diff options
Diffstat (limited to 'sound/pci')
35 files changed, 2547 insertions, 278 deletions
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 61e35ecc57b8..c6b44102aa5b 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -33,6 +33,7 @@ config SND_ALS4000 select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM + select SND_SB_COMMON help Say Y here to include support for soundcards based on Avance Logic ALS4000 chips. @@ -215,6 +216,16 @@ config SND_CS46XX_NEW_DSP This works better than the old code, so say Y. +config SND_CS5530 + tristate "CS5530 Audio" + depends on SND && ISA_DMA_API + select SND_SB16_DSP + help + Say Y here to include support for audio on Cyrix/NatSemi CS5530 chips. + + To compile this driver as a module, choose M here: the module + will be called snd-cs5530. + config SND_CS5535AUDIO tristate "CS5535/CS5536 Audio" depends on SND && X86 && !X86_64 diff --git a/sound/pci/Makefile b/sound/pci/Makefile index e06736da9ef1..cd76e0293d06 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -12,6 +12,7 @@ snd-azt3328-objs := azt3328.o snd-bt87x-objs := bt87x.o snd-cmipci-objs := cmipci.o snd-cs4281-objs := cs4281.o +snd-cs5530-objs := cs5530.o snd-ens1370-objs := ens1370.o snd-ens1371-objs := ens1371.o snd-es1938-objs := es1938.o @@ -36,6 +37,7 @@ obj-$(CONFIG_SND_AZT3328) += snd-azt3328.o obj-$(CONFIG_SND_BT87X) += snd-bt87x.o obj-$(CONFIG_SND_CMIPCI) += snd-cmipci.o obj-$(CONFIG_SND_CS4281) += snd-cs4281.o +obj-$(CONFIG_SND_CS5530) += snd-cs5530.o obj-$(CONFIG_SND_ENS1370) += snd-ens1370.o obj-$(CONFIG_SND_ENS1371) += snd-ens1371.o obj-$(CONFIG_SND_ES1938) += snd-es1938.o diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index cb59f994c68f..05b4c8696941 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -239,7 +239,7 @@ struct snd_ali_image { struct snd_ali { - unsigned long irq; + int irq; unsigned long port; unsigned char revision; @@ -731,8 +731,7 @@ static void snd_ali_detect_spdif_rate(struct snd_ali *codec) return; } - count = 0; - while (count++ <= 50000) { + for (count = 0; count <= 50000; count++) { snd_ali_delay(codec, 6); bval = inb(ALI_REG(codec,ALI_SPDIF_CTRL + 1)); R2 = bval & 0x1F; @@ -2218,7 +2217,7 @@ static int __devinit snd_ali_create(struct snd_card *card, codec->card = card; codec->pci = pci; codec->irq = -1; - pci_read_config_byte(pci, PCI_REVISION_ID, &codec->revision); + codec->revision = pci->revision; codec->spdif_support = spdif_support; if (pcm_streams < 1) @@ -2343,7 +2342,7 @@ static int __devinit snd_ali_probe(struct pci_dev *pci, strcpy(card->driver, "ALI5451"); strcpy(card->shortname, "ALI 5451"); - sprintf(card->longname, "%s at 0x%lx, irq %li", + sprintf(card->longname, "%s at 0x%lx, irq %i", card->shortname, codec->port, codec->irq); snd_ali_printk("register card.\n"); diff --git a/sound/pci/als300.c b/sound/pci/als300.c index 8afcb98ca7bb..48cc39b771d9 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -88,8 +88,8 @@ #define PLAYBACK_BLOCK_COUNTER 0x9A #define RECORD_BLOCK_COUNTER 0x9B -#define DEBUG_CALLS 1 -#define DEBUG_PLAY_REC 1 +#define DEBUG_CALLS 0 +#define DEBUG_PLAY_REC 0 #if DEBUG_CALLS #define snd_als300_dbgcalls(format, args...) printk(format, ##args) @@ -733,7 +733,8 @@ static int __devinit snd_als300_create(struct snd_card *card, snd_als300_init(chip); - if (snd_als300_ac97(chip) < 0) { + err = snd_als300_ac97(chip); + if (err < 0) { snd_printk(KERN_WARNING "Could not create ac97\n"); snd_als300_free(chip); return err; diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index 7d8053b5e8d5..89184a424140 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -1639,15 +1639,12 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci, { struct snd_card *card; struct atiixp *chip; - unsigned char revision; int err; card = snd_card_new(index, id, THIS_MODULE, 0); if (card == NULL) return -ENOMEM; - pci_read_config_byte(pci, PCI_REVISION_ID, &revision); - strcpy(card->driver, spdif_aclink ? "ATIIXP" : "ATIIXP-SPDMA"); strcpy(card->shortname, "ATI IXP"); if ((err = snd_atiixp_create(card, pci, &chip)) < 0) @@ -1670,7 +1667,8 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci, snd_atiixp_chip_start(chip); snprintf(card->longname, sizeof(card->longname), - "%s rev %x with %s at %#lx, irq %i", card->shortname, revision, + "%s rev %x with %s at %#lx, irq %i", card->shortname, + pci->revision, chip->ac97[0] ? snd_ac97_get_short_name(chip->ac97[0]) : "?", chip->addr, chip->irq); diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index 904023fe4f26..ce752f84457a 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -1283,15 +1283,12 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci, { struct snd_card *card; struct atiixp_modem *chip; - unsigned char revision; int err; card = snd_card_new(index, id, THIS_MODULE, 0); if (card == NULL) return -ENOMEM; - pci_read_config_byte(pci, PCI_REVISION_ID, &revision); - strcpy(card->driver, "ATIIXP-MODEM"); strcpy(card->shortname, "ATI IXP Modem"); if ((err = snd_atiixp_create(card, pci, &chip)) < 0) @@ -1312,7 +1309,7 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci, snd_atiixp_chip_start(chip); sprintf(card->longname, "%s rev %x at 0x%lx, irq %i", - card->shortname, revision, chip->addr, chip->irq); + card->shortname, pci->revision, chip->addr, chip->irq); if ((err = snd_card_register(card)) < 0) goto __error; diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index 238154bb7a25..5ec1b6fcd548 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -341,11 +341,7 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) snd_card_free(card); return err; } - if ((err = pci_read_config_byte(pci, PCI_REVISION_ID, - &(chip->rev))) < 0) { - snd_card_free(card); - return err; - } + chip->rev = pci->revision; #ifdef CHIP_AU8830 if ((chip->rev) != 0xfe && (chip->rev) != 0xfa) { printk(KERN_ALERT diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h index aaac6e5b4767..a0420bc63f0b 100644 --- a/sound/pci/ca0106/ca0106.h +++ b/sound/pci/ca0106/ca0106.h @@ -590,7 +590,6 @@ struct snd_ca0106 { struct resource *res_port; int irq; - unsigned char revision; /* chip revision */ unsigned int serial; /* serial number */ unsigned short model; /* subsystem id */ diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 48f3f17c5170..fcab8fb97e38 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -168,6 +168,25 @@ MODULE_PARM_DESC(subsystem, "Force card subsystem model."); #include "ca0106.h" static struct snd_ca0106_details ca0106_chip_details[] = { + /* Sound Blaster X-Fi Extreme Audio. This does not have an AC97. 53SB079000000 */ + /* It is really just a normal SB Live 24bit. */ + /* + * CTRL:CA0111-WTLF + * ADC: WM8775SEDS + * DAC: CS4382-KQZ + */ + /* Tested: + * Playback on front, rear, center/lfe speakers + * Capture from Mic in. + * Not-Tested: + * Capture from Line in. + * Playback to digital out. + */ + { .serial = 0x10121102, + .name = "X-Fi Extreme Audio [SB0790]", + .gpio_type = 1, + .i2c_adc = 1 } , + /* New Dell Sound Blaster Live! 7.1 24bit. This does not have an AC97. */ /* AudigyLS[SB0310] */ { .serial = 0x10021102, .name = "AudigyLS [SB0310]", @@ -1293,13 +1312,12 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, } pci_set_master(pci); - /* read revision & serial */ - pci_read_config_byte(pci, PCI_REVISION_ID, &chip->revision); + /* read serial */ pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial); pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model); #if 1 printk(KERN_INFO "snd-ca0106: Model %04x Rev %08x Serial %08x\n", chip->model, - chip->revision, chip->serial); + pci->revision, chip->serial); #endif strcpy(card->driver, "CA0106"); strcpy(card->shortname, "CA0106"); diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index bef1f6d1859c..71d7aab9d869 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2897,6 +2897,10 @@ static int snd_cs46xx_free(struct snd_cs46xx *chip) } #endif +#ifdef CONFIG_PM + kfree(chip->saved_regs); +#endif + pci_disable_device(chip->pci); kfree(chip); return 0; @@ -3140,6 +3144,23 @@ static int snd_cs46xx_chip_init(struct snd_cs46xx *chip) /* * start and load DSP */ + +static void cs46xx_enable_stream_irqs(struct snd_cs46xx *chip) +{ + unsigned int tmp; + + snd_cs46xx_pokeBA0(chip, BA0_HICR, HICR_IEV | HICR_CHGM); + + tmp = snd_cs46xx_peek(chip, BA1_PFIE); + tmp &= ~0x0000f03f; + snd_cs46xx_poke(chip, BA1_PFIE, tmp); /* playback interrupt enable */ + + tmp = snd_cs46xx_peek(chip, BA1_CIE); + tmp &= ~0x0000003f; + tmp |= 0x00000001; + snd_cs46xx_poke(chip, BA1_CIE, tmp); /* capture interrupt enable */ +} + int __devinit snd_cs46xx_start_dsp(struct snd_cs46xx *chip) { unsigned int tmp; @@ -3214,19 +3235,7 @@ int __devinit snd_cs46xx_start_dsp(struct snd_cs46xx *chip) snd_cs46xx_proc_start(chip); - /* - * Enable interrupts on the part. - */ - snd_cs46xx_pokeBA0(chip, BA0_HICR, HICR_IEV | HICR_CHGM); - - tmp = snd_cs46xx_peek(chip, BA1_PFIE); - tmp &= ~0x0000f03f; - snd_cs46xx_poke(chip, BA1_PFIE, tmp); /* playback interrupt enable */ - - tmp = snd_cs46xx_peek(chip, BA1_CIE); - tmp &= ~0x0000003f; - tmp |= 0x00000001; - snd_cs46xx_poke(chip, BA1_CIE, tmp); /* capture interrupt enable */ + cs46xx_enable_stream_irqs(chip); #ifndef CONFIG_SND_CS46XX_NEW_DSP /* set the attenuation to 0dB */ @@ -3665,11 +3674,19 @@ static struct cs_card_type __devinitdata cards[] = { * APM support */ #ifdef CONFIG_PM +static unsigned int saved_regs[] = { + BA0_ACOSV, + BA0_ASER_FADDR, + BA0_ASER_MASTER, + BA1_PVOL, + BA1_CVOL, +}; + int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state) { struct snd_card *card = pci_get_drvdata(pci); struct snd_cs46xx *chip = card->private_data; - int amp_saved; + int i, amp_saved; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); chip->in_suspend = 1; @@ -3680,6 +3697,10 @@ int snd_cs46xx_suspend(struct pci_dev *pci, pm_message_t state) snd_ac97_suspend(chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]); snd_ac97_suspend(chip->ac97[CS46XX_SECONDARY_CODEC_INDEX]); + /* save some registers */ + for (i = 0; i < ARRAY_SIZE(saved_regs); i++) + chip->saved_regs[i] = snd_cs46xx_peekBA0(chip, saved_regs[i]); + amp_saved = chip->amplifier; /* turn off amp */ chip->amplifier_ctrl(chip, -chip->amplifier); @@ -3698,7 +3719,7 @@ int snd_cs46xx_resume(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); struct snd_cs46xx *chip = card->private_data; - int amp_saved; + int i, amp_saved; pci_set_power_state(pci, PCI_D0); pci_restore_state(pci); @@ -3716,6 +3737,16 @@ int snd_cs46xx_resume(struct pci_dev *pci) snd_cs46xx_chip_init(chip); + snd_cs46xx_reset(chip); +#ifdef CONFIG_SND_CS46XX_NEW_DSP + cs46xx_dsp_resume(chip); + /* restore some registers */ + for (i = 0; i < ARRAY_SIZE(saved_regs); i++) + snd_cs46xx_pokeBA0(chip, saved_regs[i], chip->saved_regs[i]); +#else + snd_cs46xx_download_image(chip); +#endif + #if 0 snd_cs46xx_codec_write(chip, BA0_AC97_GENERAL_PURPOSE, chip->ac97_general_purpose); @@ -3730,6 +3761,13 @@ 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]); + /* reset playback/capture */ + snd_cs46xx_set_play_sample_rate(chip, 8000); + snd_cs46xx_set_capture_sample_rate(chip, 8000); + snd_cs46xx_proc_start(chip); + + cs46xx_enable_stream_irqs(chip); + if (amp_saved) chip->amplifier_ctrl(chip, 1); /* turn amp on */ else @@ -3896,6 +3934,15 @@ int __devinit snd_cs46xx_create(struct snd_card *card, snd_cs46xx_proc_init(card, chip); +#ifdef CONFIG_PM + chip->saved_regs = kmalloc(sizeof(*chip->saved_regs) * + ARRAY_SIZE(saved_regs), GFP_KERNEL); + if (!chip->saved_regs) { + snd_cs46xx_free(chip); + return -ENOMEM; + } +#endif + chip->active_ctrl(chip, -1); /* disable CLKRUN */ snd_card_set_dev(card, &pci->dev); diff --git a/sound/pci/cs46xx/cs46xx_lib.h b/sound/pci/cs46xx/cs46xx_lib.h index f75750c2bd24..20dcd72f06c1 100644 --- a/sound/pci/cs46xx/cs46xx_lib.h +++ b/sound/pci/cs46xx/cs46xx_lib.h @@ -86,6 +86,9 @@ static inline unsigned int snd_cs46xx_peekBA0(struct snd_cs46xx *chip, unsigned struct dsp_spos_instance *cs46xx_dsp_spos_create (struct snd_cs46xx * chip); void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip); int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * module); +#ifdef CONFIG_PM +int cs46xx_dsp_resume(struct snd_cs46xx * chip); +#endif struct dsp_symbol_entry *cs46xx_dsp_lookup_symbol (struct snd_cs46xx * chip, char * symbol_name, int symbol_type); #ifdef CONFIG_PROC_FS diff --git a/sound/pci/cs46xx/dsp_spos.c b/sound/pci/cs46xx/dsp_spos.c index 336e77e2600c..590b35d91df2 100644 --- a/sound/pci/cs46xx/dsp_spos.c +++ b/sound/pci/cs46xx/dsp_spos.c @@ -306,13 +306,59 @@ void cs46xx_dsp_spos_destroy (struct snd_cs46xx * chip) mutex_unlock(&chip->spos_mutex); } +static int dsp_load_parameter(struct snd_cs46xx *chip, + struct dsp_segment_desc *parameter) +{ + u32 doffset, dsize; + + if (!parameter) { + snd_printdd("dsp_spos: module got no parameter segment\n"); + return 0; + } + + doffset = (parameter->offset * 4 + DSP_PARAMETER_BYTE_OFFSET); + dsize = parameter->size * 4; + + snd_printdd("dsp_spos: " + "downloading parameter data to chip (%08x-%08x)\n", + doffset,doffset + dsize); + if (snd_cs46xx_download (chip, parameter->data, doffset, dsize)) { + snd_printk(KERN_ERR "dsp_spos: " + "failed to download parameter data to DSP\n"); + return -EINVAL; + } + return 0; +} + +static int dsp_load_sample(struct snd_cs46xx *chip, + struct dsp_segment_desc *sample) +{ + u32 doffset, dsize; + + if (!sample) { + snd_printdd("dsp_spos: module got no sample segment\n"); + return 0; + } + + doffset = (sample->offset * 4 + DSP_SAMPLE_BYTE_OFFSET); + dsize = sample->size * 4; + + snd_printdd("dsp_spos: downloading sample data to chip (%08x-%08x)\n", + doffset,doffset + dsize); + + if (snd_cs46xx_download (chip,sample->data,doffset,dsize)) { + snd_printk(KERN_ERR "dsp_spos: failed to sample data to DSP\n"); + return -EINVAL; + } + return 0; +} + int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * module) { struct dsp_spos_instance * ins = chip->dsp_spos_instance; struct dsp_segment_desc * code = get_segment_desc (module,SEGTYPE_SP_PROGRAM); - struct dsp_segment_desc * parameter = get_segment_desc (module,SEGTYPE_SP_PARAMETER); - struct dsp_segment_desc * sample = get_segment_desc (module,SEGTYPE_SP_SAMPLE); u32 doffset, dsize; + int err; if (ins->nmodules == DSP_MAX_MODULES - 1) { snd_printk(KERN_ERR "dsp_spos: to many modules loaded into DSP\n"); @@ -326,49 +372,20 @@ int cs46xx_dsp_load_module (struct snd_cs46xx * chip, struct dsp_module_desc * m snd_cs46xx_clear_BA1(chip, DSP_PARAMETER_BYTE_OFFSET, DSP_PARAMETER_BYTE_SIZE); } - if (parameter == NULL) { - snd_printdd("dsp_spos: module got no parameter segment\n"); - } else { - if (ins->nmodules > 0) { - snd_printk(KERN_WARNING "dsp_spos: WARNING current parameter data may be overwriten!\n"); - } - - doffset = (parameter->offset * 4 + DSP_PARAMETER_BYTE_OFFSET); - dsize = parameter->size * 4; - - snd_printdd("dsp_spos: downloading parameter data to chip (%08x-%08x)\n", - doffset,doffset + dsize); - - if (snd_cs46xx_download (chip, parameter->data, doffset, dsize)) { - snd_printk(KERN_ERR "dsp_spos: failed to download parameter data to DSP\n"); - return -EINVAL; - } - } + err = dsp_load_parameter(chip, get_segment_desc(module, + SEGTYPE_SP_PARAMETER)); + if (err < 0) + return err; if (ins->nmodules == 0) { snd_printdd("dsp_spos: clearing sample area\n"); snd_cs46xx_clear_BA1(chip, DSP_SAMPLE_BYTE_OFFSET, DSP_SAMPLE_BYTE_SIZE); } - if (sample == NULL) { - snd_printdd("dsp_spos: module got no sample segment\n"); - } else { - if (ins->nmodules > 0) { - snd_printk(KERN_WARNING "dsp_spos: WARNING current sample data may be overwriten\n"); - } - - doffset = (sample->offset * 4 + DSP_SAMPLE_BYTE_OFFSET); - dsize = sample->size * 4; - - snd_printdd("dsp_spos: downloading sample data to chip (%08x-%08x)\n", - doffset,doffset + dsize); - - if (snd_cs46xx_download (chip,sample->data,doffset,dsize)) { - snd_printk(KERN_ERR "dsp_spos: failed to sample data to DSP\n"); - return -EINVAL; - } - } - + err = dsp_load_sample(chip, get_segment_desc(module, + SEGTYPE_SP_SAMPLE)); + if (err < 0) + return err; if (ins->nmodules == 0) { snd_printdd("dsp_spos: clearing code area\n"); @@ -986,7 +1003,10 @@ _map_task_tree (struct snd_cs46xx *chip, char * name, u32 dest, u32 size) return NULL; } - strcpy(ins->tasks[ins->ntask].task_name,name); + if (name) + strcpy(ins->tasks[ins->ntask].task_name, name); + else + strcpy(ins->tasks[ins->ntask].task_name, "(NULL)"); ins->tasks[ins->ntask].address = dest; ins->tasks[ins->ntask].size = size; @@ -995,7 +1015,8 @@ _map_task_tree (struct snd_cs46xx *chip, char * name, u32 dest, u32 size) desc = (ins->tasks + ins->ntask); ins->ntask++; - add_symbol (chip,name,dest,SYMBOL_PARAMETER); + if (name) + add_symbol (chip,name,dest,SYMBOL_PARAMETER); return desc; } @@ -1006,6 +1027,7 @@ cs46xx_dsp_create_scb (struct snd_cs46xx *chip, char * name, u32 * scb_data, u32 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"); @@ -1023,6 +1045,7 @@ cs46xx_dsp_create_task_tree (struct snd_cs46xx *chip, char * name, u32 * task_da desc = _map_task_tree (chip,name,dest,size); if (desc) { + desc->data = task_data; _dsp_create_task_tree(chip,task_data,dest,size); } else { snd_printk(KERN_ERR "dsp_spos: failed to map TASK\n"); @@ -1320,8 +1343,10 @@ int cs46xx_dsp_scb_and_task_init (struct snd_cs46xx *chip) 0x0000ffff }; - /* dirty hack ... */ - _dsp_create_task_tree (chip,(u32 *)&mix2_ostream_spb,WRITE_BACK_SPB,2); + if (!cs46xx_dsp_create_task_tree(chip, NULL, + (u32 *)&mix2_ostream_spb, + WRITE_BACK_SPB, 2)) + goto _fail_end; } /* input sample converter */ @@ -1622,7 +1647,6 @@ static int cs46xx_dsp_async_init (struct snd_cs46xx *chip, return 0; } - static void cs46xx_dsp_disable_spdif_hw (struct snd_cs46xx *chip) { struct dsp_spos_instance * ins = chip->dsp_spos_instance; @@ -1894,3 +1918,61 @@ int cs46xx_dsp_set_iec958_volume (struct snd_cs46xx * chip, u16 left, u16 right) return 0; } + +#ifdef CONFIG_PM +int cs46xx_dsp_resume(struct snd_cs46xx * chip) +{ + struct dsp_spos_instance * ins = chip->dsp_spos_instance; + int i, err; + + /* clear parameter, sample and code areas */ + snd_cs46xx_clear_BA1(chip, DSP_PARAMETER_BYTE_OFFSET, + DSP_PARAMETER_BYTE_SIZE); + snd_cs46xx_clear_BA1(chip, DSP_SAMPLE_BYTE_OFFSET, + DSP_SAMPLE_BYTE_SIZE); + snd_cs46xx_clear_BA1(chip, DSP_CODE_BYTE_OFFSET, DSP_CODE_BYTE_SIZE); + + for (i = 0; i < ins->nmodules; i++) { + struct dsp_module_desc *module = &ins->modules[i]; + struct dsp_segment_desc *seg; + u32 doffset, dsize; + + seg = get_segment_desc(module, SEGTYPE_SP_PARAMETER); + err = dsp_load_parameter(chip, seg); + if (err < 0) + return err; + + seg = get_segment_desc(module, SEGTYPE_SP_SAMPLE); + err = dsp_load_sample(chip, seg); + if (err < 0) + return err; + + seg = get_segment_desc(module, SEGTYPE_SP_PROGRAM); + if (!seg) + continue; + + doffset = seg->offset * 4 + module->load_address * 4 + + DSP_CODE_BYTE_OFFSET; + dsize = seg->size * 4; + err = snd_cs46xx_download(chip, + ins->code.data + module->load_address, + doffset, dsize); + if (err < 0) + return err; + } + + for (i = 0; i < ins->ntask; i++) { + struct dsp_task_descriptor *t = &ins->tasks[i]; + _dsp_create_task_tree(chip, t->data, t->address, t->size); + } + + for (i = 0; i < ins->nscb; i++) { + struct dsp_scb_descriptor *s = &ins->scbs[i]; + if (s->deleted) + continue; + _dsp_create_scb(chip, s->data, s->address); + } + + return 0; +} +#endif diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c new file mode 100644 index 000000000000..240a0a462209 --- /dev/null +++ b/sound/pci/cs5530.c @@ -0,0 +1,306 @@ +/* + * cs5530.c - Initialisation code for Cyrix/NatSemi VSA1 softaudio + * + * (C) Copyright 2007 Ash Willis <ashwillis@programmer.net> + * (C) Copyright 2003 Red Hat Inc <alan@redhat.com> + * + * This driver was ported (shamelessly ripped ;) from oss/kahlua.c but I did + * mess with it a bit. The chip seems to have to have trouble with full duplex + * mode. If we're recording in 8bit 8000kHz, say, and we then attempt to + * simultaneously play back audio at 16bit 44100kHz, the device actually plays + * back in the same format in which it is capturing. By forcing the chip to + * always play/capture in 16/44100, we can let alsa-lib convert the samples and + * that way we can hack up some full duplex audio. + * + * XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems. + * The older version (VSA1) provides fairly good soundblaster emulation + * although there are a couple of bugs: large DMA buffers break record, + * and the MPU event handling seems suspect. VSA2 allows the native driver + * to control the AC97 audio engine directly and requires a different driver. + * + * Thanks to National Semiconductor for providing the needed information + * on the XpressAudio(tm) internals. + * + * 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, 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. + * + * TO DO: + * Investigate whether we can portably support Cognac (5520) in the + * same manner. + */ + +#include <sound/driver.h> +#include <linux/delay.h> +#include <linux/moduleparam.h> +#include <linux/pci.h> +#include <sound/core.h> +#include <sound/sb.h> +#include <sound/initval.h> + +MODULE_AUTHOR("Ash Willis"); +MODULE_DESCRIPTION("CS5530 Audio"); +MODULE_LICENSE("GPL"); + +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; + +struct snd_cs5530 { + struct snd_card *card; + struct pci_dev *pci; + struct snd_sb *sb; + unsigned long pci_base; +}; + +static struct pci_device_id snd_cs5530_ids[] = { + {PCI_VENDOR_ID_CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO, PCI_ANY_ID, + PCI_ANY_ID, 0, 0}, + {0,} +}; + +MODULE_DEVICE_TABLE(pci, snd_cs5530_ids); + +static int snd_cs5530_free(struct snd_cs5530 *chip) +{ + pci_release_regions(chip->pci); + pci_disable_device(chip->pci); + kfree(chip); + return 0; +} + +static int snd_cs5530_dev_free(struct snd_device *device) +{ + struct snd_cs5530 *chip = device->device_data; + return snd_cs5530_free(chip); +} + +static void __devexit snd_cs5530_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static u8 __devinit snd_cs5530_mixer_read(unsigned long io, u8 reg) +{ + outb(reg, io + 4); + udelay(20); + reg = inb(io + 5); + udelay(20); + return reg; +} + +static int __devinit snd_cs5530_create(struct snd_card *card, + struct pci_dev *pci, + struct snd_cs5530 **rchip) +{ + struct snd_cs5530 *chip; + unsigned long sb_base; + u8 irq, dma8, dma16 = 0; + u16 map; + void __iomem *mem; + int err; + + static struct snd_device_ops ops = { + .dev_free = snd_cs5530_dev_free, + }; + *rchip = NULL; + + err = pci_enable_device(pci); + if (err < 0) + return err; + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + pci_disable_device(pci); + return -ENOMEM; + } + + chip->card = card; + chip->pci = pci; + + err = pci_request_regions(pci, "CS5530"); + if (err < 0) { + kfree(chip); + pci_disable_device(pci); + return err; + } + chip->pci_base = pci_resource_start(pci, 0); + + mem = ioremap_nocache(chip->pci_base, pci_resource_len(pci, 0)); + if (mem == NULL) { + kfree(chip); + pci_disable_device(pci); + return -EBUSY; + } + + map = readw(mem + 0x18); + iounmap(mem); + + /* Map bits + 0:1 * 0x20 + 0x200 = sb base + 2 sb enable + 3 adlib enable + 5 MPU enable 0x330 + 6 MPU enable 0x300 + + The other bits may be used internally so must be masked */ + + sb_base = 0x220 + 0x20 * (map & 3); + + if (map & (1<<2)) + printk(KERN_INFO "CS5530: XpressAudio at 0x%lx\n", sb_base); + else { + printk(KERN_ERR "Could not find XpressAudio!\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (map & (1<<5)) + printk(KERN_INFO "CS5530: MPU at 0x300\n"); + else if (map & (1<<6)) + printk(KERN_INFO "CS5530: MPU at 0x330\n"); + + irq = snd_cs5530_mixer_read(sb_base, 0x80) & 0x0F; + dma8 = snd_cs5530_mixer_read(sb_base, 0x81); + + if (dma8 & 0x20) + dma16 = 5; + else if (dma8 & 0x40) + dma16 = 6; + else if (dma8 & 0x80) + dma16 = 7; + else { + printk(KERN_ERR "CS5530: No 16bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (dma8 & 0x01) + dma8 = 0; + else if (dma8 & 02) + dma8 = 1; + else if (dma8 & 0x08) + dma8 = 3; + else { + printk(KERN_ERR "CS5530: No 8bit DMA enabled\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + if (irq & 1) + irq = 9; + else if (irq & 2) + irq = 5; + else if (irq & 4) + irq = 7; + else if (irq & 8) + irq = 10; + else { + printk(KERN_ERR "CS5530: SoundBlaster IRQ not set\n"); + snd_cs5530_free(chip); + return -ENODEV; + } + + printk(KERN_INFO "CS5530: IRQ: %d DMA8: %d DMA16: %d\n", irq, dma8, + dma16); + + err = snd_sbdsp_create(card, sb_base, irq, snd_sb16dsp_interrupt, dma8, + dma16, SB_HW_CS5530, &chip->sb); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create SoundBlaster\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_sb16dsp_pcm(chip->sb, 0, &chip->sb->pcm); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create PCM\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_sbmixer_new(chip->sb); + if (err < 0) { + printk(KERN_ERR "CS5530: Could not create Mixer\n"); + snd_cs5530_free(chip); + return err; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + snd_cs5530_free(chip); + return err; + } + + snd_card_set_dev(card, &pci->dev); + *rchip = chip; + return 0; +} + +static int __devinit snd_cs5530_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + static int dev; + struct snd_card *card; + struct snd_cs5530 *chip = NULL; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) { + dev++; + return -ENOENT; + } + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + + if (card == NULL) + return -ENOMEM; + + err = snd_cs5530_create(card, pci, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + + strcpy(card->driver, "CS5530"); + strcpy(card->shortname, "CS5530 Audio"); + sprintf(card->longname, "%s at 0x%lx", card->shortname, chip->pci_base); + + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + pci_set_drvdata(pci, card); + dev++; + return 0; +} + +static struct pci_driver driver = { + .name = "CS5530_Audio", + .id_table = snd_cs5530_ids, + .probe = snd_cs5530_probe, + .remove = __devexit_p(snd_cs5530_remove), +}; + +static int __init alsa_card_cs5530_init(void) +{ + return pci_register_driver(&driver); +} + +static void __exit alsa_card_cs5530_exit(void) +{ + pci_unregister_driver(&driver); +} + +module_init(alsa_card_cs5530_init) +module_exit(alsa_card_cs5530_exit) + diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index dbc805c33fc4..404ae1be0a4b 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -51,9 +51,15 @@ #define HANA_FILENAME "emu/hana.fw" #define DOCK_FILENAME "emu/audio_dock.fw" +#define EMU1010B_FILENAME "emu/emu1010b.fw" +#define MICRO_DOCK_FILENAME "emu/micro_dock.fw" +#define EMU1010_NOTEBOOK_FILENAME "emu/emu1010_notebook.fw" MODULE_FIRMWARE(HANA_FILENAME); MODULE_FIRMWARE(DOCK_FILENAME); +MODULE_FIRMWARE(EMU1010B_FILENAME); +MODULE_FIRMWARE(MICRO_DOCK_FILENAME); +MODULE_FIRMWARE(EMU1010_NOTEBOOK_FILENAME); /************************************************************************* @@ -660,10 +666,12 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file return err; } snd_printk(KERN_INFO "firmware size=0x%zx\n", fw_entry->size); +#if 0 if (fw_entry->size != 0x133a4) { snd_printk(KERN_ERR "firmware: %s wrong size.\n",filename); return -EINVAL; } +#endif /* The FPGA is a Xilinx Spartan IIE XC2S50E */ /* GPIO7 -> FPGA PGMN @@ -694,6 +702,37 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 * emu, const char * file return 0; } +/* + * EMU-1010 - details found out from this driver, official MS Win drivers, + * testing the card: + * + * Audigy2 (aka Alice2): + * --------------------- + * * communication over PCI + * * conversion of 32-bit data coming over EMU32 links from HANA FPGA + * to 2 x 16-bit, using internal DSP instructions + * * slave mode, clock supplied by HANA + * * linked to HANA using: + * 32 x 32-bit serial EMU32 output channels + * 16 x EMU32 input channels + * (?) x I2S I/O channels (?) + * + * FPGA (aka HANA): + * --------------- + * * provides all (?) physical inputs and outputs of the card + * (ADC, DAC, SPDIF I/O, ADAT I/O, etc.) + * * provides clock signal for the card and Alice2 + * * two crystals - for 44.1kHz and 48kHz multiples + * * provides internal routing of signal sources to signal destinations + * * inputs/outputs to Alice2 - see above + * + * Current status of the driver: + * ---------------------------- + * * only 44.1/48kHz supported (the MS Win driver supports up to 192 kHz) + * * PCM device nb. 2: + * 16 x 16-bit playback - snd_emu10k1_fx8010_playback_ops + * 16 x 32-bit capture - snd_emu10k1_capture_efx_ops + */ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) { unsigned int i; @@ -727,7 +766,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) /* ID, should read & 0x7f = 0x55. (Bit 7 is the IRQ bit) */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); snd_printdd("reg1=0x%x\n",reg); - if (reg == 0x55) { + if ((reg & 0x3f) == 0x15) { /* FPGA netlist already present so clear it */ /* Return to programming mode */ @@ -735,19 +774,32 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) } snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); snd_printdd("reg2=0x%x\n",reg); - if (reg == 0x55) { + if ((reg & 0x3f) == 0x15) { /* FPGA failed to return to programming mode */ + snd_printk(KERN_INFO "emu1010: FPGA failed to return to programming mode\n"); return -ENODEV; } snd_printk(KERN_INFO "emu1010: EMU_HANA_ID=0x%x\n",reg); - if ((err = snd_emu1010_load_firmware(emu, HANA_FILENAME)) != 0) { - snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file %s failed\n", HANA_FILENAME); - return err; + if (emu->card_capabilities->emu1010 == 1) { + if ((err = snd_emu1010_load_firmware(emu, HANA_FILENAME)) != 0) { + snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file %s failed\n", HANA_FILENAME); + return err; + } + } else if (emu->card_capabilities->emu1010 == 2) { + if ((err = snd_emu1010_load_firmware(emu, EMU1010B_FILENAME)) != 0) { + snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", EMU1010B_FILENAME); + return err; + } + } else if (emu->card_capabilities->emu1010 == 3) { + if ((err = snd_emu1010_load_firmware(emu, EMU1010_NOTEBOOK_FILENAME)) != 0) { + snd_printk(KERN_INFO "emu1010: Loading Firmware file %s failed\n", EMU1010_NOTEBOOK_FILENAME); + return err; + } } /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); - if (reg != 0x55) { + if ((reg & 0x3f) != 0x15) { /* FPGA failed to be programmed */ snd_printk(KERN_INFO "emu1010: Loading Hana Firmware file failed, reg=0x%x\n", reg); return -ENODEV; @@ -850,6 +902,27 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) EMU_DST_ALICE2_EMU32_6, EMU_SRC_DOCK_ADC2_LEFT1); snd_emu1010_fpga_link_dst_src_write(emu, EMU_DST_ALICE2_EMU32_7, EMU_SRC_DOCK_ADC2_RIGHT1); + /* Pavel Hofman - setting defaults for 8 more capture channels + * Defaults only, users will set their own values anyways, let's + * just copy/paste. + */ + + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_8, EMU_SRC_DOCK_MIC_A1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_9, EMU_SRC_DOCK_MIC_B1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_A, EMU_SRC_HAMOA_ADC_LEFT2); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_B, EMU_SRC_HAMOA_ADC_LEFT2); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_C, EMU_SRC_DOCK_ADC1_LEFT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_D, EMU_SRC_DOCK_ADC1_RIGHT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_E, EMU_SRC_DOCK_ADC2_LEFT1); + snd_emu1010_fpga_link_dst_src_write(emu, + EMU_DST_ALICE2_EMU32_F, EMU_SRC_DOCK_ADC2_RIGHT1); #endif #if 0 /* Original */ @@ -943,16 +1016,27 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 * emu) /* Return to Audio Dock programming mode */ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware\n"); snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, EMU_HANA_FPGA_CONFIG_AUDIODOCK ); - if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) { - return err; + if (emu->card_capabilities->emu1010 == 1) { + if ((err = snd_emu1010_load_firmware(emu, DOCK_FILENAME)) != 0) { + return err; + } + } else if (emu->card_capabilities->emu1010 == 2) { + if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { + return err; + } + } else if (emu->card_capabilities->emu1010 == 3) { + if ((err = snd_emu1010_load_firmware(emu, MICRO_DOCK_FILENAME)) != 0) { + return err; + } } + snd_emu1010_fpga_write(emu, EMU_HANA_FPGA_CONFIG, 0 ); snd_emu1010_fpga_read(emu, EMU_HANA_IRQ_STATUS, ® ); snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_IRQ_STATUS=0x%x\n",reg); /* ID, should read & 0x7f = 0x55 when FPGA programmed. */ snd_emu1010_fpga_read(emu, EMU_HANA_ID, ® ); snd_printk(KERN_INFO "emu1010: EMU_HANA+DOCK_ID=0x%x\n",reg); - if (reg != 0x55) { + if ((reg & 0x3f) != 0x15) { /* FPGA failed to be programmed */ snd_printk(KERN_INFO "emu1010: Loading Audio Dock Firmware file failed, reg=0x%x\n", reg); return 0; @@ -1227,9 +1311,15 @@ static struct snd_emu_chip_details emu_chip_details[] = { .emu10k2_chip = 1, .ca0108_chip = 1, .ca_cardbus_chip = 1, - .spi_dac = 1, - .i2c_adc = 1, - .spk71 = 1} , + .spk71 = 1 , + .emu1010 = 3} , + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40041102, + .driver = "Audigy2", .name = "E-mu 1010b PCI [MAEM????]", + .id = "EMU1010", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1 , + .emu1010 = 2} , {.vendor = 0x1102, .device = 0x0008, .driver = "Audigy2", .name = "Audigy 2 Value [Unknown]", .id = "Audigy2", @@ -1511,7 +1601,6 @@ int __devinit snd_emu10k1_create(struct snd_card *card, struct snd_emu10k1 *emu; int idx, err; int is_audigy; - unsigned char revision; unsigned int silent_page; const struct snd_emu_chip_details *c; static struct snd_device_ops ops = { @@ -1543,8 +1632,7 @@ int __devinit snd_emu10k1_create(struct snd_card *card, emu->synth = NULL; emu->get_synth_voice = NULL; /* read revision & serial */ - pci_read_config_byte(pci, PCI_REVISION_ID, &revision); - emu->revision = revision; + emu->revision = pci->revision; pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &emu->serial); pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &emu->model); snd_printdd("vendor=0x%x, device=0x%x, subsystem_vendor_id=0x%x, subsystem_id=0x%x\n",pci->vendor, pci->device, emu->serial, emu->model); @@ -1665,12 +1753,13 @@ int __devinit snd_emu10k1_create(struct snd_card *card, emu->fx8010.extout_mask = extout_mask; emu->enable_ir = enable_ir; + if (emu->card_capabilities->ca_cardbus_chip) { + if ((err = snd_emu10k1_cardbus_init(emu)) < 0) + goto error; + } if (emu->card_capabilities->ecard) { if ((err = snd_emu10k1_ecard_init(emu)) < 0) goto error; - } else if (emu->card_capabilities->ca_cardbus_chip) { - if ((err = snd_emu10k1_cardbus_init(emu)) < 0) - goto error; } else if (emu->card_capabilities->emu1010) { if ((err = snd_emu10k1_emu1010_init(emu)) < 0) { snd_emu10k1_free(emu); @@ -1816,10 +1905,10 @@ void snd_emu10k1_suspend_regs(struct snd_emu10k1 *emu) void snd_emu10k1_resume_init(struct snd_emu10k1 *emu) { + if (emu->card_capabilities->ca_cardbus_chip) + snd_emu10k1_cardbus_init(emu); if (emu->card_capabilities->ecard) snd_emu10k1_ecard_init(emu); - else if (emu->card_capabilities->ca_cardbus_chip) - snd_emu10k1_cardbus_init(emu); else if (emu->card_capabilities->emu1010) snd_emu10k1_emu1010_init(emu); else diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index bb0fec7f7e1b..e4af7a9b808c 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -942,7 +942,7 @@ static int __devinit snd_emu10k1x_create(struct snd_card *card, pci_set_master(pci); /* read revision & serial */ - pci_read_config_byte(pci, PCI_REVISION_ID, &chip->revision); + chip->revision = pci->revision; pci_read_config_dword(pci, PCI_SUBSYSTEM_VENDOR_ID, &chip->serial); pci_read_config_word(pci, PCI_SUBSYSTEM_ID, &chip->model); snd_printk(KERN_INFO "Model %04x Rev %08x Serial %08x\n", chip->model, diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index c02012cccd8e..7206c0fa06f2 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1123,6 +1123,11 @@ snd_emu10k1_init_stereo_onoff_control(struct snd_emu10k1_fx8010_control_gpr *ctl ctl->translation = EMU10K1_GPR_TRANSLATION_ONOFF; } +/* + * Used for emu1010 - conversion from 32-bit capture inputs from HANA + * to 2 x 16-bit registers in audigy - their values are read via DMA. + * Conversion is performed by Audigy DSP instructions of FX8010. + */ static int snd_emu10k1_audigy_dsp_convert_32_to_2x16( struct snd_emu10k1_fx8010_code *icode, u32 *ptr, int tmp, int bit_shifter16, @@ -1193,7 +1198,11 @@ static int __devinit _snd_emu10k1_audigy_init_efx(struct snd_emu10k1 *emu) snd_emu10k1_ptr_write(emu, A_DBG, 0, (emu->fx8010.dbg = 0) | A_DBG_SINGLE_STEP); #if 1 - /* PCM front Playback Volume (independent from stereo mix) */ + /* PCM front Playback Volume (independent from stereo mix) + * playback = 0 + ( gpr * FXBUS_PCM_LEFT_FRONT >> 31) + * where gpr contains attenuation from corresponding mixer control + * (snd_emu10k1_init_stereo_control) + */ A_OP(icode, &ptr, iMAC0, A_GPR(playback), A_C_00000000, A_GPR(gpr), A_FXBUS(FXBUS_PCM_LEFT_FRONT)); A_OP(icode, &ptr, iMAC0, A_GPR(playback+1), A_C_00000000, A_GPR(gpr+1), A_FXBUS(FXBUS_PCM_RIGHT_FRONT)); snd_emu10k1_init_stereo_control(&controls[nctl++], "PCM Front Playback Volume", gpr, 100); @@ -1549,7 +1558,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) if (emu->card_capabilities->emu1010) { snd_printk("EMU inputs on\n"); - /* Capture 8 channels of S32_LE sound */ + /* Capture 16 (originally 8) channels of S32_LE sound */ /* printk("emufx.c: gpr=0x%x, tmp=0x%x\n",gpr, tmp); */ /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */ @@ -1560,6 +1569,11 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) ); /* Right ADC in 1 of 2 */ gpr_map[gpr++] = 0x00000000; + /* Delaying by one sample: instead of copying the input + * value A_P16VIN to output A_FXBUS2 as in the first channel, + * we use an auxiliary register, delaying the value by one + * sample + */ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) ); A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000); gpr_map[gpr++] = 0x00000000; @@ -1583,6 +1597,66 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) gpr_map[gpr++] = 0x00000000; snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) ); A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000); + /* Pavel Hofman - we still have voices, A_FXBUS2s, and + * A_P16VINs available - + * let's add 8 more capture channels - total of 16 + */ + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x10)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x12)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x14)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x16)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x18)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x1a)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x1c)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe), + A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, + bit_shifter16, + A_GPR(gpr - 1), + A_FXBUS2(0x1e)); + A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf), + A_C_00000000, A_C_00000000); #if 0 for (z = 4; z < 8; z++) { diff --git a/sound/pci/emu10k1/emumixer.c b/sound/pci/emu10k1/emumixer.c index 4db6e1ca1665..7b2c1dcc5337 100644 --- a/sound/pci/emu10k1/emumixer.c +++ b/sound/pci/emu10k1/emumixer.c @@ -77,6 +77,10 @@ static int snd_emu10k1_spdif_get_mask(struct snd_kcontrol *kcontrol, return 0; } +/* + * Items labels in enum mixer controls assigning source data to + * each destination + */ static char *emu1010_src_texts[] = { "Silence", "Dock Mic A", @@ -133,6 +137,9 @@ static char *emu1010_src_texts[] = { "DSP 31", }; +/* + * List of data sources available for each destination + */ static unsigned int emu1010_src_regs[] = { EMU_SRC_SILENCE,/* 0 */ EMU_SRC_DOCK_MIC_A1, /* 1 */ @@ -189,6 +196,10 @@ static unsigned int emu1010_src_regs[] = { EMU_SRC_ALICE_EMU32B+0xf, /* 52 */ }; +/* + * Data destinations - physical EMU outputs. + * Each destination has an enum mixer control to choose a data source + */ static unsigned int emu1010_output_dst[] = { EMU_DST_DOCK_DAC1_LEFT1, /* 0 */ EMU_DST_DOCK_DAC1_RIGHT1, /* 1 */ @@ -216,6 +227,11 @@ static unsigned int emu1010_output_dst[] = { EMU_DST_HANA_ADAT+7, /* 23 */ }; +/* + * Data destinations - HANA outputs going to Alice2 (audigy) for + * capture (EMU32 + I2S links) + * Each destination has an enum mixer control to choose a data source + */ static unsigned int emu1010_input_dst[] = { EMU_DST_ALICE2_EMU32_0, EMU_DST_ALICE2_EMU32_1, diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index ab4f5df5241b..eda5cb373ded 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -1233,24 +1233,26 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream) runtime->hw.rate_min = runtime->hw.rate_max = 48000; spin_lock_irq(&emu->reg_lock); if (emu->card_capabilities->emu1010) { - /* TODO + /* Nb. of channels has been increased to 16 */ + /* TODO * SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE * SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | * SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | * SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000 * rate_min = 44100, * rate_max = 192000, - * channels_min = 8, - * channels_max = 8, + * channels_min = 16, + * channels_max = 16, * Need to add mixer control to fix sample rate * - * There are 16 mono channels of 16bits each. + * There are 32 mono channels of 16bits each. * 24bit Audio uses 2x channels over 16bit * 96kHz uses 2x channels over 48kHz * 192kHz uses 4x channels over 48kHz - * So, for 48kHz 24bit, one has 8 channels - * for 96kHz 24bit, one has 4 channels - * for 192kHz 24bit, one has 2 channels + * So, for 48kHz 24bit, one has 16 channels + * for 96kHz 24bit, one has 8 channels + * for 192kHz 24bit, one has 4 channels + * */ #if 1 switch (emu->emu1010.internal_clock) { @@ -1258,13 +1260,15 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream) /* For 44.1kHz */ runtime->hw.rates = SNDRV_PCM_RATE_44100; runtime->hw.rate_min = runtime->hw.rate_max = 44100; - runtime->hw.channels_min = runtime->hw.channels_max = 8; + runtime->hw.channels_min = + runtime->hw.channels_max = 16; break; case 1: /* For 48kHz */ runtime->hw.rates = SNDRV_PCM_RATE_48000; runtime->hw.rate_min = runtime->hw.rate_max = 48000; - runtime->hw.channels_min = runtime->hw.channels_max = 8; + runtime->hw.channels_min = + runtime->hw.channels_max = 16; break; }; #endif @@ -1282,7 +1286,7 @@ static int snd_emu10k1_capture_efx_open(struct snd_pcm_substream *substream) #endif runtime->hw.formats = SNDRV_PCM_FMTBIT_S32_LE; /* efx_voices_mask[0] is expected to be zero - * efx_voices_mask[1] is expected to have 16bits set + * efx_voices_mask[1] is expected to have 32bits set */ } else { runtime->hw.channels_min = runtime->hw.channels_max = 0; @@ -1787,11 +1791,24 @@ int __devinit snd_emu10k1_pcm_efx(struct snd_emu10k1 * emu, int device, struct s /* emu->efx_voices_mask[0] = FXWC_DEFAULTROUTE_C | FXWC_DEFAULTROUTE_A; */ if (emu->audigy) { emu->efx_voices_mask[0] = 0; - emu->efx_voices_mask[1] = 0xffff; + if (emu->card_capabilities->emu1010) + /* Pavel Hofman - 32 voices will be used for + * capture (write mode) - + * each bit = corresponding voice + */ + emu->efx_voices_mask[1] = 0xffffffff; + else + emu->efx_voices_mask[1] = 0xffff; } else { emu->efx_voices_mask[0] = 0xffff0000; emu->efx_voices_mask[1] = 0; } + /* For emu1010, the control has to set 32 upper bits (voices) + * out of the 64 bits (voices) to true for the 16-channels capture + * to work correctly. Correct A_FXWC2 initial value (0xffffffff) + * is already defined but the snd_emu10k1_pcm_efx_voices_mask + * control can override this register's value. + */ kctl = snd_ctl_new1(&snd_emu10k1_pcm_efx_voices_mask, emu); if (!kctl) return -ENOMEM; diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index 6a0ddcf00884..21cb4268a59b 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -1607,8 +1607,8 @@ struct es1371_quirk { unsigned char rev; /* revision */ }; -static int __devinit es1371_quirk_lookup(struct ensoniq *ensoniq, - struct es1371_quirk *list) +static int es1371_quirk_lookup(struct ensoniq *ensoniq, + struct es1371_quirk *list) { while (list->vid != (unsigned short)PCI_ANY_ID) { if (ensoniq->pci->vendor == list->vid && @@ -2110,7 +2110,6 @@ static int __devinit snd_ensoniq_create(struct snd_card *card, struct ensoniq ** rensoniq) { struct ensoniq *ensoniq; - unsigned char cmdb; int err; static struct snd_device_ops ops = { .dev_free = snd_ensoniq_dev_free, @@ -2151,8 +2150,7 @@ static int __devinit snd_ensoniq_create(struct snd_card *card, } #endif pci_set_master(pci); - pci_read_config_byte(pci, PCI_REVISION_ID, &cmdb); - ensoniq->rev = cmdb; + ensoniq->rev = pci->revision; #ifdef CHIP1370 #if 0 ensoniq->ctrl = ES_1370_CDC_EN | ES_1370_SERR_DISABLE | diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 6dc578bbeec9..11015178e207 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1369,7 +1369,6 @@ static int __devinit snd_fm801_create(struct snd_card *card, struct fm801 ** rchip) { struct fm801 *chip; - unsigned char rev; int err; static struct snd_device_ops ops = { .dev_free = snd_fm801_dev_free, @@ -1405,8 +1404,7 @@ static int __devinit snd_fm801_create(struct snd_card *card, pci_set_master(pci); } - pci_read_config_byte(pci, PCI_REVISION_ID, &rev); - if (rev >= 0xb1) /* FM801-AU */ + if (pci->revision >= 0xb1) /* FM801-AU */ chip->multichannel = 1; snd_fm801_chip_init(chip, 0); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2fa281cbef91..92bc8b3fa2a0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -341,6 +341,9 @@ struct azx { unsigned int single_cmd :1; unsigned int polling_mode :1; unsigned int msi :1; + + /* for debugging */ + unsigned int last_cmd; /* last issued command (to sync) */ }; /* driver types */ @@ -466,18 +469,10 @@ static void azx_free_cmd_io(struct azx *chip) } /* send a command */ -static int azx_corb_send_cmd(struct hda_codec *codec, hda_nid_t nid, int direct, - unsigned int verb, unsigned int para) +static int azx_corb_send_cmd(struct hda_codec *codec, u32 val) { struct azx *chip = codec->bus->private_data; unsigned int wp; - u32 val; - - val = (u32)(codec->addr & 0x0f) << 28; - val |= (u32)direct << 27; - val |= (u32)nid << 20; - val |= verb << 8; - val |= para; /* add command to corb */ wp = azx_readb(chip, CORBWP); @@ -538,12 +533,12 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) } if (! chip->rirb.cmds) return chip->rirb.res; /* the last value */ - schedule_timeout_interruptible(1); + schedule_timeout(1); } while (time_after_eq(timeout, jiffies)); if (chip->msi) { snd_printk(KERN_WARNING "hda_intel: No response from codec, " - "disabling MSI...\n"); + "disabling MSI: last cmd=0x%08x\n", chip->last_cmd); free_irq(chip->irq, chip); chip->irq = -1; pci_disable_msi(chip->pci); @@ -555,13 +550,15 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) if (!chip->polling_mode) { snd_printk(KERN_WARNING "hda_intel: azx_get_response timeout, " - "switching to polling mode...\n"); + "switching to polling mode: last cmd=0x%08x\n", + chip->last_cmd); chip->polling_mode = 1; goto again; } snd_printk(KERN_ERR "hda_intel: azx_get_response timeout, " - "switching to single_cmd mode...\n"); + "switching to single_cmd mode: last cmd=0x%08x\n", + chip->last_cmd); chip->rirb.rp = azx_readb(chip, RIRBWP); chip->rirb.cmds = 0; /* switch to single_cmd mode */ @@ -581,20 +578,11 @@ static unsigned int azx_rirb_get_response(struct hda_codec *codec) */ /* send a command */ -static int azx_single_send_cmd(struct hda_codec *codec, hda_nid_t nid, - int direct, unsigned int verb, - unsigned int para) +static int azx_single_send_cmd(struct hda_codec *codec, u32 val) { struct azx *chip = codec->bus->private_data; - u32 val; int timeout = 50; - val = (u32)(codec->addr & 0x0f) << 28; - val |= (u32)direct << 27; - val |= (u32)nid << 20; - val |= verb << 8; - val |= para; - while (timeout--) { /* check ICB busy bit */ if (! (azx_readw(chip, IRS) & ICH6_IRS_BUSY)) { @@ -639,10 +627,19 @@ static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid, unsigned int para) { struct azx *chip = codec->bus->private_data; + u32 val; + + val = (u32)(codec->addr & 0x0f) << 28; + val |= (u32)direct << 27; + val |= (u32)nid << 20; + val |= verb << 8; + val |= para; + chip->last_cmd = val; + if (chip->single_cmd) - return azx_single_send_cmd(codec, nid, direct, verb, para); + return azx_single_send_cmd(codec, val); else - return azx_corb_send_cmd(codec, nid, direct, verb, para); + return azx_corb_send_cmd(codec, val); } /* get a response */ @@ -1788,6 +1785,12 @@ static struct pci_device_id azx_ids[] = { { 0x10de, 0x044b, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP65 */ { 0x10de, 0x055c, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP67 */ { 0x10de, 0x055d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP67 */ + { 0x10de, 0x07fc, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP73 */ + { 0x10de, 0x07fd, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP73 */ + { 0x10de, 0x0774, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ + { 0x10de, 0x0775, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ + { 0x10de, 0x0776, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ + { 0x10de, 0x0777, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index e313e685f161..ac15066fd300 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -250,6 +250,12 @@ static void print_codec_info(struct snd_info_entry *entry, struct snd_info_buffe snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id); snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id); snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id); + + if (codec->mfg) + snd_iprintf(buffer, "Modem Function Group: 0x%x\n", codec->mfg); + else + snd_iprintf(buffer, "No Modem Function Group found\n"); + if (! codec->afg) return; snd_iprintf(buffer, "Default PCM:\n"); diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 0e1a879663fa..4d7f8d11ad75 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1,7 +1,8 @@ /* - * HD audio interface patch for AD1981HD, AD1983, AD1986A, AD1988 + * HD audio interface patch for AD1882, AD1884, AD1981HD, AD1983, AD1984, + * AD1986A, AD1988 * - * Copyright (c) 2005 Takashi Iwai <tiwai@suse.de> + * Copyright (c) 2005-2007 Takashi Iwai <tiwai@suse.de> * * This driver is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by @@ -61,7 +62,7 @@ struct ad198x_spec { int num_channel_mode; /* PCM information */ - struct hda_pcm pcm_rec[2]; /* used in alc_build_pcms() */ + struct hda_pcm pcm_rec[3]; /* used in alc_build_pcms() */ struct mutex amp_mutex; /* PCM volume/mute control mutex */ unsigned int spdif_route; @@ -2775,11 +2776,634 @@ static int patch_ad1988(struct hda_codec *codec) /* + * AD1884 / AD1984 + * + * port-B - front line/mic-in + * port-E - aux in/out + * port-F - aux in/out + * port-C - rear line/mic-in + * port-D - rear line/hp-out + * port-A - front line/hp-out + * + * AD1984 = AD1884 + two digital mic-ins + * + * FIXME: + * For simplicity, we share the single DAC for both HP and line-outs + * right now. The inidividual playbacks could be easily implemented, + * but no build-up framework is given, so far. + */ + +static hda_nid_t ad1884_dac_nids[1] = { + 0x04, +}; + +static hda_nid_t ad1884_adc_nids[2] = { + 0x08, 0x09, +}; + +static hda_nid_t ad1884_capsrc_nids[2] = { + 0x0c, 0x0d, +}; + +#define AD1884_SPDIF_OUT 0x02 + +static struct hda_input_mux ad1884_capture_source = { + .num_items = 4, + .items = { + { "Front Mic", 0x0 }, + { "Mic", 0x1 }, + { "CD", 0x2 }, + { "Mix", 0x3 }, + }, +}; + +static struct snd_kcontrol_new ad1884_base_mixers[] = { + HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */ + HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT), + /* + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Digital Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), + */ + HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + /* SPDIF controls */ + HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", + /* identical with ad1983 */ + .info = ad1983_spdif_route_info, + .get = ad1983_spdif_route_get, + .put = ad1983_spdif_route_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1984_dmic_mixers[] = { + HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x05, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x05, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Capture Volume", 1, 0x06, 0x0, + HDA_INPUT), + HDA_CODEC_MUTE_IDX("Digital Mic Capture Switch", 1, 0x06, 0x0, + HDA_INPUT), + { } /* end */ +}; + +/* + * initialization verbs + */ +static struct hda_verb ad1884_init_verbs[] = { + /* DACs; mute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* Port-A (HP) mixer */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-A pin */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* HP selector - select DAC2 */ + {0x22, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Port-D (Line-out) mixer */ + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-D pin */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Mono-out mixer */ + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Mono-out pin */ + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Mono selector */ + {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Port-B (front mic) pin */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-C (rear mic) pin */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ + /* SPDIF output selector */ + {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ + { } /* end */ +}; + +static int patch_ad1884(struct hda_codec *codec) +{ + struct ad198x_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + mutex_init(&spec->amp_mutex); + codec->spec = spec; + + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = ARRAY_SIZE(ad1884_dac_nids); + spec->multiout.dac_nids = ad1884_dac_nids; + spec->multiout.dig_out_nid = AD1884_SPDIF_OUT; + spec->num_adc_nids = ARRAY_SIZE(ad1884_adc_nids); + spec->adc_nids = ad1884_adc_nids; + spec->capsrc_nids = ad1884_capsrc_nids; + spec->input_mux = &ad1884_capture_source; + spec->num_mixers = 1; + spec->mixers[0] = ad1884_base_mixers; + spec->num_init_verbs = 1; + spec->init_verbs[0] = ad1884_init_verbs; + spec->spdif_route = 0; + + codec->patch_ops = ad198x_patch_ops; + + return 0; +} + +/* + * Lenovo Thinkpad T61/X61 + */ +static struct hda_input_mux ad1984_thinkpad_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Internal Mic", 0x1 }, + { "Mix", 0x3 }, + }, +}; + +static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { + HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */ + HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Docking Mic Playback Volume", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("Docking Mic Playback Switch", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Docking Mic Boost", 0x25, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + { } /* end */ +}; + +/* additional verbs */ +static struct hda_verb ad1984_thinkpad_init_verbs[] = { + /* Port-E (docking station mic) pin */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* docking mic boost */ + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer - docking mic; mute as default */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + /* enable EAPD bit */ + {0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + { } /* end */ +}; + +/* Digial MIC ADC NID 0x05 + 0x06 */ +static int ad1984_pcm_dmic_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + snd_hda_codec_setup_stream(codec, 0x05 + substream->number, + stream_tag, 0, format); + return 0; +} + +static int ad1984_pcm_dmic_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + snd_hda_codec_setup_stream(codec, 0x05 + substream->number, + 0, 0, 0); + return 0; +} + +static struct hda_pcm_stream ad1984_pcm_dmic_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x05, + .ops = { + .prepare = ad1984_pcm_dmic_prepare, + .cleanup = ad1984_pcm_dmic_cleanup + }, +}; + +static int ad1984_build_pcms(struct hda_codec *codec) +{ + struct ad198x_spec *spec = codec->spec; + struct hda_pcm *info; + int err; + + err = ad198x_build_pcms(codec); + if (err < 0) + return err; + + info = spec->pcm_rec + codec->num_pcms; + codec->num_pcms++; + info->name = "AD1984 Digital Mic"; + info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad1984_pcm_dmic_capture; + return 0; +} + +/* models */ +enum { + AD1984_BASIC, + AD1984_THINKPAD, + AD1984_MODELS +}; + +static const char *ad1984_models[AD1984_MODELS] = { + [AD1984_BASIC] = "basic", + [AD1984_THINKPAD] = "thinkpad", +}; + +static struct snd_pci_quirk ad1984_cfg_tbl[] = { + /* Lenovo Thinkpad T61/X61 */ + SND_PCI_QUIRK(0x17aa, 0, "Lenovo Thinkpad", AD1984_THINKPAD), + {} +}; + +static int patch_ad1984(struct hda_codec *codec) +{ + struct ad198x_spec *spec; + int board_config, err; + + err = patch_ad1884(codec); + if (err < 0) + return err; + spec = codec->spec; + board_config = snd_hda_check_board_config(codec, AD1984_MODELS, + ad1984_models, ad1984_cfg_tbl); + switch (board_config) { + case AD1984_BASIC: + /* additional digital mics */ + spec->mixers[spec->num_mixers++] = ad1984_dmic_mixers; + codec->patch_ops.build_pcms = ad1984_build_pcms; + break; + case AD1984_THINKPAD: + spec->multiout.dig_out_nid = 0; + spec->input_mux = &ad1984_thinkpad_capture_source; + spec->mixers[0] = ad1984_thinkpad_mixers; + spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs; + break; + } + return 0; +} + + +/* + * AD1882 + * + * port-A - front hp-out + * port-B - front mic-in + * port-C - rear line-in, shared surr-out (3stack) + * port-D - rear line-out + * port-E - rear mic-in, shared clfe-out (3stack) + * port-F - rear surr-out (6stack) + * port-G - rear clfe-out (6stack) + */ + +static hda_nid_t ad1882_dac_nids[3] = { + 0x04, 0x03, 0x05 +}; + +static hda_nid_t ad1882_adc_nids[2] = { + 0x08, 0x09, +}; + +static hda_nid_t ad1882_capsrc_nids[2] = { + 0x0c, 0x0d, +}; + +#define AD1882_SPDIF_OUT 0x02 + +/* list: 0x11, 0x39, 0x3a, 0x18, 0x3c, 0x3b, 0x12, 0x20 */ +static struct hda_input_mux ad1882_capture_source = { + .num_items = 5, + .items = { + { "Front Mic", 0x1 }, + { "Mic", 0x4 }, + { "Line", 0x2 }, + { "CD", 0x3 }, + { "Mix", 0x7 }, + }, +}; + +static struct snd_kcontrol_new ad1882_base_mixers[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT), + HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x3c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x39, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Line-In Boost", 0x3a, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = ad198x_mux_enum_info, + .get = ad198x_mux_enum_get, + .put = ad198x_mux_enum_put, + }, + /* SPDIF controls */ + HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", + /* identical with ad1983 */ + .info = ad1983_spdif_route_info, + .get = ad1983_spdif_route_get, + .put = ad1983_spdif_route_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1882_3stack_mixers[] = { + HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x17, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x17, 2, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = ad198x_ch_mode_info, + .get = ad198x_ch_mode_get, + .put = ad198x_ch_mode_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new ad1882_6stack_mixers[] = { + HDA_CODEC_MUTE("Surround Playback Switch", 0x16, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x24, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x24, 2, 0x0, HDA_OUTPUT), + { } /* end */ +}; + +static struct hda_verb ad1882_ch2_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + { } /* end */ +}; + +static struct hda_verb ad1882_ch4_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + { } /* end */ +}; + +static struct hda_verb ad1882_ch6_init[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + { } /* end */ +}; + +static struct hda_channel_mode ad1882_modes[3] = { + { 2, ad1882_ch2_init }, + { 4, ad1882_ch4_init }, + { 6, ad1882_ch6_init }, +}; + +/* + * initialization verbs + */ +static struct hda_verb ad1882_init_verbs[] = { + /* DACs; mute as default */ + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + /* Port-A (HP) mixer */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-A pin */ + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* HP selector - select DAC2 */ + {0x37, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* Port-D (Line-out) mixer */ + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Port-D pin */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Mono-out mixer */ + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* Mono-out pin */ + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-B (front mic) pin */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */ + /* Port-C (line-in) pin */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */ + /* Port-C mixer - mute as input */ + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Port-E (mic-in) pin */ + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */ + /* Port-E mixer - mute as input */ + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + /* Port-F (surround) */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Port-G (CLFE) */ + {0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog mixer; mute as default */ + /* list: 0x39, 0x3a, 0x11, 0x12, 0x3c, 0x3b, 0x18, 0x1a */ + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ + /* SPDIF output selector */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ + {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ + { } /* end */ +}; + +/* models */ +enum { + AD1882_3STACK, + AD1882_6STACK, + AD1882_MODELS +}; + +static const char *ad1882_models[AD1986A_MODELS] = { + [AD1882_3STACK] = "3stack", + [AD1882_6STACK] = "6stack", +}; + + +static int patch_ad1882(struct hda_codec *codec) +{ + struct ad198x_spec *spec; + int board_config; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + mutex_init(&spec->amp_mutex); + codec->spec = spec; + + spec->multiout.max_channels = 6; + spec->multiout.num_dacs = 3; + spec->multiout.dac_nids = ad1882_dac_nids; + spec->multiout.dig_out_nid = AD1882_SPDIF_OUT; + spec->num_adc_nids = ARRAY_SIZE(ad1882_adc_nids); + spec->adc_nids = ad1882_adc_nids; + spec->capsrc_nids = ad1882_capsrc_nids; + spec->input_mux = &ad1882_capture_source; + spec->num_mixers = 1; + spec->mixers[0] = ad1882_base_mixers; + spec->num_init_verbs = 1; + spec->init_verbs[0] = ad1882_init_verbs; + spec->spdif_route = 0; + + codec->patch_ops = ad198x_patch_ops; + + /* override some parameters */ + board_config = snd_hda_check_board_config(codec, AD1882_MODELS, + ad1882_models, NULL); + switch (board_config) { + default: + case AD1882_3STACK: + spec->num_mixers = 2; + spec->mixers[1] = ad1882_3stack_mixers; + spec->channel_mode = ad1882_modes; + spec->num_channel_mode = ARRAY_SIZE(ad1882_modes); + spec->need_dac_fix = 1; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = 1; + break; + case AD1882_6STACK: + spec->num_mixers = 2; + spec->mixers[1] = ad1882_6stack_mixers; + break; + } + return 0; +} + + +/* * patch entries */ struct hda_codec_preset snd_hda_preset_analog[] = { + { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 }, + { .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 }, { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 }, { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 }, + { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 }, { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a }, { .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 }, { .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 }, diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c index f8eb4c90f801..72d3ab9751ac 100644 --- a/sound/pci/hda/patch_atihdmi.c +++ b/sound/pci/hda/patch_atihdmi.c @@ -172,6 +172,7 @@ static int patch_atihdmi(struct hda_codec *codec) */ struct hda_codec_preset snd_hda_preset_atihdmi[] = { { .id = 0x1002793c, .name = "ATI RS600 HDMI", .patch = patch_atihdmi }, + { .id = 0x10027919, .name = "ATI RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x1002791a, .name = "ATI RS690/780 HDMI", .patch = patch_atihdmi }, { .id = 0x1002aa01, .name = "ATI R600 HDMI", .patch = patch_atihdmi }, {} /* terminator */ diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index bef214bcdddf..4d8e8af5c819 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -801,7 +801,9 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV Series", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2120", CXT5045_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP), + SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_FUJITSU), + SND_PCI_QUIRK(0x1734, 0x10cb, "Fujitsu Si3515", CXT5045_LAPTOP), SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP), {} }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4776de93928b..9a47eec5a27b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -94,10 +94,18 @@ enum { ALC262_HP_BPC_D7000_WF, ALC262_BENQ_ED8, ALC262_SONY_ASSAMD, + ALC262_BENQ_T31, ALC262_AUTO, ALC262_MODEL_LAST /* last tag */ }; +/* ALC268 models */ +enum { + ALC268_3ST, + ALC268_AUTO, + ALC268_MODEL_LAST /* last tag */ +}; + /* ALC861 models */ enum { ALC861_3ST, @@ -115,6 +123,7 @@ enum { /* ALC861-VD models */ enum { ALC660VD_3ST, + ALC660VD_3ST_DIG, ALC861VD_3ST, ALC861VD_3ST_DIG, ALC861VD_6ST_DIG, @@ -144,6 +153,7 @@ enum { ALC882_TARGA, ALC882_ASUS_A7J, ALC885_MACPRO, + ALC885_IMAC24, ALC882_AUTO, ALC882_MODEL_LAST, }; @@ -163,6 +173,8 @@ enum { ALC883_LENOVO_101E_2ch, ALC883_LENOVO_NB0763, ALC888_LENOVO_MS7195_DIG, + ALC888_6ST_HP, + ALC888_3ST_HP, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -713,6 +725,38 @@ static void alc_subsystem_id(struct hda_codec *codec, } /* + * Fix-up pin default configurations + */ + +struct alc_pincfg { + hda_nid_t nid; + u32 val; +}; + +static void alc_fix_pincfg(struct hda_codec *codec, + const struct snd_pci_quirk *quirk, + const struct alc_pincfg **pinfix) +{ + const struct alc_pincfg *cfg; + + quirk = snd_pci_quirk_lookup(codec->bus->pci, quirk); + if (!quirk) + return; + + cfg = pinfix[quirk->value]; + for (; cfg->nid; cfg++) { + int i; + u32 val = cfg->val; + for (i = 0; i < 4; i++) { + snd_hda_codec_write(codec, cfg->nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 + i, + val & 0xff); + val >>= 8; + } + } +} + +/* * ALC880 3-stack model * * DAC: Front = 0x02 (0x0c), Surr = 0x05 (0x0f), CLFE = 0x04 (0x0e) @@ -1878,31 +1922,53 @@ static void alc880_lg_unsol_event(struct hda_codec *codec, unsigned int res) * Pin assignment: * Speaker-out: 0x14 * Mic-In: 0x18 - * Built-in Mic-In: 0x19 (?) - * HP-Out: 0x1b + * Built-in Mic-In: 0x19 + * Line-In: 0x1b + * HP-Out: 0x1a * SPDIF-Out: 0x1e */ -/* seems analog CD is not working */ static struct hda_input_mux alc880_lg_lw_capture_source = { - .num_items = 2, + .num_items = 3, .items = { { "Mic", 0x0 }, { "Internal Mic", 0x1 }, + { "Line In", 0x2 }, }, }; +#define alc880_lg_lw_modes alc880_threestack_modes + static struct snd_kcontrol_new alc880_lg_lw_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Channel Mode", + .info = alc_ch_mode_info, + .get = alc_ch_mode_get, + .put = alc_ch_mode_put, + }, { } /* end */ }; static struct hda_verb alc880_lg_lw_init_verbs[] = { + {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ + {0x10, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */ + {0x12, AC_VERB_SET_CONNECT_SEL, 0x03}, /* line/surround */ + /* set capture source to mic-in */ {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -1912,7 +1978,6 @@ static struct hda_verb alc880_lg_lw_init_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* HP-out */ - {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* mic-in to input */ @@ -2856,11 +2921,11 @@ static struct alc_config_preset alc880_presets[] = { .mixers = { alc880_lg_lw_mixer }, .init_verbs = { alc880_volume_init_verbs, alc880_lg_lw_init_verbs }, - .num_dacs = 1, + .num_dacs = ARRAY_SIZE(alc880_dac_nids), .dac_nids = alc880_dac_nids, .dig_out_nid = ALC880_DIGOUT_NID, - .num_channel_mode = ARRAY_SIZE(alc880_2_jack_modes), - .channel_mode = alc880_2_jack_modes, + .num_channel_mode = ARRAY_SIZE(alc880_lg_lw_modes), + .channel_mode = alc880_lg_lw_modes, .input_mux = &alc880_lg_lw_capture_source, .unsol_event = alc880_lg_lw_unsol_event, .init_hook = alc880_lg_lw_automute, @@ -5054,6 +5119,60 @@ static struct hda_verb alc882_macpro_init_verbs[] = { { } }; +/* iMac 24 mixer. */ +static struct snd_kcontrol_new alc885_imac24_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x00, HDA_INPUT), + { } /* end */ +}; + +/* iMac 24 init verbs. */ +static struct hda_verb alc885_imac24_init_verbs[] = { + /* Internal speakers: output 0 (0x0c) */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Internal speakers: output 0 (0x0c) */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Headphone: output 0 (0x0c) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + /* Front Mic: input vref at 80% */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + { } +}; + +/* Toggle speaker-output according to the hp-jack state */ +static void alc885_imac24_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_update(codec, 0x18, 0, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x18, 1, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); +} + +/* Processes unsolicited events. */ +static void alc885_imac24_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + /* Headphone insertion or removal. */ + if ((res >> 26) == ALC880_HP_EVENT) + alc885_imac24_automute(codec); +} + static struct hda_verb alc882_targa_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -5274,6 +5393,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC882_ARIMA] = "arima", [ALC882_W2JC] = "w2jc", [ALC885_MACPRO] = "macpro", + [ALC885_IMAC24] = "imac24", [ALC882_AUTO] = "auto", }; @@ -5284,6 +5404,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */ SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J), + SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC), {} @@ -5345,6 +5466,19 @@ static struct alc_config_preset alc882_presets[] = { .channel_mode = alc882_ch_modes, .input_mux = &alc882_capture_source, }, + [ALC885_IMAC24] = { + .mixers = { alc885_imac24_mixer }, + .init_verbs = { alc885_imac24_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), + .channel_mode = alc882_ch_modes, + .input_mux = &alc882_capture_source, + .unsol_event = alc885_imac24_unsol_event, + .init_hook = alc885_imac24_automute, + }, [ALC882_TARGA] = { .mixers = { alc882_targa_mixer, alc882_chmode_mixer, alc882_capture_mixer }, @@ -5379,6 +5513,29 @@ static struct alc_config_preset alc882_presets[] = { /* + * Pin config fixes + */ +enum { + PINFIX_ABIT_AW9D_MAX +}; + +static struct alc_pincfg alc882_abit_aw9d_pinfix[] = { + { 0x15, 0x01080104 }, /* side */ + { 0x16, 0x01011012 }, /* rear */ + { 0x17, 0x01016011 }, /* clfe */ + { } +}; + +static const struct alc_pincfg *alc882_pin_fixes[] = { + [PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix, +}; + +static struct snd_pci_quirk alc882_pinfix_tbl[] = { + SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), + {} +}; + +/* * BIOS auto configuration */ static void alc882_auto_set_output_and_unmute(struct hda_codec *codec, @@ -5494,6 +5651,9 @@ static int patch_alc882(struct hda_codec *codec) case 0x106b0c00: /* Mac Pro */ board_config = ALC885_MACPRO; break; + case 0x106b1000: /* iMac 24 */ + board_config = ALC885_IMAC24; + break; default: printk(KERN_INFO "hda_codec: Unknown model for ALC882, " "trying auto-probe from BIOS...\n"); @@ -5501,6 +5661,8 @@ static int patch_alc882(struct hda_codec *codec) } } + alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes); + if (board_config == ALC882_AUTO) { /* automatic parse from the BIOS config */ err = alc882_parse_auto_config(codec); @@ -5518,7 +5680,7 @@ static int patch_alc882(struct hda_codec *codec) if (board_config != ALC882_AUTO) setup_preset(spec, &alc882_presets[board_config]); - if (board_config == ALC885_MACPRO) { + if (board_config == ALC885_MACPRO || board_config == ALC885_IMAC24) { alc882_gpio_mute(codec, 0, 0); alc882_gpio_mute(codec, 1, 0); } @@ -5995,6 +6157,84 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc888_6st_hp_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new alc888_3st_hp_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Surround Playback Volume", 0x0e, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Surround Playback Switch", 0x0e, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0d, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0d, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0d, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -6126,6 +6366,42 @@ static struct hda_verb alc888_lenovo_ms7195_verbs[] = { { } /* end */ }; +static struct hda_verb alc888_6st_hp_verbs[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x02}, /* Rear : output 2 (0x0e) */ + {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* CLFE : output 1 (0x0d) */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0x03}, /* Side : output 3 (0x0f) */ + { } +}; + +static struct hda_verb alc888_3st_hp_verbs[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ + {0x18, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ + {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ + { } +}; + +static struct hda_verb alc888_3st_hp_2ch_init[] = { + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } +}; + +static struct hda_verb alc888_3st_hp_6ch_init[] = { + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { } +}; + +static struct hda_channel_mode alc888_3st_hp_modes[2] = { + { 2, alc888_3st_hp_2ch_init }, + { 6, alc888_3st_hp_6ch_init }, +}; + /* toggle front-jack and RCA according to the hp-jack state */ static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec) { @@ -6368,11 +6644,14 @@ static const char *alc883_models[ALC883_MODEL_LAST] = { [ALC883_LENOVO_101E_2ch] = "lenovo-101e", [ALC883_LENOVO_NB0763] = "lenovo-nb0763", [ALC888_LENOVO_MS7195_DIG] = "lenovo-ms7195-dig", + [ALC888_6ST_HP] = "6stack-hp", + [ALC888_3ST_HP] = "3stack-hp", [ALC883_AUTO] = "auto", }; static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_DIG), + SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch), SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), @@ -6381,6 +6660,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1462, 0x7187, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7250, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x7280, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG), @@ -6400,6 +6681,9 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), + SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC888_6ST_HP), + SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), + SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), {} }; @@ -6584,6 +6868,31 @@ static struct alc_config_preset alc883_presets[] = { .unsol_event = alc883_lenovo_ms7195_unsol_event, .init_hook = alc888_lenovo_ms7195_front_automute, }, + [ALC888_6ST_HP] = { + .mixers = { alc888_6st_hp_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc888_6st_hp_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .dig_in_nid = ALC883_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), + .channel_mode = alc883_sixstack_modes, + .input_mux = &alc883_capture_source, + }, + [ALC888_3ST_HP] = { + .mixers = { alc888_3st_hp_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc888_3st_hp_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc888_3st_hp_modes), + .channel_mode = alc888_3st_hp_modes, + .need_dac_fix = 1, + .input_mux = &alc883_capture_source, + }, }; @@ -6857,7 +7166,16 @@ static struct snd_kcontrol_new alc262_sony_mixer[] = { { } /* end */ }; - +static struct snd_kcontrol_new alc262_benq_t31_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + { } /* end */ +}; #define alc262_capture_mixer alc882_capture_mixer #define alc262_capture_alt_mixer alc882_capture_alt_mixer @@ -7189,6 +7507,15 @@ static struct hda_verb alc262_EAPD_verbs[] = { {} }; +static struct hda_verb alc262_benq_t31_EAPD_verbs[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + + {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, + {0x20, AC_VERB_SET_PROC_COEF, 0x3050}, + {} +}; + /* add playback controls from the parsed DAC table */ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) @@ -7584,7 +7911,8 @@ static const char *alc262_models[ALC262_MODEL_LAST] = { [ALC262_HP_BPC] = "hp-bpc", [ALC262_HP_BPC_D7000_WL]= "hp-bpc-d7000", [ALC262_BENQ_ED8] = "benq", - [ALC262_BENQ_ED8] = "sony-assamd", + [ALC262_BENQ_T31] = "benq-t31", + [ALC262_SONY_ASSAMD] = "sony-assamd", [ALC262_AUTO] = "auto", }; @@ -7592,8 +7920,12 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO), SND_PCI_QUIRK(0x103c, 0x12fe, "HP xw9400", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x280c, "HP xw4400", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x12ff, "HP xw4550", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x1308, "HP xw4600", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x3014, "HP xw6400", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x1307, "HP xw6600", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x3015, "HP xw8400", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x1306, "HP xw8600", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL), SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL), SND_PCI_QUIRK(0x103c, 0x2804, "HP D7000", ALC262_HP_BPC_D7000_WL), @@ -7606,6 +7938,7 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x17ff, 0x058f, "Benq Hippo", ALC262_HIPPO_1), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), + SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31), SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), @@ -7710,6 +8043,17 @@ static struct alc_config_preset alc262_presets[] = { .channel_mode = alc262_modes, .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, + }, + [ALC262_BENQ_T31] = { + .mixers = { alc262_benq_t31_mixer }, + .init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs, alc262_hippo_unsol_verbs }, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_capture_source, + .unsol_event = alc262_hippo_unsol_event, }, }; @@ -7800,6 +8144,515 @@ static int patch_alc262(struct hda_codec *codec) } /* + * ALC268 channel source setting (2 channel) + */ +#define ALC268_DIGOUT_NID ALC880_DIGOUT_NID +#define alc268_modes alc260_modes + +static hda_nid_t alc268_dac_nids[2] = { + /* front, hp */ + 0x02, 0x03 +}; + +static hda_nid_t alc268_adc_nids[2] = { + /* ADC0-1 */ + 0x08, 0x07 +}; + +static hda_nid_t alc268_adc_nids_alt[1] = { + /* ADC0 */ + 0x08 +}; + +static struct snd_kcontrol_new alc268_base_mixer[] = { + /* output mixer control */ + HDA_CODEC_VOLUME("Front Playback Volume", 0x2, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x3, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + { } +}; + +/* + * generic initialization of ADC, input mixers and output mixers + */ +static struct hda_verb alc268_base_init_verbs[] = { + /* Unmute DAC0-1 and set vol = 0 */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + /* + * Set up output mixers (0x0c - 0x0e) + */ + /* set vol=0 to output mixers */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00}, + + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + + /* FIXME: use matrix-type input source selection */ + /* Mixer elements: 0x18, 19, 1a, 1c, 14, 15, 0b */ + /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ + /* Input mixer2 */ + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, + + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, + { } +}; + +/* + * generic initialization of ADC, input mixers and output mixers + */ +static struct hda_verb alc268_volume_init_verbs[] = { + /* set output DAC */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, + + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + /* set PCBEEP vol = 0 */ + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, (0xb000 | (0x00 << 8))}, + + { } +}; + +#define alc268_mux_enum_info alc_mux_enum_info +#define alc268_mux_enum_get alc_mux_enum_get + +static int alc268_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct alc_spec *spec = codec->spec; + const struct hda_input_mux *imux = spec->input_mux; + unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + static hda_nid_t capture_mixers[3] = { 0x23, 0x24 }; + hda_nid_t nid = capture_mixers[adc_idx]; + unsigned int *cur_val = &spec->cur_mux[adc_idx]; + unsigned int i, idx; + + idx = ucontrol->value.enumerated.item[0]; + if (idx >= imux->num_items) + idx = imux->num_items - 1; + if (*cur_val == idx && !codec->in_resume) + return 0; + for (i = 0; i < imux->num_items; i++) { + unsigned int v = (i == idx) ? 0x7000 : 0x7080; + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + v | (imux->items[i].index << 8)); + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, + idx ); + } + *cur_val = idx; + return 1; +} + +static struct snd_kcontrol_new alc268_capture_alt_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 1, + .info = alc268_mux_enum_info, + .get = alc268_mux_enum_get, + .put = alc268_mux_enum_put, + }, + { } /* end */ +}; + +static struct snd_kcontrol_new alc268_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x24, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x24, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + * FIXME: the controls appear in the "playback" view! + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc268_mux_enum_info, + .get = alc268_mux_enum_get, + .put = alc268_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_input_mux alc268_capture_source = { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Front Mic", 0x1 }, + { "Line", 0x2 }, + { "CD", 0x3 }, + }, +}; + +/* create input playback/capture controls for the given pin */ +static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid, + const char *ctlname, int idx) +{ + char name[32]; + int err; + + sprintf(name, "%s Playback Volume", ctlname); + if (nid == 0x14) { + err = add_control(spec, ALC_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(0x02, 3, idx, + HDA_OUTPUT)); + if (err < 0) + return err; + } else if (nid == 0x15) { + err = add_control(spec, ALC_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(0x03, 3, idx, + HDA_OUTPUT)); + if (err < 0) + return err; + } else + return -1; + sprintf(name, "%s Playback Switch", ctlname); + err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid, 3, idx, HDA_OUTPUT)); + if (err < 0) + return err; + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, + const struct auto_pin_cfg *cfg) +{ + hda_nid_t nid; + int err; + + spec->multiout.num_dacs = 2; /* only use one dac */ + spec->multiout.dac_nids = spec->private_dac_nids; + spec->multiout.dac_nids[0] = 2; + spec->multiout.dac_nids[1] = 3; + + nid = cfg->line_out_pins[0]; + if (nid) + alc268_new_analog_output(spec, nid, "Front", 0); + + nid = cfg->speaker_pins[0]; + if (nid == 0x1d) { + err = add_control(spec, ALC_CTL_WIDGET_VOL, + "Speaker Playback Volume", + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + } + nid = cfg->hp_pins[0]; + if (nid) + alc268_new_analog_output(spec, nid, "Headphone", 0); + + nid = cfg->line_out_pins[1] | cfg->line_out_pins[2]; + if (nid == 0x16) { + err = add_control(spec, ALC_CTL_WIDGET_MUTE, + "Mono Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_INPUT)); + if (err < 0) + return err; + } + return 0; +} + +/* create playback/capture controls for input pins */ +static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, + const struct auto_pin_cfg *cfg) +{ + struct hda_input_mux *imux = &spec->private_imux; + int i, idx1; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + switch(cfg->input_pins[i]) { + case 0x18: + idx1 = 0; /* Mic 1 */ + break; + case 0x19: + idx1 = 1; /* Mic 2 */ + break; + case 0x1a: + idx1 = 2; /* Line In */ + break; + case 0x1c: + idx1 = 3; /* CD */ + break; + default: + continue; + } + imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; + imux->items[imux->num_items].index = idx1; + imux->num_items++; + } + return 0; +} + +static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + hda_nid_t speaker_nid = spec->autocfg.speaker_pins[0]; + hda_nid_t hp_nid = spec->autocfg.hp_pins[0]; + hda_nid_t line_nid = spec->autocfg.line_out_pins[0]; + unsigned int dac_vol1, dac_vol2; + + if (speaker_nid) { + snd_hda_codec_write(codec, speaker_nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + snd_hda_codec_write(codec, 0x0f, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(1)); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(1)); + } else { + snd_hda_codec_write(codec, 0x0f, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)); + snd_hda_codec_write(codec, 0x10, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)); + } + + dac_vol1 = dac_vol2 = 0xb000 | 0x40; /* set max volume */ + if (line_nid == 0x14) + dac_vol2 = AMP_OUT_ZERO; + else if (line_nid == 0x15) + dac_vol1 = AMP_OUT_ZERO; + if (hp_nid == 0x14) + dac_vol2 = AMP_OUT_ZERO; + else if (hp_nid == 0x15) + dac_vol1 = AMP_OUT_ZERO; + if (line_nid != 0x16 || hp_nid != 0x16 || + spec->autocfg.line_out_pins[1] != 0x16 || + spec->autocfg.line_out_pins[2] != 0x16) + dac_vol1 = dac_vol2 = AMP_OUT_ZERO; + + snd_hda_codec_write(codec, 0x02, 0, + AC_VERB_SET_AMP_GAIN_MUTE, dac_vol1); + snd_hda_codec_write(codec, 0x03, 0, + AC_VERB_SET_AMP_GAIN_MUTE, dac_vol2); +} + +/* pcm configuration: identiacal with ALC880 */ +#define alc268_pcm_analog_playback alc880_pcm_analog_playback +#define alc268_pcm_analog_capture alc880_pcm_analog_capture +#define alc268_pcm_digital_playback alc880_pcm_digital_playback + +/* + * BIOS auto configuration + */ +static int alc268_parse_auto_config(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + int err; + static hda_nid_t alc268_ignore[] = { 0 }; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, + alc268_ignore); + if (err < 0) + return err; + if (!spec->autocfg.line_outs) + return 0; /* can't find valid BIOS pin config */ + + err = alc268_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = alc268_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = 2; + + /* digital only support output */ + if (spec->autocfg.dig_out_pin) + spec->multiout.dig_out_nid = ALC268_DIGOUT_NID; + + if (spec->kctl_alloc) + spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + + spec->init_verbs[spec->num_init_verbs++] = alc268_volume_init_verbs; + spec->num_mux_defs = 1; + spec->input_mux = &spec->private_imux; + + return 1; +} + +#define alc268_auto_init_multi_out alc882_auto_init_multi_out +#define alc268_auto_init_hp_out alc882_auto_init_hp_out +#define alc268_auto_init_analog_input alc882_auto_init_analog_input + +/* init callback for auto-configuration model -- overriding the default init */ +static void alc268_auto_init(struct hda_codec *codec) +{ + alc268_auto_init_multi_out(codec); + alc268_auto_init_hp_out(codec); + alc268_auto_init_mono_speaker_out(codec); + alc268_auto_init_analog_input(codec); +} + +/* + * configuration and preset + */ +static const char *alc268_models[ALC268_MODEL_LAST] = { + [ALC268_3ST] = "3stack", + [ALC268_AUTO] = "auto", +}; + +static struct snd_pci_quirk alc268_cfg_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), + {} +}; + +static struct alc_config_preset alc268_presets[] = { + [ALC268_3ST] = { + .mixers = { alc268_base_mixer, alc268_capture_alt_mixer }, + .init_verbs = { alc268_base_init_verbs }, + .num_dacs = ARRAY_SIZE(alc268_dac_nids), + .dac_nids = alc268_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt), + .adc_nids = alc268_adc_nids_alt, + .hp_nid = 0x03, + .dig_out_nid = ALC268_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc268_modes), + .channel_mode = alc268_modes, + .input_mux = &alc268_capture_source, + }, +}; + +static int patch_alc268(struct hda_codec *codec) +{ + struct alc_spec *spec; + int board_config; + int err; + + spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + board_config = snd_hda_check_board_config(codec, ALC268_MODEL_LAST, + alc268_models, + alc268_cfg_tbl); + + if (board_config < 0 || board_config >= ALC268_MODEL_LAST) { + printk(KERN_INFO "hda_codec: Unknown model for ALC268, " + "trying auto-probe from BIOS...\n"); + board_config = ALC268_AUTO; + } + + if (board_config == ALC268_AUTO) { + /* automatic parse from the BIOS config */ + err = alc268_parse_auto_config(codec); + if (err < 0) { + alc_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO + "hda_codec: Cannot set up configuration " + "from BIOS. Using base mode...\n"); + board_config = ALC268_3ST; + } + } + + if (board_config != ALC268_AUTO) + setup_preset(spec, &alc268_presets[board_config]); + + spec->stream_name_analog = "ALC268 Analog"; + spec->stream_analog_playback = &alc268_pcm_analog_playback; + spec->stream_analog_capture = &alc268_pcm_analog_capture; + + spec->stream_name_digital = "ALC268 Digital"; + spec->stream_digital_playback = &alc268_pcm_digital_playback; + + if (board_config == ALC268_AUTO) { + if (!spec->adc_nids && spec->input_mux) { + /* check whether NID 0x07 is valid */ + unsigned int wcap = get_wcaps(codec, 0x07); + + /* get type */ + wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + if (wcap != AC_WID_AUD_IN) { + spec->adc_nids = alc268_adc_nids_alt; + spec->num_adc_nids = + ARRAY_SIZE(alc268_adc_nids_alt); + spec->mixers[spec->num_mixers] = + alc268_capture_alt_mixer; + spec->num_mixers++; + } else { + spec->adc_nids = alc268_adc_nids; + spec->num_adc_nids = + ARRAY_SIZE(alc268_adc_nids); + spec->mixers[spec->num_mixers] = + alc268_capture_mixer; + spec->num_mixers++; + } + } + } + codec->patch_ops = alc_patch_ops; + if (board_config == ALC268_AUTO) + spec->init_hook = alc268_auto_init; + + return 0; +} + +/* * ALC861 channel source setting (2/6 channel selection for 3-stack) */ @@ -8767,13 +9620,21 @@ static struct snd_pci_quirk alc861_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1335, "ASUS F2/3", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x13d7, "ASUS A9rp", ALC861_ASUS_LAPTOP), + SND_PCI_QUIRK(0x1584, 0x9075, "Airis Praxis N1212", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS), + SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS P1-AH2", ALC861_3ST_DIG), SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA), - SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), + /* FIXME: the entry below breaks Toshiba A100 (model=auto works!) + * Any other models that need this preset? + */ + /* SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), */ SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31), + SND_PCI_QUIRK(0x1584, 0x9075, "Uniwill", ALC861_UNIWILL_M31), SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31), SND_PCI_QUIRK(0x1849, 0x0660, "Asrock 939SLI32", ALC660_3ST), SND_PCI_QUIRK(0x8086, 0xd600, "Intel", ALC861_3ST), + SND_PCI_QUIRK(0x1462, 0x7254, "HP dx2200 (MSI MS-7254)", ALC861_3ST), + SND_PCI_QUIRK(0x1462, 0x7297, "HP dx2250 (MSI MS-7297)", ALC861_3ST), {} }; @@ -9464,6 +10325,7 @@ static void alc861vd_dallas_unsol_event(struct hda_codec *codec, unsigned int re */ static const char *alc861vd_models[ALC861VD_MODEL_LAST] = { [ALC660VD_3ST] = "3stack-660", + [ALC660VD_3ST_DIG]= "3stack-660-digout", [ALC861VD_3ST] = "3stack", [ALC861VD_3ST_DIG] = "3stack-digout", [ALC861VD_6ST_DIG] = "6stack-digout", @@ -9475,7 +10337,7 @@ static const char *alc861vd_models[ALC861VD_MODEL_LAST] = { static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x12e2, "Asus z35m", ALC660VD_3ST), SND_PCI_QUIRK(0x1043, 0x1339, "Asus G1", ALC660VD_3ST), - SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST), + SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS", ALC660VD_3ST_DIG), SND_PCI_QUIRK(0x10de, 0x03f0, "Realtek ALC660 demo", ALC660VD_3ST), SND_PCI_QUIRK(0x1019, 0xa88d, "Realtek ALC660 demo", ALC660VD_3ST), @@ -9483,6 +10345,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff01, "DALLAS", ALC861VD_DALLAS), SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO), + SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba A135", ALC861VD_LENOVO), {} }; @@ -9499,6 +10362,19 @@ static struct alc_config_preset alc861vd_presets[] = { .channel_mode = alc861vd_3stack_2ch_modes, .input_mux = &alc861vd_capture_source, }, + [ALC660VD_3ST_DIG] = { + .mixers = { alc861vd_3st_mixer }, + .init_verbs = { alc861vd_volume_init_verbs, + alc861vd_3stack_init_verbs }, + .num_dacs = ARRAY_SIZE(alc660vd_dac_nids), + .dac_nids = alc660vd_dac_nids, + .dig_out_nid = ALC861VD_DIGOUT_NID, + .num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids), + .adc_nids = alc861vd_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc861vd_3stack_2ch_modes), + .channel_mode = alc861vd_3stack_2ch_modes, + .input_mux = &alc861vd_capture_source, + }, [ALC861VD_3ST] = { .mixers = { alc861vd_3st_mixer }, .init_verbs = { alc861vd_volume_init_verbs, @@ -10420,7 +11296,7 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec, for (i = 0; i < cfg->line_outs; i++) { if (!spec->multiout.dac_nids[i]) continue; - nid = alc880_idx_to_dac(i); + nid = alc880_idx_to_mixer(i); if (i == 2) { /* Center/LFE */ err = add_control(spec, ALC_CTL_WIDGET_VOL, @@ -10643,14 +11519,10 @@ static int alc662_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; - if (err < 0) - return err; - else if (err > 0) - /* hack - override the init verbs */ - spec->init_verbs[0] = alc662_auto_init_verbs; + spec->init_verbs[spec->num_init_verbs++] = alc662_auto_init_verbs; spec->mixers[spec->num_mixers] = alc662_capture_mixer; spec->num_mixers++; - return err; + return 1; } /* additional initialization for auto-configuration model */ @@ -10687,7 +11559,7 @@ static int patch_alc662(struct hda_codec *codec) if (err < 0) { alc_free(codec); return err; - } else if (err) { + } else if (!err) { printk(KERN_INFO "hda_codec: Cannot set up configuration " "from BIOS. Using base mode...\n"); @@ -10724,6 +11596,7 @@ static int patch_alc662(struct hda_codec *codec) struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 }, + { .id = 0x10ec0268, .name = "ALC268", .patch = patch_alc268 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", .patch = patch_alc861 }, { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd }, diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c index 43f537ef40bf..6d2ecc38905c 100644 --- a/sound/pci/hda/patch_si3054.c +++ b/sound/pci/hda/patch_si3054.c @@ -304,8 +304,12 @@ struct hda_codec_preset snd_hda_preset_si3054[] = { { .id = 0x10573055, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x10573057, .name = "Si3054", .patch = patch_si3054 }, { .id = 0x10573155, .name = "Si3054", .patch = patch_si3054 }, + /* VIA HDA on Clevo m540 */ + { .id = 0x11063288, .name = "Si3054", .patch = patch_si3054 }, /* Asus A8J Modem (SM56) */ { .id = 0x15433155, .name = "Si3054", .patch = patch_si3054 }, + /* LG LW20 modem */ + { .id = 0x18540018, .name = "Si3054", .patch = patch_si3054 }, {} }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e3964fc4c405..3f25de72966b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -44,6 +44,7 @@ enum { enum { STAC_9205_REF, + STAC_M43xx, STAC_9205_MODELS }; @@ -59,11 +60,19 @@ enum { STAC_D945_REF, STAC_D945GTP3, STAC_D945GTP5, + STAC_922X_DELL, + STAC_INTEL_MAC_V1, + STAC_INTEL_MAC_V2, + STAC_INTEL_MAC_V3, + STAC_INTEL_MAC_V4, + STAC_INTEL_MAC_V5, + /* for backward compitability */ STAC_MACMINI, STAC_MACBOOK, STAC_MACBOOK_PRO_V1, STAC_MACBOOK_PRO_V2, STAC_IMAC_INTEL, + STAC_IMAC_INTEL_20, STAC_922X_MODELS }; @@ -210,7 +219,6 @@ static hda_nid_t stac9205_pin_nids[12] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x14, 0x16, 0x17, 0x18, 0x21, 0x22, - }; static int stac92xx_dmux_enum_info(struct snd_kcontrol *kcontrol, @@ -326,8 +334,6 @@ static struct snd_kcontrol_new stac9200_mixer[] = { }; static struct snd_kcontrol_new stac925x_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0xe, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0xe, 0, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Input Source", @@ -549,44 +555,78 @@ static unsigned int d945gtp5_pin_configs[10] = { 0x02a19320, 0x40000100, }; -static unsigned int macbook_pro_v1_pin_configs[10] = { - 0x0321e230, 0x03a1e020, 0x9017e110, 0x01014010, - 0x01a19021, 0x0381e021, 0x1345e240, 0x13c5e22e, - 0x02a19320, 0x400000fb +static unsigned int intel_mac_v1_pin_configs[10] = { + 0x0121e21f, 0x400000ff, 0x9017e110, 0x400000fd, + 0x400000fe, 0x0181e020, 0x1145e030, 0x11c5e240, + 0x400000fc, 0x400000fb, +}; + +static unsigned int intel_mac_v2_pin_configs[10] = { + 0x0121e21f, 0x90a7012e, 0x9017e110, 0x400000fd, + 0x400000fe, 0x0181e020, 0x1145e230, 0x500000fa, + 0x400000fc, 0x400000fb, +}; + +static unsigned int intel_mac_v3_pin_configs[10] = { + 0x0121e21f, 0x90a7012e, 0x9017e110, 0x400000fd, + 0x400000fe, 0x0181e020, 0x1145e230, 0x11c5e240, + 0x400000fc, 0x400000fb, }; -static unsigned int macbook_pro_v2_pin_configs[10] = { - 0x0221401f, 0x90a70120, 0x01813024, 0x01014010, - 0x400000fd, 0x01016011, 0x1345e240, 0x13c5e22e, +static unsigned int intel_mac_v4_pin_configs[10] = { + 0x0321e21f, 0x03a1e02e, 0x9017e110, 0x9017e11f, + 0x400000fe, 0x0381e020, 0x1345e230, 0x13c5e240, 0x400000fc, 0x400000fb, }; -static unsigned int imac_intel_pin_configs[10] = { - 0x0121e230, 0x90a70120, 0x9017e110, 0x400000fe, - 0x400000fd, 0x0181e021, 0x1145e040, 0x400000fa, +static unsigned int intel_mac_v5_pin_configs[10] = { + 0x0321e21f, 0x03a1e02e, 0x9017e110, 0x9017e11f, + 0x400000fe, 0x0381e020, 0x1345e230, 0x13c5e240, 0x400000fc, 0x400000fb, }; +static unsigned int stac922x_dell_pin_configs[10] = { + 0x0221121e, 0x408103ff, 0x02a1123e, 0x90100310, + 0x408003f1, 0x0221122f, 0x03451340, 0x40c003f2, + 0x50a003f3, 0x405003f4 +}; + static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { [STAC_D945_REF] = ref922x_pin_configs, [STAC_D945GTP3] = d945gtp3_pin_configs, [STAC_D945GTP5] = d945gtp5_pin_configs, - [STAC_MACMINI] = macbook_pro_v1_pin_configs, - [STAC_MACBOOK] = macbook_pro_v1_pin_configs, - [STAC_MACBOOK_PRO_V1] = macbook_pro_v1_pin_configs, - [STAC_MACBOOK_PRO_V2] = macbook_pro_v2_pin_configs, - [STAC_IMAC_INTEL] = imac_intel_pin_configs, + [STAC_922X_DELL] = stac922x_dell_pin_configs, + [STAC_INTEL_MAC_V1] = intel_mac_v1_pin_configs, + [STAC_INTEL_MAC_V2] = intel_mac_v2_pin_configs, + [STAC_INTEL_MAC_V3] = intel_mac_v3_pin_configs, + [STAC_INTEL_MAC_V4] = intel_mac_v4_pin_configs, + [STAC_INTEL_MAC_V5] = intel_mac_v5_pin_configs, + /* for backward compitability */ + [STAC_MACMINI] = intel_mac_v3_pin_configs, + [STAC_MACBOOK] = intel_mac_v5_pin_configs, + [STAC_MACBOOK_PRO_V1] = intel_mac_v3_pin_configs, + [STAC_MACBOOK_PRO_V2] = intel_mac_v3_pin_configs, + [STAC_IMAC_INTEL] = intel_mac_v2_pin_configs, + [STAC_IMAC_INTEL_20] = intel_mac_v3_pin_configs, }; static const char *stac922x_models[STAC_922X_MODELS] = { [STAC_D945_REF] = "ref", [STAC_D945GTP5] = "5stack", [STAC_D945GTP3] = "3stack", + [STAC_922X_DELL] = "dell", + [STAC_INTEL_MAC_V1] = "intel-mac-v1", + [STAC_INTEL_MAC_V2] = "intel-mac-v2", + [STAC_INTEL_MAC_V3] = "intel-mac-v3", + [STAC_INTEL_MAC_V4] = "intel-mac-v4", + [STAC_INTEL_MAC_V5] = "intel-mac-v5", + /* for backward compitability */ [STAC_MACMINI] = "macmini", [STAC_MACBOOK] = "macbook", [STAC_MACBOOK_PRO_V1] = "macbook-pro-v1", [STAC_MACBOOK_PRO_V2] = "macbook-pro", [STAC_IMAC_INTEL] = "imac-intel", + [STAC_IMAC_INTEL_20] = "imac-intel-20", }; static struct snd_pci_quirk stac922x_cfg_tbl[] = { @@ -649,7 +689,10 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { /* other systems */ /* Apple Mac Mini (early 2006) */ SND_PCI_QUIRK(0x8384, 0x7680, - "Mac Mini", STAC_MACMINI), + "Mac Mini", STAC_INTEL_MAC_V3), + /* Dell */ + SND_PCI_QUIRK(0x1028, 0x01d7, "Dell XPS M1210", STAC_922X_DELL), + {} /* terminator */ }; @@ -730,7 +773,8 @@ static unsigned int ref9205_pin_configs[12] = { }; static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = { - ref9205_pin_configs, + [STAC_REF] = ref9205_pin_configs, + [STAC_M43xx] = NULL, }; static const char *stac9205_models[STAC_9205_MODELS] = { @@ -741,6 +785,10 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_9205_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x01f8, + "Dell Precision", STAC_M43xx), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x01ff, + "Dell Precision", STAC_M43xx), {} /* terminator */ }; @@ -770,33 +818,56 @@ static int stac92xx_save_bios_config_regs(struct hda_codec *codec) return 0; } +static void stac92xx_set_config_reg(struct hda_codec *codec, + hda_nid_t pin_nid, unsigned int pin_config) +{ + int i; + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, + pin_config & 0x000000ff); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, + (pin_config & 0x0000ff00) >> 8); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, + (pin_config & 0x00ff0000) >> 16); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + pin_config >> 24); + i = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, + 0x00); + snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n", + pin_nid, i); +} + static void stac92xx_set_config_regs(struct hda_codec *codec) { int i; struct sigmatel_spec *spec = codec->spec; - unsigned int pin_cfg; - if (! spec->pin_nids || ! spec->pin_configs) - return; + if (!spec->pin_configs) + return; - for (i = 0; i < spec->num_pins; i++) { - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, - spec->pin_configs[i] & 0x000000ff); - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, - (spec->pin_configs[i] & 0x0000ff00) >> 8); - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, - (spec->pin_configs[i] & 0x00ff0000) >> 16); - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, - spec->pin_configs[i] >> 24); - pin_cfg = snd_hda_codec_read(codec, spec->pin_nids[i], 0, - AC_VERB_GET_CONFIG_DEFAULT, - 0x00); - snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n", spec->pin_nids[i], pin_cfg); - } + for (i = 0; i < spec->num_pins; i++) + stac92xx_set_config_reg(codec, spec->pin_nids[i], + spec->pin_configs[i]); +} + +static void stac92xx_enable_gpio_mask(struct hda_codec *codec, + int gpio_mask, int gpio_data) +{ + /* Configure GPIOx as output */ + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_DIRECTION, gpio_mask); + /* Configure GPIOx as CMOS */ + snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0x00000000); + /* Assert GPIOx */ + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_DATA, gpio_data); + /* Enable GPIOx */ + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_MASK, gpio_mask); } /* @@ -1168,7 +1239,7 @@ static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) * and 9202/925x. For those, dac_nids[] must be hard-coded. */ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, - const struct auto_pin_cfg *cfg) + struct auto_pin_cfg *cfg) { struct sigmatel_spec *spec = codec->spec; int i, j, conn_len = 0; @@ -1193,6 +1264,13 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec, } if (j == conn_len) { + if (spec->multiout.num_dacs > 0) { + /* we have already working output pins, + * so let's drop the broken ones again + */ + cfg->line_outs = spec->multiout.num_dacs; + break; + } /* error out, no available DAC found */ snd_printk(KERN_ERR "%s: No available DAC for pin 0x%x\n", @@ -1334,7 +1412,15 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, continue; add_spec_dacs(spec, nid); } - + for (i = 0; i < cfg->line_outs; i++) { + nid = snd_hda_codec_read(codec, cfg->line_out_pins[i], 0, + AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + if (check_in_dac_nids(spec, nid)) + nid = 0; + if (! nid) + continue; + add_spec_dacs(spec, nid); + } for (i = old_num_dacs; i < spec->multiout.num_dacs; i++) { static const char *pfxs[] = { "Speaker", "External Speaker", "Speaker2", @@ -1891,7 +1977,7 @@ static int patch_stac9200(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; - spec->num_pins = 8; + spec->num_pins = ARRAY_SIZE(stac9200_pin_nids); spec->pin_nids = stac9200_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_9200_MODELS, stac9200_models, @@ -1941,7 +2027,7 @@ static int patch_stac925x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; - spec->num_pins = 8; + spec->num_pins = ARRAY_SIZE(stac925x_pin_nids); spec->pin_nids = stac925x_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_925x_MODELS, stac925x_models, @@ -2013,29 +2099,41 @@ static int patch_stac922x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; - spec->num_pins = 10; + spec->num_pins = ARRAY_SIZE(stac922x_pin_nids); spec->pin_nids = stac922x_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_922X_MODELS, stac922x_models, stac922x_cfg_tbl); - if (spec->board_config == STAC_MACMINI) { + if (spec->board_config == STAC_INTEL_MAC_V3) { spec->gpio_mute = 1; /* Intel Macs have all same PCI SSID, so we need to check * codec SSID to distinguish the exact models */ printk(KERN_INFO "hda_codec: STAC922x, Apple subsys_id=%x\n", codec->subsystem_id); switch (codec->subsystem_id) { - case 0x106b0a00: /* MacBook First generatoin */ - spec->board_config = STAC_MACBOOK; + + case 0x106b0800: + spec->board_config = STAC_INTEL_MAC_V1; + break; + case 0x106b0600: + case 0x106b0700: + spec->board_config = STAC_INTEL_MAC_V2; break; - case 0x106b0200: /* MacBook Pro first generation */ - spec->board_config = STAC_MACBOOK_PRO_V1; + case 0x106b0e00: + case 0x106b0f00: + case 0x106b1600: + case 0x106b1700: + case 0x106b0200: + case 0x106b1e00: + spec->board_config = STAC_INTEL_MAC_V3; break; - case 0x106b1e00: /* MacBook Pro second generation */ - spec->board_config = STAC_MACBOOK_PRO_V2; + case 0x106b1a00: + case 0x00000100: + spec->board_config = STAC_INTEL_MAC_V4; break; - case 0x106b0700: /* Intel-based iMac */ - spec->board_config = STAC_IMAC_INTEL; + case 0x106b0a00: + case 0x106b2200: + spec->board_config = STAC_INTEL_MAC_V5; break; } } @@ -2082,6 +2180,13 @@ static int patch_stac922x(struct hda_codec *codec) codec->patch_ops = stac92xx_patch_ops; + /* Fix Mux capture level; max to 2 */ + snd_hda_override_amp_caps(codec, 0x12, HDA_OUTPUT, + (0 << AC_AMPCAP_OFFSET_SHIFT) | + (2 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x27 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (0 << AC_AMPCAP_MUTE_SHIFT)); + return 0; } @@ -2095,7 +2200,7 @@ static int patch_stac927x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; - spec->num_pins = 14; + spec->num_pins = ARRAY_SIZE(stac927x_pin_nids); spec->pin_nids = stac927x_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_927X_MODELS, stac927x_models, @@ -2141,7 +2246,9 @@ static int patch_stac927x(struct hda_codec *codec) } spec->multiout.dac_nids = spec->dac_nids; - + /* GPIO0 High = Enable EAPD */ + stac92xx_enable_gpio_mask(codec, 0x00000001, 0x00000001); + err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); if (!err) { if (spec->board_config < 0) { @@ -2159,27 +2266,20 @@ static int patch_stac927x(struct hda_codec *codec) codec->patch_ops = stac92xx_patch_ops; - /* Fix Mux capture level; max to 2 */ - snd_hda_override_amp_caps(codec, 0x12, HDA_OUTPUT, - (0 << AC_AMPCAP_OFFSET_SHIFT) | - (2 << AC_AMPCAP_NUM_STEPS_SHIFT) | - (0x27 << AC_AMPCAP_STEP_SIZE_SHIFT) | - (0 << AC_AMPCAP_MUTE_SHIFT)); - return 0; } static int patch_stac9205(struct hda_codec *codec) { struct sigmatel_spec *spec; - int err; + int err, gpio_mask, gpio_data; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; codec->spec = spec; - spec->num_pins = 14; + spec->num_pins = ARRAY_SIZE(stac9205_pin_nids); spec->pin_nids = stac9205_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_9205_MODELS, stac9205_models, @@ -2209,19 +2309,21 @@ static int patch_stac9205(struct hda_codec *codec) spec->mixer = stac9205_mixer; spec->multiout.dac_nids = spec->dac_nids; + + if (spec->board_config == STAC_M43xx) { + /* Enable SPDIF in/out */ + stac92xx_set_config_reg(codec, 0x1f, 0x01441030); + stac92xx_set_config_reg(codec, 0x20, 0x1c410030); + + gpio_mask = 0x00000007; /* GPIO0-2 */ + /* GPIO0 High = EAPD, GPIO1 Low = DRM, + * GPIO2 High = Headphone Mute + */ + gpio_data = 0x00000005; + } else + gpio_mask = gpio_data = 0x00000001; /* GPIO0 High = EAPD */ - /* Configure GPIO0 as EAPD output */ - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DIRECTION, 0x00000001); - /* Configure GPIO0 as CMOS */ - snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0x00000000); - /* Assert GPIO0 high */ - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DATA, 0x00000001); - /* Enable GPIO0 */ - snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_MASK, 0x00000001); - + stac92xx_enable_gpio_mask(codec, gpio_mask, gpio_data); err = stac92xx_parse_auto_config(codec, 0x1f, 0x20); if (!err) { if (spec->board_config < 0) { @@ -2256,8 +2358,8 @@ static struct hda_input_mux vaio_mux = { .num_items = 2, .items = { /* { "HP", 0x0 }, */ - { "Line", 0x1 }, - { "Mic", 0x2 }, + { "Mic Jack", 0x1 }, + { "Internal Mic", 0x2 }, { "PCM", 0x3 }, } }; @@ -2268,7 +2370,7 @@ static struct hda_verb vaio_init[] = { {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */ {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x2}, /* mic-sel: 0a,0d,14,02 */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */ @@ -2284,7 +2386,7 @@ static struct hda_verb vaio_ar_init[] = { {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ /* {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },*/ /* Optical Out */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x2}, /* mic-sel: 0a,0d,14,02 */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ /* {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},*/ /* Optical Out */ diff --git a/sound/pci/ice1712/revo.c b/sound/pci/ice1712/revo.c index 690ceb340644..d18a31e188a9 100644 --- a/sound/pci/ice1712/revo.c +++ b/sound/pci/ice1712/revo.c @@ -186,7 +186,12 @@ static int revo51_i2c_init(struct snd_ice1712 *ice, #define AK_DAC(xname,xch) { .name = xname, .num_channels = xch } static const struct snd_akm4xxx_dac_channel revo71_front[] = { - AK_DAC("PCM Playback Volume", 2) + { + .name = "PCM Playback Volume", + .num_channels = 2, + /* front channels DAC supports muting */ + .switch_name = "PCM Playback Switch", + }, }; static const struct snd_akm4xxx_dac_channel revo71_surround[] = { diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 1d9232d2db34..170781a72292 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -24,6 +24,7 @@ #include <linux/interrupt.h> #include <linux/pci.h> #include <linux/firmware.h> +#include <linux/vmalloc.h> #include <asm/io.h> #include <sound/core.h> #include "mixart.h" diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c index 03b3a4792f73..c7621bd770a6 100644 --- a/sound/pci/nm256/nm256.c +++ b/sound/pci/nm256/nm256.c @@ -1533,7 +1533,8 @@ snd_nm256_create(struct snd_card *card, struct pci_dev *pci, printk(KERN_ERR " force the driver to load by " "passing in the module parameter\n"); printk(KERN_ERR " force_ac97=1\n"); - printk(KERN_ERR " or try sb16 or cs423x drivers instead.\n"); + printk(KERN_ERR " or try sb16, opl3sa2, or " + "cs423x drivers instead.\n"); err = -ENXIO; goto __error; } diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index bd7dbd267ed1..2de27405a0bd 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -406,7 +406,7 @@ static snd_pcm_uframes_t rme9652_hw_pointer(struct snd_rme9652 *rme9652) } else if (!frag) return 0; offset -= rme9652->max_jitter; - if (offset < 0) + if ((int)offset < 0) offset += period_size * 2; } else { if (offset > period_size + rme9652->max_jitter) { diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index a28992269f5e..6ea09df0c73a 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -2098,7 +2098,7 @@ static int snd_via82xx_chip_init(struct via82xx *chip) pci_read_config_byte(chip->pci, VIA_ACLINK_STAT, &pval); if (pval & VIA_ACLINK_C00_READY) /* primary codec ready */ break; - schedule_timeout_uninterruptible(1); + schedule_timeout(1); } while (time_before(jiffies, end_time)); if ((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_BUSY) @@ -2117,7 +2117,7 @@ static int snd_via82xx_chip_init(struct via82xx *chip) chip->ac97_secondary = 1; goto __ac97_ok2; } - schedule_timeout_interruptible(1); + schedule_timeout(1); } while (time_before(jiffies, end_time)); /* This is ok, the most of motherboards have only one codec */ @@ -2431,7 +2431,6 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci, { struct snd_card *card; struct via82xx *chip; - unsigned char revision; int chip_type = 0, card_type; unsigned int i; int err; @@ -2441,18 +2440,17 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci, return -ENOMEM; card_type = pci_id->driver_data; - pci_read_config_byte(pci, PCI_REVISION_ID, &revision); switch (card_type) { case TYPE_CARD_VIA686: strcpy(card->driver, "VIA686A"); - sprintf(card->shortname, "VIA 82C686A/B rev%x", revision); + sprintf(card->shortname, "VIA 82C686A/B rev%x", pci->revision); chip_type = TYPE_VIA686; break; case TYPE_CARD_VIA8233: chip_type = TYPE_VIA8233; - sprintf(card->shortname, "VIA 823x rev%x", revision); + sprintf(card->shortname, "VIA 823x rev%x", pci->revision); for (i = 0; i < ARRAY_SIZE(via823x_cards); i++) { - if (revision == via823x_cards[i].revision) { + if (pci->revision == via823x_cards[i].revision) { chip_type = via823x_cards[i].type; strcpy(card->shortname, via823x_cards[i].name); break; @@ -2460,7 +2458,7 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci, } if (chip_type != TYPE_VIA8233A) { if (dxs_support == VIA_DXS_AUTO) - dxs_support = check_dxs_list(pci, revision); + dxs_support = check_dxs_list(pci, pci->revision); /* force to use VIA8233 or 8233A model according to * dxs_support module option */ @@ -2471,7 +2469,7 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci, } if (chip_type == TYPE_VIA8233A) strcpy(card->driver, "VIA8233A"); - else if (revision >= VIA_REV_8237) + else if (pci->revision >= VIA_REV_8237) strcpy(card->driver, "VIA8237"); /* no slog assignment */ else strcpy(card->driver, "VIA8233"); @@ -2482,7 +2480,7 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci, goto __error; } - if ((err = snd_via82xx_create(card, pci, chip_type, revision, + if ((err = snd_via82xx_create(card, pci, chip_type, pci->revision, ac97_clock, &chip)) < 0) goto __error; card->private_data = chip; diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index b338e15db0d9..72425e73abae 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -983,7 +983,7 @@ static int snd_via82xx_chip_init(struct via82xx_modem *chip) pci_read_config_byte(chip->pci, VIA_ACLINK_STAT, &pval); if (pval & VIA_ACLINK_C00_READY) /* primary codec ready */ break; - schedule_timeout_uninterruptible(1); + schedule_timeout(1); } while (time_before(jiffies, end_time)); if ((val = snd_via82xx_codec_xread(chip)) & VIA_REG_AC97_BUSY) @@ -1001,7 +1001,7 @@ static int snd_via82xx_chip_init(struct via82xx_modem *chip) chip->ac97_secondary = 1; goto __ac97_ok2; } - schedule_timeout_interruptible(1); + schedule_timeout(1); } while (time_before(jiffies, end_time)); /* This is ok, the most of motherboards have only one codec */ @@ -1162,7 +1162,6 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci, { struct snd_card *card; struct via82xx_modem *chip; - unsigned char revision; int chip_type = 0, card_type; unsigned int i; int err; @@ -1172,7 +1171,6 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci, return -ENOMEM; card_type = pci_id->driver_data; - pci_read_config_byte(pci, PCI_REVISION_ID, &revision); switch (card_type) { case TYPE_CARD_VIA82XX_MODEM: strcpy(card->driver, "VIA82XX-MODEM"); @@ -1184,7 +1182,7 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci, goto __error; } - if ((err = snd_via82xx_create(card, pci, chip_type, revision, + if ((err = snd_via82xx_create(card, pci, chip_type, pci->revision, ac97_clock, &chip)) < 0) goto __error; card->private_data = chip; diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index ea861bceaddf..ab7a81c35705 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -2404,7 +2404,7 @@ int __devinit snd_ymfpci_create(struct snd_card *card, chip->pci = pci; chip->irq = -1; chip->device_id = pci->device; - pci_read_config_byte(pci, PCI_REVISION_ID, &chip->rev); + chip->rev = pci->revision; chip->reg_area_phys = pci_resource_start(pci, 0); chip->reg_area_virt = ioremap_nocache(chip->reg_area_phys, 0x8000); pci_set_master(pci); |