From a1e21c9078fb8005e5accb921696ec9e2f38176e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 17 Jun 2009 09:33:52 +0200
Subject: ALSA: hda - Don't call snd_hda_codec_configure in snd_hda_codec_new()

The codec setup call via snd_hda_codec_configure() isn't necessarily
called in snd_hda_codec_new().  For the later added feature, it's better
to change the code flow like:
 - create all codec instances
 - configure each codec

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c |  8 ++------
 sound/pci/hda/hda_codec.h |  3 ++-
 sound/pci/hda/hda_intel.c | 21 +++++++++++++++++----
 sound/pci/hda/hda_local.h |  1 -
 4 files changed, 21 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 462e2cedaa6..506f46ef030 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -885,7 +885,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
  * Returns 0 if successful, or a negative error code.
  */
 int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
-				    int do_init, struct hda_codec **codecp)
+				    struct hda_codec **codecp)
 {
 	struct hda_codec *codec;
 	char component[31];
@@ -978,11 +978,6 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr
 			    codec->afg ? codec->afg : codec->mfg,
 			    AC_PWRST_D0);
 
-	if (do_init) {
-		err = snd_hda_codec_configure(codec);
-		if (err < 0)
-			goto error;
-	}
 	snd_hda_codec_proc_new(codec);
 
 	snd_hda_create_hwdep(codec);
@@ -1036,6 +1031,7 @@ int snd_hda_codec_configure(struct hda_codec *codec)
 		err = init_unsol_queue(codec->bus);
 	return err;
 }
+EXPORT_SYMBOL_HDA(snd_hda_codec_configure);
 
 /**
  * snd_hda_codec_setup_stream - set up the codec for streaming
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index cad79efaabc..b7ca7d5bbe8 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -830,7 +830,8 @@ enum {
 int snd_hda_bus_new(struct snd_card *card, const struct hda_bus_template *temp,
 		    struct hda_bus **busp);
 int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
-		      int do_init, struct hda_codec **codecp);
+		      struct hda_codec **codecp);
+int snd_hda_codec_configure(struct hda_codec *codec);
 
 /*
  * low level functions
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 4e9ea708027..da58f2ca915 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1286,8 +1286,7 @@ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] __devinitdata = {
 	[AZX_DRIVER_TERA] = 1,
 };
 
-static int __devinit azx_codec_create(struct azx *chip, const char *model,
-				      int no_init)
+static int __devinit azx_codec_create(struct azx *chip, const char *model)
 {
 	struct hda_bus_template bus_temp;
 	int c, codecs, err;
@@ -1346,7 +1345,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model,
 	for (c = 0; c < max_slots; c++) {
 		if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) {
 			struct hda_codec *codec;
-			err = snd_hda_codec_new(chip->bus, c, !no_init, &codec);
+			err = snd_hda_codec_new(chip->bus, c, &codec);
 			if (err < 0)
 				continue;
 			codecs++;
@@ -1356,7 +1355,16 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model,
 		snd_printk(KERN_ERR SFX "no codecs initialized\n");
 		return -ENXIO;
 	}
+	return 0;
+}
 
+/* configure each codec instance */
+static int __devinit azx_codec_configure(struct azx *chip)
+{
+	struct hda_codec *codec;
+	list_for_each_entry(codec, &chip->bus->codec_list, list) {
+		snd_hda_codec_configure(codec);
+	}
 	return 0;
 }
 
@@ -2466,9 +2474,14 @@ static int __devinit azx_probe(struct pci_dev *pci,
 	card->private_data = chip;
 
 	/* create codec instances */
-	err = azx_codec_create(chip, model[dev], probe_only[dev]);
+	err = azx_codec_create(chip, model[dev]);
 	if (err < 0)
 		goto out_free;
+	if (!probe_only[dev]) {
+		err = azx_codec_configure(chip);
+		if (err < 0)
+			goto out_free;
+	}
 
 	/* create PCM streams */
 	err = snd_hda_build_pcms(chip->bus);
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 83349013b4d..75aa3785212 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -99,7 +99,6 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
 int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
 			unsigned int *tlv, const char **slaves);
 int snd_hda_codec_reset(struct hda_codec *codec);
-int snd_hda_codec_configure(struct hda_codec *codec);
 
 /* amp value bits */
 #define HDA_AMP_MUTE	0x80
-- 
cgit v1.2.3


From 4ea6fbc8eb23c3ae5fd2fb55a340ab85c8649bce Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 17 Jun 2009 09:52:54 +0200
Subject: ALSA: hda - Add patch module option

Added the patch module option to apply a "patch" as a firmware to
modify pin configurations or give additional hints to the driver
before actually initializing and configuring the codec.

This can be used as a workaround when the BIOS doesn't give sufficient
information or give wrong information that doesn't match with the real
hardware setup, until it's fixed statically in the driver via a quirk.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/Kconfig     |  14 +++
 sound/pci/hda/hda_codec.h |   7 ++
 sound/pci/hda/hda_hwdep.c | 236 ++++++++++++++++++++++++++++++++++++++++++----
 sound/pci/hda/hda_intel.c |  21 ++++-
 4 files changed, 257 insertions(+), 21 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index 04438f1d682..b8a77f9b082 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -46,6 +46,20 @@ config SND_HDA_INPUT_JACK
 	  Say Y here to enable the jack plugging notification via
 	  input layer.
 
+config SND_HDA_PATCH_LOADER
+	bool "Support initialization patch loading for HD-audio"
+	depends on EXPERIMENTAL
+	select FW_LOADER
+	select SND_HDA_HWDEP
+	select SND_HDA_RECONFIG
+	help
+	  Say Y here to allow the HD-audio driver to load a pseudo
+	  firmware file ("patch") for overriding the BIOS setup at
+	  start up.  The "patch" file can be specified via patch module
+	  option, such as patch=hda-init.
+
+	  This option turns on hwdep and reconfig features automatically.
+
 config SND_HDA_CODEC_REALTEK
 	bool "Build Realtek HD-audio codec support"
 	default y
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index b7ca7d5bbe8..72c997592ee 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -939,6 +939,13 @@ static inline void snd_hda_power_down(struct hda_codec *codec) {}
 #define snd_hda_codec_needs_resume(codec) 1
 #endif
 
+#ifdef CONFIG_SND_HDA_PATCH_LOADER
+/*
+ * patch firmware
+ */
+int snd_hda_load_patch(struct hda_bus *bus, const char *patch);
+#endif
+
 /*
  * Codec modularization
  */
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 6812fbe80fa..cc24e6721d7 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -24,6 +24,7 @@
 #include <linux/compat.h>
 #include <linux/mutex.h>
 #include <linux/ctype.h>
+#include <linux/firmware.h>
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
@@ -312,12 +313,8 @@ static ssize_t init_verbs_show(struct device *dev,
 	return len;
 }
 
-static ssize_t init_verbs_store(struct device *dev,
-				struct device_attribute *attr,
-				const char *buf, size_t count)
+static int parse_init_verbs(struct hda_codec *codec, const char *buf)
 {
-	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
-	struct hda_codec *codec = hwdep->private_data;
 	struct hda_verb *v;
 	int nid, verb, param;
 
@@ -331,6 +328,18 @@ static ssize_t init_verbs_store(struct device *dev,
 	v->nid = nid;
 	v->verb = verb;
 	v->param = param;
+	return 0;
+}
+
+static ssize_t init_verbs_store(struct device *dev,
+				struct device_attribute *attr,
+				const char *buf, size_t count)
+{
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+	struct hda_codec *codec = hwdep->private_data;
+	int err = parse_init_verbs(codec, buf);
+	if (err < 0)
+		return err;
 	return count;
 }
 
@@ -376,19 +385,15 @@ static void remove_trail_spaces(char *str)
 
 #define MAX_HINTS	1024
 
-static ssize_t hints_store(struct device *dev,
-			   struct device_attribute *attr,
-			   const char *buf, size_t count)
+static int parse_hints(struct hda_codec *codec, const char *buf)
 {
-	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
-	struct hda_codec *codec = hwdep->private_data;
 	char *key, *val;
 	struct hda_hint *hint;
 
 	while (isspace(*buf))
 		buf++;
 	if (!*buf || *buf == '#' || *buf == '\n')
-		return count;
+		return 0;
 	if (*buf == '=')
 		return -EINVAL;
 	key = kstrndup_noeol(buf, 1024);
@@ -411,7 +416,7 @@ static ssize_t hints_store(struct device *dev,
 		kfree(hint->key);
 		hint->key = key;
 		hint->val = val;
-		return count;
+		return 0;
 	}
 	/* allocate a new hint entry */
 	if (codec->hints.used >= MAX_HINTS)
@@ -424,6 +429,18 @@ static ssize_t hints_store(struct device *dev,
 	}
 	hint->key = key;
 	hint->val = val;
+	return 0;
+}
+
+static ssize_t hints_store(struct device *dev,
+			   struct device_attribute *attr,
+			   const char *buf, size_t count)
+{
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+	struct hda_codec *codec = hwdep->private_data;
+	int err = parse_hints(codec, buf);
+	if (err < 0)
+		return err;
 	return count;
 }
 
@@ -469,20 +486,24 @@ static ssize_t driver_pin_configs_show(struct device *dev,
 
 #define MAX_PIN_CONFIGS		32
 
-static ssize_t user_pin_configs_store(struct device *dev,
-				      struct device_attribute *attr,
-				      const char *buf, size_t count)
+static int parse_user_pin_configs(struct hda_codec *codec, const char *buf)
 {
-	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
-	struct hda_codec *codec = hwdep->private_data;
 	int nid, cfg;
-	int err;
 
 	if (sscanf(buf, "%i %i", &nid, &cfg) != 2)
 		return -EINVAL;
 	if (!nid)
 		return -EINVAL;
-	err = snd_hda_add_pincfg(codec, &codec->user_pins, nid, cfg);
+	return snd_hda_add_pincfg(codec, &codec->user_pins, nid, cfg);
+}
+
+static ssize_t user_pin_configs_store(struct device *dev,
+				      struct device_attribute *attr,
+				      const char *buf, size_t count)
+{
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+	struct hda_codec *codec = hwdep->private_data;
+	int err = parse_user_pin_configs(codec, buf);
 	if (err < 0)
 		return err;
 	return count;
@@ -553,3 +574,180 @@ int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key)
 EXPORT_SYMBOL_HDA(snd_hda_get_bool_hint);
 
 #endif /* CONFIG_SND_HDA_RECONFIG */
+
+#ifdef CONFIG_SND_HDA_PATCH_LOADER
+
+/* parser mode */
+enum {
+	LINE_MODE_NONE,
+	LINE_MODE_CODEC,
+	LINE_MODE_MODEL,
+	LINE_MODE_PINCFG,
+	LINE_MODE_VERB,
+	LINE_MODE_HINT,
+	NUM_LINE_MODES,
+};
+
+static inline int strmatch(const char *a, const char *b)
+{
+	return strnicmp(a, b, strlen(b)) == 0;
+}
+
+/* parse the contents after the line "[codec]"
+ * accept only the line with three numbers, and assign the current codec
+ */
+static void parse_codec_mode(char *buf, struct hda_bus *bus,
+			     struct hda_codec **codecp)
+{
+	unsigned int vendorid, subid, caddr;
+	struct hda_codec *codec;
+
+	*codecp = NULL;
+	if (sscanf(buf, "%i %i %i", &vendorid, &subid, &caddr) == 3) {
+		list_for_each_entry(codec, &bus->codec_list, list) {
+			if (codec->addr == caddr) {
+				*codecp = codec;
+				break;
+			}
+		}
+	}
+}
+
+/* parse the contents after the other command tags, [pincfg], [verb],
+ * [hint] and [model]
+ * just pass to the sysfs helper (only when any codec was specified)
+ */
+static void parse_pincfg_mode(char *buf, struct hda_bus *bus,
+			      struct hda_codec **codecp)
+{
+	if (!*codecp)
+		return;
+	parse_user_pin_configs(*codecp, buf);
+}
+
+static void parse_verb_mode(char *buf, struct hda_bus *bus,
+			    struct hda_codec **codecp)
+{
+	if (!*codecp)
+		return;
+	parse_init_verbs(*codecp, buf);
+}
+
+static void parse_hint_mode(char *buf, struct hda_bus *bus,
+			    struct hda_codec **codecp)
+{
+	if (!*codecp)
+		return;
+	parse_hints(*codecp, buf);
+}
+
+static void parse_model_mode(char *buf, struct hda_bus *bus,
+			     struct hda_codec **codecp)
+{
+	if (!*codecp)
+		return;
+	kfree((*codecp)->modelname);
+	(*codecp)->modelname = kstrdup(buf, GFP_KERNEL);
+}
+
+struct hda_patch_item {
+	const char *tag;
+	void (*parser)(char *buf, struct hda_bus *bus, struct hda_codec **retc);
+};
+
+static struct hda_patch_item patch_items[NUM_LINE_MODES] = {
+	[LINE_MODE_CODEC] = { "[codec]", parse_codec_mode },
+	[LINE_MODE_MODEL] = { "[model]", parse_model_mode },
+	[LINE_MODE_VERB] = { "[verb]", parse_verb_mode },
+	[LINE_MODE_PINCFG] = { "[pincfg]", parse_pincfg_mode },
+	[LINE_MODE_HINT] = { "[hint]", parse_hint_mode },
+};
+
+/* check the line starting with '[' -- change the parser mode accodingly */
+static int parse_line_mode(char *buf, struct hda_bus *bus)
+{
+	int i;
+	for (i = 0; i < ARRAY_SIZE(patch_items); i++) {
+		if (!patch_items[i].tag)
+			continue;
+		if (strmatch(buf, patch_items[i].tag))
+			return i;
+	}
+	return LINE_MODE_NONE;
+}
+
+/* copy one line from the buffer in fw, and update the fields in fw
+ * return zero if it reaches to the end of the buffer, or non-zero
+ * if successfully copied a line
+ *
+ * the spaces at the beginning and the end of the line are stripped
+ */
+static int get_line_from_fw(char *buf, int size, struct firmware *fw)
+{
+	int len;
+	const char *p = fw->data;
+	while (isspace(*p) && fw->size) {
+		p++;
+		fw->size--;
+	}
+	if (!fw->size)
+		return 0;
+	if (size < fw->size)
+		size = fw->size;
+
+	for (len = 0; len < fw->size; len++) {
+		if (!*p)
+			break;
+		if (*p == '\n') {
+			p++;
+			len++;
+			break;
+		}
+		if (len < size)
+			*buf++ = *p++;
+	}
+	*buf = 0;
+	fw->size -= len;
+	fw->data = p;
+	remove_trail_spaces(buf);
+	return 1;
+}
+
+/*
+ * load a "patch" firmware file and parse it
+ */
+int snd_hda_load_patch(struct hda_bus *bus, const char *patch)
+{
+	int err;
+	const struct firmware *fw;
+	struct firmware tmp;
+	char buf[128];
+	struct hda_codec *codec;
+	int line_mode;
+	struct device *dev = bus->card->dev;
+
+	if (snd_BUG_ON(!dev))
+		return -ENODEV;
+	err = request_firmware(&fw, patch, dev);
+	if (err < 0) {
+		printk(KERN_ERR "hda-codec: Cannot load the patch '%s'\n",
+		       patch);
+		return err;
+	}
+
+	tmp = *fw;
+	line_mode = LINE_MODE_NONE;
+	codec = NULL;
+	while (get_line_from_fw(buf, sizeof(buf) - 1, &tmp)) {
+		if (!*buf || *buf == '#' || *buf == '\n')
+			continue;
+		if (*buf == '[')
+			line_mode = parse_line_mode(buf, bus);
+		else if (patch_items[line_mode].parser)
+			patch_items[line_mode].parser(buf, bus, &codec);
+	}
+	release_firmware(fw);
+	return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_load_patch);
+#endif /* CONFIG_SND_HDA_PATCH_LOADER */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index da58f2ca915..a2f4a116f87 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -61,6 +61,9 @@ static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1};
 static int probe_only[SNDRV_CARDS];
 static int single_cmd;
 static int enable_msi;
+#ifdef CONFIG_SND_HDA_PATCH_LOADER
+static char *patch[SNDRV_CARDS];
+#endif
 
 module_param_array(index, int, NULL, 0444);
 MODULE_PARM_DESC(index, "Index value for Intel HD audio interface.");
@@ -84,6 +87,10 @@ MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs "
 		 "(for debugging only).");
 module_param(enable_msi, int, 0444);
 MODULE_PARM_DESC(enable_msi, "Enable Message Signaled Interrupt (MSI)");
+#ifdef CONFIG_SND_HDA_PATCH_LOADER
+module_param_array(patch, charp, NULL, 0444);
+MODULE_PARM_DESC(patch, "Patch file for Intel HD audio interface.");
+#endif
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT;
@@ -2468,6 +2475,9 @@ static int __devinit azx_probe(struct pci_dev *pci,
 		return err;
 	}
 
+	/* set this here since it's referred in snd_hda_load_patch() */
+	snd_card_set_dev(card, &pci->dev);
+
 	err = azx_create(card, pci, dev, pci_id->driver_data, &chip);
 	if (err < 0)
 		goto out_free;
@@ -2477,6 +2487,15 @@ static int __devinit azx_probe(struct pci_dev *pci,
 	err = azx_codec_create(chip, model[dev]);
 	if (err < 0)
 		goto out_free;
+#ifdef CONFIG_SND_HDA_PATCH_LOADER
+	if (patch[dev]) {
+		snd_printk(KERN_ERR SFX "Applying patch firmware '%s'\n",
+			   patch[dev]);
+		err = snd_hda_load_patch(chip->bus, patch[dev]);
+		if (err < 0)
+			goto out_free;
+	}
+#endif
 	if (!probe_only[dev]) {
 		err = azx_codec_configure(chip);
 		if (err < 0)
@@ -2493,8 +2512,6 @@ static int __devinit azx_probe(struct pci_dev *pci,
 	if (err < 0)
 		goto out_free;
 
-	snd_card_set_dev(card, &pci->dev);
-
 	err = snd_card_register(card);
 	if (err < 0)
 		goto out_free;
-- 
cgit v1.2.3


From b7b51141b4fb6f9059a20c03dd2a5bf77c466c7e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 29 Jun 2009 08:34:06 +0200
Subject: ALSA: hda - Check "beep" hint

Check the hint "beep" in snd_hda_attach_beep_device() to avoid the beep
device creation if user doesn't want.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_beep.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 29272f2e95a..08fe6592ad4 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -24,6 +24,7 @@
 #include <linux/workqueue.h>
 #include <sound/core.h>
 #include "hda_beep.h"
+#include "hda_local.h"
 
 enum {
 	DIGBEEP_HZ_STEP = 46875,	/* 46.875 Hz */
@@ -115,6 +116,9 @@ int snd_hda_attach_beep_device(struct hda_codec *codec, int nid)
 	struct hda_beep *beep;
 	int err;
 
+	if (!snd_hda_get_bool_hint(codec, "beep"))
+		return 0; /* disabled explicitly */
+
 	beep = kzalloc(sizeof(*beep), GFP_KERNEL);
 	if (beep == NULL)
 		return -ENOMEM;
-- 
cgit v1.2.3


From 4953550a6ca399b644ef057626617465d8be9a7b Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 30 Jun 2009 15:28:30 +0200
Subject: ALSA: hda - Merge patch_alc882() and patch_alc883()

Merge patch_alc882() and patch_alc883() to the former one since both
codecs have fairly similar connections but just a slight difference.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 1979 ++++++++++++++++-------------------------
 1 file changed, 762 insertions(+), 1217 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3a8e58c483d..6a899e8fdd0 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -208,12 +208,6 @@ enum {
 	ALC885_MBP3,
 	ALC885_MB5,
 	ALC885_IMAC24,
-	ALC882_AUTO,
-	ALC882_MODEL_LAST,
-};
-
-/* ALC883 models */
-enum {
 	ALC883_3ST_2ch_DIG,
 	ALC883_3ST_6ch_DIG,
 	ALC883_3ST_6ch,
@@ -246,8 +240,8 @@ enum {
 	ALC889A_MB31,
 	ALC1200_ASUS_P5Q,
 	ALC883_SONY_VAIO_TT,
-	ALC883_AUTO,
-	ALC883_MODEL_LAST,
+	ALC882_AUTO,
+	ALC882_MODEL_LAST,
 };
 
 /* for GPIO Poll */
@@ -320,6 +314,8 @@ struct alc_spec {
 	struct snd_array kctls;
 	struct hda_input_mux private_imux[3];
 	hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
+	hda_nid_t private_adc_nids[AUTO_CFG_MAX_OUTS];
+	hda_nid_t private_capsrc_nids[AUTO_CFG_MAX_OUTS];
 
 	/* hooks */
 	void (*init_hook)(struct hda_codec *codec);
@@ -6295,7 +6291,7 @@ static int patch_alc260(struct hda_codec *codec)
 
 
 /*
- * ALC882 support
+ * ALC882/883/885/888/889 support
  *
  * ALC882 is almost identical with ALC880 but has cleaner and more flexible
  * configuration.  Each pin widget can choose any input DACs and a mixer.
@@ -6307,22 +6303,35 @@ static int patch_alc260(struct hda_codec *codec)
  */
 #define ALC882_DIGOUT_NID	0x06
 #define ALC882_DIGIN_NID	0x0a
+#define ALC883_DIGOUT_NID	ALC882_DIGOUT_NID
+#define ALC883_DIGIN_NID	ALC882_DIGIN_NID
+#define ALC1200_DIGOUT_NID	0x10
+
 
 static struct hda_channel_mode alc882_ch_modes[1] = {
 	{ 8, NULL }
 };
 
+/* DACs */
 static hda_nid_t alc882_dac_nids[4] = {
 	/* front, rear, clfe, rear_surr */
 	0x02, 0x03, 0x04, 0x05
 };
+#define alc883_dac_nids		alc882_dac_nids
 
-/* identical with ALC880 */
+/* ADCs */
 #define alc882_adc_nids		alc880_adc_nids
 #define alc882_adc_nids_alt	alc880_adc_nids_alt
+#define alc883_adc_nids		alc882_adc_nids_alt
+static hda_nid_t alc883_adc_nids_alt[1] = { 0x08 };
+static hda_nid_t alc883_adc_nids_rev[2] = { 0x09, 0x08 };
+#define alc889_adc_nids		alc880_adc_nids
 
 static hda_nid_t alc882_capsrc_nids[3] = { 0x24, 0x23, 0x22 };
 static hda_nid_t alc882_capsrc_nids_alt[2] = { 0x23, 0x22 };
+#define alc883_capsrc_nids	alc882_capsrc_nids_alt
+static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 };
+#define alc889_capsrc_nids	alc882_capsrc_nids
 
 /* input MUX */
 /* FIXME: should be a matrix-type input source selection */
@@ -6337,6 +6346,8 @@ static struct hda_input_mux alc882_capture_source = {
 	},
 };
 
+#define alc883_capture_source	alc882_capture_source
+
 static struct hda_input_mux mb5_capture_source = {
 	.num_items = 3,
 	.items = {
@@ -6346,6 +6357,77 @@ static struct hda_input_mux mb5_capture_source = {
 	},
 };
 
+static struct hda_input_mux alc883_3stack_6ch_intel = {
+	.num_items = 4,
+	.items = {
+		{ "Mic", 0x1 },
+		{ "Front Mic", 0x0 },
+		{ "Line", 0x2 },
+		{ "CD", 0x4 },
+	},
+};
+
+static struct hda_input_mux alc883_lenovo_101e_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Mic", 0x1 },
+		{ "Line", 0x2 },
+	},
+};
+
+static struct hda_input_mux alc883_lenovo_nb0763_capture_source = {
+	.num_items = 4,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "iMic", 0x1 },
+		{ "Line", 0x2 },
+		{ "CD", 0x4 },
+	},
+};
+
+static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Int Mic", 0x1 },
+	},
+};
+
+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 },
+	},
+};
+
+static struct hda_input_mux alc889A_mb31_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Mic", 0x0 },
+		/* Front Mic (0x01) unused */
+		{ "Line", 0x2 },
+		/* Line 2 (0x03) unused */
+		/* CD (0x04) unsused? */
+	},
+};
+
+/*
+ * 2ch mode
+ */
+static struct hda_channel_mode alc883_3ST_2ch_modes[1] = {
+	{ 2, NULL }
+};
+
 /*
  * 2ch mode
  */
@@ -6357,6 +6439,18 @@ static struct hda_verb alc882_3ST_ch2_init[] = {
 	{ } /* end */
 };
 
+/*
+ * 4ch mode
+ */
+static struct hda_verb alc882_3ST_ch4_init[] = {
+	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+	{ } /* end */
+};
+
 /*
  * 6ch mode
  */
@@ -6370,11 +6464,14 @@ static struct hda_verb alc882_3ST_ch6_init[] = {
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc882_3ST_6ch_modes[2] = {
+static struct hda_channel_mode alc882_3ST_6ch_modes[3] = {
 	{ 2, alc882_3ST_ch2_init },
+	{ 4, alc882_3ST_ch4_init },
 	{ 6, alc882_3ST_ch6_init },
 };
 
+#define alc883_3ST_6ch_modes	alc882_3ST_6ch_modes
+
 /*
  * 6ch mode
  */
@@ -6462,6 +6559,143 @@ static struct hda_channel_mode alc885_mb5_6ch_modes[2] = {
 	{ 6, alc885_mb5_ch6_init },
 };
 
+
+/*
+ * 2ch mode
+ */
+static struct hda_verb alc883_4ST_ch2_init[] = {
+	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+static struct hda_verb alc883_4ST_ch4_init[] = {
+	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+	{ } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc883_4ST_ch6_init[] = {
+	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+	{ } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+static struct hda_verb alc883_4ST_ch8_init[] = {
+	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
+	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+	{ } /* end */
+};
+
+static struct hda_channel_mode alc883_4ST_8ch_modes[4] = {
+	{ 2, alc883_4ST_ch2_init },
+	{ 4, alc883_4ST_ch4_init },
+	{ 6, alc883_4ST_ch6_init },
+	{ 8, alc883_4ST_ch8_init },
+};
+
+
+/*
+ * 2ch mode
+ */
+static struct hda_verb alc883_3ST_ch2_intel_init[] = {
+	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+	{ 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+static struct hda_verb alc883_3ST_ch4_intel_init[] = {
+	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+	{ 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+	{ } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc883_3ST_ch6_intel_init[] = {
+	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x19, AC_VERB_SET_CONNECT_SEL, 0x02 },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+	{ } /* end */
+};
+
+static struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = {
+	{ 2, alc883_3ST_ch2_intel_init },
+	{ 4, alc883_3ST_ch4_intel_init },
+	{ 6, alc883_3ST_ch6_intel_init },
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc883_sixstack_ch6_init[] = {
+	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
+	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+static struct hda_verb alc883_sixstack_ch8_init[] = {
+	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ } /* end */
+};
+
+static struct hda_channel_mode alc883_sixstack_modes[2] = {
+	{ 6, alc883_sixstack_ch6_init },
+	{ 8, alc883_sixstack_ch8_init },
+};
+
+
 /* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
  *                 Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
  */
@@ -6597,7 +6831,7 @@ static struct snd_kcontrol_new alc882_chmode_mixer[] = {
 	{ } /* end */
 };
 
-static struct hda_verb alc882_init_verbs[] = {
+static struct hda_verb alc882_base_init_verbs[] = {
 	/* Front mixer: unmute input/output amp left and right (volume = 0) */
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
 	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
@@ -6649,11 +6883,6 @@ static struct hda_verb alc882_init_verbs[] = {
 
 	/* 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 */
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
 	/* Input mixer2 */
 	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
@@ -6664,9 +6893,6 @@ static struct hda_verb alc882_init_verbs[] = {
 	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
 	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
 	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	/* ADC1: mute amp left and right */
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
 	/* ADC2: mute amp left and right */
 	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
@@ -6677,6 +6903,21 @@ static struct hda_verb alc882_init_verbs[] = {
 	{ }
 };
 
+static struct hda_verb alc882_adc1_init_verbs[] = {
+	/* 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(3)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+	/* ADC1: mute amp left and right */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{ }
+};
+
+/* HACK - expand to two elements */
+#define alc882_init_verbs	alc882_base_init_verbs, alc882_adc1_init_verbs
+
 static struct hda_verb alc882_eapd_verbs[] = {
 	/* change to EAPD mode */
 	{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
@@ -6684,6 +6925,8 @@ static struct hda_verb alc882_eapd_verbs[] = {
 	{ }
 };
 
+#define alc883_init_verbs	alc882_base_init_verbs
+
 /* Mac Pro test */
 static struct snd_kcontrol_new alc882_macpro_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
@@ -7034,882 +7277,62 @@ static void alc885_imac24_init_hook(struct hda_codec *codec)
 /*
  * generic initialization of ADC, input mixers and output mixers
  */
-static struct hda_verb alc882_auto_init_verbs[] = {
+static struct hda_verb alc883_auto_init_verbs[] = {
 	/*
 	 * Unmute ADC0-2 and set the default input to mic-in
 	 */
-	{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-	/* Mute 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
-	 * 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 - 0x0f)
-	 */
-	/* set vol=0 to output mixers */
-	{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},
-	{0x0f, 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)},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
-	/* 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 */
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
-	/* Input mixer2 */
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
-	/* Input mixer3 */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
-
-	{ }
-};
-
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-#define alc882_loopbacks	alc880_loopbacks
-#endif
-
-/* pcm configuration: identical with ALC880 */
-#define alc882_pcm_analog_playback	alc880_pcm_analog_playback
-#define alc882_pcm_analog_capture	alc880_pcm_analog_capture
-#define alc882_pcm_digital_playback	alc880_pcm_digital_playback
-#define alc882_pcm_digital_capture	alc880_pcm_digital_capture
-
-/*
- * configuration and preset
- */
-static const char *alc882_models[ALC882_MODEL_LAST] = {
-	[ALC882_3ST_DIG]	= "3stack-dig",
-	[ALC882_6ST_DIG]	= "6stack-dig",
-	[ALC882_ARIMA]		= "arima",
-	[ALC882_W2JC]		= "w2jc",
-	[ALC882_TARGA]		= "targa",
-	[ALC882_ASUS_A7J]	= "asus-a7j",
-	[ALC882_ASUS_A7M]	= "asus-a7m",
-	[ALC885_MACPRO]		= "macpro",
-	[ALC885_MB5]		= "mb5",
-	[ALC885_MBP3]		= "mbp3",
-	[ALC885_IMAC24]		= "imac24",
-	[ALC882_AUTO]		= "auto",
-};
-
-static struct snd_pci_quirk alc882_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC882_6ST_DIG),
-	SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J),
-	SND_PCI_QUIRK(0x1043, 0x1243, "Asus A7J", ALC882_ASUS_A7J),
-	SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_ASUS_A7M),
-	SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC),
-	SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG),
-	SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG),
-	SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG),
-	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG),
-	SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8  */
-	SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG),
-	SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA),
-	{}
-};
-
-static struct alc_config_preset alc882_presets[] = {
-	[ALC882_3ST_DIG] = {
-		.mixers = { alc882_base_mixer },
-		.init_verbs = { alc882_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
-		.dac_nids = alc882_dac_nids,
-		.dig_out_nid = ALC882_DIGOUT_NID,
-		.dig_in_nid = ALC882_DIGIN_NID,
-		.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
-		.channel_mode = alc882_ch_modes,
-		.need_dac_fix = 1,
-		.input_mux = &alc882_capture_source,
-	},
-	[ALC882_6ST_DIG] = {
-		.mixers = { alc882_base_mixer, alc882_chmode_mixer },
-		.init_verbs = { alc882_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
-		.dac_nids = alc882_dac_nids,
-		.dig_out_nid = ALC882_DIGOUT_NID,
-		.dig_in_nid = ALC882_DIGIN_NID,
-		.num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
-		.channel_mode = alc882_sixstack_modes,
-		.input_mux = &alc882_capture_source,
-	},
-	[ALC882_ARIMA] = {
-		.mixers = { alc882_base_mixer, alc882_chmode_mixer },
-		.init_verbs = { alc882_init_verbs, alc882_eapd_verbs },
-		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
-		.dac_nids = alc882_dac_nids,
-		.num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
-		.channel_mode = alc882_sixstack_modes,
-		.input_mux = &alc882_capture_source,
-	},
-	[ALC882_W2JC] = {
-		.mixers = { alc882_w2jc_mixer, alc882_chmode_mixer },
-		.init_verbs = { alc882_init_verbs, alc882_eapd_verbs,
-				alc880_gpio1_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
-		.dac_nids = alc882_dac_nids,
-		.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
-		.channel_mode = alc880_threestack_modes,
-		.need_dac_fix = 1,
-		.input_mux = &alc882_capture_source,
-		.dig_out_nid = ALC882_DIGOUT_NID,
-	},
-	[ALC885_MBP3] = {
-		.mixers = { alc885_mbp3_mixer, alc882_chmode_mixer },
-		.init_verbs = { alc885_mbp3_init_verbs,
-				alc880_gpio1_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
-		.dac_nids = alc882_dac_nids,
-		.channel_mode = alc885_mbp_6ch_modes,
-		.num_channel_mode = ARRAY_SIZE(alc885_mbp_6ch_modes),
-		.input_mux = &alc882_capture_source,
-		.dig_out_nid = ALC882_DIGOUT_NID,
-		.dig_in_nid = ALC882_DIGIN_NID,
-		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc885_mbp3_init_hook,
-	},
-	[ALC885_MB5] = {
-		.mixers = { alc885_mb5_mixer, alc882_chmode_mixer },
-		.init_verbs = { alc885_mb5_init_verbs,
-				alc880_gpio1_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
-		.dac_nids = alc882_dac_nids,
-		.channel_mode = alc885_mb5_6ch_modes,
-		.num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes),
-		.input_mux = &mb5_capture_source,
-		.dig_out_nid = ALC882_DIGOUT_NID,
-		.dig_in_nid = ALC882_DIGIN_NID,
-	},
-	[ALC885_MACPRO] = {
-		.mixers = { alc882_macpro_mixer },
-		.init_verbs = { alc882_macpro_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
-		.dac_nids = alc882_dac_nids,
-		.dig_out_nid = ALC882_DIGOUT_NID,
-		.dig_in_nid = ALC882_DIGIN_NID,
-		.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
-		.channel_mode = alc882_ch_modes,
-		.input_mux = &alc882_capture_source,
-		.init_hook = alc885_macpro_init_hook,
-	},
-	[ALC885_IMAC24] = {
-		.mixers = { alc885_imac24_mixer },
-		.init_verbs = { alc885_imac24_init_verbs },
-		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
-		.dac_nids = alc882_dac_nids,
-		.dig_out_nid = ALC882_DIGOUT_NID,
-		.dig_in_nid = ALC882_DIGIN_NID,
-		.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
-		.channel_mode = alc882_ch_modes,
-		.input_mux = &alc882_capture_source,
-		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc885_imac24_init_hook,
-	},
-	[ALC882_TARGA] = {
-		.mixers = { alc882_targa_mixer, alc882_chmode_mixer },
-		.init_verbs = { alc882_init_verbs, alc882_targa_verbs},
-		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
-		.dac_nids = alc882_dac_nids,
-		.dig_out_nid = ALC882_DIGOUT_NID,
-		.num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
-		.adc_nids = alc882_adc_nids,
-		.capsrc_nids = alc882_capsrc_nids,
-		.num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
-		.channel_mode = alc882_3ST_6ch_modes,
-		.need_dac_fix = 1,
-		.input_mux = &alc882_capture_source,
-		.unsol_event = alc882_targa_unsol_event,
-		.init_hook = alc882_targa_init_hook,
-	},
-	[ALC882_ASUS_A7J] = {
-		.mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer },
-		.init_verbs = { alc882_init_verbs, alc882_asus_a7j_verbs},
-		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
-		.dac_nids = alc882_dac_nids,
-		.dig_out_nid = ALC882_DIGOUT_NID,
-		.num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
-		.adc_nids = alc882_adc_nids,
-		.capsrc_nids = alc882_capsrc_nids,
-		.num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
-		.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,
-				alc880_gpio1_init_verbs,
-				alc882_asus_a7m_verbs },
-		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
-		.dac_nids = alc882_dac_nids,
-		.dig_out_nid = ALC882_DIGOUT_NID,
-		.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
-		.channel_mode = alc880_threestack_modes,
-		.need_dac_fix = 1,
-		.input_mux = &alc882_capture_source,
-	},
-};
-
-
-/*
- * Pin config fixes
- */
-enum {
-	PINFIX_ABIT_AW9D_MAX
-};
-
-static struct alc_pincfg alc882_abit_aw9d_pinfix[] = {
-	{ 0x15, 0x01080104 }, /* side */
-	{ 0x16, 0x01011012 }, /* rear */
-	{ 0x17, 0x01016011 }, /* clfe */
-	{ }
-};
-
-static const struct alc_pincfg *alc882_pin_fixes[] = {
-	[PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix,
-};
-
-static struct snd_pci_quirk alc882_pinfix_tbl[] = {
-	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
-	{}
-};
-
-/*
- * BIOS auto configuration
- */
-static void alc882_auto_set_output_and_unmute(struct hda_codec *codec,
-					      hda_nid_t nid, int pin_type,
-					      int dac_idx)
-{
-	/* set as output */
-	struct alc_spec *spec = codec->spec;
-	int idx;
-
-	alc_set_pin_output(codec, nid, pin_type);
-	if (spec->multiout.dac_nids[dac_idx] == 0x25)
-		idx = 4;
-	else
-		idx = spec->multiout.dac_nids[dac_idx] - 2;
-	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
-
-}
-
-static void alc882_auto_init_multi_out(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	int i;
-
-	for (i = 0; i <= HDA_SIDE; i++) {
-		hda_nid_t nid = spec->autocfg.line_out_pins[i];
-		int pin_type = get_pin_type(spec->autocfg.line_out_type);
-		if (nid)
-			alc882_auto_set_output_and_unmute(codec, nid, pin_type,
-							  i);
-	}
-}
-
-static void alc882_auto_init_hp_out(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	hda_nid_t pin;
-
-	pin = spec->autocfg.hp_pins[0];
-	if (pin) /* connect to front */
-		/* use dac 0 */
-		alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
-	pin = spec->autocfg.speaker_pins[0];
-	if (pin)
-		alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
-}
-
-#define alc882_is_input_pin(nid)	alc880_is_input_pin(nid)
-#define ALC882_PIN_CD_NID		ALC880_PIN_CD_NID
-
-static void alc882_auto_init_analog_input(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	int i;
-
-	for (i = 0; i < AUTO_PIN_LAST; i++) {
-		hda_nid_t nid = spec->autocfg.input_pins[i];
-		if (!nid)
-			continue;
-		alc_set_input_pin(codec, nid, AUTO_PIN_FRONT_MIC /*i*/);
-		if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_AMP_GAIN_MUTE,
-					    AMP_OUT_MUTE);
-	}
-}
-
-static void alc882_auto_init_input_src(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	int c;
-
-	for (c = 0; c < spec->num_adc_nids; c++) {
-		hda_nid_t conn_list[HDA_MAX_NUM_INPUTS];
-		hda_nid_t nid = spec->capsrc_nids[c];
-		unsigned int mux_idx;
-		const struct hda_input_mux *imux;
-		int conns, mute, idx, item;
-
-		conns = snd_hda_get_connections(codec, nid, conn_list,
-						ARRAY_SIZE(conn_list));
-		if (conns < 0)
-			continue;
-		mux_idx = c >= spec->num_mux_defs ? 0 : c;
-		imux = &spec->input_mux[mux_idx];
-		for (idx = 0; idx < conns; idx++) {
-			/* if the current connection is the selected one,
-			 * unmute it as default - otherwise mute it
-			 */
-			mute = AMP_IN_MUTE(idx);
-			for (item = 0; item < imux->num_items; item++) {
-				if (imux->items[item].index == idx) {
-					if (spec->cur_mux[c] == item)
-						mute = AMP_IN_UNMUTE(idx);
-					break;
-				}
-			}
-			/* check if we have a selector or mixer
-			 * we could check for the widget type instead, but
-			 * just check for Amp-In presence (in case of mixer
-			 * without amp-in there is something wrong, this
-			 * function shouldn't be used or capsrc nid is wrong)
-			 */
-			if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)
-				snd_hda_codec_write(codec, nid, 0,
-						    AC_VERB_SET_AMP_GAIN_MUTE,
-						    mute);
-			else if (mute != AMP_IN_MUTE(idx))
-				snd_hda_codec_write(codec, nid, 0,
-						    AC_VERB_SET_CONNECT_SEL,
-						    idx);
-		}
-	}
-}
-
-/* add mic boosts if needed */
-static int alc_auto_add_mic_boost(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	int err;
-	hda_nid_t nid;
-
-	nid = spec->autocfg.input_pins[AUTO_PIN_MIC];
-	if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) {
-		err = add_control(spec, ALC_CTL_WIDGET_VOL,
-				  "Mic Boost",
-				  HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
-		if (err < 0)
-			return err;
-	}
-	nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC];
-	if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) {
-		err = add_control(spec, ALC_CTL_WIDGET_VOL,
-				  "Front Mic Boost",
-				  HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
-/* almost identical with ALC880 parser... */
-static int alc882_parse_auto_config(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	int err = alc880_parse_auto_config(codec);
-
-	if (err < 0)
-		return err;
-	else if (!err)
-		return 0; /* no config found */
-
-	err = alc_auto_add_mic_boost(codec);
-	if (err < 0)
-		return err;
-
-	/* hack - override the init verbs */
-	spec->init_verbs[0] = alc882_auto_init_verbs;
-
-	return 1; /* config found */
-}
-
-/* additional initialization for auto-configuration model */
-static void alc882_auto_init(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	alc882_auto_init_multi_out(codec);
-	alc882_auto_init_hp_out(codec);
-	alc882_auto_init_analog_input(codec);
-	alc882_auto_init_input_src(codec);
-	if (spec->unsol_event)
-		alc_inithook(codec);
-}
-
-static int patch_alc883(struct hda_codec *codec); /* called in patch_alc882() */
-
-static int patch_alc882(struct hda_codec *codec)
-{
-	struct alc_spec *spec;
-	int err, board_config;
-
-	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
-	if (spec == NULL)
-		return -ENOMEM;
-
-	codec->spec = spec;
-
-	board_config = snd_hda_check_board_config(codec, ALC882_MODEL_LAST,
-						  alc882_models,
-						  alc882_cfg_tbl);
-
-	if (board_config < 0 || board_config >= ALC882_MODEL_LAST) {
-		/* Pick up systems that don't supply PCI SSID */
-		switch (codec->subsystem_id) {
-		case 0x106b0c00: /* Mac Pro */
-			board_config = ALC885_MACPRO;
-			break;
-		case 0x106b1000: /* iMac 24 */
-		case 0x106b2800: /* AppleTV */
-		case 0x106b3e00: /* iMac 24 Aluminium */
-			board_config = ALC885_IMAC24;
-			break;
-		case 0x106b00a0: /* MacBookPro3,1 - Another revision */
-		case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */
-		case 0x106b00a4: /* MacbookPro4,1 */
-		case 0x106b2c00: /* Macbook Pro rev3 */
-		/* Macbook 3.1 (0x106b3600) is handled by patch_alc883() */
-		case 0x106b3800: /* MacbookPro4,1 - latter revision */
-			board_config = ALC885_MBP3;
-			break;
-		case 0x106b3f00: /* Macbook 5,1 */
-		case 0x106b4000: /* Macbook Pro 5,1 - FIXME: HP jack sense
-				  *   seems not working, so apparently
-				  *   no perfect solution yet
-				  */
-			board_config = ALC885_MB5;
-			break;
-		default:
-			/* ALC889A is handled better as ALC888-compatible */
-			if (codec->revision_id == 0x100101 ||
-			    codec->revision_id == 0x100103) {
-				alc_free(codec);
-				return patch_alc883(codec);
-			}
-			printk(KERN_INFO "hda_codec: Unknown model for %s, "
-			       "trying auto-probe from BIOS...\n",
-			       codec->chip_name);
-			board_config = ALC882_AUTO;
-		}
-	}
-
-	alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes);
-
-	if (board_config == ALC882_AUTO) {
-		/* automatic parse from the BIOS config */
-		err = alc882_parse_auto_config(codec);
-		if (err < 0) {
-			alc_free(codec);
-			return err;
-		} else if (!err) {
-			printk(KERN_INFO
-			       "hda_codec: Cannot set up configuration "
-			       "from BIOS.  Using base mode...\n");
-			board_config = ALC882_3ST_DIG;
-		}
-	}
-
-	err = snd_hda_attach_beep_device(codec, 0x1);
-	if (err < 0) {
-		alc_free(codec);
-		return err;
-	}
-
-	if (board_config != ALC882_AUTO)
-		setup_preset(spec, &alc882_presets[board_config]);
-
-	spec->stream_analog_playback = &alc882_pcm_analog_playback;
-	spec->stream_analog_capture = &alc882_pcm_analog_capture;
-	/* FIXME: setup DAC5 */
-	/*spec->stream_analog_alt_playback = &alc880_pcm_analog_alt_playback;*/
-	spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture;
-
-	spec->stream_digital_playback = &alc882_pcm_digital_playback;
-	spec->stream_digital_capture = &alc882_pcm_digital_capture;
-
-	if (!spec->adc_nids && spec->input_mux) {
-		/* check whether NID 0x07 is valid */
-		unsigned int wcap = get_wcaps(codec, 0x07);
-		/* get type */
-		wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
-		if (wcap != AC_WID_AUD_IN) {
-			spec->adc_nids = alc882_adc_nids_alt;
-			spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids_alt);
-			spec->capsrc_nids = alc882_capsrc_nids_alt;
-		} else {
-			spec->adc_nids = alc882_adc_nids;
-			spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids);
-			spec->capsrc_nids = alc882_capsrc_nids;
-		}
-	}
-	set_capture_mixer(spec);
-	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
-
-	spec->vmaster_nid = 0x0c;
-
-	codec->patch_ops = alc_patch_ops;
-	if (board_config == ALC882_AUTO)
-		spec->init_hook = alc882_auto_init;
-#ifdef CONFIG_SND_HDA_POWER_SAVE
-	if (!spec->loopback.amplist)
-		spec->loopback.amplist = alc882_loopbacks;
-#endif
-	codec->proc_widget_hook = print_realtek_coef;
-
-	return 0;
-}
-
-/*
- * ALC883 support
- *
- * ALC883 is almost identical with ALC880 but has cleaner and more flexible
- * configuration.  Each pin widget can choose any input DACs and a mixer.
- * Each ADC is connected from a mixer of all inputs.  This makes possible
- * 6-channel independent captures.
- *
- * In addition, an independent DAC for the multi-playback (not used in this
- * driver yet).
- */
-#define ALC883_DIGOUT_NID	0x06
-#define ALC883_DIGIN_NID	0x0a
-
-#define ALC1200_DIGOUT_NID	0x10
-
-static hda_nid_t alc883_dac_nids[4] = {
-	/* front, rear, clfe, rear_surr */
-	0x02, 0x03, 0x04, 0x05
-};
-
-static hda_nid_t alc883_adc_nids[2] = {
-	/* ADC1-2 */
-	0x08, 0x09,
-};
-
-static hda_nid_t alc883_adc_nids_alt[1] = {
-	/* ADC1 */
-	0x08,
-};
-
-static hda_nid_t alc883_adc_nids_rev[2] = {
-	/* ADC2-1 */
-	0x09, 0x08
-};
-
-#define alc889_adc_nids		alc880_adc_nids
-
-static hda_nid_t alc883_capsrc_nids[2] = { 0x23, 0x22 };
-
-static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 };
-
-#define alc889_capsrc_nids	alc882_capsrc_nids
-
-/* input MUX */
-/* FIXME: should be a matrix-type input source selection */
-
-static struct hda_input_mux alc883_capture_source = {
-	.num_items = 4,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Front Mic", 0x1 },
-		{ "Line", 0x2 },
-		{ "CD", 0x4 },
-	},
-};
-
-static struct hda_input_mux alc883_3stack_6ch_intel = {
-	.num_items = 4,
-	.items = {
-		{ "Mic", 0x1 },
-		{ "Front Mic", 0x0 },
-		{ "Line", 0x2 },
-		{ "CD", 0x4 },
-	},
-};
-
-static struct hda_input_mux alc883_lenovo_101e_capture_source = {
-	.num_items = 2,
-	.items = {
-		{ "Mic", 0x1 },
-		{ "Line", 0x2 },
-	},
-};
-
-static struct hda_input_mux alc883_lenovo_nb0763_capture_source = {
-	.num_items = 4,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "iMic", 0x1 },
-		{ "Line", 0x2 },
-		{ "CD", 0x4 },
-	},
-};
-
-static struct hda_input_mux alc883_fujitsu_pi2515_capture_source = {
-	.num_items = 2,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Int Mic", 0x1 },
-	},
-};
-
-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 },
-	},
-};
-
-static struct hda_input_mux alc889A_mb31_capture_source = {
-	.num_items = 2,
-	.items = {
-		{ "Mic", 0x0 },
-		/* Front Mic (0x01) unused */
-		{ "Line", 0x2 },
-		/* Line 2 (0x03) unused */
-		/* CD (0x04) unsused? */
-	},
-};
-
-/*
- * 2ch mode
- */
-static struct hda_channel_mode alc883_3ST_2ch_modes[1] = {
-	{ 2, NULL }
-};
-
-/*
- * 2ch mode
- */
-static struct hda_verb alc883_3ST_ch2_init[] = {
-	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
-	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
-	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ } /* end */
-};
-
-/*
- * 4ch mode
- */
-static struct hda_verb alc883_3ST_ch4_init[] = {
-	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
-	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
-	{ } /* end */
-};
-
-/*
- * 6ch mode
- */
-static struct hda_verb alc883_3ST_ch6_init[] = {
-	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
-	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
-	{ } /* end */
-};
-
-static struct hda_channel_mode alc883_3ST_6ch_modes[3] = {
-	{ 2, alc883_3ST_ch2_init },
-	{ 4, alc883_3ST_ch4_init },
-	{ 6, alc883_3ST_ch6_init },
-};
-
-
-/*
- * 2ch mode
- */
-static struct hda_verb alc883_4ST_ch2_init[] = {
-	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
-	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
-	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ } /* end */
-};
-
-/*
- * 4ch mode
- */
-static struct hda_verb alc883_4ST_ch4_init[] = {
-	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
-	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
-	{ } /* end */
-};
-
-/*
- * 6ch mode
- */
-static struct hda_verb alc883_4ST_ch6_init[] = {
-	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
-	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
-	{ } /* end */
-};
-
-/*
- * 8ch mode
- */
-static struct hda_verb alc883_4ST_ch8_init[] = {
-	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
-	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
-	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
-	{ } /* end */
-};
-
-static struct hda_channel_mode alc883_4ST_8ch_modes[4] = {
-	{ 2, alc883_4ST_ch2_init },
-	{ 4, alc883_4ST_ch4_init },
-	{ 6, alc883_4ST_ch6_init },
-	{ 8, alc883_4ST_ch8_init },
-};
-
-
-/*
- * 2ch mode
- */
-static struct hda_verb alc883_3ST_ch2_intel_init[] = {
-	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
-	{ 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
-	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ } /* end */
-};
-
-/*
- * 4ch mode
- */
-static struct hda_verb alc883_3ST_ch4_intel_init[] = {
-	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
-	{ 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
-	{ } /* end */
-};
-
-/*
- * 6ch mode
- */
-static struct hda_verb alc883_3ST_ch6_intel_init[] = {
-	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x19, AC_VERB_SET_CONNECT_SEL, 0x02 },
-	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
-	{ } /* end */
-};
+	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 
-static struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = {
-	{ 2, alc883_3ST_ch2_intel_init },
-	{ 4, alc883_3ST_ch4_intel_init },
-	{ 6, alc883_3ST_ch6_intel_init },
-};
+	/* Mute 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
+	 * 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)},
 
-/*
- * 6ch mode
- */
-static struct hda_verb alc883_sixstack_ch6_init[] = {
-	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00 },
-	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ } /* end */
-};
+	/*
+	 * Set up output mixers (0x0c - 0x0f)
+	 */
+	/* set vol=0 to output mixers */
+	{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},
+	{0x0f, 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)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 
-/*
- * 8ch mode
- */
-static struct hda_verb alc883_sixstack_ch8_init[] = {
-	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ } /* end */
-};
+	/* FIXME: use matrix-type input source selection */
+	/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
+	/* Input mixer2 */
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
+	/* Input mixer3 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x03 << 8))},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x02 << 8))},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x04 << 8))},
 
-static struct hda_channel_mode alc883_sixstack_modes[2] = {
-	{ 6, alc883_sixstack_ch6_init },
-	{ 8, alc883_sixstack_ch8_init },
+	{ }
 };
 
 /* 2ch mode (Speaker:front, Subwoofer:CLFE, Line:input, Headphones:front) */
@@ -7962,34 +7385,7 @@ static struct hda_verb alc883_medion_eapd_verbs[] = {
 	{ }
 };
 
-/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
- *                 Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
- */
-
-static struct snd_kcontrol_new alc883_base_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", 0x0d, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
-	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
-	HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Side Playback Switch", 0x0f, 2, HDA_INPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	{ } /* end */
-};
+#define alc883_base_mixer	alc882_base_mixer
 
 static struct snd_kcontrol_new alc883_mitac_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
@@ -8309,113 +7705,35 @@ static struct snd_kcontrol_new alc883_asus_eee1601_mixer[] = {
 	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),
-	{ } /* end */
-};
-
-static struct snd_kcontrol_new alc883_asus_eee1601_cap_mixer[] = {
-	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 = alc_mux_enum_info,
-		.get = alc_mux_enum_get,
-		.put = alc_mux_enum_put,
-	},
-	{ } /* end */
-};
-
-static struct snd_kcontrol_new alc883_chmode_mixer[] = {
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Channel Mode",
-		.info = alc_ch_mode_info,
-		.get = alc_ch_mode_get,
-		.put = alc_ch_mode_put,
-	},
-	{ } /* end */
-};
-
-static struct hda_verb alc883_init_verbs[] = {
-	/* ADC1: mute amp left and right */
-	{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)},
-	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
-	/* Front mixer: unmute input/output amp left and right (volume = 0) */
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	/* Rear mixer */
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	/* CLFE mixer */
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	/* Side mixer */
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-
-	/* mute analog input loopbacks */
-	{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)},
+	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
 
-	/* Front Pin: output 0 (0x0c) */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
-	/* Rear Pin: output 1 (0x0d) */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
-	/* CLFE Pin: output 2 (0x0e) */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
-	/* Side Pin: output 3 (0x0f) */
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
-	/* Mic (rear) pin: input vref at 80% */
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Front Mic pin: input vref at 80% */
-	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Line In pin: input */
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Line-2 In: Headphone output (output 0 - 0x0c) */
-	{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},
-	/* CD pin widget for input */
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+static struct snd_kcontrol_new alc883_asus_eee1601_cap_mixer[] = {
+	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 = alc_mux_enum_info,
+		.get = alc_mux_enum_get,
+		.put = alc_mux_enum_put,
+	},
+	{ } /* end */
+};
 
-	/* FIXME: use matrix-type input source selection */
-	/* 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_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_MUTE(1)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{ }
+static struct snd_kcontrol_new alc883_chmode_mixer[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = alc_ch_mode_info,
+		.get = alc_ch_mode_get,
+		.put = alc_ch_mode_put,
+	},
+	{ } /* end */
 };
 
 /* toggle speaker-output according to the hp-jack state */
@@ -8850,69 +8168,6 @@ static void alc883_vaiott_init_hook(struct hda_codec *codec)
 	alc_automute_amp(codec);
 }
 
-/*
- * generic initialization of ADC, input mixers and output mixers
- */
-static struct hda_verb alc883_auto_init_verbs[] = {
-	/*
-	 * Unmute ADC0-2 and set the default input to mic-in
-	 */
-	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-
-	/* Mute 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
-	 * 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 - 0x0f)
-	 */
-	/* set vol=0 to output mixers */
-	{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},
-	{0x0f, 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)},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-
-	/* FIXME: use matrix-type input source selection */
-	/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
-	/* Input mixer1 */
-	{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(3)}, */
-	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
-	/* Input mixer2 */
-	{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(3)}, */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)},
-
-	{ }
-};
-
 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)},
@@ -9023,25 +8278,44 @@ static void alc889A_mb31_unsol_event(struct hda_codec *codec, unsigned int res)
 		alc889A_mb31_automute(codec);
 }
 
+
 #ifdef CONFIG_SND_HDA_POWER_SAVE
-#define alc883_loopbacks	alc880_loopbacks
+#define alc882_loopbacks	alc880_loopbacks
 #endif
 
 /* pcm configuration: identical with ALC880 */
-#define alc883_pcm_analog_playback	alc880_pcm_analog_playback
-#define alc883_pcm_analog_capture	alc880_pcm_analog_capture
-#define alc883_pcm_analog_alt_capture	alc880_pcm_analog_alt_capture
-#define alc883_pcm_digital_playback	alc880_pcm_digital_playback
-#define alc883_pcm_digital_capture	alc880_pcm_digital_capture
+#define alc882_pcm_analog_playback	alc880_pcm_analog_playback
+#define alc882_pcm_analog_capture	alc880_pcm_analog_capture
+#define alc882_pcm_digital_playback	alc880_pcm_digital_playback
+#define alc882_pcm_digital_capture	alc880_pcm_digital_capture
+
+static hda_nid_t alc883_slave_dig_outs[] = {
+	ALC1200_DIGOUT_NID, 0,
+};
+
+static hda_nid_t alc1200_slave_dig_outs[] = {
+	ALC883_DIGOUT_NID, 0,
+};
 
 /*
  * configuration and preset
  */
-static const char *alc883_models[ALC883_MODEL_LAST] = {
-	[ALC883_3ST_2ch_DIG]	= "3stack-dig",
+static const char *alc882_models[ALC882_MODEL_LAST] = {
+	[ALC882_3ST_DIG]	= "3stack-dig",
+	[ALC882_6ST_DIG]	= "6stack-dig",
+	[ALC882_ARIMA]		= "arima",
+	[ALC882_W2JC]		= "w2jc",
+	[ALC882_TARGA]		= "targa",
+	[ALC882_ASUS_A7J]	= "asus-a7j",
+	[ALC882_ASUS_A7M]	= "asus-a7m",
+	[ALC885_MACPRO]		= "macpro",
+	[ALC885_MB5]		= "mb5",
+	[ALC885_MBP3]		= "mbp3",
+	[ALC885_IMAC24]		= "imac24",
+	[ALC883_3ST_2ch_DIG]	= "3stack-2ch-dig",
 	[ALC883_3ST_6ch_DIG]	= "3stack-6ch-dig",
 	[ALC883_3ST_6ch]	= "3stack-6ch",
-	[ALC883_6ST_DIG]	= "6stack-dig",
+	[ALC883_6ST_DIG]	= "alc883-6stack-dig",
 	[ALC883_TARGA_DIG]	= "targa-dig",
 	[ALC883_TARGA_2ch_DIG]	= "targa-2ch-dig",
 	[ALC883_TARGA_8ch_DIG]	= "targa-8ch-dig",
@@ -9068,11 +8342,12 @@ static const char *alc883_models[ALC883_MODEL_LAST] = {
 	[ALC1200_ASUS_P5Q]	= "asus-p5q",
 	[ALC889A_MB31]		= "mb31",
 	[ALC883_SONY_VAIO_TT]	= "sony-vaio-tt",
-	[ALC883_AUTO]		= "auto",
+	[ALC882_AUTO]		= "auto",
 };
 
-static struct snd_pci_quirk alc883_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_DIG),
+static struct snd_pci_quirk alc882_cfg_tbl[] = {
+	SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC882_6ST_DIG),
+
 	SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x010a, "Acer Ferrari 5000", ALC883_ACER_ASPIRE),
@@ -9087,8 +8362,8 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 		ALC888_ACER_ASPIRE_8930G),
 	SND_PCI_QUIRK(0x1025, 0x0146, "Acer Aspire 6935G",
 		ALC888_ACER_ASPIRE_8930G),
-	SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC883_AUTO),
-	SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC883_AUTO),
+	SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC882_AUTO),
+	SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC882_AUTO),
 	SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G",
 		ALC888_ACER_ASPIRE_6530G),
 	SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G",
@@ -9097,30 +8372,44 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	 *    model=auto should work fine now
 	 */
 	/* SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_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(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x103c, 0x2a72, "HP Educ.ar", ALC888_3ST_HP),
+
+	SND_PCI_QUIRK(0x1043, 0x060d, "Asus A7J", ALC882_ASUS_A7J),
+	SND_PCI_QUIRK(0x1043, 0x1243, "Asus A7J", ALC882_ASUS_A7J),
+	SND_PCI_QUIRK(0x1043, 0x13c2, "Asus A7M", ALC882_ASUS_A7M),
 	SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V),
+	SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_W2JC),
+	SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG),
+	SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG),
 	SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q),
 	SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601),
+
+	SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC883_SONY_VAIO_TT),
 	SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG),
-	SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG),
+	SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG),
 	SND_PCI_QUIRK(0x1071, 0x8227, "Mitac 82801H", ALC883_MITAC),
 	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(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG),
+
 	SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG),
 	SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG),
 	SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG),
+	SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8  */
 	SND_PCI_QUIRK(0x1462, 0x2fb3, "MSI", ALC883_TARGA_2ch_DIG),
+	SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG),
 	SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG),
 	SND_PCI_QUIRK(0x1462, 0x3783, "NEC S970", ALC883_TARGA_DIG),
 	SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG),
@@ -9142,11 +8431,13 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG),
+
 	SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720),
 	SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720),
 	SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC883_LAPTOP_EAPD),
 	SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch),
+	/* SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), */
 	SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION),
 	SND_PCI_QUIRK_MASK(0x1734, 0xfff0, 0x1100, "FSC AMILO Xi/Pi25xx",
 		      ALC883_FUJITSU_PI2515),
@@ -9161,24 +8452,175 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x17c0, 0x4085, "MEDION MD96630", ALC888_LENOVO_MS7195_DIG),
 	SND_PCI_QUIRK(0x17f2, 0x5000, "Albatron KI690-AM2", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66),
+
 	SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL),
 	SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL),
 	SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC),
 	SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC883_3ST_6ch_INTEL),
 	SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch),
-	SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC883_SONY_VAIO_TT),
-	{}
-};
 
-static hda_nid_t alc883_slave_dig_outs[] = {
-	ALC1200_DIGOUT_NID, 0,
+	{}
 };
 
-static hda_nid_t alc1200_slave_dig_outs[] = {
-	ALC883_DIGOUT_NID, 0,
+/* codec SSID table for Intel Mac */
+static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
+	SND_PCI_QUIRK(0x106b, 0x00a0, "MacBookPro 3,1", ALC885_MBP3),
+	SND_PCI_QUIRK(0x106b, 0x00a1, "Macbook", ALC885_MBP3),
+	SND_PCI_QUIRK(0x106b, 0x00a4, "MacbookPro 4,1", ALC885_MBP3),
+	SND_PCI_QUIRK(0x106b, 0x0c00, "Mac Pro", ALC885_MACPRO),
+	SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_IMAC24),
+	SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_IMAC24),
+	SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3),
+	SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31),
+	SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3),
+	SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24),
+	SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5),
+	/* FIXME: HP jack sense seems not working for MBP 5,1, so apparently
+	 * no perfect solution yet
+	 */
+	SND_PCI_QUIRK(0x106b, 0x4000, "MacbookPro 5,1", ALC885_MB5),
+	{} /* terminator */
 };
 
-static struct alc_config_preset alc883_presets[] = {
+static struct alc_config_preset alc882_presets[] = {
+	[ALC882_3ST_DIG] = {
+		.mixers = { alc882_base_mixer },
+		.init_verbs = { alc882_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.dig_in_nid = ALC882_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
+		.channel_mode = alc882_ch_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc882_capture_source,
+	},
+	[ALC882_6ST_DIG] = {
+		.mixers = { alc882_base_mixer, alc882_chmode_mixer },
+		.init_verbs = { alc882_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.dig_in_nid = ALC882_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
+		.channel_mode = alc882_sixstack_modes,
+		.input_mux = &alc882_capture_source,
+	},
+	[ALC882_ARIMA] = {
+		.mixers = { alc882_base_mixer, alc882_chmode_mixer },
+		.init_verbs = { alc882_init_verbs, alc882_eapd_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
+		.channel_mode = alc882_sixstack_modes,
+		.input_mux = &alc882_capture_source,
+	},
+	[ALC882_W2JC] = {
+		.mixers = { alc882_w2jc_mixer, alc882_chmode_mixer },
+		.init_verbs = { alc882_init_verbs, alc882_eapd_verbs,
+				alc880_gpio1_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+		.channel_mode = alc880_threestack_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc882_capture_source,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+	},
+	[ALC885_MBP3] = {
+		.mixers = { alc885_mbp3_mixer, alc882_chmode_mixer },
+		.init_verbs = { alc885_mbp3_init_verbs,
+				alc880_gpio1_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.channel_mode = alc885_mbp_6ch_modes,
+		.num_channel_mode = ARRAY_SIZE(alc885_mbp_6ch_modes),
+		.input_mux = &alc882_capture_source,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.dig_in_nid = ALC882_DIGIN_NID,
+		.unsol_event = alc_automute_amp_unsol_event,
+		.init_hook = alc885_mbp3_init_hook,
+	},
+	[ALC885_MB5] = {
+		.mixers = { alc885_mb5_mixer, alc882_chmode_mixer },
+		.init_verbs = { alc885_mb5_init_verbs,
+				alc880_gpio1_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.channel_mode = alc885_mb5_6ch_modes,
+		.num_channel_mode = ARRAY_SIZE(alc885_mb5_6ch_modes),
+		.input_mux = &mb5_capture_source,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.dig_in_nid = ALC882_DIGIN_NID,
+	},
+	[ALC885_MACPRO] = {
+		.mixers = { alc882_macpro_mixer },
+		.init_verbs = { alc882_macpro_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.dig_in_nid = ALC882_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
+		.channel_mode = alc882_ch_modes,
+		.input_mux = &alc882_capture_source,
+		.init_hook = alc885_macpro_init_hook,
+	},
+	[ALC885_IMAC24] = {
+		.mixers = { alc885_imac24_mixer },
+		.init_verbs = { alc885_imac24_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.dig_in_nid = ALC882_DIGIN_NID,
+		.num_channel_mode = ARRAY_SIZE(alc882_ch_modes),
+		.channel_mode = alc882_ch_modes,
+		.input_mux = &alc882_capture_source,
+		.unsol_event = alc_automute_amp_unsol_event,
+		.init_hook = alc885_imac24_init_hook,
+	},
+	[ALC882_TARGA] = {
+		.mixers = { alc882_targa_mixer, alc882_chmode_mixer },
+		.init_verbs = { alc882_init_verbs, alc882_targa_verbs},
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
+		.adc_nids = alc882_adc_nids,
+		.capsrc_nids = alc882_capsrc_nids,
+		.num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
+		.channel_mode = alc882_3ST_6ch_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc882_capture_source,
+		.unsol_event = alc882_targa_unsol_event,
+		.init_hook = alc882_targa_init_hook,
+	},
+	[ALC882_ASUS_A7J] = {
+		.mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer },
+		.init_verbs = { alc882_init_verbs, alc882_asus_a7j_verbs},
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.num_adc_nids = ARRAY_SIZE(alc882_adc_nids),
+		.adc_nids = alc882_adc_nids,
+		.capsrc_nids = alc882_capsrc_nids,
+		.num_channel_mode = ARRAY_SIZE(alc882_3ST_6ch_modes),
+		.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,
+				alc880_gpio1_init_verbs,
+				alc882_asus_a7m_verbs },
+		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
+		.dac_nids = alc882_dac_nids,
+		.dig_out_nid = ALC882_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
+		.channel_mode = alc880_threestack_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc882_capture_source,
+	},
 	[ALC883_3ST_2ch_DIG] = {
 		.mixers = { alc883_3ST_2ch_mixer },
 		.init_verbs = { alc883_init_verbs },
@@ -9612,10 +9054,33 @@ static struct alc_config_preset alc883_presets[] = {
 };
 
 
+/*
+ * Pin config fixes
+ */
+enum {
+	PINFIX_ABIT_AW9D_MAX
+};
+
+static struct alc_pincfg alc882_abit_aw9d_pinfix[] = {
+	{ 0x15, 0x01080104 }, /* side */
+	{ 0x16, 0x01011012 }, /* rear */
+	{ 0x17, 0x01016011 }, /* clfe */
+	{ }
+};
+
+static const struct alc_pincfg *alc882_pin_fixes[] = {
+	[PINFIX_ABIT_AW9D_MAX] = alc882_abit_aw9d_pinfix,
+};
+
+static struct snd_pci_quirk alc882_pinfix_tbl[] = {
+	SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX),
+	{}
+};
+
 /*
  * BIOS auto configuration
  */
-static void alc883_auto_set_output_and_unmute(struct hda_codec *codec,
+static void alc882_auto_set_output_and_unmute(struct hda_codec *codec,
 					      hda_nid_t nid, int pin_type,
 					      int dac_idx)
 {
@@ -9632,7 +9097,7 @@ static void alc883_auto_set_output_and_unmute(struct hda_codec *codec,
 
 }
 
-static void alc883_auto_init_multi_out(struct hda_codec *codec)
+static void alc882_auto_init_multi_out(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	int i;
@@ -9641,12 +9106,12 @@ static void alc883_auto_init_multi_out(struct hda_codec *codec)
 		hda_nid_t nid = spec->autocfg.line_out_pins[i];
 		int pin_type = get_pin_type(spec->autocfg.line_out_type);
 		if (nid)
-			alc883_auto_set_output_and_unmute(codec, nid, pin_type,
+			alc882_auto_set_output_and_unmute(codec, nid, pin_type,
 							  i);
 	}
 }
 
-static void alc883_auto_init_hp_out(struct hda_codec *codec)
+static void alc882_auto_init_hp_out(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	hda_nid_t pin;
@@ -9654,42 +9119,114 @@ static void alc883_auto_init_hp_out(struct hda_codec *codec)
 	pin = spec->autocfg.hp_pins[0];
 	if (pin) /* connect to front */
 		/* use dac 0 */
-		alc883_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
+		alc882_auto_set_output_and_unmute(codec, pin, PIN_HP, 0);
 	pin = spec->autocfg.speaker_pins[0];
 	if (pin)
-		alc883_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
+		alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
 }
 
-#define alc883_is_input_pin(nid)	alc880_is_input_pin(nid)
-#define ALC883_PIN_CD_NID		ALC880_PIN_CD_NID
+#define alc882_is_input_pin(nid)	alc880_is_input_pin(nid)
+#define ALC882_PIN_CD_NID		ALC880_PIN_CD_NID
 
-static void alc883_auto_init_analog_input(struct hda_codec *codec)
+static void alc882_auto_init_analog_input(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	int i;
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
-		if (alc883_is_input_pin(nid)) {
-			alc_set_input_pin(codec, nid, i);
-			if (nid != ALC883_PIN_CD_NID &&
-			    (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
+		if (!nid)
+			continue;
+		alc_set_input_pin(codec, nid, AUTO_PIN_FRONT_MIC /*i*/);
+		if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
+			snd_hda_codec_write(codec, nid, 0,
+					    AC_VERB_SET_AMP_GAIN_MUTE,
+					    AMP_OUT_MUTE);
+	}
+}
+
+static void alc882_auto_init_input_src(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	int c;
+
+	for (c = 0; c < spec->num_adc_nids; c++) {
+		hda_nid_t conn_list[HDA_MAX_NUM_INPUTS];
+		hda_nid_t nid = spec->capsrc_nids[c];
+		unsigned int mux_idx;
+		const struct hda_input_mux *imux;
+		int conns, mute, idx, item;
+
+		conns = snd_hda_get_connections(codec, nid, conn_list,
+						ARRAY_SIZE(conn_list));
+		if (conns < 0)
+			continue;
+		mux_idx = c >= spec->num_mux_defs ? 0 : c;
+		imux = &spec->input_mux[mux_idx];
+		for (idx = 0; idx < conns; idx++) {
+			/* if the current connection is the selected one,
+			 * unmute it as default - otherwise mute it
+			 */
+			mute = AMP_IN_MUTE(idx);
+			for (item = 0; item < imux->num_items; item++) {
+				if (imux->items[item].index == idx) {
+					if (spec->cur_mux[c] == item)
+						mute = AMP_IN_UNMUTE(idx);
+					break;
+				}
+			}
+			/* check if we have a selector or mixer
+			 * we could check for the widget type instead, but
+			 * just check for Amp-In presence (in case of mixer
+			 * without amp-in there is something wrong, this
+			 * function shouldn't be used or capsrc nid is wrong)
+			 */
+			if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)
 				snd_hda_codec_write(codec, nid, 0,
 						    AC_VERB_SET_AMP_GAIN_MUTE,
-						    AMP_OUT_MUTE);
+						    mute);
+			else if (mute != AMP_IN_MUTE(idx))
+				snd_hda_codec_write(codec, nid, 0,
+						    AC_VERB_SET_CONNECT_SEL,
+						    idx);
 		}
 	}
 }
 
-#define alc883_auto_init_input_src	alc882_auto_init_input_src
+/* add mic boosts if needed */
+static int alc_auto_add_mic_boost(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	int err;
+	hda_nid_t nid;
+
+	nid = spec->autocfg.input_pins[AUTO_PIN_MIC];
+	if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) {
+		err = add_control(spec, ALC_CTL_WIDGET_VOL,
+				  "Mic Boost",
+				  HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
+		if (err < 0)
+			return err;
+	}
+	nid = spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC];
+	if (nid && (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)) {
+		err = add_control(spec, ALC_CTL_WIDGET_VOL,
+				  "Front Mic Boost",
+				  HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
+		if (err < 0)
+			return err;
+	}
+	return 0;
+}
 
 /* almost identical with ALC880 parser... */
-static int alc883_parse_auto_config(struct hda_codec *codec)
+static int alc882_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	int err = alc880_parse_auto_config(codec);
-	struct auto_pin_cfg *cfg = &spec->autocfg;
+	struct auto_pin_cfg *autocfg = &spec->autocfg;
+	unsigned int wcap;
 	int i;
+	int err = alc880_parse_auto_config(codec);
 
 	if (err < 0)
 		return err;
@@ -9702,43 +9239,45 @@ static int alc883_parse_auto_config(struct hda_codec *codec)
 
 	/* hack - override the init verbs */
 	spec->init_verbs[0] = alc883_auto_init_verbs;
+	/* if ADC 0x07 is available, initialize it, too */
+	wcap = get_wcaps(codec, 0x07);
+	wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+	if (wcap == AC_WID_AUD_IN)
+		add_verb(spec, alc882_adc1_init_verbs);
 
-	/* setup input_mux for ALC889 */
-	if (codec->vendor_id == 0x10ec0889) {
-		/* digital-mic input pin is excluded in alc880_auto_create..()
-		 * because it's under 0x18
-		 */
-		if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 ||
-		    cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) {
-			struct hda_input_mux *imux = &spec->private_imux[0];
-			for (i = 1; i < 3; i++)
-				memcpy(&spec->private_imux[i],
-				       &spec->private_imux[0],
-				       sizeof(spec->private_imux[0]));
-			imux->items[imux->num_items].label = "Int DMic";
-			imux->items[imux->num_items].index = 0x0b;
-			imux->num_items++;
-			spec->num_mux_defs = 3;
-			spec->input_mux = spec->private_imux;
-		}
+	/* digital-mic input pin is excluded in alc880_auto_create..()
+	 * because it's under 0x18
+	 */
+	if (autocfg->input_pins[AUTO_PIN_MIC] == 0x12 ||
+	    autocfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) {
+		struct hda_input_mux *imux = &spec->private_imux[0];
+		for (i = 1; i < 3; i++)
+			memcpy(&spec->private_imux[i],
+			       &spec->private_imux[0],
+			       sizeof(spec->private_imux[0]));
+		imux->items[imux->num_items].label = "Int DMic";
+		imux->items[imux->num_items].index = 0x0b;
+		imux->num_items++;
+		spec->num_mux_defs = 3;
+		spec->input_mux = spec->private_imux;
 	}
 
 	return 1; /* config found */
 }
 
 /* additional initialization for auto-configuration model */
-static void alc883_auto_init(struct hda_codec *codec)
+static void alc882_auto_init(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	alc883_auto_init_multi_out(codec);
-	alc883_auto_init_hp_out(codec);
-	alc883_auto_init_analog_input(codec);
-	alc883_auto_init_input_src(codec);
+	alc882_auto_init_multi_out(codec);
+	alc882_auto_init_hp_out(codec);
+	alc882_auto_init_analog_input(codec);
+	alc882_auto_init_input_src(codec);
 	if (spec->unsol_event)
 		alc_inithook(codec);
 }
 
-static int patch_alc883(struct hda_codec *codec)
+static int patch_alc882(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
 	int err, board_config;
@@ -9749,28 +9288,36 @@ static int patch_alc883(struct hda_codec *codec)
 
 	codec->spec = spec;
 
-	alc_fix_pll_init(codec, 0x20, 0x0a, 10);
+	switch (codec->vendor_id) {
+	case 0x10ec0882:
+	case 0x10ec0885:
+		break;
+	default:
+		/* ALC883 and variants */
+		alc_fix_pll_init(codec, 0x20, 0x0a, 10);
+		break;
+	}
 
-	board_config = snd_hda_check_board_config(codec, ALC883_MODEL_LAST,
-						  alc883_models,
-						  alc883_cfg_tbl);
-	if (board_config < 0 || board_config >= ALC883_MODEL_LAST) {
-		/* Pick up systems that don't supply PCI SSID */
-		switch (codec->subsystem_id) {
-		case 0x106b3600: /* Macbook 3.1 */
-			board_config = ALC889A_MB31;
-			break;
-		default:
-			printk(KERN_INFO
-				"hda_codec: Unknown model for %s, trying "
-				"auto-probe from BIOS...\n", codec->chip_name);
-			board_config = ALC883_AUTO;
-		}
+	board_config = snd_hda_check_board_config(codec, ALC882_MODEL_LAST,
+						  alc882_models,
+						  alc882_cfg_tbl);
+
+	if (board_config < 0 || board_config >= ALC882_MODEL_LAST)
+		board_config = snd_hda_check_board_codec_sid_config(codec,
+			ALC882_MODEL_LAST, alc882_models, alc882_ssid_cfg_tbl);
+
+	if (board_config < 0 || board_config >= ALC882_MODEL_LAST) {
+		printk(KERN_INFO "hda_codec: Unknown model for %s, "
+		       "trying auto-probe from BIOS...\n",
+		       codec->chip_name);
+		board_config = ALC882_AUTO;
 	}
 
-	if (board_config == ALC883_AUTO) {
+	alc_fix_pincfg(codec, alc882_pinfix_tbl, alc882_pin_fixes);
+
+	if (board_config == ALC882_AUTO) {
 		/* automatic parse from the BIOS config */
-		err = alc883_parse_auto_config(codec);
+		err = alc882_parse_auto_config(codec);
 		if (err < 0) {
 			alc_free(codec);
 			return err;
@@ -9778,7 +9325,7 @@ static int patch_alc883(struct hda_codec *codec)
 			printk(KERN_INFO
 			       "hda_codec: Cannot set up configuration "
 			       "from BIOS.  Using base mode...\n");
-			board_config = ALC883_3ST_2ch_DIG;
+			board_config = ALC882_3ST_DIG;
 		}
 	}
 
@@ -9788,63 +9335,61 @@ static int patch_alc883(struct hda_codec *codec)
 		return err;
 	}
 
-	if (board_config != ALC883_AUTO)
-		setup_preset(spec, &alc883_presets[board_config]);
+	if (board_config != ALC882_AUTO)
+		setup_preset(spec, &alc882_presets[board_config]);
 
-	switch (codec->vendor_id) {
-	case 0x10ec0888:
-		if (!spec->num_adc_nids) {
-			spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
-			spec->adc_nids = alc883_adc_nids;
-		}
-		if (!spec->capsrc_nids)
-			spec->capsrc_nids = alc883_capsrc_nids;
+	spec->stream_analog_playback = &alc882_pcm_analog_playback;
+	spec->stream_analog_capture = &alc882_pcm_analog_capture;
+	/* FIXME: setup DAC5 */
+	/*spec->stream_analog_alt_playback = &alc880_pcm_analog_alt_playback;*/
+	spec->stream_analog_alt_capture = &alc880_pcm_analog_alt_capture;
+
+	spec->stream_digital_playback = &alc882_pcm_digital_playback;
+	spec->stream_digital_capture = &alc882_pcm_digital_capture;
+
+	if (codec->vendor_id == 0x10ec0888)
 		spec->init_amp = ALC_INIT_DEFAULT; /* always initialize */
-		break;
-	case 0x10ec0889:
-		if (!spec->num_adc_nids) {
-			spec->num_adc_nids = ARRAY_SIZE(alc889_adc_nids);
-			spec->adc_nids = alc889_adc_nids;
-		}
-		if (!spec->capsrc_nids)
-			spec->capsrc_nids = alc889_capsrc_nids;
-		break;
-	default:
-		if (!spec->num_adc_nids) {
-			spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
-			spec->adc_nids = alc883_adc_nids;
+
+	if (!spec->adc_nids && spec->input_mux) {
+		int i;
+		spec->num_adc_nids = 0;
+		for (i = 0; i < ARRAY_SIZE(alc882_adc_nids); i++) {
+			hda_nid_t cap;
+			hda_nid_t nid = alc882_adc_nids[i];
+			unsigned int wcap = get_wcaps(codec, nid);
+			/* get type */
+			wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+			if (wcap != AC_WID_AUD_IN)
+				continue;
+			spec->private_adc_nids[spec->num_adc_nids] = nid;
+			err = snd_hda_get_connections(codec, nid, &cap, 1);
+			if (err < 0)
+				continue;
+			spec->private_capsrc_nids[spec->num_adc_nids] = cap;
+			spec->num_adc_nids++;
 		}
-		if (!spec->capsrc_nids)
-			spec->capsrc_nids = alc883_capsrc_nids;
-		break;
+		spec->adc_nids = spec->private_adc_nids;
+		spec->capsrc_nids = spec->private_capsrc_nids;
 	}
 
-	spec->stream_analog_playback = &alc883_pcm_analog_playback;
-	spec->stream_analog_capture = &alc883_pcm_analog_capture;
-	spec->stream_analog_alt_capture = &alc883_pcm_analog_alt_capture;
-
-	spec->stream_digital_playback = &alc883_pcm_digital_playback;
-	spec->stream_digital_capture = &alc883_pcm_digital_capture;
-
-	if (!spec->cap_mixer)
-		set_capture_mixer(spec);
+	set_capture_mixer(spec);
 	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
 	spec->vmaster_nid = 0x0c;
 
 	codec->patch_ops = alc_patch_ops;
-	if (board_config == ALC883_AUTO)
-		spec->init_hook = alc883_auto_init;
-
+	if (board_config == ALC882_AUTO)
+		spec->init_hook = alc882_auto_init;
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	if (!spec->loopback.amplist)
-		spec->loopback.amplist = alc883_loopbacks;
+		spec->loopback.amplist = alc882_loopbacks;
 #endif
 	codec->proc_widget_hook = print_realtek_coef;
 
 	return 0;
 }
 
+
 /*
  * ALC262 support
  */
@@ -17546,23 +17091,23 @@ static struct hda_codec_preset snd_hda_preset_realtek[] = {
 	{ .id = 0x10ec0861, .name = "ALC861", .patch = patch_alc861 },
 	{ .id = 0x10ec0862, .name = "ALC861-VD", .patch = patch_alc861vd },
 	{ .id = 0x10ec0662, .rev = 0x100002, .name = "ALC662 rev2",
-	  .patch = patch_alc883 },
+	  .patch = patch_alc882 },
 	{ .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1",
 	  .patch = patch_alc662 },
 	{ .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 },
 	{ .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
 	{ .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
-	{ .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc883 },
+	{ .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 },
 	{ .id = 0x10ec0885, .rev = 0x100101, .name = "ALC889A",
-	  .patch = patch_alc882 }, /* should be patch_alc883() in future */
+	  .patch = patch_alc882 },
 	{ .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A",
-	  .patch = patch_alc882 }, /* should be patch_alc883() in future */
+	  .patch = patch_alc882 },
 	{ .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
-	{ .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc883 },
+	{ .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 },
 	{ .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200",
-	  .patch = patch_alc883 },
-	{ .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc883 },
-	{ .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc883 },
+	  .patch = patch_alc882 },
+	{ .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc882 },
+	{ .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 },
 	{} /* terminator */
 };
 
-- 
cgit v1.2.3


From 0d971c9fcf06d22663040570c3cfe08b137c4b2f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 30 Jun 2009 16:11:11 +0200
Subject: ALSA: hda - Fix input pinctl for ALC882 auto mode

alc882_auto_init_analog_input() sets the input pins to VREF-80 regardless
of the input pin types although it shouldn't be for line-in pins.

This patch fixes the behavior to follow other codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 5 +----
 1 file changed, 1 insertion(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6a899e8fdd0..0f6b6a6f72e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -9125,9 +9125,6 @@ static void alc882_auto_init_hp_out(struct hda_codec *codec)
 		alc882_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
 }
 
-#define alc882_is_input_pin(nid)	alc880_is_input_pin(nid)
-#define ALC882_PIN_CD_NID		ALC880_PIN_CD_NID
-
 static void alc882_auto_init_analog_input(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -9137,7 +9134,7 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec)
 		hda_nid_t nid = spec->autocfg.input_pins[i];
 		if (!nid)
 			continue;
-		alc_set_input_pin(codec, nid, AUTO_PIN_FRONT_MIC /*i*/);
+		alc_set_input_pin(codec, nid, i);
 		if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
 			snd_hda_codec_write(codec, nid, 0,
 					    AC_VERB_SET_AMP_GAIN_MUTE,
-- 
cgit v1.2.3


From b5025c50b5e817b3e509ad7e569f131b80d7c223 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 1 Jul 2009 18:05:27 +0200
Subject: ALSA: hda - Allow FLOAT PCM format

So far, the FLOAT PCM format is used only exclusivley set.  But
this can be a combination with other formats.

This patch changes the parser to allow the FLOAT format in addition
to other PCM formats.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 506f46ef030..263d124de61 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2690,11 +2690,11 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
 					bps = 20;
 			}
 		}
-		else if (streams == AC_SUPFMT_FLOAT32) {
-			/* should be exclusive */
+		if (streams & AC_SUPFMT_FLOAT32) {
 			formats |= SNDRV_PCM_FMTBIT_FLOAT_LE;
 			bps = 32;
-		} else if (streams == AC_SUPFMT_AC3) {
+		}
+		if (streams == AC_SUPFMT_AC3) {
 			/* should be exclusive */
 			/* temporary hack: we have still no proper support
 			 * for the direct AC3 stream...
-- 
cgit v1.2.3


From e5f1424807f4fa7aeddc376575e3b413c71c6fe1 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 1 Jul 2009 18:11:44 +0200
Subject: ALSA: hda - Add Cirrus Logic CS420x support

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/Kconfig        |  13 +
 sound/pci/hda/Makefile       |   4 +
 sound/pci/hda/hda_codec.c    |   1 +
 sound/pci/hda/patch_cirrus.c | 937 +++++++++++++++++++++++++++++++++++++++++++
 4 files changed, 955 insertions(+)
 create mode 100644 sound/pci/hda/patch_cirrus.c

(limited to 'sound')

diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index b8a77f9b082..55545e0818b 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -148,6 +148,19 @@ config SND_HDA_ELD
 	def_bool y
 	depends on SND_HDA_CODEC_INTELHDMI
 
+config SND_HDA_CODEC_CIRRUS
+	bool "Build Cirrus Logic codec support"
+	depends on SND_HDA_INTEL
+	default y
+	help
+	  Say Y here to include Cirrus Logic codec support in
+	  snd-hda-intel driver, such as CS4206.
+
+	  When the HD-audio driver is built as a module, the codec
+	  support code is also built as another module,
+	  snd-hda-codec-cirrus.
+	  This module is automatically loaded at probing.
+
 config SND_HDA_CODEC_CONEXANT
 	bool "Build Conexant HD-audio codec support"
 	default y
diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
index e3081d4586c..315a1c4f899 100644
--- a/sound/pci/hda/Makefile
+++ b/sound/pci/hda/Makefile
@@ -13,6 +13,7 @@ snd-hda-codec-analog-objs :=	patch_analog.o
 snd-hda-codec-idt-objs :=	patch_sigmatel.o
 snd-hda-codec-si3054-objs :=	patch_si3054.o
 snd-hda-codec-atihdmi-objs :=	patch_atihdmi.o
+snd-hda-codec-cirrus-objs :=	patch_cirrus.o
 snd-hda-codec-ca0110-objs :=	patch_ca0110.o
 snd-hda-codec-conexant-objs :=	patch_conexant.o
 snd-hda-codec-via-objs :=	patch_via.o
@@ -41,6 +42,9 @@ endif
 ifdef CONFIG_SND_HDA_CODEC_ATIHDMI
 obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-atihdmi.o
 endif
+ifdef CONFIG_SND_HDA_CODEC_CIRRUS
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-cirrus.o
+endif
 ifdef CONFIG_SND_HDA_CODEC_CA0110
 obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-codec-ca0110.o
 endif
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 263d124de61..eea91c3bd42 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -44,6 +44,7 @@ struct hda_vendor_id {
 /* codec vendor labels */
 static struct hda_vendor_id hda_vendor_ids[] = {
 	{ 0x1002, "ATI" },
+	{ 0x1013, "Cirrus Logic" },
 	{ 0x1057, "Motorola" },
 	{ 0x1095, "Silicon Image" },
 	{ 0x10de, "Nvidia" },
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
new file mode 100644
index 00000000000..c4dc12e8573
--- /dev/null
+++ b/sound/pci/hda/patch_cirrus.c
@@ -0,0 +1,937 @@
+/*
+ * HD audio interface patch for Cirrus Logic CS420x chip
+ *
+ * Copyright (c) 2009 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
+ *  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 <linux/pci.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+
+/*
+ */
+
+struct cs_spec {
+	struct auto_pin_cfg autocfg;
+	struct hda_multi_out multiout;
+	struct snd_kcontrol *vmaster_sw;
+	struct snd_kcontrol *vmaster_vol;
+
+	hda_nid_t dac_nid[AUTO_CFG_MAX_OUTS];
+	hda_nid_t slave_dig_outs[2];
+
+	unsigned int input_idx[AUTO_PIN_LAST];
+	unsigned int capsrc_idx[AUTO_PIN_LAST];
+	hda_nid_t adc_nid[AUTO_PIN_LAST];
+	unsigned int adc_idx[AUTO_PIN_LAST];
+	unsigned int num_inputs;
+	unsigned int cur_input;
+	unsigned int automic_idx;
+	hda_nid_t cur_adc;
+	unsigned int cur_adc_stream_tag;
+	unsigned int cur_adc_format;
+	hda_nid_t dig_in;
+
+	struct hda_bind_ctls *capture_bind[2];
+
+	struct hda_pcm pcm_rec[2];	/* PCM information */
+
+	unsigned int hp_detect:1;
+	unsigned int mic_detect:1;
+	unsigned int built_up:1;
+};
+
+#define HP_EVENT	1
+#define MIC_EVENT	2
+
+/*
+ * PCM callbacks
+ */
+static int cs_playback_pcm_open(struct hda_pcm_stream *hinfo,
+				struct hda_codec *codec,
+				struct snd_pcm_substream *substream)
+{
+	struct cs_spec *spec = codec->spec;
+	return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
+					     hinfo);
+}
+
+static int cs_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 cs_spec *spec = codec->spec;
+	return snd_hda_multi_out_analog_prepare(codec, &spec->multiout,
+						stream_tag, format, substream);
+}
+
+static int cs_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+				   struct hda_codec *codec,
+				   struct snd_pcm_substream *substream)
+{
+	struct cs_spec *spec = codec->spec;
+	return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
+}
+
+/*
+ * Digital out
+ */
+static int cs_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+				    struct hda_codec *codec,
+				    struct snd_pcm_substream *substream)
+{
+	struct cs_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int cs_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+				     struct hda_codec *codec,
+				     struct snd_pcm_substream *substream)
+{
+	struct cs_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+static int cs_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 cs_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
+					     format, substream);
+}
+
+static int cs_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+				       struct hda_codec *codec,
+				       struct snd_pcm_substream *substream)
+{
+	struct cs_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
+}
+
+/*
+ * Analog capture
+ */
+static int cs_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+				  struct hda_codec *codec,
+				  unsigned int stream_tag,
+				  unsigned int format,
+				  struct snd_pcm_substream *substream)
+{
+	struct cs_spec *spec = codec->spec;
+	spec->cur_adc = spec->adc_nid[spec->cur_input];
+	spec->cur_adc_stream_tag = stream_tag;
+	spec->cur_adc_format = format;
+	snd_hda_codec_setup_stream(codec, spec->cur_adc, stream_tag, 0, format);
+	return 0;
+}
+
+static int cs_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+				  struct hda_codec *codec,
+				  struct snd_pcm_substream *substream)
+{
+	struct cs_spec *spec = codec->spec;
+	snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
+	spec->cur_adc = 0;
+	return 0;
+}
+
+/*
+ */
+static struct hda_pcm_stream cs_pcm_analog_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.ops = {
+		.open = cs_playback_pcm_open,
+		.prepare = cs_playback_pcm_prepare,
+		.cleanup = cs_playback_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream cs_pcm_analog_capture = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.ops = {
+		.prepare = cs_capture_pcm_prepare,
+		.cleanup = cs_capture_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream cs_pcm_digital_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.ops = {
+		.open = cs_dig_playback_pcm_open,
+		.close = cs_dig_playback_pcm_close,
+		.prepare = cs_dig_playback_pcm_prepare,
+		.cleanup = cs_dig_playback_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream cs_pcm_digital_capture = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+};
+
+static int cs_build_pcms(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	struct hda_pcm *info = spec->pcm_rec;
+
+	codec->pcm_info = info;
+	codec->num_pcms = 0;
+
+	info->name = "Cirrus Analog";
+	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = cs_pcm_analog_playback;
+	info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->dac_nid[0];
+	info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
+		spec->multiout.max_channels;
+	info->stream[SNDRV_PCM_STREAM_CAPTURE] = cs_pcm_analog_capture;
+	info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_inputs;
+	info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
+		spec->adc_nid[spec->cur_input];
+	codec->num_pcms++;
+
+	if (!spec->multiout.dig_out_nid && !spec->dig_in)
+		return 0;
+
+	info++;
+	info->name = "Cirrus Digital";
+	info->pcm_type = spec->autocfg.dig_out_type[0];
+	if (!info->pcm_type)
+		info->pcm_type = HDA_PCM_TYPE_SPDIF;
+	if (spec->multiout.dig_out_nid) {
+		info->stream[SNDRV_PCM_STREAM_PLAYBACK] =
+			cs_pcm_digital_playback;
+		info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid =
+			spec->multiout.dig_out_nid;
+	}
+	if (spec->dig_in) {
+		info->stream[SNDRV_PCM_STREAM_CAPTURE] =
+			cs_pcm_digital_capture;
+		info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in;
+	}
+	codec->num_pcms++;
+
+	return 0;
+}
+
+static hda_nid_t get_dac(struct hda_codec *codec, hda_nid_t pin)
+{
+	hda_nid_t dac;
+	if (!pin)
+		return 0;
+	if (snd_hda_get_connections(codec, pin, &dac, 1) != 1)
+		return 0;
+	return dac;
+}
+
+static const char *dir_sfx[2] = { "Playback", "Capture" };
+
+static int add_mute(struct hda_codec *codec, const char *name, int index,
+		    unsigned int pval, int dir, struct snd_kcontrol **kctlp)
+{
+	char tmp[32];
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_MUTE_IDX(tmp, index, 0, 0, HDA_OUTPUT);
+	knew.private_value = pval;
+	snprintf(tmp, sizeof(tmp), "%s %s Switch", name, dir_sfx[dir]);
+	*kctlp = snd_ctl_new1(&knew, codec);
+	return snd_hda_ctl_add(codec, *kctlp);
+}
+
+static int add_volume(struct hda_codec *codec, const char *name,
+		      int index, unsigned int pval, int dir,
+		      struct snd_kcontrol **kctlp)
+{
+	char tmp[32];
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_VOLUME_IDX(tmp, index, 0, 0, HDA_OUTPUT);
+	knew.private_value = pval;
+	snprintf(tmp, sizeof(tmp), "%s %s Volume", name, dir_sfx[dir]);
+	*kctlp = snd_ctl_new1(&knew, codec);
+	return snd_hda_ctl_add(codec, *kctlp);
+}
+
+static void fix_volume_caps(struct hda_codec *codec, hda_nid_t dac)
+{
+	unsigned int caps;
+
+	/* set the upper-limit for mixer amp to 0dB */
+	caps = query_amp_caps(codec, dac, HDA_OUTPUT);
+	caps &= ~(0x7f << AC_AMPCAP_NUM_STEPS_SHIFT);
+	caps |= ((caps >> AC_AMPCAP_OFFSET_SHIFT) & 0x7f)
+		<< AC_AMPCAP_NUM_STEPS_SHIFT;
+	snd_hda_override_amp_caps(codec, dac, HDA_OUTPUT, caps);
+}
+
+static int add_vmaster(struct hda_codec *codec, hda_nid_t dac)
+{
+	struct cs_spec *spec = codec->spec;
+	unsigned int tlv[4];
+	int err;
+
+	spec->vmaster_sw =
+		snd_ctl_make_virtual_master("Master Playback Switch", NULL);
+	err = snd_hda_ctl_add(codec, spec->vmaster_sw);
+	if (err < 0)
+		return err;
+
+	snd_hda_set_vmaster_tlv(codec, dac, HDA_OUTPUT, tlv);
+	spec->vmaster_vol =
+		snd_ctl_make_virtual_master("Master Playback Volume", tlv);
+	err = snd_hda_ctl_add(codec, spec->vmaster_vol);
+	if (err < 0)
+		return err;
+	return 0;
+}
+
+static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx,
+		      int num_ctls, int type)
+{
+	struct cs_spec *spec = codec->spec;
+	const char *name;
+	int err, index;
+	struct snd_kcontrol *kctl;
+	static char *speakers[] = {
+		"Front Speaker", "Surround Speaker", "Bass Speaker"
+	};
+	static char *line_outs[] = {
+		"Front Line-Out", "Surround Line-Out", "Bass Line-Out"
+	};
+
+	fix_volume_caps(codec, dac);
+	if (!spec->vmaster_sw) {
+		err = add_vmaster(codec, dac);
+		if (err < 0)
+			return err;
+	}
+
+	index = 0;
+	switch (type) {
+	case AUTO_PIN_HP_OUT:
+		name = "Headphone";
+		index = idx;
+		break;
+	case AUTO_PIN_SPEAKER_OUT:
+		if (num_ctls > 1)
+			name = speakers[idx];
+		else
+			name = "Speaker";
+		break;
+	default:
+		if (num_ctls > 1)
+			name = line_outs[idx];
+		else
+			name = "Line-Out";
+		break;
+	}
+
+	err = add_mute(codec, name, index,
+		       HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl);
+	if (err < 0)
+		return err;
+	err = snd_ctl_add_slave(spec->vmaster_sw, kctl);
+	if (err < 0)
+		return err;
+
+	err = add_volume(codec, name, index,
+			 HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl);
+	if (err < 0)
+		return err;
+	err = snd_ctl_add_slave(spec->vmaster_vol, kctl);
+	if (err < 0)
+		return err;
+
+	return 0;
+}		
+
+static int build_output(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i, err, extra_nids;
+	hda_nid_t dac;
+
+	for (i = 0; i < cfg->line_outs; i++) {
+		dac = get_dac(codec, cfg->line_out_pins[i]);
+		if (!dac)
+			break;
+		spec->dac_nid[i] = dac;
+		err = add_output(codec, dac, i, cfg->line_outs,
+				 cfg->line_out_type);
+		if (err < 0)
+			return err;
+	}
+	spec->multiout.num_dacs = i;
+	spec->multiout.dac_nids = spec->dac_nid;
+	spec->multiout.max_channels = i * 2;
+
+	/* add HP and speakers */
+	extra_nids = 0;
+	for (i = 0; i < cfg->hp_outs; i++) {
+		dac = get_dac(codec, cfg->hp_pins[i]);
+		if (!dac)
+			break;
+		if (!i)
+			spec->multiout.hp_nid = dac;
+		else
+			spec->multiout.extra_out_nid[extra_nids++] = dac;
+		err = add_output(codec, dac, i, cfg->hp_outs, AUTO_PIN_HP_OUT);
+		if (err < 0)
+			return err;
+	}
+	for (i = 0; i < cfg->speaker_outs; i++) {
+		dac = get_dac(codec, cfg->speaker_pins[i]);
+		if (!dac)
+			break;
+		spec->multiout.extra_out_nid[extra_nids++] = dac;
+		err = add_output(codec, dac, i, cfg->speaker_outs,
+				 AUTO_PIN_SPEAKER_OUT);
+		if (err < 0)
+			return err;
+	}
+
+	if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
+		cfg->speaker_outs = cfg->line_outs;
+		memcpy(cfg->speaker_pins, cfg->line_out_pins,
+		       sizeof(cfg->speaker_pins));
+		cfg->line_outs = 0;
+	}
+
+	return 0;
+}
+
+/*
+ */
+
+static struct snd_kcontrol_new cs_capture_ctls[] = {
+	HDA_BIND_SW("Capture Switch", 0),
+	HDA_BIND_VOL("Capture Volume", 0),
+};
+
+static int change_cur_input(struct hda_codec *codec, unsigned int idx)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	
+	if (spec->cur_input == idx)
+		return 0;
+	if (spec->cur_adc && spec->cur_adc != spec->adc_nid[idx]) {
+		/* stream is running, let's swap the current ADC */
+		snd_hda_codec_cleanup_stream(codec, spec->cur_adc);
+		spec->cur_adc = spec->adc_nid[idx];
+		snd_hda_codec_setup_stream(codec, spec->cur_adc,
+					   spec->cur_adc_stream_tag, 0,
+					   spec->cur_adc_format);
+	}
+	snd_hda_codec_write(codec, spec->cur_adc, 0,
+			    AC_VERB_SET_CONNECT_SEL,
+			    spec->adc_idx[idx]);
+	spec->cur_input = idx;
+	return 1;
+}
+
+static int cs_capture_source_info(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_info *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cs_spec *spec = codec->spec;
+	unsigned int idx;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = spec->num_inputs;
+	if (uinfo->value.enumerated.item >= spec->num_inputs)
+		uinfo->value.enumerated.item = spec->num_inputs - 1;
+	idx = spec->input_idx[uinfo->value.enumerated.item];
+	strcpy(uinfo->value.enumerated.name, auto_pin_cfg_labels[idx]);
+	return 0;
+}
+
+static int cs_capture_source_get(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cs_spec *spec = codec->spec;
+	ucontrol->value.enumerated.item[0] = spec->capsrc_idx[spec->cur_input];
+	return 0;
+}
+
+static int cs_capture_source_put(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cs_spec *spec = codec->spec;
+	unsigned int idx = ucontrol->value.enumerated.item[0];
+
+	if (idx >= spec->num_inputs)
+		return -EINVAL;
+	idx = spec->input_idx[idx];
+	return change_cur_input(codec, idx);
+}
+
+static struct snd_kcontrol_new cs_capture_source = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Capture Source",
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.info = cs_capture_source_info,
+	.get = cs_capture_source_get,
+	.put = cs_capture_source_put,
+};
+
+
+static int is_ext_mic(struct hda_codec *codec, unsigned int idx)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	hda_nid_t pin = cfg->input_pins[idx];
+	unsigned int val = snd_hda_query_pin_caps(codec, pin);
+	if (!(val & AC_PINCAP_PRES_DETECT))
+		return 0;
+	val = snd_hda_codec_get_pincfg(codec, pin);
+	return (get_defcfg_connect(val) == AC_JACK_PORT_COMPLEX);
+}
+
+static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
+			 unsigned int *idxp)
+{
+	int i;
+	hda_nid_t nid;
+
+	nid = codec->start_nid;
+	for (i = 0; i < codec->num_nodes; i++, nid++) {
+		hda_nid_t pins[2];
+		unsigned int type;
+		int j, nums;
+		type = (get_wcaps(codec, nid) & AC_WCAP_TYPE)
+			>> AC_WCAP_TYPE_SHIFT;
+		if (type != AC_WID_AUD_IN)
+			continue;
+		nums = snd_hda_get_connections(codec, nid, pins,
+					       ARRAY_SIZE(pins));
+		if (nums <= 0)
+			continue;
+		for (j = 0; j < nums; j++) {
+			if (pins[j] == pin) {
+				*idxp = j;
+				return nid;
+			}
+		}
+	}
+	return 0;
+}
+
+static struct hda_bind_ctls *make_bind_capture(struct hda_codec *codec,
+					       struct hda_ctl_ops *ops)
+{
+	struct cs_spec *spec = codec->spec;
+	struct hda_bind_ctls *bind;
+	int i, n;
+
+	bind = kzalloc(sizeof(*bind) + sizeof(long) * (spec->num_inputs + 1),
+		       GFP_KERNEL);
+	if (!bind)
+		return NULL;
+	bind->ops = ops;
+	n = 0;
+	for (i = 0; i < AUTO_PIN_LAST; i++) {
+		if (!spec->adc_nid[i])
+			continue;
+		bind->values[n++] =
+			HDA_COMPOSE_AMP_VAL(spec->adc_nid[i], 3,
+					    spec->adc_idx[i], HDA_INPUT);
+	}
+	return bind;
+}
+
+static int build_input(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i, n, err;
+
+	for (i = 0; i < AUTO_PIN_LAST; i++) {
+		hda_nid_t pin = cfg->input_pins[i];
+		struct snd_kcontrol *kctl;
+		if (!pin)
+			continue;
+		spec->input_idx[spec->num_inputs] = i;
+		spec->capsrc_idx[i] = spec->num_inputs++;
+		spec->cur_input = i;
+		spec->adc_nid[i] = get_adc(codec, pin, &spec->adc_idx[i]);
+	}
+	if (!spec->num_inputs)
+		return 0;
+
+	/* check whether the automatic mic switch is available */
+	if (spec->num_inputs == 2 &&
+	    spec->adc_nid[AUTO_PIN_MIC] && spec->adc_nid[AUTO_PIN_FRONT_MIC]) {
+		if (is_ext_mic(codec, cfg->input_pins[AUTO_PIN_FRONT_MIC])) {
+			if (!is_ext_mic(codec, cfg->input_pins[AUTO_PIN_MIC])) {
+				spec->mic_detect = 1;
+				spec->automic_idx = AUTO_PIN_FRONT_MIC;
+			}
+		} else {
+			if (is_ext_mic(codec, cfg->input_pins[AUTO_PIN_MIC])) {
+				spec->mic_detect = 1;
+				spec->automic_idx = AUTO_PIN_MIC;
+			}
+		}
+	}
+
+	/* make bind-capture */
+	spec->capture_bind[0] = make_bind_capture(codec, &snd_hda_bind_sw);
+	spec->capture_bind[1] = make_bind_capture(codec, &snd_hda_bind_vol);
+	for (i = 0; i < 2; i++) {
+		struct snd_kcontrol *kctl;
+		if (!spec->capture_bind[i])
+			return -ENOMEM;
+		kctl = snd_ctl_new1(&cs_capture_ctls[i], codec);
+		if (!kctl)
+			return -ENOMEM;
+		kctl->private_value = (long)spec->capture_bind[i];
+		err = snd_hda_ctl_add(codec, kctl);
+		if (err < 0)
+			return err;
+	}
+	
+	if (spec->num_inputs > 1 && !spec->mic_detect) {
+		err = snd_hda_ctl_add(codec,
+				      snd_ctl_new1(&cs_capture_source, codec));
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+static int build_digital_output(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	hda_nid_t nid;
+	int err;
+
+	if (!cfg->dig_outs)
+		return 0;
+	if (snd_hda_get_connections(codec, cfg->dig_out_pins[0], &nid, 1) < 1)
+		return 0;
+	spec->multiout.dig_out_nid = nid;
+	err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+	if (err < 0)
+		return err;
+	err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);
+	if (err < 0)
+		return err;
+	spec->multiout.share_spdif = 1;
+	if (cfg->dig_outs > 1 &&
+	    snd_hda_get_connections(codec, cfg->dig_out_pins[1], &nid, 1) > 0) {
+		spec->slave_dig_outs[0] = nid;
+		codec->slave_dig_outs = spec->slave_dig_outs;
+	}
+	return 0;
+}
+
+static int build_digital_input(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int idx;
+
+	if (!cfg->dig_in_pin)
+		return 0;
+	spec->dig_in = get_adc(codec, cfg->dig_in_pin, &idx);
+	if (!spec->dig_in)
+		return 0;
+	return snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
+}
+
+static void cs_automute(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	unsigned int caps, present, hp_present;
+	hda_nid_t nid;
+	int i;
+
+	hp_present = 0;
+	for (i = 0; i < cfg->hp_outs; i++) {
+		nid = cfg->hp_pins[i];
+		caps = snd_hda_query_pin_caps(codec, nid);
+		if (!(caps & AC_PINCAP_PRES_DETECT))
+			continue;
+		if (caps & AC_PINCAP_TRIG_REQ)
+			snd_hda_codec_read(codec, nid, 0,
+					   AC_VERB_SET_PIN_SENSE, 0);
+		present = snd_hda_codec_read(codec, nid, 0,
+					     AC_VERB_GET_PIN_SENSE, 0);
+		hp_present |= (present & AC_PINSENSE_PRESENCE) != 0;
+		if (hp_present)
+			break;
+	}
+	for (i = 0; i < cfg->speaker_outs; i++) {
+		nid = cfg->speaker_pins[i];
+		snd_hda_codec_write(codec, nid, 0,
+				    AC_VERB_SET_PIN_WIDGET_CONTROL,
+				    hp_present ? 0 : PIN_OUT);
+	}
+}
+
+static void cs_automic(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	hda_nid_t nid;
+	unsigned int caps, present;
+	
+	nid = cfg->input_pins[spec->automic_idx];
+	caps = snd_hda_query_pin_caps(codec, nid);
+	if (caps & AC_PINCAP_TRIG_REQ)
+		snd_hda_codec_read(codec, nid, 0, AC_VERB_SET_PIN_SENSE, 0);
+	present = snd_hda_codec_read(codec, nid, 0,
+				     AC_VERB_GET_PIN_SENSE, 0);
+	if (present & AC_PINSENSE_PRESENCE)
+		change_cur_input(codec, spec->automic_idx);
+	else {
+		unsigned int imic = (spec->automic_idx == AUTO_PIN_MIC) ?
+			AUTO_PIN_FRONT_MIC : AUTO_PIN_MIC;
+		change_cur_input(codec, imic);
+	}
+}
+
+/*
+ */
+
+static void init_output(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i;
+
+	/* mute first */
+	for (i = 0; i < spec->multiout.num_dacs; i++)
+		snd_hda_codec_write(codec, spec->multiout.dac_nids[i], 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+	if (spec->multiout.hp_nid)
+		snd_hda_codec_write(codec, spec->multiout.hp_nid, 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+	for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++) {
+		if (!spec->multiout.extra_out_nid[i])
+			break;
+		snd_hda_codec_write(codec, spec->multiout.extra_out_nid[i], 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+	}
+
+	/* set appropriate pin controls */
+	for (i = 0; i < cfg->line_outs; i++)
+		snd_hda_codec_write(codec, cfg->line_out_pins[i], 0,
+				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+	for (i = 0; i < cfg->hp_outs; i++) {
+		hda_nid_t nid = cfg->hp_pins[i];
+		snd_hda_codec_write(codec, nid, 0,
+				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP);
+		if (!cfg->speaker_outs)
+			continue;
+		if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) {
+			snd_hda_codec_write(codec, nid, 0,
+					    AC_VERB_SET_UNSOLICITED_ENABLE,
+					    AC_USRSP_EN | HP_EVENT);
+			spec->hp_detect = 1;
+		}
+	}
+	for (i = 0; i < cfg->speaker_outs; i++)
+		snd_hda_codec_write(codec, cfg->speaker_pins[i], 0,
+				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+	if (spec->hp_detect)
+		cs_automute(codec);
+}
+
+static void init_input(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i;
+
+	for (i = 0; i < AUTO_PIN_LAST; i++) {
+		unsigned int ctl;
+		hda_nid_t pin = cfg->input_pins[i];
+		if (!pin || !spec->adc_nid[i])
+			continue;
+		/* set appropriate pin control and mute first */
+		ctl = PIN_IN;
+		if (i <= AUTO_PIN_FRONT_MIC) {
+			unsigned int caps = snd_hda_query_pin_caps(codec, pin);
+			caps >>= AC_PINCAP_VREF_SHIFT;
+			if (caps & AC_PINCAP_VREF_80)
+				ctl = PIN_VREF80;
+		}
+		snd_hda_codec_write(codec, pin, 0,
+				    AC_VERB_SET_PIN_WIDGET_CONTROL, ctl);
+		snd_hda_codec_write(codec, spec->adc_nid[i], 0,
+				    AC_VERB_SET_AMP_GAIN_MUTE,
+				    AMP_IN_MUTE(spec->adc_idx[i]));
+		if (spec->mic_detect && spec->automic_idx == i)
+			snd_hda_codec_write(codec, pin, 0,
+					    AC_VERB_SET_UNSOLICITED_ENABLE,
+					    AC_USRSP_EN | MIC_EVENT);
+	}
+	if (spec->mic_detect)
+		cs_automic(codec);
+}
+
+static int cs_init(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+
+	if (!spec->built_up)
+		return 0;
+	init_output(codec);
+	init_input(codec);
+	return 0;
+}
+
+static int cs_build_controls(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	int err;
+
+	err = build_output(codec);
+	if (err < 0)
+		return err;
+	err = build_input(codec);
+	if (err < 0)
+		return err;
+	err = build_digital_output(codec);
+	if (err < 0)
+		return err;
+	err = build_digital_input(codec);
+	if (err < 0)
+		return err;
+	spec->built_up = 1;
+	return cs_init(codec);
+}
+
+static void cs_free(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	kfree(spec->capture_bind[0]);
+	kfree(spec->capture_bind[1]);
+	kfree(codec->spec);
+}
+
+static void cs_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+	switch ((res >> 26) & 0x7f) {
+	case HP_EVENT:
+		cs_automute(codec);
+		break;
+	case MIC_EVENT:
+		cs_automic(codec);
+		break;
+	}
+}
+
+static struct hda_codec_ops cs_patch_ops = {
+	.build_controls = cs_build_controls,
+	.build_pcms = cs_build_pcms,
+	.init = cs_init,
+	.free = cs_free,
+	.unsol_event = cs_unsol_event,
+};
+
+static int cs_parse_auto_config(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	int err;
+
+	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
+	if (err < 0)
+		return err;
+	return 0;
+}
+
+
+static int patch_cs420x(struct hda_codec *codec)
+{
+	struct cs_spec *spec;
+	int err;
+
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (!spec)
+		return -ENOMEM;
+	codec->spec = spec;
+
+	err = cs_parse_auto_config(codec);
+	if (err < 0)
+		goto error;
+
+	codec->patch_ops = cs_patch_ops;
+
+	return 0;
+
+ error:
+	kfree(codec->spec);
+	codec->spec = NULL;
+	return err;
+}
+
+
+/*
+ * patch entries
+ */
+static struct hda_codec_preset snd_hda_preset_cirrus[] = {
+	{ .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x },
+	{ .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x },
+	{} /* terminator */
+};
+
+MODULE_ALIAS("snd-hda-codec-id:10134206");
+MODULE_ALIAS("snd-hda-codec-id:10134207");
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Cirrus Logic HD-audio codec");
+
+static struct hda_codec_preset_list cirrus_list = {
+	.preset = snd_hda_preset_cirrus,
+	.owner = THIS_MODULE,
+};
+
+static int __init patch_cirrus_init(void)
+{
+	return snd_hda_add_codec_preset(&cirrus_list);
+}
+
+static void __exit patch_cirrus_exit(void)
+{
+	snd_hda_delete_codec_preset(&cirrus_list);
+}
+
+module_init(patch_cirrus_init)
+module_exit(patch_cirrus_exit)
-- 
cgit v1.2.3


From cb6381225a8064b0911dced3eb10f00bd5520c85 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 3 Jul 2009 10:56:10 +0200
Subject: ALSA: hda - Add missing mixer amp initialization for ALC882

After merting patch_alc882() and patch_alc883(), the initialization of
mixer amp 0x0b was missing in alc882_base_init_verbs[].
This is usually no critical problem, but it can disable the power-saving
as the default state, so better to put to mute these channels.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 7 +++++++
 1 file changed, 7 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 0f6b6a6f72e..d15d83ee4ad 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6849,6 +6849,13 @@ static struct hda_verb alc882_base_init_verbs[] = {
 	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
 
+	/* mute analog input loopbacks */
+	{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)},
+
 	/* Front Pin: output 0 (0x0c) */
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
 	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-- 
cgit v1.2.3


From 8ab9e0af6d7709a781b60a51711ddf1d43bd22bb Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 3 Jul 2009 10:58:12 +0200
Subject: ALSA: hda - Manually expand alc882_init_verbs

Instead of expanding alc882_init_verbs to two elements via a macro,
manually expand to each entry.  This makes clear that some have already
the full slot for init_verbs array (currently 5).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 26 ++++++++++++++------------
 1 file changed, 14 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d15d83ee4ad..07e260a5ddf 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6922,9 +6922,6 @@ static struct hda_verb alc882_adc1_init_verbs[] = {
 	{ }
 };
 
-/* HACK - expand to two elements */
-#define alc882_init_verbs	alc882_base_init_verbs, alc882_adc1_init_verbs
-
 static struct hda_verb alc882_eapd_verbs[] = {
 	/* change to EAPD mode */
 	{0x20, AC_VERB_SET_COEF_INDEX, 0x07},
@@ -8492,7 +8489,8 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
 static struct alc_config_preset alc882_presets[] = {
 	[ALC882_3ST_DIG] = {
 		.mixers = { alc882_base_mixer },
-		.init_verbs = { alc882_init_verbs },
+		.init_verbs = { alc882_base_init_verbs,
+				alc882_adc1_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
 		.dac_nids = alc882_dac_nids,
 		.dig_out_nid = ALC882_DIGOUT_NID,
@@ -8504,7 +8502,8 @@ static struct alc_config_preset alc882_presets[] = {
 	},
 	[ALC882_6ST_DIG] = {
 		.mixers = { alc882_base_mixer, alc882_chmode_mixer },
-		.init_verbs = { alc882_init_verbs },
+		.init_verbs = { alc882_base_init_verbs,
+				alc882_adc1_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
 		.dac_nids = alc882_dac_nids,
 		.dig_out_nid = ALC882_DIGOUT_NID,
@@ -8515,7 +8514,8 @@ static struct alc_config_preset alc882_presets[] = {
 	},
 	[ALC882_ARIMA] = {
 		.mixers = { alc882_base_mixer, alc882_chmode_mixer },
-		.init_verbs = { alc882_init_verbs, alc882_eapd_verbs },
+		.init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+				alc882_eapd_verbs },
 		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
 		.dac_nids = alc882_dac_nids,
 		.num_channel_mode = ARRAY_SIZE(alc882_sixstack_modes),
@@ -8524,8 +8524,8 @@ static struct alc_config_preset alc882_presets[] = {
 	},
 	[ALC882_W2JC] = {
 		.mixers = { alc882_w2jc_mixer, alc882_chmode_mixer },
-		.init_verbs = { alc882_init_verbs, alc882_eapd_verbs,
-				alc880_gpio1_init_verbs },
+		.init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+				alc882_eapd_verbs, alc880_gpio1_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
 		.dac_nids = alc882_dac_nids,
 		.num_channel_mode = ARRAY_SIZE(alc880_threestack_modes),
@@ -8587,7 +8587,8 @@ static struct alc_config_preset alc882_presets[] = {
 	},
 	[ALC882_TARGA] = {
 		.mixers = { alc882_targa_mixer, alc882_chmode_mixer },
-		.init_verbs = { alc882_init_verbs, alc882_targa_verbs},
+		.init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+				alc882_targa_verbs},
 		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
 		.dac_nids = alc882_dac_nids,
 		.dig_out_nid = ALC882_DIGOUT_NID,
@@ -8603,7 +8604,8 @@ static struct alc_config_preset alc882_presets[] = {
 	},
 	[ALC882_ASUS_A7J] = {
 		.mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer },
-		.init_verbs = { alc882_init_verbs, alc882_asus_a7j_verbs},
+		.init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+				alc882_asus_a7j_verbs},
 		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
 		.dac_nids = alc882_dac_nids,
 		.dig_out_nid = ALC882_DIGOUT_NID,
@@ -8617,8 +8619,8 @@ static struct alc_config_preset alc882_presets[] = {
 	},
 	[ALC882_ASUS_A7M] = {
 		.mixers = { alc882_asus_a7m_mixer, alc882_chmode_mixer },
-		.init_verbs = { alc882_init_verbs, alc882_eapd_verbs,
-				alc880_gpio1_init_verbs,
+		.init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
+				alc882_eapd_verbs, alc880_gpio1_init_verbs,
 				alc882_asus_a7m_verbs },
 		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
 		.dac_nids = alc882_dac_nids,
-- 
cgit v1.2.3


From b0bb3aa6233dccfccd040793d0d9ce838e4890e1 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 3 Jul 2009 23:25:37 +0200
Subject: ALSA: hda - Don't override maxbps for FLOAT sharing with linear
 formats

When FLOAT PCM format is available but together with other linear
PCM formats, don't override maxbps value.  For FLOAT format, it's always
32, thus it can be better checked in snd_hda_calc_stream_format().
Otherwise the maxbps 32 might be used wrongly even if the linear PCM
doesn't support it.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 263d124de61..fb5760c64e7 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2563,7 +2563,7 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate,
 	case 20:
 	case 24:
 	case 32:
-		if (maxbps >= 32)
+		if (maxbps >= 32 || format == SNDRV_PCM_FORMAT_FLOAT_LE)
 			val |= 0x40;
 		else if (maxbps >= 24)
 			val |= 0x30;
@@ -2692,7 +2692,8 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
 		}
 		if (streams & AC_SUPFMT_FLOAT32) {
 			formats |= SNDRV_PCM_FMTBIT_FLOAT_LE;
-			bps = 32;
+			if (!bps)
+				bps = 32;
 		}
 		if (streams == AC_SUPFMT_AC3) {
 			/* should be exclusive */
-- 
cgit v1.2.3


From 21a4dc43acdd547335f59ad52efeef5160247736 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 6 Jul 2009 12:55:46 +0200
Subject: ALSA: hda - Fix cirrus codec parsing

The parser wasn't called in the proper order.
Split now the parser to be called in patch_cirrus(), and the rest
are just for building PCMs and controls.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_cirrus.c | 331 ++++++++++++++++++++++++++-----------------
 1 file changed, 198 insertions(+), 133 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index c4dc12e8573..051302e7834 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -56,7 +56,6 @@ struct cs_spec {
 
 	unsigned int hp_detect:1;
 	unsigned int mic_detect:1;
-	unsigned int built_up:1;
 };
 
 #define HP_EVENT	1
@@ -213,7 +212,6 @@ static int cs_build_pcms(struct hda_codec *codec)
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max =
 		spec->multiout.max_channels;
 	info->stream[SNDRV_PCM_STREAM_CAPTURE] = cs_pcm_analog_capture;
-	info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_inputs;
 	info->stream[SNDRV_PCM_STREAM_CAPTURE].nid =
 		spec->adc_nid[spec->cur_input];
 	codec->num_pcms++;
@@ -242,6 +240,10 @@ static int cs_build_pcms(struct hda_codec *codec)
 	return 0;
 }
 
+/*
+ * parse codec topology
+ */
+
 static hda_nid_t get_dac(struct hda_codec *codec, hda_nid_t pin)
 {
 	hda_nid_t dac;
@@ -252,6 +254,169 @@ static hda_nid_t get_dac(struct hda_codec *codec, hda_nid_t pin)
 	return dac;
 }
 
+static int is_ext_mic(struct hda_codec *codec, unsigned int idx)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	hda_nid_t pin = cfg->input_pins[idx];
+	unsigned int val = snd_hda_query_pin_caps(codec, pin);
+	if (!(val & AC_PINCAP_PRES_DETECT))
+		return 0;
+	val = snd_hda_codec_get_pincfg(codec, pin);
+	return (get_defcfg_connect(val) == AC_JACK_PORT_COMPLEX);
+}
+
+static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
+			 unsigned int *idxp)
+{
+	int i;
+	hda_nid_t nid;
+
+	nid = codec->start_nid;
+	for (i = 0; i < codec->num_nodes; i++, nid++) {
+		hda_nid_t pins[2];
+		unsigned int type;
+		int j, nums;
+		type = (get_wcaps(codec, nid) & AC_WCAP_TYPE)
+			>> AC_WCAP_TYPE_SHIFT;
+		if (type != AC_WID_AUD_IN)
+			continue;
+		nums = snd_hda_get_connections(codec, nid, pins,
+					       ARRAY_SIZE(pins));
+		if (nums <= 0)
+			continue;
+		for (j = 0; j < nums; j++) {
+			if (pins[j] == pin) {
+				*idxp = j;
+				return nid;
+			}
+		}
+	}
+	return 0;
+}
+
+static int parse_output(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i, err, extra_nids;
+	hda_nid_t dac;
+
+	for (i = 0; i < cfg->line_outs; i++) {
+		dac = get_dac(codec, cfg->line_out_pins[i]);
+		if (!dac)
+			break;
+		spec->dac_nid[i] = dac;
+	}
+	spec->multiout.num_dacs = i;
+	spec->multiout.dac_nids = spec->dac_nid;
+	spec->multiout.max_channels = i * 2;
+
+	/* add HP and speakers */
+	extra_nids = 0;
+	for (i = 0; i < cfg->hp_outs; i++) {
+		dac = get_dac(codec, cfg->hp_pins[i]);
+		if (!dac)
+			break;
+		if (!i)
+			spec->multiout.hp_nid = dac;
+		else
+			spec->multiout.extra_out_nid[extra_nids++] = dac;
+	}
+	for (i = 0; i < cfg->speaker_outs; i++) {
+		dac = get_dac(codec, cfg->speaker_pins[i]);
+		if (!dac)
+			break;
+		spec->multiout.extra_out_nid[extra_nids++] = dac;
+	}
+
+	if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
+		cfg->speaker_outs = cfg->line_outs;
+		memcpy(cfg->speaker_pins, cfg->line_out_pins,
+		       sizeof(cfg->speaker_pins));
+		cfg->line_outs = 0;
+	}
+
+	return 0;
+}
+
+static int parse_input(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i, n, err;
+
+	for (i = 0; i < AUTO_PIN_LAST; i++) {
+		hda_nid_t pin = cfg->input_pins[i];
+		struct snd_kcontrol *kctl;
+		if (!pin)
+			continue;
+		spec->input_idx[spec->num_inputs] = i;
+		spec->capsrc_idx[i] = spec->num_inputs++;
+		spec->cur_input = i;
+		spec->adc_nid[i] = get_adc(codec, pin, &spec->adc_idx[i]);
+	}
+	if (!spec->num_inputs)
+		return 0;
+
+	/* check whether the automatic mic switch is available */
+	if (spec->num_inputs == 2 &&
+	    spec->adc_nid[AUTO_PIN_MIC] && spec->adc_nid[AUTO_PIN_FRONT_MIC]) {
+		if (is_ext_mic(codec, cfg->input_pins[AUTO_PIN_FRONT_MIC])) {
+			if (!is_ext_mic(codec, cfg->input_pins[AUTO_PIN_MIC])) {
+				spec->mic_detect = 1;
+				spec->automic_idx = AUTO_PIN_FRONT_MIC;
+			}
+		} else {
+			if (is_ext_mic(codec, cfg->input_pins[AUTO_PIN_MIC])) {
+				spec->mic_detect = 1;
+				spec->automic_idx = AUTO_PIN_MIC;
+			}
+		}
+	}
+	return 0;
+}
+
+
+static int parse_digital_output(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	hda_nid_t nid;
+	int err;
+
+	if (!cfg->dig_outs)
+		return 0;
+	if (snd_hda_get_connections(codec, cfg->dig_out_pins[0], &nid, 1) < 1)
+		return 0;
+	spec->multiout.dig_out_nid = nid;
+	spec->multiout.share_spdif = 1;
+	if (cfg->dig_outs > 1 &&
+	    snd_hda_get_connections(codec, cfg->dig_out_pins[1], &nid, 1) > 0) {
+		spec->slave_dig_outs[0] = nid;
+		codec->slave_dig_outs = spec->slave_dig_outs;
+	}
+	return 0;
+}
+
+static int parse_digital_input(struct hda_codec *codec)
+{
+	struct cs_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int idx;
+
+	if (!cfg->dig_in_pin)
+		return 0;
+	spec->dig_in = get_adc(codec, cfg->dig_in_pin, &idx);
+	if (!spec->dig_in)
+		return 0;
+	return snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
+}
+
+/*
+ * create mixer controls
+ */
+
 static const char *dir_sfx[2] = { "Playback", "Capture" };
 
 static int add_mute(struct hda_codec *codec, const char *name, int index,
@@ -376,55 +541,26 @@ static int build_output(struct hda_codec *codec)
 {
 	struct cs_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
-	int i, err, extra_nids;
-	hda_nid_t dac;
+	int i, err;
 
 	for (i = 0; i < cfg->line_outs; i++) {
-		dac = get_dac(codec, cfg->line_out_pins[i]);
-		if (!dac)
-			break;
-		spec->dac_nid[i] = dac;
-		err = add_output(codec, dac, i, cfg->line_outs,
-				 cfg->line_out_type);
+		err = add_output(codec, get_dac(codec, cfg->line_out_pins[i]),
+				 i, cfg->line_outs, cfg->line_out_type);
 		if (err < 0)
 			return err;
 	}
-	spec->multiout.num_dacs = i;
-	spec->multiout.dac_nids = spec->dac_nid;
-	spec->multiout.max_channels = i * 2;
-
-	/* add HP and speakers */
-	extra_nids = 0;
 	for (i = 0; i < cfg->hp_outs; i++) {
-		dac = get_dac(codec, cfg->hp_pins[i]);
-		if (!dac)
-			break;
-		if (!i)
-			spec->multiout.hp_nid = dac;
-		else
-			spec->multiout.extra_out_nid[extra_nids++] = dac;
-		err = add_output(codec, dac, i, cfg->hp_outs, AUTO_PIN_HP_OUT);
+		err = add_output(codec, get_dac(codec, cfg->hp_pins[i]),
+				 i, cfg->hp_outs, AUTO_PIN_HP_OUT);
 		if (err < 0)
 			return err;
 	}
 	for (i = 0; i < cfg->speaker_outs; i++) {
-		dac = get_dac(codec, cfg->speaker_pins[i]);
-		if (!dac)
-			break;
-		spec->multiout.extra_out_nid[extra_nids++] = dac;
-		err = add_output(codec, dac, i, cfg->speaker_outs,
-				 AUTO_PIN_SPEAKER_OUT);
+		err = add_output(codec, get_dac(codec, cfg->speaker_pins[i]),
+				 i, cfg->speaker_outs, AUTO_PIN_SPEAKER_OUT);
 		if (err < 0)
 			return err;
 	}
-
-	if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
-		cfg->speaker_outs = cfg->line_outs;
-		memcpy(cfg->speaker_pins, cfg->line_out_pins,
-		       sizeof(cfg->speaker_pins));
-		cfg->line_outs = 0;
-	}
-
 	return 0;
 }
 
@@ -506,48 +642,6 @@ static struct snd_kcontrol_new cs_capture_source = {
 	.put = cs_capture_source_put,
 };
 
-
-static int is_ext_mic(struct hda_codec *codec, unsigned int idx)
-{
-	struct cs_spec *spec = codec->spec;
-	struct auto_pin_cfg *cfg = &spec->autocfg;
-	hda_nid_t pin = cfg->input_pins[idx];
-	unsigned int val = snd_hda_query_pin_caps(codec, pin);
-	if (!(val & AC_PINCAP_PRES_DETECT))
-		return 0;
-	val = snd_hda_codec_get_pincfg(codec, pin);
-	return (get_defcfg_connect(val) == AC_JACK_PORT_COMPLEX);
-}
-
-static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
-			 unsigned int *idxp)
-{
-	int i;
-	hda_nid_t nid;
-
-	nid = codec->start_nid;
-	for (i = 0; i < codec->num_nodes; i++, nid++) {
-		hda_nid_t pins[2];
-		unsigned int type;
-		int j, nums;
-		type = (get_wcaps(codec, nid) & AC_WCAP_TYPE)
-			>> AC_WCAP_TYPE_SHIFT;
-		if (type != AC_WID_AUD_IN)
-			continue;
-		nums = snd_hda_get_connections(codec, nid, pins,
-					       ARRAY_SIZE(pins));
-		if (nums <= 0)
-			continue;
-		for (j = 0; j < nums; j++) {
-			if (pins[j] == pin) {
-				*idxp = j;
-				return nid;
-			}
-		}
-	}
-	return 0;
-}
-
 static struct hda_bind_ctls *make_bind_capture(struct hda_codec *codec,
 					       struct hda_ctl_ops *ops)
 {
@@ -574,38 +668,11 @@ static struct hda_bind_ctls *make_bind_capture(struct hda_codec *codec,
 static int build_input(struct hda_codec *codec)
 {
 	struct cs_spec *spec = codec->spec;
-	struct auto_pin_cfg *cfg = &spec->autocfg;
-	int i, n, err;
+	int i, err;
 
-	for (i = 0; i < AUTO_PIN_LAST; i++) {
-		hda_nid_t pin = cfg->input_pins[i];
-		struct snd_kcontrol *kctl;
-		if (!pin)
-			continue;
-		spec->input_idx[spec->num_inputs] = i;
-		spec->capsrc_idx[i] = spec->num_inputs++;
-		spec->cur_input = i;
-		spec->adc_nid[i] = get_adc(codec, pin, &spec->adc_idx[i]);
-	}
 	if (!spec->num_inputs)
 		return 0;
 
-	/* check whether the automatic mic switch is available */
-	if (spec->num_inputs == 2 &&
-	    spec->adc_nid[AUTO_PIN_MIC] && spec->adc_nid[AUTO_PIN_FRONT_MIC]) {
-		if (is_ext_mic(codec, cfg->input_pins[AUTO_PIN_FRONT_MIC])) {
-			if (!is_ext_mic(codec, cfg->input_pins[AUTO_PIN_MIC])) {
-				spec->mic_detect = 1;
-				spec->automic_idx = AUTO_PIN_FRONT_MIC;
-			}
-		} else {
-			if (is_ext_mic(codec, cfg->input_pins[AUTO_PIN_MIC])) {
-				spec->mic_detect = 1;
-				spec->automic_idx = AUTO_PIN_MIC;
-			}
-		}
-	}
-
 	/* make bind-capture */
 	spec->capture_bind[0] = make_bind_capture(codec, &snd_hda_bind_sw);
 	spec->capture_bind[1] = make_bind_capture(codec, &snd_hda_bind_vol);
@@ -632,47 +699,35 @@ static int build_input(struct hda_codec *codec)
 	return 0;
 }
 
+/*
+ */
+
 static int build_digital_output(struct hda_codec *codec)
 {
 	struct cs_spec *spec = codec->spec;
-	struct auto_pin_cfg *cfg = &spec->autocfg;
-	hda_nid_t nid;
 	int err;
 
-	if (!cfg->dig_outs)
-		return 0;
-	if (snd_hda_get_connections(codec, cfg->dig_out_pins[0], &nid, 1) < 1)
-		return 0;
-	spec->multiout.dig_out_nid = nid;
 	err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
 	if (err < 0)
 		return err;
 	err = snd_hda_create_spdif_share_sw(codec, &spec->multiout);
 	if (err < 0)
 		return err;
-	spec->multiout.share_spdif = 1;
-	if (cfg->dig_outs > 1 &&
-	    snd_hda_get_connections(codec, cfg->dig_out_pins[1], &nid, 1) > 0) {
-		spec->slave_dig_outs[0] = nid;
-		codec->slave_dig_outs = spec->slave_dig_outs;
-	}
 	return 0;
 }
 
 static int build_digital_input(struct hda_codec *codec)
 {
 	struct cs_spec *spec = codec->spec;
-	struct auto_pin_cfg *cfg = &spec->autocfg;
-	int idx;
-
-	if (!cfg->dig_in_pin)
-		return 0;
-	spec->dig_in = get_adc(codec, cfg->dig_in_pin, &idx);
-	if (!spec->dig_in)
-		return 0;
-	return snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
+	if (spec->dig_in)
+		return snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
+	return 0;
 }
 
+/*
+ * auto-mute and auto-mic switching
+ */
+
 static void cs_automute(struct hda_codec *codec)
 {
 	struct cs_spec *spec = codec->spec;
@@ -810,8 +865,6 @@ static int cs_init(struct hda_codec *codec)
 {
 	struct cs_spec *spec = codec->spec;
 
-	if (!spec->built_up)
-		return 0;
 	init_output(codec);
 	init_input(codec);
 	return 0;
@@ -834,7 +887,6 @@ static int cs_build_controls(struct hda_codec *codec)
 	err = build_digital_input(codec);
 	if (err < 0)
 		return err;
-	spec->built_up = 1;
 	return cs_init(codec);
 }
 
@@ -892,6 +944,19 @@ static int patch_cs420x(struct hda_codec *codec)
 	if (err < 0)
 		goto error;
 
+	err = parse_output(codec);
+	if (err < 0)
+		goto error;
+	err = parse_input(codec);
+	if (err < 0)
+		goto error;
+	err = parse_digital_output(codec);
+	if (err < 0)
+		goto error;
+	err = parse_digital_input(codec);
+	if (err < 0)
+		goto error;
+
 	codec->patch_ops = cs_patch_ops;
 
 	return 0;
-- 
cgit v1.2.3


From ea35929b886975a240660b3ba6c61826761731ad Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 6 Jul 2009 12:58:59 +0200
Subject: ALSA: hda - Force to initialize input mixer setup for CS420x

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_cirrus.c | 12 +++++++-----
 1 file changed, 7 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 051302e7834..8e33214abeb 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -572,12 +572,13 @@ static struct snd_kcontrol_new cs_capture_ctls[] = {
 	HDA_BIND_VOL("Capture Volume", 0),
 };
 
-static int change_cur_input(struct hda_codec *codec, unsigned int idx)
+static int change_cur_input(struct hda_codec *codec, unsigned int idx,
+			    int force)
 {
 	struct cs_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
 	
-	if (spec->cur_input == idx)
+	if (spec->cur_input == idx && !force)
 		return 0;
 	if (spec->cur_adc && spec->cur_adc != spec->adc_nid[idx]) {
 		/* stream is running, let's swap the current ADC */
@@ -630,7 +631,7 @@ static int cs_capture_source_put(struct snd_kcontrol *kcontrol,
 	if (idx >= spec->num_inputs)
 		return -EINVAL;
 	idx = spec->input_idx[idx];
-	return change_cur_input(codec, idx);
+	return change_cur_input(codec, idx, 0);
 }
 
 static struct snd_kcontrol_new cs_capture_source = {
@@ -773,11 +774,11 @@ static void cs_automic(struct hda_codec *codec)
 	present = snd_hda_codec_read(codec, nid, 0,
 				     AC_VERB_GET_PIN_SENSE, 0);
 	if (present & AC_PINSENSE_PRESENCE)
-		change_cur_input(codec, spec->automic_idx);
+		change_cur_input(codec, spec->automic_idx, 0);
 	else {
 		unsigned int imic = (spec->automic_idx == AUTO_PIN_MIC) ?
 			AUTO_PIN_FRONT_MIC : AUTO_PIN_MIC;
-		change_cur_input(codec, imic);
+		change_cur_input(codec, imic, 0);
 	}
 }
 
@@ -857,6 +858,7 @@ static void init_input(struct hda_codec *codec)
 					    AC_VERB_SET_UNSOLICITED_ENABLE,
 					    AC_USRSP_EN | MIC_EVENT);
 	}
+	change_cur_input(codec, spec->cur_input, 1);
 	if (spec->mic_detect)
 		cs_automic(codec);
 }
-- 
cgit v1.2.3


From 40c20fa05a29766565f56ede17d0ffa539e1c9a9 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 6 Jul 2009 13:00:57 +0200
Subject: ALSA: hda - Add CS420x-specific coef setup

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_cirrus.c | 110 +++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 110 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 8e33214abeb..2187383cdfd 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -58,6 +58,64 @@ struct cs_spec {
 	unsigned int mic_detect:1;
 };
 
+/* Vendor-specific processing widget */
+#define CS420X_VENDOR_NID	0x11
+#define CS_DIG_OUT1_PIN_NID	0x10
+#define CS_DIG_OUT2_PIN_NID	0x15
+#define CS_DMIC1_PIN_NID	0x12
+#define CS_DMIC2_PIN_NID	0x0e
+
+/* coef indices */
+#define IDX_SPDIF_STAT		0x0000
+#define IDX_SPDIF_CTL		0x0001
+#define IDX_ADC_CFG		0x0002
+/* SZC bitmask, 4 modes below:
+ * 0 = immediate,
+ * 1 = digital immediate, analog zero-cross
+ * 2 = digtail & analog soft-ramp
+ * 3 = digital soft-ramp, analog zero-cross
+ */
+#define   CS_COEF_ADC_SZC_MASK		(3 << 0)
+#define   CS_COEF_ADC_MIC_SZC_MODE	(3 << 0) /* SZC setup for mic */
+#define   CS_COEF_ADC_LI_SZC_MODE	(3 << 0) /* SZC setup for line-in */
+/* PGA mode: 0 = differential, 1 = signle-ended */
+#define   CS_COEF_ADC_MIC_PGA_MODE	(1 << 5) /* PGA setup for mic */
+#define   CS_COEF_ADC_LI_PGA_MODE	(1 << 6) /* PGA setup for line-in */
+#define IDX_DAC_CFG		0x0003
+/* SZC bitmask, 4 modes below:
+ * 0 = Immediate
+ * 1 = zero-cross
+ * 2 = soft-ramp
+ * 3 = soft-ramp on zero-cross
+ */
+#define   CS_COEF_DAC_HP_SZC_MODE	(3 << 0) /* nid 0x02 */
+#define   CS_COEF_DAC_LO_SZC_MODE	(3 << 2) /* nid 0x03 */
+#define   CS_COEF_DAC_SPK_SZC_MODE	(3 << 4) /* nid 0x04 */
+
+#define IDX_BEEP_CFG		0x0004
+/* 0x0008 - test reg key */
+/* 0x0009 - 0x0014 -> 12 test regs */
+/* 0x0015 - visibility reg */
+
+
+static int cs_vendor_coef_get(struct hda_codec *codec, unsigned int idx)
+{
+	snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0,
+			    AC_VERB_SET_COEF_INDEX, idx);
+	return snd_hda_codec_read(codec, CS420X_VENDOR_NID, 0,
+				  AC_VERB_GET_PROC_COEF, 0);
+}
+
+static void cs_vendor_coef_set(struct hda_codec *codec, unsigned int idx,
+			       unsigned int coef)
+{
+	snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0,
+			    AC_VERB_SET_COEF_INDEX, idx);
+	snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0,
+			    AC_VERB_SET_PROC_COEF, coef);
+}
+
+
 #define HP_EVENT	1
 #define MIC_EVENT	2
 
@@ -295,6 +353,14 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
 	return 0;
 }
 
+static int is_active_pin(struct hda_codec *codec, hda_nid_t nid)
+{
+	struct cs_spec *spec = codec->spec;
+	unsigned int val;
+	val = snd_hda_codec_get_pincfg(codec, nid);
+	return (get_defcfg_connect(val) != AC_JACK_PORT_NONE);
+}
+
 static int parse_output(struct hda_codec *codec)
 {
 	struct cs_spec *spec = codec->spec;
@@ -833,6 +899,7 @@ static void init_input(struct hda_codec *codec)
 {
 	struct cs_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
+	unsigned int coef;
 	int i;
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
@@ -861,14 +928,57 @@ static void init_input(struct hda_codec *codec)
 	change_cur_input(codec, spec->cur_input, 1);
 	if (spec->mic_detect)
 		cs_automic(codec);
+
+	coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */
+	if (is_active_pin(codec, CS_DMIC2_PIN_NID))
+		coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */
+	if (is_active_pin(codec, CS_DMIC1_PIN_NID))
+		coef |= 0x1800; /* DMIC1 enable 2 channels, disable GPIO0 
+				 * No effect if SPDIF_OUT2 is slected in 
+				 * IDX_SPDIF_CTL.
+				  */
+	cs_vendor_coef_set(codec, IDX_ADC_CFG, coef);
+}
+
+static struct hda_verb cs_coef_init_verbs[] = {
+	{0x11, AC_VERB_SET_PROC_STATE, 1},
+	{0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG},
+	{0x11, AC_VERB_SET_PROC_COEF,
+	 (0x002a /* DAC1/2/3 SZCMode Soft Ramp */
+	  | 0x0040 /* Mute DACs on FIFO error */
+	  | 0x1000 /* Enable DACs High Pass Filter */
+	  | 0x0400 /* Disable Coefficient Auto increment */
+	  )},
+	/* Beep */
+	{0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG},
+	{0x11, AC_VERB_SET_PROC_COEF, 0x0007}, /* Enable Beep thru DAC1/2/3 */
+
+	{} /* terminator */
+};
+
+/* SPDIF setup */
+static void init_digital(struct hda_codec *codec)
+{
+	unsigned int coef;
+
+	coef = 0x0002; /* SRC_MUTE soft-mute on SPDIF (if no lock) */
+	coef |= 0x0008; /* Replace with mute on error */
+	if (is_active_pin(codec, CS_DIG_OUT2_PIN_NID))
+		coef |= 0x4000; /* RX to TX1 or TX2 Loopthru / SPDIF2
+				 * SPDIF_OUT2 is shared with GPIO1 and
+				 * DMIC_SDA2.
+				 */
+	cs_vendor_coef_set(codec, IDX_SPDIF_CTL, coef);
 }
 
 static int cs_init(struct hda_codec *codec)
 {
 	struct cs_spec *spec = codec->spec;
 
+	snd_hda_sequence_write(codec, cs_coef_init_verbs);
 	init_output(codec);
 	init_input(codec);
+	init_digital(codec);
 	return 0;
 }
 
-- 
cgit v1.2.3


From 60e53882ace216cf58fc847a94d839f32b8eb7bf Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 6 Jul 2009 15:01:09 +0200
Subject: ALSA: hda - Fix double creation of SPDIF input controls

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_cirrus.c | 9 +++------
 1 file changed, 3 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 2187383cdfd..404c120991b 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -471,12 +471,9 @@ static int parse_digital_input(struct hda_codec *codec)
 	struct auto_pin_cfg *cfg = &spec->autocfg;
 	int idx;
 
-	if (!cfg->dig_in_pin)
-		return 0;
-	spec->dig_in = get_adc(codec, cfg->dig_in_pin, &idx);
-	if (!spec->dig_in)
-		return 0;
-	return snd_hda_create_spdif_in_ctls(codec, spec->dig_in);
+	if (cfg->dig_in_pin)
+		spec->dig_in = get_adc(codec, cfg->dig_in_pin, &idx);
+	return 0;
 }
 
 /*
-- 
cgit v1.2.3


From a6bae20559bb0371e89ebc46689e9cf4e7816813 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 6 Jul 2009 15:15:22 +0200
Subject: ALSA: hda - Add quirk for MacBook Pro 5,5 with CS4206

Add the default pin configs for MBP55.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_cirrus.c | 57 ++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 57 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 404c120991b..b1fd183d760 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -30,6 +30,7 @@
  */
 
 struct cs_spec {
+	int board_config;
 	struct auto_pin_cfg autocfg;
 	struct hda_multi_out multiout;
 	struct snd_kcontrol *vmaster_sw;
@@ -58,6 +59,13 @@ struct cs_spec {
 	unsigned int mic_detect:1;
 };
 
+/* available models */
+enum {
+	CS420X_MBP55,
+	CS420X_AUTO,
+	CS420X_MODELS
+};
+
 /* Vendor-specific processing widget */
 #define CS420X_VENDOR_NID	0x11
 #define CS_DIG_OUT1_PIN_NID	0x10
@@ -1038,6 +1046,49 @@ static int cs_parse_auto_config(struct hda_codec *codec)
 	return 0;
 }
 
+static const char *cs420x_models[CS420X_MODELS] = {
+	[CS420X_MBP55] = "mbp55",
+	[CS420X_AUTO] = "auto",
+};
+
+
+static struct snd_pci_quirk cs420x_cfg_tbl[] = {
+	SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55),
+	{} /* terminator */
+};
+
+struct cs_pincfg {
+	hda_nid_t nid;
+	u32 val;
+};
+
+static struct cs_pincfg mbp55_pincfgs[] = {
+	{ 0x09, 0x012b4030 },
+	{ 0x0a, 0x90100121 },
+	{ 0x0b, 0x90100120 },
+	{ 0x0c, 0x400000f0 },
+	{ 0x0d, 0x90a00110 },
+	{ 0x0e, 0x400000f0 },
+	{ 0x0f, 0x400000f0 },
+	{ 0x10, 0x014be040 },
+	{ 0x12, 0x400000f0 },
+	{ 0x15, 0x400000f0 },
+	{} /* terminator */
+};
+
+static struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = {
+	[CS420X_MBP55] = mbp55_pincfgs,
+};
+
+static void fix_pincfg(struct hda_codec *codec, int model)
+{
+	const struct cs_pincfg *cfg = cs_pincfgs[model];
+	if (!cfg)
+		return;
+	for (; cfg->nid; cfg++)
+		snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
+}
+
 
 static int patch_cs420x(struct hda_codec *codec)
 {
@@ -1049,6 +1100,12 @@ static int patch_cs420x(struct hda_codec *codec)
 		return -ENOMEM;
 	codec->spec = spec;
 
+	spec->board_config =
+		snd_hda_check_board_config(codec, CS420X_MODELS,
+					   cs420x_models, cs420x_cfg_tbl);
+	if (spec->board_config >= 0)
+		fix_pincfg(codec, spec->board_config);
+
 	err = cs_parse_auto_config(codec);
 	if (err < 0)
 		goto error;
-- 
cgit v1.2.3


From ed208255e700073c34b0ef8d2045ac030ac28971 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 Jul 2009 09:04:26 +0200
Subject: ALSA: hda - Add GPIO setup for MacBook pro 5,5 with CS420x

GPIO3 seems corresponding to EAPD that is required for the speaker
output.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_cirrus.c | 49 ++++++++++++++++++++++++++++++++------------
 1 file changed, 36 insertions(+), 13 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index b1fd183d760..12173448009 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -53,6 +53,10 @@ struct cs_spec {
 
 	struct hda_bind_ctls *capture_bind[2];
 
+	unsigned int gpio_mask;
+	unsigned int gpio_dir;
+	unsigned int gpio_data;
+
 	struct hda_pcm pcm_rec[2];	/* PCM information */
 
 	unsigned int hp_detect:1;
@@ -981,6 +985,16 @@ static int cs_init(struct hda_codec *codec)
 	struct cs_spec *spec = codec->spec;
 
 	snd_hda_sequence_write(codec, cs_coef_init_verbs);
+
+	if (spec->gpio_mask) {
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK,
+				    spec->gpio_mask);
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION,
+				    spec->gpio_dir);
+		snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+				    spec->gpio_data);
+	}
+
 	init_output(codec);
 	init_input(codec);
 	init_digital(codec);
@@ -1041,6 +1055,19 @@ static int cs_parse_auto_config(struct hda_codec *codec)
 	int err;
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL);
+	if (err < 0)
+		return err;
+
+	err = parse_output(codec);
+	if (err < 0)
+		return err;
+	err = parse_input(codec);
+	if (err < 0)
+		return err;
+	err = parse_digital_output(codec);
+	if (err < 0)
+		return err;
+	err = parse_digital_input(codec);
 	if (err < 0)
 		return err;
 	return 0;
@@ -1106,20 +1133,16 @@ static int patch_cs420x(struct hda_codec *codec)
 	if (spec->board_config >= 0)
 		fix_pincfg(codec, spec->board_config);
 
-	err = cs_parse_auto_config(codec);
-	if (err < 0)
-		goto error;
+	switch (spec->board_config) {
+	case CS420X_MBP55:
+		/* GPIO3 = EAPD? */
+		spec->gpio_mask = 0x08;
+		spec->gpio_dir = 0x08;
+		spec->gpio_data = 0x08;
+		break;
+	}
 
-	err = parse_output(codec);
-	if (err < 0)
-		goto error;
-	err = parse_input(codec);
-	if (err < 0)
-		goto error;
-	err = parse_digital_output(codec);
-	if (err < 0)
-		goto error;
-	err = parse_digital_input(codec);
+	err = cs_parse_auto_config(codec);
 	if (err < 0)
 		goto error;
 
-- 
cgit v1.2.3


From b4dabfc452a1011882983db9e464f8ee1e3c01e6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 7 Jul 2009 09:05:07 +0200
Subject: ALSA: hda - Fix the speaker volume control name

Increase the name string buffer size so that "Surround Speaker Playback
Volume" won't be truncated.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_cirrus.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 12173448009..54a7dc8345a 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -497,7 +497,7 @@ static const char *dir_sfx[2] = { "Playback", "Capture" };
 static int add_mute(struct hda_codec *codec, const char *name, int index,
 		    unsigned int pval, int dir, struct snd_kcontrol **kctlp)
 {
-	char tmp[32];
+	char tmp[44];
 	struct snd_kcontrol_new knew =
 		HDA_CODEC_MUTE_IDX(tmp, index, 0, 0, HDA_OUTPUT);
 	knew.private_value = pval;
-- 
cgit v1.2.3


From 277a57c710ff73af1213f5d4b42bf6458c1ebca3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 8 Jul 2009 12:42:08 +0200
Subject: ALSA: hda - Fix compile warnings in patch_cirrus.c

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_cirrus.c | 15 +++++----------
 1 file changed, 5 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 54a7dc8345a..57251d72b12 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -110,7 +110,7 @@ enum {
 /* 0x0015 - visibility reg */
 
 
-static int cs_vendor_coef_get(struct hda_codec *codec, unsigned int idx)
+static inline int cs_vendor_coef_get(struct hda_codec *codec, unsigned int idx)
 {
 	snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0,
 			    AC_VERB_SET_COEF_INDEX, idx);
@@ -118,8 +118,8 @@ static int cs_vendor_coef_get(struct hda_codec *codec, unsigned int idx)
 				  AC_VERB_GET_PROC_COEF, 0);
 }
 
-static void cs_vendor_coef_set(struct hda_codec *codec, unsigned int idx,
-			       unsigned int coef)
+static inline void cs_vendor_coef_set(struct hda_codec *codec, unsigned int idx,
+				      unsigned int coef)
 {
 	snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0,
 			    AC_VERB_SET_COEF_INDEX, idx);
@@ -367,7 +367,6 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin,
 
 static int is_active_pin(struct hda_codec *codec, hda_nid_t nid)
 {
-	struct cs_spec *spec = codec->spec;
 	unsigned int val;
 	val = snd_hda_codec_get_pincfg(codec, nid);
 	return (get_defcfg_connect(val) != AC_JACK_PORT_NONE);
@@ -377,7 +376,7 @@ static int parse_output(struct hda_codec *codec)
 {
 	struct cs_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
-	int i, err, extra_nids;
+	int i, extra_nids;
 	hda_nid_t dac;
 
 	for (i = 0; i < cfg->line_outs; i++) {
@@ -422,11 +421,10 @@ static int parse_input(struct hda_codec *codec)
 {
 	struct cs_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
-	int i, n, err;
+	int i;
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t pin = cfg->input_pins[i];
-		struct snd_kcontrol *kctl;
 		if (!pin)
 			continue;
 		spec->input_idx[spec->num_inputs] = i;
@@ -461,7 +459,6 @@ static int parse_digital_output(struct hda_codec *codec)
 	struct cs_spec *spec = codec->spec;
 	struct auto_pin_cfg *cfg = &spec->autocfg;
 	hda_nid_t nid;
-	int err;
 
 	if (!cfg->dig_outs)
 		return 0;
@@ -651,7 +648,6 @@ static int change_cur_input(struct hda_codec *codec, unsigned int idx,
 			    int force)
 {
 	struct cs_spec *spec = codec->spec;
-	struct auto_pin_cfg *cfg = &spec->autocfg;
 	
 	if (spec->cur_input == idx && !force)
 		return 0;
@@ -1003,7 +999,6 @@ static int cs_init(struct hda_codec *codec)
 
 static int cs_build_controls(struct hda_codec *codec)
 {
-	struct cs_spec *spec = codec->spec;
 	int err;
 
 	err = build_output(codec);
-- 
cgit v1.2.3


From b1e4422f96a9b4830ffd8119b05211abb830ef4a Mon Sep 17 00:00:00 2001
From: William Weston <weston@sysex.net>
Date: Wed, 8 Jul 2009 01:10:05 -0700
Subject: ALSA: hda - Add quirks for RTL888 & RV630/M76 based MSI GX710

Signed-off-by: William Weston <weston@sysex.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index de72b72f58f..0227b0c8a72 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8422,6 +8422,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1462, 0x3fc3, "MSI", ALC883_TARGA_DIG),
 	SND_PCI_QUIRK(0x1462, 0x3fcc, "MSI", ALC883_TARGA_DIG),
 	SND_PCI_QUIRK(0x1462, 0x3fdf, "MSI", ALC883_TARGA_DIG),
+	SND_PCI_QUIRK(0x1462, 0x42cd, "MSI", ALC883_TARGA_DIG),
 	SND_PCI_QUIRK(0x1462, 0x4314, "MSI", ALC883_TARGA_DIG),
 	SND_PCI_QUIRK(0x1462, 0x4319, "MSI", ALC883_TARGA_DIG),
 	SND_PCI_QUIRK(0x1462, 0x4324, "MSI", ALC883_TARGA_DIG),
@@ -8435,6 +8436,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1462, 0x7350, "MSI", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1462, 0xa422, "MSI", ALC883_TARGA_2ch_DIG),
+	SND_PCI_QUIRK(0x1462, 0xaa08, "MSI", ALC883_TARGA_2ch_DIG),
 
 	SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720),
-- 
cgit v1.2.3


From 63b2413b2ff446e30286c2bd775fd10b12d188c6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 9 Jul 2009 11:45:59 +0200
Subject: ALSA: hda - don't build digital output controls if not exist

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_cirrus.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 57251d72b12..f5527380a24 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -779,6 +779,9 @@ static int build_digital_output(struct hda_codec *codec)
 	struct cs_spec *spec = codec->spec;
 	int err;
 
+	if (!spec->multiout.dig_out_nid)
+		return 0;
+
 	err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
 	if (err < 0)
 		return err;
-- 
cgit v1.2.3


From 31909b83ea5b4035ce129bb0df622ec2d2fb3e96 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 10 Jul 2009 12:33:48 +0200
Subject: ALSA: hda - Fix the merge error

Fix the merge error at the commit 305355aad89f1b7eb27cb210fad2f9d3c67b2572,
an addition of the missing alc880_gpio3_init_verbs to ALC882_TARGA model.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 88a9da4afe5..7967f8f1a34 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -8587,7 +8587,7 @@ static struct alc_config_preset alc882_presets[] = {
 	[ALC882_TARGA] = {
 		.mixers = { alc882_targa_mixer, alc882_chmode_mixer },
 		.init_verbs = { alc882_base_init_verbs, alc882_adc1_init_verbs,
-				alc882_targa_verbs},
+				alc880_gpio3_init_verbs, alc882_targa_verbs},
 		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
 		.dac_nids = alc882_dac_nids,
 		.dig_out_nid = ALC882_DIGOUT_NID,
-- 
cgit v1.2.3


From 3c6aae4489e7c05f3685e9eaa538249be7c79ebe Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 10 Jul 2009 12:52:27 +0200
Subject: ALSA: hda - Check codec errors in snd_hda_get_connections()

The codec read errors in snd_hda_get_connections() are ignored so far,
and it causes a problem like the bug in the commit
    9d30937accf2c01e8b0bd59787409a7348cbbcb7
    ALSA: hda_intel: more strict alc880_parse_auto_config dig_nid checking

Better to check errors in the function and returns a proper error code
rather than passing bogus NID values.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 7 ++++++-
 1 file changed, 6 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 94d848e9871..ec352c6ae49 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -316,6 +316,8 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
 		/* single connection */
 		parm = snd_hda_codec_read(codec, nid, 0,
 					  AC_VERB_GET_CONNECT_LIST, 0);
+		if (parm == -1 && codec->bus->rirb_error)
+			return -EIO;
 		conn_list[0] = parm & mask;
 		return 1;
 	}
@@ -327,9 +329,12 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
 		int range_val;
 		hda_nid_t val, n;
 
-		if (i % num_elems == 0)
+		if (i % num_elems == 0) {
 			parm = snd_hda_codec_read(codec, nid, 0,
 						  AC_VERB_GET_CONNECT_LIST, i);
+			if (parm == -1 && codec->bus->rirb_error)
+				return -EIO;
+		}
 		range_val = !!(parm & (1 << (shift-1))); /* ranges */
 		val = parm & mask;
 		parm >>= shift;
-- 
cgit v1.2.3


From 0fb67e982a1749b2cd477709838bf866028970ce Mon Sep 17 00:00:00 2001
From: Daniel Drake <dsd@laptop.org>
Date: Thu, 16 Jul 2009 14:46:57 +0100
Subject: ALSA: hda - Add CX20582 and OLPC XO-1.5 support

This adds support for the Conexant CX20582 codec, based on code from
http://www.linuxant.com/alsa-driver/alsa-driver-linuxant-1.0.19ppch12-1.noarch.rpm.zip

This is the codec to be shipped in the OLPC XO-1.5, so this patch also
includes an XO-specific profile. Resultant configuration:
http://dev.laptop.org/~dsd/20090713/codec0.txt
http://dev.laptop.org/~dsd/20090713/codec0.svg

As the Linuxant code is structured differently to the other codecs,
I was unable to cleanly reimplement everything in the generic and Dell
profiles as more info is needed (e.g. codec graphs). I simplified those
profiles so that hopefully it will not break anyone's audio. If it does,
it may be worth returning -ENODEV from patch_cx5066 on non-OLPC systems,
and then fixing snd_hda_codec_configure() to fall back on the generic
parser, at least until support for other systems is figured out.

Signed-off-by: Daniel Drake <dsd@laptop.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_conexant.c | 479 +++++++++++++++++++++++++++++++++++++++++
 1 file changed, 479 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index ac868c59f9e..9d899eda44d 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -108,6 +108,8 @@ struct conexant_spec {
 	struct hda_input_mux private_imux;
 	hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
 
+	unsigned int dell_automute;
+	unsigned int port_d_mode;
 };
 
 static int conexant_playback_pcm_open(struct hda_pcm_stream *hinfo,
@@ -1908,6 +1910,480 @@ static int patch_cxt5051(struct hda_codec *codec)
 	return 0;
 }
 
+/* Conexant 5066 specific */
+
+static hda_nid_t cxt5066_dac_nids[1] = { 0x10 };
+static hda_nid_t cxt5066_adc_nids[3] = { 0x14, 0x15, 0x16 };
+static hda_nid_t cxt5066_capsrc_nids[1] = { 0x17 };
+#define CXT5066_SPDIF_OUT	0x21
+
+static struct hda_channel_mode cxt5066_modes[1] = {
+	{ 2, NULL },
+};
+
+static void cxt5066_update_speaker(struct hda_codec *codec)
+{
+	struct conexant_spec *spec = codec->spec;
+	unsigned int pinctl;
+
+	snd_printdd("CXT5066: update speaker, hp_present=%d\n",
+		spec->hp_present);
+
+	/* Port A (HP) */
+	pinctl = ((spec->hp_present & 1) && spec->cur_eapd) ? PIN_HP : 0;
+	snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+			pinctl);
+
+	/* Port D (HP/LO) */
+	pinctl = ((spec->hp_present & 2) && spec->cur_eapd)
+		? spec->port_d_mode : 0;
+	snd_hda_codec_write(codec, 0x1c, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+			pinctl);
+
+	/* CLASS_D AMP */
+	pinctl = (!spec->hp_present && spec->cur_eapd) ? PIN_OUT : 0;
+	snd_hda_codec_write(codec, 0x1f, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+			pinctl);
+
+	if (spec->dell_automute) {
+		/* DELL AIO Port Rule: PortA > PortD > IntSpk */
+		pinctl = (!(spec->hp_present & 1) && spec->cur_eapd)
+			? PIN_OUT : 0;
+		snd_hda_codec_write(codec, 0x1c, 0,
+			AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl);
+	}
+}
+
+/* turn on/off EAPD (+ mute HP) as a master switch */
+static int cxt5066_hp_master_sw_put(struct snd_kcontrol *kcontrol,
+				    struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	if (!cxt_eapd_put(kcontrol, ucontrol))
+		return 0;
+
+	cxt5066_update_speaker(codec);
+	return 1;
+}
+
+/* toggle input of built-in and mic jack appropriately */
+static void cxt5066_automic(struct hda_codec *codec)
+{
+	static struct hda_verb ext_mic_present[] = {
+		/* enable external mic, port B */
+		{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+
+		/* switch to external mic input */
+		{0x17, AC_VERB_SET_CONNECT_SEL, 0},
+
+		/* disable internal mic, port C */
+		{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+		{}
+	};
+	static struct hda_verb ext_mic_absent[] = {
+		/* enable internal mic, port C */
+		{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+
+		/* switch to internal mic input */
+		{0x17, AC_VERB_SET_CONNECT_SEL, 1},
+
+		/* disable external mic, port B */
+		{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+		{}
+	};
+	unsigned int present;
+
+	present = snd_hda_codec_read(codec, 0x1a, 0,
+				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	if (present) {
+		snd_printdd("CXT5066: external microphone detected\n");
+		snd_hda_sequence_write(codec, ext_mic_present);
+	} else {
+		snd_printdd("CXT5066: external microphone absent\n");
+		snd_hda_sequence_write(codec, ext_mic_absent);
+	}
+}
+
+/* mute internal speaker if HP is plugged */
+static void cxt5066_hp_automute(struct hda_codec *codec)
+{
+	struct conexant_spec *spec = codec->spec;
+	unsigned int portA, portD;
+
+	/* Port A */
+	portA = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0)
+		& AC_PINSENSE_PRESENCE;
+
+	/* Port D */
+	portD = (snd_hda_codec_read(codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0)
+		& AC_PINSENSE_PRESENCE) << 1;
+
+	spec->hp_present = !!(portA | portD);
+	snd_printdd("CXT5066: hp automute portA=%x portD=%x present=%d\n",
+		portA, portD, spec->hp_present);
+	cxt5066_update_speaker(codec);
+}
+
+/* unsolicited event for jack sensing */
+static void cxt5066_unsol_event(struct hda_codec *codec, unsigned int res)
+{
+	snd_printdd("CXT5066: unsol event %x (%x)\n", res, res >> 26);
+	switch (res >> 26) {
+	case CONEXANT_HP_EVENT:
+		cxt5066_hp_automute(codec);
+		break;
+	case CONEXANT_MIC_EVENT:
+		cxt5066_automic(codec);
+		break;
+	}
+}
+
+static const struct hda_input_mux cxt5066_analog_mic_boost = {
+	.num_items = 5,
+	.items = {
+		{ "0dB",  0 },
+		{ "10dB", 1 },
+		{ "20dB", 2 },
+		{ "30dB", 3 },
+		{ "40dB", 4 },
+	},
+};
+
+static int cxt5066_mic_boost_mux_enum_info(struct snd_kcontrol *kcontrol,
+					   struct snd_ctl_elem_info *uinfo)
+{
+	return snd_hda_input_mux_info(&cxt5066_analog_mic_boost, uinfo);
+}
+
+static int cxt5066_mic_boost_mux_enum_get(struct snd_kcontrol *kcontrol,
+					  struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	int val;
+
+	val = snd_hda_codec_read(codec, 0x17, 0,
+		AC_VERB_GET_AMP_GAIN_MUTE, AC_AMP_GET_OUTPUT);
+
+	ucontrol->value.enumerated.item[0] = val & AC_AMP_GAIN;
+	return 0;
+}
+
+static int cxt5066_mic_boost_mux_enum_put(struct snd_kcontrol *kcontrol,
+					  struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	const struct hda_input_mux *imux = &cxt5066_analog_mic_boost;
+	unsigned int idx;
+
+	if (!imux->num_items)
+		return 0;
+	idx = ucontrol->value.enumerated.item[0];
+	if (idx >= imux->num_items)
+		idx = imux->num_items - 1;
+
+	snd_hda_codec_write_cache(codec, 0x17, 0,
+		AC_VERB_SET_AMP_GAIN_MUTE,
+		AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | AC_AMP_SET_OUTPUT |
+			imux->items[idx].index);
+
+	return 1;
+}
+
+static struct hda_input_mux cxt5066_capture_source = {
+	.num_items = 4,
+	.items = {
+		{ "Mic B", 0 },
+		{ "Mic C", 1 },
+		{ "Mic E", 2 },
+		{ "Mic F", 3 },
+	},
+};
+
+static struct hda_bind_ctls cxt5066_bind_capture_vol_others = {
+	.ops = &snd_hda_bind_vol,
+	.values = {
+		HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT),
+		HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT),
+		0
+	},
+};
+
+static struct hda_bind_ctls cxt5066_bind_capture_sw_others = {
+	.ops = &snd_hda_bind_sw,
+	.values = {
+		HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_INPUT),
+		HDA_COMPOSE_AMP_VAL(0x14, 3, 2, HDA_INPUT),
+		0
+	},
+};
+
+static struct snd_kcontrol_new cxt5066_mixer_master[] = {
+	HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT),
+	{}
+};
+
+static struct snd_kcontrol_new cxt5066_mixer_master_olpc[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Master Playback Volume",
+		.access = SNDRV_CTL_ELEM_ACCESS_READWRITE |
+				  SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+				  SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK,
+		.info = snd_hda_mixer_amp_volume_info,
+		.get = snd_hda_mixer_amp_volume_get,
+		.put = snd_hda_mixer_amp_volume_put,
+		.tlv = { .c = snd_hda_mixer_amp_tlv },
+		/* offset by 28 volume steps to limit minimum gain to -46dB */
+		.private_value =
+			HDA_COMPOSE_AMP_VAL_OFS(0x10, 3, 0, HDA_OUTPUT, 28),
+	},
+	{}
+};
+
+static struct snd_kcontrol_new cxt5066_mixers[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Master Playback Switch",
+		.info = cxt_eapd_info,
+		.get = cxt_eapd_get,
+		.put = cxt5066_hp_master_sw_put,
+		.private_value = 0x1d,
+	},
+
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Analog Mic Boost Capture Enum",
+		.info = cxt5066_mic_boost_mux_enum_info,
+		.get = cxt5066_mic_boost_mux_enum_get,
+		.put = cxt5066_mic_boost_mux_enum_put,
+	},
+
+	HDA_BIND_VOL("Capture Volume", &cxt5066_bind_capture_vol_others),
+	HDA_BIND_SW("Capture Switch", &cxt5066_bind_capture_sw_others),
+	{}
+};
+
+static struct hda_verb cxt5066_init_verbs[] = {
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port B */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Port C */
+	{0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port F */
+	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, /* Port E */
+
+	/* Speakers  */
+	{0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
+
+	/* HP, Amp  */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
+
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x1c, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
+
+	/* DAC1 */
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	/* Node 14 connections: 0x17 0x18 0x23 0x24 0x27 */
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2) | 0x50},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+
+	/* no digital microphone support yet */
+	{0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+
+	/* Audio input selector */
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3},
+
+	/* SPDIF route: PCM */
+	{0x20, AC_VERB_SET_CONNECT_SEL, 0x0},
+	{0x22, AC_VERB_SET_CONNECT_SEL, 0x0},
+
+	{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+
+	/* EAPD */
+	{0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
+
+	/* not handling these yet */
+	{0x19, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
+	{0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
+	{0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
+	{0x1d, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
+	{0x1e, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
+	{0x20, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
+	{0x22, AC_VERB_SET_UNSOLICITED_ENABLE, 0},
+	{ } /* end */
+};
+
+static struct hda_verb cxt5066_init_verbs_olpc[] = {
+	/* Port A: headphones */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
+
+	/* Port B: external microphone */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+
+	/* Port C: internal microphone */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+
+	/* Port D: unused */
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+
+	/* Port E: unused, but has primary EAPD */
+	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x1d, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
+
+	/* Port F: unused */
+	{0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+
+	/* Port G: internal speakers */
+	{0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1f, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC1 */
+
+	/* DAC1 */
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+
+	/* DAC2: unused */
+	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x50},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+
+	/* Disable digital microphone port */
+	{0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+
+	/* Audio input selectors */
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x3},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+
+	/* Disable SPDIF */
+	{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+
+	/* enable unsolicited events for Port A and B */
+	{0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
+	{0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
+	{ } /* end */
+};
+
+static struct hda_verb cxt5066_init_verbs_portd_lo[] = {
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{ } /* end */
+};
+
+/* initialize jack-sensing, too */
+static int cxt5066_init(struct hda_codec *codec)
+{
+	snd_printdd("CXT5066: init\n");
+	conexant_init(codec);
+	if (codec->patch_ops.unsol_event) {
+		cxt5066_hp_automute(codec);
+		cxt5066_automic(codec);
+	}
+	return 0;
+}
+
+enum {
+	CXT5066_LAPTOP,			/* Laptops w/ EAPD support */
+	CXT5066_DELL_LAPTOP,	/* Dell Laptop */
+	CXT5066_OLPC_XO_1_5,	/* OLPC XO 1.5 */
+	CXT5066_MODELS
+};
+
+static const char *cxt5066_models[CXT5066_MODELS] = {
+	[CXT5066_LAPTOP]		= "laptop",
+	[CXT5066_DELL_LAPTOP]	= "dell-laptop",
+	[CXT5066_OLPC_XO_1_5]	= "olpc-xo-1_5",
+};
+
+static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
+	SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board",
+		      CXT5066_LAPTOP),
+	SND_PCI_QUIRK(0x1028, 0x02f5, "Dell",
+		      CXT5066_DELL_LAPTOP),
+	{}
+};
+
+static int patch_cxt5066(struct hda_codec *codec)
+{
+	struct conexant_spec *spec;
+	int board_config;
+
+	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
+	if (!spec)
+		return -ENOMEM;
+	codec->spec = spec;
+
+	codec->patch_ops = conexant_patch_ops;
+	codec->patch_ops.init = cxt5066_init;
+
+	spec->dell_automute = 0;
+	spec->multiout.max_channels = 2;
+	spec->multiout.num_dacs = ARRAY_SIZE(cxt5066_dac_nids);
+	spec->multiout.dac_nids = cxt5066_dac_nids;
+	spec->multiout.dig_out_nid = CXT5066_SPDIF_OUT;
+	spec->num_adc_nids = 1;
+	spec->adc_nids = cxt5066_adc_nids;
+	spec->capsrc_nids = cxt5066_capsrc_nids;
+	spec->input_mux = &cxt5066_capture_source;
+
+	spec->port_d_mode = PIN_HP;
+
+	spec->num_init_verbs = 1;
+	spec->init_verbs[0] = cxt5066_init_verbs;
+	spec->num_channel_mode = ARRAY_SIZE(cxt5066_modes);
+	spec->channel_mode = cxt5066_modes;
+	spec->cur_adc = 0;
+	spec->cur_adc_idx = 0;
+
+	board_config = snd_hda_check_board_config(codec, CXT5066_MODELS,
+						  cxt5066_models, cxt5066_cfg_tbl);
+	switch (board_config) {
+	default:
+	case CXT5066_LAPTOP:
+		spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
+		spec->mixers[spec->num_mixers++] = cxt5066_mixers;
+		break;
+	case CXT5066_DELL_LAPTOP:
+		spec->mixers[spec->num_mixers++] = cxt5066_mixer_master;
+		spec->mixers[spec->num_mixers++] = cxt5066_mixers;
+
+		spec->port_d_mode = PIN_OUT;
+		spec->init_verbs[spec->num_init_verbs] = cxt5066_init_verbs_portd_lo;
+		spec->num_init_verbs++;
+		spec->dell_automute = 1;
+		break;
+	case CXT5066_OLPC_XO_1_5:
+		codec->patch_ops.unsol_event = cxt5066_unsol_event;
+		spec->init_verbs[0] = cxt5066_init_verbs_olpc;
+		spec->mixers[spec->num_mixers++] = cxt5066_mixer_master_olpc;
+		spec->mixers[spec->num_mixers++] = cxt5066_mixers;
+		spec->port_d_mode = 0;
+
+		/* no S/PDIF out */
+		spec->multiout.dig_out_nid = 0;
+
+		/* input source automatically selected */
+		spec->input_mux = NULL;
+		break;
+	}
+
+	return 0;
+}
 
 /*
  */
@@ -1919,12 +2395,15 @@ static struct hda_codec_preset snd_hda_preset_conexant[] = {
 	  .patch = patch_cxt5047 },
 	{ .id = 0x14f15051, .name = "CX20561 (Hermosa)",
 	  .patch = patch_cxt5051 },
+	{ .id = 0x14f15066, .name = "CX20582 (Pebble)",
+	  .patch = patch_cxt5066 },
 	{} /* terminator */
 };
 
 MODULE_ALIAS("snd-hda-codec-id:14f15045");
 MODULE_ALIAS("snd-hda-codec-id:14f15047");
 MODULE_ALIAS("snd-hda-codec-id:14f15051");
+MODULE_ALIAS("snd-hda-codec-id:14f15066");
 
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("Conexant HD-audio codec");
-- 
cgit v1.2.3


From 9176b672c29baaa94cdff4eedf1350a3b553d9ea Mon Sep 17 00:00:00 2001
From: Andiry Brienza <andiry.xu@amd.com>
Date: Fri, 17 Jul 2009 11:32:32 +0800
Subject: ALSA: hda - Add support for new AMD HD audio devices

Add support for new AMD HD audio devices. Use generic driver to detect HD audio
devices with Vendor ID AMD.

Signed-off-by: Andiry Xu <andiry.xu@amd.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 6 +++++-
 1 file changed, 5 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index a7cf3496522..4db854b43e6 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2634,11 +2634,15 @@ static struct pci_device_id azx_ids[] = {
 	/* this entry seems still valid -- i.e. without emu20kx chip */
 	{ PCI_DEVICE(0x1102, 0x0009), .driver_data = AZX_DRIVER_GENERIC },
 #endif
-	/* AMD Generic, PCI class code and Vendor ID for HD Audio */
+	/* AMD/ATI Generic, PCI class code and Vendor ID for HD Audio */
 	{ PCI_DEVICE(PCI_VENDOR_ID_ATI, PCI_ANY_ID),
 	  .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
 	  .class_mask = 0xffffff,
 	  .driver_data = AZX_DRIVER_GENERIC },
+	{ PCI_DEVICE(PCI_VENDOR_ID_AMD, PCI_ANY_ID),
+	  .class = PCI_CLASS_MULTIMEDIA_HD_AUDIO << 8,
+	  .class_mask = 0xffffff,
+	  .driver_data = AZX_DRIVER_GENERIC },
 	{ 0, }
 };
 MODULE_DEVICE_TABLE(pci, azx_ids);
-- 
cgit v1.2.3


From 6430aeeb30b478d4ef25f988b1fde6f6ae83adb5 Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Fri, 17 Jul 2009 16:49:19 +0800
Subject: ALSA: hda - add bounds checking for the codec command fields

A recent bug involves passing auto detected >0x7f NID to codec command,
creating an invalid codec addr field, and finally lead to cmd timeout
and fall back into single command mode. Jaroslav fixed that bug in
alc880_parse_auto_config().

It would be safer to further check the bounds of all cmd fields.

Cc: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 12 +++++++++++-
 1 file changed, 11 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index ec352c6ae49..d9d32629783 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -150,7 +150,14 @@ make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct,
 {
 	u32 val;
 
-	val = (u32)(codec->addr & 0x0f) << 28;
+	if ((codec->addr & ~0xf) | (direct & ~1) | (nid & ~0x7f) |
+	    (verb & ~0xfff) | (parm & ~0xff)) {
+		printk(KERN_ERR "hda-codec: out of range cmd %x:%x:%x:%x:%x\n",
+		       codec->addr, direct, nid, verb, parm);
+		return ~0;
+	}
+
+	val = (u32)codec->addr << 28;
 	val |= (u32)direct << 27;
 	val |= (u32)nid << 20;
 	val |= verb << 8;
@@ -167,6 +174,9 @@ static int codec_exec_verb(struct hda_codec *codec, unsigned int cmd,
 	struct hda_bus *bus = codec->bus;
 	int err;
 
+	if (cmd == ~0)
+		return -1;
+
 	if (res)
 		*res = -1;
  again:
-- 
cgit v1.2.3


From 82e1b804b03defe46fb69ffd2c8b19e6649bcb0d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 17 Jul 2009 12:47:34 +0200
Subject: ALSA: hda - Fix the previous sanity check in make_codec_cmd()

The newly added sanity-check for a codec verb can be better written
with logical ORs.  Also, the parameter can be more than 8bit.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index d9d32629783..35f0f223f85 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -150,8 +150,8 @@ make_codec_cmd(struct hda_codec *codec, hda_nid_t nid, int direct,
 {
 	u32 val;
 
-	if ((codec->addr & ~0xf) | (direct & ~1) | (nid & ~0x7f) |
-	    (verb & ~0xfff) | (parm & ~0xff)) {
+	if ((codec->addr & ~0xf) || (direct & ~1) || (nid & ~0x7f) ||
+	    (verb & ~0xfff) || (parm & ~0xffff)) {
 		printk(KERN_ERR "hda-codec: out of range cmd %x:%x:%x:%x:%x\n",
 		       codec->addr, direct, nid, verb, parm);
 		return ~0;
-- 
cgit v1.2.3


From 3f3b7c1aed70fa25c6811f830c5fb1a7054681ae Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 17 Jul 2009 14:36:59 +0200
Subject: ALSA: hda - Fix ALC268 parser for mono speaker

- Parse the mono output pin 0x16 correctly even as the primary output
- Create "Speaker" volume control if the primary output is a speaker
- Fix the wrong direction of (optional) "Mono" switch

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 62 ++++++++++++++++++++++++++++++-------------
 1 file changed, 43 insertions(+), 19 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7967f8f1a34..507a7720013 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -11816,26 +11816,38 @@ static int alc268_new_analog_output(struct alc_spec *spec, hda_nid_t nid,
 				    const char *ctlname, int idx)
 {
 	char name[32];
+	hda_nid_t dac;
 	int err;
 
 	sprintf(name, "%s Playback Volume", ctlname);
-	if (nid == 0x14) {
-		err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
-				  HDA_COMPOSE_AMP_VAL(0x02, 3, idx,
-						      HDA_OUTPUT));
-		if (err < 0)
-			return err;
-	} else if (nid == 0x15) {
+	switch (nid) {
+	case 0x14:
+	case 0x16:
+		dac = 0x02;
+		break;
+	case 0x15:
+		dac = 0x03;
+		break;
+	default:
+		return 0;
+	}
+	if (spec->multiout.dac_nids[0] != dac &&
+	    spec->multiout.dac_nids[1] != dac) {
 		err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
-				  HDA_COMPOSE_AMP_VAL(0x03, 3, idx,
+				  HDA_COMPOSE_AMP_VAL(dac, 3, idx,
 						      HDA_OUTPUT));
 		if (err < 0)
 			return err;
-	} else
-		return -1;
+		spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac;
+	}
+
 	sprintf(name, "%s Playback Switch", ctlname);
-	err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+	if (nid != 0x16)
+		err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
 			  HDA_COMPOSE_AMP_VAL(nid, 3, idx, HDA_OUTPUT));
+	else /* mono */
+		err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
+			  HDA_COMPOSE_AMP_VAL(nid, 2, idx, HDA_OUTPUT));
 	if (err < 0)
 		return err;
 	return 0;
@@ -11848,14 +11860,19 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec,
 	hda_nid_t nid;
 	int err;
 
-	spec->multiout.num_dacs = 2;	/* only use one dac */
 	spec->multiout.dac_nids = spec->private_dac_nids;
-	spec->multiout.dac_nids[0] = 2;
-	spec->multiout.dac_nids[1] = 3;
 
 	nid = cfg->line_out_pins[0];
-	if (nid)
-		alc268_new_analog_output(spec, nid, "Front", 0);
+	if (nid) {
+		const char *name;
+		if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+			name = "Speaker";
+		else
+			name = "Front";
+		err = alc268_new_analog_output(spec, nid, name, 0);
+		if (err < 0)
+			return err;
+	}
 
 	nid = cfg->speaker_pins[0];
 	if (nid == 0x1d) {
@@ -11864,16 +11881,23 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec,
 				  HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
 		if (err < 0)
 			return err;
+	} else {
+		err = alc268_new_analog_output(spec, nid, "Speaker", 0);
+		if (err < 0)
+			return err;
 	}
 	nid = cfg->hp_pins[0];
-	if (nid)
-		alc268_new_analog_output(spec, nid, "Headphone", 0);
+	if (nid) {
+		err = alc268_new_analog_output(spec, nid, "Headphone", 0);
+		if (err < 0)
+			return err;
+	}
 
 	nid = cfg->line_out_pins[1] | cfg->line_out_pins[2];
 	if (nid == 0x16) {
 		err = add_control(spec, ALC_CTL_WIDGET_MUTE,
 				  "Mono Playback Switch",
-				  HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_INPUT));
+				  HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT));
 		if (err < 0)
 			return err;
 	}
-- 
cgit v1.2.3


From 05ff7e11b78f18ff6819d2c260b7bcc7da0c8f46 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 22 Jul 2009 12:39:24 +0200
Subject: ALSA: hda - Reduce click noise at power-saving

Add some tricks to reduce the click noise at powering down to D3
in the power saving mode on STAC/IDT codecs.
The key seems to be to reset PINs before the power-down, and some
delay before entering D3.  The needed delay is significantly long,
but I don't know why.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c      |  9 +++++++--
 sound/pci/hda/patch_sigmatel.c | 14 ++++++++++++++
 2 files changed, 21 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index d686f4fcd78..3ecb45ddadf 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -2367,9 +2367,14 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 	hda_nid_t nid;
 	int i;
 
-	snd_hda_codec_write(codec, fg, 0, AC_VERB_SET_POWER_STATE,
+	/* this delay seems necessary to avoid click noise at power-down */
+	if (power_state == AC_PWRST_D3)
+		msleep(100);
+	snd_hda_codec_read(codec, fg, 0, AC_VERB_SET_POWER_STATE,
 			    power_state);
-	msleep(10); /* partial workaround for "azx_get_response timeout" */
+	/* partial workaround for "azx_get_response timeout" */
+	if (power_state == AC_PWRST_D0)
+		msleep(10);
 
 	nid = codec->start_nid;
 	for (i = 0; i < codec->num_nodes; i++, nid++) {
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 41b5b3a18c1..00c702df721 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4745,6 +4745,20 @@ static int stac92xx_hp_check_power_status(struct hda_codec *codec,
 static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
 {
 	struct sigmatel_spec *spec = codec->spec;
+	int i;
+	hda_nid_t nid;
+
+	/* reset each pin before powering down DAC/ADC to avoid click noise */
+	nid = codec->start_nid;
+	for (i = 0; i < codec->num_nodes; i++, nid++) {
+		unsigned int wcaps = get_wcaps(codec, nid);
+		unsigned int wid_type = (wcaps & AC_WCAP_TYPE) >>
+			AC_WCAP_TYPE_SHIFT;
+		if (wid_type == AC_WID_PIN)
+			snd_hda_codec_read(codec, nid, 0,
+				AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
+	}
+
 	if (spec->eapd_mask)
 		stac_gpio_set(codec, spec->gpio_mask,
 				spec->gpio_dir, spec->gpio_data &
-- 
cgit v1.2.3


From 1c20930a41c623a2281e754f9321cd0feb01ceb9 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 22 Jul 2009 15:17:45 +0200
Subject: ALSA: hda - Fix ALC861 auto-mode parser

Fix the logic of ALC861 auto-mode parser for the outputs.
Instead of assuming the fixed DAC list, parse the conection and assign
the DAC dynamically.

Also, unmute the unused output connections to avoid noises on inputs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 169 +++++++++++++++++++++++++++---------------
 1 file changed, 111 insertions(+), 58 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 454df58e983..b2ea43206f1 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -13685,23 +13685,23 @@ static struct hda_verb alc861_auto_init_verbs[] = {
 	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
 	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb00c},
 
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
 
 	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(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(2)},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
 	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
 
 	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},	/* set Mic 1 */
 
@@ -13773,64 +13773,97 @@ static struct hda_input_mux alc861_capture_source = {
 	},
 };
 
+static hda_nid_t alc861_look_for_dac(struct hda_codec *codec, hda_nid_t pin)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t mix, srcs[5];
+	int i, j, num;
+
+	if (snd_hda_get_connections(codec, pin, &mix, 1) != 1)
+		return 0;
+	num = snd_hda_get_connections(codec, mix, srcs, ARRAY_SIZE(srcs));
+	if (num < 0)
+		return 0;
+	for (i = 0; i < num; i++) {
+		unsigned int type;
+		type = (get_wcaps(codec, srcs[i]) & AC_WCAP_TYPE)
+			>> AC_WCAP_TYPE_SHIFT;
+		if (type != AC_WID_AUD_OUT)
+			continue;
+		for (j = 0; j < spec->multiout.num_dacs; j++)
+			if (spec->multiout.dac_nids[j] == srcs[i])
+				break;
+		if (j >= spec->multiout.num_dacs)
+			return srcs[i];
+	}
+	return 0;
+}
+
 /* fill in the dac_nids table from the parsed pin configuration */
-static int alc861_auto_fill_dac_nids(struct alc_spec *spec,
+static int alc861_auto_fill_dac_nids(struct hda_codec *codec,
 				     const struct auto_pin_cfg *cfg)
 {
+	struct alc_spec *spec = codec->spec;
 	int i;
-	hda_nid_t nid;
+	hda_nid_t nid, dac;
 
 	spec->multiout.dac_nids = spec->private_dac_nids;
 	for (i = 0; i < cfg->line_outs; i++) {
 		nid = cfg->line_out_pins[i];
-		if (nid) {
-			if (i >= ARRAY_SIZE(alc861_dac_nids))
-				continue;
-			spec->multiout.dac_nids[i] = alc861_dac_nids[i];
-		}
+		dac = alc861_look_for_dac(codec, nid);
+		if (!dac)
+			continue;
+		spec->multiout.dac_nids[spec->multiout.num_dacs++] = dac;
 	}
-	spec->multiout.num_dacs = cfg->line_outs;
 	return 0;
 }
 
+static int alc861_create_out_sw(struct hda_codec *codec, const char *pfx,
+				hda_nid_t nid, unsigned int chs)
+{
+	char name[32];
+	snprintf(name, sizeof(name), "%s Playback Switch", pfx);
+	return add_control(codec->spec, ALC_CTL_WIDGET_MUTE, name,
+			   HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
+}
+
 /* add playback controls from the parsed DAC table */
-static int alc861_auto_create_multi_out_ctls(struct alc_spec *spec,
+static int alc861_auto_create_multi_out_ctls(struct hda_codec *codec,
 					     const struct auto_pin_cfg *cfg)
 {
-	char name[32];
+	struct alc_spec *spec = codec->spec;
 	static const char *chname[4] = {
 		"Front", "Surround", NULL /*CLFE*/, "Side"
 	};
 	hda_nid_t nid;
-	int i, idx, err;
+	int i, err;
+
+	if (cfg->line_outs == 1) {
+		const char *pfx = NULL;
+		if (!cfg->hp_outs)
+			pfx = "Master";
+		else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+			pfx = "Speaker";
+		if (pfx) {
+			nid = spec->multiout.dac_nids[0];
+			return alc861_create_out_sw(codec, pfx, nid, 3);
+		}
+	}
 
 	for (i = 0; i < cfg->line_outs; i++) {
 		nid = spec->multiout.dac_nids[i];
 		if (!nid)
 			continue;
-		if (nid == 0x05) {
+		if (i == 2) {
 			/* Center/LFE */
-			err = add_control(spec, ALC_CTL_BIND_MUTE,
-					  "Center Playback Switch",
-					  HDA_COMPOSE_AMP_VAL(nid, 1, 0,
-							      HDA_OUTPUT));
+			err = alc861_create_out_sw(codec, "Center", nid, 1);
 			if (err < 0)
 				return err;
-			err = add_control(spec, ALC_CTL_BIND_MUTE,
-					  "LFE Playback Switch",
-					  HDA_COMPOSE_AMP_VAL(nid, 2, 0,
-							      HDA_OUTPUT));
+			err = alc861_create_out_sw(codec, "LFE", nid, 2);
 			if (err < 0)
 				return err;
 		} else {
-			for (idx = 0; idx < ARRAY_SIZE(alc861_dac_nids) - 1;
-			     idx++)
-				if (nid == alc861_dac_nids[idx])
-					break;
-			sprintf(name, "%s Playback Switch", chname[idx]);
-			err = add_control(spec, ALC_CTL_BIND_MUTE, name,
-					  HDA_COMPOSE_AMP_VAL(nid, 3, 0,
-							      HDA_OUTPUT));
+			err = alc861_create_out_sw(codec, chname[i], nid, 3);
 			if (err < 0)
 				return err;
 		}
@@ -13838,8 +13871,9 @@ static int alc861_auto_create_multi_out_ctls(struct alc_spec *spec,
 	return 0;
 }
 
-static int alc861_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin)
+static int alc861_auto_create_hp_ctls(struct hda_codec *codec, hda_nid_t pin)
 {
+	struct alc_spec *spec = codec->spec;
 	int err;
 	hda_nid_t nid;
 
@@ -13847,21 +13881,22 @@ static int alc861_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin)
 		return 0;
 
 	if ((pin >= 0x0b && pin <= 0x10) || pin == 0x1f || pin == 0x20) {
-		nid = 0x03;
-		err = add_control(spec, ALC_CTL_WIDGET_MUTE,
-				  "Headphone Playback Switch",
-				  HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
-		if (err < 0)
-			return err;
-		spec->multiout.hp_nid = nid;
+		nid = alc861_look_for_dac(codec, pin);
+		if (nid) {
+			err = alc861_create_out_sw(codec, "Headphone", nid, 3);
+			if (err < 0)
+				return err;
+			spec->multiout.hp_nid = nid;
+		}
 	}
 	return 0;
 }
 
 /* create playback/capture controls for input pins */
-static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec,
+static int alc861_auto_create_analog_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
+	struct alc_spec *spec = codec->spec;
 	struct hda_input_mux *imux = &spec->private_imux[0];
 	int i, err, idx, idx1;
 
@@ -13905,12 +13940,29 @@ static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec,
 
 static void alc861_auto_set_output_and_unmute(struct hda_codec *codec,
 					      hda_nid_t nid,
-					      int pin_type, int dac_idx)
+					      int pin_type, hda_nid_t dac)
 {
+	hda_nid_t mix, srcs[5];
+	int i, num;
+
 	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
 			    pin_type);
-	snd_hda_codec_write(codec, dac_idx, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+	snd_hda_codec_write(codec, dac, 0, AC_VERB_SET_AMP_GAIN_MUTE,
 			    AMP_OUT_UNMUTE);
+	if (snd_hda_get_connections(codec, nid, &mix, 1) != 1)
+		return;
+	num = snd_hda_get_connections(codec, mix, srcs, ARRAY_SIZE(srcs));
+	if (num < 0)
+		return;
+	for (i = 0; i < num; i++) {
+		unsigned int mute;
+		if (srcs[i] == dac || srcs[i] == 0x15)
+			mute = AMP_IN_UNMUTE(i);
+		else
+			mute = AMP_IN_MUTE(i);
+		snd_hda_codec_write(codec, mix, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+				    mute);
+	}
 }
 
 static void alc861_auto_init_multi_out(struct hda_codec *codec)
@@ -13933,12 +13985,13 @@ static void alc861_auto_init_hp_out(struct hda_codec *codec)
 	hda_nid_t pin;
 
 	pin = spec->autocfg.hp_pins[0];
-	if (pin) /* connect to front */
+	if (pin)
 		alc861_auto_set_output_and_unmute(codec, pin, PIN_HP,
-						  spec->multiout.dac_nids[0]);
+						  spec->multiout.hp_nid);
 	pin = spec->autocfg.speaker_pins[0];
 	if (pin)
-		alc861_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
+		alc861_auto_set_output_and_unmute(codec, pin, PIN_OUT,
+						  spec->multiout.dac_nids[0]);
 }
 
 static void alc861_auto_init_analog_input(struct hda_codec *codec)
@@ -13970,16 +14023,16 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
 	if (!spec->autocfg.line_outs)
 		return 0; /* can't find valid BIOS pin config */
 
-	err = alc861_auto_fill_dac_nids(spec, &spec->autocfg);
+	err = alc861_auto_fill_dac_nids(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
-	err = alc861_auto_create_multi_out_ctls(spec, &spec->autocfg);
+	err = alc861_auto_create_multi_out_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
-	err = alc861_auto_create_hp_ctls(spec, spec->autocfg.hp_pins[0]);
+	err = alc861_auto_create_hp_ctls(codec, spec->autocfg.hp_pins[0]);
 	if (err < 0)
 		return err;
-	err = alc861_auto_create_analog_input_ctls(spec, &spec->autocfg);
+	err = alc861_auto_create_analog_input_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
 
-- 
cgit v1.2.3


From 052960483833f81603450e2a3cfa7a5ee98eedd4 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Wed, 22 Jul 2009 14:38:58 +0200
Subject: ALSA: hda_generic: do not read connections for widged with an unknown
 type

Reading node connections for an unknown widget can confuse HDA codec bus.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_generic.c | 18 +++++++++++-------
 1 file changed, 11 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 1d5797a9668..da3e1ed29b3 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -121,11 +121,17 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid
 	if (node == NULL)
 		return -ENOMEM;
 	node->nid = nid;
-	nconns = snd_hda_get_connections(codec, nid, conn_list,
-					 HDA_MAX_CONNECTIONS);
-	if (nconns < 0) {
-		kfree(node);
-		return nconns;
+	node->wid_caps = get_wcaps(codec, nid);
+	node->type = (node->wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+	if (node->type <= AC_WID_BEEP || node->type == AC_WID_VENDOR) {
+		nconns = snd_hda_get_connections(codec, nid, conn_list,
+						 HDA_MAX_CONNECTIONS);
+		if (nconns < 0) {
+			kfree(node);
+			return nconns;
+		}
+	} else {
+		nconns = 0;
 	}
 	if (nconns <= ARRAY_SIZE(node->slist))
 		node->conn_list = node->slist;
@@ -140,8 +146,6 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid
 	}
 	memcpy(node->conn_list, conn_list, nconns * sizeof(hda_nid_t));
 	node->nconns = nconns;
-	node->wid_caps = get_wcaps(codec, nid);
-	node->type = (node->wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
 
 	if (node->type == AC_WID_PIN) {
 		node->pin_caps = snd_hda_query_pin_caps(codec, node->nid);
-- 
cgit v1.2.3


From 254da007f93f79ee773788cbb38f24dfab9590f5 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Wed, 22 Jul 2009 15:31:03 +0200
Subject: ALSA: hda_generic: use AC_WCAP_CONN_LIST check for widget connections

Previous patch used widget type, but the presence flag of the connection
list is in the widget capabilities.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_generic.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index da3e1ed29b3..bccb62a643c 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -123,7 +123,7 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid
 	node->nid = nid;
 	node->wid_caps = get_wcaps(codec, nid);
 	node->type = (node->wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
-	if (node->type <= AC_WID_BEEP || node->type == AC_WID_VENDOR) {
+	if (node->wid_caps & AC_WCAP_CONN_LIST) {
 		nconns = snd_hda_get_connections(codec, nid, conn_list,
 						 HDA_MAX_CONNECTIONS);
 		if (nconns < 0) {
-- 
cgit v1.2.3


From 16a433d8b38720a816736d59a62e705c19e31600 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Wed, 22 Jul 2009 16:20:40 +0200
Subject: ALSA: hda-intel: Cleanups for widget connection list handling

This patch adds a check to snd_hda_get_connections() routine for
presence of AC_WCAP_CONN_LIST. Also, make sure that negative error
codes from noted route are handled on all places as errors.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c      | 6 ++++++
 sound/pci/hda/patch_realtek.c  | 6 ------
 sound/pci/hda/patch_sigmatel.c | 8 +++++---
 3 files changed, 11 insertions(+), 9 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 3ecb45ddadf..aa554078611 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -306,6 +306,12 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
 	if (snd_BUG_ON(!conn_list || max_conns <= 0))
 		return -EINVAL;
 
+	if ((get_wcaps(codec, nid) & AC_WCAP_CONN_LIST) == 0) {
+		snd_printk(KERN_WARNING "hda_codec: "
+			   "connection list not available for 0x%x\n", nid);
+		return -EINVAL;
+	}
+
 	parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN);
 	if (parm & AC_CLIST_LONG) {
 		/* long form */
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b2ea43206f1..728f2a72f9b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4501,12 +4501,6 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 					      &dig_nid, 1);
 		if (err < 0)
 			continue;
-		if (dig_nid > 0x7f) {
-			printk(KERN_ERR "alc880_auto: invalid dig_nid "
-				"connection 0x%x for NID 0x%x\n", dig_nid,
-				spec->autocfg.dig_out_pins[i]);
-			continue;
-		}
 		if (!i)
 			spec->multiout.dig_out_nid = dig_nid;
 		else {
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 00c702df721..948600dd3b9 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -3389,7 +3389,7 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec)
 				spec->mono_nid,
 				con_lst,
 				HDA_MAX_NUM_INPUTS);
-	if (!num_cons || num_cons > ARRAY_SIZE(stac92xx_mono_labels))
+	if (num_cons <= 0 || num_cons > ARRAY_SIZE(stac92xx_mono_labels))
 		return -EINVAL;
 
 	for (i = 0; i < num_cons; i++) {
@@ -3535,7 +3535,7 @@ static int stac92xx_auto_create_spdif_mux_ctls(struct hda_codec *codec)
 				spec->smux_nids[0],
 				con_lst,
 				HDA_MAX_NUM_INPUTS);
-	if (!num_cons)
+	if (num_cons <= 0)
 		return -EINVAL;
 
 	if (!labels)
@@ -3742,7 +3742,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 		if (snd_hda_get_connections(codec,
 				spec->autocfg.mono_out_pin, conn_list, 1) &&
 				snd_hda_get_connections(codec, conn_list[0],
-				conn_list, 1)) {
+				conn_list, 1) > 0) {
 
 				int wcaps = get_wcaps(codec, conn_list[0]);
 				int wid_type = (wcaps & AC_WCAP_TYPE)
@@ -5169,6 +5169,8 @@ again:
 
 	num_dacs = snd_hda_get_connections(codec, nid,
 				conn, STAC92HD83_DAC_COUNT + 1) - 1;
+	if (num_dacs < 0)
+		num_dacs = STAC92HD83_DAC_COUNT;
 
 	/* set port X to select the last DAC
 	 */
-- 
cgit v1.2.3


From 87a8c3702e40028468a2d0ef6dab7627dab60c86 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Thu, 23 Jul 2009 10:58:29 +0200
Subject: ALSA: hda - Add better Intel IbexPeak platform support

Here are the new sound enabling patches for IbexPeak.

Summary of tested features:

  - playback
    - Front Headphone: OK
    - 8 channel audio: Front/Rear/CLFE/Side all OK

  - recording
    - Front Mic/Rear Mic: both OK
      (front/rear/line mics are selectable in the "Input source" alsamixer control)
    - Line In: not working
      (in 6ch mode, its amp/mute, direction and route all looks fine,
       so I'm a little puzzled)
      (hopefully no one will care this feature)

  - digital SPDIF input/output: not tested (no equipment)

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_intelhdmi.c |   2 +
 sound/pci/hda/patch_realtek.c   | 232 ++++++++++++++++++++++++++++++++++++++--
 2 files changed, 224 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c
index fcc77fec448..032850eba36 100644
--- a/sound/pci/hda/patch_intelhdmi.c
+++ b/sound/pci/hda/patch_intelhdmi.c
@@ -685,6 +685,7 @@ static struct hda_codec_preset snd_hda_preset_intelhdmi[] = {
 	{ .id = 0x80862802, .name = "G45 DEVCTG", .patch = patch_intel_hdmi },
 	{ .id = 0x80862803, .name = "G45 DEVELK", .patch = patch_intel_hdmi },
 	{ .id = 0x80862804, .name = "G45 DEVIBX", .patch = patch_intel_hdmi },
+	{ .id = 0x80860054, .name = "G45 DEVIBX", .patch = patch_intel_hdmi },
 	{ .id = 0x10951392, .name = "SiI1392 HDMI",     .patch = patch_intel_hdmi },
 	{} /* terminator */
 };
@@ -694,6 +695,7 @@ MODULE_ALIAS("snd-hda-codec-id:80862801");
 MODULE_ALIAS("snd-hda-codec-id:80862802");
 MODULE_ALIAS("snd-hda-codec-id:80862803");
 MODULE_ALIAS("snd-hda-codec-id:80862804");
+MODULE_ALIAS("snd-hda-codec-id:80860054");
 MODULE_ALIAS("snd-hda-codec-id:10951392");
 
 MODULE_LICENSE("GPL");
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 728f2a72f9b..b5b36b893b1 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -235,6 +235,8 @@ enum {
 	ALC883_FUJITSU_PI2515,
 	ALC888_FUJITSU_XA3530,
 	ALC883_3ST_6ch_INTEL,
+	ALC889A_INTEL,
+	ALC889_INTEL,
 	ALC888_ASUS_M90V,
 	ALC888_ASUS_EEE1601,
 	ALC889A_MB31,
@@ -1027,6 +1029,16 @@ static void alc888_coef_init(struct hda_codec *codec)
 				    AC_VERB_SET_PROC_COEF, 0x3030);
 }
 
+static void alc889_coef_init(struct hda_codec *codec)
+{
+	unsigned int tmp;
+
+	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7);
+	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);
+	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, tmp|0x2010);
+}
+
 static void alc_auto_init_amp(struct hda_codec *codec, int type)
 {
 	unsigned int tmp;
@@ -1084,15 +1096,7 @@ static void alc_auto_init_amp(struct hda_codec *codec, int type)
 		case 0x10ec0885:
 		case 0x10ec0887:
 		case 0x10ec0889:
-			snd_hda_codec_write(codec, 0x20, 0,
-					    AC_VERB_SET_COEF_INDEX, 7);
-			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);
-			snd_hda_codec_write(codec, 0x20, 0,
-					    AC_VERB_SET_PROC_COEF,
-					    tmp | 0x2010);
+			alc889_coef_init(codec);
 			break;
 		case 0x10ec0888:
 			alc888_coef_init(codec);
@@ -6348,6 +6352,15 @@ static struct hda_input_mux alc882_capture_source = {
 
 #define alc883_capture_source	alc882_capture_source
 
+static struct hda_input_mux alc889_capture_source = {
+	.num_items = 3,
+	.items = {
+		{ "Front Mic", 0x0 },
+		{ "Mic", 0x3 },
+		{ "Line", 0x2 },
+	},
+};
+
 static struct hda_input_mux mb5_capture_source = {
 	.num_items = 3,
 	.items = {
@@ -6668,6 +6681,46 @@ static struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = {
 	{ 6, alc883_3ST_ch6_intel_init },
 };
 
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc889_ch6_intel_init[] = {
+	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ } /* end */
+};
+
+/*
+ * 8ch mode
+ */
+static struct hda_verb alc889_ch8_intel_init[] = {
+	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x03 },
+	{ } /* end */
+};
+
+static struct hda_channel_mode alc889_8ch_intel_modes[2] = {
+	{ 6, alc889_ch6_intel_init },
+	{ 8, alc889_ch8_intel_init },
+};
+
 /*
  * 6ch mode
  */
@@ -6929,6 +6982,103 @@ static struct hda_verb alc882_eapd_verbs[] = {
 	{ }
 };
 
+static struct hda_verb alc889_eapd_verbs[] = {
+	{0x14, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{0x15, AC_VERB_SET_EAPD_BTLENABLE, 2},
+	{ }
+};
+
+
+static struct hda_verb alc885_init_verbs[] = {
+	/* Front mixer: unmute input/output amp left and right (volume = 0) */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* Rear mixer */
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* CLFE mixer */
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* Side mixer */
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+
+	/* mute analog input loopbacks */
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+
+	/* Front HP Pin: output 0 (0x0c) */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* Front Pin: output 0 (0x0c) */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* Rear Pin: output 1 (0x0d) */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x19, AC_VERB_SET_CONNECT_SEL, 0x01},
+	/* CLFE Pin: output 2 (0x0e) */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
+	/* Side Pin: output 3 (0x0f) */
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+	/* Mic (rear) pin: input vref at 80% */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Front Mic pin: input vref at 80% */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+	/* Line In pin: input */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
+
+	/* Mixer elements: 0x18, , 0x1a, 0x1b */
+	/* Input mixer1 */
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	/* Input mixer2 */
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
+	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	/* Input mixer3 */
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
+	/* ADC2: mute amp left and right */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	/* ADC3: mute amp left and right */
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+
+	{ }
+};
+
+static struct hda_verb alc885_init_input_verbs[] = {
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)},
+	{ }
+};
+
+
+/* Unmute Selector 24h and set the default input to front mic */
+static struct hda_verb alc889_init_input_verbs[] = {
+	{0x24, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{ }
+};
+
+
 #define alc883_init_verbs	alc882_base_init_verbs
 
 /* Mac Pro test */
@@ -7497,6 +7647,30 @@ static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = {
 	{ } /* end */
 };
 
+static struct snd_kcontrol_new alc885_8ch_intel_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", 0x0d, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Surround Playback Switch", 0x0d, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0,
+			      HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT),
+	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Speaker Playback Switch", 0x0f, 2, HDA_INPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 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, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost", 0x1b, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	{ } /* end */
+};
+
 static struct snd_kcontrol_new alc883_fivestack_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT),
@@ -8340,6 +8514,8 @@ static const char *alc882_models[ALC882_MODEL_LAST] = {
 	[ALC883_FUJITSU_PI2515] = "fujitsu-pi2515",
 	[ALC888_FUJITSU_XA3530] = "fujitsu-xa3530",
 	[ALC883_3ST_6ch_INTEL]	= "3stack-6ch-intel",
+	[ALC889A_INTEL]		= "intel-alc889a",
+	[ALC889_INTEL]		= "intel-x58",
 	[ALC1200_ASUS_P5Q]	= "asus-p5q",
 	[ALC889A_MB31]		= "mb31",
 	[ALC883_SONY_VAIO_TT]	= "sony-vaio-tt",
@@ -8459,7 +8635,9 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL),
 	SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL),
 	SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC),
-	SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC883_3ST_6ch_INTEL),
+	SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC889_INTEL),
+	SND_PCI_QUIRK(0x8086, 0x0021, "Intel IbexPeak", ALC889A_INTEL),
+	SND_PCI_QUIRK(0x8086, 0x3b56, "Intel IbexPeak", ALC889A_INTEL),
 	SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch),
 
 	{}
@@ -8675,6 +8853,40 @@ static struct alc_config_preset alc882_presets[] = {
 		.need_dac_fix = 1,
 		.input_mux = &alc883_3stack_6ch_intel,
 	},
+	[ALC889A_INTEL] = {
+		.mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc885_init_verbs, alc885_init_input_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
+		.adc_nids = alc889_adc_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.dig_in_nid = ALC883_DIGIN_NID,
+		.slave_dig_outs = alc883_slave_dig_outs,
+		.num_channel_mode = ARRAY_SIZE(alc889_8ch_intel_modes),
+		.channel_mode = alc889_8ch_intel_modes,
+		.capsrc_nids = alc889_capsrc_nids,
+		.input_mux = &alc889_capture_source,
+		.need_dac_fix = 1,
+	},
+	[ALC889_INTEL] = {
+		.mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc885_init_verbs, alc889_init_input_verbs,
+				alc889_eapd_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
+		.adc_nids = alc889_adc_nids,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.dig_in_nid = ALC883_DIGIN_NID,
+		.slave_dig_outs = alc883_slave_dig_outs,
+		.num_channel_mode = ARRAY_SIZE(alc889_8ch_intel_modes),
+		.channel_mode = alc889_8ch_intel_modes,
+		.capsrc_nids = alc889_capsrc_nids,
+		.input_mux = &alc889_capture_source,
+		.init_hook = alc889_coef_init,
+		.need_dac_fix = 1,
+	},
 	[ALC883_6ST_DIG] = {
 		.mixers = { alc883_base_mixer, alc883_chmode_mixer },
 		.init_verbs = { alc883_init_verbs },
-- 
cgit v1.2.3


From a22d543a95e82e5ad0ee1a44aad54fd6b6bf52a8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 27 Jul 2009 12:54:26 +0200
Subject: ALSA: hda - Introduce get_wcaps_type() macro

Add a helper macro to retrieve the widget type from wiget cap bits.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c      |  9 +++------
 sound/pci/hda/hda_generic.c    |  2 +-
 sound/pci/hda/hda_local.h      |  3 +++
 sound/pci/hda/hda_proc.c       |  3 +--
 sound/pci/hda/patch_ca0110.c   |  3 +--
 sound/pci/hda/patch_realtek.c  | 17 ++++++++---------
 sound/pci/hda/patch_sigmatel.c |  8 +++-----
 sound/pci/hda/patch_via.c      |  3 +--
 8 files changed, 21 insertions(+), 27 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index aa554078611..b8c656c21fe 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -748,8 +748,7 @@ static int read_pin_defaults(struct hda_codec *codec)
 	for (i = 0; i < codec->num_nodes; i++, nid++) {
 		struct hda_pincfg *pin;
 		unsigned int wcaps = get_wcaps(codec, nid);
-		unsigned int wid_type = (wcaps & AC_WCAP_TYPE) >>
-				AC_WCAP_TYPE_SHIFT;
+		unsigned int wid_type = get_wcaps_type(wcaps);
 		if (wid_type != AC_WID_PIN)
 			continue;
 		pin = snd_array_new(&codec->init_pins);
@@ -2386,8 +2385,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 	for (i = 0; i < codec->num_nodes; i++, nid++) {
 		unsigned int wcaps = get_wcaps(codec, nid);
 		if (wcaps & AC_WCAP_POWER) {
-			unsigned int wid_type = (wcaps & AC_WCAP_TYPE) >>
-				AC_WCAP_TYPE_SHIFT;
+			unsigned int wid_type = get_wcaps_type(wcaps);
 			if (power_state == AC_PWRST_D3 &&
 			    wid_type == AC_WID_PIN) {
 				unsigned int pincap;
@@ -3678,8 +3676,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 	end_nid = codec->start_nid + codec->num_nodes;
 	for (nid = codec->start_nid; nid < end_nid; nid++) {
 		unsigned int wid_caps = get_wcaps(codec, nid);
-		unsigned int wid_type =
-			(wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+		unsigned int wid_type = get_wcaps_type(wid_caps);
 		unsigned int def_conf;
 		short assoc, loc;
 
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index bccb62a643c..b36f6c5a92d 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -122,7 +122,7 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid
 		return -ENOMEM;
 	node->nid = nid;
 	node->wid_caps = get_wcaps(codec, nid);
-	node->type = (node->wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+	node->type = get_wcaps_type(node->wid_caps);
 	if (node->wid_caps & AC_WCAP_CONN_LIST) {
 		nconns = snd_hda_get_connections(codec, nid, conn_list,
 						 HDA_MAX_CONNECTIONS);
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 75aa3785212..fa57cb93b44 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -407,6 +407,9 @@ static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid)
 	return codec->wcaps[nid - codec->start_nid];
 }
 
+/* get the widget type from widget capability bits */
+#define get_wcaps_type(wcaps) (((wcaps) & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT)
+
 u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction);
 int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
 			      unsigned int caps);
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 418c5d1bada..a721eb08a29 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -508,8 +508,7 @@ static void print_codec_info(struct snd_info_entry *entry,
 		unsigned int wid_caps =
 			snd_hda_param_read(codec, nid,
 					   AC_PAR_AUDIO_WIDGET_CAP);
-		unsigned int wid_type =
-			(wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+		unsigned int wid_type = get_wcaps_type(wid_caps);
 		hda_nid_t conn[HDA_MAX_CONNECTIONS];
 		int conn_len = 0;
 
diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
index 019ca7cb56d..d08353d3bb7 100644
--- a/sound/pci/hda/patch_ca0110.c
+++ b/sound/pci/hda/patch_ca0110.c
@@ -459,8 +459,7 @@ static void parse_input(struct hda_codec *codec)
 	nid = codec->start_nid;
 	for (i = 0; i < codec->num_nodes; i++, nid++) {
 		unsigned int wcaps = get_wcaps(codec, nid);
-		unsigned int type = (wcaps & AC_WCAP_TYPE) >>
-			AC_WCAP_TYPE_SHIFT;
+		unsigned int type = get_wcaps_type(wcaps);
 		if (type != AC_WID_AUD_IN)
 			continue;
 		if (snd_hda_get_connections(codec, nid, &pin, 1) != 1)
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e1563f48243..df87c01e27d 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -415,7 +415,7 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol,
 	mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx;
 	imux = &spec->input_mux[mux_idx];
 
-	type = (get_wcaps(codec, nid) & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+	type = get_wcaps_type(get_wcaps(codec, nid));
 	if (type == AC_WID_AUD_MIX) {
 		/* Matrix-mixer style (e.g. ALC882) */
 		unsigned int *cur_val = &spec->cur_mux[adc_idx];
@@ -4623,7 +4623,7 @@ static int patch_alc880(struct hda_codec *codec)
 		/* check whether NID 0x07 is valid */
 		unsigned int wcap = get_wcaps(codec, alc880_adc_nids[0]);
 		/* get type */
-		wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+		wcap = get_wcaps_type(wcap);
 		if (wcap != AC_WID_AUD_IN) {
 			spec->adc_nids = alc880_adc_nids_alt;
 			spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids_alt);
@@ -6266,7 +6266,7 @@ static int patch_alc260(struct hda_codec *codec)
 	if (!spec->adc_nids && spec->input_mux) {
 		/* check whether NID 0x04 is valid */
 		unsigned int wcap = get_wcaps(codec, 0x04);
-		wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+		wcap = get_wcaps_type(wcap);
 		/* get type */
 		if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
 			spec->adc_nids = alc260_adc_nids_alt;
@@ -9460,7 +9460,7 @@ static int alc882_parse_auto_config(struct hda_codec *codec)
 	spec->init_verbs[0] = alc883_auto_init_verbs;
 	/* if ADC 0x07 is available, initialize it, too */
 	wcap = get_wcaps(codec, 0x07);
-	wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+	wcap = get_wcaps_type(wcap);
 	if (wcap == AC_WID_AUD_IN)
 		add_verb(spec, alc882_adc1_init_verbs);
 
@@ -9577,7 +9577,7 @@ static int patch_alc882(struct hda_codec *codec)
 			hda_nid_t nid = alc882_adc_nids[i];
 			unsigned int wcap = get_wcaps(codec, nid);
 			/* get type */
-			wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+			wcap = get_wcaps_type(wcap);
 			if (wcap != AC_WID_AUD_IN)
 				continue;
 			spec->private_adc_nids[spec->num_adc_nids] = nid;
@@ -11460,7 +11460,7 @@ static int patch_alc262(struct hda_codec *codec)
 			unsigned int wcap = get_wcaps(codec, 0x07);
 
 			/* get type */
-			wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+			wcap = get_wcaps_type(wcap);
 			if (wcap != AC_WID_AUD_IN) {
 				spec->adc_nids = alc262_adc_nids_alt;
 				spec->num_adc_nids =
@@ -12536,7 +12536,7 @@ static int patch_alc268(struct hda_codec *codec)
 		int i;
 
 		/* get type */
-		wcap = (wcap & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+		wcap = get_wcaps_type(wcap);
 		if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
 			spec->adc_nids = alc268_adc_nids_alt;
 			spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt);
@@ -13991,8 +13991,7 @@ static hda_nid_t alc861_look_for_dac(struct hda_codec *codec, hda_nid_t pin)
 		return 0;
 	for (i = 0; i < num; i++) {
 		unsigned int type;
-		type = (get_wcaps(codec, srcs[i]) & AC_WCAP_TYPE)
-			>> AC_WCAP_TYPE_SHIFT;
+		type = get_wcaps_type(get_wcaps(codec, srcs[i]));
 		if (type != AC_WID_AUD_OUT)
 			continue;
 		for (j = 0; j < spec->multiout.num_dacs; j++)
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 8c48f868d99..9de97f30699 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -3058,7 +3058,7 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid)
 					   HDA_MAX_CONNECTIONS);
 	for (j = 0; j < conn_len; j++) {
 		wcaps = get_wcaps(codec, conn[j]);
-		wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+		wtype = get_wcaps_type(wcaps);
 		/* we check only analog outputs */
 		if (wtype != AC_WID_AUD_OUT || (wcaps & AC_WCAP_DIGITAL))
 			continue;
@@ -3746,8 +3746,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 				conn_list, 1) > 0) {
 
 				int wcaps = get_wcaps(codec, conn_list[0]);
-				int wid_type = (wcaps & AC_WCAP_TYPE)
-					>> AC_WCAP_TYPE_SHIFT;
+				int wid_type = get_wcaps_type(wcaps);
 				/* LR swap check, some stac925x have a mux that
  				 * changes the DACs output path instead of the
  				 * mono-mux path.
@@ -4753,8 +4752,7 @@ static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
 	nid = codec->start_nid;
 	for (i = 0; i < codec->num_nodes; i++, nid++) {
 		unsigned int wcaps = get_wcaps(codec, nid);
-		unsigned int wid_type = (wcaps & AC_WCAP_TYPE) >>
-			AC_WCAP_TYPE_SHIFT;
+		unsigned int wid_type = get_wcaps_type(wcaps);
 		if (wid_type == AC_WID_PIN)
 			snd_hda_codec_read(codec, nid, 0,
 				AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index 9008b4b013a..ab90abb04cc 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1339,8 +1339,7 @@ static int get_mux_nids(struct hda_codec *codec)
 	for (i = 0; i < spec->num_adc_nids; i++) {
 		nid = spec->adc_nids[i];
 		while (nid) {
-			type = (get_wcaps(codec, nid) & AC_WCAP_TYPE)
-				>> AC_WCAP_TYPE_SHIFT;
+			type = get_wcaps_type(get_wcaps(codec, nid));
 			if (type == AC_WID_PIN)
 				break;
 			n = snd_hda_get_connections(codec, nid, conn,
-- 
cgit v1.2.3


From 1ba7a7c650bd8ee567e25d9b68e9081d131c70f6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 27 Jul 2009 12:56:26 +0200
Subject: ALSA: hda - Add exception for volume-knob in
 snd_hda_get_connections()

Volume-knob widgets may have connections even if they have no CONN_LIST
cap bit.  Allow the query exceptionally in snd_hda_get_connections().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 5 ++++-
 1 file changed, 4 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index b8c656c21fe..3a603cde8cc 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -301,12 +301,15 @@ int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
 	unsigned int parm;
 	int i, conn_len, conns;
 	unsigned int shift, num_elems, mask;
+	unsigned int wcaps;
 	hda_nid_t prev_nid;
 
 	if (snd_BUG_ON(!conn_list || max_conns <= 0))
 		return -EINVAL;
 
-	if ((get_wcaps(codec, nid) & AC_WCAP_CONN_LIST) == 0) {
+	wcaps = get_wcaps(codec, nid);
+	if (!(wcaps & AC_WCAP_CONN_LIST) &&
+	    get_wcaps_type(wcaps) != AC_WID_VOL_KNB) {
 		snd_printk(KERN_WARNING "hda_codec: "
 			   "connection list not available for 0x%x\n", nid);
 		return -EINVAL;
-- 
cgit v1.2.3


From 9a11f1aa8e14798037d0c9ac134696fa3af6eb2a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 28 Jul 2009 16:01:20 +0200
Subject: ALSA: hda - Reword information messages for BIOS auto-probing mode

The sentense "Unknown model for xxx, ..." makes people too nervous
and drives them to a direction to a wrong "fix" by giving any
mismatching model option.

Let's rephrase the messages to be more nice and easy (at least that
won't make people suspect conspiracies).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c   |  3 ++-
 sound/pci/hda/patch_cmedia.c   |  3 ++-
 sound/pci/hda/patch_realtek.c  | 34 ++++++++++++++++------------------
 sound/pci/hda/patch_sigmatel.c | 34 ++++++++++++++++++----------------
 4 files changed, 38 insertions(+), 36 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 3da85caf8af..ab3bcb78ace 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -2982,7 +2982,8 @@ static int patch_ad1988(struct hda_codec *codec)
 	board_config = snd_hda_check_board_config(codec, AD1988_MODEL_LAST,
 						  ad1988_models, ad1988_cfg_tbl);
 	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: Unknown model for AD1988, trying auto-probe from BIOS...\n");
+		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+		       codec->chip_name);
 		board_config = AD1988_AUTO;
 	}
 
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index c921264bbd7..780e1a72114 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -635,7 +635,8 @@ static int patch_cmi9880(struct hda_codec *codec)
 							cmi9880_models,
 							cmi9880_cfg_tbl);
 	if (spec->board_config < 0) {
-		snd_printdd(KERN_INFO "hda_codec: Unknown model for CMI9880\n");
+		snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+			    codec->chip_name);
 		spec->board_config = CMI_AUTO; /* try everything */
 	}
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index df87c01e27d..6794d028119 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4584,8 +4584,8 @@ static int patch_alc880(struct hda_codec *codec)
 						  alc880_models,
 						  alc880_cfg_tbl);
 	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: Unknown model for %s, "
-		       "trying auto-probe from BIOS...\n", codec->chip_name);
+		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+		       codec->chip_name);
 		board_config = ALC880_AUTO;
 	}
 
@@ -6228,8 +6228,7 @@ static int patch_alc260(struct hda_codec *codec)
 						  alc260_models,
 						  alc260_cfg_tbl);
 	if (board_config < 0) {
-		snd_printd(KERN_INFO "hda_codec: Unknown model for %s, "
-			   "trying auto-probe from BIOS...\n",
+		snd_printd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
 			   codec->chip_name);
 		board_config = ALC260_AUTO;
 	}
@@ -9526,8 +9525,7 @@ static int patch_alc882(struct hda_codec *codec)
 			ALC882_MODEL_LAST, alc882_models, alc882_ssid_cfg_tbl);
 
 	if (board_config < 0 || board_config >= ALC882_MODEL_LAST) {
-		printk(KERN_INFO "hda_codec: Unknown model for %s, "
-		       "trying auto-probe from BIOS...\n",
+		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
 		       codec->chip_name);
 		board_config = ALC882_AUTO;
 	}
@@ -11406,8 +11404,8 @@ static int patch_alc262(struct hda_codec *codec)
 						  alc262_cfg_tbl);
 
 	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: Unknown model for %s, "
-		       "trying auto-probe from BIOS...\n", codec->chip_name);
+		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+		       codec->chip_name);
 		board_config = ALC262_AUTO;
 	}
 
@@ -12479,8 +12477,8 @@ static int patch_alc268(struct hda_codec *codec)
 						  alc268_cfg_tbl);
 
 	if (board_config < 0 || board_config >= ALC268_MODEL_LAST) {
-		printk(KERN_INFO "hda_codec: Unknown model for %s, "
-		       "trying auto-probe from BIOS...\n", codec->chip_name);
+		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+		       codec->chip_name);
 		board_config = ALC268_AUTO;
 	}
 
@@ -13297,8 +13295,8 @@ static int patch_alc269(struct hda_codec *codec)
 						  alc269_cfg_tbl);
 
 	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: Unknown model for %s, "
-		       "trying auto-probe from BIOS...\n", codec->chip_name);
+		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+		       codec->chip_name);
 		board_config = ALC269_AUTO;
 	}
 
@@ -14448,8 +14446,8 @@ static int patch_alc861(struct hda_codec *codec)
 						  alc861_cfg_tbl);
 
 	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: Unknown model for %s, "
-		       "trying auto-probe from BIOS...\n", codec->chip_name);
+		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+		       codec->chip_name);
 		board_config = ALC861_AUTO;
 	}
 
@@ -15372,8 +15370,8 @@ static int patch_alc861vd(struct hda_codec *codec)
 						  alc861vd_cfg_tbl);
 
 	if (board_config < 0 || board_config >= ALC861VD_MODEL_LAST) {
-		printk(KERN_INFO "hda_codec: Unknown model for %s, "
-		       "trying auto-probe from BIOS...\n", codec->chip_name);
+		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+		       codec->chip_name);
 		board_config = ALC861VD_AUTO;
 	}
 
@@ -17296,8 +17294,8 @@ static int patch_alc662(struct hda_codec *codec)
 						  alc662_models,
 			  	                  alc662_cfg_tbl);
 	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: Unknown model for %s, "
-		       "trying auto-probe from BIOS...\n", codec->chip_name);
+		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+		       codec->chip_name);
 		board_config = ALC662_AUTO;
 	}
 
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 9de97f30699..c6dc625c66b 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4794,7 +4794,8 @@ static int patch_stac9200(struct hda_codec *codec)
 							stac9200_models,
 							stac9200_cfg_tbl);
 	if (spec->board_config < 0)
-		snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9200, using BIOS defaults\n");
+		snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+			    codec->chip_name);
 	else
 		stac92xx_set_config_regs(codec,
 					 stac9200_brd_tbl[spec->board_config]);
@@ -4866,8 +4867,8 @@ static int patch_stac925x(struct hda_codec *codec)
 							stac925x_cfg_tbl);
  again:
 	if (spec->board_config < 0)
-		snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC925x,"
-				      "using BIOS defaults\n");
+		snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+			    codec->chip_name);
 	else
 		stac92xx_set_config_regs(codec,
 					 stac925x_brd_tbl[spec->board_config]);
@@ -4949,8 +4950,8 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
 							stac92hd73xx_cfg_tbl);
 again:
 	if (spec->board_config < 0)
-		snd_printdd(KERN_INFO "hda_codec: Unknown model for"
-			" STAC92HD73XX, using BIOS defaults\n");
+		snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+			    codec->chip_name);
 	else
 		stac92xx_set_config_regs(codec,
 				stac92hd73xx_brd_tbl[spec->board_config]);
@@ -5125,8 +5126,8 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 							stac92hd83xxx_cfg_tbl);
 again:
 	if (spec->board_config < 0)
-		snd_printdd(KERN_INFO "hda_codec: Unknown model for"
-			" STAC92HD83XXX, using BIOS defaults\n");
+		snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+			    codec->chip_name);
 	else
 		stac92xx_set_config_regs(codec,
 				stac92hd83xxx_brd_tbl[spec->board_config]);
@@ -5291,8 +5292,8 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 							stac92hd71bxx_cfg_tbl);
 again:
 	if (spec->board_config < 0)
-		snd_printdd(KERN_INFO "hda_codec: Unknown model for"
-			" STAC92HD71BXX, using BIOS defaults\n");
+		snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+			    codec->chip_name);
 	else
 		stac92xx_set_config_regs(codec,
 				stac92hd71bxx_brd_tbl[spec->board_config]);
@@ -5547,8 +5548,8 @@ static int patch_stac922x(struct hda_codec *codec)
 
  again:
 	if (spec->board_config < 0)
-		snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, "
-			"using BIOS defaults\n");
+		snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+			    codec->chip_name);
 	else
 		stac92xx_set_config_regs(codec,
 				stac922x_brd_tbl[spec->board_config]);
@@ -5610,8 +5611,8 @@ static int patch_stac927x(struct hda_codec *codec)
 							stac927x_cfg_tbl);
  again:
 	if (spec->board_config < 0)
-		snd_printdd(KERN_INFO "hda_codec: Unknown model for"
-			    "STAC927x, using BIOS defaults\n");
+		snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+			    codec->chip_name);
 	else
 		stac92xx_set_config_regs(codec,
 				stac927x_brd_tbl[spec->board_config]);
@@ -5738,7 +5739,8 @@ static int patch_stac9205(struct hda_codec *codec)
 							stac9205_cfg_tbl);
  again:
 	if (spec->board_config < 0)
-		snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9205, using BIOS defaults\n");
+		snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+			    codec->chip_name);
 	else
 		stac92xx_set_config_regs(codec,
 					 stac9205_brd_tbl[spec->board_config]);
@@ -5890,8 +5892,8 @@ static int patch_stac9872(struct hda_codec *codec)
 							stac9872_models,
 							stac9872_cfg_tbl);
 	if (spec->board_config < 0)
-		snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9872, "
-			    "using BIOS defaults\n");
+		snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
+			    codec->chip_name);
 	else
 		stac92xx_set_config_regs(codec,
 					 stac9872_brd_tbl[spec->board_config]);
-- 
cgit v1.2.3


From 4417932315d185b59c0089091de7fa509c59fd5a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 28 Jul 2009 17:03:49 +0200
Subject: ALSA: hda - Don't create unneeded digital input source for IDT
 92HD71x

The current driver creates always the digital input source mixer
elements for IDT 92HD71x codecs no matter whether digital mics are
present.  This patch adds the proper check to avoid the creation of
these controls if unnecessary.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 16 ++++++++++------
 1 file changed, 10 insertions(+), 6 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index c6dc625c66b..4e3531b42e1 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5355,7 +5355,6 @@ again:
 		/* 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];
@@ -5366,8 +5365,11 @@ again:
 		spec->num_dmics = stac92hd71bxx_connected_ports(codec,
 					stac92hd71bxx_dmic_nids,
 					STAC92HD71BXX_NUM_DMICS - 1);
-		spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
-		ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 2;
+		if (spec->num_dmics) {
+			spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
+			spec->dinput_mux = &spec->private_dimux;
+			ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 2;
+		}
 		break;
 	case 0x111d7603: /* 6 Port with Analog Mixer */
 		if ((codec->revision_id & 0xf) == 1)
@@ -5379,15 +5381,17 @@ again:
 	default:
 		memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer,
 		       sizeof(stac92hd71bxx_dmux_amixer));
-		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->num_dmics = stac92hd71bxx_connected_ports(codec,
 					stac92hd71bxx_dmic_nids,
 					STAC92HD71BXX_NUM_DMICS);
-		spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
-		ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1;
+		if (spec->num_dmics) {
+			spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
+			spec->dinput_mux = &spec->private_dimux;
+			ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1;
+		}
 	}
 
 	if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP)
-- 
cgit v1.2.3


From 6479c63188290beae83ade3243b9d6eb47d394b6 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 28 Jul 2009 18:20:25 +0200
Subject: ALSA: hda - Create Capture controls dynamically

Instead of static snd_kcontrol_new arrays, create "Capture Volume"
and "Capture Switch" controls dynamically based on the mixer attr
values (made via HDA_COMPOSE_AMP_VAL()).
This reduces the code size and gives more flexibility to change
the number of controls later.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 220 +++++++++++++++++++++++------------------
 1 file changed, 125 insertions(+), 95 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 4e3531b42e1..be6cf2cfb19 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -238,6 +238,11 @@ struct sigmatel_spec {
 	unsigned int num_dmuxes;
 	hda_nid_t *smux_nids;
 	unsigned int num_smuxes;
+
+	unsigned long *capvols; /* amp-volume attr: HDA_COMPOSE_AMP_VAL() */
+	unsigned long *capsws; /* amp-mute attr: HDA_COMPOSE_AMP_VAL() */
+	unsigned int num_caps; /* number of capture volume/switch elements */
+
 	const char **spdif_labels;
 
 	hda_nid_t dig_in_nid;
@@ -334,6 +339,13 @@ static hda_nid_t stac92hd73xx_smux_nids[2] = {
 	0x22, 0x23,
 };
 
+#define STAC92HD73XX_NUM_CAPS	2
+static unsigned long stac92hd73xx_capvols[] = {
+	HDA_COMPOSE_AMP_VAL(0x20, 3, 0, HDA_OUTPUT),
+	HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+};
+#define stac92hd73xx_capsws	stac92hd73xx_capvols
+
 #define STAC92HD83XXX_NUM_DMICS	2
 static hda_nid_t stac92hd83xxx_dmic_nids[STAC92HD83XXX_NUM_DMICS + 1] = {
 	0x11, 0x12, 0
@@ -365,6 +377,13 @@ static hda_nid_t stac92hd83xxx_amp_nids[1] = {
 	0xc,
 };
 
+#define STAC92HD83XXX_NUM_CAPS	2
+static unsigned long stac92hd83xxx_capvols[] = {
+	HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
+	HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_OUTPUT),
+};
+#define stac92hd83xxx_capsws	stac92hd83xxx_capvols
+
 static hda_nid_t stac92hd71bxx_pwr_nids[3] = {
 	0x0a, 0x0d, 0x0f
 };
@@ -394,6 +413,13 @@ static hda_nid_t stac92hd71bxx_slave_dig_outs[2] = {
 	0x22, 0
 };
 
+#define STAC92HD71BXX_NUM_CAPS		2
+static unsigned long stac92hd71bxx_capvols[] = {
+	HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_OUTPUT),
+	HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT),
+};
+#define stac92hd71bxx_capsws	stac92hd71bxx_capvols
+
 static hda_nid_t stac925x_adc_nids[1] = {
         0x03,
 };
@@ -415,6 +441,13 @@ static hda_nid_t stac925x_dmux_nids[1] = {
 	0x14,
 };
 
+static unsigned long stac925x_capvols[] = {
+	HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_OUTPUT),
+};
+static unsigned long stac925x_capsws[] = {
+	HDA_COMPOSE_AMP_VAL(0x14, 3, 0, HDA_OUTPUT),
+};
+
 static hda_nid_t stac922x_adc_nids[2] = {
         0x06, 0x07,
 };
@@ -423,6 +456,13 @@ static hda_nid_t stac922x_mux_nids[2] = {
         0x12, 0x13,
 };
 
+#define STAC922X_NUM_CAPS	2
+static unsigned long stac922x_capvols[] = {
+	HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_INPUT),
+	HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_INPUT),
+};
+#define stac922x_capsws		stac922x_capvols
+
 static hda_nid_t stac927x_slave_dig_outs[2] = {
 	0x1f, 0,
 };
@@ -452,6 +492,18 @@ static hda_nid_t stac927x_dmic_nids[STAC927X_NUM_DMICS + 1] = {
 	0x13, 0x14, 0
 };
 
+#define STAC927X_NUM_CAPS	3
+static unsigned long stac927x_capvols[] = {
+	HDA_COMPOSE_AMP_VAL(0x18, 3, 0, HDA_INPUT),
+	HDA_COMPOSE_AMP_VAL(0x19, 3, 0, HDA_INPUT),
+	HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_INPUT),
+};
+static unsigned long stac927x_capsws[] = {
+	HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
+	HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_OUTPUT),
+	HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT),
+};
+
 static const char *stac927x_spdif_labels[5] = {
 	"Digital Playback", "ADAT", "Analog Mux 1",
 	"Analog Mux 2", "Analog Mux 3"
@@ -478,6 +530,16 @@ static hda_nid_t stac9205_dmic_nids[STAC9205_NUM_DMICS + 1] = {
         0x17, 0x18, 0
 };
 
+#define STAC9205_NUM_CAPS	2
+static unsigned long stac9205_capvols[] = {
+	HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_INPUT),
+	HDA_COMPOSE_AMP_VAL(0x1c, 3, 0, HDA_INPUT),
+};
+static unsigned long stac9205_capsws[] = {
+	HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT),
+	HDA_COMPOSE_AMP_VAL(0x1e, 3, 0, HDA_OUTPUT),
+};
+
 static hda_nid_t stac9200_pin_nids[8] = {
 	0x08, 0x09, 0x0d, 0x0e, 
 	0x0f, 0x10, 0x11, 0x12,
@@ -1069,12 +1131,6 @@ static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = {
 	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_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 */
 };
 
@@ -1094,12 +1150,6 @@ static struct snd_kcontrol_new stac92hd73xx_10ch_loopback[] = {
 };
 
 static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = {
-	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),
-
 	HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT),
 
@@ -1118,12 +1168,6 @@ static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = {
 };
 
 static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = {
-	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),
-
 	HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT),
 
@@ -1143,12 +1187,6 @@ static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = {
 
 
 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, 0x3, HDA_INPUT),
 	HDA_CODEC_MUTE("DAC0 Capture Switch", 0x1b, 0x3, HDA_INPUT),
 
@@ -1169,17 +1207,6 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = {
 };
 
 static struct snd_kcontrol_new stac92hd71bxx_analog_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 Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 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),
 
@@ -1198,29 +1225,9 @@ static struct snd_kcontrol_new stac92hd71bxx_loopback[] = {
 	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2)
 };
 
-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 Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
 static struct snd_kcontrol_new stac925x_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x0e, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-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("Capture Volume", 0x1, 0x1c, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1e, 0x0, HDA_OUTPUT),
 	{ } /* end */
 };
 
@@ -1229,29 +1236,6 @@ static struct snd_kcontrol_new stac9205_loopback[] = {
 	{}
 };
 
-/* This needs to be generated dynamically based on sequence */
-static struct snd_kcontrol_new stac922x_mixer[] = {
-	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("Capture Volume", 0x1, 0x18, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x18, 0x0, HDA_INPUT),
-	{ } /* end */
-};
-
-
-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("Capture Volume", 0x1, 0x19, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1c, 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),
-	{ } /* end */
-};
-
 static struct snd_kcontrol_new stac927x_loopback[] = {
 	STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1),
 	{}
@@ -1309,9 +1293,11 @@ static int stac92xx_build_controls(struct hda_codec *codec)
 	int err;
 	int i;
 
-	err = snd_hda_add_new_ctls(codec, spec->mixer);
-	if (err < 0)
-		return err;
+	if (spec->mixer) {
+		err = snd_hda_add_new_ctls(codec, spec->mixer);
+		if (err < 0)
+			return err;
+	}
 
 	for (i = 0; i < spec->num_mixers; i++) {
 		err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
@@ -3317,6 +3303,21 @@ static int create_multi_out_ctls(struct hda_codec *codec, int num_outs,
 	return 0;
 }
 
+static int stac92xx_add_capvol_ctls(struct hda_codec *codec, unsigned long vol,
+				    unsigned long sw, int idx)
+{
+	int err;
+	err = stac92xx_add_control_idx(codec->spec, STAC_CTL_WIDGET_VOL, idx,
+				       "Captuer Volume", vol);
+	if (err < 0)
+		return err;
+	err = stac92xx_add_control_idx(codec->spec, STAC_CTL_WIDGET_MUTE, idx,
+				       "Captuer Switch", sw);
+	if (err < 0)
+		return err;
+	return 0;
+}
+
 /* add playback controls from the parsed DAC table */
 static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
 					       const struct auto_pin_cfg *cfg)
@@ -3703,7 +3704,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 {
 	struct sigmatel_spec *spec = codec->spec;
 	int hp_swap = 0;
-	int err;
+	int i, err;
 
 	if ((err = snd_hda_parse_pin_def_config(codec,
 						&spec->autocfg,
@@ -3837,6 +3838,13 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 		spec->autocfg.line_outs = 0;
 	}
 
+	for (i = 0; i < spec->num_caps; i++) {
+		err = stac92xx_add_capvol_ctls(codec, spec->capvols[i],
+					       spec->capsws[i], i);
+		if (err < 0)
+			return err;
+	}
+
 	err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
@@ -4898,6 +4906,9 @@ static int patch_stac925x(struct hda_codec *codec)
 
 	spec->init = stac925x_core_init;
 	spec->mixer = stac925x_mixer;
+	spec->num_caps = 1;
+	spec->capvols = stac925x_capvols;
+	spec->capsws = stac925x_capsws;
 
 	err = stac92xx_parse_auto_config(codec, 0x8, 0x7);
 	if (!err) {
@@ -5001,6 +5012,10 @@ again:
 	memcpy(&spec->private_dimux, &stac92hd73xx_dmux,
 			sizeof(stac92hd73xx_dmux));
 
+	spec->num_caps = STAC92HD73XX_NUM_CAPS;
+	spec->capvols = stac92hd73xx_capvols;
+	spec->capsws = stac92hd73xx_capsws;
+
 	switch (spec->board_config) {
 	case STAC_DELL_EQ:
 		spec->init = dell_eq_core_init;
@@ -5120,6 +5135,10 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 	spec->num_dmics = STAC92HD83XXX_NUM_DMICS;
 	spec->dinput_mux = &stac92hd83xxx_dmux;
 	spec->pin_nids = stac92hd83xxx_pin_nids;
+	spec->num_caps = STAC92HD83XXX_NUM_CAPS;
+	spec->capvols = stac92hd83xxx_capvols;
+	spec->capsws = stac92hd83xxx_capsws;
+
 	spec->board_config = snd_hda_check_board_config(codec,
 							STAC_92HD83XXX_MODELS,
 							stac92hd83xxx_models,
@@ -5308,6 +5327,10 @@ again:
 	spec->dmic_nids = stac92hd71bxx_dmic_nids;
 	spec->dmux_nids = stac92hd71bxx_dmux_nids;
 
+	spec->num_caps = STAC92HD71BXX_NUM_CAPS;
+	spec->capvols = stac92hd71bxx_capvols;
+	spec->capsws = stac92hd71bxx_capsws;
+
 	switch (codec->vendor_id) {
 	case 0x111d76b6: /* 4 Port without Analog Mixer */
 	case 0x111d76b7:
@@ -5317,7 +5340,6 @@ again:
 	case 0x111d76b5:
 		memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_nomixer,
 		       sizeof(stac92hd71bxx_dmux_nomixer));
-		spec->mixer = stac92hd71bxx_mixer;
 		spec->init = stac92hd71bxx_core_init;
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
 		spec->num_dmics = stac92hd71bxx_connected_ports(codec,
@@ -5566,7 +5588,10 @@ static int patch_stac922x(struct hda_codec *codec)
 	spec->num_pwrs = 0;
 
 	spec->init = stac922x_core_init;
-	spec->mixer = stac922x_mixer;
+
+	spec->num_caps = STAC922X_NUM_CAPS;
+	spec->capvols = stac922x_capvols;
+	spec->capsws = stac922x_capsws;
 
 	spec->multiout.dac_nids = spec->dac_nids;
 	
@@ -5641,7 +5666,6 @@ static int patch_stac927x(struct hda_codec *codec)
 		spec->num_dmics = 0;
 
 		spec->init = d965_core_init;
-		spec->mixer = stac927x_mixer;
 		break;
 	case STAC_DELL_BIOS:
 		switch (codec->subsystem_id) {
@@ -5666,7 +5690,6 @@ static int patch_stac927x(struct hda_codec *codec)
 		spec->num_dmics = STAC927X_NUM_DMICS;
 
 		spec->init = d965_core_init;
-		spec->mixer = stac927x_mixer;
 		spec->dmux_nids = stac927x_dmux_nids;
 		spec->num_dmuxes = ARRAY_SIZE(stac927x_dmux_nids);
 		break;
@@ -5679,9 +5702,12 @@ static int patch_stac927x(struct hda_codec *codec)
 		spec->num_dmics = 0;
 
 		spec->init = stac927x_core_init;
-		spec->mixer = stac927x_mixer;
 	}
 
+	spec->num_caps = STAC927X_NUM_CAPS;
+	spec->capvols = stac927x_capvols;
+	spec->capsws = stac927x_capsws;
+
 	spec->num_pwrs = 0;
 	spec->aloopback_ctl = stac927x_loopback;
 	spec->aloopback_mask = 0x40;
@@ -5763,9 +5789,12 @@ static int patch_stac9205(struct hda_codec *codec)
 	spec->num_pwrs = 0;
 
 	spec->init = stac9205_core_init;
-	spec->mixer = stac9205_mixer;
 	spec->aloopback_ctl = stac9205_loopback;
 
+	spec->num_caps = STAC9205_NUM_CAPS;
+	spec->capvols = stac9205_capvols;
+	spec->capsws = stac9205_capsws;
+
 	spec->aloopback_mask = 0x40;
 	spec->aloopback_shift = 0;
 	/* Turn on/off EAPD per HP plugging */
@@ -5840,12 +5869,6 @@ static struct hda_verb stac9872_core_init[] = {
 	{}
 };
 
-static struct snd_kcontrol_new stac9872_mixer[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT),
-	{ } /* end */
-};
-
 static hda_nid_t stac9872_pin_nids[] = {
 	0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f,
 	0x11, 0x13, 0x14,
@@ -5859,6 +5882,11 @@ static hda_nid_t stac9872_mux_nids[] = {
 	0x15
 };
 
+static unsigned long stac9872_capvols[] = {
+	HDA_COMPOSE_AMP_VAL(0x09, 3, 0, HDA_INPUT),
+};
+#define stac9872_capsws		stac9872_capvols
+
 static unsigned int stac9872_vaio_pin_configs[9] = {
 	0x03211020, 0x411111f0, 0x411111f0, 0x03a15030,
 	0x411111f0, 0x90170110, 0x411111f0, 0x411111f0,
@@ -5907,8 +5935,10 @@ static int patch_stac9872(struct hda_codec *codec)
 	spec->adc_nids = stac9872_adc_nids;
 	spec->num_muxes = ARRAY_SIZE(stac9872_mux_nids);
 	spec->mux_nids = stac9872_mux_nids;
-	spec->mixer = stac9872_mixer;
 	spec->init = stac9872_core_init;
+	spec->num_caps = 1;
+	spec->capvols = stac9872_capvols;
+	spec->capsws = stac9872_capsws;
 
 	err = stac92xx_parse_auto_config(codec, 0x10, 0x12);
 	if (err < 0) {
-- 
cgit v1.2.3


From 50c62f068ee67b5a0178855f502f4ea2ee931eed Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 28 Jul 2009 18:25:29 +0200
Subject: ALSA: hda - Don't create analog mixer for IDT92HD71bxx

The analog mixer unit on IDT 92HD71Bxx codecs is almost useless
since we use only the direct connections from DAC to pin.

Remove the controls to avoid unneeded confusion as default now.
This can be still back via "analog_mixer = 1" hint.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index be6cf2cfb19..66b1f3cfc87 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5376,7 +5376,8 @@ again:
 
 		/* no output amps */
 		spec->num_pwrs = 0;
-		spec->mixer = stac92hd71bxx_analog_mixer;
+		if (snd_hda_get_bool_hint(codec, "analog_mixer") == 1)
+			spec->mixer = stac92hd71bxx_analog_mixer;
 
 		/* disable VSW */
 		spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF];
@@ -5403,7 +5404,8 @@ again:
 	default:
 		memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer,
 		       sizeof(stac92hd71bxx_dmux_amixer));
-		spec->mixer = stac92hd71bxx_analog_mixer;
+		if (snd_hda_get_bool_hint(codec, "analog_mixer") == 1)
+			spec->mixer = stac92hd71bxx_analog_mixer;
 		spec->init = stac92hd71bxx_analog_core_init;
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
 		spec->num_dmics = stac92hd71bxx_connected_ports(codec,
-- 
cgit v1.2.3


From 62558ce15759ee93223132258588320967e1e521 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 29 Jul 2009 14:23:09 +0200
Subject: ALSA: hda - Avoid overwrite of jack events with STAC/IDT

Since only one event can be associated to a (pin) widget, it's safer
to avoid the multiple mapping.  This patch fixes the behavior of the
STAC/IDT codec driver.

Now stac_get_event() doesn't take the type argument but simply returns
the first hit element.  Then enable_pin_detect() checks the validity
of the type, and returns non-zero only if a valid entry.  The caller
can call stac_issue_unsol_event() after checking the return value.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 59 +++++++++++++++++++++---------------------
 1 file changed, 30 insertions(+), 29 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 66b1f3cfc87..307e86ceede 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2620,8 +2620,7 @@ static int stac92xx_hp_switch_get(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid,
-				   unsigned char type);
+static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid);
 
 static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol,
 			struct snd_ctl_elem_value *ucontrol)
@@ -2635,7 +2634,7 @@ static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol,
 	/* check to be sure that the ports are upto date with
 	 * switch changes
 	 */
-	stac_issue_unsol_event(codec, nid, STAC_HP_EVENT);
+	stac_issue_unsol_event(codec, nid);
 
 	return 1;
 }
@@ -2768,7 +2767,7 @@ static int stac92xx_io_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
 	 * appropriately according to the pin direction
 	 */
 	if (spec->hp_detect)
-		stac_issue_unsol_event(codec, nid, STAC_HP_EVENT);
+		stac_issue_unsol_event(codec, nid);
 
         return 1;
 }
@@ -4107,14 +4106,14 @@ static int stac_add_event(struct sigmatel_spec *spec, hda_nid_t nid,
 }
 
 static struct sigmatel_event *stac_get_event(struct hda_codec *codec,
-					     hda_nid_t nid, unsigned char type)
+					     hda_nid_t nid)
 {
 	struct sigmatel_spec *spec = codec->spec;
 	struct sigmatel_event *event = spec->events.list;
 	int i;
 
 	for (i = 0; i < spec->events.used; i++, event++) {
-		if (event->nid == nid && event->type == type)
+		if (event->nid == nid)
 			return event;
 	}
 	return NULL;
@@ -4134,24 +4133,32 @@ static struct sigmatel_event *stac_get_event_from_tag(struct hda_codec *codec,
 	return NULL;
 }
 
-static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid,
-			      unsigned int type)
+/* check if given nid is a valid pin and no other events are assigned
+ * to it.  If OK, assign the event, set the unsol flag, and returns 1.
+ * Otherwise, returns zero.
+ */
+static int enable_pin_detect(struct hda_codec *codec, hda_nid_t nid,
+			     unsigned int type)
 {
 	struct sigmatel_event *event;
 	int tag;
 
 	if (!(get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP))
-		return;
-	event = stac_get_event(codec, nid, type);
-	if (event)
+		return 0;
+	event = stac_get_event(codec, nid);
+	if (event) {
+		if (event->type != type)
+			return 0;
 		tag = event->tag;
-	else
+	} else {
 		tag = stac_add_event(codec->spec, nid, type, 0);
-	if (tag < 0)
-		return;
+		if (tag < 0)
+			return 0;
+	}
 	snd_hda_codec_write_cache(codec, nid, 0,
 				  AC_VERB_SET_UNSOLICITED_ENABLE,
 				  AC_USRSP_EN | tag);
+	return 1;
 }
 
 static int is_nid_hp_pin(struct auto_pin_cfg *cfg, hda_nid_t nid)
@@ -4250,8 +4257,7 @@ static int stac92xx_init(struct hda_codec *codec)
 		stac92xx_auto_set_pinctl(codec, spec->autocfg.line_out_pins[0],
 				AC_PINCTL_OUT_EN);
 		/* fake event to set up pins */
-		stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0],
-				       STAC_HP_EVENT);
+		stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0]);
 	} else {
 		stac92xx_auto_init_multi_out(codec);
 		stac92xx_auto_init_hp_out(codec);
@@ -4284,10 +4290,9 @@ static int stac92xx_init(struct hda_codec *codec)
 			}
 			conf = snd_hda_codec_get_pincfg(codec, nid);
 			if (get_defcfg_connect(conf) != AC_JACK_PORT_FIXED) {
-				enable_pin_detect(codec, nid,
-						  STAC_INSERT_EVENT);
-				stac_issue_unsol_event(codec, nid,
-						       STAC_INSERT_EVENT);
+				if (enable_pin_detect(codec, nid,
+						      STAC_INSERT_EVENT))
+					stac_issue_unsol_event(codec, nid);
 			}
 		}
 	}
@@ -4332,10 +4337,8 @@ static int stac92xx_init(struct hda_codec *codec)
 				stac_toggle_power_map(codec, nid, 1);
 			continue;
 		}
-		if (!stac_get_event(codec, nid, STAC_INSERT_EVENT)) {
-			enable_pin_detect(codec, nid, STAC_PWR_EVENT);
-			stac_issue_unsol_event(codec, nid, STAC_PWR_EVENT);
-		}
+		if (enable_pin_detect(codec, nid, STAC_PWR_EVENT))
+			stac_issue_unsol_event(codec, nid);
 	}
 	if (spec->dac_list)
 		stac92xx_power_down(codec);
@@ -4598,10 +4601,9 @@ static void stac92xx_report_jack(struct hda_codec *codec, hda_nid_t nid)
 	}
 }
 
-static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid,
-				   unsigned char type)
+static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid)
 {
-	struct sigmatel_event *event = stac_get_event(codec, nid, type);
+	struct sigmatel_event *event = stac_get_event(codec, nid);
 	if (!event)
 		return;
 	codec->patch_ops.unsol_event(codec, (unsigned)event->tag << 26);
@@ -4712,8 +4714,7 @@ static int stac92xx_resume(struct hda_codec *codec)
 	snd_hda_codec_resume_cache(codec);
 	/* fake event to set up pins again to override cached values */
 	if (spec->hp_detect)
-		stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0],
-				       STAC_HP_EVENT);
+		stac_issue_unsol_event(codec, spec->autocfg.hp_pins[0]);
 	return 0;
 }
 
-- 
cgit v1.2.3


From 3d21d3f7e7032619f5c5b47d3ee23bbe45de5993 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 29 Jul 2009 14:32:56 +0200
Subject: ALSA: hda - Support auto-mic switching with IDT/STAC codec

Support the automatic mic-switching with some devices with IDT/STAC
codecs.  The condition is that the device has only two inputs, one
for an external mic and one for an internal mic.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 170 +++++++++++++++++++++++++++++++++++++----
 1 file changed, 154 insertions(+), 16 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 307e86ceede..f39dc98519a 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -40,6 +40,7 @@ enum {
 	STAC_INSERT_EVENT,
 	STAC_PWR_EVENT,
 	STAC_HP_EVENT,
+	STAC_MIC_EVENT,
 };
 
 enum {
@@ -176,6 +177,12 @@ struct sigmatel_jack {
 	struct snd_jack *jack;
 };
 
+struct sigmatel_mic_route {
+	hda_nid_t pin;
+	unsigned char mux_idx;
+	unsigned char dmux_idx;
+};
+
 struct sigmatel_spec {
 	struct snd_kcontrol_new *mixers[4];
 	unsigned int num_mixers;
@@ -187,6 +194,7 @@ struct sigmatel_spec {
 	unsigned int hp_detect: 1;
 	unsigned int spdif_mute: 1;
 	unsigned int check_volume_offset:1;
+	unsigned int auto_mic:1;
 
 	/* gpio lines */
 	unsigned int eapd_mask;
@@ -243,6 +251,9 @@ struct sigmatel_spec {
 	unsigned long *capsws; /* amp-mute attr: HDA_COMPOSE_AMP_VAL() */
 	unsigned int num_caps; /* number of capture volume/switch elements */
 
+	struct sigmatel_mic_route ext_mic;
+	struct sigmatel_mic_route int_mic;
+
 	const char **spdif_labels;
 
 	hda_nid_t dig_in_nid;
@@ -1304,7 +1315,7 @@ static int stac92xx_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 	}
-	if (spec->num_dmuxes > 0) {
+	if (!spec->auto_mic && spec->num_dmuxes > 0) {
 		stac_dmux_mixer.count = spec->num_dmuxes;
 		err = snd_hda_ctl_add(codec,
 				  snd_ctl_new1(&stac_dmux_mixer, codec));
@@ -2950,6 +2961,8 @@ static int stac92xx_add_input_source(struct sigmatel_spec *spec)
 	struct snd_kcontrol_new *knew;
 	struct hda_input_mux *imux = &spec->private_imux;
 
+	if (spec->auto_mic)
+		return 0; /* no need for input source */
 	if (!spec->num_adcs || imux->num_items <= 1)
 		return 0; /* no need for input source control */
 	knew = stac_control_new(spec, &stac_input_src_temp,
@@ -3557,14 +3570,26 @@ static const char *stac92xx_dmic_labels[5] = {
 	"Digital Mic 3", "Digital Mic 4"
 };
 
+static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
+				hda_nid_t nid)
+{
+	hda_nid_t conn[HDA_MAX_NUM_INPUTS];
+	int i, nums;
+
+	nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
+	for (i = 0; i < nums; i++)
+		if (conn[i] == nid)
+			return i;
+	return -1;
+}
+
 /* create playback/capture controls for input pins on dmic capable codecs */
 static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
 	struct sigmatel_spec *spec = codec->spec;
 	struct hda_input_mux *dimux = &spec->private_dimux;
-	hda_nid_t con_lst[HDA_MAX_NUM_INPUTS];
-	int err, i, j;
+	int err, i;
 	char name[32];
 
 	dimux->items[dimux->num_items].label = stac92xx_dmic_labels[0];
@@ -3574,7 +3599,6 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
 	for (i = 0; i < spec->num_dmics; i++) {
 		hda_nid_t nid;
 		int index;
-		int num_cons;
 		unsigned int wcaps;
 		unsigned int def_conf;
 
@@ -3583,17 +3607,10 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
 			continue;
 
 		nid = spec->dmic_nids[i];
-		num_cons = snd_hda_get_connections(codec,
-				spec->dmux_nids[0],
-				con_lst,
-				HDA_MAX_NUM_INPUTS);
-		for (j = 0; j < num_cons; j++)
-			if (con_lst[j] == nid) {
-				index = j;
-				goto found;
-			}
-		continue;
-found:
+		index = get_connection_index(codec, spec->dmux_nids[0], nid);
+		if (index < 0)
+			continue;
+
 		wcaps = get_wcaps(codec, nid) &
 			(AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP);
 
@@ -3620,6 +3637,88 @@ found:
 	return 0;
 }
 
+static int check_mic_pin(struct hda_codec *codec, hda_nid_t nid,
+			 hda_nid_t *fixed, hda_nid_t *ext)
+{
+	unsigned int cfg;
+
+	if (!nid)
+		return 0;
+	cfg = snd_hda_codec_get_pincfg(codec, nid);
+	switch (get_defcfg_connect(cfg)) {
+	case AC_JACK_PORT_FIXED:
+		if (*fixed)
+			return 1; /* already occupied */
+		*fixed = nid;
+		break;
+	case AC_JACK_PORT_COMPLEX:
+		if (*ext)
+			return 1; /* already occupied */
+		*ext = nid;
+		break;
+	}
+	return 0;
+}
+
+static int set_mic_route(struct hda_codec *codec,
+			 struct sigmatel_mic_route *mic,
+			 hda_nid_t pin)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i;
+
+	mic->pin = pin;
+	for (i = AUTO_PIN_MIC; i <= AUTO_PIN_FRONT_MIC; i++)
+		if (pin == cfg->input_pins[i])
+			break;
+	if (i <= AUTO_PIN_FRONT_MIC) {
+		/* analog pin */
+		mic->dmux_idx = 0;
+		i = get_connection_index(codec, spec->mux_nids[0], pin);
+		if (i < 0)
+			return -1;
+		mic->mux_idx = i;
+	}  else {
+		/* digital pin */
+		mic->mux_idx = 0;
+		i = get_connection_index(codec, spec->dmux_nids[0], pin);
+		if (i < 0)
+			return -1;
+		mic->dmux_idx = i;
+	}
+	return 0;
+}
+
+/* return non-zero if the device is for automatic mic switch */
+static int stac_check_auto_mic(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	hda_nid_t fixed, ext;
+	int i;
+
+	for (i = AUTO_PIN_LINE; i < AUTO_PIN_LAST; i++) {
+		if (cfg->input_pins[i])
+			return 0; /* must be exclusively mics */
+	}
+	fixed = ext = 0;
+	for (i = AUTO_PIN_MIC; i <= AUTO_PIN_FRONT_MIC; i++)
+		if (check_mic_pin(codec, cfg->input_pins[i], &fixed, &ext))
+			return 0;
+	for (i = 0; i < spec->num_dmics; i++)
+		if (check_mic_pin(codec, spec->dmic_nids[i], &fixed, &ext))
+			return 0;
+	if (!fixed || !ext)
+		return 0;
+	if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP))
+		return 0; /* no unsol support */
+	if (set_mic_route(codec, &spec->ext_mic, ext) ||
+	    set_mic_route(codec, &spec->int_mic, fixed))
+		return 0; /* something is wrong */
+	return 1;
+}
+
 /* create playback/capture controls for input pins */
 static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg)
 {
@@ -3837,6 +3936,14 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 		spec->autocfg.line_outs = 0;
 	}
 
+	if (stac_check_auto_mic(codec)) {
+		spec->auto_mic = 1;
+		/* only one capture for auto-mic */
+		spec->num_adcs = 1;
+		spec->num_caps = 1;
+		spec->num_muxes = 1;
+	}
+
 	for (i = 0; i < spec->num_caps; i++) {
 		err = stac92xx_add_capvol_ctls(codec, spec->capvols[i],
 					       spec->capsws[i], i);
@@ -4264,6 +4371,10 @@ static int stac92xx_init(struct hda_codec *codec)
 		for (i = 0; i < cfg->hp_outs; i++)
 			stac_toggle_power_map(codec, cfg->hp_pins[i], 1);
 	}
+	if (spec->auto_mic) {
+		if (enable_pin_detect(codec, spec->ext_mic.pin, STAC_MIC_EVENT))
+			stac_issue_unsol_event(codec, spec->ext_mic.pin);
+	}
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = cfg->input_pins[i];
 		if (nid) {
@@ -4601,6 +4712,25 @@ static void stac92xx_report_jack(struct hda_codec *codec, hda_nid_t nid)
 	}
 }
 
+static void stac92xx_mic_detect(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	struct sigmatel_mic_route *mic;
+
+	if (get_pin_presence(codec, spec->ext_mic.pin))
+		mic = &spec->ext_mic;
+	else
+		mic = &spec->int_mic;
+	if (mic->dmux_idx)
+		snd_hda_codec_write_cache(codec, spec->dmux_nids[0], 0,
+					  AC_VERB_SET_CONNECT_SEL,
+					  mic->dmux_idx);
+	else
+		snd_hda_codec_write_cache(codec, spec->mux_nids[0], 0,
+					  AC_VERB_SET_CONNECT_SEL,
+					  mic->mux_idx);
+}
+
 static void stac_issue_unsol_event(struct hda_codec *codec, hda_nid_t nid)
 {
 	struct sigmatel_event *event = stac_get_event(codec, nid);
@@ -4623,7 +4753,15 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
 	switch (event->type) {
 	case STAC_HP_EVENT:
 		stac92xx_hp_detect(codec);
-		/* fallthru */
+		break;
+	case STAC_MIC_EVENT:
+		stac92xx_mic_detect(codec);
+		break;
+	}
+
+	switch (event->type) {
+	case STAC_HP_EVENT:
+	case STAC_MIC_EVENT:
 	case STAC_INSERT_EVENT:
 	case STAC_PWR_EVENT:
 		if (spec->num_pwrs > 0)
-- 
cgit v1.2.3


From 26a2798053519d766ddae1ca1b03c8295fddf1a8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 29 Jul 2009 16:28:09 +0200
Subject: ALSA: hda - Remove static connection in IDT 92HD71bxx

We don't need any more static connection to the port F (which is often
used for docking stations) since its connection is done dynamically via
DAC assignment now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 17 ++---------------
 1 file changed, 2 insertions(+), 15 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index f39dc98519a..abc44dbe0ab 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -997,19 +997,6 @@ static struct hda_verb stac92hd71bxx_core_init[] = {
 	{}
 };
 
-#define HD_DISABLE_PORTF 1
-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},
-	/* start of config #2 */
-
-	/* set master volume and direct control */
-	{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
-	{}
-};
-
 static struct hda_verb stac92hd71bxx_unmute_core_init[] = {
 	/* unmute right and left channels for nodes 0x0f, 0xa, 0x0d */
 	{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -5519,7 +5506,7 @@ again:
 			spec->mixer = stac92hd71bxx_analog_mixer;
 
 		/* disable VSW */
-		spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF];
+		spec->init = stac92hd71bxx_core_init;
 		unmute_init++;
 		snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0);
 		snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3);
@@ -5545,7 +5532,7 @@ again:
 		       sizeof(stac92hd71bxx_dmux_amixer));
 		if (snd_hda_get_bool_hint(codec, "analog_mixer") == 1)
 			spec->mixer = stac92hd71bxx_analog_mixer;
-		spec->init = stac92hd71bxx_analog_core_init;
+		spec->init = stac92hd71bxx_core_init;
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
 		spec->num_dmics = stac92hd71bxx_connected_ports(codec,
 					stac92hd71bxx_dmic_nids,
-- 
cgit v1.2.3


From 15b4f296fce683497ecc815b2f9b6f121fb3fef8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 29 Jul 2009 16:32:55 +0200
Subject: ALSA: hda - Add missing DMUX initialization for auto-mic with
 STAC/IDT

Added the missing initialization of DMUX connection (to analog input)
for auto-mic mode with STAC/IDT codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 3 +++
 1 file changed, 3 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index abc44dbe0ab..2405f8418a7 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4359,6 +4359,9 @@ static int stac92xx_init(struct hda_codec *codec)
 			stac_toggle_power_map(codec, cfg->hp_pins[i], 1);
 	}
 	if (spec->auto_mic) {
+		/* initialize connection to analog input */
+		snd_hda_codec_write_cache(codec, spec->dmux_nids[0], 0,
+					  AC_VERB_SET_CONNECT_SEL, 0);
 		if (enable_pin_detect(codec, spec->ext_mic.pin, STAC_MIC_EVENT))
 			stac_issue_unsol_event(codec, spec->ext_mic.pin);
 	}
-- 
cgit v1.2.3


From 71443b0b745f7d2c775c8a228e09ef2dc98d42f3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 29 Jul 2009 18:41:29 +0200
Subject: ALSA: hda - No analog mix input source as default for IDT92HD71bxx

The analog mix is disabled now as default (unless "analog_mixer" hint
is given), so it shoudn't appear in the digital input source as well.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 23 ++++++++++++++++-------
 1 file changed, 16 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 2405f8418a7..883e2aa1007 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5481,8 +5481,6 @@ again:
 		}
 		break;
 	case 0x111d7608: /* 5 Port with Analog Mixer */
-		memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer,
-		       sizeof(stac92hd71bxx_dmux_amixer));
 		spec->private_dimux.num_items--;
 		switch (spec->board_config) {
 		case STAC_HP_M4:
@@ -5505,9 +5503,15 @@ again:
 
 		/* no output amps */
 		spec->num_pwrs = 0;
-		if (snd_hda_get_bool_hint(codec, "analog_mixer") == 1)
+		if (snd_hda_get_bool_hint(codec, "analog_mixer") == 1) {
 			spec->mixer = stac92hd71bxx_analog_mixer;
-
+			memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer,
+			       sizeof(stac92hd71bxx_dmux_amixer));
+		} else {
+			memcpy(&spec->private_dimux,
+			       &stac92hd71bxx_dmux_nomixer,
+			       sizeof(stac92hd71bxx_dmux_nomixer));
+		}
 		/* disable VSW */
 		spec->init = stac92hd71bxx_core_init;
 		unmute_init++;
@@ -5531,10 +5535,15 @@ again:
 		spec->num_pwrs = 0;
 		/* fallthru */
 	default:
-		memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer,
-		       sizeof(stac92hd71bxx_dmux_amixer));
-		if (snd_hda_get_bool_hint(codec, "analog_mixer") == 1)
+		if (snd_hda_get_bool_hint(codec, "analog_mixer") == 1) {
 			spec->mixer = stac92hd71bxx_analog_mixer;
+			memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer,
+			       sizeof(stac92hd71bxx_dmux_amixer));
+		} else {
+			memcpy(&spec->private_dimux,
+			       &stac92hd71bxx_dmux_nomixer,
+			       sizeof(stac92hd71bxx_dmux_nomixer));
+		}
 		spec->init = stac92hd71bxx_core_init;
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
 		spec->num_dmics = stac92hd71bxx_connected_ports(codec,
-- 
cgit v1.2.3


From dd7714c92248943b778e8199c7dec7ef859dcac2 Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Thu, 30 Jul 2009 14:36:35 +0800
Subject: ALSA: hda: add 2-channel mode to Intel ALC889/ALC889A models

This 2-channel mode is useful in that it will broadcast
a 2-channel audio stream to all front/side/... ports.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 44 ++++++++++++++++++++++++-------------------
 1 file changed, 25 insertions(+), 19 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6794d028119..8c5d342bd4e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6680,19 +6680,28 @@ static struct hda_channel_mode alc883_3ST_6ch_intel_modes[3] = {
 	{ 6, alc883_3ST_ch6_intel_init },
 };
 
+/*
+ * 2ch mode
+ */
+static struct hda_verb alc889_ch2_intel_init[] = {
+	{ 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	{ 0x19, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	{ 0x16, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ } /* end */
+};
+
 /*
  * 6ch mode
  */
 static struct hda_verb alc889_ch6_intel_init[] = {
-	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+	{ 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	{ 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 },
+	{ 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 },
+	{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
 	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
 	{ } /* end */
 };
@@ -6701,21 +6710,18 @@ static struct hda_verb alc889_ch6_intel_init[] = {
  * 8ch mode
  */
 static struct hda_verb alc889_ch8_intel_init[] = {
-	{ 0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x14, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	{ 0x19, AC_VERB_SET_CONNECT_SEL, 0x01 },
+	{ 0x16, AC_VERB_SET_CONNECT_SEL, 0x02 },
+	{ 0x17, AC_VERB_SET_CONNECT_SEL, 0x03 },
+	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x03 },
 	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x03 },
 	{ } /* end */
 };
 
-static struct hda_channel_mode alc889_8ch_intel_modes[2] = {
+static struct hda_channel_mode alc889_8ch_intel_modes[3] = {
+	{ 2, alc889_ch2_intel_init },
 	{ 6, alc889_ch6_intel_init },
 	{ 8, alc889_ch8_intel_init },
 };
-- 
cgit v1.2.3


From 6732bd0d15893d20359396309adc238e0af10b58 Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Thu, 30 Jul 2009 09:19:14 +0200
Subject: ALSA: hda: add HP automute support to Intel ALC889/ALC889A models

It auto mutes all 8-channel outputs at rear panel when
the front panel headphone is connected.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 47 ++++++++++++++++++++++++++++++++-----------
 1 file changed, 35 insertions(+), 12 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8c5d342bd4e..587d94f869b 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1436,6 +1436,25 @@ static void alc_automute_amp_unsol_event(struct hda_codec *codec,
 		alc_automute_amp(codec);
 }
 
+static void alc889_automute_init(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	spec->autocfg.hp_pins[0] = 0x15;
+	spec->autocfg.speaker_pins[0] = 0x14;
+	spec->autocfg.speaker_pins[1] = 0x16;
+	spec->autocfg.speaker_pins[2] = 0x17;
+	spec->autocfg.speaker_pins[3] = 0x19;
+	spec->autocfg.speaker_pins[4] = 0x1a;
+	alc_automute_amp(codec);
+}
+
+static void alc889_intel_init_hook(struct hda_codec *codec)
+{
+	alc889_coef_init(codec);
+	alc889_automute_init(codec);
+}
+
 static void alc888_fujitsu_xa3530_init_hook(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -6993,6 +7012,11 @@ static struct hda_verb alc889_eapd_verbs[] = {
 	{ }
 };
 
+static struct hda_verb alc_hp15_unsol_verbs[] = {
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{}
+};
 
 static struct hda_verb alc885_init_verbs[] = {
 	/* Front mixer: unmute input/output amp left and right (volume = 0) */
@@ -7018,7 +7042,7 @@ static struct hda_verb alc885_init_verbs[] = {
 	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
 
 	/* Front HP Pin: output 0 (0x0c) */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{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, 0x00},
 	/* Front Pin: output 0 (0x0c) */
@@ -8860,7 +8884,8 @@ static struct alc_config_preset alc882_presets[] = {
 	},
 	[ALC889A_INTEL] = {
 		.mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer },
-		.init_verbs = { alc885_init_verbs, alc885_init_input_verbs },
+		.init_verbs = { alc885_init_verbs, alc885_init_input_verbs,
+				alc_hp15_unsol_verbs },
 		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
 		.dac_nids = alc883_dac_nids,
 		.num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
@@ -8872,12 +8897,14 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc889_8ch_intel_modes,
 		.capsrc_nids = alc889_capsrc_nids,
 		.input_mux = &alc889_capture_source,
+		.init_hook = alc889_automute_init,
+		.unsol_event = alc_automute_amp_unsol_event,
 		.need_dac_fix = 1,
 	},
 	[ALC889_INTEL] = {
 		.mixers = { alc885_8ch_intel_mixer, alc883_chmode_mixer },
 		.init_verbs = { alc885_init_verbs, alc889_init_input_verbs,
-				alc889_eapd_verbs },
+				alc889_eapd_verbs, alc_hp15_unsol_verbs},
 		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
 		.dac_nids = alc883_dac_nids,
 		.num_adc_nids = ARRAY_SIZE(alc889_adc_nids),
@@ -8889,7 +8916,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc889_8ch_intel_modes,
 		.capsrc_nids = alc889_capsrc_nids,
 		.input_mux = &alc889_capture_source,
-		.init_hook = alc889_coef_init,
+		.init_hook = alc889_intel_init_hook,
+		.unsol_event = alc_automute_amp_unsol_event,
 		.need_dac_fix = 1,
 	},
 	[ALC883_6ST_DIG] = {
@@ -10121,12 +10149,6 @@ static struct hda_verb alc262_eapd_verbs[] = {
 	{ }
 };
 
-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},
-	{}
-};
-
 static struct hda_verb alc262_hippo1_unsol_verbs[] = {
 	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0},
 	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
@@ -11168,7 +11190,7 @@ static struct alc_config_preset alc262_presets[] = {
 	},
 	[ALC262_HIPPO] = {
 		.mixers = { alc262_hippo_mixer },
-		.init_verbs = { alc262_init_verbs, alc262_hippo_unsol_verbs},
+		.init_verbs = { alc262_init_verbs, alc_hp15_unsol_verbs},
 		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
 		.dac_nids = alc262_dac_nids,
 		.hp_nid = 0x03,
@@ -11288,7 +11310,8 @@ static struct alc_config_preset alc262_presets[] = {
 	},
 	[ALC262_BENQ_T31] = {
 		.mixers = { alc262_benq_t31_mixer },
-		.init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs, alc262_hippo_unsol_verbs },
+		.init_verbs = { alc262_init_verbs, alc262_benq_t31_EAPD_verbs,
+				alc_hp15_unsol_verbs },
 		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
 		.dac_nids = alc262_dac_nids,
 		.hp_nid = 0x03,
-- 
cgit v1.2.3


From bf677bd8fbd5dc06ead5511cf9fa76d090856c6a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 30 Jul 2009 09:24:29 +0200
Subject: ALSA: hda - Fix typos of Capture controls.

The commit 6479c63188290beae83ade3243b9d6eb47d394b6
    ALSA: hda - Create Capture controls dynamically
introduced typos of "Capture".  Fixed now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 883e2aa1007..9e192529e80 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -3307,11 +3307,11 @@ static int stac92xx_add_capvol_ctls(struct hda_codec *codec, unsigned long vol,
 {
 	int err;
 	err = stac92xx_add_control_idx(codec->spec, STAC_CTL_WIDGET_VOL, idx,
-				       "Captuer Volume", vol);
+				       "Capture Volume", vol);
 	if (err < 0)
 		return err;
 	err = stac92xx_add_control_idx(codec->spec, STAC_CTL_WIDGET_MUTE, idx,
-				       "Captuer Switch", sw);
+				       "Capture Switch", sw);
 	if (err < 0)
 		return err;
 	return 0;
-- 
cgit v1.2.3


From 5207e10ed49c515e7432d0b1a7561ccc7b71f9df Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 30 Jul 2009 13:09:08 +0200
Subject: ALSA: hda - Integrate Digital Input Source to Input Source

STAC/IDT codecs provide both "Input Source" and "Digital Input Source"
controls to choose the analog input source and the digital input source.
But this is far user-unfriendly.

This patch merges the input source selections into one "Input Source"
control.  To have separate digital and analog input source controls,
you can pass "separate_dmux = 1 " hint string.

At the same time, this patch gets rid of analog mixer stuff that was
already disabled in previous patches.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 168 +++++++++++++++--------------------------
 1 file changed, 59 insertions(+), 109 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 883e2aa1007..3457f619bad 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -246,6 +246,7 @@ struct sigmatel_spec {
 	unsigned int num_dmuxes;
 	hda_nid_t *smux_nids;
 	unsigned int num_smuxes;
+	unsigned int num_analog_muxes;
 
 	unsigned long *capvols; /* amp-volume attr: HDA_COMPOSE_AMP_VAL() */
 	unsigned long *capsws; /* amp-mute attr: HDA_COMPOSE_AMP_VAL() */
@@ -765,9 +766,35 @@ static int stac92xx_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct sigmatel_spec *spec = codec->spec;
 	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
-
-	return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol,
-				     spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]);
+	const struct hda_input_mux *imux = spec->input_mux;
+	unsigned int idx, prev_idx;
+
+	idx = ucontrol->value.enumerated.item[0];
+	if (idx >= imux->num_items)
+		idx = imux->num_items - 1;
+	prev_idx = spec->cur_mux[adc_idx];
+	if (prev_idx == idx)
+		return 0;
+	if (idx < spec->num_analog_muxes) {
+		snd_hda_codec_write_cache(codec, spec->mux_nids[adc_idx], 0,
+					  AC_VERB_SET_CONNECT_SEL,
+					  imux->items[idx].index);
+		if (prev_idx >= spec->num_analog_muxes) {
+			imux = spec->dinput_mux;
+			/* 0 = analog */
+			snd_hda_codec_write_cache(codec,
+						  spec->dmux_nids[adc_idx], 0,
+						  AC_VERB_SET_CONNECT_SEL,
+						  imux->items[0].index);
+		}
+	} else {
+		imux = spec->dinput_mux;
+		snd_hda_codec_write_cache(codec, spec->dmux_nids[adc_idx], 0,
+					  AC_VERB_SET_CONNECT_SEL,
+					  imux->items[idx - 1].index);
+	}
+	spec->cur_mux[adc_idx] = idx;
+	return 1;
 }
 
 static int stac92xx_mono_mux_enum_info(struct snd_kcontrol *kcontrol,
@@ -1204,21 +1231,6 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
-	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("DAC1 Capture Switch", 0x17, 0x4, HDA_INPUT),
-	HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x17, 0x4, HDA_INPUT),
-	{ } /* end */
-};
-
 static struct snd_kcontrol_new stac92hd71bxx_loopback[] = {
 	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2)
 };
@@ -1302,7 +1314,8 @@ static int stac92xx_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 	}
-	if (!spec->auto_mic && spec->num_dmuxes > 0) {
+	if (!spec->auto_mic && spec->num_dmuxes > 0 &&
+	    snd_hda_get_bool_hint(codec, "separate_dmux") == 1) {
 		stac_dmux_mixer.count = spec->num_dmuxes;
 		err = snd_hda_ctl_add(codec,
 				  snd_ctl_new1(&stac_dmux_mixer, codec));
@@ -3575,19 +3588,28 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
 	struct sigmatel_spec *spec = codec->spec;
+	struct hda_input_mux *imux = &spec->private_imux;
 	struct hda_input_mux *dimux = &spec->private_dimux;
-	int err, i;
+	int err, i, active_mics;
+	unsigned int def_conf;
 	char name[32];
 
 	dimux->items[dimux->num_items].label = stac92xx_dmic_labels[0];
 	dimux->items[dimux->num_items].index = 0;
 	dimux->num_items++;
 
+	active_mics = 0;
+	for (i = 0; i < spec->num_dmics; i++) {
+		def_conf = snd_hda_codec_get_pincfg(codec, spec->dmic_nids[i]);
+		if (get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE)
+			active_mics++;
+	}
+
 	for (i = 0; i < spec->num_dmics; i++) {
 		hda_nid_t nid;
 		int index;
 		unsigned int wcaps;
-		unsigned int def_conf;
+		const char *label;
 
 		def_conf = snd_hda_codec_get_pincfg(codec, spec->dmic_nids[i]);
 		if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
@@ -3598,12 +3620,16 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
 		if (index < 0)
 			continue;
 
+		if (active_mics == 1)
+			label = "Digital Mic";
+		else
+			label = stac92xx_dmic_labels[dimux->num_items];
+
 		wcaps = get_wcaps(codec, nid) &
 			(AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP);
 
 		if (wcaps) {
-			sprintf(name, "%s Capture Volume",
-				stac92xx_dmic_labels[dimux->num_items]);
+			sprintf(name, "%s Capture Volume", label);
 
 			err = stac92xx_add_control(spec,
 				STAC_CTL_WIDGET_VOL,
@@ -3615,10 +3641,14 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
 				return err;
 		}
 
-		dimux->items[dimux->num_items].label =
-			stac92xx_dmic_labels[dimux->num_items];
+		dimux->items[dimux->num_items].label = label;
 		dimux->items[dimux->num_items].index = index;
 		dimux->num_items++;
+		if (snd_hda_get_bool_hint(codec, "separate_dmux") != 1) {
+			imux->items[imux->num_items].label = label;
+			imux->items[imux->num_items].index = index;
+			imux->num_items++;
+		}
 	}
 
 	return 0;
@@ -3738,6 +3768,7 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const
 		imux->items[imux->num_items].index = index;
 		imux->num_items++;
 	}
+	spec->num_analog_muxes = imux->num_items;
 
 	if (imux->num_items) {
 		/*
@@ -5059,16 +5090,6 @@ static int patch_stac925x(struct hda_codec *codec)
 	return 0;
 }
 
-static struct hda_input_mux stac92hd73xx_dmux = {
-	.num_items = 4,
-	.items = {
-		{ "Analog Inputs", 0x0b },
-		{ "Digital Mic 1", 0x09 },
-		{ "Digital Mic 2", 0x0a },
-		{ "CD", 0x08 },
-	}
-};
-
 static int patch_stac92hd73xx(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
@@ -5138,8 +5159,6 @@ again:
 	spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids);
 	spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids);
 	spec->num_dmuxes = ARRAY_SIZE(stac92hd73xx_dmux_nids);
-	memcpy(&spec->private_dimux, &stac92hd73xx_dmux,
-			sizeof(stac92hd73xx_dmux));
 
 	spec->num_caps = STAC92HD73XX_NUM_CAPS;
 	spec->capvols = stac92hd73xx_capvols;
@@ -5164,18 +5183,15 @@ again:
 		case STAC_DELL_M6_AMIC: /* Analog Mics */
 			snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170);
 			spec->num_dmics = 0;
-			spec->private_dimux.num_items = 1;
 			break;
 		case STAC_DELL_M6_DMIC: /* Digital Mics */
 			snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160);
 			spec->num_dmics = 1;
-			spec->private_dimux.num_items = 2;
 			break;
 		case STAC_DELL_M6_BOTH: /* Both */
 			snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170);
 			snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160);
 			spec->num_dmics = 1;
-			spec->private_dimux.num_items = 2;
 			break;
 		}
 		break;
@@ -5183,13 +5199,13 @@ again:
 		spec->num_dmics = STAC92HD73XX_NUM_DMICS;
 		spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids);
 		spec->eapd_switch = 1;
+		break;
 	}
 	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;
@@ -5221,15 +5237,6 @@ 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;
@@ -5262,7 +5269,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 	spec->num_adcs = ARRAY_SIZE(stac92hd83xxx_adc_nids);
 	spec->num_amps = ARRAY_SIZE(stac92hd83xxx_amp_nids);
 	spec->num_dmics = STAC92HD83XXX_NUM_DMICS;
-	spec->dinput_mux = &stac92hd83xxx_dmux;
 	spec->pin_nids = stac92hd83xxx_pin_nids;
 	spec->num_caps = STAC92HD83XXX_NUM_CAPS;
 	spec->capvols = stac92hd83xxx_capvols;
@@ -5332,25 +5338,6 @@ again:
 	return 0;
 }
 
-static struct hda_input_mux stac92hd71bxx_dmux_nomixer = {
-	.num_items = 3,
-	.items = {
-		{ "Analog Inputs", 0x00 },
-		{ "Digital Mic 1", 0x02 },
-		{ "Digital Mic 2", 0x03 },
-	}
-};
-
-static struct hda_input_mux stac92hd71bxx_dmux_amixer = {
-	.num_items = 4,
-	.items = {
-		{ "Analog Inputs", 0x00 },
-		{ "Mixer", 0x01 },
-		{ "Digital Mic 1", 0x02 },
-		{ "Digital Mic 2", 0x03 },
-	}
-};
-
 /* get the pin connection (fixed, none, etc) */
 static unsigned int stac_get_defcfg_connect(struct hda_codec *codec, int idx)
 {
@@ -5411,7 +5398,6 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 	struct sigmatel_spec *spec;
 	struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init;
 	int err = 0;
-	unsigned int ndmic_nids = 0;
 
 	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -5467,21 +5453,13 @@ again:
 		/* fallthru */
 	case 0x111d76b4: /* 6 Port without Analog Mixer */
 	case 0x111d76b5:
-		memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_nomixer,
-		       sizeof(stac92hd71bxx_dmux_nomixer));
 		spec->init = stac92hd71bxx_core_init;
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
 		spec->num_dmics = stac92hd71bxx_connected_ports(codec,
 					stac92hd71bxx_dmic_nids,
 					STAC92HD71BXX_NUM_DMICS);
-		if (spec->num_dmics) {
-			spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
-			spec->dinput_mux = &spec->private_dimux;
-			ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1;
-		}
 		break;
 	case 0x111d7608: /* 5 Port with Analog Mixer */
-		spec->private_dimux.num_items--;
 		switch (spec->board_config) {
 		case STAC_HP_M4:
 			/* Enable VREF power saving on GPIO1 detect */
@@ -5503,15 +5481,6 @@ again:
 
 		/* no output amps */
 		spec->num_pwrs = 0;
-		if (snd_hda_get_bool_hint(codec, "analog_mixer") == 1) {
-			spec->mixer = stac92hd71bxx_analog_mixer;
-			memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer,
-			       sizeof(stac92hd71bxx_dmux_amixer));
-		} else {
-			memcpy(&spec->private_dimux,
-			       &stac92hd71bxx_dmux_nomixer,
-			       sizeof(stac92hd71bxx_dmux_nomixer));
-		}
 		/* disable VSW */
 		spec->init = stac92hd71bxx_core_init;
 		unmute_init++;
@@ -5521,11 +5490,6 @@ again:
 		spec->num_dmics = stac92hd71bxx_connected_ports(codec,
 					stac92hd71bxx_dmic_nids,
 					STAC92HD71BXX_NUM_DMICS - 1);
-		if (spec->num_dmics) {
-			spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
-			spec->dinput_mux = &spec->private_dimux;
-			ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 2;
-		}
 		break;
 	case 0x111d7603: /* 6 Port with Analog Mixer */
 		if ((codec->revision_id & 0xf) == 1)
@@ -5535,25 +5499,12 @@ again:
 		spec->num_pwrs = 0;
 		/* fallthru */
 	default:
-		if (snd_hda_get_bool_hint(codec, "analog_mixer") == 1) {
-			spec->mixer = stac92hd71bxx_analog_mixer;
-			memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer,
-			       sizeof(stac92hd71bxx_dmux_amixer));
-		} else {
-			memcpy(&spec->private_dimux,
-			       &stac92hd71bxx_dmux_nomixer,
-			       sizeof(stac92hd71bxx_dmux_nomixer));
-		}
 		spec->init = stac92hd71bxx_core_init;
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
 		spec->num_dmics = stac92hd71bxx_connected_ports(codec,
 					stac92hd71bxx_dmic_nids,
 					STAC92HD71BXX_NUM_DMICS);
-		if (spec->num_dmics) {
-			spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
-			spec->dinput_mux = &spec->private_dimux;
-			ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1;
-		}
+		break;
 	}
 
 	if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP)
@@ -5581,6 +5532,7 @@ again:
 
 	spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids);
 	spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids);
+	spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
 	spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e);
 
 	switch (spec->board_config) {
@@ -5635,8 +5587,6 @@ again:
 #endif	
 
 	spec->multiout.dac_nids = spec->dac_nids;
-	if (spec->dinput_mux)
-		spec->private_dimux.num_items += spec->num_dmics - ndmic_nids;
 
 	err = stac92xx_parse_auto_config(codec, 0x21, 0);
 	if (!err) {
-- 
cgit v1.2.3


From 3a38516750e176a18f76d605b401fbab2c72d648 Mon Sep 17 00:00:00 2001
From: Stelian Pop <stelian@popies.net>
Date: Thu, 30 Jul 2009 14:44:27 +0200
Subject: ALSA: hda - Enable HP output with Macbook Pro 5, 5

The patch below, to be applied on the latest sound-unstable-2.6.git,
enables headphones output on my MacBookPro 5,5, together with the
automuting feature.

Here is the exact soundcard id:
	Vendor Id: 0x10134206
	Subsystem Id: 0x106b4d00
	Revision Id: 0x100301

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_cirrus.c | 13 +++++++++----
 1 file changed, 9 insertions(+), 4 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index f5527380a24..8ba306856d3 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -832,6 +832,11 @@ static void cs_automute(struct hda_codec *codec)
 				    AC_VERB_SET_PIN_WIDGET_CONTROL,
 				    hp_present ? 0 : PIN_OUT);
 	}
+	if (spec->board_config == CS420X_MBP55) {
+		unsigned int gpio = hp_present ? 0x02 : 0x08;
+		snd_hda_codec_write(codec, 0x01, 0,
+				    AC_VERB_SET_GPIO_DATA, gpio);
+	}
 }
 
 static void cs_automic(struct hda_codec *codec)
@@ -1133,10 +1138,10 @@ static int patch_cs420x(struct hda_codec *codec)
 
 	switch (spec->board_config) {
 	case CS420X_MBP55:
-		/* GPIO3 = EAPD? */
-		spec->gpio_mask = 0x08;
-		spec->gpio_dir = 0x08;
-		spec->gpio_data = 0x08;
+		/* GPIO1 = headphones */
+		/* GPIO3 = speakers */
+		spec->gpio_mask = 0x0a;
+		spec->gpio_dir = 0x0a;
 		break;
 	}
 
-- 
cgit v1.2.3


From fefd67f31ee7f5259344e36a237d59b47e8715cf Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 30 Jul 2009 18:03:05 +0200
Subject: ALSA: hda - Add line-out jack detection on IDT/STAC codecs

Add the automatic mute of speakers via line-out jack plugging on
STAC/IDT codecs.  The feature is enabled when the HP detect is present.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 70 +++++++++++++++++++++++++++++++++---------
 1 file changed, 56 insertions(+), 14 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 9e192529e80..c7e5de6e394 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -40,6 +40,7 @@ enum {
 	STAC_INSERT_EVENT,
 	STAC_PWR_EVENT,
 	STAC_HP_EVENT,
+	STAC_LO_EVENT,
 	STAC_MIC_EVENT,
 };
 
@@ -4345,6 +4346,14 @@ static int stac92xx_init(struct hda_codec *codec)
 			hda_nid_t nid = cfg->hp_pins[i];
 			enable_pin_detect(codec, nid, STAC_HP_EVENT);
 		}
+		if (cfg->line_out_type == AUTO_PIN_LINE_OUT) {
+			/* enable pin-detect for line-outs as well */
+			for (i = 0; i < cfg->hp_outs; i++) {
+				hda_nid_t nid = cfg->hp_pins[i];
+				enable_pin_detect(codec, nid, STAC_LO_EVENT);
+			}
+		}
+
 		/* force to enable the first line-out; the others are set up
 		 * in unsol_event
 		 */
@@ -4543,6 +4552,48 @@ static int get_pin_presence(struct hda_codec *codec, hda_nid_t nid)
 	return 0;
 }
 
+static void stac92xx_line_out_detect(struct hda_codec *codec,
+				     int presence)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i;
+
+	for (i = 0; i < cfg->line_outs; i++) {
+		if (presence)
+			break;
+		presence = get_pin_presence(codec, cfg->line_out_pins[i]);
+		if (presence) {
+			unsigned int pinctl;
+			pinctl = snd_hda_codec_read(codec,
+						    cfg->line_out_pins[i], 0,
+					    AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+			if (pinctl & AC_PINCTL_IN_EN)
+				presence = 0; /* mic- or line-input */
+		}
+	}
+
+	if (presence) {
+		/* disable speakers */
+		for (i = 0; i < cfg->speaker_outs; i++)
+			stac92xx_reset_pinctl(codec, cfg->speaker_pins[i],
+						AC_PINCTL_OUT_EN);
+		if (spec->eapd_mask && spec->eapd_switch)
+			stac_gpio_set(codec, spec->gpio_mask,
+				spec->gpio_dir, spec->gpio_data &
+				~spec->eapd_mask);
+	} else {
+		/* enable speakers */
+		for (i = 0; i < cfg->speaker_outs; i++)
+			stac92xx_set_pinctl(codec, cfg->speaker_pins[i],
+						AC_PINCTL_OUT_EN);
+		if (spec->eapd_mask && spec->eapd_switch)
+			stac_gpio_set(codec, spec->gpio_mask,
+				spec->gpio_dir, spec->gpio_data |
+				spec->eapd_mask);
+	}
+} 
+
 /* return non-zero if the hp-pin of the given array index isn't
  * a jack-detection target
  */
@@ -4595,13 +4646,6 @@ static void stac92xx_hp_detect(struct hda_codec *codec)
 		for (i = 0; i < cfg->line_outs; i++)
 			stac92xx_reset_pinctl(codec, cfg->line_out_pins[i],
 						AC_PINCTL_OUT_EN);
-		for (i = 0; i < cfg->speaker_outs; i++)
-			stac92xx_reset_pinctl(codec, cfg->speaker_pins[i],
-						AC_PINCTL_OUT_EN);
-		if (spec->eapd_mask && spec->eapd_switch)
-			stac_gpio_set(codec, spec->gpio_mask,
-				spec->gpio_dir, spec->gpio_data &
-				~spec->eapd_mask);
 	} else {
 		/* enable lineouts */
 		if (spec->hp_switch)
@@ -4610,14 +4654,8 @@ static void stac92xx_hp_detect(struct hda_codec *codec)
 		for (i = 0; i < cfg->line_outs; i++)
 			stac92xx_set_pinctl(codec, cfg->line_out_pins[i],
 						AC_PINCTL_OUT_EN);
-		for (i = 0; i < cfg->speaker_outs; i++)
-			stac92xx_set_pinctl(codec, cfg->speaker_pins[i],
-						AC_PINCTL_OUT_EN);
-		if (spec->eapd_mask && spec->eapd_switch)
-			stac_gpio_set(codec, spec->gpio_mask,
-				spec->gpio_dir, spec->gpio_data |
-				spec->eapd_mask);
 	}
+	stac92xx_line_out_detect(codec, presence);
 	/* toggle hp outs */
 	for (i = 0; i < cfg->hp_outs; i++) {
 		unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN;
@@ -4744,6 +4782,9 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
 	case STAC_HP_EVENT:
 		stac92xx_hp_detect(codec);
 		break;
+	case STAC_LO_EVENT:
+		stac92xx_line_out_detect(codec, 0);
+		break;
 	case STAC_MIC_EVENT:
 		stac92xx_mic_detect(codec);
 		break;
@@ -4751,6 +4792,7 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
 
 	switch (event->type) {
 	case STAC_HP_EVENT:
+	case STAC_LO_EVENT:
 	case STAC_MIC_EVENT:
 	case STAC_INSERT_EVENT:
 	case STAC_PWR_EVENT:
-- 
cgit v1.2.3


From 559059b27f1768e89adc204f3ba2cb82882c15ca Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Sun, 2 Aug 2009 16:48:55 +0800
Subject: ALSA: hda: add IbexPeak/Clarkdale HDMI model with static cvt/pin
 number

The new IbexPeak HDMI codec has 3 pin nodes and 2 converter nodes.
Here we assume only the first ones will be used.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_intelhdmi.c | 104 ++++++++++++++++++++--------------------
 1 file changed, 52 insertions(+), 52 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c
index 032850eba36..01a18ed475a 100644
--- a/sound/pci/hda/patch_intelhdmi.c
+++ b/sound/pci/hda/patch_intelhdmi.c
@@ -33,8 +33,8 @@
 #include "hda_codec.h"
 #include "hda_local.h"
 
-#define CVT_NID		0x02	/* audio converter */
-#define PIN_NID		0x03	/* HDMI output pin */
+static hda_nid_t cvt_nid;	/* audio converter */
+static hda_nid_t pin_nid;	/* HDMI output pin */
 
 #define INTEL_HDMI_EVENT_TAG		0x08
 
@@ -44,30 +44,6 @@ struct intel_hdmi_spec {
 	struct hdmi_eld sink_eld;
 };
 
-static struct hda_verb pinout_enable_verb[] = {
-	{PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{} /* terminator */
-};
-
-static struct hda_verb unsolicited_response_verb[] = {
-	{PIN_NID, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN |
-						  INTEL_HDMI_EVENT_TAG},
-	{}
-};
-
-static struct hda_verb def_chan_map[] = {
-	{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x00},
-	{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x11},
-	{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x22},
-	{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x33},
-	{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x44},
-	{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x55},
-	{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x66},
-	{CVT_NID, AC_VERB_SET_HDMI_CHAN_SLOT, 0x77},
-	{}
-};
-
-
 struct hdmi_audio_infoframe {
 	u8 type; /* 0x84 */
 	u8 ver;  /* 0x01 */
@@ -244,11 +220,12 @@ static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t nid,
 static void hdmi_enable_output(struct hda_codec *codec)
 {
 	/* Unmute */
-	if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP)
-		snd_hda_codec_write(codec, PIN_NID, 0,
+	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);
 	/* Enable pin out */
-	snd_hda_sequence_write(codec, pinout_enable_verb);
+	snd_hda_codec_write(codec, pin_nid, 0,
+			    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
 }
 
 /*
@@ -256,8 +233,8 @@ static void hdmi_enable_output(struct hda_codec *codec)
  */
 static void hdmi_start_infoframe_trans(struct hda_codec *codec)
 {
-	hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
-	snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT,
+	hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0);
+	snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT,
 						AC_DIPXMIT_BEST);
 }
 
@@ -266,20 +243,20 @@ static void hdmi_start_infoframe_trans(struct hda_codec *codec)
  */
 static void hdmi_stop_infoframe_trans(struct hda_codec *codec)
 {
-	hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
-	snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT,
+	hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0);
+	snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_HDMI_DIP_XMIT,
 						AC_DIPXMIT_DISABLE);
 }
 
 static int hdmi_get_channel_count(struct hda_codec *codec)
 {
-	return 1 + snd_hda_codec_read(codec, CVT_NID, 0,
+	return 1 + snd_hda_codec_read(codec, cvt_nid, 0,
 					AC_VERB_GET_CVT_CHAN_COUNT, 0);
 }
 
 static void hdmi_set_channel_count(struct hda_codec *codec, int chs)
 {
-	snd_hda_codec_write(codec, CVT_NID, 0,
+	snd_hda_codec_write(codec, cvt_nid, 0,
 					AC_VERB_SET_CVT_CHAN_COUNT, chs - 1);
 
 	if (chs != hdmi_get_channel_count(codec))
@@ -294,7 +271,7 @@ static void hdmi_debug_channel_mapping(struct hda_codec *codec)
 	int slot;
 
 	for (i = 0; i < 8; i++) {
-		slot = snd_hda_codec_read(codec, CVT_NID, 0,
+		slot = snd_hda_codec_read(codec, cvt_nid, 0,
 						AC_VERB_GET_HDMI_CHAN_SLOT, i);
 		printk(KERN_DEBUG "HDMI: ASP channel %d => slot %d\n",
 						slot >> 4, slot & 0x7);
@@ -307,7 +284,7 @@ static void hdmi_parse_eld(struct hda_codec *codec)
 	struct intel_hdmi_spec *spec = codec->spec;
 	struct hdmi_eld *eld = &spec->sink_eld;
 
-	if (!snd_hdmi_get_eld(eld, codec, PIN_NID))
+	if (!snd_hdmi_get_eld(eld, codec, pin_nid))
 		snd_hdmi_show_eld(eld);
 }
 
@@ -322,11 +299,11 @@ static void hdmi_debug_dip_size(struct hda_codec *codec)
 	int i;
 	int size;
 
-	size = snd_hdmi_get_eld_size(codec, PIN_NID);
+	size = snd_hdmi_get_eld_size(codec, pin_nid);
 	printk(KERN_DEBUG "HDMI: ELD buf size is %d\n", size);
 
 	for (i = 0; i < 8; i++) {
-		size = snd_hda_codec_read(codec, PIN_NID, 0,
+		size = snd_hda_codec_read(codec, pin_nid, 0,
 						AC_VERB_GET_HDMI_DIP_SIZE, i);
 		printk(KERN_DEBUG "HDMI: DIP GP[%d] buf size is %d\n", i, size);
 	}
@@ -340,15 +317,15 @@ static void hdmi_clear_dip_buffers(struct hda_codec *codec)
 	int size;
 	int pi, bi;
 	for (i = 0; i < 8; i++) {
-		size = snd_hda_codec_read(codec, PIN_NID, 0,
+		size = snd_hda_codec_read(codec, pin_nid, 0,
 						AC_VERB_GET_HDMI_DIP_SIZE, i);
 		if (size == 0)
 			continue;
 
-		hdmi_set_dip_index(codec, PIN_NID, i, 0x0);
+		hdmi_set_dip_index(codec, pin_nid, i, 0x0);
 		for (j = 1; j < 1000; j++) {
-			hdmi_write_dip_byte(codec, PIN_NID, 0x0);
-			hdmi_get_dip_index(codec, PIN_NID, &pi, &bi);
+			hdmi_write_dip_byte(codec, pin_nid, 0x0);
+			hdmi_get_dip_index(codec, pin_nid, &pi, &bi);
 			if (pi != i)
 				snd_printd(KERN_INFO "dip index %d: %d != %d\n",
 						bi, pi, i);
@@ -376,9 +353,9 @@ static void hdmi_fill_audio_infoframe(struct hda_codec *codec,
 		sum += params[i];
 	ai->checksum = - sum;
 
-	hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0);
+	hdmi_set_dip_index(codec, pin_nid, 0x0, 0x0);
 	for (i = 0; i < sizeof(ai); i++)
-		hdmi_write_dip_byte(codec, PIN_NID, params[i]);
+		hdmi_write_dip_byte(codec, pin_nid, params[i]);
 }
 
 /*
@@ -465,6 +442,8 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec,
 static void hdmi_setup_channel_mapping(struct hda_codec *codec,
 					struct hdmi_audio_infoframe *ai)
 {
+	int i;
+
 	if (!ai->CA)
 		return;
 
@@ -473,7 +452,11 @@ static void hdmi_setup_channel_mapping(struct hda_codec *codec,
 	 * ALSA sequence is front/surr/clfe/side?
 	 */
 
-	snd_hda_sequence_write(codec, def_chan_map);
+	for (i = 0; i < 8; i++)
+		snd_hda_codec_write(codec, cvt_nid, 0,
+				    AC_VERB_SET_HDMI_CHAN_SLOT,
+				    (i << 4) | i);
+
 	hdmi_debug_channel_mapping(codec);
 }
 
@@ -597,7 +580,6 @@ static struct hda_pcm_stream intel_hdmi_pcm_playback = {
 	.substreams = 1,
 	.channels_min = 2,
 	.channels_max = 8,
-	.nid = CVT_NID, /* NID to query formats and rates and setup streams */
 	.ops = {
 		.open    = intel_hdmi_playback_pcm_open,
 		.close   = intel_hdmi_playback_pcm_close,
@@ -613,6 +595,9 @@ static int intel_hdmi_build_pcms(struct hda_codec *codec)
 	codec->num_pcms = 1;
 	codec->pcm_info = info;
 
+	/* NID to query formats and rates and setup streams */
+	intel_hdmi_pcm_playback.nid = cvt_nid;
+
 	info->name = "INTEL HDMI";
 	info->pcm_type = HDA_PCM_TYPE_HDMI;
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = intel_hdmi_pcm_playback;
@@ -636,8 +621,9 @@ static int intel_hdmi_init(struct hda_codec *codec)
 {
 	hdmi_enable_output(codec);
 
-	snd_hda_sequence_write(codec, unsolicited_response_verb);
-
+	snd_hda_codec_write(codec, pin_nid, 0,
+			    AC_VERB_SET_UNSOLICITED_ENABLE,
+			    AC_USRSP_EN | INTEL_HDMI_EVENT_TAG);
 	return 0;
 }
 
@@ -657,7 +643,7 @@ static struct hda_codec_ops intel_hdmi_patch_ops = {
 	.unsol_event		= intel_hdmi_unsol_event,
 };
 
-static int patch_intel_hdmi(struct hda_codec *codec)
+static int do_patch_intel_hdmi(struct hda_codec *codec)
 {
 	struct intel_hdmi_spec *spec;
 
@@ -667,7 +653,7 @@ static int patch_intel_hdmi(struct hda_codec *codec)
 
 	spec->multiout.num_dacs = 0;	  /* no analog */
 	spec->multiout.max_channels = 8;
-	spec->multiout.dig_out_nid = CVT_NID;
+	spec->multiout.dig_out_nid = cvt_nid;
 
 	codec->spec = spec;
 	codec->patch_ops = intel_hdmi_patch_ops;
@@ -679,13 +665,27 @@ static int patch_intel_hdmi(struct hda_codec *codec)
 	return 0;
 }
 
+static int patch_intel_hdmi(struct hda_codec *codec)
+{
+	cvt_nid = 0x02;
+	pin_nid = 0x03;
+	return do_patch_intel_hdmi(codec);
+}
+
+static int patch_intel_hdmi_ibexpeak(struct hda_codec *codec)
+{
+	cvt_nid = 0x02;
+	pin_nid = 0x04;
+	return do_patch_intel_hdmi(codec);
+}
+
 static struct hda_codec_preset snd_hda_preset_intelhdmi[] = {
 	{ .id = 0x808629fb, .name = "G45 DEVCL",  .patch = patch_intel_hdmi },
 	{ .id = 0x80862801, .name = "G45 DEVBLC", .patch = patch_intel_hdmi },
 	{ .id = 0x80862802, .name = "G45 DEVCTG", .patch = patch_intel_hdmi },
 	{ .id = 0x80862803, .name = "G45 DEVELK", .patch = patch_intel_hdmi },
 	{ .id = 0x80862804, .name = "G45 DEVIBX", .patch = patch_intel_hdmi },
-	{ .id = 0x80860054, .name = "G45 DEVIBX", .patch = patch_intel_hdmi },
+	{ .id = 0x80860054, .name = "Q57 DEVIBX", .patch = patch_intel_hdmi_ibexpeak },
 	{ .id = 0x10951392, .name = "SiI1392 HDMI",     .patch = patch_intel_hdmi },
 	{} /* terminator */
 };
-- 
cgit v1.2.3


From 15cfa2b3db8b64d4e66d36fb88d21bae9ea3dfba Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 3 Aug 2009 14:23:33 +0200
Subject: ALSA: hda - Fix line-out jack detection

The commit fefd67f31ee7f5259344e36a237d59b47e8715cf
    ALSA: hda - Add line-out jack detection on IDT/STAC codecs
enabled wrong pins for jack detections.  Fixed to the correct ones.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index b3c53f7a668..9961fe19769 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4350,8 +4350,8 @@ static int stac92xx_init(struct hda_codec *codec)
 		}
 		if (cfg->line_out_type == AUTO_PIN_LINE_OUT) {
 			/* enable pin-detect for line-outs as well */
-			for (i = 0; i < cfg->hp_outs; i++) {
-				hda_nid_t nid = cfg->hp_pins[i];
+			for (i = 0; i < cfg->line_outs; i++) {
+				hda_nid_t nid = cfg->line_out_pins[i];
 				enable_pin_detect(codec, nid, STAC_LO_EVENT);
 			}
 		}
-- 
cgit v1.2.3


From 16ffe32c773dab387cfc5e6bdf181367ff944376 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 4 Aug 2009 13:40:54 +0200
Subject: ALSA: hda - Fix line-out jack handling with STAC/IDT codec

When the line-out jack is plugged/unplugged, the driver needs to check
the headphone plug, not only the line-out jack itself.  Otherwise the
headphone or the speaker may be wrongly muted/unmuted.

As a result, both STAC_HP_EVENT and STAC_LO_EVENT need to call the
same function, stac92xx_hp_detect().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 4 +---
 1 file changed, 1 insertion(+), 3 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 9961fe19769..edeeb568696 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4782,10 +4782,8 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)
 
 	switch (event->type) {
 	case STAC_HP_EVENT:
-		stac92xx_hp_detect(codec);
-		break;
 	case STAC_LO_EVENT:
-		stac92xx_line_out_detect(codec, 0);
+		stac92xx_hp_detect(codec);
 		break;
 	case STAC_MIC_EVENT:
 		stac92xx_mic_detect(codec);
-- 
cgit v1.2.3


From 1972d02504b142a6f1442bd987a8831146031d3d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 6 Aug 2009 08:44:43 +0200
Subject: ALSA: hda - Add quirks for some HP laptops

The new HP laptops have PCI SSID 103c:701x and requires model=hp-dv5.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index edeeb568696..163aad599b6 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -1906,6 +1906,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 		      "HP mini 1000", STAC_HP_M4),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361b,
 		      "HP HDX", STAC_HP_HDX),  /* HDX16 */
+	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010,
+		      "HP", STAC_HP_DV5),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233,
 				"unknown Dell", STAC_DELL_M4_1),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234,
-- 
cgit v1.2.3


From da2a2aaa8e3dddb41678af7e5295d9e3e6c030a4 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 10 Aug 2009 07:44:09 +0200
Subject: ALSA: hda - Fix Oops due to STAC/IDT auto-mic changes

The previous auto-mic patch for STAC/IDT codecs causes the Oops on
machines without digital mic pins.  This patch fixes the problem.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 5 +++--
 1 file changed, 3 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 163aad599b6..27607306057 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -3671,7 +3671,7 @@ static int set_mic_route(struct hda_codec *codec,
 		if (i < 0)
 			return -1;
 		mic->mux_idx = i;
-	}  else {
+	}  else if (spec->dmux_nids) {
 		/* digital pin */
 		mic->mux_idx = 0;
 		i = get_connection_index(codec, spec->dmux_nids[0], pin);
@@ -4373,7 +4373,8 @@ static int stac92xx_init(struct hda_codec *codec)
 	}
 	if (spec->auto_mic) {
 		/* initialize connection to analog input */
-		snd_hda_codec_write_cache(codec, spec->dmux_nids[0], 0,
+		if (spec->dmux_nids)
+			snd_hda_codec_write_cache(codec, spec->dmux_nids[0], 0,
 					  AC_VERB_SET_CONNECT_SEL, 0);
 		if (enable_pin_detect(codec, spec->ext_mic.pin, STAC_MIC_EVENT))
 			stac_issue_unsol_event(codec, spec->ext_mic.pin);
-- 
cgit v1.2.3


From 6c8194922739138f046a4d0924519dd43b48e1f3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 10 Aug 2009 18:47:44 +0200
Subject: ALSA: hda - Add auto-mic support for Realtek codecs

Added the support for automatic mic selection via plugging for
Realtek codecs (in auto-probing mode).  The auto-mic mode is enabled
only when one internal mic and one external mic are present.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 177 ++++++++++++++++++++++++++++++++----------
 1 file changed, 135 insertions(+), 42 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b0d30fced3f..678c2d7b7f9 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -258,6 +258,14 @@ enum {
 	ALC_INIT_GPIO3,
 };
 
+struct alc_mic_route {
+	hda_nid_t pin;
+	unsigned char mux_idx;
+	unsigned char amix_idx;
+};
+
+#define MUX_IDX_UNDEF	((unsigned char)-1)
+
 struct alc_spec {
 	/* codec parameterization */
 	struct snd_kcontrol_new *mixers[5];	/* mixer arrays */
@@ -300,6 +308,8 @@ struct alc_spec {
 	unsigned int num_mux_defs;
 	const struct hda_input_mux *input_mux;
 	unsigned int cur_mux[3];
+	struct alc_mic_route ext_mic;
+	struct alc_mic_route int_mic;
 
 	/* channel model */
 	const struct hda_channel_mode *channel_mode;
@@ -327,6 +337,7 @@ struct alc_spec {
 	unsigned int sense_updated: 1;
 	unsigned int jack_present: 1;
 	unsigned int master_sw: 1;
+	unsigned int auto_mic:1;
 
 	/* other flags */
 	unsigned int no_analog :1; /* digital I/O only */
@@ -963,30 +974,79 @@ static void alc_automute_pin(struct hda_codec *codec)
 	}
 }
 
-#if 0 /* it's broken in some cases -- temporarily disabled */
+static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
+				hda_nid_t nid)
+{
+	hda_nid_t conn[HDA_MAX_NUM_INPUTS];
+	int i, nums;
+
+	nums = snd_hda_get_connections(codec, mux, conn, ARRAY_SIZE(conn));
+	for (i = 0; i < nums; i++)
+		if (conn[i] == nid)
+			return i;
+	return -1;
+}
+
+static int set_mic_mux_idx(struct hda_codec *codec, hda_nid_t cap,
+			   struct alc_mic_route *mic)
+{
+	int idx = get_connection_index(codec, cap, mic->pin);
+	if (idx < 0)
+		return 1; /* invalid */
+	mic->mux_idx = idx;
+	return 0;
+}
+
 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);
+	struct alc_mic_route *dead, *alive;
+	unsigned int present, type;
+	hda_nid_t cap_nid;
+
+	if (!spec->int_mic.pin || !spec->ext_mic.pin)
+		return;
+	if (snd_BUG_ON(!spec->adc_nids))
+		return;
+
+	cap_nid = spec->capsrc_nids ? spec->capsrc_nids[0] : spec->adc_nids[0];
+
+	present = snd_hda_codec_read(codec, spec->ext_mic.pin, 0,
+				     AC_VERB_GET_PIN_SENSE, 0);
+	present &= AC_PINSENSE_PRESENCE;
+	if (present) {
+		alive = &spec->ext_mic;
+		dead = &spec->int_mic;
+	} else {
+		alive = &spec->int_mic;
+		dead = &spec->ext_mic;
+	}
+
+	if (alive->mux_idx == MUX_IDX_UNDEF &&
+	    set_mic_mux_idx(codec, cap_nid, alive))
+		return;
+	if (dead->mux_idx == MUX_IDX_UNDEF &&
+	    set_mic_mux_idx(codec, cap_nid, dead))
+		return;
+
+	type = get_wcaps_type(get_wcaps(codec, cap_nid));
+	if (type == AC_WID_AUD_MIX) {
+		/* Matrix-mixer style (e.g. ALC882) */
+		snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT,
+					 alive->mux_idx,
+					 HDA_AMP_MUTE, 0);
+		snd_hda_codec_amp_stereo(codec, cap_nid, HDA_INPUT,
+					 dead->mux_idx,
+					 HDA_AMP_MUTE, HDA_AMP_MUTE);
+	} else {
+		/* MUX style (e.g. ALC880) */
+		snd_hda_codec_write_cache(codec, cap_nid, 0,
+					  AC_VERB_SET_CONNECT_SEL,
+					  alive->mux_idx);
+	}
+
+	/* FIXME: analog mixer */
 }
-#else
-#define alc_mic_automute(codec) do {} while(0) /* NOP */
-#endif /* disabled */
 
 /* unsolicited event for HP jack sensing */
 static void alc_sku_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -1142,6 +1202,55 @@ static void alc_init_auto_hp(struct hda_codec *codec)
 	spec->unsol_event = alc_sku_unsol_event;
 }
 
+static void alc_init_auto_mic(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	hda_nid_t fixed, ext;
+	int i;
+
+	/* there must be only two mic inputs exclusively */
+	for (i = AUTO_PIN_LINE; i < AUTO_PIN_LAST; i++)
+		if (cfg->input_pins[i])
+			return;
+
+	fixed = ext = 0;
+	for (i = AUTO_PIN_MIC; i <= AUTO_PIN_FRONT_MIC; i++) {
+		hda_nid_t nid = cfg->input_pins[i];
+		unsigned int defcfg;
+		if (!nid)
+			return;
+		defcfg = snd_hda_codec_get_pincfg(codec, nid);
+		switch (get_defcfg_connect(defcfg)) {
+		case AC_JACK_PORT_FIXED:
+			if (fixed)
+				return; /* already occupied */
+			fixed = nid;
+			break;
+		case AC_JACK_PORT_COMPLEX:
+			if (ext)
+				return; /* already occupied */
+			ext = nid;
+			break;
+		default:
+			return; /* invalid entry */
+		}
+	}
+	if (!(get_wcaps(codec, ext) & AC_WCAP_UNSOL_CAP))
+		return; /* no unsol support */
+	snd_printdd("realtek: Enable auto-mic switch on NID 0x%x/0x%x\n",
+		    ext, fixed);
+	spec->ext_mic.pin = ext;
+	spec->int_mic.pin = fixed;
+	spec->ext_mic.mux_idx = MUX_IDX_UNDEF; /* set later */
+	spec->int_mic.mux_idx = MUX_IDX_UNDEF; /* set later */
+	spec->auto_mic = 1;
+	snd_hda_codec_write_cache(codec, spec->ext_mic.pin, 0,
+				  AC_VERB_SET_UNSOLICITED_ENABLE,
+				  AC_USRSP_EN | ALC880_MIC_EVENT);
+	spec->unsol_event = alc_sku_unsol_event;
+}
+
 /* check subsystem ID and set up device-specific initialization;
  * return 1 if initialized, 0 if invalid SSID
  */
@@ -1243,6 +1352,7 @@ do_sku:
 	}
 
 	alc_init_auto_hp(codec);
+	alc_init_auto_mic(codec);
 	return 1;
 }
 
@@ -1255,6 +1365,7 @@ static void alc_ssid_check(struct hda_codec *codec,
 			   "Enable default setup for auto mode as fallback\n");
 		spec->init_amp = ALC_INIT_DEFAULT;
 		alc_init_auto_hp(codec);
+		alc_init_auto_mic(codec);
 	}
 }
 
@@ -4572,7 +4683,8 @@ static void set_capture_mixer(struct alc_spec *spec)
 	};
 	if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) {
 		int mux;
-		if (spec->input_mux && spec->input_mux->num_items > 1)
+		if (spec->input_mux && spec->input_mux->num_items > 1 &&
+		    !spec->auto_mic)
 			mux = 1;
 		else
 			mux = 0;
@@ -17118,25 +17230,6 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
 	return 0;
 }
 
-/* return the index of the src widget from the connection list of the nid.
- * return -1 if not found
- */
-static int alc662_input_pin_idx(struct hda_codec *codec, hda_nid_t nid,
-				hda_nid_t src)
-{
-	hda_nid_t conn_list[HDA_MAX_CONNECTIONS];
-	int i, conns;
-
-	conns = snd_hda_get_connections(codec, nid, conn_list,
-					ARRAY_SIZE(conn_list));
-	if (conns < 0)
-		return -1;
-	for (i = 0; i < conns; i++)
-		if (conn_list[i] == src)
-			return i;
-	return -1;
-}
-
 static int alc662_is_input_pin(struct hda_codec *codec, hda_nid_t nid)
 {
 	unsigned int pincap = snd_hda_query_pin_caps(codec, nid);
@@ -17153,7 +17246,7 @@ static int alc662_auto_create_analog_input_ctls(struct hda_codec *codec,
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		if (alc662_is_input_pin(codec, cfg->input_pins[i])) {
-			idx = alc662_input_pin_idx(codec, 0x0b,
+			idx = get_connection_index(codec, 0x0b,
 						   cfg->input_pins[i]);
 			if (idx >= 0) {
 				err = new_analog_input(spec, cfg->input_pins[i],
@@ -17162,7 +17255,7 @@ static int alc662_auto_create_analog_input_ctls(struct hda_codec *codec,
 				if (err < 0)
 					return err;
 			}
-			idx = alc662_input_pin_idx(codec, 0x22,
+			idx = get_connection_index(codec, 0x22,
 						   cfg->input_pins[i]);
 			if (idx >= 0) {
 				imux->items[imux->num_items].label =
-- 
cgit v1.2.3


From 2a22d3f81283253ab9dae9244f3ca58a5ac66b36 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 10 Aug 2009 18:54:38 +0200
Subject: ALSA: hda - Use only one capture stream for auto-mic

When the auto-mic feature is enabled, we should support only one
capture stream.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 6 ++++--
 1 file changed, 4 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 678c2d7b7f9..7ecf929b97a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4683,8 +4683,10 @@ static void set_capture_mixer(struct alc_spec *spec)
 	};
 	if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) {
 		int mux;
-		if (spec->input_mux && spec->input_mux->num_items > 1 &&
-		    !spec->auto_mic)
+		if (spec->auto_mic) {
+			mux = 0;
+			spec->num_adc_nids = 1; /* support only one ADC */
+		} else if (spec->input_mux && spec->input_mux->num_items > 1)
 			mux = 1;
 		else
 			mux = 0;
-- 
cgit v1.2.3


From b59bdf3b0ca11bfc6a539012431d751deaa65b32 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 11 Aug 2009 09:47:30 +0200
Subject: ALSA: hda - Check connectivity for auto-mic of Realtek codecs

Some Realtek codecs don't provide the full connections for certain pins
from each ADC; e.g. ACL662/ALC272 gives only one of two digital-mic pins
for each ADC.  Thus, depending on the digital mic pin, the ADC/MUX to be
used has to be chosen properly.

This patch adds the check of the connectivity of pins at auto-mic mode.
If no proper connectivity is found, auto_mic flag is turned off to be
sure.

Also the mux_idx is determined during this check so it won't be checked
in the unsol event any more.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 83 ++++++++++++++++++++++++++-----------------
 1 file changed, 51 insertions(+), 32 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 2c348e1339d..5156c4f14f8 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -987,16 +987,6 @@ static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
 	return -1;
 }
 
-static int set_mic_mux_idx(struct hda_codec *codec, hda_nid_t cap,
-			   struct alc_mic_route *mic)
-{
-	int idx = get_connection_index(codec, cap, mic->pin);
-	if (idx < 0)
-		return 1; /* invalid */
-	mic->mux_idx = idx;
-	return 0;
-}
-
 static void alc_mic_automute(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -1004,6 +994,8 @@ static void alc_mic_automute(struct hda_codec *codec)
 	unsigned int present, type;
 	hda_nid_t cap_nid;
 
+	if (!spec->auto_mic)
+		return;
 	if (!spec->int_mic.pin || !spec->ext_mic.pin)
 		return;
 	if (snd_BUG_ON(!spec->adc_nids))
@@ -1022,13 +1014,6 @@ static void alc_mic_automute(struct hda_codec *codec)
 		dead = &spec->ext_mic;
 	}
 
-	if (alive->mux_idx == MUX_IDX_UNDEF &&
-	    set_mic_mux_idx(codec, cap_nid, alive))
-		return;
-	if (dead->mux_idx == MUX_IDX_UNDEF &&
-	    set_mic_mux_idx(codec, cap_nid, dead))
-		return;
-
 	type = get_wcaps_type(get_wcaps(codec, cap_nid));
 	if (type == AC_WID_AUD_MIX) {
 		/* Matrix-mixer style (e.g. ALC882) */
@@ -4671,8 +4656,42 @@ static void alc880_auto_init(struct hda_codec *codec)
 		alc_inithook(codec);
 }
 
-static void set_capture_mixer(struct alc_spec *spec)
+/* check the ADC/MUX contains all input pins; some ADC/MUX contains only
+ * one of two digital mic pins, e.g. on ALC272
+ */
+static void fixup_automic_adc(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	int i;
+
+	for (i = 0; i < spec->num_adc_nids; i++) {
+		hda_nid_t cap = spec->capsrc_nids ?
+			spec->capsrc_nids[i] : spec->adc_nids[i];
+		int iidx, eidx;
+
+		iidx = get_connection_index(codec, cap, spec->int_mic.pin);
+		if (iidx < 0)
+			continue;
+		eidx = get_connection_index(codec, cap, spec->ext_mic.pin);
+		if (eidx < 0)
+			continue;
+		spec->int_mic.mux_idx = iidx;
+		spec->ext_mic.mux_idx = eidx;
+		if (spec->capsrc_nids)
+			spec->capsrc_nids += i;
+		spec->adc_nids += i;
+		spec->num_adc_nids = 1;
+		return;
+	}
+	snd_printd(KERN_INFO "hda_codec: %s: "
+		   "No ADC/MUX containing both 0x%x and 0x%x pins\n",
+		   codec->chip_name, spec->int_mic.pin, spec->ext_mic.pin);
+	spec->auto_mic = 0; /* disable auto-mic to be sure */
+}
+
+static void set_capture_mixer(struct hda_codec *codec)
 {
+	struct alc_spec *spec = codec->spec;
 	static struct snd_kcontrol_new *caps[2][3] = {
 		{ alc_capture_mixer_nosrc1,
 		  alc_capture_mixer_nosrc2,
@@ -4685,7 +4704,7 @@ static void set_capture_mixer(struct alc_spec *spec)
 		int mux;
 		if (spec->auto_mic) {
 			mux = 0;
-			spec->num_adc_nids = 1; /* support only one ADC */
+			fixup_automic_adc(codec);
 		} else if (spec->input_mux && spec->input_mux->num_items > 1)
 			mux = 1;
 		else
@@ -4765,7 +4784,7 @@ static int patch_alc880(struct hda_codec *codec)
 			spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids);
 		}
 	}
-	set_capture_mixer(spec);
+	set_capture_mixer(codec);
 	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
 	spec->vmaster_nid = 0x0c;
@@ -6408,7 +6427,7 @@ static int patch_alc260(struct hda_codec *codec)
 			spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids);
 		}
 	}
-	set_capture_mixer(spec);
+	set_capture_mixer(codec);
 	set_beep_amp(spec, 0x07, 0x05, HDA_INPUT);
 
 	spec->vmaster_nid = 0x08;
@@ -9737,7 +9756,7 @@ static int patch_alc882(struct hda_codec *codec)
 		spec->capsrc_nids = spec->private_capsrc_nids;
 	}
 
-	set_capture_mixer(spec);
+	set_capture_mixer(codec);
 	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
 	spec->vmaster_nid = 0x0c;
@@ -11616,7 +11635,7 @@ static int patch_alc262(struct hda_codec *codec)
 		}
 	}
 	if (!spec->cap_mixer && !spec->no_analog)
-		set_capture_mixer(spec);
+		set_capture_mixer(codec);
 	if (!spec->no_analog)
 		set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
@@ -13285,7 +13304,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 		return err;
 
 	if (!spec->cap_mixer && !spec->no_analog)
-		set_capture_mixer(spec);
+		set_capture_mixer(codec);
 
 	alc_ssid_check(codec, 0x15, 0x1b, 0x14);
 
@@ -13483,7 +13502,7 @@ static int patch_alc269(struct hda_codec *codec)
 	spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids);
 	spec->capsrc_nids = alc269_capsrc_nids;
 	if (!spec->cap_mixer)
-		set_capture_mixer(spec);
+		set_capture_mixer(codec);
 	set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
 
 	spec->vmaster_nid = 0x02;
@@ -14398,7 +14417,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
 
 	spec->adc_nids = alc861_adc_nids;
 	spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids);
-	set_capture_mixer(spec);
+	set_capture_mixer(codec);
 
 	alc_ssid_check(codec, 0x0e, 0x0f, 0x0b);
 
@@ -15561,7 +15580,7 @@ static int patch_alc861vd(struct hda_codec *codec)
 	if (!spec->capsrc_nids)
 		spec->capsrc_nids = alc861vd_capsrc_nids;
 
-	set_capture_mixer(spec);
+	set_capture_mixer(codec);
 	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
 	spec->vmaster_nid = 0x02;
@@ -15602,9 +15621,9 @@ static hda_nid_t alc272_dac_nids[2] = {
 	0x02, 0x03
 };
 
-static hda_nid_t alc662_adc_nids[1] = {
+static hda_nid_t alc662_adc_nids[2] = {
 	/* ADC1-2 */
-	0x09,
+	0x09, 0x08
 };
 
 static hda_nid_t alc272_adc_nids[1] = {
@@ -15612,7 +15631,7 @@ static hda_nid_t alc272_adc_nids[1] = {
 	0x08,
 };
 
-static hda_nid_t alc662_capsrc_nids[1] = { 0x22 };
+static hda_nid_t alc662_capsrc_nids[2] = { 0x22, 0x23 };
 static hda_nid_t alc272_capsrc_nids[1] = { 0x23 };
 
 
@@ -17099,7 +17118,7 @@ static struct alc_config_preset alc662_presets[] = {
 		.dac_nids = alc662_dac_nids,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.adc_nids = alc662_adc_nids,
-		.num_adc_nids = ARRAY_SIZE(alc662_adc_nids),
+		.num_adc_nids = 1,
 		.capsrc_nids = alc662_capsrc_nids,
 		.channel_mode = alc662_3ST_2ch_modes,
 		.input_mux = &alc663_m51va_capture_source,
@@ -17465,7 +17484,7 @@ static int patch_alc662(struct hda_codec *codec)
 		spec->capsrc_nids = alc662_capsrc_nids;
 
 	if (!spec->cap_mixer)
-		set_capture_mixer(spec);
+		set_capture_mixer(codec);
 	if (codec->vendor_id == 0x10ec0662)
 		set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 	else
-- 
cgit v1.2.3


From 4d8e22e0f6aacb365db2cb9bf69e38310b37f418 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 11 Aug 2009 14:21:26 +0200
Subject: ALSA: hda - Add a white-list for MSI option

Created a white-list to enable MSI since some devices require MSI
explicitly due to BIOS/ACPI problems.  Simply using a quirk list.
As the first case, take HP Compaq CQ40.

Reference: Novell bnc#529971
	https://bugzilla.novell.com/show_bug.cgi?id=529971

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_intel.c | 26 +++++++++++++++++++++++++-
 1 file changed, 25 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 754de4f8201..20a66f85f0a 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2299,6 +2299,30 @@ static void __devinit check_probe_mask(struct azx *chip, int dev)
 	}
 }
 
+/*
+ * white-list for enable_msi
+ */
+static struct snd_pci_quirk msi_white_list[] __devinitdata = {
+	SND_PCI_QUIRK(0x103c, 0x3607, "HP Compa CQ40", 1),
+	{}
+};
+
+static void __devinit check_msi(struct azx *chip)
+{
+	const struct snd_pci_quirk *q;
+
+	chip->msi = enable_msi;
+	if (chip->msi)
+		return;
+	q = snd_pci_quirk_lookup(chip->pci, msi_white_list);
+	if (q) {
+		printk(KERN_INFO
+		       "hda_intel: msi for device %04x:%04x set to %d\n",
+		       q->subvendor, q->subdevice, q->value);
+		chip->msi = q->value;
+	}
+}
+
 
 /*
  * constructor
@@ -2333,7 +2357,7 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
 	chip->pci = pci;
 	chip->irq = -1;
 	chip->driver_type = driver_type;
-	chip->msi = enable_msi;
+	check_msi(chip);
 	chip->dev_index = dev;
 	INIT_WORK(&chip->irq_pending_work, azx_irq_pending_work);
 
-- 
cgit v1.2.3


From e9c364c04f59f7fb250f9689198eaa59872cf8af Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 11 Aug 2009 17:16:13 +0200
Subject: ALSA: hda - Add setup hook to ALC preset struct

Added setup hook to ALC preset struct to be called at in the parser
but not at each init callback.
This can be used for setting up the static pins, etc, while the
init hook should be used for updating the status again.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 25 +++++++++++++++----------
 1 file changed, 15 insertions(+), 10 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 5156c4f14f8..b2e097bdc59 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -379,6 +379,7 @@ struct alc_config_preset {
 	unsigned int num_mux_defs;
 	const struct hda_input_mux *input_mux;
 	void (*unsol_event)(struct hda_codec *, unsigned int);
+	void (*setup)(struct hda_codec *);
 	void (*init_hook)(struct hda_codec *);
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	struct hda_amp_list *loopbacks;
@@ -851,9 +852,10 @@ static void print_realtek_coef(struct snd_info_buffer *buffer,
 /*
  * set up from the preset table
  */
-static void setup_preset(struct alc_spec *spec,
+static void setup_preset(struct hda_codec *codec,
 			 const struct alc_config_preset *preset)
 {
+	struct alc_spec *spec = codec->spec;
 	int i;
 
 	for (i = 0; i < ARRAY_SIZE(preset->mixers) && preset->mixers[i]; i++)
@@ -895,6 +897,9 @@ static void setup_preset(struct alc_spec *spec,
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	spec->loopback.amplist = preset->loopbacks;
 #endif
+
+	if (preset->setup)
+		preset->setup(codec);
 }
 
 /* Enable GPIO mask and set output */
@@ -4762,7 +4767,7 @@ static int patch_alc880(struct hda_codec *codec)
 	}
 
 	if (board_config != ALC880_AUTO)
-		setup_preset(spec, &alc880_presets[board_config]);
+		setup_preset(codec, &alc880_presets[board_config]);
 
 	spec->stream_analog_playback = &alc880_pcm_analog_playback;
 	spec->stream_analog_capture = &alc880_pcm_analog_capture;
@@ -6406,7 +6411,7 @@ static int patch_alc260(struct hda_codec *codec)
 	}
 
 	if (board_config != ALC260_AUTO)
-		setup_preset(spec, &alc260_presets[board_config]);
+		setup_preset(codec, &alc260_presets[board_config]);
 
 	spec->stream_analog_playback = &alc260_pcm_analog_playback;
 	spec->stream_analog_capture = &alc260_pcm_analog_capture;
@@ -9720,7 +9725,7 @@ static int patch_alc882(struct hda_codec *codec)
 	}
 
 	if (board_config != ALC882_AUTO)
-		setup_preset(spec, &alc882_presets[board_config]);
+		setup_preset(codec, &alc882_presets[board_config]);
 
 	spec->stream_analog_playback = &alc882_pcm_analog_playback;
 	spec->stream_analog_capture = &alc882_pcm_analog_capture;
@@ -11594,7 +11599,7 @@ static int patch_alc262(struct hda_codec *codec)
 	}
 
 	if (board_config != ALC262_AUTO)
-		setup_preset(spec, &alc262_presets[board_config]);
+		setup_preset(codec, &alc262_presets[board_config]);
 
 	spec->stream_analog_playback = &alc262_pcm_analog_playback;
 	spec->stream_analog_capture = &alc262_pcm_analog_capture;
@@ -12659,7 +12664,7 @@ static int patch_alc268(struct hda_codec *codec)
 	}
 
 	if (board_config != ALC268_AUTO)
-		setup_preset(spec, &alc268_presets[board_config]);
+		setup_preset(codec, &alc268_presets[board_config]);
 
 	spec->stream_analog_playback = &alc268_pcm_analog_playback;
 	spec->stream_analog_capture = &alc268_pcm_analog_capture;
@@ -13483,7 +13488,7 @@ static int patch_alc269(struct hda_codec *codec)
 	}
 
 	if (board_config != ALC269_AUTO)
-		setup_preset(spec, &alc269_presets[board_config]);
+		setup_preset(codec, &alc269_presets[board_config]);
 
 	if (codec->subsystem_id == 0x17aa3bf8) {
 		/* Due to a hardware problem on Lenovo Ideadpad, we need to
@@ -14636,7 +14641,7 @@ static int patch_alc861(struct hda_codec *codec)
 	}
 
 	if (board_config != ALC861_AUTO)
-		setup_preset(spec, &alc861_presets[board_config]);
+		setup_preset(codec, &alc861_presets[board_config]);
 
 	spec->stream_analog_playback = &alc861_pcm_analog_playback;
 	spec->stream_analog_capture = &alc861_pcm_analog_capture;
@@ -15560,7 +15565,7 @@ static int patch_alc861vd(struct hda_codec *codec)
 	}
 
 	if (board_config != ALC861VD_AUTO)
-		setup_preset(spec, &alc861vd_presets[board_config]);
+		setup_preset(codec, &alc861vd_presets[board_config]);
 
 	if (codec->vendor_id == 0x10ec0660) {
 		/* always turn on EAPD */
@@ -17468,7 +17473,7 @@ static int patch_alc662(struct hda_codec *codec)
 	}
 
 	if (board_config != ALC662_AUTO)
-		setup_preset(spec, &alc662_presets[board_config]);
+		setup_preset(codec, &alc662_presets[board_config]);
 
 	spec->stream_analog_playback = &alc662_pcm_analog_playback;
 	spec->stream_analog_capture = &alc662_pcm_analog_capture;
-- 
cgit v1.2.3


From 4f5d17062085ee1c74ff44a37ffb09f94ab4db1c Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 11 Aug 2009 18:17:46 +0200
Subject: ALSA: hda - Clean up init and setup hooks for Realtek codecs

Move static codes to setup from init_hook for each model.

Also, use the common auto-mic selection helper for devices that support
auto-mic selection.  They just need to set up ext_mic, int_mic and
auto_mic flag in the setup section.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 724 ++++++++++++++++++------------------------
 1 file changed, 310 insertions(+), 414 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index b2e097bdc59..ac3207efe10 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1537,7 +1537,7 @@ static void alc_automute_amp_unsol_event(struct hda_codec *codec,
 		alc_automute_amp(codec);
 }
 
-static void alc889_automute_init(struct hda_codec *codec)
+static void alc889_automute_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
@@ -1547,16 +1547,15 @@ static void alc889_automute_init(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[2] = 0x17;
 	spec->autocfg.speaker_pins[3] = 0x19;
 	spec->autocfg.speaker_pins[4] = 0x1a;
-	alc_automute_amp(codec);
 }
 
 static void alc889_intel_init_hook(struct hda_codec *codec)
 {
 	alc889_coef_init(codec);
-	alc889_automute_init(codec);
+	alc_automute_amp(codec);
 }
 
-static void alc888_fujitsu_xa3530_init_hook(struct hda_codec *codec)
+static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
@@ -1564,7 +1563,6 @@ static void alc888_fujitsu_xa3530_init_hook(struct hda_codec *codec)
 	spec->autocfg.hp_pins[1] = 0x1b; /* hp */
 	spec->autocfg.speaker_pins[0] = 0x14; /* speaker */
 	spec->autocfg.speaker_pins[1] = 0x15; /* bass */
-	alc_automute_amp(codec);
 }
 
 /*
@@ -1763,16 +1761,15 @@ static struct snd_kcontrol_new alc888_base_mixer[] = {
 	{ } /* end */
 };
 
-static void alc888_acer_aspire_4930g_init_hook(struct hda_codec *codec)
+static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
-	alc_automute_amp(codec);
 }
 
-static void alc888_acer_aspire_6530g_init_hook(struct hda_codec *codec)
+static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
@@ -1780,10 +1777,9 @@ static void alc888_acer_aspire_6530g_init_hook(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x16;
 	spec->autocfg.speaker_pins[2] = 0x17;
-	alc_automute_amp(codec);
 }
 
-static void alc889_acer_aspire_8930g_init_hook(struct hda_codec *codec)
+static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
@@ -1791,7 +1787,6 @@ static void alc889_acer_aspire_8930g_init_hook(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x16;
 	spec->autocfg.speaker_pins[2] = 0x1b;
-	alc_automute_amp(codec);
 }
 
 /*
@@ -2771,13 +2766,17 @@ static void alc880_uniwill_mic_automute(struct hda_codec *codec)
 	snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits);
 }
 
-static void alc880_uniwill_init_hook(struct hda_codec *codec)
+static void alc880_uniwill_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x16;
+}
+
+static void alc880_uniwill_init_hook(struct hda_codec *codec)
+{
 	alc_automute_amp(codec);
 	alc880_uniwill_mic_automute(codec);
 }
@@ -2798,13 +2797,12 @@ static void alc880_uniwill_unsol_event(struct hda_codec *codec,
 	}
 }
 
-static void alc880_uniwill_p53_init_hook(struct hda_codec *codec)
+static void alc880_uniwill_p53_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x15;
-	alc_automute_amp(codec);
 }
 
 static void alc880_uniwill_p53_dcvol_automute(struct hda_codec *codec)
@@ -3067,13 +3065,12 @@ static struct hda_verb alc880_lg_init_verbs[] = {
 };
 
 /* toggle speaker-output according to the hp-jack state */
-static void alc880_lg_init_hook(struct hda_codec *codec)
+static void alc880_lg_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.speaker_pins[0] = 0x17;
-	alc_automute_amp(codec);
 }
 
 /*
@@ -3152,13 +3149,12 @@ static struct hda_verb alc880_lg_lw_init_verbs[] = {
 };
 
 /* toggle speaker-output according to the hp-jack state */
-static void alc880_lg_lw_init_hook(struct hda_codec *codec)
+static void alc880_lg_lw_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.speaker_pins[0] = 0x14;
-	alc_automute_amp(codec);
 }
 
 static struct snd_kcontrol_new alc880_medion_rim_mixer[] = {
@@ -3224,13 +3220,12 @@ static void alc880_medion_rim_unsol_event(struct hda_codec *codec,
 		alc880_medion_rim_automute(codec);
 }
 
-static void alc880_medion_rim_init_hook(struct hda_codec *codec)
+static void alc880_medion_rim_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x1b;
-	alc880_medion_rim_automute(codec);
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -4097,7 +4092,8 @@ static struct alc_config_preset alc880_presets[] = {
 		.channel_mode = alc880_2_jack_modes,
 		.input_mux = &alc880_f1734_capture_source,
 		.unsol_event = alc880_uniwill_p53_unsol_event,
-		.init_hook = alc880_uniwill_p53_init_hook,
+		.setup = alc880_uniwill_p53_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC880_ASUS] = {
 		.mixers = { alc880_asus_mixer },
@@ -4174,6 +4170,7 @@ static struct alc_config_preset alc880_presets[] = {
 		.need_dac_fix = 1,
 		.input_mux = &alc880_capture_source,
 		.unsol_event = alc880_uniwill_unsol_event,
+		.setup = alc880_uniwill_setup,
 		.init_hook = alc880_uniwill_init_hook,
 	},
 	[ALC880_UNIWILL_P53] = {
@@ -4186,7 +4183,8 @@ static struct alc_config_preset alc880_presets[] = {
 		.channel_mode = alc880_threestack_modes,
 		.input_mux = &alc880_capture_source,
 		.unsol_event = alc880_uniwill_p53_unsol_event,
-		.init_hook = alc880_uniwill_p53_init_hook,
+		.setup = alc880_uniwill_p53_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC880_FUJITSU] = {
 		.mixers = { alc880_fujitsu_mixer },
@@ -4200,7 +4198,8 @@ static struct alc_config_preset alc880_presets[] = {
 		.channel_mode = alc880_2_jack_modes,
 		.input_mux = &alc880_capture_source,
 		.unsol_event = alc880_uniwill_p53_unsol_event,
-		.init_hook = alc880_uniwill_p53_init_hook,
+		.setup = alc880_uniwill_p53_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC880_CLEVO] = {
 		.mixers = { alc880_three_stack_mixer },
@@ -4226,7 +4225,8 @@ static struct alc_config_preset alc880_presets[] = {
 		.need_dac_fix = 1,
 		.input_mux = &alc880_lg_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc880_lg_init_hook,
+		.setup = alc880_lg_setup,
+		.init_hook = alc_automute_amp,
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 		.loopbacks = alc880_lg_loopbacks,
 #endif
@@ -4242,7 +4242,8 @@ static struct alc_config_preset alc880_presets[] = {
 		.channel_mode = alc880_lg_lw_modes,
 		.input_mux = &alc880_lg_lw_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc880_lg_lw_init_hook,
+		.setup = alc880_lg_lw_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC880_MEDION_RIM] = {
 		.mixers = { alc880_medion_rim_mixer },
@@ -4256,7 +4257,8 @@ static struct alc_config_preset alc880_presets[] = {
 		.channel_mode = alc880_2_jack_modes,
 		.input_mux = &alc880_medion_rim_capture_source,
 		.unsol_event = alc880_medion_rim_unsol_event,
-		.init_hook = alc880_medion_rim_init_hook,
+		.setup = alc880_medion_rim_setup,
+		.init_hook = alc880_medion_rim_automute,
 	},
 #ifdef CONFIG_SND_DEBUG
 	[ALC880_TEST] = {
@@ -7451,23 +7453,21 @@ static struct hda_verb alc885_imac24_init_verbs[] = {
 };
 
 /* Toggle speaker-output according to the hp-jack state */
-static void alc885_imac24_automute_init_hook(struct hda_codec *codec)
+static void alc885_imac24_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x18;
 	spec->autocfg.speaker_pins[1] = 0x1a;
-	alc_automute_amp(codec);
 }
 
-static void alc885_mbp3_init_hook(struct hda_codec *codec)
+static void alc885_mbp3_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
-	alc_automute_amp(codec);
 }
 
 
@@ -7495,13 +7495,12 @@ static void alc882_targa_automute(struct hda_codec *codec)
 				  spec->jack_present ? 1 : 3);
 }
 
-static void alc882_targa_init_hook(struct hda_codec *codec)
+static void alc882_targa_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x1b;
-	alc882_targa_automute(codec);
 }
 
 static void alc882_targa_unsol_event(struct hda_codec *codec, unsigned int res)
@@ -7589,7 +7588,7 @@ static void alc885_macpro_init_hook(struct hda_codec *codec)
 static void alc885_imac24_init_hook(struct hda_codec *codec)
 {
 	alc885_macpro_init_hook(codec);
-	alc885_imac24_automute_init_hook(codec);
+	alc_automute_amp(codec);
 }
 
 /*
@@ -8079,14 +8078,13 @@ static struct snd_kcontrol_new alc883_chmode_mixer[] = {
 };
 
 /* toggle speaker-output according to the hp-jack state */
-static void alc883_mitac_init_hook(struct hda_codec *codec)
+static void alc883_mitac_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x17;
-	alc_automute_amp(codec);
 }
 
 /* auto-toggle front mic */
@@ -8241,7 +8239,7 @@ static struct hda_verb alc883_vaiott_verbs[] = {
 	{ } /* end */
 };
 
-static void alc888_3st_hp_init_hook(struct hda_codec *codec)
+static void alc888_3st_hp_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
@@ -8249,7 +8247,6 @@ static void alc888_3st_hp_init_hook(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x16;
 	spec->autocfg.speaker_pins[2] = 0x18;
-	alc_automute_amp(codec);
 }
 
 static struct hda_verb alc888_3st_hp_verbs[] = {
@@ -8346,13 +8343,12 @@ static struct hda_verb alc883_medion_md2_verbs[] = {
 };
 
 /* toggle speaker-output according to the hp-jack state */
-static void alc883_medion_md2_init_hook(struct hda_codec *codec)
+static void alc883_medion_md2_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x15;
-	alc_automute_amp(codec);
 }
 
 /* toggle speaker-output according to the hp-jack state */
@@ -8369,12 +8365,16 @@ static void alc883_clevo_m720_mic_automute(struct hda_codec *codec)
 				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
 }
 
-static void alc883_clevo_m720_init_hook(struct hda_codec *codec)
+static void alc883_clevo_m720_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
+}
+
+static void alc883_clevo_m720_init_hook(struct hda_codec *codec)
+{
 	alc_automute_amp(codec);
 	alc883_clevo_m720_mic_automute(codec);
 }
@@ -8393,22 +8393,20 @@ static void alc883_clevo_m720_unsol_event(struct hda_codec *codec,
 }
 
 /* toggle speaker-output according to the hp-jack state */
-static void alc883_2ch_fujitsu_pi2515_init_hook(struct hda_codec *codec)
+static void alc883_2ch_fujitsu_pi2515_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x15;
-	alc_automute_amp(codec);
 }
 
-static void alc883_haier_w66_init_hook(struct hda_codec *codec)
+static void alc883_haier_w66_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.speaker_pins[0] = 0x14;
-	alc_automute_amp(codec);
 }
 
 static void alc883_lenovo_101e_ispeaker_automute(struct hda_codec *codec)
@@ -8447,14 +8445,13 @@ static void alc883_lenovo_101e_unsol_event(struct hda_codec *codec,
 }
 
 /* toggle speaker-output according to the hp-jack state */
-static void alc883_acer_aspire_init_hook(struct hda_codec *codec)
+static void alc883_acer_aspire_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[1] = 0x16;
-	alc_automute_amp(codec);
 }
 
 static struct hda_verb alc883_acer_eapd_verbs[] = {
@@ -8475,7 +8472,7 @@ static struct hda_verb alc883_acer_eapd_verbs[] = {
 	{ }
 };
 
-static void alc888_6st_dell_init_hook(struct hda_codec *codec)
+static void alc888_6st_dell_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
@@ -8484,10 +8481,9 @@ static void alc888_6st_dell_init_hook(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[1] = 0x15;
 	spec->autocfg.speaker_pins[2] = 0x16;
 	spec->autocfg.speaker_pins[3] = 0x17;
-	alc_automute_amp(codec);
 }
 
-static void alc888_lenovo_sky_init_hook(struct hda_codec *codec)
+static void alc888_lenovo_sky_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
@@ -8497,17 +8493,15 @@ static void alc888_lenovo_sky_init_hook(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[2] = 0x16;
 	spec->autocfg.speaker_pins[3] = 0x17;
 	spec->autocfg.speaker_pins[4] = 0x1a;
-	alc_automute_amp(codec);
 }
 
-static void alc883_vaiott_init_hook(struct hda_codec *codec)
+static void alc883_vaiott_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x17;
-	alc_automute_amp(codec);
 }
 
 static struct hda_verb alc888_asus_m90v_verbs[] = {
@@ -8520,19 +8514,7 @@ static struct hda_verb alc888_asus_m90v_verbs[] = {
 	{ } /* 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_init_hook(struct hda_codec *codec)
+static void alc883_mode2_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
@@ -8540,26 +8522,11 @@ static void alc883_M90V_init_hook(struct hda_codec *codec)
 	spec->autocfg.speaker_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[1] = 0x15;
 	spec->autocfg.speaker_pins[2] = 0x16;
-	alc_automute_pin(codec);
-}
-
-static void alc883_mode2_unsol_event(struct hda_codec *codec,
-					   unsigned int res)
-{
-	switch (res >> 26) {
-	case ALC880_MIC_EVENT:
-		alc883_nb_mic_automute(codec);
-		break;
-	default:
-		alc_sku_unsol_event(codec, res);
-		break;
-	}
-}
-
-static void alc883_mode2_inithook(struct hda_codec *codec)
-{
-	alc883_M90V_init_hook(codec);
-	alc883_nb_mic_automute(codec);
+	spec->ext_mic.pin = 0x18;
+	spec->int_mic.pin = 0x19;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.mux_idx = 1;
+	spec->auto_mic = 1;
 }
 
 static struct hda_verb alc888_asus_eee1601_verbs[] = {
@@ -8890,7 +8857,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.dig_out_nid = ALC882_DIGOUT_NID,
 		.dig_in_nid = ALC882_DIGIN_NID,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc885_mbp3_init_hook,
+		.setup = alc885_mbp3_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC885_MB5] = {
 		.mixers = { alc885_mb5_mixer, alc882_chmode_mixer },
@@ -8927,6 +8895,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc882_ch_modes,
 		.input_mux = &alc882_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
+		.setup = alc885_imac24_setup,
 		.init_hook = alc885_imac24_init_hook,
 	},
 	[ALC882_TARGA] = {
@@ -8944,7 +8913,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.need_dac_fix = 1,
 		.input_mux = &alc882_capture_source,
 		.unsol_event = alc882_targa_unsol_event,
-		.init_hook = alc882_targa_init_hook,
+		.setup = alc882_targa_setup,
+		.init_hook = alc882_targa_automute,
 	},
 	[ALC882_ASUS_A7J] = {
 		.mixers = { alc882_asus_a7j_mixer, alc882_chmode_mixer },
@@ -9035,7 +9005,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc889_8ch_intel_modes,
 		.capsrc_nids = alc889_capsrc_nids,
 		.input_mux = &alc889_capture_source,
-		.init_hook = alc889_automute_init,
+		.setup = alc889_automute_setup,
+		.init_hook = alc_automute_amp,
 		.unsol_event = alc_automute_amp_unsol_event,
 		.need_dac_fix = 1,
 	},
@@ -9054,6 +9025,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc889_8ch_intel_modes,
 		.capsrc_nids = alc889_capsrc_nids,
 		.input_mux = &alc889_capture_source,
+		.setup = alc889_automute_setup,
 		.init_hook = alc889_intel_init_hook,
 		.unsol_event = alc_automute_amp_unsol_event,
 		.need_dac_fix = 1,
@@ -9081,7 +9053,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.need_dac_fix = 1,
 		.input_mux = &alc883_capture_source,
 		.unsol_event = alc883_targa_unsol_event,
-		.init_hook = alc883_targa_init_hook,
+		.setup = alc882_targa_setup,
+		.init_hook = alc882_targa_automute,
 	},
 	[ALC883_TARGA_2ch_DIG] = {
 		.mixers = { alc883_targa_2ch_mixer},
@@ -9096,7 +9069,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc883_3ST_2ch_modes,
 		.input_mux = &alc883_capture_source,
 		.unsol_event = alc883_targa_unsol_event,
-		.init_hook = alc883_targa_init_hook,
+		.setup = alc882_targa_setup,
+		.init_hook = alc882_targa_automute,
 	},
 	[ALC883_TARGA_8ch_DIG] = {
 		.mixers = { alc883_base_mixer, alc883_chmode_mixer },
@@ -9114,7 +9088,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.need_dac_fix = 1,
 		.input_mux = &alc883_capture_source,
 		.unsol_event = alc883_targa_unsol_event,
-		.init_hook = alc883_targa_init_hook,
+		.setup = alc882_targa_setup,
+		.init_hook = alc882_targa_automute,
 	},
 	[ALC883_ACER] = {
 		.mixers = { alc883_base_mixer },
@@ -9140,7 +9115,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc883_3ST_2ch_modes,
 		.input_mux = &alc883_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc883_acer_aspire_init_hook,
+		.setup = alc883_acer_aspire_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC888_ACER_ASPIRE_4930G] = {
 		.mixers = { alc888_base_mixer,
@@ -9160,7 +9136,8 @@ static struct alc_config_preset alc882_presets[] = {
 			ARRAY_SIZE(alc888_2_capture_sources),
 		.input_mux = alc888_2_capture_sources,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc888_acer_aspire_4930g_init_hook,
+		.setup = alc888_acer_aspire_4930g_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC888_ACER_ASPIRE_6530G] = {
 		.mixers = { alc888_acer_aspire_6530_mixer },
@@ -9178,7 +9155,8 @@ static struct alc_config_preset alc882_presets[] = {
 			ARRAY_SIZE(alc888_2_capture_sources),
 		.input_mux = alc888_acer_aspire_6530_sources,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc888_acer_aspire_6530g_init_hook,
+		.setup = alc888_acer_aspire_6530g_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC888_ACER_ASPIRE_8930G] = {
 		.mixers = { alc888_base_mixer,
@@ -9199,7 +9177,8 @@ static struct alc_config_preset alc882_presets[] = {
 			ARRAY_SIZE(alc889_capture_sources),
 		.input_mux = alc889_capture_sources,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc889_acer_aspire_8930g_init_hook,
+		.setup = alc889_acer_aspire_8930g_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC883_MEDION] = {
 		.mixers = { alc883_fivestack_mixer,
@@ -9224,7 +9203,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc883_3ST_2ch_modes,
 		.input_mux = &alc883_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc883_medion_md2_init_hook,
+		.setup = alc883_medion_md2_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC883_LAPTOP_EAPD] = {
 		.mixers = { alc883_base_mixer },
@@ -9245,6 +9225,7 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc883_3ST_2ch_modes,
 		.input_mux = &alc883_capture_source,
 		.unsol_event = alc883_clevo_m720_unsol_event,
+		.setup = alc883_clevo_m720_setup,
 		.init_hook = alc883_clevo_m720_init_hook,
 	},
 	[ALC883_LENOVO_101E_2ch] = {
@@ -9270,7 +9251,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.need_dac_fix = 1,
 		.input_mux = &alc883_lenovo_nb0763_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc883_medion_md2_init_hook,
+		.setup = alc883_medion_md2_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC888_LENOVO_MS7195_DIG] = {
 		.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
@@ -9295,7 +9277,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc883_3ST_2ch_modes,
 		.input_mux = &alc883_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc883_haier_w66_init_hook,
+		.setup = alc883_haier_w66_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC888_3ST_HP] = {
 		.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
@@ -9307,7 +9290,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.need_dac_fix = 1,
 		.input_mux = &alc883_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc888_3st_hp_init_hook,
+		.setup = alc888_3st_hp_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC888_6ST_DELL] = {
 		.mixers = { alc883_base_mixer, alc883_chmode_mixer },
@@ -9320,7 +9304,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc883_sixstack_modes,
 		.input_mux = &alc883_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc888_6st_dell_init_hook,
+		.setup = alc888_6st_dell_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC883_MITAC] = {
 		.mixers = { alc883_mitac_mixer },
@@ -9331,7 +9316,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc883_3ST_2ch_modes,
 		.input_mux = &alc883_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc883_mitac_init_hook,
+		.setup = alc883_mitac_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC883_FUJITSU_PI2515] = {
 		.mixers = { alc883_2ch_fujitsu_pi2515_mixer },
@@ -9344,7 +9330,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc883_3ST_2ch_modes,
 		.input_mux = &alc883_fujitsu_pi2515_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc883_2ch_fujitsu_pi2515_init_hook,
+		.setup = alc883_2ch_fujitsu_pi2515_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC888_FUJITSU_XA3530] = {
 		.mixers = { alc888_base_mixer, alc883_chmode_mixer },
@@ -9362,7 +9349,8 @@ static struct alc_config_preset alc882_presets[] = {
 			ARRAY_SIZE(alc888_2_capture_sources),
 		.input_mux = alc888_2_capture_sources,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc888_fujitsu_xa3530_init_hook,
+		.setup = alc888_fujitsu_xa3530_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC888_LENOVO_SKY] = {
 		.mixers = { alc888_lenovo_sky_mixer, alc883_chmode_mixer },
@@ -9375,7 +9363,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.need_dac_fix = 1,
 		.input_mux = &alc883_lenovo_sky_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc888_lenovo_sky_init_hook,
+		.setup = alc888_lenovo_sky_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC888_ASUS_M90V] = {
 		.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
@@ -9388,8 +9377,9 @@ static struct alc_config_preset alc882_presets[] = {
 		.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,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc883_mode2_setup,
+		.init_hook = alc_inithook,
 	},
 	[ALC888_ASUS_EEE1601] = {
 		.mixers = { alc883_asus_eee1601_mixer },
@@ -9442,7 +9432,8 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc883_3ST_2ch_modes,
 		.input_mux = &alc883_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc883_vaiott_init_hook,
+		.setup = alc883_vaiott_setup,
+		.init_hook = alc_automute_amp,
 	},
 };
 
@@ -9954,13 +9945,12 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_option_mixer[] = {
 };
 
 /* mute/unmute internal speaker according to the hp jack and mute state */
-static void alc262_hp_t5735_init_hook(struct hda_codec *codec)
+static void alc262_hp_t5735_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x0c; /* HACK: not actually a pin */
-	alc_automute_amp(codec);
 }
 
 static struct snd_kcontrol_new alc262_hp_t5735_mixer[] = {
@@ -10117,22 +10107,20 @@ static void alc262_hippo_unsol_event(struct hda_codec *codec, unsigned int res)
 	alc262_hippo_automute(codec);
 }
 
-static void alc262_hippo_init_hook(struct hda_codec *codec)
+static void alc262_hippo_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
-	alc262_hippo_automute(codec);
 }
 
-static void alc262_hippo1_init_hook(struct hda_codec *codec)
+static void alc262_hippo1_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.speaker_pins[0] = 0x14;
-	alc262_hippo_automute(codec);
 }
 
 
@@ -10189,13 +10177,12 @@ static struct hda_verb alc262_tyan_verbs[] = {
 };
 
 /* unsolicited event for HP jack sensing */
-static void alc262_tyan_init_hook(struct hda_codec *codec)
+static void alc262_tyan_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.speaker_pins[0] = 0x15;
-	alc_automute_amp(codec);
 }
 
 
@@ -10307,14 +10294,6 @@ 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),
@@ -10336,35 +10315,17 @@ static struct hda_verb alc262_toshiba_s06_verbs[] = {
 	{}
 };
 
-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);
-}
-
-
-/* unsolicited event for HP jack sensing */
-static void alc262_toshiba_s06_unsol_event(struct hda_codec *codec,
-				       unsigned int res)
-{
-	if ((res >> 26) == ALC880_MIC_EVENT)
-		alc262_dmic_automute(codec);
-	else
-		alc_sku_unsol_event(codec, res);
-}
-
-static void alc262_toshiba_s06_init_hook(struct hda_codec *codec)
+static void alc262_toshiba_s06_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
-	alc_automute_pin(codec);
-	alc262_dmic_automute(codec);
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x12;
+	spec->int_mic.mux_idx = 9;
+	spec->auto_mic = 1;
 }
 
 /*
@@ -11337,7 +11298,8 @@ static struct alc_config_preset alc262_presets[] = {
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_capture_source,
 		.unsol_event = alc262_hippo_unsol_event,
-		.init_hook = alc262_hippo_init_hook,
+		.setup = alc262_hippo_setup,
+		.init_hook = alc262_hippo_automute,
 	},
 	[ALC262_HIPPO_1] = {
 		.mixers = { alc262_hippo1_mixer },
@@ -11350,7 +11312,8 @@ static struct alc_config_preset alc262_presets[] = {
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_capture_source,
 		.unsol_event = alc262_hippo_unsol_event,
-		.init_hook = alc262_hippo1_init_hook,
+		.setup = alc262_hippo1_setup,
+		.init_hook = alc262_hippo_automute,
 	},
 	[ALC262_FUJITSU] = {
 		.mixers = { alc262_fujitsu_mixer },
@@ -11413,7 +11376,8 @@ static struct alc_config_preset alc262_presets[] = {
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc262_hp_t5735_init_hook,
+		.setup = alc262_hp_t5735_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC262_HP_RP5700] = {
 		.mixers = { alc262_hp_rp5700_mixer },
@@ -11444,7 +11408,8 @@ static struct alc_config_preset alc262_presets[] = {
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_capture_source,
 		.unsol_event = alc262_hippo_unsol_event,
-		.init_hook = alc262_hippo_init_hook,
+		.setup = alc262_hippo_setup,
+		.init_hook = alc262_hippo_automute,
 	},
 	[ALC262_BENQ_T31] = {
 		.mixers = { alc262_benq_t31_mixer },
@@ -11457,7 +11422,8 @@ static struct alc_config_preset alc262_presets[] = {
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_capture_source,
 		.unsol_event = alc262_hippo_unsol_event,
-		.init_hook = alc262_hippo_init_hook,
+		.setup = alc262_hippo_setup,
+		.init_hook = alc262_hippo_automute,
 	},
 	[ALC262_ULTRA] = {
 		.mixers = { alc262_ultra_mixer },
@@ -11509,9 +11475,9 @@ static struct alc_config_preset alc262_presets[] = {
 		.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,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc262_toshiba_s06_setup,
+		.init_hook = alc_inithook,
 	},
 	[ALC262_TOSHIBA_RX1] = {
 		.mixers = { alc262_toshiba_rx1_mixer },
@@ -11523,7 +11489,8 @@ static struct alc_config_preset alc262_presets[] = {
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_capture_source,
 		.unsol_event = alc262_hippo_unsol_event,
-		.init_hook = alc262_hippo_init_hook,
+		.setup = alc262_hippo_setup,
+		.init_hook = alc262_hippo_automute,
 	},
 	[ALC262_TYAN] = {
 		.mixers = { alc262_tyan_mixer },
@@ -11536,7 +11503,8 @@ static struct alc_config_preset alc262_presets[] = {
 		.channel_mode = alc262_modes,
 		.input_mux = &alc262_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc262_tyan_init_hook,
+		.setup = alc262_tyan_setup,
+		.init_hook = alc_automute_amp,
 	},
 };
 
@@ -11732,14 +11700,6 @@ 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 = {
@@ -11858,7 +11818,8 @@ static struct hda_verb alc268_acer_verbs[] = {
 
 /* unsolicited event for HP jack sensing */
 #define alc268_toshiba_unsol_event	alc262_hippo_unsol_event
-#define alc268_toshiba_init_hook	alc262_hippo_init_hook
+#define alc268_toshiba_setup		alc262_hippo_setup
+#define alc268_toshiba_automute		alc262_hippo_automute
 
 static void alc268_acer_unsol_event(struct hda_codec *codec,
 				       unsigned int res)
@@ -11888,30 +11849,33 @@ static void alc268_aspire_one_speaker_automute(struct hda_codec *codec)
 				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)
+	switch (res >> 26) {
+	case ALC880_HP_EVENT:
 		alc268_aspire_one_speaker_automute(codec);
-	if ((res >> 26) == ALC880_MIC_EVENT)
-		alc268_acer_mic_automute(codec);
+		break;
+	case ALC880_MIC_EVENT:
+		alc_mic_automute(codec);
+		break;
+	}
+}
+
+static void alc268_acer_lc_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x12;
+	spec->int_mic.mux_idx = 6;
+	spec->auto_mic = 1;
 }
 
 static void alc268_acer_lc_init_hook(struct hda_codec *codec)
 {
 	alc268_aspire_one_speaker_automute(codec);
-	alc268_acer_mic_automute(codec);
+	alc_mic_automute(codec);
 }
 
 static struct snd_kcontrol_new alc268_dell_mixer[] = {
@@ -11929,17 +11893,22 @@ static struct hda_verb alc268_dell_verbs[] = {
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{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, ALC880_MIC_EVENT | AC_USRSP_EN},
 	{ }
 };
 
 /* mute/unmute internal speaker according to the hp jack and mute state */
-static void alc268_dell_init_hook(struct hda_codec *codec)
+static void alc268_dell_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
-	alc_automute_pin(codec);
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x19;
+	spec->int_mic.mux_idx = 1;
+	spec->auto_mic = 1;
 }
 
 static struct snd_kcontrol_new alc267_quanta_il1_mixer[] = {
@@ -11960,38 +11929,16 @@ static struct hda_verb alc267_quanta_il1_verbs[] = {
 	{ }
 };
 
-static void alc267_quanta_il1_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 ? 0x00 : 0x01);
-}
-
-static void alc267_quanta_il1_init_hook(struct hda_codec *codec)
+static void alc267_quanta_il1_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
-	alc_automute_pin(codec);
-	alc267_quanta_il1_mic_automute(codec);
-}
-
-static void alc267_quanta_il1_unsol_event(struct hda_codec *codec,
-					   unsigned int res)
-{
-	switch (res >> 26) {
-	case ALC880_MIC_EVENT:
-		alc267_quanta_il1_mic_automute(codec);
-		break;
-	default:
-		alc_sku_unsol_event(codec, res);
-		break;
-	}
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x19;
+	spec->int_mic.mux_idx = 1;
+	spec->auto_mic = 1;
 }
 
 /*
@@ -12490,8 +12437,9 @@ static struct alc_config_preset alc268_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc268_modes),
 		.channel_mode = alc268_modes,
 		.input_mux = &alc268_capture_source,
-		.unsol_event = alc267_quanta_il1_unsol_event,
-		.init_hook = alc267_quanta_il1_init_hook,
+		.unsol_event = alc_sku_unsol_event,
+		.setup = alc267_quanta_il1_setup,
+		.init_hook = alc_inithook,
 	},
 	[ALC268_3ST] = {
 		.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
@@ -12523,7 +12471,8 @@ static struct alc_config_preset alc268_presets[] = {
 		.channel_mode = alc268_modes,
 		.input_mux = &alc268_capture_source,
 		.unsol_event = alc268_toshiba_unsol_event,
-		.init_hook = alc268_toshiba_init_hook,
+		.setup = alc268_toshiba_setup,
+		.init_hook = alc268_toshiba_automute,
 	},
 	[ALC268_ACER] = {
 		.mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
@@ -12573,8 +12522,8 @@ static struct alc_config_preset alc268_presets[] = {
 		.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,
+		.setup = alc268_acer_lc_setup,
 		.init_hook = alc268_acer_lc_init_hook,
 	},
 	[ALC268_DELL] = {
@@ -12587,8 +12536,8 @@ static struct alc_config_preset alc268_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc268_modes),
 		.channel_mode = alc268_modes,
 		.unsol_event = alc_sku_unsol_event,
-		.init_hook = alc268_dell_init_hook,
-		.input_mux = &alc268_capture_source,
+		.setup = alc268_dell_setup,
+		.init_hook = alc_inithook,
 	},
 	[ALC268_ZEPTO] = {
 		.mixers = { alc268_base_mixer, alc268_capture_alt_mixer,
@@ -12605,8 +12554,8 @@ static struct alc_config_preset alc268_presets[] = {
 		.num_channel_mode = ARRAY_SIZE(alc268_modes),
 		.channel_mode = alc268_modes,
 		.input_mux = &alc268_capture_source,
-		.unsol_event = alc268_toshiba_unsol_event,
-		.init_hook = alc268_toshiba_init_hook
+		.setup = alc268_toshiba_setup,
+		.init_hook = alc268_toshiba_automute,
 	},
 #ifdef CONFIG_SND_DEBUG
 	[ALC268_TEST] = {
@@ -12750,22 +12699,6 @@ static hda_nid_t alc269_capsrc_nids[1] = {
  *       not a mux!
  */
 
-static struct hda_input_mux alc269_eeepc_dmic_capture_source = {
-	.num_items = 2,
-	.items = {
-		{ "i-Mic", 0x5 },
-		{ "e-Mic", 0x0 },
-	},
-};
-
-static struct hda_input_mux alc269_eeepc_amic_capture_source = {
-	.num_items = 2,
-	.items = {
-		{ "i-Mic", 0x1 },
-		{ "e-Mic", 0x0 },
-	},
-};
-
 #define alc269_modes		alc260_modes
 #define alc269_capture_source	alc880_lg_lw_capture_source
 
@@ -12927,16 +12860,6 @@ static void alc269_lifebook_speaker_automute(struct hda_codec *codec)
 			AC_VERB_SET_PROC_COEF, 0x480);
 }
 
-static void alc269_quanta_fl1_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 : 0x1);
-}
-
 static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec)
 {
 	unsigned int present_laptop;
@@ -12963,10 +12886,14 @@ static void alc269_lifebook_mic_autoswitch(struct hda_codec *codec)
 static void alc269_quanta_fl1_unsol_event(struct hda_codec *codec,
 				    unsigned int res)
 {
-	if ((res >> 26) == ALC880_HP_EVENT)
+	switch (res >> 26) {
+	case ALC880_HP_EVENT:
 		alc269_quanta_fl1_speaker_automute(codec);
-	if ((res >> 26) == ALC880_MIC_EVENT)
-		alc269_quanta_fl1_mic_automute(codec);
+		break;
+	case ALC880_MIC_EVENT:
+		alc_mic_automute(codec);
+		break;
+	}
 }
 
 static void alc269_lifebook_unsol_event(struct hda_codec *codec,
@@ -12978,10 +12905,20 @@ static void alc269_lifebook_unsol_event(struct hda_codec *codec,
 		alc269_lifebook_mic_autoswitch(codec);
 }
 
+static void alc269_quanta_fl1_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x19;
+	spec->int_mic.mux_idx = 1;
+	spec->auto_mic = 1;
+}
+
 static void alc269_quanta_fl1_init_hook(struct hda_codec *codec)
 {
 	alc269_quanta_fl1_speaker_automute(codec);
-	alc269_quanta_fl1_mic_automute(codec);
+	alc_mic_automute(codec);
 }
 
 static void alc269_lifebook_init_hook(struct hda_codec *codec)
@@ -13026,60 +12963,44 @@ static void alc269_speaker_automute(struct hda_codec *codec)
 				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) & 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) & 0x80000000;
-	snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-				0x7000 | (0x00 << 8) | (present ? 0 : 0x80));
-	snd_hda_codec_write(codec, 0x24, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-				0x7000 | (0x01 << 8) | (present ? 0x80 : 0));
-}
-
 /* unsolicited event for HP jack sensing */
-static void alc269_eeepc_dmic_unsol_event(struct hda_codec *codec,
+static void alc269_eeepc_unsol_event(struct hda_codec *codec,
 				     unsigned int res)
 {
-	if ((res >> 26) == ALC880_HP_EVENT)
+	switch (res >> 26) {
+	case ALC880_HP_EVENT:
 		alc269_speaker_automute(codec);
-
-	if ((res >> 26) == ALC880_MIC_EVENT)
-		alc269_eeepc_dmic_automute(codec);
+		break;
+	case ALC880_MIC_EVENT:
+		alc_mic_automute(codec);
+		break;
+	}
 }
 
-static void alc269_eeepc_dmic_inithook(struct hda_codec *codec)
+static void alc269_eeepc_dmic_setup(struct hda_codec *codec)
 {
-	alc269_speaker_automute(codec);
-	alc269_eeepc_dmic_automute(codec);
+	struct alc_spec *spec = codec->spec;
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x12;
+	spec->int_mic.mux_idx = 5;
+	spec->auto_mic = 1;
 }
 
-/* unsolicited event for HP jack sensing */
-static void alc269_eeepc_amic_unsol_event(struct hda_codec *codec,
-				     unsigned int res)
+static void alc269_eeepc_amic_setup(struct hda_codec *codec)
 {
-	if ((res >> 26) == ALC880_HP_EVENT)
-		alc269_speaker_automute(codec);
-
-	if ((res >> 26) == ALC880_MIC_EVENT)
-		alc269_eeepc_amic_automute(codec);
+	struct alc_spec *spec = codec->spec;
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x19;
+	spec->int_mic.mux_idx = 1;
+	spec->auto_mic = 1;
 }
 
-static void alc269_eeepc_amic_inithook(struct hda_codec *codec)
+static void alc269_eeepc_inithook(struct hda_codec *codec)
 {
 	alc269_speaker_automute(codec);
-	alc269_eeepc_amic_automute(codec);
+	alc_mic_automute(codec);
 }
 
 /*
@@ -13385,6 +13306,7 @@ static struct alc_config_preset alc269_presets[] = {
 		.channel_mode = alc269_modes,
 		.input_mux = &alc269_capture_source,
 		.unsol_event = alc269_quanta_fl1_unsol_event,
+		.setup = alc269_quanta_fl1_setup,
 		.init_hook = alc269_quanta_fl1_init_hook,
 	},
 	[ALC269_ASUS_EEEPC_P703] = {
@@ -13397,9 +13319,9 @@ static struct alc_config_preset alc269_presets[] = {
 		.hp_nid = 0x03,
 		.num_channel_mode = ARRAY_SIZE(alc269_modes),
 		.channel_mode = alc269_modes,
-		.input_mux = &alc269_eeepc_amic_capture_source,
-		.unsol_event = alc269_eeepc_amic_unsol_event,
-		.init_hook = alc269_eeepc_amic_inithook,
+		.unsol_event = alc269_eeepc_unsol_event,
+		.setup = alc269_eeepc_amic_setup,
+		.init_hook = alc269_eeepc_inithook,
 	},
 	[ALC269_ASUS_EEEPC_P901] = {
 		.mixers = { alc269_eeepc_mixer },
@@ -13411,9 +13333,9 @@ static struct alc_config_preset alc269_presets[] = {
 		.hp_nid = 0x03,
 		.num_channel_mode = ARRAY_SIZE(alc269_modes),
 		.channel_mode = alc269_modes,
-		.input_mux = &alc269_eeepc_dmic_capture_source,
-		.unsol_event = alc269_eeepc_dmic_unsol_event,
-		.init_hook = alc269_eeepc_dmic_inithook,
+		.unsol_event = alc269_eeepc_unsol_event,
+		.setup = alc269_eeepc_dmic_setup,
+		.init_hook = alc269_eeepc_inithook,
 	},
 	[ALC269_FUJITSU] = {
 		.mixers = { alc269_fujitsu_mixer },
@@ -13425,9 +13347,9 @@ static struct alc_config_preset alc269_presets[] = {
 		.hp_nid = 0x03,
 		.num_channel_mode = ARRAY_SIZE(alc269_modes),
 		.channel_mode = alc269_modes,
-		.input_mux = &alc269_eeepc_dmic_capture_source,
-		.unsol_event = alc269_eeepc_dmic_unsol_event,
-		.init_hook = alc269_eeepc_dmic_inithook,
+		.unsol_event = alc269_eeepc_unsol_event,
+		.setup = alc269_eeepc_dmic_setup,
+		.init_hook = alc269_eeepc_inithook,
 	},
 	[ALC269_LIFEBOOK] = {
 		.mixers = { alc269_lifebook_mixer },
@@ -15044,12 +14966,15 @@ static void alc861vd_lenovo_mic_automute(struct hda_codec *codec)
 				 HDA_AMP_MUTE, bits);
 }
 
-static void alc861vd_lenovo_init_hook(struct hda_codec *codec)
+static void alc861vd_lenovo_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-
 	spec->autocfg.hp_pins[0] = 0x1b;
 	spec->autocfg.speaker_pins[0] = 0x14;
+}
+
+static void alc861vd_lenovo_init_hook(struct hda_codec *codec)
+{
 	alc_automute_amp(codec);
 	alc861vd_lenovo_mic_automute(codec);
 }
@@ -15113,13 +15038,12 @@ static struct hda_verb alc861vd_dallas_verbs[] = {
 };
 
 /* toggle speaker-output according to the hp-jack state */
-static void alc861vd_dallas_init_hook(struct hda_codec *codec)
+static void alc861vd_dallas_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x15;
 	spec->autocfg.speaker_pins[0] = 0x14;
-	alc_automute_amp(codec);
 }
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -15233,6 +15157,7 @@ static struct alc_config_preset alc861vd_presets[] = {
 		.channel_mode = alc861vd_3stack_2ch_modes,
 		.input_mux = &alc861vd_capture_source,
 		.unsol_event = alc861vd_lenovo_unsol_event,
+		.setup = alc861vd_lenovo_setup,
 		.init_hook = alc861vd_lenovo_init_hook,
 	},
 	[ALC861VD_DALLAS] = {
@@ -15244,7 +15169,8 @@ static struct alc_config_preset alc861vd_presets[] = {
 		.channel_mode = alc861vd_3stack_2ch_modes,
 		.input_mux = &alc861vd_dallas_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc861vd_dallas_init_hook,
+		.setup = alc861vd_dallas_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC861VD_HP] = {
 		.mixers = { alc861vd_hp_mixer },
@@ -15256,7 +15182,8 @@ static struct alc_config_preset alc861vd_presets[] = {
 		.channel_mode = alc861vd_3stack_2ch_modes,
 		.input_mux = &alc861vd_hp_capture_source,
 		.unsol_event = alc_automute_amp_unsol_event,
-		.init_hook = alc861vd_dallas_init_hook,
+		.setup = alc861vd_dallas_setup,
+		.init_hook = alc_automute_amp,
 	},
 	[ALC660VD_ASUS_V1S] = {
 		.mixers = { alc861vd_lenovo_mixer },
@@ -15271,6 +15198,7 @@ static struct alc_config_preset alc861vd_presets[] = {
 		.channel_mode = alc861vd_3stack_2ch_modes,
 		.input_mux = &alc861vd_capture_source,
 		.unsol_event = alc861vd_lenovo_unsol_event,
+		.setup = alc861vd_lenovo_setup,
 		.init_hook = alc861vd_lenovo_init_hook,
 	},
 };
@@ -15660,14 +15588,6 @@ static struct hda_input_mux alc662_lenovo_101e_capture_source = {
 	},
 };
 
-static struct hda_input_mux alc662_eeepc_capture_source = {
-	.num_items = 2,
-	.items = {
-		{ "i-Mic", 0x1 },
-		{ "e-Mic", 0x0 },
-	},
-};
-
 static struct hda_input_mux alc663_capture_source = {
 	.num_items = 3,
 	.items = {
@@ -15677,23 +15597,7 @@ static struct hda_input_mux alc663_capture_source = {
 	},
 };
 
-static struct hda_input_mux alc663_m51va_capture_source = {
-	.num_items = 2,
-	.items = {
-		{ "Ext-Mic", 0x0 },
-		{ "D-Mic", 0x9 },
-	},
-};
-
-#if 1 /* set to 0 for testing other input sources below */
-static struct hda_input_mux alc272_nc10_capture_source = {
-	.num_items = 2,
-	.items = {
-		{ "Autoselect Mic", 0x0 },
-		{ "Internal Mic", 0x1 },
-	},
-};
-#else
+#if 0 /* set to 1 for testing other input sources below */
 static struct hda_input_mux alc272_nc10_capture_source = {
 	.num_items = 16,
 	.items = {
@@ -16362,47 +16266,44 @@ static void alc662_lenovo_101e_unsol_event(struct hda_codec *codec,
 		alc662_lenovo_101e_ispeaker_automute(codec);
 }
 
-static void alc662_eeepc_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, 0x22, 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 | (0x00 << 8) | (present ? 0 : 0x80));
-	snd_hda_codec_write(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-			    0x7000 | (0x01 << 8) | (present ? 0x80 : 0));
-	snd_hda_codec_write(codec, 0x23, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-			    0x7000 | (0x01 << 8) | (present ? 0x80 : 0));
-}
-
 /* unsolicited event for HP jack sensing */
 static void alc662_eeepc_unsol_event(struct hda_codec *codec,
 				     unsigned int res)
 {
 	if ((res >> 26) == ALC880_MIC_EVENT)
-		alc662_eeepc_mic_automute(codec);
+		alc_mic_automute(codec);
 	else
 		alc262_hippo_unsol_event(codec, res);
 }
 
+static void alc662_eeepc_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+
+	alc262_hippo1_setup(codec);
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x19;
+	spec->int_mic.mux_idx = 1;
+	spec->auto_mic = 1;
+}
+
 static void alc662_eeepc_inithook(struct hda_codec *codec)
 {
-	alc262_hippo1_init_hook(codec);
-	alc662_eeepc_mic_automute(codec);
+	alc262_hippo_automute(codec);
+	alc_mic_automute(codec);
 }
 
-static void alc662_eeepc_ep20_inithook(struct hda_codec *codec)
+static void alc662_eeepc_ep20_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x14;
 	spec->autocfg.speaker_pins[0] = 0x1b;
-	alc262_hippo_master_update(codec);
 }
 
+#define alc662_eeepc_ep20_inithook	alc262_hippo_master_update
+
 static void alc663_m51va_speaker_automute(struct hda_codec *codec)
 {
 	unsigned int present;
@@ -16513,23 +16414,6 @@ static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec)
 	}
 }
 
-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;
-	snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-			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));
-	snd_hda_codec_write_cache(codec, 0x22, 0, AC_VERB_SET_AMP_GAIN_MUTE,
-			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));
-}
-
 static void alc663_m51va_unsol_event(struct hda_codec *codec,
 					   unsigned int res)
 {
@@ -16538,36 +16422,32 @@ static void alc663_m51va_unsol_event(struct hda_codec *codec,
 		alc663_m51va_speaker_automute(codec);
 		break;
 	case ALC880_MIC_EVENT:
-		alc663_m51va_mic_automute(codec);
+		alc_mic_automute(codec);
 		break;
 	}
 }
 
+static void alc663_m51va_setup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	spec->ext_mic.pin = 0x18;
+	spec->ext_mic.mux_idx = 0;
+	spec->int_mic.pin = 0x12;
+	spec->int_mic.mux_idx = 1;
+	spec->auto_mic = 1;
+}
+
 static void alc663_m51va_inithook(struct hda_codec *codec)
 {
 	alc663_m51va_speaker_automute(codec);
-	alc663_m51va_mic_automute(codec);
+	alc_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;
-	}
-}
+#define alc663_mode1_unsol_event	alc663_m51va_unsol_event
+#define alc663_mode1_setup		alc663_m51va_setup
+#define alc663_mode1_inithook		alc663_m51va_inithook
 
-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)
@@ -16577,15 +16457,17 @@ static void alc662_mode2_unsol_event(struct hda_codec *codec,
 		alc662_f5z_speaker_automute(codec);
 		break;
 	case ALC880_MIC_EVENT:
-		alc662_eeepc_mic_automute(codec);
+		alc_mic_automute(codec);
 		break;
 	}
 }
 
+#define alc662_mode2_setup	alc663_m51va_setup
+
 static void alc662_mode2_inithook(struct hda_codec *codec)
 {
 	alc662_f5z_speaker_automute(codec);
-	alc662_eeepc_mic_automute(codec);
+	alc_mic_automute(codec);
 }
 /* ***************** Mode3 ******************************/
 static void alc663_mode3_unsol_event(struct hda_codec *codec,
@@ -16596,15 +16478,17 @@ static void alc663_mode3_unsol_event(struct hda_codec *codec,
 		alc663_two_hp_m1_speaker_automute(codec);
 		break;
 	case ALC880_MIC_EVENT:
-		alc662_eeepc_mic_automute(codec);
+		alc_mic_automute(codec);
 		break;
 	}
 }
 
+#define alc663_mode3_setup	alc663_m51va_setup
+
 static void alc663_mode3_inithook(struct hda_codec *codec)
 {
 	alc663_two_hp_m1_speaker_automute(codec);
-	alc662_eeepc_mic_automute(codec);
+	alc_mic_automute(codec);
 }
 /* ***************** Mode4 ******************************/
 static void alc663_mode4_unsol_event(struct hda_codec *codec,
@@ -16615,15 +16499,17 @@ static void alc663_mode4_unsol_event(struct hda_codec *codec,
 		alc663_21jd_two_speaker_automute(codec);
 		break;
 	case ALC880_MIC_EVENT:
-		alc662_eeepc_mic_automute(codec);
+		alc_mic_automute(codec);
 		break;
 	}
 }
 
+#define alc663_mode4_setup	alc663_m51va_setup
+
 static void alc663_mode4_inithook(struct hda_codec *codec)
 {
 	alc663_21jd_two_speaker_automute(codec);
-	alc662_eeepc_mic_automute(codec);
+	alc_mic_automute(codec);
 }
 /* ***************** Mode5 ******************************/
 static void alc663_mode5_unsol_event(struct hda_codec *codec,
@@ -16634,15 +16520,17 @@ static void alc663_mode5_unsol_event(struct hda_codec *codec,
 		alc663_15jd_two_speaker_automute(codec);
 		break;
 	case ALC880_MIC_EVENT:
-		alc662_eeepc_mic_automute(codec);
+		alc_mic_automute(codec);
 		break;
 	}
 }
 
+#define alc663_mode5_setup	alc663_m51va_setup
+
 static void alc663_mode5_inithook(struct hda_codec *codec)
 {
 	alc663_15jd_two_speaker_automute(codec);
-	alc662_eeepc_mic_automute(codec);
+	alc_mic_automute(codec);
 }
 /* ***************** Mode6 ******************************/
 static void alc663_mode6_unsol_event(struct hda_codec *codec,
@@ -16653,15 +16541,17 @@ static void alc663_mode6_unsol_event(struct hda_codec *codec,
 		alc663_two_hp_m2_speaker_automute(codec);
 		break;
 	case ALC880_MIC_EVENT:
-		alc662_eeepc_mic_automute(codec);
+		alc_mic_automute(codec);
 		break;
 	}
 }
 
+#define alc663_mode6_setup	alc663_m51va_setup
+
 static void alc663_mode6_inithook(struct hda_codec *codec)
 {
 	alc663_two_hp_m2_speaker_automute(codec);
-	alc662_eeepc_mic_automute(codec);
+	alc_mic_automute(codec);
 }
 
 static void alc663_g71v_hp_automute(struct hda_codec *codec)
@@ -16703,16 +16593,18 @@ static void alc663_g71v_unsol_event(struct hda_codec *codec,
 		alc663_g71v_front_automute(codec);
 		break;
 	case ALC880_MIC_EVENT:
-		alc662_eeepc_mic_automute(codec);
+		alc_mic_automute(codec);
 		break;
 	}
 }
 
+#define alc663_g71v_setup	alc663_m51va_setup
+
 static void alc663_g71v_inithook(struct hda_codec *codec)
 {
 	alc663_g71v_front_automute(codec);
 	alc663_g71v_hp_automute(codec);
-	alc662_eeepc_mic_automute(codec);
+	alc_mic_automute(codec);
 }
 
 static void alc663_g50v_unsol_event(struct hda_codec *codec,
@@ -16723,15 +16615,17 @@ static void alc663_g50v_unsol_event(struct hda_codec *codec,
 		alc663_m51va_speaker_automute(codec);
 		break;
 	case ALC880_MIC_EVENT:
-		alc662_eeepc_mic_automute(codec);
+		alc_mic_automute(codec);
 		break;
 	}
 }
 
+#define alc663_g50v_setup	alc663_m51va_setup
+
 static void alc663_g50v_inithook(struct hda_codec *codec)
 {
 	alc663_m51va_speaker_automute(codec);
-	alc662_eeepc_mic_automute(codec);
+	alc_mic_automute(codec);
 }
 
 static struct snd_kcontrol_new alc662_ecs_mixer[] = {
@@ -16935,8 +16829,8 @@ static struct alc_config_preset alc662_presets[] = {
 		.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,
+		.setup = alc662_eeepc_setup,
 		.init_hook = alc662_eeepc_inithook,
 	},
 	[ALC662_ASUS_EEEPC_EP20] = {
@@ -16950,6 +16844,7 @@ static struct alc_config_preset alc662_presets[] = {
 		.channel_mode = alc662_3ST_6ch_modes,
 		.input_mux = &alc662_lenovo_101e_capture_source,
 		.unsol_event = alc662_eeepc_unsol_event,
+		.setup = alc662_eeepc_ep20_setup,
 		.init_hook = alc662_eeepc_ep20_inithook,
 	},
 	[ALC662_ECS] = {
@@ -16960,8 +16855,8 @@ static struct alc_config_preset alc662_presets[] = {
 		.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,
+		.setup = alc662_eeepc_setup,
 		.init_hook = alc662_eeepc_inithook,
 	},
 	[ALC663_ASUS_M51VA] = {
@@ -16972,8 +16867,8 @@ static struct alc_config_preset alc662_presets[] = {
 		.dig_out_nid = ALC662_DIGOUT_NID,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.channel_mode = alc662_3ST_2ch_modes,
-		.input_mux = &alc663_m51va_capture_source,
 		.unsol_event = alc663_m51va_unsol_event,
+		.setup = alc663_m51va_setup,
 		.init_hook = alc663_m51va_inithook,
 	},
 	[ALC663_ASUS_G71V] = {
@@ -16984,8 +16879,8 @@ static struct alc_config_preset alc662_presets[] = {
 		.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_g71v_unsol_event,
+		.setup = alc663_g71v_setup,
 		.init_hook = alc663_g71v_inithook,
 	},
 	[ALC663_ASUS_H13] = {
@@ -16995,7 +16890,6 @@ static struct alc_config_preset alc662_presets[] = {
 		.dac_nids = alc662_dac_nids,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.channel_mode = alc662_3ST_2ch_modes,
-		.input_mux = &alc663_m51va_capture_source,
 		.unsol_event = alc663_m51va_unsol_event,
 		.init_hook = alc663_m51va_inithook,
 	},
@@ -17009,6 +16903,7 @@ static struct alc_config_preset alc662_presets[] = {
 		.channel_mode = alc662_3ST_6ch_modes,
 		.input_mux = &alc663_capture_source,
 		.unsol_event = alc663_g50v_unsol_event,
+		.setup = alc663_g50v_setup,
 		.init_hook = alc663_g50v_inithook,
 	},
 	[ALC663_ASUS_MODE1] = {
@@ -17022,8 +16917,8 @@ static struct alc_config_preset alc662_presets[] = {
 		.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,
+		.setup = alc663_mode1_setup,
 		.init_hook = alc663_mode1_inithook,
 	},
 	[ALC662_ASUS_MODE2] = {
@@ -17036,8 +16931,8 @@ static struct alc_config_preset alc662_presets[] = {
 		.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,
+		.setup = alc662_mode2_setup,
 		.init_hook = alc662_mode2_inithook,
 	},
 	[ALC663_ASUS_MODE3] = {
@@ -17051,8 +16946,8 @@ static struct alc_config_preset alc662_presets[] = {
 		.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,
+		.setup = alc663_mode3_setup,
 		.init_hook = alc663_mode3_inithook,
 	},
 	[ALC663_ASUS_MODE4] = {
@@ -17066,8 +16961,8 @@ static struct alc_config_preset alc662_presets[] = {
 		.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,
+		.setup = alc663_mode4_setup,
 		.init_hook = alc663_mode4_inithook,
 	},
 	[ALC663_ASUS_MODE5] = {
@@ -17081,8 +16976,8 @@ static struct alc_config_preset alc662_presets[] = {
 		.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,
+		.setup = alc663_mode5_setup,
 		.init_hook = alc663_mode5_inithook,
 	},
 	[ALC663_ASUS_MODE6] = {
@@ -17096,8 +16991,8 @@ static struct alc_config_preset alc662_presets[] = {
 		.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,
+		.setup = alc663_mode6_setup,
 		.init_hook = alc663_mode6_inithook,
 	},
 	[ALC272_DELL] = {
@@ -17111,8 +17006,8 @@ static struct alc_config_preset alc662_presets[] = {
 		.num_adc_nids = ARRAY_SIZE(alc272_adc_nids),
 		.capsrc_nids = alc272_capsrc_nids,
 		.channel_mode = alc662_3ST_2ch_modes,
-		.input_mux = &alc663_m51va_capture_source,
 		.unsol_event = alc663_m51va_unsol_event,
+		.setup = alc663_m51va_setup,
 		.init_hook = alc663_m51va_inithook,
 	},
 	[ALC272_DELL_ZM1] = {
@@ -17126,8 +17021,8 @@ static struct alc_config_preset alc662_presets[] = {
 		.num_adc_nids = 1,
 		.capsrc_nids = alc662_capsrc_nids,
 		.channel_mode = alc662_3ST_2ch_modes,
-		.input_mux = &alc663_m51va_capture_source,
 		.unsol_event = alc663_m51va_unsol_event,
+		.setup = alc663_m51va_setup,
 		.init_hook = alc663_m51va_inithook,
 	},
 	[ALC272_SAMSUNG_NC10] = {
@@ -17138,8 +17033,9 @@ static struct alc_config_preset alc662_presets[] = {
 		.dac_nids = alc272_dac_nids,
 		.num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes),
 		.channel_mode = alc662_3ST_2ch_modes,
-		.input_mux = &alc272_nc10_capture_source,
+		/*.input_mux = &alc272_nc10_capture_source,*/
 		.unsol_event = alc663_mode4_unsol_event,
+		.setup = alc663_mode4_setup,
 		.init_hook = alc663_mode4_inithook,
 	},
 };
-- 
cgit v1.2.3


From 5908589f318835f515bf94829650d36dcb2d9024 Mon Sep 17 00:00:00 2001
From: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Date: Tue, 11 Aug 2009 22:33:09 -0300
Subject: ALSA: hda - fix noise issue when recording from digital mic with
 alc268

With auto config model of alc268 realtek codec, it allows to select any
of possible available digital microphone inputs when only one is
available. For example, when only digital mic in nid 0x12 is available,
on second input source it will allow you to select unavailable digital
mic in nid 0x13. The problem is that selecting unavailable digital mic
creates a source of noise when recording (I'm not sure if this happens
on all machines with alc268 and only one digital mic input, but testing
on a quanta uw1 netbook a lot of noise is introduced in recording from
digital mic 0x12/first input source, when you select the unavailable
digital mic 0x13 for capture source 0x24 in the second input source in
mixer).

Then to avoid noise when recording from digital mic with auto model in
this case, prevent a digital mic input source to be selected if
microphone is not available.

Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 62 +++++++++++++++++++++++++------------------
 1 file changed, 36 insertions(+), 26 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ac3207efe10..7ff293458f6 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12232,32 +12232,40 @@ static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec,
 						const struct auto_pin_cfg *cfg)
 {
 	struct hda_input_mux *imux = &spec->private_imux[0];
-	int i, idx1;
-
-	for (i = 0; i < AUTO_PIN_LAST; i++) {
-		switch(cfg->input_pins[i]) {
-		case 0x18:
-			idx1 = 0;	/* Mic 1 */
-			break;
-		case 0x19:
-			idx1 = 1;	/* Mic 2 */
-			break;
-		case 0x1a:
-			idx1 = 2;	/* Line In */
-			break;
-		case 0x1c:
-			idx1 = 3;	/* CD */
-			break;
-		case 0x12:
-		case 0x13:
-			idx1 = 6;	/* digital mics */
-			break;
-		default:
-			continue;
+	int i, idx1, dmic_nid;
+
+	dmic_nid = 0x12;
+	while (dmic_nid <= 0x13) {
+		for (i = 0; i < AUTO_PIN_LAST; i++) {
+			switch (cfg->input_pins[i]) {
+			case 0x18:
+				idx1 = 0;	/* Mic 1 */
+				break;
+			case 0x19:
+				idx1 = 1;	/* Mic 2 */
+				break;
+			case 0x1a:
+				idx1 = 2;	/* Line In */
+				break;
+			case 0x1c:
+				idx1 = 3;	/* CD */
+				break;
+			case 0x12:
+			case 0x13:
+				if (cfg->input_pins[i] != dmic_nid)
+					continue;
+				idx1 = 6;	/* digital mics */
+				break;
+			default:
+				continue;
+			}
+			imux->items[imux->num_items].label =
+				auto_pin_cfg_labels[i];
+			imux->items[imux->num_items].index = idx1;
+			imux->num_items++;
 		}
-		imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
-		imux->items[imux->num_items].index = idx1;
-		imux->num_items++;
+		imux++;
+		dmic_nid++;
 	}
 	return 0;
 }
@@ -12355,7 +12363,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
 		add_mixer(spec, alc268_beep_mixer);
 
 	add_verb(spec, alc268_volume_init_verbs);
-	spec->num_mux_defs = 1;
+	spec->num_mux_defs = 2;
 	spec->input_mux = &spec->private_imux[0];
 
 	err = alc_auto_add_mic_boost(codec);
@@ -12665,6 +12673,8 @@ static int patch_alc268(struct hda_codec *codec)
 		for (i = 0; i < spec->num_adc_nids; i++)
 			snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i],
 				0, AC_VERB_SET_CONNECT_SEL,
+				i < spec->num_mux_defs ?
+				spec->input_mux[i].items[0].index :
 				spec->input_mux->items[0].index);
 	}
 
-- 
cgit v1.2.3


From 1c4bdf9be010ae7c2324c0a90dd2296e0d1a775e Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 13 Aug 2009 08:23:24 +0200
Subject: ALSA: hda - Enable line-out detection only with speakers

Enable line-out detection for IDT/STAC codecs only when speaker pins
exist.  In some cases, the speaker itself is identified as line-out,
and this confuses the situation.  Only the extra line-outs should do
auto-muting.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 3 ++-
 1 file changed, 2 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 27607306057..3ac0112566b 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -4350,7 +4350,8 @@ static int stac92xx_init(struct hda_codec *codec)
 			hda_nid_t nid = cfg->hp_pins[i];
 			enable_pin_detect(codec, nid, STAC_HP_EVENT);
 		}
-		if (cfg->line_out_type == AUTO_PIN_LINE_OUT) {
+		if (cfg->line_out_type == AUTO_PIN_LINE_OUT &&
+		    cfg->speaker_outs > 0) {
 			/* enable pin-detect for line-outs as well */
 			for (i = 0; i < cfg->line_outs; i++) {
 				hda_nid_t nid = cfg->line_out_pins[i];
-- 
cgit v1.2.3


From 667067d8980249a71ccf82a55202fff2cd1cd54f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 13 Aug 2009 18:14:42 +0200
Subject: ALSA: hda - Fix / clean up IDT92HD83xxx codec parser

A few improvements for IDT 92HD83xxx codec pareser:
- Remove unused / deprecated mixer-amp controls
- Handle d-mics as normal inputs since this codec has no separate
  MUXes for analog and digital
- Don't create duplicated controls for capture volumes with Mux
  capture volumes

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 149 +++++++++++++++++++----------------------
 1 file changed, 69 insertions(+), 80 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index f6d7ef452d7..7d33e0a76c1 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -359,14 +359,9 @@ static unsigned long stac92hd73xx_capvols[] = {
 };
 #define stac92hd73xx_capsws	stac92hd73xx_capvols
 
-#define STAC92HD83XXX_NUM_DMICS	2
-static hda_nid_t stac92hd83xxx_dmic_nids[STAC92HD83XXX_NUM_DMICS + 1] = {
-	0x11, 0x12, 0
-};
-
 #define STAC92HD83_DAC_COUNT 3
 
-static hda_nid_t stac92hd83xxx_dmux_nids[2] = {
+static hda_nid_t stac92hd83xxx_mux_nids[2] = {
 	0x17, 0x18,
 };
 
@@ -386,10 +381,6 @@ static unsigned int stac92hd83xxx_pwr_mapping[4] = {
 	0x03, 0x0c, 0x20, 0x40,
 };
 
-static hda_nid_t stac92hd83xxx_amp_nids[1] = {
-	0xc,
-};
-
 #define STAC92HD83XXX_NUM_CAPS	2
 static unsigned long stac92hd83xxx_capvols[] = {
 	HDA_COMPOSE_AMP_VAL(0x17, 3, 0, HDA_OUTPUT),
@@ -1212,26 +1203,6 @@ static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = {
 };
 
 
-static struct snd_kcontrol_new stac92hd83xxx_mixer[] = {
-	HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x1b, 0x3, HDA_INPUT),
-	HDA_CODEC_MUTE("DAC0 Capture Switch", 0x1b, 0x3, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x1b, 0x4, HDA_INPUT),
-	HDA_CODEC_MUTE("DAC1 Capture Switch", 0x1b, 0x4, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Front Mic Capture Volume", 0x1b, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Capture Switch", 0x1b, 0x0, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Line In Capture Volume", 0x1b, 0x2, HDA_INPUT),
-	HDA_CODEC_MUTE("Line In Capture Switch", 0x1b, 0x2, HDA_INPUT),
-
-	/*
-	HDA_CODEC_VOLUME("Mic Capture Volume", 0x1b, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Capture Switch", 0x1b 0x1, HDA_INPUT),
-	*/
-	{ } /* end */
-};
-
 static struct snd_kcontrol_new stac92hd71bxx_loopback[] = {
 	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2)
 };
@@ -3521,19 +3492,33 @@ static int stac92xx_beep_switch_ctl(struct hda_codec *codec)
 static int stac92xx_auto_create_mux_input_ctls(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	int wcaps, nid, i, err = 0;
+	int i, j, err = 0;
 
 	for (i = 0; i < spec->num_muxes; i++) {
+		hda_nid_t nid;
+		unsigned int wcaps;
+		unsigned long val;
+
 		nid = spec->mux_nids[i];
 		wcaps = get_wcaps(codec, nid);
+		if (!(wcaps & AC_WCAP_OUT_AMP))
+			continue;
 
-		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;
+		/* check whether already the same control was created as
+		 * normal Capture Volume.
+		 */
+		val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
+		for (j = 0; j < spec->num_caps; j++) {
+			if (spec->capvols[j] == val)
+				break;
 		}
+		if (j < spec->num_caps)
+			continue;
+
+		err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_VOL, i,
+					       "Mux Capture Volume", val);
+		if (err < 0)
+			return err;
 	}
 	return 0;
 };
@@ -3588,6 +3573,24 @@ static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
 	return -1;
 }
 
+/* create a volume assigned to the given pin (only if supported) */
+static int create_elem_capture_vol(struct hda_codec *codec, hda_nid_t nid,
+				   const char *label)
+{
+	unsigned int caps, nums;
+	char name[32];
+
+	if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP))
+		return 0;
+	caps = query_amp_caps(codec, nid, HDA_OUTPUT);
+	nums = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT;
+	if (!nums)
+		return 0;
+	snprintf(name, sizeof(name), "%s Capture Volume", label);
+	return stac92xx_add_control(codec->spec, STAC_CTL_WIDGET_VOL, name,
+				    HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
+}
+
 /* create playback/capture controls for input pins on dmic capable codecs */
 static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
@@ -3597,7 +3600,6 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
 	struct hda_input_mux *dimux = &spec->private_dimux;
 	int err, i, active_mics;
 	unsigned int def_conf;
-	char name[32];
 
 	dimux->items[dimux->num_items].label = stac92xx_dmic_labels[0];
 	dimux->items[dimux->num_items].index = 0;
@@ -3605,6 +3607,10 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
 
 	active_mics = 0;
 	for (i = 0; i < spec->num_dmics; i++) {
+		/* check the validity: sometimes it's a dead vendor-spec node */
+		if (get_wcaps_type(get_wcaps(codec, spec->dmic_nids[i]))
+		    != AC_WID_PIN)
+			continue;
 		def_conf = snd_hda_codec_get_pincfg(codec, spec->dmic_nids[i]);
 		if (get_defcfg_connect(def_conf) != AC_JACK_PORT_NONE)
 			active_mics++;
@@ -3613,14 +3619,15 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
 	for (i = 0; i < spec->num_dmics; i++) {
 		hda_nid_t nid;
 		int index;
-		unsigned int wcaps;
 		const char *label;
 
-		def_conf = snd_hda_codec_get_pincfg(codec, spec->dmic_nids[i]);
+		nid = spec->dmic_nids[i];
+		if (get_wcaps_type(get_wcaps(codec, nid)) != AC_WID_PIN)
+			continue;
+		def_conf = snd_hda_codec_get_pincfg(codec, nid);
 		if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
 			continue;
 
-		nid = spec->dmic_nids[i];
 		index = get_connection_index(codec, spec->dmux_nids[0], nid);
 		if (index < 0)
 			continue;
@@ -3630,21 +3637,9 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
 		else
 			label = stac92xx_dmic_labels[dimux->num_items];
 
-		wcaps = get_wcaps(codec, nid) &
-			(AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP);
-
-		if (wcaps) {
-			sprintf(name, "%s Capture Volume", label);
-
-			err = stac92xx_add_control(spec,
-				STAC_CTL_WIDGET_VOL,
-				name,
-				HDA_COMPOSE_AMP_VAL(nid, 3, 0,
-				(wcaps & AC_WCAP_OUT_AMP) ?
-				HDA_OUTPUT : HDA_INPUT));
-			if (err < 0)
-				return err;
-		}
+		err = create_elem_capture_vol(codec, nid, label);
+		if (err < 0)
+			return err;
 
 		dimux->items[dimux->num_items].label = label;
 		dimux->items[dimux->num_items].index = index;
@@ -3746,29 +3741,29 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const
 {
 	struct sigmatel_spec *spec = codec->spec;
 	struct hda_input_mux *imux = &spec->private_imux;
-	hda_nid_t con_lst[HDA_MAX_NUM_INPUTS];
-	int i, j, k;
+	int i, j;
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
-		int index;
+		hda_nid_t nid = cfg->input_pins[i];
+		int index, err;
 
-		if (!cfg->input_pins[i])
+		if (!nid)
 			continue;
 		index = -1;
 		for (j = 0; j < spec->num_muxes; j++) {
-			int num_cons;
-			num_cons = snd_hda_get_connections(codec,
-							   spec->mux_nids[j],
-							   con_lst,
-							   HDA_MAX_NUM_INPUTS);
-			for (k = 0; k < num_cons; k++)
-				if (con_lst[k] == cfg->input_pins[i]) {
-					index = k;
-					goto found;
-				}
+			index = get_connection_index(codec, spec->mux_nids[j],
+						     nid);
+			if (index >= 0)
+				break;
 		}
-		continue;
-	found:
+		if (index < 0)
+			continue;
+
+		err = create_elem_capture_vol(codec, nid,
+					      auto_pin_cfg_labels[i]);
+		if (err < 0)
+			return err;
+
 		imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
 		imux->items[imux->num_items].index = index;
 		imux->num_items++;
@@ -5299,22 +5294,16 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 	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->mux_nids = stac92hd83xxx_mux_nids;
+	spec->num_muxes = ARRAY_SIZE(stac92hd83xxx_mux_nids);
 	spec->adc_nids = stac92hd83xxx_adc_nids;
 	spec->pwr_nids = stac92hd83xxx_pwr_nids;
-	spec->amp_nids = stac92hd83xxx_amp_nids;
 	spec->pwr_mapping = stac92hd83xxx_pwr_mapping;
 	spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids);
 	spec->multiout.dac_nids = spec->dac_nids;
 
 	spec->init = stac92hd83xxx_core_init;
-	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_amps = ARRAY_SIZE(stac92hd83xxx_amp_nids);
-	spec->num_dmics = STAC92HD83XXX_NUM_DMICS;
 	spec->pin_nids = stac92hd83xxx_pin_nids;
 	spec->num_caps = STAC92HD83XXX_NUM_CAPS;
 	spec->capvols = stac92hd83xxx_capvols;
-- 
cgit v1.2.3


From 7570ef18349d4f3a9336397952b05ec3e95ec4b7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sat, 15 Aug 2009 11:57:53 +0200
Subject: ALSA: hda - Add missing num_adc_nids definition for IDT92HD8xxx

The previous fix removed the definition of num_adc_nids wrongly, and
this resulted in the missing input-source control.  Now readded again.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 7d33e0a76c1..eb9ca6f7a77 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5297,6 +5297,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 	spec->mux_nids = stac92hd83xxx_mux_nids;
 	spec->num_muxes = ARRAY_SIZE(stac92hd83xxx_mux_nids);
 	spec->adc_nids = stac92hd83xxx_adc_nids;
+	spec->num_adcs = ARRAY_SIZE(stac92hd83xxx_adc_nids);
 	spec->pwr_nids = stac92hd83xxx_pwr_nids;
 	spec->pwr_mapping = stac92hd83xxx_pwr_mapping;
 	spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids);
-- 
cgit v1.2.3


From fdbc66266c21976027938642f60e0f047149a61a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 19 Aug 2009 00:18:10 +0200
Subject: ALSA: hda - Fix invalid capture mixers with some ALC268 models

The auto-mic clean-up patches caused regressions on some ALC268 models
that have no proper input_mux but with "Input Source" mixer elements.
Such a combination results in Oops when accessed.

[A reason why set_capture_mixer() isn't used in patch_alc268() is that
ALC268 codec have HDA_OUTPUT direction for capture volumes unlike other
codecs.  Thus it needs own definitions of capture elements.]

This patch fixes the issues:
- Add a capture mixer definition without input-source
- Use the new capture mixer appropriately

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 54 ++++++++++++++++++-------------------------
 1 file changed, 23 insertions(+), 31 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7ff293458f6..efb92c83c7a 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12018,21 +12018,16 @@ static struct hda_verb alc268_volume_init_verbs[] = {
 	{ }
 };
 
+static struct snd_kcontrol_new alc268_capture_nosrc_mixer[] = {
+	HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
+	{ } /* end */
+};
+
 static struct snd_kcontrol_new alc268_capture_alt_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x23, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 1,
-		.info = alc_mux_enum_info,
-		.get = alc_mux_enum_get,
-		.put = alc_mux_enum_put,
-	},
+	_DEFINE_CAPSRC(1),
 	{ } /* end */
 };
 
@@ -12041,18 +12036,7 @@ static struct snd_kcontrol_new alc268_capture_mixer[] = {
 	HDA_CODEC_MUTE("Capture Switch", 0x23, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x24, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x24, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = alc_mux_enum_info,
-		.get = alc_mux_enum_get,
-		.put = alc_mux_enum_put,
-	},
+	_DEFINE_CAPSRC(2),
 	{ } /* end */
 };
 
@@ -12434,7 +12418,8 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
 
 static struct alc_config_preset alc268_presets[] = {
 	[ALC267_QUANTA_IL1] = {
-		.mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer },
+		.mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer,
+			    alc268_capture_nosrc_mixer },
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc267_quanta_il1_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -12444,7 +12429,6 @@ static struct alc_config_preset alc268_presets[] = {
 		.hp_nid = 0x03,
 		.num_channel_mode = ARRAY_SIZE(alc268_modes),
 		.channel_mode = alc268_modes,
-		.input_mux = &alc268_capture_source,
 		.unsol_event = alc_sku_unsol_event,
 		.setup = alc267_quanta_il1_setup,
 		.init_hook = alc_inithook,
@@ -12483,7 +12467,7 @@ static struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc268_toshiba_automute,
 	},
 	[ALC268_ACER] = {
-		.mixers = { alc268_acer_mixer, alc268_capture_alt_mixer,
+		.mixers = { alc268_acer_mixer, alc268_capture_nosrc_mixer,
 			    alc268_beep_mixer },
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_acer_verbs },
@@ -12519,7 +12503,7 @@ static struct alc_config_preset alc268_presets[] = {
 	[ALC268_ACER_ASPIRE_ONE] = {
 		.mixers = { alc268_acer_aspire_one_mixer,
 			    alc268_beep_mixer,
-			    alc268_capture_alt_mixer },
+			    alc268_capture_nosrc_mixer },
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_acer_aspire_one_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -12535,11 +12519,15 @@ static struct alc_config_preset alc268_presets[] = {
 		.init_hook = alc268_acer_lc_init_hook,
 	},
 	[ALC268_DELL] = {
-		.mixers = { alc268_dell_mixer, alc268_beep_mixer },
+		.mixers = { alc268_dell_mixer, alc268_beep_mixer,
+			    alc268_capture_nosrc_mixer },
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_dell_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 = 0x02,
 		.num_channel_mode = ARRAY_SIZE(alc268_modes),
 		.channel_mode = alc268_modes,
@@ -12659,10 +12647,14 @@ static int patch_alc268(struct hda_codec *codec)
 
 		/* get type */
 		wcap = get_wcaps_type(wcap);
-		if (wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
+		if (spec->auto_mic ||
+		    wcap != AC_WID_AUD_IN || spec->input_mux->num_items == 1) {
 			spec->adc_nids = alc268_adc_nids_alt;
 			spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids_alt);
-			add_mixer(spec, alc268_capture_alt_mixer);
+			if (spec->auto_mic || spec->input_mux->num_items == 1)
+				add_mixer(spec, alc268_capture_nosrc_mixer);
+			else
+				add_mixer(spec, alc268_capture_alt_mixer);
 		} else {
 			spec->adc_nids = alc268_adc_nids;
 			spec->num_adc_nids = ARRAY_SIZE(alc268_adc_nids);
-- 
cgit v1.2.3


From fd72d008465d45db63d7f8931eea6a04e05916d6 Mon Sep 17 00:00:00 2001
From: Wu Fengguang <fengguang.wu@intel.com>
Date: Mon, 24 Aug 2009 09:50:46 +0800
Subject: ALSA: hda: move open coded tricks into get_wcaps_channels()

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_local.h     | 10 ++++++++++
 sound/pci/hda/hda_proc.c      |  4 +---
 sound/pci/hda/patch_atihdmi.c |  3 +--
 3 files changed, 12 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index fa57cb93b44..5f1dcc59002 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -410,6 +410,16 @@ static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid)
 /* get the widget type from widget capability bits */
 #define get_wcaps_type(wcaps) (((wcaps) & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT)
 
+static inline unsigned int get_wcaps_channels(u32 wcaps)
+{
+	unsigned int chans;
+
+	chans = (wcaps & AC_WCAP_CHAN_CNT_EXT) >> 13;
+	chans = ((chans << 1) | 1) + 1;
+
+	return chans;
+}
+
 u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction);
 int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
 			      unsigned int caps);
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index a721eb08a29..95f24e4729f 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -515,9 +515,7 @@ 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) {
-			unsigned int chans;
-			chans = (wid_caps & AC_WCAP_CHAN_CNT_EXT) >> 13;
-			chans = ((chans << 1) | 1) + 1;
+			unsigned int chans = get_wcaps_channels(wid_caps);
 			if (chans == 2)
 				snd_iprintf(buffer, " Stereo");
 			else
diff --git a/sound/pci/hda/patch_atihdmi.c b/sound/pci/hda/patch_atihdmi.c
index 233e4778bba..fb684f00156 100644
--- a/sound/pci/hda/patch_atihdmi.c
+++ b/sound/pci/hda/patch_atihdmi.c
@@ -141,8 +141,7 @@ static int atihdmi_build_pcms(struct hda_codec *codec)
 	/* 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;
+	chans = get_wcaps_channels(chans);
 	info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = chans;
 
 	return 0;
-- 
cgit v1.2.3


From 9d0b71b1cf7b7ff856d200841de3f3d7c64eb170 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 24 Aug 2009 14:10:30 +0200
Subject: ALSA: hda - Reuse ALC268 parser for ALC269

Reuse a part of the code of ALC268 parser for ALC269.
This will change the default output volume either to Front or Speaker
depending on the pin configuration.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 83 ++-----------------------------------------
 1 file changed, 2 insertions(+), 81 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 084e291e12d..dd25db93677 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -13086,87 +13086,8 @@ static struct hda_verb alc269_init_verbs[] = {
 	{ }
 };
 
-/* 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)
-{
-	hda_nid_t nid;
-	int err;
-
-	spec->multiout.num_dacs = 1;	/* only use one dac */
-	spec->multiout.dac_nids = spec->private_dac_nids;
-	spec->multiout.dac_nids[0] = 2;
-
-	nid = cfg->line_out_pins[0];
-	if (nid) {
-		err = add_control(spec, ALC_CTL_WIDGET_VOL,
-				  "Front Playback Volume",
-				  HDA_COMPOSE_AMP_VAL(0x02, 3, 0, HDA_OUTPUT));
-		if (err < 0)
-			return err;
-		err = add_control(spec, ALC_CTL_WIDGET_MUTE,
-				  "Front Playback Switch",
-				  HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
-		if (err < 0)
-			return err;
-	}
-
-	nid = cfg->speaker_pins[0];
-	if (nid) {
-		if (!cfg->line_out_pins[0]) {
-			err = add_control(spec, ALC_CTL_WIDGET_VOL,
-					  "Speaker Playback Volume",
-					  HDA_COMPOSE_AMP_VAL(0x02, 3, 0,
-							      HDA_OUTPUT));
-			if (err < 0)
-				return err;
-		}
-		if (nid == 0x16) {
-			err = add_control(spec, ALC_CTL_WIDGET_MUTE,
-					  "Speaker Playback Switch",
-					  HDA_COMPOSE_AMP_VAL(nid, 2, 0,
-							      HDA_OUTPUT));
-			if (err < 0)
-				return err;
-		} else {
-			err = add_control(spec, ALC_CTL_WIDGET_MUTE,
-					  "Speaker Playback Switch",
-					  HDA_COMPOSE_AMP_VAL(nid, 3, 0,
-							      HDA_OUTPUT));
-			if (err < 0)
-				return err;
-		}
-	}
-	nid = cfg->hp_pins[0];
-	if (nid) {
-		/* spec->multiout.hp_nid = 2; */
-		if (!cfg->line_out_pins[0] && !cfg->speaker_pins[0]) {
-			err = add_control(spec, ALC_CTL_WIDGET_VOL,
-					  "Headphone Playback Volume",
-					  HDA_COMPOSE_AMP_VAL(0x02, 3, 0,
-							      HDA_OUTPUT));
-			if (err < 0)
-				return err;
-		}
-		if (nid == 0x16) {
-			err = add_control(spec, ALC_CTL_WIDGET_MUTE,
-					  "Headphone Playback Switch",
-					  HDA_COMPOSE_AMP_VAL(nid, 2, 0,
-							      HDA_OUTPUT));
-			if (err < 0)
-				return err;
-		} else {
-			err = add_control(spec, ALC_CTL_WIDGET_MUTE,
-					  "Headphone Playback Switch",
-					  HDA_COMPOSE_AMP_VAL(nid, 3, 0,
-							      HDA_OUTPUT));
-			if (err < 0)
-				return err;
-		}
-	}
-	return 0;
-}
-
+#define alc269_auto_create_multi_out_ctls \
+	alc268_auto_create_multi_out_ctls
 #define alc269_auto_create_analog_input_ctls \
 	alc262_auto_create_analog_input_ctls
 
-- 
cgit v1.2.3


From 05f5f477082f0b192dab5ac25ccc5d468d98ecbd Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 25 Aug 2009 13:10:18 +0200
Subject: ALSA: hda - Generalize input pin parsing in patch_realtek.c

Provide a standard parser for input pins to create the input mixer
and input source controls instead of having a difference one for each
Realtek codec.  The new helper parses the codec connections dynamically
isntead of fixed indicies.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 356 ++++++++++++++++--------------------------
 1 file changed, 134 insertions(+), 222 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index dd25db93677..3c8773a7ac9 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4311,8 +4311,6 @@ static int add_control(struct alc_spec *spec, int type, const char *name,
 #define alc880_fixed_pin_idx(nid)	((nid) - 0x14)
 #define alc880_is_multi_pin(nid)	((nid) >= 0x18)
 #define alc880_multi_pin_idx(nid)	((nid) - 0x18)
-#define alc880_is_input_pin(nid)	((nid) >= 0x18)
-#define alc880_input_pin_idx(nid)	((nid) - 0x18)
 #define alc880_idx_to_dac(nid)		((nid) + 0x02)
 #define alc880_dac_to_idx(nid)		((nid) - 0x02)
 #define alc880_idx_to_mixer(nid)	((nid) + 0x0c)
@@ -4480,31 +4478,61 @@ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin,
 	return 0;
 }
 
+static int alc_is_input_pin(struct hda_codec *codec, hda_nid_t nid)
+{
+	unsigned int pincap = snd_hda_query_pin_caps(codec, nid);
+	return (pincap & AC_PINCAP_IN) != 0;
+}
+
 /* create playback/capture controls for input pins */
-static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec,
-						const struct auto_pin_cfg *cfg)
+static int alc_auto_create_input_ctls(struct hda_codec *codec,
+				      const struct auto_pin_cfg *cfg,
+				      hda_nid_t mixer,
+				      hda_nid_t cap1, hda_nid_t cap2)
 {
+	struct alc_spec *spec = codec->spec;
 	struct hda_input_mux *imux = &spec->private_imux[0];
 	int i, err, idx;
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
-		if (alc880_is_input_pin(cfg->input_pins[i])) {
-			idx = alc880_input_pin_idx(cfg->input_pins[i]);
-			err = new_analog_input(spec, cfg->input_pins[i],
-					       auto_pin_cfg_labels[i],
-					       idx, 0x0b);
-			if (err < 0)
-				return err;
+		hda_nid_t pin;
+
+		pin = cfg->input_pins[i];
+		if (!alc_is_input_pin(codec, pin))
+			continue;
+
+		if (mixer) {
+			idx = get_connection_index(codec, mixer, pin);
+			if (idx >= 0) {
+				err = new_analog_input(spec, pin,
+						       auto_pin_cfg_labels[i],
+						       idx, mixer);
+				if (err < 0)
+					return err;
+			}
+		}
+
+		if (!cap1)
+			continue;
+		idx = get_connection_index(codec, cap1, pin);
+		if (idx < 0 && cap2)
+			idx = get_connection_index(codec, cap2, pin);
+		if (idx >= 0) {
 			imux->items[imux->num_items].label =
 				auto_pin_cfg_labels[i];
-			imux->items[imux->num_items].index =
-				alc880_input_pin_idx(cfg->input_pins[i]);
+			imux->items[imux->num_items].index = idx;
 			imux->num_items++;
 		}
 	}
 	return 0;
 }
 
+static int alc880_auto_create_input_ctls(struct hda_codec *codec,
+						const struct auto_pin_cfg *cfg)
+{
+	return alc_auto_create_input_ctls(codec, cfg, 0x0b, 0x08, 0x09);
+}
+
 static void alc_set_pin_output(struct hda_codec *codec, hda_nid_t nid,
 			       unsigned int pin_type)
 {
@@ -4570,7 +4598,7 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec)
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
-		if (alc880_is_input_pin(nid)) {
+		if (alc_is_input_pin(codec, nid)) {
 			alc_set_input_pin(codec, nid, i);
 			if (nid != ALC880_PIN_CD_NID &&
 			    (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
@@ -4613,7 +4641,7 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 					   "Headphone");
 	if (err < 0)
 		return err;
-	err = alc880_auto_create_analog_input_ctls(spec, &spec->autocfg);
+	err = alc880_auto_create_input_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
 
@@ -6006,39 +6034,10 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
 }
 
 /* create playback/capture controls for input pins */
-static int alc260_auto_create_analog_input_ctls(struct alc_spec *spec,
+static int alc260_auto_create_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
-	struct hda_input_mux *imux = &spec->private_imux[0];
-	int i, err, idx;
-
-	for (i = 0; i < AUTO_PIN_LAST; i++) {
-		if (cfg->input_pins[i] >= 0x12) {
-			idx = cfg->input_pins[i] - 0x12;
-			err = new_analog_input(spec, cfg->input_pins[i],
-					       auto_pin_cfg_labels[i], idx,
-					       0x07);
-			if (err < 0)
-				return err;
-			imux->items[imux->num_items].label =
-				auto_pin_cfg_labels[i];
-			imux->items[imux->num_items].index = idx;
-			imux->num_items++;
-		}
-		if (cfg->input_pins[i] >= 0x0f && cfg->input_pins[i] <= 0x10){
-			idx = cfg->input_pins[i] - 0x09;
-			err = new_analog_input(spec, cfg->input_pins[i],
-					       auto_pin_cfg_labels[i], idx,
-					       0x07);
-			if (err < 0)
-				return err;
-			imux->items[imux->num_items].label =
-				auto_pin_cfg_labels[i];
-			imux->items[imux->num_items].index = idx;
-			imux->num_items++;
-		}
-	}
-	return 0;
+	return alc_auto_create_input_ctls(codec, cfg, 0x07, 0x04, 0x05);
 }
 
 static void alc260_auto_set_output_and_unmute(struct hda_codec *codec,
@@ -6152,7 +6151,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
 		return err;
 	if (!spec->kctls.list)
 		return 0; /* can't find valid BIOS pin config */
-	err = alc260_auto_create_analog_input_ctls(spec, &spec->autocfg);
+	err = alc260_auto_create_input_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
 
@@ -9464,6 +9463,12 @@ static struct snd_pci_quirk alc882_pinfix_tbl[] = {
 /*
  * BIOS auto configuration
  */
+static int alc882_auto_create_input_ctls(struct hda_codec *codec,
+						const struct auto_pin_cfg *cfg)
+{
+	return alc_auto_create_input_ctls(codec, cfg, 0x0b, 0x23, 0x22);
+}
+
 static void alc882_auto_set_output_and_unmute(struct hda_codec *codec,
 					      hda_nid_t nid, int pin_type,
 					      int dac_idx)
@@ -9604,44 +9609,73 @@ static int alc_auto_add_mic_boost(struct hda_codec *codec)
 static int alc882_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	struct auto_pin_cfg *autocfg = &spec->autocfg;
-	unsigned int wcap;
-	int i;
-	int err = alc880_parse_auto_config(codec);
+	static hda_nid_t alc882_ignore[] = { 0x1d, 0 };
+	int i, err;
 
+	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
+					   alc882_ignore);
 	if (err < 0)
 		return err;
-	else if (!err)
-		return 0; /* no config found */
+	if (!spec->autocfg.line_outs)
+		return 0; /* can't find valid BIOS pin config */
 
-	err = alc_auto_add_mic_boost(codec);
+	err = alc880_auto_fill_dac_nids(spec, &spec->autocfg);
+	if (err < 0)
+		return err;
+	err = alc880_auto_create_multi_out_ctls(spec, &spec->autocfg);
+	if (err < 0)
+		return err;
+	err = alc880_auto_create_extra_out(spec,
+					   spec->autocfg.speaker_pins[0],
+					   "Speaker");
+	if (err < 0)
+		return err;
+	err = alc880_auto_create_extra_out(spec, spec->autocfg.hp_pins[0],
+					   "Headphone");
+	if (err < 0)
+		return err;
+	err = alc882_auto_create_input_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
 
-	/* hack - override the init verbs */
-	spec->init_verbs[0] = alc883_auto_init_verbs;
+	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
+
+	/* check multiple SPDIF-out (for recent codecs) */
+	for (i = 0; i < spec->autocfg.dig_outs; i++) {
+		hda_nid_t dig_nid;
+		err = snd_hda_get_connections(codec,
+					      spec->autocfg.dig_out_pins[i],
+					      &dig_nid, 1);
+		if (err < 0)
+			continue;
+		if (!i)
+			spec->multiout.dig_out_nid = dig_nid;
+		else {
+			spec->multiout.slave_dig_outs = spec->slave_dig_outs;
+			spec->slave_dig_outs[i - 1] = dig_nid;
+			if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1)
+				break;
+		}
+	}
+	if (spec->autocfg.dig_in_pin)
+		spec->dig_in_nid = ALC880_DIGIN_NID;
+
+	if (spec->kctls.list)
+		add_mixer(spec, spec->kctls.list);
+
+	add_verb(spec, alc883_auto_init_verbs);
 	/* if ADC 0x07 is available, initialize it, too */
-	wcap = get_wcaps(codec, 0x07);
-	wcap = get_wcaps_type(wcap);
-	if (wcap == AC_WID_AUD_IN)
+	if (get_wcaps_type(get_wcaps(codec, 0x07)) == AC_WID_AUD_IN)
 		add_verb(spec, alc882_adc1_init_verbs);
 
-	/* digital-mic input pin is excluded in alc880_auto_create..()
-	 * because it's under 0x18
-	 */
-	if (autocfg->input_pins[AUTO_PIN_MIC] == 0x12 ||
-	    autocfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) {
-		struct hda_input_mux *imux = &spec->private_imux[0];
-		for (i = 1; i < 3; i++)
-			memcpy(&spec->private_imux[i],
-			       &spec->private_imux[0],
-			       sizeof(spec->private_imux[0]));
-		imux->items[imux->num_items].label = "Int DMic";
-		imux->items[imux->num_items].index = 0x0b;
-		imux->num_items++;
-		spec->num_mux_defs = 3;
-		spec->input_mux = spec->private_imux;
-	}
+	spec->num_mux_defs = 1;
+	spec->input_mux = &spec->private_imux[0];
+
+	alc_ssid_check(codec, 0x15, 0x1b, 0x14);
+
+	err = alc_auto_add_mic_boost(codec);
+	if (err < 0)
+		return err;
 
 	return 1; /* config found */
 }
@@ -10820,27 +10854,8 @@ static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec,
 	return 0;
 }
 
-static int alc262_auto_create_analog_input_ctls(struct alc_spec *spec,
-						const struct auto_pin_cfg *cfg)
-{
-	int err;
-
-	err = alc880_auto_create_analog_input_ctls(spec, cfg);
-	if (err < 0)
-		return err;
-	/* digital-mic input pin is excluded in alc880_auto_create..()
-	 * because it's under 0x18
-	 */
-	if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 ||
-	    cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) {
-		struct hda_input_mux *imux = &spec->private_imux[0];
-		imux->items[imux->num_items].label = "Int Mic";
-		imux->items[imux->num_items].index = 0x09;
-		imux->num_items++;
-	}
-	return 0;
-}
-
+#define alc262_auto_create_input_ctls \
+	alc880_auto_create_input_ctls
 
 /*
  * generic initialization of ADC, input mixers and output mixers
@@ -11158,7 +11173,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 	err = alc262_auto_create_multi_out_ctls(spec, &spec->autocfg);
 	if (err < 0)
 		return err;
-	err = alc262_auto_create_analog_input_ctls(spec, &spec->autocfg);
+	err = alc262_auto_create_input_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
 
@@ -12212,46 +12227,10 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec,
 }
 
 /* create playback/capture controls for input pins */
-static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec,
+static int alc268_auto_create_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
-	struct hda_input_mux *imux = &spec->private_imux[0];
-	int i, idx1, dmic_nid;
-
-	dmic_nid = 0x12;
-	while (dmic_nid <= 0x13) {
-		for (i = 0; i < AUTO_PIN_LAST; i++) {
-			switch (cfg->input_pins[i]) {
-			case 0x18:
-				idx1 = 0;	/* Mic 1 */
-				break;
-			case 0x19:
-				idx1 = 1;	/* Mic 2 */
-				break;
-			case 0x1a:
-				idx1 = 2;	/* Line In */
-				break;
-			case 0x1c:
-				idx1 = 3;	/* CD */
-				break;
-			case 0x12:
-			case 0x13:
-				if (cfg->input_pins[i] != dmic_nid)
-					continue;
-				idx1 = 6;	/* digital mics */
-				break;
-			default:
-				continue;
-			}
-			imux->items[imux->num_items].label =
-				auto_pin_cfg_labels[i];
-			imux->items[imux->num_items].index = idx1;
-			imux->num_items++;
-		}
-		imux++;
-		dmic_nid++;
-	}
-	return 0;
+	return alc_auto_create_input_ctls(codec, cfg, 0, 0x23, 0x24);
 }
 
 static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec)
@@ -12328,7 +12307,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
 	err = alc268_auto_create_multi_out_ctls(spec, &spec->autocfg);
 	if (err < 0)
 		return err;
-	err = alc268_auto_create_analog_input_ctls(spec, &spec->autocfg);
+	err = alc268_auto_create_input_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
 
@@ -13088,8 +13067,8 @@ static struct hda_verb alc269_init_verbs[] = {
 
 #define alc269_auto_create_multi_out_ctls \
 	alc268_auto_create_multi_out_ctls
-#define alc269_auto_create_analog_input_ctls \
-	alc262_auto_create_analog_input_ctls
+#define alc269_auto_create_input_ctls \
+	alc268_auto_create_input_ctls
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 #define alc269_loopbacks	alc880_loopbacks
@@ -13139,7 +13118,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	err = alc269_auto_create_multi_out_ctls(spec, &spec->autocfg);
 	if (err < 0)
 		return err;
-	err = alc269_auto_create_analog_input_ctls(spec, &spec->autocfg);
+	err = alc269_auto_create_input_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
 
@@ -14120,49 +14099,10 @@ static int alc861_auto_create_hp_ctls(struct hda_codec *codec, hda_nid_t pin)
 }
 
 /* create playback/capture controls for input pins */
-static int alc861_auto_create_analog_input_ctls(struct hda_codec *codec,
+static int alc861_auto_create_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
-	struct alc_spec *spec = codec->spec;
-	struct hda_input_mux *imux = &spec->private_imux[0];
-	int i, err, idx, idx1;
-
-	for (i = 0; i < AUTO_PIN_LAST; i++) {
-		switch (cfg->input_pins[i]) {
-		case 0x0c:
-			idx1 = 1;
-			idx = 2;	/* Line In */
-			break;
-		case 0x0f:
-			idx1 = 2;
-			idx = 2;	/* Line In */
-			break;
-		case 0x0d:
-			idx1 = 0;
-			idx = 1;	/* Mic In */
-			break;
-		case 0x10:
-			idx1 = 3;
-			idx = 1;	/* Mic In */
-			break;
-		case 0x11:
-			idx1 = 4;
-			idx = 0;	/* CD */
-			break;
-		default:
-			continue;
-		}
-
-		err = new_analog_input(spec, cfg->input_pins[i],
-				       auto_pin_cfg_labels[i], idx, 0x15);
-		if (err < 0)
-			return err;
-
-		imux->items[imux->num_items].label = auto_pin_cfg_labels[i];
-		imux->items[imux->num_items].index = idx1;
-		imux->num_items++;
-	}
-	return 0;
+	return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x08, 0);
 }
 
 static void alc861_auto_set_output_and_unmute(struct hda_codec *codec,
@@ -14259,7 +14199,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
 	err = alc861_auto_create_hp_ctls(codec, spec->autocfg.hp_pins[0]);
 	if (err < 0)
 		return err;
-	err = alc861_auto_create_analog_input_ctls(codec, &spec->autocfg);
+	err = alc861_auto_create_input_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
 
@@ -15140,6 +15080,13 @@ static struct alc_config_preset alc861vd_presets[] = {
 /*
  * BIOS auto configuration
  */
+static int alc861vd_auto_create_input_ctls(struct hda_codec *codec,
+						const struct auto_pin_cfg *cfg)
+{
+	return alc_auto_create_input_ctls(codec, cfg, 0x15, 0x09, 0);
+}
+
+
 static void alc861vd_auto_set_output_and_unmute(struct hda_codec *codec,
 				hda_nid_t nid, int pin_type, int dac_idx)
 {
@@ -15174,7 +15121,6 @@ static void alc861vd_auto_init_hp_out(struct hda_codec *codec)
 		alc861vd_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
 }
 
-#define alc861vd_is_input_pin(nid)	alc880_is_input_pin(nid)
 #define ALC861VD_PIN_CD_NID		ALC880_PIN_CD_NID
 
 static void alc861vd_auto_init_analog_input(struct hda_codec *codec)
@@ -15184,7 +15130,7 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec)
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
-		if (alc861vd_is_input_pin(nid)) {
+		if (alc_is_input_pin(codec, nid)) {
 			alc_set_input_pin(codec, nid, i);
 			if (nid != ALC861VD_PIN_CD_NID &&
 			    (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
@@ -15347,7 +15293,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
 					     "Headphone");
 	if (err < 0)
 		return err;
-	err = alc880_auto_create_analog_input_ctls(spec, &spec->autocfg);
+	err = alc861vd_auto_create_input_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
 
@@ -17089,43 +17035,9 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
 	return 0;
 }
 
-static int alc662_is_input_pin(struct hda_codec *codec, hda_nid_t nid)
-{
-	unsigned int pincap = snd_hda_query_pin_caps(codec, nid);
-	return (pincap & AC_PINCAP_IN) != 0;
-}
-
 /* create playback/capture controls for input pins */
-static int alc662_auto_create_analog_input_ctls(struct hda_codec *codec,
-						const struct auto_pin_cfg *cfg)
-{
-	struct alc_spec *spec = codec->spec;
-	struct hda_input_mux *imux = &spec->private_imux[0];
-	int i, err, idx;
-
-	for (i = 0; i < AUTO_PIN_LAST; i++) {
-		if (alc662_is_input_pin(codec, cfg->input_pins[i])) {
-			idx = get_connection_index(codec, 0x0b,
-						   cfg->input_pins[i]);
-			if (idx >= 0) {
-				err = new_analog_input(spec, cfg->input_pins[i],
-						       auto_pin_cfg_labels[i],
-						       idx, 0x0b);
-				if (err < 0)
-					return err;
-			}
-			idx = get_connection_index(codec, 0x22,
-						   cfg->input_pins[i]);
-			if (idx >= 0) {
-				imux->items[imux->num_items].label =
-					auto_pin_cfg_labels[i];
-				imux->items[imux->num_items].index = idx;
-				imux->num_items++;
-			}
-		}
-	}
-	return 0;
-}
+#define alc662_auto_create_input_ctls \
+	alc880_auto_create_input_ctls
 
 static void alc662_auto_set_output_and_unmute(struct hda_codec *codec,
 					      hda_nid_t nid, int pin_type,
@@ -17179,7 +17091,7 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec)
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
-		if (alc662_is_input_pin(codec, nid)) {
+		if (alc_is_input_pin(codec, nid)) {
 			alc_set_input_pin(codec, nid, i);
 			if (nid != ALC662_PIN_CD_NID &&
 			    (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
@@ -17220,7 +17132,7 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 					   "Headphone");
 	if (err < 0)
 		return err;
-	err = alc662_auto_create_analog_input_ctls(codec, &spec->autocfg);
+	err = alc662_auto_create_input_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
 
-- 
cgit v1.2.3


From cb162b6bf2f445d67a45f5485d4ce32addaf8fe0 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 25 Aug 2009 16:05:03 +0200
Subject: ALSA: hda - Improve auto-cfg mixer name for ALC880

When there is only one DAC is used for ALC880, try to assign a better
name, either Speaker or Front, depending on the output pin type.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 10 ++++++++--
 1 file changed, 8 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 3c8773a7ac9..07fbf7db69e 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -4398,13 +4398,19 @@ static int alc880_auto_create_multi_out_ctls(struct alc_spec *spec,
 			if (err < 0)
 				return err;
 		} else {
-			sprintf(name, "%s Playback Volume", chname[i]);
+			const char *pfx;
+			if (cfg->line_outs == 1 &&
+			    cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+				pfx = "Speaker";
+			else
+				pfx = chname[i];
+			sprintf(name, "%s Playback Volume", pfx);
 			err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
 					  HDA_COMPOSE_AMP_VAL(nid, 3, 0,
 							      HDA_OUTPUT));
 			if (err < 0)
 				return err;
-			sprintf(name, "%s Playback Switch", chname[i]);
+			sprintf(name, "%s Playback Switch", pfx);
 			err = add_control(spec, ALC_CTL_BIND_MUTE, name,
 					  HDA_COMPOSE_AMP_VAL(nid, 3, 2,
 							      HDA_INPUT));
-- 
cgit v1.2.3


From 23112d6d2d9b265c959ecb671366c7c3b9c83611 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 25 Aug 2009 16:07:08 +0200
Subject: ALSA: hda - Improve auto-cfg mixer name for ALC260

Instead of fixed "Front" mixer name, try to assign a better name, e.g.
"Master" or "Speaker" fot the primary output volume controls of ALC260
codec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 9 ++++++++-
 1 file changed, 8 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 07fbf7db69e..00ed1268f27 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -6017,7 +6017,14 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
 
 	nid = cfg->line_out_pins[0];
 	if (nid) {
-		err = alc260_add_playback_controls(spec, nid, "Front", &vols);
+		const char *pfx;
+		if (!cfg->speaker_pins[0] && !cfg->hp_pins[0])
+			pfx = "Master";
+		else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+			pfx = "Speaker";
+		else
+			pfx = "Front";
+		err = alc260_add_playback_controls(spec, nid, pfx, &vols);
 		if (err < 0)
 			return err;
 	}
-- 
cgit v1.2.3


From c3fc1f502a1f1a751c891d3b757a24dc76943539 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 25 Aug 2009 16:08:47 +0200
Subject: ALSA: hda - Improve auto-cfg mixer name for ALC262

Similar improvements for ALC262 codec like previous two commits:
assign a better name, either Master or Speaker, for the primary output
controls.

However, in the case of ALC262 codec, the necessary changes are larger
than others because we need to check the possibility of different mixer
amps depending on the pins.  The pin 0x16 is mono, and bound with the
dedicated mixer 0x0e while other pins are bound with 0x0c.  Thus, there
are two possible volumes.

When only one of them is used, we can name it as "Master".  OTOH, when
both are used at the same time, they have to be named uniquely.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 152 +++++++++++++++++++++++++-----------------
 1 file changed, 89 insertions(+), 63 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 00ed1268f27..407475941fd 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -10790,80 +10790,106 @@ static struct snd_kcontrol_new alc262_ultra_capture_mixer[] = {
 	{ } /* end */
 };
 
+/* We use two mixers depending on the output pin; 0x16 is a mono output
+ * and thus it's bound with a different mixer.
+ * This function returns which mixer amp should be used.
+ */
+static int alc262_check_volbit(hda_nid_t nid)
+{
+	if (!nid)
+		return 0;
+	else if (nid == 0x16)
+		return 2;
+	else
+		return 1;
+}
+
+static int alc262_add_out_vol_ctl(struct alc_spec *spec, hda_nid_t nid,
+				  const char *pfx, int *vbits)
+{
+	char name[32];
+	unsigned long val;
+	int vbit;
+
+	vbit = alc262_check_volbit(nid);
+	if (!vbit)
+		return 0;
+	if (*vbits & vbit) /* a volume control for this mixer already there */
+		return 0;
+	*vbits |= vbit;
+	snprintf(name, sizeof(name), "%s Playback Volume", pfx);
+	if (vbit == 2)
+		val = HDA_COMPOSE_AMP_VAL(0x0e, 2, 0, HDA_OUTPUT);
+	else
+		val = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT);
+	return add_control(spec, ALC_CTL_WIDGET_VOL, name, val);
+}
+
+static int alc262_add_out_sw_ctl(struct alc_spec *spec, hda_nid_t nid,
+				 const char *pfx)
+{
+	char name[32];
+	unsigned long val;
+
+	if (!nid)
+		return 0;
+	snprintf(name, sizeof(name), "%s Playback Switch", pfx);
+	if (nid == 0x16)
+		val = HDA_COMPOSE_AMP_VAL(nid, 2, 0, HDA_OUTPUT);
+	else
+		val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT);
+	return add_control(spec, ALC_CTL_WIDGET_MUTE, name, val);
+}
+
 /* add playback controls from the parsed DAC table */
 static int alc262_auto_create_multi_out_ctls(struct alc_spec *spec,
 					     const struct auto_pin_cfg *cfg)
 {
-	hda_nid_t nid;
+	const char *pfx;
+	int vbits;
 	int err;
 
 	spec->multiout.num_dacs = 1;	/* only use one dac */
 	spec->multiout.dac_nids = spec->private_dac_nids;
 	spec->multiout.dac_nids[0] = 2;
 
-	nid = cfg->line_out_pins[0];
-	if (nid) {
-		err = add_control(spec, ALC_CTL_WIDGET_VOL,
-				  "Front Playback Volume",
-				  HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT));
-		if (err < 0)
-			return err;
-		err = add_control(spec, ALC_CTL_WIDGET_MUTE,
-				  "Front Playback Switch",
-				  HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT));
-		if (err < 0)
-			return err;
-	}
+	if (!cfg->speaker_pins[0] && !cfg->hp_pins[0])
+		pfx = "Master";
+	else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+		pfx = "Speaker";
+	else
+		pfx = "Front";
+	err = alc262_add_out_sw_ctl(spec, cfg->line_out_pins[0], pfx);
+	if (err < 0)
+		return err;
+	err = alc262_add_out_sw_ctl(spec, cfg->speaker_pins[0], "Speaker");
+	if (err < 0)
+		return err;
+	err = alc262_add_out_sw_ctl(spec, cfg->hp_pins[0], "Headphone");
+	if (err < 0)
+		return err;
 
-	nid = cfg->speaker_pins[0];
-	if (nid) {
-		if (nid == 0x16) {
-			err = add_control(spec, ALC_CTL_WIDGET_VOL,
-					  "Speaker Playback Volume",
-					  HDA_COMPOSE_AMP_VAL(0x0e, 2, 0,
-							      HDA_OUTPUT));
-			if (err < 0)
-				return err;
-			err = add_control(spec, ALC_CTL_WIDGET_MUTE,
-					  "Speaker Playback Switch",
-					  HDA_COMPOSE_AMP_VAL(nid, 2, 0,
-							      HDA_OUTPUT));
-			if (err < 0)
-				return err;
-		} else {
-			err = add_control(spec, ALC_CTL_WIDGET_MUTE,
-					  "Speaker Playback Switch",
-					  HDA_COMPOSE_AMP_VAL(nid, 3, 0,
-							      HDA_OUTPUT));
-			if (err < 0)
-				return err;
-		}
-	}
-	nid = cfg->hp_pins[0];
-	if (nid) {
-		/* spec->multiout.hp_nid = 2; */
-		if (nid == 0x16) {
-			err = add_control(spec, ALC_CTL_WIDGET_VOL,
-					  "Headphone Playback Volume",
-					  HDA_COMPOSE_AMP_VAL(0x0e, 2, 0,
-							      HDA_OUTPUT));
-			if (err < 0)
-				return err;
-			err = add_control(spec, ALC_CTL_WIDGET_MUTE,
-					  "Headphone Playback Switch",
-					  HDA_COMPOSE_AMP_VAL(nid, 2, 0,
-							      HDA_OUTPUT));
-			if (err < 0)
-				return err;
-		} else {
-			err = add_control(spec, ALC_CTL_WIDGET_MUTE,
-					  "Headphone Playback Switch",
-					  HDA_COMPOSE_AMP_VAL(nid, 3, 0,
-							      HDA_OUTPUT));
-			if (err < 0)
-				return err;
-		}
-	}
+	vbits = alc262_check_volbit(cfg->line_out_pins[0]) |
+		alc262_check_volbit(cfg->speaker_pins[0]) |
+		alc262_check_volbit(cfg->hp_pins[0]);
+	if (vbits == 1 || vbits == 2)
+		pfx = "Master"; /* only one mixer is used */
+	else if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+		pfx = "Speaker";
+	else
+		pfx = "Front";
+	vbits = 0;
+	err = alc262_add_out_vol_ctl(spec, cfg->line_out_pins[0], pfx, &vbits);
+	if (err < 0)
+		return err;
+	err = alc262_add_out_vol_ctl(spec, cfg->speaker_pins[0], "Speaker",
+				     &vbits);
+	if (err < 0)
+		return err;
+	err = alc262_add_out_vol_ctl(spec, cfg->hp_pins[0], "Headphone",
+				     &vbits);
+	if (err < 0)
+		return err;
 	return 0;
 }
 
-- 
cgit v1.2.3


From a4fcd491098603356e7600b021bdb6acbb53adb4 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 25 Aug 2009 16:12:15 +0200
Subject: ALSA: hda - Improve auto-cfg mixer name for ALC861-VD

One more patch to give a better name for the primary output controls,
this time for ALC861-VD codec.  The change is simple, just checking the
pin connection whether it's a speaker-out.  When both speaker and HP
are assigned, we name the volume as "PCM" as this influences on both
outputs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 16 ++++++++++++++--
 1 file changed, 14 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 407475941fd..206fcb930a6 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -15233,13 +15233,25 @@ static int alc861vd_auto_create_multi_out_ctls(struct alc_spec *spec,
 			if (err < 0)
 				return err;
 		} else {
-			sprintf(name, "%s Playback Volume", chname[i]);
+			const char *pfx;
+			if (cfg->line_outs == 1 &&
+			    cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
+				if (!cfg->hp_pins)
+					pfx = "Speaker";
+				else
+					pfx = "PCM";
+			} else
+				pfx = chname[i];
+			sprintf(name, "%s Playback Volume", pfx);
 			err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
 					  HDA_COMPOSE_AMP_VAL(nid_v, 3, 0,
 							      HDA_OUTPUT));
 			if (err < 0)
 				return err;
-			sprintf(name, "%s Playback Switch", chname[i]);
+			if (cfg->line_outs == 1 &&
+			    cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+				pfx = "Speaker";
+			sprintf(name, "%s Playback Switch", pfx);
 			err = add_control(spec, ALC_CTL_BIND_MUTE, name,
 					  HDA_COMPOSE_AMP_VAL(nid_s, 3, 2,
 							      HDA_INPUT));
-- 
cgit v1.2.3


From 0d884cb936e8e36c47bc7ee86eb6750261bd5438 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 25 Aug 2009 16:14:35 +0200
Subject: ALSA: hda - Improve auto-cfg mixer name for ALC662

The last patch in this series is for ALC662; pretty similar as the
previous patch for ALC861-VD.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 16 ++++++++++++++--
 1 file changed, 14 insertions(+), 2 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 206fcb930a6..ce098b20cdf 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -17018,13 +17018,25 @@ static int alc662_auto_create_multi_out_ctls(struct alc_spec *spec,
 			if (err < 0)
 				return err;
 		} else {
-			sprintf(name, "%s Playback Volume", chname[i]);
+			const char *pfx;
+			if (cfg->line_outs == 1 &&
+			    cfg->line_out_type == AUTO_PIN_SPEAKER_OUT) {
+				if (!cfg->hp_pins)
+					pfx = "Speaker";
+				else
+					pfx = "PCM";
+			} else
+				pfx = chname[i];
+			sprintf(name, "%s Playback Volume", pfx);
 			err = add_control(spec, ALC_CTL_WIDGET_VOL, name,
 					  HDA_COMPOSE_AMP_VAL(nid, 3, 0,
 							      HDA_OUTPUT));
 			if (err < 0)
 				return err;
-			sprintf(name, "%s Playback Switch", chname[i]);
+			if (cfg->line_outs == 1 &&
+			    cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+				pfx = "Speaker";
+			sprintf(name, "%s Playback Switch", pfx);
 			err = add_control(spec, ALC_CTL_WIDGET_MUTE, name,
 				HDA_COMPOSE_AMP_VAL(alc880_idx_to_mixer(i),
 						    3, 0, HDA_INPUT));
-- 
cgit v1.2.3


From fc86f95415d14c74fd71f5e70173a4def7da68f6 Mon Sep 17 00:00:00 2001
From: Denis Kuplyakov <dener.kup@gmail.com>
Date: Tue, 25 Aug 2009 18:15:59 +0200
Subject: ALSA: hda - Add full audio support on Acer Aspire 7730G notebook

1) Added support of internal subwoofer (it sounds!!!)
2) Auto muting front speakers and internal subwoofer on headphones plug.
3) Internal mic works.
4) 3 channel mods (jack maps):
       black  pink         blue
2ch: front   ext mic     line in
4ch: front   ext mic     surround
6ch: front   CLFE        surround
  Can be changed in mixer.
5) Sound can be recorded from:
 Internal mic
 Ext mic
 Cd
 Line in
6) 2 separate capture channels.

Signed-off-by: Denis Kuplyakov <dener.kup@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 31 +++++++++++++++++++++++++++++++
 1 file changed, 31 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index ce098b20cdf..d572db619fb 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -220,6 +220,7 @@ enum {
 	ALC888_ACER_ASPIRE_4930G,
 	ALC888_ACER_ASPIRE_6530G,
 	ALC888_ACER_ASPIRE_8930G,
+	ALC888_ACER_ASPIRE_7730G,
 	ALC883_MEDION,
 	ALC883_MEDION_MD2,
 	ALC883_LAPTOP_EAPD,
@@ -8484,6 +8485,13 @@ static struct hda_verb alc883_acer_eapd_verbs[] = {
 	{ }
 };
 
+static struct hda_verb alc888_acer_aspire_7730G_verbs[] = {
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x17, AC_VERB_SET_CONNECT_SEL, 0x02},
+	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+	{ } /* end */
+};
+
 static void alc888_6st_dell_setup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -8645,6 +8653,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = {
 	[ALC888_ACER_ASPIRE_4930G]	= "acer-aspire-4930g",
 	[ALC888_ACER_ASPIRE_6530G]	= "acer-aspire-6530g",
 	[ALC888_ACER_ASPIRE_8930G]	= "acer-aspire-8930g",
+	[ALC888_ACER_ASPIRE_7730G]	= "acer-aspire-7730g",
 	[ALC883_MEDION]		= "medion",
 	[ALC883_MEDION_MD2]	= "medion-md2",
 	[ALC883_LAPTOP_EAPD]	= "laptop-eapd",
@@ -8691,6 +8700,8 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
 		ALC888_ACER_ASPIRE_6530G),
 	SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G",
 		ALC888_ACER_ASPIRE_6530G),
+	SND_PCI_QUIRK(0x1025, 0x0142, "Acer Aspire 7730G",
+		ALC888_ACER_ASPIRE_7730G),
 	/* default Acer -- disabled as it causes more problems.
 	 *    model=auto should work fine now
 	 */
@@ -9192,6 +9203,26 @@ static struct alc_config_preset alc882_presets[] = {
 		.setup = alc889_acer_aspire_8930g_setup,
 		.init_hook = alc_automute_amp,
 	},
+	[ALC888_ACER_ASPIRE_7730G] = {
+		.mixers = { alc883_3ST_6ch_mixer,
+				alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs, alc880_gpio1_init_verbs,
+				alc888_acer_aspire_7730G_verbs },
+		.num_dacs = ARRAY_SIZE(alc883_dac_nids),
+		.dac_nids = alc883_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc883_adc_nids_rev),
+		.adc_nids = alc883_adc_nids_rev,
+		.capsrc_nids = alc883_capsrc_nids_rev,
+		.dig_out_nid = ALC883_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_modes),
+		.channel_mode = alc883_3ST_6ch_modes,
+		.need_dac_fix = 1,
+		.const_channel_count = 6,
+		.input_mux = &alc883_capture_source,
+		.unsol_event = alc_automute_amp_unsol_event,
+		.setup = alc888_acer_aspire_6530g_setup,
+		.init_hook = alc_automute_amp,
+	},
 	[ALC883_MEDION] = {
 		.mixers = { alc883_fivestack_mixer,
 			    alc883_chmode_mixer },
-- 
cgit v1.2.3


From 286f5875ca5c4e58432e3e6ca386ea4294275d84 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Thu, 27 Aug 2009 14:37:51 +0200
Subject: ALSA: hda - Add more quirk for HP laptops with AD1984A

More entries for HP laptops to get them working properly.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 1 +
 1 file changed, 1 insertion(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 41ec0f865ee..f22c3c08a9f 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -4009,6 +4009,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
 	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30d0, "HP laptop", AD1884A_LAPTOP),
 	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP),
 	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP),
+	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE),
 	SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
 	{}
 };
-- 
cgit v1.2.3


From 96f845de89be6be12112d7b388cdf366dccfe12d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sat, 29 Aug 2009 00:49:36 +0200
Subject: ALSA: hda - Create "Digital Mic Capture Volume" correctly for IDT
 codecs

So far, the digital mic capture volume wasn't created.  This is because
IDT codecs have output amps for digital mics, not input amps, while
input amps should be used for other analog pins.  Thus the automatic
capture volume creation should check both directions for digital mics.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 30 +++++++++++++++++++++++-------
 1 file changed, 23 insertions(+), 7 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 742b9c8b25e..c878dfa369b 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -3580,21 +3580,30 @@ static int get_connection_index(struct hda_codec *codec, hda_nid_t mux,
 }
 
 /* create a volume assigned to the given pin (only if supported) */
+/* return 1 if the volume control is created */
 static int create_elem_capture_vol(struct hda_codec *codec, hda_nid_t nid,
-				   const char *label)
+				   const char *label, int direction)
 {
 	unsigned int caps, nums;
 	char name[32];
+	int err;
 
-	if (!(get_wcaps(codec, nid) & AC_WCAP_IN_AMP))
+	if (direction == HDA_OUTPUT)
+		caps = AC_WCAP_OUT_AMP;
+	else
+		caps = AC_WCAP_IN_AMP;
+	if (!(get_wcaps(codec, nid) & caps))
 		return 0;
-	caps = query_amp_caps(codec, nid, HDA_OUTPUT);
+	caps = query_amp_caps(codec, nid, direction);
 	nums = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT;
 	if (!nums)
 		return 0;
 	snprintf(name, sizeof(name), "%s Capture Volume", label);
-	return stac92xx_add_control(codec->spec, STAC_CTL_WIDGET_VOL, name,
-				    HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_INPUT));
+	err = stac92xx_add_control(codec->spec, STAC_CTL_WIDGET_VOL, name,
+				    HDA_COMPOSE_AMP_VAL(nid, 3, 0, direction));
+	if (err < 0)
+		return err;
+	return 1;
 }
 
 /* create playback/capture controls for input pins on dmic capable codecs */
@@ -3643,9 +3652,15 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
 		else
 			label = stac92xx_dmic_labels[dimux->num_items];
 
-		err = create_elem_capture_vol(codec, nid, label);
+		err = create_elem_capture_vol(codec, nid, label, HDA_INPUT);
 		if (err < 0)
 			return err;
+		if (!err) {
+			err = create_elem_capture_vol(codec, nid, label,
+						      HDA_OUTPUT);
+			if (err < 0)
+				return err;
+		}
 
 		dimux->items[dimux->num_items].label = label;
 		dimux->items[dimux->num_items].index = index;
@@ -3766,7 +3781,8 @@ static int stac92xx_auto_create_analog_input_ctls(struct hda_codec *codec, const
 			continue;
 
 		err = create_elem_capture_vol(codec, nid,
-					      auto_pin_cfg_labels[i]);
+					      auto_pin_cfg_labels[i],
+					      HDA_INPUT);
 		if (err < 0)
 			return err;
 
-- 
cgit v1.2.3


From e9af4f365fee8065c0c1b2389fe10e540dd9d60a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sat, 29 Aug 2009 23:23:08 +0200
Subject: ALSA: hda - Fix ALC268/ALC269 headphone pin routing

Fix the headphone pin routing of ALC268/ALC269 codecs.  Using alc882
routine doesn't work because alc268/alc269 parser assumes the
independent DACs for both HP and speaker outputs.  Need to assign the
DAC depending on the pin.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 46 ++++++++++++++++++++++++++++++++++++++-----
 1 file changed, 41 insertions(+), 5 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index d572db619fb..bc17a96a794 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12303,6 +12303,42 @@ static int alc268_auto_create_input_ctls(struct hda_codec *codec,
 	return alc_auto_create_input_ctls(codec, cfg, 0, 0x23, 0x24);
 }
 
+static void alc268_auto_set_output_and_unmute(struct hda_codec *codec,
+					      hda_nid_t nid, int pin_type)
+{
+	int idx;
+
+	alc_set_pin_output(codec, nid, pin_type);
+	if (nid == 0x14 || nid == 0x16)
+		idx = 0;
+	else
+		idx = 1;
+	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx);
+}
+
+static void alc268_auto_init_multi_out(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t nid = spec->autocfg.line_out_pins[0];
+	if (nid) {
+		int pin_type = get_pin_type(spec->autocfg.line_out_type);
+		alc268_auto_set_output_and_unmute(codec, nid, pin_type);
+	}
+}
+
+static void alc268_auto_init_hp_out(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t pin;
+
+	pin = spec->autocfg.hp_pins[0];
+	if (pin)
+		alc268_auto_set_output_and_unmute(codec, pin, PIN_HP);
+	pin = spec->autocfg.speaker_pins[0];
+	if (pin)
+		alc268_auto_set_output_and_unmute(codec, pin, PIN_OUT);
+}
+
 static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -12311,9 +12347,10 @@ static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec)
 	hda_nid_t line_nid = spec->autocfg.line_out_pins[0];
 	unsigned int	dac_vol1, dac_vol2;
 
-	if (speaker_nid) {
+	if (line_nid == 0x1d || speaker_nid == 0x1d) {
 		snd_hda_codec_write(codec, speaker_nid, 0,
 				    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+		/* mute mixer inputs from 0x1d */
 		snd_hda_codec_write(codec, 0x0f, 0,
 				    AC_VERB_SET_AMP_GAIN_MUTE,
 				    AMP_IN_UNMUTE(1));
@@ -12321,6 +12358,7 @@ static void alc268_auto_init_mono_speaker_out(struct hda_codec *codec)
 				    AC_VERB_SET_AMP_GAIN_MUTE,
 				    AMP_IN_UNMUTE(1));
 	} else {
+		/* unmute mixer inputs from 0x1d */
 		snd_hda_codec_write(codec, 0x0f, 0,
 				    AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1));
 		snd_hda_codec_write(codec, 0x10, 0,
@@ -12408,8 +12446,6 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
 	return 1;
 }
 
-#define alc268_auto_init_multi_out	alc882_auto_init_multi_out
-#define alc268_auto_init_hp_out		alc882_auto_init_hp_out
 #define alc268_auto_init_analog_input	alc882_auto_init_analog_input
 
 /* init callback for auto-configuration model -- overriding the default init */
@@ -13220,8 +13256,8 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	return 1;
 }
 
-#define alc269_auto_init_multi_out	alc882_auto_init_multi_out
-#define alc269_auto_init_hp_out		alc882_auto_init_hp_out
+#define alc269_auto_init_multi_out	alc268_auto_init_multi_out
+#define alc269_auto_init_hp_out		alc268_auto_init_hp_out
 #define alc269_auto_init_analog_input	alc882_auto_init_analog_input
 
 
-- 
cgit v1.2.3


From 17bbaa6f600a36930879fd58e4e38a8c25bf9b60 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Sun, 30 Aug 2009 12:15:59 +0200
Subject: ALSA: hda - Add support of docking auto-mute/mic for AD1984A laptop
 model

Add the support of automatic mute and mic-switching of the docking
station HP and mic plugs for AD1984A laptop model for some HP machines.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 121 +++++++++++++++++++++++++++++--------------
 1 file changed, 81 insertions(+), 40 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index f22c3c08a9f..570e072938c 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3703,19 +3703,29 @@ static struct hda_amp_list ad1884a_loopbacks[] = {
  * Port F: Internal speakers
  */
 
-static struct hda_input_mux ad1884a_laptop_capture_source = {
-	.num_items = 4,
-	.items = {
-		{ "Mic", 0x0 },		/* port-B */
-		{ "Internal Mic", 0x1 }, /* port-C */
-		{ "Dock Mic", 0x4 },	/* port-E */
-		{ "Mix", 0x3 },
-	},
-};
+static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
+	int mute = (!ucontrol->value.integer.value[0] &&
+		    !ucontrol->value.integer.value[1]);
+	/* toggle GPIO1 according to the mute state */
+	snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
+			    mute ? 0x02 : 0x0);
+	return ret;
+}
 
 static struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 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 = ad1884a_mobile_master_sw_put,
+		.private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
+	},
 	HDA_CODEC_MUTE("Dock Playback Switch", 0x12, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
 	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
@@ -3730,36 +3740,9 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
 	HDA_CODEC_VOLUME("Dock Mic Boost", 0x25, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
 	{ } /* end */
 };
 
-static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
-				        struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
-	int mute = (!ucontrol->value.integer.value[0] &&
-		    !ucontrol->value.integer.value[1]);
-	/* toggle GPIO1 according to the mute state */
-	snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
-			    mute ? 0x02 : 0x0);
-	return ret;
-}
-
 static struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
 	/*HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
@@ -3829,6 +3812,63 @@ static int ad1884a_hp_init(struct hda_codec *codec)
 	return 0;
 }
 
+/* mute internal speaker if HP or docking HP is plugged */
+static void ad1884a_laptop_automute(struct hda_codec *codec)
+{
+	unsigned int present;
+
+	present = snd_hda_codec_read(codec, 0x11, 0, AC_VERB_GET_PIN_SENSE, 0);
+	present &= AC_PINSENSE_PRESENCE;
+	if (!present) {
+		present = snd_hda_codec_read(codec, 0x12, 0,
+					     AC_VERB_GET_PIN_SENSE, 0);
+		present &= AC_PINSENSE_PRESENCE;
+	}
+	snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
+	snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE,
+			    present ? 0x00 : 0x02);
+}
+
+/* switch to external mic if plugged */
+static void ad1884a_laptop_automic(struct hda_codec *codec)
+{
+	unsigned int idx;
+
+	if (snd_hda_codec_read(codec, 0x14, 0, AC_VERB_GET_PIN_SENSE, 0) &
+	    AC_PINSENSE_PRESENCE)
+		idx = 0;
+	else if (snd_hda_codec_read(codec, 0x1c, 0, AC_VERB_GET_PIN_SENSE, 0) &
+		 AC_PINSENSE_PRESENCE)
+		idx = 4;
+	else
+		idx = 1;
+	snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, idx);
+}
+
+/* unsolicited event for HP jack sensing */
+static void ad1884a_laptop_unsol_event(struct hda_codec *codec,
+				       unsigned int res)
+{
+	switch (res >> 26) {
+	case AD1884A_HP_EVENT:
+		ad1884a_laptop_automute(codec);
+		break;
+	case AD1884A_MIC_EVENT:
+		ad1884a_laptop_automic(codec);
+		break;
+	}
+}
+
+/* initialize jack-sensing, too */
+static int ad1884a_laptop_init(struct hda_codec *codec)
+{
+	ad198x_init(codec);
+	ad1884a_laptop_automute(codec);
+	ad1884a_laptop_automic(codec);
+	return 0;
+}
+
 /* additional verbs for laptop model */
 static struct hda_verb ad1884a_laptop_verbs[] = {
 	/* Port-A (HP) pin - always unmuted */
@@ -3849,7 +3889,9 @@ static struct hda_verb ad1884a_laptop_verbs[] = {
 	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
 	/* unsolicited event for pin-sense */
 	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
+	{0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
 	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
+	{0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
 	{ } /* end */
 };
 
@@ -4059,9 +4101,8 @@ static int patch_ad1884a(struct hda_codec *codec)
 		spec->mixers[0] = ad1884a_laptop_mixers;
 		spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs;
 		spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1884a_laptop_capture_source;
-		codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
-		codec->patch_ops.init = ad1884a_hp_init;
+		codec->patch_ops.unsol_event = ad1884a_laptop_unsol_event;
+		codec->patch_ops.init = ad1884a_laptop_init;
 		/* set the upper-limit for mixer amp to 0dB for avoiding the
 		 * possible damage by overloading
 		 */
-- 
cgit v1.2.3


From fe7e56814cb82eb28698c2a74bdb830ae0592a45 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Mon, 31 Aug 2009 08:37:46 +0200
Subject: ALSA: hda - Add missing GPIO initialization for AD1984A laptop model

A similar initialization of GPIO1 pin like mobile model is needed
for laptop model, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index 570e072938c..db3f3b15f0e 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3892,6 +3892,10 @@ static struct hda_verb ad1884a_laptop_verbs[] = {
 	{0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
 	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
 	{0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
+	/* allow to touch GPIO1 (for mute control) */
+	{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
+	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
+	{0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
 	{ } /* end */
 };
 
-- 
cgit v1.2.3


From f8ff035e38514ea60aae8752af45ed90e7598fcd Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 1 Sep 2009 08:53:19 +0200
Subject: ALSA: hda - Add another entry for Nvidia HDMI device

Added another entry for Nvidia HDMI device (10de:0003).

Reference: kernel bug#14097
	http://bugzilla.kernel.org/show_bug.cgi?id=14097

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_nvhdmi.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
index f5792e2eea8..c8435c9a97f 100644
--- a/sound/pci/hda/patch_nvhdmi.c
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -377,6 +377,7 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec)
  */
 static struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
 	{ .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
+	{ .id = 0x10de0003, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
 	{ .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi_8ch },
 	{ .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi_8ch },
 	{ .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch },
@@ -385,6 +386,7 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = {
 };
 
 MODULE_ALIAS("snd-hda-codec-id:10de0002");
+MODULE_ALIAS("snd-hda-codec-id:10de0003");
 MODULE_ALIAS("snd-hda-codec-id:10de0006");
 MODULE_ALIAS("snd-hda-codec-id:10de0007");
 MODULE_ALIAS("snd-hda-codec-id:10de0067");
-- 
cgit v1.2.3


From 2ad81ba0147919edc0cf56b80c6224775656b78d Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 1 Sep 2009 09:09:26 +0200
Subject: ALSA: hda - Unmute docking line-out as default with AD1984A codec

Unmute the docking-station line-out as default on machines with
AD1984A codec chip.  It can be still muted via "Dock" mixer switch.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_analog.c | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index db3f3b15f0e..215e72a8711 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -3885,6 +3885,8 @@ static struct hda_verb ad1884a_laptop_verbs[] = {
 	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
 	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
 	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
+	/* Port-D (docking line-out) pin - default unmuted */
+	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	/* analog mix */
 	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
 	/* unsolicited event for pin-sense */
-- 
cgit v1.2.3


From d94ff6b7cab20cc014e22b2fd5f0ff35d3e2308f Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 2 Sep 2009 00:20:21 +0200
Subject: ALSA: hda - Fix obsolete CONFIG_SND_DEBUG_DETECT

Fix the old dead CONFIG_SND_DEBUG_DETECT to CONFIG_SND_DEBUG_VERBOSE.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/hda_codec.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 24401d5d351..af989f660cc 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -3127,7 +3127,7 @@ int snd_hda_check_board_codec_sid_config(struct hda_codec *codec,
 	tbl = q;
 
 	if (tbl->value >= 0 && tbl->value < num_configs) {
-#ifdef CONFIG_SND_DEBUG_DETECT
+#ifdef CONFIG_SND_DEBUG_VERBOSE
 		char tmp[10];
 		const char *model = NULL;
 		if (models)
-- 
cgit v1.2.3


From e2aec17100331669ecae8550a83f551f5e8fc4b8 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 2 Sep 2009 01:00:05 +0200
Subject: ALSA: hda - Fix input source selection of IDT92HD73xx

Fix the mux_nids to select directly the input source instead of mux
mixers so that it works with the current mux enum handler for IDT
92HD73xx codecs.

Also, clean up useless / unnecessary mixer controls and init verbs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 151 ++---------------------------------------
 1 file changed, 4 insertions(+), 147 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index c878dfa369b..4cce92cf8ea 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -329,11 +329,6 @@ 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
@@ -341,8 +336,8 @@ static hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = {
 
 #define STAC92HD73_DAC_COUNT 5
 
-static hda_nid_t stac92hd73xx_mux_nids[4] = {
-	0x28, 0x29, 0x2a, 0x2b,
+static hda_nid_t stac92hd73xx_mux_nids[2] = {
+	0x20, 0x21,
 };
 
 static hda_nid_t stac92hd73xx_dmux_nids[2] = {
@@ -920,84 +915,16 @@ static struct hda_verb stac9200_eapd_init[] = {
 	{}
 };
 
-static struct hda_verb stac92hd73xx_6ch_core_init[] = {
-	/* set master volume and direct control */
-	{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
-	/* setup adcs to point to mixer */
-	{ 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
-	{ 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
-	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* setup import muxs */
-	{ 0x28, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x29, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{}
-};
-
 static struct hda_verb dell_eq_core_init[] = {
 	/* set master volume to max value without distortion
 	 * and direct control */
 	{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec},
-	/* setup adcs to point to mixer */
-	{ 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
-	{ 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
-	/* setup import muxs */
-	{ 0x28, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x29, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{}
-};
-
-static struct hda_verb dell_m6_core_init[] = {
-	{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
-	/* setup adcs to point to mixer */
-	{ 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
-	{ 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
-	/* setup import muxs */
-	{ 0x28, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x29, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{}
-};
-
-static struct hda_verb stac92hd73xx_8ch_core_init[] = {
-	/* set master volume and direct control */
-	{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
-	/* setup adcs to point to mixer */
-	{ 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
-	{ 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
-	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* setup import muxs */
-	{ 0x28, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x29, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x2b, AC_VERB_SET_CONNECT_SEL, 0x03},
 	{}
 };
 
-static struct hda_verb stac92hd73xx_10ch_core_init[] = {
+static struct hda_verb stac92hd73xx_core_init[] = {
 	/* set master volume and direct control */
 	{ 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
-	/* dac3 is connected to import3 mux */
-	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb07f},
-	/* setup adcs to point to mixer */
-	{ 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b},
-	{ 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b},
-	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* setup import muxs */
-	{ 0x28, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x29, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{ 0x2b, AC_VERB_SET_CONNECT_SEL, 0x03},
 	{}
 };
 
@@ -1130,28 +1057,6 @@ static struct snd_kcontrol_new stac9200_mixer[] = {
 	{ } /* end */
 };
 
-#define DELL_M6_MIXER 6
-static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = {
-	/* 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("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),
-
-	{ } /* end */
-};
-
 static struct snd_kcontrol_new stac92hd73xx_6ch_loopback[] = {
 	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3),
 	{}
@@ -1167,42 +1072,6 @@ static struct snd_kcontrol_new stac92hd73xx_10ch_loopback[] = {
 	{}
 };
 
-static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = {
-	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("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),
-	{ } /* end */
-};
-
-static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = {
-	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("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),
-	{ } /* end */
-};
-
 
 static struct snd_kcontrol_new stac92hd71bxx_loopback[] = {
 	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2)
@@ -5188,20 +5057,15 @@ again:
 		       "number of channels defaulting to DAC count\n");
 		num_dacs = STAC92HD73_DAC_COUNT;
 	}
+	spec->init = stac92hd73xx_core_init;
 	switch (num_dacs) {
 	case 0x3: /* 6 Channel */
-		spec->mixer = stac92hd73xx_6ch_mixer;
-		spec->init = stac92hd73xx_6ch_core_init;
 		spec->aloopback_ctl = stac92hd73xx_6ch_loopback;
 		break;
 	case 0x4: /* 8 Channel */
-		spec->mixer = stac92hd73xx_8ch_mixer;
-		spec->init = stac92hd73xx_8ch_core_init;
 		spec->aloopback_ctl = stac92hd73xx_8ch_loopback;
 		break;
 	case 0x5: /* 10 Channel */
-		spec->mixer = stac92hd73xx_10ch_mixer;
-		spec->init = stac92hd73xx_10ch_core_init;
 		spec->aloopback_ctl = stac92hd73xx_10ch_loopback;
 		break;
 	}
@@ -5216,8 +5080,6 @@ again:
 	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);
@@ -5235,13 +5097,8 @@ again:
 	case STAC_DELL_M6_DMIC:
 	case STAC_DELL_M6_BOTH:
 		spec->num_smuxes = 0;
-		spec->mixer = &stac92hd73xx_6ch_mixer[DELL_M6_MIXER];
-		spec->amp_nids = &stac92hd73xx_amp_nids[DELL_M6_AMP];
 		spec->eapd_switch = 0;
-		spec->num_amps = 1;
 
-		if (spec->board_config != STAC_DELL_EQ)
-			spec->init = dell_m6_core_init;
 		switch (spec->board_config) {
 		case STAC_DELL_M6_AMIC: /* Analog Mics */
 			snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170);
-- 
cgit v1.2.3


From 4a9678909b366b1f95e1f93220c61340ab866151 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 2 Sep 2009 01:04:24 +0200
Subject: ALSA: hda - Remove dead codes from patch_sigmatel.c

Due to the previous fix of input source for IDT92HD73xx, the amp mux
and amp vol stuff became unused.  Let's rip off dead codes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 132 -----------------------------------------
 1 file changed, 132 deletions(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 4cce92cf8ea..83a338b9c08 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -228,7 +228,6 @@ struct sigmatel_spec {
 
 	/* 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];
@@ -281,7 +280,6 @@ struct sigmatel_spec {
 	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 */
@@ -300,7 +298,6 @@ struct sigmatel_spec {
 	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;
 };
 
@@ -590,34 +587,6 @@ 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)
 {
@@ -813,41 +782,6 @@ 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,
@@ -1002,31 +936,6 @@ 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_ANALOG_LOOPBACK(verb_read, verb_write, cnt) \
 	{ \
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
@@ -2698,8 +2607,6 @@ 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,
@@ -2710,8 +2617,6 @@ 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),
@@ -3268,37 +3173,6 @@ 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)
@@ -3869,11 +3743,6 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 		if (err < 0)
 			return err;
 	}
-	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)
@@ -3910,7 +3779,6 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 		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;
 }
 
-- 
cgit v1.2.3


From 842ae63800bc2be62085d7ce5b3a2298c014d37a Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Wed, 2 Sep 2009 07:43:08 +0200
Subject: ALSA: hda - Add support of Alienware M17x laptop

Added the quirk for Alienware M17x with IDT 92HD73* codec chip.
It has two HP and one line-out jack, one mic jack, a built-in
speaker and a built-in mic.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_sigmatel.c | 27 +++++++++++++++++++++++++++
 1 file changed, 27 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 83a338b9c08..e31e53dc696 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -83,6 +83,7 @@ enum {
 	STAC_DELL_M6_DMIC,
 	STAC_DELL_M6_BOTH,
 	STAC_DELL_EQ,
+	STAC_ALIENWARE_M17X,
 	STAC_92HD73XX_MODELS
 };
 
@@ -1513,12 +1514,20 @@ static unsigned int dell_m6_pin_configs[13] = {
 	0x4f0000f0,
 };
 
+static unsigned int alienware_m17x_pin_configs[13] = {
+	0x0321101f, 0x0321101f, 0x03a11020, 0x03014020,
+	0x90170110, 0x4f0000f0, 0x4f0000f0, 0x4f0000f0,
+	0x4f0000f0, 0x90a60160, 0x4f0000f0, 0x4f0000f0,
+	0x904601b0,
+};
+
 static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = {
 	[STAC_92HD73XX_REF]	= ref92hd73xx_pin_configs,
 	[STAC_DELL_M6_AMIC]	= dell_m6_pin_configs,
 	[STAC_DELL_M6_DMIC]	= dell_m6_pin_configs,
 	[STAC_DELL_M6_BOTH]	= dell_m6_pin_configs,
 	[STAC_DELL_EQ]	= dell_m6_pin_configs,
+	[STAC_ALIENWARE_M17X]	= alienware_m17x_pin_configs,
 };
 
 static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = {
@@ -1530,6 +1539,7 @@ static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = {
 	[STAC_DELL_M6_DMIC] = "dell-m6-dmic",
 	[STAC_DELL_M6_BOTH] = "dell-m6",
 	[STAC_DELL_EQ] = "dell-eq",
+	[STAC_ALIENWARE_M17X] = "alienware",
 };
 
 static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
@@ -1567,6 +1577,12 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 	{} /* terminator */
 };
 
+static struct snd_pci_quirk stac92hd73xx_codec_id_cfg_tbl[] = {
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02a1,
+		      "Alienware M17x", STAC_ALIENWARE_M17X),
+	{} /* terminator */
+};
+
 static unsigned int ref92hd83xxx_pin_configs[10] = {
 	0x02214030, 0x02211010, 0x02a19020, 0x02170130,
 	0x01014050, 0x01819040, 0x01014020, 0x90a3014e,
@@ -4909,6 +4925,12 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
 							STAC_92HD73XX_MODELS,
 							stac92hd73xx_models,
 							stac92hd73xx_cfg_tbl);
+	/* check codec subsystem id if not found */
+	if (spec->board_config < 0)
+		spec->board_config =
+			snd_hda_check_board_codec_sid_config(codec,
+				STAC_92HD73XX_MODELS, stac92hd73xx_models,
+				stac92hd73xx_codec_id_cfg_tbl);
 again:
 	if (spec->board_config < 0)
 		snd_printdd(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
@@ -4983,6 +5005,11 @@ again:
 			break;
 		}
 		break;
+	case STAC_ALIENWARE_M17X:
+		spec->num_dmics = STAC92HD73XX_NUM_DMICS;
+		spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids);
+		spec->eapd_switch = 0;
+		break;
 	default:
 		spec->num_dmics = STAC92HD73XX_NUM_DMICS;
 		spec->num_smuxes = ARRAY_SIZE(stac92hd73xx_smux_nids);
-- 
cgit v1.2.3


From a65cc60f6349fe91539f736dad8fb2398488e431 Mon Sep 17 00:00:00 2001
From: "ddiaz@cenditel.gob.ve" <ddiaz@cenditel.gob.ve>
Date: Sat, 5 Sep 2009 16:28:06 -0430
Subject: ALSA: hda/realtek: Added support for CLEVO M540R subsystem, 6 channel
 + digital

The model clevo-m540r was created with 6-channel and digital support. All
functions verified except spdif. Tested with a VIT D2000 laptop which has:

[lspci extract]
 Audio device [0403]: Intel Corporation 82801H (ICH8 Family) HD Audio
Controller [8086:284b] (rev 03)
        Subsystem: CLEVO/KAPOK Computer Device [1558:5409]

[/proc/asound/card0/codec\#0 header]
Codec: Realtek ALC883
Address: 0
Function Id: 0x1
Vendor Id: 0x10ec0883
Subsystem Id: 0x15585409
Revision Id: 0x100002

[Added a comment about HP mute and the model description by tiwai]

Signed-off-by: Dhionel Diaz <ddiaz@cenditel.gob.ve>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 80 +++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 80 insertions(+)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index f3d186a1e3d..e58d4c5d6c1 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -232,6 +232,7 @@ enum {
 	ALC888_3ST_HP,
 	ALC888_6ST_DELL,
 	ALC883_MITAC,
+	ALC883_CLEVO_M540R,
 	ALC883_CLEVO_M720,
 	ALC883_FUJITSU_PI2515,
 	ALC888_FUJITSU_XA3530,
@@ -6656,6 +6657,52 @@ static struct hda_channel_mode alc882_3ST_6ch_modes[3] = {
 
 #define alc883_3ST_6ch_modes	alc882_3ST_6ch_modes
 
+/*
+ * 2ch mode
+ */
+static struct hda_verb alc883_3ST_ch2_clevo_init[] = {
+	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
+	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+static struct hda_verb alc883_3ST_ch4_clevo_init[] = {
+	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+	{ } /* end */
+};
+
+/*
+ * 6ch mode
+ */
+static struct hda_verb alc883_3ST_ch6_clevo_init[] = {
+	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 },
+	{ } /* end */
+};
+
+static struct hda_channel_mode alc883_3ST_6ch_clevo_modes[3] = {
+	{ 2, alc883_3ST_ch2_clevo_init },
+	{ 4, alc883_3ST_ch4_clevo_init },
+	{ 6, alc883_3ST_ch6_clevo_init },
+};
+
+
 /*
  * 6ch mode
  */
@@ -8134,6 +8181,22 @@ static struct hda_verb alc883_mitac_verbs[] = {
 	{ } /* end */
 };
 
+static struct hda_verb alc883_clevo_m540r_verbs[] = {
+	/* HP */
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	/* Int speaker */
+	/*{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},*/
+
+	/* enable unsolicited event */
+	/*
+	{0x15, 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 struct hda_verb alc883_clevo_m720_verbs[] = {
 	/* HP */
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
@@ -8670,6 +8733,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = {
 	[ALC888_3ST_HP]		= "3stack-hp",
 	[ALC888_6ST_DELL]	= "6stack-dell",
 	[ALC883_MITAC]		= "mitac",
+	[ALC883_CLEVO_M540R]	= "clevo-m540r",
 	[ALC883_CLEVO_M720]	= "clevo-m720",
 	[ALC883_FUJITSU_PI2515] = "fujitsu-pi2515",
 	[ALC888_FUJITSU_XA3530] = "fujitsu-xa3530",
@@ -8776,6 +8840,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720),
 	SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720),
+	SND_PCI_QUIRK(0x1558, 0x5409, "Clevo laptop M540R", ALC883_CLEVO_M540R),
 	SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC883_LAPTOP_EAPD),
 	SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch),
 	/* SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), */
@@ -9264,6 +9329,21 @@ static struct alc_config_preset alc882_presets[] = {
 		.channel_mode = alc883_3ST_2ch_modes,
 		.input_mux = &alc883_capture_source,
 	},
+	[ALC883_CLEVO_M540R] = {
+		.mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer },
+		.init_verbs = { alc883_init_verbs, alc883_clevo_m540r_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_clevo_modes),
+		.channel_mode = alc883_3ST_6ch_clevo_modes,
+		.need_dac_fix = 1,
+		.input_mux = &alc883_capture_source,
+		/* This machine has the hardware HP auto-muting, thus
+		 * we need no software mute via unsol event
+		 */
+	},
 	[ALC883_CLEVO_M720] = {
 		.mixers = { alc883_clevo_m720_mixer },
 		.init_verbs = { alc883_init_verbs, alc883_clevo_m720_verbs },
-- 
cgit v1.2.3


From 33d78674586aeb6a623b1e612e6f92dd83015ed3 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Tue, 8 Sep 2009 11:03:41 +0200
Subject: ALSA: hda - Use auto model for HP laptops with ALC268 codec

The HP laptops with ALC268 codec seem working better with model=auto
than model=toshiba; e.g. the auto model fixes missing digital outputs.
Let's fix quirk entry to choose auto model explicitly.

Tested-by: Jens Jorgensen <jbj1@ultraemail.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 sound/pci/hda/patch_realtek.c | 5 ++++-
 1 file changed, 4 insertions(+), 1 deletion(-)

(limited to 'sound')

diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index e58d4c5d6c1..7ed47f66ddd 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -12574,8 +12574,11 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
 						ALC268_ACER_ASPIRE_ONE),
 	SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL),
 	SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron Mini9", ALC268_DELL),
+	/* almost compatible with toshiba but with optional digital outs;
+	 * auto-probing seems working fine
+	 */
 	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP TX25xx series",
-			   ALC268_TOSHIBA),
+			   ALC268_AUTO),
 	SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST),
 	SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO),
 	SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA),
-- 
cgit v1.2.3