diff options
Diffstat (limited to 'sound/pci')
30 files changed, 2073 insertions, 5010 deletions
diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index 4cc315daeda..bc86cb726d7 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com> + Copyright (C) 1997-2012 AudioScience Inc. <support@audioscience.com> This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as diff --git a/sound/pci/asihpi/hpios.c b/sound/pci/asihpi/hpios.c index 2d7d1c2e1d0..5ef4fe96436 100644 --- a/sound/pci/asihpi/hpios.c +++ b/sound/pci/asihpi/hpios.c @@ -1,7 +1,7 @@ /****************************************************************************** AudioScience HPI driver - Copyright (C) 1997-2011 AudioScience Inc. <support@audioscience.com> + Copyright (C) 1997-2012 AudioScience Inc. <support@audioscience.com> This program is free software; you can redistribute it and/or modify it under the terms of version 2 of the GNU General Public License as @@ -39,9 +39,9 @@ void hpios_delay_micro_seconds(u32 num_micro_sec) } -/** Allocated an area of locked memory for bus master DMA operations. +/** Allocate an area of locked memory for bus master DMA operations. -On error, return -ENOMEM, and *pMemArea.size = 0 +If allocation fails, return 1, and *pMemArea.size = 0 */ u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size, struct pci_dev *pdev) @@ -62,7 +62,7 @@ u16 hpios_locked_mem_alloc(struct consistent_dma_area *p_mem_area, u32 size, HPI_DEBUG_LOG(WARNING, "failed to allocate %d bytes locked memory\n", size); p_mem_area->size = 0; - return -ENOMEM; + return 1; } } diff --git a/sound/pci/asihpi/hpios.h b/sound/pci/asihpi/hpios.h index c5cef113c20..d3fbd0d76c3 100644 --- a/sound/pci/asihpi/hpios.h +++ b/sound/pci/asihpi/hpios.h @@ -30,7 +30,6 @@ HPI Operating System Specific macros for Linux Kernel driver #define HPI_BUILD_KERNEL_MODE #include <linux/io.h> -#include <asm/system.h> #include <linux/ioctl.h> #include <linux/kernel.h> #include <linux/string.h> diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index bb938153a96..466a5c8e835 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -26,7 +26,7 @@ #include <sound/mpu401.h> #include <sound/hwdep.h> #include <sound/ac97_codec.h> - +#include <sound/tlv.h> #endif #ifndef CHIP_AU8820 @@ -107,6 +107,14 @@ #define NR_WTPB 0x20 /* WT channels per each bank. */ #define NR_PCM 0x10 +struct pcm_vol { + struct snd_kcontrol *kctl; + int active; + int dma; + int mixin[4]; + int vol[4]; +}; + /* Structs */ typedef struct { //int this_08; /* Still unknown */ @@ -168,6 +176,7 @@ struct snd_vortex { /* Xtalk canceler */ int xt_mode; /* 1: speakers, 0:headphones. */ #endif + struct pcm_vol pcm_vol[NR_PCM]; int isquad; /* cache of extended ID codec flag. */ @@ -239,7 +248,7 @@ static int vortex_alsafmt_aspfmt(int alsafmt); /* Connection stuff. */ static void vortex_connect_default(vortex_t * vortex, int en); static int vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, - int dir, int type); + int dir, int type, int subdev); static char vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype); #ifndef CHIP_AU8810 diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index 6933a27a5d7..525f881f040 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -2050,8 +2050,6 @@ vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype) } /* Default Connections */ -static int -vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type); static void vortex_connect_default(vortex_t * vortex, int en) { @@ -2111,15 +2109,13 @@ static void vortex_connect_default(vortex_t * vortex, int en) Return: Return allocated DMA or same DMA passed as "dma" when dma >= 0. */ static int -vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type) +vortex_adb_allocroute(vortex_t *vortex, int dma, int nr_ch, int dir, + int type, int subdev) { stream_t *stream; int i, en; + struct pcm_vol *p; - if ((nr_ch == 3) - || ((dir == SNDRV_PCM_STREAM_CAPTURE) && (nr_ch > 2))) - return -EBUSY; - if (dma >= 0) { en = 0; vortex_adb_checkinout(vortex, @@ -2250,6 +2246,14 @@ vortex_adb_allocroute(vortex_t * vortex, int dma, int nr_ch, int dir, int type) MIX_DEFIGAIN); #endif } + if (stream->type == VORTEX_PCM_ADB && en) { + p = &vortex->pcm_vol[subdev]; + p->dma = dma; + for (i = 0; i < nr_ch; i++) + p->mixin[i] = mix[i]; + for (i = 0; i < ch_top; i++) + p->vol[i] = 0; + } } #ifndef CHIP_AU8820 else { @@ -2473,7 +2477,7 @@ static irqreturn_t vortex_interrupt(int irq, void *dev_id) hwread(vortex->mmio, VORTEX_IRQ_STAT); handled = 1; } - if (source & IRQ_MIDI) { + if ((source & IRQ_MIDI) && vortex->rmidi) { snd_mpu401_uart_interrupt(vortex->irq, vortex->rmidi->private_data); handled = 1; diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 0ef2f971220..e59f120742a 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -122,6 +122,18 @@ static struct snd_pcm_hw_constraint_list hw_constraints_au8830_channels = { .mask = 0, }; #endif + +static void vortex_notify_pcm_vol_change(struct snd_card *card, + struct snd_kcontrol *kctl, int activate) +{ + if (activate) + kctl->vd[0].access &= ~SNDRV_CTL_ELEM_ACCESS_INACTIVE; + else + kctl->vd[0].access |= SNDRV_CTL_ELEM_ACCESS_INACTIVE; + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE | + SNDRV_CTL_EVENT_MASK_INFO, &(kctl->id)); +} + /* open callback */ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream) { @@ -230,12 +242,14 @@ snd_vortex_pcm_hw_params(struct snd_pcm_substream *substream, if (stream != NULL) vortex_adb_allocroute(chip, stream->dma, stream->nr_ch, stream->dir, - stream->type); + stream->type, + substream->number); /* Alloc routes. */ dma = vortex_adb_allocroute(chip, -1, params_channels(hw_params), - substream->stream, type); + substream->stream, type, + substream->number); if (dma < 0) { spin_unlock_irq(&chip->lock); return dma; @@ -246,6 +260,11 @@ snd_vortex_pcm_hw_params(struct snd_pcm_substream *substream, vortex_adbdma_setbuffers(chip, dma, params_period_bytes(hw_params), params_periods(hw_params)); + if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB) { + chip->pcm_vol[substream->number].active = 1; + vortex_notify_pcm_vol_change(chip->card, + chip->pcm_vol[substream->number].kctl, 1); + } } #ifndef CHIP_AU8810 else { @@ -275,10 +294,18 @@ static int snd_vortex_pcm_hw_free(struct snd_pcm_substream *substream) spin_lock_irq(&chip->lock); // Delete audio routes. if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) { - if (stream != NULL) + if (stream != NULL) { + if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_ADB) { + chip->pcm_vol[substream->number].active = 0; + vortex_notify_pcm_vol_change(chip->card, + chip->pcm_vol[substream->number].kctl, + 0); + } vortex_adb_allocroute(chip, stream->dma, stream->nr_ch, stream->dir, - stream->type); + stream->type, + substream->number); + } } #ifndef CHIP_AU8810 else { @@ -506,6 +533,83 @@ static struct snd_kcontrol_new snd_vortex_mixer_spdif[] __devinitdata = { }, }; +/* subdevice PCM Volume control */ + +static int snd_vortex_pcm_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + vortex_t *vortex = snd_kcontrol_chip(kcontrol); + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = (VORTEX_IS_QUAD(vortex) ? 4 : 2); + uinfo->value.integer.min = -128; + uinfo->value.integer.max = 32; + return 0; +} + +static int snd_vortex_pcm_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int i; + vortex_t *vortex = snd_kcontrol_chip(kcontrol); + int subdev = kcontrol->id.subdevice; + struct pcm_vol *p = &vortex->pcm_vol[subdev]; + int max_chn = (VORTEX_IS_QUAD(vortex) ? 4 : 2); + for (i = 0; i < max_chn; i++) + ucontrol->value.integer.value[i] = p->vol[i]; + return 0; +} + +static int snd_vortex_pcm_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int i; + int changed = 0; + int mixin; + unsigned char vol; + vortex_t *vortex = snd_kcontrol_chip(kcontrol); + int subdev = kcontrol->id.subdevice; + struct pcm_vol *p = &vortex->pcm_vol[subdev]; + int max_chn = (VORTEX_IS_QUAD(vortex) ? 4 : 2); + for (i = 0; i < max_chn; i++) { + if (p->vol[i] != ucontrol->value.integer.value[i]) { + p->vol[i] = ucontrol->value.integer.value[i]; + if (p->active) { + switch (vortex->dma_adb[p->dma].nr_ch) { + case 1: + mixin = p->mixin[0]; + break; + case 2: + default: + mixin = p->mixin[(i < 2) ? i : (i - 2)]; + break; + case 4: + mixin = p->mixin[i]; + break; + }; + vol = p->vol[i]; + vortex_mix_setinputvolumebyte(vortex, + vortex->mixplayb[i], mixin, vol); + } + changed = 1; + } + } + return changed; +} + +static const DECLARE_TLV_DB_MINMAX(vortex_pcm_vol_db_scale, -9600, 2400); + +static struct snd_kcontrol_new snd_vortex_pcm_vol __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = "PCM Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ | + SNDRV_CTL_ELEM_ACCESS_INACTIVE, + .info = snd_vortex_pcm_vol_info, + .get = snd_vortex_pcm_vol_get, + .put = snd_vortex_pcm_vol_put, + .tlv = { .p = vortex_pcm_vol_db_scale }, +}; + /* create a pcm device */ static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) { @@ -555,5 +659,20 @@ static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) return err; } } + if (VORTEX_PCM_TYPE(pcm) == VORTEX_PCM_ADB) { + for (i = 0; i < NR_PCM; i++) { + chip->pcm_vol[i].active = 0; + chip->pcm_vol[i].dma = -1; + kctl = snd_ctl_new1(&snd_vortex_pcm_vol, chip); + if (!kctl) + return -ENOMEM; + chip->pcm_vol[i].kctl = kctl; + kctl->id.device = 0; + kctl->id.subdevice = i; + err = snd_ctl_add(chip->card, kctl); + if (err < 0) + return err; + } + } return 0; } diff --git a/sound/pci/aw2/aw2-saa7146.c b/sound/pci/aw2/aw2-saa7146.c index 8afd8b5d1ac..4439636971e 100644 --- a/sound/pci/aw2/aw2-saa7146.c +++ b/sound/pci/aw2/aw2-saa7146.c @@ -27,7 +27,6 @@ #include <linux/pci.h> #include <linux/interrupt.h> #include <linux/delay.h> -#include <asm/system.h> #include <asm/io.h> #include <sound/core.h> #include <sound/initval.h> diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c index b78f3fc3c33..6109490b83e 100644 --- a/sound/pci/ctxfi/ctvmem.c +++ b/sound/pci/ctxfi/ctvmem.c @@ -36,7 +36,7 @@ get_vm_block(struct ct_vm *vm, unsigned int size) size = CT_PAGE_ALIGN(size); if (size > vm->size) { - printk(KERN_ERR "ctxfi: Fail! No sufficient device virtural " + printk(KERN_ERR "ctxfi: Fail! No sufficient device virtual " "memory space available!\n"); return NULL; } diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index cb557c603a8..a8faae1c85e 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -142,6 +142,7 @@ static int enable_mpu[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = 2}; #ifdef SUPPORT_JOYSTICK static bool joystick[SNDRV_CARDS]; #endif +static int radio_nr[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -1}; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); @@ -165,6 +166,9 @@ MODULE_PARM_DESC(enable_mpu, "Enable MPU401. (0 = off, 1 = on, 2 = auto)"); module_param_array(joystick, bool, NULL, 0444); MODULE_PARM_DESC(joystick, "Enable joystick."); #endif +module_param_array(radio_nr, int, NULL, 0444); +MODULE_PARM_DESC(radio_nr, "Radio device numbers"); + #define NR_APUS 64 @@ -558,6 +562,7 @@ struct es1968 { struct work_struct hwvol_work; #ifdef CONFIG_SND_ES1968_RADIO + struct v4l2_device v4l2_dev; struct snd_tea575x tea; #endif }; @@ -2613,6 +2618,7 @@ static int snd_es1968_free(struct es1968 *chip) #ifdef CONFIG_SND_ES1968_RADIO snd_tea575x_exit(&chip->tea); + v4l2_device_unregister(&chip->v4l2_dev); #endif if (chip->irq >= 0) @@ -2655,6 +2661,7 @@ static int __devinit snd_es1968_create(struct snd_card *card, int capt_streams, int chip_type, int do_pm, + int radio_nr, struct es1968 **chip_ret) { static struct snd_device_ops ops = { @@ -2751,7 +2758,14 @@ static int __devinit snd_es1968_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); #ifdef CONFIG_SND_ES1968_RADIO + err = v4l2_device_register(&pci->dev, &chip->v4l2_dev); + if (err < 0) { + snd_es1968_free(chip); + return err; + } + chip->tea.v4l2_dev = &chip->v4l2_dev; chip->tea.private_data = chip; + chip->tea.radio_nr = radio_nr; chip->tea.ops = &snd_es1968_tea_ops; strlcpy(chip->tea.card, "SF64-PCE2", sizeof(chip->tea.card)); sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); @@ -2797,6 +2811,7 @@ static int __devinit snd_es1968_probe(struct pci_dev *pci, pcm_substreams_c[dev], pci_id->driver_data, use_pm[dev], + radio_nr[dev], &chip)) < 0) { snd_card_free(card); return err; diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index 9597ef1eccc..a416ea8af3e 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -58,6 +58,7 @@ static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card * High 16-bits are video (radio) device number + 1 */ static int tea575x_tuner[SNDRV_CARDS]; +static int radio_nr[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -1}; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for the FM801 soundcard."); @@ -67,6 +68,9 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable FM801 soundcard."); module_param_array(tea575x_tuner, int, NULL, 0444); MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (0 = auto, 1 = SF256-PCS, 2=SF256-PCP, 3=SF64-PCR, 8=disable, +16=tuner-only)."); +module_param_array(radio_nr, int, NULL, 0444); +MODULE_PARM_DESC(radio_nr, "Radio device numbers"); + #define TUNER_DISABLED (1<<3) #define TUNER_ONLY (1<<4) @@ -197,6 +201,7 @@ struct fm801 { struct snd_info_entry *proc_entry; #ifdef CONFIG_SND_FM801_TEA575X_BOOL + struct v4l2_device v4l2_dev; struct snd_tea575x tea; #endif @@ -1154,8 +1159,10 @@ static int snd_fm801_free(struct fm801 *chip) __end_hw: #ifdef CONFIG_SND_FM801_TEA575X_BOOL - if (!(chip->tea575x_tuner & TUNER_DISABLED)) + if (!(chip->tea575x_tuner & TUNER_DISABLED)) { snd_tea575x_exit(&chip->tea); + v4l2_device_unregister(&chip->v4l2_dev); + } #endif if (chip->irq >= 0) free_irq(chip->irq, chip); @@ -1175,6 +1182,7 @@ static int snd_fm801_dev_free(struct snd_device *device) static int __devinit snd_fm801_create(struct snd_card *card, struct pci_dev * pci, int tea575x_tuner, + int radio_nr, struct fm801 ** rchip) { struct fm801 *chip; @@ -1234,6 +1242,13 @@ static int __devinit snd_fm801_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); #ifdef CONFIG_SND_FM801_TEA575X_BOOL + err = v4l2_device_register(&pci->dev, &chip->v4l2_dev); + if (err < 0) { + snd_fm801_free(chip); + return err; + } + chip->tea.v4l2_dev = &chip->v4l2_dev; + chip->tea.radio_nr = radio_nr; chip->tea.private_data = chip; chip->tea.ops = &snd_fm801_tea_ops; sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); @@ -1241,6 +1256,7 @@ static int __devinit snd_fm801_create(struct snd_card *card, (tea575x_tuner & TUNER_TYPE_MASK) < 4) { if (snd_tea575x_init(&chip->tea)) { snd_printk(KERN_ERR "TEA575x radio not found\n"); + snd_fm801_free(chip); return -ENODEV; } } else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) { @@ -1287,7 +1303,7 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci, err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); if (err < 0) return err; - if ((err = snd_fm801_create(card, pci, tea575x_tuner[dev], &chip)) < 0) { + if ((err = snd_fm801_create(card, pci, tea575x_tuner[dev], radio_nr[dev], &chip)) < 0) { snd_card_free(card); return err; } diff --git a/sound/pci/hda/alc260_quirks.c b/sound/pci/hda/alc260_quirks.c deleted file mode 100644 index 3b5170b9700..00000000000 --- a/sound/pci/hda/alc260_quirks.c +++ /dev/null @@ -1,968 +0,0 @@ -/* - * ALC260 quirk models - * included by patch_realtek.c - */ - -/* ALC260 models */ -enum { - ALC260_AUTO, - ALC260_BASIC, - ALC260_FUJITSU_S702X, - ALC260_ACER, - ALC260_WILL, - ALC260_REPLACER_672V, - ALC260_FAVORIT100, -#ifdef CONFIG_SND_DEBUG - ALC260_TEST, -#endif - ALC260_MODEL_LAST /* last tag */ -}; - -static const hda_nid_t alc260_dac_nids[1] = { - /* front */ - 0x02, -}; - -static const hda_nid_t alc260_adc_nids[1] = { - /* ADC0 */ - 0x04, -}; - -static const hda_nid_t alc260_adc_nids_alt[1] = { - /* ADC1 */ - 0x05, -}; - -/* NIDs used when simultaneous access to both ADCs makes sense. Note that - * alc260_capture_mixer assumes ADC0 (nid 0x04) is the first ADC. - */ -static const hda_nid_t alc260_dual_adc_nids[2] = { - /* ADC0, ADC1 */ - 0x04, 0x05 -}; - -#define ALC260_DIGOUT_NID 0x03 -#define ALC260_DIGIN_NID 0x06 - -static const struct hda_input_mux alc260_capture_source = { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Front Mic", 0x1 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - }, -}; - -/* On Fujitsu S702x laptops capture only makes sense from Mic/LineIn jack, - * headphone jack and the internal CD lines since these are the only pins at - * which audio can appear. For flexibility, also allow the option of - * recording the mixer output on the second ADC (ADC0 doesn't have a - * connection to the mixer output). - */ -static const struct hda_input_mux alc260_fujitsu_capture_sources[2] = { - { - .num_items = 3, - .items = { - { "Mic/Line", 0x0 }, - { "CD", 0x4 }, - { "Headphone", 0x2 }, - }, - }, - { - .num_items = 4, - .items = { - { "Mic/Line", 0x0 }, - { "CD", 0x4 }, - { "Headphone", 0x2 }, - { "Mixer", 0x5 }, - }, - }, - -}; - -/* Acer TravelMate(/Extensa/Aspire) notebooks have similar configuration to - * the Fujitsu S702x, but jacks are marked differently. - */ -static const struct hda_input_mux alc260_acer_capture_sources[2] = { - { - .num_items = 4, - .items = { - { "Mic", 0x0 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - { "Headphone", 0x5 }, - }, - }, - { - .num_items = 5, - .items = { - { "Mic", 0x0 }, - { "Line", 0x2 }, - { "CD", 0x4 }, - { "Headphone", 0x6 }, - { "Mixer", 0x5 }, - }, - }, -}; - -/* Maxdata Favorit 100XS */ -static const struct hda_input_mux alc260_favorit100_capture_sources[2] = { - { - .num_items = 2, - .items = { - { "Line/Mic", 0x0 }, - { "CD", 0x4 }, - }, - }, - { - .num_items = 3, - .items = { - { "Line/Mic", 0x0 }, - { "CD", 0x4 }, - { "Mixer", 0x5 }, - }, - }, -}; - -/* - * This is just place-holder, so there's something for alc_build_pcms to look - * at when it calculates the maximum number of channels. ALC260 has no mixer - * element which allows changing the channel mode, so the verb list is - * never used. - */ -static const struct hda_channel_mode alc260_modes[1] = { - { 2, NULL }, -}; - - -/* Mixer combinations - * - * basic: base_output + input + pc_beep + capture - * fujitsu: fujitsu + capture - * acer: acer + capture - */ - -static const struct snd_kcontrol_new alc260_base_output_mixer[] = { - HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x08, 2, HDA_INPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x09, 2, HDA_INPUT), - HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Mono Playback Switch", 0x0a, 1, 2, HDA_INPUT), - { } /* end */ -}; - -static const struct snd_kcontrol_new alc260_input_mixer[] = { - HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT), - HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT), - { } /* end */ -}; - -/* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12, - * HP jack = 0x14, CD audio = 0x16, internal speaker = 0x10. - */ -static const struct snd_kcontrol_new alc260_fujitsu_mixer[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Headphone Playback Switch", 0x08, 2, HDA_INPUT), - ALC_PIN_MODE("Headphone Jack Mode", 0x14, ALC_PIN_DIR_INOUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT), - ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT), - { } /* end */ -}; - -/* Mixer for Acer TravelMate(/Extensa/Aspire) notebooks. Note that current - * versions of the ALC260 don't act on requests to enable mic bias from NID - * 0x0f (used to drive the headphone jack in these laptops). The ALC260 - * datasheet doesn't mention this restriction. At this stage it's not clear - * whether this behaviour is intentional or is a hardware bug in chip - * revisions available in early 2006. Therefore for now allow the - * "Headphone Jack Mode" control to span all choices, but if it turns out - * that the lack of mic bias for this NID is intentional we could change the - * mode from ALC_PIN_DIR_INOUT to ALC_PIN_DIR_INOUT_NOMICBIAS. - * - * In addition, Acer TravelMate(/Extensa/Aspire) notebooks in early 2006 - * don't appear to make the mic bias available from the "line" jack, even - * though the NID used for this jack (0x14) can supply it. The theory is - * that perhaps Acer have included blocking capacitors between the ALC260 - * and the output jack. If this turns out to be the case for all such - * models the "Line Jack Mode" mode could be changed from ALC_PIN_DIR_INOUT - * to ALC_PIN_DIR_INOUT_NOMICBIAS. - * - * The C20x Tablet series have a mono internal speaker which is controlled - * via the chip's Mono sum widget and pin complex, so include the necessary - * controls for such models. On models without a "mono speaker" the control - * won't do anything. - */ -static const struct snd_kcontrol_new alc260_acer_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT), - ALC_PIN_MODE("Headphone Jack Mode", 0x0f, ALC_PIN_DIR_INOUT), - HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0a, 1, 0x0, - HDA_OUTPUT), - HDA_BIND_MUTE_MONO("Speaker Playback Switch", 0x0a, 1, 2, - HDA_INPUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), - ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), - HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), - ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT), - { } /* end */ -}; - -/* Maxdata Favorit 100XS: one output and one input (0x12) jack - */ -static const struct snd_kcontrol_new alc260_favorit100_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT), - ALC_PIN_MODE("Output Jack Mode", 0x0f, ALC_PIN_DIR_INOUT), - HDA_CODEC_VOLUME("Line/Mic Playback Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Line/Mic Playback Switch", 0x07, 0x0, HDA_INPUT), - ALC_PIN_MODE("Line/Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), - { } /* end */ -}; - -/* Packard bell V7900 ALC260 pin usage: HP = 0x0f, Mic jack = 0x12, - * Line In jack = 0x14, CD audio = 0x16, pc beep = 0x17. - */ -static const struct snd_kcontrol_new alc260_will_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), - ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), - HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), - ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT), - HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), - { } /* end */ -}; - -/* Replacer 672V ALC260 pin usage: Mic jack = 0x12, - * Line In jack = 0x14, ATAPI Mic = 0x13, speaker = 0x0f. - */ -static const struct snd_kcontrol_new alc260_replacer_672v_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Master Playback Switch", 0x08, 0x2, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT), - ALC_PIN_MODE("Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), - HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x07, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("ATATI Mic Playback Switch", 0x07, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), - HDA |