e39b37fd4d484ea3dc39ab456 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Sun, 20 Dec 2009 11:47:57 +0100 Subject: ALSA: pcm_lib - add possibility to log last 10 DMA ring buffer positions In some debug cases, it might be usefull to see previous ring buffer positions to determine position problems from the lowlevel drivers. Signed-off-by: Jaroslav Kysela --- include/sound/pcm.h | 6 +++ sound/core/pcm.c | 4 ++ sound/core/pcm_lib.c | 140 ++++++++++++++++++++++++++++++++++++++++++--------- 3 files changed, 127 insertions(+), 23 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index c83a4a7..4e18a6d 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -262,6 +262,8 @@ struct snd_pcm_hw_constraint_list { unsigned int mask; }; +struct snd_pcm_hwptr_log; + struct snd_pcm_runtime { /* -- Status -- */ struct snd_pcm_substream *trigger_master; @@ -340,6 +342,10 @@ struct snd_pcm_runtime { /* -- OSS things -- */ struct snd_pcm_oss_runtime oss; #endif + +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + struct snd_pcm_hwptr_log *hwptr_log; +#endif }; struct snd_pcm_group { /* keep linked substreams */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 6884ae0..df57a0e 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -921,6 +921,10 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) snd_free_pages((void*)runtime->control, PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control))); kfree(runtime->hw_constraints.rules); +#ifdef CONFIG_SND_PCM_XRUN_DEBUG + if (runtime->hwptr_log) + kfree(runtime->hwptr_log); +#endif kfree(runtime); substream->runtime = NULL; put_pid(substream->pid); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 9621236..1990afb 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -126,34 +126,34 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram } } +static void pcm_debug_name(struct snd_pcm_substream *substream, + char *name, size_t len) +{ + snprintf(name, len, "pcmC%dD%d%c:%d", + substream->pcm->card->number, + substream->pcm->device, + substream->stream ? 'c' : 'p', + substream->number); +} + #define XRUN_DEBUG_BASIC (1<<0) #define XRUN_DEBUG_STACK (1<<1) /* dump also stack */ #define XRUN_DEBUG_JIFFIESCHECK (1<<2) /* do jiffies check */ #define XRUN_DEBUG_PERIODUPDATE (1<<3) /* full period update info */ #define XRUN_DEBUG_HWPTRUPDATE (1<<4) /* full hwptr update info */ +#define XRUN_DEBUG_LOG (1<<5) /* show last 10 positions on err */ +#define XRUN_DEBUG_LOGONCE (1<<6) /* do above only once */ #ifdef CONFIG_SND_PCM_XRUN_DEBUG + #define xrun_debug(substream, mask) \ ((substream)->pstr->xrun_debug & (mask)) -#else -#define xrun_debug(substream, mask) 0 -#endif #define dump_stack_on_xrun(substream) do { \ if (xrun_debug(substream, XRUN_DEBUG_STACK)) \ dump_stack(); \ } while (0) -static void pcm_debug_name(struct snd_pcm_substream *substream, - char *name, size_t len) -{ - snprintf(name, len, "pcmC%dD%d%c:%d", - substream->pcm->card->number, - substream->pcm->device, - substream->stream ? 'c' : 'p', - substream->number); -} - static void xrun(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -169,6 +169,102 @@ static void xrun(struct snd_pcm_substream *substream) } } +#define hw_ptr_error(substream, fmt, args...) \ + do { \ + if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ + if (printk_ratelimit()) { \ + snd_printd("PCM: " fmt, ##args); \ + } \ + dump_stack_on_xrun(substream); \ + } \ + } while (0) + +#define XRUN_LOG_CNT 10 + +struct hwptr_log_entry { + unsigned long jiffies; + snd_pcm_uframes_t pos; + snd_pcm_uframes_t period_size; + snd_pcm_uframes_t buffer_size; + snd_pcm_uframes_t old_hw_ptr; + snd_pcm_uframes_t hw_ptr_base; + snd_pcm_uframes_t hw_ptr_interrupt; +}; + +struct snd_pcm_hwptr_log { + unsigned int idx; + unsigned int hit: 1; + struct hwptr_log_entry entries[XRUN_LOG_CNT]; +}; + +static void xrun_log(struct snd_pcm_substream *substream, + snd_pcm_uframes_t pos) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_pcm_hwptr_log *log = runtime->hwptr_log; + struct hwptr_log_entry *entry; + + if (log == NULL) { + log = kzalloc(sizeof(*log), GFP_ATOMIC); + if (log == NULL) + return; + runtime->hwptr_log = log; + } else { + if (xrun_debug(substream, XRUN_DEBUG_LOGONCE) && log->hit) + return; + } + entry = &log->entries[log->idx]; + entry->jiffies = jiffies; + entry->pos = pos; + entry->period_size = runtime->period_size; + entry->buffer_size = runtime->buffer_size;; + entry->old_hw_ptr = runtime->status->hw_ptr; + entry->hw_ptr_base = runtime->hw_ptr_base; + entry->hw_ptr_interrupt = runtime->hw_ptr_interrupt;; + log->idx = (log->idx + 1) % XRUN_LOG_CNT; +} + +static void xrun_log_show(struct snd_pcm_substream *substream) +{ + struct snd_pcm_hwptr_log *log = substream->runtime->hwptr_log; + struct hwptr_log_entry *entry; + char name[16]; + unsigned int idx; + int cnt; + + if (log == NULL) + return; + if (xrun_debug(substream, XRUN_DEBUG_LOGONCE) && log->hit) + return; + pcm_debug_name(substream, name, sizeof(name)); + for (cnt = 0, idx = log->idx; cnt < XRUN_LOG_CNT; cnt++) { + entry = &log->entries[idx]; + if (entry->period_size == 0) + break; + snd_printd("hwptr log: %s: j=%lu, pos=0x%lx/0x%lx/0x%lx, " + "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", + name, entry->jiffies, (unsigned long)entry->pos, + (unsigned long)entry->period_size, + (unsigned long)entry->buffer_size, + (unsigned long)entry->old_hw_ptr, + (unsigned long)entry->hw_ptr_base, + (unsigned long)entry->hw_ptr_interrupt); + idx++; + idx %= XRUN_LOG_CNT; + } + log->hit = 1; +} + +#else /* ! CONFIG_SND_PCM_XRUN_DEBUG */ + +#define xrun_debug(substream, mask) 0 +#define xrun(substream) do { } while (0) +#define hw_ptr_error(substream, fmt, args...) do { } while (0) +#define xrun_log(substream, pos) do { } while (0) +#define xrun_log_show(substream) do { } while (0) + +#endif + static snd_pcm_uframes_t snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime) @@ -182,6 +278,7 @@ snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, if (printk_ratelimit()) { char name[16]; pcm_debug_name(substream, name, sizeof(name)); + xrun_log_show(substream); snd_printd(KERN_ERR "BUG: %s, pos = 0x%lx, " "buffer size = 0x%lx, period size = 0x%lx\n", name, pos, runtime->buffer_size, @@ -190,6 +287,8 @@ snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, pos = 0; } pos -= pos % runtime->min_align; + if (xrun_debug(substream, XRUN_DEBUG_LOG)) + xrun_log(substream, pos); return pos; } @@ -220,16 +319,6 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, return 0; } -#define hw_ptr_error(substream, fmt, args...) \ - do { \ - if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ - if (printk_ratelimit()) { \ - snd_printd("PCM: " fmt, ##args); \ - } \ - dump_stack_on_xrun(substream); \ - } \ - } while (0) - static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -270,6 +359,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) if (runtime->periods == 1 || new_hw_ptr < old_hw_ptr) delta += runtime->buffer_size; if (delta < 0) { + xrun_log_show(substream); hw_ptr_error(substream, "Unexpected hw_pointer value " "(stream=%i, pos=%ld, intr_ptr=%ld)\n", @@ -315,6 +405,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) delta = jdelta / (((runtime->period_size * HZ) / runtime->rate) + HZ/100); + xrun_log_show(substream); hw_ptr_error(substream, "hw_ptr skipping! [Q] " "(pos=%ld, delta=%ld, period=%ld, " @@ -334,6 +425,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) } no_jiffies_check: if (delta > runtime->period_size + runtime->period_size / 2) { + xrun_log_show(substream); hw_ptr_error(substream, "Lost interrupts? " "(stream=%i, delta=%ld, intr_ptr=%ld)\n", @@ -397,6 +489,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) if (delta < 0) { delta += runtime->buffer_size; if (delta < 0) { + xrun_log_show(substream); hw_ptr_error(substream, "Unexpected hw_pointer value [2] " "(stream=%i, pos=%ld, old_ptr=%ld, jdelta=%li)\n", @@ -416,6 +509,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) goto no_jiffies_check; delta -= runtime->delay; if (((delta * HZ) / runtime->rate) > jdelta + HZ/100) { + xrun_log_show(substream); hw_ptr_error(substream, "hw_ptr skipping! " "(pos=%ld, delta=%ld, period=%ld, jdelta=%lu/%lu)\n", -- cgit v1.1 From f240406babfe1526998e10583ea5eccc2676a433 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 5 Jan 2010 17:19:34 +0100 Subject: ALSA: pcm_lib - cleanup & merge hw_ptr update functions Do general cleanup in snd_pcm_update_hw_ptr*() routines and merge them. The main change is hw_ptr_interrupt variable removal to simplify code logic. This variable can be computed directly from hw_ptr. Ensure that updated hw_ptr is not lower than previous one (it was possible with old code in some obscure situations when interrupt was delayed or the lowlevel driver returns wrong ring buffer position value). Signed-off-by: Jaroslav Kysela --- include/sound/pcm.h | 1 - include/sound/pcm_oss.h | 2 +- sound/core/oss/pcm_oss.c | 32 ++++-- sound/core/pcm_lib.c | 279 ++++++++++++++++------------------------------- sound/core/pcm_native.c | 2 - 5 files changed, 121 insertions(+), 195 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 4e18a6d..fe1b131 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -271,7 +271,6 @@ struct snd_pcm_runtime { int overrange; snd_pcm_uframes_t avail_max; snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ - snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time */ unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */ snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ diff --git a/include/sound/pcm_oss.h b/include/sound/pcm_oss.h index cc4e226..760c969 100644 --- a/include/sound/pcm_oss.h +++ b/include/sound/pcm_oss.h @@ -61,7 +61,7 @@ struct snd_pcm_oss_runtime { struct snd_pcm_plugin *plugin_first; struct snd_pcm_plugin *plugin_last; #endif - unsigned int prev_hw_ptr_interrupt; + unsigned int prev_hw_ptr_period; }; struct snd_pcm_oss_file { diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index d9c9635..255ad91 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -632,6 +632,13 @@ static long snd_pcm_alsa_frames(struct snd_pcm_substream *substream, long bytes) return bytes_to_frames(runtime, (buffer_size * bytes) / runtime->oss.buffer_bytes); } +static inline +snd_pcm_uframes_t get_hw_ptr_period(struct snd_pcm_runtime *runtime) +{ + snd_pcm_uframes_t ptr = runtime->status->hw_ptr; + return ptr - (ptr % runtime->period_size); +} + /* define extended formats in the recent OSS versions (if any) */ /* linear formats */ #define AFMT_S32_LE 0x00001000 @@ -1102,7 +1109,7 @@ static int snd_pcm_oss_prepare(struct snd_pcm_substream *substream) return err; } runtime->oss.prepare = 0; - runtime->oss.prev_hw_ptr_interrupt = 0; + runtime->oss.prev_hw_ptr_period = 0; runtime->oss.period_ptr = 0; runtime->oss.buffer_used = 0; @@ -1950,7 +1957,8 @@ static int snd_pcm_oss_get_caps(struct snd_pcm_oss_file *pcm_oss_file) return result; } -static void snd_pcm_oss_simulate_fill(struct snd_pcm_substream *substream, snd_pcm_uframes_t hw_ptr) +static void snd_pcm_oss_simulate_fill(struct snd_pcm_substream *substream, + snd_pcm_uframes_t hw_ptr) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t appl_ptr; @@ -1986,7 +1994,8 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr if (runtime->oss.trigger) goto _skip1; if (atomic_read(&psubstream->mmap_count)) - snd_pcm_oss_simulate_fill(psubstream, runtime->hw_ptr_interrupt); + snd_pcm_oss_simulate_fill(psubstream, + get_hw_ptr_period(runtime)); runtime->oss.trigger = 1; runtime->start_threshold = 1; cmd = SNDRV_PCM_IOCTL_START; @@ -2105,11 +2114,12 @@ static int snd_pcm_oss_get_ptr(struct snd_pcm_oss_file *pcm_oss_file, int stream info.ptr = snd_pcm_oss_bytes(substream, runtime->status->hw_ptr % runtime->buffer_size); if (atomic_read(&substream->mmap_count)) { snd_pcm_sframes_t n; - n = (delay = runtime->hw_ptr_interrupt) - runtime->oss.prev_hw_ptr_interrupt; + delay = get_hw_ptr_period(runtime); + n = delay - runtime->oss.prev_hw_ptr_period; if (n < 0) n += runtime->boundary; info.blocks = n / runtime->period_size; - runtime->oss.prev_hw_ptr_interrupt = delay; + runtime->oss.prev_hw_ptr_period = delay; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) snd_pcm_oss_simulate_fill(substream, delay); info.bytes = snd_pcm_oss_bytes(substream, runtime->status->hw_ptr) & INT_MAX; @@ -2673,18 +2683,22 @@ static int snd_pcm_oss_playback_ready(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; if (atomic_read(&substream->mmap_count)) - return runtime->oss.prev_hw_ptr_interrupt != runtime->hw_ptr_interrupt; + return runtime->oss.prev_hw_ptr_period != + get_hw_ptr_period(runtime); else - return snd_pcm_playback_avail(runtime) >= runtime->oss.period_frames; + return snd_pcm_playback_avail(runtime) >= + runtime->oss.period_frames; } static int snd_pcm_oss_capture_ready(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; if (atomic_read(&substream->mmap_count)) - return runtime->oss.prev_hw_ptr_interrupt != runtime->hw_ptr_interrupt; + return runtime->oss.prev_hw_ptr_period != + get_hw_ptr_period(runtime); else - return snd_pcm_capture_avail(runtime) >= runtime->oss.period_frames; + return snd_pcm_capture_avail(runtime) >= + runtime->oss.period_frames; } static unsigned int snd_pcm_oss_poll(struct file *file, poll_table * wait) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 1990afb..70a4f74 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -172,6 +172,7 @@ static void xrun(struct snd_pcm_substream *substream) #define hw_ptr_error(substream, fmt, args...) \ do { \ if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ + xrun_log_show(substream); \ if (printk_ratelimit()) { \ snd_printd("PCM: " fmt, ##args); \ } \ @@ -188,7 +189,6 @@ struct hwptr_log_entry { snd_pcm_uframes_t buffer_size; snd_pcm_uframes_t old_hw_ptr; snd_pcm_uframes_t hw_ptr_base; - snd_pcm_uframes_t hw_ptr_interrupt; }; struct snd_pcm_hwptr_log { @@ -220,7 +220,6 @@ static void xrun_log(struct snd_pcm_substream *substream, entry->buffer_size = runtime->buffer_size;; entry->old_hw_ptr = runtime->status->hw_ptr; entry->hw_ptr_base = runtime->hw_ptr_base; - entry->hw_ptr_interrupt = runtime->hw_ptr_interrupt;; log->idx = (log->idx + 1) % XRUN_LOG_CNT; } @@ -241,14 +240,13 @@ static void xrun_log_show(struct snd_pcm_substream *substream) entry = &log->entries[idx]; if (entry->period_size == 0) break; - snd_printd("hwptr log: %s: j=%lu, pos=0x%lx/0x%lx/0x%lx, " - "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", + snd_printd("hwptr log: %s: j=%lu, pos=%ld/%ld/%ld, " + "hwptr=%ld/%ld\n", name, entry->jiffies, (unsigned long)entry->pos, (unsigned long)entry->period_size, (unsigned long)entry->buffer_size, (unsigned long)entry->old_hw_ptr, - (unsigned long)entry->hw_ptr_base, - (unsigned long)entry->hw_ptr_interrupt); + (unsigned long)entry->hw_ptr_base); idx++; idx %= XRUN_LOG_CNT; } @@ -265,33 +263,6 @@ static void xrun_log_show(struct snd_pcm_substream *substream) #endif -static snd_pcm_uframes_t -snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, - struct snd_pcm_runtime *runtime) -{ - snd_pcm_uframes_t pos; - - pos = substream->ops->pointer(substream); - if (pos == SNDRV_PCM_POS_XRUN) - return pos; /* XRUN */ - if (pos >= runtime->buffer_size) { - if (printk_ratelimit()) { - char name[16]; - pcm_debug_name(substream, name, sizeof(name)); - xrun_log_show(substream); - snd_printd(KERN_ERR "BUG: %s, pos = 0x%lx, " - "buffer size = 0x%lx, period size = 0x%lx\n", - name, pos, runtime->buffer_size, - runtime->period_size); - } - pos = 0; - } - pos -= pos % runtime->min_align; - if (xrun_debug(substream, XRUN_DEBUG_LOG)) - xrun_log(substream, pos); - return pos; -} - static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime) { @@ -319,72 +290,88 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, return 0; } -static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) +static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, + unsigned int in_interrupt) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t pos; - snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_ptr_interrupt, hw_base; + snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_base; snd_pcm_sframes_t hdelta, delta; unsigned long jdelta; old_hw_ptr = runtime->status->hw_ptr; - pos = snd_pcm_update_hw_ptr_pos(substream, runtime); + pos = substream->ops->pointer(substream); if (pos == SNDRV_PCM_POS_XRUN) { xrun(substream); return -EPIPE; } - if (xrun_debug(substream, XRUN_DEBUG_PERIODUPDATE)) { - char name[16]; - pcm_debug_name(substream, name, sizeof(name)); - snd_printd("period_update: %s: pos=0x%x/0x%x/0x%x, " - "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", - name, (unsigned int)pos, - (unsigned int)runtime->period_size, - (unsigned int)runtime->buffer_size, - (unsigned long)old_hw_ptr, - (unsigned long)runtime->hw_ptr_base, - (unsigned long)runtime->hw_ptr_interrupt); + if (pos >= runtime->buffer_size) { + if (printk_ratelimit()) { + char name[16]; + pcm_debug_name(substream, name, sizeof(name)); + xrun_log_show(substream); + snd_printd(KERN_ERR "BUG: %s, pos = %ld, " + "buffer size = %ld, period size = %ld\n", + name, pos, runtime->buffer_size, + runtime->period_size); + } + pos = 0; } + pos -= pos % runtime->min_align; + if (xrun_debug(substream, XRUN_DEBUG_LOG)) + xrun_log(substream, pos); hw_base = runtime->hw_ptr_base; new_hw_ptr = hw_base + pos; - hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size; - delta = new_hw_ptr - hw_ptr_interrupt; - if (hw_ptr_interrupt >= runtime->boundary) { - hw_ptr_interrupt -= runtime->boundary; - if (hw_base < runtime->boundary / 2) - /* hw_base was already lapped; recalc delta */ - delta = new_hw_ptr - hw_ptr_interrupt; - } - if (delta < 0) { - if (runtime->periods == 1 || new_hw_ptr < old_hw_ptr) - delta += runtime->buffer_size; - if (delta < 0) { - xrun_log_show(substream); - hw_ptr_error(substream, - "Unexpected hw_pointer value " - "(stream=%i, pos=%ld, intr_ptr=%ld)\n", - substream->stream, (long)pos, - (long)hw_ptr_interrupt); -#if 1 - /* simply skipping the hwptr update seems more - * robust in some cases, e.g. on VMware with - * inaccurate timer source - */ - return 0; /* skip this update */ -#else - /* rebase to interrupt position */ - hw_base = new_hw_ptr = hw_ptr_interrupt; - /* align hw_base to buffer_size */ - hw_base -= hw_base % runtime->buffer_size; - delta = 0; -#endif - } else { + if (in_interrupt) { + /* we know that one period was processed */ + /* delta = "expected next hw_ptr" for in_interrupt != 0 */ + delta = old_hw_ptr - (old_hw_ptr % runtime->period_size) + + runtime->period_size; + if (delta > new_hw_ptr) { hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) hw_base = 0; new_hw_ptr = hw_base + pos; + goto __delta; } } + /* new_hw_ptr might be lower than old_hw_ptr in case when */ + /* pointer crosses the end of the ring buffer */ + if (new_hw_ptr < old_hw_ptr) { + hw_base += runtime->buffer_size; + if (hw_base >= runtime->boundary) + hw_base = 0; + new_hw_ptr = hw_base + pos; + } + __delta: + delta = (new_hw_ptr - old_hw_ptr) % runtime->boundary; + if (xrun_debug(substream, in_interrupt ? + XRUN_DEBUG_PERIODUPDATE : XRUN_DEBUG_HWPTRUPDATE)) { + char name[16]; + pcm_debug_name(substream, name, sizeof(name)); + snd_printd("%s_update: %s: pos=%u/%u/%u, " + "hwptr=%ld/%ld/%ld/%ld\n", + in_interrupt ? "period" : "hwptr", + name, + (unsigned int)pos, + (unsigned int)runtime->period_size, + (unsigned int)runtime->buffer_size, + (unsigned long)delta, + (unsigned long)old_hw_ptr, + (unsigned long)new_hw_ptr, + (unsigned long)runtime->hw_ptr_base); + } + /* something must be really wrong */ + if (delta >= runtime->buffer_size) { + hw_ptr_error(substream, + "Unexpected hw_pointer value %s" + "(stream=%i, pos=%ld, new_hw_ptr=%ld, " + "old_hw_ptr=%ld)\n", + in_interrupt ? "[Q] " : "[P]", + substream->stream, (long)pos, + (long)new_hw_ptr, (long)old_hw_ptr); + return 0; + } /* Do jiffies check only in xrun_debug mode */ if (!xrun_debug(substream, XRUN_DEBUG_JIFFIESCHECK)) @@ -396,7 +383,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) */ if (runtime->hw.info & SNDRV_PCM_INFO_BATCH) goto no_jiffies_check; - hdelta = new_hw_ptr - old_hw_ptr; + hdelta = delta; if (hdelta < runtime->delay) goto no_jiffies_check; hdelta -= runtime->delay; @@ -405,45 +392,49 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) delta = jdelta / (((runtime->period_size * HZ) / runtime->rate) + HZ/100); - xrun_log_show(substream); + /* move new_hw_ptr according jiffies not pos variable */ + new_hw_ptr = old_hw_ptr; + /* use loop to avoid checks for delta overflows */ + /* the delta value is small or zero in most cases */ + while (delta > 0) { + new_hw_ptr += runtime->period_size; + if (new_hw_ptr >= runtime->boundary) + new_hw_ptr -= runtime->boundary; + delta--; + } + /* align hw_base to buffer_size */ + hw_base = new_hw_ptr - (new_hw_ptr % runtime->buffer_size); + delta = 0; hw_ptr_error(substream, - "hw_ptr skipping! [Q] " + "hw_ptr skipping! %s" "(pos=%ld, delta=%ld, period=%ld, " - "jdelta=%lu/%lu/%lu)\n", + "jdelta=%lu/%lu/%lu, hw_ptr=%ld/%ld)\n", + in_interrupt ? "[Q] " : "", (long)pos, (long)hdelta, (long)runtime->period_size, jdelta, - ((hdelta * HZ) / runtime->rate), delta); - hw_ptr_interrupt = runtime->hw_ptr_interrupt + - runtime->period_size * delta; - if (hw_ptr_interrupt >= runtime->boundary) - hw_ptr_interrupt -= runtime->boundary; - /* rebase to interrupt position */ - hw_base = new_hw_ptr = hw_ptr_interrupt; - /* align hw_base to buffer_size */ - hw_base -= hw_base % runtime->buffer_size; - delta = 0; + ((hdelta * HZ) / runtime->rate), delta, + (unsigned long)old_hw_ptr, + (unsigned long)new_hw_ptr); } no_jiffies_check: if (delta > runtime->period_size + runtime->period_size / 2) { - xrun_log_show(substream); hw_ptr_error(substream, - "Lost interrupts? " - "(stream=%i, delta=%ld, intr_ptr=%ld)\n", + "Lost interrupts? %s" + "(stream=%i, delta=%ld, new_hw_ptr=%ld, " + "old_hw_ptr=%ld)\n", + in_interrupt ? "[Q] " : "", substream->stream, (long)delta, - (long)hw_ptr_interrupt); - /* rebase hw_ptr_interrupt */ - hw_ptr_interrupt = - new_hw_ptr - new_hw_ptr % runtime->period_size; + (long)new_hw_ptr, + (long)old_hw_ptr); } - runtime->hw_ptr_interrupt = hw_ptr_interrupt; + + if (runtime->status->hw_ptr == new_hw_ptr) + return 0; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); - if (runtime->status->hw_ptr == new_hw_ptr) - return 0; - runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; runtime->hw_ptr_jiffies = jiffies; @@ -456,83 +447,7 @@ static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) /* CAUTION: call it with irq disabled */ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) { - struct snd_pcm_runtime *runtime = substream->runtime; - snd_pcm_uframes_t pos; - snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_base; - snd_pcm_sframes_t delta; - unsigned long jdelta; - - old_hw_ptr = runtime->status->hw_ptr; - pos = snd_pcm_update_hw_ptr_pos(substream, runtime); - if (pos == SNDRV_PCM_POS_XRUN) { - xrun(substream); - return -EPIPE; - } - if (xrun_debug(substream, XRUN_DEBUG_HWPTRUPDATE)) { - char name[16]; - pcm_debug_name(substream, name, sizeof(name)); - snd_printd("hw_update: %s: pos=0x%x/0x%x/0x%x, " - "hwptr=0x%lx, hw_base=0x%lx, hw_intr=0x%lx\n", - name, (unsigned int)pos, - (unsigned int)runtime->period_size, - (unsigned int)runtime->buffer_size, - (unsigned long)old_hw_ptr, - (unsigned long)runtime->hw_ptr_base, - (unsigned long)runtime->hw_ptr_interrupt); - } - - hw_base = runtime->hw_ptr_base; - new_hw_ptr = hw_base + pos; - - delta = new_hw_ptr - old_hw_ptr; - jdelta = jiffies - runtime->hw_ptr_jiffies; - if (delta < 0) { - delta += runtime->buffer_size; - if (delta < 0) { - xrun_log_show(substream); - hw_ptr_error(substream, - "Unexpected hw_pointer value [2] " - "(stream=%i, pos=%ld, old_ptr=%ld, jdelta=%li)\n", - substream->stream, (long)pos, - (long)old_hw_ptr, jdelta); - return 0; - } - hw_base += runtime->buffer_size; - if (hw_base >= runtime->boundary) - hw_base = 0; - new_hw_ptr = hw_base + pos; - } - /* Do jiffies check only in xrun_debug mode */ - if (!xrun_debug(substream, XRUN_DEBUG_JIFFIESCHECK)) - goto no_jiffies_check; - if (delta < runtime->delay) - goto no_jiffies_check; - delta -= runtime->delay; - if (((delta * HZ) / runtime->rate) > jdelta + HZ/100) { - xrun_log_show(substream); - hw_ptr_error(substream, - "hw_ptr skipping! " - "(pos=%ld, delta=%ld, period=%ld, jdelta=%lu/%lu)\n", - (long)pos, (long)delta, - (long)runtime->period_size, jdelta, - ((delta * HZ) / runtime->rate)); - return 0; - } - no_jiffies_check: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - runtime->silence_size > 0) - snd_pcm_playback_silence(substream, new_hw_ptr); - - if (runtime->status->hw_ptr == new_hw_ptr) - return 0; - - runtime->hw_ptr_base = hw_base; - runtime->status->hw_ptr = new_hw_ptr; - runtime->hw_ptr_jiffies = jiffies; - if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) - snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); - - return snd_pcm_update_hw_ptr_post(substream, runtime); + return snd_pcm_update_hw_ptr0(substream, 0); } /** @@ -1744,7 +1659,7 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream) snd_pcm_stream_lock_irqsave(substream, flags); if (!snd_pcm_running(substream) || - snd_pcm_update_hw_ptr_interrupt(substream) < 0) + snd_pcm_update_hw_ptr0(substream, 1) < 0) goto _end; if (substream->timer_running) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 29ab46a1..8e777f7 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1247,8 +1247,6 @@ static int snd_pcm_do_reset(struct snd_pcm_substream *substream, int state) if (err < 0) return err; runtime->hw_ptr_base = 0; - runtime->hw_ptr_interrupt = runtime->status->hw_ptr - - runtime->status->hw_ptr % runtime->period_size; runtime->silence_start = runtime->status->hw_ptr; runtime->silence_filled = 0; return 0; -- cgit v1.1 From 1250932e48d3b698415b1f04775433cf1da688d6 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 7 Jan 2010 15:36:31 +0100 Subject: ALSA: pcm_lib - optimize wake_up() calls for PCM I/O As noted by pl bossart , the PCM I/O routines (snd_pcm_lib_write1, snd_pcm_lib_read1) should block wake_up() calls until all samples are not processed. Signed-off-by: Jaroslav Kysela --- include/sound/pcm.h | 3 +++ sound/core/pcm_lib.c | 30 ++++++++++++++++++++---------- sound/core/pcm_native.c | 6 ++++-- 3 files changed, 27 insertions(+), 12 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index fe1b131..e26fb3c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -311,6 +311,7 @@ struct snd_pcm_runtime { struct snd_pcm_mmap_control *control; /* -- locking / scheduling -- */ + unsigned int nowake: 1; /* no wakeup (data-copy in progress) */ wait_queue_head_t sleep; struct fasync_struct *fasync; @@ -839,6 +840,8 @@ void snd_pcm_set_sync(struct snd_pcm_substream *substream); int snd_pcm_lib_interleave_len(struct snd_pcm_substream *substream); int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream, unsigned int cmd, void *arg); +int snd_pcm_update_state(struct snd_pcm_substream *substream, + struct snd_pcm_runtime *runtime); int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream); int snd_pcm_playback_xrun_check(struct snd_pcm_substream *substream); int snd_pcm_capture_xrun_check(struct snd_pcm_substream *substream); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 70a4f74..a632262 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -263,8 +263,8 @@ static void xrun_log_show(struct snd_pcm_substream *substream) #endif -static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, - struct snd_pcm_runtime *runtime) +int snd_pcm_update_state(struct snd_pcm_substream *substream, + struct snd_pcm_runtime *runtime) { snd_pcm_uframes_t avail; @@ -285,7 +285,7 @@ static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, return -EPIPE; } } - if (avail >= runtime->control->avail_min) + if (!runtime->nowake && avail >= runtime->control->avail_min) wake_up(&runtime->sleep); return 0; } @@ -441,7 +441,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE) snd_pcm_gettime(runtime, (struct timespec *)&runtime->status->tstamp); - return snd_pcm_update_hw_ptr_post(substream, runtime); + return snd_pcm_update_state(substream, runtime); } /* CAUTION: call it with irq disabled */ @@ -1792,6 +1792,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, goto _end_unlock; } + runtime->nowake = 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -1813,15 +1814,17 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, if (frames > cont) frames = cont; if (snd_BUG_ON(!frames)) { + runtime->nowake = 0; snd_pcm_stream_unlock_irq(substream); return -EINVAL; } appl_ptr = runtime->control->appl_ptr; appl_ofs = appl_ptr % runtime->buffer_size; snd_pcm_stream_unlock_irq(substream); - if ((err = transfer(substream, appl_ofs, data, offset, frames)) < 0) - goto _end; + err = transfer(substream, appl_ofs, data, offset, frames); snd_pcm_stream_lock_irq(substream); + if (err < 0) + goto _end_unlock; switch (runtime->status->state) { case SNDRV_PCM_STATE_XRUN: err = -EPIPE; @@ -1850,8 +1853,10 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, } } _end_unlock: + runtime->nowake = 0; + if (xfer > 0 && err >= 0) + snd_pcm_update_state(substream, runtime); snd_pcm_stream_unlock_irq(substream); - _end: return xfer > 0 ? (snd_pcm_sframes_t)xfer : err; } @@ -2009,6 +2014,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, goto _end_unlock; } + runtime->nowake = 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -2037,15 +2043,17 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, if (frames > cont) frames = cont; if (snd_BUG_ON(!frames)) { + runtime->nowake = 0; snd_pcm_stream_unlock_irq(substream); return -EINVAL; } appl_ptr = runtime->control->appl_ptr; appl_ofs = appl_ptr % runtime->buffer_size; snd_pcm_stream_unlock_irq(substream); - if ((err = transfer(substream, appl_ofs, data, offset, frames)) < 0) - goto _end; + err = transfer(substream, appl_ofs, data, offset, frames); snd_pcm_stream_lock_irq(substream); + if (err < 0) + goto _end_unlock; switch (runtime->status->state) { case SNDRV_PCM_STATE_XRUN: err = -EPIPE; @@ -2068,8 +2076,10 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, xfer += frames; } _end_unlock: + runtime->nowake = 0; + if (xfer > 0 && err >= 0) + snd_pcm_update_state(substream, runtime); snd_pcm_stream_unlock_irq(substream); - _end: return xfer > 0 ? (snd_pcm_sframes_t)xfer : err; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 8e777f7..27284f6 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -516,6 +516,7 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, struct snd_pcm_sw_params *params) { struct snd_pcm_runtime *runtime; + int err; if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; @@ -540,6 +541,7 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, if (params->silence_threshold > runtime->buffer_size) return -EINVAL; } + err = 0; snd_pcm_stream_lock_irq(substream); runtime->tstamp_mode = params->tstamp_mode; runtime->period_step = params->period_step; @@ -553,10 +555,10 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, ULONG_MAX); - wake_up(&runtime->sleep); + err = snd_pcm_update_state(substream, runtime); } snd_pcm_stream_unlock_irq(substream); - return 0; + return err; } static int snd_pcm_sw_params_user(struct snd_pcm_substream *substream, -- cgit v1.1 From 7b3a177b0d4f92b3431b8dca777313a07533a710 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Fri, 8 Jan 2010 08:43:01 +0100 Subject: ALSA: pcm_lib: fix "something must be really wrong" condition When runtime->periods == 1 or when pointer crosses end of ring buffer, the delta might be greater than buffer_size. Signed-off-by: Jaroslav Kysela --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a632262..c7b35b2 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -362,7 +362,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, (unsigned long)runtime->hw_ptr_base); } /* something must be really wrong */ - if (delta >= runtime->buffer_size) { + if (delta >= runtime->buffer_size + runtime->period_size) { hw_ptr_error(substream, "Unexpected hw_pointer value %s" "(stream=%i, pos=%ld, new_hw_ptr=%ld, " -- cgit v1.1 From dd3533eca859a6debb1565503ec03e68354e08e0 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Fri, 1 Jan 2010 19:05:43 +0100 Subject: ALSA: ac97_codec: merge WM9703 and WM9705 ops The WM9705 and WM9703 ops are the same actually so use the same code for both. Signed-off-by: Krzysztof Helt Signed-off-by: Jaroslav Kysela --- sound/pci/ac97/ac97_patch.c | 17 +---------------- 1 file changed, 1 insertion(+), 16 deletions(-) diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 139cf3b..e288a55 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -544,25 +544,10 @@ static int patch_wolfson04(struct snd_ac97 * ac97) return 0; } -static int patch_wolfson_wm9705_specific(struct snd_ac97 * ac97) -{ - int err, i; - for (i = 0; i < ARRAY_SIZE(wm97xx_snd_ac97_controls); i++) { - if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&wm97xx_snd_ac97_controls[i], ac97))) < 0) - return err; - } - snd_ac97_write_cache(ac97, 0x72, 0x0808); - return 0; -} - -static struct snd_ac97_build_ops patch_wolfson_wm9705_ops = { - .build_specific = patch_wolfson_wm9705_specific, -}; - static int patch_wolfson05(struct snd_ac97 * ac97) { /* WM9705, WM9710 */ - ac97->build_ops = &patch_wolfson_wm9705_ops; + ac97->build_ops = &patch_wolfson_wm9703_ops; #ifdef CONFIG_TOUCHSCREEN_WM9705 /* WM9705 touchscreen uses AUX and VIDEO for touch */ ac97->flags |= AC97_HAS_NO_VIDEO | AC97_HAS_NO_AUX; -- cgit v1.1 From cd9d95a55550555da8e587ead9cbba5f98a371a3 Mon Sep 17 00:00:00 2001 From: Ken Prox Date: Fri, 8 Jan 2010 09:01:47 +0100 Subject: ALSA: hda - conexant - Fixed microphone mixer for HP Compaq Presario F700 Added patch for Hewlett-Packard Company Device Subsystem id - 103c:30ea. Signed-off-by: Ken Prox Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 50 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 50 insertions(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1ab2958..b20c640 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1720,6 +1720,22 @@ static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { {} }; +static struct snd_kcontrol_new cxt5051_f700_mixers[] = { + HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Switch", 0x14, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5051_hp_master_sw_put, + .private_value = 0x1a, + }, + + {} +}; + static struct hda_verb cxt5051_init_verbs[] = { /* Line in, Mic */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, @@ -1810,6 +1826,32 @@ static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { { } /* end */ }; +static struct hda_verb cxt5051_f700_init_verbs[] = { + /* Line in, Mic */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x03}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, + /* SPK */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP, Amp */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Record selector: Int mic */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* SPDIF route: PCM */ + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* EAPD */ + {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5051_init(struct hda_codec *codec) { @@ -1829,6 +1871,7 @@ enum { CXT5051_HP, /* no docking */ CXT5051_HP_DV6736, /* HP without mic switch */ CXT5051_LENOVO_X200, /* Lenovo X200 laptop */ + CXT5051_F700, /* HP Compaq Presario F700 */ CXT5051_MODELS }; @@ -1837,6 +1880,7 @@ static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_HP] = "hp", [CXT5051_HP_DV6736] = "hp-dv6736", [CXT5051_LENOVO_X200] = "lenovo-x200", + [CXT5051_F700] = "hp 700" }; static struct snd_pci_quirk cxt5051_cfg_tbl[] = { @@ -1846,6 +1890,7 @@ static struct snd_pci_quirk cxt5051_cfg_tbl[] = { CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), + SND_PCI_QUIRK(0x103c, 0x30ea, "Compaq Presario F700", CXT5051_F700), {} }; @@ -1896,6 +1941,11 @@ static int patch_cxt5051(struct hda_codec *codec) case CXT5051_LENOVO_X200: spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs; break; + case CXT5051_F700: + spec->init_verbs[0] = cxt5051_f700_init_verbs; + spec->mixers[0] = cxt5051_f700_mixers; + spec->no_auto_mic = 1; + break; } return 0; -- cgit v1.1 From 75f8991d0e6969407d51501d5a0537f104075c99 Mon Sep 17 00:00:00 2001 From: Daniel Drake Date: Thu, 7 Jan 2010 13:46:25 +0100 Subject: ALSA: hda - Configure XO-1.5 microphones at capture time The XO-1.5 has a microphone LED designed to indicate to the user when something is being recorded. This light is controlled by the microphone bias voltage and it is currently coming on all the time. This patch defers the microphone port configuration until when recording is actually taking place, fixing the behaviour of the LED. Signed-off-by: Daniel Drake Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 125 ++++++++++++++++++++++++++++------------- 1 file changed, 85 insertions(+), 40 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 01e46ba..3521f33 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -111,8 +111,12 @@ struct conexant_spec { unsigned int dell_automute; unsigned int port_d_mode; - unsigned char ext_mic_bias; unsigned int dell_vostro; + + unsigned int ext_mic_present; + unsigned int recording; + void (*capture_prepare)(struct hda_codec *codec); + void (*capture_cleanup)(struct hda_codec *codec); }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -185,6 +189,8 @@ static int conexant_capture_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct conexant_spec *spec = codec->spec; + if (spec->capture_prepare) + spec->capture_prepare(codec); snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], stream_tag, 0, format); return 0; @@ -196,6 +202,8 @@ static int conexant_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, { struct conexant_spec *spec = codec->spec; snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); + if (spec->capture_cleanup) + spec->capture_cleanup(codec); return 0; } @@ -2016,53 +2024,53 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol, return 1; } -/* toggle input of built-in and mic jack appropriately */ -static void cxt5066_automic(struct hda_codec *codec) +/* OLPC defers mic widget control until when capture is started because the + * microphone LED comes on as soon as these settings are put in place. if we + * did this before recording, it would give the false indication that recording + * is happening when it is not. */ +static void cxt5066_olpc_select_mic(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; - struct hda_verb ext_mic_present[] = { - /* enable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias}, - - /* switch to external mic input */ - {0x17, AC_VERB_SET_CONNECT_SEL, 0}, + if (!spec->recording) + return; - /* disable internal mic, port C */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; - static struct hda_verb ext_mic_absent[] = { - /* enable internal mic, port C */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + /* external mic, port B */ + snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + spec->ext_mic_present ? CXT5066_OLPC_EXT_MIC_BIAS : 0); - /* switch to internal mic input */ - {0x17, AC_VERB_SET_CONNECT_SEL, 1}, + /* internal mic, port C */ + snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + spec->ext_mic_present ? 0 : PIN_VREF80); +} - /* disable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, - {} - }; +/* toggle input of built-in and mic jack appropriately */ +static void cxt5066_olpc_automic(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; unsigned int present; - present = snd_hda_jack_detect(codec, 0x1a); - if (present) { + present = snd_hda_codec_read(codec, 0x1a, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + if (present) snd_printdd("CXT5066: external microphone detected\n"); - snd_hda_sequence_write(codec, ext_mic_present); - } else { + else snd_printdd("CXT5066: external microphone absent\n"); - snd_hda_sequence_write(codec, ext_mic_absent); - } + + snd_hda_codec_write(codec, 0x17, 0, AC_VERB_SET_CONNECT_SEL, + present ? 0 : 1); + spec->ext_mic_present = !!present; + + cxt5066_olpc_select_mic(codec); } /* toggle input of built-in digital mic and mic jack appropriately */ static void cxt5066_vostro_automic(struct hda_codec *codec) { - struct conexant_spec *spec = codec->spec; unsigned int present; struct hda_verb ext_mic_present[] = { /* enable external mic, port B */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_bias}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* switch to external mic input */ {0x17, AC_VERB_SET_CONNECT_SEL, 0}, @@ -2113,7 +2121,7 @@ static void cxt5066_hp_automute(struct hda_codec *codec) } /* unsolicited event for jack sensing */ -static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) +static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res) { snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); switch (res >> 26) { @@ -2121,7 +2129,7 @@ static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res) cxt5066_hp_automute(codec); break; case CONEXANT_MIC_EVENT: - cxt5066_automic(codec); + cxt5066_olpc_automic(codec); break; } } @@ -2197,6 +2205,31 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol, return 1; } +static void cxt5066_olpc_capture_prepare(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + /* mark as recording and configure the microphone widget so that the + * recording LED comes on. */ + spec->recording = 1; + cxt5066_olpc_select_mic(codec); +} + +static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + const struct hda_verb disable_mics[] = { + /* disable external mic, port B */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* disble internal mic, port C */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {}, + }; + + snd_hda_sequence_write(codec, disable_mics); + spec->recording = 0; +} + static struct hda_input_mux cxt5066_capture_source = { .num_items = 4, .items = { @@ -2347,10 +2380,10 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = { {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */ /* Port B: external microphone */ - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, CXT5066_OLPC_EXT_MIC_BIAS}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, /* Port C: internal microphone */ - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, /* Port D: unused */ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, @@ -2479,12 +2512,19 @@ static int cxt5066_init(struct hda_codec *codec) cxt5066_hp_automute(codec); if (spec->dell_vostro) cxt5066_vostro_automic(codec); - else - cxt5066_automic(codec); } return 0; } +static int cxt5066_olpc_init(struct hda_codec *codec) +{ + snd_printdd("CXT5066: init\n"); + conexant_init(codec); + cxt5066_hp_automute(codec); + cxt5066_olpc_automic(codec); + return 0; +} + enum { CXT5066_LAPTOP, /* Laptops w/ EAPD support */ CXT5066_DELL_LAPTOP, /* Dell Laptop */ @@ -2521,7 +2561,7 @@ static int patch_cxt5066(struct hda_codec *codec) codec->spec = spec; codec->patch_ops = conexant_patch_ops; - codec->patch_ops.init = cxt5066_init; + codec->patch_ops.init = conexant_init; spec->dell_automute = 0; spec->multiout.max_channels = 2; @@ -2534,7 +2574,6 @@ static int patch_cxt5066(struct hda_codec *codec) spec->input_mux = &cxt5066_capture_source; spec->port_d_mode = PIN_HP; - spec->ext_mic_bias = PIN_VREF80; spec->num_init_verbs = 1; spec->init_verbs[0] = cxt5066_init_verbs; @@ -2561,20 +2600,26 @@ static int patch_cxt5066(struct hda_codec *codec) spec->dell_automute = 1; break; case CXT5066_OLPC_XO_1_5: - codec->patch_ops.unsol_event = cxt5066_unsol_event; + codec->patch_ops.init = cxt5066_olpc_init; + codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; spec->init_verbs[0] = cxt5066_init_verbs_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->port_d_mode = 0; - spec->ext_mic_bias = CXT5066_OLPC_EXT_MIC_BIAS; /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; /* input source automatically selected */ spec->input_mux = NULL; + + /* our capture hooks which allow us to turn on the microphone LED + * at the right time */ + spec->capture_prepare = cxt5066_olpc_capture_prepare; + spec->capture_cleanup = cxt5066_olpc_capture_cleanup; break; case CXT5066_DELL_VOSTO: + codec->patch_ops.init = cxt5066_init; codec->patch_ops.unsol_event = cxt5066_vostro_event; spec->init_verbs[0] = cxt5066_init_verbs_vostro; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; -- cgit v1.1 From c4cfe66c4c2d5a91b3734ffb4e2bad0badd5c874 Mon Sep 17 00:00:00 2001 From: Daniel Drake Date: Thu, 7 Jan 2010 13:47:04 +0100 Subject: ALSA: hda - support OLPC XO-1.5 DC input The XO's audio hardware is wired up to allow DC sensors (e.g. light sensors, thermistors, etc) to be plugged in through the microphone jack. Add sound mixer controls to allow this mode to be enabled and tweaked. Signed-off-by: Daniel Drake Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_conexant.c | 213 ++++++++++++++++++++++++++++++++++++----- 1 file changed, 190 insertions(+), 23 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 3521f33..685015a 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -117,6 +117,16 @@ struct conexant_spec { unsigned int recording; void (*capture_prepare)(struct hda_codec *codec); void (*capture_cleanup)(struct hda_codec *codec); + + /* OLPC XO-1.5 supports DC input mode (e.g. for use with analog sensors) + * through the microphone jack. + * When the user enables this through a mixer switch, both internal and + * external microphones are disabled. Gain is fixed at 0dB. In this mode, + * we also allow the bias to be configured through a separate mixer + * control. */ + unsigned int dc_enable; + unsigned int dc_input_bias; /* offset into cxt5066_olpc_dc_bias */ + unsigned int mic_boost; /* offset into cxt5066_analog_mic_boost */ }; static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo, @@ -2024,6 +2034,26 @@ static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol, return 1; } +static const struct hda_input_mux cxt5066_olpc_dc_bias = { + .num_items = 3, + .items = { + { "Off", PIN_IN }, + { "50%", PIN_VREF50 }, + { "80%", PIN_VREF80 }, + }, +}; + +static int cxt5066_set_olpc_dc_bias(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + /* Even though port F is the DC input, the bias is controlled on port B. + * we also leave that port as an active input (but unselected) in DC mode + * just in case that is necessary to make the bias setting take effect. */ + return snd_hda_codec_write_cache(codec, 0x1a, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + cxt5066_olpc_dc_bias.items[spec->dc_input_bias].index); +} + /* OLPC defers mic widget control until when capture is started because the * microphone LED comes on as soon as these settings are put in place. if we * did this before recording, it would give the false indication that recording @@ -2034,6 +2064,27 @@ static void cxt5066_olpc_select_mic(struct hda_codec *codec) if (!spec->recording) return; + if (spec->dc_enable) { + /* in DC mode we ignore presence detection and just use the jack + * through our special DC port */ + const struct hda_verb enable_dc_mode[] = { + /* disble internal mic, port C */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* enable DC capture, port F */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {}, + }; + + snd_hda_sequence_write(codec, enable_dc_mode); + /* port B input disabled (and bias set) through the following call */ + cxt5066_set_olpc_dc_bias(codec); + return; + } + + /* disable DC (port F) */ + snd_hda_codec_write(codec, 0x1e, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + /* external mic, port B */ snd_hda_codec_write(codec, 0x1a, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, spec->ext_mic_present ? CXT5066_OLPC_EXT_MIC_BIAS : 0); @@ -2049,6 +2100,9 @@ static void cxt5066_olpc_automic(struct hda_codec *codec) struct conexant_spec *spec = codec->spec; unsigned int present; + if (spec->dc_enable) /* don't do presence detection in DC mode */ + return; + present = snd_hda_codec_read(codec, 0x1a, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; if (present) @@ -2123,13 +2177,16 @@ static void cxt5066_hp_automute(struct hda_codec *codec) /* unsolicited event for jack sensing */ static void cxt5066_olpc_unsol_event(struct hda_codec *codec, unsigned int res) { + struct conexant_spec *spec = codec->spec; snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26); switch (res >> 26) { case CONEXANT_HP_EVENT: cxt5066_hp_automute(codec); break; case CONEXANT_MIC_EVENT: - cxt5066_olpc_automic(codec); + /* ignore mic events in DC mode; we're always using the jack */ + if (!spec->dc_enable) + cxt5066_olpc_automic(codec); break; } } @@ -2159,6 +2216,15 @@ static const struct hda_input_mux cxt5066_analog_mic_boost = { }, }; +static int cxt5066_set_mic_boost(struct hda_codec *codec) +{ + struct conexant_spec *spec = codec->spec; + return snd_hda_codec_write_cache(codec, 0x17, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT | + cxt5066_analog_mic_boost.items[spec->mic_boost].index); +} + static int cxt5066_mic_boost_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -2169,15 +2235,8 @@ static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - int val; - hda_nid_t nid = kcontrol->private_value & 0xff; - int inout = (kcontrol->private_value & 0x100) ? - AC_AMP_GET_INPUT : AC_AMP_GET_OUTPUT; - - val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_AMP_GAIN_MUTE, inout); - - ucontrol->value.enumerated.item[0] = val & AC_AMP_GAIN; + struct conexant_spec *spec = codec->spec; + ucontrol->value.enumerated.item[0] = spec->mic_boost; return 0; } @@ -2185,23 +2244,101 @@ static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; const struct hda_input_mux *imux = &cxt5066_analog_mic_boost; unsigned int idx; - hda_nid_t nid = kcontrol->private_value & 0xff; - int inout = (kcontrol->private_value & 0x100) ? - AC_AMP_SET_INPUT : AC_AMP_SET_OUTPUT; + idx = ucontrol->value.enumerated.item[0]; + if (idx >= imux->num_items) + idx = imux->num_items - 1; + + spec->mic_boost = idx; + if (!spec->dc_enable) + cxt5066_set_mic_boost(codec); + return 1; +} + +static void cxt5066_enable_dc(struct hda_codec *codec) +{ + const struct hda_verb enable_dc_mode[] = { + /* disable gain */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* switch to DC input */ + {0x17, AC_VERB_SET_CONNECT_SEL, 3}, + {} + }; + + /* configure as input source */ + snd_hda_sequence_write(codec, enable_dc_mode); + cxt5066_olpc_select_mic(codec); /* also sets configured bias */ +} + +static void cxt5066_disable_dc(struct hda_codec *codec) +{ + /* reconfigure input source */ + cxt5066_set_mic_boost(codec); + /* automic also selects the right mic if we're recording */ + cxt5066_olpc_automic(codec); +} + +static int cxt5066_olpc_dc_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + ucontrol->value.integer.value[0] = spec->dc_enable; + return 0; +} - if (!imux->num_items) +static int cxt5066_olpc_dc_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + int dc_enable = !!ucontrol->value.integer.value[0]; + + if (dc_enable == spec->dc_enable) return 0; + + spec->dc_enable = dc_enable; + if (dc_enable) + cxt5066_enable_dc(codec); + else + cxt5066_disable_dc(codec); + + return 1; +} + +static int cxt5066_olpc_dc_bias_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + return snd_hda_input_mux_info(&cxt5066_olpc_dc_bias, uinfo); +} + +static int cxt5066_olpc_dc_bias_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + ucontrol->value.enumerated.item[0] = spec->dc_input_bias; + return 0; +} + +static int cxt5066_olpc_dc_bias_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct conexant_spec *spec = codec->spec; + const struct hda_input_mux *imux = &cxt5066_analog_mic_boost; + unsigned int idx; + idx = ucontrol->value.enumerated.item[0]; if (idx >= imux->num_items) idx = imux->num_items - 1; - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | inout | - imux->items[idx].index); - + spec->dc_input_bias = idx; + if (spec->dc_enable) + cxt5066_set_olpc_dc_bias(codec); return 1; } @@ -2223,6 +2360,9 @@ static void cxt5066_olpc_capture_cleanup(struct hda_codec *codec) /* disble internal mic, port C */ {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* disable DC capture, port F */ + {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, {}, }; @@ -2282,6 +2422,24 @@ static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = { {} }; +static struct snd_kcontrol_new cxt5066_mixer_olpc_dc[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DC Mode Enable Switch", + .info = snd_ctl_boolean_mono_info, + .get = cxt5066_olpc_dc_get, + .put = cxt5066_olpc_dc_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DC Input Bias Enum", + .info = cxt5066_olpc_dc_bias_enum_info, + .get = cxt5066_olpc_dc_bias_enum_get, + .put = cxt5066_olpc_dc_bias_enum_put, + }, + {} +}; + static struct snd_kcontrol_new cxt5066_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -2294,11 +2452,10 @@ static struct snd_kcontrol_new cxt5066_mixers[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Ext Mic Boost Capture Enum", + .name = "Analog Mic Boost Capture Enum", .info = cxt5066_mic_boost_mux_enum_info, .get = cxt5066_mic_boost_mux_enum_get, .put = cxt5066_mic_boost_mux_enum_put, - .private_value = 0x17, }, HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others), @@ -2392,7 +2549,7 @@ static struct hda_verb cxt5066_init_verbs_olpc[] = { {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, {0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ - /* Port F: unused */ + /* Port F: external DC input through microphone port */ {0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, /* Port G: internal speakers */ @@ -2513,15 +2670,22 @@ static int cxt5066_init(struct hda_codec *codec) if (spec->dell_vostro) cxt5066_vostro_automic(codec); } + cxt5066_set_mic_boost(codec); return 0; } static int cxt5066_olpc_init(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; snd_printdd("CXT5066: init\n"); conexant_init(codec); cxt5066_hp_automute(codec); - cxt5066_olpc_automic(codec); + if (!spec->dc_enable) { + cxt5066_set_mic_boost(codec); + cxt5066_olpc_automic(codec); + } else { + cxt5066_enable_dc(codec); + } return 0; } @@ -2604,8 +2768,10 @@ static int patch_cxt5066(struct hda_codec *codec) codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; spec->init_verbs[0] = cxt5066_init_verbs_olpc; spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_olpc_dc; spec->mixers[spec->num_mixers++] = cxt5066_mixers; spec->port_d_mode = 0; + spec->mic_boost = 3; /* default 30dB gain */ /* no S/PDIF out */ spec->multiout.dig_out_nid = 0; @@ -2627,6 +2793,7 @@ static int patch_cxt5066(struct hda_codec *codec) spec->mixers[spec->num_mixers++] = cxt5066_vostro_mixers; spec->port_d_mode = 0; spec->dell_vostro = 1; + spec->mic_boost = 3; /* default 30dB gain */ snd_hda_attach_beep_device(codec, 0x13); /* no S/PDIF out */ -- cgit v1.1 From 6b98515a620592636d2f8e0d3e2942d1cb4847ec Mon Sep 17 00:00:00 2001 From: Alan Cox Date: Mon, 4 Jan 2010 16:22:59 +0000 Subject: sound_oss: remove use of old BKL ioctl path Signed-off-by: Alan Cox Signed-off-by: Takashi Iwai --- sound/oss/soundcard.c | 35 ++++++++++++++++++++++------------- 1 file changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index 61aaeda..6c3267b 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -328,11 +328,11 @@ static int sound_mixer_ioctl(int mixdev, unsigned int cmd, void __user *arg) return mixer_devs[mixdev]->ioctl(mixdev, cmd, arg); } -static int sound_ioctl(struct inode *inode, struct file *file, - unsigned int cmd, unsigned long arg) +static long sound_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { int len = 0, dtype; - int dev = iminor(inode); + int dev = iminor(file->f_dentry->d_inode); + long ret = -EINVAL; void __user *p = (void __user *)arg; if (_SIOC_DIR(cmd) != _SIOC_NONE && _SIOC_DIR(cmd) != 0) { @@ -353,6 +353,7 @@ static int sound_ioctl(struct inode *inode, struct file *file, if (cmd == OSS_GETVERSION) return __put_user(SOUND_VERSION, (int __user *)p); + lock_kernel(); if (_IOC_TYPE(cmd) == 'M' && num_mixers > 0 && /* Mixer ioctl */ (dev & 0x0f) != SND_DEV_CTL) { dtype = dev & 0x0f; @@ -360,24 +361,31 @@ static int sound_ioctl(struct inode *inode, struct file *file, case SND_DEV_DSP: case SND_DEV_DSP16: case SND_DEV_AUDIO: - return sound_mixer_ioctl(audio_devs[dev >> 4]->mixer_dev, + ret = sound_mixer_ioctl(audio_devs[dev >> 4]->mixer_dev, cmd, p); - + break; default: - return sound_mixer_ioctl(dev >> 4, cmd, p); + ret = sound_mixer_ioctl(dev >> 4, cmd, p); + break; } + unlock_kernel(); + return ret; } + switch (dev & 0x0f) { case SND_DEV_CTL: if (cmd == SOUND_MIXER_GETLEVELS) - return get_mixer_levels(p); - if (cmd == SOUND_MIXER_SETLEVELS) - return set_mixer_levels(p); - return sound_mixer_ioctl(dev >> 4, cmd, p); + ret = get_mixer_levels(p); + else if (cmd == SOUND_MIXER_SETLEVELS) + ret = set_mixer_levels(p); + else + ret = sound_mixer_ioctl(dev >> 4, cmd, p); + break; case SND_DEV_SEQ: case SND_DEV_SEQ2: - return sequencer_ioctl(dev, file, cmd, p); + ret = sequencer_ioctl(dev, file, cmd, p); + break; case SND_DEV_DSP: case SND_DEV_DSP16: @@ -390,7 +398,8 @@ static int sound_ioctl(struct inode *inode, struct file *file, break; } - return -EINVAL; + unlock_kernel(); + return ret; } static unsigned int sound_poll(struct file *file, poll_table * wait) @@ -490,7 +499,7 @@ const struct file_operations oss_sound_fops = { .read = sound_read, .write = sound_write, .poll = sound_poll, - .ioctl = sound_ioctl, + .unlocked_ioctl = sound_ioctl, .mmap = sound_mmap, .open = sound_open, .release = sound_release, -- cgit v1.1 From ed69c6a8eef679f2783848ed624897a937a434ac Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 13 Jan 2010 08:12:31 +0100 Subject: ALSA: pcm_lib - fix wrong delta print for jiffies check The previous jiffies delta was 0 in all cases. Use hw_ptr variable to store and print original value. Signed-off-by: Jaroslav Kysela --- sound/core/pcm_lib.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 0ee7e80..5417f7d 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -394,6 +394,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, + HZ/100); /* move new_hw_ptr according jiffies not pos variable */ new_hw_ptr = old_hw_ptr; + hw_base = delta; /* use loop to avoid checks for delta overflows */ /* the delta value is small or zero in most cases */ while (delta > 0) { @@ -403,8 +404,6 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, delta--; } /* align hw_base to buffer_size */ - hw_base = new_hw_ptr - (new_hw_ptr % runtime->buffer_size); - delta = 0; hw_ptr_error(substream, "hw_ptr skipping! %s" "(pos=%ld, delta=%ld, period=%ld, " @@ -412,9 +411,12 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, in_interrupt ? "[Q] " : "", (long)pos, (long)hdelta, (long)runtime->period_size, jdelta, - ((hdelta * HZ) / runtime->rate), delta, + ((hdelta * HZ) / runtime->rate), hw_base, (unsigned long)old_hw_ptr, (unsigned long)new_hw_ptr); + /* reset values to proper state */ + delta = 0; + hw_base = new_hw_ptr - (new_hw_ptr % runtime->buffer_size); } no_jiffies_check: if (delta > runtime->period_size + runtime->period_size / 2) { -- cgit v1.1 From d1458279bf9c575a52fd22818ca19c463f380aba Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Jan 2010 09:16:52 +0100 Subject: ALSA: Add snd_pci_quirk_lookup_id() Added a new function to look up a quirk entry with the given PCI SSID instead of a pci device pointer. This can be used when the searched ID is overridden for debugging or such a purpose. Signed-off-by: Takashi Iwai --- include/sound/core.h | 3 +++ sound/core/misc.c | 32 +++++++++++++++++++++++++++----- 2 files changed, 30 insertions(+), 5 deletions(-) diff --git a/include/sound/core.h b/include/sound/core.h index a61499c..89e0ac1 100644 --- a/include/sound/core.h +++ b/include/sound/core.h @@ -458,5 +458,8 @@ struct snd_pci_quirk { const struct snd_pci_quirk * snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list); +const struct snd_pci_quirk * +snd_pci_quirk_lookup_id(u16 vendor, u16 device, + const struct snd_pci_quirk *list); #endif /* __SOUND_CORE_H */ diff --git a/sound/core/misc.c b/sound/core/misc.c index 23a032c..3da4f92 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -101,8 +101,9 @@ EXPORT_SYMBOL_GPL(__snd_printk); #ifdef CONFIG_PCI #include /** - * snd_pci_quirk_lookup - look up a PCI SSID quirk list - * @pci: pci_dev handle + * snd_pci_quirk_lookup_id - look up a PCI SSID quirk list + * @vendor: PCI SSV id + * @device: PCI SSD id * @list: quirk list, terminated by a null entry * * Look through the given quirk list and finds a matching entry @@ -112,18 +113,39 @@ EXPORT_SYMBOL_GPL(__snd_printk); * Returns the matched entry pointer, or NULL if nothing matched. */ const struct snd_pci_quirk * -snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) +snd_pci_quirk_lookup_id(u16 vendor, u16 device, + const struct snd_pci_quirk *list) { const struct snd_pci_quirk *q; for (q = list; q->subvendor; q++) { - if (q->subvendor != pci->subsystem_vendor) + if (q->subvendor != vendor) continue; if (!q->subdevice || - (pci->subsystem_device & q->subdevice_mask) == q->subdevice) + (device & q->subdevice_mask) == q->subdevice) return q; } return NULL; } +EXPORT_SYMBOL(snd_pci_quirk_lookup_id); + +/** + * snd_pci_quirk_lookup - look up a PCI SSID quirk list + * @pci: pci_dev handle + * @list: quirk list, terminated by a null entry + * + * Look through the given quirk list and finds a matching entry + * with the same PCI SSID. When subdevice is 0, all subdevice + * values may match. + * + * Returns the matched entry pointer, or NULL if nothing matched. + */ +const struct snd_pci_quirk * +snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) +{ + return snd_pci_quirk_lookup_id(pci->subsystem_vendor, + pci->subsystem_device, + list); +} EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif -- cgit v1.1 From 408bffd01cfcda2907b07fb86b3666e3db86fd82 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 14 Jan 2010 09:19:46 +0100 Subject: ALSA: ctxfi - Add subsystem option Added a new option "subsystem" to override the PCI SSID for identifying the card type. Signed-off-by: Takashi Iwai --- Documentation/sound/alsa/ALSA-Configuration.txt | 3 +++ sound/pci/ctxfi/ctatc.c | 23 +++++++++++++++-------- sound/pci/ctxfi/ctatc.h | 2 +- sound/pci/ctxfi/xfi.c | 5 ++++- 4 files changed, 23 insertions(+), 10 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index c540637..c83fd7b 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -482,6 +482,9 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. reference_rate - reference sample rate, 44100 or 48000 (default) multiple - multiple to ref. sample rate, 1 or 2 (default) + subsystem - override the PCI SSID for probing; the value + consists of SSVID << 16 | SSDID. The default is + zero, which means no override. This module supports multiple cards. diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index cb65bd0..903594e 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1225,10 +1225,11 @@ static int atc_dev_free(struct snd_device *dev) return ct_atc_destroy(atc); } -static int __devinit atc_identify_card(struct ct_atc *atc) +static int __devinit atc_identify_card(struct ct_atc *atc, unsigned int ssid) { const struct snd_pci_quirk *p; const struct snd_pci_quirk *list; + u16 vendor_id, device_id; switch (atc->chip_type) { case ATC20K1: @@ -1242,13 +1243,19 @@ static int __devinit atc_identify_card(struct ct_atc *atc) default: return -ENOENT; } - p = snd_pci_quirk_lookup(atc->pci, list); + if (ssid) { + vendor_id = ssid >> 16; + device_id = ssid & 0xffff; + } else { + vendor_id = atc->pci->subsystem_vendor; + device_id = atc->pci->subsystem_device; + } + p = snd_pci_quirk_lookup_id(vendor_id, device_id, list); if (p) { if (p->value < 0) { printk(KERN_ERR "ctxfi: " "Device %04x:%04x is black-listed\n", - atc->pci->subsystem_vendor, - atc->pci->subsystem_device); + vendor_id, device_id); return -ENOENT; } atc->model = p->value; @@ -1261,8 +1268,7 @@ static int __devinit atc_identify_card(struct ct_atc *atc) atc->model_name = ct_subsys_name[atc->model]; snd_printd("ctxfi: chip %s model %s (%04x:%04x) is found\n", atc->chip_name, atc->model_name, - atc->pci->subsystem_vendor, - atc->pci->subsystem_device); + vendor_id, device_id); return 0; } @@ -1636,7 +1642,8 @@ static struct ct_atc atc_preset __devinitdata = { int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, unsigned int rsr, unsigned int msr, - int chip_type, struct ct_atc **ratc) + int chip_type, unsigned int ssid, + struct ct_atc **ratc) { struct ct_atc *atc; static struct snd_device_ops ops = { @@ -1662,7 +1669,7 @@ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, mutex_init(&atc->atc_mutex); /* Find card model */ - err = atc_identify_card(atc); + err = atc_identify_card(atc, ssid); if (err < 0) { printk(KERN_ERR "ctatc: Card not recognised\n"); goto error1; diff --git a/sound/pci/ctxfi/ctatc.h b/sound/pci/ctxfi/ctatc.h index 9fd8a57..7167c01 100644 --- a/sound/pci/ctxfi/ctatc.h +++ b/sound/pci/ctxfi/ctatc.h @@ -148,7 +148,7 @@ struct ct_atc { int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, unsigned int rsr, unsigned int msr, int chip_type, - struct ct_atc **ratc); + unsigned int subsysid, struct ct_atc **ratc); int __devinit ct_atc_create_alsa_devs(struct ct_atc *atc); #endif /* CTATC_H */ diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index 7654174..ed44ed7 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -32,6 +32,7 @@ module_param(multiple, uint, S_IRUGO); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +static unsigned int subsystem[SNDRV_CARDS]; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Creative X-Fi driver"); @@ -39,6 +40,8 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for Creative X-Fi driver"); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Creative X-Fi driver"); +module_param_array(subsystem, int, NULL, 0444); +MODULE_PARM_DESC(subsystem, "Override subsystem ID for Creative X-Fi driver"); static struct pci_device_id ct_pci_dev_ids[] = { /* only X-Fi is supported, so... */ @@ -85,7 +88,7 @@ ct_card_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) multiple = 2; } err = ct_atc_create(card, pci, reference_rate, multiple, - pci_id->driver_data, &atc); + pci_id->driver_data, subsystem[dev], &atc); if (err < 0) goto error; -- cgit v1.1 From 3e879d7bac705be4813a0ec9560cbe31db4b269f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Jan 2010 14:49:50 +0100 Subject: ALSA: pcm - Remove unneeded ifdef pgprot_noncached Signed-off-by: Takashi Iwai --- sound/core/pcm_native.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a870fe6..5df0d21 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3162,9 +3162,7 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, long size; unsigned long offset; -#ifdef pgprot_noncached area->vm_page_prot = pgprot_noncached(area->vm_page_prot); -#endif area->vm_flags |= VM_IO; size = area->vm_end - area->vm_start; offset = area->vm_pgoff << PAGE_SHIFT; -- cgit v1.1 From c32d977b8157bf67cdf47729ce7dd054a26eb534 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 18 Jan 2010 14:58:57 +0100 Subject: ALSA: pcm - Call pgprot_noncached() for vmalloc'ed buffers pgprot_noncached() can be set for vmalloc'ed buffers safely, and we'd need non-cached behavior more or less, even for the intermediate ring- buffers. Now snd_pcm_lib_mmap_vmalloc() is added as the common PCM mmap callback that is coupled with snd_pcm_lib_alloc_vmalloc_buffer() & co. Signed-off-by: Takashi Iwai --- include/sound/pcm.h | 4 ++++ sound/core/pcm_native.c | 9 +++++++++ sound/drivers/vx/vx_pcm.c | 2 ++ sound/mips/sgio2audio.c | 3 +++ sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 1 + sound/usb/ua101.c | 2 ++ sound/usb/usbaudio.c | 2 ++ 7 files changed, 23 insertions(+) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 1d4ca2a..aabf48b 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1021,6 +1021,10 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, struct vm_area_s #define snd_pcm_lib_mmap_iomem NULL #endif +int snd_pcm_lib_mmap_noncached(struct snd_pcm_substream *substream, + struct vm_area_struct *area); +#define snd_pcm_lib_mmap_vmalloc snd_pcm_lib_mmap_noncached + static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max) { *max = dma < 4 ? 64 * 1024 : 128 * 1024; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 5df0d21..88fff44 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3176,6 +3176,15 @@ int snd_pcm_lib_mmap_iomem(struct snd_pcm_substream *substream, EXPORT_SYMBOL(snd_pcm_lib_mmap_iomem); #endif /* SNDRV_PCM_INFO_MMAP */ +/* mmap callback with pgprot_noncached */ +int snd_pcm_lib_mmap_noncached(struct snd_pcm_substream *substream, + struct vm_area_struct *area) +{ + area->vm_page_prot = pgprot_noncached(area->vm_page_prot); + return snd_pcm_default_mmap(substream, area); +} +EXPORT_SYMBOL(snd_pcm_lib_mmap_noncached); + /* * mmap DMA buffer */ diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index c8385d2..35a2f71 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -905,6 +905,7 @@ static struct snd_pcm_ops vx_pcm_playback_ops = { .trigger = vx_pcm_trigger, .pointer = vx_pcm_playback_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; @@ -1125,6 +1126,7 @@ static struct snd_pcm_ops vx_pcm_capture_ops = { .trigger = vx_pcm_trigger, .pointer = vx_pcm_capture_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 9b486be..6aff217 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -691,6 +691,7 @@ static struct snd_pcm_ops snd_sgio2audio_playback1_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static struct snd_pcm_ops snd_sgio2audio_playback2_ops = { @@ -703,6 +704,7 @@ static struct snd_pcm_ops snd_sgio2audio_playback2_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static struct snd_pcm_ops snd_sgio2audio_capture_ops = { @@ -715,6 +717,7 @@ static struct snd_pcm_ops snd_sgio2audio_capture_ops = { .trigger = snd_sgio2audio_pcm_trigger, .pointer = snd_sgio2audio_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; /* diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 0afa683..0d668f4 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -277,6 +277,7 @@ static struct snd_pcm_ops pdacf_pcm_capture_ops = { .trigger = pdacf_pcm_trigger, .pointer = pdacf_pcm_capture_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; diff --git a/sound/usb/ua101.c b/sound/usb/ua101.c index 16dc7bd..4f4ccdf 100644 --- a/sound/usb/ua101.c +++ b/sound/usb/ua101.c @@ -911,6 +911,7 @@ static struct snd_pcm_ops capture_pcm_ops = { .trigger = capture_pcm_trigger, .pointer = capture_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static struct snd_pcm_ops playback_pcm_ops = { @@ -923,6 +924,7 @@ static struct snd_pcm_ops playback_pcm_ops = { .trigger = playback_pcm_trigger, .pointer = playback_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static const struct uac_format_type_i_discrete_descriptor * diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 4ada98e..b8e0b8f 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -1997,6 +1997,7 @@ static struct snd_pcm_ops snd_usb_playback_ops = { .trigger = snd_usb_pcm_playback_trigger, .pointer = snd_usb_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; static struct snd_pcm_ops snd_usb_capture_ops = { @@ -2009,6 +2010,7 @@ static struct snd_pcm_ops snd_usb_capture_ops = { .trigger = snd_usb_pcm_capture_trigger, .pointer = snd_usb_pcm_pointer, .page = snd_pcm_lib_get_vmalloc_page, + .mmap = snd_pcm_lib_mmap_vmalloc, }; -- cgit v1.1 From a32f66746c635ebf2341d99b3d4c0cc1c11b2cbf Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 18 Jan 2010 15:40:56 +0100 Subject: sound: seq_timer: simplify snd_seq_timer_set_tick_resolution() parameters As snd_seq_timer_set_tick_resolution() is always called with the same three fields of struct snd_seq_timer, it suffices to give that as the only parameter. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/core/seq/seq_timer.c | 27 +++++++++++++-------------- 1 file changed, 13 insertions(+), 14 deletions(-) diff --git a/sound/core/seq/seq_timer.c b/sound/core/seq/seq_timer.c index f745c31..160b1bd 100644 --- a/sound/core/seq/seq_timer.c +++ b/sound/core/seq/seq_timer.c @@ -33,22 +33,21 @@ #define SKEW_BASE 0x10000 /* 16bit shift */ -static void snd_seq_timer_set_tick_resolution(struct snd_seq_timer_tick *tick, - int tempo, int ppq) +static void snd_seq_timer_set_tick_resolution(struct snd_seq_timer *tmr) { - if (tempo < 1000000) - tick->resolution = (tempo * 1000) / ppq; + if (tmr->tempo < 1000000) + tmr->tick.resolution = (tmr->tempo * 1000) / tmr->ppq; else { /* might overflow.. */ unsigned int s; - s = tempo % ppq; - s = (s * 1000) / ppq; - tick->resolution = (tempo / ppq) * 1000; - tick->resolution += s; + s = tmr->tempo % tmr->ppq; + s = (s * 1000) / tmr->ppq; + tmr->tick.resolution = (tmr->tempo / tmr->ppq) * 1000; + tmr->tick.resolution += s; } - if (tick->resolution <= 0) - tick->resolution = 1; - snd_seq_timer_update_tick(tick, 0); + if (tmr->tick.resolution <= 0) + tmr->tick.resolution = 1; + snd_seq_timer_update_tick(&tmr->tick, 0); } /* create new timer (constructor) */ @@ -96,7 +95,7 @@ void snd_seq_timer_defaults(struct snd_seq_timer * tmr) /* setup defaults */ tmr->ppq = 96; /* 96 PPQ */ tmr->tempo = 500000; /* 120 BPM */ - snd_seq_timer_set_tick_resolution(&tmr->tick, tmr->tempo, tmr->ppq); + snd_seq_timer_set_tick_resolution(tmr); tmr->running = 0; tmr->type = SNDRV_SEQ_TIMER_ALSA; @@ -180,7 +179,7 @@ int snd_seq_timer_set_tempo(struct snd_seq_timer * tmr, int tempo) spin_lock_irqsave(&tmr->lock, flags); if ((unsigned int)tempo != tmr->tempo) { tmr->tempo = tempo; - snd_seq_timer_set_tick_resolution(&tmr->tick, tmr->tempo, tmr->ppq); + snd_seq_timer_set_tick_resolution(tmr); } spin_unlock_irqrestore(&tmr->lock, flags); return 0; @@ -205,7 +204,7 @@ int snd_seq_timer_set_ppq(struct snd_seq_timer * tmr, int ppq) } tmr->ppq = ppq; - snd_seq_timer_set_tick_resolution(&tmr->tick, tmr->tempo, tmr->ppq); + snd_seq_timer_set_tick_resolution(tmr); spin_unlock_irqrestore(&tmr->lock, flags); return 0; } -- cgit v1.1 From d1db38c015a392b0ea8c15ab95abb3ee768b8d47 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 18 Jan 2010 15:44:04 +0100 Subject: sound: virtuoso: add Xonar DS support Add experimental support for the Asus Xonar DS. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- Documentation/sound/alsa/ALSA-Configuration.txt | 2 +- sound/pci/Kconfig | 1 + sound/pci/oxygen/Makefile | 2 +- sound/pci/oxygen/virtuoso.c | 3 + sound/pci/oxygen/wm8766.h | 73 ++ sound/pci/oxygen/wm8776.h | 177 ++++ sound/pci/oxygen/xonar.h | 2 + sound/pci/oxygen/xonar_wm87x6.c | 1021 +++++++++++++++++++++++ 8 files changed, 1279 insertions(+), 2 deletions(-) create mode 100644 sound/pci/oxygen/wm8766.h create mode 100644 sound/pci/oxygen/wm8776.h create mode 100644 sound/pci/oxygen/xonar_wm87x6.c diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 8923597..3579e82 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1923,7 +1923,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. ------------------- Module for sound cards based on the Asus AV100/AV200 chips, - i.e., Xonar D1, DX, D2, D2X, HDAV1.3 (Deluxe), Essence ST + i.e., Xonar D1, DX, D2, D2X, DS, HDAV1.3 (Deluxe), Essence ST (Deluxe) and Essence STX. This module supports autoprobe and multiple cards. diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 351654c..1298c68 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -789,6 +789,7 @@ config SND_VIRTUOSO Say Y here to include support for sound cards based on the Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, Essence ST (Deluxe), and Essence STX. + Support for the DS is experimental. Support for the HDAV1.3 (Deluxe) is very experimental. To compile this driver as a module, choose M here: the module diff --git a/sound/pci/oxygen/Makefile b/sound/pci/oxygen/Makefile index 389941c..acd8f15 100644 --- a/sound/pci/oxygen/Makefile +++ b/sound/pci/oxygen/Makefile @@ -2,7 +2,7 @@ snd-oxygen-lib-objs := oxygen_io.o oxygen_lib.o oxygen_mixer.o oxygen_pcm.o snd-hifier-objs := hifier.o snd-oxygen-objs := oxygen.o snd-virtuoso-objs := virtuoso.o xonar_lib.o \ - xonar_pcm179x.o xonar_cs43xx.o xonar_hdmi.o + xonar_pcm179x.o xonar_cs43xx.o xonar_wm87x6.o xonar_hdmi.o obj-$(CONFIG_SND_OXYGEN_LIB) += snd-oxygen-lib.o obj-$(CONFIG_SND_HIFIER) += snd-hifier.o diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 6accaf9..563b6f5 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -49,6 +49,7 @@ static struct pci_device_id xonar_ids[] __devinitdata = { { OXYGEN_PCI_SUBID(0x1043, 0x834f) }, { OXYGEN_PCI_SUBID(0x1043, 0x835c) }, { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, + { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; @@ -61,6 +62,8 @@ static int __devinit get_xonar_model(struct oxygen *chip, return 0; if (get_xonar_cs43xx_model(chip, id) >= 0) return 0; + if (get_xonar_wm87x6_model(chip, id) >= 0) + return 0; return -EINVAL; } diff --git a/sound/pci/oxygen/wm8766.h b/sound/pci/oxygen/wm8766.h new file mode 100644 index 0000000..e0e849a --- /dev/null +++ b/sound/pci/oxygen/wm8766.h @@ -0,0 +1,73 @@ +#ifndef WM8766_H_INCLUDED +#define WM8766_H_INCLUDED + +#define WM8766_LDA1 0x00 +#define WM8766_RDA1 0x01 +#define WM8766_DAC_CTRL 0x02 +#define WM8766_INT_CTRL 0x03 +#define WM8766_LDA2 0x04 +#define WM8766_RDA2 0x05 +#define WM8766_LDA3 0x06 +#define WM8766_RDA3 0x07 +#define WM8766_MASTDA 0x08 +#define WM8766_DAC_CTRL2 0x09 +#define WM8766_DAC_CTRL3 0x0a +#define WM8766_MUTE1 0x0c +#define WM8766_MUTE2 0x0f +#define WM8766_RESET 0x1f + +/* LDAx/RDAx/MASTDA */ +#define WM8766_ATT_MASK 0x0ff +#define WM8766_UPDATE 0x100 +/* DAC_CTRL */ +#define WM8766_MUTEALL 0x001 +#define WM8766_DEEMPALL 0x002 +#define WM8766_PWDN 0x004 +#define WM8766_ATC 0x008 +#define WM8766_IZD 0x010 +#define WM8766_PL_LEFT_MASK 0x060 +#define WM8766_PL_LEFT_MUTE 0x000 +#define WM8766_PL_LEFT_LEFT 0x020 +#define WM8766_PL_LEFT_RIGHT 0x040 +#define WM8766_PL_LEFT_LRMIX 0x060 +#define WM8766_PL_RIGHT_MASK 0x180 +#define WM8766_PL_RIGHT_MUTE 0x000 +#define WM8766_PL_RIGHT_LEFT 0x080 +#define WM8766_PL_RIGHT_RIGHT 0x100 +#define WM8766_PL_RIGHT_LRMIX 0x180 +/* INT_CTRL */ +#define WM8766_FMT_MASK 0x003 +#define WM8766_FMT_RJUST 0x000 +#define WM8766_FMT_LJUST 0x001 +#define WM8766_FMT_I2S 0x002 +#define WM8766_FMT_DSP 0x003 +#define WM8766_LRP 0x004 +#define WM8766_BCP 0x008 +#define WM8766_IWL_MASK 0x030 +#define WM8766_IWL_16 0x000 +#define WM8766_IWL_20 0x010 +#define WM8766_IWL_24 0x020 +#define WM8766_IWL_32 0x030 +#define WM8766_PHASE_MASK 0x1c0 +/* DAC_CTRL2 */ +#define WM8766_ZCD 0x001 +#define WM8766_DZFM_MASK 0x006 +#define WM8766_DMUTE_MASK 0x038 +#define WM8766_DEEMP_MASK 0x1c0 +/* DAC_CTRL3 */ +#define WM8766_DACPD_MASK 0x00e +#define WM8766_PWRDNALL 0x010 +#define WM8766_MS 0x020 +#define WM8766_RATE_MASK 0x1c0 +#define WM8766_RATE_128 0x000 +#define WM8766_RATE_192 0x040 +#define WM8766_RATE_256 0x080 +#define WM8766_RATE_384 0x0c0 +#define WM8766_RATE_512 0x100 +#define WM8766_RATE_768 0x140 +/* MUTE1 */ +#define WM8766_MPD1 0x040 +/* MUTE2 */ +#define WM8766_MPD2 0x020 + +#endif diff --git a/sound/pci/oxygen/wm8776.h b/sound/pci/oxygen/wm8776.h new file mode 100644 index 0000000..1a96f56 --- /dev/null +++ b/sound/pci/oxygen/wm8776.h @@ -0,0 +1,177 @@ +#ifndef WM8776_H_INCLUDED +#define WM8776_H_INCLUDED + +/* + * the following register names are from: + * wm8776.h -- WM8776 ASoC driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#define WM8776_HPLVOL 0x00 +#define WM8776_HPRVOL 0x01 +#define WM8776_HPMASTER 0x02 +#define WM8776_DACLVOL 0x03 +#define WM8776_DACRVOL 0x04 +#define WM8776_DACMASTER 0x05 +#define WM8776_PHASESWAP 0x06 +#define WM8776_DACCTRL1 0x07 +#define WM8776_DACMUTE 0x08 +#define WM8776_DACCTRL2 0x09 +#define WM8776_DACIFCTRL 0x0a +#define WM8776_ADCIFCTRL 0x0b +#define WM8776_MSTRCTRL 0x0c +#define WM8776_PWRDOWN 0x0d +#define WM8776_ADCLVOL 0x0e +#define WM8776_ADCRVOL 0x0f +#define WM8776_ALCCTRL1 0x10 +#define WM8776_ALCCTRL2 0x11 +#define WM8776_ALCCTRL3 0x12 +#define WM8776_NOISEGATE 0x13 +#define WM8776_LIMITER 0x14 +#define WM8776_ADCMUX 0x15 +#define WM8776_OUTMUX 0x16 +#define WM8776_RESET 0x17 + + +/* HPLVOL/HPRVOL/HPMASTER */ +#define WM8776_HPATT_MASK 0x07f +#define WM8776_HPZCEN 0x080 +#define WM8776_UPDATE 0x100 + +/* DACLVOL/DACRVOL/DACMASTER */ +#define WM8776_DATT_MASK 0x0ff +/*#define WM8776_UPDATE 0x100*/ + +/* PHASESWAP */ +#define WM8776_PH_MASK 0x003 + +/* DACCTRL1 */ +#define WM8776_DZCEN 0x001 +#define WM8776_ATC 0x002 +#define WM8776_IZD 0x004 +#define WM8776_TOD 0x008 +#define WM8776_PL_LEFT_MASK 0x030 +#define WM8776_PL_LEFT_MUTE 0x000 +#define WM8776_PL_LEFT_LEFT 0x010 +#define WM8776_PL_LEFT_RIGHT 0x020 +#define WM8776_PL_LEFT_LRMIX 0x030 +#define WM8776_PL_RIGHT_MASK 0x0c0 +#define WM8776_PL_RIGHT_MUTE 0x000 +#define WM8776_PL_RIGHT_LEFT 0x040 +#define WM8776_PL_RIGHT_RIGHT 0x080 +#define WM8776_PL_RIGHT_LRMIX 0x0c0 + +/* DACMUTE */ +#define WM8776_DMUTE 0x001 + +/* DACCTRL2 */ +#define WM8776_DEEMPH 0x001 +#define WM8776_DZFM_MASK 0x006 +#define WM8776_DZFM_NONE 0x000 +#define WM8776_DZFM_LR 0x002 +#define WM8776_DZFM_BOTH 0x004 +#define WM8776_DZFM_EITHER 0x006 + +/* DACIFCTRL */ +#define WM8776_DACFMT_MASK 0x003 +#define WM8776_DACFMT_RJUST 0x000 +#define WM8776_DACFMT_LJUST 0x001 +#define WM8776_DACFMT_I2S 0x002 +#define WM8776_DACFMT_DSP 0x003 +#define WM8776_DACLRP 0x004 +#define WM8776_DACBCP 0x008 +#define WM8776_DACWL_MASK 0x030 +#define WM8776_DACWL_16 0x000 +#define WM8776_DACWL_20 0x010 +#define WM8776_DACWL_24 0x020 +#define WM8776_DACWL_32 0x030 + +/* ADCIFCTRL */ +#define WM8776_ADCFMT_MASK 0x003 +#define WM8776_ADCFMT_RJUST 0x000 +#define WM8776_ADCFMT_LJUST 0x001 +#define WM8776_ADCFMT_I2S 0x002 +#define WM8776_ADCFMT_DSP 0x003 +#define WM8776_ADCLRP 0x004 +#define WM8776_ADCBCP 0x008 +#define WM8776_ADCWL_MASK 0x030 +#define WM8776_ADCWL_16 0x000 +#define WM8776_ADCWL_20 0x010 +#define WM8776_ADCWL_24 0x020 +#define WM8776_ADCWL_32 0x030 +#define WM8776_ADCMCLK 0x040 +#define WM8776_ADCHPD 0x100 + +/* MSTRCTRL */ +#define WM8776_ADCRATE_MASK 0x007 +#define WM8776_ADCRATE_256 0x002 +#define WM8776_ADCRATE_384 0x003 +#define WM8776_ADCRATE_512 0x004 +#define WM8776_ADCRATE_768 0x005 +#define WM8776_ADCOSR 0x008 +#define WM8776_DACRATE_MASK 0x070 +#define WM8776_DACRATE_128 0x000 +#define WM8776_DACRATE_192 0x010 +#define WM8776_DACRATE_256 0x020 +#define WM8776_DACRATE_384 0x030 +#define WM8776_DACRATE_512 0x040 +#define WM8776_DACRATE_768 0x050 +#define WM8776_DACMS 0x080 +#define WM8776_ADCMS 0x100 + +/* PWRDOWN */ +#define WM8776_PDWN 0x001 +#define WM8776_ADCPD 0x002 +#define WM8776_DACPD 0x004 +#define WM8776_HPPD 0x008 +#define WM8776_AINPD 0x040 + +/* ADCLVOL/ADCRVOL */ +#define WM8776_AGMASK 0x0ff +#define WM8776_ZCA 0x100 + +/* ALCCTRL1 */ +#define WM8776_LCT_MASK 0x00f +#define WM8776_MAXGAIN_MASK 0x070 +#define WM8776_LCSEL_MASK 0x180 +#define WM8776_LCSEL_LIMITER 0x000 +#define WM8776_LCSEL_ALC_RIGHT 0x080 +#define WM8776_LCSEL_ALC_LEFT 0x100 +#define WM8776_LCSEL_ALC_STEREO 0x180 + +/* ALCCTRL2 */ +#define WM8776_HLD_MASK 0x00f +#define WM8776_ALCZC 0x080 +#define WM8776_LCEN 0x100 + +/* ALCCTRL3 */ +#define WM8776_ATK_MASK 0x00f +#define WM8776_DCY_MASK 0x0f0 + +/* NOISEGATE */ +#define WM8776_NGAT 0x001 +#define WM8776_NGTH_MASK 0x01c + +/* LIMITER */ +#define WM8776_MAXATTEN_MASK 0x00f +#define WM8776_TRANWIN_MASK 0x070 + +/* ADCMUX */ +#define WM8776_AMX_MASK 0x01f +#define WM8776_MUTERA 0x040 +#define WM8776_MUTELA 0x080 +#define WM8776_LRBOTH 0x100 + +/* OUTMUX */ +#define WM8776_MX_DAC 0x001 +#define WM8776_MX_AUX 0x002 +#define WM8776_MX_BYPASS 0x004 + +#endif diff --git a/sound/pci/oxygen/xonar.h b/sound/pci/oxygen/xonar.h index 89b3ed8..b35343b 100644 --- a/sound/pci/oxygen/xonar.h +++ b/sound/pci/oxygen/xonar.h @@ -35,6 +35,8 @@ int get_xonar_pcm179x_model(struct oxygen *chip, const struct pci_device_id *id); int get_xonar_cs43xx_model(struct oxygen *chip, const struct pci_device_id *id); +int get_xonar_wm87x6_model(struct oxygen *chip, + const struct pci_device_id *id); /* HDMI helper functions */ diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c new file mode 100644 index 0000000..7754db1 --- /dev/null +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -0,0 +1,1021 @@ +/* + * card driver for models with WM8776/WM8766 DACs (Xonar DS) + * + * Copyright (c) Clemens Ladisch + * + * + * This driver is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License, version 2. + * + * This driver 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 driver; if not, see . + */ + +/* + * Xonar DS + * -------- + * + * CMI8788: + * + * SPI 0 -> WM8766 (surround, center/LFE, back) + * SPI 1 -> WM8776 (front, input) + * + * GPIO 4 <- headphone detect + * GPIO 6 -> route input jack to input 1/2 (1/0) + * GPIO 7 -> enable output to speakers + * GPIO 8 -> enable output to speakers + */ + +#include +#include +#include +#include +#include +#include +#include +#include "xonar.h" +#include "wm8776.h" +#include "wm8766.h" + +#define GPIO_DS_HP_DETECT 0x0010 +#define GPIO_DS_INPUT_ROUTE 0x0040 +#define GPIO_DS_OUTPUT_ENABLE 0x0180 + +#define LC_CONTROL_LIMITER 0x40000000 +#define LC_CONTROL_ALC 0x20000000 + +struct xonar_wm87x6 { + struct xonar_generic generic; + u16 wm8776_regs[0x17]; + u16 wm8766_regs[0x10]; + struct snd_kcontrol *lc_controls[13]; +}; + +static void wm8776_write(struct oxygen *chip, + unsigned int reg, unsigned int value) +{ + struct xonar_wm87x6 *data = chip->model_data; + + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | + OXYGEN_SPI_DATA_LENGTH_2 | + OXYGEN_SPI_CLOCK_160 | + (1 << OXYGEN_SPI_CODEC_SHIFT) | + OXYGEN_SPI_CEN_LATCH_CLOCK_LO, + (reg << 9) | value); + if (reg < ARRAY_SIZE(data->wm8776_regs)) { + if (reg >= WM8776_HPLVOL || reg <= WM8776_DACMASTER) + value &= ~WM8776_UPDATE; + data->wm8776_regs[reg] = value; + } +} + +static void wm8776_write_cached(struct oxygen *chip, + unsigned int reg, unsigned int value) +{ + struct xonar_wm87x6 *data = chip->model_data; + + if (reg >= ARRAY_SIZE(data->wm8776_regs) || + value != data->wm8776_regs[reg]) + wm8776_write(chip, reg, value); +} + +static void wm8766_write(struct oxygen *chip, + unsigned int reg, unsigned int value) +{ + struct xonar_wm87x6 *data = chip->model_data; + + oxygen_write_spi(chip, OXYGEN_SPI_TRIGGER | + OXYGEN_SPI_DATA_LENGTH_2 | + OXYGEN_SPI_CLOCK_160 | + (0 << OXYGEN_SPI_CODEC_SHIFT) | + OXYGEN_SPI_CEN_LATCH_CLOCK_LO, + (reg << 9) | value); + if (reg < ARRAY_SIZE(data->wm8766_regs)) + data->wm8766_regs[reg] = value; +} + +static void wm8766_write_cached(struct oxygen *chip, + unsigned int reg, unsigned int value) +{ + struct xonar_wm87x6 *data = chip->model_data; + + if (reg >= ARRAY_SIZE(data->wm8766_regs) || + value != data->wm8766_regs[reg]) { + if ((reg >= WM8766_LDA1 && reg <= WM8766_RDA1) || + (reg >= WM8766_LDA2 && reg <= WM8766_MASTDA)) + value &= ~WM8766_UPDATE; + wm8766_write(chip, reg, value); + } +} + +static void wm8776_registers_init(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + + wm8776_write(chip, WM8776_RESET, 0); + wm8776_write(chip, WM8776_DACCTRL1, WM8776_DZCEN | + WM8776_PL_LEFT_LEFT | WM8776_PL_RIGHT_RIGHT); + wm8776_write(chip, WM8776_DACMUTE, chip->dac_mute ? WM8776_DMUTE : 0); + wm8776_write(chip, WM8776_DACIFCTRL, + WM8776_DACFMT_LJUST | WM8776_DACWL_24); + wm8776_write(chip, WM8776_ADCIFCTRL, + data->wm8776_regs[WM8776_ADCIFCTRL]); + wm8776_write(chip, WM8776_MSTRCTRL, data->wm8776_regs[WM8776_MSTRCTRL]); + wm8776_write(chip, WM8776_PWRDOWN, data->wm8776_regs[WM8776_PWRDOWN]); + wm8776_write(chip, WM8776_HPLVOL, data->wm8776_regs[WM8776_HPLVOL]); + wm8776_write(chip, WM8776_HPRVOL, data->wm8776_regs[WM8776_HPRVOL] | + WM8776_UPDATE); + wm8776_write(chip, WM8776_ADCLVOL, data->wm8776_regs[WM8776_ADCLVOL]); + wm8776_write(chip, WM8776_ADCRVOL, data->wm8776_regs[WM8776_ADCRVOL]); + wm8776_write(chip, WM8776_ADCMUX, data->wm8776_regs[WM8776_ADCMUX]); + wm8776_write(chip, WM8776_DACLVOL, chip->dac_volume[0]); + wm8776_write(chip, WM8776_DACRVOL, chip->dac_volume[1] | WM8776_UPDATE); +} + +static void wm8766_registers_init(struct oxygen *chip) +{ + wm8766_write(chip, WM8766_RESET, 0); + wm8766_write(chip, WM8766_INT_CTRL, WM8766_FMT_LJUST | WM8766_IWL_24); + wm8766_write(chip, WM8766_DAC_CTRL2, + WM8766_ZCD | (chip->dac_mute ? WM8766_DMUTE_MASK : 0)); + wm8766_write(chip, WM8766_LDA1, chip->dac_volume[2]); + wm8766_write(chip, WM8766_RDA1, chip->dac_volume[3]); + wm8766_write(chip, WM8766_LDA2, chip->dac_volume[4]); + wm8766_write(chip, WM8766_RDA2, chip->dac_volume[5]); + wm8766_write(chip, WM8766_LDA3, chip->dac_volume[6]); + wm8766_write(chip, WM8766_RDA3, chip->dac_volume[7] | WM8766_UPDATE); +} + +static void wm8776_init(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + + data->wm8776_regs[WM8776_HPLVOL] = (0x79 - 60) | WM8776_HPZCEN; + data->wm8776_regs[WM8776_HPRVOL] = (0x79 - 60) | WM8776_HPZCEN; + data->wm8776_regs[WM8776_ADCIFCTRL] = + WM8776_ADCFMT_LJUST | WM8776_ADCWL_24 | WM8776_ADCMCLK; + data->wm8776_regs[WM8776_MSTRCTRL] = + WM8776_ADCRATE_256 | WM8776_DACRATE_256; + data->wm8776_regs[WM8776_PWRDOWN] = WM8776_HPPD; + data->wm8776_regs[WM8776_ADCLVOL] = 0xa5 | WM8776_ZCA; + data->wm8776_regs[WM8776_ADCRVOL] = 0xa5 | WM8776_ZCA; + data->wm8776_regs[WM8776_ADCMUX] = 0x001; + wm8776_registers_init(chip); +} + +static void xonar_ds_init(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + + data->generic.anti_pop_delay = 300; + data->generic.output_enable_bit = GPIO_DS_OUTPUT_ENABLE; + + wm8776_init(chip); + wm8766_registers_init(chip); + + oxygen_write16_masked(chip, OXYGEN_GPIO_CONTROL, GPIO_DS_INPUT_ROUTE, + GPIO_DS_HP_DETECT | GPIO_DS_INPUT_ROUTE); + oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_DS_INPUT_ROUTE); + oxygen_set_bits16(chip, OXYGEN_GPIO_INTERRUPT_MASK, GPIO_DS_HP_DETECT); + chip->interrupt_mask |= OXYGEN_INT_GPIO; + + xonar_enable_output(chip); + + snd_component_add(chip->card, "WM8776"); + snd_component_add(chip->card, "WM8766"); +} + +static void xonar_ds_cleanup(struct oxygen *chip) +{ + xonar_disable_output(chip); +} + +static void xonar_ds_suspend(struct oxygen *chip) +{ + xonar_ds_cleanup(chip); +} + +static void xonar_ds_resume(struct oxygen *chip) +{ + wm8776_registers_init(chip); + wm8766_registers_init(chip); + xonar_enable_output(chip); +} + +static void wm8776_adc_hardware_filter(unsigned int channel, + struct snd_pcm_hardware *hardware) +{ + if (channel == PCM_A) { + hardware->rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000; + hardware->rate_max = 96000; + } +} + +static void set_wm87x6_dac_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ +} + +static void set_wm8776_adc_params(struct oxygen *chip, + struct snd_pcm_hw_params *params) +{ + u16 reg; + + reg = WM8776_ADCRATE_256 | WM8776_DACRATE_256; + if (params_rate(params) > 48000) + reg |= WM8776_ADCOSR; + wm8776_write_cached(chip, WM8776_MSTRCTRL, reg); +} + +static void update_wm8776_volume(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + u8 to_change; + + if (chip->dac_volume[0] == chip->dac_volume[1]) { + if (chip->dac_volume[0] != data->wm8776_regs[WM8776_DACLVOL] || + chip->dac_volume[1] != data->wm8776_regs[WM8776_DACRVOL]) { + wm8776_write(chip, WM8776_DACMASTER, + chip->dac_volume[0] | WM8776_UPDATE); + data->wm8776_regs[WM8776_DACLVOL] = chip->dac_volume[0]; + data->wm8776_regs[WM8776_DACRVOL] = chip->dac_volume[0]; + } + } else { + to_change = (chip->dac_volume[0] != + data->wm8776_regs[WM8776_DACLVOL]) << 0; + to_change |= (chip->dac_volume[1] != + data->wm8776_regs[WM8776_DACLVOL]) << 1; + if (to_change & 1) + wm8776_write(chip, WM8776_DACLVOL, chip->dac_volume[0] | + ((to_change & 2) ? 0 : WM8776_UPDATE)); + if (to_change & 2) + wm8776_write(chip, WM8776_DACRVOL, + chip->dac_volume[1] | WM8776_UPDATE); + } +} + +static void update_wm87x6_volume(struct oxygen *chip) +{ + static const u8 wm8766_regs[6] = { + WM8766_LDA1, WM8766_RDA1, + WM8766_LDA2, WM8766_RDA2, + WM8766_LDA3, WM8766_RDA3, + }; + struct xonar_wm87x6 *data = chip->model_data; + unsigned int i; + u8 to_change; + + update_wm8776_volume(chip); + if (chip->dac_volume[2] == chip->dac_volume[3] && + chip->dac_volume[2] == chip->dac_volume[4] && + chip->dac_volume[2] == chip->dac_volume[5] && + chip->dac_volume[2] == chip->dac_volume[6] && + chip->dac_volume[2] == chip->dac_volume[7]) { + to_change = 0; + for (i = 0; i < 6; ++i) + if (chip->dac_volume[2] != + data->wm8766_regs[wm8766_regs[i]]) + to_change = 1; + if (to_change) { + wm8766_write(chip, WM8766_MASTDA, + chip->dac_volume[2] | WM8766_UPDATE); + for (i = 0; i < 6; ++i) + data->wm8766_regs[wm8766_regs[i]] = + chip->dac_volume[2]; + } + } else { + to_change = 0; + for (i = 0; i < 6; ++i) + to_change |= (chip->dac_volume[2 + i] != + data->wm8766_regs[wm8766_regs[i]]) << i; + for (i = 0; i < 6; ++i) + if (to_change & (1 << i)) + wm8766_write(chip, wm8766_regs[i], + chip->dac_volume[2 + i] | + ((to_change & (0x3e << i)) + ? 0 : WM8766_UPDATE)); + } +} + +static void update_wm8776_mute(struct oxygen *chip) +{ + wm8776_write_cached(chip, WM8776_DACMUTE, + chip->dac_mute ? WM8776_DMUTE : 0); +} + +static void update_wm87x6_mute(struct oxygen *chip) +{ + update_wm8776_mute(chip); + wm8766_write_cached(chip, WM8766_DAC_CTRL2, WM8766_ZCD | + (chip->dac_mute ? WM8766_DMUTE_MASK : 0)); +} + +static void xonar_ds_gpio_changed(struct oxygen *chip) +{ + u16 bits; + + bits = oxygen_read16(chip, OXYGEN_GPIO_DATA); + snd_printk(KERN_INFO "HP detect: %d\n", !!(bits & GPIO_DS_HP_DETECT)); +} + +static int wm8776_bit_switch_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + u16 bit = ctl->private_value & 0xffff; + unsigned int reg_index = (ctl->private_value >> 16) & 0xff; + bool invert = (ctl->private_value >> 24) & 1; + + value->value.integer.value[0] = + ((data->wm8776_regs[reg_index] & bit) != 0) ^ invert; + return 0; +} + +static int wm8776_bit_switch_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + u16 bit = ctl->private_value & 0xffff; + u16 reg_value; + unsigned int reg_index = (ctl->private_value >> 16) & 0xff; + bool invert = (ctl->private_value >> 24) & 1; + int changed; + + mutex_lock(&chip->mutex); + reg_value = data->wm8776_regs[reg_index] & ~bit; + if (value->value.integer.value[0] ^ invert) + reg_value |= bit; + changed = reg_value != data->wm8776_regs[reg_index]; + if (changed) + wm8776_write(chip, reg_index, reg_value); + mutex_unlock(&chip->mutex); + return changed; +} + +static int wm8776_field_enum_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const hld[16] = { + "0 ms", "2.67 ms", "5.33 ms", "10.6 ms", + "21.3 ms", "42.7 ms", "85.3 ms", "171 ms", + "341 ms", "683 ms", "1.37 s", "2.73 s", + "5.46 s", "10.9 s", "21.8 s", "43.7 s", + }; + static const char *const atk_lim[11] = { + "0.25 ms", "0.5 ms", "1 ms", "2 ms", + "4 ms", "8 ms", "16 ms", "32 ms", + "64 ms", "128 ms", "256 ms", + }; + static const char *const atk_alc[11] = { + "8.40 ms", "16.8 ms", "33.6 ms", "67.2 ms", + "134 ms", "269 ms", "538 ms", "1.08 s", + "2.15 s", "4.3 s", "8.6 s", + }; + static const char *const dcy_lim[11] = { + "1.2 ms", "2.4 ms", "4.8 ms", "9.6 ms", + "19.2 ms", "38.4 ms", "76.8 ms", "154 ms", + "307 ms", "614 ms", "1.23 s", + }; + static const char *const dcy_alc[11] = { + "33.5 ms", "67.0 ms", "134 ms", "268 ms", + "536 ms", "1.07 s", "2.14 s", "4.29 s", + "8.58 s", "17.2 s", "34.3 s", + }; + static const char *const tranwin[8] = { + "0 us", "62.5 us", "125 us", "250 us", + "500 us", "1 ms", "2 ms", "4 ms", + }; + u8 max; + const char *const *names; + + max = (ctl->private_value >> 12) & 0xf; + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = max + 1; + if (info->value.enumerated.item > max) + info->value.enumerated.item = max; + switch ((ctl->private_value >> 24) & 0x1f) { + case WM8776_ALCCTRL2: + names = hld; + break; + case WM8776_ALCCTRL3: + if (((ctl->private_value >> 20) & 0xf) == 0) { + if (ctl->private_value & LC_CONTROL_LIMITER) + names = atk_lim; + else + names = atk_alc; + } else { + if (ctl->private_value & LC_CONTROL_LIMITER) + names = dcy_lim; + else + names = dcy_alc; + } + break; + case WM8776_LIMITER: + names = tranwin; + break; + default: + return -ENXIO; + } + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int wm8776_field_volume_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 1; + info->value.integer.min = (ctl->private_value >> 8) & 0xf; + info->value.integer.max = (ctl->private_value >> 12) & 0xf; + return 0; +} + +static void wm8776_field_set_from_ctl(struct snd_kcontrol *ctl) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + unsigned int value, reg_index, mode; + u8 min, max, shift; + u16 mask, reg_value; + bool invert; + + if ((data->wm8776_regs[WM8776_ALCCTRL1] & WM8776_LCSEL_MASK) == + WM8776_LCSEL_LIMITER) + mode = LC_CONTROL_LIMITER; + else + mode = LC_CONTROL_ALC; + if (!(ctl->private_value & mode)) + return; + + value = ctl->private_value & 0xf; + min = (ctl->private_value >> 8) & 0xf; + max = (ctl->private_value >> 12) & 0xf; + mask = (ctl->private_value >> 16) & 0xf; + shift = (ctl->private_value >> 20) & 0xf; + reg_index = (ctl->private_value >> 24) & 0x1f; + invert = (ctl->private_value >> 29) & 0x1; + + if (invert) + value = max - (value - min); + reg_value = data->wm8776_regs[reg_index]; + reg_value &= ~(mask << shift); + reg_value |= value << shift; + wm8776_write_cached(chip, reg_index, reg_value); +} + +static int wm8776_field_set(struct snd_kcontrol *ctl, unsigned int value) +{ + struct oxygen *chip = ctl->private_data; + u8 min, max; + int changed; + + min = (ctl->private_value >> 8) & 0xf; + max = (ctl->private_value >> 12) & 0xf; + if (value < min || value > max) + return -EINVAL; + mutex_lock(&chip->mutex); + changed = value != (ctl->private_value & 0xf); + if (changed) { + ctl->private_value = (ctl->private_value & ~0xf) | value; + wm8776_field_set_from_ctl(ctl); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int wm8776_field_enum_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + value->value.enumerated.item[0] = ctl->private_value & 0xf; + return 0; +} + +static int wm8776_field_volume_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + value->value.integer.value[0] = ctl->private_value & 0xf; + return 0; +} + +static int wm8776_field_enum_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + return wm8776_field_set(ctl, value->value.enumerated.item[0]); +} + +static int wm8776_field_volume_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + return wm8776_field_set(ctl, value->value.integer.value[0]); +} + +static int wm8776_hp_vol_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 2; + info->value.integer.min = 0x79 - 60; + info->value.integer.max = 0x7f; + return 0; +} + +static int wm8776_hp_vol_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + + mutex_lock(&chip->mutex); + value->value.integer.value[0] = + data->wm8776_regs[WM8776_HPLVOL] & WM8776_HPATT_MASK; + value->value.integer.value[1] = + data->wm8776_regs[WM8776_HPRVOL] & WM8776_HPATT_MASK; + mutex_unlock(&chip->mutex); + return 0; +} + +static int wm8776_hp_vol_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + u8 to_update; + + mutex_lock(&chip->mutex); + to_update = (value->value.integer.value[0] != + (data->wm8776_regs[WM8776_HPLVOL] & WM8776_HPATT_MASK)) + << 0; + to_update |= (value->value.integer.value[1] != + (data->wm8776_regs[WM8776_HPRVOL] & WM8776_HPATT_MASK)) + << 1; + if (value->value.integer.value[0] == value->value.integer.value[1]) { + if (to_update) { + wm8776_write(chip, WM8776_HPMASTER, + value->value.integer.value[0] | + WM8776_HPZCEN | WM8776_UPDATE); + data->wm8776_regs[WM8776_HPLVOL] = + value->value.integer.value[0] | WM8776_HPZCEN; + data->wm8776_regs[WM8776_HPRVOL] = + value->value.integer.value[0] | WM8776_HPZCEN; + } + } else { + if (to_update & 1) + wm8776_write(chip, WM8776_HPLVOL, + value->value.integer.value[0] | + WM8776_HPZCEN | + ((to_update & 2) ? 0 : WM8776_UPDATE)); + if (to_update & 2) + wm8776_write(chip, WM8776_HPRVOL, + value->value.integer.value[1] | + WM8776_HPZCEN | WM8776_UPDATE); + } + mutex_unlock(&chip->mutex); + return to_update != 0; +} + +static int wm8776_input_mux_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + unsigned int mux_bit = ctl->private_value; + + value->value.integer.value[0] = + !!(data->wm8776_regs[WM8776_ADCMUX] & mux_bit); + return 0; +} + +static int wm8776_input_mux_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + unsigned int mux_bit = ctl->private_value; + u16 reg; + int changed; + + mutex_lock(&chip->mutex); + reg = data->wm8776_regs[WM8776_ADCMUX]; + if (value->value.integer.value[0]) { + reg &= ~0x003; + reg |= mux_bit; + } else + reg &= ~mux_bit; + changed = reg != data->wm8776_regs[WM8776_ADCMUX]; + if (changed) { + oxygen_write16_masked(chip, OXYGEN_GPIO_DATA, + reg & 1 ? GPIO_DS_INPUT_ROUTE : 0, + GPIO_DS_INPUT_ROUTE); + wm8776_write(chip, WM8776_ADCMUX, reg); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int wm8776_input_vol_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 2; + info->value.integer.min = 0xa5; + info->value.integer.max = 0xff; + return 0; +} + +static int wm8776_input_vol_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + + mutex_lock(&chip->mutex); + value->value.integer.value[0] = + data->wm8776_regs[WM8776_ADCLVOL] & WM8776_AGMASK; + value->value.integer.value[1] = + data->wm8776_regs[WM8776_ADCRVOL] & WM8776_AGMASK; + mutex_unlock(&chip->mutex); + return 0; +} + +static int wm8776_input_vol_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + int changed = 0; + + mutex_lock(&chip->mutex); + changed = (value->value.integer.value[0] != + (data->wm8776_regs[WM8776_ADCLVOL] & WM8776_AGMASK)) || + (value->value.integer.value[1] != + (data->wm8776_regs[WM8776_ADCRVOL] & WM8776_AGMASK)); + wm8776_write_cached(chip, WM8776_ADCLVOL, + value->value.integer.value[0] | WM8776_ZCA); + wm8776_write_cached(chip, WM8776_ADCRVOL, + value->value.integer.value[1] | WM8776_ZCA); + mutex_unlock(&chip->mutex); + return changed; +} + +static int wm8776_level_control_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[3] = { + "None", "Peak Limiter", "Automatic Level Control" + }; + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 3; + if (info->value.enumerated.item >= 3) + info->value.enumerated.item = 2; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int wm8776_level_control_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + + if (!(data->wm8776_regs[WM8776_ALCCTRL2] & WM8776_LCEN)) + value->value.enumerated.item[0] = 0; + else if ((data->wm8776_regs[WM8776_ALCCTRL1] & WM8776_LCSEL_MASK) == + WM8776_LCSEL_LIMITER) + value->value.enumerated.item[0] = 1; + else + value->value.enumerated.item[0] = 2; + return 0; +} + +static void activate_control(struct oxygen *chip, + struct snd_kcontrol *ctl, unsigned int mode) +{ + unsigned int access; + + if (ctl->private_value & mode) + access = 0; + else + access = SNDRV_CTL_ELEM_ACCESS_INACTIVE; + if ((ctl->vd[0].access & SNDRV_CTL_ELEM_ACCESS_INACTIVE) != access) { + ctl->vd[0].access ^= SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(chip->card, SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); + } +} + +static int wm8776_level_control_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + unsigned int mode = 0, i; + u16 ctrl1, ctrl2; + int changed; + + if (value->value.enumerated.item[0] >= 3) + return -EINVAL; + mutex_lock(&chip->mutex); + changed = value->value.enumerated.item[0] != ctl->private_value; + if (changed) { + ctl->private_value = value->value.enumerated.item[0]; + ctrl1 = data->wm8776_regs[WM8776_ALCCTRL1]; + ctrl2 = data->wm8776_regs[WM8776_ALCCTRL2]; + switch (value->value.enumerated.item[0]) { + default: + wm8776_write_cached(chip, WM8776_ALCCTRL2, + ctrl2 & ~WM8776_LCEN); + break; + case 1: + wm8776_write_cached(chip, WM8776_ALCCTRL1, + (ctrl1 & ~WM8776_LCSEL_MASK) | + WM8776_LCSEL_LIMITER); + wm8776_write_cached(chip, WM8776_ALCCTRL2, + ctrl2 | WM8776_LCEN); + mode = LC_CONTROL_LIMITER; + break; + case 2: + wm8776_write_cached(chip, WM8776_ALCCTRL1, + (ctrl1 & ~WM8776_LCSEL_MASK) | + WM8776_LCSEL_ALC_STEREO); + wm8776_write_cached(chip, WM8776_ALCCTRL2, + ctrl2 | WM8776_LCEN); + mode = LC_CONTROL_ALC; + break; + } + for (i = 0; i < ARRAY_SIZE(data->lc_controls); ++i) + activate_control(chip, data->lc_controls[i], mode); + } + mutex_unlock(&chip->mutex); + return changed; +} + +static int hpf_info(struct snd_kcontrol *ctl, struct snd_ctl_elem_info *info) +{ + static const char *const names[2] = { + "None", "High-pass Filter" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 2; + if (info->value.enumerated.item >= 2) + info->value.enumerated.item = 1; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int hpf_get(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + + value->value.enumerated.item[0] = + !(data->wm8776_regs[WM8776_ADCIFCTRL] & WM8776_ADCHPD); + return 0; +} + +static int hpf_put(struct snd_kcontrol *ctl, struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + struct xonar_wm87x6 *data = chip->model_data; + unsigned int reg; + int changed; + + mutex_lock(&chip->mutex); + reg = data->wm8776_regs[WM8776_ADCIFCTRL] & ~WM8776_ADCHPD; + if (!value->value.enumerated.item[0]) + reg |= WM8776_ADCHPD; + changed = reg != data->wm8776_regs[WM8776_ADCIFCTRL]; + if (changed) + wm8776_write(chip, WM8776_ADCIFCTRL, reg); + mutex_unlock(&chip->mutex); + return changed; +} + +#define WM8776_BIT_SWITCH(xname, reg, bit, invert, flags) { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .info = snd_ctl_boolean_mono_info, \ + .get = wm8776_bit_switch_get, \ + .put = wm8776_bit_switch_put, \ + .private_value = ((reg) << 16) | (bit) | ((invert) << 24) | (flags), \ +} +#define _WM8776_FIELD_CTL(xname, reg, shift, initval, min, max, mask, flags) \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .private_value = (initval) | ((min) << 8) | ((max) << 12) | \ + ((mask) << 16) | ((shift) << 20) | ((reg) << 24) | (flags) +#define WM8776_FIELD_CTL_ENUM(xname, reg, shift, init, min, max, mask, flags) {\ + _WM8776_FIELD_CTL(xname " Capture Enum", \ + reg, shift, init, min, max, mask, flags), \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_INACTIVE, \ + .info = wm8776_field_enum_info, \ + .get = wm8776_field_enum_get, \ + .put = wm8776_field_enum_put, \ +} +#define WM8776_FIELD_CTL_VOLUME(a, b, c, d, e, f, g, h, tlv_p) { \ + _WM8776_FIELD_CTL(a " Capture Volume", b, c, d, e, f, g, h), \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_INACTIVE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_READ, \ + .info = wm8776_field_volume_info, \ + .get = wm8776_field_volume_get, \ + .put = wm8776_field_volume_put, \ + .tlv = { .p = tlv_p }, \ +} + +static const DECLARE_TLV_DB_SCALE(wm87x6_dac_db_scale, -6000, 50, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_adc_db_scale, -2100, 50, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_hp_db_scale, -6000, 100, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_lct_db_scale, -1600, 100, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_maxgain_db_scale, 0, 400, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_ngth_db_scale, -7800, 600, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_maxatten_lim_db_scale, -1200, 100, 0); +static const DECLARE_TLV_DB_SCALE(wm8776_maxatten_alc_db_scale, -2100, 400, 0); + +static const struct snd_kcontrol_new ds_controls[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Playback Volume", + .info = wm8776_hp_vol_info, + .get = wm8776_hp_vol_get, + .put = wm8776_hp_vol_put, + .tlv = { .p = wm8776_hp_db_scale }, + }, + WM8776_BIT_SWITCH("Headphone Playback Switch", + WM8776_PWRDOWN, WM8776_HPPD, 1, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Capture Volume", + .info = wm8776_input_vol_info, + .get = wm8776_input_vol_get, + .put = wm8776_input_vol_put, + .tlv = { .p = wm8776_adc_db_scale }, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Capture Switch", + .info = snd_ctl_boolean_mono_info, + .get = wm8776_input_mux_get, + .put = wm8776_input_mux_put, + .private_value = 1 << 0, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Capture Switch", + .info = snd_ctl_boolean_mono_info, + .get = wm8776_input_mux_get, + .put = wm8776_input_mux_put, + .private_value = 1 << 1, + }, + WM8776_BIT_SWITCH("Aux", WM8776_ADCMUX, 1 << 2, 0, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "ADC Filter Capture Enum", + .info = hpf_info, + .get = hpf_get, + .put = hpf_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Level Control Capture Enum", + .info = wm8776_level_control_info, + .get = wm8776_level_control_get, + .put = wm8776_level_control_put, + .private_value = 0, + }, +}; +static const struct snd_kcontrol_new lc_controls[] = { + WM8776_FIELD_CTL_VOLUME("Limiter Threshold", + WM8776_ALCCTRL1, 0, 11, 0, 15, 0xf, + LC_CONTROL_LIMITER, wm8776_lct_db_scale), + WM8776_FIELD_CTL_ENUM("Limiter Attack Time", + WM8776_ALCCTRL3, 0, 2, 0, 10, 0xf, + LC_CONTROL_LIMITER), + WM8776_FIELD_CTL_ENUM("Limiter Decay Time", + WM8776_ALCCTRL3, 4, 3, 0, 10, 0xf, + LC_CONTROL_LIMITER), + WM8776_FIELD_CTL_ENUM("Limiter Transient Window", + WM8776_LIMITER, 4, 2, 0, 7, 0x7, + LC_CONTROL_LIMITER), + WM8776_FIELD_CTL_VOLUME("Limiter Maximum Attenuation", + WM8776_LIMITER, 0, 6, 3, 12, 0xf, + LC_CONTROL_LIMITER, + wm8776_maxatten_lim_db_scale), + WM8776_FIELD_CTL_VOLUME("ALC Target Level", + WM8776_ALCCTRL1, 0, 11, 0, 15, 0xf, + LC_CONTROL_ALC, wm8776_lct_db_scale), + WM8776_FIELD_CTL_ENUM("ALC Attack Time", + WM8776_ALCCTRL3, 0, 2, 0, 10, 0xf, + LC_CONTROL_ALC), + WM8776_FIELD_CTL_ENUM("ALC Decay Time", + WM8776_ALCCTRL3, 4, 3, 0, 10, 0xf, + LC_CONTROL_ALC), + WM8776_FIELD_CTL_VOLUME("ALC Maximum Gain", + WM8776_ALCCTRL1, 4, 7, 1, 7, 0x7, + LC_CONTROL_ALC, wm8776_maxgain_db_scale), + WM8776_FIELD_CTL_VOLUME("ALC Maximum Attenuation", + WM8776_LIMITER, 0, 10, 10, 15, 0xf, + LC_CONTROL_ALC, wm8776_maxatten_alc_db_scale), + WM8776_FIELD_CTL_ENUM("ALC Hold Time", + WM8776_ALCCTRL2, 0, 0, 0, 15, 0xf, + LC_CONTROL_ALC), + WM8776_BIT_SWITCH("Noise Gate Capture Switch", + WM8776_NOISEGATE, WM8776_NGAT, 0, + LC_CONTROL_ALC), + WM8776_FIELD_CTL_VOLUME("Noise Gate Threshold", + WM8776_NOISEGATE, 2, 0, 0, 7, 0x7, + LC_CONTROL_ALC, wm8776_ngth_db_scale), +}; + +static int xonar_ds_control_filter(struct snd_kcontrol_new *template) +{ + if (!strncmp(template->name, "CD Capture ", 11)) + return 1; /* no CD input */ + return 0; +} + +static int xonar_ds_mixer_init(struct oxygen *chip) +{ + struct xonar_wm87x6 *data = chip->model_data; + unsigned int i; + struct snd_kcontrol *ctl; + int err; + + for (i = 0; i < ARRAY_SIZE(ds_controls); ++i) { + ctl = snd_ctl_new1(&ds_controls[i], chip); + if (!ctl) + return -ENOMEM; + err = snd_ctl_add(chip->card, ctl); + if (err < 0) + return err; + } + BUILD_BUG_ON(ARRAY_SIZE(lc_controls) != ARRAY_SIZE(data->lc_controls)); + for (i = 0; i < ARRAY_SIZE(lc_controls); ++i) { + ctl = snd_ctl_new1(&lc_controls[i], chip); + if (!ctl) + return -ENOMEM; + err = snd_ctl_add(chip->card, ctl); + if (err < 0) + return err; + data->lc_controls[i] = ctl; + } + return 0; +} + +static const struct oxygen_model model_xonar_ds = { + .shortname = "Xonar DS", + .longname = "Asus Virtuoso 200", + .chip = "AV200", + .init = xonar_ds_init, + .control_filter = xonar_ds_control_filter, + .mixer_init = xonar_ds_mixer_init, + .cleanup = xonar_ds_cleanup, + .suspend = xonar_ds_suspend, + .resume = xonar_ds_resume, + .pcm_hardware_filter = wm8776_adc_hardware_filter, + .get_i2s_mclk = oxygen_default_i2s_mclk, + .set_dac_params = set_wm87x6_dac_params, + .set_adc_params = set_wm8776_adc_params, + .update_dac_volume = update_wm87x6_volume, + .update_dac_mute = update_wm87x6_mute, + .gpio_changed = xonar_ds_gpio_changed, + .dac_tlv = wm87x6_dac_db_scale, + .model_data_size = sizeof(struct xonar_wm87x6), + .device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_1, + .dac_channels = 8, + .dac_volume_min = 255 - 2*60, + .dac_volume_max = 255, + .function_flags = OXYGEN_FUNCTION_SPI, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +}; + +int __devinit get_xonar_wm87x6_model(struct oxygen *chip, + const struct pci_device_id *id) +{ + switch (id->subdevice) { + case 0x838e: + chip->model = model_xonar_ds; + break; + default: + return -EINVAL; + } + return 0; +} -- cgit v1.1 From c91a988dc6551c66418690e36b2a23cdb0255da8 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 21 Jan 2010 10:32:15 +0100 Subject: ALSA: pcm_core: Fix wake_up() optimization This change fixes the "ALSA: pcm_lib - optimize wake_up() calls for PCM I/O" commit. New sleeping queue is introduced to separate user space and kernel space wake_ups. runtime->nowake is renamed to twake (transfer wake). Signed-off-by: Jaroslav Kysela --- include/sound/pcm.h | 5 +++-- sound/core/pcm.c | 1 + sound/core/pcm_lib.c | 20 ++++++++++---------- sound/core/pcm_native.c | 3 +++ 4 files changed, 17 insertions(+), 12 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e26fb3c..3bc9bca 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -311,8 +311,9 @@ struct snd_pcm_runtime { struct snd_pcm_mmap_control *control; /* -- locking / scheduling -- */ - unsigned int nowake: 1; /* no wakeup (data-copy in progress) */ - wait_queue_head_t sleep; + unsigned int twake: 1; /* do transfer (!poll) wakeup */ + wait_queue_head_t sleep; /* poll sleep */ + wait_queue_head_t tsleep; /* transfer sleep */ struct fasync_struct *fasync; /* -- private section -- */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index df57a0e..0d428d0 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -894,6 +894,7 @@ int snd_pcm_attach_substream(struct snd_pcm *pcm, int stream, memset((void*)runtime->control, 0, size); init_waitqueue_head(&runtime->sleep); + init_waitqueue_head(&runtime->tsleep); runtime->status->state = SNDRV_PCM_STATE_OPEN; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 5417f7d..e2a817e 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -285,8 +285,8 @@ int snd_pcm_update_state(struct snd_pcm_substream *substream, return -EPIPE; } } - if (!runtime->nowake && avail >= runtime->control->avail_min) - wake_up(&runtime->sleep); + if (avail >= runtime->control->avail_min) + wake_up(runtime->twake ? &runtime->tsleep : &runtime->sleep); return 0; } @@ -1692,7 +1692,7 @@ static int wait_for_avail_min(struct snd_pcm_substream *substream, long tout; init_waitqueue_entry(&wait, current); - add_wait_queue(&runtime->sleep, &wait); + add_wait_queue(&runtime->tsleep, &wait); for (;;) { if (signal_pending(current)) { err = -ERESTARTSYS; @@ -1735,7 +1735,7 @@ static int wait_for_avail_min(struct snd_pcm_substream *substream, break; } _endloop: - remove_wait_queue(&runtime->sleep, &wait); + remove_wait_queue(&runtime->tsleep, &wait); *availp = avail; return err; } @@ -1794,7 +1794,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, goto _end_unlock; } - runtime->nowake = 1; + runtime->twake = 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -1816,7 +1816,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, if (frames > cont) frames = cont; if (snd_BUG_ON(!frames)) { - runtime->nowake = 0; + runtime->twake = 0; snd_pcm_stream_unlock_irq(substream); return -EINVAL; } @@ -1855,7 +1855,7 @@ static snd_pcm_sframes_t snd_pcm_lib_write1(struct snd_pcm_substream *substream, } } _end_unlock: - runtime->nowake = 0; + runtime->twake = 0; if (xfer > 0 && err >= 0) snd_pcm_update_state(substream, runtime); snd_pcm_stream_unlock_irq(substream); @@ -2016,7 +2016,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, goto _end_unlock; } - runtime->nowake = 1; + runtime->twake = 1; while (size > 0) { snd_pcm_uframes_t frames, appl_ptr, appl_ofs; snd_pcm_uframes_t avail; @@ -2045,7 +2045,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, if (frames > cont) frames = cont; if (snd_BUG_ON(!frames)) { - runtime->nowake = 0; + runtime->twake = 0; snd_pcm_stream_unlock_irq(substream); return -EINVAL; } @@ -2078,7 +2078,7 @@ static snd_pcm_sframes_t snd_pcm_lib_read1(struct snd_pcm_substream *substream, xfer += frames; } _end_unlock: - runtime->nowake = 0; + runtime->twake = 0; if (xfer > 0 && err >= 0) snd_pcm_update_state(substream, runtime); snd_pcm_stream_unlock_irq(substream); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 27284f6..56ec35e 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -919,6 +919,7 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, int state) runtime->status->state = state; } wake_up(&runtime->sleep); + wake_up(&runtime->tsleep); } static struct action_ops snd_pcm_action_stop = { @@ -1004,6 +1005,7 @@ static void snd_pcm_post_pause(struct snd_pcm_substream *substream, int push) SNDRV_TIMER_EVENT_MPAUSE, &runtime->trigger_tstamp); wake_up(&runtime->sleep); + wake_up(&runtime->tsleep); } else { runtime->status->state = SNDRV_PCM_STATE_RUNNING; if (substream->timer) @@ -1061,6 +1063,7 @@ static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, int state) runtime->status->suspended_state = runtime->status->state; runtime->status->state = SNDRV_PCM_STATE_SUSPENDED; wake_up(&runtime->sleep); + wake_up(&runtime->tsleep); } static struct action_ops snd_pcm_action_suspend = { -- cgit v1.1 From fd0b092a7b14559e2ff17ef3aaefb5d8adc7e15f Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 21 Jan 2010 14:54:38 +0100 Subject: ALSA: hda - AD1988 codec - fix SPDIF-input mixer initialization (unmute) The SPDIF-input pin 0x1c is muted by default in hardware. Unmute appropriate pin to get captured samples instead zeros. Tested on Lenovo Thinkstation. Signed-off-by: Jaroslav Kysela --- sound/pci/hda/patch_analog.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index cecd3c1..865715e 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2458,6 +2458,12 @@ static struct hda_verb ad1988_spdif_init_verbs[] = { { } }; +static struct hda_verb ad1988_spdif_in_init_verbs[] = { + /* unmute SPDIF input pin */ + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { } +}; + /* AD1989 has no ADC -> SPDIF route */ static struct hda_verb ad1989_spdif_init_verbs[] = { /* SPDIF-1 out pin */ @@ -3193,8 +3199,11 @@ static int patch_ad1988(struct hda_codec *codec) ad1988_spdif_init_verbs; } } - if (spec->dig_in_nid && codec->vendor_id < 0x11d4989a) + if (spec->dig_in_nid && codec->vendor_id < 0x11d4989a) { spec->mixers[spec->num_mixers++] = ad1988_spdif_in_mixers; + spec->init_verbs[spec->num_init_verbs++] = + ad1988_spdif_in_init_verbs; + } codec->patch_ops = ad198x_patch_ops; switch (board_config) { -- cgit v1.1 From cf944ee55cc318bdb1d4b2f3f5cce3257f7c07b3 Mon Sep 17 00:00:00 2001 From: Florian Zumbiehl Date: Tue, 26 Jan 2010 09:06:14 +0100 Subject: ALSA: cs46xx: Fix cpu idling with resume Make sure that capture DMA doesn't stay enabled after system resume as that potentially prevents the processor from entering deep sleep states. Signed-off-by: Florian Zumbiehl Signed-off-by: Takashi Iwai --- sound/pci/cs46xx/cs46xx_lib.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index e6b4a87..56fcf00 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -3644,6 +3644,7 @@ int snd_cs46xx_resume(struct pci_dev *pci) #ifdef CONFIG_SND_CS46XX_NEW_DSP int i; #endif + unsigned int tmp; pci_set_power_state(pci, PCI_D0); pci_restore_state(pci); @@ -3685,6 +3686,15 @@ int snd_cs46xx_resume(struct pci_dev *pci) snd_ac97_resume(chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]); snd_ac97_resume(chip->ac97[CS46XX_SECONDARY_CODEC_INDEX]); + /* + * Stop capture DMA. + */ + tmp = snd_cs46xx_peek(chip, BA1_CCTL); + chip->capt.ctl = tmp & 0x0000ffff; + snd_cs46xx_poke(chip, BA1_CCTL, tmp & 0xffff0000); + + mdelay(5); + /* reset playback/capture */ snd_cs46xx_set_play_sample_rate(chip, 8000); snd_cs46xx_set_capture_sample_rate(chip, 8000); -- cgit v1.1 From e7636925789b042ff9d98c51d48392e8c5549480 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 26 Jan 2010 17:08:24 +0100 Subject: ALSA: pcm_lib - return back hw_ptr_interrupt Clemens Ladisch noted for hw_ptr_removal in "cleanup & merge hw_ptr update functions" commit: "It is possible for the status/delay ioctls to be called when the sound card's pointer register alreay shows a position at the beginning of the new period, but immediately before the interrupt is actually executed. (This happens regularly on a SMP machine with mplayer.) When that happens, the code thinks that the position must be at least one period ahead of the current position and drops an entire buffer of data." Return back the hw_ptr_interrupt variable. The last interrupt pointer is always computed from the latest hw_ptr instead of tracking it separately (in this case all hw_ptr checks and modifications might influence also hw_ptr_interrupt and it is difficult to keep it consistent). Signed-off-by: Jaroslav Kysela --- include/sound/pcm.h | 1 + sound/core/oss/pcm_oss.c | 3 +-- sound/core/pcm_lib.c | 7 +++++-- sound/core/pcm_native.c | 2 ++ 4 files changed, 9 insertions(+), 4 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 3bc9bca..13bc83c 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -271,6 +271,7 @@ struct snd_pcm_runtime { int overrange; snd_pcm_uframes_t avail_max; snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */ + snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time */ unsigned long hw_ptr_jiffies; /* Time when hw_ptr is updated */ snd_pcm_sframes_t delay; /* extra delay; typically FIFO size */ diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 255ad91..82d4e33 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -635,8 +635,7 @@ static long snd_pcm_alsa_frames(struct snd_pcm_substream *substream, long bytes) static inline snd_pcm_uframes_t get_hw_ptr_period(struct snd_pcm_runtime *runtime) { - snd_pcm_uframes_t ptr = runtime->status->hw_ptr; - return ptr - (ptr % runtime->period_size); + return runtime->hw_ptr_interrupt; } /* define extended formats in the recent OSS versions (if any) */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index e2a817e..aa54195 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -325,8 +325,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (in_interrupt) { /* we know that one period was processed */ /* delta = "expected next hw_ptr" for in_interrupt != 0 */ - delta = old_hw_ptr - (old_hw_ptr % runtime->period_size) - + runtime->period_size; + delta = runtime->hw_ptr_interrupt + runtime->period_size; if (delta > new_hw_ptr) { hw_base += runtime->buffer_size; if (hw_base >= runtime->boundary) @@ -437,6 +436,10 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); + if (in_interrupt) { + runtime->hw_ptr_interrupt = new_hw_ptr - + (new_hw_ptr % runtime->period_size); + } runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; runtime->hw_ptr_jiffies = jiffies; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 56ec35e..7a002db 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1252,6 +1252,8 @@ static int snd_pcm_do_reset(struct snd_pcm_substream *substream, int state) if (err < 0) return err; runtime->hw_ptr_base = 0; + runtime->hw_ptr_interrupt = runtime->status->hw_ptr - + runtime->status->hw_ptr % runtime->period_size; runtime->silence_start = runtime->status->hw_ptr; runtime->silence_filled = 0; return 0; -- cgit v1.1 From 7910b4a1db63fefc3d291853d33c34c5b6352e8e Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 27 Jan 2010 18:10:13 +0100 Subject: ALSA: pcm_native - fix runtime->boundary calculation The code in pcm_lib updating runtime->hw_ptr_interrupt expects that runtime->boundary is divisible with runtime->period_size. Thanks are going to Clemens Ladisch for the notice. Fix the runtime->boundary calculation using buffer_size * period_size as base and find a least common multiple for 32bit platforms when the expression might overflow. Signed-off-by: Jaroslav Kysela --- sound/core/pcm_native.c | 39 ++++++++++++++++++++++++++++++++++++--- 1 file changed, 36 insertions(+), 3 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 7a002db..9cbaf90 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -27,6 +27,7 @@ #include #include #include +#include #include #include #include @@ -366,6 +367,38 @@ static int period_to_usecs(struct snd_pcm_runtime *runtime) return usecs; } +static int calc_boundary(struct snd_pcm_runtime *runtime) +{ + u_int64_t boundary; + + boundary = (u_int64_t)runtime->buffer_size * + (u_int64_t)runtime->period_size; +#if BITS_PER_LONG < 64 + /* try to find lowest common multiple for buffer and period */ + if (boundary > LONG_MAX - runtime->buffer_size) { + u_int32_t remainder = -1; + u_int32_t divident = runtime->buffer_size; + u_int32_t divisor = runtime->period_size; + while (remainder) { + remainder = divident % divisor; + if (remainder) { + divident = divisor; + divisor = remainder; + } + } + boundary = div_u64(boundary, divisor); + if (boundary > LONG_MAX - runtime->buffer_size) + return -ERANGE; + } +#endif + if (boundary == 0) + return -ERANGE; + runtime->boundary = boundary; + while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size) + runtime->boundary *= 2; + return 0; +} + static int snd_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -441,9 +474,9 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, runtime->stop_threshold = runtime->buffer_size; runtime->silence_threshold = 0; runtime->silence_size = 0; - runtime->boundary = runtime->buffer_size; - while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size) - runtime->boundary *= 2; + err = calc_boundary(runtime); + if (err < 0) + goto _error; snd_pcm_timer_resolution_change(substream); runtime->status->state = SNDRV_PCM_STATE_SETUP; -- cgit v1.1 From a75d7a4cf50d1cee14d8c9aaa2b971232d10f2c1 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 1 Feb 2010 13:29:50 +0100 Subject: sound: control: actually allow TLV command access Creating a control with TLV_COMMAND access was not possible because snd_ctl_new1() forgot to include it in the mask of allowable access bits. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/core/control.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/core/control.c b/sound/core/control.c index 268ab74..6a4764d 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -237,8 +237,9 @@ struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new *ncontrol, access = ncontrol->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : (ncontrol->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| SNDRV_CTL_ELEM_ACCESS_INACTIVE| - SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE| - SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)); + SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE| + SNDRV_CTL_ELEM_ACCESS_TLV_COMMAND| + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)); kctl.info = ncontrol->info; kctl.get = ncontrol->get; kctl.put = ncontrol->put; -- cgit v1.1 From 6123637fafbf445cc9ce5774dc9516da0b2daa88 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 1 Feb 2010 13:30:56 +0100 Subject: sound: control: fix minimum TLV length Allow TLV blocks that do not have any values; the smallest possible TLV is an empty container or one where the information is only in the tag. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/core/control.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/control.c b/sound/core/control.c index 6a4764d..439ce64 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1100,7 +1100,7 @@ static int snd_ctl_tlv_ioctl(struct snd_ctl_file *file, if (copy_from_user(&tlv, _tlv, sizeof(tlv))) return -EFAULT; - if (tlv.length < sizeof(unsigned int) * 3) + if (tlv.length < sizeof(unsigned int) * 2) return -EINVAL; down_read(&card->controls_rwsem); kctl = snd_ctl_find_numid(card, tlv.numid); -- cgit v1.1 From c85a400499093b2025238413198e48e4d825723e Mon Sep 17 00:00:00 2001 From: Thadeu Lima de Souza Cascardo Date: Mon, 1 Feb 2010 16:17:01 -0200 Subject: ALSA: trivial: sound seq ioctl dbg: print hexadecimal value padded with 0s Instead of padding with blanks and printing "number=0x a", print "number=0x0a". Signed-off-by: Thadeu Lima de Souza Cascardo Signed-off-by: Takashi Iwai --- sound/core/seq/seq_clientmgr.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 8ca2be3..48eca9f 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -2190,7 +2190,7 @@ static int snd_seq_do_ioctl(struct snd_seq_client *client, unsigned int cmd, if (p->cmd == cmd) return p->func(client, arg); } - snd_printd("seq unknown ioctl() 0x%x (type='%c', number=0x%2x)\n", + snd_printd("seq unknown ioctl() 0x%x (type='%c', number=0x%02x)\n", cmd, _IOC_TYPE(cmd), _IOC_NR(cmd)); return -ENOTTY; } -- cgit v1.1 From d5e1ca05f758fec2845a97fd7aa1eeca91c51a21 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 2 Feb 2010 17:48:51 +0100 Subject: ALSA: dummy driver - add model parameter This is a cleanup for the dummy driver. The model kernel module parameter is introduced to select the soundcard emulation. Signed-off-by: Jaroslav Kysela --- sound/drivers/dummy.c | 290 +++++++++++++++++++++++++++++++------------------- 1 file changed, 180 insertions(+), 110 deletions(-) diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c index 252e04c..7f41990 100644 --- a/sound/drivers/dummy.c +++ b/sound/drivers/dummy.c @@ -45,109 +45,23 @@ MODULE_SUPPORTED_DEVICE("{{ALSA,Dummy soundcard}}"); #define MAX_PCM_SUBSTREAMS 128 #define MAX_MIDI_DEVICES 2 -#if 0 /* emu10k1 emulation */ -#define MAX_BUFFER_SIZE (128 * 1024) -static int emu10k1_playback_constraints(struct snd_pcm_runtime *runtime) -{ - int err; - err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); - if (err < 0) - return err; - err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 256, UINT_MAX); - if (err < 0) - return err; - return 0; -} -#define add_playback_constraints emu10k1_playback_constraints -#endif - -#if 0 /* RME9652 emulation */ -#define MAX_BUFFER_SIZE (26 * 64 * 1024) -#define USE_FORMATS SNDRV_PCM_FMTBIT_S32_LE -#define USE_CHANNELS_MIN 26 -#define USE_CHANNELS_MAX 26 -#define USE_PERIODS_MIN 2 -#define USE_PERIODS_MAX 2 -#endif - -#if 0 /* ICE1712 emulation */ -#define MAX_BUFFER_SIZE (256 * 1024) -#define USE_FORMATS SNDRV_PCM_FMTBIT_S32_LE -#define USE_CHANNELS_MIN 10 -#define USE_CHANNELS_MAX 10 -#define USE_PERIODS_MIN 1 -#define USE_PERIODS_MAX 1024 -#endif - -#if 0 /* UDA1341 emulation */ -#define MAX_BUFFER_SIZE (16380) -#define USE_FORMATS SNDRV_PCM_FMTBIT_S16_LE -#define USE_CHANNELS_MIN 2 -#define USE_CHANNELS_MAX 2 -#define USE_PERIODS_MIN 2 -#define USE_PERIODS_MAX 255 -#endif - -#if 0 /* simple AC97 bridge (intel8x0) with 48kHz AC97 only codec */ -#define USE_FORMATS SNDRV_PCM_FMTBIT_S16_LE -#define USE_CHANNELS_MIN 2 -#define USE_CHANNELS_MAX 2 -#define USE_RATE SNDRV_PCM_RATE_48000 -#define USE_RATE_MIN 48000 -#define USE_RATE_MAX 48000 -#endif - -#if 0 /* CA0106 */ -#define USE_FORMATS SNDRV_PCM_FMTBIT_S16_LE -#define USE_CHANNELS_MIN 2 -#define USE_CHANNELS_MAX 2 -#define USE_RATE (SNDRV_PCM_RATE_48000|SNDRV_PCM_RATE_96000|SNDRV_PCM_RATE_192000) -#define USE_RATE_MIN 48000 -#define USE_RATE_MAX 192000 -#define MAX_BUFFER_SIZE ((65536-64)*8) -#define MAX_PERIOD_SIZE (65536-64) -#define USE_PERIODS_MIN 2 -#define USE_PERIODS_MAX 8 -#endif - - /* defaults */ -#ifndef MAX_BUFFER_SIZE #define MAX_BUFFER_SIZE (64*1024) -#endif -#ifndef MAX_PERIOD_SIZE +#define MIN_PERIOD_SIZE 64 #define MAX_PERIOD_SIZE MAX_BUFFER_SIZE -#endif -#ifndef USE_FORMATS #define USE_FORMATS (SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE) -#endif -#ifndef USE_RATE #define USE_RATE SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_8000_48000 #define USE_RATE_MIN 5500 #define USE_RATE_MAX 48000 -#endif -#ifndef USE_CHANNELS_MIN #define USE_CHANNELS_MIN 1 -#endif -#ifndef USE_CHANNELS_MAX #define USE_CHANNELS_MAX 2 -#endif -#ifndef USE_PERIODS_MIN #define USE_PERIODS_MIN 1 -#endif -#ifndef USE_PERIODS_MAX #define USE_PERIODS_MAX 1024 -#endif -#ifndef add_playback_constraints -#define add_playback_constraints(x) 0 -#endif -#ifndef add_capture_constraints -#define add_capture_constraints(x) 0 -#endif 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] = {1, [1 ... (SNDRV_CARDS - 1)] = 0}; +static char *model[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = NULL}; static int pcm_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 1}; static int pcm_substreams[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 8}; //static int midi_devs[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; @@ -162,6 +76,8 @@ module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for dummy soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable this dummy soundcard."); +module_param_array(model, charp, NULL, 0444); +MODULE_PARM_DESC(model, "Soundcard model."); module_param_array(pcm_devs, int, NULL, 0444); MODULE_PARM_DESC(pcm_devs, "PCM devices # (0-4) for dummy driver."); module_param_array(pcm_substreams, int, NULL, 0444); @@ -193,9 +109,28 @@ struct dummy_timer_ops { snd_pcm_uframes_t (*pointer)(struct snd_pcm_substream *); }; +struct dummy_model { + const char *name; + int (*playback_constraints)(struct snd_pcm_runtime *runtime); + int (*capture_constraints)(struct snd_pcm_runtime *runtime); + u64 formats; + size_t buffer_bytes_max; + size_t period_bytes_min; + size_t period_bytes_max; + unsigned int periods_min; + unsigned int periods_max; + unsigned int rates; + unsigned int rate_min; + unsigned int rate_max; + unsigned int channels_min; + unsigned int channels_max; +}; + struct snd_dummy { struct snd_card *card; + struct dummy_model *model; struct snd_pcm *pcm; + struct snd_pcm_hardware pcm_hw; spinlock_t mixer_lock; int mixer_volume[MIXER_ADDR_LAST+1][2]; int capture_source[MIXER_ADDR_LAST+1][2]; @@ -203,6 +138,92 @@ struct snd_dummy { }; /* + * card models + */ + +static int emu10k1_playback_constraints(struct snd_pcm_runtime *runtime) +{ + int err; + err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) + return err; + err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 256, UINT_MAX); + if (err < 0) + return err; + return 0; +} + +struct dummy_model model_emu10k1 = { + .name = "emu10k1", + .playback_constraints = emu10k1_playback_constraints, + .buffer_bytes_max = 128 * 1024, +}; + +struct dummy_model model_rme9652 = { + .name = "rme9652", + .buffer_bytes_max = 26 * 64 * 1024, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 26, + .channels_max = 26, + .periods_min = 2, + .periods_max = 2, +}; + +struct dummy_model model_ice1712 = { + .name = "ice1712", + .buffer_bytes_max = 256 * 1024, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 10, + .channels_max = 10, + .periods_min = 1, + .periods_max = 1024, +}; + +struct dummy_model model_uda1341 = { + .name = "uda1341", + .buffer_bytes_max = 16380, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 2, + .channels_max = 2, + .periods_min = 2, + .periods_max = 255, +}; + +struct dummy_model model_ac97 = { + .name = "ac97", + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, +}; + +struct dummy_model model_ca0106 = { + .name = "ca0106", + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .buffer_bytes_max = ((65536-64)*8), + .period_bytes_max = (65536-64), + .periods_min = 2, + .periods_max = 8, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000|SNDRV_PCM_RATE_96000|SNDRV_PCM_RATE_192000, + .rate_min = 48000, + .rate_max = 192000, +}; + +struct dummy_model *dummy_models[] = { + &model_emu10k1, + &model_rme9652, + &model_ice1712, + &model_uda1341, + &model_ac97, + &model_ca0106, + NULL +}; + +/* * system timer interface */ @@ -509,7 +530,7 @@ static struct snd_pcm_hardware dummy_pcm_hardware = { .channels_min = USE_CHANNELS_MIN, .channels_max = USE_CHANNELS_MAX, .buffer_bytes_max = MAX_BUFFER_SIZE, - .period_bytes_min = 64, + .period_bytes_min = MIN_PERIOD_SIZE, .period_bytes_max = MAX_PERIOD_SIZE, .periods_min = USE_PERIODS_MIN, .periods_max = USE_PERIODS_MAX, @@ -538,6 +559,7 @@ static int dummy_pcm_hw_free(struct snd_pcm_substream *substream) static int dummy_pcm_open(struct snd_pcm_substream *substream) { struct snd_dummy *dummy = snd_pcm_substream_chip(substream); + struct dummy_model *model = dummy->model; struct snd_pcm_runtime *runtime = substream->runtime; int err; @@ -551,7 +573,7 @@ static int dummy_pcm_open(struct snd_pcm_substream *substream) if (err < 0) return err; - runtime->hw = dummy_pcm_hardware; + runtime->hw = dummy->pcm_hw; if (substream->pcm->device & 1) { runtime->hw.info &= ~SNDRV_PCM_INFO_INTERLEAVED; runtime->hw.info |= SNDRV_PCM_INFO_NONINTERLEAVED; @@ -560,10 +582,16 @@ static int dummy_pcm_open(struct snd_pcm_substream *substream) runtime->hw.info &= ~(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - err = add_playback_constraints(substream->runtime); - else - err = add_capture_constraints(substream->runtime); + if (model == NULL) + return 0; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (model->playback_constraints) + err = model->playback_constraints(substream->runtime); + } else { + if (model->capture_constraints) + err = model->capture_constraints(substream->runtime); + } if (err < 0) { dummy->timer_ops->free(substream); return err; @@ -823,17 +851,19 @@ static int __devinit snd_card_dummy_new_mixer(struct snd_dummy *dummy) /* * proc interface */ -static void print_formats(struct snd_info_buffer *buffer) +static void print_formats(struct snd_dummy *dummy, + struct snd_info_buffer *buffer) { int i; for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { - if (dummy_pcm_hardware.formats & (1ULL << i)) + if (dummy->pcm_hw.formats & (1ULL << i)) snd_iprintf(buffer, " %s", snd_pcm_format_name(i)); } } -static void print_rates(struct snd_info_buffer *buffer) +static void print_rates(struct snd_dummy *dummy, + struct snd_info_buffer *buffer) { static int rates[] = { 5512, 8000, 11025, 16000, 22050, 32000, 44100, 48000, @@ -841,19 +871,19 @@ static void print_rates(struct snd_info_buffer *buffer) }; int i; - if (dummy_pcm_hardware.rates & SNDRV_PCM_RATE_CONTINUOUS) + if (dummy->pcm_hw.rates & SNDRV_PCM_RATE_CONTINUOUS) snd_iprintf(buffer, " continuous"); - if (dummy_pcm_hardware.rates & SNDRV_PCM_RATE_KNOT) + if (dummy->pcm_hw.rates & SNDRV_PCM_RATE_KNOT) snd_iprintf(buffer, " knot"); for (i = 0; i < ARRAY_SIZE(rates); i++) - if (dummy_pcm_hardware.rates & (1 << i)) + if (dummy->pcm_hw.rates & (1 << i)) snd_iprintf(buffer, " %d", rates[i]); } -#define get_dummy_int_ptr(ofs) \ - (unsigned int *)((char *)&dummy_pcm_hardware + (ofs)) -#define get_dummy_ll_ptr(ofs) \ - (unsigned long long *)((char *)&dummy_pcm_hardware + (ofs)) +#define get_dummy_int_ptr(dummy, ofs) \ + (unsigned int *)((char *)&((dummy)->pcm_hw) + (ofs)) +#define get_dummy_ll_ptr(dummy, ofs) \ + (unsigned long long *)((char *)&((dummy)->pcm_hw) + (ofs)) struct dummy_hw_field { const char *name; @@ -884,20 +914,21 @@ static struct dummy_hw_field fields[] = { static void dummy_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { + struct snd_dummy *dummy = entry->private_data; int i; for (i = 0; i < ARRAY_SIZE(fields); i++) { snd_iprintf(buffer, "%s ", fields[i].name); if (fields[i].size == sizeof(int)) snd_iprintf(buffer, fields[i].format, - *get_dummy_int_ptr(fields[i].offset)); + *get_dummy_int_ptr(dummy, fields[i].offset)); else snd_iprintf(buffer, fields[i].format, - *get_dummy_ll_ptr(fields[i].offset)); + *get_dummy_ll_ptr(dummy, fields[i].offset)); if (!strcmp(fields[i].name, "formats")) - print_formats(buffer); + print_formats(dummy, buffer); else if (!strcmp(fields[i].name, "rates")) - print_rates(buffer); + print_rates(dummy, buffer); snd_iprintf(buffer, "\n"); } } @@ -905,6 +936,7 @@ static void dummy_proc_read(struct snd_info_entry *entry, static void dummy_proc_write(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { + struct snd_dummy *dummy = entry->private_data; char line[64]; while (!snd_info_get_line(buffer, line, sizeof(line))) { @@ -924,9 +956,9 @@ static void dummy_proc_write(struct snd_info_entry *entry, if (strict_strtoull(item, 0, &val)) continue; if (fields[i].size == sizeof(int)) - *get_dummy_int_ptr(fields[i].offset) = val; + *get_dummy_int_ptr(dummy, fields[i].offset) = val; else - *get_dummy_ll_ptr(fields[i].offset) = val; + *get_dummy_ll_ptr(dummy, fields[i].offset) = val; } } @@ -938,6 +970,7 @@ static void __devinit dummy_proc_init(struct snd_dummy *chip) snd_info_set_text_ops(entry, chip, dummy_proc_read); entry->c.text.write = dummy_proc_write; entry->mode |= S_IWUSR; + entry->private_data = chip; } } #else @@ -948,6 +981,7 @@ static int __devinit snd_dummy_probe(struct platform_device *devptr) { struct snd_card *card; struct snd_dummy *dummy; + struct dummy_model *m = NULL, **mdl; int idx, err; int dev = devptr->id; @@ -957,6 +991,15 @@ static int __devinit snd_dummy_probe(struct platform_device *devptr) return err; dummy = card->private_data; dummy->card = card; + for (mdl = dummy_models; *mdl && model[dev]; mdl++) { + if (strcmp(model[dev], (*mdl)->name) == 0) { + printk(KERN_INFO + "snd-dummy: Using model '%s' for card %i\n", + (*mdl)->name, card->number); + m = dummy->model = *mdl; + break; + } + } for (idx = 0; idx < MAX_PCM_DEVICES && idx < pcm_devs[dev]; idx++) { if (pcm_substreams[dev] < 1) pcm_substreams[dev] = 1; @@ -966,6 +1009,33 @@ static int __devinit snd_dummy_probe(struct platform_device *devptr) if (err < 0) goto __nodev; } + + dummy->pcm_hw = dummy_pcm_hardware; + if (m) { + if (m->formats) + dummy->pcm_hw.formats = m->formats; + if (m->buffer_bytes_max) + dummy->pcm_hw.buffer_bytes_max = m->buffer_bytes_max; + if (m->period_bytes_min) + dummy->pcm_hw.period_bytes_min = m->period_bytes_min; + if (m->period_bytes_max) + dummy->pcm_hw.period_bytes_max = m->period_bytes_max; + if (m->periods_min) + dummy->pcm_hw.periods_min = m->periods_min; + if (m->periods_max) + dummy->pcm_hw.periods_max = m->periods_max; + if (m->rates) + dummy->pcm_hw.rates = m->rates; + if (m->rate_min) + dummy->pcm_hw.rate_min = m->rate_min; + if (m->rate_max) + dummy->pcm_hw.rate_max = m->rate_max; + if (m->channels_min) + dummy->pcm_hw.channels_min = m->channels_min; + if (m->channels_max) + dummy->pcm_hw.channels_max = m->channels_max; + } + err = snd_card_dummy_new_mixer(dummy); if (err < 0) goto __nodev; -- cgit v1.1 From 350a514787a4516746f738f69bff6aa0d4ac70e9 Mon Sep 17 00:00:00 2001 From: Sebastien Alaiwan Date: Fri, 5 Feb 2010 08:58:20 +0100 Subject: ALSA: ice1712: fix: lock samplerate when samplerate locking is enabled I found that the sampling rate locking setting of the ice1712 sound driver was only half-respected : when the driver was locked to, let's say, 44100Hz, and a usermode app was requesting 48000Hz playback, the request was succesful although the soundcard would continue to run at 44100Hz. Here's a patch that will make those requests to fail. Signed-off-by: Sebastien Alaiwan Signed-off-by: Takashi Iwai --- sound/pci/ice1712/ice1712.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index c7cff6f..fb61943 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -1180,6 +1180,10 @@ static int snd_ice1712_playback_pro_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); + if (is_pro_rate_locked(ice)) { + runtime->hw.rate_min = PRO_RATE_DEFAULT; + runtime->hw.rate_max = PRO_RATE_DEFAULT; + } if (ice->spdif.ops.open) ice->spdif.ops.open(ice, substream); @@ -1197,6 +1201,11 @@ static int snd_ice1712_capture_pro_open(struct snd_pcm_substream *substream) snd_pcm_set_sync(substream); snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); + if (is_pro_rate_locked(ice)) { + runtime->hw.rate_min = PRO_RATE_DEFAULT; + runtime->hw.rate_max = PRO_RATE_DEFAULT; + } + return 0; } -- cgit v1.1 From cebe41d4b8f8092359de31e241815fcb4b4dc0be Mon Sep 17 00:00:00 2001 From: Alexey Dobriyan Date: Sat, 6 Feb 2010 00:21:03 +0200 Subject: sound: use DEFINE_PCI_DEVICE_TABLE Use DEFINE_PCI_DEVICE_TABLE() to make PCI device ids go to .devinit.rodata section, so they can be discarded in some cases, and make them const. Signed-off-by: Alexey Dobriyan Signed-off-by: Takashi Iwai --- sound/oss/kahlua.c | 2 +- sound/pci/ad1889.c | 2 +- sound/pci/ali5451/ali5451.c | 2 +- sound/pci/als300.c | 2 +- sound/pci/als4000.c | 2 +- sound/pci/atiixp.c | 2 +- sound/pci/atiixp_modem.c | 2 +- sound/pci/au88x0/au8810.c | 2 +- sound/pci/au88x0/au8820.c | 2 +- sound/pci/au88x0/au8830.c | 2 +- sound/pci/aw2/aw2-alsa.c | 2 +- sound/pci/azt3328.c | 2 +- sound/pci/bt87x.c | 4 ++-- sound/pci/ca0106/ca0106_main.c | 2 +- sound/pci/cmipci.c | 4 ++-- sound/pci/cs4281.c | 2 +- sound/pci/cs46xx/cs46xx.c | 2 +- sound/pci/cs5530.c | 2 +- sound/pci/cs5535audio/cs5535audio.c | 2 +- sound/pci/ctxfi/xfi.c | 2 +- sound/pci/echoaudio/darla20.c | 2 +- sound/pci/echoaudio/darla24.c | 2 +- sound/pci/echoaudio/echo3g.c | 2 +- sound/pci/echoaudio/gina20.c | 2 +- sound/pci/echoaudio/gina24.c | 2 +- sound/pci/echoaudio/indigo.c | 2 +- sound/pci/echoaudio/indigodj.c | 2 +- sound/pci/echoaudio/indigodjx.c | 2 +- sound/pci/echoaudio/indigoio.c | 2 +- sound/pci/echoaudio/indigoiox.c | 2 +- sound/pci/echoaudio/layla20.c | 2 +- sound/pci/echoaudio/layla24.c | 2 +- sound/pci/echoaudio/mia.c | 2 +- sound/pci/echoaudio/mona.c | 2 +- sound/pci/emu10k1/emu10k1.c | 2 +- sound/pci/emu10k1/emu10k1x.c | 2 +- sound/pci/ens1370.c | 2 +- sound/pci/es1938.c | 2 +- sound/pci/es1968.c | 2 +- sound/pci/fm801.c | 2 +- sound/pci/hda/hda_intel.c | 2 +- sound/pci/ice1712/ice1712.c | 2 +- sound/pci/ice1712/ice1724.c | 2 +- sound/pci/intel8x0.c | 2 +- sound/pci/intel8x0m.c | 2 +- sound/pci/korg1212/korg1212.c | 2 +- sound/pci/lx6464es/lx6464es.c | 2 +- sound/pci/maestro3.c | 2 +- sound/pci/mixart/mixart.c | 2 +- sound/pci/nm256/nm256.c | 2 +- sound/pci/oxygen/hifier.c | 2 +- sound/pci/oxygen/oxygen.c | 2 +- sound/pci/oxygen/virtuoso.c | 2 +- sound/pci/pcxhr/pcxhr.c | 2 +- sound/pci/riptide/riptide.c | 4 ++-- sound/pci/rme32.c | 2 +- sound/pci/rme96.c | 2 +- sound/pci/rme9652/hdsp.c | 2 +- sound/pci/rme9652/hdspm.c | 2 +- sound/pci/rme9652/rme9652.c | 2 +- sound/pci/sis7019.c | 2 +- sound/pci/sonicvibes.c | 2 +- sound/pci/trident/trident.c | 2 +- sound/pci/via82xx.c | 2 +- sound/pci/via82xx_modem.c | 2 +- sound/pci/vx222/vx222.c | 2 +- sound/pci/ymfpci/ymfpci.c | 2 +- 67 files changed, 70 insertions(+), 70 deletions(-) diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c index 89466b0..24d152c 100644 --- a/sound/oss/kahlua.c +++ b/sound/oss/kahlua.c @@ -198,7 +198,7 @@ MODULE_LICENSE("GPL"); * 5530 only. The 5510/5520 decode is different. */ -static struct pci_device_id id_tbl[] = { +static DEFINE_PCI_DEVICE_TABLE(id_tbl) = { { PCI_VDEVICE(CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO), 0 }, { } }; diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 8f5098f..4382d0f 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -1048,7 +1048,7 @@ snd_ad1889_remove(struct pci_dev *pci) pci_set_drvdata(pci, NULL); } -static struct pci_device_id snd_ad1889_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ad1889_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_ANALOG_DEVICES, PCI_DEVICE_ID_AD1889JS) }, { 0, }, }; diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index aaf4da6..5c6e322 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -275,7 +275,7 @@ struct snd_ali { #endif }; -static struct pci_device_id snd_ali_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ali_ids) = { {PCI_DEVICE(PCI_VENDOR_ID_AL, PCI_DEVICE_ID_AL_M5451), 0, 0, 0}, {0, } }; diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 3aa35af..d7653cb 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -145,7 +145,7 @@ struct snd_als300_substream_data { int block_counter_register; }; -static struct pci_device_id snd_als300_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_als300_ids) = { { 0x4005, 0x0300, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALS300 }, { 0x4005, 0x0308, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALS300_PLUS }, { 0, } diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 3dbacde..d75cf7b 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -117,7 +117,7 @@ struct snd_card_als4000 { #endif }; -static struct pci_device_id snd_als4000_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_als4000_ids) = { { 0x4005, 0x4000, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, }, /* ALS4000 */ { 0, } }; diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index d6752df..81e2bfc 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -286,7 +286,7 @@ struct atiixp { /* */ -static struct pci_device_id snd_atiixp_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_atiixp_ids) = { { PCI_VDEVICE(ATI, 0x4341), 0 }, /* SB200 */ { PCI_VDEVICE(ATI, 0x4361), 0 }, /* SB300 */ { PCI_VDEVICE(ATI, 0x4370), 0 }, /* SB400 */ diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index e7e147b..91d7036 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -261,7 +261,7 @@ struct atiixp_modem { /* */ -static struct pci_device_id snd_atiixp_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_atiixp_ids) = { { PCI_VDEVICE(ATI, 0x434d), 0 }, /* SB200 */ { PCI_VDEVICE(ATI, 0x4378), 0 }, /* SB400 */ { 0, } diff --git a/sound/pci/au88x0/au8810.c b/sound/pci/au88x0/au8810.c index c0e8c6b..aa51cc7 100644 --- a/sound/pci/au88x0/au8810.c +++ b/sound/pci/au88x0/au8810.c @@ -1,6 +1,6 @@ #include "au8810.h" #include "au88x0.h" -static struct pci_device_id snd_vortex_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vortex_ids) = { {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_ADVANTAGE), 1,}, {0,} }; diff --git a/sound/pci/au88x0/au8820.c b/sound/pci/au88x0/au8820.c index a652733..2f321e7 100644 --- a/sound/pci/au88x0/au8820.c +++ b/sound/pci/au88x0/au8820.c @@ -1,6 +1,6 @@ #include "au8820.h" #include "au88x0.h" -static struct pci_device_id snd_vortex_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vortex_ids) = { {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_1), 0,}, {0,} }; diff --git a/sound/pci/au88x0/au8830.c b/sound/pci/au88x0/au8830.c index 6c702ad..279b78f 100644 --- a/sound/pci/au88x0/au8830.c +++ b/sound/pci/au88x0/au8830.c @@ -1,6 +1,6 @@ #include "au8830.h" #include "au88x0.h" -static struct pci_device_id snd_vortex_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vortex_ids) = { {PCI_VDEVICE(AUREAL, PCI_DEVICE_ID_AUREAL_VORTEX_2), 0,}, {0,} }; diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index 4d34bb0..67921f9 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -164,7 +164,7 @@ MODULE_PARM_DESC(id, "ID string for the Audiowerk2 soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard."); -static struct pci_device_id snd_aw2_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_aw2_ids) = { {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0, 0, 0, 0}, {0} diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 69867ac..4679ed8 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -350,7 +350,7 @@ struct snd_azf3328 { #endif }; -static const struct pci_device_id snd_azf3328_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_azf3328_ids) = { { 0x122D, 0x50DC, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* PCI168/3328 */ { 0x122D, 0x80DA, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* 3328 */ { 0, } diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c index 4e2b925..37e1b5d 100644 --- a/sound/pci/bt87x.c +++ b/sound/pci/bt87x.c @@ -795,7 +795,7 @@ fail: .driver_data = SND_BT87X_BOARD_ ## id } /* driver_data is the card id for that device */ -static struct pci_device_id snd_bt87x_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_bt87x_ids) = { /* Hauppauge WinTV series */ BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0x13eb, GENERIC), /* Hauppauge WinTV series */ @@ -964,7 +964,7 @@ static void __devexit snd_bt87x_remove(struct pci_dev *pci) /* default entries for all Bt87x cards - it's not exported */ /* driver_data is set to 0 to call detection */ -static struct pci_device_id snd_bt87x_default_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(snd_bt87x_default_ids) = { BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN), BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, PCI_ANY_ID, PCI_ANY_ID, UNKNOWN), { } diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 15e4138..0a3d3d6 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -1875,7 +1875,7 @@ static int snd_ca0106_resume(struct pci_dev *pci) #endif // PCI IDs -static struct pci_device_id snd_ca0106_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ca0106_ids) = { { PCI_VDEVICE(CREATIVE, 0x0007), 0 }, /* Audigy LS or Live 24bit */ { 0, } }; diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index a312bae..1ded64e 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -2796,7 +2796,7 @@ static inline void snd_cmipci_proc_init(struct cmipci *cm) {} #endif -static struct pci_device_id snd_cmipci_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cmipci_ids) = { {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338A), 0}, {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8338B), 0}, {PCI_VDEVICE(CMEDIA, PCI_DEVICE_ID_CMEDIA_CM8738), 0}, @@ -3018,7 +3018,7 @@ static int __devinit snd_cmipci_create(struct snd_card *card, struct pci_dev *pc int integrated_midi = 0; char modelstr[16]; int pcm_index, pcm_spdif_index; - static struct pci_device_id intel_82437vx[] = { + static DEFINE_PCI_DEVICE_TABLE(intel_82437vx) = { { PCI_DEVICE(PCI_VENDOR_ID_INTEL, PCI_DEVICE_ID_INTEL_82437VX) }, { }, }; diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index e2e0359..9edc650 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -494,7 +494,7 @@ struct cs4281 { static irqreturn_t snd_cs4281_interrupt(int irq, void *dev_id); -static struct pci_device_id snd_cs4281_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cs4281_ids) = { { PCI_VDEVICE(CIRRUS, 0x6005), 0, }, /* CS4281 */ { 0, } }; diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c index 033aec4..767fa7f 100644 --- a/sound/pci/cs46xx/cs46xx.c +++ b/sound/pci/cs46xx/cs46xx.c @@ -64,7 +64,7 @@ MODULE_PARM_DESC(thinkpad, "Force to enable Thinkpad's CLKRUN control."); module_param_array(mmap_valid, bool, NULL, 0444); MODULE_PARM_DESC(mmap_valid, "Support OSS mmap."); -static struct pci_device_id snd_cs46xx_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cs46xx_ids) = { { PCI_VDEVICE(CIRRUS, 0x6001), 0, }, /* CS4280 */ { PCI_VDEVICE(CIRRUS, 0x6003), 0, }, /* CS4612 */ { PCI_VDEVICE(CIRRUS, 0x6004), 0, }, /* CS4615 */ diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index dc46432..207479a 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -58,7 +58,7 @@ struct snd_cs5530 { unsigned long pci_base; }; -static struct pci_device_id snd_cs5530_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cs5530_ids) = { {PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, PCI_ANY_ID, 0, 0}, {0,} diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index 91e7faf..afb8037 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -66,7 +66,7 @@ MODULE_PARM_DESC(id, "ID string for " DRIVER_NAME); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable " DRIVER_NAME); -static struct pci_device_id snd_cs5535audio_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_cs5535audio_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_NS, PCI_DEVICE_ID_NS_CS5535_AUDIO) }, { PCI_DEVICE(PCI_VENDOR_ID_AMD, PCI_DEVICE_ID_AMD_CS5536_AUDIO) }, {} diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index ed44ed7..f42e7e1 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -43,7 +43,7 @@ MODULE_PARM_DESC(enable, "Enable Creative X-Fi driver"); module_param_array(subsystem, int, NULL, 0444); MODULE_PARM_DESC(subsystem, "Override subsystem ID for Creative X-Fi driver"); -static struct pci_device_id ct_pci_dev_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(ct_pci_dev_ids) = { /* only X-Fi is supported, so... */ { PCI_DEVICE(PCI_VENDOR_ID_CREATIVE, PCI_DEVICE_ID_CREATIVE_20K1), .driver_data = ATC20K1, diff --git a/sound/pci/echoaudio/darla20.c b/sound/pci/echoaudio/darla20.c index 8c6db3a..a65bafe 100644 --- a/sound/pci/echoaudio/darla20.c +++ b/sound/pci/echoaudio/darla20.c @@ -63,7 +63,7 @@ static const struct firmware card_fw[] = { {0, "darla20_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0010, 0, 0, 0}, /* DSP 56301 Darla20 rev.0 */ {0,} }; diff --git a/sound/pci/echoaudio/darla24.c b/sound/pci/echoaudio/darla24.c index 04cbf3e..0a6c50b 100644 --- a/sound/pci/echoaudio/darla24.c +++ b/sound/pci/echoaudio/darla24.c @@ -67,7 +67,7 @@ static const struct firmware card_fw[] = { {0, "darla24_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0040, 0, 0, 0}, /* DSP 56301 Darla24 rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0041, 0, 0, 0}, /* DSP 56301 Darla24 rev.1 */ {0,} diff --git a/sound/pci/echoaudio/echo3g.c b/sound/pci/echoaudio/echo3g.c index 4022e43..f514279 100644 --- a/sound/pci/echoaudio/echo3g.c +++ b/sound/pci/echoaudio/echo3g.c @@ -81,7 +81,7 @@ static const struct firmware card_fw[] = { {0, "3g_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x0100, 0, 0, 0}, /* Echo 3G */ {0,} }; diff --git a/sound/pci/echoaudio/gina20.c b/sound/pci/echoaudio/gina20.c index c0e64b8..2364f8a 100644 --- a/sound/pci/echoaudio/gina20.c +++ b/sound/pci/echoaudio/gina20.c @@ -67,7 +67,7 @@ static const struct firmware card_fw[] = { {0, "gina20_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0020, 0, 0, 0}, /* DSP 56301 Gina20 rev.0 */ {0,} }; diff --git a/sound/pci/echoaudio/gina24.c b/sound/pci/echoaudio/gina24.c index c36a78d..616b558 100644 --- a/sound/pci/echoaudio/gina24.c +++ b/sound/pci/echoaudio/gina24.c @@ -85,7 +85,7 @@ static const struct firmware card_fw[] = { {0, "gina24_361_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0050, 0, 0, 0}, /* DSP 56301 Gina24 rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0051, 0, 0, 0}, /* DSP 56301 Gina24 rev.1 */ {0x1057, 0x3410, 0xECC0, 0x0050, 0, 0, 0}, /* DSP 56361 Gina24 rev.0 */ diff --git a/sound/pci/echoaudio/indigo.c b/sound/pci/echoaudio/indigo.c index 0a58a7c..776175c 100644 --- a/sound/pci/echoaudio/indigo.c +++ b/sound/pci/echoaudio/indigo.c @@ -68,7 +68,7 @@ static const struct firmware card_fw[] = { {0, "indigo_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x0090, 0, 0, 0}, /* Indigo */ {0,} }; diff --git a/sound/pci/echoaudio/indigodj.c b/sound/pci/echoaudio/indigodj.c index 2db24d2..8816b0b 100644 --- a/sound/pci/echoaudio/indigodj.c +++ b/sound/pci/echoaudio/indigodj.c @@ -68,7 +68,7 @@ static const struct firmware card_fw[] = { {0, "indigo_dj_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x00B0, 0, 0, 0}, /* Indigo DJ*/ {0,} }; diff --git a/sound/pci/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c index 2e44316..b1e3652 100644 --- a/sound/pci/echoaudio/indigodjx.c +++ b/sound/pci/echoaudio/indigodjx.c @@ -68,7 +68,7 @@ static const struct firmware card_fw[] = { {0, "indigo_djx_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x00E0, 0, 0, 0}, /* Indigo DJx*/ {0,} }; diff --git a/sound/pci/echoaudio/indigoio.c b/sound/pci/echoaudio/indigoio.c index a60c0a0..1035125 100644 --- a/sound/pci/echoaudio/indigoio.c +++ b/sound/pci/echoaudio/indigoio.c @@ -69,7 +69,7 @@ static const struct firmware card_fw[] = { {0, "indigo_io_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x00A0, 0, 0, 0}, /* Indigo IO*/ {0,} }; diff --git a/sound/pci/echoaudio/indigoiox.c b/sound/pci/echoaudio/indigoiox.c index eb3819f..60b7cb2 100644 --- a/sound/pci/echoaudio/indigoiox.c +++ b/sound/pci/echoaudio/indigoiox.c @@ -69,7 +69,7 @@ static const struct firmware card_fw[] = { {0, "indigo_iox_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x00D0, 0, 0, 0}, /* Indigo IOx */ {0,} }; diff --git a/sound/pci/echoaudio/layla20.c b/sound/pci/echoaudio/layla20.c index 5061946..8c3f5c5 100644 --- a/sound/pci/echoaudio/layla20.c +++ b/sound/pci/echoaudio/layla20.c @@ -76,7 +76,7 @@ static const struct firmware card_fw[] = { {0, "layla20_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0030, 0, 0, 0}, /* DSP 56301 Layla20 rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0031, 0, 0, 0}, /* DSP 56301 Layla20 rev.1 */ {0,} diff --git a/sound/pci/echoaudio/layla24.c b/sound/pci/echoaudio/layla24.c index e09e3ea..ed1cc0a 100644 --- a/sound/pci/echoaudio/layla24.c +++ b/sound/pci/echoaudio/layla24.c @@ -87,7 +87,7 @@ static const struct firmware card_fw[] = { {0, "layla24_2S_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x0060, 0, 0, 0}, /* DSP 56361 Layla24 rev.0 */ {0,} }; diff --git a/sound/pci/echoaudio/mia.c b/sound/pci/echoaudio/mia.c index f05c8c0..cc2bbfc 100644 --- a/sound/pci/echoaudio/mia.c +++ b/sound/pci/echoaudio/mia.c @@ -77,7 +77,7 @@ static const struct firmware card_fw[] = { {0, "mia_dsp.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x3410, 0xECC0, 0x0080, 0, 0, 0}, /* DSP 56361 Mia rev.0 */ {0x1057, 0x3410, 0xECC0, 0x0081, 0, 0, 0}, /* DSP 56361 Mia rev.1 */ {0,} diff --git a/sound/pci/echoaudio/mona.c b/sound/pci/echoaudio/mona.c index b05bad9..3e7e018 100644 --- a/sound/pci/echoaudio/mona.c +++ b/sound/pci/echoaudio/mona.c @@ -92,7 +92,7 @@ static const struct firmware card_fw[] = { {0, "mona_2_asic.fw"} }; -static struct pci_device_id snd_echo_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_echo_ids) = { {0x1057, 0x1801, 0xECC0, 0x0070, 0, 0, 0}, /* DSP 56301 Mona rev.0 */ {0x1057, 0x1801, 0xECC0, 0x0071, 0, 0, 0}, /* DSP 56301 Mona rev.1 */ {0x1057, 0x1801, 0xECC0, 0x0072, 0, 0, 0}, /* DSP 56301 Mona rev.2 */ diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c index 168af67..4203782 100644 --- a/sound/pci/emu10k1/emu10k1.c +++ b/sound/pci/emu10k1/emu10k1.c @@ -76,7 +76,7 @@ MODULE_PARM_DESC(subsystem, "Force card subsystem model."); /* * Class 0401: 1102:0008 (rev 00) Subsystem: 1102:1001 -> Audigy2 Value Model:SB0400 */ -static struct pci_device_id snd_emu10k1_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_emu10k1_ids) = { { PCI_VDEVICE(CREATIVE, 0x0002), 0 }, /* EMU10K1 */ { PCI_VDEVICE(CREATIVE, 0x0004), 1 }, /* Audigy */ { PCI_VDEVICE(CREATIVE, 0x0008), 1 }, /* Audigy 2 Value SB0400 */ diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 1d369ff..df47f73 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -1605,7 +1605,7 @@ static void __devexit snd_emu10k1x_remove(struct pci_dev *pci) } // PCI IDs -static struct pci_device_id snd_emu10k1x_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_emu10k1x_ids) = { { PCI_VDEVICE(CREATIVE, 0x0006), 0 }, /* Dell OEM version (EMU10K1) */ { 0, } }; diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 2b82c5c..c7fba53 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -443,7 +443,7 @@ struct ensoniq { static irqreturn_t snd_audiopci_interrupt(int irq, void *dev_id); -static struct pci_device_id snd_audiopci_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_audiopci_ids) = { #ifdef CHIP1370 { PCI_VDEVICE(ENSONIQ, 0x5000), 0, }, /* ES1370 */ #endif diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index fb83e1f..553b752 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -243,7 +243,7 @@ struct es1938 { static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id); -static struct pci_device_id snd_es1938_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_es1938_ids) = { { PCI_VDEVICE(ESS, 0x1969), 0, }, /* Solo-1 */ { 0, } }; diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index a11f453..ecaea9f 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -551,7 +551,7 @@ struct es1968 { static irqreturn_t snd_es1968_interrupt(int irq, void *dev_id); -static struct pci_device_id snd_es1968_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_es1968_ids) = { /* Maestro 1 */ { 0x1285, 0x0100, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, TYPE_MAESTRO }, /* Maestro 2 */ diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 83508b3..e1baad7 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -205,7 +205,7 @@ struct fm801 { #endif }; -static struct pci_device_id snd_fm801_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_fm801_ids) = { { 0x1319, 0x0801, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0, }, /* FM801 */ { 0x5213, 0x0510, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0, }, /* Gallant Odyssey Sound 4 */ { 0, } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1f516e6..ac05bef 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2664,7 +2664,7 @@ static void __devexit azx_remove(struct pci_dev *pci) } /* PCI IDs */ -static struct pci_device_id azx_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* ICH 6..10 */ { PCI_DEVICE(0x8086, 0x2668), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x27d8), .driver_data = AZX_DRIVER_ICH }, diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index fb61943..4fc6d8b 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -106,7 +106,7 @@ module_param_array(dxr_enable, int, NULL, 0444); MODULE_PARM_DESC(dxr_enable, "Enable DXR support for Terratec DMX6FIRE."); -static const struct pci_device_id snd_ice1712_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ice1712_ids) = { { PCI_VDEVICE(ICE, PCI_DEVICE_ID_ICE_1712), 0 }, /* ICE1712 */ { 0, } }; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index ae29073..c1498fa 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -94,7 +94,7 @@ MODULE_PARM_DESC(model, "Use the given board model."); /* Both VT1720 and VT1724 have the same PCI IDs */ -static const struct pci_device_id snd_vt1724_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vt1724_ids) = { { PCI_VDEVICE(ICE, PCI_DEVICE_ID_VT1724), 0 }, { 0, } }; diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index b990143..6433e65 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -420,7 +420,7 @@ struct intel8x0 { u32 int_sta_mask; /* interrupt status mask */ }; -static struct pci_device_id snd_intel8x0_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_intel8x0_ids) = { { PCI_VDEVICE(INTEL, 0x2415), DEVICE_INTEL }, /* 82801AA */ { PCI_VDEVICE(INTEL, 0x2425), DEVICE_INTEL }, /* 82901AB */ { PCI_VDEVICE(INTEL, 0x2445), DEVICE_INTEL }, /* 82801BA */ diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 9e7d12e..13cec1e 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -219,7 +219,7 @@ struct intel8x0m { unsigned int pcm_pos_shift; }; -static struct pci_device_id snd_intel8x0m_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_intel8x0m_ids) = { { PCI_VDEVICE(INTEL, 0x2416), DEVICE_INTEL }, /* 82801AA */ { PCI_VDEVICE(INTEL, 0x2426), DEVICE_INTEL }, /* 82901AB */ { PCI_VDEVICE(INTEL, 0x2446), DEVICE_INTEL }, /* 82801BA */ diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 7cc38a1..6d79570 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -418,7 +418,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Korg 1212 soundcard."); MODULE_AUTHOR("Haroldo Gamal "); -static struct pci_device_id snd_korg1212_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_korg1212_ids) = { { .vendor = 0x10b5, .device = 0x906d, diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 11b8c65..0cca560 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -55,7 +55,7 @@ static const char card_name[] = "LX6464ES"; #define PCI_DEVICE_ID_PLX_LX6464ES PCI_DEVICE_ID_PLX_9056 -static struct pci_device_id snd_lx6464es_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_lx6464es_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_PLX, PCI_DEVICE_ID_PLX_LX6464ES), .subvendor = PCI_VENDOR_ID_DIGIGRAM, .subdevice = PCI_SUBDEVICE_ID_DIGIGRAM_LX6464ES_SERIAL_SUBSYSTEM diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 75283fbb..b64e781 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -861,7 +861,7 @@ struct snd_m3 { /* * pci ids */ -static struct pci_device_id snd_m3_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_m3_ids) = { {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_ALLEGRO_1, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0}, {PCI_VENDOR_ID_ESS, PCI_DEVICE_ID_ESS_ALLEGRO, PCI_ANY_ID, PCI_ANY_ID, diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index a83d196..7e8e7da 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -60,7 +60,7 @@ MODULE_PARM_DESC(enable, "Enable Digigram " CARD_NAME " soundcard."); /* */ -static struct pci_device_id snd_mixart_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_mixart_ids) = { { PCI_VDEVICE(MOTOROLA, 0x0003), 0, }, /* MC8240 */ { 0, } }; diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 97a0731..5a60492 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -262,7 +262,7 @@ struct nm256 { /* * PCI ids */ -static struct pci_device_id snd_nm256_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_nm256_ids) = { {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256AV_AUDIO), 0}, {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256ZX_AUDIO), 0}, {PCI_VDEVICE(NEOMAGIC, PCI_DEVICE_ID_NEOMAGIC_NM256XL_PLUS_AUDIO), 0}, diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index e3c229b..5a87d68 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -48,7 +48,7 @@ MODULE_PARM_DESC(id, "ID string"); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "enable card"); -static struct pci_device_id hifier_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(hifier_ids) = { { OXYGEN_PCI_SUBID(0x14c3, 0x1710) }, { OXYGEN_PCI_SUBID(0x14c3, 0x1711) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index acbedeb..289cb4d 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -72,7 +72,7 @@ enum { MODEL_CLARO_HALO, /* HT-Omega Claro halo */ }; -static struct pci_device_id oxygen_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(oxygen_ids) = { { OXYGEN_PCI_SUBID(0x10b0, 0x0216), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x10b0, 0x0218), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x10b0, 0x0219), .driver_data = MODEL_CMEDIA_REF }, diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 563b6f5..f03a2f2 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -40,7 +40,7 @@ MODULE_PARM_DESC(id, "ID string"); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "enable card"); -static struct pci_device_id xonar_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x8269) }, { OXYGEN_PCI_SUBID(0x1043, 0x8275) }, { OXYGEN_PCI_SUBID(0x1043, 0x82b7) }, diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 833e9c7..95cfde2 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -94,7 +94,7 @@ enum { PCI_ID_LAST }; -static struct pci_device_id pcxhr_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(pcxhr_ids) = { { 0x10b5, 0x9656, 0x1369, 0xb001, 0, 0, PCI_ID_VX882HR, }, { 0x10b5, 0x9656, 0x1369, 0xb101, 0, 0, PCI_ID_PCX882HR, }, { 0x10b5, 0x9656, 0x1369, 0xb201, 0, 0, PCI_ID_VX881HR, }, diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index b5ca02e..bb08a28 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -506,7 +506,7 @@ static int riptide_reset(struct cmdif *cif, struct snd_riptide *chip); /* */ -static struct pci_device_id snd_riptide_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_riptide_ids) = { { PCI_DEVICE(0x127a, 0x4310) }, { PCI_DEVICE(0x127a, 0x4320) }, { PCI_DEVICE(0x127a, 0x4330) }, @@ -515,7 +515,7 @@ static struct pci_device_id snd_riptide_ids[] = { }; #ifdef SUPPORT_JOYSTICK -static struct pci_device_id snd_riptide_joystick_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(snd_riptide_joystick_ids) = { { PCI_DEVICE(0x127a, 0x4312) }, { PCI_DEVICE(0x127a, 0x4322) }, { PCI_DEVICE(0x127a, 0x4332) }, diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index f977dba..d5e1c6e 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -226,7 +226,7 @@ struct rme32 { struct snd_kcontrol *spdif_ctl; }; -static struct pci_device_id snd_rme32_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_rme32_ids) = { {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32), 0,}, {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_8), 0,}, {PCI_VDEVICE(XILINX_RME, PCI_DEVICE_ID_RME_DIGI32_PRO), 0,}, diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 2ba5c0f..9d5252b 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -231,7 +231,7 @@ struct rme96 { struct snd_kcontrol *spdif_ctl; }; -static struct pci_device_id snd_rme96_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_rme96_ids) = { { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96), 0, }, { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8), 0, }, { PCI_VDEVICE(XILINX, PCI_DEVICE_ID_RME_DIGI96_8_PRO), 0, }, diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 7bb827c..52c6eb5 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -585,7 +585,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d } -static struct pci_device_id snd_hdsp_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_hdsp_ids) = { { .vendor = PCI_VENDOR_ID_XILINX, .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index a1b10d1..3d72c1e 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -512,7 +512,7 @@ static char channel_map_madi_ss[HDSPM_MAX_CHANNELS] = { }; -static struct pci_device_id snd_hdspm_ids[] __devinitdata = { +static DEFINE_PCI_DEVICE_TABLE(snd_hdspm_ids) = { { .vendor = PCI_VENDOR_ID_XILINX, .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP_MADI, diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index bc539ab..44a3e2d 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -314,7 +314,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d } -static struct pci_device_id snd_rme9652_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_rme9652_ids) = { { .vendor = 0x10ee, .device = 0x3fc4, diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 1a5ff06..7e3e8fb 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -48,7 +48,7 @@ MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator."); module_param(enable, bool, 0444); MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); -static struct pci_device_id snd_sis7019_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_sis7019_ids) = { { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) }, { 0, } }; diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index 1f6406c..337b9fa 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -242,7 +242,7 @@ struct sonicvibes { #endif }; -static struct pci_device_id snd_sonic_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_sonic_ids) = { { PCI_VDEVICE(S3, 0xca00), 0, }, { 0, } }; diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c index 21cef97..6d05818 100644 --- a/sound/pci/trident/trident.c +++ b/sound/pci/trident/trident.c @@ -62,7 +62,7 @@ MODULE_PARM_DESC(pcm_channels, "Number of hardware channels assigned for PCM."); module_param_array(wavetable_size, int, NULL, 0444); MODULE_PARM_DESC(wavetable_size, "Maximum memory size in kB for wavetable synth."); -static struct pci_device_id snd_trident_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_trident_ids) = { {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_DX), PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0}, {PCI_DEVICE(PCI_VENDOR_ID_TRIDENT, PCI_DEVICE_ID_TRIDENT_4DWAVE_NX), diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index 8a332d2..9595b5b 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -401,7 +401,7 @@ struct via82xx { #endif }; -static struct pci_device_id snd_via82xx_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_via82xx_ids) = { /* 0x1106, 0x3058 */ { PCI_VDEVICE(VIA, PCI_DEVICE_ID_VIA_82C686_5), TYPE_CARD_VIA686, }, /* 686A */ /* 0x1106, 0x3059 */ diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index 47eb615..f7e8bbbe 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -260,7 +260,7 @@ struct via82xx_modem { struct snd_info_entry *proc_entry; }; -static struct pci_device_id snd_via82xx_modem_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_via82xx_modem_ids) = { { PCI_VDEVICE(VIA, 0x3068), TYPE_CARD_VIA82XX_MODEM, }, { 0, } }; diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c index fc9136c..99a9a81 100644 --- a/sound/pci/vx222/vx222.c +++ b/sound/pci/vx222/vx222.c @@ -60,7 +60,7 @@ enum { VX_PCI_VX222_NEW }; -static struct pci_device_id snd_vx222_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_vx222_ids) = { { 0x10b5, 0x9050, 0x1369, PCI_ANY_ID, 0, 0, VX_PCI_VX222_OLD, }, /* PLX */ { 0x10b5, 0x9030, 0x1369, PCI_ANY_ID, 0, 0, VX_PCI_VX222_NEW, }, /* PLX */ { 0, } diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c index e6b18b9..80c6821 100644 --- a/sound/pci/ymfpci/ymfpci.c +++ b/sound/pci/ymfpci/ymfpci.c @@ -66,7 +66,7 @@ MODULE_PARM_DESC(joystick_port, "Joystick port address"); module_param_array(rear_switch, bool, NULL, 0444); MODULE_PARM_DESC(rear_switch, "Enable shared rear/line-in switch"); -static struct pci_device_id snd_ymfpci_ids[] = { +static DEFINE_PCI_DEVICE_TABLE(snd_ymfpci_ids) = { { PCI_VDEVICE(YAMAHA, 0x0004), 0, }, /* YMF724 */ { PCI_VDEVICE(YAMAHA, 0x000d), 0, }, /* YMF724F */ { PCI_VDEVICE(YAMAHA, 0x000a), 0, }, /* YMF740 */ -- cgit v1.1 From c3a3e040f01457d2ea4f199f75ca205401001a3b Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Thu, 11 Feb 2010 17:50:44 +0100 Subject: ALSA: usbmixer - add possibility to remap dB values USB devices tends to represent dB ranges in different way than ALSA expects. Add possibility to override these values and add guessed values for SoundBlaster MP3+. Also rename 'Capture Input Source' control to 'Capture Source' for SoundBlaster MP3+ and Extigy. Signed-off-by: Jaroslav Kysela --- sound/usb/usbmixer.c | 125 ++++++++++++++++++++++++++++------------------ sound/usb/usbmixer_maps.c | 23 ++++++--- 2 files changed, 93 insertions(+), 55 deletions(-) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index c998220..c72ad0c 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -123,6 +123,7 @@ struct usb_mixer_elem_info { int channels; int val_type; int min, max, res; + int dBmin, dBmax; int cached; int cache_val[MAX_CHANNELS]; u8 initialized; @@ -194,42 +195,50 @@ enum { */ #include "usbmixer_maps.c" -/* get the mapped name if the unit matches */ -static int check_mapped_name(struct mixer_build *state, int unitid, int control, char *buf, int buflen) +static const struct usbmix_name_map * +find_map(struct mixer_build *state, int unitid, int control) { - const struct usbmix_name_map *p; + const struct usbmix_name_map *p = state->map; - if (! state->map) - return 0; + if (!p) + return NULL; for (p = state->map; p->id; p++) { - if (p->id == unitid && p->name && - (! control || ! p->control || control == p->control)) { - buflen--; - return strlcpy(buf, p->name, buflen); - } + if (p->id == unitid && + (!control || !p->control || control == p->control)) + return p; } - return 0; + return NULL; } -/* check whether the control should be ignored */ -static int check_ignored_ctl(struct mixer_build *state, int unitid, int control) +/* get the mapped name if the unit matches */ +static int +check_mapped_name(const struct usbmix_name_map *p, char *buf, int buflen) { - const struct usbmix_name_map *p; + if (!p || !p->name) + return 0; - if (! state->map) + buflen--; + return strlcpy(buf, p->name, buflen); +} + +/* check whether the control should be ignored */ +static inline int +check_ignored_ctl(const struct usbmix_name_map *p) +{ + if (!p || p->name || p->dB) return 0; - for (p = state->map; p->id; p++) { - if (p->id == unitid && ! p->name && - (! control || ! p->control || control == p->control)) { - /* - printk(KERN_DEBUG "ignored control %d:%d\n", - unitid, control); - */ - return 1; - } + return 1; +} + +/* dB mapping */ +static inline void check_mapped_dB(const struct usbmix_name_map *p, + struct usb_mixer_elem_info *cval) +{ + if (p && p->dB) { + cval->dBmin = p->dB->min; + cval->dBmax = p->dB->max; } - return 0; } /* get the mapped selector source name */ @@ -466,20 +475,8 @@ static int mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag, if (size < sizeof(scale)) return -ENOMEM; - /* USB descriptions contain the dB scale in 1/256 dB unit - * while ALSA TLV contains in 1/100 dB unit - */ - scale[2] = (convert_signed_value(cval, cval->min) * 100) / 256; - scale[3] = (convert_signed_value(cval, cval->max) * 100) / 256; - if (scale[3] <= scale[2]) { - /* something is wrong; assume it's either from/to 0dB */ - if (scale[2] < 0) - scale[3] = 0; - else if (scale[2] > 0) - scale[2] = 0; - else /* totally crap, return an error */ - return -EINVAL; - } + scale[2] = cval->dBmin; + scale[3] = cval->dBmax; if (copy_to_user(_tlv, scale, sizeof(scale))) return -EFAULT; return 0; @@ -720,6 +717,7 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) cval->min = default_min; cval->max = cval->min + 1; cval->res = 1; + cval->dBmin = cval->dBmax = 0; if (cval->val_type == USB_MIXER_BOOLEAN || cval->val_type == USB_MIXER_INV_BOOLEAN) { @@ -787,6 +785,24 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) cval->initialized = 1; } + + /* USB descriptions contain the dB scale in 1/256 dB unit + * while ALSA TLV contains in 1/100 dB unit + */ + cval->dBmin = (convert_signed_value(cval, cval->min) * 100) / 256; + cval->dBmax = (convert_signed_value(cval, cval->max) * 100) / 256; + if (cval->dBmin > cval->dBmax) { + /* something is wrong; assume it's either from/to 0dB */ + if (cval->dBmin < 0) + cval->dBmax = 0; + else if (cval->dBmin > 0) + cval->dBmin = 0; + if (cval->dBmin > cval->dBmax) { + /* totally crap, return an error */ + return -EINVAL; + } + } + return 0; } @@ -912,6 +928,7 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, int nameid = desc[desc[0] - 1]; struct snd_kcontrol *kctl; struct usb_mixer_elem_info *cval; + const struct usbmix_name_map *map; control++; /* change from zero-based to 1-based value */ @@ -920,7 +937,8 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, return; } - if (check_ignored_ctl(state, unitid, control)) + map = find_map(state, unitid, control); + if (check_ignored_ctl(map)) return; cval = kzalloc(sizeof(*cval), GFP_KERNEL); @@ -954,10 +972,11 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, } kctl->private_free = usb_mixer_elem_free; - len = check_mapped_name(state, unitid, control, kctl->id.name, sizeof(kctl->id.name)); + len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); mapped_name = len != 0; if (! len && nameid) - len = snd_usb_copy_string_desc(state, nameid, kctl->id.name, sizeof(kctl->id.name)); + len = snd_usb_copy_string_desc(state, nameid, + kctl->id.name, sizeof(kctl->id.name)); switch (control) { case USB_FEATURE_MUTE: @@ -995,6 +1014,7 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK; + check_mapped_dB(map, cval); } break; @@ -1122,8 +1142,10 @@ static void build_mixer_unit_ctl(struct mixer_build *state, unsigned char *desc, unsigned int num_outs = desc[5 + input_pins]; unsigned int i, len; struct snd_kcontrol *kctl; + const struct usbmix_name_map *map; - if (check_ignored_ctl(state, unitid, 0)) + map = find_map(state, unitid, 0); + if (check_ignored_ctl(map)) return; cval = kzalloc(sizeof(*cval), GFP_KERNEL); @@ -1152,7 +1174,7 @@ static void build_mixer_unit_ctl(struct mixer_build *state, unsigned char *desc, } kctl->private_free = usb_mixer_elem_free; - len = check_mapped_name(state, unitid, 0, kctl->id.name, sizeof(kctl->id.name)); + len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); if (! len) len = get_term_name(state, iterm, kctl->id.name, sizeof(kctl->id.name), 0); if (! len) @@ -1342,6 +1364,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned int i, err, nameid, type, len; struct procunit_info *info; struct procunit_value_info *valinfo; + const struct usbmix_name_map *map; static struct procunit_value_info default_value_info[] = { { 0x01, "Switch", USB_MIXER_BOOLEAN }, { 0 } @@ -1371,7 +1394,8 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned /* FIXME: bitmap might be longer than 8bit */ if (! (dsc[12 + num_ins] & (1 << (valinfo->control - 1)))) continue; - if (check_ignored_ctl(state, unitid, valinfo->control)) + map = find_map(state, unitid, valinfo->control); + if (check_ignored_ctl(map)) continue; cval = kzalloc(sizeof(*cval), GFP_KERNEL); if (! cval) { @@ -1402,8 +1426,9 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, unsigned } kctl->private_free = usb_mixer_elem_free; - if (check_mapped_name(state, unitid, cval->control, kctl->id.name, sizeof(kctl->id.name))) - ; + if (check_mapped_name(map, kctl->id.name, + sizeof(kctl->id.name))) + /* nothing */ ; else if (info->name) strlcpy(kctl->id.name, info->name, sizeof(kctl->id.name)); else { @@ -1542,6 +1567,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi int err; struct usb_mixer_elem_info *cval; struct snd_kcontrol *kctl; + const struct usbmix_name_map *map; char **namelist; if (! num_ins || desc[0] < 5 + num_ins) { @@ -1557,7 +1583,8 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi if (num_ins == 1) /* only one ? nonsense! */ return 0; - if (check_ignored_ctl(state, unitid, 0)) + map = find_map(state, unitid, 0); + if (check_ignored_ctl(map)) return 0; cval = kzalloc(sizeof(*cval), GFP_KERNEL); @@ -1612,7 +1639,7 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, unsi kctl->private_free = usb_mixer_selector_elem_free; nameid = desc[desc[0] - 1]; - len = check_mapped_name(state, unitid, 0, kctl->id.name, sizeof(kctl->id.name)); + len = check_mapped_name(map, kctl->id.name, sizeof(kctl->id.name)); if (len) ; else if (nameid) diff --git a/sound/usb/usbmixer_maps.c b/sound/usb/usbmixer_maps.c index 77c3588..79e903a 100644 --- a/sound/usb/usbmixer_maps.c +++ b/sound/usb/usbmixer_maps.c @@ -19,11 +19,16 @@ * */ +struct usbmix_dB_map { + u32 min; + u32 max; +}; struct usbmix_name_map { int id; const char *name; int control; + struct usbmix_dB_map *dB; }; struct usbmix_selector_map { @@ -72,7 +77,7 @@ static struct usbmix_name_map extigy_map[] = { { 8, "Line Playback" }, /* FU */ /* 9: IT mic */ { 10, "Mic Playback" }, /* FU */ - { 11, "Capture Input Source" }, /* SU */ + { 11, "Capture Source" }, /* SU */ { 12, "Capture" }, /* FU */ /* 13: OT pcm capture */ /* 14: MU (w/o controls) */ @@ -102,6 +107,9 @@ static struct usbmix_name_map extigy_map[] = { * e.g. no Master and fake PCM volume * Pavel Mihaylov */ +static struct usbmix_dB_map mp3plus_dB_1 = {-4781, 0}; /* just guess */ +static struct usbmix_dB_map mp3plus_dB_2 = {-1781, 618}; /* just guess */ + static struct usbmix_name_map mp3plus_map[] = { /* 1: IT pcm */ /* 2: IT mic */ @@ -110,16 +118,19 @@ static struct usbmix_name_map mp3plus_map[] = { /* 5: OT digital out */ /* 6: OT speaker */ /* 7: OT pcm capture */ - { 8, "Capture Input Source" }, /* FU, default PCM Capture Source */ + { 8, "Capture Source" }, /* FU, default PCM Capture Source */ /* (Mic, Input 1 = Line input, Input 2 = Optical input) */ { 9, "Master Playback" }, /* FU, default Speaker 1 */ /* { 10, "Mic Capture", 1 }, */ /* FU, Mic Capture */ - /* { 10, "Mic Capture", 2 }, */ /* FU, Mic Capture */ + { 10, /* "Mic Capture", */ NULL, 2, .dB = &mp3plus_dB_2 }, + /* FU, Mic Capture */ { 10, "Mic Boost", 7 }, /* FU, default Auto Gain Input */ - { 11, "Line Capture" }, /* FU, default PCM Capture */ + { 11, "Line Capture", .dB = &mp3plus_dB_2 }, + /* FU, default PCM Capture */ { 12, "Digital In Playback" }, /* FU, default PCM 1 */ - /* { 13, "Mic Playback" }, */ /* FU, default Mic Playback */ - { 14, "Line Playback" }, /* FU, default Speaker */ + { 13, /* "Mic Playback", */ .dB = &mp3plus_dB_1 }, + /* FU, default Mic Playback */ + { 14, "Line Playback", .dB = &mp3plus_dB_1 }, /* FU, default Speaker */ /* 15: MU */ { 0 } /* terminator */ }; -- cgit v1.1 From 19b50063780953563e3c3a2867c39aad7b9e64cf Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Sun, 14 Feb 2010 18:15:34 +0100 Subject: ALSA: Echoaudio - Add firmware cache #1 Changes the way the firmware is passed through functions. When CONFIG_PM is enabled the firmware cannot be released because the driver will need it again to resume the card. With this patch the firmware is passed as an index of the struct firmware card_fw[] in place of a pointer. That same index is then used to locate the firmware in the firmware cache. Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/darla20_dsp.c | 2 +- sound/pci/echoaudio/darla24_dsp.c | 2 +- sound/pci/echoaudio/echo3g_dsp.c | 2 +- sound/pci/echoaudio/echoaudio.c | 8 +++++++- sound/pci/echoaudio/echoaudio.h | 6 +++--- sound/pci/echoaudio/echoaudio_3g.c | 5 ++--- sound/pci/echoaudio/echoaudio_dsp.c | 12 +++++++----- sound/pci/echoaudio/gina20_dsp.c | 2 +- sound/pci/echoaudio/gina24_dsp.c | 18 ++++++++--------- sound/pci/echoaudio/indigo_dsp.c | 2 +- sound/pci/echoaudio/indigodj_dsp.c | 2 +- sound/pci/echoaudio/indigodjx_dsp.c | 2 +- sound/pci/echoaudio/indigoio_dsp.c | 2 +- sound/pci/echoaudio/indigoiox_dsp.c | 2 +- sound/pci/echoaudio/layla20_dsp.c | 7 +++---- sound/pci/echoaudio/layla24_dsp.c | 19 +++++++++--------- sound/pci/echoaudio/mia_dsp.c | 2 +- sound/pci/echoaudio/mona_dsp.c | 39 ++++++++++++++++++------------------- 18 files changed, 69 insertions(+), 65 deletions(-) diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c index 2904330..a44135d 100644 --- a/sound/pci/echoaudio/darla20_dsp.c +++ b/sound/pci/echoaudio/darla20_dsp.c @@ -45,7 +45,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_DARLA20_DSP]; + chip->dsp_code_to_load = FW_DARLA20_DSP; chip->spdif_status = GD_SPDIF_STATUS_UNDEF; chip->clock_state = GD_CLOCK_UNDEF; /* Since this card has no ASIC, mark it as loaded so everything diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c index 6022873..d681da1 100644 --- a/sound/pci/echoaudio/darla24_dsp.c +++ b/sound/pci/echoaudio/darla24_dsp.c @@ -45,7 +45,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_DARLA24_DSP]; + chip->dsp_code_to_load = FW_DARLA24_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index 57967e5..f007193 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -61,7 +61,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; chip->has_midi = TRUE; - chip->dsp_code_to_load = &card_fw[FW_ECHO3G_DSP]; + chip->dsp_code_to_load = FW_ECHO3G_DSP; /* Load the DSP code and the ASIC on the PCI card and get what type of external box is attached */ diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 1305f7c..78fc263 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -36,11 +36,15 @@ MODULE_PARM_DESC(enable, "Enable " ECHOCARD_NAME " soundcard."); static unsigned int channels_list[10] = {1, 2, 4, 6, 8, 10, 12, 14, 16, 999999}; static const DECLARE_TLV_DB_SCALE(db_scale_output_gain, -12800, 100, 1); + + static int get_firmware(const struct firmware **fw_entry, - const struct firmware *frm, struct echoaudio *chip) + struct echoaudio *chip, const short fw_index) { int err; char name[30]; + const struct firmware *frm = &card_fw[fw_index]; + DE_ACT(("firmware requested: %s\n", frm->data)); snprintf(name, sizeof(name), "ea/%s", frm->data); if ((err = request_firmware(fw_entry, name, pci_device(chip))) < 0) @@ -48,6 +52,8 @@ static int get_firmware(const struct firmware **fw_entry, return err; } + + static void free_firmware(const struct firmware *fw_entry) { release_firmware(fw_entry); diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index f9490ae..be76ef3 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -442,8 +442,8 @@ struct echoaudio { u16 device_id, subdevice_id; u16 *dsp_code; /* Current DSP code loaded, * NULL if nothing loaded */ - const struct firmware *dsp_code_to_load;/* DSP code to load */ - const struct firmware *asic_code; /* Current ASIC code */ + short dsp_code_to_load; /* DSP code to load */ + short asic_code; /* Current ASIC code */ u32 comm_page_phys; /* Physical address of the * memory seen by DSP */ volatile u32 __iomem *dsp_registers; /* DSP's register base */ @@ -464,7 +464,7 @@ static int load_firmware(struct echoaudio *chip); static int wait_handshake(struct echoaudio *chip); static int send_vector(struct echoaudio *chip, u32 command); static int get_firmware(const struct firmware **fw_entry, - const struct firmware *frm, struct echoaudio *chip); + struct echoaudio *chip, const short fw_index); static void free_firmware(const struct firmware *fw_entry); #ifdef ECHOCARD_HAS_MIDI diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index e32a748..658db44 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -227,12 +227,11 @@ static int load_asic(struct echoaudio *chip) /* Give the DSP a few milliseconds to settle down */ mdelay(2); - err = load_asic_generic(chip, DSP_FNC_LOAD_3G_ASIC, - &card_fw[FW_3G_ASIC]); + err = load_asic_generic(chip, DSP_FNC_LOAD_3G_ASIC, FW_3G_ASIC); if (err < 0) return err; - chip->asic_code = &card_fw[FW_3G_ASIC]; + chip->asic_code = FW_3G_ASIC; /* Now give the new ASIC some time to set up */ msleep(1000); diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 4df51ef..031ef7e 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -175,15 +175,15 @@ static inline int check_asic_status(struct echoaudio *chip) #ifdef ECHOCARD_HAS_ASIC /* Load ASIC code - done after the DSP is loaded */ -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic) +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic) { const struct firmware *fw; int err; u32 i, size; u8 *code; - if ((err = get_firmware(&fw, asic, chip)) < 0) { + err = get_firmware(&fw, chip, asic); + if (err < 0) { snd_printk(KERN_WARNING "Firmware not found !\n"); return err; } @@ -245,7 +245,8 @@ static int install_resident_loader(struct echoaudio *chip) return 0; } - if ((i = get_firmware(&fw, &card_fw[FW_361_LOADER], chip)) < 0) { + i = get_firmware(&fw, chip, FW_361_LOADER); + if (i < 0) { snd_printk(KERN_WARNING "Firmware not found !\n"); return i; } @@ -485,7 +486,8 @@ static int load_firmware(struct echoaudio *chip) chip->dsp_code = NULL; } - if ((err = get_firmware(&fw, chip->dsp_code_to_load, chip)) < 0) + err = get_firmware(&fw, chip, chip->dsp_code_to_load); + if (err < 0) return err; err = load_dsp(chip, (u16 *)fw->data); free_firmware(fw); diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index 3f1e747..c5de88b 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -49,7 +49,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_GINA20_DSP]; + chip->dsp_code_to_load = FW_GINA20_DSP; chip->spdif_status = GD_SPDIF_STATUS_UNDEF; chip->clock_state = GD_CLOCK_UNDEF; /* Since this card has no ASIC, mark it as loaded so everything diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c index 2fef37a..093dd7b 100644 --- a/sound/pci/echoaudio/gina24_dsp.c +++ b/sound/pci/echoaudio/gina24_dsp.c @@ -33,8 +33,7 @@ static int write_control_reg(struct echoaudio *chip, u32 value, char force); static int set_input_clock(struct echoaudio *chip, u16 clock); static int set_professional_spdif(struct echoaudio *chip, char prof); static int set_digital_mode(struct echoaudio *chip, u8 mode); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); @@ -64,13 +63,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) /* Gina24 comes in both '301 and '361 flavors */ if (chip->device_id == DEVICE_ID_56361) { - chip->dsp_code_to_load = &card_fw[FW_GINA24_361_DSP]; + chip->dsp_code_to_load = FW_GINA24_361_DSP; chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; } else { - chip->dsp_code_to_load = &card_fw[FW_GINA24_301_DSP]; + chip->dsp_code_to_load = FW_GINA24_301_DSP; chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | @@ -125,7 +124,7 @@ static int load_asic(struct echoaudio *chip) { u32 control_reg; int err; - const struct firmware *fw; + short asic; if (chip->asic_loaded) return 1; @@ -135,14 +134,15 @@ static int load_asic(struct echoaudio *chip) /* Pick the correct ASIC for '301 or '361 Gina24 */ if (chip->device_id == DEVICE_ID_56361) - fw = &card_fw[FW_GINA24_361_ASIC]; + asic = FW_GINA24_361_ASIC; else - fw = &card_fw[FW_GINA24_301_ASIC]; + asic = FW_GINA24_301_ASIC; - if ((err = load_asic_generic(chip, DSP_FNC_LOAD_GINA24_ASIC, fw)) < 0) + err = load_asic_generic(chip, DSP_FNC_LOAD_GINA24_ASIC, asic); + if (err < 0) return err; - chip->asic_code = fw; + chip->asic_code = asic; /* Now give the new ASIC a little time to set up */ mdelay(10); diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index 0b2cd9c..8799d2e 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -50,7 +50,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DSP]; + chip->dsp_code_to_load = FW_INDIGO_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index 0839291..cb1c92c 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -50,7 +50,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJ_DSP]; + chip->dsp_code_to_load = FW_INDIGO_DJ_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c index f591fc2..91dbfeb 100644 --- a/sound/pci/echoaudio/indigodjx_dsp.c +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -48,7 +48,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJX_DSP]; + chip->dsp_code_to_load = FW_INDIGO_DJX_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index 0604c8a..134e783 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -50,7 +50,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_IO_DSP]; + chip->dsp_code_to_load = FW_INDIGO_IO_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c index f357521..766cf50 100644 --- a/sound/pci/echoaudio/indigoiox_dsp.c +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -48,7 +48,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_INDIGO_IOX_DSP]; + chip->dsp_code_to_load = FW_INDIGO_IOX_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index 83750e9..07f3245 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -31,8 +31,7 @@ static int read_dsp(struct echoaudio *chip, u32 *data); static int set_professional_spdif(struct echoaudio *chip, char prof); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); static int update_flags(struct echoaudio *chip); @@ -54,7 +53,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; chip->has_midi = TRUE; - chip->dsp_code_to_load = &card_fw[FW_LAYLA20_DSP]; + chip->dsp_code_to_load = FW_LAYLA20_DSP; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_SUPER; @@ -144,7 +143,7 @@ static int load_asic(struct echoaudio *chip) return 0; err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA_ASIC, - &card_fw[FW_LAYLA20_ASIC]); + FW_LAYLA20_ASIC); if (err < 0) return err; diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c index d61b5cb..12dc00a 100644 --- a/sound/pci/echoaudio/layla24_dsp.c +++ b/sound/pci/echoaudio/layla24_dsp.c @@ -32,8 +32,7 @@ static int write_control_reg(struct echoaudio *chip, u32 value, char force); static int set_input_clock(struct echoaudio *chip, u16 clock); static int set_professional_spdif(struct echoaudio *chip, char prof); static int set_digital_mode(struct echoaudio *chip, u8 mode); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); @@ -54,7 +53,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; chip->has_midi = TRUE; - chip->dsp_code_to_load = &card_fw[FW_LAYLA24_DSP]; + chip->dsp_code_to_load = FW_LAYLA24_DSP; chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_WORD | ECHO_CLOCK_BIT_ADAT; @@ -123,18 +122,18 @@ static int load_asic(struct echoaudio *chip) /* Load the ASIC for the PCI card */ err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_PCI_CARD_ASIC, - &card_fw[FW_LAYLA24_1_ASIC]); + FW_LAYLA24_1_ASIC); if (err < 0) return err; - chip->asic_code = &card_fw[FW_LAYLA24_2S_ASIC]; + chip->asic_code = FW_LAYLA24_2S_ASIC; /* Now give the new ASIC a little time to set up */ mdelay(10); /* Do the external one */ err = load_asic_generic(chip, DSP_FNC_LOAD_LAYLA24_EXTERNAL_ASIC, - &card_fw[FW_LAYLA24_2S_ASIC]); + FW_LAYLA24_2S_ASIC); if (err < 0) return FALSE; @@ -299,7 +298,7 @@ static int set_input_clock(struct echoaudio *chip, u16 clock) /* Depending on what digital mode you want, Layla24 needs different ASICs loaded. This function checks the ASIC needed for the new mode and sees if it matches the one already loaded. */ -static int switch_asic(struct echoaudio *chip, const struct firmware *asic) +static int switch_asic(struct echoaudio *chip, short asic) { s8 *monitors; @@ -335,7 +334,7 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) { u32 control_reg; int err, incompatible_clock; - const struct firmware *asic; + short asic; /* Set clock to "internal" if it's not compatible with the new mode */ incompatible_clock = FALSE; @@ -344,12 +343,12 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) case DIGITAL_MODE_SPDIF_RCA: if (chip->input_clock == ECHO_CLOCK_ADAT) incompatible_clock = TRUE; - asic = &card_fw[FW_LAYLA24_2S_ASIC]; + asic = FW_LAYLA24_2S_ASIC; break; case DIGITAL_MODE_ADAT: if (chip->input_clock == ECHO_CLOCK_SPDIF) incompatible_clock = TRUE; - asic = &card_fw[FW_LAYLA24_2A_ASIC]; + asic = FW_LAYLA24_2A_ASIC; break; default: DE_ACT(("Digital mode not supported: %d\n", mode)); diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c index 5514051..d0302f2 100644 --- a/sound/pci/echoaudio/mia_dsp.c +++ b/sound/pci/echoaudio/mia_dsp.c @@ -53,7 +53,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; - chip->dsp_code_to_load = &card_fw[FW_MIA_DSP]; + chip->dsp_code_to_load = FW_MIA_DSP; /* Since this card has no ASIC, mark it as loaded so everything works OK */ chip->asic_loaded = TRUE; diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c index eaa619b..b28b8e4 100644 --- a/sound/pci/echoaudio/mona_dsp.c +++ b/sound/pci/echoaudio/mona_dsp.c @@ -33,8 +33,7 @@ static int write_control_reg(struct echoaudio *chip, u32 value, char force); static int set_input_clock(struct echoaudio *chip, u16 clock); static int set_professional_spdif(struct echoaudio *chip, char prof); static int set_digital_mode(struct echoaudio *chip, u8 mode); -static int load_asic_generic(struct echoaudio *chip, u32 cmd, - const struct firmware *asic); +static int load_asic_generic(struct echoaudio *chip, u32 cmd, short asic); static int check_asic_status(struct echoaudio *chip); @@ -64,9 +63,9 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) /* Mona comes in both '301 and '361 flavors */ if (chip->device_id == DEVICE_ID_56361) - chip->dsp_code_to_load = &card_fw[FW_MONA_361_DSP]; + chip->dsp_code_to_load = FW_MONA_361_DSP; else - chip->dsp_code_to_load = &card_fw[FW_MONA_301_DSP]; + chip->dsp_code_to_load = FW_MONA_301_DSP; chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; chip->professional_spdif = FALSE; @@ -120,7 +119,7 @@ static int load_asic(struct echoaudio *chip) { u32 control_reg; int err; - const struct firmware *asic; + short asic; if (chip->asic_loaded) return 0; @@ -128,9 +127,9 @@ static int load_asic(struct echoaudio *chip) mdelay(10); if (chip->device_id == DEVICE_ID_56361) - asic = &card_fw[FW_MONA_361_1_ASIC48]; + asic = FW_MONA_361_1_ASIC48; else - asic = &card_fw[FW_MONA_301_1_ASIC48]; + asic = FW_MONA_301_1_ASIC48; err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_PCI_CARD_ASIC, asic); if (err < 0) @@ -141,7 +140,7 @@ static int load_asic(struct echoaudio *chip) /* Do the external one */ err = load_asic_generic(chip, DSP_FNC_LOAD_MONA_EXTERNAL_ASIC, - &card_fw[FW_MONA_2_ASIC]); + FW_MONA_2_ASIC); if (err < 0) return err; @@ -165,22 +164,22 @@ loaded. This function checks the ASIC needed for the new mode and sees if it matches the one already loaded. */ static int switch_asic(struct echoaudio *chip, char double_speed) { - const struct firmware *asic; int err; + short asic; /* Check the clock detect bits to see if this is a single-speed clock or a double-speed clock; load a new ASIC if necessary. */ if (chip->device_id == DEVICE_ID_56361) { if (double_speed) - asic = &card_fw[FW_MONA_361_1_ASIC96]; + asic = FW_MONA_361_1_ASIC96; else - asic = &card_fw[FW_MONA_361_1_ASIC48]; + asic = FW_MONA_361_1_ASIC48; } else { if (double_speed) - asic = &card_fw[FW_MONA_301_1_ASIC96]; + asic = FW_MONA_301_1_ASIC96; else - asic = &card_fw[FW_MONA_301_1_ASIC48]; + asic = FW_MONA_301_1_ASIC48; } if (asic != chip->asic_code) { @@ -200,7 +199,7 @@ static int switch_asic(struct echoaudio *chip, char double_speed) static int set_sample_rate(struct echoaudio *chip, u32 rate) { u32 control_reg, clock; - const struct firmware *asic; + short asic; char force_write; /* Only set the clock for internal mode. */ @@ -218,14 +217,14 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) if (chip->digital_mode == DIGITAL_MODE_ADAT) return -EINVAL; if (chip->device_id == DEVICE_ID_56361) - asic = &card_fw[FW_MONA_361_1_ASIC96]; + asic = FW_MONA_361_1_ASIC96; else - asic = &card_fw[FW_MONA_301_1_ASIC96]; + asic = FW_MONA_301_1_ASIC96; } else { if (chip->device_id == DEVICE_ID_56361) - asic = &card_fw[FW_MONA_361_1_ASIC48]; + asic = FW_MONA_361_1_ASIC48; else - asic = &card_fw[FW_MONA_301_1_ASIC48]; + asic = FW_MONA_301_1_ASIC48; } force_write = 0; @@ -410,8 +409,8 @@ static int dsp_set_digital_mode(struct echoaudio *chip, u8 mode) case DIGITAL_MODE_ADAT: /* If the current ASIC is the 96KHz ASIC, switch the ASIC and set to 48 KHz */ - if (chip->asic_code == &card_fw[FW_MONA_361_1_ASIC96] || - chip->asic_code == &card_fw[FW_MONA_301_1_ASIC96]) { + if (chip->asic_code == FW_MONA_361_1_ASIC96 || + chip->asic_code == FW_MONA_301_1_ASIC96) { set_sample_rate(chip, 48000); } control_reg |= GML_ADAT_MODE; -- cgit v1.1 From 4f8ada444cc7a7ea70cdc81f098b34c5f1f2df41 Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Sun, 14 Feb 2010 18:15:51 +0100 Subject: ALSA: Echoaudio - Add firmware cache #2 This patch implements a simple cache for the firmware files when CONFIG_PM is defined. This patch changes get_firmware(), free_firmware() and adds free_firmware_cache(). The first two functions implement a very simple cache and the latter is used to actually release all the stored firmwares when the module is unloaded. When CONFIG_PM is not enabled those functions act as before, that is free_firmware() releases the firmware immediately and free_firmware_cache() does nothing. Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio.c | 42 +++++++++++++++++++++++++++++++++++++---- sound/pci/echoaudio/echoaudio.h | 3 +++ 2 files changed, 41 insertions(+), 4 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 78fc263..79dde95 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -43,12 +43,24 @@ static int get_firmware(const struct firmware **fw_entry, { int err; char name[30]; - const struct firmware *frm = &card_fw[fw_index]; - DE_ACT(("firmware requested: %s\n", frm->data)); - snprintf(name, sizeof(name), "ea/%s", frm->data); - if ((err = request_firmware(fw_entry, name, pci_device(chip))) < 0) +#ifdef CONFIG_PM + if (chip->fw_cache[fw_index]) { + DE_ACT(("firmware requested: %s is cached\n", card_fw[fw_index].data)); + *fw_entry = chip->fw_cache[fw_index]; + return 0; + } +#endif + + DE_ACT(("firmware requested: %s\n", card_fw[fw_index].data)); + snprintf(name, sizeof(name), "ea/%s", card_fw[fw_index].data); + err = request_firmware(fw_entry, name, pci_device(chip)); + if (err < 0) snd_printk(KERN_ERR "get_firmware(): Firmware not available (%d)\n", err); +#ifdef CONFIG_PM + else + chip->fw_cache[fw_index] = *fw_entry; +#endif return err; } @@ -56,8 +68,29 @@ static int get_firmware(const struct firmware **fw_entry, static void free_firmware(const struct firmware *fw_entry) { +#ifdef CONFIG_PM + DE_ACT(("firmware not released (kept in cache)\n")); +#else release_firmware(fw_entry); DE_ACT(("firmware released\n")); +#endif +} + + + +static void free_firmware_cache(struct echoaudio *chip) +{ +#ifdef CONFIG_PM + int i; + + for (i = 0; i < 8 ; i++) + if (chip->fw_cache[i]) { + release_firmware(chip->fw_cache[i]); + DE_ACT(("release_firmware(%d)\n", i)); + } + + DE_ACT(("firmware_cache released\n")); +#endif } @@ -1880,6 +1913,7 @@ static int snd_echo_free(struct echoaudio *chip) pci_disable_device(chip->pci); /* release chip data */ + free_firmware_cache(chip); kfree(chip); DE_INIT(("Chip freed.\n")); return 0; diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index be76ef3..a84c0d1 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -449,6 +449,9 @@ struct echoaudio { volatile u32 __iomem *dsp_registers; /* DSP's register base */ u32 active_mask; /* Chs. active mask or * punks out */ +#ifdef CONFIG_PM + const struct firmware *fw_cache[8]; /* Cached firmwares */ +#endif #ifdef ECHOCARD_HAS_MIDI u16 mtc_state; /* State for MIDI input parsing state machine */ -- cgit v1.1 From ad3499f4668f684ef6e5d0222ae14d5e4ade1fdd Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Sun, 14 Feb 2010 18:15:59 +0100 Subject: ALSA: Echoaudio - Add suspend support #1 Move the controls init code outside the init_hw() function because is must not be called during resume. This patch moves the code that initializes the card's controls with default valued from the init_hw() function into a separated set_mixer_defaults() function (one for each of the 16 supported cards). This change is necessary because during resume we must resurrect the hardware without losing the previous settings. set_mixer_defaults() must be called only once when the module is loaded. Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/darla20_dsp.c | 10 +++++++--- sound/pci/echoaudio/darla24_dsp.c | 10 +++++++--- sound/pci/echoaudio/echo3g_dsp.c | 26 ++++++++++++-------------- sound/pci/echoaudio/gina20_dsp.c | 13 ++++++++----- sound/pci/echoaudio/gina24_dsp.c | 20 ++++++++++---------- sound/pci/echoaudio/indigo_dsp.c | 10 +++++++--- sound/pci/echoaudio/indigo_express_dsp.c | 1 + sound/pci/echoaudio/indigodj_dsp.c | 10 +++++++--- sound/pci/echoaudio/indigodjx_dsp.c | 11 +++++++---- sound/pci/echoaudio/indigoio_dsp.c | 10 +++++++--- sound/pci/echoaudio/indigoiox_dsp.c | 11 +++++++---- sound/pci/echoaudio/layla20_dsp.c | 13 ++++++++----- sound/pci/echoaudio/layla24_dsp.c | 18 ++++++++++-------- sound/pci/echoaudio/mia_dsp.c | 10 +++++++--- sound/pci/echoaudio/mona_dsp.c | 22 ++++++++++------------ 15 files changed, 115 insertions(+), 80 deletions(-) diff --git a/sound/pci/echoaudio/darla20_dsp.c b/sound/pci/echoaudio/darla20_dsp.c index a44135d..20c7cbc 100644 --- a/sound/pci/echoaudio/darla20_dsp.c +++ b/sound/pci/echoaudio/darla20_dsp.c @@ -57,15 +57,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + /* The Darla20 has no external clock sources */ static u32 detect_input_clocks(const struct echoaudio *chip) { diff --git a/sound/pci/echoaudio/darla24_dsp.c b/sound/pci/echoaudio/darla24_dsp.c index d681da1..6da6663 100644 --- a/sound/pci/echoaudio/darla24_dsp.c +++ b/sound/pci/echoaudio/darla24_dsp.c @@ -56,15 +56,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index f007193..3cdc2ee 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -97,20 +97,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) chip->digital_modes = ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->non_audio_spdif = FALSE; - chip->bad_board = FALSE; - - if ((err = init_line_levels(chip)) < 0) - return err; - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_phantom_power(chip, 0); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); DE_INIT(("init_hw done\n")); return err; @@ -118,6 +104,18 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->non_audio_spdif = FALSE; + chip->bad_board = FALSE; + chip->phantom_power = FALSE; + return init_line_levels(chip); +} + + + static int set_phantom_power(struct echoaudio *chip, char on) { u32 control_reg = le32_to_cpu(chip->comm_page->control_register); diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index c5de88b..d1615a0 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -62,17 +62,20 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->professional_spdif = FALSE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff --git a/sound/pci/echoaudio/gina24_dsp.c b/sound/pci/echoaudio/gina24_dsp.c index 093dd7b..98f7cfa 100644 --- a/sound/pci/echoaudio/gina24_dsp.c +++ b/sound/pci/echoaudio/gina24_dsp.c @@ -57,9 +57,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) ECHO_CLOCK_BIT_INTERNAL | ECHO_CLOCK_BIT_SPDIF | ECHO_CLOCK_BIT_ESYNC | ECHO_CLOCK_BIT_ESYNC96 | ECHO_CLOCK_BIT_ADAT; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; /* Gina24 comes in both '301 and '361 flavors */ if (chip->device_id == DEVICE_ID_56361) { @@ -81,19 +78,22 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->digital_in_automute = TRUE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index 8799d2e..5e85f14 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -60,15 +60,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { return ECHO_CLOCK_BIT_INTERNAL; diff --git a/sound/pci/echoaudio/indigo_express_dsp.c b/sound/pci/echoaudio/indigo_express_dsp.c index 9ab625e..2e4ab3e 100644 --- a/sound/pci/echoaudio/indigo_express_dsp.c +++ b/sound/pci/echoaudio/indigo_express_dsp.c @@ -61,6 +61,7 @@ static int set_sample_rate(struct echoaudio *chip, u32 rate) control_reg |= clock; if (control_reg != old_control_reg) { + DE_ACT(("set_sample_rate: %d clock %d\n", rate, clock)); chip->comm_page->control_register = cpu_to_le32(control_reg); chip->sample_rate = rate; clear_handshake(chip); diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index cb1c92c..68f3c8c 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -60,15 +60,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { return ECHO_CLOCK_BIT_INTERNAL; diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c index 91dbfeb..bb9632c 100644 --- a/sound/pci/echoaudio/indigodjx_dsp.c +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -59,10 +59,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - err = init_line_levels(chip); - if (err < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } + + + +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index 134e783..beb9a5b 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -60,15 +60,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { return ECHO_CLOCK_BIT_INTERNAL; diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c index 766cf50..394c6e7 100644 --- a/sound/pci/echoaudio/indigoiox_dsp.c +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -59,10 +59,13 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - err = init_line_levels(chip); - if (err < 0) - return err; - DE_INIT(("init_hw done\n")); return err; } + + + +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index 07f3245..53ce946 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -64,17 +64,20 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->professional_spdif = FALSE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff --git a/sound/pci/echoaudio/layla24_dsp.c b/sound/pci/echoaudio/layla24_dsp.c index 12dc00a..8c04164 100644 --- a/sound/pci/echoaudio/layla24_dsp.c +++ b/sound/pci/echoaudio/layla24_dsp.c @@ -61,9 +61,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_RCA | ECHOCAPS_HAS_DIGITAL_MODE_SPDIF_OPTICAL | ECHOCAPS_HAS_DIGITAL_MODE_ADAT; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; if ((err = load_firmware(chip)) < 0) return err; @@ -72,17 +69,22 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->digital_in_automute = TRUE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c index d0302f2..6ebfa6e 100644 --- a/sound/pci/echoaudio/mia_dsp.c +++ b/sound/pci/echoaudio/mia_dsp.c @@ -66,15 +66,19 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip))) - return err; - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; diff --git a/sound/pci/echoaudio/mona_dsp.c b/sound/pci/echoaudio/mona_dsp.c index b28b8e4..6e6a7eb 100644 --- a/sound/pci/echoaudio/mona_dsp.c +++ b/sound/pci/echoaudio/mona_dsp.c @@ -67,28 +67,26 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) else chip->dsp_code_to_load = FW_MONA_301_DSP; - chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; - chip->professional_spdif = FALSE; - chip->digital_in_automute = TRUE; - if ((err = load_firmware(chip)) < 0) return err; chip->bad_board = FALSE; - if ((err = init_line_levels(chip)) < 0) - return err; - - err = set_digital_mode(chip, DIGITAL_MODE_SPDIF_RCA); - if (err < 0) - return err; - err = set_professional_spdif(chip, TRUE); - DE_INIT(("init_hw done\n")); return err; } +static int set_mixer_defaults(struct echoaudio *chip) +{ + chip->digital_mode = DIGITAL_MODE_SPDIF_RCA; + chip->professional_spdif = FALSE; + chip->digital_in_automute = TRUE; + return init_line_levels(chip); +} + + + static u32 detect_input_clocks(const struct echoaudio *chip) { u32 clocks_from_dsp, clock_bits; -- cgit v1.1 From 47b5d028fdce8f809bf22852ac900338fb90e8aa Mon Sep 17 00:00:00 2001 From: Giuliano Pochini Date: Sun, 14 Feb 2010 18:16:10 +0100 Subject: ALSA: Echoaudio - Add suspend support #2 This patch adds rearranges parts of the initialization code and adds suspend and resume callbacks. This patch adds suspend and resume callbacks. It also rearranges parts of the initialization code so it can be used in both the first initialization (when the module is loaded we also have to load default settings) and the resume callback (where we have to restore the previous settings). Signed-off-by: Giuliano Pochini Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio.c | 153 ++++++++++++++++++++++++++++++++---- sound/pci/echoaudio/echoaudio.h | 2 + sound/pci/echoaudio/echoaudio_dsp.c | 145 +++++++++++++++++++--------------- 3 files changed, 222 insertions(+), 78 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 79dde95..2783ce6 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -753,6 +753,8 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) spin_lock(&chip->lock); switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + DE_ACT(("pcm_trigger resume\n")); case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: DE_ACT(("pcm_trigger start\n")); @@ -776,6 +778,8 @@ static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) err = start_transport(chip, channelmask, chip->pipe_cyclic_mask); break; + case SNDRV_PCM_TRIGGER_SUSPEND: + DE_ACT(("pcm_trigger suspend\n")); case SNDRV_PCM_TRIGGER_STOP: DE_ACT(("pcm_trigger stop\n")); for (i = 0; i < DSP_MAXPIPES; i++) { @@ -1951,18 +1955,27 @@ static __devinit int snd_echo_create(struct snd_card *card, return err; pci_set_master(pci); - /* allocate a chip-specific data */ - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (!chip) { - pci_disable_device(pci); - return -ENOMEM; + /* Allocate chip if needed */ + if (!*rchip) { + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (!chip) { + pci_disable_device(pci); + return -ENOMEM; + } + DE_INIT(("chip=%p\n", chip)); + spin_lock_init(&chip->lock); + chip->card = card; + chip->pci = pci; + chip->irq = -1; + atomic_set(&chip->opencount, 0); + mutex_init(&chip->mode_mutex); + chip->can_set_rate = 1; + } else { + /* If this was called from the resume function, chip is + * already allocated and it contains current card settings. + */ + chip = *rchip; } - DE_INIT(("chip=%p\n", chip)); - - spin_lock_init(&chip->lock); - chip->card = card; - chip->pci = pci; - chip->irq = -1; /* PCI resource allocation */ chip->dsp_registers_phys = pci_resource_start(pci, 0); @@ -2002,7 +2015,9 @@ static __devinit int snd_echo_create(struct snd_card *card, chip->comm_page = (struct comm_page *)chip->commpage_dma_buf.area; err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); - if (err) { + if (err >= 0) + err = set_mixer_defaults(chip); + if (err < 0) { DE_INIT(("init_hw err=%d\n", err)); snd_echo_free(chip); return err; @@ -2013,9 +2028,6 @@ static __devinit int snd_echo_create(struct snd_card *card, snd_echo_free(chip); return err; } - atomic_set(&chip->opencount, 0); - mutex_init(&chip->mode_mutex); - chip->can_set_rate = 1; *rchip = chip; /* Init done ! */ return 0; @@ -2048,6 +2060,7 @@ static int __devinit snd_echo_probe(struct pci_dev *pci, snd_card_set_dev(card, &pci->dev); + chip = NULL; /* Tells snd_echo_create to allocate chip */ if ((err = snd_echo_create(card, pci, &chip)) < 0) { snd_card_free(card); return err; @@ -2187,6 +2200,112 @@ ctl_error: +#if defined(CONFIG_PM) + +static int snd_echo_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct echoaudio *chip = pci_get_drvdata(pci); + + DE_INIT(("suspend start\n")); + snd_pcm_suspend_all(chip->analog_pcm); + snd_pcm_suspend_all(chip->digital_pcm); + +#ifdef ECHOCARD_HAS_MIDI + /* This call can sleep */ + if (chip->midi_out) + snd_echo_midi_output_trigger(chip->midi_out, 0); +#endif + spin_lock_irq(&chip->lock); + if (wait_handshake(chip)) { + spin_unlock_irq(&chip->lock); + return -EIO; + } + clear_handshake(chip); + if (send_vector(chip, DSP_VC_GO_COMATOSE) < 0) { + spin_unlock_irq(&chip->lock); + return -EIO; + } + spin_unlock_irq(&chip->lock); + + chip->dsp_code = NULL; + free_irq(chip->irq, chip); + chip->irq = -1; + pci_save_state(pci); + pci_disable_device(pci); + + DE_INIT(("suspend done\n")); + return 0; +} + + + +static int snd_echo_resume(struct pci_dev *pci) +{ + struct echoaudio *chip = pci_get_drvdata(pci); + struct comm_page *commpage, *commpage_bak; + u32 pipe_alloc_mask; + int err; + + DE_INIT(("resume start\n")); + pci_restore_state(pci); + commpage_bak = kmalloc(sizeof(struct echoaudio), GFP_KERNEL); + commpage = chip->comm_page; + memcpy(commpage_bak, commpage, sizeof(struct comm_page)); + + err = init_hw(chip, chip->pci->device, chip->pci->subsystem_device); + if (err < 0) { + kfree(commpage_bak); + DE_INIT(("resume init_hw err=%d\n", err)); + snd_echo_free(chip); + return err; + } + DE_INIT(("resume init OK\n")); + + /* Temporarily set chip->pipe_alloc_mask=0 otherwise + * restore_dsp_settings() fails. + */ + pipe_alloc_mask = chip->pipe_alloc_mask; + chip->pipe_alloc_mask = 0; + err = restore_dsp_rettings(chip); + chip->pipe_alloc_mask = pipe_alloc_mask; + if (err < 0) { + kfree(commpage_bak); + return err; + } + DE_INIT(("resume restore OK\n")); + + memcpy(&commpage->audio_format, &commpage_bak->audio_format, + sizeof(commpage->audio_format)); + memcpy(&commpage->sglist_addr, &commpage_bak->sglist_addr, + sizeof(commpage->sglist_addr)); + memcpy(&commpage->midi_output, &commpage_bak->midi_output, + sizeof(commpage->midi_output)); + kfree(commpage_bak); + + if (request_irq(pci->irq, snd_echo_interrupt, IRQF_SHARED, + ECHOCARD_NAME, chip)) { + snd_echo_free(chip); + snd_printk(KERN_ERR "cannot grab irq\n"); + return -EBUSY; + } + chip->irq = pci->irq; + DE_INIT(("resume irq=%d\n", chip->irq)); + +#ifdef ECHOCARD_HAS_MIDI + if (chip->midi_input_enabled) + enable_midi_input(chip, TRUE); + if (chip->midi_out) + snd_echo_midi_output_trigger(chip->midi_out, 1); +#endif + + DE_INIT(("resume done\n")); + return 0; +} + +#endif /* CONFIG_PM */ + + + static void __devexit snd_echo_remove(struct pci_dev *pci) { struct echoaudio *chip; @@ -2209,6 +2328,10 @@ static struct pci_driver driver = { .id_table = snd_echo_ids, .probe = snd_echo_probe, .remove = __devexit_p(snd_echo_remove), +#ifdef CONFIG_PM + .suspend = snd_echo_suspend, + .resume = snd_echo_resume, +#endif /* CONFIG_PM */ }; diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index a84c0d1..1df974d 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -472,6 +472,8 @@ static void free_firmware(const struct firmware *fw_entry); #ifdef ECHOCARD_HAS_MIDI static int enable_midi_input(struct echoaudio *chip, char enable); +static void snd_echo_midi_output_trigger( + struct snd_rawmidi_substream *substream, int up); static int midi_service_irq(struct echoaudio *chip); static int __devinit snd_echo_midi_create(struct snd_card *card, struct echoaudio *chip); diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index 031ef7e..64417a7 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -497,9 +497,6 @@ static int load_firmware(struct echoaudio *chip) if ((box_type = load_asic(chip)) < 0) return box_type; /* error */ - if ((err = restore_dsp_rettings(chip)) < 0) - return err; - return box_type; } @@ -659,51 +656,106 @@ static void get_audio_meters(struct echoaudio *chip, long *meters) static int restore_dsp_rettings(struct echoaudio *chip) { - int err; + int i, o, err; DE_INIT(("restore_dsp_settings\n")); if ((err = check_asic_status(chip)) < 0) return err; - /* @ Gina20/Darla20 only. Should be harmless for other cards. */ + /* Gina20/Darla20 only. Should be harmless for other cards. */ chip->comm_page->gd_clock_state = GD_CLOCK_UNDEF; chip->comm_page->gd_spdif_status = GD_SPDIF_STATUS_UNDEF; chip->comm_page->handshake = 0xffffffff; - if ((err = set_sample_rate(chip, chip->sample_rate)) < 0) + /* Restore output busses */ + for (i = 0; i < num_busses_out(chip); i++) { + err = set_output_gain(chip, i, chip->output_gain[i]); + if (err < 0) + return err; + } + +#ifdef ECHOCARD_HAS_VMIXER + for (i = 0; i < num_pipes_out(chip); i++) + for (o = 0; o < num_busses_out(chip); o++) { + err = set_vmixer_gain(chip, o, i, + chip->vmixer_gain[o][i]); + if (err < 0) + return err; + } + if (update_vmixer_level(chip) < 0) + return -EIO; +#endif /* ECHOCARD_HAS_VMIXER */ + +#ifdef ECHOCARD_HAS_MONITOR + for (o = 0; o < num_busses_out(chip); o++) + for (i = 0; i < num_busses_in(chip); i++) { + err = set_monitor_gain(chip, o, i, + chip->monitor_gain[o][i]); + if (err < 0) + return err; + } +#endif /* ECHOCARD_HAS_MONITOR */ + +#ifdef ECHOCARD_HAS_INPUT_GAIN + for (i = 0; i < num_busses_in(chip); i++) { + err = set_input_gain(chip, i, chip->input_gain[i]); + if (err < 0) + return err; + } +#endif /* ECHOCARD_HAS_INPUT_GAIN */ + + err = update_output_line_level(chip); + if (err < 0) return err; - if (chip->meters_enabled) - if (send_vector(chip, DSP_VC_METERS_ON) < 0) - return -EIO; + err = update_input_line_level(chip); + if (err < 0) + return err; -#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK - if (set_input_clock(chip, chip->input_clock) < 0) + err = set_sample_rate(chip, chip->sample_rate); + if (err < 0) + return err; + + if (chip->meters_enabled) { + err = send_vector(chip, DSP_VC_METERS_ON); + if (err < 0) + return err; + } + +#ifdef ECHOCARD_HAS_DIGITAL_MODE_SWITCH + if (set_digital_mode(chip, chip->digital_mode) < 0) return -EIO; #endif -#ifdef ECHOCARD_HAS_OUTPUT_CLOCK_SWITCH - if (set_output_clock(chip, chip->output_clock) < 0) +#ifdef ECHOCARD_HAS_DIGITAL_IO + if (set_professional_spdif(chip, chip->professional_spdif) < 0) return -EIO; #endif - if (update_output_line_level(chip) < 0) +#ifdef ECHOCARD_HAS_PHANTOM_POWER + if (set_phantom_power(chip, chip->phantom_power) < 0) return -EIO; +#endif - if (update_input_line_level(chip) < 0) +#ifdef ECHOCARD_HAS_EXTERNAL_CLOCK + /* set_input_clock() also restores automute setting */ + if (set_input_clock(chip, chip->input_clock) < 0) return -EIO; +#endif -#ifdef ECHOCARD_HAS_VMIXER - if (update_vmixer_level(chip) < 0) +#ifdef ECHOCARD_HAS_OUTPUT_CLOCK_SWITCH + if (set_output_clock(chip, chip->output_clock) < 0) return -EIO; #endif if (wait_handshake(chip) < 0) return -EIO; clear_handshake(chip); + if (send_vector(chip, DSP_VC_UPDATE_FLAGS) < 0) + return -EIO; DE_INIT(("restore_dsp_rettings done\n")); - return send_vector(chip, DSP_VC_UPDATE_FLAGS); + return 0; } @@ -920,9 +972,6 @@ static int init_dsp_comm_page(struct echoaudio *chip) chip->card_name = ECHOCARD_NAME; chip->bad_board = TRUE; /* Set TRUE until DSP loaded */ chip->dsp_code = NULL; /* Current DSP code not loaded */ - chip->digital_mode = DIGITAL_MODE_NONE; - chip->input_clock = ECHO_CLOCK_INTERNAL; - chip->output_clock = ECHO_CLOCK_WORD; chip->asic_loaded = FALSE; memset(chip->comm_page, 0, sizeof(struct comm_page)); @@ -933,7 +982,6 @@ static int init_dsp_comm_page(struct echoaudio *chip) chip->comm_page->midi_out_free_count = cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE); chip->comm_page->sample_rate = cpu_to_le32(44100); - chip->sample_rate = 44100; /* Set line levels so we don't blast any inputs on startup */ memset(chip->comm_page->monitors, ECHOGAIN_MUTED, MONITOR_ARRAY_SIZE); @@ -944,50 +992,21 @@ static int init_dsp_comm_page(struct echoaudio *chip) -/* This function initializes the several volume controls for busses and pipes. -This MUST be called after the DSP is up and running ! */ +/* This function initializes the chip structure with default values, ie. all + * muted and internal clock source. Then it copies the settings to the DSP. + * This MUST be called after the DSP is up and running ! + */ static int init_line_levels(struct echoaudio *chip) { - int st, i, o; - DE_INIT(("init_line_levels\n")); - - /* Mute output busses */ - for (i = 0; i < num_busses_out(chip); i++) - if ((st = set_output_gain(chip, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_output_line_level(chip))) - return st; - -#ifdef ECHOCARD_HAS_VMIXER - /* Mute the Vmixer */ - for (i = 0; i < num_pipes_out(chip); i++) - for (o = 0; o < num_busses_out(chip); o++) - if ((st = set_vmixer_gain(chip, o, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_vmixer_level(chip))) - return st; -#endif /* ECHOCARD_HAS_VMIXER */ - -#ifdef ECHOCARD_HAS_MONITOR - /* Mute the monitor mixer */ - for (o = 0; o < num_busses_out(chip); o++) - for (i = 0; i < num_busses_in(chip); i++) - if ((st = set_monitor_gain(chip, o, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_output_line_level(chip))) - return st; -#endif /* ECHOCARD_HAS_MONITOR */ - -#ifdef ECHOCARD_HAS_INPUT_GAIN - for (i = 0; i < num_busses_in(chip); i++) - if ((st = set_input_gain(chip, i, ECHOGAIN_MUTED))) - return st; - if ((st = update_input_line_level(chip))) - return st; -#endif /* ECHOCARD_HAS_INPUT_GAIN */ - - return 0; + memset(chip->output_gain, ECHOGAIN_MUTED, sizeof(chip->output_gain)); + memset(chip->input_gain, ECHOGAIN_MUTED, sizeof(chip->input_gain)); + memset(chip->monitor_gain, ECHOGAIN_MUTED, sizeof(chip->monitor_gain)); + memset(chip->vmixer_gain, ECHOGAIN_MUTED, sizeof(chip->vmixer_gain)); + chip->input_clock = ECHO_CLOCK_INTERNAL; + chip->output_clock = ECHO_CLOCK_WORD; + chip->sample_rate = 44100; + return restore_dsp_rettings(chip); } -- cgit v1.1 From d39e82db73eb876c60d00f00219d767b3be30307 Mon Sep 17 00:00:00 2001 From: Sebastien Alaiwan Date: Tue, 16 Feb 2010 08:55:08 +0100 Subject: ALSA: USB MIDI support for Access Music VirusTI Here's a patch that adds MIDI support through USB for one of the Access Music synths, the VirusTI. The synth uses standard USBMIDI protocol on its USB interface 3, although it does signal "vendor specific" class. A magic string has to be sent on interface 3 to enable the sending of MIDI from the synth (this string was found by sniffing usb communication of the Windows driver). This is all my patch does, and it works on my computer. Please note that the synth can also do standard usb audio I/O on its interfaces 2&3, which already works with the current snd-usb-audio driver, except for the audio input from the synth. I'm going to work on it when I have some time. Signed-off-by: Sebastien Alaiwan Signed-off-by: Clemens Ladisch (cosmetics, list terminator) Signed-off-by: Jaroslav Kysela --- sound/usb/usbaudio.c | 32 ++++++++++++++++++++++++++++++++ sound/usb/usbmidi.c | 6 ++++++ sound/usb/usbquirks.h | 27 +++++++++++++++++++++++++++ 3 files changed, 65 insertions(+) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 4963def..d01ec18 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3327,6 +3327,32 @@ static int snd_usb_cm6206_boot_quirk(struct usb_device *dev) } /* + * This call will put the synth in "USB send" mode, i.e it will send MIDI + * messages through USB (this is disabled at startup). The synth will + * acknowledge by sending a sysex on endpoint 0x85 and by displaying a USB + * sign on its LCD. Values here are chosen based on sniffing USB traffic + * under Windows. + */ +static int snd_usb_accessmusic_boot_quirk(struct usb_device *dev) +{ + int err, actual_length; + + /* "midi send" enable */ + static const u8 seq[] = { 0x4e, 0x73, 0x52, 0x01 }; + + void *buf = kmemdup(seq, ARRAY_SIZE(seq), GFP_KERNEL); + if (!buf) + return -ENOMEM; + err = usb_interrupt_msg(dev, usb_sndintpipe(dev, 0x05), buf, + ARRAY_SIZE(seq), &actual_length, 1000); + kfree(buf); + if (err < 0) + return err; + + return 0; +} + +/* * Setup quirks */ #define AUDIOPHILE_SET 0x01 /* if set, parse device_setup */ @@ -3624,6 +3650,12 @@ static void *snd_usb_audio_probe(struct usb_device *dev, goto __err_val; } + /* Access Music VirusTI Desktop */ + if (id == USB_ID(0x133e, 0x0815)) { + if (snd_usb_accessmusic_boot_quirk(dev) < 0) + goto __err_val; + } + /* * found a config. now register to ALSA */ diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 6e89b83..8f5bc1e 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1407,6 +1407,12 @@ static struct port_info { EXTERNAL_PORT(0x086a, 0x0001, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0002, 8, "%s Broadcast"), EXTERNAL_PORT(0x086a, 0x0003, 4, "%s Broadcast"), + /* Access Music Virus TI */ + EXTERNAL_PORT(0x133e, 0x0815, 0, "%s MIDI"), + PORT_INFO(0x133e, 0x0815, 1, "%s Synth", 0, + SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC | + SNDRV_SEQ_PORT_TYPE_HARDWARE | + SNDRV_SEQ_PORT_TYPE_SYNTHESIZER), }; static struct port_info *find_port_info(struct snd_usb_midi* umidi, int number) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index a892bda..406b74b 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -2073,6 +2073,33 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Access Music devices */ +{ + /* VirusTI Desktop */ + USB_DEVICE_VENDOR_SPEC(0x133e, 0x0815), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = &(const struct snd_usb_audio_quirk[]) { + { + .ifnum = 3, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = &(const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0003, + .in_cables = 0x0003 + } + }, + { + .ifnum = 4, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = -1 + } + } + } +}, + /* */ { /* aka. Serato Scratch Live DJ Box */ -- cgit v1.1 From ebfdeea3df2b8c265975b6acc47996a0b7c507e8 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 16 Feb 2010 11:17:09 +0100 Subject: ALSA: usbmixer - introduce /proc/asound/card#/usbmixer file The usbmixer proc file contains mapping between ALSA control API and USB mixer control units. The purpose of this file is for debugging and a problem diagnostics. Signed-off-by: Jaroslav Kysela --- sound/usb/usbmixer.c | 75 ++++++++++++++++++++++++++++++++++++++++++---------- 1 file changed, 61 insertions(+), 14 deletions(-) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index dd0c1d7..170bfd4 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -69,13 +69,16 @@ static const struct rc_config { { USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */ }; +#define MAX_ID_ELEMS 256 + struct usb_mixer_interface { struct snd_usb_audio *chip; unsigned int ctrlif; struct list_head list; unsigned int ignore_ctl_error; struct urb *urb; - struct usb_mixer_elem_info **id_elems; /* array[256], indexed by unit id */ + /* array[MAX_ID_ELEMS], indexed by unit id */ + struct usb_mixer_elem_info **id_elems; /* Sound Blaster remote control stuff */ const struct rc_config *rc_cfg; @@ -1825,6 +1828,45 @@ static void snd_usb_mixer_notify_id(struct usb_mixer_interface *mixer, info->elem_id); } +static void snd_usb_mixer_dump_cval(struct snd_info_buffer *buffer, + int unitid, + struct usb_mixer_elem_info *cval) +{ + static char *val_types[] = {"BOOLEAN", "INV_BOOLEAN", + "S8", "U8", "S16", "U16"}; + snd_iprintf(buffer, " Unit: %i\n", unitid); + if (cval->elem_id) + snd_iprintf(buffer, " Control: name=\"%s\", index=%i\n", + cval->elem_id->name, cval->elem_id->index); + snd_iprintf(buffer, " Info: id=%i, control=%i, cmask=0x%x, " + "channels=%i, type=\"%s\"\n", cval->id, + cval->control, cval->cmask, cval->channels, + val_types[cval->val_type]); + snd_iprintf(buffer, " Volume: min=%i, max=%i, dBmin=%i, dBmax=%i\n", + cval->min, cval->max, cval->dBmin, cval->dBmax); +} + +static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_usb_audio *chip = entry->private_data; + struct usb_mixer_interface *mixer; + struct usb_mixer_elem_info *cval; + int unitid; + + list_for_each_entry(mixer, &chip->mixer_list, list) { + snd_iprintf(buffer, + "USB Mixer: ctrlif=%i, ctlerr=%i\n", + mixer->ctrlif, mixer->ignore_ctl_error); + snd_iprintf(buffer, "Card: %s\n", chip->card->longname); + for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) { + for (cval = mixer->id_elems[unitid]; cval; + cval = cval->next_id_elem) + snd_usb_mixer_dump_cval(buffer, unitid, cval); + } + } +} + static void snd_usb_mixer_memory_change(struct usb_mixer_interface *mixer, int unitid) { @@ -2187,20 +2229,21 @@ static int snd_xonar_u1_controls_create(struct usb_mixer_interface *mixer) } void snd_emuusb_set_samplerate(struct snd_usb_audio *chip, - unsigned char samplerate_id) + unsigned char samplerate_id) { - struct usb_mixer_interface *mixer; - struct usb_mixer_elem_info *cval; - int unitid = 12; /* SamleRate ExtensionUnit ID */ + struct usb_mixer_interface *mixer; + struct usb_mixer_elem_info *cval; + int unitid = 12; /* SamleRate ExtensionUnit ID */ - list_for_each_entry(mixer, &chip->mixer_list, list) { - cval = mixer->id_elems[unitid]; - if (cval) { - set_cur_ctl_value(cval, cval->control << 8, samplerate_id); + list_for_each_entry(mixer, &chip->mixer_list, list) { + cval = mixer->id_elems[unitid]; + if (cval) { + set_cur_ctl_value(cval, cval->control << 8, + samplerate_id); snd_usb_mixer_notify_id(mixer, unitid); - } - break; - } + } + break; + } } int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, @@ -2210,6 +2253,7 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, .dev_free = snd_usb_mixer_dev_free }; struct usb_mixer_interface *mixer; + struct snd_info_entry *entry; int err; strcpy(chip->card->mixername, "USB Mixer"); @@ -2236,8 +2280,6 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020) || mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) { - struct snd_info_entry *entry; - if ((err = snd_audigy2nx_controls_create(mixer)) < 0) goto _error; if (!snd_card_proc_new(chip->card, "audigy2nx", &entry)) @@ -2255,6 +2297,11 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, err = snd_device_new(chip->card, SNDRV_DEV_LOWLEVEL, mixer, &dev_ops); if (err < 0) goto _error; + + if (list_empty(&chip->mixer_list) && + !snd_card_proc_new(chip->card, "usbmixer", &entry)) + snd_info_set_text_ops(entry, chip, snd_usb_mixer_proc_read); + list_add(&mixer->list, &chip->mixer_list); return 0; -- cgit v1.1 From 7affdc17d49b5d9e9c350d5d99ee34ab8655c7b4 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 16 Feb 2010 11:52:27 +0100 Subject: ALSA: usbmixer - add usb_id value to usbmixer proc file Signed-off-by: Jaroslav Kysela --- sound/usb/usbmixer.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 170bfd4..03f125d 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -1856,8 +1856,9 @@ static void snd_usb_mixer_proc_read(struct snd_info_entry *entry, list_for_each_entry(mixer, &chip->mixer_list, list) { snd_iprintf(buffer, - "USB Mixer: ctrlif=%i, ctlerr=%i\n", - mixer->ctrlif, mixer->ignore_ctl_error); + "USB Mixer: usb_id=0x%08x, ctrlif=%i, ctlerr=%i\n", + chip->usb_id, mixer->ctrlif, + mixer->ignore_ctl_error); snd_iprintf(buffer, "Card: %s\n", chip->card->longname); for (unitid = 0; unitid < MAX_ID_ELEMS; unitid++) { for (cval = mixer->id_elems[unitid]; cval; -- cgit v1.1 From 291186e049d7f8178ad31d43c38a53889f25d79e Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Tue, 16 Feb 2010 11:55:18 +0100 Subject: ALSA: usbmixer - use MAX_ID_ELEMS where possible Signed-off-by: Jaroslav Kysela --- sound/usb/usbmixer.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 03f125d..35b4830 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -108,7 +108,7 @@ struct mixer_build { struct usb_mixer_interface *mixer; unsigned char *buffer; unsigned int buflen; - DECLARE_BITMAP(unitbitmap, 256); + DECLARE_BITMAP(unitbitmap, MAX_ID_ELEMS); struct usb_audio_term oterm; const struct usbmix_name_map *map; const struct usbmix_selector_map *selector_map; @@ -2265,7 +2265,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, mixer->chip = chip; mixer->ctrlif = ctrlif; mixer->ignore_ctl_error = ignore_error; - mixer->id_elems = kcalloc(256, sizeof(*mixer->id_elems), GFP_KERNEL); + mixer->id_elems = kcalloc(MAX_ID_ELEMS, sizeof(*mixer->id_elems), + GFP_KERNEL); if (!mixer->id_elems) { kfree(mixer); return -ENOMEM; -- cgit v1.1 From 7fb2d723e65cc793213515fa1da092b7c92a5b48 Mon Sep 17 00:00:00 2001 From: Florian Zumbiehl Date: Thu, 18 Feb 2010 07:01:20 +0100 Subject: ALSA: cs46xx - Do test writes to register AC97_REC_GAIN in snd_cs46xx_codec_reset() bypassing the register cache, so as to not clobber the cached register value during resume. Signed-off-by: Florian Zumbiehl Signed-off-by: Takashi Iwai --- sound/pci/cs46xx/cs46xx_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 56fcf00..9fea5bb 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2266,7 +2266,7 @@ static void snd_cs46xx_codec_reset (struct snd_ac97 * ac97) return; /* test if we can write to the record gain volume register */ - snd_ac97_write_cache(ac97, AC97_REC_GAIN, 0x8a05); + snd_ac97_write(ac97, AC97_REC_GAIN, 0x8a05); if ((err = snd_ac97_read(ac97, AC97_REC_GAIN)) == 0x8a05) return; -- cgit v1.1 From 04510a74bfbcbfd53dd48b3094aad89d5eca1d28 Mon Sep 17 00:00:00 2001 From: Florian Zumbiehl Date: Thu, 18 Feb 2010 07:03:55 +0100 Subject: ALSA: cs46xx - fix some typos Signed-off-by: Florian Zumbiehl Signed-off-by: Takashi Iwai --- sound/pci/cs46xx/cs46xx_lib.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 9fea5bb..3f99a5e 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2238,11 +2238,11 @@ static void snd_cs46xx_codec_reset (struct snd_ac97 * ac97) /* set the desired CODEC mode */ if (ac97->num == CS46XX_PRIMARY_CODEC_INDEX) { - snd_printdd("cs46xx: CODOEC1 mode %04x\n",0x0); - snd_cs46xx_ac97_write(ac97,AC97_CSR_ACMODE,0x0); + snd_printdd("cs46xx: CODEC1 mode %04x\n", 0x0); + snd_cs46xx_ac97_write(ac97, AC97_CSR_ACMODE, 0x0); } else if (ac97->num == CS46XX_SECONDARY_CODEC_INDEX) { - snd_printdd("cs46xx: CODOEC2 mode %04x\n",0x3); - snd_cs46xx_ac97_write(ac97,AC97_CSR_ACMODE,0x3); + snd_printdd("cs46xx: CODEC2 mode %04x\n", 0x3); + snd_cs46xx_ac97_write(ac97, AC97_CSR_ACMODE, 0x3); } else { snd_BUG(); /* should never happen ... */ } -- cgit v1.1 From 40717382e0c1f572553e4fdefb489db4b95a5e7e Mon Sep 17 00:00:00 2001 From: Chris J Arges Date: Wed, 17 Feb 2010 12:12:52 -0600 Subject: ALSA: usbaudio Mbox support, output only Signed-off-by: Chris J Arges Signed-off-by: Takashi Iwai --- sound/usb/usbquirks.h | 45 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 45 insertions(+) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index e691eba..fc1d2cd 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -2215,6 +2215,51 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Digidesign Mbox */ +{ + /* Thanks to Clemens Ladisch */ + USB_DEVICE(0x0dba, 0x1000), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Digidesign", + .product_name = "MBox", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]){ + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .format = SNDRV_PCM_FORMAT_S24_3BE, + .channels = 2, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .attributes = EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x02, + .ep_attr = 0x01, + .maxpacksize = 0x130, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = -1 + } + } + + } +}, + { /* * Some USB MIDI devices don't have an audio control interface, -- cgit v1.1 From 28e1b773083d349d5223f586a39fa30f5d0f1c36 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:09 +0100 Subject: ALSA: usbaudio: parse USB descriptors with structs In preparation of support for v2.0 audio class, use the structs from linux/usb/audio.h and add some new ones to describe the fields that are actually parsed by the descriptor decoders. Also, factor out code from usb_create_streams(). This makes it easier to adopt the new iteration logic needed for v2.0. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- include/linux/usb/audio.h | 32 +++++++- sound/usb/usbaudio.c | 198 ++++++++++++++++++++++++++++------------------ sound/usb/usbmixer.c | 37 +++++---- 3 files changed, 168 insertions(+), 99 deletions(-) diff --git a/include/linux/usb/audio.h b/include/linux/usb/audio.h index eaf9dff..44f82d8 100644 --- a/include/linux/usb/audio.h +++ b/include/linux/usb/audio.h @@ -81,7 +81,7 @@ /* Terminal Control Selectors */ /* 4.3.2 Class-Specific AC Interface Descriptor */ -struct uac_ac_header_descriptor { +struct uac_ac_header_descriptor_v1 { __u8 bLength; /* 8 + n */ __u8 bDescriptorType; /* USB_DT_CS_INTERFACE */ __u8 bDescriptorSubtype; /* UAC_MS_HEADER */ @@ -95,7 +95,7 @@ struct uac_ac_header_descriptor { /* As above, but more useful for defining your own descriptors: */ #define DECLARE_UAC_AC_HEADER_DESCRIPTOR(n) \ -struct uac_ac_header_descriptor_##n { \ +struct uac_ac_header_descriptor_v1_##n { \ __u8 bLength; \ __u8 bDescriptorType; \ __u8 bDescriptorSubtype; \ @@ -131,7 +131,7 @@ struct uac_input_terminal_descriptor { #define UAC_INPUT_TERMINAL_PROC_MICROPHONE_ARRAY 0x206 /* 4.3.2.2 Output Terminal Descriptor */ -struct uac_output_terminal_descriptor { +struct uac_output_terminal_descriptor_v1 { __u8 bLength; /* in bytes: 9 */ __u8 bDescriptorType; /* CS_INTERFACE descriptor type */ __u8 bDescriptorSubtype; /* OUTPUT_TERMINAL descriptor subtype */ @@ -171,7 +171,7 @@ struct uac_feature_unit_descriptor_##ch { \ } __attribute__ ((packed)) /* 4.5.2 Class-Specific AS Interface Descriptor */ -struct uac_as_header_descriptor { +struct uac_as_header_descriptor_v1 { __u8 bLength; /* in bytes: 7 */ __u8 bDescriptorType; /* USB_DT_CS_INTERFACE */ __u8 bDescriptorSubtype; /* AS_GENERAL */ @@ -232,6 +232,19 @@ struct uac_format_type_i_discrete_descriptor_##n { \ #define UAC_FORMAT_TYPE_I_DISCRETE_DESC_SIZE(n) (8 + (n * 3)) +/* Formats - Audio Data Format Type I Codes */ + +struct uac_format_type_ii_discrete_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bFormatType; + __le16 wMaxBitRate; + __le16 wSamplesPerFrame; + __u8 bSamFreqType; + __u8 tSamFreq[][3]; +} __attribute__((packed)); + /* Formats - A.2 Format Type Codes */ #define UAC_FORMAT_TYPE_UNDEFINED 0x0 #define UAC_FORMAT_TYPE_I 0x1 @@ -253,6 +266,17 @@ struct uac_iso_endpoint_descriptor { #define UAC_EP_CS_ATTR_FILL_MAX 0x80 /* A.10.2 Feature Unit Control Selectors */ + +struct uac_feature_unit_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bUnitID; + __u8 bSourceID; + __u8 bControlSize; + __u8 controls[0]; /* variable length */ +} __attribute__((packed)); + #define UAC_FU_CONTROL_UNDEFINED 0x00 #define UAC_MUTE_CONTROL 0x01 #define UAC_VOLUME_CONTROL 0x02 diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index c6b9c8c..f833dea 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -46,6 +46,8 @@ #include #include #include +#include + #include #include #include @@ -2421,15 +2423,17 @@ static int is_big_endian_format(struct snd_usb_audio *chip, struct audioformat * * @fmt: the format type descriptor */ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt) + int format, void *fmt_raw) { int pcm_format; int sample_width, sample_bytes; + struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; /* FIXME: correct endianess and sign? */ pcm_format = -1; - sample_width = fmt[6]; - sample_bytes = fmt[5]; + sample_width = fmt->bBitResolution; + sample_bytes = fmt->bSubframeSize; + switch (format) { case 0: /* some devices don't define this correctly... */ snd_printdd(KERN_INFO "%d:%u:%d : format type 0 is detected, processed as PCM\n", @@ -2442,7 +2446,7 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor sample_width, sample_bytes); } /* check the format byte size */ - switch (fmt[5]) { + switch (sample_bytes) { case 1: pcm_format = SNDRV_PCM_FORMAT_S8; break; @@ -2463,8 +2467,8 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor break; default: snd_printk(KERN_INFO "%d:%u:%d : unsupported sample bitwidth %d in %d bytes\n", - chip->dev->devnum, fp->iface, - fp->altsetting, sample_width, sample_bytes); + chip->dev->devnum, fp->iface, fp->altsetting, + sample_width, sample_bytes); break; } break; @@ -2564,11 +2568,12 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform * parse the format type I and III descriptors */ static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt) + int format, void *fmt_raw) { int pcm_format; + struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; - if (fmt[3] == USB_FORMAT_TYPE_III) { + if (fmt->bFormatType == USB_FORMAT_TYPE_III) { /* FIXME: the format type is really IECxxx * but we give normal PCM format to get the existing * apps working... @@ -2590,23 +2595,27 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat * if (pcm_format < 0) return -1; } + fp->format = pcm_format; - fp->channels = fmt[4]; + fp->channels = fmt->bNrChannels; + if (fp->channels < 1) { snd_printk(KERN_ERR "%d:%u:%d : invalid channels %d\n", chip->dev->devnum, fp->iface, fp->altsetting, fp->channels); return -1; } - return parse_audio_format_rates(chip, fp, fmt, 7); + return parse_audio_format_rates(chip, fp, fmt_raw, 7); } /* - * prase the format type II descriptor + * parse the format type II descriptor */ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt) + int format, void *fmt_raw) { int brate, framesize; + struct uac_format_type_ii_discrete_descriptor *fmt = fmt_raw; + switch (format) { case USB_AUDIO_FORMAT_AC3: /* FIXME: there is no AC3 format defined yet */ @@ -2622,20 +2631,25 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat fp->format = SNDRV_PCM_FORMAT_MPEG; break; } + fp->channels = 1; - brate = combine_word(&fmt[4]); /* fmt[4,5] : wMaxBitRate (in kbps) */ - framesize = combine_word(&fmt[6]); /* fmt[6,7]: wSamplesPerFrame */ + + brate = le16_to_cpu(fmt->wMaxBitRate); + framesize = le16_to_cpu(fmt->wSamplesPerFrame); snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize); fp->frame_size = framesize; - return parse_audio_format_rates(chip, fp, fmt, 8); /* fmt[8..] sample rates */ + return parse_audio_format_rates(chip, fp, fmt_raw, 8); /* fmt[8..] sample rates */ } static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, - int format, unsigned char *fmt, int stream) + int format, void *fmt_raw, int stream) { int err; + /* we only parse the common header of all format types here, + * so it is safe to take a type_i struct */ + struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; - switch (fmt[3]) { + switch (fmt->bFormatType) { case USB_FORMAT_TYPE_I: case USB_FORMAT_TYPE_III: err = parse_audio_format_i(chip, fp, format, fmt); @@ -2645,10 +2659,10 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp break; default: snd_printd(KERN_INFO "%d:%u:%d : format type %d is not supported yet\n", - chip->dev->devnum, fp->iface, fp->altsetting, fmt[3]); + chip->dev->devnum, fp->iface, fp->altsetting, fmt->bFormatType); return -1; } - fp->fmt_type = fmt[3]; + fp->fmt_type = fmt->bFormatType; if (err < 0) return err; #if 1 @@ -2659,7 +2673,7 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp if (chip->usb_id == USB_ID(0x041e, 0x3000) || chip->usb_id == USB_ID(0x041e, 0x3020) || chip->usb_id == USB_ID(0x041e, 0x3061)) { - if (fmt[3] == USB_FORMAT_TYPE_I && + if (fmt->bFormatType == USB_FORMAT_TYPE_I && fp->rates != SNDRV_PCM_RATE_48000 && fp->rates != SNDRV_PCM_RATE_96000) return -1; @@ -2708,6 +2722,8 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) num = 4; for (i = 0; i < num; i++) { + struct uac_as_header_descriptor_v1 *as; + alts = &iface->altsetting[i]; altsd = get_iface_desc(alts); /* skip invalid one */ @@ -2726,7 +2742,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; altno = altsd->bAlternateSetting; - + /* audiophile usb: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x0763, 0x2003) && @@ -2734,20 +2750,21 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) continue; /* get audio formats */ - fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); - if (!fmt) { + as = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + + if (!as) { snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", dev->devnum, iface_no, altno); continue; } - if (fmt[0] < 7) { + if (as->bLength < sizeof(*as)) { snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", dev->devnum, iface_no, altno); continue; } - format = (fmt[6] << 8) | fmt[5]; /* remember the format value */ + format = le16_to_cpu(as->wFormatTag); /* remember the format value */ /* get format type */ fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, FORMAT_TYPE); @@ -2875,6 +2892,65 @@ static void snd_usb_stream_disconnect(struct list_head *head) } } +static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int interface) +{ + struct usb_device *dev = chip->dev; + struct usb_host_interface *alts; + struct usb_interface_descriptor *altsd; + struct usb_interface *iface = usb_ifnum_to_if(dev, interface); + + if (!iface) { + snd_printk(KERN_ERR "%d:%u:%d : does not exist\n", + dev->devnum, ctrlif, interface); + return -EINVAL; + } + + if (usb_interface_claimed(iface)) { + snd_printdd(KERN_INFO "%d:%d:%d: skipping, already claimed\n", + dev->devnum, ctrlif, interface); + return -EINVAL; + } + + alts = &iface->altsetting[0]; + altsd = get_iface_desc(alts); + if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || + altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && + altsd->bInterfaceSubClass == USB_SUBCLASS_MIDI_STREAMING) { + int err = snd_usbmidi_create(chip->card, iface, + &chip->midi_list, NULL); + if (err < 0) { + snd_printk(KERN_ERR "%d:%u:%d: cannot create sequencer device\n", + dev->devnum, ctrlif, interface); + return -EINVAL; + } + usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); + + return 0; + } + + if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && + altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || + altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING) { + snd_printdd(KERN_ERR "%d:%u:%d: skipping non-supported interface %d\n", + dev->devnum, ctrlif, interface, altsd->bInterfaceClass); + /* skip non-supported classes */ + return -EINVAL; + } + + if (snd_usb_get_speed(dev) == USB_SPEED_LOW) { + snd_printk(KERN_ERR "low speed audio streaming not supported\n"); + return -EINVAL; + } + + if (! parse_audio_endpoints(chip, interface)) { + usb_set_interface(dev, interface, 0); /* reset the current interface */ + usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); + return -EINVAL; + } + + return 0; +} + /* * parse audio control descriptor and create pcm/midi streams */ @@ -2882,69 +2958,36 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) { struct usb_device *dev = chip->dev; struct usb_host_interface *host_iface; - struct usb_interface *iface; - unsigned char *p1; - int i, j; + struct uac_ac_header_descriptor_v1 *h1; + void *control_header; + int i; /* find audiocontrol interface */ host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0]; - if (!(p1 = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, NULL, HEADER))) { + control_header = snd_usb_find_csint_desc(host_iface->extra, + host_iface->extralen, + NULL, HEADER); + + if (!control_header) { snd_printk(KERN_ERR "cannot find HEADER\n"); return -EINVAL; } - if (! p1[7] || p1[0] < 8 + p1[7]) { - snd_printk(KERN_ERR "invalid HEADER\n"); + + h1 = control_header; + + if (!h1->bInCollection) { + snd_printk(KERN_INFO "skipping empty audio interface (v1)\n"); return -EINVAL; } - /* - * parse all USB audio streaming interfaces - */ - for (i = 0; i < p1[7]; i++) { - struct usb_host_interface *alts; - struct usb_interface_descriptor *altsd; - j = p1[8 + i]; - iface = usb_ifnum_to_if(dev, j); - if (!iface) { - snd_printk(KERN_ERR "%d:%u:%d : does not exist\n", - dev->devnum, ctrlif, j); - continue; - } - if (usb_interface_claimed(iface)) { - snd_printdd(KERN_INFO "%d:%d:%d: skipping, already claimed\n", dev->devnum, ctrlif, j); - continue; - } - alts = &iface->altsetting[0]; - altsd = get_iface_desc(alts); - if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || - altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && - altsd->bInterfaceSubClass == USB_SUBCLASS_MIDI_STREAMING) { - int err = snd_usbmidi_create(chip->card, iface, - &chip->midi_list, NULL); - if (err < 0) { - snd_printk(KERN_ERR "%d:%u:%d: cannot create sequencer device\n", dev->devnum, ctrlif, j); - continue; - } - usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); - continue; - } - if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && - altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || - altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING) { - snd_printdd(KERN_ERR "%d:%u:%d: skipping non-supported interface %d\n", dev->devnum, ctrlif, j, altsd->bInterfaceClass); - /* skip non-supported classes */ - continue; - } - if (snd_usb_get_speed(dev) == USB_SPEED_LOW) { - snd_printk(KERN_ERR "low speed audio streaming not supported\n"); - continue; - } - if (! parse_audio_endpoints(chip, j)) { - usb_set_interface(dev, j, 0); /* reset the current interface */ - usb_driver_claim_interface(&usb_audio_driver, iface, (void *)-1L); - } + if (h1->bLength < sizeof(*h1) + h1->bInCollection) { + snd_printk(KERN_ERR "invalid HEADER (v1)\n"); + return -EINVAL; } + for (i = 0; i < h1->bInCollection; i++) + snd_usb_create_stream(chip, ctrlif, h1->baInterfaceNr[i]); + return 0; } @@ -3607,7 +3650,6 @@ static void *snd_usb_audio_probe(struct usb_device *dev, ifnum = get_iface_desc(alts)->bInterfaceNumber; id = USB_ID(le16_to_cpu(dev->descriptor.idVendor), le16_to_cpu(dev->descriptor.idProduct)); - if (quirk && quirk->ifnum >= 0 && ifnum != quirk->ifnum) goto __err_val; diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 35b4830..11636a6 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -32,6 +32,8 @@ #include #include #include +#include + #include #include #include @@ -1086,29 +1088,30 @@ static void build_feature_ctl(struct mixer_build *state, unsigned char *desc, * * most of controlls are defined here. */ -static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unsigned char *ftr) +static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void *_ftr) { int channels, i, j; struct usb_audio_term iterm; unsigned int master_bits, first_ch_bits; int err, csize; + struct uac_feature_unit_descriptor *ftr = _ftr; - if (ftr[0] < 7 || ! (csize = ftr[5]) || ftr[0] < 7 + csize) { + if (ftr->bLength < 7 || ! (csize = ftr->bControlSize) || ftr->bLength < 7 + csize) { snd_printk(KERN_ERR "usbaudio: unit %u: invalid FEATURE_UNIT descriptor\n", unitid); return -EINVAL; } /* parse the source unit */ - if ((err = parse_audio_unit(state, ftr[4])) < 0) + if ((err = parse_audio_unit(state, ftr->bSourceID)) < 0) return err; /* determine the input source type and name */ - if (check_input_term(state, ftr[4], &iterm) < 0) + if (check_input_term(state, ftr->bSourceID, &iterm) < 0) return -EINVAL; - channels = (ftr[0] - 7) / csize - 1; + channels = (ftr->bLength - 7) / csize - 1; - master_bits = snd_usb_combine_bytes(ftr + 6, csize); + master_bits = snd_usb_combine_bytes(ftr->controls, csize); /* master configuration quirks */ switch (state->chip->usb_id) { case USB_ID(0x08bb, 0x2702): @@ -1119,21 +1122,21 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, unsig break; } if (channels > 0) - first_ch_bits = snd_usb_combine_bytes(ftr + 6 + csize, csize); + first_ch_bits = snd_usb_combine_bytes(ftr->controls + csize, csize); else first_ch_bits = 0; /* check all control types */ for (i = 0; i < 10; i++) { unsigned int ch_bits = 0; for (j = 0; j < channels; j++) { - unsigned int mask = snd_usb_combine_bytes(ftr + 6 + csize * (j+1), csize); + unsigned int mask = snd_usb_combine_bytes(ftr->controls + csize * (j+1), csize); if (mask & (1 << i)) ch_bits |= (1 << j); } if (ch_bits & 1) /* the first channel must be set (for ease of programming) */ - build_feature_ctl(state, ftr, ch_bits, i, &iterm, unitid); + build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid); if (master_bits & (1 << i)) - build_feature_ctl(state, ftr, 0, i, &iterm, unitid); + build_feature_ctl(state, _ftr, 0, i, &iterm, unitid); } return 0; @@ -1780,7 +1783,7 @@ static int snd_usb_mixer_dev_free(struct snd_device *device) */ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) { - unsigned char *desc; + struct uac_output_terminal_descriptor_v1 *desc; struct mixer_build state; int err; const struct usbmix_ctl_map *map; @@ -1805,13 +1808,13 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) desc = NULL; while ((desc = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, desc, OUTPUT_TERMINAL)) != NULL) { - if (desc[0] < 9) + if (desc->bLength < 9) continue; /* invalid descriptor? */ - set_bit(desc[3], state.unitbitmap); /* mark terminal ID as visited */ - state.oterm.id = desc[3]; - state.oterm.type = combine_word(&desc[4]); - state.oterm.name = desc[8]; - err = parse_audio_unit(&state, desc[7]); + set_bit(desc->bTerminalID, state.unitbitmap); /* mark terminal ID as visited */ + state.oterm.id = desc->bTerminalID; + state.oterm.type = le16_to_cpu(desc->wTerminalType); + state.oterm.name = desc->iTerminal; + err = parse_audio_unit(&state, desc->bSourceID); if (err < 0) return err; } -- cgit v1.1 From 8fee4aff8c89c229593b76a6ab172a9cad24b412 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:10 +0100 Subject: ALSA: usbaudio: introduce new types for audio class v2 This patch adds some definitions for audio class v2. Unfortunately, the UNIT types PROCESSING_UNIT and EXTENSION_UNIT have different numerical representations in both standards, so there is need for a _V1 add-on now. usbmixer.c is changed accordingly. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- include/linux/usb/audio.h | 57 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/usbaudio.h | 19 +++++++++++++--- sound/usb/usbmixer.c | 14 ++++++------ 3 files changed, 80 insertions(+), 10 deletions(-) diff --git a/include/linux/usb/audio.h b/include/linux/usb/audio.h index 44f82d8..fb1a97b 100644 --- a/include/linux/usb/audio.h +++ b/include/linux/usb/audio.h @@ -25,6 +25,9 @@ #define USB_SUBCLASS_AUDIOSTREAMING 0x02 #define USB_SUBCLASS_MIDISTREAMING 0x03 +#define UAC_VERSION_1 0x00 +#define UAC_VERSION_2 0x20 + /* A.5 Audio Class-Specific AC Interface Descriptor Subtypes */ #define UAC_HEADER 0x01 #define UAC_INPUT_TERMINAL 0x02 @@ -180,6 +183,19 @@ struct uac_as_header_descriptor_v1 { __le16 wFormatTag; /* The Audio Data Format */ } __attribute__ ((packed)); +struct uac_as_header_descriptor_v2 { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bTerminalLink; + __u8 bmControls; + __u8 bFormatType; + __u32 bmFormats; + __u8 bNrChannels; + __u32 bmChannelConfig; + __u8 iChannelNames; +} __attribute__((packed)); + #define UAC_DT_AS_HEADER_SIZE 7 /* Formats - A.1.1 Audio Data Format Type I Codes */ @@ -232,6 +248,19 @@ struct uac_format_type_i_discrete_descriptor_##n { \ #define UAC_FORMAT_TYPE_I_DISCRETE_DESC_SIZE(n) (8 + (n * 3)) +struct uac_format_type_i_ext_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bSubslotSize; + __u8 bFormatType; + __u8 bBitResolution; + __u8 bHeaderLength; + __u8 bControlSize; + __u8 bSideBandProtocol; +} __attribute__((packed)); + + /* Formats - Audio Data Format Type I Codes */ struct uac_format_type_ii_discrete_descriptor { @@ -245,11 +274,26 @@ struct uac_format_type_ii_discrete_descriptor { __u8 tSamFreq[][3]; } __attribute__((packed)); +struct uac_format_type_ii_ext_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bFormatType; + __u16 wMaxBitRate; + __u16 wSamplesPerFrame; + __u8 bHeaderLength; + __u8 bSideBandProtocol; +} __attribute__((packed)); + + /* Formats - A.2 Format Type Codes */ #define UAC_FORMAT_TYPE_UNDEFINED 0x0 #define UAC_FORMAT_TYPE_I 0x1 #define UAC_FORMAT_TYPE_II 0x2 #define UAC_FORMAT_TYPE_III 0x3 +#define UAC_EXT_FORMAT_TYPE_I 0x81 +#define UAC_EXT_FORMAT_TYPE_II 0x82 +#define UAC_EXT_FORMAT_TYPE_III 0x83 struct uac_iso_endpoint_descriptor { __u8 bLength; /* in bytes: 7 */ @@ -265,6 +309,19 @@ struct uac_iso_endpoint_descriptor { #define UAC_EP_CS_ATTR_PITCH_CONTROL 0x02 #define UAC_EP_CS_ATTR_FILL_MAX 0x80 +/* Audio class v2.0: CLOCK_SOURCE descriptor */ + +struct uac_clock_source_descriptor { + __u8 bLength; + __u8 bDescriptorType; + __u8 bDescriptorSubtype; + __u8 bClockID; + __u8 bmAttributes; + __u8 bmControls; + __u8 bAssocTerminal; + __u8 iClockSource; +} __attribute__((packed)); + /* A.10.2 Feature Unit Control Selectors */ struct uac_feature_unit_descriptor { diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 9d8cea4..4f48293 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -36,8 +36,17 @@ #define MIXER_UNIT 0x04 #define SELECTOR_UNIT 0x05 #define FEATURE_UNIT 0x06 -#define PROCESSING_UNIT 0x07 -#define EXTENSION_UNIT 0x08 +#define PROCESSING_UNIT_V1 0x07 +#define EXTENSION_UNIT_V1 0x08 + +/* audio class v2 */ +#define EFFECT_UNIT 0x07 +#define PROCESSING_UNIT_V2 0x08 +#define EXTENSION_UNIT_V2 0x09 +#define CLOCK_SOURCE 0x0a +#define CLOCK_SELECTOR 0x0b +#define CLOCK_MULTIPLIER 0x0c +#define SAMPLE_RATE_CONVERTER 0x0d #define AS_GENERAL 0x01 #define FORMAT_TYPE 0x02 @@ -60,7 +69,7 @@ #define EP_CS_ATTR_PITCH_CONTROL 0x02 #define EP_CS_ATTR_FILL_MAX 0x80 -/* Audio Class specific Request Codes */ +/* Audio Class specific Request Codes (v1) */ #define SET_CUR 0x01 #define GET_CUR 0x81 @@ -74,6 +83,10 @@ #define GET_MEM 0x85 #define GET_STAT 0xff +/* Audio Class specific Request Codes (v2) */ +#define CS_CUR 0x01 +#define CS_RANGE 0x02 + /* Terminal Control Selectors */ #define COPY_PROTECT_CONTROL 0x01 diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 11636a6..ca79495 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -286,7 +286,7 @@ static void *find_audio_control_unit(struct mixer_build *state, unsigned char un p = NULL; while ((p = snd_usb_find_desc(state->buffer, state->buflen, p, USB_DT_CS_INTERFACE)) != NULL) { - if (p[0] >= 4 && p[2] >= INPUT_TERMINAL && p[2] <= EXTENSION_UNIT && p[3] == unit) + if (p[0] >= 4 && p[2] >= INPUT_TERMINAL && p[2] <= EXTENSION_UNIT_V1 && p[3] == unit) return p; } return NULL; @@ -607,9 +607,9 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm switch (iterm->type >> 16) { case SELECTOR_UNIT: strcpy(name, "Selector"); return 8; - case PROCESSING_UNIT: + case PROCESSING_UNIT_V1: strcpy(name, "Process Unit"); return 12; - case EXTENSION_UNIT: + case EXTENSION_UNIT_V1: strcpy(name, "Ext Unit"); return 8; case MIXER_UNIT: strcpy(name, "Mixer"); return 5; @@ -673,8 +673,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ term->id = id; term->name = p1[9 + p1[0] - 1]; return 0; - case PROCESSING_UNIT: - case EXTENSION_UNIT: + case PROCESSING_UNIT_V1: + case EXTENSION_UNIT_V1: if (p1[6] == 1) { id = p1[7]; break; /* continue to parse */ @@ -1747,9 +1747,9 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) return parse_audio_selector_unit(state, unitid, p1); case FEATURE_UNIT: return parse_audio_feature_unit(state, unitid, p1); - case PROCESSING_UNIT: + case PROCESSING_UNIT_V1: return parse_audio_processing_unit(state, unitid, p1); - case EXTENSION_UNIT: + case EXTENSION_UNIT_V1: return parse_audio_extension_unit(state, unitid, p1); default: snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]); -- cgit v1.1 From 53ee98fe8ac77d00bacc1c814d450d83cbd193d4 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:11 +0100 Subject: ALSA: usbaudio: implement basic set of class v2.0 parser This adds a number of parsers for audio class v2.0. In particular, the following internals are different and now handled by the code: * the number of streaming interfaces is now reported by an interface association descriptor. The old approach using a proprietary descriptor is deprecated. * The number of channels per interface is now stored in the AS_GENERAL descriptor (used to be part of the FORMAT_TYPE descriptor). * The list of supported sample rates is no longer stored in a variable length appendix of the format_type descriptor but is retrieved from the device using a class specific GET_RANGE command. * Supported sample formats are now reported as 32bit bitmap rather than a fixed value. For now, this is worked around by choosing just one of them. * A devices needs to have at least one CLOCK_SOURCE descriptor which denotes a clockID that is needed im the class request command. * Many descriptors (format_type, ...) have changed their layout. Handle this by casting the descriptors to the appropriate structs. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 352 ++++++++++++++++++++++++++++++++++++++++++--------- sound/usb/usbaudio.h | 3 + 2 files changed, 292 insertions(+), 63 deletions(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index f833dea..411a6cf 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2422,17 +2422,53 @@ static int is_big_endian_format(struct snd_usb_audio *chip, struct audioformat * * @format: the format tag (wFormatTag) * @fmt: the format type descriptor */ -static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audioformat *fp, - int format, void *fmt_raw) +static int parse_audio_format_i_type(struct snd_usb_audio *chip, + struct audioformat *fp, + int format, void *_fmt, + int protocol) { - int pcm_format; + int pcm_format, i; int sample_width, sample_bytes; - struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; + + switch (protocol) { + case UAC_VERSION_1: { + struct uac_format_type_i_discrete_descriptor *fmt = _fmt; + sample_width = fmt->bBitResolution; + sample_bytes = fmt->bSubframeSize; + break; + } + + case UAC_VERSION_2: { + struct uac_format_type_i_ext_descriptor *fmt = _fmt; + sample_width = fmt->bBitResolution; + sample_bytes = fmt->bSubslotSize; + + /* + * FIXME + * USB audio class v2 devices specify a bitmap of possible + * audio formats rather than one fix value. For now, we just + * pick one of them and report that as the only possible + * value for this setting. + * The bit allocation map is in fact compatible to the + * wFormatTag of the v1 AS streaming descriptors, which is why + * we can simply map the matrix. + */ + + for (i = 0; i < 5; i++) + if (format & (1UL << i)) { + format = i + 1; + break; + } + + break; + } + + default: + return -EINVAL; + } /* FIXME: correct endianess and sign? */ pcm_format = -1; - sample_width = fmt->bBitResolution; - sample_bytes = fmt->bSubframeSize; switch (format) { case 0: /* some devices don't define this correctly... */ @@ -2446,6 +2482,7 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor sample_width, sample_bytes); } /* check the format byte size */ + printk(" XXXXX SAMPLE BYTES %d\n", sample_bytes); switch (sample_bytes) { case 1: pcm_format = SNDRV_PCM_FORMAT_S8; @@ -2500,7 +2537,7 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor /* * parse the format descriptor and stores the possible sample rates - * on the audioformat table. + * on the audioformat table (audio class v1). * * @dev: usb device * @fp: audioformat record @@ -2508,8 +2545,8 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, struct audiofor * @offset: the start offset of descriptor pointing the rate type * (7 for type I and II, 8 for type II) */ -static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioformat *fp, - unsigned char *fmt, int offset) +static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audioformat *fp, + unsigned char *fmt, int offset) { int nr_rates = fmt[offset]; @@ -2565,13 +2602,85 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform } /* + * parse the format descriptor and stores the possible sample rates + * on the audioformat table (audio class v2). + */ +static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, + struct audioformat *fp, + struct usb_host_interface *iface) +{ + struct usb_device *dev = chip->dev; + unsigned char tmp[2], *data; + int i, nr_rates, data_size, ret = 0; + + /* get the number of sample rates first by only fetching 2 bytes */ + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), CS_RANGE, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + 0x0100, chip->clock_id << 8, tmp, sizeof(tmp), 1000); + + if (ret < 0) { + snd_printk(KERN_ERR "unable to retrieve number of sample rates\n"); + goto err; + } + + nr_rates = (tmp[1] << 8) | tmp[0]; + data_size = 2 + 12 * nr_rates; + data = kzalloc(data_size, GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto err; + } + + /* now get the full information */ + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), CS_RANGE, + USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, + 0x0100, chip->clock_id << 8, data, data_size, 1000); + + if (ret < 0) { + snd_printk(KERN_ERR "unable to retrieve sample rate range\n"); + ret = -EINVAL; + goto err_free; + } + + fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL); + if (!fp->rate_table) { + ret = -ENOMEM; + goto err_free; + } + + fp->nr_rates = 0; + fp->rate_min = fp->rate_max = 0; + + for (i = 0; i < nr_rates; i++) { + int rate = combine_quad(&data[2 + 12 * i]); + + fp->rate_table[fp->nr_rates] = rate; + if (!fp->rate_min || rate < fp->rate_min) + fp->rate_min = rate; + if (!fp->rate_max || rate > fp->rate_max) + fp->rate_max = rate; + fp->rates |= snd_pcm_rate_to_rate_bit(rate); + fp->nr_rates++; + } + +err_free: + kfree(data); +err: + return ret; +} + +/* * parse the format type I and III descriptors */ -static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat *fp, - int format, void *fmt_raw) +static int parse_audio_format_i(struct snd_usb_audio *chip, + struct audioformat *fp, + int format, void *_fmt, + struct usb_host_interface *iface) { - int pcm_format; - struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; + struct usb_interface_descriptor *altsd = get_iface_desc(iface); + struct uac_format_type_i_discrete_descriptor *fmt = _fmt; + int protocol = altsd->bInterfaceProtocol; + int pcm_format, ret; if (fmt->bFormatType == USB_FORMAT_TYPE_III) { /* FIXME: the format type is really IECxxx @@ -2591,30 +2700,49 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, struct audioformat * pcm_format = SNDRV_PCM_FORMAT_S16_LE; } } else { - pcm_format = parse_audio_format_i_type(chip, fp, format, fmt); + pcm_format = parse_audio_format_i_type(chip, fp, format, fmt, protocol); if (pcm_format < 0) return -1; } fp->format = pcm_format; - fp->channels = fmt->bNrChannels; + + /* gather possible sample rates */ + /* audio class v1 reports possible sample rates as part of the + * proprietary class specific descriptor. + * audio class v2 uses class specific EP0 range requests for that. + */ + switch (protocol) { + case UAC_VERSION_1: + fp->channels = fmt->bNrChannels; + ret = parse_audio_format_rates_v1(chip, fp, _fmt, 7); + break; + case UAC_VERSION_2: + /* fp->channels is already set in this case */ + ret = parse_audio_format_rates_v2(chip, fp, iface); + break; + } if (fp->channels < 1) { snd_printk(KERN_ERR "%d:%u:%d : invalid channels %d\n", chip->dev->devnum, fp->iface, fp->altsetting, fp->channels); return -1; } - return parse_audio_format_rates(chip, fp, fmt_raw, 7); + + return ret; } /* * parse the format type II descriptor */ -static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat *fp, - int format, void *fmt_raw) +static int parse_audio_format_ii(struct snd_usb_audio *chip, + struct audioformat *fp, + int format, void *_fmt, + struct usb_host_interface *iface) { - int brate, framesize; - struct uac_format_type_ii_discrete_descriptor *fmt = fmt_raw; + int brate, framesize, ret; + struct usb_interface_descriptor *altsd = get_iface_desc(iface); + int protocol = altsd->bInterfaceProtocol; switch (format) { case USB_AUDIO_FORMAT_AC3: @@ -2634,35 +2762,50 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat fp->channels = 1; - brate = le16_to_cpu(fmt->wMaxBitRate); - framesize = le16_to_cpu(fmt->wSamplesPerFrame); - snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize); - fp->frame_size = framesize; - return parse_audio_format_rates(chip, fp, fmt_raw, 8); /* fmt[8..] sample rates */ + switch (protocol) { + case UAC_VERSION_1: { + struct uac_format_type_ii_discrete_descriptor *fmt = _fmt; + brate = le16_to_cpu(fmt->wMaxBitRate); + framesize = le16_to_cpu(fmt->wSamplesPerFrame); + snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize); + fp->frame_size = framesize; + ret = parse_audio_format_rates_v1(chip, fp, _fmt, 8); /* fmt[8..] sample rates */ + break; + } + case UAC_VERSION_2: { + struct uac_format_type_ii_ext_descriptor *fmt = _fmt; + brate = le16_to_cpu(fmt->wMaxBitRate); + framesize = le16_to_cpu(fmt->wSamplesPerFrame); + snd_printd(KERN_INFO "found format II with max.bitrate = %d, frame size=%d\n", brate, framesize); + fp->frame_size = framesize; + ret = parse_audio_format_rates_v2(chip, fp, iface); + break; + } + } + + return ret; } static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp, - int format, void *fmt_raw, int stream) + int format, unsigned char *fmt, int stream, + struct usb_host_interface *iface) { int err; - /* we only parse the common header of all format types here, - * so it is safe to take a type_i struct */ - struct uac_format_type_i_discrete_descriptor *fmt = fmt_raw; - switch (fmt->bFormatType) { + switch (fmt[3]) { case USB_FORMAT_TYPE_I: case USB_FORMAT_TYPE_III: - err = parse_audio_format_i(chip, fp, format, fmt); + err = parse_audio_format_i(chip, fp, format, fmt, iface); break; case USB_FORMAT_TYPE_II: - err = parse_audio_format_ii(chip, fp, format, fmt); + err = parse_audio_format_ii(chip, fp, format, fmt, iface); break; default: snd_printd(KERN_INFO "%d:%u:%d : format type %d is not supported yet\n", - chip->dev->devnum, fp->iface, fp->altsetting, fmt->bFormatType); + chip->dev->devnum, fp->iface, fp->altsetting, fmt[3]); return -1; } - fp->fmt_type = fmt->bFormatType; + fp->fmt_type = fmt[3]; if (err < 0) return err; #if 1 @@ -2673,7 +2816,7 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp if (chip->usb_id == USB_ID(0x041e, 0x3000) || chip->usb_id == USB_ID(0x041e, 0x3020) || chip->usb_id == USB_ID(0x041e, 0x3061)) { - if (fmt->bFormatType == USB_FORMAT_TYPE_I && + if (fmt[3] == USB_FORMAT_TYPE_I && fp->rates != SNDRV_PCM_RATE_48000 && fp->rates != SNDRV_PCM_RATE_96000) return -1; @@ -2702,10 +2845,10 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) struct usb_host_interface *alts; struct usb_interface_descriptor *altsd; int i, altno, err, stream; - int format; + int format = 0, num_channels = 0; struct audioformat *fp = NULL; unsigned char *fmt, *csep; - int num; + int num, protocol; dev = chip->dev; @@ -2722,10 +2865,9 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) num = 4; for (i = 0; i < num; i++) { - struct uac_as_header_descriptor_v1 *as; - alts = &iface->altsetting[i]; altsd = get_iface_desc(alts); + protocol = altsd->bInterfaceProtocol; /* skip invalid one */ if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || @@ -2742,7 +2884,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) stream = (get_endpoint(alts, 0)->bEndpointAddress & USB_DIR_IN) ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; altno = altsd->bAlternateSetting; - + /* audiophile usb: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x0763, 0x2003) && @@ -2750,21 +2892,54 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) continue; /* get audio formats */ - as = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + switch (protocol) { + case UAC_VERSION_1: { + struct uac_as_header_descriptor_v1 *as = + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + + if (!as) { + snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", + dev->devnum, iface_no, altno); + continue; + } - if (!as) { - snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", - dev->devnum, iface_no, altno); - continue; + if (as->bLength < sizeof(*as)) { + snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", + dev->devnum, iface_no, altno); + continue; + } + + format = le16_to_cpu(as->wFormatTag); /* remember the format value */ + break; } - if (as->bLength < sizeof(*as)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", - dev->devnum, iface_no, altno); - continue; + case UAC_VERSION_2: { + struct uac_as_header_descriptor_v2 *as = + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + + if (!as) { + snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", + dev->devnum, iface_no, altno); + continue; + } + + if (as->bLength < sizeof(*as)) { + snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", + dev->devnum, iface_no, altno); + continue; + } + + num_channels = as->bNrChannels; + format = le32_to_cpu(as->bmFormats); + + break; } - format = le16_to_cpu(as->wFormatTag); /* remember the format value */ + default: + snd_printk(KERN_ERR "%d:%u:%d : unknown interface protocol %04x\n", + dev->devnum, iface_no, altno, protocol); + continue; + } /* get format type */ fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, FORMAT_TYPE); @@ -2773,7 +2948,8 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) dev->devnum, iface_no, altno); continue; } - if (fmt[0] < 8) { + if (((protocol == UAC_VERSION_1) && (fmt[0] < 8)) || + ((protocol == UAC_VERSION_2) && (fmt[0] != 6))) { snd_printk(KERN_ERR "%d:%u:%d : invalid FORMAT_TYPE desc\n", dev->devnum, iface_no, altno); continue; @@ -2787,6 +2963,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) if (fmt[4] == 1 && fmt[5] == 2 && altno == 2 && num == 3 && fp && fp->altsetting == 1 && fp->channels == 1 && fp->format == SNDRV_PCM_FORMAT_S16_LE && + protocol == UAC_VERSION_1 && le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == fp->maxpacksize * 2) continue; @@ -2815,6 +2992,8 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; fp->datainterval = parse_datainterval(chip, alts); fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); + /* num_channels is only set for v2 interfaces */ + fp->channels = num_channels; if (snd_usb_get_speed(dev) == USB_SPEED_HIGH) fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) * (fp->maxpacksize & 0x7ff); @@ -2850,7 +3029,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) } /* ok, let's parse further... */ - if (parse_audio_format(chip, fp, format, fmt, stream) < 0) { + if (parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) { kfree(fp->rate_table); kfree(fp); continue; @@ -2958,35 +3137,82 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) { struct usb_device *dev = chip->dev; struct usb_host_interface *host_iface; - struct uac_ac_header_descriptor_v1 *h1; + struct usb_interface_descriptor *altsd; void *control_header; - int i; + int i, protocol; /* find audiocontrol interface */ host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0]; control_header = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, NULL, HEADER); + altsd = get_iface_desc(host_iface); + protocol = altsd->bInterfaceProtocol; if (!control_header) { snd_printk(KERN_ERR "cannot find HEADER\n"); return -EINVAL; } - h1 = control_header; + switch (protocol) { + case UAC_VERSION_1: { + struct uac_ac_header_descriptor_v1 *h1 = control_header; - if (!h1->bInCollection) { - snd_printk(KERN_INFO "skipping empty audio interface (v1)\n"); - return -EINVAL; + if (!h1->bInCollection) { + snd_printk(KERN_INFO "skipping empty audio interface (v1)\n"); + return -EINVAL; + } + + if (h1->bLength < sizeof(*h1) + h1->bInCollection) { + snd_printk(KERN_ERR "invalid HEADER (v1)\n"); + return -EINVAL; + } + + for (i = 0; i < h1->bInCollection; i++) + snd_usb_create_stream(chip, ctrlif, h1->baInterfaceNr[i]); + + break; } - if (h1->bLength < sizeof(*h1) + h1->bInCollection) { - snd_printk(KERN_ERR "invalid HEADER (v1)\n"); - return -EINVAL; + case UAC_VERSION_2: { + struct uac_clock_source_descriptor *cs; + struct usb_interface_assoc_descriptor *assoc = + usb_ifnum_to_if(dev, ctrlif)->intf_assoc; + + if (!assoc) { + snd_printk(KERN_ERR "Audio class v2 interfaces need an interface association\n"); + return -EINVAL; + } + + /* FIXME: for now, we expect there is at least one clock source + * descriptor and we always take the first one. + * We should properly support devices with multiple clock sources, + * clock selectors and sample rate conversion units. */ + + cs = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, + NULL, CLOCK_SOURCE); + + if (!cs) { + snd_printk(KERN_ERR "CLOCK_SOURCE descriptor not found\n"); + return -EINVAL; + } + + chip->clock_id = cs->bClockID; + + for (i = 0; i < assoc->bInterfaceCount; i++) { + int intf = assoc->bFirstInterface + i; + + if (intf != ctrlif) + snd_usb_create_stream(chip, ctrlif, intf); + } + + break; } - for (i = 0; i < h1->bInCollection; i++) - snd_usb_create_stream(chip, ctrlif, h1->baInterfaceNr[i]); + default: + snd_printk(KERN_ERR "unknown protocol version 0x%02x\n", protocol); + return -EINVAL; + } return 0; } diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 4f48293..26daf68 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -142,6 +142,9 @@ struct snd_usb_audio { int num_interfaces; int num_suspended_intf; + /* for audio class v2 */ + int clock_id; + struct list_head pcm_list; /* list of pcm streams */ int pcm_devs; -- cgit v1.1 From 7b8a043f2686af9f41e313a78ed5e98233e5fded Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:12 +0100 Subject: ALSA: usbmixer: bail out early when parsing audio class v2 descriptors This is just a quick hack that needs to be removed once the new units defined by the audio class v2.0 standard are supported. However, it allows using these devices for now, without mixer support. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/usbmixer.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index ca79495..42bb95c 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -2258,7 +2258,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, }; struct usb_mixer_interface *mixer; struct snd_info_entry *entry; - int err; + struct usb_host_interface *host_iface; + int err, protocol; strcpy(chip->card->mixername, "USB Mixer"); @@ -2275,6 +2276,16 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, return -ENOMEM; } + host_iface = &usb_ifnum_to_if(chip->dev, ctrlif)->altsetting[0]; + protocol = host_iface->desc.bInterfaceProtocol; + + /* FIXME! */ + if (protocol != UAC_VERSION_1) { + snd_printk(KERN_WARNING "mixer interface protocol 0x%02x not yet supported\n", + protocol); + return 0; + } + if ((err = snd_usb_mixer_controls(mixer)) < 0 || (err = snd_usb_mixer_status_create(mixer)) < 0) goto _error; -- cgit v1.1 From de48c7bc6f93c6c8e0be8612c9d72a2dc92eaa01 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Feb 2010 23:49:13 +0100 Subject: ALSA: usbaudio: consolidate header files Use the definitions from linux/usb/audio.h all over the ALSA USB audio driver and add some missing definitions there as well. Use the endpoint attribute macros from linux/usb/ch9 and remove the own things from sound/usb/usbaudio.h. Now things are also nicely prefixed which makes understanding the code easier. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- include/linux/usb/audio.h | 31 +++++++++++- sound/usb/usbaudio.c | 125 +++++++++++++++++++++++----------------------- sound/usb/usbaudio.h | 100 ------------------------------------- sound/usb/usbmidi.c | 10 ++-- sound/usb/usbmixer.c | 62 +++++++++++------------ sound/usb/usbquirks.h | 34 ++++++------- sound/usb/usx2y/us122l.c | 6 ++- 7 files changed, 150 insertions(+), 218 deletions(-) diff --git a/include/linux/usb/audio.h b/include/linux/usb/audio.h index fb1a97b..6bb2936 100644 --- a/include/linux/usb/audio.h +++ b/include/linux/usb/audio.h @@ -35,8 +35,17 @@ #define UAC_MIXER_UNIT 0x04 #define UAC_SELECTOR_UNIT 0x05 #define UAC_FEATURE_UNIT 0x06 -#define UAC_PROCESSING_UNIT 0x07 -#define UAC_EXTENSION_UNIT 0x08 +#define UAC_PROCESSING_UNIT_V1 0x07 +#define UAC_EXTENSION_UNIT_V1 0x08 + +/* UAC v2.0 types */ +#define UAC_EFFECT_UNIT 0x07 +#define UAC_PROCESSING_UNIT_V2 0x08 +#define UAC_EXTENSION_UNIT_V2 0x09 +#define UAC_CLOCK_SOURCE 0x0a +#define UAC_CLOCK_SELECTOR 0x0b +#define UAC_CLOCK_MULTIPLIER 0x0c +#define UAC_SAMPLE_RATE_CONVERTER 0x0d /* A.6 Audio Class-Specific AS Interface Descriptor Subtypes */ #define UAC_AS_GENERAL 0x01 @@ -69,6 +78,10 @@ #define UAC_GET_STAT 0xff +/* Audio class v2.0 handles all the parameter calls differently */ +#define UAC2_CS_CUR 0x01 +#define UAC2_CS_RANGE 0x02 + /* MIDI - A.1 MS Class-Specific Interface Descriptor Subtypes */ #define UAC_MS_HEADER 0x01 #define UAC_MIDI_IN_JACK 0x02 @@ -133,6 +146,10 @@ struct uac_input_terminal_descriptor { #define UAC_INPUT_TERMINAL_MICROPHONE_ARRAY 0x205 #define UAC_INPUT_TERMINAL_PROC_MICROPHONE_ARRAY 0x206 +/* Terminals - control selectors */ + +#define UAC_TERMINAL_CS_COPY_PROTECT_CONTROL 0x01 + /* 4.3.2.2 Output Terminal Descriptor */ struct uac_output_terminal_descriptor_v1 { __u8 bLength; /* in bytes: 9 */ @@ -263,6 +280,9 @@ struct uac_format_type_i_ext_descriptor { /* Formats - Audio Data Format Type I Codes */ +#define UAC_FORMAT_TYPE_II_MPEG 0x1001 +#define UAC_FORMAT_TYPE_II_AC3 0x1002 + struct uac_format_type_ii_discrete_descriptor { __u8 bLength; __u8 bDescriptorType; @@ -285,6 +305,13 @@ struct uac_format_type_ii_ext_descriptor { __u8 bSideBandProtocol; } __attribute__((packed)); +/* type III */ +#define UAC_FORMAT_TYPE_III_IEC1937_AC3 0x2001 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG1_LAYER1 0x2002 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG2_NOEXT 0x2003 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG2_EXT 0x2004 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG2_LAYER1_LS 0x2005 +#define UAC_FORMAT_TYPE_III_IEC1937_MPEG2_LAYER23_LS 0x2006 /* Formats - A.2 Format Type Codes */ #define UAC_FORMAT_TYPE_UNDEFINED 0x0 diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 411a6cf..c539f7f 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -47,6 +47,7 @@ #include #include #include +#include #include #include @@ -598,7 +599,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, if (subs->transfer_done >= runtime->period_size) { subs->transfer_done -= runtime->period_size; period_elapsed = 1; - if (subs->fmt_type == USB_FORMAT_TYPE_II) { + if (subs->fmt_type == UAC_FORMAT_TYPE_II) { if (subs->transfer_done > 0) { /* FIXME: fill-max mode is not * supported yet */ @@ -1106,7 +1107,7 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri u->packets = (i + 1) * total_packs / subs->nurbs - i * total_packs / subs->nurbs; u->buffer_size = maxsize * u->packets; - if (subs->fmt_type == USB_FORMAT_TYPE_II) + if (subs->fmt_type == UAC_FORMAT_TYPE_II) u->packets++; /* for transfer delimiter */ u->urb = usb_alloc_urb(u->packets, GFP_KERNEL); if (!u->urb) @@ -1182,7 +1183,7 @@ static struct audioformat *find_format(struct snd_usb_substream *subs, unsigned if (i >= fp->nr_rates) continue; } - attr = fp->ep_attr & EP_ATTR_MASK; + attr = fp->ep_attr & USB_ENDPOINT_SYNCTYPE; if (! found) { found = fp; cur_attr = attr; @@ -1194,14 +1195,14 @@ static struct audioformat *find_format(struct snd_usb_substream *subs, unsigned * M-audio audiophile USB. */ if (attr != cur_attr) { - if ((attr == EP_ATTR_ASYNC && + if ((attr == USB_ENDPOINT_SYNC_ASYNC && subs->direction == SNDRV_PCM_STREAM_PLAYBACK) || - (attr == EP_ATTR_ADAPTIVE && + (attr == USB_ENDPOINT_SYNC_ADAPTIVE && subs->direction == SNDRV_PCM_STREAM_CAPTURE)) continue; - if ((cur_attr == EP_ATTR_ASYNC && + if ((cur_attr == USB_ENDPOINT_SYNC_ASYNC && subs->direction == SNDRV_PCM_STREAM_PLAYBACK) || - (cur_attr == EP_ATTR_ADAPTIVE && + (cur_attr == USB_ENDPOINT_SYNC_ADAPTIVE && subs->direction == SNDRV_PCM_STREAM_CAPTURE)) { found = fp; cur_attr = attr; @@ -1231,11 +1232,11 @@ static int init_usb_pitch(struct usb_device *dev, int iface, ep = get_endpoint(alts, 0)->bEndpointAddress; /* if endpoint has pitch control, enable it */ - if (fmt->attributes & EP_CS_ATTR_PITCH_CONTROL) { + if (fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL) { data[0] = 1; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR, + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, - PITCH_CONTROL << 8, ep, data, 1, 1000)) < 0) { + UAC_EP_CS_ATTR_PITCH_CONTROL << 8, ep, data, 1, 1000)) < 0) { snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH\n", dev->devnum, iface, ep); return err; @@ -1254,21 +1255,21 @@ static int init_usb_sample_rate(struct usb_device *dev, int iface, ep = get_endpoint(alts, 0)->bEndpointAddress; /* if endpoint has sampling rate control, set it */ - if (fmt->attributes & EP_CS_ATTR_SAMPLE_RATE) { + if (fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE) { int crate; data[0] = rate; data[1] = rate >> 8; data[2] = rate >> 16; - if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR, + if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, - SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) { + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, data, 3, 1000)) < 0) { snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n", dev->devnum, iface, fmt->altsetting, rate, ep); return err; } - if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), GET_CUR, + if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_IN, - SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) { + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, data, 3, 1000)) < 0) { snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n", dev->devnum, iface, fmt->altsetting, ep); return 0; /* some devices don't support reading */ @@ -1386,9 +1387,9 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) * descriptors which fool us. if it has only one EP, * assume it as adaptive-out or sync-in. */ - attr = fmt->ep_attr & EP_ATTR_MASK; - if (((is_playback && attr == EP_ATTR_ASYNC) || - (! is_playback && attr == EP_ATTR_ADAPTIVE)) && + attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE; + if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) || + (! is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) && altsd->bNumEndpoints >= 2) { /* check sync-pipe endpoint */ /* ... and check descriptor size before accessing bSynchAddress @@ -1428,7 +1429,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) } /* always fill max packet size */ - if (fmt->attributes & EP_CS_ATTR_FILL_MAX) + if (fmt->attributes & UAC_EP_CS_ATTR_FILL_MAX) subs->fill_max = 1; if ((err = init_usb_pitch(dev, subs->interface, alts, fmt)) < 0) @@ -1886,7 +1887,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre runtime->hw.channels_min = fp->channels; if (runtime->hw.channels_max < fp->channels) runtime->hw.channels_max = fp->channels; - if (fp->fmt_type == USB_FORMAT_TYPE_II && fp->frame_size > 0) { + if (fp->fmt_type == UAC_FORMAT_TYPE_II && fp->frame_size > 0) { /* FIXME: there might be more than one audio formats... */ runtime->hw.period_bytes_min = runtime->hw.period_bytes_max = fp->frame_size; @@ -2120,7 +2121,7 @@ static struct usb_device_id usb_audio_ids [] = { #include "usbquirks.h" { .match_flags = (USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS), .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL }, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { } /* Terminating entry */ }; @@ -2159,7 +2160,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s snd_iprintf(buffer, " Endpoint: %d %s (%s)\n", fp->endpoint & USB_ENDPOINT_NUMBER_MASK, fp->endpoint & USB_DIR_IN ? "IN" : "OUT", - sync_types[(fp->ep_attr & EP_ATTR_MASK) >> 2]); + sync_types[(fp->ep_attr & USB_ENDPOINT_SYNCTYPE) >> 2]); if (fp->rates & SNDRV_PCM_RATE_CONTINUOUS) { snd_iprintf(buffer, " Rates: %d - %d (continuous)\n", fp->rate_min, fp->rate_max); @@ -2471,11 +2472,11 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, pcm_format = -1; switch (format) { - case 0: /* some devices don't define this correctly... */ + case UAC_FORMAT_TYPE_I_UNDEFINED: /* some devices don't define this correctly... */ snd_printdd(KERN_INFO "%d:%u:%d : format type 0 is detected, processed as PCM\n", chip->dev->devnum, fp->iface, fp->altsetting); /* fall-through */ - case USB_AUDIO_FORMAT_PCM: + case UAC_FORMAT_TYPE_I_PCM: if (sample_width > sample_bytes * 8) { snd_printk(KERN_INFO "%d:%u:%d : sample bitwidth %d in over sample bytes %d\n", chip->dev->devnum, fp->iface, fp->altsetting, @@ -2509,7 +2510,7 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, break; } break; - case USB_AUDIO_FORMAT_PCM8: + case UAC_FORMAT_TYPE_I_PCM8: pcm_format = SNDRV_PCM_FORMAT_U8; /* Dallas DS4201 workaround: it advertises U8 format, but really @@ -2517,13 +2518,13 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, if (chip->usb_id == USB_ID(0x04fa, 0x4201)) pcm_format = SNDRV_PCM_FORMAT_S8; break; - case USB_AUDIO_FORMAT_IEEE_FLOAT: + case UAC_FORMAT_TYPE_I_IEEE_FLOAT: pcm_format = SNDRV_PCM_FORMAT_FLOAT_LE; break; - case USB_AUDIO_FORMAT_ALAW: + case UAC_FORMAT_TYPE_I_ALAW: pcm_format = SNDRV_PCM_FORMAT_A_LAW; break; - case USB_AUDIO_FORMAT_MU_LAW: + case UAC_FORMAT_TYPE_I_MULAW: pcm_format = SNDRV_PCM_FORMAT_MU_LAW; break; default: @@ -2551,7 +2552,7 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof int nr_rates = fmt[offset]; if (fmt[0] < offset + 1 + 3 * (nr_rates ? nr_rates : 2)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid FORMAT_TYPE desc\n", + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", chip->dev->devnum, fp->iface, fp->altsetting); return -1; } @@ -2614,7 +2615,7 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, int i, nr_rates, data_size, ret = 0; /* get the number of sample rates first by only fetching 2 bytes */ - ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), CS_RANGE, + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, 0x0100, chip->clock_id << 8, tmp, sizeof(tmp), 1000); @@ -2632,7 +2633,7 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip, } /* now get the full information */ - ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), CS_RANGE, + ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE, USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN, 0x0100, chip->clock_id << 8, data, data_size, 1000); @@ -2682,7 +2683,7 @@ static int parse_audio_format_i(struct snd_usb_audio *chip, int protocol = altsd->bInterfaceProtocol; int pcm_format, ret; - if (fmt->bFormatType == USB_FORMAT_TYPE_III) { + if (fmt->bFormatType == UAC_FORMAT_TYPE_III) { /* FIXME: the format type is really IECxxx * but we give normal PCM format to get the existing * apps working... @@ -2745,12 +2746,12 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, int protocol = altsd->bInterfaceProtocol; switch (format) { - case USB_AUDIO_FORMAT_AC3: + case UAC_FORMAT_TYPE_II_AC3: /* FIXME: there is no AC3 format defined yet */ // fp->format = SNDRV_PCM_FORMAT_AC3; fp->format = SNDRV_PCM_FORMAT_U8; /* temporarily hack to receive byte streams */ break; - case USB_AUDIO_FORMAT_MPEG: + case UAC_FORMAT_TYPE_II_MPEG: fp->format = SNDRV_PCM_FORMAT_MPEG; break; default: @@ -2793,11 +2794,11 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp int err; switch (fmt[3]) { - case USB_FORMAT_TYPE_I: - case USB_FORMAT_TYPE_III: + case UAC_FORMAT_TYPE_I: + case UAC_FORMAT_TYPE_III: err = parse_audio_format_i(chip, fp, format, fmt, iface); break; - case USB_FORMAT_TYPE_II: + case UAC_FORMAT_TYPE_II: err = parse_audio_format_ii(chip, fp, format, fmt, iface); break; default: @@ -2816,7 +2817,7 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp if (chip->usb_id == USB_ID(0x041e, 0x3000) || chip->usb_id == USB_ID(0x041e, 0x3020) || chip->usb_id == USB_ID(0x041e, 0x3061)) { - if (fmt[3] == USB_FORMAT_TYPE_I && + if (fmt[3] == UAC_FORMAT_TYPE_I && fp->rates != SNDRV_PCM_RATE_48000 && fp->rates != SNDRV_PCM_RATE_96000) return -1; @@ -2871,7 +2872,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) /* skip invalid one */ if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || - (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING && + (altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING && altsd->bInterfaceSubClass != USB_SUBCLASS_VENDOR_SPEC) || altsd->bNumEndpoints < 1 || le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize) == 0) @@ -2895,16 +2896,16 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) switch (protocol) { case UAC_VERSION_1: { struct uac_as_header_descriptor_v1 *as = - snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); if (!as) { - snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", + snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", dev->devnum, iface_no, altno); continue; } if (as->bLength < sizeof(*as)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", dev->devnum, iface_no, altno); continue; } @@ -2915,16 +2916,16 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) case UAC_VERSION_2: { struct uac_as_header_descriptor_v2 *as = - snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, AS_GENERAL); + snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL); if (!as) { - snd_printk(KERN_ERR "%d:%u:%d : AS_GENERAL descriptor not found\n", + snd_printk(KERN_ERR "%d:%u:%d : UAC_AS_GENERAL descriptor not found\n", dev->devnum, iface_no, altno); continue; } if (as->bLength < sizeof(*as)) { - snd_printk(KERN_ERR "%d:%u:%d : invalid AS_GENERAL desc\n", + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_AS_GENERAL desc\n", dev->devnum, iface_no, altno); continue; } @@ -2942,15 +2943,15 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) } /* get format type */ - fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, FORMAT_TYPE); + fmt = snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_FORMAT_TYPE); if (!fmt) { - snd_printk(KERN_ERR "%d:%u:%d : no FORMAT_TYPE desc\n", + snd_printk(KERN_ERR "%d:%u:%d : no UAC_FORMAT_TYPE desc\n", dev->devnum, iface_no, altno); continue; } if (((protocol == UAC_VERSION_1) && (fmt[0] < 8)) || ((protocol == UAC_VERSION_2) && (fmt[0] != 6))) { - snd_printk(KERN_ERR "%d:%u:%d : invalid FORMAT_TYPE desc\n", + snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n", dev->devnum, iface_no, altno); continue; } @@ -2972,7 +2973,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) /* Creamware Noah has this descriptor after the 2nd endpoint */ if (!csep && altsd->bNumEndpoints >= 2) csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT); - if (!csep || csep[0] < 7 || csep[2] != EP_GENERAL) { + if (!csep || csep[0] < 7 || csep[2] != UAC_EP_GENERAL) { snd_printk(KERN_WARNING "%d:%u:%d : no or invalid" " class specific endpoint descriptor\n", dev->devnum, iface_no, altno); @@ -3006,12 +3007,12 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) /* Optoplay sets the sample rate attribute although * it seems not supporting it in fact. */ - fp->attributes &= ~EP_CS_ATTR_SAMPLE_RATE; + fp->attributes &= ~UAC_EP_CS_ATTR_SAMPLE_RATE; break; case USB_ID(0x041e, 0x3020): /* Creative SB Audigy 2 NX */ case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ /* doesn't set the sample rate attribute, but supports it */ - fp->attributes |= EP_CS_ATTR_SAMPLE_RATE; + fp->attributes |= UAC_EP_CS_ATTR_SAMPLE_RATE; break; case USB_ID(0x047f, 0x0ca1): /* plantronics headset */ case USB_ID(0x077d, 0x07af): /* Griffin iMic (note that there is @@ -3020,11 +3021,11 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) * plantronics headset and Griffin iMic have set adaptive-in * although it's really not... */ - fp->ep_attr &= ~EP_ATTR_MASK; + fp->ep_attr &= ~USB_ENDPOINT_SYNCTYPE; if (stream == SNDRV_PCM_STREAM_PLAYBACK) - fp->ep_attr |= EP_ATTR_ADAPTIVE; + fp->ep_attr |= USB_ENDPOINT_SYNC_ADAPTIVE; else - fp->ep_attr |= EP_ATTR_SYNC; + fp->ep_attr |= USB_ENDPOINT_SYNC_SYNC; break; } @@ -3094,7 +3095,7 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int altsd = get_iface_desc(alts); if ((altsd->bInterfaceClass == USB_CLASS_AUDIO || altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC) && - altsd->bInterfaceSubClass == USB_SUBCLASS_MIDI_STREAMING) { + altsd->bInterfaceSubClass == USB_SUBCLASS_MIDISTREAMING) { int err = snd_usbmidi_create(chip->card, iface, &chip->midi_list, NULL); if (err < 0) { @@ -3109,7 +3110,7 @@ static int snd_usb_create_stream(struct snd_usb_audio *chip, int ctrlif, int int if ((altsd->bInterfaceClass != USB_CLASS_AUDIO && altsd->bInterfaceClass != USB_CLASS_VENDOR_SPEC) || - altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIO_STREAMING) { + altsd->bInterfaceSubClass != USB_SUBCLASS_AUDIOSTREAMING) { snd_printdd(KERN_ERR "%d:%u:%d: skipping non-supported interface %d\n", dev->devnum, ctrlif, interface, altsd->bInterfaceClass); /* skip non-supported classes */ @@ -3145,12 +3146,12 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) host_iface = &usb_ifnum_to_if(dev, ctrlif)->altsetting[0]; control_header = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, - NULL, HEADER); + NULL, UAC_HEADER); altsd = get_iface_desc(host_iface); protocol = altsd->bInterfaceProtocol; if (!control_header) { - snd_printk(KERN_ERR "cannot find HEADER\n"); + snd_printk(KERN_ERR "cannot find UAC_HEADER\n"); return -EINVAL; } @@ -3164,7 +3165,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) } if (h1->bLength < sizeof(*h1) + h1->bInCollection) { - snd_printk(KERN_ERR "invalid HEADER (v1)\n"); + snd_printk(KERN_ERR "invalid UAC_HEADER (v1)\n"); return -EINVAL; } @@ -3190,7 +3191,7 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif) * clock selectors and sample rate conversion units. */ cs = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen, - NULL, CLOCK_SOURCE); + NULL, UAC_CLOCK_SOURCE); if (!cs) { snd_printk(KERN_ERR "CLOCK_SOURCE descriptor not found\n"); @@ -3302,7 +3303,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, static const struct audioformat ua_format = { .format = SNDRV_PCM_FORMAT_S24_3LE, .channels = 2, - .fmt_type = USB_FORMAT_TYPE_I, + .fmt_type = UAC_FORMAT_TYPE_I, .altsetting = 1, .altset_idx = 1, .rates = SNDRV_PCM_RATE_CONTINUOUS, @@ -3394,7 +3395,7 @@ static int create_ua1000_quirk(struct snd_usb_audio *chip, { static const struct audioformat ua1000_format = { .format = SNDRV_PCM_FORMAT_S32_LE, - .fmt_type = USB_FORMAT_TYPE_I, + .fmt_type = UAC_FORMAT_TYPE_I, .altsetting = 1, .altset_idx = 1, .attributes = 0, diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 26daf68..6b016d4 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -21,106 +21,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ - -/* - */ - -#define USB_SUBCLASS_AUDIO_CONTROL 0x01 -#define USB_SUBCLASS_AUDIO_STREAMING 0x02 -#define USB_SUBCLASS_MIDI_STREAMING 0x03 -#define USB_SUBCLASS_VENDOR_SPEC 0xff - -#define HEADER 0x01 -#define INPUT_TERMINAL 0x02 -#define OUTPUT_TERMINAL 0x03 -#define MIXER_UNIT 0x04 -#define SELECTOR_UNIT 0x05 -#define FEATURE_UNIT 0x06 -#define PROCESSING_UNIT_V1 0x07 -#define EXTENSION_UNIT_V1 0x08 - -/* audio class v2 */ -#define EFFECT_UNIT 0x07 -#define PROCESSING_UNIT_V2 0x08 -#define EXTENSION_UNIT_V2 0x09 -#define CLOCK_SOURCE 0x0a -#define CLOCK_SELECTOR 0x0b -#define CLOCK_MULTIPLIER 0x0c -#define SAMPLE_RATE_CONVERTER 0x0d - -#define AS_GENERAL 0x01 -#define FORMAT_TYPE 0x02 -#define FORMAT_SPECIFIC 0x03 - -#define EP_GENERAL 0x01 - -#define MS_GENERAL 0x01 -#define MIDI_IN_JACK 0x02 -#define MIDI_OUT_JACK 0x03 - -/* endpoint attributes */ -#define EP_ATTR_MASK 0x0c -#define EP_ATTR_ASYNC 0x04 -#define EP_ATTR_ADAPTIVE 0x08 -#define EP_ATTR_SYNC 0x0c - -/* cs endpoint attributes */ -#define EP_CS_ATTR_SAMPLE_RATE 0x01 -#define EP_CS_ATTR_PITCH_CONTROL 0x02 -#define EP_CS_ATTR_FILL_MAX 0x80 - -/* Audio Class specific Request Codes (v1) */ - -#define SET_CUR 0x01 -#define GET_CUR 0x81 -#define SET_MIN 0x02 -#define GET_MIN 0x82 -#define SET_MAX 0x03 -#define GET_MAX 0x83 -#define SET_RES 0x04 -#define GET_RES 0x84 -#define SET_MEM 0x05 -#define GET_MEM 0x85 -#define GET_STAT 0xff - -/* Audio Class specific Request Codes (v2) */ -#define CS_CUR 0x01 -#define CS_RANGE 0x02 - -/* Terminal Control Selectors */ - -#define COPY_PROTECT_CONTROL 0x01 - -/* Endpoint Control Selectors */ - -#define SAMPLING_FREQ_CONTROL 0x01 -#define PITCH_CONTROL 0x02 - -/* Format Types */ -#define USB_FORMAT_TYPE_I 0x01 -#define USB_FORMAT_TYPE_II 0x02 -#define USB_FORMAT_TYPE_III 0x03 - -/* type I */ -#define USB_AUDIO_FORMAT_PCM 0x01 -#define USB_AUDIO_FORMAT_PCM8 0x02 -#define USB_AUDIO_FORMAT_IEEE_FLOAT 0x03 -#define USB_AUDIO_FORMAT_ALAW 0x04 -#define USB_AUDIO_FORMAT_MU_LAW 0x05 - -/* type II */ -#define USB_AUDIO_FORMAT_MPEG 0x1001 -#define USB_AUDIO_FORMAT_AC3 0x1002 - -/* type III */ -#define USB_AUDIO_FORMAT_IEC1937_AC3 0x2001 -#define USB_AUDIO_FORMAT_IEC1937_MPEG1_LAYER1 0x2002 -#define USB_AUDIO_FORMAT_IEC1937_MPEG2_NOEXT 0x2003 -#define USB_AUDIO_FORMAT_IEC1937_MPEG2_EXT 0x2004 -#define USB_AUDIO_FORMAT_IEC1937_MPEG2_LAYER1_LS 0x2005 -#define USB_AUDIO_FORMAT_IEC1937_MPEG2_LAYER23_LS 0x2006 - - /* maximum number of endpoints per interface */ #define MIDI_MAX_ENDPOINTS 2 diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index b2da478..2c59afd 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -46,6 +46,8 @@ #include #include #include +#include + #include #include #include @@ -1540,7 +1542,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, if (hostif->extralen >= 7 && ms_header->bLength >= 7 && ms_header->bDescriptorType == USB_DT_CS_INTERFACE && - ms_header->bDescriptorSubtype == HEADER) + ms_header->bDescriptorSubtype == UAC_HEADER) snd_printdd(KERN_INFO "MIDIStreaming version %02x.%02x\n", ms_header->bcdMSC[1], ms_header->bcdMSC[0]); else @@ -1556,7 +1558,7 @@ static int snd_usbmidi_get_ms_info(struct snd_usb_midi* umidi, if (hostep->extralen < 4 || ms_ep->bLength < 4 || ms_ep->bDescriptorType != USB_DT_CS_ENDPOINT || - ms_ep->bDescriptorSubtype != MS_GENERAL) + ms_ep->bDescriptorSubtype != UAC_MS_GENERAL) continue; if (usb_endpoint_dir_out(ep)) { if (endpoints[epidx].out_ep) { @@ -1768,9 +1770,9 @@ static int snd_usbmidi_detect_yamaha(struct snd_usb_midi* umidi, cs_desc < hostif->extra + hostif->extralen && cs_desc[0] >= 2; cs_desc += cs_desc[0]) { if (cs_desc[1] == USB_DT_CS_INTERFACE) { - if (cs_desc[2] == MIDI_IN_JACK) + if (cs_desc[2] == UAC_MIDI_IN_JACK) endpoint->in_cables = (endpoint->in_cables << 1) | 1; - else if (cs_desc[2] == MIDI_OUT_JACK) + else if (cs_desc[2] == UAC_MIDI_OUT_JACK) endpoint->out_cables = (endpoint->out_cables << 1) | 1; } } diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 42bb95c..8e8f871b 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -286,7 +286,7 @@ static void *find_audio_control_unit(struct mixer_build *state, unsigned char un p = NULL; while ((p = snd_usb_find_desc(state->buffer, state->buflen, p, USB_DT_CS_INTERFACE)) != NULL) { - if (p[0] >= 4 && p[2] >= INPUT_TERMINAL && p[2] <= EXTENSION_UNIT_V1 && p[3] == unit) + if (p[0] >= 4 && p[2] >= UAC_INPUT_TERMINAL && p[2] <= UAC_EXTENSION_UNIT_V1 && p[3] == unit) return p; } return NULL; @@ -407,14 +407,14 @@ static int get_ctl_value(struct usb_mixer_elem_info *cval, int request, int vali static int get_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int *value) { - return get_ctl_value(cval, GET_CUR, validx, value); + return get_ctl_value(cval, UAC_GET_CUR, validx, value); } /* channel = 0: master, 1 = first channel */ static inline int get_cur_mix_raw(struct usb_mixer_elem_info *cval, int channel, int *value) { - return get_ctl_value(cval, GET_CUR, (cval->control << 8) | channel, value); + return get_ctl_value(cval, UAC_GET_CUR, (cval->control << 8) | channel, value); } static int get_cur_mix_value(struct usb_mixer_elem_info *cval, @@ -468,14 +468,14 @@ static int set_ctl_value(struct usb_mixer_elem_info *cval, int request, int vali static int set_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int value) { - return set_ctl_value(cval, SET_CUR, validx, value); + return set_ctl_value(cval, UAC_SET_CUR, validx, value); } static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int index, int value) { int err; - err = set_ctl_value(cval, SET_CUR, (cval->control << 8) | channel, + err = set_ctl_value(cval, UAC_SET_CUR, (cval->control << 8) | channel, value); if (err < 0) return err; @@ -605,13 +605,13 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm if (term_only) return 0; switch (iterm->type >> 16) { - case SELECTOR_UNIT: + case UAC_SELECTOR_UNIT: strcpy(name, "Selector"); return 8; - case PROCESSING_UNIT_V1: + case UAC_PROCESSING_UNIT_V1: strcpy(name, "Process Unit"); return 12; - case EXTENSION_UNIT_V1: + case UAC_EXTENSION_UNIT_V1: strcpy(name, "Ext Unit"); return 8; - case MIXER_UNIT: + case UAC_MIXER_UNIT: strcpy(name, "Mixer"); return 5; default: return sprintf(name, "Unit %d", iterm->id); @@ -650,22 +650,22 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ while ((p1 = find_audio_control_unit(state, id)) != NULL) { term->id = id; switch (p1[2]) { - case INPUT_TERMINAL: + case UAC_INPUT_TERMINAL: term->type = combine_word(p1 + 4); term->channels = p1[7]; term->chconfig = combine_word(p1 + 8); term->name = p1[11]; return 0; - case FEATURE_UNIT: + case UAC_FEATURE_UNIT: id = p1[4]; break; /* continue to parse */ - case MIXER_UNIT: + case UAC_MIXER_UNIT: term->type = p1[2] << 16; /* virtual type */ term->channels = p1[5 + p1[4]]; term->chconfig = combine_word(p1 + 6 + p1[4]); term->name = p1[p1[0] - 1]; return 0; - case SELECTOR_UNIT: + case UAC_SELECTOR_UNIT: /* call recursively to retrieve the channel info */ if (check_input_term(state, p1[5], term) < 0) return -ENODEV; @@ -673,8 +673,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_ term->id = id; term->name = p1[9 + p1[0] - 1]; return 0; - case PROCESSING_UNIT_V1: - case EXTENSION_UNIT_V1: + case UAC_PROCESSING_UNIT_V1: + case UAC_EXTENSION_UNIT_V1: if (p1[6] == 1) { id = p1[7]; break; /* continue to parse */ @@ -752,23 +752,23 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) break; } } - if (get_ctl_value(cval, GET_MAX, (cval->control << 8) | minchn, &cval->max) < 0 || - get_ctl_value(cval, GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) { + if (get_ctl_value(cval, UAC_GET_MAX, (cval->control << 8) | minchn, &cval->max) < 0 || + get_ctl_value(cval, UAC_GET_MIN, (cval->control << 8) | minchn, &cval->min) < 0) { snd_printd(KERN_ERR "%d:%d: cannot get min/max values for control %d (id %d)\n", cval->id, cval->mixer->ctrlif, cval->control, cval->id); return -EINVAL; } - if (get_ctl_value(cval, GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) { + if (get_ctl_value(cval, UAC_GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) { cval->res = 1; } else { int last_valid_res = cval->res; while (cval->res > 1) { - if (set_ctl_value(cval, SET_RES, (cval->control << 8) | minchn, cval->res / 2) < 0) + if (set_ctl_value(cval, UAC_SET_RES, (cval->control << 8) | minchn, cval->res / 2) < 0) break; cval->res /= 2; } - if (get_ctl_value(cval, GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) + if (get_ctl_value(cval, UAC_GET_RES, (cval->control << 8) | minchn, &cval->res) < 0) cval->res = last_valid_res; } if (cval->res == 0) @@ -1097,7 +1097,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void struct uac_feature_unit_descriptor *ftr = _ftr; if (ftr->bLength < 7 || ! (csize = ftr->bControlSize) || ftr->bLength < 7 + csize) { - snd_printk(KERN_ERR "usbaudio: unit %u: invalid FEATURE_UNIT descriptor\n", unitid); + snd_printk(KERN_ERR "usbaudio: unit %u: invalid UAC_FEATURE_UNIT descriptor\n", unitid); return -EINVAL; } @@ -1739,17 +1739,17 @@ static int parse_audio_unit(struct mixer_build *state, int unitid) } switch (p1[2]) { - case INPUT_TERMINAL: + case UAC_INPUT_TERMINAL: return 0; /* NOP */ - case MIXER_UNIT: + case UAC_MIXER_UNIT: return parse_audio_mixer_unit(state, unitid, p1); - case SELECTOR_UNIT: + case UAC_SELECTOR_UNIT: return parse_audio_selector_unit(state, unitid, p1); - case FEATURE_UNIT: + case UAC_FEATURE_UNIT: return parse_audio_feature_unit(state, unitid, p1); - case PROCESSING_UNIT_V1: + case UAC_PROCESSING_UNIT_V1: return parse_audio_processing_unit(state, unitid, p1); - case EXTENSION_UNIT_V1: + case UAC_EXTENSION_UNIT_V1: return parse_audio_extension_unit(state, unitid, p1); default: snd_printk(KERN_ERR "usbaudio: unit %u: unexpected type 0x%02x\n", unitid, p1[2]); @@ -1779,7 +1779,7 @@ static int snd_usb_mixer_dev_free(struct snd_device *device) /* * create mixer controls * - * walk through all OUTPUT_TERMINAL descriptors to search for mixers + * walk through all UAC_OUTPUT_TERMINAL descriptors to search for mixers */ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) { @@ -1807,7 +1807,7 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) } desc = NULL; - while ((desc = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, desc, OUTPUT_TERMINAL)) != NULL) { + while ((desc = snd_usb_find_csint_desc(hostif->extra, hostif->extralen, desc, UAC_OUTPUT_TERMINAL)) != NULL) { if (desc->bLength < 9) continue; /* invalid descriptor? */ set_bit(desc->bTerminalID, state.unitbitmap); /* mark terminal ID as visited */ @@ -2047,7 +2047,7 @@ static int snd_usb_soundblaster_remote_init(struct usb_mixer_interface *mixer) } mixer->rc_setup_packet->bRequestType = USB_DIR_IN | USB_TYPE_CLASS | USB_RECIP_INTERFACE; - mixer->rc_setup_packet->bRequest = GET_MEM; + mixer->rc_setup_packet->bRequest = UAC_GET_MEM; mixer->rc_setup_packet->wValue = cpu_to_le16(0); mixer->rc_setup_packet->wIndex = cpu_to_le16(0); mixer->rc_setup_packet->wLength = cpu_to_le16(len); @@ -2170,7 +2170,7 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, snd_iprintf(buffer, "%s: ", jacks[i].name); err = snd_usb_ctl_msg(mixer->chip->dev, usb_rcvctrlpipe(mixer->chip->dev, 0), - GET_MEM, USB_DIR_IN | USB_TYPE_CLASS | + UAC_GET_MEM, USB_DIR_IN | USB_TYPE_CLASS | USB_RECIP_INTERFACE, 0, jacks[i].unitid << 8, buf, 3, 100); if (err == 3 && (buf[0] == 3 || buf[0] == 6)) diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index fc1d2cd..f06faf7 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -91,7 +91,7 @@ .idVendor = 0x046d, .idProduct = 0x0850, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -100,7 +100,7 @@ .idVendor = 0x046d, .idProduct = 0x08ae, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -109,7 +109,7 @@ .idVendor = 0x046d, .idProduct = 0x08c6, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -118,7 +118,7 @@ .idVendor = 0x046d, .idProduct = 0x08f0, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -127,7 +127,7 @@ .idVendor = 0x046d, .idProduct = 0x08f5, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { .match_flags = USB_DEVICE_ID_MATCH_DEVICE | @@ -136,7 +136,7 @@ .idVendor = 0x046d, .idProduct = 0x08f6, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL }, { USB_DEVICE(0x046d, 0x0990), @@ -301,7 +301,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .iface = 1, .altsetting = 1, .altset_idx = 1, - .attributes = EP_CS_ATTR_FILL_MAX, + .attributes = UAC_EP_CS_ATTR_FILL_MAX, .endpoint = 0x81, .ep_attr = 0x05, .rates = SNDRV_PCM_RATE_CONTINUOUS, @@ -2108,7 +2108,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2122,7 +2122,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2136,7 +2136,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2150,7 +2150,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2164,7 +2164,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2178,7 +2178,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2192,7 +2192,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-950Q", @@ -2206,7 +2206,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL, + .bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { .vendor_name = "Hauppauge", .product_name = "HVR-850", @@ -2238,7 +2238,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .iface = 1, .altsetting = 1, .altset_idx = 1, - .attributes = EP_CS_ATTR_SAMPLE_RATE, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .endpoint = 0x02, .ep_attr = 0x01, .maxpacksize = 0x130, @@ -2268,7 +2268,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .match_flags = USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .bInterfaceClass = USB_CLASS_AUDIO, - .bInterfaceSubClass = USB_SUBCLASS_MIDI_STREAMING, + .bInterfaceSubClass = USB_SUBCLASS_MIDISTREAMING, .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_MIDI_STANDARD_INTERFACE diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 91bb296..44deb21 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -16,6 +16,8 @@ * Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ +#include +#include #include #include #include @@ -315,9 +317,9 @@ static int us122l_set_sample_rate(struct usb_device *dev, int rate) data[0] = rate; data[1] = rate >> 8; data[2] = rate >> 16; - err = us122l_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR, + err = us122l_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, - SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000); + UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep, data, 3, 1000); if (err < 0) snd_printk(KERN_ERR "%d: cannot set freq %d to ep 0x%x\n", dev->devnum, rate, ep); -- cgit v1.1 From aefbd3e823d4fe219bb6420b0cac505847270507 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Tue, 23 Feb 2010 10:30:00 +0100 Subject: usb/gadget/{f_audio,gmidi}.c: follow recent changes in audio.h Some structs in linux/usb/audio.h have got new names to mark them as part of version 1.0 of the USB audio standard. Follow these changes in the gadget drivers. Note that this header and the ALSA USB driver will undergo some refactoring soon, so there might be another update to the gadgets as well. Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai --- drivers/usb/gadget/f_audio.c | 6 +++--- drivers/usb/gadget/gmidi.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/usb/gadget/f_audio.c b/drivers/usb/gadget/f_audio.c index df77f61..f1e3aad 100644 --- a/drivers/usb/gadget/f_audio.c +++ b/drivers/usb/gadget/f_audio.c @@ -60,7 +60,7 @@ DECLARE_UAC_AC_HEADER_DESCRIPTOR(2); #define UAC_DT_TOTAL_LENGTH (UAC_DT_AC_HEADER_LENGTH + UAC_DT_INPUT_TERMINAL_SIZE \ + UAC_DT_OUTPUT_TERMINAL_SIZE + UAC_DT_FEATURE_UNIT_SIZE(0)) /* B.3.2 Class-Specific AC Interface Descriptor */ -static struct uac_ac_header_descriptor_2 ac_header_desc = { +static struct uac_ac_header_descriptor_v1_2 ac_header_desc = { .bLength = UAC_DT_AC_HEADER_LENGTH, .bDescriptorType = USB_DT_CS_INTERFACE, .bDescriptorSubtype = UAC_HEADER, @@ -124,7 +124,7 @@ static struct usb_audio_control_selector feature_unit = { }; #define OUTPUT_TERMINAL_ID 3 -static struct uac_output_terminal_descriptor output_terminal_desc = { +static struct uac_output_terminal_descriptor_v1 output_terminal_desc = { .bLength = UAC_DT_OUTPUT_TERMINAL_SIZE, .bDescriptorType = USB_DT_CS_INTERFACE, .bDescriptorSubtype = UAC_OUTPUT_TERMINAL, @@ -154,7 +154,7 @@ static struct usb_interface_descriptor as_interface_alt_1_desc = { }; /* B.4.2 Class-Specific AS Interface Descriptor */ -static struct uac_as_header_descriptor as_header_desc = { +static struct uac_as_header_descriptor_v1 as_header_desc = { .bLength = UAC_DT_AS_HEADER_SIZE, .bDescriptorType = USB_DT_CS_INTERFACE, .bDescriptorSubtype = UAC_AS_GENERAL, diff --git a/drivers/usb/gadget/gmidi.c b/drivers/usb/gadget/gmidi.c index d0b1e83..5f6a2e0 100644 --- a/drivers/usb/gadget/gmidi.c +++ b/drivers/usb/gadget/gmidi.c @@ -237,7 +237,7 @@ static const struct usb_interface_descriptor ac_interface_desc = { }; /* B.3.2 Class-Specific AC Interface Descriptor */ -static const struct uac_ac_header_descriptor_1 ac_header_desc = { +static const struct uac_ac_header_descriptor_v1_1 ac_header_desc = { .bLength = UAC_DT_AC_HEADER_SIZE(1), .bDescriptorType = USB_DT_CS_INTERFACE, .bDescriptorSubtype = USB_MS_HEADER, -- cgit v1.1 From e584bc3cf6865e005bbb4dbabae0bf4b3df59500 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 1 Mar 2010 16:20:37 +0100 Subject: ALSA: ua101: add Edirol UA-1000 support Add support for the Edirol UA-1000 to the UA-101 driver. Both devices behave the same, so we just have to shuffle around some interface numbers and name strings. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- Documentation/sound/alsa/ALSA-Configuration.txt | 2 +- sound/usb/Kconfig | 6 +-- sound/usb/ua101.c | 45 +++++++++++++++------ sound/usb/usbaudio.c | 53 ------------------------- sound/usb/usbaudio.h | 3 +- sound/usb/usbquirks.h | 30 -------------- 6 files changed, 38 insertions(+), 101 deletions(-) diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 33df82e..bfcbbf8 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -1812,7 +1812,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. Module snd-ua101 ---------------- - Module for the Edirol UA-101 audio/MIDI interface. + Module for the Edirol UA-101/UA-1000 audio/MIDI interfaces. This module supports multiple devices, autoprobe and hotplugging. diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 8c29258..c570ae3 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -22,13 +22,13 @@ config SND_USB_AUDIO will be called snd-usb-audio. config SND_USB_UA101 - tristate "Edirol UA-101 driver (EXPERIMENTAL)" + tristate "Edirol UA-101/UA-1000 driver (EXPERIMENTAL)" depends on EXPERIMENTAL select SND_PCM select SND_RAWMIDI help - Say Y here to include support for the Edirol UA-101 audio/MIDI - interface. + Say Y here to include support for the Edirol UA-101 and UA-1000 + audio/MIDI interfaces. To compile this driver as a module, choose M here: the module will be called snd-ua101. diff --git a/sound/usb/ua101.c b/sound/usb/ua101.c index 4f4ccdf..047dc1c 100644 --- a/sound/usb/ua101.c +++ b/sound/usb/ua101.c @@ -1,5 +1,5 @@ /* - * Edirol UA-101 driver + * Edirol UA-101/UA-1000 driver * Copyright (c) Clemens Ladisch * * This driver is free software: you can redistribute it and/or modify @@ -25,10 +25,10 @@ #include #include "usbaudio.h" -MODULE_DESCRIPTION("Edirol UA-101 driver"); +MODULE_DESCRIPTION("Edirol UA-101/1000 driver"); MODULE_AUTHOR("Clemens Ladisch "); MODULE_LICENSE("GPL v2"); -MODULE_SUPPORTED_DEVICE("{{Edirol,UA-101}}"); +MODULE_SUPPORTED_DEVICE("{{Edirol,UA-101},{Edirol,UA-1000}}"); /* I use my UA-1A for testing because I don't have a UA-101 ... */ #define UA1A_HACK @@ -1200,13 +1200,30 @@ static int ua101_probe(struct usb_interface *interface, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = &midi_ep }; + static const int intf_numbers[2][3] = { + { /* UA-101 */ + [INTF_PLAYBACK] = 0, + [INTF_CAPTURE] = 1, + [INTF_MIDI] = 2, + }, + { /* UA-1000 */ + [INTF_CAPTURE] = 1, + [INTF_PLAYBACK] = 2, + [INTF_MIDI] = 3, + }, + }; struct snd_card *card; struct ua101 *ua; unsigned int card_index, i; + int is_ua1000; + const char *name; char usb_path[32]; int err; - if (interface->altsetting->desc.bInterfaceNumber != 0) + is_ua1000 = usb_id->idProduct == 0x0044; + + if (interface->altsetting->desc.bInterfaceNumber != + intf_numbers[is_ua1000][0]) return -ENODEV; mutex_lock(&devices_mutex); @@ -1250,9 +1267,11 @@ static int ua101_probe(struct usb_interface *interface, #endif ua->intf[0] = interface; for (i = 1; i < ARRAY_SIZE(ua->intf); ++i) { - ua->intf[i] = usb_ifnum_to_if(ua->dev, i); + ua->intf[i] = usb_ifnum_to_if(ua->dev, + intf_numbers[is_ua1000][i]); if (!ua->intf[i]) { - dev_err(&ua->dev->dev, "interface %u not found\n", i); + dev_err(&ua->dev->dev, "interface %u not found\n", + intf_numbers[is_ua1000][i]); err = -ENXIO; goto probe_error; } @@ -1292,11 +1311,12 @@ static int ua101_probe(struct usb_interface *interface, } #endif + name = usb_id->idProduct == 0x0044 ? "UA-1000" : "UA-101"; strcpy(card->driver, "UA-101"); - strcpy(card->shortname, "UA-101"); + strcpy(card->shortname, name); usb_make_path(ua->dev, usb_path, sizeof(usb_path)); snprintf(ua->card->longname, sizeof(ua->card->longname), - "EDIROL UA-101 (serial %s), %u Hz at %s, %s speed", + "EDIROL %s (serial %s), %u Hz at %s, %s speed", name, ua->dev->serial ? ua->dev->serial : "?", ua->rate, usb_path, ua->dev->speed == USB_SPEED_HIGH ? "high" : "full"); @@ -1314,11 +1334,11 @@ static int ua101_probe(struct usb_interface *interface, if (err < 0) goto probe_error; - err = snd_pcm_new(card, "UA-101", 0, 1, 1, &ua->pcm); + err = snd_pcm_new(card, name, 0, 1, 1, &ua->pcm); if (err < 0) goto probe_error; ua->pcm->private_data = ua; - strcpy(ua->pcm->name, "UA-101"); + strcpy(ua->pcm->name, name); snd_pcm_set_ops(ua->pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_pcm_ops); snd_pcm_set_ops(ua->pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_pcm_ops); @@ -1389,8 +1409,9 @@ static struct usb_device_id ua101_ids[] = { #ifdef UA1A_HACK { USB_DEVICE(0x0582, 0x0018) }, #endif - { USB_DEVICE(0x0582, 0x007d) }, - { USB_DEVICE(0x0582, 0x008d) }, + { USB_DEVICE(0x0582, 0x0044) }, /* UA-1000 high speed */ + { USB_DEVICE(0x0582, 0x007d) }, /* UA-101 high speed */ + { USB_DEVICE(0x0582, 0x008d) }, /* UA-101 full speed */ { } }; MODULE_DEVICE_TABLE(usb, ua101_ids); diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 8a8f625..7ad8089 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -3116,58 +3116,6 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, return 0; } -/* - * Create a stream for an Edirol UA-1000 interface. - */ -static int create_ua1000_quirk(struct snd_usb_audio *chip, - struct usb_interface *iface, - const struct snd_usb_audio_quirk *quirk) -{ - static const struct audioformat ua1000_format = { - .format = SNDRV_PCM_FORMAT_S32_LE, - .fmt_type = USB_FORMAT_TYPE_I, - .altsetting = 1, - .altset_idx = 1, - .attributes = 0, - .rates = SNDRV_PCM_RATE_CONTINUOUS, - }; - struct usb_host_interface *alts; - struct usb_interface_descriptor *altsd; - struct audioformat *fp; - int stream, err; - - if (iface->num_altsetting != 2) - return -ENXIO; - alts = &iface->altsetting[1]; - altsd = get_iface_desc(alts); - if (alts->extralen != 11 || alts->extra[1] != USB_DT_CS_INTERFACE || - altsd->bNumEndpoints != 1) - return -ENXIO; - - fp = kmemdup(&ua1000_format, sizeof(*fp), GFP_KERNEL); - if (!fp) - return -ENOMEM; - - fp->channels = alts->extra[4]; - fp->iface = altsd->bInterfaceNumber; - fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; - fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; - fp->datainterval = parse_datainterval(chip, alts); - fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); - fp->rate_max = fp->rate_min = combine_triple(&alts->extra[8]); - - stream = (fp->endpoint & USB_DIR_IN) - ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; - err = add_audio_endpoint(chip, stream, fp); - if (err < 0) { - kfree(fp); - return err; - } - /* FIXME: playback must be synchronized to capture */ - usb_set_interface(chip->dev, fp->iface, 0); - return 0; -} - static int snd_usb_create_quirk(struct snd_usb_audio *chip, struct usb_interface *iface, const struct snd_usb_audio_quirk *quirk); @@ -3416,7 +3364,6 @@ static int snd_usb_create_quirk(struct snd_usb_audio *chip, [QUIRK_MIDI_CME] = create_any_midi_quirk, [QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk, [QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk, - [QUIRK_AUDIO_EDIROL_UA1000] = create_ua1000_quirk, [QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk, [QUIRK_AUDIO_ALIGN_TRANSFER] = create_align_transfer_quirk }; diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 9d8cea4..96c558a 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -159,7 +159,6 @@ enum quirk_type { QUIRK_MIDI_US122L, QUIRK_AUDIO_STANDARD_INTERFACE, QUIRK_AUDIO_FIXED_ENDPOINT, - QUIRK_AUDIO_EDIROL_UA1000, QUIRK_AUDIO_EDIROL_UAXX, QUIRK_AUDIO_ALIGN_TRANSFER, @@ -196,7 +195,7 @@ struct snd_usb_midi_endpoint_info { /* for QUIRK_AUDIO/MIDI_STANDARD_INTERFACE, data is NULL */ -/* for QUIRK_AUDIO_EDIROL_UA700_UA25/UA1000, data is NULL */ +/* for QUIRK_AUDIO_EDIROL_UAXX, data is NULL */ /* for QUIRK_IGNORE_INTERFACE, data is NULL */ diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index e691eba..977d980 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -1016,36 +1016,6 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, { - USB_DEVICE(0x0582, 0x0044), - .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { - .vendor_name = "Roland", - .product_name = "UA-1000", - .ifnum = QUIRK_ANY_INTERFACE, - .type = QUIRK_COMPOSITE, - .data = (const struct snd_usb_audio_quirk[]) { - { - .ifnum = 1, - .type = QUIRK_AUDIO_EDIROL_UA1000 - }, - { - .ifnum = 2, - .type = QUIRK_AUDIO_EDIROL_UA1000 - }, - { - .ifnum = 3, - .type = QUIRK_MIDI_FIXED_ENDPOINT, - .data = & (const struct snd_usb_midi_endpoint_info) { - .out_cables = 0x0003, - .in_cables = 0x0003 - } - }, - { - .ifnum = -1 - } - } - } -}, -{ /* has ID 0x0049 when not in "Advanced Driver" mode */ USB_DEVICE(0x0582, 0x0047), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { -- cgit v1.1 From 76b53774c51c4eaec646578a2e1b3716befedf1c Mon Sep 17 00:00:00 2001 From: Andrea Gelmini Date: Sat, 27 Feb 2010 17:51:03 +0100 Subject: sound/oss/v_midi.h: Checkpatch cleanup sound/oss/v_midi.h:5: ERROR: code indent should use tabs where possible sound/oss/v_midi.h:7: ERROR: trailing whitespace Signed-off-by: Andrea Gelmini Signed-off-by: Takashi Iwai --- sound/oss/v_midi.h | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/oss/v_midi.h b/sound/oss/v_midi.h index 1b86cb4..08e2185 100644 --- a/sound/oss/v_midi.h +++ b/sound/oss/v_midi.h @@ -2,9 +2,9 @@ typedef struct vmidi_devc { int dev; /* State variables */ - int opened; + int opened; spinlock_t lock; - + /* MIDI fields */ int my_mididev; int pair_mididev; @@ -12,4 +12,3 @@ typedef struct vmidi_devc { int intr_active; void (*midi_input_intr) (int dev, unsigned char data); } vmidi_devc; - -- cgit v1.1 From 3ea49652f679c2b571ca214c605ec80cb056ec10 Mon Sep 17 00:00:00 2001 From: Andrea Gelmini Date: Sat, 27 Feb 2010 17:51:19 +0100 Subject: sound/oss/coproc.h: Checkpatch cleanup sound/oss/coproc.h:7: ERROR: trailing whitespace Signed-off-by: Andrea Gelmini Signed-off-by: Takashi Iwai --- sound/oss/coproc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/coproc.h b/sound/oss/coproc.h index 7306346..7bec21b 100644 --- a/sound/oss/coproc.h +++ b/sound/oss/coproc.h @@ -4,7 +4,7 @@ */ /* - * Coprocessor access types + * Coprocessor access types */ #define COPR_CUSTOM 0x0001 /* Custom applications */ #define COPR_MIDI 0x0002 /* MIDI (MPU-401) emulation */ -- cgit v1.1 From 7f9320d415fab5c05097c77eea7a77f2f6341f24 Mon Sep 17 00:00:00 2001 From: Andrea Gelmini Date: Sat, 27 Feb 2010 17:51:29 +0100 Subject: ALSA: sound/usb/caiaq/midi.h: Checkpatch cleanup sound/usb/caiaq/midi.h:6: ERROR: "foo* bar" should be "foo *bar" Signed-off-by: Andrea Gelmini Acked-by: Daniel Mack Signed-off-by: Takashi Iwai --- sound/usb/caiaq/midi.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/caiaq/midi.h b/sound/usb/caiaq/midi.h index 9d16db0..380f984 100644 --- a/sound/usb/caiaq/midi.h +++ b/sound/usb/caiaq/midi.h @@ -3,6 +3,6 @@ int snd_usb_caiaq_midi_init(struct snd_usb_caiaqdev *dev); void snd_usb_caiaq_midi_handle_input(struct snd_usb_caiaqdev *dev, int port, const char *buf, int len); -void snd_usb_caiaq_midi_output_done(struct urb* urb); +void snd_usb_caiaq_midi_output_done(struct urb *urb); #endif /* CAIAQ_MIDI_H */ -- cgit v1.1 From 0a566ec25627bdd360f7294aa2e52f9d121233b4 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 2 Mar 2010 08:47:20 +0100 Subject: ALSA: ua101: removing debugging code Remove some code that is no longer needed now that the relevant parts of the driver have been tested. Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela --- sound/usb/ua101.c | 55 ------------------------------------------------------- 1 file changed, 55 deletions(-) diff --git a/sound/usb/ua101.c b/sound/usb/ua101.c index 047dc1c..3d458d3 100644 --- a/sound/usb/ua101.c +++ b/sound/usb/ua101.c @@ -30,9 +30,6 @@ MODULE_AUTHOR("Clemens Ladisch "); MODULE_LICENSE("GPL v2"); MODULE_SUPPORTED_DEVICE("{{Edirol,UA-101},{Edirol,UA-1000}}"); -/* I use my UA-1A for testing because I don't have a UA-101 ... */ -#define UA1A_HACK - /* * Should not be lower than the minimum scheduling delay of the host * controller. Some Intel controllers need more than one frame; as long as @@ -132,9 +129,6 @@ struct ua101 { dma_addr_t dma; } buffers[MAX_MEMORY_BUFFERS]; } capture, playback; - - unsigned int fps[10]; - unsigned int frame_counter; }; static DEFINE_MUTEX(devices_mutex); @@ -424,16 +418,6 @@ static void capture_urb_complete(struct urb *urb) if (do_period_elapsed) snd_pcm_period_elapsed(stream->substream); - /* for debugging: measure the sample rate relative to the USB clock */ - ua->fps[ua->frame_counter++ / ua->packets_per_second] += frames; - if (ua->frame_counter >= ARRAY_SIZE(ua->fps) * ua->packets_per_second) { - printk(KERN_DEBUG "capture rate:"); - for (frames = 0; frames < ARRAY_SIZE(ua->fps); ++frames) - printk(KERN_CONT " %u", ua->fps[frames]); - printk(KERN_CONT "\n"); - memset(ua->fps, 0, sizeof(ua->fps)); - ua->frame_counter = 0; - } return; stream_stopped: @@ -1256,15 +1240,6 @@ static int ua101_probe(struct usb_interface *interface, init_waitqueue_head(&ua->rate_feedback_wait); init_waitqueue_head(&ua->alsa_playback_wait); -#ifdef UA1A_HACK - if (ua->dev->descriptor.idProduct == cpu_to_le16(0x0018)) { - ua->intf[2] = interface; - ua->intf[0] = usb_ifnum_to_if(ua->dev, 1); - ua->intf[1] = usb_ifnum_to_if(ua->dev, 2); - usb_driver_claim_interface(&ua101_driver, ua->intf[0], ua); - usb_driver_claim_interface(&ua101_driver, ua->intf[1], ua); - } else { -#endif ua->intf[0] = interface; for (i = 1; i < ARRAY_SIZE(ua->intf); ++i) { ua->intf[i] = usb_ifnum_to_if(ua->dev, @@ -1283,33 +1258,12 @@ static int ua101_probe(struct usb_interface *interface, goto probe_error; } } -#ifdef UA1A_HACK - } -#endif snd_card_set_dev(card, &interface->dev); -#ifdef UA1A_HACK - if (ua->dev->descriptor.idProduct == cpu_to_le16(0x0018)) { - ua->format_bit = SNDRV_PCM_FMTBIT_S16_LE; - ua->rate = 44100; - ua->packets_per_second = 1000; - ua->capture.channels = 2; - ua->playback.channels = 2; - ua->capture.frame_bytes = 4; - ua->playback.frame_bytes = 4; - ua->capture.usb_pipe = usb_rcvisocpipe(ua->dev, 2); - ua->playback.usb_pipe = usb_sndisocpipe(ua->dev, 1); - ua->capture.max_packet_bytes = 192; - ua->playback.max_packet_bytes = 192; - } else { -#endif err = detect_usb_format(ua); if (err < 0) goto probe_error; -#ifdef UA1A_HACK - } -#endif name = usb_id->idProduct == 0x0044 ? "UA-1000" : "UA-101"; strcpy(card->driver, "UA-101"); @@ -1342,16 +1296,10 @@ static int ua101_probe(struct usb_interface *interface, snd_pcm_set_ops(ua->pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_pcm_ops); snd_pcm_set_ops(ua->pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_pcm_ops); -#ifdef UA1A_HACK - if (ua->dev->descriptor.idProduct != cpu_to_le16(0x0018)) { -#endif err = snd_usbmidi_create(card, ua->intf[INTF_MIDI], &ua->midi_list, &midi_quirk); if (err < 0) goto probe_error; -#ifdef UA1A_HACK - } -#endif err = snd_card_register(card); if (err < 0) @@ -1406,9 +1354,6 @@ static void ua101_disconnect(struct usb_interface *interface) } static struct usb_device_id ua101_ids[] = { -#ifdef UA1A_HACK - { USB_DEVICE(0x0582, 0x0018) }, -#endif { USB_DEVICE(0x0582, 0x0044) }, /* UA-1000 high speed */ { USB_DEVICE(0x0582, 0x007d) }, /* UA-101 high speed */ { USB_DEVICE(0x0582, 0x008d) }, /* UA-101 full speed */ -- cgit v1.1 From 864c11080cf365720103042444534a1e94d42bac Mon Sep 17 00:00:00 2001 From: Arseniy Lartsev Date: Tue, 2 Mar 2010 14:52:28 +0300 Subject: ALSA: usbaudio: Fix wrong bitrate for Creative Creative VF0470 Live Cam This patch works around misbehaviour of Creative Creative VF0470 Live Cam which reports 16 kHz sample rate for audio capture while actually producing 8 kHz stream. Signed-off-by: Arseniy Lartsev Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index 20b656e..ea3eaa5 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2581,6 +2581,9 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof chip->usb_id == USB_ID(0x0d8c, 0x0102)) && fp->altsetting == 5 && fp->maxpacksize == 392) rate = 96000; + /* Creative VF0470 Live Cam reports 16 kHz instead of 8kHz */ + if (rate == 16000 && chip->usb_id == USB_ID(0x041e, 0x4068)) + rate = 8000; fp->rate_table[fp->nr_rates] = rate; if (!fp->rate_min || rate < fp->rate_min) fp->rate_min = rate; -- cgit v1.1 From fd8d47351d2e241f3168eeb697ce55cc28c75b78 Mon Sep 17 00:00:00 2001 From: Krzysztof Helt Date: Wed, 3 Mar 2010 19:41:44 +0100 Subject: ALSA: opti92x: use PnP data to select Master Control port The Master Control port (MC) is available as the last PnP resource (OPT005). Use this value instead fo guessing. Also, add some comments to the code. Signed-off-by: Krzysztof Helt Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/miro.c | 2 +- sound/isa/opti9xx/opti92x-ad1848.c | 120 ++++++++++++++++++++++++------------- 2 files changed, 79 insertions(+), 43 deletions(-) diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index b865e45..5913717 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -1558,7 +1558,7 @@ static int __devinit snd_card_miro_pnp(struct snd_miro *chip, err = pnp_activate_dev(devmc); if (err < 0) { - snd_printk(KERN_ERR "OPL syntg pnp configure failure: %d\n", + snd_printk(KERN_ERR "MC pnp configure failure: %d\n", err); return err; } diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index a4af53b..becd90d 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -144,12 +144,8 @@ struct snd_opti9xx { spinlock_t lock; + long wss_base; int irq; - -#ifdef CONFIG_PNP - struct pnp_dev *dev; - struct pnp_dev *devmpu; -#endif /* CONFIG_PNP */ }; static int snd_opti9xx_pnp_is_probed; @@ -159,12 +155,17 @@ static int snd_opti9xx_pnp_is_probed; static struct pnp_card_device_id snd_opti9xx_pnpids[] = { #ifndef OPTi93X /* OPTi 82C924 */ - { .id = "OPT0924", .devs = { { "OPT0000" }, { "OPT0002" } }, .driver_data = 0x0924 }, + { .id = "OPT0924", + .devs = { { "OPT0000" }, { "OPT0002" }, { "OPT0005" } }, + .driver_data = 0x0924 }, /* OPTi 82C925 */ - { .id = "OPT0925", .devs = { { "OPT9250" }, { "OPT0002" } }, .driver_data = 0x0925 }, + { .id = "OPT0925", + .devs = { { "OPT9250" }, { "OPT0002" }, { "OPT0005" } }, + .driver_data = 0x0925 }, #else /* OPTi 82C931/3 */ - { .id = "OPT0931", .devs = { { "OPT9310" }, { "OPT0002" } }, .driver_data = 0x0931 }, + { .id = "OPT0931", .devs = { { "OPT9310" }, { "OPT0002" } }, + .driver_data = 0x0931 }, #endif /* OPTi93X */ { .id = "" } }; @@ -207,24 +208,34 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, chip->hardware = hardware; strcpy(chip->name, snd_opti9xx_names[hardware]); - chip->mc_base_size = opti9xx_mc_size[hardware]; - spin_lock_init(&chip->lock); chip->irq = -1; +#ifndef OPTi93X +#ifdef CONFIG_PNP + if (isapnp && chip->mc_base) + /* PnP resource gives the least 10 bits */ + chip->mc_base |= 0xc00; +#endif /* CONFIG_PNP */ + else { + chip->mc_base = 0xf8c; + chip->mc_base_size = opti9xx_mc_size[hardware]; + } +#else + chip->mc_base_size = opti9xx_mc_size[hardware]; +#endif + switch (hardware) { #ifndef OPTi93X case OPTi9XX_HW_82C928: case OPTi9XX_HW_82C929: - chip->mc_base = 0xf8c; chip->password = (hardware == OPTi9XX_HW_82C928) ? 0xe2 : 0xe3; chip->pwd_reg = 3; break; case OPTi9XX_HW_82C924: case OPTi9XX_HW_82C925: - chip->mc_base = 0xf8c; chip->password = 0xe5; chip->pwd_reg = 3; break; @@ -292,7 +303,7 @@ static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip, spin_unlock_irqrestore(&chip->lock, flags); return retval; } - + static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, unsigned char value) { @@ -341,7 +352,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, - long wss_base, + long port, int irq, int dma1, int dma2, long mpu_port, int mpu_irq) { @@ -354,16 +365,23 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, switch (chip->hardware) { #ifndef OPTi93X case OPTi9XX_HW_82C924: + /* opti 929 mode (?), OPL3 clock output, audio enable */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(4), 0xf0, 0xfc); + /* enable wave audio */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(6), 0x02, 0x02); case OPTi9XX_HW_82C925: + /* enable WSS mode */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), 0x80, 0x80); + /* OPL3 FM synthesis */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(2), 0x00, 0x20); + /* disable Sound Blaster IRQ and DMA */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(3), 0xf0, 0xff); #ifdef CS4231 + /* cs4231/4248 fix enabled */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x02, 0x02); #else + /* cs4231/4248 fix disabled */ snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x00, 0x02); #endif /* CS4231 */ break; @@ -411,21 +429,26 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip, return -EINVAL; } - switch (wss_base) { - case 0x530: + /* PnP resource says it decodes only 10 bits of address */ + switch (port & 0x3ff) { + case 0x130: + chip->wss_base = 0x530; wss_base_bits = 0x00; break; - case 0x604: + case 0x204: + chip->wss_base = 0x604; wss_base_bits = 0x03; break; - case 0xe80: + case 0x280: + chip->wss_base = 0xe80; wss_base_bits = 0x01; break; - case 0xf40: + case 0x340: + chip->wss_base = 0xf40; wss_base_bits = 0x02; break; default: - snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", wss_base); + snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", port); goto __skip_base; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), wss_base_bits << 4, 0x30); @@ -487,7 +510,7 @@ __skip_base: #endif /* CS4231 || OPTi93X */ #ifndef OPTi93X - outb(irq_bits << 3 | dma_bits, wss_base); + outb(irq_bits << 3 | dma_bits, chip->wss_base); #else /* OPTi93X */ snd_opti9xx_write(chip, OPTi9XX_MC_REG(3), (irq_bits << 3 | dma_bits)); #endif /* OPTi93X */ @@ -729,15 +752,15 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, { struct pnp_dev *pdev; int err; + struct pnp_dev *devmpu; +#ifndef OPTi93X + struct pnp_dev *devmc; +#endif - chip->dev = pnp_request_card_device(card, pid->devs[0].id, NULL); - if (chip->dev == NULL) + pdev = pnp_request_card_device(card, pid->devs[0].id, NULL); + if (pdev == NULL) return -EBUSY; - chip->devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); - - pdev = chip->dev; - err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR "AUDIO pnp configure failure: %d\n", err); @@ -750,9 +773,24 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, chip->mc_indir_index = pnp_port_start(pdev, 3) + 2; chip->mc_indir_size = pnp_port_len(pdev, 3) - 2; #else - if (pid->driver_data != 0x0924) - port = pnp_port_start(pdev, 1); + devmc = pnp_request_card_device(card, pid->devs[2].id, NULL); + if (devmc == NULL) + return -EBUSY; + + err = pnp_activate_dev(devmc); + if (err < 0) { + snd_printk(KERN_ERR "MC pnp configure failure: %d\n", err); + return err; + } + + port = pnp_port_start(pdev, 1); fm_port = pnp_port_start(pdev, 2) + 8; + /* + * The MC(0) is never accessed and card does not + * include it in the PnP resource range. OPTI93x include it. + */ + chip->mc_base = pnp_port_start(devmc, 0) - 1; + chip->mc_base_size = pnp_port_len(devmc, 0) + 1; #endif /* OPTi93X */ irq = pnp_irq(pdev, 0); dma1 = pnp_dma(pdev, 0); @@ -760,16 +798,16 @@ static int __devinit snd_card_opti9xx_pnp(struct snd_opti9xx *chip, dma2 = pnp_dma(pdev, 1); #endif /* CS4231 || OPTi93X */ - pdev = chip->devmpu; - if (pdev && mpu_port > 0) { - err = pnp_activate_dev(pdev); + devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); + + if (devmpu && mpu_port > 0) { + err = pnp_activate_dev(devmpu); if (err < 0) { - snd_printk(KERN_ERR "AUDIO pnp configure failure\n"); + snd_printk(KERN_ERR "MPU401 pnp configure failure\n"); mpu_port = -1; - chip->devmpu = NULL; } else { - mpu_port = pnp_port_start(pdev, 0); - mpu_irq = pnp_irq(pdev, 0); + mpu_port = pnp_port_start(devmpu, 0); + mpu_irq = pnp_irq(devmpu, 0); } } return pid->driver_data; @@ -824,7 +862,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) if (error) return error; - error = snd_wss_create(card, port + 4, -1, irq, dma1, xdma2, + error = snd_wss_create(card, chip->wss_base + 4, -1, irq, dma1, xdma2, #ifdef OPTi93X WSS_HW_OPTI93X, WSS_HWSHARE_IRQ, #else @@ -865,10 +903,11 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) sprintf(card->shortname, "OPTi %s", card->driver); #if defined(CS4231) || defined(OPTi93X) sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d&%d", - card->shortname, pcm->name, port + 4, irq, dma1, xdma2); + card->shortname, pcm->name, + chip->wss_base + 4, irq, dma1, xdma2); #else sprintf(card->longname, "%s, %s at 0x%lx, irq %d, dma %d", - card->shortname, pcm->name, port + 4, irq, dma1); + card->shortname, pcm->name, chip->wss_base + 4, irq, dma1); #endif /* CS4231 || OPTi93X */ if (mpu_port <= 0 || mpu_port == SNDRV_AUTO_PORT) @@ -1062,9 +1101,6 @@ static int __devinit snd_opti9xx_pnp_probe(struct pnp_card_link *pcard, snd_card_free(card); return error; } - if (hw <= OPTi9XX_HW_82C930) - chip->mc_base -= 0x80; - error = snd_opti9xx_read_check(chip); if (error) { snd_printk(KERN_ERR "OPTI chip not found\n"); -- cgit v1.1 From faf4eb23d5fcb9a4728766a1e7bce9c6f2b43bd8 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 3 Mar 2010 09:16:18 +0100 Subject: ALSA: oxygen: change || to && In the original code the condition was always true (hopefully) because WM8776_HPLVOL is zero. Signed-off-by: Dan Carpenter Signed-off-by: Clemens Ladisch Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/pci/oxygen/xonar_wm87x6.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/oxygen/xonar_wm87x6.c b/sound/pci/oxygen/xonar_wm87x6.c index 7754db1..dbc4b89 100644 --- a/sound/pci/oxygen/xonar_wm87x6.c +++ b/sound/pci/oxygen/xonar_wm87x6.c @@ -68,7 +68,7 @@ static void wm8776_write(struct oxygen *chip, OXYGEN_SPI_CEN_LATCH_CLOCK_LO, (reg << 9) | value); if (reg < ARRAY_SIZE(data->wm8776_regs)) { - if (reg >= WM8776_HPLVOL || reg <= WM8776_DACMASTER) + if (reg >= WM8776_HPLVOL && reg <= WM8776_DACMASTER) value &= ~WM8776_UPDATE; data->wm8776_regs[reg] = value; } -- cgit v1.1 From b30477d5e2961bfd90ad4146c517871ca8a6bebc Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 3 Mar 2010 11:05:55 +0100 Subject: ALSA: timer - pass real event in snd_timer_notify1() to instance callback Do not use hardcoded SNDRV_TIMER_EVENT_START value. Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- include/sound/asound.h | 2 +- sound/core/timer.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/asound.h b/include/sound/asound.h index 1f57bb9..0985955 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -544,7 +544,7 @@ struct snd_rawmidi_status { * Timer section - /dev/snd/timer */ -#define SNDRV_TIMER_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 5) +#define SNDRV_TIMER_VERSION SNDRV_PROTOCOL_VERSION(2, 0, 6) enum { SNDRV_TIMER_CLASS_NONE = -1, diff --git a/sound/core/timer.c b/sound/core/timer.c index 8f8b17a..7394365 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -393,7 +393,7 @@ static void snd_timer_notify1(struct snd_timer_instance *ti, int event) event == SNDRV_TIMER_EVENT_CONTINUE) resolution = snd_timer_resolution(ti); if (ti->ccallback) - ti->ccallback(ti, SNDRV_TIMER_EVENT_START, &tstamp, resolution); + ti->ccallback(ti, event, &tstamp, resolution); if (ti->flags & SNDRV_TIMER_IFLG_SLAVE) return; timer = ti->timer; -- cgit v1.1 From e61e642c2a0dc283c52cec76a223ac0699773633 Mon Sep 17 00:00:00 2001 From: Jaroslav Kysela Date: Wed, 3 Mar 2010 11:11:57 +0100 Subject: ALSA: usbaudio - remove debug "SAMPLE BYTES" printk line Signed-off-by: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/usb/usbaudio.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index ea3eaa5..11b0826 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -2483,7 +2483,6 @@ static int parse_audio_format_i_type(struct snd_usb_audio *chip, sample_width, sample_bytes); } /* check the format byte size */ - printk(" XXXXX SAMPLE BYTES %d\n", sample_bytes); switch (sample_bytes) { case 1: pcm_format = SNDRV_PCM_FORMAT_S8; -- cgit v1.1 From 282572b5ab99cf27073210ca60b80dd085e1a469 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 3 Mar 2010 10:13:49 +0300 Subject: ALSA: riptide: clean up while loop If getpaths() returned an odd number this would be a buffer under-run and an endless loop. It turns out that getpaths() can only return even numbers, but let's make it easy for people auditing code. With the new code you don't need to look at getpaths(). This silences a smatch warning. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/riptide/riptide.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index 960a227..ad44626 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -1974,9 +1974,9 @@ snd_riptide_proc_read(struct snd_info_entry *entry, } snd_iprintf(buffer, "Paths:\n"); i = getpaths(cif, p); - while (i--) { - snd_iprintf(buffer, "%x->%x ", p[i - 1], p[i]); - i--; + while (i >= 2) { + i -= 2; + snd_iprintf(buffer, "%x->%x ", p[i], p[i + 1]); } snd_iprintf(buffer, "\n"); } -- cgit v1.1 From 50152dfaa7d09da85588b66fee7e8c7f541f631d Mon Sep 17 00:00:00 2001 From: Meelis Roos Date: Thu, 4 Mar 2010 20:33:07 +0200 Subject: ALSA: fix jazz16 compile (udelay) While trying to compile jazz16 isa sound driver on alpha (2.6.33+git), I found a compile failure in jazz16.c (udelay is unknown). Fix it by including delay.h. Signed-foo-by: Meelis Roos Signed-off-by: Takashi Iwai --- sound/isa/sb/jazz16.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/isa/sb/jazz16.c b/sound/isa/sb/jazz16.c index 8d21a3f..8ccbcdd 100644 --- a/sound/isa/sb/jazz16.c +++ b/sound/isa/sb/jazz16.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include #include -- cgit v1.1 From 2b9ddcb8b2ce6a44f0f969000f16b016caa64294 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Thu, 4 Mar 2010 19:46:18 +0100 Subject: ALSA: usb/audio.h: Fix field order Signed-off-by: Daniel Mack Cc: Clemens Ladisch Signed-off-by: Takashi Iwai --- include/linux/usb/audio.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/linux/usb/audio.h b/include/linux/usb/audio.h index 6bb2936..4d3e450 100644 --- a/include/linux/usb/audio.h +++ b/include/linux/usb/audio.h @@ -269,8 +269,8 @@ struct uac_format_type_i_ext_descriptor { __u8 bLength; __u8 bDescriptorType; __u8 bDescriptorSubtype; - __u8 bSubslotSize; __u8 bFormatType; + __u8 bSubslotSize; __u8 bBitResolution; __u8 bHeaderLength; __u8 bControlSize; -- cgit v1.1 From 89c0ac7cab2440a771ba1e2ab953186bc9c29786 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Mon, 8 Mar 2010 09:32:42 -0800 Subject: sound: fix opti92x-ad1848 build Fix 'else' placement in ifdef block so that build succeeds: sound/isa/opti9xx/opti92x-ad1848.c:221: error: 'else' without a previous 'if' Signed-off-by: Randy Dunlap Signed-off-by: Takashi Iwai --- sound/isa/opti9xx/opti92x-ad1848.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index becd90d..4d2d040 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -217,8 +217,9 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, if (isapnp && chip->mc_base) /* PnP resource gives the least 10 bits */ chip->mc_base |= 0xc00; + else #endif /* CONFIG_PNP */ - else { + { chip->mc_base = 0xf8c; chip->mc_base_size = opti9xx_mc_size[hardware]; } -- cgit v1.1