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-rw-r--r--sound/pci/hda/Makefile2
-rw-r--r--sound/pci/hda/hda_beep.c134
-rw-r--r--sound/pci/hda/hda_beep.h44
-rw-r--r--sound/pci/hda/hda_codec.c113
-rw-r--r--sound/pci/hda/hda_codec.h93
-rw-r--r--sound/pci/hda/hda_generic.c3
-rw-r--r--sound/pci/hda/hda_intel.c146
-rw-r--r--sound/pci/hda/hda_local.h24
-rw-r--r--sound/pci/hda/hda_patch.h2
-rw-r--r--sound/pci/hda/hda_proc.c25
-rw-r--r--sound/pci/hda/patch_analog.c96
-rw-r--r--sound/pci/hda/patch_atihdmi.c45
-rw-r--r--sound/pci/hda/patch_nvhdmi.c165
-rw-r--r--sound/pci/hda/patch_realtek.c1900
-rw-r--r--sound/pci/hda/patch_sigmatel.c1008
-rw-r--r--sound/pci/hda/patch_via.c1407
16 files changed, 4730 insertions, 477 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..2eed2c8b98d
--- /dev/null
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -0,0 +1,165 @@
+/*
+ * 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->pcm_type = HDA_PCM_TYPE_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 f3da621f25c..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 */
};
@@ -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 */
};
generated by cgit v1.2.3-70-g09d2 (git 2.46.0) at 2024-12-30 19:43:49 +0000