Linux-2.6.12-rc2

Initial git repository build. I'm not bothering with the full history,
even though we have it. We can create a separate "historical" git
archive of that later if we want to, and in the meantime it's about
3.2GB when imported into git - space that would just make the early
git days unnecessarily complicated, when we don't have a lot of good
infrastructure for it.

Let it rip!

11 files changed, 7867 insertions, 0 deletions

diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile
new file mode 100644
index 000000000000..570a59d33b41
--- /dev/null
+++ b/sound/pci/hda/Makefile
@@ -0,0 +1,7 @@
+snd-hda-intel-objs := hda_intel.o
+snd-hda-codec-objs := hda_codec.o hda_generic.o patch_realtek.o patch_cmedia.o patch_analog.o
+ifdef CONFIG_PROC_FS
+snd-hda-codec-objs += hda_proc.o
+endif
+
+obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-intel.o snd-hda-codec.o
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
new file mode 100644
index 000000000000..9ed117ac0c09
--- /dev/null
+++ b/sound/pci/hda/hda_codec.c
@@ -0,0 +1,1856 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * Copyright (c) 2004 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 <sound/driver.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <linux/moduleparam.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include "hda_local.h"
+
+
+MODULE_AUTHOR("Takashi Iwai <tiwai@suse.de>");
+MODULE_DESCRIPTION("Universal interface for High Definition Audio Codec");
+MODULE_LICENSE("GPL");
+
+
+/*
+ * vendor / preset table
+ */
+
+struct hda_vendor_id {
+	unsigned int id;
+	const char *name;
+};
+
+/* codec vendor labels */
+static struct hda_vendor_id hda_vendor_ids[] = {
+	{ 0x10ec, "Realtek" },
+	{ 0x13f6, "C-Media" },
+	{ 0x434d, "C-Media" },
+	{} /* terminator */
+};
+
+/* codec presets */
+#include "hda_patch.h"
+
+
+/**
+ * snd_hda_codec_read - send a command and get the response
+ * @codec: the HDA codec
+ * @nid: NID to send the command
+ * @direct: direct flag
+ * @verb: the verb to send
+ * @parm: the parameter for the verb
+ *
+ * Send a single command and read the corresponding response.
+ *
+ * Returns the obtained response value, or -1 for an error.
+ */
+unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, int direct,
+				unsigned int verb, unsigned int parm)
+{
+	unsigned int res;
+	down(&codec->bus->cmd_mutex);
+	if (! codec->bus->ops.command(codec, nid, direct, verb, parm))
+		res = codec->bus->ops.get_response(codec);
+	else
+		res = (unsigned int)-1;
+	up(&codec->bus->cmd_mutex);
+	return res;
+}
+
+/**
+ * snd_hda_codec_write - send a single command without waiting for response
+ * @codec: the HDA codec
+ * @nid: NID to send the command
+ * @direct: direct flag
+ * @verb: the verb to send
+ * @parm: the parameter for the verb
+ *
+ * Send a single command without waiting for response.
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct,
+			 unsigned int verb, unsigned int parm)
+{
+	int err;
+	down(&codec->bus->cmd_mutex);
+	err = codec->bus->ops.command(codec, nid, direct, verb, parm);
+	up(&codec->bus->cmd_mutex);
+	return err;
+}
+
+/**
+ * snd_hda_sequence_write - sequence writes
+ * @codec: the HDA codec
+ * @seq: VERB array to send
+ *
+ * Send the commands sequentially from the given array.
+ * The array must be terminated with NID=0.
+ */
+void snd_hda_sequence_write(struct hda_codec *codec, const struct hda_verb *seq)
+{
+	for (; seq->nid; seq++)
+		snd_hda_codec_write(codec, seq->nid, 0, seq->verb, seq->param);
+}
+
+/**
+ * snd_hda_get_sub_nodes - get the range of sub nodes
+ * @codec: the HDA codec
+ * @nid: NID to parse
+ * @start_id: the pointer to store the start NID
+ *
+ * Parse the NID and store the start NID of its sub-nodes.
+ * Returns the number of sub-nodes.
+ */
+int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *start_id)
+{
+	unsigned int parm;
+
+	parm = snd_hda_param_read(codec, nid, AC_PAR_NODE_COUNT);
+	*start_id = (parm >> 16) & 0x7fff;
+	return (int)(parm & 0x7fff);
+}
+
+/**
+ * snd_hda_get_connections - get connection list
+ * @codec: the HDA codec
+ * @nid: NID to parse
+ * @conn_list: connection list array
+ * @max_conns: max. number of connections to store
+ *
+ * Parses the connection list of the given widget and stores the list
+ * of NIDs.
+ *
+ * Returns the number of connections, or a negative error code.
+ */
+int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid,
+			    hda_nid_t *conn_list, int max_conns)
+{
+	unsigned int parm;
+	int i, j, conn_len, num_tupples, conns;
+	unsigned int shift, num_elems, mask;
+
+	snd_assert(conn_list && max_conns > 0, return -EINVAL);
+
+	parm = snd_hda_param_read(codec, nid, AC_PAR_CONNLIST_LEN);
+	if (parm & AC_CLIST_LONG) {
+		/* long form */
+		shift = 16;
+		num_elems = 2;
+	} else {
+		/* short form */
+		shift = 8;
+		num_elems = 4;
+	}
+	conn_len = parm & AC_CLIST_LENGTH;
+	num_tupples = num_elems / 2;
+	mask = (1 << (shift-1)) - 1;
+
+	if (! conn_len)
+		return 0; /* no connection */
+
+	if (conn_len == 1) {
+		/* single connection */
+		parm = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_LIST, 0);
+		conn_list[0] = parm & mask;
+		return 1;
+	}
+
+	/* multi connection */
+	conns = 0;
+	for (i = 0; i < conn_len; i += num_elems) {
+		parm = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_LIST, i);
+		for (j = 0; j < num_tupples; j++) {
+			int range_val;
+			hda_nid_t val1, val2, n;
+			range_val = parm & (1 << (shift-1)); /* ranges */
+			val1 = parm & mask;
+			parm >>= shift;
+			val2 = parm & mask;
+			parm >>= shift;
+			if (range_val) {
+				/* ranges between val1 and val2 */
+				if (val1 > val2) {
+					snd_printk(KERN_WARNING "hda_codec: invalid dep_range_val %x:%x\n", val1, val2);
+					continue;
+				}
+				for (n = val1; n <= val2; n++) {
+					if (conns >= max_conns)
+						return -EINVAL;
+					conn_list[conns++] = n;
+				}
+			} else {
+				if (! val1)
+					break;
+				if (conns >= max_conns)
+					return -EINVAL;
+				conn_list[conns++] = val1;
+				if (! val2)
+					break;
+				if (conns >= max_conns)
+					return -EINVAL;
+				conn_list[conns++] = val2;
+			}
+		}
+	}
+	return conns;
+}
+
+
+/**
+ * snd_hda_queue_unsol_event - add an unsolicited event to queue
+ * @bus: the BUS
+ * @res: unsolicited event (lower 32bit of RIRB entry)
+ * @res_ex: codec addr and flags (upper 32bit or RIRB entry)
+ *
+ * Adds the given event to the queue.  The events are processed in
+ * the workqueue asynchronously.  Call this function in the interrupt
+ * hanlder when RIRB receives an unsolicited event.
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex)
+{
+	struct hda_bus_unsolicited *unsol;
+	unsigned int wp;
+
+	if ((unsol = bus->unsol) == NULL)
+		return 0;
+
+	wp = (unsol->wp + 1) % HDA_UNSOL_QUEUE_SIZE;
+	unsol->wp = wp;
+
+	wp <<= 1;
+	unsol->queue[wp] = res;
+	unsol->queue[wp + 1] = res_ex;
+
+	queue_work(unsol->workq, &unsol->work);
+
+	return 0;
+}
+
+/*
+ * process queueud unsolicited events
+ */
+static void process_unsol_events(void *data)
+{
+	struct hda_bus *bus = data;
+	struct hda_bus_unsolicited *unsol = bus->unsol;
+	struct hda_codec *codec;
+	unsigned int rp, caddr, res;
+
+	while (unsol->rp != unsol->wp) {
+		rp = (unsol->rp + 1) % HDA_UNSOL_QUEUE_SIZE;
+		unsol->rp = rp;
+		rp <<= 1;
+		res = unsol->queue[rp];
+		caddr = unsol->queue[rp + 1];
+		if (! (caddr & (1 << 4))) /* no unsolicited event? */
+			continue;
+		codec = bus->caddr_tbl[caddr & 0x0f];
+		if (codec && codec->patch_ops.unsol_event)
+			codec->patch_ops.unsol_event(codec, res);
+	}
+}
+
+/*
+ * initialize unsolicited queue
+ */
+static int init_unsol_queue(struct hda_bus *bus)
+{
+	struct hda_bus_unsolicited *unsol;
+
+	unsol = kcalloc(1, sizeof(*unsol), GFP_KERNEL);
+	if (! unsol) {
+		snd_printk(KERN_ERR "hda_codec: can't allocate unsolicited queue\n");
+		return -ENOMEM;
+	}
+	unsol->workq = create_workqueue("hda_codec");
+	if (! unsol->workq) {
+		snd_printk(KERN_ERR "hda_codec: can't create workqueue\n");
+		kfree(unsol);
+		return -ENOMEM;
+	}
+	INIT_WORK(&unsol->work, process_unsol_events, bus);
+	bus->unsol = unsol;
+	return 0;
+}
+
+/*
+ * destructor
+ */
+static void snd_hda_codec_free(struct hda_codec *codec);
+
+static int snd_hda_bus_free(struct hda_bus *bus)
+{
+	struct list_head *p, *n;
+
+	if (! bus)
+		return 0;
+	if (bus->unsol) {
+		destroy_workqueue(bus->unsol->workq);
+		kfree(bus->unsol);
+	}
+	list_for_each_safe(p, n, &bus->codec_list) {
+		struct hda_codec *codec = list_entry(p, struct hda_codec, list);
+		snd_hda_codec_free(codec);
+	}
+	if (bus->ops.private_free)
+		bus->ops.private_free(bus);
+	kfree(bus);
+	return 0;
+}
+
+static int snd_hda_bus_dev_free(snd_device_t *device)
+{
+	struct hda_bus *bus = device->device_data;
+	return snd_hda_bus_free(bus);
+}
+
+/**
+ * snd_hda_bus_new - create a HDA bus
+ * @card: the card entry
+ * @temp: the template for hda_bus information
+ * @busp: the pointer to store the created bus instance
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_hda_bus_new(snd_card_t *card, const struct hda_bus_template *temp,
+		    struct hda_bus **busp)
+{
+	struct hda_bus *bus;
+	int err;
+	static snd_device_ops_t dev_ops = {
+		.dev_free = snd_hda_bus_dev_free,
+	};
+
+	snd_assert(temp, return -EINVAL);
+	snd_assert(temp->ops.command && temp->ops.get_response, return -EINVAL);
+
+	if (busp)
+		*busp = NULL;
+
+	bus = kcalloc(1, sizeof(*bus), GFP_KERNEL);
+	if (bus == NULL) {
+		snd_printk(KERN_ERR "can't allocate struct hda_bus\n");
+		return -ENOMEM;
+	}
+
+	bus->card = card;
+	bus->private_data = temp->private_data;
+	bus->pci = temp->pci;
+	bus->modelname = temp->modelname;
+	bus->ops = temp->ops;
+
+	init_MUTEX(&bus->cmd_mutex);
+	INIT_LIST_HEAD(&bus->codec_list);
+
+	init_unsol_queue(bus);
+
+	if ((err = snd_device_new(card, SNDRV_DEV_BUS, bus, &dev_ops)) < 0) {
+		snd_hda_bus_free(bus);
+		return err;
+	}
+	if (busp)
+		*busp = bus;
+	return 0;
+}
+
+
+/*
+ * find a matching codec preset
+ */
+static const struct hda_codec_preset *find_codec_preset(struct hda_codec *codec)
+{
+	const struct hda_codec_preset **tbl, *preset;
+
+	for (tbl = hda_preset_tables; *tbl; tbl++) {
+		for (preset = *tbl; preset->id; preset++) {
+			u32 mask = preset->mask;
+			if (! mask)
+				mask = ~0;
+			if (preset->id == (codec->vendor_id & mask))
+				return preset;
+		}
+	}
+	return NULL;
+}
+
+/*
+ * snd_hda_get_codec_name - store the codec name
+ */
+void snd_hda_get_codec_name(struct hda_codec *codec,
+			    char *name, int namelen)
+{
+	const struct hda_vendor_id *c;
+	const char *vendor = NULL;
+	u16 vendor_id = codec->vendor_id >> 16;
+	char tmp[16];
+
+	for (c = hda_vendor_ids; c->id; c++) {
+		if (c->id == vendor_id) {
+			vendor = c->name;
+			break;
+		}
+	}
+	if (! vendor) {
+		sprintf(tmp, "Generic %04x", vendor_id);
+		vendor = tmp;
+	}
+	if (codec->preset && codec->preset->name)
+		snprintf(name, namelen, "%s %s", vendor, codec->preset->name);
+	else
+		snprintf(name, namelen, "%s ID %x", vendor, codec->vendor_id & 0xffff);
+}
+
+/*
+ * look for an AFG node
+ *
+ * return 0 if not found
+ */
+static int look_for_afg_node(struct hda_codec *codec)
+{
+	int i, total_nodes;
+	hda_nid_t nid;
+
+	total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid);
+	for (i = 0; i < total_nodes; i++, nid++) {
+		if ((snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE) & 0xff) ==
+		    AC_GRP_AUDIO_FUNCTION)
+			return nid;
+	}
+	return 0;
+}
+
+/*
+ * codec destructor
+ */
+static void snd_hda_codec_free(struct hda_codec *codec)
+{
+	if (! codec)
+		return;
+	list_del(&codec->list);
+	codec->bus->caddr_tbl[codec->addr] = NULL;
+	if (codec->patch_ops.free)
+		codec->patch_ops.free(codec);
+	kfree(codec);
+}
+
+static void init_amp_hash(struct hda_codec *codec);
+
+/**
+ * snd_hda_codec_new - create a HDA codec
+ * @bus: the bus to assign
+ * @codec_addr: the codec address
+ * @codecp: the pointer to store the generated codec
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
+		      struct hda_codec **codecp)
+{
+	struct hda_codec *codec;
+	char component[13];
+	int err;
+
+	snd_assert(bus, return -EINVAL);
+	snd_assert(codec_addr <= HDA_MAX_CODEC_ADDRESS, return -EINVAL);
+
+	if (bus->caddr_tbl[codec_addr]) {
+		snd_printk(KERN_ERR "hda_codec: address 0x%x is already occupied\n", codec_addr);
+		return -EBUSY;
+	}
+
+	codec = kcalloc(1, sizeof(*codec), GFP_KERNEL);
+	if (codec == NULL) {
+		snd_printk(KERN_ERR "can't allocate struct hda_codec\n");
+		return -ENOMEM;
+	}
+
+	codec->bus = bus;
+	codec->addr = codec_addr;
+	init_MUTEX(&codec->spdif_mutex);
+	init_amp_hash(codec);
+
+	list_add_tail(&codec->list, &bus->codec_list);
+	bus->caddr_tbl[codec_addr] = codec;
+
+	codec->vendor_id = snd_hda_param_read(codec, AC_NODE_ROOT, AC_PAR_VENDOR_ID);
+	codec->subsystem_id = snd_hda_param_read(codec, AC_NODE_ROOT, AC_PAR_SUBSYSTEM_ID);
+	codec->revision_id = snd_hda_param_read(codec, AC_NODE_ROOT, AC_PAR_REV_ID);
+
+	/* FIXME: support for multiple AFGs? */
+	codec->afg = look_for_afg_node(codec);
+	if (! codec->afg) {
+		snd_printk(KERN_ERR "hda_codec: no AFG node found\n");
+		snd_hda_codec_free(codec);
+		return -ENODEV;
+	}
+
+	codec->preset = find_codec_preset(codec);
+	if (! *bus->card->mixername)
+		snd_hda_get_codec_name(codec, bus->card->mixername,
+				       sizeof(bus->card->mixername));
+
+	if (codec->preset && codec->preset->patch)
+		err = codec->preset->patch(codec);
+	else
+		err = snd_hda_parse_generic_codec(codec);
+	if (err < 0) {
+		snd_hda_codec_free(codec);
+		return err;
+	}
+
+	snd_hda_codec_proc_new(codec);
+
+	sprintf(component, "HDA:%08x", codec->vendor_id);
+	snd_component_add(codec->bus->card, component);
+
+	if (codecp)
+		*codecp = codec;
+	return 0;
+}
+
+/**
+ * snd_hda_codec_setup_stream - set up the codec for streaming
+ * @codec: the CODEC to set up
+ * @nid: the NID to set up
+ * @stream_tag: stream tag to pass, it's between 0x1 and 0xf.
+ * @channel_id: channel id to pass, zero based.
+ * @format: stream format.
+ */
+void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stream_tag,
+				int channel_id, int format)
+{
+	snd_printdd("hda_codec_setup_stream: NID=0x%x, stream=0x%x, channel=%d, format=0x%x\n",
+		    nid, stream_tag, channel_id, format);
+	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CHANNEL_STREAMID,
+			    (stream_tag << 4) | channel_id);
+	msleep(1);
+	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_STREAM_FORMAT, format);
+}
+
+
+/*
+ * amp access functions
+ */
+
+#define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + (idx) * 32 + (dir) * 64)
+#define INFO_AMP_CAPS	(1<<0)
+#define INFO_AMP_VOL	(1<<1)
+
+/* initialize the hash table */
+static void init_amp_hash(struct hda_codec *codec)
+{
+	memset(codec->amp_hash, 0xff, sizeof(codec->amp_hash));
+	codec->num_amp_entries = 0;
+}
+
+/* query the hash.  allocate an entry if not found. */
+static struct hda_amp_info *get_alloc_amp_hash(struct hda_codec *codec, u32 key)
+{
+	u16 idx = key % (u16)ARRAY_SIZE(codec->amp_hash);
+	u16 cur = codec->amp_hash[idx];
+	struct hda_amp_info *info;
+
+	while (cur != 0xffff) {
+		info = &codec->amp_info[cur];
+		if (info->key == key)
+			return info;
+		cur = info->next;
+	}
+
+	/* add a new hash entry */
+	if (codec->num_amp_entries >= ARRAY_SIZE(codec->amp_info)) {
+		snd_printk(KERN_ERR "hda_codec: Tooooo many amps!\n");
+		return NULL;
+	}
+	cur = codec->num_amp_entries++;
+	info = &codec->amp_info[cur];
+	info->key = key;
+	info->status = 0; /* not initialized yet */
+	info->next = codec->amp_hash[idx];
+	codec->amp_hash[idx] = cur;
+
+	return info;
+}
+
+/*
+ * query AMP capabilities for the given widget and direction
+ */
+static u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction)
+{
+	struct hda_amp_info *info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, 0));
+
+	if (! info)
+		return 0;
+	if (! (info->status & INFO_AMP_CAPS)) {
+		if (!(snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP) & AC_WCAP_AMP_OVRD))
+			nid = codec->afg;
+		info->amp_caps = snd_hda_param_read(codec, nid, direction == HDA_OUTPUT ?
+						    AC_PAR_AMP_OUT_CAP : AC_PAR_AMP_IN_CAP);
+		info->status |= INFO_AMP_CAPS;
+	}
+	return info->amp_caps;
+}
+
+/*
+ * read the current volume to info
+ * if the cache exists, read from the cache.
+ */
+static void get_vol_mute(struct hda_codec *codec, struct hda_amp_info *info,
+			 hda_nid_t nid, int ch, int direction, int index)
+{
+	u32 val, parm;
+
+	if (info->status & (INFO_AMP_VOL << ch))
+		return;
+
+	parm = ch ? AC_AMP_GET_RIGHT : AC_AMP_GET_LEFT;
+	parm |= direction == HDA_OUTPUT ? AC_AMP_GET_OUTPUT : AC_AMP_GET_INPUT;
+	parm |= index;
+	val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_AMP_GAIN_MUTE, parm);
+	info->vol[ch] = val & 0xff;
+	info->status |= INFO_AMP_VOL << ch;
+}
+
+/*
+ * write the current volume in info to the h/w
+ */
+static void put_vol_mute(struct hda_codec *codec,
+			 hda_nid_t nid, int ch, int direction, int index, int val)
+{
+	u32 parm;
+
+	parm = ch ? AC_AMP_SET_RIGHT : AC_AMP_SET_LEFT;
+	parm |= direction == HDA_OUTPUT ? AC_AMP_SET_OUTPUT : AC_AMP_SET_INPUT;
+	parm |= index << AC_AMP_SET_INDEX_SHIFT;
+	parm |= val;
+	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, parm);
+}
+
+/*
+ * read/write AMP value.  The volume is between 0 to 0x7f, 0x80 = mute bit.
+ */
+int snd_hda_codec_amp_read(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int index)
+{
+	struct hda_amp_info *info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, index));
+	if (! info)
+		return 0;
+	get_vol_mute(codec, info, nid, ch, direction, index);
+	return info->vol[ch];
+}
+
+int snd_hda_codec_amp_write(struct hda_codec *codec, hda_nid_t nid, int ch, int direction, int idx, int val)
+{
+	struct hda_amp_info *info = get_alloc_amp_hash(codec, HDA_HASH_KEY(nid, direction, idx));
+	if (! info)
+		return 0;
+	get_vol_mute(codec, info, nid, ch, direction, idx);
+	if (info->vol[ch] == val && ! codec->in_resume)
+		return 0;
+	put_vol_mute(codec, nid, ch, direction, idx, val);
+	info->vol[ch] = val;
+	return 1;
+}
+
+
+/*
+ * AMP control callbacks
+ */
+/* retrieve parameters from private_value */
+#define get_amp_nid(kc)		((kc)->private_value & 0xffff)
+#define get_amp_channels(kc)	(((kc)->private_value >> 16) & 0x3)
+#define get_amp_direction(kc)	(((kc)->private_value >> 18) & 0x1)
+#define get_amp_index(kc)	(((kc)->private_value >> 19) & 0xf)
+
+/* volume */
+int snd_hda_mixer_amp_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	u16 nid = get_amp_nid(kcontrol);
+	u8 chs = get_amp_channels(kcontrol);
+	int dir = get_amp_direction(kcontrol);
+	u32 caps;
+
+	caps = query_amp_caps(codec, nid, dir);
+	caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; /* num steps */
+	if (! caps) {
+		printk(KERN_WARNING "hda_codec: num_steps = 0 for NID=0x%x\n", nid);
+		return -EINVAL;
+	}
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = chs == 3 ? 2 : 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = caps;
+	return 0;
+}
+
+int snd_hda_mixer_amp_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = get_amp_nid(kcontrol);
+	int chs = get_amp_channels(kcontrol);
+	int dir = get_amp_direction(kcontrol);
+	int idx = get_amp_index(kcontrol);
+	long *valp = ucontrol->value.integer.value;
+
+	if (chs & 1)
+		*valp++ = snd_hda_codec_amp_read(codec, nid, 0, dir, idx) & 0x7f;
+	if (chs & 2)
+		*valp = snd_hda_codec_amp_read(codec, nid, 1, dir, idx) & 0x7f;
+	return 0;
+}
+
+int snd_hda_mixer_amp_volume_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = get_amp_nid(kcontrol);
+	int chs = get_amp_channels(kcontrol);
+	int dir = get_amp_direction(kcontrol);
+	int idx = get_amp_index(kcontrol);
+	int val;
+	long *valp = ucontrol->value.integer.value;
+	int change = 0;
+
+	if (chs & 1) {
+		val = *valp & 0x7f;
+		val |= snd_hda_codec_amp_read(codec, nid, 0, dir, idx) & 0x80;
+		change = snd_hda_codec_amp_write(codec, nid, 0, dir, idx, val);
+		valp++;
+	}
+	if (chs & 2) {
+		val = *valp & 0x7f;
+		val |= snd_hda_codec_amp_read(codec, nid, 1, dir, idx) & 0x80;
+		change |= snd_hda_codec_amp_write(codec, nid, 1, dir, idx, val);
+	}
+	return change;
+}
+
+/* switch */
+int snd_hda_mixer_amp_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+	int chs = get_amp_channels(kcontrol);
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+	uinfo->count = chs == 3 ? 2 : 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 1;
+	return 0;
+}
+
+int snd_hda_mixer_amp_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = get_amp_nid(kcontrol);
+	int chs = get_amp_channels(kcontrol);
+	int dir = get_amp_direction(kcontrol);
+	int idx = get_amp_index(kcontrol);
+	long *valp = ucontrol->value.integer.value;
+
+	if (chs & 1)
+		*valp++ = (snd_hda_codec_amp_read(codec, nid, 0, dir, idx) & 0x80) ? 0 : 1;
+	if (chs & 2)
+		*valp = (snd_hda_codec_amp_read(codec, nid, 1, dir, idx) & 0x80) ? 0 : 1;
+	return 0;
+}
+
+int snd_hda_mixer_amp_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = get_amp_nid(kcontrol);
+	int chs = get_amp_channels(kcontrol);
+	int dir = get_amp_direction(kcontrol);
+	int idx = get_amp_index(kcontrol);
+	int val;
+	long *valp = ucontrol->value.integer.value;
+	int change = 0;
+
+	if (chs & 1) {
+		val = snd_hda_codec_amp_read(codec, nid, 0, dir, idx) & 0x7f;
+		val |= *valp ? 0 : 0x80;
+		change = snd_hda_codec_amp_write(codec, nid, 0, dir, idx, val);
+		valp++;
+	}
+	if (chs & 2) {
+		val = snd_hda_codec_amp_read(codec, nid, 1, dir, idx) & 0x7f;
+		val |= *valp ? 0 : 0x80;
+		change = snd_hda_codec_amp_write(codec, nid, 1, dir, idx, val);
+	}
+	return change;
+}
+
+/*
+ * SPDIF out controls
+ */
+
+static int snd_hda_spdif_mask_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+	uinfo->count = 1;
+	return 0;
+}
+
+static int snd_hda_spdif_cmask_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	ucontrol->value.iec958.status[0] = IEC958_AES0_PROFESSIONAL |
+					   IEC958_AES0_NONAUDIO |
+					   IEC958_AES0_CON_EMPHASIS_5015 |
+					   IEC958_AES0_CON_NOT_COPYRIGHT;
+	ucontrol->value.iec958.status[1] = IEC958_AES1_CON_CATEGORY |
+					   IEC958_AES1_CON_ORIGINAL;
+	return 0;
+}
+
+static int snd_hda_spdif_pmask_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	ucontrol->value.iec958.status[0] = IEC958_AES0_PROFESSIONAL |
+					   IEC958_AES0_NONAUDIO |
+					   IEC958_AES0_PRO_EMPHASIS_5015;
+	return 0;
+}
+
+static int snd_hda_spdif_default_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.iec958.status[0] = codec->spdif_status & 0xff;
+	ucontrol->value.iec958.status[1] = (codec->spdif_status >> 8) & 0xff;
+	ucontrol->value.iec958.status[2] = (codec->spdif_status >> 16) & 0xff;
+	ucontrol->value.iec958.status[3] = (codec->spdif_status >> 24) & 0xff;
+
+	return 0;
+}
+
+/* convert from SPDIF status bits to HDA SPDIF bits
+ * bit 0 (DigEn) is always set zero (to be filled later)
+ */
+static unsigned short convert_from_spdif_status(unsigned int sbits)
+{
+	unsigned short val = 0;
+
+	if (sbits & IEC958_AES0_PROFESSIONAL)
+		val |= 1 << 6;
+	if (sbits & IEC958_AES0_NONAUDIO)
+		val |= 1 << 5;
+	if (sbits & IEC958_AES0_PROFESSIONAL) {
+		if ((sbits & IEC958_AES0_PRO_EMPHASIS) == IEC958_AES0_PRO_EMPHASIS_5015)
+			val |= 1 << 3;
+	} else {
+		if ((sbits & IEC958_AES0_CON_EMPHASIS) == IEC958_AES0_CON_EMPHASIS_5015)
+			val |= 1 << 3;
+		if (! (sbits & IEC958_AES0_CON_NOT_COPYRIGHT))
+			val |= 1 << 4;
+		if (sbits & (IEC958_AES1_CON_ORIGINAL << 8))
+			val |= 1 << 7;
+		val |= sbits & (IEC958_AES1_CON_CATEGORY << 8);
+	}
+	return val;
+}
+
+/* convert to SPDIF status bits from HDA SPDIF bits
+ */
+static unsigned int convert_to_spdif_status(unsigned short val)
+{
+	unsigned int sbits = 0;
+
+	if (val & (1 << 5))
+		sbits |= IEC958_AES0_NONAUDIO;
+	if (val & (1 << 6))
+		sbits |= IEC958_AES0_PROFESSIONAL;
+	if (sbits & IEC958_AES0_PROFESSIONAL) {
+		if (sbits & (1 << 3))
+			sbits |= IEC958_AES0_PRO_EMPHASIS_5015;
+	} else {
+		if (val & (1 << 3))
+			sbits |= IEC958_AES0_CON_EMPHASIS_5015;
+		if (! (val & (1 << 4)))
+			sbits |= IEC958_AES0_CON_NOT_COPYRIGHT;
+		if (val & (1 << 7))
+			sbits |= (IEC958_AES1_CON_ORIGINAL << 8);
+		sbits |= val & (0x7f << 8);
+	}
+	return sbits;
+}
+
+static int snd_hda_spdif_default_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = kcontrol->private_value;
+	unsigned short val;
+	int change;
+
+	down(&codec->spdif_mutex);
+	codec->spdif_status = ucontrol->value.iec958.status[0] |
+		((unsigned int)ucontrol->value.iec958.status[1] << 8) |
+		((unsigned int)ucontrol->value.iec958.status[2] << 16) |
+		((unsigned int)ucontrol->value.iec958.status[3] << 24);
+	val = convert_from_spdif_status(codec->spdif_status);
+	val |= codec->spdif_ctls & 1;
+	change = codec->spdif_ctls != val;
+	codec->spdif_ctls = val;
+
+	if (change || codec->in_resume) {
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff);
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_2, val >> 8);
+	}
+
+	up(&codec->spdif_mutex);
+	return change;
+}
+
+static int snd_hda_spdif_out_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 1;
+	return 0;
+}
+
+static int snd_hda_spdif_out_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.integer.value[0] = codec->spdif_ctls & 1;
+	return 0;
+}
+
+static int snd_hda_spdif_out_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = kcontrol->private_value;
+	unsigned short val;
+	int change;
+
+	down(&codec->spdif_mutex);
+	val = codec->spdif_ctls & ~1;
+	if (ucontrol->value.integer.value[0])
+		val |= 1;
+	change = codec->spdif_ctls != val;
+	if (change || codec->in_resume) {
+		codec->spdif_ctls = val;
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff);
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+				    AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT |
+				    AC_AMP_SET_OUTPUT | ((val & 1) ? 0 : 0x80));
+	}
+	up(&codec->spdif_mutex);
+	return change;
+}
+
+static snd_kcontrol_new_t dig_mixes[] = {
+	{
+		.access = SNDRV_CTL_ELEM_ACCESS_READ,
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK),
+		.info = snd_hda_spdif_mask_info,
+		.get = snd_hda_spdif_cmask_get,
+	},
+	{
+		.access = SNDRV_CTL_ELEM_ACCESS_READ,
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,PRO_MASK),
+		.info = snd_hda_spdif_mask_info,
+		.get = snd_hda_spdif_pmask_get,
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+		.info = snd_hda_spdif_mask_info,
+		.get = snd_hda_spdif_default_get,
+		.put = snd_hda_spdif_default_put,
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH),
+		.info = snd_hda_spdif_out_switch_info,
+		.get = snd_hda_spdif_out_switch_get,
+		.put = snd_hda_spdif_out_switch_put,
+	},
+	{ } /* end */
+};
+
+/**
+ * snd_hda_create_spdif_out_ctls - create Output SPDIF-related controls
+ * @codec: the HDA codec
+ * @nid: audio out widget NID
+ *
+ * Creates controls related with the SPDIF output.
+ * Called from each patch supporting the SPDIF out.
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid)
+{
+	int err;
+	snd_kcontrol_t *kctl;
+	snd_kcontrol_new_t *dig_mix;
+
+	for (dig_mix = dig_mixes; dig_mix->name; dig_mix++) {
+		kctl = snd_ctl_new1(dig_mix, codec);
+		kctl->private_value = nid;
+		if ((err = snd_ctl_add(codec->bus->card, kctl)) < 0)
+			return err;
+	}
+	codec->spdif_ctls = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0);
+	codec->spdif_status = convert_to_spdif_status(codec->spdif_ctls);
+	return 0;
+}
+
+/*
+ * SPDIF input
+ */
+
+#define snd_hda_spdif_in_switch_info	snd_hda_spdif_out_switch_info
+
+static int snd_hda_spdif_in_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.integer.value[0] = codec->spdif_in_enable;
+	return 0;
+}
+
+static int snd_hda_spdif_in_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = kcontrol->private_value;
+	unsigned int val = !!ucontrol->value.integer.value[0];
+	int change;
+
+	down(&codec->spdif_mutex);
+	change = codec->spdif_in_enable != val;
+	if (change || codec->in_resume) {
+		codec->spdif_in_enable = val;
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val);
+	}
+	up(&codec->spdif_mutex);
+	return change;
+}
+
+static int snd_hda_spdif_in_status_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	hda_nid_t nid = kcontrol->private_value;
+	unsigned short val;
+	unsigned int sbits;
+
+	val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0);
+	sbits = convert_to_spdif_status(val);
+	ucontrol->value.iec958.status[0] = sbits;
+	ucontrol->value.iec958.status[1] = sbits >> 8;
+	ucontrol->value.iec958.status[2] = sbits >> 16;
+	ucontrol->value.iec958.status[3] = sbits >> 24;
+	return 0;
+}
+
+static snd_kcontrol_new_t dig_in_ctls[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = SNDRV_CTL_NAME_IEC958("",CAPTURE,SWITCH),
+		.info = snd_hda_spdif_in_switch_info,
+		.get = snd_hda_spdif_in_switch_get,
+		.put = snd_hda_spdif_in_switch_put,
+	},
+	{
+		.access = SNDRV_CTL_ELEM_ACCESS_READ,
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = SNDRV_CTL_NAME_IEC958("",CAPTURE,DEFAULT),
+		.info = snd_hda_spdif_mask_info,
+		.get = snd_hda_spdif_in_status_get,
+	},
+	{ } /* end */
+};
+
+/**
+ * snd_hda_create_spdif_in_ctls - create Input SPDIF-related controls
+ * @codec: the HDA codec
+ * @nid: audio in widget NID
+ *
+ * Creates controls related with the SPDIF input.
+ * Called from each patch supporting the SPDIF in.
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid)
+{
+	int err;
+	snd_kcontrol_t *kctl;
+	snd_kcontrol_new_t *dig_mix;
+
+	for (dig_mix = dig_in_ctls; dig_mix->name; dig_mix++) {
+		kctl = snd_ctl_new1(dig_mix, codec);
+		kctl->private_value = nid;
+		if ((err = snd_ctl_add(codec->bus->card, kctl)) < 0)
+			return err;
+	}
+	codec->spdif_in_enable = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0) & 1;
+	return 0;
+}
+
+
+/**
+ * snd_hda_build_controls - build mixer controls
+ * @bus: the BUS
+ *
+ * Creates mixer controls for each codec included in the bus.
+ *
+ * Returns 0 if successful, otherwise a negative error code.
+ */
+int snd_hda_build_controls(struct hda_bus *bus)
+{
+	struct list_head *p;
+
+	/* build controls */
+	list_for_each(p, &bus->codec_list) {
+		struct hda_codec *codec = list_entry(p, struct hda_codec, list);
+		int err;
+		if (! codec->patch_ops.build_controls)
+			continue;
+		err = codec->patch_ops.build_controls(codec);
+		if (err < 0)
+			return err;
+	}
+
+	/* initialize */
+	list_for_each(p, &bus->codec_list) {
+		struct hda_codec *codec = list_entry(p, struct hda_codec, list);
+		int err;
+		if (! codec->patch_ops.init)
+			continue;
+		err = codec->patch_ops.init(codec);
+		if (err < 0)
+			return err;
+	}
+	return 0;
+}
+
+
+/*
+ * stream formats
+ */
+static unsigned int rate_bits[][3] = {
+	/* rate in Hz, ALSA rate bitmask, HDA format value */
+	{ 8000, SNDRV_PCM_RATE_8000, 0x0500 }, /* 1/6 x 48 */
+	{ 11025, SNDRV_PCM_RATE_11025, 0x4300 }, /* 1/4 x 44 */
+	{ 16000, SNDRV_PCM_RATE_16000, 0x0200 }, /* 1/3 x 48 */
+	{ 22050, SNDRV_PCM_RATE_22050, 0x4100 }, /* 1/2 x 44 */
+	{ 32000, SNDRV_PCM_RATE_32000, 0x0a00 }, /* 2/3 x 48 */
+	{ 44100, SNDRV_PCM_RATE_44100, 0x4000 }, /* 44 */
+	{ 48000, SNDRV_PCM_RATE_48000, 0x0000 }, /* 48 */
+	{ 88200, SNDRV_PCM_RATE_88200, 0x4800 }, /* 2 x 44 */
+	{ 96000, SNDRV_PCM_RATE_96000, 0x0800 }, /* 2 x 48 */
+	{ 176400, SNDRV_PCM_RATE_176400, 0x5800 },/* 4 x 44 */
+	{ 192000, SNDRV_PCM_RATE_192000, 0x1800 }, /* 4 x 48 */
+	{ 0 }
+};
+
+/**
+ * snd_hda_calc_stream_format - calculate format bitset
+ * @rate: the sample rate
+ * @channels: the number of channels
+ * @format: the PCM format (SNDRV_PCM_FORMAT_XXX)
+ * @maxbps: the max. bps
+ *
+ * Calculate the format bitset from the given rate, channels and th PCM format.
+ *
+ * Return zero if invalid.
+ */
+unsigned int snd_hda_calc_stream_format(unsigned int rate,
+					unsigned int channels,
+					unsigned int format,
+					unsigned int maxbps)
+{
+	int i;
+	unsigned int val = 0;
+
+	for (i = 0; rate_bits[i][0]; i++)
+		if (rate_bits[i][0] == rate) {
+			val = rate_bits[i][2];
+			break;
+		}
+	if (! rate_bits[i][0]) {
+		snd_printdd("invalid rate %d\n", rate);
+		return 0;
+	}
+
+	if (channels == 0 || channels > 8) {
+		snd_printdd("invalid channels %d\n", channels);
+		return 0;
+	}
+	val |= channels - 1;
+
+	switch (snd_pcm_format_width(format)) {
+	case 8:  val |= 0x00; break;
+	case 16: val |= 0x10; break;
+	case 20:
+	case 24:
+	case 32:
+		if (maxbps >= 32)
+			val |= 0x40;
+		else if (maxbps >= 24)
+			val |= 0x30;
+		else
+			val |= 0x20;
+		break;
+	default:
+		snd_printdd("invalid format width %d\n", snd_pcm_format_width(format));
+		return 0;
+	}
+
+	return val;
+}
+
+/**
+ * snd_hda_query_supported_pcm - query the supported PCM rates and formats
+ * @codec: the HDA codec
+ * @nid: NID to query
+ * @ratesp: the pointer to store the detected rate bitflags
+ * @formatsp: the pointer to store the detected formats
+ * @bpsp: the pointer to store the detected format widths
+ *
+ * Queries the supported PCM rates and formats.  The NULL @ratesp, @formatsp
+ * or @bsps argument is ignored.
+ *
+ * Returns 0 if successful, otherwise a negative error code.
+ */
+int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
+				u32 *ratesp, u64 *formatsp, unsigned int *bpsp)
+{
+	int i;
+	unsigned int val, streams;
+
+	val = 0;
+	if (nid != codec->afg &&
+	    snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP) & AC_WCAP_FORMAT_OVRD) {
+		val = snd_hda_param_read(codec, nid, AC_PAR_PCM);
+		if (val == -1)
+			return -EIO;
+	}
+	if (! val)
+		val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM);
+
+	if (ratesp) {
+		u32 rates = 0;
+		for (i = 0; rate_bits[i][0]; i++) {
+			if (val & (1 << i))
+				rates |= rate_bits[i][1];
+		}
+		*ratesp = rates;
+	}
+
+	if (formatsp || bpsp) {
+		u64 formats = 0;
+		unsigned int bps;
+		unsigned int wcaps;
+
+		wcaps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP);
+		streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM);
+		if (streams == -1)
+			return -EIO;
+		if (! streams) {
+			streams = snd_hda_param_read(codec, codec->afg, AC_PAR_STREAM);
+			if (streams == -1)
+				return -EIO;
+		}
+
+		bps = 0;
+		if (streams & AC_SUPFMT_PCM) {
+			if (val & AC_SUPPCM_BITS_8) {
+				formats |= SNDRV_PCM_FMTBIT_U8;
+				bps = 8;
+			}
+			if (val & AC_SUPPCM_BITS_16) {
+				formats |= SNDRV_PCM_FMTBIT_S16_LE;
+				bps = 16;
+			}
+			if (wcaps & AC_WCAP_DIGITAL) {
+				if (val & AC_SUPPCM_BITS_32)
+					formats |= SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE;
+				if (val & (AC_SUPPCM_BITS_20|AC_SUPPCM_BITS_24))
+					formats |= SNDRV_PCM_FMTBIT_S32_LE;
+				if (val & AC_SUPPCM_BITS_24)
+					bps = 24;
+				else if (val & AC_SUPPCM_BITS_20)
+					bps = 20;
+			} else if (val & (AC_SUPPCM_BITS_20|AC_SUPPCM_BITS_24|AC_SUPPCM_BITS_32)) {
+				formats |= SNDRV_PCM_FMTBIT_S32_LE;
+				if (val & AC_SUPPCM_BITS_32)
+					bps = 32;
+				else if (val & AC_SUPPCM_BITS_20)
+					bps = 20;
+				else if (val & AC_SUPPCM_BITS_24)
+					bps = 24;
+			}
+		}
+		else if (streams == AC_SUPFMT_FLOAT32) { /* should be exclusive */
+			formats |= SNDRV_PCM_FMTBIT_FLOAT_LE;
+			bps = 32;
+		} else if (streams == AC_SUPFMT_AC3) { /* should be exclusive */
+			/* temporary hack: we have still no proper support
+			 * for the direct AC3 stream...
+			 */
+			formats |= SNDRV_PCM_FMTBIT_U8;
+			bps = 8;
+		}
+		if (formatsp)
+			*formatsp = formats;
+		if (bpsp)
+			*bpsp = bps;
+	}
+
+	return 0;
+}
+
+/**
+ * snd_hda_is_supported_format - check whether the given node supports the format val
+ *
+ * Returns 1 if supported, 0 if not.
+ */
+int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid,
+				unsigned int format)
+{
+	int i;
+	unsigned int val = 0, rate, stream;
+
+	if (nid != codec->afg &&
+	    snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP) & AC_WCAP_FORMAT_OVRD) {
+		val = snd_hda_param_read(codec, nid, AC_PAR_PCM);
+		if (val == -1)
+			return 0;
+	}
+	if (! val) {
+		val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM);
+		if (val == -1)
+			return 0;
+	}
+
+	rate = format & 0xff00;
+	for (i = 0; rate_bits[i][0]; i++)
+		if (rate_bits[i][2] == rate) {
+			if (val & (1 << i))
+				break;
+			return 0;
+		}
+	if (! rate_bits[i][0])
+		return 0;
+
+	stream = snd_hda_param_read(codec, nid, AC_PAR_STREAM);
+	if (stream == -1)
+		return 0;
+	if (! stream && nid != codec->afg)
+		stream = snd_hda_param_read(codec, codec->afg, AC_PAR_STREAM);
+	if (! stream || stream == -1)
+		return 0;
+
+	if (stream & AC_SUPFMT_PCM) {
+		switch (format & 0xf0) {
+		case 0x00:
+			if (! (val & AC_SUPPCM_BITS_8))
+				return 0;
+			break;
+		case 0x10:
+			if (! (val & AC_SUPPCM_BITS_16))
+				return 0;
+			break;
+		case 0x20:
+			if (! (val & AC_SUPPCM_BITS_20))
+				return 0;
+			break;
+		case 0x30:
+			if (! (val & AC_SUPPCM_BITS_24))
+				return 0;
+			break;
+		case 0x40:
+			if (! (val & AC_SUPPCM_BITS_32))
+				return 0;
+			break;
+		default:
+			return 0;
+		}
+	} else {
+		/* FIXME: check for float32 and AC3? */
+	}
+
+	return 1;
+}
+
+/*
+ * PCM stuff
+ */
+static int hda_pcm_default_open_close(struct hda_pcm_stream *hinfo,
+				      struct hda_codec *codec,
+				      snd_pcm_substream_t *substream)
+{
+	return 0;
+}
+
+static int hda_pcm_default_prepare(struct hda_pcm_stream *hinfo,
+				   struct hda_codec *codec,
+				   unsigned int stream_tag,
+				   unsigned int format,
+				   snd_pcm_substream_t *substream)
+{
+	snd_hda_codec_setup_stream(codec, hinfo->nid, stream_tag, 0, format);
+	return 0;
+}
+
+static int hda_pcm_default_cleanup(struct hda_pcm_stream *hinfo,
+				   struct hda_codec *codec,
+				   snd_pcm_substream_t *substream)
+{
+	snd_hda_codec_setup_stream(codec, hinfo->nid, 0, 0, 0);
+	return 0;
+}
+
+static int set_pcm_default_values(struct hda_codec *codec, struct hda_pcm_stream *info)
+{
+	if (info->nid) {
+		/* query support PCM information from the given NID */
+		if (! info->rates || ! info->formats)
+			snd_hda_query_supported_pcm(codec, info->nid,
+						    info->rates ? NULL : &info->rates,
+						    info->formats ? NULL : &info->formats,
+						    info->maxbps ? NULL : &info->maxbps);
+	}
+	if (info->ops.open == NULL)
+		info->ops.open = hda_pcm_default_open_close;
+	if (info->ops.close == NULL)
+		info->ops.close = hda_pcm_default_open_close;
+	if (info->ops.prepare == NULL) {
+		snd_assert(info->nid, return -EINVAL);
+		info->ops.prepare = hda_pcm_default_prepare;
+	}
+	if (info->ops.prepare == NULL) {
+		snd_assert(info->nid, return -EINVAL);
+		info->ops.prepare = hda_pcm_default_prepare;
+	}
+	if (info->ops.cleanup == NULL) {
+		snd_assert(info->nid, return -EINVAL);
+		info->ops.cleanup = hda_pcm_default_cleanup;
+	}
+	return 0;
+}
+
+/**
+ * snd_hda_build_pcms - build PCM information
+ * @bus: the BUS
+ *
+ * Create PCM information for each codec included in the bus.
+ *
+ * The build_pcms codec patch is requested to set up codec->num_pcms and
+ * codec->pcm_info properly.  The array is referred by the top-level driver
+ * to create its PCM instances.
+ * The allocated codec->pcm_info should be released in codec->patch_ops.free
+ * callback.
+ *
+ * At least, substreams, channels_min and channels_max must be filled for
+ * each stream.  substreams = 0 indicates that the stream doesn't exist.
+ * When rates and/or formats are zero, the supported values are queried
+ * from the given nid.  The nid is used also by the default ops.prepare
+ * and ops.cleanup callbacks.
+ *
+ * The driver needs to call ops.open in its open callback.  Similarly,
+ * ops.close is supposed to be called in the close callback.
+ * ops.prepare should be called in the prepare or hw_params callback
+ * with the proper parameters for set up.
+ * ops.cleanup should be called in hw_free for clean up of streams.
+ *
+ * This function returns 0 if successfull, or a negative error code.
+ */
+int snd_hda_build_pcms(struct hda_bus *bus)
+{
+	struct list_head *p;
+
+	list_for_each(p, &bus->codec_list) {
+		struct hda_codec *codec = list_entry(p, struct hda_codec, list);
+		unsigned int pcm, s;
+		int err;
+		if (! codec->patch_ops.build_pcms)
+			continue;
+		err = codec->patch_ops.build_pcms(codec);
+		if (err < 0)
+			return err;
+		for (pcm = 0; pcm < codec->num_pcms; pcm++) {
+			for (s = 0; s < 2; s++) {
+				struct hda_pcm_stream *info;
+				info = &codec->pcm_info[pcm].stream[s];
+				if (! info->substreams)
+					continue;
+				err = set_pcm_default_values(codec, info);
+				if (err < 0)
+					return err;
+			}
+		}
+	}
+	return 0;
+}
+
+
+/**
+ * snd_hda_check_board_config - compare the current codec with the config table
+ * @codec: the HDA codec
+ * @tbl: configuration table, terminated by null entries
+ *
+ * Compares the modelname or PCI subsystem id of the current codec with the
+ * given configuration table.  If a matching entry is found, returns its
+ * config value (supposed to be 0 or positive).
+ *
+ * If no entries are matching, the function returns a negative value.
+ */
+int snd_hda_check_board_config(struct hda_codec *codec, struct hda_board_config *tbl)
+{
+	struct hda_board_config *c;
+
+	if (codec->bus->modelname) {
+		for (c = tbl; c->modelname || c->pci_vendor; c++) {
+			if (c->modelname &&
+			    ! strcmp(codec->bus->modelname, c->modelname)) {
+				snd_printd(KERN_INFO "hda_codec: model '%s' is selected\n", c->modelname);
+				return c->config;
+			}
+		}
+	}
+
+	if (codec->bus->pci) {
+		u16 subsystem_vendor, subsystem_device;
+		pci_read_config_word(codec->bus->pci, PCI_SUBSYSTEM_VENDOR_ID, &subsystem_vendor);
+		pci_read_config_word(codec->bus->pci, PCI_SUBSYSTEM_ID, &subsystem_device);
+		for (c = tbl; c->modelname || c->pci_vendor; c++) {
+			if (c->pci_vendor == subsystem_vendor &&
+			    c->pci_device == subsystem_device)
+				return c->config;
+		}
+	}
+	return -1;
+}
+
+/**
+ * snd_hda_add_new_ctls - create controls from the array
+ * @codec: the HDA codec
+ * @knew: the array of snd_kcontrol_new_t
+ *
+ * This helper function creates and add new controls in the given array.
+ * The array must be terminated with an empty entry as terminator.
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_hda_add_new_ctls(struct hda_codec *codec, snd_kcontrol_new_t *knew)
+{
+	int err;
+
+	for (; knew->name; knew++) {
+		err = snd_ctl_add(codec->bus->card, snd_ctl_new1(knew, codec));
+		if (err < 0)
+			return err;
+	}
+	return 0;
+}
+
+
+/*
+ * input MUX helper
+ */
+int snd_hda_input_mux_info(const struct hda_input_mux *imux, snd_ctl_elem_info_t *uinfo)
+{
+	unsigned int index;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = imux->num_items;
+	index = uinfo->value.enumerated.item;
+	if (index >= imux->num_items)
+		index = imux->num_items - 1;
+	strcpy(uinfo->value.enumerated.name, imux->items[index].label);
+	return 0;
+}
+
+int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *imux,
+			  snd_ctl_elem_value_t *ucontrol, hda_nid_t nid,
+			  unsigned int *cur_val)
+{
+	unsigned int idx;
+
+	idx = ucontrol->value.enumerated.item[0];
+	if (idx >= imux->num_items)
+		idx = imux->num_items - 1;
+	if (*cur_val == idx && ! codec->in_resume)
+		return 0;
+	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL,
+			    imux->items[idx].index);
+	*cur_val = idx;
+	return 1;
+}
+
+
+/*
+ * Multi-channel / digital-out PCM helper functions
+ */
+
+/*
+ * open the digital out in the exclusive mode
+ */
+int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mout)
+{
+	down(&codec->spdif_mutex);
+	if (mout->dig_out_used) {
+		up(&codec->spdif_mutex);
+		return -EBUSY; /* already being used */
+	}
+	mout->dig_out_used = HDA_DIG_EXCLUSIVE;
+	up(&codec->spdif_mutex);
+	return 0;
+}
+
+/*
+ * release the digital out
+ */
+int snd_hda_multi_out_dig_close(struct hda_codec *codec, struct hda_multi_out *mout)
+{
+	down(&codec->spdif_mutex);
+	mout->dig_out_used = 0;
+	up(&codec->spdif_mutex);
+	return 0;
+}
+
+/*
+ * set up more restrictions for analog out
+ */
+int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout,
+				  snd_pcm_substream_t *substream)
+{
+	substream->runtime->hw.channels_max = mout->max_channels;
+	return snd_pcm_hw_constraint_step(substream->runtime, 0,
+					  SNDRV_PCM_HW_PARAM_CHANNELS, 2);
+}
+
+/*
+ * set up the i/o for analog out
+ * when the digital out is available, copy the front out to digital out, too.
+ */
+int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_out *mout,
+				     unsigned int stream_tag,
+				     unsigned int format,
+				     snd_pcm_substream_t *substream)
+{
+	hda_nid_t *nids = mout->dac_nids;
+	int chs = substream->runtime->channels;
+	int i;
+
+	down(&codec->spdif_mutex);
+	if (mout->dig_out_nid && mout->dig_out_used != HDA_DIG_EXCLUSIVE) {
+		if (chs == 2 &&
+		    snd_hda_is_supported_format(codec, mout->dig_out_nid, format) &&
+		    ! (codec->spdif_status & IEC958_AES0_NONAUDIO)) {
+			mout->dig_out_used = HDA_DIG_ANALOG_DUP;
+			/* setup digital receiver */
+			snd_hda_codec_setup_stream(codec, mout->dig_out_nid,
+						   stream_tag, 0, format);
+		} else {
+			mout->dig_out_used = 0;
+			snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0);
+		}
+	}
+	up(&codec->spdif_mutex);
+
+	/* front */
+	snd_hda_codec_setup_stream(codec, nids[HDA_FRONT], stream_tag, 0, format);
+	if (mout->hp_nid)
+		/* headphone out will just decode front left/right (stereo) */
+		snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format);
+	/* surrounds */
+	for (i = 1; i < mout->num_dacs; i++) {
+		if (i == HDA_REAR && chs == 2) /* copy front to rear */
+			snd_hda_codec_setup_stream(codec, nids[i], stream_tag, 0, format);
+		else if (chs >= (i + 1) * 2) /* independent out */
+			snd_hda_codec_setup_stream(codec, nids[i], stream_tag, i * 2,
+						   format);
+	}
+	return 0;
+}
+
+/*
+ * clean up the setting for analog out
+ */
+int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, struct hda_multi_out *mout)
+{
+	hda_nid_t *nids = mout->dac_nids;
+	int i;
+
+	for (i = 0; i < mout->num_dacs; i++)
+		snd_hda_codec_setup_stream(codec, nids[i], 0, 0, 0);
+	if (mout->hp_nid)
+		snd_hda_codec_setup_stream(codec, mout->hp_nid, 0, 0, 0);
+	down(&codec->spdif_mutex);
+	if (mout->dig_out_nid && mout->dig_out_used == HDA_DIG_ANALOG_DUP) {
+		snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0);
+		mout->dig_out_used = 0;
+	}
+	up(&codec->spdif_mutex);
+	return 0;
+}
+
+#ifdef CONFIG_PM
+/*
+ * power management
+ */
+
+/**
+ * snd_hda_suspend - suspend the codecs
+ * @bus: the HDA bus
+ * @state: suspsend state
+ *
+ * Returns 0 if successful.
+ */
+int snd_hda_suspend(struct hda_bus *bus, pm_message_t state)
+{
+	struct list_head *p;
+
+	/* FIXME: should handle power widget capabilities */
+	list_for_each(p, &bus->codec_list) {
+		struct hda_codec *codec = list_entry(p, struct hda_codec, list);
+		if (codec->patch_ops.suspend)
+			codec->patch_ops.suspend(codec, state);
+	}
+	return 0;
+}
+
+/**
+ * snd_hda_resume - resume the codecs
+ * @bus: the HDA bus
+ * @state: resume state
+ *
+ * Returns 0 if successful.
+ */
+int snd_hda_resume(struct hda_bus *bus)
+{
+	struct list_head *p;
+
+	list_for_each(p, &bus->codec_list) {
+		struct hda_codec *codec = list_entry(p, struct hda_codec, list);
+		if (codec->patch_ops.resume)
+			codec->patch_ops.resume(codec);
+	}
+	return 0;
+}
+
+/**
+ * snd_hda_resume_ctls - resume controls in the new control list
+ * @codec: the HDA codec
+ * @knew: the array of snd_kcontrol_new_t
+ *
+ * This function resumes the mixer controls in the snd_kcontrol_new_t array,
+ * originally for snd_hda_add_new_ctls().
+ * The array must be terminated with an empty entry as terminator.
+ */
+int snd_hda_resume_ctls(struct hda_codec *codec, snd_kcontrol_new_t *knew)
+{
+	snd_ctl_elem_value_t *val;
+
+	val = kmalloc(sizeof(*val), GFP_KERNEL);
+	if (! val)
+		return -ENOMEM;
+	codec->in_resume = 1;
+	for (; knew->name; knew++) {
+		int i, count;
+		count = knew->count ? knew->count : 1;
+		for (i = 0; i < count; i++) {
+			memset(val, 0, sizeof(*val));
+			val->id.iface = knew->iface;
+			val->id.device = knew->device;
+			val->id.subdevice = knew->subdevice;
+			strcpy(val->id.name, knew->name);
+			val->id.index = knew->index ? knew->index : i;
+			/* Assume that get callback reads only from cache,
+			 * not accessing to the real hardware
+			 */
+			if (snd_ctl_elem_read(codec->bus->card, val) < 0)
+				continue;
+			snd_ctl_elem_write(codec->bus->card, NULL, val);
+		}
+	}
+	codec->in_resume = 0;
+	kfree(val);
+	return 0;
+}
+
+/**
+ * snd_hda_resume_spdif_out - resume the digital out
+ * @codec: the HDA codec
+ */
+int snd_hda_resume_spdif_out(struct hda_codec *codec)
+{
+	return snd_hda_resume_ctls(codec, dig_mixes);
+}
+
+/**
+ * snd_hda_resume_spdif_in - resume the digital in
+ * @codec: the HDA codec
+ */
+int snd_hda_resume_spdif_in(struct hda_codec *codec)
+{
+	return snd_hda_resume_ctls(codec, dig_in_ctls);
+}
+#endif
+
+/*
+ * symbols exported for controller modules
+ */
+EXPORT_SYMBOL(snd_hda_codec_read);
+EXPORT_SYMBOL(snd_hda_codec_write);
+EXPORT_SYMBOL(snd_hda_sequence_write);
+EXPORT_SYMBOL(snd_hda_get_sub_nodes);
+EXPORT_SYMBOL(snd_hda_queue_unsol_event);
+EXPORT_SYMBOL(snd_hda_bus_new);
+EXPORT_SYMBOL(snd_hda_codec_new);
+EXPORT_SYMBOL(snd_hda_codec_setup_stream);
+EXPORT_SYMBOL(snd_hda_calc_stream_format);
+EXPORT_SYMBOL(snd_hda_build_pcms);
+EXPORT_SYMBOL(snd_hda_build_controls);
+#ifdef CONFIG_PM
+EXPORT_SYMBOL(snd_hda_suspend);
+EXPORT_SYMBOL(snd_hda_resume);
+#endif
+
+/*
+ *  INIT part
+ */
+
+static int __init alsa_hda_init(void)
+{
+	return 0;
+}
+
+static void __exit alsa_hda_exit(void)
+{
+}
+
+module_init(alsa_hda_init)
+module_exit(alsa_hda_exit)
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
new file mode 100644
index 000000000000..c9e9dc9c7c98
--- /dev/null
+++ b/sound/pci/hda/hda_codec.h
@@ -0,0 +1,604 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
+ *
+ *  This program is free software; you can redistribute it and/or modify it
+ *  under the terms of the GNU General Public License as published by the Free
+ *  Software Foundation; either version 2 of the License, or (at your option)
+ *  any later version.
+ *
+ *  This program is distributed in the hope that it will be useful, but WITHOUT
+ *  ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
+ *  FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License for
+ *  more details.
+ *
+ *  You should have received a copy of the GNU General Public License along with
+ *  this program; if not, write to the Free Software Foundation, Inc., 59
+ *  Temple Place - Suite 330, Boston, MA  02111-1307, USA.
+ */
+
+#ifndef __SOUND_HDA_CODEC_H
+#define __SOUND_HDA_CODEC_H
+
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+
+/*
+ * nodes
+ */
+#define	AC_NODE_ROOT		0x00
+
+/*
+ * function group types
+ */
+enum {
+	AC_GRP_AUDIO_FUNCTION = 0x01,
+	AC_GRP_MODEM_FUNCTION = 0x02,
+};
+	
+/*
+ * widget types
+ */
+enum {
+	AC_WID_AUD_OUT,		/* Audio Out */
+	AC_WID_AUD_IN,		/* Audio In */
+	AC_WID_AUD_MIX,		/* Audio Mixer */
+	AC_WID_AUD_SEL,		/* Audio Selector */
+	AC_WID_PIN,		/* Pin Complex */
+	AC_WID_POWER,		/* Power */
+	AC_WID_VOL_KNB,		/* Volume Knob */
+	AC_WID_BEEP,		/* Beep Generator */
+	AC_WID_VENDOR = 0x0f	/* Vendor specific */
+};
+
+/*
+ * GET verbs
+ */
+#define AC_VERB_GET_STREAM_FORMAT		0x0a00
+#define AC_VERB_GET_AMP_GAIN_MUTE		0x0b00
+#define AC_VERB_GET_PROC_COEF			0x0c00
+#define AC_VERB_GET_COEF_INDEX			0x0d00
+#define AC_VERB_PARAMETERS			0x0f00
+#define AC_VERB_GET_CONNECT_SEL			0x0f01
+#define AC_VERB_GET_CONNECT_LIST		0x0f02
+#define AC_VERB_GET_PROC_STATE			0x0f03
+#define AC_VERB_GET_SDI_SELECT			0x0f04
+#define AC_VERB_GET_POWER_STATE			0x0f05
+#define AC_VERB_GET_CONV			0x0f06
+#define AC_VERB_GET_PIN_WIDGET_CONTROL		0x0f07
+#define AC_VERB_GET_UNSOLICITED_RESPONSE	0x0f08
+#define AC_VERB_GET_PIN_SENSE			0x0f09
+#define AC_VERB_GET_BEEP_CONTROL		0x0f0a
+#define AC_VERB_GET_EAPD_BTLENABLE		0x0f0c
+#define AC_VERB_GET_DIGI_CONVERT		0x0f0d
+#define AC_VERB_GET_VOLUME_KNOB_CONTROL		0x0f0f
+/* f10-f1a: GPIO */
+#define AC_VERB_GET_CONFIG_DEFAULT		0x0f1c
+
+/*
+ * SET verbs
+ */
+#define AC_VERB_SET_STREAM_FORMAT		0x200
+#define AC_VERB_SET_AMP_GAIN_MUTE		0x300
+#define AC_VERB_SET_PROC_COEF			0x400
+#define AC_VERB_SET_COEF_INDEX			0x500
+#define AC_VERB_SET_CONNECT_SEL			0x701
+#define AC_VERB_SET_PROC_STATE			0x703
+#define AC_VERB_SET_SDI_SELECT			0x704
+#define AC_VERB_SET_POWER_STATE			0x705
+#define AC_VERB_SET_CHANNEL_STREAMID		0x706
+#define AC_VERB_SET_PIN_WIDGET_CONTROL		0x707
+#define AC_VERB_SET_UNSOLICITED_ENABLE		0x708
+#define AC_VERB_SET_PIN_SENSE			0x709
+#define AC_VERB_SET_BEEP_CONTROL		0x70a
+#define AC_VERB_SET_EAPD_BTLENALBE		0x70c
+#define AC_VERB_SET_DIGI_CONVERT_1		0x70d
+#define AC_VERB_SET_DIGI_CONVERT_2		0x70e
+#define AC_VERB_SET_VOLUME_KNOB_CONTROL		0x70f
+#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0	0x71c
+#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1	0x71d
+#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2	0x71e
+#define AC_VERB_SET_CONFIG_DEFAULT_BYTES_3	0x71f
+#define AC_VERB_SET_CODEC_RESET			0x7ff
+
+/*
+ * Parameter IDs
+ */
+#define AC_PAR_VENDOR_ID		0x00
+#define AC_PAR_SUBSYSTEM_ID		0x01
+#define AC_PAR_REV_ID			0x02
+#define AC_PAR_NODE_COUNT		0x04
+#define AC_PAR_FUNCTION_TYPE		0x05
+#define AC_PAR_AUDIO_FG_CAP		0x08
+#define AC_PAR_AUDIO_WIDGET_CAP		0x09
+#define AC_PAR_PCM			0x0a
+#define AC_PAR_STREAM			0x0b
+#define AC_PAR_PIN_CAP			0x0c
+#define AC_PAR_AMP_IN_CAP		0x0d
+#define AC_PAR_CONNLIST_LEN		0x0e
+#define AC_PAR_POWER_STATE		0x0f
+#define AC_PAR_PROC_CAP			0x10
+#define AC_PAR_GPIO_CAP			0x11
+#define AC_PAR_AMP_OUT_CAP		0x12
+
+/*
+ * AC_VERB_PARAMETERS results (32bit)
+ */
+
+/* Function Group Type */
+#define AC_FGT_TYPE			(0xff<<0)
+#define AC_FGT_TYPE_SHIFT		0
+#define AC_FGT_UNSOL_CAP		(1<<8)
+
+/* Audio Function Group Capabilities */
+#define AC_AFG_OUT_DELAY		(0xf<<0)
+#define AC_AFG_IN_DELAY			(0xf<<8)
+#define AC_AFG_BEEP_GEN			(1<<16)
+
+/* Audio Widget Capabilities */
+#define AC_WCAP_STEREO			(1<<0)	/* stereo I/O */
+#define AC_WCAP_IN_AMP			(1<<1)	/* AMP-in present */
+#define AC_WCAP_OUT_AMP			(1<<2)	/* AMP-out present */
+#define AC_WCAP_AMP_OVRD		(1<<3)	/* AMP-parameter override */
+#define AC_WCAP_FORMAT_OVRD		(1<<4)	/* format override */
+#define AC_WCAP_STRIPE			(1<<5)	/* stripe */
+#define AC_WCAP_PROC_WID		(1<<6)	/* Proc Widget */
+#define AC_WCAP_UNSOL_CAP		(1<<7)	/* Unsol capable */
+#define AC_WCAP_CONN_LIST		(1<<8)	/* connection list */
+#define AC_WCAP_DIGITAL			(1<<9)	/* digital I/O */
+#define AC_WCAP_POWER			(1<<10)	/* power control */
+#define AC_WCAP_LR_SWAP			(1<<11)	/* L/R swap */
+#define AC_WCAP_DELAY			(0xf<<16)
+#define AC_WCAP_DELAY_SHIFT		16
+#define AC_WCAP_TYPE			(0xf<<20)
+#define AC_WCAP_TYPE_SHIFT		20
+
+/* supported PCM rates and bits */
+#define AC_SUPPCM_RATES			(0xfff << 0)
+#define AC_SUPPCM_BITS_8		(1<<16)
+#define AC_SUPPCM_BITS_16		(1<<17)
+#define AC_SUPPCM_BITS_20		(1<<18)
+#define AC_SUPPCM_BITS_24		(1<<19)
+#define AC_SUPPCM_BITS_32		(1<<20)
+
+/* supported PCM stream format */
+#define AC_SUPFMT_PCM			(1<<0)
+#define AC_SUPFMT_FLOAT32		(1<<1)
+#define AC_SUPFMT_AC3			(1<<2)
+
+/* Pin widget capabilies */
+#define AC_PINCAP_IMP_SENSE		(1<<0)	/* impedance sense capable */
+#define AC_PINCAP_TRIG_REQ		(1<<1)	/* trigger required */
+#define AC_PINCAP_PRES_DETECT		(1<<2)	/* presence detect capable */
+#define AC_PINCAP_HP_DRV		(1<<3)	/* headphone drive capable */
+#define AC_PINCAP_OUT			(1<<4)	/* output capable */
+#define AC_PINCAP_IN			(1<<5)	/* input capable */
+#define AC_PINCAP_BALANCE		(1<<6)	/* balanced I/O capable */
+#define AC_PINCAP_VREF			(7<<8)
+#define AC_PINCAP_VREF_SHIFT		8
+#define AC_PINCAP_EAPD			(1<<16)	/* EAPD capable */
+/* Vref status (used in pin cap and pin ctl) */
+#define AC_PIN_VREF_HIZ			(1<<0)	/* Hi-Z */
+#define AC_PIN_VREF_50			(1<<1)	/* 50% */
+#define AC_PIN_VREF_GRD			(1<<2)	/* ground */
+#define AC_PIN_VREF_80			(1<<4)	/* 80% */
+#define AC_PIN_VREF_100			(1<<5)	/* 100% */
+
+
+/* Amplifier capabilities */
+#define AC_AMPCAP_OFFSET		(0x7f<<0)  /* 0dB offset */
+#define AC_AMPCAP_OFFSET_SHIFT		0
+#define AC_AMPCAP_NUM_STEPS		(0x7f<<8)  /* number of steps */
+#define AC_AMPCAP_NUM_STEPS_SHIFT	8
+#define AC_AMPCAP_STEP_SIZE		(0x7f<<16) /* step size 0-32dB in 0.25dB */
+#define AC_AMPCAP_STEP_SIZE_SHIFT	16
+#define AC_AMPCAP_MUTE			(1<<31)    /* mute capable */
+#define AC_AMPCAP_MUTE_SHIFT		31
+
+/* Connection list */
+#define AC_CLIST_LENGTH			(0x7f<<0)
+#define AC_CLIST_LONG			(1<<7)
+
+/* Supported power status */
+#define AC_PWRST_D0SUP			(1<<0)
+#define AC_PWRST_D1SUP			(1<<1)
+#define AC_PWRST_D2SUP			(1<<2)
+#define AC_PWRST_D3SUP			(1<<3)
+
+/* Processing capabilies */
+#define AC_PCAP_BENIGN			(1<<0)
+#define AC_PCAP_NUM_COEF		(0xff<<8)
+
+/* Volume knobs capabilities */
+#define AC_KNBCAP_NUM_STEPS		(0x7f<<0)
+#define AC_KNBCAP_DELTA			(1<<8)
+
+/*
+ * Control Parameters
+ */
+
+/* Amp gain/mute */
+#define AC_AMP_MUTE			(1<<8)
+#define AC_AMP_GAIN			(0x7f)
+#define AC_AMP_GET_INDEX		(0xf<<0)
+
+#define AC_AMP_GET_LEFT			(1<<13)
+#define AC_AMP_GET_RIGHT		(0<<13)
+#define AC_AMP_GET_OUTPUT		(1<<15)
+#define AC_AMP_GET_INPUT		(0<<15)
+
+#define AC_AMP_SET_INDEX		(0xf<<8)
+#define AC_AMP_SET_INDEX_SHIFT		8
+#define AC_AMP_SET_RIGHT		(1<<12)
+#define AC_AMP_SET_LEFT			(1<<13)
+#define AC_AMP_SET_INPUT		(1<<14)
+#define AC_AMP_SET_OUTPUT		(1<<15)
+
+/* DIGITAL1 bits */
+#define AC_DIG1_ENABLE			(1<<0)
+#define AC_DIG1_V			(1<<1)
+#define AC_DIG1_VCFG			(1<<2)
+#define AC_DIG1_EMPHASIS		(1<<3)
+#define AC_DIG1_COPYRIGHT		(1<<4)
+#define AC_DIG1_NONAUDIO		(1<<5)
+#define AC_DIG1_PROFESSIONAL		(1<<6)
+#define AC_DIG1_LEVEL			(1<<7)
+
+/* Pin widget control - 8bit */
+#define AC_PINCTL_VREFEN		(0x7<<0)
+#define AC_PINCTL_IN_EN			(1<<5)
+#define AC_PINCTL_OUT_EN		(1<<6)
+#define AC_PINCTL_HP_EN			(1<<7)
+
+/* configuration default - 32bit */
+#define AC_DEFCFG_SEQUENCE		(0xf<<0)
+#define AC_DEFCFG_DEF_ASSOC		(0xf<<4)
+#define AC_DEFCFG_MISC			(0xf<<8)
+#define AC_DEFCFG_COLOR			(0xf<<12)
+#define AC_DEFCFG_COLOR_SHIFT		12
+#define AC_DEFCFG_CONN_TYPE		(0xf<<16)
+#define AC_DEFCFG_CONN_TYPE_SHIFT	16
+#define AC_DEFCFG_DEVICE		(0xf<<20)
+#define AC_DEFCFG_DEVICE_SHIFT		20
+#define AC_DEFCFG_LOCATION		(0x3f<<24)
+#define AC_DEFCFG_LOCATION_SHIFT	24
+#define AC_DEFCFG_PORT_CONN		(0x3<<30)
+#define AC_DEFCFG_PORT_CONN_SHIFT	30
+
+/* device device types (0x0-0xf) */
+enum {
+	AC_JACK_LINE_OUT,
+	AC_JACK_SPEAKER,
+	AC_JACK_HP_OUT,
+	AC_JACK_CD,
+	AC_JACK_SPDIF_OUT,
+	AC_JACK_DIG_OTHER_OUT,
+	AC_JACK_MODEM_LINE_SIDE,
+	AC_JACK_MODEM_HAND_SIDE,
+	AC_JACK_LINE_IN,
+	AC_JACK_AUX,
+	AC_JACK_MIC_IN,
+	AC_JACK_TELEPHONY,
+	AC_JACK_SPDIF_IN,
+	AC_JACK_DIG_OTHER_IN,
+	AC_JACK_OTHER = 0xf,
+};
+
+/* jack connection types (0x0-0xf) */
+enum {
+	AC_JACK_CONN_UNKNOWN,
+	AC_JACK_CONN_1_8,
+	AC_JACK_CONN_1_4,
+	AC_JACK_CONN_ATAPI,
+	AC_JACK_CONN_RCA,
+	AC_JACK_CONN_OPTICAL,
+	AC_JACK_CONN_OTHER_DIGITAL,
+	AC_JACK_CONN_OTHER_ANALOG,
+	AC_JACK_CONN_DIN,
+	AC_JACK_CONN_XLR,
+	AC_JACK_CONN_RJ11,
+	AC_JACK_CONN_COMB,
+	AC_JACK_CONN_OTHER = 0xf,
+};
+
+/* jack colors (0x0-0xf) */
+enum {
+	AC_JACK_COLOR_UNKNOWN,
+	AC_JACK_COLOR_BLACK,
+	AC_JACK_COLOR_GREY,
+	AC_JACK_COLOR_BLUE,
+	AC_JACK_COLOR_GREEN,
+	AC_JACK_COLOR_RED,
+	AC_JACK_COLOR_ORANGE,
+	AC_JACK_COLOR_YELLOW,
+	AC_JACK_COLOR_PURPLE,
+	AC_JACK_COLOR_PINK,
+	AC_JACK_COLOR_WHITE = 0xe,
+	AC_JACK_COLOR_OTHER,
+};
+
+/* Jack location (0x0-0x3f) */
+/* common case */
+enum {
+	AC_JACK_LOC_NONE,
+	AC_JACK_LOC_REAR,
+	AC_JACK_LOC_FRONT,
+	AC_JACK_LOC_LEFT,
+	AC_JACK_LOC_RIGHT,
+	AC_JACK_LOC_TOP,
+	AC_JACK_LOC_BOTTOM,
+};
+/* bits 4-5 */
+enum {
+	AC_JACK_LOC_EXTERNAL = 0x00,
+	AC_JACK_LOC_INTERNAL = 0x10,
+	AC_JACK_LOC_SEPARATE = 0x20,
+	AC_JACK_LOC_OTHER    = 0x30,
+};
+enum {
+	/* external on primary chasis */
+	AC_JACK_LOC_REAR_PANEL = 0x07,
+	AC_JACK_LOC_DRIVE_BAY,
+	/* internal */
+	AC_JACK_LOC_RISER = 0x17,
+	AC_JACK_LOC_HDMI,
+	AC_JACK_LOC_ATAPI,
+	/* others */
+	AC_JACK_LOC_MOBILE_IN = 0x37,
+	AC_JACK_LOC_MOBILE_OUT,
+};
+
+/* Port connectivity (0-3) */
+enum {
+	AC_JACK_PORT_COMPLEX,
+	AC_JACK_PORT_NONE,
+	AC_JACK_PORT_FIXED,
+	AC_JACK_PORT_BOTH,
+};
+
+/* max. connections to a widget */
+#define HDA_MAX_CONNECTIONS	16
+
+/* max. codec address */
+#define HDA_MAX_CODEC_ADDRESS	0x0f
+
+/*
+ * Structures
+ */
+
+struct hda_bus;
+struct hda_codec;
+struct hda_pcm;
+struct hda_pcm_stream;
+struct hda_bus_unsolicited;
+
+/* NID type */
+typedef u16 hda_nid_t;
+
+/* bus operators */
+struct hda_bus_ops {
+	/* send a single command */
+	int (*command)(struct hda_codec *codec, hda_nid_t nid, int direct,
+		       unsigned int verb, unsigned int parm);
+	/* get a response from the last command */
+	unsigned int (*get_response)(struct hda_codec *codec);
+	/* free the private data */
+	void (*private_free)(struct hda_bus *);
+};
+
+/* template to pass to the bus constructor */
+struct hda_bus_template {
+	void *private_data;
+	struct pci_dev *pci;
+	const char *modelname;
+	struct hda_bus_ops ops;
+};
+
+/*
+ * codec bus
+ *
+ * each controller needs to creata a hda_bus to assign the accessor.
+ * A hda_bus contains several codecs in the list codec_list.
+ */
+struct hda_bus {
+	snd_card_t *card;
+
+	/* copied from template */
+	void *private_data;
+	struct pci_dev *pci;
+	const char *modelname;
+	struct hda_bus_ops ops;
+
+	/* codec linked list */
+	struct list_head codec_list;
+	struct hda_codec *caddr_tbl[HDA_MAX_CODEC_ADDRESS]; /* caddr -> codec */
+
+	struct semaphore cmd_mutex;
+
+	/* unsolicited event queue */
+	struct hda_bus_unsolicited *unsol;
+
+	snd_info_entry_t *proc;
+};
+
+/*
+ * codec preset
+ *
+ * Known codecs have the patch to build and set up the controls/PCMs
+ * better than the generic parser.
+ */
+struct hda_codec_preset {
+	unsigned int id;
+	unsigned int mask;
+	unsigned int subs;
+	unsigned int subs_mask;
+	unsigned int rev;
+	const char *name;
+	int (*patch)(struct hda_codec *codec);
+};
+	
+/* ops set by the preset patch */
+struct hda_codec_ops {
+	int (*build_controls)(struct hda_codec *codec);
+	int (*build_pcms)(struct hda_codec *codec);
+	int (*init)(struct hda_codec *codec);
+	void (*free)(struct hda_codec *codec);
+	void (*unsol_event)(struct hda_codec *codec, unsigned int res);
+#ifdef CONFIG_PM
+	int (*suspend)(struct hda_codec *codec, pm_message_t state);
+	int (*resume)(struct hda_codec *codec);
+#endif
+};
+
+/* record for amp information cache */
+struct hda_amp_info {
+	u32 key;		/* hash key */
+	u32 amp_caps;		/* amp capabilities */
+	u16 vol[2];		/* current volume & mute*/
+	u16 status;		/* update flag */
+	u16 next;		/* next link */
+};
+
+/* PCM callbacks */
+struct hda_pcm_ops {
+	int (*open)(struct hda_pcm_stream *info, struct hda_codec *codec,
+		    snd_pcm_substream_t *substream);
+	int (*close)(struct hda_pcm_stream *info, struct hda_codec *codec,
+		     snd_pcm_substream_t *substream);
+	int (*prepare)(struct hda_pcm_stream *info, struct hda_codec *codec,
+		       unsigned int stream_tag, unsigned int format,
+		       snd_pcm_substream_t *substream);
+	int (*cleanup)(struct hda_pcm_stream *info, struct hda_codec *codec,
+		       snd_pcm_substream_t *substream);
+};
+
+/* PCM information for each substream */
+struct hda_pcm_stream {
+	unsigned int substreams;	/* number of substreams, 0 = not exist */
+	unsigned int channels_min;	/* min. number of channels */
+	unsigned int channels_max;	/* max. number of channels */
+	hda_nid_t nid;	/* default NID to query rates/formats/bps, or set up */
+	u32 rates;	/* supported rates */
+	u64 formats;	/* supported formats (SNDRV_PCM_FMTBIT_) */
+	unsigned int maxbps;	/* supported max. bit per sample */
+	struct hda_pcm_ops ops;
+};
+
+/* for PCM creation */
+struct hda_pcm {
+	char *name;
+	struct hda_pcm_stream stream[2];
+};
+
+/* codec information */
+struct hda_codec {
+	struct hda_bus *bus;
+	unsigned int addr;	/* codec addr*/
+	struct list_head list;	/* list point */
+
+	hda_nid_t afg;	/* AFG node id */
+
+	/* ids */
+	u32 vendor_id;
+	u32 subsystem_id;
+	u32 revision_id;
+
+	/* detected preset */
+	const struct hda_codec_preset *preset;
+
+	/* set by patch */
+	struct hda_codec_ops patch_ops;
+
+	/* resume phase - all controls should update even if
+	 * the values are not changed
+	 */
+	unsigned int in_resume;
+
+	/* PCM to create, set by patch_ops.build_pcms callback */
+	unsigned int num_pcms;
+	struct hda_pcm *pcm_info;
+
+	/* codec specific info */
+	void *spec;
+
+	/* hash for amp access */
+	u16 amp_hash[32];
+	int num_amp_entries;
+	struct hda_amp_info amp_info[128]; /* big enough? */
+
+	struct semaphore spdif_mutex;
+	unsigned int spdif_status;	/* IEC958 status bits */
+	unsigned short spdif_ctls;	/* SPDIF control bits */
+	unsigned int spdif_in_enable;	/* SPDIF input enable? */
+};
+
+/* direction */
+enum {
+	HDA_INPUT, HDA_OUTPUT
+};
+
+
+/*
+ * constructors
+ */
+int snd_hda_bus_new(snd_card_t *card, const struct hda_bus_template *temp,
+		    struct hda_bus **busp);
+int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
+		      struct hda_codec **codecp);
+
+/*
+ * low level functions
+ */
+unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid, int direct,
+				unsigned int verb, unsigned int parm);
+int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int direct,
+			unsigned int verb, unsigned int parm);
+#define snd_hda_param_read(codec, nid, param) snd_hda_codec_read(codec, nid, 0, AC_VERB_PARAMETERS, param)
+int snd_hda_get_sub_nodes(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *start_id);
+int snd_hda_get_connections(struct hda_codec *codec, hda_nid_t nid, hda_nid_t *conn_list, int max_conns);
+
+struct hda_verb {
+	hda_nid_t nid;
+	u32 verb;
+	u32 param;
+};
+
+void snd_hda_sequence_write(struct hda_codec *codec, const struct hda_verb *seq);
+
+/* unsolicited event */
+int snd_hda_queue_unsol_event(struct hda_bus *bus, u32 res, u32 res_ex);
+
+/*
+ * Mixer
+ */
+int snd_hda_build_controls(struct hda_bus *bus);
+
+/*
+ * PCM
+ */
+int snd_hda_build_pcms(struct hda_bus *bus);
+void snd_hda_codec_setup_stream(struct hda_codec *codec, hda_nid_t nid, u32 stream_tag,
+				int channel_id, int format);
+unsigned int snd_hda_calc_stream_format(unsigned int rate, unsigned int channels,
+					unsigned int format, unsigned int maxbps);
+int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
+				u32 *ratesp, u64 *formatsp, unsigned int *bpsp);
+int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid,
+				unsigned int format);
+
+/*
+ * Misc
+ */
+void snd_hda_get_codec_name(struct hda_codec *codec, char *name, int namelen);
+
+/*
+ * power management
+ */
+#ifdef CONFIG_PM
+int snd_hda_suspend(struct hda_bus *bus, pm_message_t state);
+int snd_hda_resume(struct hda_bus *bus);
+#endif
+
+#endif /* __SOUND_HDA_CODEC_H */
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
new file mode 100644
index 000000000000..69f7b6c4cf83
--- /dev/null
+++ b/sound/pci/hda/hda_generic.c
@@ -0,0 +1,906 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * Generic widget tree parser
+ *
+ * Copyright (c) 2004 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 <sound/driver.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+#include "hda_local.h"
+
+/* widget node for parsing */
+struct hda_gnode {
+	hda_nid_t nid;		/* NID of this widget */
+	unsigned short nconns;	/* number of input connections */
+	hda_nid_t conn_list[HDA_MAX_CONNECTIONS]; /* input connections */
+	unsigned int wid_caps;	/* widget capabilities */
+	unsigned char type;	/* widget type */
+	unsigned char pin_ctl;	/* pin controls */
+	unsigned char checked;	/* the flag indicates that the node is already parsed */
+	unsigned int pin_caps;	/* pin widget capabilities */
+	unsigned int def_cfg;	/* default configuration */
+	unsigned int amp_out_caps;	/* AMP out capabilities */
+	unsigned int amp_in_caps;	/* AMP in capabilities */
+	struct list_head list;
+};
+
+/* pathc-specific record */
+struct hda_gspec {
+	struct hda_gnode *dac_node;	/* DAC node */
+	struct hda_gnode *out_pin_node;	/* Output pin (Line-Out) node */
+	struct hda_gnode *pcm_vol_node;	/* Node for PCM volume */
+	unsigned int pcm_vol_index;	/* connection of PCM volume */
+
+	struct hda_gnode *adc_node;	/* ADC node */
+	struct hda_gnode *cap_vol_node;	/* Node for capture volume */
+	unsigned int cur_cap_src;	/* current capture source */
+	struct hda_input_mux input_mux;
+	char cap_labels[HDA_MAX_NUM_INPUTS][16];
+
+	unsigned int def_amp_in_caps;
+	unsigned int def_amp_out_caps;
+
+	struct hda_pcm pcm_rec;		/* PCM information */
+
+	struct list_head nid_list;	/* list of widgets */
+};
+
+/*
+ * retrieve the default device type from the default config value
+ */
+#define get_defcfg_type(node) (((node)->def_cfg & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT)
+#define get_defcfg_location(node) (((node)->def_cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT)
+
+/*
+ * destructor
+ */
+static void snd_hda_generic_free(struct hda_codec *codec)
+{
+	struct hda_gspec *spec = codec->spec;
+	struct list_head *p, *n;
+
+	if (! spec)
+		return;
+	/* free all widgets */
+	list_for_each_safe(p, n, &spec->nid_list) {
+		struct hda_gnode *node = list_entry(p, struct hda_gnode, list);
+		kfree(node);
+	}
+	kfree(spec);
+}
+
+
+/*
+ * add a new widget node and read its attributes
+ */
+static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid_t nid)
+{
+	struct hda_gnode *node;
+	int nconns;
+
+	node = kcalloc(1, sizeof(*node), GFP_KERNEL);
+	if (node == NULL)
+		return -ENOMEM;
+	node->nid = nid;
+	nconns = snd_hda_get_connections(codec, nid, node->conn_list, HDA_MAX_CONNECTIONS);
+	if (nconns < 0) {
+		kfree(node);
+		return nconns;
+	}
+	node->nconns = nconns;
+	node->wid_caps = snd_hda_param_read(codec, nid, AC_PAR_AUDIO_WIDGET_CAP);
+	node->type = (node->wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
+
+	if (node->type == AC_WID_PIN) {
+		node->pin_caps = snd_hda_param_read(codec, node->nid, AC_PAR_PIN_CAP);
+		node->pin_ctl = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+		node->def_cfg = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+	}
+
+	if (node->wid_caps & AC_WCAP_OUT_AMP) {
+		if (node->wid_caps & AC_WCAP_AMP_OVRD)
+			node->amp_out_caps = snd_hda_param_read(codec, node->nid, AC_PAR_AMP_OUT_CAP);
+		if (! node->amp_out_caps)
+			node->amp_out_caps = spec->def_amp_out_caps;
+	}
+	if (node->wid_caps & AC_WCAP_IN_AMP) {
+		if (node->wid_caps & AC_WCAP_AMP_OVRD)
+			node->amp_in_caps = snd_hda_param_read(codec, node->nid, AC_PAR_AMP_IN_CAP);
+		if (! node->amp_in_caps)
+			node->amp_in_caps = spec->def_amp_in_caps;
+	}
+	list_add_tail(&node->list, &spec->nid_list);
+	return 0;
+}
+
+/*
+ * build the AFG subtree
+ */
+static int build_afg_tree(struct hda_codec *codec)
+{
+	struct hda_gspec *spec = codec->spec;
+	int i, nodes, err;
+	hda_nid_t nid;
+
+	snd_assert(spec, return -EINVAL);
+
+	spec->def_amp_out_caps = snd_hda_param_read(codec, codec->afg, AC_PAR_AMP_OUT_CAP);
+	spec->def_amp_in_caps = snd_hda_param_read(codec, codec->afg, AC_PAR_AMP_IN_CAP);
+
+	nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid);
+	if (! nid || nodes < 0) {
+		printk(KERN_ERR "Invalid AFG subtree\n");
+		return -EINVAL;
+	}
+
+	/* parse all nodes belonging to the AFG */
+	for (i = 0; i < nodes; i++, nid++) {
+		if ((err = add_new_node(codec, spec, nid)) < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+
+/*
+ * look for the node record for the given NID
+ */
+/* FIXME: should avoid the braindead linear search */
+static struct hda_gnode *hda_get_node(struct hda_gspec *spec, hda_nid_t nid)
+{
+	struct list_head *p;
+	struct hda_gnode *node;
+
+	list_for_each(p, &spec->nid_list) {
+		node = list_entry(p, struct hda_gnode, list);
+		if (node->nid == nid)
+			return node;
+	}
+	return NULL;
+}
+
+/*
+ * unmute (and set max vol) the output amplifier
+ */
+static int unmute_output(struct hda_codec *codec, struct hda_gnode *node)
+{
+	unsigned int val, ofs;
+	snd_printdd("UNMUTE OUT: NID=0x%x\n", node->nid);
+	val = (node->amp_out_caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT;
+	ofs = (node->amp_out_caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT;
+	if (val >= ofs)
+		val -= ofs;
+	val |= AC_AMP_SET_LEFT | AC_AMP_SET_RIGHT;
+	val |= AC_AMP_SET_OUTPUT;
+	return snd_hda_codec_write(codec, node->nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, val);
+}
+
+/*
+ * unmute (and set max vol) the input amplifier
+ */
+static int unmute_input(struct hda_codec *codec, struct hda_gnode *node, unsigned int index)
+{
+	unsigned int val, ofs;
+	snd_printdd("UNMUTE IN: NID=0x%x IDX=0x%x\n", node->nid, index);
+	val = (node->amp_in_caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT;
+	ofs = (node->amp_in_caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT;
+	if (val >= ofs)
+		val -= ofs;
+	val |= AC_AMP_SET_LEFT | AC_AMP_SET_RIGHT;
+	val |= AC_AMP_SET_INPUT;
+	// awk added - fixed to allow unmuting of indexed amps
+	val |= index << AC_AMP_SET_INDEX_SHIFT;
+	return snd_hda_codec_write(codec, node->nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, val);
+}
+
+/*
+ * select the input connection of the given node.
+ */
+static int select_input_connection(struct hda_codec *codec, struct hda_gnode *node,
+				   unsigned int index)
+{
+	snd_printdd("CONNECT: NID=0x%x IDX=0x%x\n", node->nid, index);
+	return snd_hda_codec_write(codec, node->nid, 0, AC_VERB_SET_CONNECT_SEL, index);
+}
+
+/*
+ * clear checked flag of each node in the node list
+ */
+static void clear_check_flags(struct hda_gspec *spec)
+{
+	struct list_head *p;
+	struct hda_gnode *node;
+
+	list_for_each(p, &spec->nid_list) {
+		node = list_entry(p, struct hda_gnode, list);
+		node->checked = 0;
+	}
+}
+
+/*
+ * parse the output path recursively until reach to an audio output widget
+ *
+ * returns 0 if not found, 1 if found, or a negative error code.
+ */
+static int parse_output_path(struct hda_codec *codec, struct hda_gspec *spec,
+			     struct hda_gnode *node)
+{
+	int i, err;
+	struct hda_gnode *child;
+
+	if (node->checked)
+		return 0;
+
+	node->checked = 1;
+	if (node->type == AC_WID_AUD_OUT) {
+		if (node->wid_caps & AC_WCAP_DIGITAL) {
+			snd_printdd("Skip Digital OUT node %x\n", node->nid);
+			return 0;
+		}
+		snd_printdd("AUD_OUT found %x\n", node->nid);
+		if (spec->dac_node) {
+			/* already DAC node is assigned, just unmute & connect */
+			return node == spec->dac_node;
+		}
+		spec->dac_node = node;
+		if (node->wid_caps & AC_WCAP_OUT_AMP) {
+			spec->pcm_vol_node = node;
+			spec->pcm_vol_index = 0;
+		}
+		return 1; /* found */
+	}
+
+	for (i = 0; i < node->nconns; i++) {
+		child = hda_get_node(spec, node->conn_list[i]);
+		if (! child)
+			continue;
+		err = parse_output_path(codec, spec, child);
+		if (err < 0)
+			return err;
+		else if (err > 0) {
+			/* found one,
+			 * select the path, unmute both input and output
+			 */
+			if (node->nconns > 1)
+				select_input_connection(codec, node, i);
+			unmute_input(codec, node, i);
+			unmute_output(codec, node);
+			if (! spec->pcm_vol_node) {
+				if (node->wid_caps & AC_WCAP_IN_AMP) {
+					spec->pcm_vol_node = node;
+					spec->pcm_vol_index = i;
+				} else if (node->wid_caps & AC_WCAP_OUT_AMP) {
+					spec->pcm_vol_node = node;
+					spec->pcm_vol_index = 0;
+				}
+			}
+			return 1;
+		}
+	}
+	return 0;
+}
+
+/*
+ * Look for the output PIN widget with the given jack type
+ * and parse the output path to that PIN.
+ *
+ * Returns the PIN node when the path to DAC is established.
+ */
+static struct hda_gnode *parse_output_jack(struct hda_codec *codec,
+					   struct hda_gspec *spec,
+					   int jack_type)
+{
+	struct list_head *p;
+	struct hda_gnode *node;
+	int err;
+
+	list_for_each(p, &spec->nid_list) {
+		node = list_entry(p, struct hda_gnode, list);
+		if (node->type != AC_WID_PIN)
+			continue;
+		/* output capable? */
+		if (! (node->pin_caps & AC_PINCAP_OUT))
+			continue;
+		if (jack_type >= 0) {
+			if (jack_type != get_defcfg_type(node))
+				continue;
+			if (node->wid_caps & AC_WCAP_DIGITAL)
+				continue; /* skip SPDIF */
+		} else {
+			/* output as default? */
+			if (! (node->pin_ctl & AC_PINCTL_OUT_EN))
+				continue;
+		}
+		clear_check_flags(spec);
+		err = parse_output_path(codec, spec, node);
+		if (err < 0)
+			return NULL;
+		else if (err > 0) {
+			/* unmute the PIN output */
+			unmute_output(codec, node);
+			/* set PIN-Out enable */
+			snd_hda_codec_write(codec, node->nid, 0,
+					    AC_VERB_SET_PIN_WIDGET_CONTROL,
+					    AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN);
+			return node;
+		}
+	}
+	return NULL;
+}
+
+
+/*
+ * parse outputs
+ */
+static int parse_output(struct hda_codec *codec)
+{
+	struct hda_gspec *spec = codec->spec;
+	struct hda_gnode *node;
+
+	/*
+	 * Look for the output PIN widget
+	 */
+	/* first, look for the line-out pin */
+	node = parse_output_jack(codec, spec, AC_JACK_LINE_OUT);
+	if (node) /* found, remember the PIN node */
+		spec->out_pin_node = node;
+	/* look for the HP-out pin */
+	node = parse_output_jack(codec, spec, AC_JACK_HP_OUT);
+	if (node) {
+		if (! spec->out_pin_node)
+			spec->out_pin_node = node;
+	}
+
+	if (! spec->out_pin_node) {
+		/* no line-out or HP pins found,
+		 * then choose for the first output pin
+		 */
+		spec->out_pin_node = parse_output_jack(codec, spec, -1);
+		if (! spec->out_pin_node)
+			snd_printd("hda_generic: no proper output path found\n");
+	}
+
+	return 0;
+}
+
+/*
+ * input MUX
+ */
+
+/* control callbacks */
+static int capture_source_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct hda_gspec *spec = codec->spec;
+	return snd_hda_input_mux_info(&spec->input_mux, uinfo);
+}
+
+static int capture_source_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct hda_gspec *spec = codec->spec;
+
+	ucontrol->value.enumerated.item[0] = spec->cur_cap_src;
+	return 0;
+}
+
+static int capture_source_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct hda_gspec *spec = codec->spec;
+	return snd_hda_input_mux_put(codec, &spec->input_mux, ucontrol,
+				     spec->adc_node->nid, &spec->cur_cap_src);
+}
+
+/*
+ * return the string name of the given input PIN widget
+ */
+static const char *get_input_type(struct hda_gnode *node, unsigned int *pinctl)
+{
+	unsigned int location = get_defcfg_location(node);
+	switch (get_defcfg_type(node)) {
+	case AC_JACK_LINE_IN:
+		if ((location & 0x0f) == AC_JACK_LOC_FRONT)
+			return "Front Line";
+		return "Line";
+	case AC_JACK_CD:
+		if (pinctl)
+			*pinctl |= AC_PIN_VREF_GRD;
+		return "CD";
+	case AC_JACK_AUX:
+		if ((location & 0x0f) == AC_JACK_LOC_FRONT)
+			return "Front Aux";
+		return "Aux";
+	case AC_JACK_MIC_IN:
+		if ((location & 0x0f) == AC_JACK_LOC_FRONT)
+			return "Front Mic";
+		return "Mic";
+	case AC_JACK_SPDIF_IN:
+		return "SPDIF";
+	case AC_JACK_DIG_OTHER_IN:
+		return "Digital";
+	}
+	return NULL;
+}
+
+/*
+ * parse the nodes recursively until reach to the input PIN
+ *
+ * returns 0 if not found, 1 if found, or a negative error code.
+ */
+static int parse_adc_sub_nodes(struct hda_codec *codec, struct hda_gspec *spec,
+			       struct hda_gnode *node)
+{
+	int i, err;
+	unsigned int pinctl;
+	char *label;
+	const char *type;
+
+	if (node->checked)
+		return 0;
+
+	node->checked = 1;
+	if (node->type != AC_WID_PIN) {
+		for (i = 0; i < node->nconns; i++) {
+			struct hda_gnode *child;
+			child = hda_get_node(spec, node->conn_list[i]);
+			if (! child)
+				continue;
+			err = parse_adc_sub_nodes(codec, spec, child);
+			if (err < 0)
+				return err;
+			if (err > 0) {
+				/* found one,
+				 * select the path, unmute both input and output
+				 */
+				if (node->nconns > 1)
+					select_input_connection(codec, node, i);
+				unmute_input(codec, node, i);
+				unmute_output(codec, node);
+				return err;
+			}
+		}
+		return 0;
+	}
+
+	/* input capable? */
+	if (! (node->pin_caps & AC_PINCAP_IN))
+		return 0;
+
+	if (node->wid_caps & AC_WCAP_DIGITAL)
+		return 0; /* skip SPDIF */
+
+	if (spec->input_mux.num_items >= HDA_MAX_NUM_INPUTS) {
+		snd_printk(KERN_ERR "hda_generic: Too many items for capture\n");
+		return -EINVAL;
+	}
+
+	pinctl = AC_PINCTL_IN_EN;
+	/* create a proper capture source label */
+	type = get_input_type(node, &pinctl);
+	if (! type) {
+		/* input as default? */
+		if (! (node->pin_ctl & AC_PINCTL_IN_EN))
+			return 0;
+		type = "Input";
+	}
+	label = spec->cap_labels[spec->input_mux.num_items];
+	strcpy(label, type);
+	spec->input_mux.items[spec->input_mux.num_items].label = label;
+
+	/* unmute the PIN external input */
+	unmute_input(codec, node, 0); /* index = 0? */
+	/* set PIN-In enable */
+	snd_hda_codec_write(codec, node->nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl);
+
+	return 1; /* found */
+}
+
+/*
+ * parse input
+ */
+static int parse_input_path(struct hda_codec *codec, struct hda_gnode *adc_node)
+{
+	struct hda_gspec *spec = codec->spec;
+	struct hda_gnode *node;
+	int i, err;
+
+	snd_printdd("AUD_IN = %x\n", adc_node->nid);
+	clear_check_flags(spec);
+
+	// awk added - fixed no recording due to muted widget
+	unmute_input(codec, adc_node, 0);
+	
+	/*
+	 * check each connection of the ADC
+	 * if it reaches to a proper input PIN, add the path as the
+	 * input path.
+	 */
+	for (i = 0; i < adc_node->nconns; i++) {
+		node = hda_get_node(spec, adc_node->conn_list[i]);
+		if (! node)
+			continue;
+		err = parse_adc_sub_nodes(codec, spec, node);
+		if (err < 0)
+			return err;
+		else if (err > 0) {
+			struct hda_input_mux_item *csrc = &spec->input_mux.items[spec->input_mux.num_items];
+			char *buf = spec->cap_labels[spec->input_mux.num_items];
+			int ocap;
+			for (ocap = 0; ocap < spec->input_mux.num_items; ocap++) {
+				if (! strcmp(buf, spec->cap_labels[ocap])) {
+					/* same label already exists,
+					 * put the index number to be unique
+					 */
+					sprintf(buf, "%s %d", spec->cap_labels[ocap],
+						spec->input_mux.num_items);
+				}
+			}
+			csrc->index = i;
+			spec->input_mux.num_items++;
+		}
+	}
+
+	if (! spec->input_mux.num_items)
+		return 0; /* no input path found... */
+
+	snd_printdd("[Capture Source] NID=0x%x, #SRC=%d\n", adc_node->nid, spec->input_mux.num_items);
+	for (i = 0; i < spec->input_mux.num_items; i++)
+		snd_printdd("  [%s] IDX=0x%x\n", spec->input_mux.items[i].label,
+			    spec->input_mux.items[i].index);
+
+	spec->adc_node = adc_node;
+	return 1;
+}
+
+/*
+ * parse input
+ */
+static int parse_input(struct hda_codec *codec)
+{
+	struct hda_gspec *spec = codec->spec;
+	struct list_head *p;
+	struct hda_gnode *node;
+	int err;
+
+	/*
+	 * At first we look for an audio input widget.
+	 * If it reaches to certain input PINs, we take it as the
+	 * input path.
+	 */
+	list_for_each(p, &spec->nid_list) {
+		node = list_entry(p, struct hda_gnode, list);
+		if (node->wid_caps & AC_WCAP_DIGITAL)
+			continue; /* skip SPDIF */
+		if (node->type == AC_WID_AUD_IN) {
+			err = parse_input_path(codec, node);
+			if (err < 0)
+				return err;
+			else if (err > 0)
+				return 0;
+		}
+	}
+	snd_printd("hda_generic: no proper input path found\n");
+	return 0;
+}
+
+/*
+ * create mixer controls if possible
+ */
+#define DIR_OUT		0x1
+#define DIR_IN		0x2
+
+static int create_mixer(struct hda_codec *codec, struct hda_gnode *node,
+			unsigned int index, const char *type, const char *dir_sfx)
+{
+	char name[32];
+	int err;
+	int created = 0;
+	snd_kcontrol_new_t knew;
+
+	if (type)
+		sprintf(name, "%s %s Switch", type, dir_sfx);
+	else
+		sprintf(name, "%s Switch", dir_sfx);
+	if ((node->wid_caps & AC_WCAP_IN_AMP) &&
+	    (node->amp_in_caps & AC_AMPCAP_MUTE)) {
+		knew = (snd_kcontrol_new_t)HDA_CODEC_MUTE(name, node->nid, index, HDA_INPUT);
+		snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index);
+		if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
+			return err;
+		created = 1;
+	} else if ((node->wid_caps & AC_WCAP_OUT_AMP) &&
+		   (node->amp_out_caps & AC_AMPCAP_MUTE)) {
+		knew = (snd_kcontrol_new_t)HDA_CODEC_MUTE(name, node->nid, 0, HDA_OUTPUT);
+		snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid);
+		if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
+			return err;
+		created = 1;
+	}
+
+	if (type)
+		sprintf(name, "%s %s Volume", type, dir_sfx);
+	else
+		sprintf(name, "%s Volume", dir_sfx);
+	if ((node->wid_caps & AC_WCAP_IN_AMP) &&
+	    (node->amp_in_caps & AC_AMPCAP_NUM_STEPS)) {
+		knew = (snd_kcontrol_new_t)HDA_CODEC_VOLUME(name, node->nid, index, HDA_INPUT);
+		snd_printdd("[%s] NID=0x%x, DIR=IN, IDX=0x%x\n", name, node->nid, index);
+		if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
+			return err;
+		created = 1;
+	} else if ((node->wid_caps & AC_WCAP_OUT_AMP) &&
+		   (node->amp_out_caps & AC_AMPCAP_NUM_STEPS)) {
+		knew = (snd_kcontrol_new_t)HDA_CODEC_VOLUME(name, node->nid, 0, HDA_OUTPUT);
+		snd_printdd("[%s] NID=0x%x, DIR=OUT\n", name, node->nid);
+		if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&knew, codec))) < 0)
+			return err;
+		created = 1;
+	}
+
+	return created;
+}
+
+/*
+ * check whether the controls with the given name and direction suffix already exist
+ */
+static int check_existing_control(struct hda_codec *codec, const char *type, const char *dir)
+{
+	snd_ctl_elem_id_t id;
+	memset(&id, 0, sizeof(id));
+	sprintf(id.name, "%s %s Volume", type, dir);
+	id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+	if (snd_ctl_find_id(codec->bus->card, &id))
+		return 1;
+	sprintf(id.name, "%s %s Switch", type, dir);
+	id.iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+	if (snd_ctl_find_id(codec->bus->card, &id))
+		return 1;
+	return 0;
+}
+
+/*
+ * build output mixer controls
+ */
+static int build_output_controls(struct hda_codec *codec)
+{
+	struct hda_gspec *spec = codec->spec;
+	int err;
+
+	err = create_mixer(codec, spec->pcm_vol_node, spec->pcm_vol_index,
+			   "PCM", "Playback");
+	if (err < 0)
+		return err;
+	return 0;
+}
+
+/* create capture volume/switch */
+static int build_input_controls(struct hda_codec *codec)
+{
+	struct hda_gspec *spec = codec->spec;
+	struct hda_gnode *adc_node = spec->adc_node;
+	int err;
+
+	if (! adc_node)
+		return 0; /* not found */
+
+	/* create capture volume and switch controls if the ADC has an amp */
+	err = create_mixer(codec, adc_node, 0, NULL, "Capture");
+
+	/* create input MUX if multiple sources are available */
+	if (spec->input_mux.num_items > 1) {
+		static snd_kcontrol_new_t cap_sel = {
+			.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+			.name = "Capture Source",
+			.info = capture_source_info,
+			.get = capture_source_get,
+			.put = capture_source_put,
+		};
+		if ((err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&cap_sel, codec))) < 0)
+			return err;
+		spec->cur_cap_src = 0;
+		select_input_connection(codec, adc_node, spec->input_mux.items[0].index);
+	}
+	return 0;
+}
+
+
+/*
+ * parse the nodes recursively until reach to the output PIN.
+ *
+ * returns 0 - if not found,
+ *         1 - if found, but no mixer is created
+ *         2 - if found and mixer was already created, (just skip)
+ *         a negative error code
+ */
+static int parse_loopback_path(struct hda_codec *codec, struct hda_gspec *spec,
+			       struct hda_gnode *node, struct hda_gnode *dest_node,
+			       const char *type)
+{
+	int i, err;
+
+	if (node->checked)
+		return 0;
+
+	node->checked = 1;
+	if (node == dest_node) {
+		/* loopback connection found */
+		return 1;
+	}
+
+	for (i = 0; i < node->nconns; i++) {
+		struct hda_gnode *child = hda_get_node(spec, node->conn_list[i]);
+		if (! child)
+			continue;
+		err = parse_loopback_path(codec, spec, child, dest_node, type);
+		if (err < 0)
+			return err;
+		else if (err >= 1) {
+			if (err == 1) {
+				err = create_mixer(codec, node, i, type, "Playback");
+				if (err < 0)
+					return err;
+				if (err > 0)
+					return 2; /* ok, created */
+				/* not created, maybe in the lower path */
+				err = 1;
+			}
+			/* connect and unmute */
+			if (node->nconns > 1)
+				select_input_connection(codec, node, i);
+			unmute_input(codec, node, i);
+			unmute_output(codec, node);
+			return err;
+		}
+	}
+	return 0;
+}
+
+/*
+ * parse the tree and build the loopback controls
+ */
+static int build_loopback_controls(struct hda_codec *codec)
+{
+	struct hda_gspec *spec = codec->spec;
+	struct list_head *p;
+	struct hda_gnode *node;
+	int err;
+	const char *type;
+
+	if (! spec->out_pin_node)
+		return 0;
+
+	list_for_each(p, &spec->nid_list) {
+		node = list_entry(p, struct hda_gnode, list);
+		if (node->type != AC_WID_PIN)
+			continue;
+		/* input capable? */
+		if (! (node->pin_caps & AC_PINCAP_IN))
+			return 0;
+		type = get_input_type(node, NULL);
+		if (type) {
+			if (check_existing_control(codec, type, "Playback"))
+				continue;
+			clear_check_flags(spec);
+			err = parse_loopback_path(codec, spec, spec->out_pin_node,
+						  node, type);
+			if (err < 0)
+				return err;
+			if (! err)
+				continue;
+		}
+	}
+	return 0;
+}
+
+/*
+ * build mixer controls
+ */
+static int build_generic_controls(struct hda_codec *codec)
+{
+	int err;
+
+	if ((err = build_input_controls(codec)) < 0 ||
+	    (err = build_output_controls(codec)) < 0 ||
+	    (err = build_loopback_controls(codec)) < 0)
+		return err;
+
+	return 0;
+}
+
+/*
+ * PCM
+ */
+static struct hda_pcm_stream generic_pcm_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+};
+
+static int build_generic_pcms(struct hda_codec *codec)
+{
+	struct hda_gspec *spec = codec->spec;
+	struct hda_pcm *info = &spec->pcm_rec;
+
+	if (! spec->dac_node && ! spec->adc_node) {
+		snd_printd("hda_generic: no PCM found\n");
+		return 0;
+	}
+
+	codec->num_pcms = 1;
+	codec->pcm_info = info;
+
+	info->name = "HDA Generic";
+	if (spec->dac_node) {
+		info->stream[0] = generic_pcm_playback;
+		info->stream[0].nid = spec->dac_node->nid;
+	}
+	if (spec->adc_node) {
+		info->stream[1] = generic_pcm_playback;
+		info->stream[1].nid = spec->adc_node->nid;
+	}
+
+	return 0;
+}
+
+
+/*
+ */
+static struct hda_codec_ops generic_patch_ops = {
+	.build_controls = build_generic_controls,
+	.build_pcms = build_generic_pcms,
+	.free = snd_hda_generic_free,
+};
+
+/*
+ * the generic parser
+ */
+int snd_hda_parse_generic_codec(struct hda_codec *codec)
+{
+	struct hda_gspec *spec;
+	int err;
+
+	spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
+	if (spec == NULL) {
+		printk(KERN_ERR "hda_generic: can't allocate spec\n");
+		return -ENOMEM;
+	}
+	codec->spec = spec;
+	INIT_LIST_HEAD(&spec->nid_list);
+
+	if ((err = build_afg_tree(codec)) < 0)
+		goto error;
+
+	if ((err = parse_input(codec)) < 0 ||
+	    (err = parse_output(codec)) < 0)
+		goto error;
+
+	codec->patch_ops = generic_patch_ops;
+
+	return 0;
+
+ error:
+	snd_hda_generic_free(codec);
+	return err;
+}
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
new file mode 100644
index 000000000000..d89647a3d449
--- /dev/null
+++ b/sound/pci/hda/hda_intel.c
@@ -0,0 +1,1449 @@
+/*
+ *
+ *  hda_intel.c - Implementation of primary alsa driver code base for Intel HD Audio.
+ *
+ *  Copyright(c) 2004 Intel Corporation. All rights reserved.
+ *
+ *  Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
+ *                     PeiSen Hou <pshou@realtek.com.tw>
+ *
+ *  This program is free software; you can redistribute it and/or modify it
+ *  under the terms of the GNU General Public License as published by the Free
+ *  Software Foundation; either version 2 of the License, or (at your option)
+ *  any later version.
+ *
+ *  This program is distributed in the hope that it will be useful, but WITHOUT
+ *  ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
+ *  FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License for
+ *  more details.
+ *
+ *  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.
+ *
+ *  CONTACTS:
+ *
+ *  Matt Jared		matt.jared@intel.com
+ *  Andy Kopp		andy.kopp@intel.com
+ *  Dan Kogan		dan.d.kogan@intel.com
+ *
+ *  CHANGES:
+ *
+ *  2004.12.01	Major rewrite by tiwai, merged the work of pshou
+ * 
+ */
+
+#include <sound/driver.h>
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include "hda_codec.h"
+
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+static char *model[SNDRV_CARDS];
+
+module_param_array(index, int, NULL, 0444);
+MODULE_PARM_DESC(index, "Index value for Intel HD audio interface.");
+module_param_array(id, charp, NULL, 0444);
+MODULE_PARM_DESC(id, "ID string for Intel HD audio interface.");
+module_param_array(enable, bool, NULL, 0444);
+MODULE_PARM_DESC(enable, "Enable Intel HD audio interface.");
+module_param_array(model, charp, NULL, 0444);
+MODULE_PARM_DESC(model, "Use the given board model.");
+
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Intel, ICH6},"
+			 "{Intel, ICH6M},"
+			 "{Intel, ICH7}}");
+MODULE_DESCRIPTION("Intel HDA driver");
+
+#define SFX	"hda-intel: "
+
+/*
+ * registers
+ */
+#define ICH6_REG_GCAP			0x00
+#define ICH6_REG_VMIN			0x02
+#define ICH6_REG_VMAJ			0x03
+#define ICH6_REG_OUTPAY			0x04
+#define ICH6_REG_INPAY			0x06
+#define ICH6_REG_GCTL			0x08
+#define ICH6_REG_WAKEEN			0x0c
+#define ICH6_REG_STATESTS		0x0e
+#define ICH6_REG_GSTS			0x10
+#define ICH6_REG_INTCTL			0x20
+#define ICH6_REG_INTSTS			0x24
+#define ICH6_REG_WALCLK			0x30
+#define ICH6_REG_SYNC			0x34	
+#define ICH6_REG_CORBLBASE		0x40
+#define ICH6_REG_CORBUBASE		0x44
+#define ICH6_REG_CORBWP			0x48
+#define ICH6_REG_CORBRP			0x4A
+#define ICH6_REG_CORBCTL		0x4c
+#define ICH6_REG_CORBSTS		0x4d
+#define ICH6_REG_CORBSIZE		0x4e
+
+#define ICH6_REG_RIRBLBASE		0x50
+#define ICH6_REG_RIRBUBASE		0x54
+#define ICH6_REG_RIRBWP			0x58
+#define ICH6_REG_RINTCNT		0x5a
+#define ICH6_REG_RIRBCTL		0x5c
+#define ICH6_REG_RIRBSTS		0x5d
+#define ICH6_REG_RIRBSIZE		0x5e
+
+#define ICH6_REG_IC			0x60
+#define ICH6_REG_IR			0x64
+#define ICH6_REG_IRS			0x68
+#define   ICH6_IRS_VALID	(1<<1)
+#define   ICH6_IRS_BUSY		(1<<0)
+
+#define ICH6_REG_DPLBASE		0x70
+#define ICH6_REG_DPUBASE		0x74
+#define   ICH6_DPLBASE_ENABLE	0x1	/* Enable position buffer */
+
+/* SD offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */
+enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 };
+
+/* stream register offsets from stream base */
+#define ICH6_REG_SD_CTL			0x00
+#define ICH6_REG_SD_STS			0x03
+#define ICH6_REG_SD_LPIB		0x04
+#define ICH6_REG_SD_CBL			0x08
+#define ICH6_REG_SD_LVI			0x0c
+#define ICH6_REG_SD_FIFOW		0x0e
+#define ICH6_REG_SD_FIFOSIZE		0x10
+#define ICH6_REG_SD_FORMAT		0x12
+#define ICH6_REG_SD_BDLPL		0x18
+#define ICH6_REG_SD_BDLPU		0x1c
+
+/* PCI space */
+#define ICH6_PCIREG_TCSEL	0x44
+
+/*
+ * other constants
+ */
+
+/* max number of SDs */
+#define MAX_ICH6_DEV		8
+/* max number of fragments - we may use more if allocating more pages for BDL */
+#define AZX_MAX_FRAG		(PAGE_SIZE / (MAX_ICH6_DEV * 16))
+/* max buffer size - no h/w limit, you can increase as you like */
+#define AZX_MAX_BUF_SIZE	(1024*1024*1024)
+/* max number of PCM devics per card */
+#define AZX_MAX_PCMS		8
+
+/* RIRB int mask: overrun[2], response[0] */
+#define RIRB_INT_RESPONSE	0x01
+#define RIRB_INT_OVERRUN	0x04
+#define RIRB_INT_MASK		0x05
+
+/* STATESTS int mask: SD2,SD1,SD0 */
+#define STATESTS_INT_MASK	0x07
+#define AZX_MAX_CODECS		3
+
+/* SD_CTL bits */
+#define SD_CTL_STREAM_RESET	0x01	/* stream reset bit */
+#define SD_CTL_DMA_START	0x02	/* stream DMA start bit */
+#define SD_CTL_STREAM_TAG_MASK	(0xf << 20)
+#define SD_CTL_STREAM_TAG_SHIFT	20
+
+/* SD_CTL and SD_STS */
+#define SD_INT_DESC_ERR		0x10	/* descriptor error interrupt */
+#define SD_INT_FIFO_ERR		0x08	/* FIFO error interrupt */
+#define SD_INT_COMPLETE		0x04	/* completion interrupt */
+#define SD_INT_MASK		(SD_INT_DESC_ERR|SD_INT_FIFO_ERR|SD_INT_COMPLETE)
+
+/* SD_STS */
+#define SD_STS_FIFO_READY	0x20	/* FIFO ready */
+
+/* INTCTL and INTSTS */
+#define ICH6_INT_ALL_STREAM	0xff		/* all stream interrupts */
+#define ICH6_INT_CTRL_EN	0x40000000	/* controller interrupt enable bit */
+#define ICH6_INT_GLOBAL_EN	0x80000000	/* global interrupt enable bit */
+
+/* GCTL reset bit */
+#define ICH6_GCTL_RESET		(1<<0)
+
+/* CORB/RIRB control, read/write pointer */
+#define ICH6_RBCTL_DMA_EN	0x02	/* enable DMA */
+#define ICH6_RBCTL_IRQ_EN	0x01	/* enable IRQ */
+#define ICH6_RBRWP_CLR		0x8000	/* read/write pointer clear */
+/* below are so far hardcoded - should read registers in future */
+#define ICH6_MAX_CORB_ENTRIES	256
+#define ICH6_MAX_RIRB_ENTRIES	256
+
+
+/*
+ * Use CORB/RIRB for communication from/to codecs.
+ * This is the way recommended by Intel (see below).
+ */
+#define USE_CORB_RIRB
+
+/*
+ * Define this if use the position buffer instead of reading SD_LPIB
+ * It's not used as default since SD_LPIB seems to give more accurate position
+ */
+/* #define USE_POSBUF */
+
+/*
+ */
+
+typedef struct snd_azx azx_t;
+typedef struct snd_azx_rb azx_rb_t;
+typedef struct snd_azx_dev azx_dev_t;
+
+struct snd_azx_dev {
+	u32 *bdl;			/* virtual address of the BDL */
+	dma_addr_t bdl_addr;		/* physical address of the BDL */
+	volatile u32 *posbuf;			/* position buffer pointer */
+
+	unsigned int bufsize;		/* size of the play buffer in bytes */
+	unsigned int fragsize;		/* size of each period in bytes */
+	unsigned int frags;		/* number for period in the play buffer */
+	unsigned int fifo_size;		/* FIFO size */
+
+	void __iomem *sd_addr;		/* stream descriptor pointer */
+
+	u32 sd_int_sta_mask;		/* stream int status mask */
+
+	/* pcm support */
+	snd_pcm_substream_t *substream;	/* assigned substream, set in PCM open */
+	unsigned int format_val;	/* format value to be set in the controller and the codec */
+	unsigned char stream_tag;	/* assigned stream */
+	unsigned char index;		/* stream index */
+
+	unsigned int opened: 1;
+	unsigned int running: 1;
+};
+
+/* CORB/RIRB */
+struct snd_azx_rb {
+	u32 *buf;		/* CORB/RIRB buffer
+				 * Each CORB entry is 4byte, RIRB is 8byte
+				 */
+	dma_addr_t addr;	/* physical address of CORB/RIRB buffer */
+	/* for RIRB */
+	unsigned short rp, wp;	/* read/write pointers */
+	int cmds;		/* number of pending requests */
+	u32 res;		/* last read value */
+};
+
+struct snd_azx {
+	snd_card_t *card;
+	struct pci_dev *pci;
+
+	/* pci resources */
+	unsigned long addr;
+	void __iomem *remap_addr;
+	int irq;
+
+	/* locks */
+	spinlock_t reg_lock;
+	struct semaphore open_mutex;
+
+	/* streams */
+	azx_dev_t azx_dev[MAX_ICH6_DEV];
+
+	/* PCM */
+	unsigned int pcm_devs;
+	snd_pcm_t *pcm[AZX_MAX_PCMS];
+
+	/* HD codec */
+	unsigned short codec_mask;
+	struct hda_bus *bus;
+
+	/* CORB/RIRB */
+	azx_rb_t corb;
+	azx_rb_t rirb;
+
+	/* BDL, CORB/RIRB and position buffers */
+	struct snd_dma_buffer bdl;
+	struct snd_dma_buffer rb;
+	struct snd_dma_buffer posbuf;
+};
+
+/*
+ * macros for easy use
+ */
+#define azx_writel(chip,reg,value) \
+	writel(value, (chip)->remap_addr + ICH6_REG_##reg)
+#define azx_readl(chip,reg) \
+	readl((chip)->remap_addr + ICH6_REG_##reg)
+#define azx_writew(chip,reg,value) \
+	writew(value, (chip)->remap_addr + ICH6_REG_##reg)
+#define azx_readw(chip,reg) \
+	readw((chip)->remap_addr + ICH6_REG_##reg)
+#define azx_writeb(chip,reg,value) \
+	writeb(value, (chip)->remap_addr + ICH6_REG_##reg)
+#define azx_readb(chip,reg) \
+	readb((chip)->remap_addr + ICH6_REG_##reg)
+
+#define azx_sd_writel(dev,reg,value) \
+	writel(value, (dev)->sd_addr + ICH6_REG_##reg)
+#define azx_sd_readl(dev,reg) \
+	readl((dev)->sd_addr + ICH6_REG_##reg)
+#define azx_sd_writew(dev,reg,value) \
+	writew(value, (dev)->sd_addr + ICH6_REG_##reg)
+#define azx_sd_readw(dev,reg) \
+	readw((dev)->sd_addr + ICH6_REG_##reg)
+#define azx_sd_writeb(dev,reg,value) \
+	writeb(value, (dev)->sd_addr + ICH6_REG_##reg)
+#define azx_sd_readb(dev,reg) \
+	readb((dev)->sd_addr + ICH6_REG_##reg)
+
+/* for pcm support */
+#define get_azx_dev(substream) (azx_dev_t*)(substream->runtime->private_data)
+
+/* Get the upper 32bit of the given dma_addr_t
+ * Compiler should optimize and eliminate the code if dma_addr_t is 32bit
+ */
+#define upper_32bit(addr) (sizeof(addr) > 4 ? (u32)((addr) >> 32) : (u32)0)
+
+
+/*
+ * Interface for HD codec
+ */
+
+#ifdef USE_CORB_RIRB
+/*
+ * CORB / RIRB interface
+ */
+static int azx_alloc_cmd_io(azx_t *chip)
+{
+	int err;
+
+	/* single page (at least 4096 bytes) must suffice for both ringbuffes */
+	err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
+				  PAGE_SIZE, &chip->rb);
+	if (err < 0) {
+		snd_printk(KERN_ERR SFX "cannot allocate CORB/RIRB\n");
+		return err;
+	}
+	return 0;
+}
+
+static void azx_init_cmd_io(azx_t *chip)
+{
+	/* CORB set up */
+	chip->corb.addr = chip->rb.addr;
+	chip->corb.buf = (u32 *)chip->rb.area;
+	azx_writel(chip, CORBLBASE, (u32)chip->corb.addr);
+	azx_writel(chip, CORBUBASE, upper_32bit(chip->corb.addr));
+
+	/* set the corb write pointer to 0 */
+	azx_writew(chip, CORBWP, 0);
+	/* reset the corb hw read pointer */
+	azx_writew(chip, CORBRP, ICH6_RBRWP_CLR);
+	/* enable corb dma */
+	azx_writeb(chip, CORBCTL, ICH6_RBCTL_DMA_EN);
+
+	/* RIRB set up */
+	chip->rirb.addr = chip->rb.addr + 2048;
+	chip->rirb.buf = (u32 *)(chip->rb.area + 2048);
+	azx_writel(chip, RIRBLBASE, (u32)chip->rirb.addr);
+	azx_writel(chip, RIRBUBASE, upper_32bit(chip->rirb.addr));
+
+	/* reset the rirb hw write pointer */
+	azx_writew(chip, RIRBWP, ICH6_RBRWP_CLR);
+	/* set N=1, get RIRB response interrupt for new entry */
+	azx_writew(chip, RINTCNT, 1);
+	/* enable rirb dma and response irq */
+#ifdef USE_CORB_RIRB
+	azx_writeb(chip, RIRBCTL, ICH6_RBCTL_DMA_EN | ICH6_RBCTL_IRQ_EN);
+#else
+	azx_writeb(chip, RIRBCTL, ICH6_RBCTL_DMA_EN);
+#endif
+	chip->rirb.rp = chip->rirb.cmds = 0;
+}
+
+static void azx_free_cmd_io(azx_t *chip)
+{
+	/* disable ringbuffer DMAs */
+	azx_writeb(chip, RIRBCTL, 0);
+	azx_writeb(chip, CORBCTL, 0);
+}
+
+/* send a command */
+static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid, int direct,
+			unsigned int verb, unsigned int para)
+{
+	azx_t *chip = codec->bus->private_data;
+	unsigned int wp;
+	u32 val;
+
+	val = (u32)(codec->addr & 0x0f) << 28;
+	val |= (u32)direct << 27;
+	val |= (u32)nid << 20;
+	val |= verb << 8;
+	val |= para;
+
+	/* add command to corb */
+	wp = azx_readb(chip, CORBWP);
+	wp++;
+	wp %= ICH6_MAX_CORB_ENTRIES;
+
+	spin_lock_irq(&chip->reg_lock);
+	chip->rirb.cmds++;
+	chip->corb.buf[wp] = cpu_to_le32(val);
+	azx_writel(chip, CORBWP, wp);
+	spin_unlock_irq(&chip->reg_lock);
+
+	return 0;
+}
+
+#define ICH6_RIRB_EX_UNSOL_EV	(1<<4)
+
+/* retrieve RIRB entry - called from interrupt handler */
+static void azx_update_rirb(azx_t *chip)
+{
+	unsigned int rp, wp;
+	u32 res, res_ex;
+
+	wp = azx_readb(chip, RIRBWP);
+	if (wp == chip->rirb.wp)
+		return;
+	chip->rirb.wp = wp;
+		
+	while (chip->rirb.rp != wp) {
+		chip->rirb.rp++;
+		chip->rirb.rp %= ICH6_MAX_RIRB_ENTRIES;
+
+		rp = chip->rirb.rp << 1; /* an RIRB entry is 8-bytes */
+		res_ex = le32_to_cpu(chip->rirb.buf[rp + 1]);
+		res = le32_to_cpu(chip->rirb.buf[rp]);
+		if (res_ex & ICH6_RIRB_EX_UNSOL_EV)
+			snd_hda_queue_unsol_event(chip->bus, res, res_ex);
+		else if (chip->rirb.cmds) {
+			chip->rirb.cmds--;
+			chip->rirb.res = res;
+		}
+	}
+}
+
+/* receive a response */
+static unsigned int azx_get_response(struct hda_codec *codec)
+{
+	azx_t *chip = codec->bus->private_data;
+	int timeout = 50;
+
+	while (chip->rirb.cmds) {
+		if (! --timeout) {
+			snd_printk(KERN_ERR "azx_get_response timeout\n");
+			chip->rirb.rp = azx_readb(chip, RIRBWP);
+			chip->rirb.cmds = 0;
+			return -1;
+		}
+		msleep(1);
+	}
+	return chip->rirb.res; /* the last value */
+}
+
+#else
+/*
+ * Use the single immediate command instead of CORB/RIRB for simplicity
+ *
+ * Note: according to Intel, this is not preferred use.  The command was
+ *       intended for the BIOS only, and may get confused with unsolicited
+ *       responses.  So, we shouldn't use it for normal operation from the
+ *       driver.
+ *       I left the codes, however, for debugging/testing purposes.
+ */
+
+#define azx_alloc_cmd_io(chip)	0
+#define azx_init_cmd_io(chip)
+#define azx_free_cmd_io(chip)
+
+/* send a command */
+static int azx_send_cmd(struct hda_codec *codec, hda_nid_t nid, int direct,
+			unsigned int verb, unsigned int para)
+{
+	azx_t *chip = codec->bus->private_data;
+	u32 val;
+	int timeout = 50;
+
+	val = (u32)(codec->addr & 0x0f) << 28;
+	val |= (u32)direct << 27;
+	val |= (u32)nid << 20;
+	val |= verb << 8;
+	val |= para;
+
+	while (timeout--) {
+		/* check ICB busy bit */
+		if (! (azx_readw(chip, IRS) & ICH6_IRS_BUSY)) {
+			/* Clear IRV valid bit */
+			azx_writew(chip, IRS, azx_readw(chip, IRS) | ICH6_IRS_VALID);
+			azx_writel(chip, IC, val);
+			azx_writew(chip, IRS, azx_readw(chip, IRS) | ICH6_IRS_BUSY);
+			return 0;
+		}
+		udelay(1);
+	}
+	snd_printd(SFX "send_cmd timeout: IRS=0x%x, val=0x%x\n", azx_readw(chip, IRS), val);
+	return -EIO;
+}
+
+/* receive a response */
+static unsigned int azx_get_response(struct hda_codec *codec)
+{
+	azx_t *chip = codec->bus->private_data;
+	int timeout = 50;
+
+	while (timeout--) {
+		/* check IRV busy bit */
+		if (azx_readw(chip, IRS) & ICH6_IRS_VALID)
+			return azx_readl(chip, IR);
+		udelay(1);
+	}
+	snd_printd(SFX "get_response timeout: IRS=0x%x\n", azx_readw(chip, IRS));
+	return (unsigned int)-1;
+}
+
+#define azx_update_rirb(chip)
+
+#endif /* USE_CORB_RIRB */
+
+/* reset codec link */
+static int azx_reset(azx_t *chip)
+{
+	int count;
+
+	/* reset controller */
+	azx_writel(chip, GCTL, azx_readl(chip, GCTL) & ~ICH6_GCTL_RESET);
+
+	count = 50;
+	while (azx_readb(chip, GCTL) && --count)
+		msleep(1);
+
+	/* delay for >= 100us for codec PLL to settle per spec
+	 * Rev 0.9 section 5.5.1
+	 */
+	msleep(1);
+
+	/* Bring controller out of reset */
+	azx_writeb(chip, GCTL, azx_readb(chip, GCTL) | ICH6_GCTL_RESET);
+
+	count = 50;
+	while (! azx_readb(chip, GCTL) && --count)
+		msleep(1);
+
+	/* Brent Chartrand said to wait >= 540us for codecs to intialize */
+	msleep(1);
+
+	/* check to see if controller is ready */
+	if (! azx_readb(chip, GCTL)) {
+		snd_printd("azx_reset: controller not ready!\n");
+		return -EBUSY;
+	}
+
+	/* detect codecs */
+	if (! chip->codec_mask) {
+		chip->codec_mask = azx_readw(chip, STATESTS);
+		snd_printdd("codec_mask = 0x%x\n", chip->codec_mask);
+	}
+
+	return 0;
+}
+
+
+/*
+ * Lowlevel interface
+ */  
+
+/* enable interrupts */
+static void azx_int_enable(azx_t *chip)
+{
+	/* enable controller CIE and GIE */
+	azx_writel(chip, INTCTL, azx_readl(chip, INTCTL) |
+		   ICH6_INT_CTRL_EN | ICH6_INT_GLOBAL_EN);
+}
+
+/* disable interrupts */
+static void azx_int_disable(azx_t *chip)
+{
+	int i;
+
+	/* disable interrupts in stream descriptor */
+	for (i = 0; i < MAX_ICH6_DEV; i++) {
+		azx_dev_t *azx_dev = &chip->azx_dev[i];
+		azx_sd_writeb(azx_dev, SD_CTL,
+			      azx_sd_readb(azx_dev, SD_CTL) & ~SD_INT_MASK);
+	}
+
+	/* disable SIE for all streams */
+	azx_writeb(chip, INTCTL, 0);
+
+	/* disable controller CIE and GIE */
+	azx_writel(chip, INTCTL, azx_readl(chip, INTCTL) &
+		   ~(ICH6_INT_CTRL_EN | ICH6_INT_GLOBAL_EN));
+}
+
+/* clear interrupts */
+static void azx_int_clear(azx_t *chip)
+{
+	int i;
+
+	/* clear stream status */
+	for (i = 0; i < MAX_ICH6_DEV; i++) {
+		azx_dev_t *azx_dev = &chip->azx_dev[i];
+		azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK);
+	}
+
+	/* clear STATESTS */
+	azx_writeb(chip, STATESTS, STATESTS_INT_MASK);
+
+	/* clear rirb status */
+	azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
+
+	/* clear int status */
+	azx_writel(chip, INTSTS, ICH6_INT_CTRL_EN | ICH6_INT_ALL_STREAM);
+}
+
+/* start a stream */
+static void azx_stream_start(azx_t *chip, azx_dev_t *azx_dev)
+{
+	/* enable SIE */
+	azx_writeb(chip, INTCTL,
+		   azx_readb(chip, INTCTL) | (1 << azx_dev->index));
+	/* set DMA start and interrupt mask */
+	azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) |
+		      SD_CTL_DMA_START | SD_INT_MASK);
+}
+
+/* stop a stream */
+static void azx_stream_stop(azx_t *chip, azx_dev_t *azx_dev)
+{
+	/* stop DMA */
+	azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) &
+		      ~(SD_CTL_DMA_START | SD_INT_MASK));
+	azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */
+	/* disable SIE */
+	azx_writeb(chip, INTCTL,
+		   azx_readb(chip, INTCTL) & ~(1 << azx_dev->index));
+}
+
+
+/*
+ * initialize the chip
+ */
+static void azx_init_chip(azx_t *chip)
+{
+	unsigned char tcsel_reg;
+
+	/* Clear bits 0-2 of PCI register TCSEL (at offset 0x44)
+	 * TCSEL == Traffic Class Select Register, which sets PCI express QOS
+	 * Ensuring these bits are 0 clears playback static on some HD Audio codecs
+	 */
+	pci_read_config_byte (chip->pci, ICH6_PCIREG_TCSEL, &tcsel_reg);
+	pci_write_config_byte(chip->pci, ICH6_PCIREG_TCSEL, tcsel_reg & 0xf8);
+
+	/* reset controller */
+	azx_reset(chip);
+
+	/* initialize interrupts */
+	azx_int_clear(chip);
+	azx_int_enable(chip);
+
+	/* initialize the codec command I/O */
+	azx_init_cmd_io(chip);
+
+#ifdef USE_POSBUF
+	/* program the position buffer */
+	azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr);
+	azx_writel(chip, DPUBASE, upper_32bit(chip->posbuf.addr));
+#endif
+}
+
+
+/*
+ * interrupt handler
+ */
+static irqreturn_t azx_interrupt(int irq, void* dev_id, struct pt_regs *regs)
+{
+	azx_t *chip = dev_id;
+	azx_dev_t *azx_dev;
+	u32 status;
+	int i;
+
+	spin_lock(&chip->reg_lock);
+
+	status = azx_readl(chip, INTSTS);
+	if (status == 0) {
+		spin_unlock(&chip->reg_lock);
+		return IRQ_NONE;
+	}
+	
+	for (i = 0; i < MAX_ICH6_DEV; i++) {
+		azx_dev = &chip->azx_dev[i];
+		if (status & azx_dev->sd_int_sta_mask) {
+			azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK);
+			if (azx_dev->substream && azx_dev->running) {
+				spin_unlock(&chip->reg_lock);
+				snd_pcm_period_elapsed(azx_dev->substream);
+				spin_lock(&chip->reg_lock);
+			}
+		}
+	}
+
+	/* clear rirb int */
+	status = azx_readb(chip, RIRBSTS);
+	if (status & RIRB_INT_MASK) {
+		if (status & RIRB_INT_RESPONSE)
+			azx_update_rirb(chip);
+		azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
+	}
+
+#if 0
+	/* clear state status int */
+	if (azx_readb(chip, STATESTS) & 0x04)
+		azx_writeb(chip, STATESTS, 0x04);
+#endif
+	spin_unlock(&chip->reg_lock);
+	
+	return IRQ_HANDLED;
+}
+
+
+/*
+ * set up BDL entries
+ */
+static void azx_setup_periods(azx_dev_t *azx_dev)
+{
+	u32 *bdl = azx_dev->bdl;
+	dma_addr_t dma_addr = azx_dev->substream->runtime->dma_addr;
+	int idx;
+
+	/* reset BDL address */
+	azx_sd_writel(azx_dev, SD_BDLPL, 0);
+	azx_sd_writel(azx_dev, SD_BDLPU, 0);
+
+	/* program the initial BDL entries */
+	for (idx = 0; idx < azx_dev->frags; idx++) {
+		unsigned int off = idx << 2; /* 4 dword step */
+		dma_addr_t addr = dma_addr + idx * azx_dev->fragsize;
+		/* program the address field of the BDL entry */
+		bdl[off] = cpu_to_le32((u32)addr);
+		bdl[off+1] = cpu_to_le32(upper_32bit(addr));
+
+		/* program the size field of the BDL entry */
+		bdl[off+2] = cpu_to_le32(azx_dev->fragsize);
+
+		/* program the IOC to enable interrupt when buffer completes */
+		bdl[off+3] = cpu_to_le32(0x01);
+	}
+}
+
+/*
+ * set up the SD for streaming
+ */
+static int azx_setup_controller(azx_t *chip, azx_dev_t *azx_dev)
+{
+	unsigned char val;
+	int timeout;
+
+	/* make sure the run bit is zero for SD */
+	azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) & ~SD_CTL_DMA_START);
+	/* reset stream */
+	azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) | SD_CTL_STREAM_RESET);
+	udelay(3);
+	timeout = 300;
+	while (!((val = azx_sd_readb(azx_dev, SD_CTL)) & SD_CTL_STREAM_RESET) &&
+	       --timeout)
+		;
+	val &= ~SD_CTL_STREAM_RESET;
+	azx_sd_writeb(azx_dev, SD_CTL, val);
+	udelay(3);
+
+	timeout = 300;
+	/* waiting for hardware to report that the stream is out of reset */
+	while (((val = azx_sd_readb(azx_dev, SD_CTL)) & SD_CTL_STREAM_RESET) &&
+	       --timeout)
+		;
+
+	/* program the stream_tag */
+	azx_sd_writel(azx_dev, SD_CTL,
+		      (azx_sd_readl(azx_dev, SD_CTL) & ~SD_CTL_STREAM_TAG_MASK) |
+		      (azx_dev->stream_tag << SD_CTL_STREAM_TAG_SHIFT));
+
+	/* program the length of samples in cyclic buffer */
+	azx_sd_writel(azx_dev, SD_CBL, azx_dev->bufsize);
+
+	/* program the stream format */
+	/* this value needs to be the same as the one programmed */
+	azx_sd_writew(azx_dev, SD_FORMAT, azx_dev->format_val);
+
+	/* program the stream LVI (last valid index) of the BDL */
+	azx_sd_writew(azx_dev, SD_LVI, azx_dev->frags - 1);
+
+	/* program the BDL address */
+	/* lower BDL address */
+	azx_sd_writel(azx_dev, SD_BDLPL, (u32)azx_dev->bdl_addr);
+	/* upper BDL address */
+	azx_sd_writel(azx_dev, SD_BDLPU, upper_32bit(azx_dev->bdl_addr));
+
+#ifdef USE_POSBUF
+	/* enable the position buffer */
+	if (! (azx_readl(chip, DPLBASE) & ICH6_DPLBASE_ENABLE))
+		azx_writel(chip, DPLBASE, (u32)chip->posbuf.addr | ICH6_DPLBASE_ENABLE);
+#endif
+	/* set the interrupt enable bits in the descriptor control register */
+	azx_sd_writel(azx_dev, SD_CTL, azx_sd_readl(azx_dev, SD_CTL) | SD_INT_MASK);
+
+	return 0;
+}
+
+
+/*
+ * Codec initialization
+ */
+
+static int __devinit azx_codec_create(azx_t *chip, const char *model)
+{
+	struct hda_bus_template bus_temp;
+	int c, codecs, err;
+
+	memset(&bus_temp, 0, sizeof(bus_temp));
+	bus_temp.private_data = chip;
+	bus_temp.modelname = model;
+	bus_temp.pci = chip->pci;
+	bus_temp.ops.command = azx_send_cmd;
+	bus_temp.ops.get_response = azx_get_response;
+
+	if ((err = snd_hda_bus_new(chip->card, &bus_temp, &chip->bus)) < 0)
+		return err;
+
+	codecs = 0;
+	for (c = 0; c < AZX_MAX_CODECS; c++) {
+		if (chip->codec_mask & (1 << c)) {
+			err = snd_hda_codec_new(chip->bus, c, NULL);
+			if (err < 0)
+				continue;
+			codecs++;
+		}
+	}
+	if (! codecs) {
+		snd_printk(KERN_ERR SFX "no codecs initialized\n");
+		return -ENXIO;
+	}
+
+	return 0;
+}
+
+
+/*
+ * PCM support
+ */
+
+/* assign a stream for the PCM */
+static inline azx_dev_t *azx_assign_device(azx_t *chip, int stream)
+{
+	int dev, i;
+	dev = stream == SNDRV_PCM_STREAM_PLAYBACK ? 4 : 0;
+	for (i = 0; i < 4; i++, dev++)
+		if (! chip->azx_dev[dev].opened) {
+			chip->azx_dev[dev].opened = 1;
+			return &chip->azx_dev[dev];
+		}
+	return NULL;
+}
+
+/* release the assigned stream */
+static inline void azx_release_device(azx_dev_t *azx_dev)
+{
+	azx_dev->opened = 0;
+}
+
+static snd_pcm_hardware_t azx_pcm_hw = {
+	.info =			(SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED |
+				 SNDRV_PCM_INFO_BLOCK_TRANSFER |
+				 SNDRV_PCM_INFO_MMAP_VALID |
+				 SNDRV_PCM_INFO_PAUSE |
+				 SNDRV_PCM_INFO_RESUME),
+	.formats =		SNDRV_PCM_FMTBIT_S16_LE,
+	.rates =		SNDRV_PCM_RATE_48000,
+	.rate_min =		48000,
+	.rate_max =		48000,
+	.channels_min =		2,
+	.channels_max =		2,
+	.buffer_bytes_max =	AZX_MAX_BUF_SIZE,
+	.period_bytes_min =	128,
+	.period_bytes_max =	AZX_MAX_BUF_SIZE / 2,
+	.periods_min =		2,
+	.periods_max =		AZX_MAX_FRAG,
+	.fifo_size =		0,
+};
+
+struct azx_pcm {
+	azx_t *chip;
+	struct hda_codec *codec;
+	struct hda_pcm_stream *hinfo[2];
+};
+
+static int azx_pcm_open(snd_pcm_substream_t *substream)
+{
+	struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
+	struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream];
+	azx_t *chip = apcm->chip;
+	azx_dev_t *azx_dev;
+	snd_pcm_runtime_t *runtime = substream->runtime;
+	unsigned long flags;
+	int err;
+
+	down(&chip->open_mutex);
+	azx_dev = azx_assign_device(chip, substream->stream);
+	if (azx_dev == NULL) {
+		up(&chip->open_mutex);
+		return -EBUSY;
+	}
+	runtime->hw = azx_pcm_hw;
+	runtime->hw.channels_min = hinfo->channels_min;
+	runtime->hw.channels_max = hinfo->channels_max;
+	runtime->hw.formats = hinfo->formats;
+	runtime->hw.rates = hinfo->rates;
+	snd_pcm_limit_hw_rates(runtime);
+	snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
+	if ((err = hinfo->ops.open(hinfo, apcm->codec, substream)) < 0) {
+		azx_release_device(azx_dev);
+		up(&chip->open_mutex);
+		return err;
+	}
+	spin_lock_irqsave(&chip->reg_lock, flags);
+	azx_dev->substream = substream;
+	azx_dev->running = 0;
+	spin_unlock_irqrestore(&chip->reg_lock, flags);
+
+	runtime->private_data = azx_dev;
+	up(&chip->open_mutex);
+	return 0;
+}
+
+static int azx_pcm_close(snd_pcm_substream_t *substream)
+{
+	struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
+	struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream];
+	azx_t *chip = apcm->chip;
+	azx_dev_t *azx_dev = get_azx_dev(substream);
+	unsigned long flags;
+
+	down(&chip->open_mutex);
+	spin_lock_irqsave(&chip->reg_lock, flags);
+	azx_dev->substream = NULL;
+	azx_dev->running = 0;
+	spin_unlock_irqrestore(&chip->reg_lock, flags);
+	azx_release_device(azx_dev);
+	hinfo->ops.close(hinfo, apcm->codec, substream);
+	up(&chip->open_mutex);
+	return 0;
+}
+
+static int azx_pcm_hw_params(snd_pcm_substream_t *substream, snd_pcm_hw_params_t *hw_params)
+{
+	return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+}
+
+static int azx_pcm_hw_free(snd_pcm_substream_t *substream)
+{
+	struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
+	azx_dev_t *azx_dev = get_azx_dev(substream);
+	struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream];
+
+	/* reset BDL address */
+	azx_sd_writel(azx_dev, SD_BDLPL, 0);
+	azx_sd_writel(azx_dev, SD_BDLPU, 0);
+	azx_sd_writel(azx_dev, SD_CTL, 0);
+
+	hinfo->ops.cleanup(hinfo, apcm->codec, substream);
+
+	return snd_pcm_lib_free_pages(substream);
+}
+
+static int azx_pcm_prepare(snd_pcm_substream_t *substream)
+{
+	struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
+	azx_t *chip = apcm->chip;
+	azx_dev_t *azx_dev = get_azx_dev(substream);
+	struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream];
+	snd_pcm_runtime_t *runtime = substream->runtime;
+
+	azx_dev->bufsize = snd_pcm_lib_buffer_bytes(substream);
+	azx_dev->fragsize = snd_pcm_lib_period_bytes(substream);
+	azx_dev->frags = azx_dev->bufsize / azx_dev->fragsize;
+	azx_dev->format_val = snd_hda_calc_stream_format(runtime->rate,
+							 runtime->channels,
+							 runtime->format,
+							 hinfo->maxbps);
+	if (! azx_dev->format_val) {
+		snd_printk(KERN_ERR SFX "invalid format_val, rate=%d, ch=%d, format=%d\n",
+			   runtime->rate, runtime->channels, runtime->format);
+		return -EINVAL;
+	}
+
+	snd_printdd("azx_pcm_prepare: bufsize=0x%x, fragsize=0x%x, format=0x%x\n",
+		    azx_dev->bufsize, azx_dev->fragsize, azx_dev->format_val);
+	azx_setup_periods(azx_dev);
+	azx_setup_controller(chip, azx_dev);
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		azx_dev->fifo_size = azx_sd_readw(azx_dev, SD_FIFOSIZE) + 1;
+	else
+		azx_dev->fifo_size = 0;
+
+	return hinfo->ops.prepare(hinfo, apcm->codec, azx_dev->stream_tag,
+				  azx_dev->format_val, substream);
+}
+
+static int azx_pcm_trigger(snd_pcm_substream_t *substream, int cmd)
+{
+	struct azx_pcm *apcm = snd_pcm_substream_chip(substream);
+	azx_dev_t *azx_dev = get_azx_dev(substream);
+	azx_t *chip = apcm->chip;
+	int err = 0;
+
+	spin_lock(&chip->reg_lock);
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_START:
+		azx_stream_start(chip, azx_dev);
+		azx_dev->running = 1;
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+	case SNDRV_PCM_TRIGGER_STOP:
+		azx_stream_stop(chip, azx_dev);
+		azx_dev->running = 0;
+		break;
+	default:
+		err = -EINVAL;
+	}
+	spin_unlock(&chip->reg_lock);
+	if (cmd == SNDRV_PCM_TRIGGER_PAUSE_PUSH ||
+	    cmd == SNDRV_PCM_TRIGGER_STOP) {
+		int timeout = 5000;
+		while (azx_sd_readb(azx_dev, SD_CTL) & SD_CTL_DMA_START && --timeout)
+			;
+	}
+	return err;
+}
+
+static snd_pcm_uframes_t azx_pcm_pointer(snd_pcm_substream_t *substream)
+{
+	azx_dev_t *azx_dev = get_azx_dev(substream);
+	unsigned int pos;
+
+#ifdef USE_POSBUF
+	/* use the position buffer */
+	pos = *azx_dev->posbuf;
+#else
+	/* read LPIB */
+	pos = azx_sd_readl(azx_dev, SD_LPIB) + azx_dev->fifo_size;
+#endif
+	if (pos >= azx_dev->bufsize)
+		pos = 0;
+	return bytes_to_frames(substream->runtime, pos);
+}
+
+static snd_pcm_ops_t azx_pcm_ops = {
+	.open = azx_pcm_open,
+	.close = azx_pcm_close,
+	.ioctl = snd_pcm_lib_ioctl,
+	.hw_params = azx_pcm_hw_params,
+	.hw_free = azx_pcm_hw_free,
+	.prepare = azx_pcm_prepare,
+	.trigger = azx_pcm_trigger,
+	.pointer = azx_pcm_pointer,
+};
+
+static void azx_pcm_free(snd_pcm_t *pcm)
+{
+	kfree(pcm->private_data);
+}
+
+static int __devinit create_codec_pcm(azx_t *chip, struct hda_codec *codec,
+				      struct hda_pcm *cpcm, int pcm_dev)
+{
+	int err;
+	snd_pcm_t *pcm;
+	struct azx_pcm *apcm;
+
+	snd_assert(cpcm->stream[0].substreams || cpcm->stream[1].substreams, return -EINVAL);
+	snd_assert(cpcm->name, return -EINVAL);
+
+	err = snd_pcm_new(chip->card, cpcm->name, pcm_dev,
+			  cpcm->stream[0].substreams, cpcm->stream[1].substreams,
+			  &pcm);
+	if (err < 0)
+		return err;
+	strcpy(pcm->name, cpcm->name);
+	apcm = kmalloc(sizeof(*apcm), GFP_KERNEL);
+	if (apcm == NULL)
+		return -ENOMEM;
+	apcm->chip = chip;
+	apcm->codec = codec;
+	apcm->hinfo[0] = &cpcm->stream[0];
+	apcm->hinfo[1] = &cpcm->stream[1];
+	pcm->private_data = apcm;
+	pcm->private_free = azx_pcm_free;
+	if (cpcm->stream[0].substreams)
+		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &azx_pcm_ops);
+	if (cpcm->stream[1].substreams)
+		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &azx_pcm_ops);
+	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+					      snd_dma_pci_data(chip->pci),
+					      1024 * 64, 1024 * 128);
+	chip->pcm[pcm_dev] = pcm;
+
+	return 0;
+}
+
+static int __devinit azx_pcm_create(azx_t *chip)
+{
+	struct list_head *p;
+	struct hda_codec *codec;
+	int c, err;
+	int pcm_dev;
+
+	if ((err = snd_hda_build_pcms(chip->bus)) < 0)
+		return err;
+
+	pcm_dev = 0;
+	list_for_each(p, &chip->bus->codec_list) {
+		codec = list_entry(p, struct hda_codec, list);
+		for (c = 0; c < codec->num_pcms; c++) {
+			if (pcm_dev >= AZX_MAX_PCMS) {
+				snd_printk(KERN_ERR SFX "Too many PCMs\n");
+				return -EINVAL;
+			}
+			err = create_codec_pcm(chip, codec, &codec->pcm_info[c], pcm_dev);
+			if (err < 0)
+				return err;
+			pcm_dev++;
+		}
+	}
+	return 0;
+}
+
+/*
+ * mixer creation - all stuff is implemented in hda module
+ */
+static int __devinit azx_mixer_create(azx_t *chip)
+{
+	return snd_hda_build_controls(chip->bus);
+}
+
+
+/*
+ * initialize SD streams
+ */
+static int __devinit azx_init_stream(azx_t *chip)
+{
+	int i;
+
+	/* initialize each stream (aka device)
+	 * assign the starting bdl address to each stream (device) and initialize
+	 */
+	for (i = 0; i < MAX_ICH6_DEV; i++) {
+		unsigned int off = sizeof(u32) * (i * AZX_MAX_FRAG * 4);
+		azx_dev_t *azx_dev = &chip->azx_dev[i];
+		azx_dev->bdl = (u32 *)(chip->bdl.area + off);
+		azx_dev->bdl_addr = chip->bdl.addr + off;
+#ifdef USE_POSBUF
+		azx_dev->posbuf = (volatile u32 *)(chip->posbuf.area + i * 8);
+#endif
+		/* offset: SDI0=0x80, SDI1=0xa0, ... SDO3=0x160 */
+		azx_dev->sd_addr = chip->remap_addr + (0x20 * i + 0x80);
+		/* int mask: SDI0=0x01, SDI1=0x02, ... SDO3=0x80 */
+		azx_dev->sd_int_sta_mask = 1 << i;
+		/* stream tag: must be non-zero and unique */
+		azx_dev->index = i;
+		azx_dev->stream_tag = i + 1;
+	}
+
+	return 0;
+}
+
+
+#ifdef CONFIG_PM
+/*
+ * power management
+ */
+static int azx_suspend(snd_card_t *card, pm_message_t state)
+{
+	azx_t *chip = card->pm_private_data;
+	int i;
+
+	for (i = 0; i < chip->pcm_devs; i++)
+		if (chip->pcm[i])
+			snd_pcm_suspend_all(chip->pcm[i]);
+	snd_hda_suspend(chip->bus, state);
+	azx_free_cmd_io(chip);
+	pci_disable_device(chip->pci);
+	return 0;
+}
+
+static int azx_resume(snd_card_t *card)
+{
+	azx_t *chip = card->pm_private_data;
+
+	pci_enable_device(chip->pci);
+	pci_set_master(chip->pci);
+	azx_init_chip(chip);
+	snd_hda_resume(chip->bus);
+	return 0;
+}
+#endif /* CONFIG_PM */
+
+
+/*
+ * destructor
+ */
+static int azx_free(azx_t *chip)
+{
+	if (chip->remap_addr) {
+		int i;
+
+		for (i = 0; i < MAX_ICH6_DEV; i++)
+			azx_stream_stop(chip, &chip->azx_dev[i]);
+
+		/* disable interrupts */
+		azx_int_disable(chip);
+		azx_int_clear(chip);
+
+		/* disable CORB/RIRB */
+		azx_free_cmd_io(chip);
+
+		/* disable position buffer */
+		azx_writel(chip, DPLBASE, 0);
+		azx_writel(chip, DPUBASE, 0);
+
+		/* wait a little for interrupts to finish */
+		msleep(1);
+
+		iounmap(chip->remap_addr);
+	}
+
+	if (chip->irq >= 0)
+		free_irq(chip->irq, (void*)chip);
+
+	if (chip->bdl.area)
+		snd_dma_free_pages(&chip->bdl);
+	if (chip->rb.area)
+		snd_dma_free_pages(&chip->rb);
+#ifdef USE_POSBUF
+	if (chip->posbuf.area)
+		snd_dma_free_pages(&chip->posbuf);
+#endif
+	pci_release_regions(chip->pci);
+	pci_disable_device(chip->pci);
+	kfree(chip);
+
+	return 0;
+}
+
+static int azx_dev_free(snd_device_t *device)
+{
+	return azx_free(device->device_data);
+}
+
+/*
+ * constructor
+ */
+static int __devinit azx_create(snd_card_t *card, struct pci_dev *pci, azx_t **rchip)
+{
+	azx_t *chip;
+	int err = 0;
+	static snd_device_ops_t ops = {
+		.dev_free = azx_dev_free,
+	};
+
+	*rchip = NULL;
+	
+	if ((err = pci_enable_device(pci)) < 0)
+		return err;
+
+	chip = kcalloc(1, sizeof(*chip), GFP_KERNEL);
+	
+	if (NULL == chip) {
+		snd_printk(KERN_ERR SFX "cannot allocate chip\n");
+		pci_disable_device(pci);
+		return -ENOMEM;
+	}
+
+	spin_lock_init(&chip->reg_lock);
+	init_MUTEX(&chip->open_mutex);
+	chip->card = card;
+	chip->pci = pci;
+	chip->irq = -1;
+
+	if ((err = pci_request_regions(pci, "ICH HD audio")) < 0) {
+		kfree(chip);
+		pci_disable_device(pci);
+		return err;
+	}
+
+	chip->addr = pci_resource_start(pci,0);
+	chip->remap_addr = ioremap_nocache(chip->addr, pci_resource_len(pci,0));
+	if (chip->remap_addr == NULL) {
+		snd_printk(KERN_ERR SFX "ioremap error\n");
+		err = -ENXIO;
+		goto errout;
+	}
+
+	if (request_irq(pci->irq, azx_interrupt, SA_INTERRUPT|SA_SHIRQ,
+			"HDA Intel", (void*)chip)) {
+		snd_printk(KERN_ERR SFX "unable to grab IRQ %d\n", pci->irq);
+		err = -EBUSY;
+		goto errout;
+	}
+	chip->irq = pci->irq;
+
+	pci_set_master(pci);
+	synchronize_irq(chip->irq);
+
+	/* allocate memory for the BDL for each stream */
+	if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
+				       PAGE_SIZE, &chip->bdl)) < 0) {
+		snd_printk(KERN_ERR SFX "cannot allocate BDL\n");
+		goto errout;
+	}
+#ifdef USE_POSBUF
+	/* allocate memory for the position buffer */
+	if ((err = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, snd_dma_pci_data(chip->pci),
+				       MAX_ICH6_DEV * 8, &chip->posbuf)) < 0) {
+		snd_printk(KERN_ERR SFX "cannot allocate posbuf\n");
+		goto errout;
+	}
+#endif
+	/* allocate CORB/RIRB */
+	if ((err = azx_alloc_cmd_io(chip)) < 0)
+		goto errout;
+
+	/* initialize streams */
+	azx_init_stream(chip);
+
+	/* initialize chip */
+	azx_init_chip(chip);
+
+	/* codec detection */
+	if (! chip->codec_mask) {
+		snd_printk(KERN_ERR SFX "no codecs found!\n");
+		err = -ENODEV;
+		goto errout;
+	}
+
+	if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) <0) {
+		snd_printk(KERN_ERR SFX "Error creating device [card]!\n");
+		goto errout;
+	}
+
+	*rchip = chip;
+	return 0;
+
+ errout:
+	azx_free(chip);
+	return err;
+}
+
+static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
+{
+	static int dev;
+	snd_card_t *card;
+	azx_t *chip;
+	int err = 0;
+
+	if (dev >= SNDRV_CARDS)
+		return -ENODEV;
+	if (! enable[dev]) {
+		dev++;
+		return -ENOENT;
+	}
+
+	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+	if (NULL == card) {
+		snd_printk(KERN_ERR SFX "Error creating card!\n");
+		return -ENOMEM;
+	}
+
+	if ((err = azx_create(card, pci, &chip)) < 0) {
+		snd_card_free(card);
+		return err;
+	}
+
+	strcpy(card->driver, "HDA-Intel");
+	strcpy(card->shortname, "HDA Intel");
+	sprintf(card->longname, "%s at 0x%lx irq %i", card->shortname, chip->addr, chip->irq);
+
+	/* create codec instances */
+	if ((err = azx_codec_create(chip, model[dev])) < 0) {
+		snd_card_free(card);
+		return err;
+	}
+
+	/* create PCM streams */
+	if ((err = azx_pcm_create(chip)) < 0) {
+		snd_card_free(card);
+		return err;
+	}
+
+	/* create mixer controls */
+	if ((err = azx_mixer_create(chip)) < 0) {
+		snd_card_free(card);
+		return err;
+	}
+
+	snd_card_set_pm_callback(card, azx_suspend, azx_resume, chip);
+	snd_card_set_dev(card, &pci->dev);
+
+	if ((err = snd_card_register(card)) < 0) {
+		snd_card_free(card);
+		return err;
+	}
+
+	pci_set_drvdata(pci, card);
+	dev++;
+
+	return err;
+}
+
+static void __devexit azx_remove(struct pci_dev *pci)
+{
+	snd_card_free(pci_get_drvdata(pci));
+	pci_set_drvdata(pci, NULL);
+}
+
+/* PCI IDs */
+static struct pci_device_id azx_ids[] = {
+	{ 0x8086, 0x2668, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ICH6 */
+	{ 0x8086, 0x27d8, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0 }, /* ICH7 */
+	{ 0, }
+};
+MODULE_DEVICE_TABLE(pci, azx_ids);
+
+/* pci_driver definition */
+static struct pci_driver driver = {
+	.name = "HDA Intel",
+	.id_table = azx_ids,
+	.probe = azx_probe,
+	.remove = __devexit_p(azx_remove),
+	SND_PCI_PM_CALLBACKS
+};
+
+static int __init alsa_card_azx_init(void)
+{
+	return pci_module_init(&driver);
+}
+
+static void __exit alsa_card_azx_exit(void)
+{
+	pci_unregister_driver(&driver);
+}
+
+module_init(alsa_card_azx_init)
+module_exit(alsa_card_azx_exit)
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
new file mode 100644
index 000000000000..7c7b849875a0
--- /dev/null
+++ b/sound/pci/hda/hda_local.h
@@ -0,0 +1,161 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * Local helper functions
+ *
+ * Copyright (c) 2004 Takashi Iwai <tiwai@suse.de>
+ *
+ *  This program is free software; you can redistribute it and/or modify it
+ *  under the terms of the GNU General Public License as published by the Free
+ *  Software Foundation; either version 2 of the License, or (at your option)
+ *  any later version.
+ *
+ *  This program is distributed in the hope that it will be useful, but WITHOUT
+ *  ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or
+ *  FITNESS FOR A PARTICULAR PURPOSE.  See the GNU General Public License for
+ *  more details.
+ *
+ *  You should have received a copy of the GNU General Public License along with
+ *  this program; if not, write to the Free Software Foundation, Inc., 59
+ *  Temple Place - Suite 330, Boston, MA  02111-1307, USA.
+ */
+
+#ifndef __SOUND_HDA_LOCAL_H
+#define __SOUND_HDA_LOCAL_H
+
+/*
+ * for mixer controls
+ */
+#define HDA_COMPOSE_AMP_VAL(nid,chs,idx,dir) ((nid) | ((chs)<<16) | ((dir)<<18) | ((idx)<<19))
+#define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \
+	{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx,  \
+	  .info = snd_hda_mixer_amp_volume_info, \
+	  .get = snd_hda_mixer_amp_volume_get, \
+	  .put = snd_hda_mixer_amp_volume_put, \
+	  .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) }
+#define HDA_CODEC_VOLUME_IDX(xname, xcidx, nid, xindex, direction) \
+	HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, 3, xindex, direction)
+#define HDA_CODEC_VOLUME_MONO(xname, nid, channel, xindex, direction) \
+	HDA_CODEC_VOLUME_MONO_IDX(xname, 0, nid, channel, xindex, direction)
+#define HDA_CODEC_VOLUME(xname, nid, xindex, direction) \
+	HDA_CODEC_VOLUME_MONO(xname, nid, 3, xindex, direction)
+#define HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \
+	{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \
+	  .info = snd_hda_mixer_amp_switch_info, \
+	  .get = snd_hda_mixer_amp_switch_get, \
+	  .put = snd_hda_mixer_amp_switch_put, \
+	  .private_value = HDA_COMPOSE_AMP_VAL(nid, channel, xindex, direction) }
+#define HDA_CODEC_MUTE_IDX(xname, xcidx, nid, xindex, direction) \
+	HDA_CODEC_MUTE_MONO_IDX(xname, xcidx, nid, 3, xindex, direction)
+#define HDA_CODEC_MUTE_MONO(xname, nid, channel, xindex, direction) \
+	HDA_CODEC_MUTE_MONO_IDX(xname, 0, nid, channel, xindex, direction)
+#define HDA_CODEC_MUTE(xname, nid, xindex, direction) \
+	HDA_CODEC_MUTE_MONO(xname, nid, 3, xindex, direction)
+
+int snd_hda_mixer_amp_volume_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo);
+int snd_hda_mixer_amp_volume_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol);
+int snd_hda_mixer_amp_volume_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol);
+int snd_hda_mixer_amp_switch_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo);
+int snd_hda_mixer_amp_switch_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol);
+int snd_hda_mixer_amp_switch_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol);
+
+int snd_hda_create_spdif_out_ctls(struct hda_codec *codec, hda_nid_t nid);
+int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid);
+
+/*
+ * input MUX helper
+ */
+#define HDA_MAX_NUM_INPUTS	8
+struct hda_input_mux_item {
+	const char *label;
+	unsigned int index;
+};
+struct hda_input_mux {
+	unsigned int num_items;
+	struct hda_input_mux_item items[HDA_MAX_NUM_INPUTS];
+};
+
+int snd_hda_input_mux_info(const struct hda_input_mux *imux, snd_ctl_elem_info_t *uinfo);
+int snd_hda_input_mux_put(struct hda_codec *codec, const struct hda_input_mux *imux,
+			  snd_ctl_elem_value_t *ucontrol, hda_nid_t nid,
+			  unsigned int *cur_val);
+
+/*
+ * Multi-channel / digital-out PCM helper
+ */
+
+enum { HDA_FRONT, HDA_REAR, HDA_CLFE, HDA_SIDE }; /* index for dac_nidx */
+enum { HDA_DIG_NONE, HDA_DIG_EXCLUSIVE, HDA_DIG_ANALOG_DUP }; /* dig_out_used */
+
+struct hda_multi_out {
+	int num_dacs;		/* # of DACs, must be more than 1 */
+	hda_nid_t *dac_nids;	/* DAC list */
+	hda_nid_t hp_nid;	/* optional DAC for HP, 0 when not exists */
+	hda_nid_t dig_out_nid;	/* digital out audio widget */
+	int max_channels;	/* currently supported analog channels */
+	int dig_out_used;	/* current usage of digital out (HDA_DIG_XXX) */
+};
+
+int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mout);
+int snd_hda_multi_out_dig_close(struct hda_codec *codec, struct hda_multi_out *mout);
+int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout,
+				  snd_pcm_substream_t *substream);
+int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_out *mout,
+				     unsigned int stream_tag,
+				     unsigned int format,
+				     snd_pcm_substream_t *substream);
+int snd_hda_multi_out_analog_cleanup(struct hda_codec *codec, struct hda_multi_out *mout);
+
+/*
+ * generic codec parser
+ */
+int snd_hda_parse_generic_codec(struct hda_codec *codec);
+
+/*
+ * generic proc interface
+ */
+#ifdef CONFIG_PROC_FS
+int snd_hda_codec_proc_new(struct hda_codec *codec);
+#else
+static inline int snd_hda_codec_proc_new(struct hda_codec *codec) { return 0; }
+#endif
+
+/*
+ * Misc
+ */
+struct hda_board_config {
+	const char *modelname;
+	int config;
+	unsigned short pci_vendor;
+	unsigned short pci_device;
+};
+
+int snd_hda_check_board_config(struct hda_codec *codec, struct hda_board_config *tbl);
+int snd_hda_add_new_ctls(struct hda_codec *codec, snd_kcontrol_new_t *knew);
+
+/*
+ * power management
+ */
+#ifdef CONFIG_PM
+int snd_hda_resume_ctls(struct hda_codec *codec, snd_kcontrol_new_t *knew);
+int snd_hda_resume_spdif_out(struct hda_codec *codec);
+int snd_hda_resume_spdif_in(struct hda_codec *codec);
+#endif
+
+/*
+ * unsolicited event handler
+ */
+
+#define HDA_UNSOL_QUEUE_SIZE	64
+
+struct hda_bus_unsolicited {
+	/* ring buffer */
+	u32 queue[HDA_UNSOL_QUEUE_SIZE * 2];
+	unsigned int rp, wp;
+
+	/* workqueue */
+	struct workqueue_struct *workq;
+	struct work_struct work;
+};
+
+#endif /* __SOUND_HDA_LOCAL_H */
diff --git a/sound/pci/hda/hda_patch.h b/sound/pci/hda/hda_patch.h
new file mode 100644
index 000000000000..cf6abce42bc9
--- /dev/null
+++ b/sound/pci/hda/hda_patch.h
@@ -0,0 +1,17 @@
+/*
+ * HDA Patches - included by hda_codec.c
+ */
+
+/* Realtek codecs */
+extern struct hda_codec_preset snd_hda_preset_realtek[];
+/* C-Media codecs */
+extern struct hda_codec_preset snd_hda_preset_cmedia[];
+/* Analog Devices codecs */
+extern struct hda_codec_preset snd_hda_preset_analog[];
+
+static const struct hda_codec_preset *hda_preset_tables[] = {
+	snd_hda_preset_realtek,
+	snd_hda_preset_cmedia,
+	snd_hda_preset_analog,
+	NULL
+};
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
new file mode 100644
index 000000000000..4d5db7faad8d
--- /dev/null
+++ b/sound/pci/hda/hda_proc.c
@@ -0,0 +1,298 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ * 
+ * Generic proc interface
+ *
+ * Copyright (c) 2004 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 <sound/driver.h>
+#include <linux/init.h>
+#include <linux/pci.h>
+#include <sound/core.h>
+#include "hda_codec.h"
+
+static const char *get_wid_type_name(unsigned int wid_value)
+{
+	static char *names[16] = {
+		[AC_WID_AUD_OUT] = "Audio Output",
+		[AC_WID_AUD_IN] = "Audio Input",
+		[AC_WID_AUD_MIX] = "Audio Mixer",
+		[AC_WID_AUD_SEL] = "Audio Selector",
+		[AC_WID_PIN] = "Pin Complex",
+		[AC_WID_POWER] = "Power Widget",
+		[AC_WID_VOL_KNB] = "Volume Knob Widget",
+		[AC_WID_BEEP] = "Beep Generator Widget",
+		[AC_WID_VENDOR] = "Vendor Defined Widget",
+	};
+	wid_value &= 0xf;
+	if (names[wid_value])
+		return names[wid_value];
+	else
+		return "UNKOWN Widget";
+}
+
+static void print_amp_caps(snd_info_buffer_t *buffer,
+			   struct hda_codec *codec, hda_nid_t nid, int dir)
+{
+	unsigned int caps;
+	if (dir == HDA_OUTPUT)
+		caps = snd_hda_param_read(codec, nid, AC_PAR_AMP_OUT_CAP);
+	else
+		caps = snd_hda_param_read(codec, nid, AC_PAR_AMP_IN_CAP);
+	if (caps == -1 || caps == 0) {
+		snd_iprintf(buffer, "N/A\n");
+		return;
+	}
+	snd_iprintf(buffer, "ofs=0x%02x, nsteps=0x%02x, stepsize=0x%02x, mute=%x\n",
+		    caps & AC_AMPCAP_OFFSET,
+		    (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT,
+		    (caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT,
+		    (caps & AC_AMPCAP_MUTE) >> AC_AMPCAP_MUTE_SHIFT);
+}
+
+static void print_amp_vals(snd_info_buffer_t *buffer,
+			   struct hda_codec *codec, hda_nid_t nid,
+			   int dir, int stereo)
+{
+	unsigned int val;
+	if (stereo) {
+		val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_AMP_GAIN_MUTE,
+					  AC_AMP_GET_LEFT |
+					 (dir == HDA_OUTPUT ? AC_AMP_GET_OUTPUT :
+					  AC_AMP_GET_INPUT));
+		snd_iprintf(buffer, "0x%02x ", val);
+	}
+	val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_AMP_GAIN_MUTE,
+				 AC_AMP_GET_RIGHT |
+				 (dir == HDA_OUTPUT ? AC_AMP_GET_OUTPUT :
+				  AC_AMP_GET_INPUT));
+	snd_iprintf(buffer, "0x%02x\n", val);
+}
+
+static void print_pcm_caps(snd_info_buffer_t *buffer,
+			   struct hda_codec *codec, hda_nid_t nid)
+{
+	unsigned int pcm = snd_hda_param_read(codec, nid, AC_PAR_PCM);
+	unsigned int stream = snd_hda_param_read(codec, nid, AC_PAR_STREAM);
+	if (pcm == -1 || stream == -1) {
+		snd_iprintf(buffer, "N/A\n");
+		return;
+	}
+	snd_iprintf(buffer, "rates 0x%03x, bits 0x%02x, types 0x%x\n",
+		    pcm & AC_SUPPCM_RATES, (pcm >> 16) & 0xff, stream & 0xf);
+}
+
+static const char *get_jack_location(u32 cfg)
+{
+	static char *bases[7] = {
+		"N/A", "Rear", "Front", "Left", "Right", "Top", "Bottom",
+	};
+	static unsigned char specials_idx[] = {
+		0x07, 0x08,
+		0x17, 0x18, 0x19,
+		0x37, 0x38
+	};
+	static char *specials[] = {
+		"Rear Panel", "Drive Bar",
+		"Riser", "HDMI", "ATAPI",
+		"Mobile-In", "Mobile-Out"
+	};
+	int i;
+	cfg = (cfg & AC_DEFCFG_LOCATION) >> AC_DEFCFG_LOCATION_SHIFT;
+	if ((cfg & 0x0f) < 7)
+		return bases[cfg & 0x0f];
+	for (i = 0; i < ARRAY_SIZE(specials_idx); i++) {
+		if (cfg == specials_idx[i])
+			return specials[i];
+	}
+	return "UNKNOWN";
+}
+
+static const char *get_jack_connection(u32 cfg)
+{
+	static char *names[16] = {
+		"Unknown", "1/8", "1/4", "ATAPI",
+		"RCA", "Optical","Digital", "Analog",
+		"DIN", "XLR", "RJ11", "Comb",
+		NULL, NULL, NULL, "Other"
+	};
+	cfg = (cfg & AC_DEFCFG_CONN_TYPE) >> AC_DEFCFG_CONN_TYPE_SHIFT;
+	if (names[cfg])
+		return names[cfg];
+	else
+		return "UNKNOWN";
+}
+
+static const char *get_jack_color(u32 cfg)
+{
+	static char *names[16] = {
+		"Unknown", "Black", "Grey", "Blue",
+		"Green", "Red", "Orange", "Yellow",
+		"Purple", "Pink", NULL, NULL,
+		NULL, NULL, "White", "Other",
+	};
+	cfg = (cfg & AC_DEFCFG_COLOR) >> AC_DEFCFG_COLOR_SHIFT;
+	if (names[cfg])
+		return names[cfg];
+	else
+		return "UNKNOWN";
+}
+
+static void print_pin_caps(snd_info_buffer_t *buffer,
+			   struct hda_codec *codec, hda_nid_t nid)
+{
+	static char *jack_types[16] = {
+		"Line Out", "Speaker", "HP Out", "CD",
+		"SPDIF Out", "Digital Out", "Modem Line", "Modem Hand",
+		"Line In", "Aux", "Mic", "Telephony",
+		"SPDIF In", "Digitial In", "Reserved", "Other"
+	};
+	static char *jack_locations[4] = { "Ext", "Int", "Sep", "Oth" };
+	unsigned int caps;
+
+	caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+	snd_iprintf(buffer, "  Pincap 0x08%x:", caps);
+	if (caps & AC_PINCAP_IN)
+		snd_iprintf(buffer, " IN");
+	if (caps & AC_PINCAP_OUT)
+		snd_iprintf(buffer, " OUT");
+	if (caps & AC_PINCAP_HP_DRV)
+		snd_iprintf(buffer, " HP");
+	snd_iprintf(buffer, "\n");
+	caps = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+	snd_iprintf(buffer, "  Pin Default 0x%08x: %s at %s %s\n", caps,
+		    jack_types[(caps & AC_DEFCFG_DEVICE) >> AC_DEFCFG_DEVICE_SHIFT],
+		    jack_locations[(caps >> (AC_DEFCFG_LOCATION_SHIFT + 4)) & 3],
+		    get_jack_location(caps));
+	snd_iprintf(buffer, "    Conn = %s, Color = %s\n",
+		    get_jack_connection(caps),
+		    get_jack_color(caps));
+}
+
+
+static void print_codec_info(snd_info_entry_t *entry, snd_info_buffer_t *buffer)
+{
+	struct hda_codec *codec = entry->private_data;
+	char buf[32];
+	hda_nid_t nid;
+	int i, nodes;
+
+	snd_hda_get_codec_name(codec, buf, sizeof(buf));
+	snd_iprintf(buffer, "Codec: %s\n", buf);
+	snd_iprintf(buffer, "Address: %d\n", codec->addr);
+	snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id);
+	snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id);
+	snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id);
+	snd_iprintf(buffer, "Default PCM: ");
+	print_pcm_caps(buffer, codec, codec->afg);
+	snd_iprintf(buffer, "Default Amp-In caps: ");
+	print_amp_caps(buffer, codec, codec->afg, HDA_INPUT);
+	snd_iprintf(buffer, "Default Amp-Out caps: ");
+	print_amp_caps(buffer, codec, codec->afg, HDA_OUTPUT);
+
+	nodes = snd_hda_get_sub_nodes(codec, codec->afg, &nid);
+	if (! nid || nodes < 0) {
+		snd_iprintf(buffer, "Invalid AFG subtree\n");
+		return;
+	}
+	for (i = 0; i < nodes; i++, nid++) {
+		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;
+		snd_iprintf(buffer, "Node 0x%02x [%s] wcaps 0x%x:", nid,
+			    get_wid_type_name(wid_type), wid_caps);
+		if (wid_caps & AC_WCAP_STEREO)
+			snd_iprintf(buffer, " Stereo");
+		else
+			snd_iprintf(buffer, " Mono");
+		if (wid_caps & AC_WCAP_DIGITAL)
+			snd_iprintf(buffer, " Digital");
+		if (wid_caps & AC_WCAP_IN_AMP)
+			snd_iprintf(buffer, " Amp-In");
+		if (wid_caps & AC_WCAP_OUT_AMP)
+			snd_iprintf(buffer, " Amp-Out");
+		snd_iprintf(buffer, "\n");
+
+		if (wid_caps & AC_WCAP_IN_AMP) {
+			snd_iprintf(buffer, "  Amp-In caps: ");
+			print_amp_caps(buffer, codec, nid, HDA_INPUT);
+			snd_iprintf(buffer, "  Amp-In vals: ");
+			print_amp_vals(buffer, codec, nid, HDA_INPUT,
+				       wid_caps & AC_WCAP_STEREO);
+		}
+		if (wid_caps & AC_WCAP_OUT_AMP) {
+			snd_iprintf(buffer, "  Amp-Out caps: ");
+			print_amp_caps(buffer, codec, nid, HDA_OUTPUT);
+			snd_iprintf(buffer, "  Amp-Out vals: ");
+			print_amp_vals(buffer, codec, nid, HDA_OUTPUT,
+				       wid_caps & AC_WCAP_STEREO);
+		}
+
+		if (wid_type == AC_WID_PIN) {
+			unsigned int pinctls;
+			print_pin_caps(buffer, codec, nid);
+			pinctls = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+			snd_iprintf(buffer, "  Pin-ctls: 0x%02x:", pinctls);
+			if (pinctls & AC_PINCTL_IN_EN)
+				snd_iprintf(buffer, " IN");
+			if (pinctls & AC_PINCTL_OUT_EN)
+				snd_iprintf(buffer, " OUT");
+			if (pinctls & AC_PINCTL_HP_EN)
+				snd_iprintf(buffer, " HP");
+			snd_iprintf(buffer, "\n");
+		}
+
+		if ((wid_type == AC_WID_AUD_OUT || wid_type == AC_WID_AUD_IN) &&
+		    (wid_caps & AC_WCAP_FORMAT_OVRD)) {
+			snd_iprintf(buffer, "  PCM: ");
+			print_pcm_caps(buffer, codec, nid);
+		}
+
+		if (wid_caps & AC_WCAP_CONN_LIST) {
+			hda_nid_t conn[HDA_MAX_CONNECTIONS];
+			int c, conn_len;
+			conn_len = snd_hda_get_connections(codec, nid, conn,
+							   HDA_MAX_CONNECTIONS);
+			snd_iprintf(buffer, "  Connection: %d\n", conn_len);
+			snd_iprintf(buffer, "    ");
+			for (c = 0; c < conn_len; c++)
+				snd_iprintf(buffer, " 0x%02x", conn[c]);
+			snd_iprintf(buffer, "\n");
+		}
+	}
+}
+
+/*
+ * create a proc read
+ */
+int snd_hda_codec_proc_new(struct hda_codec *codec)
+{
+	char name[32];
+	snd_info_entry_t *entry;
+	int err;
+
+	snprintf(name, sizeof(name), "codec#%d", codec->addr);
+	err = snd_card_proc_new(codec->bus->card, name, &entry);
+	if (err < 0)
+		return err;
+
+	snd_info_set_text_ops(entry, codec, 32 * 1024, print_codec_info);
+	return 0;
+}
+
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
new file mode 100644
index 000000000000..75d23849f71a
--- /dev/null
+++ b/sound/pci/hda/patch_analog.c
@@ -0,0 +1,445 @@
+/*
+ * HD audio interface patch for AD1986A
+ *
+ * Copyright (c) 2005 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 <sound/driver.h>
+#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 ad1986a_spec {
+	struct semaphore amp_mutex;	/* PCM volume/mute control mutex */
+	struct hda_multi_out multiout;	/* playback */
+	unsigned int cur_mux;		/* capture source */
+	struct hda_pcm pcm_rec[2];	/* PCM information */
+};
+
+#define AD1986A_SPDIF_OUT	0x02
+#define AD1986A_FRONT_DAC	0x03
+#define AD1986A_SURR_DAC	0x04
+#define AD1986A_CLFE_DAC	0x05
+#define AD1986A_ADC		0x06
+
+static hda_nid_t ad1986a_dac_nids[3] = {
+	AD1986A_FRONT_DAC, AD1986A_SURR_DAC, AD1986A_CLFE_DAC
+};
+
+static struct hda_input_mux ad1986a_capture_source = {
+	.num_items = 7,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "CD", 0x1 },
+		{ "Aux", 0x3 },
+		{ "Line", 0x4 },
+		{ "Mix", 0x5 },
+		{ "Mono", 0x6 },
+		{ "Phone", 0x7 },
+	},
+};
+
+/*
+ * PCM control
+ *
+ * bind volumes/mutes of 3 DACs as a single PCM control for simplicity
+ */
+
+#define ad1986a_pcm_amp_vol_info	snd_hda_mixer_amp_volume_info
+
+static int ad1986a_pcm_amp_vol_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ad1986a_spec *ad = codec->spec;
+
+	down(&ad->amp_mutex);
+	snd_hda_mixer_amp_volume_get(kcontrol, ucontrol);
+	up(&ad->amp_mutex);
+	return 0;
+}
+
+static int ad1986a_pcm_amp_vol_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ad1986a_spec *ad = codec->spec;
+	int i, change = 0;
+
+	down(&ad->amp_mutex);
+	for (i = 0; i < ARRAY_SIZE(ad1986a_dac_nids); i++) {
+		kcontrol->private_value = HDA_COMPOSE_AMP_VAL(ad1986a_dac_nids[i], 3, 0, HDA_OUTPUT);
+		change |= snd_hda_mixer_amp_volume_put(kcontrol, ucontrol);
+	}
+	kcontrol->private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT);
+	up(&ad->amp_mutex);
+	return change;
+}
+
+#define ad1986a_pcm_amp_sw_info		snd_hda_mixer_amp_volume_info
+
+static int ad1986a_pcm_amp_sw_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ad1986a_spec *ad = codec->spec;
+
+	down(&ad->amp_mutex);
+	snd_hda_mixer_amp_switch_get(kcontrol, ucontrol);
+	up(&ad->amp_mutex);
+	return 0;
+}
+
+static int ad1986a_pcm_amp_sw_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ad1986a_spec *ad = codec->spec;
+	int i, change = 0;
+
+	down(&ad->amp_mutex);
+	for (i = 0; i < ARRAY_SIZE(ad1986a_dac_nids); i++) {
+		kcontrol->private_value = HDA_COMPOSE_AMP_VAL(ad1986a_dac_nids[i], 3, 0, HDA_OUTPUT);
+		change |= snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
+	}
+	kcontrol->private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT);
+	up(&ad->amp_mutex);
+	return change;
+}
+
+/*
+ * input MUX handling
+ */
+static int ad1986a_mux_enum_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+	return snd_hda_input_mux_info(&ad1986a_capture_source, uinfo);
+}
+
+static int ad1986a_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ad1986a_spec *spec = codec->spec;
+
+	ucontrol->value.enumerated.item[0] = spec->cur_mux;
+	return 0;
+}
+
+static int ad1986a_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ad1986a_spec *spec = codec->spec;
+
+	return snd_hda_input_mux_put(codec, &ad1986a_capture_source, ucontrol,
+				     AD1986A_ADC, &spec->cur_mux);
+}
+
+/*
+ * mixers
+ */
+static snd_kcontrol_new_t ad1986a_mixers[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "PCM Playback Volume",
+		.info = ad1986a_pcm_amp_vol_info,
+		.get = ad1986a_pcm_amp_vol_get,
+		.put = ad1986a_pcm_amp_vol_put,
+		.private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT)
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "PCM Playback Switch",
+		.info = ad1986a_pcm_amp_sw_info,
+		.get = ad1986a_pcm_amp_sw_get,
+		.put = ad1986a_pcm_amp_sw_put,
+		.private_value = HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT)
+	},
+	HDA_CODEC_VOLUME("Front Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Front Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Surround Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Surround Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x1d, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x1d, 2, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x1d, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x1d, 2, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x1a, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1a, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x17, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x17, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Aux Playback Volume", 0x16, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Aux Playback Switch", 0x16, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x18, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x18, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Capture Source",
+		.info = ad1986a_mux_enum_info,
+		.get = ad1986a_mux_enum_get,
+		.put = ad1986a_mux_enum_put,
+	},
+	HDA_CODEC_MUTE("Stereo Downmix Switch", 0x09, 0x0, HDA_OUTPUT),
+	{ } /* end */
+};
+
+/*
+ * initialization verbs
+ */
+static struct hda_verb ad1986a_init_verbs[] = {
+	/* Front, Surround, CLFE DAC; mute as default */
+	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	/* Downmix - off */
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	/* HP, Line-Out, Surround, CLFE selectors */
+	{0x0a, AC_VERB_SET_CONNECT_SEL, 0x0},
+	{0x0b, AC_VERB_SET_CONNECT_SEL, 0x0},
+	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
+	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
+	/* Mono selector */
+	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x0},
+	/* Mic selector: Mic 1/2 pin */
+	{0x0f, AC_VERB_SET_CONNECT_SEL, 0x0},
+	/* Line-in selector: Line-in */
+	{0x10, AC_VERB_SET_CONNECT_SEL, 0x0},
+	/* Mic 1/2 swap */
+	{0x11, AC_VERB_SET_CONNECT_SEL, 0x0},
+	/* Record selector: mic */
+	{0x12, AC_VERB_SET_CONNECT_SEL, 0x0},
+	/* Mic, Phone, CD, Aux, Line-In amp; mute as default */
+	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	/* PC beep */
+	{0x18, AC_VERB_SET_CONNECT_SEL, 0x0},
+	/* HP, Line-Out, Surround, CLFE, Mono pins; mute as default */
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	{ } /* end */
+};
+
+
+static int ad1986a_init(struct hda_codec *codec)
+{
+	snd_hda_sequence_write(codec, ad1986a_init_verbs);
+	return 0;
+}
+
+static int ad1986a_build_controls(struct hda_codec *codec)
+{
+	int err;
+
+	err = snd_hda_add_new_ctls(codec, ad1986a_mixers);
+	if (err < 0)
+		return err;
+	err = snd_hda_create_spdif_out_ctls(codec, AD1986A_SPDIF_OUT);
+	if (err < 0)
+		return err;
+	return 0;
+}
+
+/*
+ * Analog playback callbacks
+ */
+static int ad1986a_playback_pcm_open(struct hda_pcm_stream *hinfo,
+				     struct hda_codec *codec,
+				     snd_pcm_substream_t *substream)
+{
+	struct ad1986a_spec *spec = codec->spec;
+	return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream);
+}
+
+static int ad1986a_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+					struct hda_codec *codec,
+					unsigned int stream_tag,
+					unsigned int format,
+					snd_pcm_substream_t *substream)
+{
+	struct ad1986a_spec *spec = codec->spec;
+	return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
+						format, substream);
+}
+
+static int ad1986a_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+					struct hda_codec *codec,
+					snd_pcm_substream_t *substream)
+{
+	struct ad1986a_spec *spec = codec->spec;
+	return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
+}
+
+/*
+ * Digital out
+ */
+static int ad1986a_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+					 struct hda_codec *codec,
+					 snd_pcm_substream_t *substream)
+{
+	struct ad1986a_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int ad1986a_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+					  struct hda_codec *codec,
+					  snd_pcm_substream_t *substream)
+{
+	struct ad1986a_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+/*
+ * Analog capture
+ */
+static int ad1986a_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+				       struct hda_codec *codec,
+				       unsigned int stream_tag,
+				       unsigned int format,
+				       snd_pcm_substream_t *substream)
+{
+	snd_hda_codec_setup_stream(codec, AD1986A_ADC, stream_tag, 0, format);
+	return 0;
+}
+
+static int ad1986a_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+				       struct hda_codec *codec,
+				       snd_pcm_substream_t *substream)
+{
+	snd_hda_codec_setup_stream(codec, AD1986A_ADC, 0, 0, 0);
+	return 0;
+}
+
+
+/*
+ */
+static struct hda_pcm_stream ad1986a_pcm_analog_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 6,
+	.nid = AD1986A_FRONT_DAC, /* NID to query formats and rates */
+	.ops = {
+		.open = ad1986a_playback_pcm_open,
+		.prepare = ad1986a_playback_pcm_prepare,
+		.cleanup = ad1986a_playback_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream ad1986a_pcm_analog_capture = {
+	.substreams = 2,
+	.channels_min = 2,
+	.channels_max = 2,
+	.nid = AD1986A_ADC, /* NID to query formats and rates */
+	.ops = {
+		.prepare = ad1986a_capture_pcm_prepare,
+		.cleanup = ad1986a_capture_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream ad1986a_pcm_digital_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.nid = AD1986A_SPDIF_OUT, 
+	.ops = {
+		.open = ad1986a_dig_playback_pcm_open,
+		.close = ad1986a_dig_playback_pcm_close
+	},
+};
+
+static int ad1986a_build_pcms(struct hda_codec *codec)
+{
+	struct ad1986a_spec *spec = codec->spec;
+	struct hda_pcm *info = spec->pcm_rec;
+
+	codec->num_pcms = 2;
+	codec->pcm_info = info;
+
+	info->name = "AD1986A Analog";
+	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad1986a_pcm_analog_playback;
+	info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad1986a_pcm_analog_capture;
+	info++;
+
+	info->name = "AD1986A Digital";
+	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad1986a_pcm_digital_playback;
+
+	return 0;
+}
+
+static void ad1986a_free(struct hda_codec *codec)
+{
+	kfree(codec->spec);
+}
+
+#ifdef CONFIG_PM
+static int ad1986a_resume(struct hda_codec *codec)
+{
+	ad1986a_init(codec);
+	snd_hda_resume_ctls(codec, ad1986a_mixers);
+	snd_hda_resume_spdif_out(codec);
+	return 0;
+}
+#endif
+
+static struct hda_codec_ops ad1986a_patch_ops = {
+	.build_controls = ad1986a_build_controls,
+	.build_pcms = ad1986a_build_pcms,
+	.init = ad1986a_init,
+	.free = ad1986a_free,
+#ifdef CONFIG_PM
+	.resume = ad1986a_resume,
+#endif
+};
+
+static int patch_ad1986a(struct hda_codec *codec)
+{
+	struct ad1986a_spec *spec;
+
+	spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
+	if (spec == NULL)
+		return -ENOMEM;
+
+	init_MUTEX(&spec->amp_mutex);
+	codec->spec = spec;
+
+	spec->multiout.max_channels = 6;
+	spec->multiout.num_dacs = ARRAY_SIZE(ad1986a_dac_nids);
+	spec->multiout.dac_nids = ad1986a_dac_nids;
+	spec->multiout.dig_out_nid = AD1986A_SPDIF_OUT;
+
+	codec->patch_ops = ad1986a_patch_ops;
+
+	return 0;
+}
+
+/*
+ * patch entries
+ */
+struct hda_codec_preset snd_hda_preset_analog[] = {
+	{ .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a },
+	{} /* terminator */
+};
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
new file mode 100644
index 000000000000..b7cc8e4bffb7
--- /dev/null
+++ b/sound/pci/hda/patch_cmedia.c
@@ -0,0 +1,621 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * HD audio interface patch for C-Media CMI9880
+ *
+ * Copyright (c) 2004 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 <sound/driver.h>
+#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"
+
+
+/* board config type */
+enum {
+	CMI_MINIMAL,	/* back 3-jack */
+	CMI_MIN_FP,	/* back 3-jack + front-panel 2-jack */
+	CMI_FULL,	/* back 6-jack + front-panel 2-jack */
+	CMI_FULL_DIG,	/* back 6-jack + front-panel 2-jack + digital I/O */
+	CMI_ALLOUT,	/* back 5-jack + front-panel 2-jack + digital out */
+};
+
+struct cmi_spec {
+	int board_config;
+	unsigned int surr_switch: 1;	/* switchable line,mic */
+	unsigned int no_line_in: 1;	/* no line-in (5-jack) */
+	unsigned int front_panel: 1;	/* has front-panel 2-jack */
+
+	/* playback */
+	struct hda_multi_out multiout;
+
+	/* capture */
+	hda_nid_t *adc_nids;
+	hda_nid_t dig_in_nid;
+
+	/* capture source */
+	const struct hda_input_mux *input_mux;
+	unsigned int cur_mux[2];
+
+	/* channel mode */
+	unsigned int num_ch_modes;
+	unsigned int cur_ch_mode;
+	const struct cmi_channel_mode *channel_modes;
+
+	struct hda_pcm pcm_rec[2];	/* PCM information */
+};
+
+/*
+ * input MUX
+ */
+static int cmi_mux_enum_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cmi_spec *spec = codec->spec;
+	return snd_hda_input_mux_info(spec->input_mux, uinfo);
+}
+
+static int cmi_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cmi_spec *spec = codec->spec;
+	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+	ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx];
+	return 0;
+}
+
+static int cmi_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cmi_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->adc_nids[adc_idx], &spec->cur_mux[adc_idx]);
+}
+
+/*
+ * shared line-in, mic for surrounds
+ */
+
+/* 3-stack / 2 channel */
+static struct hda_verb cmi9880_ch2_init[] = {
+	/* set line-in PIN for input */
+	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+	/* set mic PIN for input, also enable vref */
+	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+	/* route front PCM (DAC1) to HP */
+	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	{}
+};
+
+/* 3-stack / 6 channel */
+static struct hda_verb cmi9880_ch6_init[] = {
+	/* set line-in PIN for output */
+	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	/* set mic PIN for output */
+	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	/* route front PCM (DAC1) to HP */
+	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	{}
+};
+
+/* 3-stack+front / 8 channel */
+static struct hda_verb cmi9880_ch8_init[] = {
+	/* set line-in PIN for output */
+	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	/* set mic PIN for output */
+	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	/* route rear-surround PCM (DAC4) to HP */
+	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x03 },
+	{}
+};
+
+struct cmi_channel_mode {
+	unsigned int channels;
+	const struct hda_verb *sequence;
+};
+
+static struct cmi_channel_mode cmi9880_channel_modes[3] = {
+	{ 2, cmi9880_ch2_init },
+	{ 6, cmi9880_ch6_init },
+	{ 8, cmi9880_ch8_init },
+};
+
+static int cmi_ch_mode_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cmi_spec *spec = codec->spec;
+
+	snd_assert(spec->channel_modes, return -EINVAL);
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = spec->num_ch_modes;
+	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
+		uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1;
+	sprintf(uinfo->value.enumerated.name, "%dch",
+		spec->channel_modes[uinfo->value.enumerated.item].channels);
+	return 0;
+}
+
+static int cmi_ch_mode_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cmi_spec *spec = codec->spec;
+
+	ucontrol->value.enumerated.item[0] = spec->cur_ch_mode;
+	return 0;
+}
+
+static int cmi_ch_mode_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cmi_spec *spec = codec->spec;
+
+	snd_assert(spec->channel_modes, return -EINVAL);
+	if (ucontrol->value.enumerated.item[0] >= spec->num_ch_modes)
+		ucontrol->value.enumerated.item[0] = spec->num_ch_modes;
+	if (ucontrol->value.enumerated.item[0] == spec->cur_ch_mode &&
+	    ! codec->in_resume)
+		return 0;
+
+	spec->cur_ch_mode = ucontrol->value.enumerated.item[0];
+	snd_hda_sequence_write(codec, spec->channel_modes[spec->cur_ch_mode].sequence);
+	spec->multiout.max_channels = spec->channel_modes[spec->cur_ch_mode].channels;
+	return 1;
+}
+
+/*
+ */
+static snd_kcontrol_new_t cmi9880_basic_mixer[] = {
+	/* CMI9880 has no playback volumes! */
+	HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT), /* front */
+	HDA_CODEC_MUTE("Surround Playback Switch", 0x04, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x05, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x05, 2, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Side Playback Switch", 0x06, 0x0, HDA_OUTPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		/* The multiple "Capture Source" controls confuse alsamixer
+		 * So call somewhat different..
+		 * FIXME: the controls appear in the "playback" view!
+		 */
+		/* .name = "Capture Source", */
+		.name = "Input Source",
+		.count = 2,
+		.info = cmi_mux_enum_info,
+		.get = cmi_mux_enum_get,
+		.put = cmi_mux_enum_put,
+	},
+	HDA_CODEC_VOLUME("Capture Volume", 0x08, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x08, 0, HDA_INPUT),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x23, 0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x23, 0, HDA_OUTPUT),
+	{ } /* end */
+};
+
+/*
+ * shared I/O pins
+ */
+static snd_kcontrol_new_t cmi9880_ch_mode_mixer[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = cmi_ch_mode_info,
+		.get = cmi_ch_mode_get,
+		.put = cmi_ch_mode_put,
+	},
+	{ } /* end */
+};
+
+/* AUD-in selections:
+ * 0x0b 0x0c 0x0d 0x0e 0x0f 0x10 0x11 0x1f 0x20
+ */
+static struct hda_input_mux cmi9880_basic_mux = {
+	.num_items = 4,
+	.items = {
+		{ "Front Mic", 0x5 },
+		{ "Rear Mic", 0x2 },
+		{ "Line", 0x1 },
+		{ "CD", 0x7 },
+	}
+};
+
+static struct hda_input_mux cmi9880_no_line_mux = {
+	.num_items = 3,
+	.items = {
+		{ "Front Mic", 0x5 },
+		{ "Rear Mic", 0x2 },
+		{ "CD", 0x7 },
+	}
+};
+
+/* front, rear, clfe, rear_surr */
+static hda_nid_t cmi9880_dac_nids[4] = {
+	0x03, 0x04, 0x05, 0x06
+};
+/* ADC0, ADC1 */
+static hda_nid_t cmi9880_adc_nids[2] = {
+	0x08, 0x09
+};
+
+#define CMI_DIG_OUT_NID	0x07
+#define CMI_DIG_IN_NID	0x0a
+
+/*
+ */
+static struct hda_verb cmi9880_basic_init[] = {
+	/* port-D for line out (rear panel) */
+	{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+	/* port-E for HP out (front panel) */
+	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+	/* route front PCM to HP */
+	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	/* port-A for surround (rear panel) */
+	{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+	/* port-G for CLFE (rear panel) */
+	{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+	/* port-H for side (rear panel) */
+	{ 0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+	/* port-C for line-in (rear panel) */
+	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+	/* port-B for mic-in (rear panel) with vref */
+	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+	/* port-F for mic-in (front panel) with vref */
+	{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+	/* CD-in */
+	{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+	/* route front mic to ADC1/2 */
+	{ 0x08, AC_VERB_SET_CONNECT_SEL, 0x05 },
+	{ 0x09, AC_VERB_SET_CONNECT_SEL, 0x05 },
+	{} /* terminator */
+};
+
+static struct hda_verb cmi9880_allout_init[] = {
+	/* port-D for line out (rear panel) */
+	{ 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+	/* port-E for HP out (front panel) */
+	{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+	/* route front PCM to HP */
+	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 },
+	/* port-A for side (rear panel) */
+	{ 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+	/* port-G for CLFE (rear panel) */
+	{ 0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+	/* port-C for surround (rear panel) */
+	{ 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
+	/* port-B for mic-in (rear panel) with vref */
+	{ 0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+	/* port-F for mic-in (front panel) with vref */
+	{ 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+	/* CD-in */
+	{ 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+	/* route front mic to ADC1/2 */
+	{ 0x08, AC_VERB_SET_CONNECT_SEL, 0x05 },
+	{ 0x09, AC_VERB_SET_CONNECT_SEL, 0x05 },
+	{} /* terminator */
+};
+
+/*
+ */
+static int cmi9880_build_controls(struct hda_codec *codec)
+{
+	struct cmi_spec *spec = codec->spec;
+	int err;
+
+	err = snd_hda_add_new_ctls(codec, cmi9880_basic_mixer);
+	if (err < 0)
+		return err;
+	if (spec->surr_switch) {
+		err = snd_hda_add_new_ctls(codec, cmi9880_ch_mode_mixer);
+		if (err < 0)
+			return err;
+	}
+	if (spec->multiout.dig_out_nid) {
+		err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+		if (err < 0)
+			return err;
+	}
+	if (spec->dig_in_nid) {
+		err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
+		if (err < 0)
+			return err;
+	}
+	return 0;
+}
+
+static int cmi9880_init(struct hda_codec *codec)
+{
+	struct cmi_spec *spec = codec->spec;
+	if (spec->board_config == CMI_ALLOUT)
+		snd_hda_sequence_write(codec, cmi9880_allout_init);
+	else
+		snd_hda_sequence_write(codec, cmi9880_basic_init);
+	return 0;
+}
+
+#ifdef CONFIG_PM
+/*
+ * resume
+ */
+static int cmi9880_resume(struct hda_codec *codec)
+{
+	struct cmi_spec *spec = codec->spec;
+
+	cmi9880_init(codec);
+	snd_hda_resume_ctls(codec, cmi9880_basic_mixer);
+	if (spec->surr_switch)
+		snd_hda_resume_ctls(codec, cmi9880_ch_mode_mixer);
+	if (spec->multiout.dig_out_nid)
+		snd_hda_resume_spdif_out(codec);
+	if (spec->dig_in_nid)
+		snd_hda_resume_spdif_in(codec);
+
+	return 0;
+}
+#endif
+
+/*
+ * Analog playback callbacks
+ */
+static int cmi9880_playback_pcm_open(struct hda_pcm_stream *hinfo,
+				     struct hda_codec *codec,
+				     snd_pcm_substream_t *substream)
+{
+	struct cmi_spec *spec = codec->spec;
+	return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream);
+}
+
+static int cmi9880_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+					struct hda_codec *codec,
+					unsigned int stream_tag,
+					unsigned int format,
+					snd_pcm_substream_t *substream)
+{
+	struct cmi_spec *spec = codec->spec;
+	return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
+						format, substream);
+}
+
+static int cmi9880_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+				       struct hda_codec *codec,
+				       snd_pcm_substream_t *substream)
+{
+	struct cmi_spec *spec = codec->spec;
+	return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
+}
+
+/*
+ * Digital out
+ */
+static int cmi9880_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+					 struct hda_codec *codec,
+					 snd_pcm_substream_t *substream)
+{
+	struct cmi_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int cmi9880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+					  struct hda_codec *codec,
+					  snd_pcm_substream_t *substream)
+{
+	struct cmi_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+/*
+ * Analog capture
+ */
+static int cmi9880_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+				      struct hda_codec *codec,
+				      unsigned int stream_tag,
+				      unsigned int format,
+				      snd_pcm_substream_t *substream)
+{
+	struct cmi_spec *spec = codec->spec;
+
+	snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
+				   stream_tag, 0, format);
+	return 0;
+}
+
+static int cmi9880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+				      struct hda_codec *codec,
+				      snd_pcm_substream_t *substream)
+{
+	struct cmi_spec *spec = codec->spec;
+
+	snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], 0, 0, 0);
+	return 0;
+}
+
+
+/*
+ */
+static struct hda_pcm_stream cmi9880_pcm_analog_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 8,
+	.nid = 0x03, /* NID to query formats and rates */
+	.ops = {
+		.open = cmi9880_playback_pcm_open,
+		.prepare = cmi9880_playback_pcm_prepare,
+		.cleanup = cmi9880_playback_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream cmi9880_pcm_analog_capture = {
+	.substreams = 2,
+	.channels_min = 2,
+	.channels_max = 2,
+	.nid = 0x08, /* NID to query formats and rates */
+	.ops = {
+		.prepare = cmi9880_capture_pcm_prepare,
+		.cleanup = cmi9880_capture_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream cmi9880_pcm_digital_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	/* NID is set in cmi9880_build_pcms */
+	.ops = {
+		.open = cmi9880_dig_playback_pcm_open,
+		.close = cmi9880_dig_playback_pcm_close
+	},
+};
+
+static struct hda_pcm_stream cmi9880_pcm_digital_capture = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	/* NID is set in cmi9880_build_pcms */
+};
+
+static int cmi9880_build_pcms(struct hda_codec *codec)
+{
+	struct cmi_spec *spec = codec->spec;
+	struct hda_pcm *info = spec->pcm_rec;
+
+	codec->num_pcms = 1;
+	codec->pcm_info = info;
+
+	info->name = "CMI9880";
+	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = cmi9880_pcm_analog_playback;
+	info->stream[SNDRV_PCM_STREAM_CAPTURE] = cmi9880_pcm_analog_capture;
+
+	if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
+		codec->num_pcms++;
+		info++;
+		info->name = "CMI9880 Digital";
+		if (spec->multiout.dig_out_nid) {
+			info->stream[SNDRV_PCM_STREAM_PLAYBACK] = cmi9880_pcm_digital_playback;
+			info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
+		}
+		if (spec->dig_in_nid) {
+			info->stream[SNDRV_PCM_STREAM_CAPTURE] = cmi9880_pcm_digital_capture;
+			info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid;
+		}
+	}
+
+	return 0;
+}
+
+static void cmi9880_free(struct hda_codec *codec)
+{
+	kfree(codec->spec);
+}
+
+/*
+ */
+
+static struct hda_board_config cmi9880_cfg_tbl[] = {
+	{ .modelname = "minimal", .config = CMI_MINIMAL },
+	{ .modelname = "min_fp", .config = CMI_MIN_FP },
+	{ .modelname = "full", .config = CMI_FULL },
+	{ .modelname = "full_dig", .config = CMI_FULL_DIG },
+	{ .modelname = "allout", .config = CMI_ALLOUT },
+	{} /* terminator */
+};
+
+static struct hda_codec_ops cmi9880_patch_ops = {
+	.build_controls = cmi9880_build_controls,
+	.build_pcms = cmi9880_build_pcms,
+	.init = cmi9880_init,
+	.free = cmi9880_free,
+#ifdef CONFIG_PM
+	.resume = cmi9880_resume,
+#endif
+};
+
+static int patch_cmi9880(struct hda_codec *codec)
+{
+	struct cmi_spec *spec;
+
+	spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
+	if (spec == NULL)
+		return -ENOMEM;
+
+	codec->spec = spec;
+	spec->board_config = snd_hda_check_board_config(codec, cmi9880_cfg_tbl);
+	if (spec->board_config < 0) {
+		snd_printd(KERN_INFO "hda_codec: Unknown model for CMI9880\n");
+		spec->board_config = CMI_FULL_DIG; /* try everything */
+	}
+
+	switch (spec->board_config) {
+	case CMI_MINIMAL:
+	case CMI_MIN_FP:
+		spec->surr_switch = 1;
+		if (spec->board_config == CMI_MINIMAL)
+			spec->num_ch_modes = 2;
+		else {
+			spec->front_panel = 1;
+			spec->num_ch_modes = 3;
+		}
+		spec->channel_modes = cmi9880_channel_modes;
+		spec->multiout.max_channels = cmi9880_channel_modes[0].channels;
+		spec->input_mux = &cmi9880_basic_mux;
+		break;
+	case CMI_FULL:
+	case CMI_FULL_DIG:
+		spec->front_panel = 1;
+		spec->multiout.max_channels = 8;
+		spec->input_mux = &cmi9880_basic_mux;
+		if (spec->board_config == CMI_FULL_DIG) {
+			spec->multiout.dig_out_nid = CMI_DIG_OUT_NID;
+			spec->dig_in_nid = CMI_DIG_IN_NID;
+		}
+		break;
+	case CMI_ALLOUT:
+		spec->front_panel = 1;
+		spec->multiout.max_channels = 8;
+		spec->no_line_in = 1;
+		spec->input_mux = &cmi9880_no_line_mux;
+		spec->multiout.dig_out_nid = CMI_DIG_OUT_NID;
+		break;
+	}
+
+	spec->multiout.num_dacs = 4;
+	spec->multiout.dac_nids = cmi9880_dac_nids;
+
+	spec->adc_nids = cmi9880_adc_nids;
+
+	codec->patch_ops = cmi9880_patch_ops;
+
+	return 0;
+}
+
+/*
+ * patch entries
+ */
+struct hda_codec_preset snd_hda_preset_cmedia[] = {
+	{ .id = 0x13f69880, .name = "CMI9880", .patch = patch_cmi9880 },
+ 	{ .id = 0x434d4980, .name = "CMI9880", .patch = patch_cmi9880 },
+	{} /* terminator */
+};
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
new file mode 100644
index 000000000000..17c5062423ae
--- /dev/null
+++ b/sound/pci/hda/patch_realtek.c
@@ -0,0 +1,1503 @@
+/*
+ * Universal Interface for Intel High Definition Audio Codec
+ *
+ * HD audio interface patch for ALC 260/880/882 codecs
+ *
+ * Copyright (c) 2004 PeiSen Hou <pshou@realtek.com.tw>
+ *                    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 <sound/driver.h>
+#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"
+
+
+/* ALC880 board config type */
+enum {
+	ALC880_MINIMAL,
+	ALC880_3ST,
+	ALC880_3ST_DIG,
+	ALC880_5ST,
+	ALC880_5ST_DIG,
+	ALC880_W810,
+};
+
+struct alc_spec {
+	/* codec parameterization */
+	unsigned int front_panel: 1;
+
+	snd_kcontrol_new_t* mixers[2];
+	unsigned int num_mixers;
+
+	struct hda_verb *init_verbs;
+
+	char* stream_name_analog;
+	struct hda_pcm_stream *stream_analog_playback;
+	struct hda_pcm_stream *stream_analog_capture;
+
+	char* stream_name_digital;
+	struct hda_pcm_stream *stream_digital_playback;
+	struct hda_pcm_stream *stream_digital_capture;
+
+	/* playback */
+	struct hda_multi_out multiout;
+
+	/* capture */
+	unsigned int num_adc_nids;
+	hda_nid_t *adc_nids;
+	hda_nid_t dig_in_nid;
+
+	/* capture source */
+	const struct hda_input_mux *input_mux;
+	unsigned int cur_mux[3];
+
+	/* channel model */
+	const struct alc_channel_mode *channel_mode;
+	int num_channel_mode;
+
+	/* PCM information */
+	struct hda_pcm pcm_rec[2];
+};
+
+/* DAC/ADC assignment */
+
+static hda_nid_t alc880_dac_nids[4] = {
+	/* front, rear, clfe, rear_surr */
+	0x02, 0x05, 0x04, 0x03
+};
+
+static hda_nid_t alc880_w810_dac_nids[3] = {
+	/* front, rear/surround, clfe */
+	0x02, 0x03, 0x04
+};
+
+static hda_nid_t alc880_adc_nids[3] = {
+	/* ADC0-2 */
+	0x07, 0x08, 0x09,
+};
+
+#define ALC880_DIGOUT_NID	0x06
+#define ALC880_DIGIN_NID	0x0a
+
+static hda_nid_t alc260_dac_nids[1] = {
+	/* front */
+	0x02,
+};
+
+static hda_nid_t alc260_adc_nids[2] = {
+	/* ADC0-1 */
+	0x04, 0x05,
+};
+
+#define ALC260_DIGOUT_NID	0x03
+#define ALC260_DIGIN_NID	0x06
+
+static struct hda_input_mux alc880_capture_source = {
+	.num_items = 4,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Front Mic", 0x3 },
+		{ "Line", 0x2 },
+		{ "CD", 0x4 },
+	},
+};
+
+static struct hda_input_mux alc260_capture_source = {
+	.num_items = 4,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Front Mic", 0x1 },
+		{ "Line", 0x2 },
+		{ "CD", 0x4 },
+	},
+};
+
+/*
+ * input MUX handling
+ */
+static int alc_mux_enum_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+	return snd_hda_input_mux_info(spec->input_mux, uinfo);
+}
+
+static int alc_mux_enum_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+
+	ucontrol->value.enumerated.item[0] = spec->cur_mux[adc_idx];
+	return 0;
+}
+
+static int alc_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_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->adc_nids[adc_idx], &spec->cur_mux[adc_idx]);
+}
+
+/*
+ * channel mode setting
+ */
+struct alc_channel_mode {
+	int channels;
+	const struct hda_verb *sequence;
+};
+
+
+/*
+ * channel source setting (2/6 channel selection for 3-stack)
+ */
+
+/*
+ * set the path ways for 2 channel output
+ * need to set the codec line out and mic 1 pin widgets to inputs
+ */
+static struct hda_verb alc880_threestack_ch2_init[] = {
+	/* set pin widget 1Ah (line in) for input */
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+	/* set pin widget 18h (mic1) for input, for mic also enable the vref */
+	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
+	/* mute the output for Line In PW */
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
+	/* mute for Mic1 PW */
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
+	{ } /* end */
+};
+
+/*
+ * 6ch mode
+ * need to set the codec line out and mic 1 pin widgets to outputs
+ */
+static struct hda_verb alc880_threestack_ch6_init[] = {
+	/* set pin widget 1Ah (line in) for output */
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	/* set pin widget 18h (mic1) for output */
+	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	/* unmute the output for Line In PW */
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 },
+	/* unmute for Mic1 PW */
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 },
+	/* for rear channel output using Line In 1
+	 * set select widget connection (nid = 0x12) - to summer node
+	 * for rear NID = 0x0f...offset 3 in connection list
+	 */
+	{ 0x12, AC_VERB_SET_CONNECT_SEL, 0x3 },
+	/* for Mic1 - retask for center/lfe */
+	/* set select widget connection (nid = 0x10) - to summer node for
+	 * front CLFE NID = 0x0e...offset 2 in connection list
+	 */
+	{ 0x10, AC_VERB_SET_CONNECT_SEL, 0x2 },
+	{ } /* end */
+};
+
+static struct alc_channel_mode alc880_threestack_modes[2] = {
+	{ 2, alc880_threestack_ch2_init },
+	{ 6, alc880_threestack_ch6_init },
+};
+
+
+/*
+ * channel source setting (6/8 channel selection for 5-stack)
+ */
+
+/* set the path ways for 6 channel output
+ * need to set the codec line out and mic 1 pin widgets to inputs
+ */
+static struct hda_verb alc880_fivestack_ch6_init[] = {
+	/* set pin widget 1Ah (line in) for input */
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
+	/* mute the output for Line In PW */
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
+	{ } /* end */
+};
+
+/* need to set the codec line out and mic 1 pin widgets to outputs */
+static struct hda_verb alc880_fivestack_ch8_init[] = {
+	/* set pin widget 1Ah (line in) for output */
+	{ 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
+	/* unmute the output for Line In PW */
+	{ 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000 },
+	/* output for surround channel output using Line In 1 */
+	/* set select widget connection (nid = 0x12) - to summer node
+	 * for surr_rear NID = 0x0d...offset 1 in connection list
+	 */
+	{ 0x12, AC_VERB_SET_CONNECT_SEL, 0x1 },
+	{ } /* end */
+};
+
+static struct alc_channel_mode alc880_fivestack_modes[2] = {
+	{ 6, alc880_fivestack_ch6_init },
+	{ 8, alc880_fivestack_ch8_init },
+};
+
+/*
+ * channel source setting for W810 system
+ *
+ * W810 has rear IO for:
+ * Front (DAC 02)
+ * Surround (DAC 03)
+ * Center/LFE (DAC 04)
+ * Digital out (06)
+ *
+ * The system also has a pair of internal speakers, and a headphone jack.
+ * These are both connected to Line2 on the codec, hence to DAC 02.
+ * 
+ * There is a variable resistor to control the speaker or headphone
+ * volume. This is a hardware-only device without a software API.
+ *
+ * Plugging headphones in will disable the internal speakers. This is
+ * implemented in hardware, not via the driver using jack sense. In
+ * a similar fashion, plugging into the rear socket marked "front" will
+ * disable both the speakers and headphones.
+ *
+ * For input, there's a microphone jack, and an "audio in" jack.
+ * These may not do anything useful with this driver yet, because I
+ * haven't setup any initialization verbs for these yet...
+ */
+
+static struct alc_channel_mode alc880_w810_modes[1] = {
+	{ 6, NULL }
+};
+
+/*
+ */
+static int alc880_ch_mode_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t *uinfo)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+
+	snd_assert(spec->channel_mode, return -ENXIO);
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = 2;
+	if (uinfo->value.enumerated.item >= 2)
+		uinfo->value.enumerated.item = 1;
+	sprintf(uinfo->value.enumerated.name, "%dch",
+		spec->channel_mode[uinfo->value.enumerated.item].channels);
+	return 0;
+}
+
+static int alc880_ch_mode_get(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+
+	snd_assert(spec->channel_mode, return -ENXIO);
+	ucontrol->value.enumerated.item[0] =
+		(spec->multiout.max_channels == spec->channel_mode[0].channels) ? 0 : 1;
+	return 0;
+}
+
+static int alc880_ch_mode_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+	int mode;
+
+	snd_assert(spec->channel_mode, return -ENXIO);
+	mode = ucontrol->value.enumerated.item[0] ? 1 : 0;
+	if (spec->multiout.max_channels == spec->channel_mode[mode].channels &&
+	    ! codec->in_resume)
+		return 0;
+
+	/* change the current channel setting */
+	spec->multiout.max_channels = spec->channel_mode[mode].channels;
+	if (spec->channel_mode[mode].sequence)
+		snd_hda_sequence_write(codec, spec->channel_mode[mode].sequence);
+
+	return 1;
+}
+
+
+/*
+ */
+
+/* 3-stack mode
+ * Pin assignment: Front=0x14, Line-In/Rear=0x1a, Mic/CLFE=0x18, F-Mic=0x1b
+ *                 HP=0x19
+ */
+static snd_kcontrol_new_t alc880_base_mixer[] = {
+	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Surround Playback Switch", 0x1a, 0x0, HDA_OUTPUT),
+	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_CODEC_MUTE_MONO("Center Playback Switch", 0x18, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x18, 2, 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_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		/* The multiple "Capture Source" controls confuse alsamixer
+		 * So call somewhat different..
+		 * FIXME: the controls appear in the "playback" view!
+		 */
+		/* .name = "Capture Source", */
+		.name = "Input Source",
+		.count = 2,
+		.info = alc_mux_enum_info,
+		.get = alc_mux_enum_get,
+		.put = alc_mux_enum_put,
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = alc880_ch_mode_info,
+		.get = alc880_ch_mode_get,
+		.put = alc880_ch_mode_put,
+	},
+	{ } /* end */
+};
+
+/* 5-stack mode
+ * Pin assignment: Front=0x14, Rear=0x17, CLFE=0x16
+ *                 Line-In/Side=0x1a, Mic=0x18, F-Mic=0x1b, HP=0x19
+ */
+static snd_kcontrol_new_t alc880_five_stack_mixer[] = {
+	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Surround Playback Switch", 0x17, 0x0, HDA_OUTPUT),
+	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_CODEC_MUTE_MONO("Center Playback Switch", 0x16, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Side Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Side Playback Switch", 0x1a, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
+	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		/* The multiple "Capture Source" controls confuse alsamixer
+		 * So call somewhat different..
+		 * FIXME: the controls appear in the "playback" view!
+		 */
+		/* .name = "Capture Source", */
+		.name = "Input Source",
+		.count = 2,
+		.info = alc_mux_enum_info,
+		.get = alc_mux_enum_get,
+		.put = alc_mux_enum_put,
+	},
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Channel Mode",
+		.info = alc880_ch_mode_info,
+		.get = alc880_ch_mode_get,
+		.put = alc880_ch_mode_put,
+	},
+	{ } /* end */
+};
+
+static snd_kcontrol_new_t alc880_w810_base_mixer[] = {
+	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	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_CODEC_MUTE_MONO("Center Playback Switch", 0x16, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		/* The multiple "Capture Source" controls confuse alsamixer
+		 * So call somewhat different..
+		 * FIXME: the controls appear in the "playback" view!
+		 */
+		/* .name = "Capture Source", */
+		.name = "Input Source",
+		.count = 3,
+		.info = alc_mux_enum_info,
+		.get = alc_mux_enum_get,
+		.put = alc_mux_enum_put,
+	},
+	{ } /* end */
+};
+
+/*
+ */
+static int alc_build_controls(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	int err;
+	int i;
+
+	for (i = 0; i < spec->num_mixers; i++) {
+		err = snd_hda_add_new_ctls(codec, spec->mixers[i]);
+		if (err < 0)
+			return err;
+	}
+
+	if (spec->multiout.dig_out_nid) {
+		err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid);
+		if (err < 0)
+			return err;
+	}
+	if (spec->dig_in_nid) {
+		err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
+		if (err < 0)
+			return err;
+	}
+	return 0;
+}
+
+/*
+ * initialize the codec volumes, etc
+ */
+
+static struct hda_verb alc880_init_verbs_three_stack[] = {
+	/* Line In pin widget for input */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+	/* CD pin widget for input */
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+	/* Mic1 (rear panel) pin widget for input and vref at 80% */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	/* Mic2 (front panel) pin widget for input and vref at 80% */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	/* unmute amp left and right */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
+	/* set connection select to line in (default select for this ADC) */
+	{0x07, AC_VERB_SET_CONNECT_SEL, 0x02},
+	/* unmute front mixer amp left (volume = 0) */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* mute pin widget amp left and right (no gain on this amp) */
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+	/* unmute rear mixer amp left and right (volume = 0) */
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* mute pin widget amp left and right (no gain on this amp) */
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+	/* unmute rear mixer amp left and right (volume = 0) */
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* mute pin widget amp left and right (no gain on this amp) */
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+
+	/* using rear surround as the path for headphone output */
+	/* unmute rear surround mixer amp left and right (volume = 0) */
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* PASD 3 stack boards use the Mic 2 as the headphone output */
+	/* need to program the selector associated with the Mic 2 pin widget to
+	 * surround path (index 0x01) for headphone output */
+	{0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
+	/* mute pin widget amp left and right (no gain on this amp) */
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+	/* need to retask the Mic 2 pin widget to output */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+
+	/* Unmute input amps (CD, Line In, Mic 1 & Mic 2) for mixer widget(nid=0x0B)
+	 * to support the input path of analog loopback
+	 * Note: PASD motherboards uses the Line In 2 as the input for front panel
+	 * mic (mic 2)
+	 */
+	/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03 */
+	/* unmute CD */
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+	/* unmute Line In */
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
+	/* unmute Mic 1 */
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	/* unmute Line In 2 (for PASD boards Mic 2) */
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
+
+	/* Unmute input amps for the line out paths to support the output path of
+	 * analog loopback
+	 * the mixers on the output path has 2 inputs, one from the DAC and one
+	 * from the mixer
+	 */
+	/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
+	/* Unmute Front out path */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	/* Unmute Surround (used as HP) out path */
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	/* Unmute C/LFE out path */
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))}, /* mute */
+	/* Unmute rear Surround out path */
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+
+	{ }
+};
+
+static struct hda_verb alc880_init_verbs_five_stack[] = {
+	/* Line In pin widget for input */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+	/* CD pin widget for input */
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+	/* Mic1 (rear panel) pin widget for input and vref at 80% */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	/* Mic2 (front panel) pin widget for input and vref at 80% */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	/* unmute amp left and right */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
+	/* set connection select to line in (default select for this ADC) */
+	{0x07, AC_VERB_SET_CONNECT_SEL, 0x02},
+	/* unmute front mixer amp left and right (volume = 0) */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* mute pin widget amp left and right (no gain on this amp) */
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+	/* five rear and clfe */
+	/* unmute rear mixer amp left and right (volume = 0)  */
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* mute pin widget amp left and right (no gain on this amp) */
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+	/* unmute clfe mixer amp left and right (volume = 0) */
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* mute pin widget amp left and right (no gain on this amp) */
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+
+	/* using rear surround as the path for headphone output */
+	/* unmute rear surround mixer amp left and right (volume = 0) */
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* PASD 3 stack boards use the Mic 2 as the headphone output */
+	/* need to program the selector associated with the Mic 2 pin widget to
+	 * surround path (index 0x01) for headphone output
+	 */
+	{0x11, AC_VERB_SET_CONNECT_SEL, 0x01},
+	/* mute pin widget amp left and right (no gain on this amp) */
+	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+	/* need to retask the Mic 2 pin widget to output */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+
+	/* Unmute input amps (CD, Line In, Mic 1 & Mic 2) for mixer
+	 * widget(nid=0x0B) to support the input path of analog loopback
+	 */
+	/* Note: PASD motherboards uses the Line In 2 as the input for front panel mic (mic 2) */
+	/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03*/
+	/* unmute CD */
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+	/* unmute Line In */
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
+	/* unmute Mic 1 */
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	/* unmute Line In 2 (for PASD boards Mic 2) */
+	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))},
+
+	/* Unmute input amps for the line out paths to support the output path of
+	 * analog loopback
+	 * the mixers on the output path has 2 inputs, one from the DAC and
+	 * one from the mixer
+	 */
+	/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
+	/* Unmute Front out path */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	/* Unmute Surround (used as HP) out path */
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	/* Unmute C/LFE out path */
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))}, /* mute */
+	/* Unmute rear Surround out path */
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+
+	{ }
+};
+
+static struct hda_verb alc880_w810_init_verbs[] = {
+	/* front channel selector/amp: input 0: DAC: unmuted, (no volume selection) */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
+
+	/* front channel selector/amp: input 1: capture mix: muted, (no volume selection) */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0x7180},
+
+	/* front channel selector/amp: output 0: unmuted, max volume */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+
+	/* front out pin: muted, (no volume selection)  */
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+
+	/* front out pin: NOT headphone enable, out enable, vref disabled */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+
+
+	/* surround channel selector/amp: input 0: DAC: unmuted, (no volume selection) */
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
+
+	/* surround channel selector/amp: input 1: capture mix: muted, (no volume selection) */
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0x7180},
+
+	/* surround channel selector/amp: output 0: unmuted, max volume */
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+
+	/* surround out pin: muted, (no volume selection)  */
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+
+	/* surround out pin: NOT headphone enable, out enable, vref disabled */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+
+
+	/* c/lfe channel selector/amp: input 0: DAC: unmuted, (no volume selection) */
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
+
+	/* c/lfe channel selector/amp: input 1: capture mix: muted, (no volume selection) */
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0x7180},
+
+	/* c/lfe channel selector/amp: output 0: unmuted, max volume */
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+
+	/* c/lfe out pin: muted, (no volume selection)  */
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+
+	/* c/lfe out pin: NOT headphone enable, out enable, vref disabled */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+
+
+	/* hphone/speaker input selector: front DAC */
+	{0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
+
+	/* hphone/speaker out pin: muted, (no volume selection)  */
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+
+	/* hphone/speaker out pin: NOT headphone enable, out enable, vref disabled */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+
+
+	{ }
+};
+
+static int alc_init(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	snd_hda_sequence_write(codec, spec->init_verbs);
+	return 0;
+}
+
+#ifdef CONFIG_PM
+/*
+ * resume
+ */
+static int alc_resume(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	int i;
+
+	alc_init(codec);
+	for (i = 0; i < spec->num_mixers; i++) {
+		snd_hda_resume_ctls(codec, spec->mixers[i]);
+	}
+	if (spec->multiout.dig_out_nid)
+		snd_hda_resume_spdif_out(codec);
+	if (spec->dig_in_nid)
+		snd_hda_resume_spdif_in(codec);
+
+	return 0;
+}
+#endif
+
+/*
+ * Analog playback callbacks
+ */
+static int alc880_playback_pcm_open(struct hda_pcm_stream *hinfo,
+				    struct hda_codec *codec,
+				    snd_pcm_substream_t *substream)
+{
+	struct alc_spec *spec = codec->spec;
+	return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream);
+}
+
+static int alc880_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
+				       struct hda_codec *codec,
+				       unsigned int stream_tag,
+				       unsigned int format,
+				       snd_pcm_substream_t *substream)
+{
+	struct alc_spec *spec = codec->spec;
+	return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
+						format, substream);
+}
+
+static int alc880_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+				       struct hda_codec *codec,
+				       snd_pcm_substream_t *substream)
+{
+	struct alc_spec *spec = codec->spec;
+	return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
+}
+
+/*
+ * Digital out
+ */
+static int alc880_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
+					struct hda_codec *codec,
+					snd_pcm_substream_t *substream)
+{
+	struct alc_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_open(codec, &spec->multiout);
+}
+
+static int alc880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
+					 struct hda_codec *codec,
+					 snd_pcm_substream_t *substream)
+{
+	struct alc_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_close(codec, &spec->multiout);
+}
+
+/*
+ * Analog capture
+ */
+static int alc880_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
+				      struct hda_codec *codec,
+				      unsigned int stream_tag,
+				      unsigned int format,
+				      snd_pcm_substream_t *substream)
+{
+	struct alc_spec *spec = codec->spec;
+
+	snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
+				   stream_tag, 0, format);
+	return 0;
+}
+
+static int alc880_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
+				      struct hda_codec *codec,
+				      snd_pcm_substream_t *substream)
+{
+	struct alc_spec *spec = codec->spec;
+
+	snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number], 0, 0, 0);
+	return 0;
+}
+
+
+/*
+ */
+static struct hda_pcm_stream alc880_pcm_analog_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 8,
+	.nid = 0x02, /* NID to query formats and rates */
+	.ops = {
+		.open = alc880_playback_pcm_open,
+		.prepare = alc880_playback_pcm_prepare,
+		.cleanup = alc880_playback_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream alc880_pcm_analog_capture = {
+	.substreams = 2,
+	.channels_min = 2,
+	.channels_max = 2,
+	.nid = 0x07, /* NID to query formats and rates */
+	.ops = {
+		.prepare = alc880_capture_pcm_prepare,
+		.cleanup = alc880_capture_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream alc880_pcm_digital_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	/* NID is set in alc_build_pcms */
+	.ops = {
+		.open = alc880_dig_playback_pcm_open,
+		.close = alc880_dig_playback_pcm_close
+	},
+};
+
+static struct hda_pcm_stream alc880_pcm_digital_capture = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	/* NID is set in alc_build_pcms */
+};
+
+static int alc_build_pcms(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	struct hda_pcm *info = spec->pcm_rec;
+	int i;
+
+	codec->num_pcms = 1;
+	codec->pcm_info = info;
+
+	info->name = spec->stream_name_analog;
+	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_analog_playback);
+	info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_analog_capture);
+
+	info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = 0;
+	for (i = 0; i < spec->num_channel_mode; i++) {
+		if (spec->channel_mode[i].channels > info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max) {
+		    info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->channel_mode[i].channels;
+		}
+	}
+
+	if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
+		codec->num_pcms++;
+		info++;
+		info->name = spec->stream_name_digital;
+		if (spec->multiout.dig_out_nid) {
+			info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback);
+			info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
+		}
+		if (spec->dig_in_nid) {
+			info->stream[SNDRV_PCM_STREAM_CAPTURE] = *(spec->stream_digital_capture);
+			info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid;
+		}
+	}
+
+	return 0;
+}
+
+static void alc_free(struct hda_codec *codec)
+{
+	kfree(codec->spec);
+}
+
+/*
+ */
+static struct hda_codec_ops alc_patch_ops = {
+	.build_controls = alc_build_controls,
+	.build_pcms = alc_build_pcms,
+	.init = alc_init,
+	.free = alc_free,
+#ifdef CONFIG_PM
+	.resume = alc_resume,
+#endif
+};
+
+/*
+ */
+
+static struct hda_board_config alc880_cfg_tbl[] = {
+	/* Back 3 jack, front 2 jack */
+	{ .modelname = "3stack", .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe200, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe201, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe202, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe203, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe204, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe205, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe206, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe207, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe208, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe209, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe20a, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe20b, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe20c, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe20d, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe20e, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe20f, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe210, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe211, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe214, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe302, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe303, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe304, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe306, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe307, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe404, .config = ALC880_3ST },
+	{ .pci_vendor = 0x8086, .pci_device = 0xa101, .config = ALC880_3ST },
+	{ .pci_vendor = 0x107b, .pci_device = 0x3031, .config = ALC880_3ST },
+	{ .pci_vendor = 0x107b, .pci_device = 0x4036, .config = ALC880_3ST },
+	{ .pci_vendor = 0x107b, .pci_device = 0x4037, .config = ALC880_3ST },
+	{ .pci_vendor = 0x107b, .pci_device = 0x4038, .config = ALC880_3ST },
+	{ .pci_vendor = 0x107b, .pci_device = 0x4040, .config = ALC880_3ST },
+	{ .pci_vendor = 0x107b, .pci_device = 0x4041, .config = ALC880_3ST },
+
+	/* Back 3 jack, front 2 jack (Internal add Aux-In) */
+	{ .pci_vendor = 0x1025, .pci_device = 0xe310, .config = ALC880_3ST },
+
+	/* Back 3 jack plus 1 SPDIF out jack, front 2 jack */
+	{ .modelname = "3stack-digout", .config = ALC880_3ST_DIG },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe308, .config = ALC880_3ST_DIG },
+
+	/* Back 3 jack plus 1 SPDIF out jack, front 2 jack (Internal add Aux-In)*/
+	{ .pci_vendor = 0x8086, .pci_device = 0xe305, .config = ALC880_3ST_DIG },
+	{ .pci_vendor = 0x8086, .pci_device = 0xd402, .config = ALC880_3ST_DIG },
+	{ .pci_vendor = 0x1025, .pci_device = 0xe309, .config = ALC880_3ST_DIG },
+
+	/* Back 5 jack, front 2 jack */
+	{ .modelname = "5stack", .config = ALC880_5ST },
+	{ .pci_vendor = 0x107b, .pci_device = 0x3033, .config = ALC880_5ST },
+	{ .pci_vendor = 0x107b, .pci_device = 0x4039, .config = ALC880_5ST },
+	{ .pci_vendor = 0x107b, .pci_device = 0x3032, .config = ALC880_5ST },
+	{ .pci_vendor = 0x103c, .pci_device = 0x2a09, .config = ALC880_5ST },
+
+	/* Back 5 jack plus 1 SPDIF out jack, front 2 jack */
+	{ .modelname = "5stack-digout", .config = ALC880_5ST_DIG },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe224, .config = ALC880_5ST_DIG },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe400, .config = ALC880_5ST_DIG },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe401, .config = ALC880_5ST_DIG },
+	{ .pci_vendor = 0x8086, .pci_device = 0xe402, .config = ALC880_5ST_DIG },
+	{ .pci_vendor = 0x8086, .pci_device = 0xd400, .config = ALC880_5ST_DIG },
+	{ .pci_vendor = 0x8086, .pci_device = 0xd401, .config = ALC880_5ST_DIG },
+	{ .pci_vendor = 0x8086, .pci_device = 0xa100, .config = ALC880_5ST_DIG },
+	{ .pci_vendor = 0x1565, .pci_device = 0x8202, .config = ALC880_5ST_DIG },
+
+	{ .modelname = "w810", .config = ALC880_W810 },
+	{ .pci_vendor = 0x161f, .pci_device = 0x203d, .config = ALC880_W810 },
+
+	{}
+};
+
+static int patch_alc880(struct hda_codec *codec)
+{
+	struct alc_spec *spec;
+	int board_config;
+
+	spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
+	if (spec == NULL)
+		return -ENOMEM;
+
+	codec->spec = spec;
+
+	board_config = snd_hda_check_board_config(codec, alc880_cfg_tbl);
+	if (board_config < 0) {
+		snd_printd(KERN_INFO "hda_codec: Unknown model for ALC880\n");
+		board_config = ALC880_MINIMAL;
+	}
+
+	switch (board_config) {
+	case ALC880_W810:
+		spec->mixers[spec->num_mixers] = alc880_w810_base_mixer;
+		spec->num_mixers++;
+		break;
+	case ALC880_5ST:
+	case ALC880_5ST_DIG:
+		spec->mixers[spec->num_mixers] = alc880_five_stack_mixer;
+		spec->num_mixers++;
+		break;
+	default:
+		spec->mixers[spec->num_mixers] = alc880_base_mixer;
+		spec->num_mixers++;
+		break;
+	}
+
+	switch (board_config) {
+	case ALC880_3ST_DIG:
+	case ALC880_5ST_DIG:
+	case ALC880_W810:
+		spec->multiout.dig_out_nid = ALC880_DIGOUT_NID;
+		break;
+	default:
+		break;
+	}
+
+	switch (board_config) {
+	case ALC880_3ST:
+	case ALC880_3ST_DIG:
+	case ALC880_5ST:
+	case ALC880_5ST_DIG:
+	case ALC880_W810:
+		spec->front_panel = 1;
+		break;
+	default:
+		break;
+	}
+
+	switch (board_config) {
+	case ALC880_5ST:
+	case ALC880_5ST_DIG:
+		spec->init_verbs = alc880_init_verbs_five_stack;
+		spec->channel_mode = alc880_fivestack_modes;
+		spec->num_channel_mode = ARRAY_SIZE(alc880_fivestack_modes);
+		break;
+	case ALC880_W810:
+		spec->init_verbs = alc880_w810_init_verbs;
+		spec->channel_mode = alc880_w810_modes;
+		spec->num_channel_mode = ARRAY_SIZE(alc880_w810_modes);
+		break;
+	default:
+		spec->init_verbs = alc880_init_verbs_three_stack;
+		spec->channel_mode = alc880_threestack_modes;
+		spec->num_channel_mode = ARRAY_SIZE(alc880_threestack_modes);
+		break;
+	}
+
+	spec->stream_name_analog = "ALC880 Analog";
+	spec->stream_analog_playback = &alc880_pcm_analog_playback;
+	spec->stream_analog_capture = &alc880_pcm_analog_capture;
+
+	spec->stream_name_digital = "ALC880 Digital";
+	spec->stream_digital_playback = &alc880_pcm_digital_playback;
+	spec->stream_digital_capture = &alc880_pcm_digital_capture;
+
+	spec->multiout.max_channels = spec->channel_mode[0].channels;
+
+	switch (board_config) {
+	case ALC880_W810:
+		spec->multiout.num_dacs = ARRAY_SIZE(alc880_w810_dac_nids);
+		spec->multiout.dac_nids = alc880_w810_dac_nids;
+		// No dedicated headphone socket - it's shared with built-in speakers.
+		break;
+	default:
+		spec->multiout.num_dacs = ARRAY_SIZE(alc880_dac_nids);
+		spec->multiout.dac_nids = alc880_dac_nids;
+		spec->multiout.hp_nid = 0x03; /* rear-surround NID */
+		break;
+	}
+
+	spec->input_mux = &alc880_capture_source;
+	spec->num_adc_nids = ARRAY_SIZE(alc880_adc_nids);
+	spec->adc_nids = alc880_adc_nids;
+
+	codec->patch_ops = alc_patch_ops;
+
+	return 0;
+}
+
+/*
+ * ALC260 support
+ */
+
+/*
+ * This is just place-holder, so there's something for alc_build_pcms to look
+ * at when it calculates the maximum number of channels. ALC260 has no mixer
+ * element which allows changing the channel mode, so the verb list is
+ * never used.
+ */
+static struct alc_channel_mode alc260_modes[1] = {
+	{ 2, NULL },
+};
+
+snd_kcontrol_new_t alc260_base_mixer[] = {
+	HDA_CODEC_VOLUME("Front Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+	/* use LINE2 for the output */
+	/* HDA_CODEC_MUTE("Front Playback Switch", 0x0f, 0x0, HDA_OUTPUT), */
+	HDA_CODEC_MUTE("Front Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
+	HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x07, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x07, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x07, 0x05, HDA_INPUT),
+	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x07, 0x05, HDA_INPUT),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x09, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Headphone Playback Switch", 0x10, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x11, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Capture Volume", 0x04, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x04, 0x0, HDA_INPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Capture Source",
+		.info = alc_mux_enum_info,
+		.get = alc_mux_enum_get,
+		.put = alc_mux_enum_put,
+	},
+	{ } /* end */
+};
+
+static struct hda_verb alc260_init_verbs[] = {
+	/* Line In pin widget for input */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+	/* CD pin widget for input */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+	/* Mic1 (rear panel) pin widget for input and vref at 80% */
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	/* Mic2 (front panel) pin widget for input and vref at 80% */
+	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	/* LINE-2 is used for line-out in rear */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	/* select line-out */
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* LINE-OUT pin */
+	{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	/* enable HP */
+	{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	/* enable Mono */
+	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	/* unmute amp left and right */
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
+	/* set connection select to line in (default select for this ADC) */
+	{0x04, AC_VERB_SET_CONNECT_SEL, 0x02},
+	/* unmute Line-Out mixer amp left and right (volume = 0) */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* mute pin widget amp left and right (no gain on this amp) */
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* unmute HP mixer amp left and right (volume = 0) */
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* mute pin widget amp left and right (no gain on this amp) */
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	/* unmute Mono mixer amp left and right (volume = 0) */
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	/* mute pin widget amp left and right (no gain on this amp) */
+	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	/* mute LINE-2 out */
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
+	/* Amp Indexes: CD = 0x04, Line In 1 = 0x02, Mic 1 = 0x00 & Line In 2 = 0x03 */
+	/* unmute CD */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x04 << 8))},
+	/* unmute Line In */
+	{0x07,  AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))},
+	/* unmute Mic */
+	{0x07,  AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	/* Amp Indexes: DAC = 0x01 & mixer = 0x00 */
+	/* Unmute Front out path */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	/* Unmute Headphone out path */
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	/* Unmute Mono out path */
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	{ }
+};
+
+static struct hda_pcm_stream alc260_pcm_analog_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.nid = 0x2,
+};
+
+static struct hda_pcm_stream alc260_pcm_analog_capture = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.nid = 0x4,
+};
+
+static int patch_alc260(struct hda_codec *codec)
+{
+	struct alc_spec *spec;
+
+	spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
+	if (spec == NULL)
+		return -ENOMEM;
+
+	codec->spec = spec;
+
+	spec->mixers[spec->num_mixers] = alc260_base_mixer;
+	spec->num_mixers++;
+
+	spec->init_verbs = alc260_init_verbs;
+	spec->channel_mode = alc260_modes;
+	spec->num_channel_mode = ARRAY_SIZE(alc260_modes);
+
+	spec->stream_name_analog = "ALC260 Analog";
+	spec->stream_analog_playback = &alc260_pcm_analog_playback;
+	spec->stream_analog_capture = &alc260_pcm_analog_capture;
+
+	spec->multiout.max_channels = spec->channel_mode[0].channels;
+	spec->multiout.num_dacs = ARRAY_SIZE(alc260_dac_nids);
+	spec->multiout.dac_nids = alc260_dac_nids;
+
+	spec->input_mux = &alc260_capture_source;
+	spec->num_adc_nids = ARRAY_SIZE(alc260_adc_nids);
+	spec->adc_nids = alc260_adc_nids;
+
+	codec->patch_ops = alc_patch_ops;
+
+	return 0;
+}
+
+/*
+ * ALC882 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.
+ * 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).
+ */
+
+static struct alc_channel_mode alc882_ch_modes[1] = {
+	{ 8, NULL }
+};
+
+static hda_nid_t alc882_dac_nids[4] = {
+	/* front, rear, clfe, rear_surr */
+	0x02, 0x03, 0x04, 0x05
+};
+
+static hda_nid_t alc882_adc_nids[3] = {
+	/* ADC0-2 */
+	0x07, 0x08, 0x09,
+};
+
+/* input MUX */
+/* FIXME: should be a matrix-type input source selection */
+
+static struct hda_input_mux alc882_capture_source = {
+	.num_items = 4,
+	.items = {
+		{ "Mic", 0x0 },
+		{ "Front Mic", 0x1 },
+		{ "Line", 0x2 },
+		{ "CD", 0x4 },
+	},
+};
+
+#define alc882_mux_enum_info alc_mux_enum_info
+#define alc882_mux_enum_get alc_mux_enum_get
+
+static int alc882_mux_enum_put(snd_kcontrol_t *kcontrol, snd_ctl_elem_value_t *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct alc_spec *spec = codec->spec;
+	const struct hda_input_mux *imux = spec->input_mux;
+	unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+	static hda_nid_t capture_mixers[3] = { 0x24, 0x23, 0x22 };
+	hda_nid_t nid = capture_mixers[adc_idx];
+	unsigned int *cur_val = &spec->cur_mux[adc_idx];
+	unsigned int i, idx;
+
+	idx = ucontrol->value.enumerated.item[0];
+	if (idx >= imux->num_items)
+		idx = imux->num_items - 1;
+	if (*cur_val == idx && ! codec->in_resume)
+		return 0;
+	for (i = 0; i < imux->num_items; i++) {
+		unsigned int v = (i == idx) ? 0x7000 : 0x7080;
+		snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE,
+				    v | (imux->items[i].index << 8));
+	}
+	*cur_val = idx;
+	return 1;
+}
+
+/* Pin assignment: Front=0x14, Rear=0x15, CLFE=0x16, Side=0x17
+ *                 Mic=0x18, Front Mic=0x19, Line-In=0x1a, HP=0x1b
+ */
+static snd_kcontrol_new_t alc882_base_mixer[] = {
+	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Front Playback Switch", 0x14, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0d, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT),
+	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_CODEC_MUTE_MONO("Center Playback Switch", 0x16, 1, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x16, 2, 0x0, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Side Playback Volume", 0x0f, 0x0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("Side Playback Switch", 0x17, 0x0, HDA_OUTPUT),
+	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_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
+	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
+	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
+	HDA_CODEC_VOLUME("Capture Volume", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Capture Switch", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x08, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x08, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x09, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x09, 0x0, HDA_INPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		/* .name = "Capture Source", */
+		.name = "Input Source",
+		.count = 3,
+		.info = alc882_mux_enum_info,
+		.get = alc882_mux_enum_get,
+		.put = alc882_mux_enum_put,
+	},
+	{ } /* end */
+};
+
+static struct hda_verb alc882_init_verbs[] = {
+	/* Front mixer: unmute input/output amp left and right (volume = 0) */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	/* Rear mixer */
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	/* CLFE mixer */
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+	/* Side mixer */
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))},
+
+	/* Front Pin: to output mode */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	/* Front Pin: mute amp left and right (no volume) */
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+	/* select Front mixer (0x0c, index 0) */
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* Rear Pin */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	/* Rear Pin: mute amp left and right (no volume) */
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+	/* select Rear mixer (0x0d, index 1) */
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
+	/* CLFE Pin */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	/* CLFE Pin: mute amp left and right (no volume) */
+	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+	/* select CLFE mixer (0x0e, index 2) */
+	{0x16, AC_VERB_SET_CONNECT_SEL, 0x02},
+	/* Side Pin */
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	/* Side Pin: mute amp left and right (no volume) */
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+	/* select Side mixer (0x0f, index 3) */
+	{0x17, AC_VERB_SET_CONNECT_SEL, 0x03},
+	/* Headphone Pin */
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40},
+	/* Headphone Pin: mute amp left and right (no volume) */
+	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, 0x0000},
+	/* select Front mixer (0x0c, index 0) */
+	{0x1b, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* Mic (rear) pin widget for input and vref at 80% */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	/* Front Mic pin widget for input and vref at 80% */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
+	/* Line In pin widget for input */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+	/* CD pin widget for input */
+	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
+
+	/* 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))},
+	/* ADC1: unmute amp left and right */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
+	/* ADC2: unmute amp left and right */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
+	/* ADC3: unmute amp left and right */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0x7000},
+
+	/* Unmute front loopback */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	/* Unmute rear loopback */
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+	/* Mute CLFE loopback */
+	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, (0x7080 | (0x01 << 8))},
+	/* Unmute side loopback */
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))},
+
+	{ }
+};
+
+static int patch_alc882(struct hda_codec *codec)
+{
+	struct alc_spec *spec;
+
+	spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
+	if (spec == NULL)
+		return -ENOMEM;
+
+	codec->spec = spec;
+
+	spec->mixers[spec->num_mixers] = alc882_base_mixer;
+	spec->num_mixers++;
+
+	spec->multiout.dig_out_nid = ALC880_DIGOUT_NID;
+	spec->dig_in_nid = ALC880_DIGIN_NID;
+	spec->front_panel = 1;
+	spec->init_verbs = alc882_init_verbs;
+	spec->channel_mode = alc882_ch_modes;
+	spec->num_channel_mode = ARRAY_SIZE(alc882_ch_modes);
+
+	spec->stream_name_analog = "ALC882 Analog";
+	spec->stream_analog_playback = &alc880_pcm_analog_playback;
+	spec->stream_analog_capture = &alc880_pcm_analog_capture;
+
+	spec->stream_name_digital = "ALC882 Digital";
+	spec->stream_digital_playback = &alc880_pcm_digital_playback;
+	spec->stream_digital_capture = &alc880_pcm_digital_capture;
+
+	spec->multiout.max_channels = spec->channel_mode[0].channels;
+	spec->multiout.num_dacs = ARRAY_SIZE(alc882_dac_nids);
+	spec->multiout.dac_nids = alc882_dac_nids;
+
+	spec->input_mux = &alc882_capture_source;
+	spec->num_adc_nids = ARRAY_SIZE(alc882_adc_nids);
+	spec->adc_nids = alc882_adc_nids;
+
+	codec->patch_ops = alc_patch_ops;
+
+	return 0;
+}
+
+/*
+ * patch entries
+ */
+struct hda_codec_preset snd_hda_preset_realtek[] = {
+	{ .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 },
+ 	{ .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
+	{ .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
+	{} /* terminator */
+};