diff options
Diffstat (limited to 'sound/pci/hda')
| -rw-r--r-- | sound/pci/hda/Makefile | 2 | ||||
| -rw-r--r-- | sound/pci/hda/hda_beep.c | 134 | ||||
| -rw-r--r-- | sound/pci/hda/hda_beep.h | 44 | ||||
| -rw-r--r-- | sound/pci/hda/hda_codec.c | 113 | ||||
| -rw-r--r-- | sound/pci/hda/hda_codec.h | 93 | ||||
| -rw-r--r-- | sound/pci/hda/hda_generic.c | 3 | ||||
| -rw-r--r-- | sound/pci/hda/hda_intel.c | 146 | ||||
| -rw-r--r-- | sound/pci/hda/hda_local.h | 24 | ||||
| -rw-r--r-- | sound/pci/hda/hda_patch.h | 2 | ||||
| -rw-r--r-- | sound/pci/hda/hda_proc.c | 25 | ||||
| -rw-r--r-- | sound/pci/hda/patch_analog.c | 96 | ||||
| -rw-r--r-- | sound/pci/hda/patch_atihdmi.c | 45 | ||||
| -rw-r--r-- | sound/pci/hda/patch_nvhdmi.c | 164 | ||||
| -rw-r--r-- | sound/pci/hda/patch_realtek.c | 1900 | ||||
| -rw-r--r-- | sound/pci/hda/patch_sigmatel.c | 1010 | ||||
| -rw-r--r-- | sound/pci/hda/patch_via.c | 1407 |
16 files changed, 4730 insertions, 478 deletions
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index ab0c726d648..1980c6d207e 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -5,6 +5,7 @@ snd-hda-intel-y := hda_intel.o snd-hda-intel-y += hda_codec.o snd-hda-intel-$(CONFIG_PROC_FS) += hda_proc.o snd-hda-intel-$(CONFIG_SND_HDA_HWDEP) += hda_hwdep.o +snd-hda-intel-$(CONFIG_SND_HDA_INPUT_BEEP) += hda_beep.o snd-hda-intel-$(CONFIG_SND_HDA_GENERIC) += hda_generic.o snd-hda-intel-$(CONFIG_SND_HDA_CODEC_REALTEK) += patch_realtek.o snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CMEDIA) += patch_cmedia.o @@ -14,5 +15,6 @@ snd-hda-intel-$(CONFIG_SND_HDA_CODEC_SI3054) += patch_si3054.o snd-hda-intel-$(CONFIG_SND_HDA_CODEC_ATIHDMI) += patch_atihdmi.o snd-hda-intel-$(CONFIG_SND_HDA_CODEC_CONEXANT) += patch_conexant.o snd-hda-intel-$(CONFIG_SND_HDA_CODEC_VIA) += patch_via.o +snd-hda-intel-$(CONFIG_SND_HDA_CODEC_NVHDMI) += patch_nvhdmi.o obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-intel.o diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c new file mode 100644 index 00000000000..9b77b3e0fa9 --- /dev/null +++ b/sound/pci/hda/hda_beep.c @@ -0,0 +1,134 @@ +/* + * Digital Beep Input Interface for HD-audio codec + * + * Author: Matthew Ranostay <mranostay@embeddedalley.com> + * Copyright (c) 2008 Embedded Alley Solutions Inc + * + * 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 + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/input.h> +#include <linux/pci.h> +#include <linux/workqueue.h> +#include <sound/core.h> +#include "hda_beep.h" + +enum { + DIGBEEP_HZ_STEP = 46875, /* 46.875 Hz */ + DIGBEEP_HZ_MIN = 93750, /* 93.750 Hz */ + DIGBEEP_HZ_MAX = 12000000, /* 12 KHz */ +}; + +static void snd_hda_generate_beep(struct work_struct *work) +{ + struct hda_beep *beep = + container_of(work, struct hda_beep, beep_work); + struct hda_codec *codec = beep->codec; + + /* generate tone */ + snd_hda_codec_write_cache(codec, beep->nid, 0, + AC_VERB_SET_BEEP_CONTROL, beep->tone); +} + +static int snd_hda_beep_event(struct input_dev *dev, unsigned int type, + unsigned int code, int hz) +{ + struct hda_beep *beep = input_get_drvdata(dev); + + switch (code) { + case SND_BELL: + if (hz) + hz = 1000; + case SND_TONE: + hz *= 1000; /* fixed point */ + hz = hz - DIGBEEP_HZ_MIN; + if (hz < 0) + hz = 0; /* turn off PC beep*/ + else if (hz >= (DIGBEEP_HZ_MAX - DIGBEEP_HZ_MIN)) + hz = 0xff; + else { + hz /= DIGBEEP_HZ_STEP; + hz++; + } + break; + default: + return -1; + } + beep->tone = hz; + + /* schedule beep event */ + schedule_work(&beep->beep_work); + return 0; +} + +int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) +{ + struct input_dev *input_dev; + struct hda_beep *beep; + int err; + + beep = kzalloc(sizeof(*beep), GFP_KERNEL); + if (beep == NULL) + return -ENOMEM; + snprintf(beep->phys, sizeof(beep->phys), + "card%d/codec#%d/beep0", codec->bus->card->number, codec->addr); + input_dev = input_allocate_device(); + + /* setup digital beep device */ + input_dev->name = "HDA Digital PCBeep"; + input_dev->phys = beep->phys; + input_dev->id.bustype = BUS_PCI; + + input_dev->id.vendor = codec->vendor_id >> 16; + input_dev->id.product = codec->vendor_id & 0xffff; + input_dev->id.version = 0x01; + + input_dev->evbit[0] = BIT_MASK(EV_SND); + input_dev->sndbit[0] = BIT_MASK(SND_BELL) | BIT_MASK(SND_TONE); + input_dev->event = snd_hda_beep_event; + input_dev->dev.parent = &codec->bus->pci->dev; + input_set_drvdata(input_dev, beep); + + err = input_register_device(input_dev); + if (err < 0) { + input_free_device(input_dev); + kfree(beep); + return err; + } + + /* enable linear scale */ + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_DIGI_CONVERT_2, 0x01); + + beep->nid = nid; + beep->dev = input_dev; + beep->codec = codec; + codec->beep = beep; + + INIT_WORK(&beep->beep_work, &snd_hda_generate_beep); + return 0; +} + +void snd_hda_detach_beep_device(struct hda_codec *codec) +{ + struct hda_beep *beep = codec->beep; + if (beep) { + cancel_work_sync(&beep->beep_work); + flush_scheduled_work(); + + input_unregister_device(beep->dev); + kfree(beep); + } +} diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h new file mode 100644 index 00000000000..de4036e6e71 --- /dev/null +++ b/sound/pci/hda/hda_beep.h @@ -0,0 +1,44 @@ +/* + * Digital Beep Input Interface for HD-audio codec + * + * Author: Matthew Ranostay <mranostay@embeddedalley.com> + * Copyright (c) 2008 Embedded Alley Solutions Inc + * + * 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 + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef __SOUND_HDA_BEEP_H +#define __SOUND_HDA_BEEP_H + +#include "hda_codec.h" + +/* beep information */ +struct hda_beep { + struct input_dev *dev; + struct hda_codec *codec; + char phys[32]; + int tone; + int nid; + struct work_struct beep_work; /* scheduled task for beep event */ +}; + +#ifdef CONFIG_SND_HDA_INPUT_BEEP +int snd_hda_attach_beep_device(struct hda_codec *codec, int nid); +void snd_hda_detach_beep_device(struct hda_codec *codec); +#else +#define snd_hda_attach_beep_device(...) +#define snd_hda_detach_beep_device(...) +#endif +#endif diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d2e1093f8e9..6447754ae56 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -94,6 +94,9 @@ static const struct hda_codec_preset *hda_preset_tables[] = { #ifdef CONFIG_SND_HDA_CODEC_VIA snd_hda_preset_via, #endif +#ifdef CONFIG_SND_HDA_CODEC_NVHDMI + snd_hda_preset_nvhdmi, +#endif NULL }; @@ -211,7 +214,8 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, unsigned int shift, num_elems, mask; hda_nid_t prev_nid; - snd_assert(conn_list && max_conns > 0, return -EINVAL); + if (snd_BUG_ON(!conn_list || max_conns <= 0)) + return -EINVAL; parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN); if (parm & AC_CLIST_LONG) { @@ -313,7 +317,7 @@ int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex) } /* - * process queueud unsolicited events + * process queued unsolicited events */ static void process_unsol_events(struct work_struct *work) { @@ -407,8 +411,10 @@ int __devinit snd_hda_bus_new(struct snd_card *card, .dev_free = snd_hda_bus_dev_free, }; - snd_assert(temp, return -EINVAL); - snd_assert(temp->ops.command && temp->ops.get_response, return -EINVAL); + if (snd_BUG_ON(!temp)) + return -EINVAL; + if (snd_BUG_ON(!temp->ops.command || !temp->ops.get_response)) + return -EINVAL; if (busp) *busp = NULL; @@ -585,11 +591,13 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, struct hda_codec **codecp) { struct hda_codec *codec; - char component[13]; + char component[31]; int err; - snd_assert(bus, return -EINVAL); - snd_assert(codec_addr <= HDA_MAX_CODEC_ADDRESS, return -EINVAL); + if (snd_BUG_ON(!bus)) + return -EINVAL; + if (snd_BUG_ON(codec_addr > HDA_MAX_CODEC_ADDRESS)) + return -EINVAL; if (bus->caddr_tbl[codec_addr]) { snd_printk(KERN_ERR "hda_codec: " @@ -688,7 +696,7 @@ int __devinit snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr, snd_hda_create_hwdep(codec); #endif - sprintf(component, "HDA:%08x", codec->vendor_id); + sprintf(component, "HDA:%08x,%08x,%08x", codec->vendor_id, codec->subsystem_id, codec->revision_id); snd_component_add(codec->bus->card, component); if (codecp) @@ -956,15 +964,6 @@ void snd_hda_codec_resume_amp(struct hda_codec *codec) } #endif /* SND_HDA_NEEDS_RESUME */ -/* - * AMP control callbacks - */ -/* retrieve parameters from private_value */ -#define get_amp_nid(kc) ((kc)->private_value & 0xffff) -#define get_amp_channels(kc) (((kc)->private_value >> 16) & 0x3) -#define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1) -#define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf) - /* volume */ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -1430,6 +1429,29 @@ static unsigned int convert_to_spdif_status(unsigned short val) return sbits; } +/* set digital convert verbs both for the given NID and its slaves */ +static void set_dig_out(struct hda_codec *codec, hda_nid_t nid, + int verb, int val) +{ + hda_nid_t *d; + + snd_hda_codec_write(codec, nid, 0, verb, val); + d = codec->slave_dig_outs; + if (!d) + return; + for (; *d; d++) + snd_hda_codec_write(codec, *d, 0, verb, val); +} + +static inline void set_dig_out_convert(struct hda_codec *codec, hda_nid_t nid, + int dig1, int dig2) +{ + if (dig1 != -1) + set_dig_out(codec, nid, AC_VERB_SET_DIGI_CONVERT_1, dig1); + if (dig2 != -1) + set_dig_out(codec, nid, AC_VERB_SET_DIGI_CONVERT_2, dig2); +} + static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1448,14 +1470,8 @@ static int snd_hda_spdif_default_put(struct snd_kcontrol *kcontrol, change = codec->spdif_ctls != val; codec->spdif_ctls = val; - if (change) { - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_DIGI_CONVERT_1, - val & 0xff); - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_DIGI_CONVERT_2, - val >> 8); - } + if (change) + set_dig_out_convert(codec, nid, val & 0xff, (val >> 8) & 0xff); mutex_unlock(&codec->spdif_mutex); return change; @@ -1487,9 +1503,7 @@ static int snd_hda_spdif_out_switch_put(struct snd_kcontrol *kcontrol, change = codec->spdif_ctls != val; if (change) { codec->spdif_ctls = val; - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_DIGI_CONVERT_1, - val & 0xff); + set_dig_out_convert(codec, nid, val & 0xff, -1); /* unmute amp switch (if any) */ if ((get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) && (val & AC_DIG1_ENABLE)) @@ -2236,11 +2250,13 @@ static int __devinit set_pcm_default_values(struct hda_codec *codec, if (info->ops.close == NULL) info->ops.close = hda_pcm_default_open_close; if (info->ops.prepare == NULL) { - snd_assert(info->nid, return -EINVAL); + if (snd_BUG_ON(!info->nid)) + return -EINVAL; info->ops.prepare = hda_pcm_default_prepare; } if (info->ops.cleanup == NULL) { - snd_assert(info->nid, return -EINVAL); + if (snd_BUG_ON(!info->nid)) + return -EINVAL; info->ops.cleanup = hda_pcm_default_cleanup; } return 0; @@ -2583,14 +2599,31 @@ static void setup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid, unsigned int stream_tag, unsigned int format) { /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ - if (codec->spdif_ctls & AC_DIG1_ENABLE) - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, - codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); + if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) + set_dig_out_convert(codec, nid, + codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff, + -1); snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); + if (codec->slave_dig_outs) { + hda_nid_t *d; + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_setup_stream(codec, *d, stream_tag, 0, + format); + } /* turn on again (if needed) */ - if (codec->spdif_ctls & AC_DIG1_ENABLE) - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, - codec->spdif_ctls & 0xff); + if (codec->spdif_status_reset && (codec->spdif_ctls & AC_DIG1_ENABLE)) + set_dig_out_convert(codec, nid, + codec->spdif_ctls & 0xff, -1); +} + +static void cleanup_dig_out_stream(struct hda_codec *codec, hda_nid_t nid) +{ + snd_hda_codec_cleanup_stream(codec, nid); + if (codec->slave_dig_outs) { + hda_nid_t *d; + for (d = codec->slave_dig_outs; *d; d++) + snd_hda_codec_cleanup_stream(codec, *d); + } } /* @@ -2602,7 +2635,7 @@ int snd_hda_multi_out_dig_open(struct hda_codec *codec, mutex_lock(&codec->spdif_mutex); if (mout->dig_out_used == HDA_DIG_ANALOG_DUP) /* already opened as analog dup; reset it once */ - snd_hda_codec_cleanup_stream(codec, mout->dig_out_nid); + cleanup_dig_out_stream(codec, mout->dig_out_nid); mout->dig_out_used = HDA_DIG_EXCLUSIVE; mutex_unlock(&codec->spdif_mutex); return 0; @@ -2697,7 +2730,7 @@ int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, stream_tag, format); } else { mout->dig_out_used = 0; - snd_hda_codec_cleanup_stream(codec, mout->dig_out_nid); + cleanup_dig_out_stream(codec, mout->dig_out_nid); } } mutex_unlock(&codec->spdif_mutex); @@ -2748,7 +2781,7 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, mout->extra_out_nid[i]); mutex_lock(&codec->spdif_mutex); if (mout->dig_out_nid && mout->dig_out_used == HDA_DIG_ANALOG_DUP) { - snd_hda_codec_cleanup_stream(codec, mout->dig_out_nid); + cleanup_dig_out_stream(codec, mout->dig_out_nid); mout->dig_out_used = 0; } mutex_unlock(&codec->spdif_mutex); @@ -2756,7 +2789,7 @@ int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, } /* - * Helper for automatic ping configuration + * Helper for automatic pin configuration */ static int is_in_nid_list(hda_nid_t nid, hda_nid_t *list) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index efc682888b3..60468f56240 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -90,6 +90,14 @@ enum { #define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c /* f20: AFG/MFG */ #define AC_VERB_GET_SUBSYSTEM_ID 0x0f20 +#define AC_VERB_GET_CVT_CHAN_COUNT 0x0f2d +#define AC_VERB_GET_HDMI_DIP_SIZE 0x0f2e +#define AC_VERB_GET_HDMI_ELDD 0x0f2f +#define AC_VERB_GET_HDMI_DIP_INDEX 0x0f30 +#define AC_VERB_GET_HDMI_DIP_DATA 0x0f31 +#define AC_VERB_GET_HDMI_DIP_XMIT 0x0f32 +#define AC_VERB_GET_HDMI_CP_CTRL 0x0f33 +#define AC_VERB_GET_HDMI_CHAN_SLOT 0x0f34 /* * SET verbs @@ -121,7 +129,14 @@ enum { #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_3 0x71f +#define AC_VERB_SET_EAPD 0x788 #define AC_VERB_SET_CODEC_RESET 0x7ff +#define AC_VERB_SET_CVT_CHAN_COUNT 0x72d +#define AC_VERB_SET_HDMI_DIP_INDEX 0x730 +#define AC_VERB_SET_HDMI_DIP_DATA 0x731 +#define AC_VERB_SET_HDMI_DIP_XMIT 0x732 +#define AC_VERB_SET_HDMI_CP_CTRL 0x733 +#define AC_VERB_SET_HDMI_CHAN_SLOT 0x734 /* * Parameter IDs @@ -143,6 +158,7 @@ enum { #define AC_PAR_GPIO_CAP 0x11 #define AC_PAR_AMP_OUT_CAP 0x12 #define AC_PAR_VOL_KNB_CAP 0x13 +#define AC_PAR_HDMI_LPCM_CAP 0x20 /* * AC_VERB_PARAMETERS results (32bit) @@ -171,6 +187,8 @@ enum { #define AC_WCAP_DIGITAL (1<<9) /* digital I/O */ #define AC_WCAP_POWER (1<<10) /* power control */ #define AC_WCAP_LR_SWAP (1<<11) /* L/R swap */ +#define AC_WCAP_CP_CAPS (1<<12) /* content protection */ +#define AC_WCAP_CHAN_CNT_EXT (7<<13) /* channel count ext */ #define AC_WCAP_DELAY (0xf<<16) #define AC_WCAP_DELAY_SHIFT 16 #define AC_WCAP_TYPE (0xf<<20) @@ -206,9 +224,20 @@ enum { /* Input converter SDI select */ #define AC_SDI_SELECT (0xf<<0) -/* Unsolicited response */ +/* Unsolicited response control */ #define AC_UNSOL_TAG (0x3f<<0) #define AC_UNSOL_ENABLED (1<<7) +#define AC_USRSP_EN AC_UNSOL_ENABLED + +/* Unsolicited responses */ +#define AC_UNSOL_RES_TAG (0x3f<<26) +#define AC_UNSOL_RES_TAG_SHIFT 26 +#define AC_UNSOL_RES_SUBTAG (0x1f<<21) +#define AC_UNSOL_RES_SUBTAG_SHIFT 21 +#define AC_UNSOL_RES_ELDV (1<<1) /* ELD Data valid (for HDMI) */ +#define AC_UNSOL_RES_PD (1<<0) /* pinsense detect */ +#define AC_UNSOL_RES_CP_STATE (1<<1) /* content protection */ +#define AC_UNSOL_RES_CP_READY (1<<0) /* content protection */ /* Pin widget capabilies */ #define AC_PINCAP_IMP_SENSE (1<<0) /* impedance sense capable */ @@ -222,6 +251,10 @@ enum { * but is marked reserved in the Intel HDA specification. */ #define AC_PINCAP_LR_SWAP (1<<7) /* L/R swap */ +/* Note: The same bit as LR_SWAP is newly defined as HDMI capability + * in HD-audio specification + */ +#define AC_PINCAP_HDMI (1<<7) /* HDMI pin */ #define AC_PINCAP_VREF (0x37<<8) #define AC_PINCAP_VREF_SHIFT 8 #define AC_PINCAP_EAPD (1<<16) /* EAPD capable */ @@ -272,6 +305,22 @@ enum { #define AC_KNBCAP_NUM_STEPS (0x7f<<0) #define AC_KNBCAP_DELTA (1<<7) +/* HDMI LPCM capabilities */ +#define AC_LPCMCAP_48K_CP_CHNS (0x0f<<0) /* max channels w/ CP-on */ +#define AC_LPCMCAP_48K_NO_CHNS (0x0f<<4) /* max channels w/o CP-on */ +#define AC_LPCMCAP_48K_20BIT (1<<8) /* 20b bitrate supported */ +#define AC_LPCMCAP_48K_24BIT (1<<9) /* 24b bitrate supported */ +#define AC_LPCMCAP_96K_CP_CHNS (0x0f<<10) /* max channels w/ CP-on */ +#define AC_LPCMCAP_96K_NO_CHNS (0x0f<<14) /* max channels w/o CP-on */ +#define AC_LPCMCAP_96K_20BIT (1<<18) /* 20b bitrate supported */ +#define AC_LPCMCAP_96K_24BIT (1<<19) /* 24b bitrate supported */ +#define AC_LPCMCAP_192K_CP_CHNS (0x0f<<20) /* max channels w/ CP-on */ +#define AC_LPCMCAP_192K_NO_CHNS (0x0f<<24) /* max channels w/o CP-on */ +#define AC_LPCMCAP_192K_20BIT (1<<28) /* 20b bitrate supported */ +#define AC_LPCMCAP_192K_24BIT (1<<29) /* 24b bitrate supported */ +#define AC_LPCMCAP_44K (1<<30) /* 44.1kHz support */ +#define AC_LPCMCAP_44K_MS (1<<31) /* 44.1kHz-multiplies support */ + /* * Control Parameters */ @@ -317,18 +366,44 @@ enum { #define AC_PINCTL_OUT_EN (1<<6) #define AC_PINCTL_HP_EN (1<<7) -/* Unsolicited response - 8bit */ -#define AC_USRSP_EN (1<<7) - /* Pin sense - 32bit */ #define AC_PINSENSE_IMPEDANCE_MASK (0x7fffffff) #define AC_PINSENSE_PRESENCE (1<<31) +#define AC_PINSENSE_ELDV (1<<30) /* ELD valid (HDMI) */ /* EAPD/BTL enable - 32bit */ #define AC_EAPDBTL_BALANCED (1<<0) #define AC_EAPDBTL_EAPD (1<<1) #define AC_EAPDBTL_LR_SWAP (1<<2) +/* HDMI ELD data */ +#define AC_ELDD_ELD_VALID (1<<31) +#define AC_ELDD_ELD_DATA 0xff + +/* HDMI DIP size */ +#define AC_DIPSIZE_ELD_BUF (1<<3) /* ELD buf size of packet size */ +#define AC_DIPSIZE_PACK_IDX (0x07<<0) /* packet index */ + +/* HDMI DIP index */ +#define AC_DIPIDX_PACK_IDX (0x07<<5) /* packet idnex */ +#define AC_DIPIDX_BYTE_IDX (0x1f<<0) /* byte index */ + +/* HDMI DIP xmit (transmit) control */ +#define AC_DIPXMIT_MASK (0x3<<6) +#define AC_DIPXMIT_DISABLE (0x0<<6) /* disable xmit */ +#define AC_DIPXMIT_ONCE (0x2<<6) /* xmit once then disable */ +#define AC_DIPXMIT_BEST (0x3<<6) /* best effort */ + +/* HDMI content protection (CP) control */ +#define AC_CPCTRL_CES (1<<9) /* current encryption state */ +#define AC_CPCTRL_READY (1<<8) /* ready bit */ +#define AC_CPCTRL_SUBTAG (0x1f<<3) /* subtag for unsol-resp */ +#define AC_CPCTRL_STATE (3<<0) /* current CP request state */ + +/* Converter channel <-> HDMI slot mapping */ +#define AC_CVTMAP_HDMI_SLOT (0xf<<0) /* HDMI slot number */ +#define AC_CVTMAP_CHAN (0xf<<4) /* converter channel number */ + /* configuration default - 32bit */ #define AC_DEFCFG_SEQUENCE (0xf<<0) #define AC_DEFCFG_DEF_ASSOC (0xf<<4) @@ -449,6 +524,7 @@ enum { */ struct hda_bus; +struct hda_beep; struct hda_codec; struct hda_pcm; struct hda_pcm_stream; @@ -634,6 +710,9 @@ struct hda_codec { /* codec specific info */ void *spec; + /* beep device */ + struct hda_beep *beep; + /* widget capabilities cache */ unsigned int num_nodes; hda_nid_t start_nid; @@ -646,9 +725,15 @@ struct hda_codec { unsigned int spdif_status; /* IEC958 status bits */ unsigned short spdif_ctls; /* SPDIF control bits */ unsigned int spdif_in_enable; /* SPDIF input enable? */ + hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ struct snd_hwdep *hwdep; /* assigned hwdep device */ + /* misc flags */ + unsigned int spdif_status_reset :1; /* needs to toggle SPDIF for each + * status change + * (e.g. Realtek codecs) + */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ unsigned int power_transition :1; /* power-state in transition */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 59e4389c94a..0ca30894f7c 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -174,7 +174,8 @@ static int build_afg_tree(struct hda_codec *codec) int i, nodes, err; hda_nid_t nid; - snd_assert(spec, return -EINVAL); + if (snd_BUG_ON(!spec)) + return -EINVAL; spec->def_amp_out_caps = snd_hda_param_read(codec, codec->afg, AC_PAR_AMP_OUT_CAP); spec->def_amp_in_caps = snd_hda_param_read(codec, codec->afg, AC_PAR_AMP_IN_CAP); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1c53e337ecb..9f316c1b279 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -222,9 +222,9 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define RIRB_INT_OVERRUN 0x04 #define RIRB_INT_MASK 0x05 -/* STATESTS int mask: SD2,SD1,SD0 */ -#define AZX_MAX_CODECS 3 -#define STATESTS_INT_MASK 0x07 +/* STATESTS int mask: S3,SD2,SD1,SD0 */ +#define AZX_MAX_CODECS 4 +#define STATESTS_INT_MASK 0x0f /* SD_CTL bits */ #define SD_CTL_STREAM_RESET 0x01 /* stream reset bit */ @@ -286,6 +286,11 @@ enum { #define INTEL_SCH_HDA_DEVC 0x78 #define INTEL_SCH_HDA_DEVC_NOSNOOP (0x1<<11) +/* Define IN stream 0 FIFO size offset in VIA controller */ +#define VIA_IN_STREAM0_FIFO_SIZE_OFFSET 0x90 +/* Define VIA HD Audio Device ID*/ +#define VIA_HDAC_DEVICE_ID 0x3288 + /* */ @@ -317,6 +322,12 @@ struct azx_dev { unsigned int running :1; unsigned int irq_pending :1; unsigned int irq_ignore :1; + /* + * For VIA: + * A flag to ensure DMA position is 0 + * when link position is not greater than FIFO size + */ + unsigned int insufficient :1; }; /* CORB/RIRB */ @@ -379,6 +390,7 @@ struct azx { unsigned int polling_mode :1; unsigned int msi :1; unsigned int irq_pending_warned :1; + unsigned int via_dmapos_patch :1; /* enable DMA-position fix for VIA */ /* for debugging */ unsigned int last_cmd; /* last issued command (to sync) */ @@ -398,6 +410,7 @@ enum { AZX_DRIVER_ULI, AZX_DRIVER_NVIDIA, AZX_DRIVER_TERA, + AZX_NUM_DRIVERS, /* keep this as last entry */ }; static char *driver_short_names[] __devinitdata = { @@ -818,6 +831,11 @@ static void azx_int_clear(struct azx *chip) /* start a stream */ static void azx_stream_start(struct azx *chip, struct azx_dev *azx_dev) { + /* + * Before stream start, initialize parameter + */ + azx_dev->insufficient = 1; + /* enable SIE */ azx_writeb(chip, INTCTL, azx_readb(chip, INTCTL) | (1 << azx_dev->index)); @@ -998,7 +1016,6 @@ static int setup_bdle(struct snd_pcm_substream *substream, struct azx_dev *azx_dev, u32 **bdlp, int ofs, int size, int with_ioc) { - struct snd_sg_buf *sgbuf = snd_pcm_substream_sgbuf(substream); u32 *bdl = *bdlp; while (size > 0) { @@ -1008,14 +1025,12 @@ static int setup_bdle(struct snd_pcm_substream *substream, if (azx_dev->frags >= AZX_MAX_BDL_ENTRIES) return -EINVAL; - addr = snd_pcm_sgbuf_get_addr(sgbuf, ofs); + addr = snd_pcm_sgbuf_get_addr(substream, ofs); /* program the address field of the BDL entry */ bdl[0] = cpu_to_le32((u32)addr); bdl[1] = cpu_to_le32(upper_32_bits(addr)); /* program the size field of the BDL entry */ - chunk = PAGE_SIZE - (ofs % PAGE_SIZE); - if (size < chunk) - chunk = size; + chunk = snd_pcm_sgbuf_get_chunk_size(substream, ofs, size); bdl[2] = cpu_to_le32(chunk); /* program the IOC to enable interrupt * only when the whole fragment is processed @@ -1151,7 +1166,8 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) /* enable the position buffer */ if (chip->position_fix == POS_FIX_POSBUF || - chip->position_fix == POS_FIX_AUTO) { + chip->position_fix == POS_FIX_AUTO || + chip->via_dmapos_patch) { if (!(azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE)) azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr | ICH6_DPLBASE_ENABLE); @@ -1169,23 +1185,26 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) * Codec initialization */ -static unsigned int azx_max_codecs[] __devinitdata = { - [AZX_DRIVER_ICH] = 4, /* Some ICH9 boards use SD3 */ - [AZX_DRIVER_SCH] = 3, - [AZX_DRIVER_ATI] = 4, - [AZX_DRIVER_ATIHDMI] = 4, - [AZX_DRIVER_VIA] = 3, /* FIXME: correct? */ - [AZX_DRIVER_SIS] = 3, /* FIXME: correct? */ - [AZX_DRIVER_ULI] = 3, /* FIXME: correct? */ - [AZX_DRIVER_NVIDIA] = 3, /* FIXME: correct? */ +/* number of codec slots for each chipset: 0 = default slots (i.e. 4) */ +static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] __devinitdata = { [AZX_DRIVER_TERA] = 1, }; +/* number of slots to probe as default + * this can be different from azx_max_codecs[] -- e.g. some boards + * report wrongly the non-existing 4th slot availability + */ +static unsigned int azx_default_codecs[AZX_NUM_DRIVERS] __devinitdata = { + [AZX_DRIVER_ICH] = 3, + [AZX_DRIVER_ATI] = 3, +}; + static int __devinit azx_codec_create(struct azx *chip, const char *model, unsigned int codec_probe_mask) { struct hda_bus_template bus_temp; int c, codecs, audio_codecs, err; + int def_slots, max_slots; memset(&bus_temp, 0, sizeof(bus_temp)); bus_temp.private_data = chip; @@ -1201,8 +1220,17 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, if (err < 0) return err; + if (chip->driver_type == AZX_DRIVER_NVIDIA) + chip->bus->needs_damn_long_delay = 1; + codecs = audio_codecs = 0; - for (c = 0; c < AZX_MAX_CODECS; c++) { + max_slots = azx_max_codecs[chip->driver_type]; + if (!max_slots) + max_slots = AZX_MAX_CODECS; + def_slots = azx_default_codecs[chip->driver_type]; + if (!def_slots) + def_slots = max_slots; + for (c = 0; c < def_slots; c++) { if ((chip->codec_mask & (1 << c)) & codec_probe_mask) { struct hda_codec *codec; err = snd_hda_codec_new(chip->bus, c, &codec); @@ -1215,7 +1243,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, } if (!audio_codecs) { /* probe additional slots if no codec is found */ - for (; c < azx_max_codecs[chip->driver_type]; c++) { + for (; c < max_slots; c++) { if ((chip->codec_mask & (1 << c)) & codec_probe_mask) { err = snd_hda_codec_new(chip->bus, c, NULL); if (err < 0) @@ -1507,13 +1535,71 @@ static int azx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } +/* get the current DMA position with correction on VIA chips */ +static unsigned int azx_via_get_position(struct azx *chip, + struct azx_dev *azx_dev) +{ + unsigned int link_pos, mini_pos, bound_pos; + unsigned int mod_link_pos, mod_dma_pos, mod_mini_pos; + unsigned int fifo_size; + + link_pos = azx_sd_readl(azx_dev, SD_LPIB); + if (azx_dev->index >= 4) { + /* Playback, no problem using link position */ + return link_pos; + } + + /* Capture */ + /* For new chipset, + * use mod to get the DMA position just like old chipset + */ + mod_dma_pos = le32_to_cpu(*azx_dev->posbuf); + mod_dma_pos %= azx_dev->period_bytes; + + /* azx_dev->fifo_size can't get FIFO size of in stream. + * Get from base address + offset. + */ + fifo_size = readw(chip->remap_addr + VIA_IN_STREAM0_FIFO_SIZE_OFFSET); + + if (azx_dev->insufficient) { + /* Link position never gather than FIFO size */ + if (link_pos <= fifo_size) + return 0; + + azx_dev->insufficient = 0; + } + + if (link_pos <= fifo_size) + mini_pos = azx_dev->bufsize + link_pos - fifo_size; + else + mini_pos = link_pos - fifo_size; + + /* Find nearest previous boudary */ + mod_mini_pos = mini_pos % azx_dev->period_bytes; + mod_link_pos = link_pos % azx_dev->period_bytes; + if (mod_link_pos >= fifo_size) + bound_pos = link_pos - mod_link_pos; + else if (mod_dma_pos >= mod_mini_pos) + bound_pos = mini_pos - mod_mini_pos; + else { + bound_pos = mini_pos - mod_mini_pos + azx_dev->period_bytes; + if (bound_pos >= azx_dev->bufsize) + bound_pos = 0; + } + + /* Calculate real DMA position we want */ + return bound_pos + mod_dma_pos; +} + static unsigned int azx_get_position(struct azx *chip, struct azx_dev *azx_dev) { unsigned int pos; - if (chip->position_fix == POS_FIX_POSBUF || - chip->position_fix == POS_FIX_AUTO) { + if (chip->via_dmapos_patch) + pos = azx_via_get_position(chip, azx_dev); + else if (chip->position_fix == POS_FIX_POSBUF || + chip->position_fix == POS_FIX_AUTO) { /* use the position buffer */ pos = le32_to_cpu(*azx_dev->posbuf); } else { @@ -1559,6 +1645,8 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev) chip->position_fix = POS_FIX_POSBUF; } + if (!bdl_pos_adj[chip->dev_index]) + return 1; /* no delayed ack */ if (pos % azx_dev->period_bytes > azx_dev->period_bytes / 2) return 0; /* NG - it's below the period boundary */ return 1; /* OK, it's fine */ @@ -1646,7 +1734,8 @@ static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec, if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams) return 0; - snd_assert(cpcm->name, return -EINVAL); + if (snd_BUG_ON(!cpcm->name)) + return -EINVAL; err = snd_pcm_new(chip->card, cpcm->name, cpcm->device, cpcm->stream[0].substreams, @@ -1670,7 +1759,7 @@ static int __devinit create_codec_pcm(struct azx *chip, struct hda_codec *codec, snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &azx_pcm_ops); snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(chip->pci), - 1024 * 64, 1024 * 1024); + 1024 * 64, 32 * 1024 * 1024); chip->pcm[cpcm->device] = pcm; return 0; } @@ -1946,6 +2035,15 @@ static int __devinit check_position_fix(struct azx *chip, int fix) { const struct snd_pci_quirk *q; + /* Check VIA HD Audio Controller exist */ + if (chip->pci->vendor == PCI_VENDOR_ID_VIA && + chip->pci->device == VIA_HDAC_DEVICE_ID) { + chip->via_dmapos_patch = 1; + /* Use link position directly, avoid any transfer problem. */ + return POS_FIX_LPIB; + } + chip->via_dmapos_patch = 0; + if (fix == POS_FIX_AUTO) { q = snd_pci_quirk_lookup(chip->pci, position_fix_list); if (q) { diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 5c9e578f7f2..7957fefda73 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -368,12 +368,15 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, #define AMP_OUT_UNMUTE 0xb000 #define AMP_OUT_ZERO 0xb000 /* pinctl values */ -#define PIN_IN 0x20 -#define PIN_VREF80 0x24 -#define PIN_VREF50 0x21 -#define PIN_OUT 0x40 -#define PIN_HP 0xc0 -#define PIN_HP_AMP 0x80 +#define PIN_IN (AC_PINCTL_IN_EN) +#define PIN_VREFHIZ (AC_PINCTL_IN_EN | AC_PINCTL_VREF_HIZ) +#define PIN_VREF50 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_50) +#define PIN_VREFGRD (AC_PINCTL_IN_EN | AC_PINCTL_VREF_GRD) +#define PIN_VREF80 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_80) +#define PIN_VREF100 (AC_PINCTL_IN_EN | AC_PINCTL_VREF_100) +#define PIN_OUT (AC_PINCTL_OUT_EN) +#define PIN_HP (AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN) +#define PIN_HP_AMP (AC_PINCTL_HP_EN) /* * get widget capabilities @@ -418,4 +421,13 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, hda_nid_t nid); #endif /* CONFIG_SND_HDA_POWER_SAVE */ +/* + * AMP control callbacks + */ +/* retrieve parameters from private_value */ +#define get_amp_nid(kc) ((kc)->private_value & 0xffff) +#define get_amp_channels(kc) (((kc)->private_value >> 16) & 0x3) +#define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1) +#define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf) + #endif /* __SOUND_HDA_LOCAL_H */ diff --git a/sound/pci/hda/hda_patch.h b/sound/pci/hda/hda_patch.h index 2fdf2358dbc..dfbcfa88da4 100644 --- a/sound/pci/hda/hda_patch.h +++ b/sound/pci/hda/hda_patch.h @@ -18,3 +18,5 @@ extern struct hda_codec_preset snd_hda_preset_atihdmi[]; extern struct hda_codec_preset snd_hda_preset_conexant[]; /* VIA codecs */ extern struct hda_codec_preset snd_hda_preset_via[]; +/* NVIDIA HDMI codecs */ +extern struct hda_codec_preset snd_hda_preset_nvhdmi[]; diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 1e5aff5c48d..743d77922bc 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -216,7 +216,7 @@ static void print_pin_caps(struct snd_info_buffer *buffer, unsigned int caps, val; caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); - snd_iprintf(buffer, " Pincap 0x08%x:", caps); + snd_iprintf(buffer, " Pincap 0x%08x:", caps); if (caps & AC_PINCAP_IN) snd_iprintf(buffer, " IN"); if (caps & AC_PINCAP_OUT) @@ -229,8 +229,13 @@ static void print_pin_caps(struct snd_info_buffer *buffer, snd_iprintf(buffer, " Detect"); if (caps & AC_PINCAP_BALANCE) snd_iprintf(buffer, " Balanced"); - if (caps & AC_PINCAP_LR_SWAP) - snd_iprintf(buffer, " R/L"); + if (caps & AC_PINCAP_HDMI) { + /* Realtek uses this bit as a different meaning */ + if ((codec->vendor_id >> 16) == 0x10ec) + snd_iprintf(buffer, " R/L"); + else + snd_iprintf(buffer, " HDMI"); + } if (caps & AC_PINCAP_TRIG_REQ) snd_iprintf(buffer, " Trigger"); if (caps & AC_PINCAP_IMP_SENSE) @@ -552,9 +557,15 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, "Node 0x%02x [%s] wcaps 0x%x:", nid, get_wid_type_name(wid_type), wid_caps); - if (wid_caps & AC_WCAP_STEREO) - snd_iprintf(buffer, " Stereo"); - else + if (wid_caps & AC_WCAP_STEREO) { + unsigned int chans; + chans = (wid_caps & AC_WCAP_CHAN_CNT_EXT) >> 13; + chans = ((chans << 1) | 1) + 1; + if (chans == 2) + snd_iprintf(buffer, " Stereo"); + else + snd_iprintf(buffer, " %d-Channels", chans); + } else snd_iprintf(buffer, " Mono"); if (wid_caps & AC_WCAP_DIGITAL) snd_iprintf(buffer, " Digital"); @@ -566,6 +577,8 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, " Stripe"); if (wid_caps & AC_WCAP_LR_SWAP) snd_iprintf(buffer, " R/L"); + if (wid_caps & AC_WCAP_CP_CAPS) + snd_iprintf(buffer, " CP"); snd_iprintf(buffer, "\n"); /* volume knob is a special widget that always have connection diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index e8003d99f0b..2b00c4afdf9 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1826,9 +1826,14 @@ static hda_nid_t ad1988_capsrc_nids[3] = { 0x0c, 0x0d, 0x0e }; -#define AD1988_SPDIF_OUT 0x02 +#define AD1988_SPDIF_OUT 0x02 +#define AD1988_SPDIF_OUT_HDMI 0x0b #define AD1988_SPDIF_IN 0x07 +static hda_nid_t ad1989b_slave_dig_outs[2] = { + AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI +}; + static struct hda_input_mux ad1988_6stack_capture_source = { .num_items = 5, .items = { @@ -2143,6 +2148,7 @@ static struct snd_kcontrol_new ad1988_spdif_in_mixers[] = { static struct snd_kcontrol_new ad1989_spdif_out_mixers[] = { HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("HDMI Playback Volume", 0x1d, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -2207,6 +2213,8 @@ static struct hda_verb ad1988_6stack_init_verbs[] = { {0x34, AC_VERB_SET_CONNECT_SEL, 0x0}, /* Analog CD Input */ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ { } }; @@ -2247,8 +2255,12 @@ static struct hda_verb ad1988_spdif_init_verbs[] = { /* AD1989 has no ADC -> SPDIF route */ static struct hda_verb ad1989_spdif_init_verbs[] = { - /* SPDIF out pin */ + /* SPDIF-1 out pin */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ + /* SPDIF-2/HDMI out pin */ + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ { } }; @@ -2336,6 +2348,8 @@ static struct hda_verb ad1988_3stack_init_verbs[] = { {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ { } }; @@ -2409,6 +2423,8 @@ static struct hda_verb ad1988_laptop_init_verbs[] = { {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* Analog Mix output amp */ + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ { } }; @@ -2868,6 +2884,7 @@ static struct snd_pci_quirk ad1988_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG), SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG), + SND_PCI_QUIRK(0x1043, 0x8311, "Asus P5Q-Premium/Pro", AD1988_6STACK_DIG), {} }; @@ -2975,6 +2992,7 @@ static int patch_ad1988(struct hda_codec *codec) ad1989_spdif_out_mixers; spec->init_verbs[spec->num_init_verbs++] = ad1989_spdif_init_verbs; + codec->slave_dig_outs = ad1989b_slave_dig_outs; } else { spec->mixers[spec->num_mixers++] = ad1988_spdif_out_mixers; @@ -3911,7 +3929,7 @@ static int patch_ad1884a(struct hda_codec *codec) /* - * AD1882 + * AD1882 / AD1882A * * port-A - front hp-out * port-B - front mic-in @@ -3948,6 +3966,18 @@ static struct hda_input_mux ad1882_capture_source = { }, }; +/* list: 0x11, 0x39, 0x3a, 0x3c, 0x18, 0x1f, 0x12, 0x20 */ +static struct hda_input_mux ad1882a_capture_source = { + .num_items = 5, + .items = { + { "Front Mic", 0x1 }, + { "Mic", 0x4}, + { "Line", 0x2 }, + { "Digital Mic", 0x06 }, + { "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), @@ -3957,16 +3987,7 @@ static struct snd_kcontrol_new ad1882_base_mixers[] = { 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), @@ -3999,6 +4020,35 @@ static struct snd_kcontrol_new ad1882_base_mixers[] = { { } /* end */ }; +static struct snd_kcontrol_new ad1882_loopback_mixers[] = { + 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), + { } /* end */ +}; + +static struct snd_kcontrol_new ad1882a_loopback_mixers[] = { + 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, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, 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("Digital Mic Boost", 0x1f, 0x0, HDA_INPUT), + { } /* 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), @@ -4168,9 +4218,16 @@ static int patch_ad1882(struct hda_codec *codec) 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; + if (codec->vendor_id == 0x11d1882) + spec->input_mux = &ad1882_capture_source; + else + spec->input_mux = &ad1882a_capture_source; + spec->num_mixers = 2; spec->mixers[0] = ad1882_base_mixers; + if (codec->vendor_id == 0x11d1882) + spec->mixers[1] = ad1882_loopback_mixers; + else + spec->mixers[1] = ad1882a_loopback_mixers; spec->num_init_verbs = 1; spec->init_verbs[0] = ad1882_init_verbs; spec->spdif_route = 0; @@ -4187,8 +4244,8 @@ static int patch_ad1882(struct hda_codec *codec) switch (board_config) { default: case AD1882_3STACK: - spec->num_mixers = 2; - spec->mixers[1] = ad1882_3stack_mixers; + spec->num_mixers = 3; + spec->mixers[2] = ad1882_3stack_mixers; spec->channel_mode = ad1882_modes; spec->num_channel_mode = ARRAY_SIZE(ad1882_modes); spec->need_dac_fix = 1; @@ -4196,8 +4253,8 @@ static int patch_ad1882(struct hda_codec *codec) spec->multiout.num_dacs = 1; break; case AD1882_6STACK: - spec->num_mixers = 2; - spec->mixers[1] = ad1882_6stack_mixers; + spec->num_mixers = 3; + spec->mixers[2] = ad1882_6stack_mixers; break; } return 0; @@ -4220,6 +4277,7 @@ struct hda_codec_preset snd_hda_preset_analog[] = { { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a }, { .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 }, { .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 }, + { .id = 0x11d4882a, .name = "AD1882A", .patch = patch_ad1882 }, { .id = 0x11d4989a, .name = "AD1989A", .patch = patch_ad1988 }, { .id = 0x11d4989b, .name = "AD1989B", .patch = patch_ad1988 }, {} /* terminator */ diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c index 12272508b11..ba61575983f 100644 --- a/sound/pci/hda/patch_atihdmi.c +++ b/sound/pci/hda/patch_atihdmi.c @@ -35,6 +35,9 @@ struct atihdmi_spec { struct hda_pcm pcm_rec; }; +#define CVT_NID 0x02 /* audio converter */ +#define PIN_NID 0x03 /* HDMI output pin */ + static struct hda_verb atihdmi_basic_init[] = { /* enable digital output on pin widget */ { 0x03, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, @@ -60,8 +63,9 @@ static int atihdmi_init(struct hda_codec *codec) { snd_hda_sequence_write(codec, atihdmi_basic_init); /* SI codec requires to unmute the pin */ - if (get_wcaps(codec, 0x03) & AC_WCAP_OUT_AMP) - snd_hda_codec_write(codec, 0x03, 0, AC_VERB_SET_AMP_GAIN_MUTE, + if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, PIN_NID, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); return 0; } @@ -92,15 +96,29 @@ static int atihdmi_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct atihdmi_spec *spec = codec->spec; - return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag, - format, substream); + int chans = substream->runtime->channels; + int i, err; + + err = snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag, + format, substream); + if (err < 0) + return err; + snd_hda_codec_write(codec, CVT_NID, 0, AC_VERB_SET_CVT_CHAN_COUNT, + chans - 1); + /* FIXME: XXX */ + for (i = 0; i < chans; i++) { + snd_hda_codec_write(codec, CVT_NID, 0, + AC_VERB_SET_HDMI_CHAN_SLOT, + (i << 4) | i); + } + return 0; } static struct hda_pcm_stream atihdmi_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, - .nid = 0x2, /* NID to query formats and rates and setup streams */ + .nid = CVT_NID, /* NID to query formats and rates and setup streams */ .ops = { .open = atihdmi_dig_playback_pcm_open, .close = atihdmi_dig_playback_pcm_close, @@ -112,6 +130,7 @@ static int atihdmi_build_pcms(struct hda_codec *codec) { struct atihdmi_spec *spec = codec->spec; struct hda_pcm *info = &spec->pcm_rec; + unsigned int chans; codec->num_pcms = 1; codec->pcm_info = info; @@ -120,6 +139,13 @@ static int atihdmi_build_pcms(struct hda_codec *codec) info->pcm_type = HDA_PCM_TYPE_HDMI; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = atihdmi_pcm_digital_playback; + /* FIXME: we must check ELD and change the PCM parameters dynamically + */ + chans = get_wcaps(codec, CVT_NID); + chans = (chans & AC_WCAP_CHAN_CNT_EXT) >> 13; + chans = ((chans << 1) | 1) + 1; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = chans; + return 0; } @@ -147,9 +173,11 @@ static int patch_atihdmi(struct hda_codec *codec) spec->multiout.num_dacs = 0; /* no analog */ spec->multiout.max_channels = 2; - spec->multiout.dig_out_nid = 0x2; /* NID for copying analog to digital, - * seems to be unused in pure-digital - * case. */ + /* NID for copying analog to digital, + * seems to be unused in pure-digital + * case. + */ + spec->multiout.dig_out_nid = CVT_NID; codec->patch_ops = atihdmi_patch_ops; @@ -164,6 +192,7 @@ struct hda_codec_preset snd_hda_preset_atihdmi[] = { { .id = 0x10027919, .name = "ATI RS600 HDMI", .patch = patch_atihdmi }, { .id = 0x1002791a, .name = "ATI RS690/780 HDMI", .patch = patch_atihdmi }, { .id = 0x1002aa01, .name = "ATI R6xx HDMI", .patch = patch_atihdmi }, + { .id = 0x10951390, .name = "SiI1390 HDMI", .patch = patch_atihdmi }, { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_atihdmi }, { .id = 0x17e80047, .name = "Chrontel HDMI", .patch = patch_atihdmi }, {} /* terminator */ diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c new file mode 100644 index 00000000000..1a65775d28e --- /dev/null +++ b/sound/pci/hda/patch_nvhdmi.c @@ -0,0 +1,164 @@ +/* + * Universal Interface for Intel High Definition Audio Codec + * + * HD audio interface patch for NVIDIA HDMI codecs + * + * Copyright (c) 2008 NVIDIA Corp. All rights reserved. + * Copyright (c) 2008 Wei Ni <wni@nvidia.com> + * + * + * 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 + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This driver is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <sound/core.h> +#include "hda_codec.h" +#include "hda_local.h" + +struct nvhdmi_spec { + struct hda_multi_out multiout; + + struct hda_pcm pcm_rec; +}; + +static struct hda_verb nvhdmi_basic_init[] = { + /* enable digital output on pin widget */ + { 0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + {} /* terminator */ +}; + +/* + * Controls + */ +static int nvhdmi_build_controls(struct hda_codec *codec) +{ + struct nvhdmi_spec *spec = codec->spec; + int err; + + err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); + if (err < 0) + return err; + + return 0; +} + +static int nvhdmi_init(struct hda_codec *codec) +{ + snd_hda_sequence_write(codec, nvhdmi_basic_init); + return 0; +} + +/* + * Digital out + */ +static int nvhdmi_dig_playback_pcm_open(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct nvhdmi_spec *spec = codec->spec; + return snd_hda_multi_out_dig_open(codec, &spec->multiout); +} + +static int nvhdmi_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct nvhdmi_spec *spec = codec->spec; + return snd_hda_multi_out_dig_close(codec, &spec->multiout); +} + +static int nvhdmi_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct nvhdmi_spec *spec = codec->spec; + return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag, + format, substream); +} + +static struct hda_pcm_stream nvhdmi_pcm_digital_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .nid = 0x4, /* NID to query formats and rates and setup streams */ + .rates = SNDRV_PCM_RATE_48000, + .maxbps = 16, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .ops = { + .open = nvhdmi_dig_playback_pcm_open, + .close = nvhdmi_dig_playback_pcm_close, + .prepare = nvhdmi_dig_playback_pcm_prepare + }, +}; + +static int nvhdmi_build_pcms(struct hda_codec *codec) +{ + struct nvhdmi_spec *spec = codec->spec; + struct hda_pcm *info = &spec->pcm_rec; + + codec->num_pcms = 1; + codec->pcm_info = info; + + info->name = "NVIDIA HDMI"; + info->stream[SNDRV_PCM_STREAM_PLAYBACK] = nvhdmi_pcm_digital_playback; + + return 0; +} + +static void nvhdmi_free(struct hda_codec *codec) +{ + kfree(codec->spec); +} + +static struct hda_codec_ops nvhdmi_patch_ops = { + .build_controls = nvhdmi_build_controls, + .build_pcms = nvhdmi_build_pcms, + .init = nvhdmi_init, + .free = nvhdmi_free, +}; + +static int patch_nvhdmi(struct hda_codec *codec) +{ + struct nvhdmi_spec *spec; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + spec->multiout.num_dacs = 0; /* no analog */ + spec->multiout.max_channels = 2; + spec->multiout.dig_out_nid = 0x4; /* NID for copying analog to digital, + * seems to be unused in pure-digital + * case. */ + + codec->patch_ops = nvhdmi_patch_ops; + + return 0; +} + +/* + * patch entries + */ +struct hda_codec_preset snd_hda_preset_nvhdmi[] = { + { .id = 0x10de0002, .name = "NVIDIA MCP78 HDMI", .patch = patch_nvhdmi }, + { .id = 0x10de0007, .name = "NVIDIA MCP7A HDMI", .patch = patch_nvhdmi }, + {} /* terminator */ +}; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 66025161bd6..e72707cb60a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -72,6 +72,7 @@ enum { enum { ALC260_BASIC, ALC260_HP, + ALC260_HP_DC7600, ALC260_HP_3013, ALC260_FUJITSU_S702X, ALC260_ACER, @@ -100,6 +101,9 @@ enum { ALC262_BENQ_T31, ALC262_ULTRA, ALC262_LENOVO_3000, + ALC262_NEC, + ALC262_TOSHIBA_S06, + ALC262_TOSHIBA_RX1, ALC262_AUTO, ALC262_MODEL_LAST /* last tag */ }; @@ -110,6 +114,7 @@ enum { ALC268_3ST, ALC268_TOSHIBA, ALC268_ACER, + ALC268_ACER_ASPIRE_ONE, ALC268_DELL, ALC268_ZEPTO, #ifdef CONFIG_SND_DEBUG @@ -122,6 +127,7 @@ enum { /* ALC269 models */ enum { ALC269_BASIC, + ALC269_QUANTA_FL1, ALC269_ASUS_EEEPC_P703, ALC269_ASUS_EEEPC_P901, ALC269_AUTO, @@ -169,6 +175,13 @@ enum { ALC663_ASUS_G71V, ALC663_ASUS_H13, ALC663_ASUS_G50V, + ALC662_ECS, + ALC663_ASUS_MODE1, + ALC662_ASUS_MODE2, + ALC663_ASUS_MODE3, + ALC663_ASUS_MODE4, + ALC663_ASUS_MODE5, + ALC663_ASUS_MODE6, ALC662_AUTO, ALC662_MODEL_LAST, }; @@ -200,18 +213,21 @@ enum { ALC883_ACER, ALC883_ACER_ASPIRE, ALC883_MEDION, - ALC883_MEDION_MD2, + ALC883_MEDION_MD2, ALC883_LAPTOP_EAPD, ALC883_LENOVO_101E_2ch, ALC883_LENOVO_NB0763, ALC888_LENOVO_MS7195_DIG, - ALC883_HAIER_W66, + ALC888_LENOVO_SKY, + ALC883_HAIER_W66, ALC888_3ST_HP, ALC888_6ST_DELL, ALC883_MITAC, ALC883_CLEVO_M720, ALC883_FUJITSU_PI2515, ALC883_3ST_6ch_INTEL, + ALC888_ASUS_M90V, + ALC888_ASUS_EEE1601, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -398,7 +414,7 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, /* * Control the mode of pin widget settings via the mixer. "pc" is used - * instead of "%" to avoid consequences of accidently treating the % as + * instead of "%" to avoid consequences of accidently treating the % as * being part of a format specifier. Maximum allowed length of a value is * 63 characters plus NULL terminator. * @@ -429,7 +445,7 @@ static unsigned char alc_pin_mode_values[] = { #define ALC_PIN_DIR_IN_NOMICBIAS 0x03 #define ALC_PIN_DIR_INOUT_NOMICBIAS 0x04 -/* Info about the pin modes supported by the different pin direction modes. +/* Info about the pin modes supported by the different pin direction modes. * For each direction the minimum and maximum values are given. */ static signed char alc_pin_mode_dir_info[5][2] = { @@ -502,7 +518,7 @@ static int alc_pin_mode_put(struct snd_kcontrol *kcontrol, AC_VERB_SET_PIN_WIDGET_CONTROL, alc_pin_mode_values[val]); - /* Also enable the retasking pin's input/output as required + /* Also enable the retasking pin's input/output as required * for the requested pin mode. Enum values of 2 or less are * input modes. * @@ -707,7 +723,7 @@ static void setup_preset(struct alc_spec *spec, i++) spec->init_verbs[spec->num_init_verbs++] = preset->init_verbs[i]; - + spec->channel_mode = preset->channel_mode; spec->num_channel_mode = preset->num_channel_mode; spec->need_dac_fix = preset->need_dac_fix; @@ -718,7 +734,7 @@ static void setup_preset(struct alc_spec *spec, spec->multiout.dac_nids = preset->dac_nids; spec->multiout.dig_out_nid = preset->dig_out_nid; spec->multiout.hp_nid = preset->hp_nid; - + spec->num_mux_defs = preset->num_mux_defs; if (!spec->num_mux_defs) spec->num_mux_defs = 1; @@ -806,6 +822,27 @@ static void alc_sku_automute(struct hda_codec *codec) spec->jack_present ? 0 : PIN_OUT); } +static void alc_mic_automute(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + unsigned int present; + unsigned int mic_nid = spec->autocfg.input_pins[AUTO_PIN_MIC]; + unsigned int fmic_nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC]; + unsigned int mix_nid = spec->capsrc_nids[0]; + unsigned int capsrc_idx_mic, capsrc_idx_fmic; + + capsrc_idx_mic = mic_nid - 0x18; + capsrc_idx_fmic = fmic_nid - 0x18; + present = snd_hda_codec_read(codec, mic_nid, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_write(codec, mix_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (capsrc_idx_mic << 8) | (present ? 0 : 0x80)); + snd_hda_codec_write(codec, mix_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (capsrc_idx_fmic << 8) | (present ? 0x80 : 0)); + snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, capsrc_idx_fmic, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + /* unsolicited event for HP jack sensing */ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) { @@ -813,10 +850,17 @@ static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res) res >>= 28; else res >>= 26; - if (res != ALC880_HP_EVENT) - return; + if (res == ALC880_HP_EVENT) + alc_sku_automute(codec); + if (res == ALC880_MIC_EVENT) + alc_mic_automute(codec); +} + +static void alc_inithook(struct hda_codec *codec) +{ alc_sku_automute(codec); + alc_mic_automute(codec); } /* additional initialization for ALC888 variants */ @@ -855,7 +899,7 @@ static void alc_subsystem_id(struct hda_codec *codec, if ((ass != codec->bus->pci->subsystem_device) && (ass & 1)) goto do_sku; - /* + /* * 31~30 : port conetcivity * 29~21 : reserve * 20 : PCBEEP input @@ -946,7 +990,7 @@ do_sku: tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0); snd_hda_codec_write(codec, 0x20, 0, - AC_VERB_SET_COEF_INDEX, 7); + AC_VERB_SET_COEF_INDEX, 7); snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, tmp | 0x2010); @@ -961,7 +1005,7 @@ do_sku: tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0); snd_hda_codec_write(codec, 0x20, 0, - AC_VERB_SET_COEF_INDEX, 7); + AC_VERB_SET_COEF_INDEX, 7); snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, tmp | 0x3000); @@ -970,7 +1014,7 @@ do_sku: default: break; } - + /* is laptop or Desktop and enable the function "Mute internal speaker * when the external headphone out jack is plugged" */ @@ -1002,10 +1046,18 @@ do_sku: else return; } + if (spec->autocfg.hp_pins[0]) + snd_hda_codec_write(codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | ALC880_HP_EVENT); + + if (spec->autocfg.input_pins[AUTO_PIN_MIC] && + spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC]) + snd_hda_codec_write(codec, + spec->autocfg.input_pins[AUTO_PIN_MIC], 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | ALC880_MIC_EVENT); - snd_hda_codec_write(codec, spec->autocfg.hp_pins[0], 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - AC_USRSP_EN | ALC880_HP_EVENT); spec->unsol_event = alc_sku_unsol_event; } @@ -1296,7 +1348,7 @@ static struct snd_kcontrol_new alc880_six_stack_mixer[] = { * * The system also has a pair of internal speakers, and a headphone jack. * These are both connected to Line2 on the codec, hence to DAC 02. - * + * * There is a variable resistor to control the speaker or headphone * volume. This is a hardware-only device without a software API. * @@ -1824,7 +1876,7 @@ static struct hda_verb alc880_pin_6stack_init_verbs[] = { {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - + { } }; @@ -1869,7 +1921,7 @@ static struct hda_verb alc880_uniwill_init_verbs[] = { /* * Uniwill P53 -* HP = 0x14, InternalSpeaker = 0x15, mic = 0x19, +* HP = 0x14, InternalSpeaker = 0x15, mic = 0x19, */ static struct hda_verb alc880_uniwill_p53_init_verbs[] = { {0x13, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ @@ -1968,7 +2020,7 @@ static void alc880_uniwill_p53_hp_automute(struct hda_codec *codec) static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec) { unsigned int present; - + present = snd_hda_codec_read(codec, 0x21, 0, AC_VERB_GET_VOLUME_KNOB_CONTROL, 0); present &= HDA_AMP_VOLMASK; @@ -2050,7 +2102,7 @@ static struct hda_verb alc880_pin_asus_init_verbs[] = { {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - + { } }; @@ -2632,12 +2684,14 @@ static int alc_build_pcms(struct hda_codec *codec) info->name = spec->stream_name_analog; if (spec->stream_analog_playback) { - snd_assert(spec->multiout.dac_nids, return -EINVAL); + if (snd_BUG_ON(!spec->multiout.dac_nids)) + return -EINVAL; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback); info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0]; } if (spec->stream_analog_capture) { - snd_assert(spec->adc_nids, return -EINVAL); + if (snd_BUG_ON(!spec->adc_nids)) + return -EINVAL; info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture); info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; } @@ -2667,6 +2721,8 @@ static int alc_build_pcms(struct hda_codec *codec) info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_digital_capture); info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid; } + /* FIXME: do we need this for all Realtek codec models? */ + codec->spdif_status_reset = 1; } /* If the use of more than one ADC is requested for the current @@ -3683,7 +3739,7 @@ static void alc880_auto_init_multi_out(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int i; - + alc_subsystem_id(codec, 0x15, 0x1b, 0x14); for (i = 0; i < spec->autocfg.line_outs; i++) { hda_nid_t nid = spec->autocfg.line_out_pins[i]; @@ -3787,7 +3843,7 @@ static void alc880_auto_init(struct hda_codec *codec) alc880_auto_init_extra_out(codec); alc880_auto_init_analog_input(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } /* @@ -4124,6 +4180,33 @@ static struct snd_kcontrol_new alc260_hp_3013_mixer[] = { { } /* end */ }; +static struct hda_bind_ctls alc260_dc7600_bind_master_vol = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x0a, 3, 0, HDA_OUTPUT), + 0 + }, +}; + +static struct hda_bind_ctls alc260_dc7600_bind_switch = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x11, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT), + 0 + }, +}; + +static struct snd_kcontrol_new alc260_hp_dc7600_mixer[] = { + HDA_BIND_VOL("Master Playback Volume", &alc260_dc7600_bind_master_vol), + HDA_BIND_SW("LineOut Playback Switch", &alc260_dc7600_bind_switch), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x0f, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x10, 0x0, HDA_OUTPUT), + { } /* end */ +}; + static struct hda_verb alc260_hp_3013_unsol_verbs[] = { {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, {}, @@ -4147,7 +4230,30 @@ static void alc260_hp_3013_unsol_event(struct hda_codec *codec, alc260_hp_3013_automute(codec); } -/* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12, +static void alc260_hp_3012_automute(struct hda_codec *codec) +{ + unsigned int present, bits; + + present = snd_hda_codec_read(codec, 0x10, 0, + AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE; + + bits = present ? 0 : PIN_OUT; + snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + bits); + snd_hda_codec_write(codec, 0x11, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + bits); + snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + bits); +} + +static void alc260_hp_3012_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc260_hp_3012_automute(codec); +} + +/* Fujitsu S702x series laptops. ALC260 pin usage: Mic/Line jack = 0x12, * HP jack = 0x14, CD audio = 0x16, internal speaker = 0x10. */ static struct snd_kcontrol_new alc260_fujitsu_mixer[] = { @@ -4478,7 +4584,7 @@ static struct hda_verb alc260_fujitsu_init_verbs[] = { {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, - /* Ensure Line1 pin widget takes its input from the OUT1 sum bus + /* Ensure Line1 pin widget takes its input from the OUT1 sum bus * when acting as an output. */ {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, @@ -4503,14 +4609,14 @@ static struct hda_verb alc260_fujitsu_init_verbs[] = { * stage. */ {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - /* Unmute input buffer of pin widget used for Line-in (no equiv + /* Unmute input buffer of pin widget used for Line-in (no equiv * mixer ctrl) */ {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Mute capture amp left and right */ {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - /* Set ADC connection select to match default mixer setting - line + /* Set ADC connection select to match default mixer setting - line * in (on mic1 pin) */ {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -4564,7 +4670,7 @@ static struct hda_verb alc260_acer_init_verbs[] = { {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, - /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum + /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum * bus when acting as outputs. */ {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, @@ -4675,6 +4781,20 @@ static void alc260_replacer_672v_unsol_event(struct hda_codec *codec, alc260_replacer_672v_automute(codec); } +static struct hda_verb alc260_hp_dc7600_verbs[] = { + {0x05, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x10, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + /* Test configuration for debugging, modelled after the ALC880 test * configuration. */ @@ -4686,7 +4806,7 @@ static hda_nid_t alc260_test_adc_nids[2] = { 0x04, 0x05, }; /* For testing the ALC260, each input MUX needs its own definition since - * the signal assignments are different. This assumes that the first ADC + * the signal assignments are different. This assumes that the first ADC * is NID 0x04. */ static struct hda_input_mux alc260_test_capture_sources[2] = { @@ -4769,7 +4889,7 @@ static struct snd_kcontrol_new alc260_test_mixer[] = { /* Switches to allow the digital IO pins to be enabled. The datasheet * is ambigious as to which NID is which; testing on laptops which - * make this output available should provide clarification. + * make this output available should provide clarification. */ ALC_SPDIF_CTRL_SWITCH("SPDIF Playback Switch", 0x03, 0x01), ALC_SPDIF_CTRL_SWITCH("SPDIF Capture Switch", 0x06, 0x01), @@ -4805,7 +4925,7 @@ static struct hda_verb alc260_test_init_verbs[] = { {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, - /* Ensure mic1, mic2, line1 and line2 pin widgets take input from the + /* Ensure mic1, mic2, line1 and line2 pin widgets take input from the * OUT1 sum bus when acting as an output. */ {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, @@ -4897,7 +5017,7 @@ static int alc260_add_playback_controls(struct alc_spec *spec, hda_nid_t nid, sw_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); } else return 0; /* N/A */ - + snprintf(name, sizeof(name), "%s Playback Volume", pfx); err = add_control(spec, ALC_CTL_WIDGET_VOL, name, vol_val); if (err < 0) @@ -5003,7 +5123,7 @@ static void alc260_auto_init_multi_out(struct hda_codec *codec) int pin_type = get_pin_type(spec->autocfg.line_out_type); alc260_auto_set_output_and_unmute(codec, nid, pin_type, 0); } - + nid = spec->autocfg.speaker_pins[0]; if (nid) alc260_auto_set_output_and_unmute(codec, nid, PIN_OUT, 0); @@ -5045,7 +5165,7 @@ static struct hda_verb alc260_volume_init_verbs[] = { {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - + /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback * mixer widget * Note: PASD motherboards uses the Line In 2 as the input for @@ -5074,7 +5194,7 @@ static struct hda_verb alc260_volume_init_verbs[] = { {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - + { } }; @@ -5134,7 +5254,7 @@ static void alc260_auto_init(struct hda_codec *codec) alc260_auto_init_multi_out(codec); alc260_auto_init_analog_input(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -5155,6 +5275,7 @@ static const char *alc260_models[ALC260_MODEL_LAST] = { [ALC260_BASIC] = "basic", [ALC260_HP] = "hp", [ALC260_HP_3013] = "hp-3013", + [ALC260_HP_DC7600] = "hp-dc7600", [ALC260_FUJITSU_S702X] = "fujitsu", [ALC260_ACER] = "acer", [ALC260_WILL] = "will", @@ -5172,7 +5293,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3011, "HP", ALC260_HP_3013), - SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_3013), + SND_PCI_QUIRK(0x103c, 0x3012, "HP", ALC260_HP_DC7600), SND_PCI_QUIRK(0x103c, 0x3013, "HP", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3014, "HP", ALC260_HP), SND_PCI_QUIRK(0x103c, 0x3015, "HP", ALC260_HP), @@ -5218,6 +5339,22 @@ static struct alc_config_preset alc260_presets[] = { .unsol_event = alc260_hp_unsol_event, .init_hook = alc260_hp_automute, }, + [ALC260_HP_DC7600] = { + .mixers = { alc260_hp_dc7600_mixer, + alc260_input_mixer, + alc260_capture_alt_mixer }, + .init_verbs = { alc260_init_verbs, + alc260_hp_dc7600_verbs }, + .num_dacs = ARRAY_SIZE(alc260_dac_nids), + .dac_nids = alc260_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc260_hp_adc_nids), + .adc_nids = alc260_hp_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .input_mux = &alc260_capture_source, + .unsol_event = alc260_hp_3012_unsol_event, + .init_hook = alc260_hp_3012_automute, + }, [ALC260_HP_3013] = { .mixers = { alc260_hp_3013_mixer, alc260_input_mixer, @@ -5933,7 +6070,7 @@ static struct hda_verb alc882_targa_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - + {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ @@ -5949,7 +6086,7 @@ static struct hda_verb alc882_targa_verbs[] = { static void alc882_targa_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_stereo(codec, 0x1b, HDA_OUTPUT, 0, @@ -5975,7 +6112,7 @@ static struct hda_verb alc882_asus_a7j_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */ @@ -5993,7 +6130,7 @@ static struct hda_verb alc882_asus_a7m_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front */ @@ -6319,7 +6456,7 @@ static struct alc_config_preset alc882_presets[] = { .channel_mode = alc882_3ST_6ch_modes, .need_dac_fix = 1, .input_mux = &alc882_capture_source, - }, + }, [ALC882_ASUS_A7M] = { .mixers = { alc882_asus_a7m_mixer, alc882_chmode_mixer }, .init_verbs = { alc882_init_verbs, alc882_eapd_verbs, @@ -6332,14 +6469,14 @@ static struct alc_config_preset alc882_presets[] = { .channel_mode = alc880_threestack_modes, .need_dac_fix = 1, .input_mux = &alc882_capture_source, - }, + }, }; /* * Pin config fixes */ -enum { +enum { PINFIX_ABIT_AW9D_MAX }; @@ -6527,7 +6664,7 @@ static void alc882_auto_init(struct hda_codec *codec) alc882_auto_init_analog_input(codec); alc882_auto_init_input_src(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } static int patch_alc883(struct hda_codec *codec); /* called in patch_alc882() */ @@ -6554,16 +6691,19 @@ static int patch_alc882(struct hda_codec *codec) board_config = ALC885_MACPRO; break; case 0x106b1000: /* iMac 24 */ + case 0x106b2800: /* AppleTV */ board_config = ALC885_IMAC24; break; case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ + case 0x106b00a4: /* MacbookPro4,1 */ case 0x106b2c00: /* Macbook Pro rev3 */ case 0x106b3600: /* Macbook 3.1 */ board_config = ALC885_MBP3; break; default: /* ALC889A is handled better as ALC888-compatible */ - if (codec->revision_id == 0x100103) { + if (codec->revision_id == 0x100101 || + codec->revision_id == 0x100103) { alc_free(codec); return patch_alc883(codec); } @@ -6718,6 +6858,23 @@ static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = { }, }; +static struct hda_input_mux alc883_lenovo_sky_capture_source = { + .num_items = 3, + .items = { + { "Mic", 0x0 }, + { "Front Mic", 0x1 }, + { "Line", 0x4 }, + }, +}; + +static struct hda_input_mux alc883_asus_eee1601_capture_source = { + .num_items = 2, + .items = { + { "Mic", 0x0 }, + { "Line", 0x2 }, + }, +}; + #define alc883_mux_enum_info alc_mux_enum_info #define alc883_mux_enum_get alc_mux_enum_get /* ALC883 has the ALC882-type input selection */ @@ -7032,13 +7189,11 @@ static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = { 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, + .count = 1, .info = alc883_mux_enum_info, .get = alc883_mux_enum_get, .put = alc883_mux_enum_put, @@ -7256,7 +7411,7 @@ static struct snd_kcontrol_new alc883_medion_md2_mixer[] = { .put = alc883_mux_enum_put, }, { } /* end */ -}; +}; static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -7283,6 +7438,87 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc888_lenovo_sky_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_MUTE("iSpeaker Playback Switch", 0x1a, 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("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 hda_bind_ctls alc883_bind_cap_vol = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT), + 0 + }, +}; + +static struct hda_bind_ctls alc883_bind_cap_switch = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x08, 3, 0, HDA_INPUT), + HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT), + 0 + }, +}; + +static struct snd_kcontrol_new alc883_asus_eee1601_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), + 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_BIND_VOL("Capture Volume", &alc883_bind_cap_vol), + HDA_BIND_SW("Capture Switch", &alc883_bind_cap_switch), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 1, + .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, @@ -7296,7 +7532,7 @@ static struct snd_kcontrol_new alc883_chmode_mixer[] = { static struct hda_verb alc883_init_verbs[] = { /* ADC1: mute amp left and right */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, /* ADC2: mute amp left and right */ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -7361,14 +7597,14 @@ static struct hda_verb alc883_init_verbs[] = { /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ /* Input mixer2 */ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* Input mixer3 */ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, - {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, { } }; @@ -7468,7 +7704,7 @@ static struct hda_verb alc883_tagra_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - + {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* mic/clfe */ {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* line/surround */ {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* HP */ @@ -7518,6 +7754,18 @@ static struct hda_verb alc883_haier_w66_verbs[] = { { } /* end */ }; +static struct hda_verb alc888_lenovo_sky_verbs[] = { + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + { } /* end */ +}; + static struct hda_verb alc888_3st_hp_verbs[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ @@ -7555,7 +7803,7 @@ static struct hda_channel_mode alc888_3st_hp_modes[2] = { static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec) { unsigned int present; - + present = snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, @@ -7568,7 +7816,7 @@ static void alc888_lenovo_ms7195_front_automute(struct hda_codec *codec) static void alc888_lenovo_ms7195_rca_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_stereo(codec, 0x15, HDA_OUTPUT, 0, @@ -7598,7 +7846,7 @@ static struct hda_verb alc883_medion_md2_verbs[] = { static void alc883_medion_md2_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_stereo(codec, 0x15, HDA_OUTPUT, 0, @@ -7753,7 +8001,7 @@ static void alc883_lenovo_101e_unsol_event(struct hda_codec *codec, static void alc883_acer_aspire_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_stereo(codec, 0x15, HDA_OUTPUT, 0, @@ -7790,7 +8038,7 @@ static struct hda_verb alc883_acer_eapd_verbs[] = { static void alc888_6st_dell_front_automute(struct hda_codec *codec) { unsigned int present; - + present = snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, @@ -7814,6 +8062,50 @@ static void alc888_6st_dell_unsol_event(struct hda_codec *codec, } } +static void alc888_lenovo_sky_front_automute(struct hda_codec *codec) +{ + unsigned int mute; + unsigned int present; + + snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0); + present = (present & 0x80000000) != 0; + if (present) { + /* mute internal speaker */ + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + } else { + /* unmute internal speaker if necessary */ + mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0); + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + } +} + +static void alc883_lenovo_sky_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc888_lenovo_sky_front_automute(codec); +} + /* * generic initialization of ADC, input mixers and output mixers */ @@ -7898,6 +8190,105 @@ static struct snd_kcontrol_new alc883_capture_mixer[] = { { } /* end */ }; +static struct hda_verb alc888_asus_m90v_verbs[] = { + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* enable unsolicited event */ + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, + { } /* end */ +}; + +static void alc883_nb_mic_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); + snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, + 0x7000 | (0x01 << 8) | (present ? 0x80 : 0)); +} + +static void alc883_M90V_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? 0 : PIN_OUT; + snd_hda_codec_write(codec, 0x14, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + bits); + snd_hda_codec_write(codec, 0x15, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + bits); + snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + bits); +} + +static void alc883_mode2_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc883_M90V_speaker_automute(codec); + break; + case ALC880_MIC_EVENT: + alc883_nb_mic_automute(codec); + break; + } +} + +static void alc883_mode2_inithook(struct hda_codec *codec) +{ + alc883_M90V_speaker_automute(codec); + alc883_nb_mic_automute(codec); +} + +static struct hda_verb alc888_asus_eee1601_verbs[] = { + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x20, AC_VERB_SET_COEF_INDEX, 0x0b}, + {0x20, AC_VERB_SET_PROC_COEF, 0x0838}, + /* enable unsolicited event */ + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + { } /* end */ +}; + +static void alc883_eee1601_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? 0 : PIN_OUT; + snd_hda_codec_write(codec, 0x1b, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, + bits); +} + +static void alc883_eee1601_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc883_eee1601_speaker_automute(codec); + break; + } +} + +static void alc883_eee1601_inithook(struct hda_codec *codec) +{ + alc883_eee1601_speaker_automute(codec); +} + #ifdef CONFIG_SND_HDA_POWER_SAVE #define alc883_loopbacks alc880_loopbacks #endif @@ -7927,6 +8318,7 @@ 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_LENOVO_SKY] = "lenovo-sky", [ALC883_HAIER_W66] = "haier-w66", [ALC888_3ST_HP] = "3stack-hp", [ALC888_6ST_DELL] = "6stack-dell", @@ -7942,18 +8334,21 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE), - SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE), + SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */ SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL), SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), + SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601), SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), + SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL), SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch), SND_PCI_QUIRK(0x1458, 0xa002, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), @@ -7989,6 +8384,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x3bfc, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763), + SND_PCI_QUIRK(0x17aa, 0x101d, "Lenovo Sky", ALC888_LENOVO_SKY), SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66), @@ -8128,7 +8524,7 @@ static struct alc_config_preset alc883_presets[] = { .input_mux = &alc883_capture_source, .unsol_event = alc883_medion_md2_unsol_event, .init_hook = alc883_medion_md2_automute, - }, + }, [ALC883_LAPTOP_EAPD] = { .mixers = { alc883_base_mixer }, .init_verbs = { alc883_init_verbs, alc882_eapd_verbs }, @@ -8245,6 +8641,49 @@ static struct alc_config_preset alc883_presets[] = { .unsol_event = alc883_2ch_fujitsu_pi2515_unsol_event, .init_hook = alc883_2ch_fujitsu_pi2515_automute, }, + [ALC888_LENOVO_SKY] = { + .mixers = { alc888_lenovo_sky_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc888_lenovo_sky_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, + .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), + .channel_mode = alc883_sixstack_modes, + .need_dac_fix = 1, + .input_mux = &alc883_lenovo_sky_capture_source, + .unsol_event = alc883_lenovo_sky_unsol_event, + .init_hook = alc888_lenovo_sky_front_automute, + }, + [ALC888_ASUS_M90V] = { + .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc888_asus_m90v_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .dig_in_nid = ALC883_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes), + .channel_mode = alc883_3ST_6ch_modes, + .need_dac_fix = 1, + .input_mux = &alc883_fujitsu_pi2515_capture_source, + .unsol_event = alc883_mode2_unsol_event, + .init_hook = alc883_mode2_inithook, + }, + [ALC888_ASUS_EEE1601] = { + .mixers = { alc883_asus_eee1601_mixer }, + .init_verbs = { alc883_init_verbs, alc888_asus_eee1601_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .dig_in_nid = ALC883_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .need_dac_fix = 1, + .input_mux = &alc883_asus_eee1601_capture_source, + .unsol_event = alc883_eee1601_unsol_event, + .init_hook = alc883_eee1601_inithook, + }, }; @@ -8354,7 +8793,7 @@ static void alc883_auto_init(struct hda_codec *codec) alc883_auto_init_analog_input(codec); alc883_auto_init_input_src(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } static int patch_alc883(struct hda_codec *codec) @@ -8398,8 +8837,13 @@ static int patch_alc883(struct hda_codec *codec) switch (codec->vendor_id) { case 0x10ec0888: - spec->stream_name_analog = "ALC888 Analog"; - spec->stream_name_digital = "ALC888 Digital"; + if (codec->revision_id == 0x100101) { + spec->stream_name_analog = "ALC1200 Analog"; + spec->stream_name_digital = "ALC1200 Digital"; + } else { + spec->stream_name_analog = "ALC888 Analog"; + spec->stream_name_digital = "ALC888 Digital"; + } break; case 0x10ec0889: spec->stream_name_analog = "ALC889 Analog"; @@ -8452,6 +8896,13 @@ static int patch_alc883(struct hda_codec *codec) #define alc262_modes alc260_modes #define alc262_capture_source alc882_capture_source +static hda_nid_t alc262_dmic_adc_nids[1] = { + /* ADC0 */ + 0x09 +}; + +static hda_nid_t alc262_dmic_capsrc_nids[1] = { 0x22 }; + static struct snd_kcontrol_new alc262_base_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT), @@ -8833,10 +9284,10 @@ static struct hda_verb alc262_init_verbs[] = { {0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000}, - + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, - + /* FIXME: use matrix-type input source selection */ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ @@ -8858,6 +9309,12 @@ static struct hda_verb alc262_init_verbs[] = { { } }; +static struct hda_verb alc262_eapd_verbs[] = { + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, + { } +}; + static struct hda_verb alc262_hippo_unsol_verbs[] = { {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, @@ -8884,6 +9341,91 @@ static struct hda_verb alc262_sony_unsol_verbs[] = { {} }; +static struct hda_input_mux alc262_dmic_capture_source = { + .num_items = 2, + .items = { + { "Int DMic", 0x9 }, + { "Mic", 0x0 }, + }, +}; + +static struct snd_kcontrol_new alc262_toshiba_s06_mixer[] = { + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker 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("Capture Volume", 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 1, + .info = alc_mux_enum_info, + .get = alc_mux_enum_get, + .put = alc_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb alc262_toshiba_s06_verbs[] = { + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x22, AC_VERB_SET_CONNECT_SEL, 0x09}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + +static void alc262_dmic_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_write(codec, 0x22, 0, + AC_VERB_SET_CONNECT_SEL, present ? 0x0 : 0x09); +} + +/* toggle speaker-output according to the hp-jack state */ +static void alc262_toshiba_s06_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + bits = present ? 0 : PIN_OUT; + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, bits); +} + + + +/* unsolicited event for HP jack sensing */ +static void alc262_toshiba_s06_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc262_toshiba_s06_speaker_automute(codec); + if ((res >> 26) == ALC880_MIC_EVENT) + alc262_dmic_automute(codec); + +} + +static void alc262_toshiba_s06_init_hook(struct hda_codec *codec) +{ + alc262_toshiba_s06_speaker_automute(codec); + alc262_dmic_automute(codec); +} + /* mute/unmute internal speaker according to the hp jack and mute state */ static void alc262_hippo_automute(struct hda_codec *codec) { @@ -8948,6 +9490,41 @@ static void alc262_hippo1_unsol_event(struct hda_codec *codec, } /* + * nec model + * 0x15 = headphone + * 0x16 = internal speaker + * 0x18 = external mic + */ + +static struct snd_kcontrol_new alc262_nec_mixer[] = { + HDA_CODEC_VOLUME_MONO("Speaker Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_MONO("Speaker Playback Switch", 0x16, 0, 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("Mic Boost", 0x18, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + { } /* end */ +}; + +static struct hda_verb alc262_nec_verbs[] = { + /* Unmute Speaker */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + + /* Headphone */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* External mic to headphone */ + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + /* External mic to speaker */ + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {} +}; + +/* * fujitsu model * 0x14 = headphone/spdif-out, 0x15 = internal speaker, * 0x1b = port replicator headphone out @@ -9179,6 +9756,25 @@ static struct snd_kcontrol_new alc262_lenovo_3000_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc262_toshiba_rx1_mixer[] = { + HDA_BIND_VOL("Master Playback Volume", &alc262_fujitsu_bind_master_vol), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = alc262_sony_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x15, 3, 0, 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("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + { } /* end */ +}; + /* additional init verbs for Benq laptops */ static struct hda_verb alc262_EAPD_verbs[] = { {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, @@ -9427,7 +10023,7 @@ static struct hda_verb alc262_volume_init_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - + /* set up input amps for analog loopback */ /* Amp Indices: DAC = 0, mixer = 1 */ {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -9482,7 +10078,7 @@ static struct hda_verb alc262_HP_BPC_init_verbs[] = { {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, - + /* * Set up output mixers (0x0c - 0x0e) */ @@ -9643,6 +10239,24 @@ static struct hda_verb alc262_HP_BPC_WildWest_init_verbs[] = { { } }; +static struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = { + + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Front Speaker */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x01}, + + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* MIC jack */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Front MIC */ + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) }, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) }, + + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP jack */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + + #ifdef CONFIG_SND_HDA_POWER_SAVE #define alc262_loopbacks alc880_loopbacks #endif @@ -9711,7 +10325,7 @@ static void alc262_auto_init(struct hda_codec *codec) alc262_auto_init_analog_input(codec); alc262_auto_init_input_src(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } /* @@ -9729,13 +10343,17 @@ static const char *alc262_models[ALC262_MODEL_LAST] = { [ALC262_BENQ_ED8] = "benq", [ALC262_BENQ_T31] = "benq-t31", [ALC262_SONY_ASSAMD] = "sony-assamd", + [ALC262_TOSHIBA_S06] = "toshiba-s06", + [ALC262_TOSHIBA_RX1] = "toshiba-rx1", [ALC262_ULTRA] = "ultra", [ALC262_LENOVO_3000] = "lenovo-3000", + [ALC262_NEC] = "nec", [ALC262_AUTO] = "auto", }; static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO), + SND_PCI_QUIRK(0x1033, 0x8895, "NEC Versa S9100", ALC262_NEC), SND_PCI_QUIRK(0x103c, 0x12fe, "HP xw9400", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x12ff, "HP xw4550", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x1306, "HP xw8600", ALC262_HP_BPC), @@ -9764,7 +10382,8 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", - ALC262_SONY_ASSAMD), + ALC262_TOSHIBA_RX1), + SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), @@ -9918,7 +10537,7 @@ static struct alc_config_preset alc262_presets[] = { .input_mux = &alc262_capture_source, .unsol_event = alc262_hippo_unsol_event, .init_hook = alc262_hippo_automute, - }, + }, [ALC262_ULTRA] = { .mixers = { alc262_ultra_mixer, alc262_ultra_capture_mixer }, .init_verbs = { alc262_ultra_verbs }, @@ -9946,6 +10565,43 @@ static struct alc_config_preset alc262_presets[] = { .input_mux = &alc262_fujitsu_capture_source, .unsol_event = alc262_lenovo_3000_unsol_event, }, + [ALC262_NEC] = { + .mixers = { alc262_nec_mixer }, + .init_verbs = { alc262_nec_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, + }, + [ALC262_TOSHIBA_S06] = { + .mixers = { alc262_toshiba_s06_mixer }, + .init_verbs = { alc262_init_verbs, alc262_toshiba_s06_verbs, + alc262_eapd_verbs }, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .capsrc_nids = alc262_dmic_capsrc_nids, + .dac_nids = alc262_dac_nids, + .adc_nids = alc262_dmic_adc_nids, /* ADC0 */ + .dig_out_nid = ALC262_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_dmic_capture_source, + .unsol_event = alc262_toshiba_s06_unsol_event, + .init_hook = alc262_toshiba_s06_init_hook, + }, + [ALC262_TOSHIBA_RX1] = { + .mixers = { alc262_toshiba_rx1_mixer }, + .init_verbs = { alc262_init_verbs, alc262_toshiba_rx1_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, + .init_hook = alc262_hippo_automute, + }, }; static int patch_alc262(struct hda_codec *codec) @@ -10004,7 +10660,7 @@ static int patch_alc262(struct hda_codec *codec) spec->stream_name_analog = "ALC262 Analog"; spec->stream_analog_playback = &alc262_pcm_analog_playback; spec->stream_analog_capture = &alc262_pcm_analog_capture; - + spec->stream_name_digital = "ALC262 Digital"; spec->stream_digital_playback = &alc262_pcm_digital_playback; spec->stream_digital_capture = &alc262_pcm_digital_capture; @@ -10040,7 +10696,7 @@ static int patch_alc262(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc262_loopbacks; #endif - + return 0; } @@ -10049,7 +10705,7 @@ static int patch_alc262(struct hda_codec *codec) */ #define ALC268_DIGOUT_NID ALC880_DIGOUT_NID #define alc268_modes alc260_modes - + static hda_nid_t alc268_dac_nids[2] = { /* front, hp */ 0x02, 0x03 @@ -10109,6 +10765,14 @@ static struct hda_verb alc268_toshiba_verbs[] = { { } /* end */ }; +static struct hda_input_mux alc268_acer_lc_capture_source = { + .num_items = 2, + .items = { + { "i-Mic", 0x6 }, + { "E-Mic", 0x0 }, + }, +}; + /* Acer specific */ /* bind volumes of both NID 0x02 and 0x03 */ static struct hda_bind_ctls alc268_acer_bind_master_vol = { @@ -10161,6 +10825,21 @@ static int alc268_acer_master_sw_put(struct snd_kcontrol *kcontrol, return change; } +static struct snd_kcontrol_new alc268_acer_aspire_one_mixer[] = { + /* output mixer control */ + HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = alc268_acer_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), + }, + HDA_CODEC_VOLUME("Mic Boost Capture Volume", 0x18, 0, HDA_INPUT), + { } +}; + static struct snd_kcontrol_new alc268_acer_mixer[] = { /* output mixer control */ HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol), @@ -10178,6 +10857,16 @@ static struct snd_kcontrol_new alc268_acer_mixer[] = { { } }; +static struct hda_verb alc268_acer_aspire_one_verbs[] = { + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x23, AC_VERB_SET_CONNECT_SEL, 0x06}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, 0xa017}, + { } +}; + static struct hda_verb alc268_acer_verbs[] = { {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* internal dmic? */ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, @@ -10185,7 +10874,6 @@ static struct hda_verb alc268_acer_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, { } }; @@ -10212,6 +10900,47 @@ static void alc268_acer_init_hook(struct hda_codec *codec) alc268_acer_automute(codec, 1); } +/* toggle speaker-output according to the hp-jack state */ +static void alc268_aspire_one_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + bits = present ? AMP_IN_MUTE(0) : 0; + snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 0, + AMP_IN_MUTE(0), bits); + snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 1, + AMP_IN_MUTE(0), bits); +} + + +static void alc268_acer_mic_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_CONNECT_SEL, + present ? 0x0 : 0x6); +} + +static void alc268_acer_lc_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc268_aspire_one_speaker_automute(codec); + if ((res >> 26) == ALC880_MIC_EVENT) + alc268_acer_mic_automute(codec); +} + +static void alc268_acer_lc_init_hook(struct hda_codec *codec) +{ + alc268_aspire_one_speaker_automute(codec); + alc268_acer_mic_automute(codec); +} + static struct snd_kcontrol_new alc268_dell_mixer[] = { /* output mixer control */ HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), @@ -10360,7 +11089,7 @@ static struct hda_verb alc268_base_init_verbs[] = { {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* Unmute Selector 23h,24h and set the default input to mic-in */ - + {0x23, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x24, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -10559,7 +11288,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, nid = cfg->line_out_pins[0]; if (nid) - alc268_new_analog_output(spec, nid, "Front", 0); + alc268_new_analog_output(spec, nid, "Front", 0); nid = cfg->speaker_pins[0]; if (nid == 0x1d) { @@ -10581,7 +11310,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, if (err < 0) return err; } - return 0; + return 0; } /* create playback/capture controls for input pins */ @@ -10602,7 +11331,7 @@ static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, case 0x1a: idx1 = 2; /* Line In */ break; - case 0x1c: + case 0x1c: idx1 = 3; /* CD */ break; case 0x12: @@ -10614,7 +11343,7 @@ static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, } imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; imux->items[imux->num_items].index = idx1; - imux->num_items++; + imux->num_items++; } return 0; } @@ -10644,11 +11373,11 @@ static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec) } dac_vol1 = dac_vol2 = 0xb000 | 0x40; /* set max volume */ - if (line_nid == 0x14) + if (line_nid == 0x14) dac_vol2 = AMP_OUT_ZERO; else if (line_nid == 0x15) dac_vol1 = AMP_OUT_ZERO; - if (hp_nid == 0x14) + if (hp_nid == 0x14) dac_vol2 = AMP_OUT_ZERO; else if (hp_nid == 0x15) dac_vol1 = AMP_OUT_ZERO; @@ -10728,7 +11457,7 @@ static void alc268_auto_init(struct hda_codec *codec) alc268_auto_init_mono_speaker_out(codec); alc268_auto_init_analog_input(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } /* @@ -10739,6 +11468,7 @@ static const char *alc268_models[ALC268_MODEL_LAST] = { [ALC268_3ST] = "3stack", [ALC268_TOSHIBA] = "toshiba", [ALC268_ACER] = "acer", + [ALC268_ACER_ASPIRE_ONE] = "acer-aspire", [ALC268_DELL] = "dell", [ALC268_ZEPTO] = "zepto", #ifdef CONFIG_SND_DEBUG @@ -10753,11 +11483,14 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x012e, "Acer Aspire 5310", ALC268_ACER), SND_PCI_QUIRK(0x1025, 0x0130, "Acer Extensa 5210", ALC268_ACER), SND_PCI_QUIRK(0x1025, 0x0136, "Acer Aspire 5315", ALC268_ACER), + SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One", + ALC268_ACER_ASPIRE_ONE), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), SND_PCI_QUIRK(0x103c, 0x30cc, "TOSHIBA", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), SND_PCI_QUIRK(0x1179, 0xff10, "TOSHIBA A205", ALC268_TOSHIBA), SND_PCI_QUIRK(0x1179, 0xff50, "TOSHIBA A305", ALC268_TOSHIBA), + SND_PCI_QUIRK(0x1179, 0xff64, "TOSHIBA L305", ALC268_TOSHIBA), SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA), SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER), SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1), @@ -10830,6 +11563,23 @@ static struct alc_config_preset alc268_presets[] = { .unsol_event = alc268_acer_unsol_event, .init_hook = alc268_acer_init_hook, }, + [ALC268_ACER_ASPIRE_ONE] = { + .mixers = { alc268_acer_aspire_one_mixer, + alc268_capture_alt_mixer }, + .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, + alc268_acer_aspire_one_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, + .capsrc_nids = alc268_capsrc_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc268_modes), + .channel_mode = alc268_modes, + .input_mux = &alc268_acer_lc_capture_source, + .unsol_event = alc268_acer_lc_unsol_event, + .init_hook = alc268_acer_lc_init_hook, + }, [ALC268_DELL] = { .mixers = { alc268_dell_mixer, alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, @@ -10974,7 +11724,7 @@ static int patch_alc268(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC268_AUTO) spec->init_hook = alc268_auto_init; - + return 0; } @@ -10990,6 +11740,14 @@ static hda_nid_t alc269_adc_nids[1] = { 0x08, }; +static hda_nid_t alc269_capsrc_nids[1] = { + 0x23, +}; + +/* NOTE: ADC2 (0x07) is connected from a recording *MIXER* (0x24), + * not a mux! + */ + static struct hda_input_mux alc269_eeepc_dmic_capture_source = { .num_items = 2, .items = { @@ -11016,6 +11774,8 @@ static struct snd_kcontrol_new alc269_base_mixer[] = { 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("Beep Playback Volume", 0x0b, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x4, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), @@ -11025,6 +11785,28 @@ static struct snd_kcontrol_new alc269_base_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { + /* output mixer control */ + HDA_BIND_VOL("Master Playback Volume", &alc268_acer_bind_master_vol), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = alc268_acer_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x14, 3, 0, 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("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT), + { } +}; + /* bind volumes of both NID 0x0c and 0x0d */ static struct hda_bind_ctls alc269_epc_bind_vol = { .ops = &snd_hda_bind_vol, @@ -11068,75 +11850,72 @@ static struct snd_kcontrol_new alc269_epc_capture_mixer[] = { { } /* end */ }; -/* - * generic initialization of ADC, input mixers and output mixers - */ -static struct hda_verb alc269_init_verbs[] = { - /* - * Unmute ADC0 and set the default input to mic-in - */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, +/* beep control */ +static struct snd_kcontrol_new alc269_beep_mixer[] = { + HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x4, HDA_INPUT), + { } /* end */ +}; - /* Mute input amps (PCBeep, Line In, Mic 1 & Mic 2) of the - * analog-loopback mixer widget - * Note: PASD motherboards uses the Line In 2 as the input for - * front panel mic (mic 2) - */ - /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, - {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, +static struct hda_verb alc269_quanta_fl1_verbs[] = { + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + { } +}; - /* - * Set up output mixers (0x0c - 0x0e) - */ - /* set vol=0 to output mixers */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, +/* toggle speaker-output according to the hp-jack state */ +static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; - /* set up input amps for analog loopback */ - /* Amp Indices: DAC = 0, mixer = 1 */ - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + bits = present ? AMP_IN_MUTE(0) : 0; + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, + AMP_IN_MUTE(0), bits); + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, + AMP_IN_MUTE(0), bits); - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, - {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_COEF_INDEX, 0x0c); + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_PROC_COEF, 0x680); - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {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}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_COEF_INDEX, 0x0c); + snd_hda_codec_write(codec, 0x20, 0, + AC_VERB_SET_PROC_COEF, 0x480); +} - {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, - {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, +static void alc269_quanta_fl1_mic_automute(struct hda_codec *codec) +{ + unsigned int present; - /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */ - /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_write(codec, 0x23, 0, + AC_VERB_SET_CONNECT_SEL, present ? 0x0 : 0x1); +} - /* set EAPD */ - {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, - {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, - { } -}; +static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) == ALC880_HP_EVENT) + alc269_quanta_fl1_speaker_automute(codec); + if ((res >> 26) == ALC880_MIC_EVENT) + alc269_quanta_fl1_mic_automute(codec); +} + +static void alc269_quanta_fl1_init_hook(struct hda_codec *codec) +{ + alc269_quanta_fl1_speaker_automute(codec); + alc269_quanta_fl1_mic_automute(codec); +} static struct hda_verb alc269_eeepc_dmic_init_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, @@ -11163,42 +11942,42 @@ static struct hda_verb alc269_eeepc_amic_init_verbs[] = { static void alc269_speaker_automute(struct hda_codec *codec) { unsigned int present; - unsigned int bits; + unsigned char bits; present = snd_hda_codec_read(codec, 0x15, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; bits = present ? AMP_IN_MUTE(0) : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + AMP_IN_MUTE(0), bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + AMP_IN_MUTE(0), bits); } static void alc269_eeepc_dmic_automute(struct hda_codec *codec) { unsigned int present; - present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; - snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_CONNECT_SEL, - present ? 0 : 5); + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_write(codec, 0x23, 0, + AC_VERB_SET_CONNECT_SEL, (present ? 0 : 5)); } static void alc269_eeepc_amic_automute(struct hda_codec *codec) { unsigned int present; - present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE, - present ? AMP_IN_UNMUTE(0) : AMP_IN_MUTE(0)); + 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE, - present ? AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1)); + 0x7000 | (0x01 << 8) | (present ? 0x80 : 0)); } /* unsolicited event for HP jack sensing */ static void alc269_eeepc_dmic_unsol_event(struct hda_codec *codec, - unsigned int res) + unsigned int res) { if ((res >> 26) == ALC880_HP_EVENT) alc269_speaker_automute(codec); @@ -11215,7 +11994,7 @@ static void alc269_eeepc_dmic_inithook(struct hda_codec *codec) /* unsolicited event for HP jack sensing */ static void alc269_eeepc_amic_unsol_event(struct hda_codec *codec, - unsigned int res) + unsigned int res) { if ((res >> 26) == ALC880_HP_EVENT) alc269_speaker_automute(codec); @@ -11230,6 +12009,76 @@ static void alc269_eeepc_amic_inithook(struct hda_codec *codec) alc269_eeepc_amic_automute(codec); } +/* + * generic initialization of ADC, input mixers and output mixers + */ +static struct hda_verb alc269_init_verbs[] = { + /* + * Unmute ADC0 and set the default input to mic-in + */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Mute input amps (PCBeep, Line In, Mic 1 & Mic 2) of the + * analog-loopback mixer widget + * Note: PASD motherboards uses the Line In 2 as the input for + * front panel mic (mic 2) + */ + /* Amp Indices: Mic1 = 0, Mic2 = 1, Line1 = 2, Line2 = 3, CD = 4 */ + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* + * Set up output mixers (0x0c - 0x0e) + */ + /* set vol=0 to output mixers */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* set up input amps for analog loopback */ + /* Amp Indices: DAC = 0, mixer = 1 */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {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}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* FIXME: use matrix-type input source selection */ + /* Mixer elements: 0x18, 19, 1a, 1b, 1d, 0b */ + /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + + /* set EAPD */ + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, + { } +}; + /* add playback controls from the parsed DAC table */ static int alc269_auto_create_multi_out_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) @@ -11330,7 +12179,7 @@ static int alc269_auto_create_multi_out_ctls(struct alc_spec *spec, static int alc269_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int err; + int i, err; static hda_nid_t alc269_ignore[] = { 0x1d, 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, @@ -11353,9 +12202,20 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (spec->kctl_alloc) spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + /* create a beep mixer control if the pin 0x1d isn't assigned */ + for (i = 0; i < ARRAY_SIZE(spec->autocfg.input_pins); i++) + if (spec->autocfg.input_pins[i] == 0x1d) + break; + if (i >= ARRAY_SIZE(spec->autocfg.input_pins)) + spec->mixers[spec->num_mixers++] = alc269_beep_mixer; + spec->init_verbs[spec->num_init_verbs++] = alc269_init_verbs; spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; + /* set default input source */ + snd_hda_codec_write_cache(codec, alc269_capsrc_nids[0], + 0, AC_VERB_SET_CONNECT_SEL, + spec->input_mux->items[0].index); err = alc_auto_add_mic_boost(codec); if (err < 0) @@ -11380,21 +12240,27 @@ static void alc269_auto_init(struct hda_codec *codec) alc269_auto_init_hp_out(codec); alc269_auto_init_analog_input(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } /* * configuration and preset */ static const char *alc269_models[ALC269_MODEL_LAST] = { - [ALC269_BASIC] = "basic", + [ALC269_BASIC] = "basic", + [ALC269_QUANTA_FL1] = "quanta", + [ALC269_ASUS_EEEPC_P703] = "eeepc-p703", + [ALC269_ASUS_EEEPC_P901] = "eeepc-p901" }; static struct snd_pci_quirk alc269_cfg_tbl[] = { + SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_QUANTA_FL1), SND_PCI_QUIRK(0x1043, 0x8330, "ASUS Eeepc P703 P900A", ALC269_ASUS_EEEPC_P703), SND_PCI_QUIRK(0x1043, 0x831a, "ASUS Eeepc P901", ALC269_ASUS_EEEPC_P901), + SND_PCI_QUIRK(0x1043, 0x834a, "ASUS Eeepc S101", + ALC269_ASUS_EEEPC_P901), {} }; @@ -11409,6 +12275,18 @@ static struct alc_config_preset alc269_presets[] = { .channel_mode = alc269_modes, .input_mux = &alc269_capture_source, }, + [ALC269_QUANTA_FL1] = { + .mixers = { alc269_quanta_fl1_mixer }, + .init_verbs = { alc269_init_verbs, alc269_quanta_fl1_verbs }, + .num_dacs = ARRAY_SIZE(alc269_dac_nids), + .dac_nids = alc269_dac_nids, + .hp_nid = 0x03, + .num_channel_mode = ARRAY_SIZE(alc269_modes), + .channel_mode = alc269_modes, + .input_mux = &alc269_capture_source, + .unsol_event = alc269_quanta_fl1_unsol_event, + .init_hook = alc269_quanta_fl1_init_hook, + }, [ALC269_ASUS_EEEPC_P703] = { .mixers = { alc269_eeepc_mixer, alc269_epc_capture_mixer }, .init_verbs = { alc269_init_verbs, @@ -11488,6 +12366,7 @@ static int patch_alc269(struct hda_codec *codec) spec->adc_nids = alc269_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids); + spec->capsrc_nids = alc269_capsrc_nids; codec->patch_ops = alc_patch_ops; if (board_config == ALC269_AUTO) @@ -11689,7 +12568,7 @@ static struct snd_kcontrol_new alc861_toshiba_mixer[] = { HDA_CODEC_MUTE("Master Playback Switch", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x15, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x15, 0x01, HDA_INPUT), - + /*Capture mixer control */ HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), @@ -11832,20 +12711,20 @@ static struct hda_verb alc861_base_init_verbs[] = { /* route front mic to ADC1*/ {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - + /* Unmute DAC0~3 & spdif out*/ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - + /* Unmute Mixer 14 (mic) 1c (Line in)*/ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - + /* Unmute Stereo Mixer 15 */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -11901,13 +12780,13 @@ static struct hda_verb alc861_threestack_init_verbs[] = { {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - + /* Unmute Mixer 14 (mic) 1c (Line in)*/ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - + /* Unmute Stereo Mixer 15 */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -11963,13 +12842,13 @@ static struct hda_verb alc861_uniwill_m31_init_verbs[] = { {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - + /* Unmute Mixer 14 (mic) 1c (Line in)*/ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - + /* Unmute Stereo Mixer 15 */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -12034,7 +12913,7 @@ static struct hda_verb alc861_asus_init_verbs[] = { {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - + /* Unmute Stereo Mixer 15 */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -12071,20 +12950,20 @@ static struct hda_verb alc861_auto_init_verbs[] = { */ /* {0x08, AC_VERB_SET_CONNECT_SEL, 0x00}, */ {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - + /* Unmute DAC0~3 & spdif out*/ {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - + /* Unmute Mixer 14 (mic) 1c (Line in)*/ {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x014, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x01c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - + /* Unmute Stereo Mixer 15 */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -12442,7 +13321,7 @@ static void alc861_auto_init(struct hda_codec *codec) alc861_auto_init_hp_out(codec); alc861_auto_init_analog_input(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -12659,7 +13538,7 @@ static int patch_alc861(struct hda_codec *codec) if (!spec->loopback.amplist) spec->loopback.amplist = alc861_loopbacks; #endif - + return 0; } @@ -12913,7 +13792,7 @@ static struct snd_kcontrol_new alc861vd_hp_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("ATAPI Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("ATAPI Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - + { } /* end */ }; @@ -13058,7 +13937,7 @@ static struct hda_verb alc861vd_lenovo_unsol_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, - {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, {} }; @@ -13120,7 +13999,7 @@ static struct hda_verb alc861vd_dallas_verbs[] = { {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, - + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -13145,7 +14024,7 @@ static struct hda_verb alc861vd_dallas_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, - {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x09, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, { } /* end */ @@ -13304,7 +14183,7 @@ static struct alc_config_preset alc861vd_presets[] = { .input_mux = &alc861vd_hp_capture_source, .unsol_event = alc861vd_dallas_unsol_event, .init_hook = alc861vd_dallas_automute, - }, + }, }; /* @@ -13554,7 +14433,7 @@ static void alc861vd_auto_init(struct hda_codec *codec) alc861vd_auto_init_analog_input(codec); alc861vd_auto_init_input_src(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } static int patch_alc861vd(struct hda_codec *codec) @@ -13883,13 +14762,120 @@ static struct snd_kcontrol_new alc662_eeepc_ep20_mixer[] = { { } /* end */ }; +static struct hda_bind_ctls alc663_asus_bind_master_vol = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x03, 3, 0, HDA_OUTPUT), + 0 + }, +}; + +static struct hda_bind_ctls alc663_asus_one_bind_switch = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), + 0 + }, +}; + static struct snd_kcontrol_new alc663_m51va_mixer[] = { + HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol), + HDA_BIND_SW("Master Playback Switch", &alc663_asus_one_bind_switch), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + { } /* end */ +}; + +static struct hda_bind_ctls alc663_asus_tree_bind_switch = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT), + 0 + }, +}; + +static struct snd_kcontrol_new alc663_two_hp_m1_mixer[] = { + HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol), + HDA_BIND_SW("Master Playback Switch", &alc663_asus_tree_bind_switch), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + + { } /* end */ +}; + +static struct hda_bind_ctls alc663_asus_four_bind_switch = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x15, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), + 0 + }, +}; + +static struct snd_kcontrol_new alc663_two_hp_m2_mixer[] = { + HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol), + HDA_BIND_SW("Master Playback Switch", &alc663_asus_four_bind_switch), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new alc662_1bjd_mixer[] = { HDA_CODEC_VOLUME("Speaker Playback Volume", 0x02, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Speaker Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 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("F-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("F-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + { } /* end */ +}; + +static struct hda_bind_ctls alc663_asus_two_bind_master_vol = { + .ops = &snd_hda_bind_vol, + .values = { + HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x04, 3, 0, HDA_OUTPUT), + 0 + }, +}; + +static struct hda_bind_ctls alc663_asus_two_bind_switch = { + .ops = &snd_hda_bind_sw, + .values = { + HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT), + HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_OUTPUT), + 0 + }, +}; + +static struct snd_kcontrol_new alc663_asus_21jd_clfe_mixer[] = { + HDA_BIND_VOL("Master Playback Volume", + &alc663_asus_two_bind_master_vol), + HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x21, 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_MUTE("DMic Playback Switch", 0x23, 0x9, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new alc663_asus_15jd_clfe_mixer[] = { + HDA_BIND_VOL("Master Playback Volume", &alc663_asus_bind_master_vol), + HDA_BIND_SW("Master Playback Switch", &alc663_asus_two_bind_switch), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 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), { } /* end */ }; @@ -14074,14 +15060,81 @@ static struct hda_verb alc663_auto_init_verbs[] = { }; static struct hda_verb alc663_m51va_init_verbs[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, - {0x21, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */ + {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(9)}, +static struct hda_verb alc663_21jd_amic_init_verbs[] = { + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; +static struct hda_verb alc662_1bjd_amic_init_verbs[] = { + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Headphone */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + +static struct hda_verb alc663_15jd_amic_init_verbs[] = { + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + +static struct hda_verb alc663_two_hp_amic_m1_init_verbs[] = { + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x21, AC_VERB_SET_CONNECT_SEL, 0x0}, /* Headphone */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x0}, /* Headphone */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, {0x21, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + +static struct hda_verb alc663_two_hp_amic_m2_init_verbs[] = { + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */ + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Headphone */ + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, {} }; @@ -14110,6 +15163,14 @@ static struct hda_verb alc663_g50v_init_verbs[] = { {} }; +static struct hda_verb alc662_ecs_init_verbs[] = { + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, + {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_MIC_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {} +}; + /* capture mixer elements */ static struct snd_kcontrol_new alc662_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), @@ -14129,6 +15190,12 @@ static struct snd_kcontrol_new alc662_capture_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc662_auto_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), + { } /* end */ +}; + static void alc662_lenovo_101e_ispeaker_automute(struct hda_codec *codec) { unsigned int present; @@ -14209,12 +15276,12 @@ static void alc662_eeepc_ep20_automute(struct hda_codec *codec) if (present) { /* mute internal speaker */ snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, HDA_AMP_MUTE); + HDA_AMP_MUTE, HDA_AMP_MUTE); } else { /* unmute internal speaker if necessary */ mute = snd_hda_codec_amp_read(codec, 0x14, 0, HDA_OUTPUT, 0); snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, - HDA_AMP_MUTE, mute); + HDA_AMP_MUTE, mute); } } @@ -14237,11 +15304,108 @@ static void alc663_m51va_speaker_automute(struct hda_codec *codec) unsigned char bits; present = snd_hda_codec_read(codec, 0x21, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; bits = present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, + AMP_IN_MUTE(0), bits); + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, + AMP_IN_MUTE(0), bits); +} + +static void alc663_21jd_two_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x21, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, + AMP_IN_MUTE(0), bits); + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, + AMP_IN_MUTE(0), bits); + snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0, + AMP_IN_MUTE(0), bits); + snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1, + AMP_IN_MUTE(0), bits); +} + +static void alc663_15jd_two_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, + AMP_IN_MUTE(0), bits); + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, + AMP_IN_MUTE(0), bits); + snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0, + AMP_IN_MUTE(0), bits); + snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1, + AMP_IN_MUTE(0), bits); +} + +static void alc662_f5z_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + bits = present ? 0 : PIN_OUT; + snd_hda_codec_write(codec, 0x14, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, bits); +} + +static void alc663_two_hp_m1_speaker_automute(struct hda_codec *codec) +{ + unsigned int present1, present2; + + present1 = snd_hda_codec_read(codec, 0x21, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + present2 = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + + if (present1 || present2) { + snd_hda_codec_write_cache(codec, 0x14, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + } else { + snd_hda_codec_write_cache(codec, 0x14, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + } +} + +static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec) +{ + unsigned int present1, present2; + + present1 = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + present2 = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; + + if (present1 || present2) { + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, + AMP_IN_MUTE(0), AMP_IN_MUTE(0)); + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, + AMP_IN_MUTE(0), AMP_IN_MUTE(0)); + } else { + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, + AMP_IN_MUTE(0), 0); + snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, + AMP_IN_MUTE(0), 0); + } } static void alc663_m51va_mic_automute(struct hda_codec *codec) @@ -14249,16 +15413,16 @@ static void alc663_m51va_mic_automute(struct hda_codec *codec) unsigned int present; present = snd_hda_codec_read(codec, 0x18, 0, - AC_VERB_GET_PIN_SENSE, 0) - & AC_PINSENSE_PRESENCE; + AC_VERB_GET_PIN_SENSE, 0) + & AC_PINSENSE_PRESENCE; snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); + 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); + 0x7000 | (0x00 << 8) | (present ? 0 : 0x80)); snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x09 << 8) | (present ? 0x80 : 0)); + 0x7000 | (0x09 << 8) | (present ? 0x80 : 0)); snd_hda_codec_write_cache(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE, - 0x7000 | (0x09 << 8) | (present ? 0x80 : 0)); + 0x7000 | (0x09 << 8) | (present ? 0x80 : 0)); } static void alc663_m51va_unsol_event(struct hda_codec *codec, @@ -14280,6 +15444,121 @@ static void alc663_m51va_inithook(struct hda_codec *codec) alc663_m51va_mic_automute(codec); } +/* ***************** Mode1 ******************************/ +static void alc663_mode1_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc663_m51va_speaker_automute(codec); + break; + case ALC880_MIC_EVENT: + alc662_eeepc_mic_automute(codec); + break; + } +} + +static void alc663_mode1_inithook(struct hda_codec *codec) +{ + alc663_m51va_speaker_automute(codec); + alc662_eeepc_mic_automute(codec); +} +/* ***************** Mode2 ******************************/ +static void alc662_mode2_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc662_f5z_speaker_automute(codec); + break; + case ALC880_MIC_EVENT: + alc662_eeepc_mic_automute(codec); + break; + } +} + +static void alc662_mode2_inithook(struct hda_codec *codec) +{ + alc662_f5z_speaker_automute(codec); + alc662_eeepc_mic_automute(codec); +} +/* ***************** Mode3 ******************************/ +static void alc663_mode3_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc663_two_hp_m1_speaker_automute(codec); + break; + case ALC880_MIC_EVENT: + alc662_eeepc_mic_automute(codec); + break; + } +} + +static void alc663_mode3_inithook(struct hda_codec *codec) +{ + alc663_two_hp_m1_speaker_automute(codec); + alc662_eeepc_mic_automute(codec); +} +/* ***************** Mode4 ******************************/ +static void alc663_mode4_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc663_21jd_two_speaker_automute(codec); + break; + case ALC880_MIC_EVENT: + alc662_eeepc_mic_automute(codec); + break; + } +} + +static void alc663_mode4_inithook(struct hda_codec *codec) +{ + alc663_21jd_two_speaker_automute(codec); + alc662_eeepc_mic_automute(codec); +} +/* ***************** Mode5 ******************************/ +static void alc663_mode5_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc663_15jd_two_speaker_automute(codec); + break; + case ALC880_MIC_EVENT: + alc662_eeepc_mic_automute(codec); + break; + } +} + +static void alc663_mode5_inithook(struct hda_codec *codec) +{ + alc663_15jd_two_speaker_automute(codec); + alc662_eeepc_mic_automute(codec); +} +/* ***************** Mode6 ******************************/ +static void alc663_mode6_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc663_two_hp_m2_speaker_automute(codec); + break; + case ALC880_MIC_EVENT: + alc662_eeepc_mic_automute(codec); + break; + } +} + +static void alc663_mode6_inithook(struct hda_codec *codec) +{ + alc663_two_hp_m2_speaker_automute(codec); + alc662_eeepc_mic_automute(codec); +} + static void alc663_g71v_hp_automute(struct hda_codec *codec) { unsigned int present; @@ -14350,6 +15629,46 @@ static void alc663_g50v_inithook(struct hda_codec *codec) alc662_eeepc_mic_automute(codec); } +/* bind hp and internal speaker mute (with plug check) */ +static int alc662_ecs_master_sw_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + long *valp = ucontrol->value.integer.value; + int change; + + change = snd_hda_codec_amp_update(codec, 0x1b, 0, HDA_OUTPUT, 0, + HDA_AMP_MUTE, + valp[0] ? 0 : HDA_AMP_MUTE); + change |= snd_hda_codec_amp_update(codec, 0x1b, 1, HDA_OUTPUT, 0, + HDA_AMP_MUTE, + valp[1] ? 0 : HDA_AMP_MUTE); + if (change) + alc262_hippo1_automute(codec); + return change; +} + +static struct snd_kcontrol_new alc662_ecs_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = alc662_ecs_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT), + }, + + HDA_CODEC_VOLUME("e-Mic/LineIn Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("e-Mic/LineIn Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("e-Mic/LineIn Playback Switch", 0x0b, 0x0, HDA_INPUT), + + HDA_CODEC_VOLUME("i-Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_VOLUME("i-Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("i-Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + { } /* end */ +}; + #ifdef CONFIG_SND_HDA_POWER_SAVE #define alc662_loopbacks alc880_loopbacks #endif @@ -14372,21 +15691,68 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { [ALC662_LENOVO_101E] = "lenovo-101e", [ALC662_ASUS_EEEPC_P701] = "eeepc-p701", [ALC662_ASUS_EEEPC_EP20] = "eeepc-ep20", + [ALC662_ECS] = "ecs", [ALC663_ASUS_M51VA] = "m51va", [ALC663_ASUS_G71V] = "g71v", [ALC663_ASUS_H13] = "h13", [ALC663_ASUS_G50V] = "g50v", + [ALC663_ASUS_MODE1] = "asus-mode1", + [ALC662_ASUS_MODE2] = "asus-mode2", + [ALC663_ASUS_MODE3] = "asus-mode3", + [ALC663_ASUS_MODE4] = "asus-mode4", + [ALC663_ASUS_MODE5] = "asus-mode5", + [ALC663_ASUS_MODE6] = "asus-mode6", [ALC662_AUTO] = "auto", }; static struct snd_pci_quirk alc662_cfg_tbl[] = { - SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS G71V", ALC663_ASUS_G71V), SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA), - SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS M51VA", ALC663_ASUS_G50V), + SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V), SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20), + SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1813, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1864, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x19f3, "ASUS NB", ALC663_ASUS_MODE4), + SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5), + SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", + ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), + SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS), + SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", + ALC662_3ST_6ch_DIG), + SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG), + SND_PCI_QUIRK(0x1849, 0x3662, "ASROCK K10N78FullHD-hSLI R3.0", + ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1854, 0x2000, "ASUS H13-2000", ALC663_ASUS_H13), SND_PCI_QUIRK(0x1854, 0x2001, "ASUS H13-2001", ALC663_ASUS_H13), SND_PCI_QUIRK(0x1854, 0x2002, "ASUS H13-2002", ALC663_ASUS_H13), @@ -14477,6 +15843,18 @@ static struct alc_config_preset alc662_presets[] = { .unsol_event = alc662_eeepc_ep20_unsol_event, .init_hook = alc662_eeepc_ep20_inithook, }, + [ALC662_ECS] = { + .mixers = { alc662_ecs_mixer, alc662_capture_mixer }, + .init_verbs = { alc662_init_verbs, + alc662_ecs_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .dac_nids = alc662_dac_nids, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc662_eeepc_capture_source, + .unsol_event = alc662_eeepc_unsol_event, + .init_hook = alc662_eeepc_inithook, + }, [ALC663_ASUS_M51VA] = { .mixers = { alc663_m51va_mixer, alc662_capture_mixer}, .init_verbs = { alc662_init_verbs, alc663_m51va_init_verbs }, @@ -14524,6 +15902,91 @@ static struct alc_config_preset alc662_presets[] = { .unsol_event = alc663_g50v_unsol_event, .init_hook = alc663_g50v_inithook, }, + [ALC663_ASUS_MODE1] = { + .mixers = { alc663_m51va_mixer, alc662_auto_capture_mixer }, + .init_verbs = { alc662_init_verbs, + alc663_21jd_amic_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .hp_nid = 0x03, + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc662_eeepc_capture_source, + .unsol_event = alc663_mode1_unsol_event, + .init_hook = alc663_mode1_inithook, + }, + [ALC662_ASUS_MODE2] = { + .mixers = { alc662_1bjd_mixer, alc662_auto_capture_mixer }, + .init_verbs = { alc662_init_verbs, + alc662_1bjd_amic_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc662_eeepc_capture_source, + .unsol_event = alc662_mode2_unsol_event, + .init_hook = alc662_mode2_inithook, + }, + [ALC663_ASUS_MODE3] = { + .mixers = { alc663_two_hp_m1_mixer, alc662_auto_capture_mixer }, + .init_verbs = { alc662_init_verbs, + alc663_two_hp_amic_m1_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .hp_nid = 0x03, + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc662_eeepc_capture_source, + .unsol_event = alc663_mode3_unsol_event, + .init_hook = alc663_mode3_inithook, + }, + [ALC663_ASUS_MODE4] = { + .mixers = { alc663_asus_21jd_clfe_mixer, + alc662_auto_capture_mixer}, + .init_verbs = { alc662_init_verbs, + alc663_21jd_amic_init_verbs}, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .hp_nid = 0x03, + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc662_eeepc_capture_source, + .unsol_event = alc663_mode4_unsol_event, + .init_hook = alc663_mode4_inithook, + }, + [ALC663_ASUS_MODE5] = { + .mixers = { alc663_asus_15jd_clfe_mixer, + alc662_auto_capture_mixer }, + .init_verbs = { alc662_init_verbs, + alc663_15jd_amic_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .hp_nid = 0x03, + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc662_eeepc_capture_source, + .unsol_event = alc663_mode5_unsol_event, + .init_hook = alc663_mode5_inithook, + }, + [ALC663_ASUS_MODE6] = { + .mixers = { alc663_two_hp_m2_mixer, alc662_auto_capture_mixer }, + .init_verbs = { alc662_init_verbs, + alc663_two_hp_amic_m2_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .hp_nid = 0x03, + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .input_mux = &alc662_eeepc_capture_source, + .unsol_event = alc663_mode6_unsol_event, + .init_hook = alc663_mode6_inithook, + }, }; @@ -14560,15 +16023,15 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec, HDA_OUTPUT)); if (err < 0) return err; - err = add_control(spec, ALC_CTL_BIND_MUTE, + err = add_control(spec, ALC_CTL_WIDGET_MUTE, "Center Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 1, 2, + HDA_COMPOSE_AMP_VAL(0x0e, 1, 0, HDA_INPUT)); if (err < 0) return err; - err = add_control(spec, ALC_CTL_BIND_MUTE, + err = add_control(spec, ALC_CTL_WIDGET_MUTE, "LFE Playback Switch", - HDA_COMPOSE_AMP_VAL(nid, 2, 2, + HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, HDA_INPUT)); if (err < 0) return err; @@ -14580,9 +16043,9 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec, if (err < 0) return err; sprintf(name, "%s Playback Switch", chname[i]); - err = add_control(spec, ALC_CTL_BIND_MUTE, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 2, - HDA_INPUT)); + err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(alc880_idx_to_mixer(i), + 3, 0, HDA_INPUT)); if (err < 0) return err; } @@ -14777,7 +16240,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec) spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux; - + spec->init_verbs[spec->num_init_verbs++] = alc662_auto_init_verbs; if (codec->vendor_id == 0x10ec0663) spec->init_verbs[spec->num_init_verbs++] = @@ -14801,7 +16264,7 @@ static void alc662_auto_init(struct hda_codec *codec) alc662_auto_init_analog_input(codec); alc662_auto_init_input_src(codec); if (spec->unsol_event) - alc_sku_automute(codec); + alc_inithook(codec); } static int patch_alc662(struct hda_codec *codec) @@ -14846,6 +16309,9 @@ static int patch_alc662(struct hda_codec *codec) if (codec->vendor_id == 0x10ec0663) { spec->stream_name_analog = "ALC663 Analog"; spec->stream_name_digital = "ALC663 Digital"; + } else if (codec->vendor_id == 0x10ec0272) { + spec->stream_name_analog = "ALC272 Analog"; + spec->stream_name_digital = "ALC272 Digital"; } else { spec->stream_name_analog = "ALC662 Analog"; spec->stream_name_digital = "ALC662 Digital"; @@ -14883,6 +16349,7 @@ struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 }, { .id = 0x10ec0268, .name = "ALC268", .patch = patch_alc268 }, { .id = 0x10ec0269, .name = "ALC269", .patch = patch_alc269 }, + { .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", .patch = patch_alc861 }, { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd }, @@ -14896,10 +16363,15 @@ struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 }, { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 }, { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc883 }, + { .id = 0x10ec0885, .rev = 0x100101, .name = "ALC889A", + .patch = patch_alc882 }, /* should be patch_alc883() in future */ { .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A", .patch = patch_alc882 }, /* should be patch_alc883() in future */ { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 }, + { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc883 }, { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 }, + { .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200", + .patch = patch_alc883 }, { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 }, {} /* terminator */ }; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ad994fcab72..a2ac7205d45 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -33,10 +33,12 @@ #include "hda_codec.h" #include "hda_local.h" #include "hda_patch.h" +#include "hda_beep.h" #define NUM_CONTROL_ALLOC 32 #define STAC_PWR_EVENT 0x20 #define STAC_HP_EVENT 0x30 +#define STAC_VREF_EVENT 0x40 enum { STAC_REF, @@ -71,9 +73,15 @@ enum { }; enum { + STAC_92HD83XXX_REF, + STAC_92HD83XXX_MODELS +}; + +enum { STAC_92HD71BXX_REF, STAC_DELL_M4_1, STAC_DELL_M4_2, + STAC_HP_M4, STAC_92HD71BXX_MODELS }; @@ -104,6 +112,7 @@ enum { STAC_MACBOOK_PRO_V2, STAC_IMAC_INTEL, STAC_IMAC_INTEL_20, + STAC_ECS_202, STAC_922X_DELL_D81, STAC_922X_DELL_D82, STAC_922X_DELL_M81, @@ -130,6 +139,7 @@ struct sigmatel_spec { unsigned int mic_switch: 1; unsigned int alt_switch: 1; unsigned int hp_detect: 1; + unsigned int spdif_mute: 1; /* gpio lines */ unsigned int eapd_mask; @@ -138,17 +148,22 @@ struct sigmatel_spec { unsigned int gpio_data; unsigned int gpio_mute; + /* stream */ + unsigned int stream_delay; + /* analog loopback */ unsigned char aloopback_mask; unsigned char aloopback_shift; /* power management */ unsigned int num_pwrs; + unsigned int *pwr_mapping; hda_nid_t *pwr_nids; hda_nid_t *dac_list; /* playback */ struct hda_input_mux *mono_mux; + struct hda_input_mux *amp_mux; unsigned int cur_mmux; struct hda_multi_out multiout; hda_nid_t dac_nids[5]; @@ -162,8 +177,14 @@ struct sigmatel_spec { unsigned int num_dmics; hda_nid_t *dmux_nids; unsigned int num_dmuxes; + hda_nid_t *smux_nids; + unsigned int num_smuxes; + const char **spdif_labels; + hda_nid_t dig_in_nid; hda_nid_t mono_nid; + hda_nid_t anabeep_nid; + hda_nid_t digbeep_nid; /* pin widgets */ hda_nid_t *pin_nids; @@ -180,6 +201,12 @@ struct sigmatel_spec { unsigned int cur_dmux[2]; struct hda_input_mux *input_mux; unsigned int cur_mux[3]; + struct hda_input_mux *sinput_mux; + unsigned int cur_smux[2]; + unsigned int cur_amux; + hda_nid_t *amp_nids; + unsigned int num_amps; + unsigned int powerdown_adcs; /* i/o switches */ unsigned int io_switch[2]; @@ -195,6 +222,8 @@ struct sigmatel_spec { struct snd_kcontrol_new *kctl_alloc; struct hda_input_mux private_dimux; struct hda_input_mux private_imux; + struct hda_input_mux private_smux; + struct hda_input_mux private_amp_mux; struct hda_input_mux private_mono_mux; }; @@ -215,10 +244,19 @@ static hda_nid_t stac92hd73xx_pwr_nids[8] = { 0x0f, 0x10, 0x11 }; +static hda_nid_t stac92hd73xx_slave_dig_outs[2] = { + 0x26, 0, +}; + static hda_nid_t stac92hd73xx_adc_nids[2] = { 0x1a, 0x1b }; +#define DELL_M6_AMP 2 +static hda_nid_t stac92hd73xx_amp_nids[3] = { + 0x0b, 0x0c, 0x0e +}; + #define STAC92HD73XX_NUM_DMICS 2 static hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = { 0x13, 0x14, 0 @@ -237,6 +275,41 @@ static hda_nid_t stac92hd73xx_dmux_nids[2] = { 0x20, 0x21, }; +static hda_nid_t stac92hd73xx_smux_nids[2] = { + 0x22, 0x23, +}; + +#define STAC92HD83XXX_NUM_DMICS 2 +static hda_nid_t stac92hd83xxx_dmic_nids[STAC92HD83XXX_NUM_DMICS + 1] = { + 0x11, 0x12, 0 +}; + +#define STAC92HD81_DAC_COUNT 2 +#define STAC92HD83_DAC_COUNT 3 +static hda_nid_t stac92hd83xxx_dac_nids[STAC92HD73_DAC_COUNT] = { + 0x13, 0x14, 0x22, +}; + +static hda_nid_t stac92hd83xxx_dmux_nids[2] = { + 0x17, 0x18, +}; + +static hda_nid_t stac92hd83xxx_adc_nids[2] = { + 0x15, 0x16, +}; + +static hda_nid_t stac92hd83xxx_pwr_nids[4] = { + 0xa, 0xb, 0xd, 0xe, +}; + +static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = { + 0x1e, 0, +}; + +static unsigned int stac92hd83xxx_pwr_mapping[4] = { + 0x03, 0x0c, 0x10, 0x40, +}; + static hda_nid_t stac92hd71bxx_pwr_nids[3] = { 0x0a, 0x0d, 0x0f }; @@ -249,8 +322,12 @@ static hda_nid_t stac92hd71bxx_mux_nids[2] = { 0x1a, 0x1b }; -static hda_nid_t stac92hd71bxx_dmux_nids[1] = { - 0x1c, +static hda_nid_t stac92hd71bxx_dmux_nids[2] = { + 0x1c, 0x1d, +}; + +static hda_nid_t stac92hd71bxx_smux_nids[2] = { + 0x24, 0x25, }; static hda_nid_t stac92hd71bxx_dac_nids[1] = { @@ -262,6 +339,10 @@ static hda_nid_t stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS + 1] = { 0x18, 0x19, 0 }; +static hda_nid_t stac92hd71bxx_slave_dig_outs[2] = { + 0x22, 0 +}; + static hda_nid_t stac925x_adc_nids[1] = { 0x03, }; @@ -299,6 +380,10 @@ static hda_nid_t stac927x_mux_nids[3] = { 0x15, 0x16, 0x17 }; +static hda_nid_t stac927x_smux_nids[1] = { + 0x21, +}; + static hda_nid_t stac927x_dac_nids[6] = { 0x02, 0x03, 0x04, 0x05, 0x06, 0 }; @@ -312,6 +397,11 @@ static hda_nid_t stac927x_dmic_nids[STAC927X_NUM_DMICS + 1] = { 0x13, 0x14, 0 }; +static const char *stac927x_spdif_labels[5] = { + "Digital Playback", "ADAT", "Analog Mux 1", + "Analog Mux 2", "Analog Mux 3" +}; + static hda_nid_t stac9205_adc_nids[2] = { 0x12, 0x13 }; @@ -324,6 +414,10 @@ static hda_nid_t stac9205_dmux_nids[1] = { 0x1d, }; +static hda_nid_t stac9205_smux_nids[1] = { + 0x21, +}; + #define STAC9205_NUM_DMICS 2 static hda_nid_t stac9205_dmic_nids[STAC9205_NUM_DMICS + 1] = { 0x17, 0x18, 0 @@ -347,12 +441,18 @@ static hda_nid_t stac922x_pin_nids[10] = { static hda_nid_t stac92hd73xx_pin_nids[13] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12, 0x13, - 0x14, 0x1e, 0x22 + 0x14, 0x22, 0x23 }; -static hda_nid_t stac92hd71bxx_pin_nids[10] = { +static hda_nid_t stac92hd83xxx_pin_nids[14] = { + 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, + 0x0f, 0x10, 0x11, 0x12, 0x13, + 0x1d, 0x1e, 0x1f, 0x20 +}; +static hda_nid_t stac92hd71bxx_pin_nids[11] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x14, 0x18, 0x19, 0x1e, + 0x1f, }; static hda_nid_t stac927x_pin_nids[14] = { @@ -367,6 +467,34 @@ static hda_nid_t stac9205_pin_nids[12] = { 0x21, 0x22, }; +#define stac92xx_amp_volume_info snd_hda_mixer_amp_volume_info + +static int stac92xx_amp_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = spec->amp_nids[spec->cur_amux]; + + kcontrol->private_value ^= get_amp_nid(kcontrol); + kcontrol->private_value |= nid; + + return snd_hda_mixer_amp_volume_get(kcontrol, ucontrol); +} + +static int stac92xx_amp_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = spec->amp_nids[spec->cur_amux]; + + kcontrol->private_value ^= get_amp_nid(kcontrol); + kcontrol->private_value |= nid; + + return snd_hda_mixer_amp_volume_put(kcontrol, ucontrol); +} + static int stac92xx_dmux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -397,6 +525,58 @@ static int stac92xx_dmux_enum_put(struct snd_kcontrol *kcontrol, spec->dmux_nids[dmux_idx], &spec->cur_dmux[dmux_idx]); } +static int stac92xx_smux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + return snd_hda_input_mux_info(spec->sinput_mux, uinfo); +} + +static int stac92xx_smux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + unsigned int smux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + + ucontrol->value.enumerated.item[0] = spec->cur_smux[smux_idx]; + return 0; +} + +static int stac92xx_smux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + struct hda_input_mux *smux = &spec->private_smux; + unsigned int smux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + int err, val; + hda_nid_t nid; + + err = snd_hda_input_mux_put(codec, spec->sinput_mux, ucontrol, + spec->smux_nids[smux_idx], &spec->cur_smux[smux_idx]); + if (err < 0) + return err; + + if (spec->spdif_mute) { + if (smux_idx == 0) + nid = spec->multiout.dig_out_nid; + else + nid = codec->slave_dig_outs[smux_idx - 1]; + if (spec->cur_smux[smux_idx] == smux->num_items - 1) + val = AMP_OUT_MUTE; + if (smux_idx == 0) + nid = spec->multiout.dig_out_nid; + else + nid = codec->slave_dig_outs[smux_idx - 1]; + /* un/mute SPDIF out */ + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, val); + } + return 0; +} + static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); @@ -452,6 +632,41 @@ static int stac92xx_mono_mux_enum_put(struct snd_kcontrol *kcontrol, spec->mono_nid, &spec->cur_mmux); } +static int stac92xx_amp_mux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + return snd_hda_input_mux_info(spec->amp_mux, uinfo); +} + +static int stac92xx_amp_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_amux; + return 0; +} + +static int stac92xx_amp_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + struct snd_kcontrol *ctl = + snd_hda_find_mixer_ctl(codec, "Amp Capture Volume"); + if (!ctl) + return -EINVAL; + + snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE | + SNDRV_CTL_EVENT_MASK_INFO, &ctl->id); + + return snd_hda_input_mux_put(codec, spec->amp_mux, ucontrol, + 0, &spec->cur_amux); +} + #define stac92xx_aloopback_info snd_ctl_boolean_mono_info static int stac92xx_aloopback_get(struct snd_kcontrol *kcontrol, @@ -546,8 +761,8 @@ static struct hda_verb dell_eq_core_init[] = { { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec}, /* setup audio connections */ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, - { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02}, + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x02}, + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x01}, /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, @@ -628,25 +843,36 @@ static struct hda_verb stac92hd73xx_10ch_core_init[] = { {} }; +static struct hda_verb stac92hd83xxx_core_init[] = { + /* start of config #1 */ + { 0xe, AC_VERB_SET_CONNECT_SEL, 0x3}, + + /* start of config #2 */ + { 0xa, AC_VERB_SET_CONNECT_SEL, 0x0}, + { 0xb, AC_VERB_SET_CONNECT_SEL, 0x0}, + { 0xd, AC_VERB_SET_CONNECT_SEL, 0x1}, + + /* power state controls amps */ + { 0x01, AC_VERB_SET_EAPD, 1 << 2}, +}; + static struct hda_verb stac92hd71bxx_core_init[] = { /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, }; -#define HD_DISABLE_PORTF 3 +#define HD_DISABLE_PORTF 2 static struct hda_verb stac92hd71bxx_analog_core_init[] = { /* start of config #1 */ /* connect port 0f to audio mixer */ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ /* unmute right and left channels for node 0x0f */ { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* start of config #2 */ @@ -655,10 +881,6 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = { { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* connect port 0d to audio mixer */ - { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2}, - /* unmute dac0 input in audio mixer */ - { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, /* unmute right and left channels for nodes 0x0a, 0xd */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -690,12 +912,16 @@ static struct hda_verb d965_core_init[] = { static struct hda_verb stac927x_core_init[] = { /* set master volume and direct control */ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* enable analog pc beep path */ + { 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, {} }; static struct hda_verb stac9205_core_init[] = { /* set master volume and direct control */ { 0x24, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* enable analog pc beep path */ + { 0x01, AC_VERB_SET_DIGI_CONVERT_2, 1 << 5}, {} }; @@ -709,6 +935,31 @@ static struct hda_verb stac9205_core_init[] = { .put = stac92xx_mono_mux_enum_put, \ } +#define STAC_AMP_MUX \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Amp Selector Capture Switch", \ + .count = 1, \ + .info = stac92xx_amp_mux_enum_info, \ + .get = stac92xx_amp_mux_enum_get, \ + .put = stac92xx_amp_mux_enum_put, \ + } + +#define STAC_AMP_VOL(xname, nid, chs, idx, dir) \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = 0, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ + .info = stac92xx_amp_volume_info, \ + .get = stac92xx_amp_volume_get, \ + .put = stac92xx_amp_volume_put, \ + .tlv = { .c = snd_hda_mixer_amp_tlv }, \ + .private_value = HDA_COMPOSE_AMP_VAL(nid, chs, idx, dir) \ + } + #define STAC_INPUT_SOURCE(cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -736,33 +987,36 @@ static struct snd_kcontrol_new stac9200_mixer[] = { STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Mux Volume", 0x0c, 0, HDA_OUTPUT), { } /* end */ }; +#define DELL_M6_MIXER 6 static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), - + /* start of config #1 */ HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), - HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT), HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT), + HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + + /* start of config #2 */ + HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + { } /* end */ }; @@ -818,22 +1072,59 @@ static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { { } /* end */ }; + +static struct snd_kcontrol_new stac92hd83xxx_mixer[] = { + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x1b, 0, HDA_INPUT), + HDA_CODEC_MUTE("DAC0 Capture Switch", 0x1b, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x1b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("DAC1 Capture Switch", 0x1b, 0x1, HDA_INPUT), + + HDA_CODEC_VOLUME("Front Mic Capture Volume", 0x1b, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Capture Switch", 0x1b, 0x2, HDA_INPUT), + + HDA_CODEC_VOLUME("Line In Capture Volume", 0x1b, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Line In Capture Switch", 0x1b, 0x3, HDA_INPUT), + + /* + HDA_CODEC_VOLUME("Mic Capture Volume", 0x1b, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("Mic Capture Switch", 0x1b 0x4, HDA_INPUT), + */ + { } /* end */ +}; + static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { STAC_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), - + /* analog pc-beep replaced with digital beep support */ + /* HDA_CODEC_VOLUME("PC Beep Volume", 0x17, 0x2, HDA_INPUT), HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT), + */ + + HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x0, HDA_INPUT), + + HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x1, HDA_INPUT), + + HDA_CODEC_MUTE("DAC0 Capture Switch", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("DAC1 Capture Switch", 0x17, 0x4, HDA_INPUT), + HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x17, 0x4, HDA_INPUT), { } /* end */ }; @@ -843,11 +1134,9 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -855,7 +1144,6 @@ static struct snd_kcontrol_new stac925x_mixer[] = { STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Mux Volume", 0x0f, 0, HDA_OUTPUT), { } /* end */ }; @@ -865,12 +1153,9 @@ static struct snd_kcontrol_new stac9205_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x19, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1c, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1e, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x1A, 0x0, HDA_OUTPUT), - { } /* end */ }; @@ -879,11 +1164,9 @@ static struct snd_kcontrol_new stac922x_mixer[] = { STAC_INPUT_SOURCE(2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x13, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -894,15 +1177,12 @@ static struct snd_kcontrol_new stac927x_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1b, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x19, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x1, 0x16, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x2, 0x1A, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x2, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x2, 0x17, 0x0, HDA_OUTPUT), { } /* end */ }; @@ -915,6 +1195,15 @@ static struct snd_kcontrol_new stac_dmux_mixer = { .put = stac92xx_dmux_enum_put, }; +static struct snd_kcontrol_new stac_smux_mixer = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "IEC958 Playback Source", + /* count set later */ + .info = stac92xx_smux_enum_info, + .get = stac92xx_smux_enum_get, + .put = stac92xx_smux_enum_put, +}; + static const char *slave_vols[] = { "Front Playback Volume", "Surround Playback Volume", @@ -966,6 +1255,22 @@ static int stac92xx_build_controls(struct hda_codec *codec) if (err < 0) return err; } + if (spec->num_smuxes > 0) { + int wcaps = get_wcaps(codec, spec->multiout.dig_out_nid); + struct hda_input_mux *smux = &spec->private_smux; + /* check for mute support on SPDIF out */ + if (wcaps & AC_WCAP_OUT_AMP) { + smux->items[smux->num_items].label = "Off"; + smux->items[smux->num_items].index = 0; + smux->num_items++; + spec->spdif_mute = 1; + } + stac_smux_mixer.count = spec->num_smuxes; + err = snd_ctl_add(codec->bus->card, + snd_ctl_new1(&stac_smux_mixer, codec)); + if (err < 0) + return err; + } if (spec->multiout.dig_out_nid) { err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); @@ -977,7 +1282,7 @@ static int stac92xx_build_controls(struct hda_codec *codec) return err; spec->multiout.share_spdif = 1; } - if (spec->dig_in_nid) { + if (spec->dig_in_nid && (!spec->gpio_dir & 0x01)) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); if (err < 0) return err; @@ -1325,40 +1630,65 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { {} /* terminator */ }; -static unsigned int ref92hd71bxx_pin_configs[10] = { +static unsigned int ref92hd83xxx_pin_configs[14] = { + 0x02214030, 0x02211010, 0x02a19020, 0x02170130, + 0x01014050, 0x01819040, 0x01014020, 0x90a3014e, + 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x40f000f0, + 0x01451160, 0x98560170, +}; + +static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { + [STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs, +}; + +static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { + [STAC_92HD83XXX_REF] = "ref", +}; + +static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { + /* SigmaTel reference board */ + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, + "DFI LanParty", STAC_92HD71BXX_REF), +}; + +static unsigned int ref92hd71bxx_pin_configs[11] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, - 0x0181302e, 0x01114010, 0x01019020, 0x90a000f0, - 0x90a000f0, 0x01452050, + 0x0181302e, 0x01014010, 0x01019020, 0x90a000f0, + 0x90a000f0, 0x01452050, 0x01452050, }; -static unsigned int dell_m4_1_pin_configs[10] = { +static unsigned int dell_m4_1_pin_configs[11] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, - 0x40f000f0, 0x4f0000f0, + 0x40f000f0, 0x4f0000f0, 0x4f0000f0, }; -static unsigned int dell_m4_2_pin_configs[10] = { +static unsigned int dell_m4_2_pin_configs[11] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, - 0x40f000f0, 0x044413b0, + 0x40f000f0, 0x044413b0, 0x044413b0, }; static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = ref92hd71bxx_pin_configs, [STAC_DELL_M4_1] = dell_m4_1_pin_configs, [STAC_DELL_M4_2] = dell_m4_2_pin_configs, + [STAC_HP_M4] = NULL, }; static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { [STAC_92HD71BXX_REF] = "ref", [STAC_DELL_M4_1] = "dell-m4-1", [STAC_DELL_M4_2] = "dell-m4-2", + [STAC_HP_M4] = "hp-m4", }; static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD71BXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a, + "unknown HP", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, "unknown Dell", STAC_DELL_M4_1), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234, @@ -1477,6 +1807,11 @@ static unsigned int intel_mac_v5_pin_configs[10] = { 0x400000fc, 0x400000fb, }; +static unsigned int ecs202_pin_configs[10] = { + 0x0221401f, 0x02a19020, 0x01a19020, 0x01114010, + 0x408000f0, 0x01813022, 0x074510a0, 0x40c400f1, + 0x9037012e, 0x40e000f2, +}; static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { [STAC_D945_REF] = ref922x_pin_configs, @@ -1495,6 +1830,7 @@ static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { [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, + [STAC_ECS_202] = ecs202_pin_configs, [STAC_922X_DELL_D81] = dell_922x_d81_pin_configs, [STAC_922X_DELL_D82] = dell_922x_d82_pin_configs, [STAC_922X_DELL_M81] = dell_922x_m81_pin_configs, @@ -1518,6 +1854,7 @@ static const char *stac922x_models[STAC_922X_MODELS] = { [STAC_MACBOOK_PRO_V2] = "macbook-pro", [STAC_IMAC_INTEL] = "imac-intel", [STAC_IMAC_INTEL_20] = "imac-intel-20", + [STAC_ECS_202] = "ecs202", [STAC_922X_DELL_D81] = "dell-d81", [STAC_922X_DELL_D82] = "dell-d82", [STAC_922X_DELL_M81] = "dell-m81", @@ -1604,6 +1941,33 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { "unknown Dell", STAC_922X_DELL_D81), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d7, "Dell XPS M1210", STAC_922X_DELL_M82), + /* ECS/PC Chips boards */ + SND_PCI_QUIRK(0x1019, 0x2144, + "ECS/PC chips", STAC_ECS_202), + SND_PCI_QUIRK(0x1019, 0x2608, + "ECS/PC chips", STAC_ECS_202), + SND_PCI_QUIRK(0x1019, 0x2633, + "ECS/PC chips P17G/1333", STAC_ECS_202), + SND_PCI_QUIRK(0x1019, 0x2811, + "ECS/PC chips", STAC_ECS_202), + SND_PCI_QUIRK(0x1019, 0x2812, + "ECS/PC chips", STAC_ECS_202), + SND_PCI_QUIRK(0x1019, 0x2813, + "ECS/PC chips", STAC_ECS_202), + SND_PCI_QUIRK(0x1019, 0x2814, + "ECS/PC chips", STAC_ECS_202), + SND_PCI_QUIRK(0x1019, 0x2815, + "ECS/PC chips", STAC_ECS_202), + SND_PCI_QUIRK(0x1019, 0x2816, + "ECS/PC chips", STAC_ECS_202), + SND_PCI_QUIRK(0x1019, 0x2817, + "ECS/PC chips", STAC_ECS_202), + SND_PCI_QUIRK(0x1019, 0x2818, + "ECS/PC chips", STAC_ECS_202), + SND_PCI_QUIRK(0x1019, 0x2819, + "ECS/PC chips", STAC_ECS_202), + SND_PCI_QUIRK(0x1019, 0x2820, + "ECS/PC chips", STAC_ECS_202), {} /* terminator */ }; @@ -1683,8 +2047,8 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { /* Dell 3 stack systems with verb table in BIOS */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0227, "Dell Vostro 1400 ", STAC_DELL_BIOS), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022f, "Dell ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022e, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022f, "Dell Inspiron 1525", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0242, "Dell ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0243, "Dell ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ff, "Dell ", STAC_DELL_BIOS), @@ -1867,6 +2231,8 @@ static int stac92xx_playback_pcm_open(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; + if (spec->stream_delay) + msleep(spec->stream_delay); return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream, hinfo); } @@ -1930,9 +2296,14 @@ static int stac92xx_capture_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = spec->adc_nids[substream->number]; - snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], - stream_tag, 0, format); + if (spec->powerdown_adcs) { + msleep(40); + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D0); + } + snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); return 0; } @@ -1941,8 +2312,12 @@ static int stac92xx_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = spec->adc_nids[substream->number]; - snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]); + snd_hda_codec_cleanup_stream(codec, nid); + if (spec->powerdown_adcs) + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); return 0; } @@ -2193,6 +2568,8 @@ enum { STAC_CTL_WIDGET_VOL, STAC_CTL_WIDGET_MUTE, STAC_CTL_WIDGET_MONO_MUX, + STAC_CTL_WIDGET_AMP_MUX, + STAC_CTL_WIDGET_AMP_VOL, STAC_CTL_WIDGET_HP_SWITCH, STAC_CTL_WIDGET_IO_SWITCH, STAC_CTL_WIDGET_CLFE_SWITCH @@ -2202,13 +2579,16 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), STAC_MONO_MUX, + STAC_AMP_MUX, + STAC_AMP_VOL(NULL, 0, 0, 0, 0), STAC_CODEC_HP_SWITCH(NULL), STAC_CODEC_IO_SWITCH(NULL, 0), STAC_CODEC_CLFE_SWITCH(NULL, 0), }; /* add dynamic controls */ -static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char *name, unsigned long val) +static int stac92xx_add_control_idx(struct sigmatel_spec *spec, int type, + int idx, const char *name, unsigned long val) { struct snd_kcontrol_new *knew; @@ -2228,6 +2608,7 @@ static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char knew = &spec->kctl_alloc[spec->num_kctl_used]; *knew = stac92xx_control_templates[type]; + knew->index = idx; knew->name = kstrdup(name, GFP_KERNEL); if (! knew->name) return -ENOMEM; @@ -2236,6 +2617,14 @@ static int stac92xx_add_control(struct sigmatel_spec *spec, int type, const char return 0; } + +/* add dynamic controls */ +static int stac92xx_add_control(struct sigmatel_spec *spec, int type, + const char *name, unsigned long val) +{ + return stac92xx_add_control_idx(spec, type, 0, name, val); +} + /* flag inputs as additional dynamic lineouts */ static int stac92xx_add_dyn_out_pins(struct hda_codec *codec, struct auto_pin_cfg *cfg) { @@ -2427,7 +2816,7 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; - hda_nid_t nid; + hda_nid_t nid = 0; int i, err; struct sigmatel_spec *spec = codec->spec; @@ -2467,6 +2856,10 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, } } + if ((spec->multiout.num_dacs - cfg->line_outs) > 0 && + cfg->hp_outs && !spec->multiout.hp_nid) + spec->multiout.hp_nid = nid; + if (cfg->hp_outs > 1) { err = stac92xx_add_control(spec, STAC_CTL_WIDGET_HP_SWITCH, @@ -2579,8 +2972,8 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, } /* labels for mono mux outputs */ -static const char *stac92xx_mono_labels[3] = { - "DAC0", "DAC1", "Mixer" +static const char *stac92xx_mono_labels[4] = { + "DAC0", "DAC1", "Mixer", "DAC2" }; /* create mono mux for mono out on capable codecs */ @@ -2609,6 +3002,116 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec) "Mono Mux", spec->mono_nid); } +/* labels for amp mux outputs */ +static const char *stac92xx_amp_labels[3] = { + "Front Microphone", "Microphone", "Line In", +}; + +/* create amp out controls mux on capable codecs */ +static int stac92xx_auto_create_amp_output_ctls(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + struct hda_input_mux *amp_mux = &spec->private_amp_mux; + int i, err; + + for (i = 0; i < spec->num_amps; i++) { + amp_mux->items[amp_mux->num_items].label = + stac92xx_amp_labels[i]; + amp_mux->items[amp_mux->num_items].index = i; + amp_mux->num_items++; + } + + if (spec->num_amps > 1) { + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_MUX, + "Amp Selector Capture Switch", 0); + if (err < 0) + return err; + } + return stac92xx_add_control(spec, STAC_CTL_WIDGET_AMP_VOL, + "Amp Capture Volume", + HDA_COMPOSE_AMP_VAL(spec->amp_nids[0], 3, 0, HDA_INPUT)); +} + + +/* create PC beep volume controls */ +static int stac92xx_auto_create_beep_ctls(struct hda_codec *codec, + hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + u32 caps = query_amp_caps(codec, nid, HDA_OUTPUT); + int err; + + /* check for mute support for the the amp */ + if ((caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT) { + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, + "PC Beep Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT)); + if (err < 0) + return err; + } + + /* check to see if there is volume support for the amp */ + if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) { + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, + "PC Beep Playback Volume", + HDA_COMPOSE_AMP_VAL(nid, 1, 0, HDA_OUTPUT)); + if (err < 0) + return err; + } + return 0; +} + +static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + int wcaps, nid, i, err = 0; + + for (i = 0; i < spec->num_muxes; i++) { + nid = spec->mux_nids[i]; + wcaps = get_wcaps(codec, nid); + + if (wcaps & AC_WCAP_OUT_AMP) { + err = stac92xx_add_control_idx(spec, + STAC_CTL_WIDGET_VOL, i, "Mux Capture Volume", + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + } + } + return 0; +}; + +static const char *stac92xx_spdif_labels[3] = { + "Digital Playback", "Analog Mux 1", "Analog Mux 2", +}; + +static int stac92xx_auto_create_spdif_mux_ctls(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + struct hda_input_mux *spdif_mux = &spec->private_smux; + const char **labels = spec->spdif_labels; + int i, num_cons; + hda_nid_t con_lst[HDA_MAX_NUM_INPUTS]; + + num_cons = snd_hda_get_connections(codec, + spec->smux_nids[0], + con_lst, + HDA_MAX_NUM_INPUTS); + if (!num_cons) + return -EINVAL; + + if (!labels) + labels = stac92xx_spdif_labels; + + for (i = 0; i < num_cons; i++) { + spdif_mux->items[spdif_mux->num_items].label = labels[i]; + spdif_mux->items[spdif_mux->num_items].index = i; + spdif_mux->num_items++; + } + + return 0; +} + /* labels for dmic mux inputs */ static const char *stac92xx_dmic_labels[5] = { "Analog Inputs", "Digital Mic 1", "Digital Mic 2", @@ -2656,16 +3159,19 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, } continue; found: - wcaps = get_wcaps(codec, nid); + wcaps = get_wcaps(codec, nid) & + (AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP); - if (wcaps & AC_WCAP_OUT_AMP) { + if (wcaps) { sprintf(name, "%s Capture Volume", stac92xx_dmic_labels[dimux->num_items]); err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + HDA_COMPOSE_AMP_VAL(nid, 3, 0, + (wcaps & AC_WCAP_OUT_AMP) ? + HDA_OUTPUT : HDA_INPUT)); if (err < 0) return err; } @@ -2789,8 +3295,8 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out hp_speaker_swap = 1; } if (spec->autocfg.mono_out_pin) { - int dir = (get_wcaps(codec, spec->autocfg.mono_out_pin) - & AC_WCAP_OUT_AMP) ? HDA_OUTPUT : HDA_INPUT; + int dir = get_wcaps(codec, spec->autocfg.mono_out_pin) & + (AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP); u32 caps = query_amp_caps(codec, spec->autocfg.mono_out_pin, dir); hda_nid_t conn_list[1]; @@ -2812,21 +3318,26 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out !(wcaps & AC_WCAP_LR_SWAP)) spec->mono_nid = conn_list[0]; } - /* all mono outs have a least a mute/unmute switch */ - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, - "Mono Playback Switch", - HDA_COMPOSE_AMP_VAL(spec->autocfg.mono_out_pin, - 1, 0, dir)); - if (err < 0) - return err; - /* check to see if there is volume support for the amp */ - if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) { - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, - "Mono Playback Volume", - HDA_COMPOSE_AMP_VAL(spec->autocfg.mono_out_pin, - 1, 0, dir)); + if (dir) { + hda_nid_t nid = spec->autocfg.mono_out_pin; + + /* most mono outs have a least a mute/unmute switch */ + dir = (dir & AC_WCAP_OUT_AMP) ? HDA_OUTPUT : HDA_INPUT; + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, + "Mono Playback Switch", + HDA_COMPOSE_AMP_VAL(nid, 1, 0, dir)); if (err < 0) return err; + /* check for volume support for the amp */ + if ((caps & AC_AMPCAP_NUM_STEPS) + >> AC_AMPCAP_NUM_STEPS_SHIFT) { + err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_VOL, + "Mono Playback Volume", + HDA_COMPOSE_AMP_VAL(nid, 1, 0, dir)); + if (err < 0) + return err; + } } stac92xx_auto_set_pinctl(codec, spec->autocfg.mono_out_pin, @@ -2844,6 +3355,28 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (err < 0) return err; + /* setup analog beep controls */ + if (spec->anabeep_nid > 0) { + err = stac92xx_auto_create_beep_ctls(codec, + spec->anabeep_nid); + if (err < 0) + return err; + } + + /* setup digital beep controls and input device */ +#ifdef CONFIG_SND_HDA_INPUT_BEEP + if (spec->digbeep_nid > 0) { + hda_nid_t nid = spec->digbeep_nid; + + err = stac92xx_auto_create_beep_ctls(codec, nid); + if (err < 0) + return err; + err = snd_hda_attach_beep_device(codec, nid); + if (err < 0) + return err; + } +#endif + if (hp_speaker_swap == 1) { /* Restore the hp_outs and line_outs */ memcpy(spec->autocfg.hp_pins, spec->autocfg.line_out_pins, @@ -2872,11 +3405,25 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (err < 0) return err; } - - if (spec->num_dmics > 0) + if (spec->num_amps > 0) { + err = stac92xx_auto_create_amp_output_ctls(codec); + if (err < 0) + return err; + } + if (spec->num_dmics > 0 && !spec->dinput_mux) if ((err = stac92xx_auto_create_dmic_input_ctls(codec, &spec->autocfg)) < 0) return err; + if (spec->num_muxes > 0) { + err = stac92xx_auto_create_mux_input_ctls(codec); + if (err < 0) + return err; + } + if (spec->num_smuxes > 0) { + err = stac92xx_auto_create_spdif_mux_ctls(codec); + if (err < 0) + return err; + } spec->multiout.max_channels = spec->multiout.num_dacs * 2; if (spec->multiout.max_channels > 2) @@ -2884,17 +3431,17 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out if (spec->autocfg.dig_out_pin) spec->multiout.dig_out_nid = dig_out; - if (spec->autocfg.dig_in_pin) + if (dig_in && spec->autocfg.dig_in_pin) spec->dig_in_nid = dig_in; if (spec->kctl_alloc) spec->mixers[spec->num_mixers++] = spec->kctl_alloc; spec->input_mux = &spec->private_imux; - if (!spec->dinput_mux) - spec->dinput_mux = &spec->private_dimux; + spec->dinput_mux = &spec->private_dimux; + spec->sinput_mux = &spec->private_smux; spec->mono_mux = &spec->private_mono_mux; - + spec->amp_mux = &spec->private_amp_mux; return 1; } @@ -3074,6 +3621,12 @@ static int stac92xx_init(struct hda_codec *codec) snd_hda_sequence_write(codec, spec->init); + /* power down adcs initially */ + if (spec->powerdown_adcs) + for (i = 0; i < spec->num_adcs; i++) + snd_hda_codec_write_cache(codec, + spec->adc_nids[i], 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); /* set up pins */ if (spec->hp_detect) { /* Enable unsolicited responses on the HP widget */ @@ -3095,7 +3648,12 @@ static int stac92xx_init(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = cfg->input_pins[i]; if (nid) { - unsigned int pinctl = AC_PINCTL_IN_EN; + unsigned int pinctl = snd_hda_codec_read(codec, nid, + 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); + /* if PINCTL already set then skip */ + if (pinctl & AC_PINCAP_IN) + continue; + pinctl = AC_PINCTL_IN_EN; if (i == AUTO_PIN_MIC || i == AUTO_PIN_FRONT_MIC) pinctl |= stac92xx_get_vref(codec, nid); stac92xx_auto_set_pinctl(codec, nid, pinctl); @@ -3158,6 +3716,7 @@ static void stac92xx_free(struct hda_codec *codec) kfree(spec->bios_pin_configs); kfree(spec); + snd_hda_detach_beep_device(codec); } static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, @@ -3279,7 +3838,12 @@ static void stac92xx_pin_sense(struct hda_codec *codec, int idx) val = snd_hda_codec_read(codec, codec->afg, 0, 0x0fec, 0x0) & 0x000000ff; presence = get_hp_pin_presence(codec, nid); - idx = 1 << idx; + + /* several codecs have two power down bits */ + if (spec->pwr_mapping) + idx = spec->pwr_mapping[idx]; + else + idx = 1 << idx; if (presence) val &= ~idx; @@ -3295,13 +3859,22 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) struct sigmatel_spec *spec = codec->spec; int idx = res >> 26 & 0x0f; - switch ((res >> 26) & 0x30) { + switch ((res >> 26) & 0x70) { case STAC_HP_EVENT: stac92xx_hp_detect(codec, res); /* fallthru */ case STAC_PWR_EVENT: if (spec->num_pwrs > 0) stac92xx_pin_sense(codec, idx); + break; + case STAC_VREF_EVENT: { + int data = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DATA, 0); + /* toggle VREF state based on GPIOx status */ + snd_hda_codec_write(codec, codec->afg, 0, 0x7e0, + !!(data & (1 << idx))); + break; + } } } @@ -3478,9 +4051,9 @@ static struct hda_input_mux stac92hd73xx_dmux = { .num_items = 4, .items = { { "Analog Inputs", 0x0b }, - { "CD", 0x08 }, { "Digital Mic 1", 0x09 }, { "Digital Mic 2", 0x0a }, + { "CD", 0x08 }, } }; @@ -3495,6 +4068,7 @@ static int patch_stac92hd73xx(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->slave_dig_outs = stac92hd73xx_slave_dig_outs; spec->num_pins = ARRAY_SIZE(stac92hd73xx_pin_nids); spec->pin_nids = stac92hd73xx_pin_nids; spec->board_config = snd_hda_check_board_config(codec, @@ -3527,17 +4101,14 @@ again: switch (spec->multiout.num_dacs) { case 0x3: /* 6 Channel */ - spec->multiout.hp_nid = 0x17; spec->mixer = stac92hd73xx_6ch_mixer; spec->init = stac92hd73xx_6ch_core_init; break; case 0x4: /* 8 Channel */ - spec->multiout.hp_nid = 0x18; spec->mixer = stac92hd73xx_8ch_mixer; spec->init = stac92hd73xx_8ch_core_init; break; case 0x5: /* 10 Channel */ - spec->multiout.hp_nid = 0x19; spec->mixer = stac92hd73xx_10ch_mixer; spec->init = stac92hd73xx_10ch_core_init; }; @@ -3546,27 +4117,34 @@ again: spec->aloopback_mask = 0x01; spec->aloopback_shift = 8; + spec->digbeep_nid = 0x1c; spec->mux_nids = stac92hd73xx_mux_nids; spec->adc_nids = stac92hd73xx_adc_nids; spec->dmic_nids = stac92hd73xx_dmic_nids; spec->dmux_nids = stac92hd73xx_dmux_nids; + spec->smux_nids = stac92hd73xx_smux_nids; + spec->amp_nids = stac92hd73xx_amp_nids; + spec->num_amps = ARRAY_SIZE(stac92hd73xx_amp_nids); spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids); spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids); - spec->dinput_mux = &stac92hd73xx_dmux; - /* GPIO0 High = Enable EAPD */ - spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; - spec->gpio_data = 0x01; + memcpy(&spec->private_dimux, &stac92hd73xx_dmux, + sizeof(stac92hd73xx_dmux)); switch (spec->board_config) { case STAC_DELL_M6: spec->init = dell_eq_core_init; + spec->num_smuxes = 0; + spec->mixer = &stac92hd73xx_6ch_mixer[DELL_M6_MIXER]; + spec->amp_nids = &stac92hd73xx_amp_nids[DELL_M6_AMP]; + spec->num_amps = 1; switch (codec->subsystem_id) { case 0x1028025e: /* Analog Mics */ case 0x1028025f: stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); spec->num_dmics = 0; + spec->private_dimux.num_items = 1; break; case 0x10280271: /* Digital Mics */ case 0x10280272: @@ -3576,23 +4154,32 @@ again: case 0x10280255: stac92xx_set_config_reg(codec, 0x13, 0x90A60160); spec->num_dmics = 1; + spec->private_dimux.num_items = 2; break; case 0x10280256: /* Both */ case 0x10280057: stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); stac92xx_set_config_reg(codec, 0x13, 0x90A60160); spec->num_dmics = 1; + spec->private_dimux.num_items = 2; break; } break; default: spec->num_dmics = STAC92HD73XX_NUM_DMICS; + spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids); } + if (spec->board_config > STAC_92HD73XX_REF) { + /* GPIO0 High = Enable EAPD */ + spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; + spec->gpio_data = 0x01; + } + spec->dinput_mux = &spec->private_dimux; spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids; - err = stac92xx_parse_auto_config(codec, 0x22, 0x24); + err = stac92xx_parse_auto_config(codec, 0x25, 0x27); if (!err) { if (spec->board_config < 0) { @@ -3614,6 +4201,146 @@ again: return 0; } +static struct hda_input_mux stac92hd83xxx_dmux = { + .num_items = 3, + .items = { + { "Analog Inputs", 0x03 }, + { "Digital Mic 1", 0x04 }, + { "Digital Mic 2", 0x05 }, + } +}; + +static int patch_stac92hd83xxx(struct hda_codec *codec) +{ + struct sigmatel_spec *spec; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + codec->slave_dig_outs = stac92hd83xxx_slave_dig_outs; + spec->mono_nid = 0x19; + spec->digbeep_nid = 0x21; + spec->dmic_nids = stac92hd83xxx_dmic_nids; + spec->dmux_nids = stac92hd83xxx_dmux_nids; + spec->adc_nids = stac92hd83xxx_adc_nids; + spec->pwr_nids = stac92hd83xxx_pwr_nids; + spec->pwr_mapping = stac92hd83xxx_pwr_mapping; + spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids); + spec->multiout.dac_nids = stac92hd83xxx_dac_nids; + + spec->init = stac92hd83xxx_core_init; + switch (codec->vendor_id) { + case 0x111d7605: + spec->multiout.num_dacs = STAC92HD81_DAC_COUNT; + break; + default: + spec->num_pwrs--; + spec->init++; /* switch to config #2 */ + spec->multiout.num_dacs = STAC92HD83_DAC_COUNT; + } + + spec->mixer = stac92hd83xxx_mixer; + spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids); + spec->num_dmuxes = ARRAY_SIZE(stac92hd83xxx_dmux_nids); + spec->num_adcs = ARRAY_SIZE(stac92hd83xxx_adc_nids); + spec->num_dmics = STAC92HD83XXX_NUM_DMICS; + spec->dinput_mux = &stac92hd83xxx_dmux; + spec->pin_nids = stac92hd83xxx_pin_nids; + spec->board_config = snd_hda_check_board_config(codec, + STAC_92HD83XXX_MODELS, + stac92hd83xxx_models, + stac92hd83xxx_cfg_tbl); +again: + if (spec->board_config < 0) { + snd_printdd(KERN_INFO "hda_codec: Unknown model for" + " STAC92HD83XXX, using BIOS defaults\n"); + err = stac92xx_save_bios_config_regs(codec); + if (err < 0) { + stac92xx_free(codec); + return err; + } + spec->pin_configs = spec->bios_pin_configs; + } else { + spec->pin_configs = stac92hd83xxx_brd_tbl[spec->board_config]; + stac92xx_set_config_regs(codec); + } + + err = stac92xx_parse_auto_config(codec, 0x1d, 0); + if (!err) { + if (spec->board_config < 0) { + printk(KERN_WARNING "hda_codec: No auto-config is " + "available, default to model=ref\n"); + spec->board_config = STAC_92HD83XXX_REF; + goto again; + } + err = -EINVAL; + } + + if (err < 0) { + stac92xx_free(codec); + return err; + } + + codec->patch_ops = stac92xx_patch_ops; + + return 0; +} + +#ifdef SND_HDA_NEEDS_RESUME +static void stac92hd71xx_set_power_state(struct hda_codec *codec, int pwr) +{ + struct sigmatel_spec *spec = codec->spec; + int i; + snd_hda_codec_write_cache(codec, codec->afg, 0, + AC_VERB_SET_POWER_STATE, pwr); + + msleep(1); + for (i = 0; i < spec->num_adcs; i++) { + snd_hda_codec_write_cache(codec, + spec->adc_nids[i], 0, + AC_VERB_SET_POWER_STATE, pwr); + } +}; + +static int stac92hd71xx_resume(struct hda_codec *codec) +{ + stac92hd71xx_set_power_state(codec, AC_PWRST_D0); + return stac92xx_resume(codec); +} + +static int stac92hd71xx_suspend(struct hda_codec *codec, pm_message_t state) +{ + stac92hd71xx_set_power_state(codec, AC_PWRST_D3); + return 0; +}; + +#endif + +static struct hda_codec_ops stac92hd71bxx_patch_ops = { + .build_controls = stac92xx_build_controls, + .build_pcms = stac92xx_build_pcms, + .init = stac92xx_init, + .free = stac92xx_free, + .unsol_event = stac92xx_unsol_event, +#ifdef SND_HDA_NEEDS_RESUME + .resume = stac92hd71xx_resume, + .suspend = stac92hd71xx_suspend, +#endif +}; + +static struct hda_input_mux stac92hd71bxx_dmux = { + .num_items = 4, + .items = { + { "Analog Inputs", 0x00 }, + { "Mixer", 0x01 }, + { "Digital Mic 1", 0x02 }, + { "Digital Mic 2", 0x03 }, + } +}; + static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; @@ -3624,9 +4351,12 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->patch_ops = stac92xx_patch_ops; spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pin_nids = stac92hd71bxx_pin_nids; + memcpy(&spec->private_dimux, &stac92hd71bxx_dmux, + sizeof(stac92hd71bxx_dmux)); spec->board_config = snd_hda_check_board_config(codec, STAC_92HD71BXX_MODELS, stac92hd71bxx_models, @@ -3653,47 +4383,101 @@ again: case 0x111d76b5: spec->mixer = stac92hd71bxx_mixer; spec->init = stac92hd71bxx_core_init; + codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; break; case 0x111d7608: /* 5 Port with Analog Mixer */ + switch (codec->subsystem_id) { + case 0x103c361a: + /* Enable VREF power saving on GPIO1 detect */ + snd_hda_codec_write(codec, codec->afg, 0, + AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, 0x02); + snd_hda_codec_write_cache(codec, codec->afg, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + (AC_USRSP_EN | STAC_VREF_EVENT | 0x01)); + spec->gpio_mask |= 0x02; + break; + } + if ((codec->revision_id & 0xf) == 0 || + (codec->revision_id & 0xf) == 1) { +#ifdef SND_HDA_NEEDS_RESUME + codec->patch_ops = stac92hd71bxx_patch_ops; +#endif + spec->stream_delay = 40; /* 40 milliseconds */ + } + /* no output amps */ spec->num_pwrs = 0; spec->mixer = stac92hd71bxx_analog_mixer; + spec->dinput_mux = &spec->private_dimux; /* disable VSW */ spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; stac92xx_set_config_reg(codec, 0xf, 0x40f000f0); break; case 0x111d7603: /* 6 Port with Analog Mixer */ + if ((codec->revision_id & 0xf) == 1) { +#ifdef SND_HDA_NEEDS_RESUME + codec->patch_ops = stac92hd71bxx_patch_ops; +#endif + spec->stream_delay = 40; /* 40 milliseconds */ + } + /* no output amps */ spec->num_pwrs = 0; /* fallthru */ default: + spec->dinput_mux = &spec->private_dimux; spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init; + codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; } - spec->aloopback_mask = 0x20; + spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; - /* GPIO0 High = EAPD */ - spec->gpio_mask = 0x01; - spec->gpio_dir = 0x01; - spec->gpio_data = 0x01; + if (spec->board_config > STAC_92HD71BXX_REF) { + /* GPIO0 = EAPD */ + spec->gpio_mask = 0x01; + spec->gpio_dir = 0x01; + spec->gpio_data = 0x01; + } + spec->powerdown_adcs = 1; + spec->digbeep_nid = 0x26; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; spec->dmic_nids = stac92hd71bxx_dmic_nids; spec->dmux_nids = stac92hd71bxx_dmux_nids; + spec->smux_nids = stac92hd71bxx_smux_nids; spec->pwr_nids = stac92hd71bxx_pwr_nids; spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); - spec->num_dmics = STAC92HD71BXX_NUM_DMICS; - spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); + + switch (spec->board_config) { + case STAC_HP_M4: + spec->num_dmics = 0; + spec->num_smuxes = 0; + spec->num_dmuxes = 0; + + /* enable internal microphone */ + stac92xx_set_config_reg(codec, 0x0e, 0x01813040); + stac92xx_auto_set_pinctl(codec, 0x0e, + AC_PINCTL_IN_EN | AC_PINCTL_VREF_80); + break; + default: + spec->num_dmics = STAC92HD71BXX_NUM_DMICS; + spec->num_smuxes = ARRAY_SIZE(stac92hd71bxx_smux_nids); + spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); + }; spec->multiout.num_dacs = 1; spec->multiout.hp_nid = 0x11; spec->multiout.dac_nids = stac92hd71bxx_dac_nids; + if (spec->dinput_mux) + spec->private_dimux.num_items += + spec->num_dmics - + (ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1); err = stac92xx_parse_auto_config(codec, 0x21, 0x23); if (!err) { @@ -3711,8 +4495,6 @@ again: return err; } - codec->patch_ops = stac92xx_patch_ops; - return 0; }; @@ -3854,10 +4636,14 @@ static int patch_stac927x(struct hda_codec *codec) stac92xx_set_config_regs(codec); } + spec->digbeep_nid = 0x23; spec->adc_nids = stac927x_adc_nids; spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); spec->mux_nids = stac927x_mux_nids; spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); + spec->smux_nids = stac927x_smux_nids; + spec->num_smuxes = ARRAY_SIZE(stac927x_smux_nids); + spec->spdif_labels = stac927x_spdif_labels; spec->dac_list = stac927x_dac_nids; spec->multiout.dac_nids = spec->dac_nids; @@ -3900,9 +4686,11 @@ static int patch_stac927x(struct hda_codec *codec) spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids); break; default: - /* GPIO0 High = Enable EAPD */ - spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; - spec->gpio_data = 0x01; + if (spec->board_config > STAC_D965_REF) { + /* GPIO0 High = Enable EAPD */ + spec->eapd_mask = spec->gpio_mask = 0x01; + spec->gpio_dir = spec->gpio_data = 0x01; + } spec->num_dmics = 0; spec->init = stac927x_core_init; @@ -3974,10 +4762,13 @@ static int patch_stac9205(struct hda_codec *codec) stac92xx_set_config_regs(codec); } + spec->digbeep_nid = 0x23; spec->adc_nids = stac9205_adc_nids; spec->num_adcs = ARRAY_SIZE(stac9205_adc_nids); spec->mux_nids = stac9205_mux_nids; spec->num_muxes = ARRAY_SIZE(stac9205_mux_nids); + spec->smux_nids = stac9205_smux_nids; + spec->num_smuxes = ARRAY_SIZE(stac9205_smux_nids); spec->dmic_nids = stac9205_dmic_nids; spec->num_dmics = STAC9205_NUM_DMICS; spec->dmux_nids = stac9205_dmux_nids; @@ -4013,6 +4804,9 @@ static int patch_stac9205(struct hda_codec *codec) */ spec->gpio_data = 0x01; break; + case STAC_9205_REF: + /* SPDIF-In enabled */ + break; default: /* GPIO0 High = EAPD */ spec->eapd_mask = spec->gpio_mask = spec->gpio_dir = 0x1; @@ -4332,6 +5126,8 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 }, { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx}, + { .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx}, + { .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx}, { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx }, diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index e7e43524f8c..63e4871e5d8 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1,10 +1,10 @@ /* * Universal Interface for Intel High Definition Audio Codec * - * HD audio interface patch for VIA VT1708 codec + * HD audio interface patch for VIA VT1702/VT1708/VT1709 codec * - * Copyright (c) 2006 Lydia Wang <lydiawang@viatech.com> - * Takashi Iwai <tiwai@suse.de> + * Copyright (c) 2006-2008 Lydia Wang <lydiawang@viatech.com> + * 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 @@ -29,6 +29,13 @@ /* 2006-09-08 Lydia Wang Fix internal loopback recording source select bug */ /* 2007-09-12 Lydia Wang Add EAPD enable during driver initialization */ /* 2007-09-17 Lydia Wang Add VT1708B codec support */ +/* 2007-11-14 Lydia Wang Add VT1708A codec HP and CD pin connect config */ +/* 2008-02-03 Lydia Wang Fix Rear channels and Back channels inverse issue */ +/* 2008-03-06 Lydia Wang Add VT1702 codec and VT1708S codec support */ +/* 2008-04-09 Lydia Wang Add mute front speaker when HP plugin */ +/* 2008-04-09 Lydia Wang Add Independent HP feature */ +/* 2008-05-28 Lydia Wang Add second S/PDIF Out support for VT1702 */ +/* 2008-09-15 Logan Li Add VT1708S Mic Boost workaround/backdoor */ /* */ /* * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * * */ @@ -37,6 +44,7 @@ #include <linux/delay.h> #include <linux/slab.h> #include <sound/core.h> +#include <sound/asoundef.h> #include "hda_codec.h" #include "hda_local.h" #include "hda_patch.h" @@ -53,6 +61,8 @@ #define VT1708_DIGOUT_NID 0x14 #define VT1708_DIGIN_NID 0x16 #define VT1708_DIGIN_PIN 0x26 +#define VT1708_HP_PIN_NID 0x20 +#define VT1708_CD_PIN_NID 0x24 #define VT1709_HP_DAC_NID 0x28 #define VT1709_DIGOUT_NID 0x13 @@ -64,12 +74,64 @@ #define VT1708B_DIGIN_NID 0x15 #define VT1708B_DIGIN_PIN 0x21 +#define VT1708S_HP_NID 0x25 +#define VT1708S_DIGOUT_NID 0x12 + +#define VT1702_HP_NID 0x17 +#define VT1702_DIGOUT_NID 0x11 + #define IS_VT1708_VENDORID(x) ((x) >= 0x11061708 && (x) <= 0x1106170b) #define IS_VT1709_10CH_VENDORID(x) ((x) >= 0x1106e710 && (x) <= 0x1106e713) #define IS_VT1709_6CH_VENDORID(x) ((x) >= 0x1106e714 && (x) <= 0x1106e717) #define IS_VT1708B_8CH_VENDORID(x) ((x) >= 0x1106e720 && (x) <= 0x1106e723) #define IS_VT1708B_4CH_VENDORID(x) ((x) >= 0x1106e724 && (x) <= 0x1106e727) +#define IS_VT1708S_VENDORID(x) ((x) >= 0x11060397 && (x) <= 0x11067397) +#define IS_VT1702_VENDORID(x) ((x) >= 0x11060398 && (x) <= 0x11067398) + +enum VIA_HDA_CODEC { + UNKNOWN = -1, + VT1708, + VT1709_10CH, + VT1709_6CH, + VT1708B_8CH, + VT1708B_4CH, + VT1708S, + VT1702, + CODEC_TYPES, +}; + +static enum VIA_HDA_CODEC get_codec_type(u32 vendor_id) +{ + u16 ven_id = vendor_id >> 16; + u16 dev_id = vendor_id & 0xffff; + enum VIA_HDA_CODEC codec_type; + + /* get codec type */ + if (ven_id != 0x1106) + codec_type = UNKNOWN; + else if (dev_id >= 0x1708 && dev_id <= 0x170b) + codec_type = VT1708; + else if (dev_id >= 0xe710 && dev_id <= 0xe713) + codec_type = VT1709_10CH; + else if (dev_id >= 0xe714 && dev_id <= 0xe717) + codec_type = VT1709_6CH; + else if (dev_id >= 0xe720 && dev_id <= 0xe723) + codec_type = VT1708B_8CH; + else if (dev_id >= 0xe724 && dev_id <= 0xe727) + codec_type = VT1708B_4CH; + else if ((dev_id & 0xfff) == 0x397 + && (dev_id >> 12) < 8) + codec_type = VT1708S; + else if ((dev_id & 0xfff) == 0x398 + && (dev_id >> 12) < 8) + codec_type = VT1702; + else + codec_type = UNKNOWN; + return codec_type; +}; +#define VIA_HP_EVENT 0x01 +#define VIA_GPIO_EVENT 0x02 enum { VIA_CTL_WIDGET_VOL, @@ -77,12 +139,54 @@ enum { }; enum { - AUTO_SEQ_FRONT, + AUTO_SEQ_FRONT = 0, AUTO_SEQ_SURROUND, AUTO_SEQ_CENLFE, AUTO_SEQ_SIDE }; +#define get_amp_nid(kc) ((kc)->private_value & 0xffff) + +/* Some VT1708S based boards gets the micboost setting wrong, so we have + * to apply some brute-force and re-write the TLV's by software. */ +static int mic_boost_tlv(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *_tlv) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = get_amp_nid(kcontrol); + + if (get_codec_type(codec->vendor_id) == VT1708S + && (nid == 0x1a || nid == 0x1e)) { + if (size < 4 * sizeof(unsigned int)) + return -ENOMEM; + if (put_user(1, _tlv)) /* SNDRV_CTL_TLVT_DB_SCALE */ + return -EFAULT; + if (put_user(2 * sizeof(unsigned int), _tlv + 1)) + return -EFAULT; + if (put_user(0, _tlv + 2)) /* offset = 0 */ + return -EFAULT; + if (put_user(1000, _tlv + 3)) /* step size = 10 dB */ + return -EFAULT; + } + return 0; +} + +static int mic_boost_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + hda_nid_t nid = get_amp_nid(kcontrol); + + if (get_codec_type(codec->vendor_id) == VT1708S + && (nid == 0x1a || nid == 0x1e)) { + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 3; + } + return 0; +} + static struct snd_kcontrol_new vt1708_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), @@ -94,7 +198,8 @@ struct via_spec { struct snd_kcontrol_new *mixers[3]; unsigned int num_mixers; - struct hda_verb *init_verbs; + struct hda_verb *init_verbs[5]; + unsigned int num_iverbs; char *stream_name_analog; struct hda_pcm_stream *stream_analog_playback; @@ -106,6 +211,7 @@ struct via_spec { /* playback */ struct hda_multi_out multiout; + hda_nid_t extra_dig_out_nid; /* capture */ unsigned int num_adc_nids; @@ -117,15 +223,19 @@ struct via_spec { unsigned int cur_mux[3]; /* PCM information */ - struct hda_pcm pcm_rec[2]; + struct hda_pcm pcm_rec[3]; /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; unsigned int num_kctl_alloc, num_kctl_used; struct snd_kcontrol_new *kctl_alloc; - struct hda_input_mux private_imux; + struct hda_input_mux private_imux[2]; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; + /* HP mode source */ + const struct hda_input_mux *hp_mux; + unsigned int hp_independent_mode; + #ifdef CONFIG_SND_HDA_POWER_SAVE struct hda_loopback_check loopback; #endif @@ -146,6 +256,16 @@ static hda_nid_t vt1708B_adc_nids[2] = { 0x13, 0x14 }; +static hda_nid_t vt1708S_adc_nids[2] = { + /* ADC1-2 */ + 0x13, 0x14 +}; + +static hda_nid_t vt1702_adc_nids[3] = { + /* ADC1-2 */ + 0x12, 0x20, 0x1F +}; + /* add dynamic controls */ static int via_add_control(struct via_spec *spec, int type, const char *name, unsigned long val) @@ -283,19 +403,108 @@ static int via_mux_enum_put(struct snd_kcontrol *kcontrol, return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, 0x18, &spec->cur_mux[adc_idx]); else if ((IS_VT1709_10CH_VENDORID(vendor_id) || - IS_VT1709_6CH_VENDORID(vendor_id)) && adc_idx == 0) + IS_VT1709_6CH_VENDORID(vendor_id)) && (adc_idx == 0)) return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, 0x19, &spec->cur_mux[adc_idx]); else if ((IS_VT1708B_8CH_VENDORID(vendor_id) || - IS_VT1708B_4CH_VENDORID(vendor_id)) && adc_idx == 0) + IS_VT1708B_4CH_VENDORID(vendor_id)) && (adc_idx == 0)) return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, 0x17, &spec->cur_mux[adc_idx]); + else if (IS_VT1702_VENDORID(vendor_id) && (adc_idx == 0)) + return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, + 0x13, &spec->cur_mux[adc_idx]); else return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, spec->adc_nids[adc_idx], &spec->cur_mux[adc_idx]); } +static int via_independent_hp_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + return snd_hda_input_mux_info(spec->hp_mux, uinfo); +} + +static int via_independent_hp_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + hda_nid_t nid = spec->autocfg.hp_pins[0]; + unsigned int pinsel = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONNECT_SEL, + 0x00); + + ucontrol->value.enumerated.item[0] = pinsel; + + return 0; +} + +static int via_independent_hp_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct via_spec *spec = codec->spec; + hda_nid_t nid = spec->autocfg.hp_pins[0]; + unsigned int pinsel = ucontrol->value.enumerated.item[0]; + unsigned int con_nid = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + + if (con_nid == spec->multiout.hp_nid) { + if (pinsel == 0) { + if (!spec->hp_independent_mode) { + if (spec->multiout.num_dacs > 1) + spec->multiout.num_dacs -= 1; + spec->hp_independent_mode = 1; + } + } else if (pinsel == 1) { + if (spec->hp_independent_mode) { + if (spec->multiout.num_dacs > 1) + spec->multiout.num_dacs += 1; + spec->hp_independent_mode = 0; + } + } + } else { + if (pinsel == 0) { + if (spec->hp_independent_mode) { + if (spec->multiout.num_dacs > 1) + spec->multiout.num_dacs += 1; + spec->hp_independent_mode = 0; + } + } else if (pinsel == 1) { + if (!spec->hp_independent_mode) { + if (spec->multiout.num_dacs > 1) + spec->multiout.num_dacs -= 1; + spec->hp_independent_mode = 1; + } + } + } + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, + pinsel); + + if (spec->multiout.hp_nid && + spec->multiout.hp_nid != spec->multiout.dac_nids[HDA_FRONT]) + snd_hda_codec_setup_stream(codec, + spec->multiout.hp_nid, + 0, 0, 0); + + return 0; +} + +static struct snd_kcontrol_new via_hp_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Independent HP", + .count = 1, + .info = via_independent_hp_info, + .get = via_independent_hp_get, + .put = via_independent_hp_put, + }, + { } /* end */ +}; + /* capture mixer elements */ static struct snd_kcontrol_new vt1708_capture_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_INPUT), @@ -380,6 +589,138 @@ static int via_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout); } + +static void playback_multi_pcm_prep_0(struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct via_spec *spec = codec->spec; + struct hda_multi_out *mout = &spec->multiout; + hda_nid_t *nids = mout->dac_nids; + int chs = substream->runtime->channels; + int i; + + mutex_lock(&codec->spdif_mutex); + if (mout->dig_out_nid && mout->dig_out_used != HDA_DIG_EXCLUSIVE) { + if (chs == 2 && + snd_hda_is_supported_format(codec, mout->dig_out_nid, + format) && + !(codec->spdif_status & IEC958_AES0_NONAUDIO)) { + mout->dig_out_used = HDA_DIG_ANALOG_DUP; + /* turn off SPDIF once; otherwise the IEC958 bits won't + * be updated */ + if (codec->spdif_ctls & AC_DIG1_ENABLE) + snd_hda_codec_write(codec, mout->dig_out_nid, 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & + ~AC_DIG1_ENABLE & 0xff); + snd_hda_codec_setup_stream(codec, mout->dig_out_nid, + stream_tag, 0, format); + /* turn on again (if needed) */ + if (codec->spdif_ctls & AC_DIG1_ENABLE) + snd_hda_codec_write(codec, mout->dig_out_nid, 0, + AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & 0xff); + } else { + mout->dig_out_used = 0; + snd_hda_codec_setup_stream(codec, mout->dig_out_nid, + 0, 0, 0); + } + } + mutex_unlock(&codec->spdif_mutex); + + /* front */ + snd_hda_codec_setup_stream(codec, nids[HDA_FRONT], stream_tag, + 0, format); + + if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] && + !spec->hp_independent_mode) + /* headphone out will just decode front left/right (stereo) */ + snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, + 0, format); + + /* extra outputs copied from front */ + for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++) + if (mout->extra_out_nid[i]) + snd_hda_codec_setup_stream(codec, + mout->extra_out_nid[i], + stream_tag, 0, format); + + /* surrounds */ + for (i = 1; i < mout->num_dacs; i++) { + if (chs >= (i + 1) * 2) /* independent out */ + snd_hda_codec_setup_stream(codec, nids[i], stream_tag, + i * 2, format); + else /* copy front */ + snd_hda_codec_setup_stream(codec, nids[i], stream_tag, + 0, format); + } +} + +static int via_playback_multi_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct via_spec *spec = codec->spec; + struct hda_multi_out *mout = &spec->multiout; + hda_nid_t *nids = mout->dac_nids; + + if (substream->number == 0) + playback_multi_pcm_prep_0(codec, stream_tag, format, + substream); + else { + if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] && + spec->hp_independent_mode) + snd_hda_codec_setup_stream(codec, mout->hp_nid, + stream_tag, 0, format); + } + + return 0; +} + +static int via_playback_multi_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct via_spec *spec = codec->spec; + struct hda_multi_out *mout = &spec->multiout; + hda_nid_t *nids = mout->dac_nids; + int i; + + if (substream->number == 0) { + for (i = 0; i < mout->num_dacs; i++) + snd_hda_codec_setup_stream(codec, nids[i], 0, 0, 0); + + if (mout->hp_nid && !spec->hp_independent_mode) + snd_hda_codec_setup_stream(codec, mout->hp_nid, + 0, 0, 0); + + for (i = 0; i < ARRAY_SIZE(mout->extra_out_nid); i++) + if (mout->extra_out_nid[i]) + snd_hda_codec_setup_stream(codec, + mout->extra_out_nid[i], + 0, 0, 0); + mutex_lock(&codec->spdif_mutex); + if (mout->dig_out_nid && + mout->dig_out_used == HDA_DIG_ANALOG_DUP) { + snd_hda_codec_setup_stream(codec, mout->dig_out_nid, + 0, 0, 0); + mout->dig_out_used = 0; + } + mutex_unlock(&codec->spdif_mutex); + } else { + if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] && + spec->hp_independent_mode) + snd_hda_codec_setup_stream(codec, mout->hp_nid, + 0, 0, 0); + } + + return 0; +} + /* * Digital out */ @@ -399,6 +740,21 @@ static int via_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, return snd_hda_multi_out_dig_close(codec, &spec->multiout); } +/* setup SPDIF output stream */ +static void setup_dig_playback_stream(struct hda_codec *codec, hda_nid_t nid, + unsigned int stream_tag, unsigned int format) +{ + /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ + if (codec->spdif_ctls & AC_DIG1_ENABLE) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & ~AC_DIG1_ENABLE & 0xff); + snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); + /* turn on again (if needed) */ + if (codec->spdif_ctls & AC_DIG1_ENABLE) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, + codec->spdif_ctls & 0xff); +} + static int via_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hda_codec *codec, unsigned int stream_tag, @@ -406,8 +762,20 @@ static int via_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct snd_pcm_substream *substream) { struct via_spec *spec = codec->spec; - return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, - stream_tag, format, substream); + hda_nid_t nid; + + /* 1st or 2nd S/PDIF */ + if (substream->number == 0) + nid = spec->multiout.dig_out_nid; + else if (substream->number == 1) + nid = spec->extra_dig_out_nid; + else + return -1; + + mutex_lock(&codec->spdif_mutex); + setup_dig_playback_stream(codec, nid, stream_tag, format); + mutex_unlock(&codec->spdif_mutex); + return 0; } /* @@ -436,14 +804,14 @@ static int via_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, } static struct hda_pcm_stream vt1708_pcm_analog_playback = { - .substreams = 1, + .substreams = 2, .channels_min = 2, .channels_max = 8, .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup }, }; @@ -515,6 +883,13 @@ static int via_build_controls(struct hda_codec *codec) if (err < 0) return err; spec->multiout.share_spdif = 1; + + if (spec->extra_dig_out_nid) { + err = snd_hda_create_spdif_out_ctls(codec, + spec->extra_dig_out_nid); + if (err < 0) + return err; + } } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -580,10 +955,89 @@ static void via_free(struct hda_codec *codec) kfree(codec->spec); } +/* mute internal speaker if HP is plugged */ +static void via_hp_automute(struct hda_codec *codec) +{ + unsigned int present; + struct via_spec *spec = codec->spec; + + present = snd_hda_codec_read(codec, spec->autocfg.hp_pins[0], 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, spec->autocfg.line_out_pins[0], + HDA_OUTPUT, 0, HDA_AMP_MUTE, + present ? HDA_AMP_MUTE : 0); +} + +static void via_gpio_control(struct hda_codec *codec) +{ + unsigned int gpio_data; + unsigned int vol_counter; + unsigned int vol; + unsigned int master_vol; + + struct via_spec *spec = codec->spec; + + gpio_data = snd_hda_codec_read(codec, codec->afg, 0, + AC_VERB_GET_GPIO_DATA, 0) & 0x03; + + vol_counter = (snd_hda_codec_read(codec, codec->afg, 0, + 0xF84, 0) & 0x3F0000) >> 16; + + vol = vol_counter & 0x1F; + master_vol = snd_hda_codec_read(codec, 0x1A, 0, + AC_VERB_GET_AMP_GAIN_MUTE, + AC_AMP_GET_INPUT); + + if (gpio_data == 0x02) { + /* unmute line out */ + snd_hda_codec_amp_stereo(codec, spec->autocfg.line_out_pins[0], + HDA_OUTPUT, 0, HDA_AMP_MUTE, 0); + + if (vol_counter & 0x20) { + /* decrease volume */ + if (vol > master_vol) + vol = master_vol; + snd_hda_codec_amp_stereo(codec, 0x1A, HDA_INPUT, + 0, HDA_AMP_VOLMASK, + master_vol-vol); + } else { + /* increase volume */ + snd_hda_codec_amp_stereo(codec, 0x1A, HDA_INPUT, 0, + HDA_AMP_VOLMASK, + ((master_vol+vol) > 0x2A) ? 0x2A : + (master_vol+vol)); + } + } else if (!(gpio_data & 0x02)) { + /* mute line out */ + snd_hda_codec_amp_stereo(codec, + spec->autocfg.line_out_pins[0], + HDA_OUTPUT, 0, HDA_AMP_MUTE, + HDA_AMP_MUTE); + } +} + +/* unsolicited event for jack sensing */ +static void via_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + res >>= 26; + if (res == VIA_HP_EVENT) + via_hp_automute(codec); + else if (res == VIA_GPIO_EVENT) + via_gpio_control(codec); +} + +static hda_nid_t slave_dig_outs[] = { + 0, +}; + static int via_init(struct hda_codec *codec) { struct via_spec *spec = codec->spec; - snd_hda_sequence_write(codec, spec->init_verbs); + int i; + for (i = 0; i < spec->num_iverbs; i++) + snd_hda_sequence_write(codec, spec->init_verbs[i]); + /* Lydia Add for EAPD enable */ if (!spec->dig_in_nid) { /* No Digital In connection */ if (IS_VT1708_VENDORID(codec->vendor_id)) { @@ -611,6 +1065,9 @@ static int via_init(struct hda_codec *codec) snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN); + /* no slave outs */ + codec->slave_dig_outs = slave_dig_outs; + return 0; } @@ -657,10 +1114,10 @@ static int vt1708_auto_fill_dac_nids(struct via_spec *spec, spec->multiout.dac_nids[i] = 0x12; break; case AUTO_SEQ_SURROUND: - spec->multiout.dac_nids[i] = 0x13; + spec->multiout.dac_nids[i] = 0x11; break; case AUTO_SEQ_SIDE: - spec->multiout.dac_nids[i] = 0x11; + spec->multiout.dac_nids[i] = 0x13; break; } } @@ -685,7 +1142,7 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec, continue; if (i != AUTO_SEQ_FRONT) - nid_vol = 0x1b - i + 1; + nid_vol = 0x18 + i; if (i == AUTO_SEQ_CENLFE) { /* Center/LFE */ @@ -760,6 +1217,24 @@ static int vt1708_auto_create_multi_out_ctls(struct via_spec *spec, return 0; } +static void create_hp_imux(struct via_spec *spec) +{ + int i; + struct hda_input_mux *imux = &spec->private_imux[1]; + static const char *texts[] = { "OFF", "ON", NULL}; + + /* for hp mode select */ + i = 0; + while (texts[i] != NULL) { + imux->items[imux->num_items].label = texts[i]; + imux->items[imux->num_items].index = i; + imux->num_items++; + i++; + } + + spec->hp_mux = &spec->private_imux[1]; +} + static int vt1708_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) { int err; @@ -780,6 +1255,8 @@ static int vt1708_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) if (err < 0) return err; + create_hp_imux(spec); + return 0; } @@ -790,7 +1267,7 @@ static int vt1708_auto_create_analog_input_ctls(struct via_spec *spec, static char *labels[] = { "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL }; - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx = 0; /* for internal loopback recording select */ @@ -840,11 +1317,36 @@ static struct hda_amp_list vt1708_loopbacks[] = { }; #endif +static void vt1708_set_pinconfig_connect(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int def_conf; + unsigned char seqassoc; + + def_conf = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0); + seqassoc = (unsigned char) get_defcfg_association(def_conf); + seqassoc = (seqassoc << 4) | get_defcfg_sequence(def_conf); + if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) { + if (seqassoc == 0xff) { + def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30)); + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + def_conf >> 24); + } + } + + return; +} + static int vt1708_parse_auto_config(struct hda_codec *codec) { struct via_spec *spec = codec->spec; int err; + /* Add HP and CD pin config connect bit re-config action */ + vt1708_set_pinconfig_connect(codec, VT1708_HP_PIN_NID); + vt1708_set_pinconfig_connect(codec, VT1708_CD_PIN_NID); + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); if (err < 0) return err; @@ -874,9 +1376,12 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) if (spec->kctl_alloc) spec->mixers[spec->num_mixers++] = spec->kctl_alloc; - spec->init_verbs = vt1708_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1708_volume_init_verbs; + + spec->input_mux = &spec->private_imux[0]; - spec->input_mux = &spec->private_imux; + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; return 1; } @@ -897,7 +1402,7 @@ static int patch_vt1708(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; @@ -966,6 +1471,11 @@ static struct snd_kcontrol_new vt1709_capture_mixer[] = { { } /* end */ }; +static struct hda_verb vt1709_uniwill_init_verbs[] = { + {0x20, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + { } +}; + /* * generic initialization of ADC, input mixers and output mixers */ @@ -1090,11 +1600,11 @@ static int vt1709_auto_fill_dac_nids(struct via_spec *spec, break; case AUTO_SEQ_SURROUND: /* AOW3 */ - spec->multiout.dac_nids[i] = 0x27; + spec->multiout.dac_nids[i] = 0x11; break; case AUTO_SEQ_SIDE: /* AOW1 */ - spec->multiout.dac_nids[i] = 0x11; + spec->multiout.dac_nids[i] = 0x27; break; default: break; @@ -1203,26 +1713,26 @@ static int vt1709_auto_create_multi_out_ctls(struct via_spec *spec, } else if (i == AUTO_SEQ_SURROUND) { sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(0x29, 3, 0, + HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT)); if (err < 0) return err; sprintf(name, "%s Playback Switch", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(0x29, 3, 0, + HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT)); if (err < 0) return err; } else if (i == AUTO_SEQ_SIDE) { sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, + HDA_COMPOSE_AMP_VAL(0x29, 3, 0, HDA_OUTPUT)); if (err < 0) return err; sprintf(name, "%s Playback Switch", chname[i]); err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, - HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, + HDA_COMPOSE_AMP_VAL(0x29, 3, 0, HDA_OUTPUT)); if (err < 0) return err; @@ -1265,7 +1775,7 @@ static int vt1709_auto_create_analog_input_ctls(struct via_spec *spec, static char *labels[] = { "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL }; - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx = 0; /* for internal loopback recording select */ @@ -1339,7 +1849,10 @@ static int vt1709_parse_auto_config(struct hda_codec *codec) if (spec->kctl_alloc) spec->mixers[spec->num_mixers++] = spec->kctl_alloc; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; return 1; } @@ -1360,7 +1873,7 @@ static int patch_vt1709_10ch(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; @@ -1375,7 +1888,8 @@ static int patch_vt1709_10ch(struct hda_codec *codec) "Using genenic mode...\n"); } - spec->init_verbs = vt1709_10ch_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1709_10ch_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1709_uniwill_init_verbs; spec->stream_name_analog = "VT1709 Analog"; spec->stream_analog_playback = &vt1709_10ch_pcm_analog_playback; @@ -1396,6 +1910,7 @@ static int patch_vt1709_10ch(struct hda_codec *codec) codec->patch_ops = via_patch_ops; codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event; #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt1709_loopbacks; #endif @@ -1451,7 +1966,7 @@ static int patch_vt1709_6ch(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; @@ -1466,7 +1981,8 @@ static int patch_vt1709_6ch(struct hda_codec *codec) "Using genenic mode...\n"); } - spec->init_verbs = vt1709_6ch_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1709_6ch_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1709_uniwill_init_verbs; spec->stream_name_analog = "VT1709 Analog"; spec->stream_analog_playback = &vt1709_6ch_pcm_analog_playback; @@ -1487,6 +2003,7 @@ static int patch_vt1709_6ch(struct hda_codec *codec) codec->patch_ops = via_patch_ops; codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event; #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt1709_loopbacks; #endif @@ -1586,27 +2103,32 @@ static struct hda_verb vt1708B_4ch_volume_init_verbs[] = { { } }; +static struct hda_verb vt1708B_uniwill_init_verbs[] = { + {0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + { } +}; + static struct hda_pcm_stream vt1708B_8ch_pcm_analog_playback = { - .substreams = 1, + .substreams = 2, .channels_min = 2, .channels_max = 8, .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup }, }; static struct hda_pcm_stream vt1708B_4ch_pcm_analog_playback = { - .substreams = 1, + .substreams = 2, .channels_min = 2, .channels_max = 4, .nid = 0x10, /* NID to query formats and rates */ .ops = { .open = via_playback_pcm_open, - .prepare = via_playback_pcm_prepare, - .cleanup = via_playback_pcm_cleanup + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup }, }; @@ -1662,10 +2184,10 @@ static int vt1708B_auto_fill_dac_nids(struct via_spec *spec, spec->multiout.dac_nids[i] = 0x24; break; case AUTO_SEQ_SURROUND: - spec->multiout.dac_nids[i] = 0x25; + spec->multiout.dac_nids[i] = 0x11; break; case AUTO_SEQ_SIDE: - spec->multiout.dac_nids[i] = 0x11; + spec->multiout.dac_nids[i] = 0x25; break; } } @@ -1680,7 +2202,7 @@ static int vt1708B_auto_create_multi_out_ctls(struct via_spec *spec, { char name[32]; static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; - hda_nid_t nid_vols[] = {0x16, 0x27, 0x26, 0x18}; + hda_nid_t nid_vols[] = {0x16, 0x18, 0x26, 0x27}; hda_nid_t nid, nid_vol = 0; int i, err; @@ -1785,6 +2307,8 @@ static int vt1708B_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) if (err < 0) return err; + create_hp_imux(spec); + return 0; } @@ -1795,7 +2319,7 @@ static int vt1708B_auto_create_analog_input_ctls(struct via_spec *spec, static char *labels[] = { "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL }; - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx = 0; /* for internal loopback recording select */ @@ -1869,7 +2393,10 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec) if (spec->kctl_alloc) spec->mixers[spec->num_mixers++] = spec->kctl_alloc; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; return 1; } @@ -1890,7 +2417,7 @@ static int patch_vt1708B_8ch(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; @@ -1906,7 +2433,8 @@ static int patch_vt1708B_8ch(struct hda_codec *codec) "from BIOS. Using genenic mode...\n"); } - spec->init_verbs = vt1708B_8ch_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1708B_8ch_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1708B_uniwill_init_verbs; spec->stream_name_analog = "VT1708B Analog"; spec->stream_analog_playback = &vt1708B_8ch_pcm_analog_playback; @@ -1926,6 +2454,7 @@ static int patch_vt1708B_8ch(struct hda_codec *codec) codec->patch_ops = via_patch_ops; codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event; #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt1708B_loopbacks; #endif @@ -1939,7 +2468,7 @@ static int patch_vt1708B_4ch(struct hda_codec *codec) int err; /* create a codec specific record */ - spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); + spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; @@ -1955,7 +2484,8 @@ static int patch_vt1708B_4ch(struct hda_codec *codec) "from BIOS. Using genenic mode...\n"); } - spec->init_verbs = vt1708B_4ch_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1708B_4ch_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1708B_uniwill_init_verbs; spec->stream_name_analog = "VT1708B Analog"; spec->stream_analog_playback = &vt1708B_4ch_pcm_analog_playback; @@ -1975,6 +2505,7 @@ static int patch_vt1708B_4ch(struct hda_codec *codec) codec->patch_ops = via_patch_ops; codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event; #ifdef CONFIG_SND_HDA_POWER_SAVE spec->loopback.amplist = vt1708B_loopbacks; #endif @@ -1982,6 +2513,752 @@ static int patch_vt1708B_4ch(struct hda_codec *codec) return 0; } +/* Patch for VT1708S */ + +/* VT1708S software backdoor based override for buggy hardware micboost + * setting */ +#define MIC_BOOST_VOLUME(xname, nid) { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = 0, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ + .info = mic_boost_volume_info, \ + .get = snd_hda_mixer_amp_volume_get, \ + .put = snd_hda_mixer_amp_volume_put, \ + .tlv = { .c = mic_boost_tlv }, \ + .private_value = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT) } + +/* capture mixer elements */ +static struct snd_kcontrol_new vt1708S_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x13, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x13, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x14, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x14, 0x0, HDA_INPUT), + MIC_BOOST_VOLUME("Mic Boost Capture Volume", 0x1A), + MIC_BOOST_VOLUME("Front Mic Boost Capture Volume", 0x1E), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 1, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb vt1708S_volume_init_verbs[] = { + /* Unmute ADC0-1 and set the default input to mic-in */ + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the + * analog-loopback mixer widget */ + /* Amp Indices: CD = 1, Mic1 = 2, Line = 3, Mic2 = 4 */ + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, + + /* Setup default input of PW4 to MW0 */ + {0x1d, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* PW9, PW10 Output enable */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + /* Enable Mic Boost Volume backdoor */ + {0x1, 0xf98, 0x1}, + { } +}; + +static struct hda_verb vt1708S_uniwill_init_verbs[] = { + {0x1D, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + { } +}; + +static struct hda_pcm_stream vt1708S_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 8, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_pcm_prepare, + .cleanup = via_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream vt1708S_pcm_analog_capture = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x13, /* NID to query formats and rates */ + .ops = { + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup + }, +}; + +static struct hda_pcm_stream vt1708S_pcm_digital_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare + }, +}; + +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt1708S_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int i; + hda_nid_t nid; + + spec->multiout.num_dacs = cfg->line_outs; + + spec->multiout.dac_nids = spec->private_dac_nids; + + for (i = 0; i < 4; i++) { + nid = cfg->line_out_pins[i]; + if (nid) { + /* config dac list */ + switch (i) { + case AUTO_SEQ_FRONT: + spec->multiout.dac_nids[i] = 0x10; + break; + case AUTO_SEQ_CENLFE: + spec->multiout.dac_nids[i] = 0x24; + break; + case AUTO_SEQ_SURROUND: + spec->multiout.dac_nids[i] = 0x11; + break; + case AUTO_SEQ_SIDE: + spec->multiout.dac_nids[i] = 0x25; + break; + } + } + } + + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int vt1708S_auto_create_multi_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + char name[32]; + static const char *chname[4] = { "Front", "Surround", "C/LFE", "Side" }; + hda_nid_t nid_vols[] = {0x10, 0x11, 0x24, 0x25}; + hda_nid_t nid_mutes[] = {0x1C, 0x18, 0x26, 0x27}; + hda_nid_t nid, nid_vol, nid_mute; + int i, err; + + for (i = 0; i <= AUTO_SEQ_SIDE; i++) { + nid = cfg->line_out_pins[i]; + + if (!nid) + continue; + + nid_vol = nid_vols[i]; + nid_mute = nid_mutes[i]; + + if (i == AUTO_SEQ_CENLFE) { + /* Center/LFE */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Center Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "LFE Playback Volume", + HDA_COMPOSE_AMP_VAL(nid_vol, 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Center Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, + 1, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "LFE Playback Switch", + HDA_COMPOSE_AMP_VAL(nid_mute, + 2, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else if (i == AUTO_SEQ_FRONT) { + /* add control to mixer index 0 */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x16, 3, 0, + HDA_INPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x16, 3, 0, + HDA_INPUT)); + if (err < 0) + return err; + + /* Front */ + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, + 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } else { + sprintf(name, "%s Playback Volume", chname[i]); + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, name, + HDA_COMPOSE_AMP_VAL(nid_vol, 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + sprintf(name, "%s Playback Switch", chname[i]); + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(nid_mute, + 3, 0, + HDA_OUTPUT)); + if (err < 0) + return err; + } + } + + return 0; +} + +static int vt1708S_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = VT1708S_HP_NID; /* AOW3 */ + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL(0x25, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt1708S_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 5; + imux->num_items++; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x1a: /* Mic */ + idx = 2; + break; + + case 0x1b: /* Line In */ + idx = 3; + break; + + case 0x1e: /* Front Mic */ + idx = 4; + break; + + case 0x1f: /* CD */ + idx = 1; + break; + } + err = via_new_analog_input(spec, cfg->input_pins[i], labels[i], + idx, 0x16); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx-1; + imux->num_items++; + } + return 0; +} + +static int vt1708S_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + static hda_nid_t vt1708s_ignore[] = {0x21, 0}; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, + vt1708s_ignore); + if (err < 0) + return err; + err = vt1708S_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) + return 0; /* can't find valid BIOS pin config */ + + err = vt1708S_auto_create_multi_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt1708S_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt1708S_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + if (spec->autocfg.dig_out_pin) + spec->multiout.dig_out_nid = VT1708S_DIGOUT_NID; + + spec->extra_dig_out_nid = 0x15; + + if (spec->kctl_alloc) + spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt1708S_loopbacks[] = { + { 0x16, HDA_INPUT, 1 }, + { 0x16, HDA_INPUT, 2 }, + { 0x16, HDA_INPUT, 3 }, + { 0x16, HDA_INPUT, 4 }, + { } /* end */ +}; +#endif + +static int patch_vt1708S(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt1708S_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + spec->init_verbs[spec->num_iverbs++] = vt1708S_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1708S_uniwill_init_verbs; + + spec->stream_name_analog = "VT1708S Analog"; + spec->stream_analog_playback = &vt1708S_pcm_analog_playback; + spec->stream_analog_capture = &vt1708S_pcm_analog_capture; + + spec->stream_name_digital = "VT1708S Digital"; + spec->stream_digital_playback = &vt1708S_pcm_digital_playback; + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt1708S_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt1708S_adc_nids); + spec->mixers[spec->num_mixers] = vt1708S_capture_mixer; + spec->num_mixers++; + } + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event; +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt1708S_loopbacks; +#endif + + return 0; +} + +/* Patch for VT1702 */ + +/* capture mixer elements */ +static struct snd_kcontrol_new vt1702_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x20, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x20, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x1F, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x1F, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Digital Mic Boost Capture Volume", 0x1E, 0x0, + HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 1, + .info = via_mux_enum_info, + .get = via_mux_enum_get, + .put = via_mux_enum_put, + }, + { } /* end */ +}; + +static struct hda_verb vt1702_volume_init_verbs[] = { + /* + * Unmute ADC0-1 and set the default input to mic-in + */ + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1F, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + + /* Unmute input amps (CD, Line In, Mic 1 & Mic 2) of the analog-loopback + * mixer widget + */ + /* Amp Indices: Mic1 = 1, Line = 1, Mic2 = 3 */ + {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, + {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, + {0x1A, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, + + /* Setup default input of PW4 to MW0 */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* PW6 PW7 Output enable */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + {0x1C, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40}, + { } +}; + +static struct hda_verb vt1702_uniwill_init_verbs[] = { + {0x01, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_GPIO_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | VIA_HP_EVENT}, + { } +}; + +static struct hda_pcm_stream vt1702_pcm_analog_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + .nid = 0x10, /* NID to query formats and rates */ + .ops = { + .open = via_playback_pcm_open, + .prepare = via_playback_multi_pcm_prepare, + .cleanup = via_playback_multi_pcm_cleanup + }, +}; + +static struct hda_pcm_stream vt1702_pcm_analog_capture = { + .substreams = 3, + .channels_min = 2, + .channels_max = 2, + .nid = 0x12, /* NID to query formats and rates */ + .ops = { + .prepare = via_capture_pcm_prepare, + .cleanup = via_capture_pcm_cleanup + }, +}; + +static struct hda_pcm_stream vt1702_pcm_digital_playback = { + .substreams = 2, + .channels_min = 2, + .channels_max = 2, + /* NID is set in via_build_pcms */ + .ops = { + .open = via_dig_playback_pcm_open, + .close = via_dig_playback_pcm_close, + .prepare = via_dig_playback_pcm_prepare + }, +}; + +/* fill in the dac_nids table from the parsed pin configuration */ +static int vt1702_auto_fill_dac_nids(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + spec->multiout.num_dacs = 1; + spec->multiout.dac_nids = spec->private_dac_nids; + + if (cfg->line_out_pins[0]) { + /* config dac list */ + spec->multiout.dac_nids[0] = 0x10; + } + + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int vt1702_auto_create_line_out_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + int err; + + if (!cfg->line_out_pins[0]) + return -1; + + /* add control to mixer index 0 */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x1A, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x1A, 3, 0, HDA_INPUT)); + if (err < 0) + return err; + + /* Front */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x10, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x16, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + return 0; +} + +static int vt1702_auto_create_hp_ctls(struct via_spec *spec, hda_nid_t pin) +{ + int err; + + if (!pin) + return 0; + + spec->multiout.hp_nid = 0x1D; + + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Headphone Playback Volume", + HDA_COMPOSE_AMP_VAL(0x1D, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Headphone Playback Switch", + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + + create_hp_imux(spec); + + return 0; +} + +/* create playback/capture controls for input pins */ +static int vt1702_auto_create_analog_input_ctls(struct via_spec *spec, + const struct auto_pin_cfg *cfg) +{ + static char *labels[] = { + "Mic", "Front Mic", "Line", "Front Line", "CD", "Aux", NULL + }; + struct hda_input_mux *imux = &spec->private_imux[0]; + int i, err, idx = 0; + + /* for internal loopback recording select */ + imux->items[imux->num_items].label = "Stereo Mixer"; + imux->items[imux->num_items].index = 3; + imux->num_items++; + + for (i = 0; i < AUTO_PIN_LAST; i++) { + if (!cfg->input_pins[i]) + continue; + + switch (cfg->input_pins[i]) { + case 0x14: /* Mic */ + idx = 1; + break; + + case 0x15: /* Line In */ + idx = 2; + break; + + case 0x18: /* Front Mic */ + idx = 3; + break; + } + err = via_new_analog_input(spec, cfg->input_pins[i], + labels[i], idx, 0x1A); + if (err < 0) + return err; + imux->items[imux->num_items].label = labels[i]; + imux->items[imux->num_items].index = idx-1; + imux->num_items++; + } + return 0; +} + +static int vt1702_parse_auto_config(struct hda_codec *codec) +{ + struct via_spec *spec = codec->spec; + int err; + static hda_nid_t vt1702_ignore[] = {0x1C, 0}; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, + vt1702_ignore); + if (err < 0) + return err; + err = vt1702_auto_fill_dac_nids(spec, &spec->autocfg); + if (err < 0) + return err; + if (!spec->autocfg.line_outs && !spec->autocfg.hp_pins[0]) + return 0; /* can't find valid BIOS pin config */ + + err = vt1702_auto_create_line_out_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + err = vt1702_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]); + if (err < 0) + return err; + err = vt1702_auto_create_analog_input_ctls(spec, &spec->autocfg); + if (err < 0) + return err; + + spec->multiout.max_channels = spec->multiout.num_dacs * 2; + + if (spec->autocfg.dig_out_pin) + spec->multiout.dig_out_nid = VT1702_DIGOUT_NID; + + spec->extra_dig_out_nid = 0x1B; + + if (spec->kctl_alloc) + spec->mixers[spec->num_mixers++] = spec->kctl_alloc; + + spec->input_mux = &spec->private_imux[0]; + + if (spec->hp_mux) + spec->mixers[spec->num_mixers++] = via_hp_mixer; + + return 1; +} + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static struct hda_amp_list vt1702_loopbacks[] = { + { 0x1A, HDA_INPUT, 1 }, + { 0x1A, HDA_INPUT, 2 }, + { 0x1A, HDA_INPUT, 3 }, + { 0x1A, HDA_INPUT, 4 }, + { } /* end */ +}; +#endif + +static int patch_vt1702(struct hda_codec *codec) +{ + struct via_spec *spec; + int err; + unsigned int response; + unsigned char control; + + /* create a codec specific record */ + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + + /* automatic parse from the BIOS config */ + err = vt1702_parse_auto_config(codec); + if (err < 0) { + via_free(codec); + return err; + } else if (!err) { + printk(KERN_INFO "hda_codec: Cannot set up configuration " + "from BIOS. Using genenic mode...\n"); + } + + spec->init_verbs[spec->num_iverbs++] = vt1702_volume_init_verbs; + spec->init_verbs[spec->num_iverbs++] = vt1702_uniwill_init_verbs; + + spec->stream_name_analog = "VT1702 Analog"; + spec->stream_analog_playback = &vt1702_pcm_analog_playback; + spec->stream_analog_capture = &vt1702_pcm_analog_capture; + + spec->stream_name_digital = "VT1702 Digital"; + spec->stream_digital_playback = &vt1702_pcm_digital_playback; + + if (!spec->adc_nids && spec->input_mux) { + spec->adc_nids = vt1702_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(vt1702_adc_nids); + spec->mixers[spec->num_mixers] = vt1702_capture_mixer; + spec->num_mixers++; + } + + codec->patch_ops = via_patch_ops; + + codec->patch_ops.init = via_auto_init; + codec->patch_ops.unsol_event = via_unsol_event; +#ifdef CONFIG_SND_HDA_POWER_SAVE + spec->loopback.amplist = vt1702_loopbacks; +#endif + + /* Open backdoor */ + response = snd_hda_codec_read(codec, codec->afg, 0, 0xF8C, 0); + control = (unsigned char)(response & 0xff); + control |= 0x3; + snd_hda_codec_write(codec, codec->afg, 0, 0xF88, control); + + /* Enable GPIO 0&1 for volume&mute control */ + /* Enable GPIO 2 for DMIC-DATA */ + response = snd_hda_codec_read(codec, codec->afg, 0, 0xF84, 0); + control = (unsigned char)((response >> 16) & 0x3f); + snd_hda_codec_write(codec, codec->afg, 0, 0xF82, control); + + return 0; +} + /* * patch entries */ @@ -2022,5 +3299,37 @@ struct hda_codec_preset snd_hda_preset_via[] = { .patch = patch_vt1708B_4ch}, { .id = 0x1106E727, .name = "VIA VT1708B 4-Ch", .patch = patch_vt1708B_4ch}, + { .id = 0x11060397, .name = "VIA VT1708S", + .patch = patch_vt1708S}, + { .id = 0x11061397, .name = "VIA VT1708S", + .patch = patch_vt1708S}, + { .id = 0x11062397, .name = "VIA VT1708S", + .patch = patch_vt1708S}, + { .id = 0x11063397, .name = "VIA VT1708S", + .patch = patch_vt1708S}, + { .id = 0x11064397, .name = "VIA VT1708S", + .patch = patch_vt1708S}, + { .id = 0x11065397, .name = "VIA VT1708S", + .patch = patch_vt1708S}, + { .id = 0x11066397, .name = "VIA VT1708S", + .patch = patch_vt1708S}, + { .id = 0x11067397, .name = "VIA VT1708S", + .patch = patch_vt1708S}, + { .id = 0x11060398, .name = "VIA VT1702", + .patch = patch_vt1702}, + { .id = 0x11061398, .name = "VIA VT1702", + .patch = patch_vt1702}, + { .id = 0x11062398, .name = "VIA VT1702", + .patch = patch_vt1702}, + { .id = 0x11063398, .name = "VIA VT1702", + .patch = patch_vt1702}, + { .id = 0x11064398, .name = "VIA VT1702", + .patch = patch_vt1702}, + { .id = 0x11065398, .name = "VIA VT1702", + .patch = patch_vt1702}, + { .id = 0x11066398, .name = "VIA VT1702", + .patch = patch_vt1702}, + { .id = 0x11067398, .name = "VIA VT1702", + .patch = patch_vt1702}, {} /* terminator */ }; |