Merge remote-tracking branch 'asoc/fix/fsl' into asoc-linus

63 files changed, 2170 insertions, 5013 deletions

diff --git a/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
new file mode 100644
index 000000000000..7d13479f9c3c
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/imx-audio-spdif.txt
@@ -0,0 +1,34 @@
+Freescale i.MX audio complex with S/PDIF transceiver
+
+Required properties:
+
+  - compatible : "fsl,imx-audio-spdif"
+
+  - model : The user-visible name of this sound complex
+
+  - spdif-controller : The phandle of the i.MX S/PDIF controller
+
+
+Optional properties:
+
+  - spdif-out : This is a boolean property. If present, the transmitting
+    function of S/PDIF will be enabled, indicating there's a physical
+    S/PDIF out connector/jack on the board or it's connecting to some
+    other IP block, such as an HDMI encoder/display-controller.
+
+  - spdif-in : This is a boolean property. If present, the receiving
+    function of S/PDIF will be enabled, indicating there's a physical
+    S/PDIF in connector/jack on the board.
+
+* Note: At least one of these two properties should be set in the DT binding.
+
+
+Example:
+
+sound-spdif {
+	compatible = "fsl,imx-audio-spdif";
+	model = "imx-spdif";
+	spdif-controller = <&spdif>;
+	spdif-out;
+	spdif-in;
+};
diff --git a/Documentation/devicetree/bindings/sound/mvebu-audio.txt b/Documentation/devicetree/bindings/sound/mvebu-audio.txt
new file mode 100644
index 000000000000..7e5fd37c1b3f
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/mvebu-audio.txt
@@ -0,0 +1,29 @@
+* mvebu (Kirkwood, Dove, Armada 370) audio controller
+
+Required properties:
+
+- compatible: "marvell,mvebu-audio"
+
+- reg: physical base address of the controller and length of memory mapped
+  region.
+
+- interrupts: list of two irq numbers.
+  The first irq is used for data flow and the second one is used for errors.
+
+- clocks: one or two phandles.
+  The first one is mandatory and defines the internal clock.
+  The second one is optional and defines an external clock.
+
+- clock-names: names associated to the clocks:
+	"internal" for the internal clock
+	"extclk" for the external clock
+
+Example:
+
+i2s1: audio-controller@b4000 {
+	compatible = "marvell,mvebu-audio";
+	reg = <0xb4000 0x2210>;
+	interrupts = <21>, <22>;
+	clocks = <&gate_clk 13>;
+	clock-names = "internal";
+};
diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt
index 809d72b8eff1..a46ddb85e83a 100644
--- a/Documentation/sound/alsa/HD-Audio-Models.txt
+++ b/Documentation/sound/alsa/HD-Audio-Models.txt
@@ -244,6 +244,7 @@ STAC9227/9228/9229/927x
   5stack-no-fp	D965 5stack without front panel
   dell-3stack	Dell Dimension E520
   dell-bios	Fixes with Dell BIOS setup
+  dell-bios-amic Fixes with Dell BIOS setup including analog mic
   volknob	Fixes with volume-knob widget 0x24
   auto		BIOS setup (default)
 
diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt
index c3c912d023cc..42a0a39b77e6 100644
--- a/Documentation/sound/alsa/HD-Audio.txt
+++ b/Documentation/sound/alsa/HD-Audio.txt
@@ -454,6 +454,8 @@ The generic parser supports the following hints:
 - need_dac_fix (bool): limits the DACs depending on the channel count
 - primary_hp (bool): probe headphone jacks as the primary outputs;
   default true
+- multi_io (bool): try probing multi-I/O config (e.g. shared
+  line-in/surround, mic/clfe jacks)
 - multi_cap_vol (bool): provide multiple capture volumes
 - inv_dmic_split (bool): provide split internal mic volume/switch for
   phase-inverted digital mics
diff --git a/MAINTAINERS b/MAINTAINERS
index b5e09128898f..a77b9440d87d 100644
--- a/MAINTAINERS
+++ b/MAINTAINERS
@@ -7676,6 +7676,17 @@ F:	include/sound/
 F:	include/uapi/sound/
 F:	sound/
 
+SOUND - COMPRESSED AUDIO
+M:	Vinod Koul <vinod.koul@intel.com>
+L:	alsa-devel@alsa-project.org (moderated for non-subscribers)
+T:	git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git
+S:	Supported
+F:	Documentation/sound/alsa/compress_offload.txt
+F:	include/sound/compress_driver.h
+F:	include/uapi/sound/compress_*
+F:	sound/core/compress_offload.c
+F:	sound/soc/soc-compress.c
+
 SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC)
 M:	Liam Girdwood <lgirdwood@gmail.com>
 M:	Mark Brown <broonie@kernel.org>
diff --git a/arch/arm/plat-samsung/s3c-dma-ops.c b/arch/arm/plat-samsung/s3c-dma-ops.c
index 0cc40aea3f5a..98b10ba67dc7 100644
--- a/arch/arm/plat-samsung/s3c-dma-ops.c
+++ b/arch/arm/plat-samsung/s3c-dma-ops.c
@@ -82,7 +82,8 @@ static int s3c_dma_config(unsigned ch, struct samsung_dma_config *param)
 static int s3c_dma_prepare(unsigned ch, struct samsung_dma_prep *param)
 {
 	struct cb_data *data;
-	int len = (param->cap == DMA_CYCLIC) ? param->period : param->len;
+	dma_addr_t pos = param->buf;
+	dma_addr_t end = param->buf + param->len;
 
 	list_for_each_entry(data, &dma_list, node)
 		if (data->ch == ch)
@@ -94,7 +95,15 @@ static int s3c_dma_prepare(unsigned ch, struct samsung_dma_prep *param)
 		data->fp_param = param->fp_param;
 	}
 
-	s3c2410_dma_enqueue(ch, (void *)data, param->buf, len);
+	if (param->cap != DMA_CYCLIC) {
+		s3c2410_dma_enqueue(ch, (void *)data, param->buf, param->len);
+		return 0;
+	}
+
+	while (pos < end) {
+		s3c2410_dma_enqueue(ch, (void *)data, pos, param->period);
+		pos += param->period;
+	}
 
 	return 0;
 }
diff --git a/include/sound/core.h b/include/sound/core.h
index c586617cfa0d..2a14f1f02d4f 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -27,6 +27,7 @@
 #include <linux/rwsem.h>		/* struct rw_semaphore */
 #include <linux/pm.h>			/* pm_message_t */
 #include <linux/stringify.h>
+#include <linux/printk.h>
 
 /* number of supported soundcards */
 #ifdef CONFIG_SND_DYNAMIC_MINORS
@@ -376,6 +377,11 @@ void __snd_printk(unsigned int level, const char *file, int line,
 #define snd_BUG()		WARN(1, "BUG?\n")
 
 /**
+ * Suppress high rates of output when CONFIG_SND_DEBUG is enabled.
+ */
+#define snd_printd_ratelimit() printk_ratelimit()
+
+/**
  * snd_BUG_ON - debugging check macro
  * @cond: condition to evaluate
  *
@@ -398,6 +404,8 @@ static inline void _snd_printd(int level, const char *format, ...) {}
 	unlikely(__ret_warn_on); \
 })
 
+static inline bool snd_printd_ratelimit(void) { return false; }
+
 #endif /* CONFIG_SND_DEBUG */
 
 #ifdef CONFIG_SND_DEBUG_VERBOSE
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index c728d28ae9a5..27a72d5d4b00 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -413,7 +413,7 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card,
 			 struct snd_soc_dapm_widget *sink);
 
 /* dapm path setup */
-int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm);
+int snd_soc_dapm_new_widgets(struct snd_soc_card *card);
 void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm);
 int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm,
 			    const struct snd_soc_dapm_route *route, int num);
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 8e2ad52078b6..d22cb0a06feb 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -697,7 +697,6 @@ struct snd_soc_codec {
 	unsigned int probed:1; /* Codec has been probed */
 	unsigned int ac97_registered:1; /* Codec has been AC97 registered */
 	unsigned int ac97_created:1; /* Codec has been created by SoC */
-	unsigned int sysfs_registered:1; /* codec has been sysfs registered */
 	unsigned int cache_init:1; /* codec cache has been initialized */
 	unsigned int using_regmap:1; /* using regmap access */
 	u32 cache_only;  /* Suppress writes to hardware */
@@ -705,7 +704,6 @@ struct snd_soc_codec {
 
 	/* codec IO */
 	void *control_data; /* codec control (i2c/3wire) data */
-	enum snd_soc_control_type control_type;
 	hw_write_t hw_write;
 	unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int);
 	unsigned int (*read)(struct snd_soc_codec *, unsigned int);
@@ -724,7 +722,6 @@ struct snd_soc_codec {
 #ifdef CONFIG_DEBUG_FS
 	struct dentry *debugfs_codec_root;
 	struct dentry *debugfs_reg;
-	struct dentry *debugfs_dapm;
 #endif
 };
 
@@ -849,7 +846,6 @@ struct snd_soc_platform {
 
 #ifdef CONFIG_DEBUG_FS
 	struct dentry *debugfs_platform_root;
-	struct dentry *debugfs_dapm;
 #endif
 };
 
@@ -934,6 +930,10 @@ struct snd_soc_dai_link {
 	/* machine stream operations */
 	const struct snd_soc_ops *ops;
 	const struct snd_soc_compr_ops *compr_ops;
+
+	/* For unidirectional dai links */
+	bool playback_only;
+	bool capture_only;
 };
 
 struct snd_soc_codec_conf {
diff --git a/include/uapi/sound/hdspm.h b/include/uapi/sound/hdspm.h
index 1f59ea2a4a76..d956c3593f65 100644
--- a/include/uapi/sound/hdspm.h
+++ b/include/uapi/sound/hdspm.h
@@ -111,7 +111,7 @@ struct hdspm_ltc {
 	enum hdspm_ltc_input_format input_format;
 };
 
-#define SNDRV_HDSPM_IOCTL_GET_LTC _IOR('H', 0x46, struct hdspm_mixer_ioctl)
+#define SNDRV_HDSPM_IOCTL_GET_LTC _IOR('H', 0x46, struct hdspm_ltc)
 
 /**
  * The status data reflects the device's current state
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 82bb029d4414..6e03b465e44e 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -184,7 +184,7 @@ static void xrun(struct snd_pcm_substream *substream)
 	do {								\
 		if (xrun_debug(substream, XRUN_DEBUG_BASIC)) {		\
 			xrun_log_show(substream);			\
-			if (printk_ratelimit()) {			\
+			if (snd_printd_ratelimit()) {			\
 				snd_printd("PCM: " fmt, ##args);	\
 			}						\
 			dump_stack_on_xrun(substream);			\
@@ -342,7 +342,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 		return -EPIPE;
 	}
 	if (pos >= runtime->buffer_size) {
-		if (printk_ratelimit()) {
+		if (snd_printd_ratelimit()) {
 			char name[16];
 			snd_pcm_debug_name(substream, name, sizeof(name));
 			xrun_log_show(substream);
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index 11048cc744d0..915b4d7fbb23 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -1022,7 +1022,7 @@ static void dummy_proc_write(struct snd_info_entry *entry,
 		if (i >= ARRAY_SIZE(fields))
 			continue;
 		snd_info_get_str(item, ptr, sizeof(item));
-		if (strict_strtoull(item, 0, &val))
+		if (kstrtoull(item, 0, &val))
 			continue;
 		if (fields[i].size == sizeof(int))
 			*get_dummy_int_ptr(dummy, fields[i].offset) = val;
diff --git a/sound/firewire/speakers.c b/sound/firewire/speakers.c
index 2c6386503940..fe9e6e2f2c5b 100644
--- a/sound/firewire/speakers.c
+++ b/sound/firewire/speakers.c
@@ -49,7 +49,6 @@ struct fwspk {
 	struct snd_card *card;
 	struct fw_unit *unit;
 	const struct device_info *device_info;
-	struct snd_pcm_substream *pcm;
 	struct mutex mutex;
 	struct cmp_connection connection;
 	struct amdtp_out_stream stream;
@@ -363,8 +362,7 @@ static int fwspk_create_pcm(struct fwspk *fwspk)
 		return err;
 	pcm->private_data = fwspk;
 	strcpy(pcm->name, fwspk->device_info->short_name);
-	fwspk->pcm = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
-	fwspk->pcm->ops = &ops;
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &ops);
 	return 0;
 }
 
diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c
index 9942691cc0ca..afef0d738078 100644
--- a/sound/isa/gus/interwave.c
+++ b/sound/isa/gus/interwave.c
@@ -443,8 +443,7 @@ static void snd_interwave_detect_memory(struct snd_gus_card *gus)
 		for (i = 0; i < 8; ++i)
 			iwave[i] = snd_gf1_peek(gus, bank_pos + i);
 #ifdef CONFIG_SND_DEBUG_ROM
-		printk(KERN_DEBUG "ROM at 0x%06x = %*phC\n", bank_pos,
-				  8, iwave);
+		printk(KERN_DEBUG "ROM at 0x%06x = %8phC\n", bank_pos, iwave);
 #endif
 		if (strncmp(iwave, "INTRWAVE", 8))
 			continue;	/* first check */
diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c
index a59c88818f48..461d94cfecbe 100644
--- a/sound/oss/dmabuf.c
+++ b/sound/oss/dmabuf.c
@@ -557,7 +557,6 @@ int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock)
 	unsigned long flags;
 	int err = 0, n = 0;
 	struct dma_buffparms *dmap = adev->dmap_in;
-	int go;
 
 	if (!(adev->open_mode & OPEN_READ))
 		return -EIO;
@@ -584,7 +583,7 @@ int DMAbuf_getrdbuffer(int dev, char **buf, int *len, int dontblock)
 			spin_unlock_irqrestore(&dmap->lock,flags);
 			return -EAGAIN;
 		}
-		if ((go = adev->go))
+		if (adev->go)
 			timeout = dmabuf_timeout(dmap);
 
 		spin_unlock_irqrestore(&dmap->lock,flags);
diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig
index 59c5e9c03d53..8de66ccd7279 100644
--- a/sound/pci/hda/Kconfig
+++ b/sound/pci/hda/Kconfig
@@ -152,14 +152,9 @@ config SND_HDA_CODEC_HDMI
 	  This module is automatically loaded at probing.
 
 config SND_HDA_I915
-	bool "Build Display HD-audio controller/codec power well support for i915 cards"
+	bool
+	default y
 	depends on DRM_I915
-	help
-	  Say Y here to include full HDMI and DisplayPort HD-audio controller/codec
-	  power-well support for Intel Haswell graphics cards based on the i915 driver.
-
-	  Note that this option must be enabled for Intel Haswell C+ stepping machines, otherwise
-	  the GPU audio controller/codecs will not be initialized or damaged when exit from S3 mode.
 
 config SND_HDA_CODEC_CIRRUS
 	bool "Build Cirrus Logic codec support"
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index 8a005f0e5ca4..5b6c4e3c92ca 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -666,6 +666,64 @@ int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
 }
 EXPORT_SYMBOL_HDA(snd_hda_get_conn_index);
 
+
+/* return DEVLIST_LEN parameter of the given widget */
+static unsigned int get_num_devices(struct hda_codec *codec, hda_nid_t nid)
+{
+	unsigned int wcaps = get_wcaps(codec, nid);
+	unsigned int parm;
+
+	if (!codec->dp_mst || !(wcaps & AC_WCAP_DIGITAL) ||
+	    get_wcaps_type(wcaps) != AC_WID_PIN)
+		return 0;
+
+	parm = snd_hda_param_read(codec, nid, AC_PAR_DEVLIST_LEN);
+	if (parm == -1 && codec->bus->rirb_error)
+		parm = 0;
+	return parm & AC_DEV_LIST_LEN_MASK;
+}
+
+/**
+ * snd_hda_get_devices - copy device list without cache
+ * @codec: the HDA codec
+ * @nid: NID of the pin to parse
+ * @dev_list: device list array
+ * @max_devices: max. number of devices to store
+ *
+ * Copy the device list. This info is dynamic and so not cached.
+ * Currently called only from hda_proc.c, so not exported.
+ */
+int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid,
+			u8 *dev_list, int max_devices)
+{
+	unsigned int parm;
+	int i, dev_len, devices;
+
+	parm = get_num_devices(codec, nid);
+	if (!parm)	/* not multi-stream capable */
+		return 0;
+
+	dev_len = parm + 1;
+	dev_len = dev_len < max_devices ? dev_len : max_devices;
+
+	devices = 0;
+	while (devices < dev_len) {
+		parm = snd_hda_codec_read(codec, nid, 0,
+					  AC_VERB_GET_DEVICE_LIST, devices);
+		if (parm == -1 && codec->bus->rirb_error)
+			break;
+
+		for (i = 0; i < 8; i++) {
+			dev_list[devices] = (u8)parm;
+			parm >>= 4;
+			devices++;
+			if (devices >= dev_len)
+				break;
+		}
+	}
+	return devices;
+}
+
 /**
  * snd_hda_queue_unsol_event - add an unsolicited event to queue
  * @bus: the BUS
@@ -1216,11 +1274,13 @@ static void hda_jackpoll_work(struct work_struct *work)
 {
 	struct hda_codec *codec =
 		container_of(work, struct hda_codec, jackpoll_work.work);
-	if (!codec->jackpoll_interval)
-		return;
 
 	snd_hda_jack_set_dirty_all(codec);
 	snd_hda_jack_poll_all(codec);
+
+	if (!codec->jackpoll_interval)
+		return;
+
 	queue_delayed_work(codec->bus->workq, &codec->jackpoll_work,
 			   codec->jackpoll_interval);
 }
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 701c2e069b10..7aa9870040c1 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -94,6 +94,8 @@ enum {
 #define AC_VERB_GET_HDMI_DIP_XMIT		0x0f32
 #define AC_VERB_GET_HDMI_CP_CTRL		0x0f33
 #define AC_VERB_GET_HDMI_CHAN_SLOT		0x0f34
+#define AC_VERB_GET_DEVICE_SEL			0xf35
+#define AC_VERB_GET_DEVICE_LIST			0xf36
 
 /*
  * SET verbs
@@ -133,6 +135,7 @@ enum {
 #define AC_VERB_SET_HDMI_DIP_XMIT		0x732
 #define AC_VERB_SET_HDMI_CP_CTRL		0x733
 #define AC_VERB_SET_HDMI_CHAN_SLOT		0x734
+#define AC_VERB_SET_DEVICE_SEL			0x735
 
 /*
  * Parameter IDs
@@ -154,6 +157,7 @@ enum {
 #define AC_PAR_GPIO_CAP			0x11
 #define AC_PAR_AMP_OUT_CAP		0x12
 #define AC_PAR_VOL_KNB_CAP		0x13
+#define AC_PAR_DEVLIST_LEN		0x15
 #define AC_PAR_HDMI_LPCM_CAP		0x20
 
 /*
@@ -251,6 +255,11 @@ enum {
 #define AC_UNSOL_RES_TAG_SHIFT		26
 #define AC_UNSOL_RES_SUBTAG		(0x1f<<21)
 #define AC_UNSOL_RES_SUBTAG_SHIFT	21
+#define AC_UNSOL_RES_DE			(0x3f<<15)  /* Device Entry
+						     * (for DP1.2 MST)
+						     */
+#define AC_UNSOL_RES_DE_SHIFT		15
+#define AC_UNSOL_RES_IA			(1<<2)	/* Inactive (for DP1.2 MST) */
 #define AC_UNSOL_RES_ELDV		(1<<1)	/* ELD Data valid (for HDMI) */
 #define AC_UNSOL_RES_PD			(1<<0)	/* pinsense detect */
 #define AC_UNSOL_RES_CP_STATE		(1<<1)	/* content protection */
@@ -352,6 +361,10 @@ enum {
 #define AC_LPCMCAP_44K			(1<<30)	/* 44.1kHz support */
 #define AC_LPCMCAP_44K_MS		(1<<31)	/* 44.1kHz-multiplies support */
 
+/* Display pin's device list length */
+#define AC_DEV_LIST_LEN_MASK		0x3f
+#define AC_MAX_DEV_LIST_LEN		64
+
 /*
  * Control Parameters
  */
@@ -460,6 +473,11 @@ enum {
 #define AC_DEFCFG_PORT_CONN		(0x3<<30)
 #define AC_DEFCFG_PORT_CONN_SHIFT	30
 
+/* Display pin's device list entry */
+#define AC_DE_PD			(1<<0)
+#define AC_DE_ELDV			(1<<1)
+#define AC_DE_IA			(1<<2)
+
 /* device device types (0x0-0xf) */
 enum {
 	AC_JACK_LINE_OUT,
@@ -885,6 +903,7 @@ struct hda_codec {
 	unsigned int pcm_format_first:1; /* PCM format must be set first */
 	unsigned int epss:1;		/* supporting EPSS? */
 	unsigned int cached_write:1;	/* write only to caches */
+	unsigned int dp_mst:1; /* support DP1.2 Multi-stream transport */
 #ifdef CONFIG_PM
 	unsigned int power_on :1;	/* current (global) power-state */
 	unsigned int d3_stop_clk:1;	/* support D3 operation without BCLK */
@@ -972,6 +991,8 @@ int snd_hda_override_conn_list(struct hda_codec *codec, hda_nid_t nid, int nums,
 			  const hda_nid_t *list);
 int snd_hda_get_conn_index(struct hda_codec *codec, hda_nid_t mux,
 			   hda_nid_t nid, int recursive);
+int snd_hda_get_devices(struct hda_codec *codec, hda_nid_t nid,
+			u8 *dev_list, int max_devices);
 int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
 				u32 *ratesp, u64 *formatsp, unsigned int *bpsp);
 
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index e3c7ba8d7582..ac41e9cdc976 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -142,6 +142,9 @@ static void parse_user_hints(struct hda_codec *codec)
 	val = snd_hda_get_bool_hint(codec, "primary_hp");
 	if (val >= 0)
 		spec->no_primary_hp = !val;
+	val = snd_hda_get_bool_hint(codec, "multi_io");
+	if (val >= 0)
+		spec->no_multi_io = !val;
 	val = snd_hda_get_bool_hint(codec, "multi_cap_vol");
 	if (val >= 0)
 		spec->multi_cap_vol = !!val;
@@ -813,6 +816,8 @@ static void resume_path_from_idx(struct hda_codec *codec, int path_idx)
 
 static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
 				  struct snd_ctl_elem_value *ucontrol);
+static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol);
 
 enum {
 	HDA_CTL_WIDGET_VOL,
@@ -830,7 +835,13 @@ static const struct snd_kcontrol_new control_templates[] = {
 		.put = hda_gen_mixer_mute_put, /* replaced */
 		.private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0),
 	},
-	HDA_BIND_MUTE(NULL, 0, 0, 0),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.info = snd_hda_mixer_amp_switch_info,
+		.get = snd_hda_mixer_bind_switch_get,
+		.put = hda_gen_bind_mute_put, /* replaced */
+		.private_value = HDA_COMPOSE_AMP_VAL(0, 3, 0, 0),
+	},
 };
 
 /* add dynamic controls from template */
@@ -937,8 +948,8 @@ static int add_stereo_sw(struct hda_codec *codec, const char *pfx,
 }
 
 /* playback mute control with the software mute bit check */
-static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
-				  struct snd_ctl_elem_value *ucontrol)
+static void sync_auto_mute_bits(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
 {
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct hda_gen_spec *spec = codec->spec;
@@ -949,10 +960,22 @@ static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
 		ucontrol->value.integer.value[0] &= enabled;
 		ucontrol->value.integer.value[1] &= enabled;
 	}
+}
 
+static int hda_gen_mixer_mute_put(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_value *ucontrol)
+{
+	sync_auto_mute_bits(kcontrol, ucontrol);
 	return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
 }
 
+static int hda_gen_bind_mute_put(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_value *ucontrol)
+{
+	sync_auto_mute_bits(kcontrol, ucontrol);
+	return snd_hda_mixer_bind_switch_put(kcontrol, ucontrol);
+}
+
 /* any ctl assigned to the path with the given index? */
 static bool path_has_mixer(struct hda_codec *codec, int path_idx, int ctl_type)
 {
@@ -1541,7 +1564,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
 					      cfg->speaker_pins,
 					      spec->multiout.extra_out_nid,
 					      spec->speaker_paths);
-			if (fill_mio_first && cfg->line_outs == 1 &&
+			if (!spec->no_multi_io &&
+			    fill_mio_first && cfg->line_outs == 1 &&
 			    cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
 				err = fill_multi_ios(codec, cfg->line_out_pins[0], true);
 				if (!err)
@@ -1554,7 +1578,7 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
 				   spec->private_dac_nids, spec->out_paths,
 				   spec->main_out_badness);
 
-	if (fill_mio_first &&
+	if (!spec->no_multi_io && fill_mio_first &&
 	    cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
 		/* try to fill multi-io first */
 		err = fill_multi_ios(codec, cfg->line_out_pins[0], false);
@@ -1582,7 +1606,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
 			return err;
 		badness += err;
 	}
-	if (cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
+	if (!spec->no_multi_io &&
+	    cfg->line_outs == 1 && cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) {
 		err = fill_multi_ios(codec, cfg->line_out_pins[0], false);
 		if (err < 0)
 			return err;
@@ -1600,7 +1625,8 @@ static int fill_and_eval_dacs(struct hda_codec *codec,
 				check_aamix_out_path(codec, spec->speaker_paths[0]);
 	}
 
-	if (cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
+	if (!spec->no_multi_io &&
+	    cfg->hp_outs && cfg->line_out_type == AUTO_PIN_SPEAKER_OUT)
 		if (count_multiio_pins(codec, cfg->hp_pins[0]) >= 2)
 			spec->multi_ios = 1; /* give badness */
 
@@ -3724,7 +3750,8 @@ static int mux_select(struct hda_codec *codec, unsigned int adc_idx,
 /* check each pin in the given array; returns true if any of them is plugged */
 static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins)
 {
-	int i, present = 0;
+	int i;
+	bool present = false;
 
 	for (i = 0; i < num_pins; i++) {
 		hda_nid_t nid = pins[i];
@@ -3733,14 +3760,15 @@ static bool detect_jacks(struct hda_codec *codec, int num_pins, hda_nid_t *pins)
 		/* don't detect pins retasked as inputs */
 		if (snd_hda_codec_get_pin_target(codec, nid) & AC_PINCTL_IN_EN)
 			continue;
-		present |= snd_hda_jack_detect(codec, nid);
+		if (snd_hda_jack_detect_state(codec, nid) == HDA_JACK_PRESENT)
+			present = true;
 	}
 	return present;
 }
 
 /* standard HP/line-out auto-mute helper */
 static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
-			bool mute)
+			int *paths, bool mute)
 {
 	struct hda_gen_spec *spec = codec->spec;
 	int i;
@@ -3752,10 +3780,19 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
 			break;
 
 		if (spec->auto_mute_via_amp) {
+			struct nid_path *path;
+			hda_nid_t mute_nid;
+
+			path = snd_hda_get_path_from_idx(codec, paths[i]);
+			if (!path)
+				continue;
+			mute_nid = get_amp_nid_(path->ctls[NID_PATH_MUTE_CTL]);
+			if (!mute_nid)
+				continue;
 			if (mute)
-				spec->mute_bits |= (1ULL << nid);
+				spec->mute_bits |= (1ULL << mute_nid);
 			else
-				spec->mute_bits &= ~(1ULL << nid);
+				spec->mute_bits &= ~(1ULL << mute_nid);
 			set_pin_eapd(codec, nid, !mute);
 			continue;
 		}
@@ -3786,14 +3823,19 @@ static void do_automute(struct hda_codec *codec, int num_pins, hda_nid_t *pins,
 void snd_hda_gen_update_outputs(struct hda_codec *codec)
 {
 	struct hda_gen_spec *spec = codec->spec;
+	int *paths;
 	int on;
 
 	/* Control HP pins/amps depending on master_mute state;
 	 * in general, HP pins/amps control should be enabled in all cases,
 	 * but currently set only for master_mute, just to be safe
 	 */
+	if (spec->autocfg.line_out_type == AUTO_PIN_HP_OUT)
+		paths = spec->out_paths;
+	else
+		paths = spec->hp_paths;
 	do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins),
-		    spec->autocfg.hp_pins, spec->master_mute);
+		    spec->autocfg.hp_pins, paths, spec->master_mute);
 
 	if (!spec->automute_speaker)
 		on = 0;
@@ -3801,8 +3843,12 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec)
 		on = spec->hp_jack_present | spec->line_jack_present;
 	on |= spec->master_mute;
 	spec->speaker_muted = on;
+	if (spec->autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT)
+		paths = spec->out_paths;
+	else
+		paths = spec->speaker_paths;
 	do_automute(codec, ARRAY_SIZE(spec->autocfg.speaker_pins),
-		    spec->autocfg.speaker_pins, on);
+		    spec->autocfg.speaker_pins, paths, on);
 
 	/* toggle line-out mutes if needed, too */
 	/* if LO is a copy of either HP or Speaker, don't need to handle it */
@@ -3815,8 +3861,9 @@ void snd_hda_gen_update_outputs(struct hda_codec *codec)
 		on = spec->hp_jack_present;
 	on |= spec->master_mute;
 	spec->line_out_muted = on;
+	paths = spec->out_paths;
 	do_automute(codec, ARRAY_SIZE(spec->autocfg.line_out_pins),
-		    spec->autocfg.line_out_pins, on);
+		    spec->autocfg.line_out_pins, paths, on);
 }
 EXPORT_SYMBOL_HDA(snd_hda_gen_update_outputs);
 
@@ -3887,7 +3934,7 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, struct hda_jack_tbl *ja
 		/* don't detect pins retasked as outputs */
 		if (snd_hda_codec_get_pin_target(codec, pin) & AC_PINCTL_OUT_EN)
 			continue;
-		if (snd_hda_jack_detect(codec, pin)) {
+		if (snd_hda_jack_detect_state(codec, pin) == HDA_JACK_PRESENT) {
 			mux_select(codec, 0, spec->am_entry[i].idx);
 			return;
 		}
diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h
index e199a852388b..48d44026705b 100644
--- a/sound/pci/hda/hda_generic.h
+++ b/sound/pci/hda/hda_generic.h
@@ -220,6 +220,7 @@ struct hda_gen_spec {
 	unsigned int hp_mic:1; /* Allow HP as a mic-in */
 	unsigned int suppress_hp_mic_detect:1; /* Don't detect HP/mic */
 	unsigned int no_primary_hp:1; /* Don't prefer HP pins to speaker pins */
+	unsigned int no_multi_io:1; /* Don't try multi I/O config */
 	unsigned int multi_cap_vol:1; /* allow multiple capture xxx volumes */
 	unsigned int inv_dmic_split:1; /* inverted dmic w/a for conexant */
 	unsigned int own_eapd_ctl:1; /* set EAPD by own function */
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index ce67608734b5..fe0bda19de15 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -295,7 +295,7 @@ static ssize_t type##_store(struct device *dev,			\
 	struct snd_hwdep *hwdep = dev_get_drvdata(dev);		\
 	struct hda_codec *codec = hwdep->private_data;		\
 	unsigned long val;					\
-	int err = strict_strtoul(buf, 0, &val);			\
+	int err = kstrtoul(buf, 0, &val);			\
 	if (err < 0)						\
 		return err;					\
 	codec->type = val;					\
@@ -654,7 +654,7 @@ int snd_hda_get_int_hint(struct hda_codec *codec, const char *key, int *valp)
 	p = snd_hda_get_hint(codec, key);
 	if (!p)
 		ret = -ENOENT;
-	else if (strict_strtoul(p, 0, &val))
+	else if (kstrtoul(p, 0, &val))
 		ret = -EINVAL;
 	else {
 		*valp = val;
@@ -751,7 +751,7 @@ static void parse_##name##_mode(char *buf, struct hda_bus *bus, \
 				 struct hda_codec **codecp) \
 { \
 	unsigned long val; \
-	if (!strict_strtoul(buf, 0, &val)) \
+	if (!kstrtoul(buf, 0, &val)) \
 		(*codecp)->name = val; \
 }
 
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 8860dd529520..c6c98298ac39 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1160,7 +1160,7 @@ static int azx_reset(struct azx *chip, int full_reset)
 		goto __skip;
 
 	/* clear STATESTS */
-	azx_writeb(chip, STATESTS, STATESTS_INT_MASK);
+	azx_writew(chip, STATESTS, STATESTS_INT_MASK);
 
 	/* reset controller */
 	azx_enter_link_reset(chip);
@@ -1242,7 +1242,7 @@ static void azx_int_clear(struct azx *chip)
 	}
 
 	/* clear STATESTS */
-	azx_writeb(chip, STATESTS, STATESTS_INT_MASK);
+	azx_writew(chip, STATESTS, STATESTS_INT_MASK);
 
 	/* clear rirb status */
 	azx_writeb(chip, RIRBSTS, RIRB_INT_MASK);
@@ -1451,8 +1451,8 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
 
 #if 0
 	/* clear state status int */
-	if (azx_readb(chip, STATESTS) & 0x04)
-		azx_writeb(chip, STATESTS, 0x04);
+	if (azx_readw(chip, STATESTS) & 0x04)
+		azx_writew(chip, STATESTS, 0x04);
 #endif
 	spin_unlock(&chip->reg_lock);
 	
@@ -2971,6 +2971,10 @@ static int azx_runtime_suspend(struct device *dev)
 	struct snd_card *card = dev_get_drvdata(dev);
 	struct azx *chip = card->private_data;
 
+	/* enable controller wake up event */
+	azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) |
+		  STATESTS_INT_MASK);
+
 	azx_stop_chip(chip);
 	azx_enter_link_reset(chip);
 	azx_clear_irq_pending(chip);
@@ -2983,11 +2987,31 @@ static int azx_runtime_resume(struct device *dev)
 {
 	struct snd_card *card = dev_get_drvdata(dev);
 	struct azx *chip = card->private_data;
+	struct hda_bus *bus;
+	struct hda_codec *codec;
+	int status;
 
 	if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL)
 		hda_display_power(true);
+
+	/* Read STATESTS before controller reset */
+	status = azx_readw(chip, STATESTS);
+
 	azx_init_pci(chip);
 	azx_init_chip(chip, 1);
+
+	bus = chip->bus;
+	if (status && bus) {
+		list_for_each_entry(codec, &bus->codec_list, list)
+			if (status & (1 << codec->addr))
+				queue_delayed_work(codec->bus->workq,
+						   &codec->jackpoll_work, codec->jackpoll_interval);
+	}
+
+	/* disable controller Wake Up event*/
+	azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) &
+			~STATESTS_INT_MASK);
+
 	return 0;
 }
 
@@ -3831,11 +3855,13 @@ static int azx_probe_continue(struct azx *chip)
 
 	/* Request power well for Haswell HDA controller and codec */
 	if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) {
+#ifdef CONFIG_SND_HDA_I915
 		err = hda_i915_init();
 		if (err < 0) {
 			snd_printk(KERN_ERR SFX "Error request power-well from i915\n");
 			goto out_free;
 		}
+#endif
 		hda_display_power(true);
 	}
 
diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c
index 3fd2973183e2..05b3e3e9108f 100644
--- a/sound/pci/hda/hda_jack.c
+++ b/sound/pci/hda/hda_jack.c
@@ -194,18 +194,24 @@ u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid)
 EXPORT_SYMBOL_HDA(snd_hda_pin_sense);
 
 /**
- * snd_hda_jack_detect - query pin Presence Detect status
+ * snd_hda_jack_detect_state - query pin Presence Detect status
  * @codec: the CODEC to sense
  * @nid: the pin NID to sense
  *
- * Query and return the pin's Presence Detect status.
+ * Query and return the pin's Presence Detect status, as either
+ * HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT or HDA_JACK_PHANTOM.
  */
-int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid)
+int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid)
 {
-	u32 sense = snd_hda_pin_sense(codec, nid);
-	return get_jack_plug_state(sense);
+	struct hda_jack_tbl *jack = snd_hda_jack_tbl_get(codec, nid);
+	if (jack && jack->phantom_jack)
+		return HDA_JACK_PHANTOM;
+	else if (snd_hda_pin_sense(codec, nid) & AC_PINSENSE_PRESENCE)
+		return HDA_JACK_PRESENT;
+	else
+		return HDA_JACK_NOT_PRESENT;
 }
-EXPORT_SYMBOL_HDA(snd_hda_jack_detect);
+EXPORT_SYMBOL_HDA(snd_hda_jack_detect_state);
 
 /**
  * snd_hda_jack_detect_enable - enable the jack-detection
@@ -247,8 +253,8 @@ EXPORT_SYMBOL_HDA(snd_hda_jack_detect_enable);
 int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid,
 				 hda_nid_t gating_nid)
 {
-	struct hda_jack_tbl *gated = snd_hda_jack_tbl_get(codec, gated_nid);
-	struct hda_jack_tbl *gating = snd_hda_jack_tbl_get(codec, gating_nid);
+	struct hda_jack_tbl *gated = snd_hda_jack_tbl_new(codec, gated_nid);
+	struct hda_jack_tbl *gating = snd_hda_jack_tbl_new(codec, gating_nid);
 
 	if (!gated || !gating)
 		return -EINVAL;
diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h
index ec12abd45263..379420c44eef 100644
--- a/sound/pci/hda/hda_jack.h
+++ b/sound/pci/hda/hda_jack.h
@@ -75,7 +75,18 @@ int snd_hda_jack_set_gating_jack(struct hda_codec *codec, hda_nid_t gated_nid,
 				 hda_nid_t gating_nid);
 
 u32 snd_hda_pin_sense(struct hda_codec *codec, hda_nid_t nid);
-int snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid);
+
+/* the jack state returned from snd_hda_jack_detect_state() */
+enum {
+	HDA_JACK_NOT_PRESENT, HDA_JACK_PRESENT, HDA_JACK_PHANTOM,
+};
+
+int snd_hda_jack_detect_state(struct hda_codec *codec, hda_nid_t nid);
+
+static inline bool snd_hda_jack_detect(struct hda_codec *codec, hda_nid_t nid)
+{
+	return snd_hda_jack_detect_state(codec, nid) != HDA_JACK_NOT_PRESENT;
+}
 
 bool is_jack_detectable(struct hda_codec *codec, hda_nid_t nid);
 
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 9760f001916d..a8cb22eec89e 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -582,6 +582,36 @@ static void print_gpio(struct snd_info_buffer *buffer,
 	print_nid_array(buffer, codec, nid, &codec->nids);
 }
 
+static void print_device_list(struct snd_info_buffer *buffer,
+			    struct hda_codec *codec, hda_nid_t nid)
+{
+	int i, curr = -1;
+	u8 dev_list[AC_MAX_DEV_LIST_LEN];
+	int devlist_len;
+
+	devlist_len = snd_hda_get_devices(codec, nid, dev_list,
+					AC_MAX_DEV_LIST_LEN);
+	snd_iprintf(buffer, "  Devices: %d\n", devlist_len);
+	if (devlist_len <= 0)
+		return;
+
+	curr = snd_hda_codec_read(codec, nid, 0,
+				AC_VERB_GET_DEVICE_SEL, 0);
+
+	for (i = 0; i < devlist_len; i++) {
+		if (i == curr)
+			snd_iprintf(buffer, "    *");
+		else
+			snd_iprintf(buffer, "     ");
+
+		snd_iprintf(buffer,
+			"Dev %02d: PD = %d, ELDV = %d, IA = %d\n", i,
+			!!(dev_list[i] & AC_DE_PD),
+			!!(dev_list[i] & AC_DE_ELDV),
+			!!(dev_list[i] & AC_DE_IA));
+	}
+}
+
 static void print_codec_info(struct snd_info_entry *entry,
 			     struct snd_info_buffer *buffer)
 {
@@ -751,6 +781,9 @@ static void print_codec_info(struct snd_info_entry *entry,
 				    (wid_caps & AC_WCAP_DELAY) >>
 				    AC_WCAP_DELAY_SHIFT);
 
+		if (wid_type == AC_WID_PIN && codec->dp_mst)
+			print_device_list(buffer, codec, nid);
+
 		if (wid_caps & AC_WCAP_CONN_LIST)
 			print_conn_list(buffer, codec, nid, wid_type,
 					conn, conn_len);
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index d97f0d61a15b..0cbdd87dde6d 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -32,7 +32,6 @@
 #include "hda_jack.h"
 #include "hda_generic.h"
 
-#define ENABLE_AD_STATIC_QUIRKS
 
 struct ad198x_spec {
 	struct hda_gen_spec gen;
@@ -43,114 +42,8 @@ struct ad198x_spec {
 	hda_nid_t eapd_nid;
 
 	unsigned int beep_amp;	/* beep amp value, set via set_beep_amp() */
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-	const struct snd_kcontrol_new *mixers[6];
-	int num_mixers;
-	const struct hda_verb *init_verbs[6];	/* initialization verbs
-						 * don't forget NULL termination!
-						 */
-	unsigned int num_init_verbs;
-
-	/* playback */
-	struct hda_multi_out multiout;	/* playback set-up
-					 * max_channels, dacs must be set
-					 * dig_out_nid and hp_nid are optional
-					 */
-	unsigned int cur_eapd;
-	unsigned int need_dac_fix;
-
-	/* capture */
-	unsigned int num_adc_nids;
-	const hda_nid_t *adc_nids;
-	hda_nid_t dig_in_nid;		/* digital-in NID; optional */
-
-	/* capture source */
-	const struct hda_input_mux *input_mux;
-	const hda_nid_t *capsrc_nids;
-	unsigned int cur_mux[3];
-
-	/* channel model */
-	const struct hda_channel_mode *channel_mode;
-	int num_channel_mode;
-
-	/* PCM information */
-	struct hda_pcm pcm_rec[3];	/* used in alc_build_pcms() */
-
-	unsigned int spdif_route;
-
-	unsigned int jack_present: 1;
-	unsigned int inv_jack_detect: 1;/* inverted jack-detection */
-	unsigned int analog_beep: 1;	/* analog beep input present */
-	unsigned int avoid_init_slave_vol:1;
-
-#ifdef CONFIG_PM
-	struct hda_loopback_check loopback;
-#endif
-	/* for virtual master */
-	hda_nid_t vmaster_nid;
-	const char * const *slave_vols;
-	const char * const *slave_sws;
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-};
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-/*
- * input MUX handling (common part)
- */
-static int ad198x_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-
-	return snd_hda_input_mux_info(spec->input_mux, uinfo);
-}
-
-static int ad198x_mux_enum_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_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 ad198x_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_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->capsrc_nids[adc_idx],
-				     &spec->cur_mux[adc_idx]);
-}
-
-/*
- * initialization (common callbacks)
- */
-static int ad198x_init(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-	int i;
-
-	for (i = 0; i < spec->num_init_verbs; i++)
-		snd_hda_sequence_write(codec, spec->init_verbs[i]);
-	return 0;
-}
-
-static const char * const ad_slave_pfxs[] = {
-	"Front", "Surround", "Center", "LFE", "Side",
-	"Headphone", "Mono", "Speaker", "IEC958",
-	NULL
 };
 
-static const char * const ad1988_6stack_fp_slave_pfxs[] = {
-	"Front", "Surround", "Center", "LFE", "Side", "IEC958",
-	NULL
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
 
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
 /* additional beep mixers; the actual parameters are overwritten at build */
@@ -160,12 +53,6 @@ static const struct snd_kcontrol_new ad_beep_mixer[] = {
 	{ } /* end */
 };
 
-static const struct snd_kcontrol_new ad_beep2_mixer[] = {
-	HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0, 0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_BEEP("Digital Beep Playback Switch", 0, 0, HDA_OUTPUT),
-	{ } /* end */
-};
-
 #define set_beep_amp(spec, nid, idx, dir) \
 	((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir)) /* mono */
 #else
@@ -181,8 +68,7 @@ static int create_beep_ctls(struct hda_codec *codec)
 	if (!spec->beep_amp)
 		return 0;
 
-	knew = spec->analog_beep ? ad_beep2_mixer : ad_beep_mixer;
-	for ( ; knew->name; knew++) {
+	for (knew = ad_beep_mixer ; knew->name; knew++) {
 		int err;
 		struct snd_kcontrol *kctl;
 		kctl = snd_ctl_new1(knew, codec);
@@ -199,268 +85,6 @@ static int create_beep_ctls(struct hda_codec *codec)
 #define create_beep_ctls(codec)		0
 #endif
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int ad198x_build_controls(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-	struct snd_kcontrol *kctl;
-	unsigned int i;
-	int err;
-
-	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,
-						    spec->multiout.dig_out_nid);
-		if (err < 0)
-			return err;
-		err = snd_hda_create_spdif_share_sw(codec,
-						    &spec->multiout);
-		if (err < 0)
-			return err;
-		spec->multiout.share_spdif = 1;
-	} 
-	if (spec->dig_in_nid) {
-		err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid);
-		if (err < 0)
-			return err;
-	}
-
-	/* create beep controls if needed */
-	err = create_beep_ctls(codec);
-	if (err < 0)
-		return err;
-
-	/* if we have no master control, let's create it */
-	if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) {
-		unsigned int vmaster_tlv[4];
-		snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid,
-					HDA_OUTPUT, vmaster_tlv);
-		err = __snd_hda_add_vmaster(codec, "Master Playback Volume",
-					  vmaster_tlv,
-					  (spec->slave_vols ?
-					   spec->slave_vols : ad_slave_pfxs),
-					  "Playback Volume",
-					  !spec->avoid_init_slave_vol, NULL);
-		if (err < 0)
-			return err;
-	}
-	if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
-		err = snd_hda_add_vmaster(codec, "Master Playback Switch",
-					  NULL,
-					  (spec->slave_sws ?
-					   spec->slave_sws : ad_slave_pfxs),
-					  "Playback Switch");
-		if (err < 0)
-			return err;
-	}
-
-	/* assign Capture Source enums to NID */
-	kctl = snd_hda_find_mixer_ctl(codec, "Capture Source");
-	if (!kctl)
-		kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
-	for (i = 0; kctl && i < kctl->count; i++) {
-		err = snd_hda_add_nid(codec, kctl, i, spec->capsrc_nids[i]);
-		if (err < 0)
-			return err;
-	}
-
-	/* assign IEC958 enums to NID */
-	kctl = snd_hda_find_mixer_ctl(codec,
-			SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source");
-	if (kctl) {
-		err = snd_hda_add_nid(codec, kctl, 0,
-				      spec->multiout.dig_out_nid);
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
-#ifdef CONFIG_PM
-static int ad198x_check_power_status(struct hda_codec *codec, hda_nid_t nid)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_check_amp_list_power(codec, &spec->loopback, nid);
-}
-#endif
-
-/*
- * Analog playback callbacks
- */
-static int ad198x_playback_pcm_open(struct hda_pcm_stream *hinfo,
-				    struct hda_codec *codec,
-				    struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_analog_open(codec, &spec->multiout, substream,
-					     hinfo);
-}
-
-static int ad198x_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
-				       struct hda_codec *codec,
-				       unsigned int stream_tag,
-				       unsigned int format,
-				       struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_analog_prepare(codec, &spec->multiout, stream_tag,
-						format, substream);
-}
-
-static int ad198x_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
-				       struct hda_codec *codec,
-				       struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_analog_cleanup(codec, &spec->multiout);
-}
-
-/*
- * Digital out
- */
-static int ad198x_dig_playback_pcm_open(struct hda_pcm_stream *hinfo,
-					struct hda_codec *codec,
-					struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_dig_open(codec, &spec->multiout);
-}
-
-static int ad198x_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
-					 struct hda_codec *codec,
-					 struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_dig_close(codec, &spec->multiout);
-}
-
-static int ad198x_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
-					   struct hda_codec *codec,
-					   unsigned int stream_tag,
-					   unsigned int format,
-					   struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, stream_tag,
-					     format, substream);
-}
-
-static int ad198x_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
-					   struct hda_codec *codec,
-					   struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
-}
-
-/*
- * Analog capture
- */
-static int ad198x_capture_pcm_prepare(struct hda_pcm_stream *hinfo,
-				      struct hda_codec *codec,
-				      unsigned int stream_tag,
-				      unsigned int format,
-				      struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	snd_hda_codec_setup_stream(codec, spec->adc_nids[substream->number],
-				   stream_tag, 0, format);
-	return 0;
-}
-
-static int ad198x_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
-				      struct hda_codec *codec,
-				      struct snd_pcm_substream *substream)
-{
-	struct ad198x_spec *spec = codec->spec;
-	snd_hda_codec_cleanup_stream(codec, spec->adc_nids[substream->number]);
-	return 0;
-}
-
-/*
- */
-static const struct hda_pcm_stream ad198x_pcm_analog_playback = {
-	.substreams = 1,
-	.channels_min = 2,
-	.channels_max = 6, /* changed later */
-	.nid = 0, /* fill later */
-	.ops = {
-		.open = ad198x_playback_pcm_open,
-		.prepare = ad198x_playback_pcm_prepare,
-		.cleanup = ad198x_playback_pcm_cleanup,
-	},
-};
-
-static const struct hda_pcm_stream ad198x_pcm_analog_capture = {
-	.substreams = 1,
-	.channels_min = 2,
-	.channels_max = 2,
-	.nid = 0, /* fill later */
-	.ops = {
-		.prepare = ad198x_capture_pcm_prepare,
-		.cleanup = ad198x_capture_pcm_cleanup
-	},
-};
-
-static const struct hda_pcm_stream ad198x_pcm_digital_playback = {
-	.substreams = 1,
-	.channels_min = 2,
-	.channels_max = 2,
-	.nid = 0, /* fill later */
-	.ops = {
-		.open = ad198x_dig_playback_pcm_open,
-		.close = ad198x_dig_playback_pcm_close,
-		.prepare = ad198x_dig_playback_pcm_prepare,
-		.cleanup = ad198x_dig_playback_pcm_cleanup
-	},
-};
-
-static const struct hda_pcm_stream ad198x_pcm_digital_capture = {
-	.substreams = 1,
-	.channels_min = 2,
-	.channels_max = 2,
-	/* NID is set in alc_build_pcms */
-};
-
-static int ad198x_build_pcms(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-	struct hda_pcm *info = spec->pcm_rec;
-
-	codec->num_pcms = 1;
-	codec->pcm_info = info;
-
-	info->name = "AD198x Analog";
-	info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_pcm_analog_playback;
-	info->stream[SNDRV_PCM_STREAM_PLAYBACK].channels_max = spec->multiout.max_channels;
-	info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dac_nids[0];
-	info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad198x_pcm_analog_capture;
-	info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adc_nids;
-	info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0];
-
-	if (spec->multiout.dig_out_nid) {
-		info++;
-		codec->num_pcms++;
-		codec->spdif_status_reset = 1;
-		info->name = "AD198x Digital";
-		info->pcm_type = HDA_PCM_TYPE_SPDIF;
-		info->stream[SNDRV_PCM_STREAM_PLAYBACK] = ad198x_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] = ad198x_pcm_digital_capture;
-			info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->dig_in_nid;
-		}
-	}
-
-	return 0;
-}
-#endif /* ENABLE_AD_STATIC_QUIRKS */
 
 static void ad198x_power_eapd_write(struct hda_codec *codec, hda_nid_t front,
 				hda_nid_t hp)
@@ -507,18 +131,6 @@ static void ad198x_shutup(struct hda_codec *codec)
 	ad198x_power_eapd(codec);
 }
 
-static void ad198x_free(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-
-	if (!spec)
-		return;
-
-	snd_hda_gen_spec_free(&spec->gen);
-	kfree(spec);
-	snd_hda_detach_beep_device(codec);
-}
-
 #ifdef CONFIG_PM
 static int ad198x_suspend(struct hda_codec *codec)
 {
@@ -527,65 +139,6 @@ static int ad198x_suspend(struct hda_codec *codec)
 }
 #endif
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const struct hda_codec_ops ad198x_patch_ops = {
-	.build_controls = ad198x_build_controls,
-	.build_pcms = ad198x_build_pcms,
-	.init = ad198x_init,
-	.free = ad198x_free,
-#ifdef CONFIG_PM
-	.check_power_status = ad198x_check_power_status,
-	.suspend = ad198x_suspend,
-#endif
-	.reboot_notify = ad198x_shutup,
-};
-
-
-/*
- * EAPD control
- * the private value = nid
- */
-#define ad198x_eapd_info	snd_ctl_boolean_mono_info
-
-static int ad198x_eapd_get(struct snd_kcontrol *kcontrol,
-			   struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-	if (codec->inv_eapd)
-		ucontrol->value.integer.value[0] = ! spec->cur_eapd;
-	else
-		ucontrol->value.integer.value[0] = spec->cur_eapd;
-	return 0;
-}
-
-static int ad198x_eapd_put(struct snd_kcontrol *kcontrol,
-			   struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-	hda_nid_t nid = kcontrol->private_value & 0xff;
-	unsigned int eapd;
-	eapd = !!ucontrol->value.integer.value[0];
-	if (codec->inv_eapd)
-		eapd = !eapd;
-	if (eapd == spec->cur_eapd)
-		return 0;
-	spec->cur_eapd = eapd;
-	snd_hda_codec_write_cache(codec, nid,
-				  0, AC_VERB_SET_EAPD_BTLENABLE,
-				  eapd ? 0x02 : 0x00);
-	return 1;
-}
-
-static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol,
-			       struct snd_ctl_elem_info *uinfo);
-static int ad198x_ch_mode_get(struct snd_kcontrol *kcontrol,
-			      struct snd_ctl_elem_value *ucontrol);
-static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol,
-			      struct snd_ctl_elem_value *ucontrol);
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 
 /*
  * Automatic parse of I/O pins from the BIOS configuration
@@ -646,537 +199,6 @@ static int ad198x_parse_auto_config(struct hda_codec *codec)
  * AD1986A specific
  */
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-#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 const hda_nid_t ad1986a_dac_nids[3] = {
-	AD1986A_FRONT_DAC, AD1986A_SURR_DAC, AD1986A_CLFE_DAC
-};
-static const hda_nid_t ad1986a_adc_nids[1] = { AD1986A_ADC };
-static const hda_nid_t ad1986a_capsrc_nids[1] = { 0x12 };
-
-static const 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 },
-	},
-};
-
-
-static const struct hda_bind_ctls ad1986a_bind_pcm_vol = {
-	.ops = &snd_hda_bind_vol,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
-static const struct hda_bind_ctls ad1986a_bind_pcm_sw = {
-	.ops = &snd_hda_bind_sw,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(AD1986A_FRONT_DAC, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(AD1986A_SURR_DAC, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(AD1986A_CLFE_DAC, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
-/*
- * mixers
- */
-static const struct snd_kcontrol_new ad1986a_mixers[] = {
-	/*
-	 * bind volumes/mutes of 3 DACs as a single PCM control for simplicity
-	 */
-	HDA_BIND_VOL("PCM Playback Volume", &ad1986a_bind_pcm_vol),
-	HDA_BIND_SW("PCM Playback Switch", &ad1986a_bind_pcm_sw),
-	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("Mic Boost Volume", 0x0f, 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 = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	HDA_CODEC_MUTE("Stereo Downmix Switch", 0x09, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-/* additional mixers for 3stack mode */
-static const struct snd_kcontrol_new ad1986a_3st_mixers[] = {
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Channel Mode",
-		.info = ad198x_ch_mode_info,
-		.get = ad198x_ch_mode_get,
-		.put = ad198x_ch_mode_put,
-	},
-	{ } /* end */
-};
-
-/* laptop model - 2ch only */
-static const hda_nid_t ad1986a_laptop_dac_nids[1] = { AD1986A_FRONT_DAC };
-
-/* master controls both pins 0x1a and 0x1b */
-static const struct hda_bind_ctls ad1986a_laptop_master_vol = {
-	.ops = &snd_hda_bind_vol,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
-		0,
-	},
-};
-
-static const struct hda_bind_ctls ad1986a_laptop_master_sw = {
-	.ops = &snd_hda_bind_sw,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x1b, 3, 0, HDA_OUTPUT),
-		0,
-	},
-};
-
-static const struct snd_kcontrol_new ad1986a_laptop_mixers[] = {
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 0x0, HDA_OUTPUT),
-	HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
-	HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
-	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("Mic Boost Volume", 0x0f, 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 = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	{ } /* end */
-};
-
-/* laptop-eapd model - 2ch only */
-
-static const struct hda_input_mux ad1986a_laptop_eapd_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Internal Mic", 0x4 },
-		{ "Mix", 0x5 },
-	},
-};
-
-static const struct hda_input_mux ad1986a_automic_capture_source = {
-	.num_items = 2,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Mix", 0x5 },
-	},
-};
-
-static const struct snd_kcontrol_new ad1986a_laptop_master_mixers[] = {
-	HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
-	HDA_BIND_SW("Master Playback Switch", &ad1986a_laptop_master_sw),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1986a_laptop_eapd_mixers[] = {
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x03, 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("Mic Boost Volume", 0x0f, 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 = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "External Amplifier",
-		.subdevice = HDA_SUBDEV_NID_FLAG | 0x1b,
-		.info = ad198x_eapd_info,
-		.get = ad198x_eapd_get,
-		.put = ad198x_eapd_put,
-		.private_value = 0x1b, /* port-D */
-	},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1986a_laptop_intmic_mixers[] = {
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x17, 0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x17, 0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-/* re-connect the mic boost input according to the jack sensing */
-static void ad1986a_automic(struct hda_codec *codec)
-{
-	unsigned int present;
-	present = snd_hda_jack_detect(codec, 0x1f);
-	/* 0 = 0x1f, 2 = 0x1d, 4 = mixed */
-	snd_hda_codec_write(codec, 0x0f, 0, AC_VERB_SET_CONNECT_SEL,
-			    present ? 0 : 2);
-}
-
-#define AD1986A_MIC_EVENT		0x36
-
-static void ad1986a_automic_unsol_event(struct hda_codec *codec,
-					    unsigned int res)
-{
-	if ((res >> 26) != AD1986A_MIC_EVENT)
-		return;
-	ad1986a_automic(codec);
-}
-
-static int ad1986a_automic_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1986a_automic(codec);
-	return 0;
-}
-
-/* laptop-automute - 2ch only */
-
-static void ad1986a_update_hp(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-	unsigned int mute;
-
-	if (spec->jack_present)
-		mute = HDA_AMP_MUTE; /* mute internal speaker */
-	else
-		/* unmute internal speaker if necessary */
-		mute = snd_hda_codec_amp_read(codec, 0x1a, 0, HDA_OUTPUT, 0);
-	snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, mute);
-}
-
-static void ad1986a_hp_automute(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-
-	spec->jack_present = snd_hda_jack_detect(codec, 0x1a);
-	if (spec->inv_jack_detect)
-		spec->jack_present = !spec->jack_present;
-	ad1986a_update_hp(codec);
-}
-
-#define AD1986A_HP_EVENT		0x37
-
-static void ad1986a_hp_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	if ((res >> 26) != AD1986A_HP_EVENT)
-		return;
-	ad1986a_hp_automute(codec);
-}
-
-static int ad1986a_hp_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1986a_hp_automute(codec);
-	return 0;
-}
-
-/* bind hp and internal speaker mute (with plug check) */
-static int ad1986a_hp_master_sw_put(struct snd_kcontrol *kcontrol,
-				    struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
-	if (change)
-		ad1986a_update_hp(codec);
-	return change;
-}
-
-static const struct snd_kcontrol_new ad1986a_automute_master_mixers[] = {
-	HDA_BIND_VOL("Master Playback Volume", &ad1986a_laptop_master_vol),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_AMP_FLAG,
-		.info = snd_hda_mixer_amp_switch_info,
-		.get = snd_hda_mixer_amp_switch_get,
-		.put = ad1986a_hp_master_sw_put,
-		.private_value = HDA_COMPOSE_AMP_VAL(0x1a, 3, 0, HDA_OUTPUT),
-	},
-	{ } /* end */
-};
-
-
-/*
- * initialization verbs
- */
-static const 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},
-	/* HP Pin */
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
-	/* Front, Surround, CLFE Pins */
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* Mono Pin */
-	{0x1e, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* Mic Pin */
-	{0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	/* Line, Aux, CD, Beep-In Pin */
-	{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	{0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	{0x22, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	{0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	{0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	{ } /* end */
-};
-
-static const struct hda_verb ad1986a_ch2_init[] = {
-	/* Surround out -> Line In */
-	{ 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
- 	/* Line-in selectors */
-	{ 0x10, AC_VERB_SET_CONNECT_SEL, 0x1 },
-	/* CLFE -> Mic in */
-	{ 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
-	/* Mic selector, mix C/LFE (backmic) and Mic (frontmic) */
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x4 },
-	{ } /* end */
-};
-
-static const struct hda_verb ad1986a_ch4_init[] = {
-	/* Surround out -> Surround */
-	{ 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 },
-	/* CLFE -> Mic in */
-	{ 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x4 },
-	{ } /* end */
-};
-
-static const struct hda_verb ad1986a_ch6_init[] = {
-	/* Surround out -> Surround out */
-	{ 0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x10, AC_VERB_SET_CONNECT_SEL, 0x0 },
-	/* CLFE -> CLFE */
-	{ 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x0 },
-	{ } /* end */
-};
-
-static const struct hda_channel_mode ad1986a_modes[3] = {
-	{ 2, ad1986a_ch2_init },
-	{ 4, ad1986a_ch4_init },
-	{ 6, ad1986a_ch6_init },
-};
-
-/* eapd initialization */
-static const struct hda_verb ad1986a_eapd_init_verbs[] = {
-	{0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 },
-	{}
-};
-
-static const struct hda_verb ad1986a_automic_verbs[] = {
-	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x1f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	/*{0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},*/
-	{0x0f, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x1f, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_MIC_EVENT},
-	{}
-};
-
-/* Ultra initialization */
-static const struct hda_verb ad1986a_ultra_init[] = {
-	/* eapd initialization */
-	{ 0x1b, AC_VERB_SET_EAPD_BTLENABLE, 0x00 },
-	/* CLFE -> Mic in */
-	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2 },
-	{ 0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	{ 0x1d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080 },
-	{ } /* end */
-};
-
-/* pin sensing on HP jack */
-static const struct hda_verb ad1986a_hp_init_verbs[] = {
-	{0x1a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1986A_HP_EVENT},
-	{}
-};
-
-static void ad1986a_samsung_p50_unsol_event(struct hda_codec *codec,
-					    unsigned int res)
-{
-	switch (res >> 26) {
-	case AD1986A_HP_EVENT:
-		ad1986a_hp_automute(codec);
-		break;
-	case AD1986A_MIC_EVENT:
-		ad1986a_automic(codec);
-		break;
-	}
-}
-
-static int ad1986a_samsung_p50_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1986a_hp_automute(codec);
-	ad1986a_automic(codec);
-	return 0;
-}
-
-
-/* models */
-enum {
-	AD1986A_AUTO,
-	AD1986A_6STACK,
-	AD1986A_3STACK,
-	AD1986A_LAPTOP,
-	AD1986A_LAPTOP_EAPD,
-	AD1986A_LAPTOP_AUTOMUTE,
-	AD1986A_ULTRA,
-	AD1986A_SAMSUNG,
-	AD1986A_SAMSUNG_P50,
-	AD1986A_MODELS
-};
-
-static const char * const ad1986a_models[AD1986A_MODELS] = {
-	[AD1986A_AUTO]		= "auto",
-	[AD1986A_6STACK]	= "6stack",
-	[AD1986A_3STACK]	= "3stack",
-	[AD1986A_LAPTOP]	= "laptop",
-	[AD1986A_LAPTOP_EAPD]	= "laptop-eapd",
-	[AD1986A_LAPTOP_AUTOMUTE] = "laptop-automute",
-	[AD1986A_ULTRA]		= "ultra",
-	[AD1986A_SAMSUNG]	= "samsung",
-	[AD1986A_SAMSUNG_P50]	= "samsung-p50",
-};
-
-static const struct snd_pci_quirk ad1986a_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x1153, "ASUS M9", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x11f7, "ASUS U5A", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x1213, "ASUS A6J", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x1263, "ASUS U5F", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x1297, "ASUS Z62F", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x12b3, "ASUS V1j", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x1302, "ASUS W3j", AD1986A_LAPTOP_EAPD),
-	SND_PCI_QUIRK(0x1043, 0x1443, "ASUS VX1", AD1986A_LAPTOP),
-	SND_PCI_QUIRK(0x1043, 0x1447, "ASUS A8J", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x1043, 0x817f, "ASUS P5", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x1043, 0x818f, "ASUS P5", AD1986A_LAPTOP),
-	SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS P5", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS M2N", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x1043, 0x8234, "ASUS M2N", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba Satellite L40-10Q", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP),
-	SND_PCI_QUIRK(0x144d, 0xc024, "Samsung P50", AD1986A_SAMSUNG_P50),
-	SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA),
-	SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_SAMSUNG),
-	SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP),
-	SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK),
-	SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_LAPTOP_AUTOMUTE),
-	SND_PCI_QUIRK(0x17c0, 0x2017, "Samsung M50", AD1986A_LAPTOP),
-	{}
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1986a_loopbacks[] = {
-	{ 0x13, HDA_OUTPUT, 0 }, /* Mic */
-	{ 0x14, HDA_OUTPUT, 0 }, /* Phone */
-	{ 0x15, HDA_OUTPUT, 0 }, /* CD */
-	{ 0x16, HDA_OUTPUT, 0 }, /* Aux */
-	{ 0x17, HDA_OUTPUT, 0 }, /* Line */
-	{ } /* end */
-};
-#endif
-
-static int is_jack_available(struct hda_codec *codec, hda_nid_t nid)
-{
-	unsigned int conf = snd_hda_codec_get_pincfg(codec, nid);
-	return get_defcfg_connect(conf) != AC_JACK_PORT_NONE;
-}
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 static int alloc_ad_spec(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
@@ -1203,6 +225,11 @@ static void ad_fixup_inv_jack_detect(struct hda_codec *codec,
 
 enum {
 	AD1986A_FIXUP_INV_JACK_DETECT,
+	AD1986A_FIXUP_ULTRA,
+	AD1986A_FIXUP_SAMSUNG,
+	AD1986A_FIXUP_3STACK,
+	AD1986A_FIXUP_LAPTOP,
+	AD1986A_FIXUP_LAPTOP_IMIC,
 };
 
 static const struct hda_fixup ad1986a_fixups[] = {
@@ -1210,16 +237,86 @@ static const struct hda_fixup ad1986a_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = ad_fixup_inv_jack_detect,
 	},
+	[AD1986A_FIXUP_ULTRA] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1b, 0x90170110 }, /* speaker */
+			{ 0x1d, 0x90a7013e }, /* int mic */
+			{}
+		},
+	},
+	[AD1986A_FIXUP_SAMSUNG] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1b, 0x90170110 }, /* speaker */
+			{ 0x1d, 0x90a7013e }, /* int mic */
+			{ 0x20, 0x411111f0 }, /* N/A */
+			{ 0x24, 0x411111f0 }, /* N/A */
+			{}
+		},
+	},
+	[AD1986A_FIXUP_3STACK] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1a, 0x02214021 }, /* headphone */
+			{ 0x1b, 0x01014011 }, /* front */
+			{ 0x1c, 0x01013012 }, /* surround */
+			{ 0x1d, 0x01019015 }, /* clfe */
+			{ 0x1e, 0x411111f0 }, /* N/A */
+			{ 0x1f, 0x02a190f0 }, /* mic */
+			{ 0x20, 0x018130f0 }, /* line-in */
+			{}
+		},
+	},
+	[AD1986A_FIXUP_LAPTOP] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1a, 0x02214021 }, /* headphone */
+			{ 0x1b, 0x90170110 }, /* speaker */
+			{ 0x1c, 0x411111f0 }, /* N/A */
+			{ 0x1d, 0x411111f0 }, /* N/A */
+			{ 0x1e, 0x411111f0 }, /* N/A */
+			{ 0x1f, 0x02a191f0 }, /* mic */
+			{ 0x20, 0x411111f0 }, /* N/A */
+			{}
+		},
+	},
+	[AD1986A_FIXUP_LAPTOP_IMIC] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1d, 0x90a7013e }, /* int mic */
+			{}
+		},
+		.chained_before = 1,
+		.chain_id = AD1986A_FIXUP_LAPTOP,
+	},
 };
 
 static const struct snd_pci_quirk ad1986a_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x103c, 0x30af, "HP B2800", AD1986A_FIXUP_LAPTOP_IMIC),
+	SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8100, "ASUS P5", AD1986A_FIXUP_3STACK),
+	SND_PCI_QUIRK_MASK(0x1043, 0xff00, 0x8200, "ASUS M2", AD1986A_FIXUP_3STACK),
+	SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_FIXUP_3STACK),
+	SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_FIXUP_LAPTOP),
+	SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_FIXUP_SAMSUNG),
+	SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_FIXUP_ULTRA),
 	SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_FIXUP_INV_JACK_DETECT),
+	SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_FIXUP_3STACK),
+	SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_FIXUP_3STACK),
+	{}
+};
+
+static const struct hda_model_fixup ad1986a_fixup_models[] = {
+	{ .id = AD1986A_FIXUP_3STACK, .name = "3stack" },
+	{ .id = AD1986A_FIXUP_LAPTOP, .name = "laptop" },
+	{ .id = AD1986A_FIXUP_LAPTOP_IMIC, .name = "laptop-imic" },
+	{ .id = AD1986A_FIXUP_LAPTOP_IMIC, .name = "laptop-eapd" }, /* alias */
 	{}
 };
 
 /*
  */
-static int ad1986a_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1986a(struct hda_codec *codec)
 {
 	int err;
 	struct ad198x_spec *spec;
@@ -1244,7 +341,8 @@ static int ad1986a_parse_auto_config(struct hda_codec *codec)
 	 */
 	spec->gen.multiout.no_share_stream = 1;
 
-	snd_hda_pick_fixup(codec, NULL, ad1986a_fixup_tbl, ad1986a_fixups);
+	snd_hda_pick_fixup(codec, ad1986a_fixup_models, ad1986a_fixup_tbl,
+			   ad1986a_fixups);
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
 
 	err = ad198x_parse_auto_config(codec);
@@ -1258,330 +356,11 @@ static int ad1986a_parse_auto_config(struct hda_codec *codec)
 	return 0;
 }
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1986a(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int err, board_config;
-
-	board_config = snd_hda_check_board_config(codec, AD1986A_MODELS,
-						  ad1986a_models,
-						  ad1986a_cfg_tbl);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1986A_AUTO;
-	}
-
-	if (board_config == AD1986A_AUTO)
-		return ad1986a_parse_auto_config(codec);
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	err = snd_hda_attach_beep_device(codec, 0x19);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x18, 0, HDA_OUTPUT);
-
-	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;
-	spec->num_adc_nids = 1;
-	spec->adc_nids = ad1986a_adc_nids;
-	spec->capsrc_nids = ad1986a_capsrc_nids;
-	spec->input_mux = &ad1986a_capture_source;
-	spec->num_mixers = 1;
-	spec->mixers[0] = ad1986a_mixers;
-	spec->num_init_verbs = 1;
-	spec->init_verbs[0] = ad1986a_init_verbs;
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1986a_loopbacks;
-#endif
-	spec->vmaster_nid = 0x1b;
-	codec->inv_eapd = 1; /* AD1986A has the inverted EAPD implementation */
-
-	codec->patch_ops = ad198x_patch_ops;
-
-	/* override some parameters */
-	switch (board_config) {
-	case AD1986A_3STACK:
-		spec->num_mixers = 2;
-		spec->mixers[1] = ad1986a_3st_mixers;
-		spec->num_init_verbs = 2;
-		spec->init_verbs[1] = ad1986a_ch2_init;
-		spec->channel_mode = ad1986a_modes;
-		spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes);
-		spec->need_dac_fix = 1;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		break;
-	case AD1986A_LAPTOP:
-		spec->mixers[0] = ad1986a_laptop_mixers;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
-		break;
-	case AD1986A_LAPTOP_EAPD:
-		spec->num_mixers = 3;
-		spec->mixers[0] = ad1986a_laptop_master_mixers;
-		spec->mixers[1] = ad1986a_laptop_eapd_mixers;
-		spec->mixers[2] = ad1986a_laptop_intmic_mixers;
-		spec->num_init_verbs = 2;
-		spec->init_verbs[1] = ad1986a_eapd_init_verbs;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
-		if (!is_jack_available(codec, 0x25))
-			spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1986a_laptop_eapd_capture_source;
-		break;
-	case AD1986A_SAMSUNG:
-		spec->num_mixers = 2;
-		spec->mixers[0] = ad1986a_laptop_master_mixers;
-		spec->mixers[1] = ad1986a_laptop_eapd_mixers;
-		spec->num_init_verbs = 3;
-		spec->init_verbs[1] = ad1986a_eapd_init_verbs;
-		spec->init_verbs[2] = ad1986a_automic_verbs;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
-		if (!is_jack_available(codec, 0x25))
-			spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1986a_automic_capture_source;
-		codec->patch_ops.unsol_event = ad1986a_automic_unsol_event;
-		codec->patch_ops.init = ad1986a_automic_init;
-		break;
-	case AD1986A_SAMSUNG_P50:
-		spec->num_mixers = 2;
-		spec->mixers[0] = ad1986a_automute_master_mixers;
-		spec->mixers[1] = ad1986a_laptop_eapd_mixers;
-		spec->num_init_verbs = 4;
-		spec->init_verbs[1] = ad1986a_eapd_init_verbs;
-		spec->init_verbs[2] = ad1986a_automic_verbs;
-		spec->init_verbs[3] = ad1986a_hp_init_verbs;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
-		if (!is_jack_available(codec, 0x25))
-			spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1986a_automic_capture_source;
-		codec->patch_ops.unsol_event = ad1986a_samsung_p50_unsol_event;
-		codec->patch_ops.init = ad1986a_samsung_p50_init;
-		break;
-	case AD1986A_LAPTOP_AUTOMUTE:
-		spec->num_mixers = 3;
-		spec->mixers[0] = ad1986a_automute_master_mixers;
-		spec->mixers[1] = ad1986a_laptop_eapd_mixers;
-		spec->mixers[2] = ad1986a_laptop_intmic_mixers;
-		spec->num_init_verbs = 3;
-		spec->init_verbs[1] = ad1986a_eapd_init_verbs;
-		spec->init_verbs[2] = ad1986a_hp_init_verbs;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
-		if (!is_jack_available(codec, 0x25))
-			spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1986a_laptop_eapd_capture_source;
-		codec->patch_ops.unsol_event = ad1986a_hp_unsol_event;
-		codec->patch_ops.init = ad1986a_hp_init;
-		/* Lenovo N100 seems to report the reversed bit
-		 * for HP jack-sensing
-		 */
-		spec->inv_jack_detect = 1;
-		break;
-	case AD1986A_ULTRA:
-		spec->mixers[0] = ad1986a_laptop_eapd_mixers;
-		spec->num_init_verbs = 2;
-		spec->init_verbs[1] = ad1986a_ultra_init;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1986a_laptop_dac_nids;
-		spec->multiout.dig_out_nid = 0;
-		break;
-	}
-
-	/* AD1986A has a hardware problem that it can't share a stream
-	 * with multiple output pins.  The copy of front to surrounds
-	 * causes noisy or silent outputs at a certain timing, e.g.
-	 * changing the volume.
-	 * So, let's disable the shared stream.
-	 */
-	spec->multiout.no_share_stream = 1;
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1986a	ad1986a_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
 
 /*
  * AD1983 specific
  */
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-#define AD1983_SPDIF_OUT	0x02
-#define AD1983_DAC		0x03
-#define AD1983_ADC		0x04
-
-static const hda_nid_t ad1983_dac_nids[1] = { AD1983_DAC };
-static const hda_nid_t ad1983_adc_nids[1] = { AD1983_ADC };
-static const hda_nid_t ad1983_capsrc_nids[1] = { 0x15 };
-
-static const struct hda_input_mux ad1983_capture_source = {
-	.num_items = 4,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Line", 0x1 },
-		{ "Mix", 0x2 },
-		{ "Mix Mono", 0x3 },
-	},
-};
-
-/*
- * SPDIF playback route
- */
-static int ad1983_spdif_route_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
-{
-	static const char * const texts[] = { "PCM", "ADC" };
-
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 2;
-	if (uinfo->value.enumerated.item > 1)
-		uinfo->value.enumerated.item = 1;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
-}
-
-static int ad1983_spdif_route_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-
-	ucontrol->value.enumerated.item[0] = spec->spdif_route;
-	return 0;
-}
-
-static int ad1983_spdif_route_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-
-	if (ucontrol->value.enumerated.item[0] > 1)
-		return -EINVAL;
-	if (spec->spdif_route != ucontrol->value.enumerated.item[0]) {
-		spec->spdif_route = ucontrol->value.enumerated.item[0];
-		snd_hda_codec_write_cache(codec, spec->multiout.dig_out_nid, 0,
-					  AC_VERB_SET_CONNECT_SEL,
-					  spec->spdif_route);
-		return 1;
-	}
-	return 0;
-}
-
-static const struct snd_kcontrol_new ad1983_mixers[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x06, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x07, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x07, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-static const struct hda_verb ad1983_init_verbs[] = {
-	/* Front, HP, Mono; mute as default */
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* Beep, PCM, Mic, Line-In: mute */
-	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* Front, HP selectors; from Mix */
-	{0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{0x06, AC_VERB_SET_CONNECT_SEL, 0x01},
-	/* Mono selector; from Mix */
-	{0x0b, AC_VERB_SET_CONNECT_SEL, 0x03},
-	/* Mic selector; Mic */
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Line-in selector: Line-in */
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Mic boost: 0dB */
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
-	/* Record selector: mic */
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* SPDIF route: PCM */
-	{0x02, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Front Pin */
-	{0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* HP Pin */
-	{0x06, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
-	/* Mono Pin */
-	{0x07, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* Mic Pin */
-	{0x08, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	/* Line Pin */
-	{0x09, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	{ } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1983_loopbacks[] = {
-	{ 0x12, HDA_OUTPUT, 0 }, /* Mic */
-	{ 0x13, HDA_OUTPUT, 0 }, /* Line */
-	{ } /* end */
-};
-#endif
-
-/* models */
-enum {
-	AD1983_AUTO,
-	AD1983_BASIC,
-	AD1983_MODELS
-};
-
-static const char * const ad1983_models[AD1983_MODELS] = {
-	[AD1983_AUTO]		= "auto",
-	[AD1983_BASIC]		= "basic",
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
 /*
  * SPDIF mux control for AD1983 auto-parser
  */
@@ -1656,7 +435,7 @@ static int ad1983_add_spdif_mux_ctl(struct hda_codec *codec)
 	return 0;
 }
 
-static int ad1983_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1983(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
 	int err;
@@ -1681,432 +460,11 @@ static int ad1983_parse_auto_config(struct hda_codec *codec)
 	return err;
 }
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1983(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int board_config;
-	int err;
-
-	board_config = snd_hda_check_board_config(codec, AD1983_MODELS,
-						  ad1983_models, NULL);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1983_AUTO;
-	}
-
-	if (board_config == AD1983_AUTO)
-		return ad1983_parse_auto_config(codec);
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	err = snd_hda_attach_beep_device(codec, 0x10);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
-	spec->multiout.max_channels = 2;
-	spec->multiout.num_dacs = ARRAY_SIZE(ad1983_dac_nids);
-	spec->multiout.dac_nids = ad1983_dac_nids;
-	spec->multiout.dig_out_nid = AD1983_SPDIF_OUT;
-	spec->num_adc_nids = 1;
-	spec->adc_nids = ad1983_adc_nids;
-	spec->capsrc_nids = ad1983_capsrc_nids;
-	spec->input_mux = &ad1983_capture_source;
-	spec->num_mixers = 1;
-	spec->mixers[0] = ad1983_mixers;
-	spec->num_init_verbs = 1;
-	spec->init_verbs[0] = ad1983_init_verbs;
-	spec->spdif_route = 0;
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1983_loopbacks;
-#endif
-	spec->vmaster_nid = 0x05;
-
-	codec->patch_ops = ad198x_patch_ops;
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1983	ad1983_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 
 /*
  * AD1981 HD specific
  */
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-#define AD1981_SPDIF_OUT	0x02
-#define AD1981_DAC		0x03
-#define AD1981_ADC		0x04
-
-static const hda_nid_t ad1981_dac_nids[1] = { AD1981_DAC };
-static const hda_nid_t ad1981_adc_nids[1] = { AD1981_ADC };
-static const hda_nid_t ad1981_capsrc_nids[1] = { 0x15 };
-
-/* 0x0c, 0x09, 0x0e, 0x0f, 0x19, 0x05, 0x18, 0x17 */
-static const struct hda_input_mux ad1981_capture_source = {
-	.num_items = 7,
-	.items = {
-		{ "Front Mic", 0x0 },
-		{ "Line", 0x1 },
-		{ "Mix", 0x2 },
-		{ "Mix Mono", 0x3 },
-		{ "CD", 0x4 },
-		{ "Mic", 0x6 },
-		{ "Aux", 0x7 },
-	},
-};
-
-static const struct snd_kcontrol_new ad1981_mixers[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x06, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x06, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x07, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x07, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Aux Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Aux Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x18, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	/* identical with AD1983 */
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-static const struct hda_verb ad1981_init_verbs[] = {
-	/* Front, HP, Mono; mute as default */
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* Beep, PCM, Front Mic, Line, Rear Mic, Aux, CD-In: mute */
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{0x13, 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},
-	/* Front, HP selectors; from Mix */
-	{0x05, AC_VERB_SET_CONNECT_SEL, 0x01},
-	{0x06, AC_VERB_SET_CONNECT_SEL, 0x01},
-	/* Mono selector; from Mix */
-	{0x0b, AC_VERB_SET_CONNECT_SEL, 0x03},
-	/* Mic Mixer; select Front Mic */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
-	{0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* Mic boost: 0dB */
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	/* Record selector: Front mic */
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	/* SPDIF route: PCM */
-	{0x02, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Front Pin */
-	{0x05, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* HP Pin */
-	{0x06, AC_VERB_SET_PIN_WIDGET_CONTROL, 0xc0 },
-	/* Mono Pin */
-	{0x07, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x40 },
-	/* Front & Rear Mic Pins */
-	{0x08, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24 },
-	/* Line Pin */
-	{0x09, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20 },
-	/* Digital Beep */
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x00},
-	/* Line-Out as Input: disabled */
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-	{ } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1981_loopbacks[] = {
-	{ 0x12, HDA_OUTPUT, 0 }, /* Front Mic */
-	{ 0x13, HDA_OUTPUT, 0 }, /* Line */
-	{ 0x1b, HDA_OUTPUT, 0 }, /* Aux */
-	{ 0x1c, HDA_OUTPUT, 0 }, /* Mic */
-	{ 0x1d, HDA_OUTPUT, 0 }, /* CD */
-	{ } /* end */
-};
-#endif
-
-/*
- * Patch for HP nx6320
- *
- * nx6320 uses EAPD in the reverse way - EAPD-on means the internal
- * speaker output enabled _and_ mute-LED off.
- */
-
-#define AD1981_HP_EVENT		0x37
-#define AD1981_MIC_EVENT	0x38
-
-static const struct hda_verb ad1981_hp_init_verbs[] = {
-	{0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x00 }, /* default off */
-	/* pin sensing on HP and Mic jacks */
-	{0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT},
-	{0x08, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_MIC_EVENT},
-	{}
-};
-
-/* turn on/off EAPD (+ mute HP) as a master switch */
-static int ad1981_hp_master_sw_put(struct snd_kcontrol *kcontrol,
-				   struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	struct ad198x_spec *spec = codec->spec;
-
-	if (! ad198x_eapd_put(kcontrol, ucontrol))
-		return 0;
-	/* change speaker pin appropriately */
-	snd_hda_set_pin_ctl(codec, 0x05, spec->cur_eapd ? PIN_OUT : 0);
-	/* toggle HP mute appropriately */
-	snd_hda_codec_amp_stereo(codec, 0x06, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE,
-				 spec->cur_eapd ? 0 : HDA_AMP_MUTE);
-	return 1;
-}
-
-/* bind volumes of both NID 0x05 and 0x06 */
-static const struct hda_bind_ctls ad1981_hp_bind_master_vol = {
-	.ops = &snd_hda_bind_vol,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x05, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x06, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
-/* mute internal speaker if HP is plugged */
-static void ad1981_hp_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x06);
-	snd_hda_codec_amp_stereo(codec, 0x05, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-/* toggle input of built-in and mic jack appropriately */
-static void ad1981_hp_automic(struct hda_codec *codec)
-{
-	static const struct hda_verb mic_jack_on[] = {
-		{0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-		{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
-		{}
-	};
-	static const struct hda_verb mic_jack_off[] = {
-		{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080},
-		{0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
-		{}
-	};
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x08);
-	if (present)
-		snd_hda_sequence_write(codec, mic_jack_on);
-	else
-		snd_hda_sequence_write(codec, mic_jack_off);
-}
-
-/* unsolicited event for HP jack sensing */
-static void ad1981_hp_unsol_event(struct hda_codec *codec,
-				  unsigned int res)
-{
-	res >>= 26;
-	switch (res) {
-	case AD1981_HP_EVENT:
-		ad1981_hp_automute(codec);
-		break;
-	case AD1981_MIC_EVENT:
-		ad1981_hp_automic(codec);
-		break;
-	}
-}
-
-static const struct hda_input_mux ad1981_hp_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Dock Mic", 0x1 },
-		{ "Mix", 0x2 },
-	},
-};
-
-static const struct snd_kcontrol_new ad1981_hp_mixers[] = {
-	HDA_BIND_VOL("Master Playback Volume", &ad1981_hp_bind_master_vol),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.subdevice = HDA_SUBDEV_NID_FLAG | 0x05,
-		.name = "Master Playback Switch",
-		.info = ad198x_eapd_info,
-		.get = ad198x_eapd_get,
-		.put = ad1981_hp_master_sw_put,
-		.private_value = 0x05,
-	},
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-#if 0
-	/* FIXME: analog mic/line loopback doesn't work with my tests...
-	 *        (although recording is OK)
-	 */
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x1c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT),
-	/* FIXME: does this laptop have analog CD connection? */
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
-#endif
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x18, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	{ } /* end */
-};
-
-/* initialize jack-sensing, too */
-static int ad1981_hp_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1981_hp_automute(codec);
-	ad1981_hp_automic(codec);
-	return 0;
-}
-
-/* configuration for Toshiba Laptops */
-static const struct hda_verb ad1981_toshiba_init_verbs[] = {
-	{0x05, AC_VERB_SET_EAPD_BTLENABLE, 0x01 }, /* default on */
-	/* pin sensing on HP and Mic jacks */
-	{0x06, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_HP_EVENT},
-	{0x08, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1981_MIC_EVENT},
-	{}
-};
-
-static const struct snd_kcontrol_new ad1981_toshiba_mixers[] = {
-	HDA_CODEC_VOLUME("Amp Volume", 0x1a, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Amp Switch", 0x1a, 0x0, HDA_OUTPUT),
-	{ }
-};
-
-/* configuration for Lenovo Thinkpad T60 */
-static const struct snd_kcontrol_new ad1981_thinkpad_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Master Playback Switch", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x08, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	/* identical with AD1983 */
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-static const struct hda_input_mux ad1981_thinkpad_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Mix", 0x2 },
-		{ "CD", 0x4 },
-	},
-};
-
-/* models */
-enum {
-	AD1981_AUTO,
-	AD1981_BASIC,
-	AD1981_HP,
-	AD1981_THINKPAD,
-	AD1981_TOSHIBA,
-	AD1981_MODELS
-};
-
-static const char * const ad1981_models[AD1981_MODELS] = {
-	[AD1981_AUTO]		= "auto",
-	[AD1981_HP]		= "hp",
-	[AD1981_THINKPAD]	= "thinkpad",
-	[AD1981_BASIC]		= "basic",
-	[AD1981_TOSHIBA]	= "toshiba"
-};
-
-static const struct snd_pci_quirk ad1981_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD),
-	SND_PCI_QUIRK(0x1014, 0x05b7, "Lenovo Z60m", AD1981_THINKPAD),
-	/* All HP models */
-	SND_PCI_QUIRK_VENDOR(0x103c, "HP nx", AD1981_HP),
-	SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba U205", AD1981_TOSHIBA),
-	/* Lenovo Thinkpad T60/X60/Z6xx */
-	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1981_THINKPAD),
-	/* HP nx6320 (reversed SSID, H/W bug) */
-	SND_PCI_QUIRK(0x30b0, 0x103c, "HP nx6320", AD1981_HP),
-	{}
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
 /* follow EAPD via vmaster hook */
 static void ad_vmaster_eapd_hook(void *private_data, int enabled)
 {
@@ -2172,7 +530,7 @@ static const struct snd_pci_quirk ad1981_fixup_tbl[] = {
 	{}
 };
 
-static int ad1981_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1981(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
 	int err;
@@ -2205,110 +563,6 @@ static int ad1981_parse_auto_config(struct hda_codec *codec)
 	return err;
 }
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1981(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int err, board_config;
-
-	board_config = snd_hda_check_board_config(codec, AD1981_MODELS,
-						  ad1981_models,
-						  ad1981_cfg_tbl);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1981_AUTO;
-	}
-
-	if (board_config == AD1981_AUTO)
-		return ad1981_parse_auto_config(codec);
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return -ENOMEM;
-	spec = codec->spec;
-
-	err = snd_hda_attach_beep_device(codec, 0x10);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x0d, 0, HDA_OUTPUT);
-
-	spec->multiout.max_channels = 2;
-	spec->multiout.num_dacs = ARRAY_SIZE(ad1981_dac_nids);
-	spec->multiout.dac_nids = ad1981_dac_nids;
-	spec->multiout.dig_out_nid = AD1981_SPDIF_OUT;
-	spec->num_adc_nids = 1;
-	spec->adc_nids = ad1981_adc_nids;
-	spec->capsrc_nids = ad1981_capsrc_nids;
-	spec->input_mux = &ad1981_capture_source;
-	spec->num_mixers = 1;
-	spec->mixers[0] = ad1981_mixers;
-	spec->num_init_verbs = 1;
-	spec->init_verbs[0] = ad1981_init_verbs;
-	spec->spdif_route = 0;
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1981_loopbacks;
-#endif
-	spec->vmaster_nid = 0x05;
-
-	codec->patch_ops = ad198x_patch_ops;
-
-	/* override some parameters */
-	switch (board_config) {
-	case AD1981_HP:
-		spec->mixers[0] = ad1981_hp_mixers;
-		spec->num_init_verbs = 2;
-		spec->init_verbs[1] = ad1981_hp_init_verbs;
-		if (!is_jack_available(codec, 0x0a))
-			spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1981_hp_capture_source;
-
-		codec->patch_ops.init = ad1981_hp_init;
-		codec->patch_ops.unsol_event = ad1981_hp_unsol_event;
-		/* set the upper-limit for mixer amp to 0dB for avoiding the
-		 * possible damage by overloading
-		 */
-		snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT,
-					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
-					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
-					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
-					  (1 << AC_AMPCAP_MUTE_SHIFT));
-		break;
-	case AD1981_THINKPAD:
-		spec->mixers[0] = ad1981_thinkpad_mixers;
-		spec->input_mux = &ad1981_thinkpad_capture_source;
-		/* set the upper-limit for mixer amp to 0dB for avoiding the
-		 * possible damage by overloading
-		 */
-		snd_hda_override_amp_caps(codec, 0x11, HDA_INPUT,
-					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
-					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
-					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
-					  (1 << AC_AMPCAP_MUTE_SHIFT));
-		break;
-	case AD1981_TOSHIBA:
-		spec->mixers[0] = ad1981_hp_mixers;
-		spec->mixers[1] = ad1981_toshiba_mixers;
-		spec->num_init_verbs = 2;
-		spec->init_verbs[1] = ad1981_toshiba_init_verbs;
-		spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1981_hp_capture_source;
-		codec->patch_ops.init = ad1981_hp_init;
-		codec->patch_ops.unsol_event = ad1981_hp_unsol_event;
-		break;
-	}
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1981	ad1981_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 
 /*
  * AD1988
@@ -2395,90 +649,7 @@ static int patch_ad1981(struct hda_codec *codec)
  *      E/F quad mic array
  */
 
-
 #ifdef ENABLE_AD_STATIC_QUIRKS
-/* models */
-enum {
-	AD1988_AUTO,
-	AD1988_6STACK,
-	AD1988_6STACK_DIG,
-	AD1988_3STACK,
-	AD1988_3STACK_DIG,
-	AD1988_LAPTOP,
-	AD1988_LAPTOP_DIG,
-	AD1988_MODEL_LAST,
-};
-
-/* reivision id to check workarounds */
-#define AD1988A_REV2		0x100200
-
-#define is_rev2(codec) \
-	((codec)->vendor_id == 0x11d41988 && \
-	 (codec)->revision_id == AD1988A_REV2)
-
-/*
- * mixers
- */
-
-static const hda_nid_t ad1988_6stack_dac_nids[4] = {
-	0x04, 0x06, 0x05, 0x0a
-};
-
-static const hda_nid_t ad1988_3stack_dac_nids[3] = {
-	0x04, 0x05, 0x0a
-};
-
-/* for AD1988A revision-2, DAC2-4 are swapped */
-static const hda_nid_t ad1988_6stack_dac_nids_rev2[4] = {
-	0x04, 0x05, 0x0a, 0x06
-};
-
-static const hda_nid_t ad1988_alt_dac_nid[1] = {
-	0x03
-};
-
-static const hda_nid_t ad1988_3stack_dac_nids_rev2[3] = {
-	0x04, 0x0a, 0x06
-};
-
-static const hda_nid_t ad1988_adc_nids[3] = {
-	0x08, 0x09, 0x0f
-};
-
-static const hda_nid_t ad1988_capsrc_nids[3] = {
-	0x0c, 0x0d, 0x0e
-};
-
-#define AD1988_SPDIF_OUT		0x02
-#define AD1988_SPDIF_OUT_HDMI	0x0b
-#define AD1988_SPDIF_IN		0x07
-
-static const hda_nid_t ad1989b_slave_dig_outs[] = {
-	AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI, 0
-};
-
-static const struct hda_input_mux ad1988_6stack_capture_source = {
-	.num_items = 5,
-	.items = {
-		{ "Front Mic", 0x1 },	/* port-B */
-		{ "Line", 0x2 },	/* port-C */
-		{ "Mic", 0x4 },		/* port-E */
-		{ "CD", 0x5 },
-		{ "Mix", 0x9 },
-	},
-};
-
-static const struct hda_input_mux ad1988_laptop_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic/Line", 0x1 },	/* port-B */
-		{ "CD", 0x5 },
-		{ "Mix", 0x9 },
-	},
-};
-
-/*
- */
 static int ad198x_ch_mode_info(struct snd_kcontrol *kcontrol,
 			       struct snd_ctl_elem_info *uinfo)
 {
@@ -2509,569 +680,6 @@ static int ad198x_ch_mode_put(struct snd_kcontrol *kcontrol,
 		spec->multiout.num_dacs = spec->multiout.max_channels / 2;
 	return err;
 }
-
-/* 6-stack mode */
-static const struct snd_kcontrol_new ad1988_6stack_mixers1[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x06, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Side Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_6stack_mixers1_rev2[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x05, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0a, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0a, 2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Side Playback Volume", 0x06, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_6stack_mixers2[] = {
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Surround Playback Switch", 0x2a, 2, HDA_INPUT),
-	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x27, 1, 2, HDA_INPUT),
-	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x27, 2, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Side Playback Switch", 0x28, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-/* 3-stack mode */
-static const struct snd_kcontrol_new ad1988_3stack_mixers1[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_3stack_mixers1_rev2[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x0a, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x06, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x06, 2, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_3stack_mixers2[] = {
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_BIND_MUTE("Front Playback Switch", 0x29, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Surround Playback Switch", 0x2c, 2, HDA_INPUT),
-	HDA_BIND_MUTE_MONO("Center Playback Switch", 0x26, 1, 2, HDA_INPUT),
-	HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x26, 2, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Headphone Playback Switch", 0x22, 2, HDA_INPUT),
-	HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Channel Mode",
-		.info = ad198x_ch_mode_info,
-		.get = ad198x_ch_mode_get,
-		.put = ad198x_ch_mode_put,
-	},
-
-	{ } /* end */
-};
-
-/* laptop mode */
-static const struct snd_kcontrol_new ad1988_laptop_mixers[] = {
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x29, 0x0, HDA_INPUT),
-	HDA_BIND_MUTE("Mono Playback Switch", 0x1e, 2, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x6, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x6, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
-
-	HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
-
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "External Amplifier",
-		.subdevice = HDA_SUBDEV_NID_FLAG | 0x12,
-		.info = ad198x_eapd_info,
-		.get = ad198x_eapd_get,
-		.put = ad198x_eapd_put,
-		.private_value = 0x12, /* port-D */
-	},
-
-	{ } /* end */
-};
-
-/* capture */
-static const struct snd_kcontrol_new ad1988_capture_mixers[] = {
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 2, 0x0e, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 2, 0x0e, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 3,
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	{ } /* end */
-};
-
-static int ad1988_spdif_playback_source_info(struct snd_kcontrol *kcontrol,
-					     struct snd_ctl_elem_info *uinfo)
-{
-	static const char * const texts[] = {
-		"PCM", "ADC1", "ADC2", "ADC3"
-	};
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = 4;
-	if (uinfo->value.enumerated.item >= 4)
-		uinfo->value.enumerated.item = 3;
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
-}
-
-static int ad1988_spdif_playback_source_get(struct snd_kcontrol *kcontrol,
-					    struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	unsigned int sel;
-
-	sel = snd_hda_codec_read(codec, 0x1d, 0, AC_VERB_GET_AMP_GAIN_MUTE,
-				 AC_AMP_GET_INPUT);
-	if (!(sel & 0x80))
-		ucontrol->value.enumerated.item[0] = 0;
-	else {
-		sel = snd_hda_codec_read(codec, 0x0b, 0,
-					 AC_VERB_GET_CONNECT_SEL, 0);
-		if (sel < 3)
-			sel++;
-		else
-			sel = 0;
-		ucontrol->value.enumerated.item[0] = sel;
-	}
-	return 0;
-}
-
-static int ad1988_spdif_playback_source_put(struct snd_kcontrol *kcontrol,
-					    struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	unsigned int val, sel;
-	int change;
-
-	val = ucontrol->value.enumerated.item[0];
-	if (val > 3)
-		return -EINVAL;
-	if (!val) {
-		sel = snd_hda_codec_read(codec, 0x1d, 0,
-					 AC_VERB_GET_AMP_GAIN_MUTE,
-					 AC_AMP_GET_INPUT);
-		change = sel & 0x80;
-		if (change) {
-			snd_hda_codec_write_cache(codec, 0x1d, 0,
-						  AC_VERB_SET_AMP_GAIN_MUTE,
-						  AMP_IN_UNMUTE(0));
-			snd_hda_codec_write_cache(codec, 0x1d, 0,
-						  AC_VERB_SET_AMP_GAIN_MUTE,
-						  AMP_IN_MUTE(1));
-		}
-	} else {
-		sel = snd_hda_codec_read(codec, 0x1d, 0,
-					 AC_VERB_GET_AMP_GAIN_MUTE,
-					 AC_AMP_GET_INPUT | 0x01);
-		change = sel & 0x80;
-		if (change) {
-			snd_hda_codec_write_cache(codec, 0x1d, 0,
-						  AC_VERB_SET_AMP_GAIN_MUTE,
-						  AMP_IN_MUTE(0));
-			snd_hda_codec_write_cache(codec, 0x1d, 0,
-						  AC_VERB_SET_AMP_GAIN_MUTE,
-						  AMP_IN_UNMUTE(1));
-		}
-		sel = snd_hda_codec_read(codec, 0x0b, 0,
-					 AC_VERB_GET_CONNECT_SEL, 0) + 1;
-		change |= sel != val;
-		if (change)
-			snd_hda_codec_write_cache(codec, 0x0b, 0,
-						  AC_VERB_SET_CONNECT_SEL,
-						  val - 1);
-	}
-	return change;
-}
-
-static const struct snd_kcontrol_new ad1988_spdif_out_mixers[] = {
-	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "IEC958 Playback Source",
-		.subdevice = HDA_SUBDEV_NID_FLAG | 0x1b,
-		.info = ad1988_spdif_playback_source_info,
-		.get = ad1988_spdif_playback_source_get,
-		.put = ad1988_spdif_playback_source_put,
-	},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1988_spdif_in_mixers[] = {
-	HDA_CODEC_VOLUME("IEC958 Capture Volume", 0x1c, 0x0, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1989_spdif_out_mixers[] = {
-	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("HDMI Playback Volume", 0x1d, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-/*
- * initialization verbs
- */
-
-/*
- * for 6-stack (+dig)
- */
-static const struct hda_verb ad1988_6stack_init_verbs[] = {
-	/* Front, Surround, CLFE, side DAC; unmute as default */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-A front headphon path */
-	{0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	/* Port-D line-out path */
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* Port-F surround path */
-	{0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x2a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* Port-G CLFE path */
-	{0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x27, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* Port-H side path */
-	{0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x28, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x25, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* Mono out path */
-	{0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
-	/* Port-B front mic-in path */
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	/* Port-C line-in path */
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Port-E mic-in path */
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x34, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Analog CD Input */
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
-
-	{ }
-};
-
-static const struct hda_verb ad1988_6stack_fp_init_verbs[] = {
-	/* Headphone; unmute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-A front headphon path */
-	{0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-
-	{ }
-};
-
-static const struct hda_verb ad1988_capture_init_verbs[] = {
-	/* mute analog mix */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
-	/* select ADCs - front-mic */
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-
-	{ }
-};
-
-static const struct hda_verb ad1988_spdif_init_verbs[] = {
-	/* SPDIF out sel */
-	{0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
-	{0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1 */
-	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	/* SPDIF out pin */
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-
-	{ }
-};
-
-static const struct hda_verb ad1988_spdif_in_init_verbs[] = {
-	/* unmute SPDIF input pin */
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{ }
-};
-
-/* AD1989 has no ADC -> SPDIF route */
-static const struct hda_verb ad1989_spdif_init_verbs[] = {
-	/* SPDIF-1 out pin */
-	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-	/* SPDIF-2/HDMI out pin */
-	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-	{ }
-};
-
-/*
- * verbs for 3stack (+dig)
- */
-static const struct hda_verb ad1988_3stack_ch2_init[] = {
-	/* set port-C to line-in */
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
-	/* set port-E to mic-in */
-	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
-	{ } /* end */
-};
-
-static const struct hda_verb ad1988_3stack_ch6_init[] = {
-	/* set port-C to surround out */
-	{ 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	/* set port-E to CLFE out */
-	{ 0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
-	{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ } /* end */
-};
-
-static const struct hda_channel_mode ad1988_3stack_modes[2] = {
-	{ 2, ad1988_3stack_ch2_init },
-	{ 6, ad1988_3stack_ch6_init },
-};
-
-static const struct hda_verb ad1988_3stack_init_verbs[] = {
-	/* Front, Surround, CLFE, side DAC; unmute as default */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-A front headphon path */
-	{0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	/* Port-D line-out path */
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* Mono out path */
-	{0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
-	/* Port-B front mic-in path */
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	/* Port-C line-in/surround path - 6ch mode as default */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x31, AC_VERB_SET_CONNECT_SEL, 0x0}, /* output sel: DAC 0x05 */
-	{0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Port-E mic-in/CLFE path - 6ch mode as default */
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x32, AC_VERB_SET_CONNECT_SEL, 0x1}, /* output sel: DAC 0x0a */
-	{0x34, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* mute analog mix */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
-	/* select ADCs - front-mic */
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
-	{ }
-};
-
-/*
- * verbs for laptop mode (+dig)
- */
-static const struct hda_verb ad1988_laptop_hp_on[] = {
-	/* unmute port-A and mute port-D */
-	{ 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ } /* end */
-};
-static const struct hda_verb ad1988_laptop_hp_off[] = {
-	/* mute port-A and unmute port-D */
-	{ 0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ 0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ } /* end */
-};
-
-#define AD1988_HP_EVENT	0x01
-
-static const struct hda_verb ad1988_laptop_init_verbs[] = {
-	/* Front, Surround, CLFE, side DAC; unmute as default */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x06, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-A front headphon path */
-	{0x37, AC_VERB_SET_CONNECT_SEL, 0x00}, /* DAC0:03h */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	/* unsolicited event for pin-sense */
-	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1988_HP_EVENT },
-	/* Port-D line-out path + EAPD */
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x00}, /* EAPD-off */
-	/* Mono out path */
-	{0x36, AC_VERB_SET_CONNECT_SEL, 0x1}, /* DAC1:04h */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, 0xb01f}, /* unmute, 0dB */
-	/* Port-B mic-in path */
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	/* Port-C docking station - try to output */
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x33, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* mute analog mix */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
-	/* select ADCs - mic */
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
-	{ }
-};
-
-static void ad1988_laptop_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	if ((res >> 26) != AD1988_HP_EVENT)
-		return;
-	if (snd_hda_jack_detect(codec, 0x11))
-		snd_hda_sequence_write(codec, ad1988_laptop_hp_on);
-	else
-		snd_hda_sequence_write(codec, ad1988_laptop_hp_off);
-} 
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1988_loopbacks[] = {
-	{ 0x20, HDA_INPUT, 0 }, /* Front Mic */
-	{ 0x20, HDA_INPUT, 1 }, /* Line */
-	{ 0x20, HDA_INPUT, 4 }, /* Mic */
-	{ 0x20, HDA_INPUT, 6 }, /* CD */
-	{ } /* end */
-};
-#endif
 #endif /* ENABLE_AD_STATIC_QUIRKS */
 
 static int ad1988_auto_smux_enum_info(struct snd_kcontrol *kcontrol,
@@ -3220,7 +828,34 @@ static int ad1988_add_spdif_mux_ctl(struct hda_codec *codec)
 /*
  */
 
-static int ad1988_parse_auto_config(struct hda_codec *codec)
+enum {
+	AD1988_FIXUP_6STACK_DIG,
+};
+
+static const struct hda_fixup ad1988_fixups[] = {
+	[AD1988_FIXUP_6STACK_DIG] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x11, 0x02214130 }, /* front-hp */
+			{ 0x12, 0x01014010 }, /* line-out */
+			{ 0x14, 0x02a19122 }, /* front-mic */
+			{ 0x15, 0x01813021 }, /* line-in */
+			{ 0x16, 0x01011012 }, /* line-out */
+			{ 0x17, 0x01a19020 }, /* mic */
+			{ 0x1b, 0x0145f1f0 }, /* SPDIF */
+			{ 0x24, 0x01016011 }, /* line-out */
+			{ 0x25, 0x01012013 }, /* line-out */
+			{ }
+		}
+	},
+};
+
+static const struct hda_model_fixup ad1988_fixup_models[] = {
+	{ .id = AD1988_FIXUP_6STACK_DIG, .name = "6stack-dig" },
+	{}
+};
+
+static int patch_ad1988(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
 	int err;
@@ -3234,12 +869,19 @@ static int ad1988_parse_auto_config(struct hda_codec *codec)
 	spec->gen.mixer_merge_nid = 0x21;
 	spec->gen.beep_nid = 0x10;
 	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
+
+	snd_hda_pick_fixup(codec, ad1988_fixup_models, NULL, ad1988_fixups);
+	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
+
 	err = ad198x_parse_auto_config(codec);
 	if (err < 0)
 		goto error;
 	err = ad1988_add_spdif_mux_ctl(codec);
 	if (err < 0)
 		goto error;
+
+	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
+
 	return 0;
 
  error:
@@ -3247,169 +889,6 @@ static int ad1988_parse_auto_config(struct hda_codec *codec)
 	return err;
 }
 
-/*
- */
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const char * const ad1988_models[AD1988_MODEL_LAST] = {
-	[AD1988_6STACK]		= "6stack",
-	[AD1988_6STACK_DIG]	= "6stack-dig",
-	[AD1988_3STACK]		= "3stack",
-	[AD1988_3STACK_DIG]	= "3stack-dig",
-	[AD1988_LAPTOP]		= "laptop",
-	[AD1988_LAPTOP_DIG]	= "laptop-dig",
-	[AD1988_AUTO]		= "auto",
-};
-
-static const struct snd_pci_quirk ad1988_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1043, 0x81ec, "Asus P5B-DLX", AD1988_6STACK_DIG),
-	SND_PCI_QUIRK(0x1043, 0x81f6, "Asus M2N-SLI", AD1988_6STACK_DIG),
-	SND_PCI_QUIRK(0x1043, 0x8277, "Asus P5K-E/WIFI-AP", AD1988_6STACK_DIG),
-	SND_PCI_QUIRK(0x1043, 0x82c0, "Asus M3N-HT Deluxe", AD1988_6STACK_DIG),
-	SND_PCI_QUIRK(0x1043, 0x8311, "Asus P5Q-Premium/Pro", AD1988_6STACK_DIG),
-	{}
-};
-
-static int patch_ad1988(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int err, board_config;
-
-	board_config = snd_hda_check_board_config(codec, AD1988_MODEL_LAST,
-						  ad1988_models, ad1988_cfg_tbl);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1988_AUTO;
-	}
-
-	if (board_config == AD1988_AUTO)
-		return ad1988_parse_auto_config(codec);
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	if (is_rev2(codec))
-		snd_printk(KERN_INFO "patch_analog: AD1988A rev.2 is detected, enable workarounds\n");
-
-	err = snd_hda_attach_beep_device(codec, 0x10);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
-	if (!spec->multiout.hp_nid)
-		spec->multiout.hp_nid = ad1988_alt_dac_nid[0];
-	switch (board_config) {
-	case AD1988_6STACK:
-	case AD1988_6STACK_DIG:
-		spec->multiout.max_channels = 8;
-		spec->multiout.num_dacs = 4;
-		if (is_rev2(codec))
-			spec->multiout.dac_nids = ad1988_6stack_dac_nids_rev2;
-		else
-			spec->multiout.dac_nids = ad1988_6stack_dac_nids;
-		spec->input_mux = &ad1988_6stack_capture_source;
-		spec->num_mixers = 2;
-		if (is_rev2(codec))
-			spec->mixers[0] = ad1988_6stack_mixers1_rev2;
-		else
-			spec->mixers[0] = ad1988_6stack_mixers1;
-		spec->mixers[1] = ad1988_6stack_mixers2;
-		spec->num_init_verbs = 1;
-		spec->init_verbs[0] = ad1988_6stack_init_verbs;
-		if (board_config == AD1988_6STACK_DIG) {
-			spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
-			spec->dig_in_nid = AD1988_SPDIF_IN;
-		}
-		break;
-	case AD1988_3STACK:
-	case AD1988_3STACK_DIG:
-		spec->multiout.max_channels = 6;
-		spec->multiout.num_dacs = 3;
-		if (is_rev2(codec))
-			spec->multiout.dac_nids = ad1988_3stack_dac_nids_rev2;
-		else
-			spec->multiout.dac_nids = ad1988_3stack_dac_nids;
-		spec->input_mux = &ad1988_6stack_capture_source;
-		spec->channel_mode = ad1988_3stack_modes;
-		spec->num_channel_mode = ARRAY_SIZE(ad1988_3stack_modes);
-		spec->num_mixers = 2;
-		if (is_rev2(codec))
-			spec->mixers[0] = ad1988_3stack_mixers1_rev2;
-		else
-			spec->mixers[0] = ad1988_3stack_mixers1;
-		spec->mixers[1] = ad1988_3stack_mixers2;
-		spec->num_init_verbs = 1;
-		spec->init_verbs[0] = ad1988_3stack_init_verbs;
-		if (board_config == AD1988_3STACK_DIG)
-			spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
-		break;
-	case AD1988_LAPTOP:
-	case AD1988_LAPTOP_DIG:
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		spec->multiout.dac_nids = ad1988_3stack_dac_nids;
-		spec->input_mux = &ad1988_laptop_capture_source;
-		spec->num_mixers = 1;
-		spec->mixers[0] = ad1988_laptop_mixers;
-		codec->inv_eapd = 1; /* inverted EAPD */
-		spec->num_init_verbs = 1;
-		spec->init_verbs[0] = ad1988_laptop_init_verbs;
-		if (board_config == AD1988_LAPTOP_DIG)
-			spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
-		break;
-	}
-
-	spec->num_adc_nids = ARRAY_SIZE(ad1988_adc_nids);
-	spec->adc_nids = ad1988_adc_nids;
-	spec->capsrc_nids = ad1988_capsrc_nids;
-	spec->mixers[spec->num_mixers++] = ad1988_capture_mixers;
-	spec->init_verbs[spec->num_init_verbs++] = ad1988_capture_init_verbs;
-	if (spec->multiout.dig_out_nid) {
-		if (codec->vendor_id >= 0x11d4989a) {
-			spec->mixers[spec->num_mixers++] =
-				ad1989_spdif_out_mixers;
-			spec->init_verbs[spec->num_init_verbs++] =
-				ad1989_spdif_init_verbs;
-			codec->slave_dig_outs = ad1989b_slave_dig_outs;
-		} else {
-			spec->mixers[spec->num_mixers++] =
-				ad1988_spdif_out_mixers;
-			spec->init_verbs[spec->num_init_verbs++] =
-				ad1988_spdif_init_verbs;
-		}
-	}
-	if (spec->dig_in_nid && codec->vendor_id < 0x11d4989a) {
-		spec->mixers[spec->num_mixers++] = ad1988_spdif_in_mixers;
-		spec->init_verbs[spec->num_init_verbs++] =
-			ad1988_spdif_in_init_verbs;
-	}
-
-	codec->patch_ops = ad198x_patch_ops;
-	switch (board_config) {
-	case AD1988_LAPTOP:
-	case AD1988_LAPTOP_DIG:
-		codec->patch_ops.unsol_event = ad1988_laptop_unsol_event;
-		break;
-	}
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1988_loopbacks;
-#endif
-	spec->vmaster_nid = 0x04;
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1988	ad1988_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 
 /*
  * AD1884 / AD1984
@@ -3423,167 +902,19 @@ static int patch_ad1988(struct hda_codec *codec)
  *
  * AD1984 = AD1884 + two digital mic-ins
  *
- * FIXME:
- * For simplicity, we share the single DAC for both HP and line-outs
- * right now.  The inidividual playbacks could be easily implemented,
- * but no build-up framework is given, so far.
- */
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const hda_nid_t ad1884_dac_nids[1] = {
-	0x04,
-};
-
-static const hda_nid_t ad1884_adc_nids[2] = {
-	0x08, 0x09,
-};
-
-static const hda_nid_t ad1884_capsrc_nids[2] = {
-	0x0c, 0x0d,
-};
-
-#define AD1884_SPDIF_OUT	0x02
-
-static const struct hda_input_mux ad1884_capture_source = {
-	.num_items = 4,
-	.items = {
-		{ "Front Mic", 0x0 },
-		{ "Mic", 0x1 },
-		{ "CD", 0x2 },
-		{ "Mix", 0x3 },
-	},
-};
-
-static const struct snd_kcontrol_new ad1884_base_mixers[] = {
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	/* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	/* SPDIF controls */
-	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		/* identical with ad1983 */
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984_dmic_mixers[] = {
-	HDA_CODEC_VOLUME("Digital Mic Capture Volume", 0x05, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Digital Mic Capture Switch", 0x05, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME_IDX("Digital Mic Capture Volume", 1, 0x06, 0x0,
-			     HDA_INPUT),
-	HDA_CODEC_MUTE_IDX("Digital Mic Capture Switch", 1, 0x06, 0x0,
-			   HDA_INPUT),
-	{ } /* end */
-};
-
-/*
- * initialization verbs
+ * AD1883 / AD1884A / AD1984A / AD1984B
+ *
+ * port-B (0x14) - front mic-in
+ * port-E (0x1c) - rear mic-in
+ * port-F (0x16) - CD / ext out
+ * port-C (0x15) - rear line-in
+ * port-D (0x12) - rear line-out
+ * port-A (0x11) - front hp-out
+ *
+ * AD1984A = AD1884A + digital-mic
+ * AD1883 = equivalent with AD1984A
+ * AD1984B = AD1984A + extra SPDIF-out
  */
-static const struct hda_verb ad1884_init_verbs[] = {
-	/* DACs; mute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	/* Port-A (HP) mixer */
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-A pin */
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* HP selector - select DAC2 */
-	{0x22, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* Port-D (Line-out) mixer */
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-D pin */
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Mono-out mixer */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Mono-out pin */
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Mono selector */
-	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* Port-B (front mic) pin */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	/* Port-C (rear mic) pin */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	/* Analog mixer; mute as default */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
-	/* SPDIF output selector */
-	{0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-	{ } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1884_loopbacks[] = {
-	{ 0x20, HDA_INPUT, 0 }, /* Front Mic */
-	{ 0x20, HDA_INPUT, 1 }, /* Mic */
-	{ 0x20, HDA_INPUT, 2 }, /* CD */
-	{ 0x20, HDA_INPUT, 4 }, /* Docking */
-	{ } /* end */
-};
-#endif
-
-static const char * const ad1884_slave_vols[] = {
-	"PCM", "Mic", "Mono", "Front Mic", "Mic", "CD",
-	"Internal Mic", "Dock Mic", /* "Beep", */ "IEC958",
-	NULL
-};
-
-enum {
-	AD1884_AUTO,
-	AD1884_BASIC,
-	AD1884_MODELS
-};
-
-static const char * const ad1884_models[AD1884_MODELS] = {
-	[AD1884_AUTO]		= "auto",
-	[AD1884_BASIC]		= "basic",
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 
 /* set the upper-limit for mixer amp to 0dB for avoiding the possible
  * damage by overloading
@@ -3599,14 +930,34 @@ static void ad1884_fixup_amp_override(struct hda_codec *codec,
 					  (1 << AC_AMPCAP_MUTE_SHIFT));
 }
 
+/* toggle GPIO1 according to the mute state */
+static void ad1884_vmaster_hp_gpio_hook(void *private_data, int enabled)
+{
+	struct hda_codec *codec = private_data;
+	struct ad198x_spec *spec = codec->spec;
+
+	if (spec->eapd_nid)
+		ad_vmaster_eapd_hook(private_data, enabled);
+	snd_hda_codec_update_cache(codec, 0x01, 0,
+				   AC_VERB_SET_GPIO_DATA,
+				   enabled ? 0x00 : 0x02);
+}
+
 static void ad1884_fixup_hp_eapd(struct hda_codec *codec,
 				 const struct hda_fixup *fix, int action)
 {
 	struct ad198x_spec *spec = codec->spec;
+	static const struct hda_verb gpio_init_verbs[] = {
+		{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
+		{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
+		{0x01, AC_VERB_SET_GPIO_DATA, 0x02},
+		{},
+	};
 
 	switch (action) {
 	case HDA_FIXUP_ACT_PRE_PROBE:
-		spec->gen.vmaster_mute.hook = ad_vmaster_eapd_hook;
+		spec->gen.vmaster_mute.hook = ad1884_vmaster_hp_gpio_hook;
+		snd_hda_sequence_write_cache(codec, gpio_init_verbs);
 		break;
 	case HDA_FIXUP_ACT_PROBE:
 		if (spec->gen.autocfg.line_out_type == AUTO_PIN_SPEAKER_OUT)
@@ -3617,9 +968,18 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec,
 	}
 }
 
+/* set magic COEFs for dmic */
+static const struct hda_verb ad1884_dmic_init_verbs[] = {
+	{0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
+	{0x01, AC_VERB_SET_PROC_COEF, 0x08},
+	{}
+};
+
 enum {
 	AD1884_FIXUP_AMP_OVERRIDE,
 	AD1884_FIXUP_HP_EAPD,
+	AD1884_FIXUP_DMIC_COEF,
+	AD1884_FIXUP_HP_TOUCHSMART,
 };
 
 static const struct hda_fixup ad1884_fixups[] = {
@@ -3633,15 +993,27 @@ static const struct hda_fixup ad1884_fixups[] = {
 		.chained = true,
 		.chain_id = AD1884_FIXUP_AMP_OVERRIDE,
 	},
+	[AD1884_FIXUP_DMIC_COEF] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = ad1884_dmic_init_verbs,
+	},
+	[AD1884_FIXUP_HP_TOUCHSMART] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = ad1884_dmic_init_verbs,
+		.chained = true,
+		.chain_id = AD1884_FIXUP_HP_EAPD,
+	},
 };
 
 static const struct snd_pci_quirk ad1884_fixup_tbl[] = {
+	SND_PCI_QUIRK(0x103c, 0x2a82, "HP Touchsmart", AD1884_FIXUP_HP_TOUCHSMART),
 	SND_PCI_QUIRK_VENDOR(0x103c, "HP", AD1884_FIXUP_HP_EAPD),
+	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1884_FIXUP_DMIC_COEF),
 	{}
 };
 
 
-static int ad1884_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1884(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
 	int err;
@@ -3674,1170 +1046,6 @@ static int ad1884_parse_auto_config(struct hda_codec *codec)
 	return err;
 }
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1884_basic(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int err;
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	err = snd_hda_attach_beep_device(codec, 0x10);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
-	spec->multiout.max_channels = 2;
-	spec->multiout.num_dacs = ARRAY_SIZE(ad1884_dac_nids);
-	spec->multiout.dac_nids = ad1884_dac_nids;
-	spec->multiout.dig_out_nid = AD1884_SPDIF_OUT;
-	spec->num_adc_nids = ARRAY_SIZE(ad1884_adc_nids);
-	spec->adc_nids = ad1884_adc_nids;
-	spec->capsrc_nids = ad1884_capsrc_nids;
-	spec->input_mux = &ad1884_capture_source;
-	spec->num_mixers = 1;
-	spec->mixers[0] = ad1884_base_mixers;
-	spec->num_init_verbs = 1;
-	spec->init_verbs[0] = ad1884_init_verbs;
-	spec->spdif_route = 0;
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1884_loopbacks;
-#endif
-	spec->vmaster_nid = 0x04;
-	/* we need to cover all playback volumes */
-	spec->slave_vols = ad1884_slave_vols;
-	/* slaves may contain input volumes, so we can't raise to 0dB blindly */
-	spec->avoid_init_slave_vol = 1;
-
-	codec->patch_ops = ad198x_patch_ops;
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-
-static int patch_ad1884(struct hda_codec *codec)
-{
-	int board_config;
-
-	board_config = snd_hda_check_board_config(codec, AD1884_MODELS,
-						  ad1884_models, NULL);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1884_AUTO;
-	}
-
-	if (board_config == AD1884_AUTO)
-		return ad1884_parse_auto_config(codec);
-	else
-		return patch_ad1884_basic(codec);
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1884	ad1884_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-/*
- * Lenovo Thinkpad T61/X61
- */
-static const struct hda_input_mux ad1984_thinkpad_capture_source = {
-	.num_items = 4,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Internal Mic", 0x1 },
-		{ "Mix", 0x3 },
-		{ "Dock Mic", 0x4 },
-	},
-};
-
-
-/*
- * Dell Precision T3400
- */
-static const struct hda_input_mux ad1984_dell_desktop_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Front Mic", 0x0 },
-		{ "Line-In", 0x1 },
-		{ "Mix", 0x3 },
-	},
-};
-
-
-static const struct snd_kcontrol_new ad1984_thinkpad_mixers[] = {
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	/* HDA_CODEC_VOLUME_IDX("PCM Playback Volume", 1, 0x03, 0x0, HDA_OUTPUT), */
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	/* SPDIF controls */
-	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		/* identical with ad1983 */
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-/* additional verbs */
-static const struct hda_verb ad1984_thinkpad_init_verbs[] = {
-	/* Port-E (docking station mic) pin */
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* docking mic boost */
-	{0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	/* Analog PC Beeper - allow firmware/ACPI beeps */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3) | 0x1a},
-	/* Analog mixer - docking mic; mute as default */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	/* enable EAPD bit */
-	{0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
-	{ } /* end */
-};
-
-/*
- * Dell Precision T3400
- */
-static const struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = {
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line-In Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Line-In Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line-In Boost Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	{ } /* end */
-};
-
-/* Digial MIC ADC NID 0x05 + 0x06 */
-static int ad1984_pcm_dmic_prepare(struct hda_pcm_stream *hinfo,
-				   struct hda_codec *codec,
-				   unsigned int stream_tag,
-				   unsigned int format,
-				   struct snd_pcm_substream *substream)
-{
-	snd_hda_codec_setup_stream(codec, 0x05 + substream->number,
-				   stream_tag, 0, format);
-	return 0;
-}
-
-static int ad1984_pcm_dmic_cleanup(struct hda_pcm_stream *hinfo,
-				   struct hda_codec *codec,
-				   struct snd_pcm_substream *substream)
-{
-	snd_hda_codec_cleanup_stream(codec, 0x05 + substream->number);
-	return 0;
-}
-
-static const struct hda_pcm_stream ad1984_pcm_dmic_capture = {
-	.substreams = 2,
-	.channels_min = 2,
-	.channels_max = 2,
-	.nid = 0x05,
-	.ops = {
-		.prepare = ad1984_pcm_dmic_prepare,
-		.cleanup = ad1984_pcm_dmic_cleanup
-	},
-};
-
-static int ad1984_build_pcms(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec = codec->spec;
-	struct hda_pcm *info;
-	int err;
-
-	err = ad198x_build_pcms(codec);
-	if (err < 0)
-		return err;
-
-	info = spec->pcm_rec + codec->num_pcms;
-	codec->num_pcms++;
-	info->name = "AD1984 Digital Mic";
-	info->stream[SNDRV_PCM_STREAM_CAPTURE] = ad1984_pcm_dmic_capture;
-	return 0;
-}
-
-/* models */
-enum {
-	AD1984_AUTO,
-	AD1984_BASIC,
-	AD1984_THINKPAD,
-	AD1984_DELL_DESKTOP,
-	AD1984_MODELS
-};
-
-static const char * const ad1984_models[AD1984_MODELS] = {
-	[AD1984_AUTO]		= "auto",
-	[AD1984_BASIC]		= "basic",
-	[AD1984_THINKPAD]	= "thinkpad",
-	[AD1984_DELL_DESKTOP]	= "dell_desktop",
-};
-
-static const struct snd_pci_quirk ad1984_cfg_tbl[] = {
-	/* Lenovo Thinkpad T61/X61 */
-	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1984_THINKPAD),
-	SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP),
-	SND_PCI_QUIRK(0x1028, 0x0233, "Dell Latitude E6400", AD1984_DELL_DESKTOP),
-	{}
-};
-
-static int patch_ad1984(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int board_config, err;
-
-	board_config = snd_hda_check_board_config(codec, AD1984_MODELS,
-						  ad1984_models, ad1984_cfg_tbl);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1984_AUTO;
-	}
-
-	if (board_config == AD1984_AUTO)
-		return ad1884_parse_auto_config(codec);
-
-	err = patch_ad1884_basic(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	switch (board_config) {
-	case AD1984_BASIC:
-		/* additional digital mics */
-		spec->mixers[spec->num_mixers++] = ad1984_dmic_mixers;
-		codec->patch_ops.build_pcms = ad1984_build_pcms;
-		break;
-	case AD1984_THINKPAD:
-		if (codec->subsystem_id == 0x17aa20fb) {
-			/* Thinpad X300 does not have the ability to do SPDIF,
-			   or attach to docking station to use SPDIF */
-			spec->multiout.dig_out_nid = 0;
-		} else
-			spec->multiout.dig_out_nid = AD1884_SPDIF_OUT;
-		spec->input_mux = &ad1984_thinkpad_capture_source;
-		spec->mixers[0] = ad1984_thinkpad_mixers;
-		spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs;
-		spec->analog_beep = 1;
-		break;
-	case AD1984_DELL_DESKTOP:
-		spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1984_dell_desktop_capture_source;
-		spec->mixers[0] = ad1984_dell_desktop_mixers;
-		break;
-	}
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1984	ad1884_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
-/*
- * AD1883 / AD1884A / AD1984A / AD1984B
- *
- * port-B (0x14) - front mic-in
- * port-E (0x1c) - rear mic-in
- * port-F (0x16) - CD / ext out
- * port-C (0x15) - rear line-in
- * port-D (0x12) - rear line-out
- * port-A (0x11) - front hp-out
- *
- * AD1984A = AD1884A + digital-mic
- * AD1883 = equivalent with AD1984A
- * AD1984B = AD1984A + extra SPDIF-out
- *
- * FIXME:
- * We share the single DAC for both HP and line-outs (see AD1884/1984).
- */
-
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const hda_nid_t ad1884a_dac_nids[1] = {
-	0x03,
-};
-
-#define ad1884a_adc_nids	ad1884_adc_nids
-#define ad1884a_capsrc_nids	ad1884_capsrc_nids
-
-#define AD1884A_SPDIF_OUT	0x02
-
-static const struct hda_input_mux ad1884a_capture_source = {
-	.num_items = 5,
-	.items = {
-		{ "Front Mic", 0x0 },
-		{ "Mic", 0x4 },
-		{ "Line", 0x1 },
-		{ "CD", 0x2 },
-		{ "Mix", 0x3 },
-	},
-};
-
-static const struct snd_kcontrol_new ad1884a_base_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Boost Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	/* SPDIF controls */
-	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		/* identical with ad1983 */
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-/*
- * initialization verbs
- */
-static const struct hda_verb ad1884a_init_verbs[] = {
-	/* DACs; unmute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	/* Port-A (HP) mixer - route only from analog mixer */
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-A pin */
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Port-D (Line-out) mixer - route only from analog mixer */
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-D pin */
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Mono-out mixer - route only from analog mixer */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Mono-out pin */
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Port-B (front mic) pin */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	/* Port-C (rear line-in) pin */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	/* Port-E (rear mic) pin */
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* no boost */
-	/* Port-F (CD) pin */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Analog mixer; mute as default */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, /* aux */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* capture sources */
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* SPDIF output amp */
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-	{ } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1884a_loopbacks[] = {
-	{ 0x20, HDA_INPUT, 0 }, /* Front Mic */
-	{ 0x20, HDA_INPUT, 1 }, /* Mic */
-	{ 0x20, HDA_INPUT, 2 }, /* CD */
-	{ 0x20, HDA_INPUT, 4 }, /* Docking */
-	{ } /* end */
-};
-#endif
-
-/*
- * Laptop model
- *
- * Port A: Headphone jack
- * Port B: MIC jack
- * Port C: Internal MIC
- * Port D: Dock Line Out (if enabled)
- * Port E: Dock Line In (if enabled)
- * Port F: Internal speakers
- */
-
-static int ad1884a_mobile_master_sw_put(struct snd_kcontrol *kcontrol,
-					struct snd_ctl_elem_value *ucontrol)
-{
-	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
-	int ret = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol);
-	int mute = (!ucontrol->value.integer.value[0] &&
-		    !ucontrol->value.integer.value[1]);
-	/* toggle GPIO1 according to the mute state */
-	snd_hda_codec_write_cache(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA,
-			    mute ? 0x02 : 0x0);
-	return ret;
-}
-
-static const struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_AMP_FLAG,
-		.info = snd_hda_mixer_amp_switch_info,
-		.get = snd_hda_mixer_amp_switch_get,
-		.put = ad1884a_mobile_master_sw_put,
-		.private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
-	},
-	HDA_CODEC_MUTE("Dock Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Dock Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	/*HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Master Playback Switch",
-		.subdevice = HDA_SUBDEV_AMP_FLAG,
-		.info = snd_hda_mixer_amp_switch_info,
-		.get = snd_hda_mixer_amp_switch_get,
-		.put = ad1884a_mobile_master_sw_put,
-		.private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
-	},
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-/* mute internal speaker if HP is plugged */
-static void ad1884a_hp_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x11);
-	snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-	snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE,
-			    present ? 0x00 : 0x02);
-}
-
-/* switch to external mic if plugged */
-static void ad1884a_hp_automic(struct hda_codec *codec)
-{
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x14);
-	snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL,
-			    present ? 0 : 1);
-}
-
-#define AD1884A_HP_EVENT		0x37
-#define AD1884A_MIC_EVENT		0x36
-
-/* unsolicited event for HP jack sensing */
-static void ad1884a_hp_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	switch (res >> 26) {
-	case AD1884A_HP_EVENT:
-		ad1884a_hp_automute(codec);
-		break;
-	case AD1884A_MIC_EVENT:
-		ad1884a_hp_automic(codec);
-		break;
-	}
-}
-
-/* initialize jack-sensing, too */
-static int ad1884a_hp_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1884a_hp_automute(codec);
-	ad1884a_hp_automic(codec);
-	return 0;
-}
-
-/* mute internal speaker if HP or docking HP is plugged */
-static void ad1884a_laptop_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x11);
-	if (!present)
-		present = snd_hda_jack_detect(codec, 0x12);
-	snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-	snd_hda_codec_write(codec, 0x16, 0, AC_VERB_SET_EAPD_BTLENABLE,
-			    present ? 0x00 : 0x02);
-}
-
-/* switch to external mic if plugged */
-static void ad1884a_laptop_automic(struct hda_codec *codec)
-{
-	unsigned int idx;
-
-	if (snd_hda_jack_detect(codec, 0x14))
-		idx = 0;
-	else if (snd_hda_jack_detect(codec, 0x1c))
-		idx = 4;
-	else
-		idx = 1;
-	snd_hda_codec_write(codec, 0x0c, 0, AC_VERB_SET_CONNECT_SEL, idx);
-}
-
-/* unsolicited event for HP jack sensing */
-static void ad1884a_laptop_unsol_event(struct hda_codec *codec,
-				       unsigned int res)
-{
-	switch (res >> 26) {
-	case AD1884A_HP_EVENT:
-		ad1884a_laptop_automute(codec);
-		break;
-	case AD1884A_MIC_EVENT:
-		ad1884a_laptop_automic(codec);
-		break;
-	}
-}
-
-/* initialize jack-sensing, too */
-static int ad1884a_laptop_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1884a_laptop_automute(codec);
-	ad1884a_laptop_automic(codec);
-	return 0;
-}
-
-/* additional verbs for laptop model */
-static const struct hda_verb ad1884a_laptop_verbs[] = {
-	/* Port-A (HP) pin - always unmuted */
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-F (int speaker) mixer - route only from analog mixer */
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-F (int speaker) pin */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* required for compaq 6530s/6531s speaker output */
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	/* Port-C pin - internal mic-in */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
-	/* Port-D (docking line-out) pin - default unmuted */
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* analog mix */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	/* unsolicited event for pin-sense */
-	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
-	{0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
-	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
-	{0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
-	/* allow to touch GPIO1 (for mute control) */
-	{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
-	{ } /* end */
-};
-
-static const struct hda_verb ad1884a_mobile_verbs[] = {
-	/* DACs; unmute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	/* Port-A (HP) mixer - route only from analog mixer */
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-A pin */
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	/* Port-A (HP) pin - always unmuted */
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-B (mic jack) pin */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
-	/* Port-C (int mic) pin */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
-	/* Port-F (int speaker) mixer - route only from analog mixer */
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-F pin */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Analog mixer; mute as default */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* capture sources */
-	/* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* unsolicited event for pin-sense */
-	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
-	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
-	/* allow to touch GPIO1 (for mute control) */
-	{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
-	{ } /* end */
-};
-
-/*
- * Thinkpad X300
- * 0x11 - HP
- * 0x12 - speaker
- * 0x14 - mic-in
- * 0x17 - built-in mic
- */
-
-static const struct hda_verb ad1984a_thinkpad_verbs[] = {
-	/* HP unmute */
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* analog mix */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	/* turn on EAPD */
-	{0x12, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
-	/* unsolicited event for pin-sense */
-	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
-	/* internal mic - dmic */
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	/* set magic COEFs for dmic */
-	{0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
-	{0x01, AC_VERB_SET_PROC_COEF, 0x08},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x14, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x17, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	{ } /* end */
-};
-
-static const struct hda_input_mux ad1984a_thinkpad_capture_source = {
-	.num_items = 3,
-	.items = {
-		{ "Mic", 0x0 },
-		{ "Internal Mic", 0x5 },
-		{ "Mix", 0x3 },
-	},
-};
-
-/* mute internal speaker if HP is plugged */
-static void ad1984a_thinkpad_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x11);
-	snd_hda_codec_amp_stereo(codec, 0x12, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-/* unsolicited event for HP jack sensing */
-static void ad1984a_thinkpad_unsol_event(struct hda_codec *codec,
-					 unsigned int res)
-{
-	if ((res >> 26) != AD1884A_HP_EVENT)
-		return;
-	ad1984a_thinkpad_automute(codec);
-}
-
-/* initialize jack-sensing, too */
-static int ad1984a_thinkpad_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1984a_thinkpad_automute(codec);
-	return 0;
-}
-
-/*
- * Precision R5500
- * 0x12 - HP/line-out
- * 0x13 - speaker (mono)
- * 0x15 - mic-in
- */
-
-static const struct hda_verb ad1984a_precision_verbs[] = {
-	/* Unmute main output path */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x1f}, /* 0dB */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5) + 0x17}, /* 0dB */
-	/* Analog mixer; mute as default */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	/* Select mic as input */
-	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE + 0x27}, /* 0dB */
-	/* Configure as mic */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */
-	/* HP unmute */
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* turn on EAPD */
-	{0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x02},
-	/* unsolicited event for pin-sense */
-	{0x12, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984a_precision_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x13, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-
-/* mute internal speaker if HP is plugged */
-static void ad1984a_precision_automute(struct hda_codec *codec)
-{
-	unsigned int present;
-
-	present = snd_hda_jack_detect(codec, 0x12);
-	snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
-}
-
-
-/* unsolicited event for HP jack sensing */
-static void ad1984a_precision_unsol_event(struct hda_codec *codec,
-					 unsigned int res)
-{
-	if ((res >> 26) != AD1884A_HP_EVENT)
-		return;
-	ad1984a_precision_automute(codec);
-}
-
-/* initialize jack-sensing, too */
-static int ad1984a_precision_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1984a_precision_automute(codec);
-	return 0;
-}
-
-
-/*
- * HP Touchsmart
- * port-A (0x11)      - front hp-out
- * port-B (0x14)      - unused
- * port-C (0x15)      - unused
- * port-D (0x12)      - rear line out
- * port-E (0x1c)      - front mic-in
- * port-F (0x16)      - Internal speakers
- * digital-mic (0x17) - Internal mic
- */
-
-static const struct hda_verb ad1984a_touchsmart_verbs[] = {
-	/* DACs; unmute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, 0x27}, /* 0dB */
-	/* Port-A (HP) mixer - route only from analog mixer */
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-A pin */
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	/* Port-A (HP) pin - always unmuted */
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	/* Port-E (int speaker) mixer - route only from analog mixer */
-	{0x25, AC_VERB_SET_AMP_GAIN_MUTE, 0x03},
-	/* Port-E pin */
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
-	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	/* Port-F (int speaker) mixer - route only from analog mixer */
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-F pin */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Analog mixer; mute as default */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* capture sources */
-	/* {0x0c, AC_VERB_SET_CONNECT_SEL, 0x0}, */ /* set via unsol */
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* unsolicited event for pin-sense */
-	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_HP_EVENT},
-	{0x1c, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1884A_MIC_EVENT},
-	/* allow to touch GPIO1 (for mute control) */
-	{0x01, AC_VERB_SET_GPIO_MASK, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x02},
-	{0x01, AC_VERB_SET_GPIO_DATA, 0x02}, /* first muted */
-	/* internal mic - dmic */
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	/* set magic COEFs for dmic */
-	{0x01, AC_VERB_SET_COEF_INDEX, 0x13f7},
-	{0x01, AC_VERB_SET_PROC_COEF, 0x08},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1984a_touchsmart_mixers[] = {
-	HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT),
-/*	HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT),*/
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.subdevice = HDA_SUBDEV_AMP_FLAG,
-		.name = "Master Playback Switch",
-		.info = snd_hda_mixer_amp_switch_info,
-		.get = snd_hda_mixer_amp_switch_get,
-		.put = ad1884a_mobile_master_sw_put,
-		.private_value = HDA_COMPOSE_AMP_VAL(0x21, 3, 0, HDA_OUTPUT),
-	},
-	HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x25, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Internal Mic Boost Volume", 0x17, 0x0, HDA_INPUT),
-	{ } /* end */
-};
-
-/* switch to external mic if plugged */
-static void ad1984a_touchsmart_automic(struct hda_codec *codec)
-{
-	if (snd_hda_jack_detect(codec, 0x1c))
-		snd_hda_codec_write(codec, 0x0c, 0,
-				     AC_VERB_SET_CONNECT_SEL, 0x4);
-	else
-		snd_hda_codec_write(codec, 0x0c, 0,
-				     AC_VERB_SET_CONNECT_SEL, 0x5);
-}
-
-
-/* unsolicited event for HP jack sensing */
-static void ad1984a_touchsmart_unsol_event(struct hda_codec *codec,
-	unsigned int res)
-{
-	switch (res >> 26) {
-	case AD1884A_HP_EVENT:
-		ad1884a_hp_automute(codec);
-		break;
-	case AD1884A_MIC_EVENT:
-		ad1984a_touchsmart_automic(codec);
-		break;
-	}
-}
-
-/* initialize jack-sensing, too */
-static int ad1984a_touchsmart_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1884a_hp_automute(codec);
-	ad1984a_touchsmart_automic(codec);
-	return 0;
-}
-
-
-/*
- */
-
-enum {
-	AD1884A_AUTO,
-	AD1884A_DESKTOP,
-	AD1884A_LAPTOP,
-	AD1884A_MOBILE,
-	AD1884A_THINKPAD,
-	AD1984A_TOUCHSMART,
-	AD1984A_PRECISION,
-	AD1884A_MODELS
-};
-
-static const char * const ad1884a_models[AD1884A_MODELS] = {
-	[AD1884A_AUTO]		= "auto",
-	[AD1884A_DESKTOP]	= "desktop",
-	[AD1884A_LAPTOP]	= "laptop",
-	[AD1884A_MOBILE]	= "mobile",
-	[AD1884A_THINKPAD]	= "thinkpad",
-	[AD1984A_TOUCHSMART]	= "touchsmart",
-	[AD1984A_PRECISION]	= "precision",
-};
-
-static const struct snd_pci_quirk ad1884a_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1028, 0x04ac, "Precision R5500", AD1984A_PRECISION),
-	SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE),
-	SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP),
-	SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE),
-	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE),
-	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30d0, "HP laptop", AD1884A_LAPTOP),
-	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP),
-	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP),
-	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x7010, "HP laptop", AD1884A_MOBILE),
-	SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
-	SND_PCI_QUIRK(0x103c, 0x2a82, "Touchsmart", AD1984A_TOUCHSMART),
-	{}
-};
-
-static int patch_ad1884a(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int err, board_config;
-
-	board_config = snd_hda_check_board_config(codec, AD1884A_MODELS,
-						  ad1884a_models,
-						  ad1884a_cfg_tbl);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1884A_AUTO;
-	}
-
-	if (board_config == AD1884A_AUTO)
-		return ad1884_parse_auto_config(codec);
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	err = snd_hda_attach_beep_device(codec, 0x10);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
-	spec->multiout.max_channels = 2;
-	spec->multiout.num_dacs = ARRAY_SIZE(ad1884a_dac_nids);
-	spec->multiout.dac_nids = ad1884a_dac_nids;
-	spec->multiout.dig_out_nid = AD1884A_SPDIF_OUT;
-	spec->num_adc_nids = ARRAY_SIZE(ad1884a_adc_nids);
-	spec->adc_nids = ad1884a_adc_nids;
-	spec->capsrc_nids = ad1884a_capsrc_nids;
-	spec->input_mux = &ad1884a_capture_source;
-	spec->num_mixers = 1;
-	spec->mixers[0] = ad1884a_base_mixers;
-	spec->num_init_verbs = 1;
-	spec->init_verbs[0] = ad1884a_init_verbs;
-	spec->spdif_route = 0;
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1884a_loopbacks;
-#endif
-	codec->patch_ops = ad198x_patch_ops;
-
-	/* override some parameters */
-	switch (board_config) {
-	case AD1884A_LAPTOP:
-		spec->mixers[0] = ad1884a_laptop_mixers;
-		spec->init_verbs[spec->num_init_verbs++] = ad1884a_laptop_verbs;
-		spec->multiout.dig_out_nid = 0;
-		codec->patch_ops.unsol_event = ad1884a_laptop_unsol_event;
-		codec->patch_ops.init = ad1884a_laptop_init;
-		/* set the upper-limit for mixer amp to 0dB for avoiding the
-		 * possible damage by overloading
-		 */
-		snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
-					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
-					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
-					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
-					  (1 << AC_AMPCAP_MUTE_SHIFT));
-		break;
-	case AD1884A_MOBILE:
-		spec->mixers[0] = ad1884a_mobile_mixers;
-		spec->init_verbs[0] = ad1884a_mobile_verbs;
-		spec->multiout.dig_out_nid = 0;
-		codec->patch_ops.unsol_event = ad1884a_hp_unsol_event;
-		codec->patch_ops.init = ad1884a_hp_init;
-		/* set the upper-limit for mixer amp to 0dB for avoiding the
-		 * possible damage by overloading
-		 */
-		snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
-					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
-					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
-					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
-					  (1 << AC_AMPCAP_MUTE_SHIFT));
-		break;
-	case AD1884A_THINKPAD:
-		spec->mixers[0] = ad1984a_thinkpad_mixers;
-		spec->init_verbs[spec->num_init_verbs++] =
-			ad1984a_thinkpad_verbs;
-		spec->multiout.dig_out_nid = 0;
-		spec->input_mux = &ad1984a_thinkpad_capture_source;
-		codec->patch_ops.unsol_event = ad1984a_thinkpad_unsol_event;
-		codec->patch_ops.init = ad1984a_thinkpad_init;
-		break;
-	case AD1984A_PRECISION:
-		spec->mixers[0] = ad1984a_precision_mixers;
-		spec->init_verbs[spec->num_init_verbs++] =
-			ad1984a_precision_verbs;
-		spec->multiout.dig_out_nid = 0;
-		codec->patch_ops.unsol_event = ad1984a_precision_unsol_event;
-		codec->patch_ops.init = ad1984a_precision_init;
-		break;
-	case AD1984A_TOUCHSMART:
-		spec->mixers[0] = ad1984a_touchsmart_mixers;
-		spec->init_verbs[0] = ad1984a_touchsmart_verbs;
-		spec->multiout.dig_out_nid = 0;
-		codec->patch_ops.unsol_event = ad1984a_touchsmart_unsol_event;
-		codec->patch_ops.init = ad1984a_touchsmart_init;
-		/* set the upper-limit for mixer amp to 0dB for avoiding the
-		 * possible damage by overloading
-		 */
-		snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT,
-					  (0x17 << AC_AMPCAP_OFFSET_SHIFT) |
-					  (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) |
-					  (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) |
-					  (1 << AC_AMPCAP_MUTE_SHIFT));
-		break;
-	}
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1884a	ad1884_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-
 /*
  * AD1882 / AD1882A
  *
@@ -4850,299 +1058,7 @@ static int patch_ad1884a(struct hda_codec *codec)
  * port-G - rear clfe-out (6stack)
  */
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static const hda_nid_t ad1882_dac_nids[3] = {
-	0x04, 0x03, 0x05
-};
-
-static const hda_nid_t ad1882_adc_nids[2] = {
-	0x08, 0x09,
-};
-
-static const hda_nid_t ad1882_capsrc_nids[2] = {
-	0x0c, 0x0d,
-};
-
-#define AD1882_SPDIF_OUT	0x02
-
-/* list: 0x11, 0x39, 0x3a, 0x18, 0x3c, 0x3b, 0x12, 0x20 */
-static const struct hda_input_mux ad1882_capture_source = {
-	.num_items = 5,
-	.items = {
-		{ "Front Mic", 0x1 },
-		{ "Mic", 0x4 },
-		{ "Line", 0x2 },
-		{ "CD", 0x3 },
-		{ "Mix", 0x7 },
-	},
-};
-
-/* list: 0x11, 0x39, 0x3a, 0x3c, 0x18, 0x1f, 0x12, 0x20 */
-static const struct hda_input_mux ad1882a_capture_source = {
-	.num_items = 5,
-	.items = {
-		{ "Front Mic", 0x1 },
-		{ "Mic", 0x4},
-		{ "Line", 0x2 },
-		{ "Digital Mic", 0x06 },
-		{ "Mix", 0x7 },
-	},
-};
-
-static const struct snd_kcontrol_new ad1882_base_mixers[] = {
-	HDA_CODEC_VOLUME("Front Playback Volume", 0x04, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Surround Playback Volume", 0x03, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x05, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x05, 2, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Front Playback Switch", 0x12, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x13, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x13, 1, 0x0, HDA_OUTPUT),
-
-	HDA_CODEC_VOLUME("Mic Boost Volume", 0x3c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Front Mic Boost Volume", 0x39, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Line-In Boost Volume", 0x3a, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x0d, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		/* The multiple "Capture Source" controls confuse alsamixer
-		 * So call somewhat different..
-		 */
-		/* .name = "Capture Source", */
-		.name = "Input Source",
-		.count = 2,
-		.info = ad198x_mux_enum_info,
-		.get = ad198x_mux_enum_get,
-		.put = ad198x_mux_enum_put,
-	},
-	/* SPDIF controls */
-	HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source",
-		/* identical with ad1983 */
-		.info = ad1983_spdif_route_info,
-		.get = ad1983_spdif_route_get,
-		.put = ad1983_spdif_route_put,
-	},
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1882_loopback_mixers[] = {
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1882a_loopback_mixers[] = {
-	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT),
-	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
-	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
-	HDA_CODEC_VOLUME("Digital Mic Boost Volume", 0x1f, 0x0, HDA_INPUT),
-	{ } /* end */
-};
-
-static const struct snd_kcontrol_new ad1882_3stack_mixers[] = {
-	HDA_CODEC_MUTE("Surround Playback Switch", 0x15, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x17, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x17, 2, 0x0, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Channel Mode",
-		.info = ad198x_ch_mode_info,
-		.get = ad198x_ch_mode_get,
-		.put = ad198x_ch_mode_put,
-	},
-	{ } /* end */
-};
-
-/* simple auto-mute control for AD1882 3-stack board */
-#define AD1882_HP_EVENT	0x01
-
-static void ad1882_3stack_automute(struct hda_codec *codec)
-{
-	bool mute = snd_hda_jack_detect(codec, 0x11);
-	snd_hda_codec_write(codec, 0x12, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
-			    mute ? 0 : PIN_OUT);
-}
-
-static int ad1882_3stack_automute_init(struct hda_codec *codec)
-{
-	ad198x_init(codec);
-	ad1882_3stack_automute(codec);
-	return 0;
-}
-
-static void ad1882_3stack_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	switch (res >> 26) {
-	case AD1882_HP_EVENT:
-		ad1882_3stack_automute(codec);
-		break;
-	}
-}
-
-static const struct snd_kcontrol_new ad1882_6stack_mixers[] = {
-	HDA_CODEC_MUTE("Surround Playback Switch", 0x16, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("Center Playback Switch", 0x24, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("LFE Playback Switch", 0x24, 2, 0x0, HDA_OUTPUT),
-	{ } /* end */
-};
-
-static const struct hda_verb ad1882_ch2_init[] = {
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{ } /* end */
-};
-
-static const struct hda_verb ad1882_ch4_init[] = {
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{ } /* end */
-};
-
-static const struct hda_verb ad1882_ch6_init[] = {
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{ } /* end */
-};
-
-static const struct hda_channel_mode ad1882_modes[3] = {
-	{ 2, ad1882_ch2_init },
-	{ 4, ad1882_ch4_init },
-	{ 6, ad1882_ch6_init },
-};
-
-/*
- * initialization verbs
- */
-static const struct hda_verb ad1882_init_verbs[] = {
-	/* DACs; mute as default */
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	/* Port-A (HP) mixer */
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-A pin */
-	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* HP selector - select DAC2 */
-	{0x37, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* Port-D (Line-out) mixer */
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x29, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Port-D pin */
-	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Mono-out mixer */
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x1e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	/* Mono-out pin */
-	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
-	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Port-B (front mic) pin */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x39, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
-	/* Port-C (line-in) pin */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x3a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
-	/* Port-C mixer - mute as input */
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x2c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	/* Port-E (mic-in) pin */
-	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* boost */
-	/* Port-E mixer - mute as input */
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x26, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	/* Port-F (surround) */
-	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Port-G (CLFE) */
-	{0x24, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
-	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Analog mixer; mute as default */
-	/* list: 0x39, 0x3a, 0x11, 0x12, 0x3c, 0x3b, 0x18, 0x1a */
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)},
-	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)},
-	/* Analog Mix output amp */
-	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
-	/* SPDIF output selector */
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-	{0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */
-	{0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */
-	{ } /* end */
-};
-
-static const struct hda_verb ad1882_3stack_automute_verbs[] = {
-	{0x11, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | AD1882_HP_EVENT},
-	{ } /* end */
-};
-
-#ifdef CONFIG_PM
-static const struct hda_amp_list ad1882_loopbacks[] = {
-	{ 0x20, HDA_INPUT, 0 }, /* Front Mic */
-	{ 0x20, HDA_INPUT, 1 }, /* Mic */
-	{ 0x20, HDA_INPUT, 4 }, /* Line */
-	{ 0x20, HDA_INPUT, 6 }, /* CD */
-	{ } /* end */
-};
-#endif
-
-/* models */
-enum {
-	AD1882_AUTO,
-	AD1882_3STACK,
-	AD1882_6STACK,
-	AD1882_3STACK_AUTOMUTE,
-	AD1882_MODELS
-};
-
-static const char * const ad1882_models[AD1986A_MODELS] = {
-	[AD1882_AUTO]		= "auto",
-	[AD1882_3STACK]		= "3stack",
-	[AD1882_6STACK]		= "6stack",
-	[AD1882_3STACK_AUTOMUTE] = "3stack-automute",
-};
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
-static int ad1882_parse_auto_config(struct hda_codec *codec)
+static int patch_ad1882(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
 	int err;
@@ -5169,110 +1085,20 @@ static int ad1882_parse_auto_config(struct hda_codec *codec)
 	return err;
 }
 
-#ifdef ENABLE_AD_STATIC_QUIRKS
-static int patch_ad1882(struct hda_codec *codec)
-{
-	struct ad198x_spec *spec;
-	int err, board_config;
-
-	board_config = snd_hda_check_board_config(codec, AD1882_MODELS,
-						  ad1882_models, NULL);
-	if (board_config < 0) {
-		printk(KERN_INFO "hda_codec: %s: BIOS auto-probing.\n",
-		       codec->chip_name);
-		board_config = AD1882_AUTO;
-	}
-
-	if (board_config == AD1882_AUTO)
-		return ad1882_parse_auto_config(codec);
-
-	err = alloc_ad_spec(codec);
-	if (err < 0)
-		return err;
-	spec = codec->spec;
-
-	err = snd_hda_attach_beep_device(codec, 0x10);
-	if (err < 0) {
-		ad198x_free(codec);
-		return err;
-	}
-	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
-
-	spec->multiout.max_channels = 6;
-	spec->multiout.num_dacs = 3;
-	spec->multiout.dac_nids = ad1882_dac_nids;
-	spec->multiout.dig_out_nid = AD1882_SPDIF_OUT;
-	spec->num_adc_nids = ARRAY_SIZE(ad1882_adc_nids);
-	spec->adc_nids = ad1882_adc_nids;
-	spec->capsrc_nids = ad1882_capsrc_nids;
-	if (codec->vendor_id == 0x11d41882)
-		spec->input_mux = &ad1882_capture_source;
-	else
-		spec->input_mux = &ad1882a_capture_source;
-	spec->num_mixers = 2;
-	spec->mixers[0] = ad1882_base_mixers;
-	if (codec->vendor_id == 0x11d41882)
-		spec->mixers[1] = ad1882_loopback_mixers;
-	else
-		spec->mixers[1] = ad1882a_loopback_mixers;
-	spec->num_init_verbs = 1;
-	spec->init_verbs[0] = ad1882_init_verbs;
-	spec->spdif_route = 0;
-#ifdef CONFIG_PM
-	spec->loopback.amplist = ad1882_loopbacks;
-#endif
-	spec->vmaster_nid = 0x04;
-
-	codec->patch_ops = ad198x_patch_ops;
-
-	/* override some parameters */
-	switch (board_config) {
-	default:
-	case AD1882_3STACK:
-	case AD1882_3STACK_AUTOMUTE:
-		spec->num_mixers = 3;
-		spec->mixers[2] = ad1882_3stack_mixers;
-		spec->channel_mode = ad1882_modes;
-		spec->num_channel_mode = ARRAY_SIZE(ad1882_modes);
-		spec->need_dac_fix = 1;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = 1;
-		if (board_config != AD1882_3STACK) {
-			spec->init_verbs[spec->num_init_verbs++] =
-				ad1882_3stack_automute_verbs;
-			codec->patch_ops.unsol_event = ad1882_3stack_unsol_event;
-			codec->patch_ops.init = ad1882_3stack_automute_init;
-		}
-		break;
-	case AD1882_6STACK:
-		spec->num_mixers = 3;
-		spec->mixers[2] = ad1882_6stack_mixers;
-		break;
-	}
-
-	codec->no_trigger_sense = 1;
-	codec->no_sticky_stream = 1;
-
-	return 0;
-}
-#else /* ENABLE_AD_STATIC_QUIRKS */
-#define patch_ad1882	ad1882_parse_auto_config
-#endif /* ENABLE_AD_STATIC_QUIRKS */
-
 
 /*
  * patch entries
  */
 static const struct hda_codec_preset snd_hda_preset_analog[] = {
-	{ .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884a },
+	{ .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884 },
 	{ .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 },
-	{ .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884a },
+	{ .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884 },
 	{ .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 },
-	{ .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884a },
-	{ .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884a },
+	{ .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884 },
+	{ .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884 },
 	{ .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 },
 	{ .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 },
-	{ .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 },
+	{ .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1884 },
 	{ .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a },
 	{ .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 },
 	{ .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 },
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index de00ce166470..4edd2d0f9a3c 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -66,6 +66,8 @@ struct conexant_spec {
 	hda_nid_t eapds[4];
 	bool dynamic_eapd;
 
+	unsigned int parse_flags; /* flag for snd_hda_parse_pin_defcfg() */
+
 #ifdef ENABLE_CXT_STATIC_QUIRKS
 	const struct snd_kcontrol_new *mixers[5];
 	int num_mixers;
@@ -3200,6 +3202,9 @@ static int cx_auto_init(struct hda_codec *codec)
 	snd_hda_gen_init(codec);
 	if (!spec->dynamic_eapd)
 		cx_auto_turn_eapd(codec, spec->num_eapds, spec->eapds, true);
+
+	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT);
+
 	return 0;
 }
 
@@ -3224,6 +3229,8 @@ enum {
 	CXT_PINCFG_LEMOTE_A1205,
 	CXT_FIXUP_STEREO_DMIC,
 	CXT_FIXUP_INC_MIC_BOOST,
+	CXT_FIXUP_HEADPHONE_MIC_PIN,
+	CXT_FIXUP_HEADPHONE_MIC,
 };
 
 static void cxt_fixup_stereo_dmic(struct hda_codec *codec,
@@ -3246,6 +3253,59 @@ static void cxt5066_increase_mic_boost(struct hda_codec *codec,
 				  (0 << AC_AMPCAP_MUTE_SHIFT));
 }
 
+static void cxt_update_headset_mode(struct hda_codec *codec)
+{
+	/* The verbs used in this function were tested on a Conexant CX20751/2 codec. */
+	int i;
+	bool mic_mode = false;
+	struct conexant_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->gen.autocfg;
+
+	hda_nid_t mux_pin = spec->gen.imux_pins[spec->gen.cur_mux[0]];
+
+	for (i = 0; i < cfg->num_inputs; i++)
+		if (cfg->inputs[i].pin == mux_pin) {
+			mic_mode = !!cfg->inputs[i].is_headphone_mic;
+			break;
+		}
+
+	if (mic_mode) {
+		snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x7c); /* enable merged mode for analog int-mic */
+		spec->gen.hp_jack_present = false;
+	} else {
+		snd_hda_codec_write_cache(codec, 0x1c, 0, 0x410, 0x54); /* disable merged mode for analog int-mic */
+		spec->gen.hp_jack_present = snd_hda_jack_detect(codec, spec->gen.autocfg.hp_pins[0]);
+	}
+
+	snd_hda_gen_update_outputs(codec);
+}
+
+static void cxt_update_headset_mode_hook(struct hda_codec *codec,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	cxt_update_headset_mode(codec);
+}
+
+static void cxt_fixup_headphone_mic(struct hda_codec *codec,
+				    const struct hda_fixup *fix, int action)
+{
+	struct conexant_spec *spec = codec->spec;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		spec->parse_flags |= HDA_PINCFG_HEADPHONE_MIC;
+		break;
+	case HDA_FIXUP_ACT_PROBE:
+		spec->gen.cap_sync_hook = cxt_update_headset_mode_hook;
+		spec->gen.automute_hook = cxt_update_headset_mode;
+		break;
+	case HDA_FIXUP_ACT_INIT:
+		cxt_update_headset_mode(codec);
+		break;
+	}
+}
+
+
 /* ThinkPad X200 & co with cxt5051 */
 static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = {
 	{ 0x16, 0x042140ff }, /* HP (seq# overridden) */
@@ -3302,6 +3362,19 @@ static const struct hda_fixup cxt_fixups[] = {
 		.type = HDA_FIXUP_FUNC,
 		.v.func = cxt5066_increase_mic_boost,
 	},
+	[CXT_FIXUP_HEADPHONE_MIC_PIN] = {
+		.type = HDA_FIXUP_PINS,
+		.chained = true,
+		.chain_id = CXT_FIXUP_HEADPHONE_MIC,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x18, 0x03a1913d }, /* use as headphone mic, without its own jack detect */
+			{ }
+		}
+	},
+	[CXT_FIXUP_HEADPHONE_MIC] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = cxt_fixup_headphone_mic,
+	},
 };
 
 static const struct snd_pci_quirk cxt5051_fixups[] = {
@@ -3311,6 +3384,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = {
 
 static const struct snd_pci_quirk cxt5066_fixups[] = {
 	SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC),
+	SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410),
 	SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410),
@@ -3395,7 +3469,8 @@ static int patch_conexant_auto(struct hda_codec *codec)
 
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE);
 
-	err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL, 0);
+	err = snd_hda_parse_pin_defcfg(codec, &spec->gen.autocfg, NULL,
+				       spec->parse_flags);
 	if (err < 0)
 		goto error;
 
@@ -3416,6 +3491,8 @@ static int patch_conexant_auto(struct hda_codec *codec)
 		codec->bus->allow_bus_reset = 1;
 	}
 
+	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
+
 	return 0;
 
  error:
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 9f3586276871..24d82d6c3464 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -959,6 +959,7 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 	int pin_nid;
 	int pin_idx;
 	struct hda_jack_tbl *jack;
+	int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT;
 
 	jack = snd_hda_jack_tbl_get_from_tag(codec, tag);
 	if (!jack)
@@ -967,8 +968,8 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 	jack->jack_dirty = 1;
 
 	_snd_printd(SND_PR_VERBOSE,
-		"HDMI hot plug event: Codec=%d Pin=%d Presence_Detect=%d ELD_Valid=%d\n",
-		codec->addr, pin_nid,
+		"HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n",
+		codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA),
 		!!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV));
 
 	pin_idx = pin_nid_to_pin_index(spec, pin_nid);
@@ -1992,8 +1993,10 @@ static int patch_generic_hdmi(struct hda_codec *codec)
 		return -EINVAL;
 	}
 	codec->patch_ops = generic_hdmi_patch_ops;
-	if (codec->vendor_id == 0x80862807)
+	if (codec->vendor_id == 0x80862807) {
 		codec->patch_ops.set_power_state = haswell_set_power_state;
+		codec->dp_mst = true;
+	}
 
 	generic_hdmi_init_per_pins(codec);
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 389db4c2801b..9e9378cde8fa 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -282,6 +282,7 @@ static void alc_eapd_shutup(struct hda_codec *codec)
 {
 	alc_auto_setup_eapd(codec, false);
 	msleep(200);
+	snd_hda_shutup_pins(codec);
 }
 
 /* generic EAPD initialization */
@@ -826,7 +827,8 @@ static inline void alc_shutup(struct hda_codec *codec)
 
 	if (spec && spec->shutup)
 		spec->shutup(codec);
-	snd_hda_shutup_pins(codec);
+	else
+		snd_hda_shutup_pins(codec);
 }
 
 #define alc_free	snd_hda_gen_free
@@ -1853,8 +1855,10 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec,
 				       const struct hda_fixup *fix, int action)
 {
 	struct alc_spec *spec = codec->spec;
-	if (action == HDA_FIXUP_ACT_PRE_PROBE)
+	if (action == HDA_FIXUP_ACT_PRE_PROBE) {
 		spec->gen.no_primary_hp = 1;
+		spec->gen.no_multi_io = 1;
+	}
 }
 
 static const struct hda_fixup alc882_fixups[] = {
@@ -2533,6 +2537,7 @@ enum {
 	ALC269_TYPE_ALC269VD,
 	ALC269_TYPE_ALC280,
 	ALC269_TYPE_ALC282,
+	ALC269_TYPE_ALC283,
 	ALC269_TYPE_ALC284,
 	ALC269_TYPE_ALC286,
 };
@@ -2558,6 +2563,7 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	case ALC269_TYPE_ALC269VB:
 	case ALC269_TYPE_ALC269VD:
 	case ALC269_TYPE_ALC282:
+	case ALC269_TYPE_ALC283:
 	case ALC269_TYPE_ALC286:
 		ssids = alc269_ssids;
 		break;
@@ -2583,15 +2589,81 @@ static void alc269_shutup(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 
-	if (spec->codec_variant != ALC269_TYPE_ALC269VB)
-		return;
-
 	if (spec->codec_variant == ALC269_TYPE_ALC269VB)
 		alc269vb_toggle_power_output(codec, 0);
 	if (spec->codec_variant == ALC269_TYPE_ALC269VB &&
 			(alc_get_coef0(codec) & 0x00ff) == 0x018) {
 		msleep(150);
 	}
+	snd_hda_shutup_pins(codec);
+}
+
+static void alc283_init(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0];
+	bool hp_pin_sense;
+	int val;
+
+	if (!hp_pin)
+		return;
+	hp_pin_sense = snd_hda_jack_detect(codec, hp_pin);
+
+	/* Index 0x43 Direct Drive HP AMP LPM Control 1 */
+	/* Headphone capless set to high power mode */
+	alc_write_coef_idx(codec, 0x43, 0x9004);
+
+	snd_hda_codec_write(codec, hp_pin, 0,
+			    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+
+	if (hp_pin_sense)
+		msleep(85);
+
+	snd_hda_codec_write(codec, hp_pin, 0,
+			    AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT);
+
+	if (hp_pin_sense)
+		msleep(85);
+	/* Index 0x46 Combo jack auto switch control 2 */
+	/* 3k pull low control for Headset jack. */
+	val = alc_read_coef_idx(codec, 0x46);
+	alc_write_coef_idx(codec, 0x46, val & ~(3 << 12));
+	/* Headphone capless set to normal mode */
+	alc_write_coef_idx(codec, 0x43, 0x9614);
+}
+
+static void alc283_shutup(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0];
+	bool hp_pin_sense;
+	int val;
+
+	if (!hp_pin) {
+		alc269_shutup(codec);
+		return;
+	}
+
+	hp_pin_sense = snd_hda_jack_detect(codec, hp_pin);
+
+	alc_write_coef_idx(codec, 0x43, 0x9004);
+
+	snd_hda_codec_write(codec, hp_pin, 0,
+			    AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
+
+	if (hp_pin_sense)
+		msleep(85);
+
+	snd_hda_codec_write(codec, hp_pin, 0,
+			    AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0);
+
+	val = alc_read_coef_idx(codec, 0x46);
+	alc_write_coef_idx(codec, 0x46, val | (3 << 12));
+
+	if (hp_pin_sense)
+		msleep(85);
+	snd_hda_shutup_pins(codec);
+	alc_write_coef_idx(codec, 0x43, 0x9614);
 }
 
 static void alc5505_coef_set(struct hda_codec *codec, unsigned int index_reg,
@@ -2722,6 +2794,7 @@ static int alc269_resume(struct hda_codec *codec)
 	hda_call_check_power_status(codec, 0x01);
 	if (spec->has_alc5505_dsp)
 		alc5505_dsp_resume(codec);
+
 	return 0;
 }
 #endif /* CONFIG_PM */
@@ -3261,6 +3334,28 @@ static void alc_fixup_headset_mode_alc668(struct hda_codec *codec,
 	alc_fixup_headset_mode(codec, fix, action);
 }
 
+/* Returns the nid of the external mic input pin, or 0 if it cannot be found. */
+static int find_ext_mic_pin(struct hda_codec *codec)
+{
+	struct alc_spec *spec = codec->spec;
+	struct auto_pin_cfg *cfg = &spec->gen.autocfg;
+	hda_nid_t nid;
+	unsigned int defcfg;
+	int i;
+
+	for (i = 0; i < cfg->num_inputs; i++) {
+		if (cfg->inputs[i].type != AUTO_PIN_MIC)
+			continue;
+		nid = cfg->inputs[i].pin;
+		defcfg = snd_hda_codec_get_pincfg(codec, nid);
+		if (snd_hda_get_input_pin_attr(defcfg) == INPUT_PIN_ATTR_INT)
+			continue;
+		return nid;
+	}
+
+	return 0;
+}
+
 static void alc271_hp_gate_mic_jack(struct hda_codec *codec,
 				    const struct hda_fixup *fix,
 				    int action)
@@ -3268,11 +3363,12 @@ static void alc271_hp_gate_mic_jack(struct hda_codec *codec,
 	struct alc_spec *spec = codec->spec;
 
 	if (action == HDA_FIXUP_ACT_PROBE) {
-		if (snd_BUG_ON(!spec->gen.am_entry[1].pin ||
-			       !spec->gen.autocfg.hp_pins[0]))
+		int mic_pin = find_ext_mic_pin(codec);
+		int hp_pin = spec->gen.autocfg.hp_pins[0];
+
+		if (snd_BUG_ON(!mic_pin || !hp_pin))
 			return;
-		snd_hda_jack_set_gating_jack(codec, spec->gen.am_entry[1].pin,
-					     spec->gen.autocfg.hp_pins[0]);
+		snd_hda_jack_set_gating_jack(codec, mic_pin, hp_pin);
 	}
 }
 
@@ -3308,6 +3404,45 @@ static void alc269_fixup_limit_int_mic_boost(struct hda_codec *codec,
 	}
 }
 
+static void alc283_hp_automute_hook(struct hda_codec *codec,
+				    struct hda_jack_tbl *jack)
+{
+	struct alc_spec *spec = codec->spec;
+	int vref;
+
+	msleep(200);
+	snd_hda_gen_hp_automute(codec, jack);
+
+	vref = spec->gen.hp_jack_present ? PIN_VREF80 : 0;
+
+	msleep(600);
+	snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL,
+			    vref);
+}
+
+static void alc283_chromebook_caps(struct hda_codec *codec)
+{
+	snd_hda_override_wcaps(codec, 0x03, 0);
+}
+
+static void alc283_fixup_chromebook(struct hda_codec *codec,
+				    const struct hda_fixup *fix, int action)
+{
+	struct alc_spec *spec = codec->spec;
+	int val;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		alc283_chromebook_caps(codec);
+		spec->gen.hp_automute_hook = alc283_hp_automute_hook;
+		/* MIC2-VREF control */
+		/* Set to manual mode */
+		val = alc_read_coef_idx(codec, 0x06);
+		alc_write_coef_idx(codec, 0x06, val & ~0x000c);
+		break;
+	}
+}
+
 enum {
 	ALC269_FIXUP_SONY_VAIO,
 	ALC275_FIXUP_SONY_VAIO_GPIO2,
@@ -3344,6 +3479,7 @@ enum {
 	ALC269_FIXUP_ACER_AC700,
 	ALC269_FIXUP_LIMIT_INT_MIC_BOOST,
 	ALC269VB_FIXUP_ORDISSIMO_EVE2,
+	ALC283_FIXUP_CHROME_BOOK,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -3595,11 +3731,20 @@ static const struct hda_fixup alc269_fixups[] = {
 			{ }
 		},
 	},
+	[ALC283_FIXUP_CHROME_BOOK] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc283_fixup_chromebook,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC),
 	SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC),
+	SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700),
+	SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
+	SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
+	SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
+	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
 	SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x05c4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE),
@@ -3637,6 +3782,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED),
 	SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
 	SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1),
+	SND_PCI_QUIRK(0x103c, 0x21ed, "HP Falco Chromebook", ALC283_FIXUP_CHROME_BOOK),
 	SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED),
 	SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -3655,11 +3801,6 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ),
 	SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO),
-	SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z),
-	SND_PCI_QUIRK(0x1025, 0x047c, "Acer AC700", ALC269_FIXUP_ACER_AC700),
-	SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK),
-	SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK),
-	SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC),
 	SND_PCI_QUIRK(0x10cf, 0x1475, "Lifebook", ALC269_FIXUP_LIFEBOOK),
 	SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
 	SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE),
@@ -3670,8 +3811,16 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK),
-	SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK),
 	SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK),
+	SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK),
+	SND_PCI_QUIRK(0x17aa, 0x220c, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x2212, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
+	SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
 	SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K),
 	SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD),
 	SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
@@ -3840,11 +3989,15 @@ static int patch_alc269(struct hda_codec *codec)
 	case 0x10ec0290:
 		spec->codec_variant = ALC269_TYPE_ALC280;
 		break;
-	case 0x10ec0233:
 	case 0x10ec0282:
-	case 0x10ec0283:
 		spec->codec_variant = ALC269_TYPE_ALC282;
 		break;
+	case 0x10ec0233:
+	case 0x10ec0283:
+		spec->codec_variant = ALC269_TYPE_ALC283;
+		spec->shutup = alc283_shutup;
+		spec->init_hook = alc283_init;
+		break;
 	case 0x10ec0284:
 	case 0x10ec0292:
 		spec->codec_variant = ALC269_TYPE_ALC284;
@@ -3872,7 +4025,8 @@ static int patch_alc269(struct hda_codec *codec)
 	codec->patch_ops.suspend = alc269_suspend;
 	codec->patch_ops.resume = alc269_resume;
 #endif
-	spec->shutup = alc269_shutup;
+	if (!spec->shutup)
+		spec->shutup = alc269_shutup;
 
 	snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PROBE);
 
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 6d1924c19abf..fba0cef1c47f 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -158,6 +158,7 @@ enum {
 	STAC_D965_VERBS,
 	STAC_DELL_3ST,
 	STAC_DELL_BIOS,
+	STAC_DELL_BIOS_AMIC,
 	STAC_DELL_BIOS_SPDIF,
 	STAC_927X_DELL_DMIC,
 	STAC_927X_VOLKNOB,
@@ -3231,8 +3232,6 @@ static const struct hda_fixup stac927x_fixups[] = {
 	[STAC_DELL_BIOS] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
-			/* configure the analog microphone on some laptops */
-			{ 0x0c, 0x90a79130 },
 			/* correct the front output jack as a hp out */
 			{ 0x0f, 0x0221101f },
 			/* correct the front input jack as a mic */
@@ -3242,6 +3241,16 @@ static const struct hda_fixup stac927x_fixups[] = {
 		.chained = true,
 		.chain_id = STAC_927X_DELL_DMIC,
 	},
+	[STAC_DELL_BIOS_AMIC] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			/* configure the analog microphone on some laptops */
+			{ 0x0c, 0x90a79130 },
+			{}
+		},
+		.chained = true,
+		.chain_id = STAC_DELL_BIOS,
+	},
 	[STAC_DELL_BIOS_SPDIF] = {
 		.type = HDA_FIXUP_PINS,
 		.v.pins = (const struct hda_pintbl[]) {
@@ -3270,6 +3279,7 @@ static const struct hda_model_fixup stac927x_models[] = {
 	{ .id = STAC_D965_5ST_NO_FP, .name = "5stack-no-fp" },
 	{ .id = STAC_DELL_3ST, .name = "dell-3stack" },
 	{ .id = STAC_DELL_BIOS, .name = "dell-bios" },
+	{ .id = STAC_DELL_BIOS_AMIC, .name = "dell-bios-amic" },
 	{ .id = STAC_927X_VOLKNOB, .name = "volknob" },
 	{}
 };
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index e2481baddc70..0bc20ef5687a 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -207,9 +207,9 @@ static void vt1708_stop_hp_work(struct hda_codec *codec)
 		return;
 	if (spec->hp_work_active) {
 		snd_hda_codec_write(codec, 0x1, 0, 0xf81, 1);
+		codec->jackpoll_interval = 0;
 		cancel_delayed_work_sync(&codec->jackpoll_work);
 		spec->hp_work_active = false;
-		codec->jackpoll_interval = 0;
 	}
 }
 
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index 2a8ad9d1a2ae..bb9ebc5543d7 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -28,6 +28,7 @@
 #include <linux/interrupt.h>
 #include <linux/pci.h>
 #include <linux/module.h>
+#include <linux/vmalloc.h>
 
 #include <sound/core.h>
 #include <sound/info.h>
@@ -198,6 +199,31 @@ MODULE_PARM_DESC(enable, "Enable RME Digi96 soundcard.");
 #define RME96_AD1852_VOL_BITS 14
 #define RME96_AD1855_VOL_BITS 10
 
+/* Defines for snd_rme96_trigger */
+#define RME96_TB_START_PLAYBACK 1
+#define RME96_TB_START_CAPTURE 2
+#define RME96_TB_STOP_PLAYBACK 4
+#define RME96_TB_STOP_CAPTURE 8
+#define RME96_TB_RESET_PLAYPOS 16
+#define RME96_TB_RESET_CAPTUREPOS 32
+#define RME96_TB_CLEAR_PLAYBACK_IRQ 64
+#define RME96_TB_CLEAR_CAPTURE_IRQ 128
+#define RME96_RESUME_PLAYBACK	(RME96_TB_START_PLAYBACK)
+#define RME96_RESUME_CAPTURE	(RME96_TB_START_CAPTURE)
+#define RME96_RESUME_BOTH	(RME96_RESUME_PLAYBACK \
+				| RME96_RESUME_CAPTURE)
+#define RME96_START_PLAYBACK	(RME96_TB_START_PLAYBACK \
+				| RME96_TB_RESET_PLAYPOS)
+#define RME96_START_CAPTURE	(RME96_TB_START_CAPTURE \
+				| RME96_TB_RESET_CAPTUREPOS)
+#define RME96_START_BOTH	(RME96_START_PLAYBACK \
+				| RME96_START_CAPTURE)
+#define RME96_STOP_PLAYBACK	(RME96_TB_STOP_PLAYBACK \
+				| RME96_TB_CLEAR_PLAYBACK_IRQ)
+#define RME96_STOP_CAPTURE	(RME96_TB_STOP_CAPTURE \
+				| RME96_TB_CLEAR_CAPTURE_IRQ)
+#define RME96_STOP_BOTH		(RME96_STOP_PLAYBACK \
+				| RME96_STOP_CAPTURE)
 
 struct rme96 {
 	spinlock_t    lock;
@@ -214,6 +240,13 @@ struct rme96 {
 
 	u8 rev; /* card revision number */
 
+#ifdef CONFIG_PM
+	u32 playback_pointer;
+	u32 capture_pointer;
+	void *playback_suspend_buffer;
+	void *capture_suspend_buffer;
+#endif
+
 	struct snd_pcm_substream *playback_substream;
 	struct snd_pcm_substream *capture_substream;
 
@@ -344,6 +377,8 @@ static struct snd_pcm_hardware snd_rme96_playback_spdif_info =
 {
 	.info =		     (SNDRV_PCM_INFO_MMAP_IOMEM |
 			      SNDRV_PCM_INFO_MMAP_VALID |
+			      SNDRV_PCM_INFO_SYNC_START |
+			      SNDRV_PCM_INFO_RESUME |
 			      SNDRV_PCM_INFO_INTERLEAVED |
 			      SNDRV_PCM_INFO_PAUSE),
 	.formats =	     (SNDRV_PCM_FMTBIT_S16_LE |
@@ -373,6 +408,8 @@ static struct snd_pcm_hardware snd_rme96_capture_spdif_info =
 {
 	.info =		     (SNDRV_PCM_INFO_MMAP_IOMEM |
 			      SNDRV_PCM_INFO_MMAP_VALID |
+			      SNDRV_PCM_INFO_SYNC_START |
+			      SNDRV_PCM_INFO_RESUME |
 			      SNDRV_PCM_INFO_INTERLEAVED |
 			      SNDRV_PCM_INFO_PAUSE),
 	.formats =	     (SNDRV_PCM_FMTBIT_S16_LE |
@@ -402,6 +439,8 @@ static struct snd_pcm_hardware snd_rme96_playback_adat_info =
 {
 	.info =		     (SNDRV_PCM_INFO_MMAP_IOMEM |
 			      SNDRV_PCM_INFO_MMAP_VALID |
+			      SNDRV_PCM_INFO_SYNC_START |
+			      SNDRV_PCM_INFO_RESUME |
 			      SNDRV_PCM_INFO_INTERLEAVED |
 			      SNDRV_PCM_INFO_PAUSE),
 	.formats =	     (SNDRV_PCM_FMTBIT_S16_LE |
@@ -427,6 +466,8 @@ static struct snd_pcm_hardware snd_rme96_capture_adat_info =
 {
 	.info =		     (SNDRV_PCM_INFO_MMAP_IOMEM |
 			      SNDRV_PCM_INFO_MMAP_VALID |
+			      SNDRV_PCM_INFO_SYNC_START |
+			      SNDRV_PCM_INFO_RESUME |
 			      SNDRV_PCM_INFO_INTERLEAVED |
 			      SNDRV_PCM_INFO_PAUSE),
 	.formats =	     (SNDRV_PCM_FMTBIT_S16_LE |
@@ -1045,54 +1086,35 @@ snd_rme96_capture_hw_params(struct snd_pcm_substream *substream,
 }
 
 static void
-snd_rme96_playback_start(struct rme96 *rme96,
-			 int from_pause)
+snd_rme96_trigger(struct rme96 *rme96,
+		  int op)
 {
-	if (!from_pause) {
+	if (op & RME96_TB_RESET_PLAYPOS)
 		writel(0, rme96->iobase + RME96_IO_RESET_PLAY_POS);
-	}
-
-	rme96->wcreg |= RME96_WCR_START;
-	writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
-}
-
-static void
-snd_rme96_capture_start(struct rme96 *rme96,
-			int from_pause)
-{
-	if (!from_pause) {
+	if (op & RME96_TB_RESET_CAPTUREPOS)
 		writel(0, rme96->iobase + RME96_IO_RESET_REC_POS);
-	}
-
-	rme96->wcreg |= RME96_WCR_START_2;
+	if (op & RME96_TB_CLEAR_PLAYBACK_IRQ) {
+		rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
+		if (rme96->rcreg & RME96_RCR_IRQ)
+			writel(0, rme96->iobase + RME96_IO_CONFIRM_PLAY_IRQ);
+	}
+	if (op & RME96_TB_CLEAR_CAPTURE_IRQ) {
+		rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
+		if (rme96->rcreg & RME96_RCR_IRQ_2)
+			writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ);
+	}
+	if (op & RME96_TB_START_PLAYBACK)
+		rme96->wcreg |= RME96_WCR_START;
+	if (op & RME96_TB_STOP_PLAYBACK)
+		rme96->wcreg &= ~RME96_WCR_START;
+	if (op & RME96_TB_START_CAPTURE)
+		rme96->wcreg |= RME96_WCR_START_2;
+	if (op & RME96_TB_STOP_CAPTURE)
+		rme96->wcreg &= ~RME96_WCR_START_2;
 	writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
 }
 
-static void
-snd_rme96_playback_stop(struct rme96 *rme96)
-{
-	/*
-	 * Check if there is an unconfirmed IRQ, if so confirm it, or else
-	 * the hardware will not stop generating interrupts
-	 */
-	rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
-	if (rme96->rcreg & RME96_RCR_IRQ) {
-		writel(0, rme96->iobase + RME96_IO_CONFIRM_PLAY_IRQ);
-	}	
-	rme96->wcreg &= ~RME96_WCR_START;
-	writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
-}
 
-static void
-snd_rme96_capture_stop(struct rme96 *rme96)
-{
-	rme96->rcreg = readl(rme96->iobase + RME96_IO_CONTROL_REGISTER);
-	if (rme96->rcreg & RME96_RCR_IRQ_2) {
-		writel(0, rme96->iobase + RME96_IO_CONFIRM_REC_IRQ);
-	}	
-	rme96->wcreg &= ~RME96_WCR_START_2;
-	writel(rme96->wcreg, rme96->iobase + RME96_IO_CONTROL_REGISTER);
-}
 
 static irqreturn_t
 snd_rme96_interrupt(int irq,
@@ -1155,6 +1177,7 @@ snd_rme96_playback_spdif_open(struct snd_pcm_substream *substream)
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
+	snd_pcm_set_sync(substream);
 	spin_lock_irq(&rme96->lock);	
         if (rme96->playback_substream != NULL) {
 		spin_unlock_irq(&rme96->lock);
@@ -1191,6 +1214,7 @@ snd_rme96_capture_spdif_open(struct snd_pcm_substream *substream)
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
+	snd_pcm_set_sync(substream);
 	runtime->hw = snd_rme96_capture_spdif_info;
         if (snd_rme96_getinputtype(rme96) != RME96_INPUT_ANALOG &&
             (rate = snd_rme96_capture_getrate(rme96, &isadat)) > 0)
@@ -1222,6 +1246,7 @@ snd_rme96_playback_adat_open(struct snd_pcm_substream *substream)
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;        
 	
+	snd_pcm_set_sync(substream);
 	spin_lock_irq(&rme96->lock);	
         if (rme96->playback_substream != NULL) {
 		spin_unlock_irq(&rme96->lock);
@@ -1253,6 +1278,7 @@ snd_rme96_capture_adat_open(struct snd_pcm_substream *substream)
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
+	snd_pcm_set_sync(substream);
 	runtime->hw = snd_rme96_capture_adat_info;
         if (snd_rme96_getinputtype(rme96) == RME96_INPUT_ANALOG) {
                 /* makes no sense to use analog input. Note that analog
@@ -1288,7 +1314,7 @@ snd_rme96_playback_close(struct snd_pcm_substream *substream)
 
 	spin_lock_irq(&rme96->lock);	
 	if (RME96_ISPLAYING(rme96)) {
-		snd_rme96_playback_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_PLAYBACK);
 	}
 	rme96->playback_substream = NULL;
 	rme96->playback_periodsize = 0;
@@ -1309,7 +1335,7 @@ snd_rme96_capture_close(struct snd_pcm_substream *substream)
 	
 	spin_lock_irq(&rme96->lock);	
 	if (RME96_ISRECORDING(rme96)) {
-		snd_rme96_capture_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_CAPTURE);
 	}
 	rme96->capture_substream = NULL;
 	rme96->capture_periodsize = 0;
@@ -1324,7 +1350,7 @@ snd_rme96_playback_prepare(struct snd_pcm_substream *substream)
 	
 	spin_lock_irq(&rme96->lock);	
 	if (RME96_ISPLAYING(rme96)) {
-		snd_rme96_playback_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_PLAYBACK);
 	}
 	writel(0, rme96->iobase + RME96_IO_RESET_PLAY_POS);
 	spin_unlock_irq(&rme96->lock);
@@ -1338,7 +1364,7 @@ snd_rme96_capture_prepare(struct snd_pcm_substream *substream)
 	
 	spin_lock_irq(&rme96->lock);	
 	if (RME96_ISRECORDING(rme96)) {
-		snd_rme96_capture_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_CAPTURE);
 	}
 	writel(0, rme96->iobase + RME96_IO_RESET_REC_POS);
 	spin_unlock_irq(&rme96->lock);
@@ -1350,41 +1376,55 @@ snd_rme96_playback_trigger(struct snd_pcm_substream *substream,
 			   int cmd)
 {
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+	struct snd_pcm_substream *s;
+	bool sync;
+
+	snd_pcm_group_for_each_entry(s, substream) {
+		if (snd_pcm_substream_chip(s) == rme96)
+			snd_pcm_trigger_done(s, substream);
+	}
+
+	sync = (rme96->playback_substream && rme96->capture_substream) &&
+	       (rme96->playback_substream->group ==
+		rme96->capture_substream->group);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		if (!RME96_ISPLAYING(rme96)) {
-			if (substream != rme96->playback_substream) {
+			if (substream != rme96->playback_substream)
 				return -EBUSY;
-			}
-			snd_rme96_playback_start(rme96, 0);
+			snd_rme96_trigger(rme96, sync ? RME96_START_BOTH
+						 : RME96_START_PLAYBACK);
 		}
 		break;
 
+	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 		if (RME96_ISPLAYING(rme96)) {
-			if (substream != rme96->playback_substream) {
+			if (substream != rme96->playback_substream)
 				return -EBUSY;
-			}
-			snd_rme96_playback_stop(rme96);
+			snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+						 :  RME96_STOP_PLAYBACK);
 		}
 		break;
 
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		if (RME96_ISPLAYING(rme96)) {
-			snd_rme96_playback_stop(rme96);
-		}
+		if (RME96_ISPLAYING(rme96))
+			snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+						 : RME96_STOP_PLAYBACK);
 		break;
 
+	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		if (!RME96_ISPLAYING(rme96)) {
-			snd_rme96_playback_start(rme96, 1);
-		}
+		if (!RME96_ISPLAYING(rme96))
+			snd_rme96_trigger(rme96, sync ? RME96_RESUME_BOTH
+						 : RME96_RESUME_PLAYBACK);
 		break;
-		
+
 	default:
 		return -EINVAL;
 	}
+
 	return 0;
 }
 
@@ -1393,38 +1433,51 @@ snd_rme96_capture_trigger(struct snd_pcm_substream *substream,
 			  int cmd)
 {
 	struct rme96 *rme96 = snd_pcm_substream_chip(substream);
+	struct snd_pcm_substream *s;
+	bool sync;
+
+	snd_pcm_group_for_each_entry(s, substream) {
+		if (snd_pcm_substream_chip(s) == rme96)
+			snd_pcm_trigger_done(s, substream);
+	}
+
+	sync = (rme96->playback_substream && rme96->capture_substream) &&
+	       (rme96->playback_substream->group ==
+		rme96->capture_substream->group);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 		if (!RME96_ISRECORDING(rme96)) {
-			if (substream != rme96->capture_substream) {
+			if (substream != rme96->capture_substream)
 				return -EBUSY;
-			}
-			snd_rme96_capture_start(rme96, 0);
+			snd_rme96_trigger(rme96, sync ? RME96_START_BOTH
+						 : RME96_START_CAPTURE);
 		}
 		break;
 
+	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_STOP:
 		if (RME96_ISRECORDING(rme96)) {
-			if (substream != rme96->capture_substream) {
+			if (substream != rme96->capture_substream)
 				return -EBUSY;
-			}
-			snd_rme96_capture_stop(rme96);
+			snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+						 : RME96_STOP_CAPTURE);
 		}
 		break;
 
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		if (RME96_ISRECORDING(rme96)) {
-			snd_rme96_capture_stop(rme96);
-		}
+		if (RME96_ISRECORDING(rme96))
+			snd_rme96_trigger(rme96, sync ? RME96_STOP_BOTH
+						 : RME96_STOP_CAPTURE);
 		break;
 
+	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		if (!RME96_ISRECORDING(rme96)) {
-			snd_rme96_capture_start(rme96, 1);
-		}
+		if (!RME96_ISRECORDING(rme96))
+			snd_rme96_trigger(rme96, sync ? RME96_RESUME_BOTH
+						 : RME96_RESUME_CAPTURE);
 		break;
-		
+
 	default:
 		return -EINVAL;
 	}
@@ -1505,8 +1558,7 @@ snd_rme96_free(void *private_data)
 	        return;
 	}
 	if (rme96->irq >= 0) {
-		snd_rme96_playback_stop(rme96);
-		snd_rme96_capture_stop(rme96);
+		snd_rme96_trigger(rme96, RME96_STOP_BOTH);
 		rme96->areg &= ~RME96_AR_DAC_EN;
 		writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
 		free_irq(rme96->irq, (void *)rme96);
@@ -1520,6 +1572,10 @@ snd_rme96_free(void *private_data)
 		pci_release_regions(rme96->pci);
 		rme96->port = 0;
 	}
+#ifdef CONFIG_PM
+	vfree(rme96->playback_suspend_buffer);
+	vfree(rme96->capture_suspend_buffer);
+#endif
 	pci_disable_device(rme96->pci);
 }
 
@@ -1606,8 +1662,7 @@ snd_rme96_create(struct rme96 *rme96)
 	rme96->capture_periodsize = 0;
 	
 	/* make sure playback/capture is stopped, if by some reason active */
-	snd_rme96_playback_stop(rme96);
-	snd_rme96_capture_stop(rme96);
+	snd_rme96_trigger(rme96, RME96_STOP_BOTH);
 	
 	/* set default values in registers */
 	rme96->wcreg =
@@ -2319,6 +2374,87 @@ snd_rme96_create_switches(struct snd_card *card,
  * Card initialisation
  */
 
+#ifdef CONFIG_PM
+
+static int
+snd_rme96_suspend(struct pci_dev *pci,
+		  pm_message_t state)
+{
+	struct snd_card *card = pci_get_drvdata(pci);
+	struct rme96 *rme96 = card->private_data;
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+	snd_pcm_suspend(rme96->playback_substream);
+	snd_pcm_suspend(rme96->capture_substream);
+
+	/* save capture & playback pointers */
+	rme96->playback_pointer = readl(rme96->iobase + RME96_IO_GET_PLAY_POS)
+				  & RME96_RCR_AUDIO_ADDR_MASK;
+	rme96->capture_pointer = readl(rme96->iobase + RME96_IO_GET_REC_POS)
+				 & RME96_RCR_AUDIO_ADDR_MASK;
+
+	/* save playback and capture buffers */
+	memcpy_fromio(rme96->playback_suspend_buffer,
+		      rme96->iobase + RME96_IO_PLAY_BUFFER, RME96_BUFFER_SIZE);
+	memcpy_fromio(rme96->capture_suspend_buffer,
+		      rme96->iobase + RME96_IO_REC_BUFFER, RME96_BUFFER_SIZE);
+
+	/* disable the DAC  */
+	rme96->areg &= ~RME96_AR_DAC_EN;
+	writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+
+	pci_disable_device(pci);
+	pci_save_state(pci);
+
+	return 0;
+}
+
+static int
+snd_rme96_resume(struct pci_dev *pci)
+{
+	struct snd_card *card = pci_get_drvdata(pci);
+	struct rme96 *rme96 = card->private_data;
+
+	pci_restore_state(pci);
+	if (pci_enable_device(pci) < 0) {
+		printk(KERN_ERR "rme96: pci_enable_device failed, disabling device\n");
+		snd_card_disconnect(card);
+		return -EIO;
+	}
+
+	/* reset playback and record buffer pointers */
+	writel(0, rme96->iobase + RME96_IO_SET_PLAY_POS
+		  + rme96->playback_pointer);
+	writel(0, rme96->iobase + RME96_IO_SET_REC_POS
+		  + rme96->capture_pointer);
+
+	/* restore playback and capture buffers */
+	memcpy_toio(rme96->iobase + RME96_IO_PLAY_BUFFER,
+		    rme96->playback_suspend_buffer, RME96_BUFFER_SIZE);
+	memcpy_toio(rme96->iobase + RME96_IO_REC_BUFFER,
+		    rme96->capture_suspend_buffer, RME96_BUFFER_SIZE);
+
+	/* reset the ADC */
+	writel(rme96->areg | RME96_AR_PD2,
+	       rme96->iobase + RME96_IO_ADDITIONAL_REG);
+	writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+
+	/* reset and enable DAC, restore analog volume */
+	snd_rme96_reset_dac(rme96);
+	rme96->areg |= RME96_AR_DAC_EN;
+	writel(rme96->areg, rme96->iobase + RME96_IO_ADDITIONAL_REG);
+	if (RME96_HAS_ANALOG_OUT(rme96)) {
+		usleep_range(3000, 10000);
+		snd_rme96_apply_dac_volume(rme96);
+	}
+
+	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
+
+	return 0;
+}
+
+#endif
+
 static void snd_rme96_card_free(struct snd_card *card)
 {
 	snd_rme96_free(card->private_data);
@@ -2355,6 +2491,23 @@ snd_rme96_probe(struct pci_dev *pci,
 		return err;
 	}
 	
+#ifdef CONFIG_PM
+	rme96->playback_suspend_buffer = vmalloc(RME96_BUFFER_SIZE);
+	if (!rme96->playback_suspend_buffer) {
+		snd_printk(KERN_ERR
+			   "Failed to allocate playback suspend buffer!\n");
+		snd_card_free(card);
+		return -ENOMEM;
+	}
+	rme96->capture_suspend_buffer = vmalloc(RME96_BUFFER_SIZE);
+	if (!rme96->capture_suspend_buffer) {
+		snd_printk(KERN_ERR
+			   "Failed to allocate capture suspend buffer!\n");
+		snd_card_free(card);
+		return -ENOMEM;
+	}
+#endif
+
 	strcpy(card->driver, "Digi96");
 	switch (rme96->pci->device) {
 	case PCI_DEVICE_ID_RME_DIGI96:
@@ -2397,6 +2550,10 @@ static struct pci_driver rme96_driver = {
 	.id_table = snd_rme96_ids,
 	.probe = snd_rme96_probe,
 	.remove = snd_rme96_remove,
+#ifdef CONFIG_PM
+	.suspend = snd_rme96_suspend,
+	.resume = snd_rme96_resume,
+#endif
 };
 
 module_pci_driver(rme96_driver);
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index bd501931ee23..3cde55b753e2 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -38,6 +38,97 @@
  *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
  *
  */
+
+/* *************    Register Documentation   *******************************************************
+ *
+ * Work in progress! Documentation is based on the code in this file.
+ *
+ * --------- HDSPM_controlRegister ---------
+ * :7654.3210:7654.3210:7654.3210:7654.3210: bit number per byte
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number
+ * :1098.7654:3210.9876:5432.1098:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit
+ * :    .    :    .    :    .    :  x .    :  HDSPM_AudioInterruptEnable \_ setting both bits
+ * :    .    :    .    :    .    :    .   x:  HDSPM_Start                /  enables audio IO
+ * :    .    :    .    :    .    :   x.    :  HDSPM_ClockModeMaster - 1: Master, 0: Slave
+ * :    .    :    .    :    .    :    .210 :  HDSPM_LatencyMask - 3 Bit value for latency
+ * :    .    :    .    :    .    :    .    :      0:64, 1:128, 2:256, 3:512,
+ * :    .    :    .    :    .    :    .    :      4:1024, 5:2048, 6:4096, 7:8192
+ * :x   .    :    .    :    .   x:xx  .    :  HDSPM_FrequencyMask
+ * :    .    :    .    :    .    :10  .    :  HDSPM_Frequency1|HDSPM_Frequency0: 1=32K,2=44.1K,3=48K,0=??
+ * :    .    :    .    :    .   x:    .    :  <MADI> HDSPM_DoubleSpeed
+ * :x   .    :    .    :    .    :    .    :  <MADI> HDSPM_QuadSpeed
+ * :    .  3 :    .  10:  2 .    :    .    :  HDSPM_SyncRefMask :
+ * :    .    :    .   x:    .    :    .    :  HDSPM_SyncRef0
+ * :    .    :    .  x :    .    :    .    :  HDSPM_SyncRef1
+ * :    .    :    .    :  x .    :    .    :  <AES32> HDSPM_SyncRef2
+ * :    .  x :    .    :    .    :    .    :  <AES32> HDSPM_SyncRef3
+ * :    .    :    .  10:    .    :    .    :  <MADI> sync ref: 0:WC, 1:Madi, 2:TCO, 3:SyncIn
+ * :    .  3 :    .  10:  2 .    :    .    :  <AES32>  0:WC, 1:AES1 ... 8:AES8, 9: TCO, 10:SyncIn?
+ * :    .  x :    .    :    .    :    .    :  <MADIe> HDSPe_FLOAT_FORMAT
+ * :    .    :    .    : x  .    :    .    :  <MADI> HDSPM_InputSelect0 : 0=optical,1=coax
+ * :    .    :    .    :x   .    :    .    :  <MADI> HDSPM_InputSelect1
+ * :    .    :    .x   :    .    :    .    :  <MADI> HDSPM_clr_tms
+ * :    .    :    .    :    . x  :    .    :  <MADI> HDSPM_TX_64ch
+ * :    .    :    .    :    . x  :    .    :  <AES32> HDSPM_Emphasis
+ * :    .    :    .    :    .x   :    .    :  <MADI> HDSPM_AutoInp
+ * :    .    :    . x  :    .    :    .    :  <MADI> HDSPM_SMUX
+ * :    .    :    .x   :    .    :    .    :  <MADI> HDSPM_clr_tms
+ * :    .    :   x.    :    .    :    .    :  <MADI> HDSPM_taxi_reset
+ * :    .   x:    .    :    .    :    .    :  <MADI> HDSPM_LineOut
+ * :    .   x:    .    :    .    :    .    :  <AES32> ??????????????????
+ * :    .    :   x.    :    .    :    .    :  <AES32> HDSPM_WCK48
+ * :    .    :    .    :    .x   :    .    :  <AES32> HDSPM_Dolby
+ * :    .    : x  .    :    .    :    .    :  HDSPM_Midi0InterruptEnable
+ * :    .    :x   .    :    .    :    .    :  HDSPM_Midi1InterruptEnable
+ * :    .    :  x .    :    .    :    .    :  HDSPM_Midi2InterruptEnable
+ * :    . x  :    .    :    .    :    .    :  <MADI> HDSPM_Midi3InterruptEnable
+ * :    . x  :    .    :    .    :    .    :  <AES32> HDSPM_DS_DoubleWire
+ * :    .x   :    .    :    .    :    .    :  <AES32> HDSPM_QS_DoubleWire
+ * :   x.    :    .    :    .    :    .    :  <AES32> HDSPM_QS_QuadWire
+ * :    .    :    .    :    .  x :    .    :  <AES32> HDSPM_Professional
+ * : x  .    :    .    :    .    :    .    :  HDSPM_wclk_sel
+ * :    .    :    .    :    .    :    .    :
+ * :7654.3210:7654.3210:7654.3210:7654.3210: bit number per byte
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number
+ * :1098.7654:3210.9876:5432.1098:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421:hex digit
+ *
+ *
+ *
+ * AIO / RayDAT only
+ *
+ * ------------ HDSPM_WR_SETTINGS ----------
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number per byte
+ * :1098.7654:3210.9876:5432.1098:7654.3210:
+ * :||||.||||:||||.||||:||||.||||:||||.||||: bit number
+ * :7654.3210:7654.3210:7654.3210:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit
+ * :    .    :    .    :    .    :    .   x: HDSPM_c0Master 1: Master, 0: Slave
+ * :    .    :    .    :    .    :    .  x : HDSPM_c0_SyncRef0
+ * :    .    :    .    :    .    :    . x  : HDSPM_c0_SyncRef1
+ * :    .    :    .    :    .    :    .x   : HDSPM_c0_SyncRef2
+ * :    .    :    .    :    .    :   x.    : HDSPM_c0_SyncRef3
+ * :    .    :    .    :    .    :   3.210 : HDSPM_c0_SyncRefMask:
+ * :    .    :    .    :    .    :    .    :  RayDat: 0:WC, 1:AES, 2:SPDIF, 3..6: ADAT1..4,
+ * :    .    :    .    :    .    :    .    :          9:TCO, 10:SyncIn
+ * :    .    :    .    :    .    :    .    :  AIO: 0:WC, 1:AES, 2: SPDIF, 3: ATAT,
+ * :    .    :    .    :    .    :    .    :          9:TCO, 10:SyncIn
+ * :    .    :    .    :    .    :    .    :
+ * :    .    :    .    :    .    :    .    :
+ * :3322.2222:2222.1111:1111.1100:0000.0000: bit number per byte
+ * :1098.7654:3210.9876:5432.1098:7654.3210:
+ * :||||.||||:||||.||||:||||.||||:||||.||||: bit number
+ * :7654.3210:7654.3210:7654.3210:7654.3210: 0..31
+ * :||||.||||:||||.||||:||||.||||:||||.||||:
+ * :8421.8421:8421.8421:8421.8421:8421.8421: hex digit
+ *
+ */
 #include <linux/init.h>
 #include <linux/delay.h>
 #include <linux/interrupt.h>
@@ -95,7 +186,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 #define HDSPM_controlRegister	     64
 #define HDSPM_interruptConfirmation  96
 #define HDSPM_control2Reg	     256  /* not in specs ???????? */
-#define HDSPM_freqReg                256  /* for AES32 */
+#define HDSPM_freqReg                256  /* for setting arbitrary clock values (DDS feature) */
 #define HDSPM_midiDataOut0	     352  /* just believe in old code */
 #define HDSPM_midiDataOut1	     356
 #define HDSPM_eeprom_wr		     384  /* for AES32 */
@@ -258,6 +349,25 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 
 #define HDSPM_wclk_sel (1<<30)
 
+/* additional control register bits for AIO*/
+#define HDSPM_c0_Wck48				0x20 /* also RayDAT */
+#define HDSPM_c0_Input0				0x1000
+#define HDSPM_c0_Input1				0x2000
+#define HDSPM_c0_Spdif_Opt			0x4000
+#define HDSPM_c0_Pro				0x8000
+#define HDSPM_c0_clr_tms			0x10000
+#define HDSPM_c0_AEB1				0x20000
+#define HDSPM_c0_AEB2				0x40000
+#define HDSPM_c0_LineOut			0x80000
+#define HDSPM_c0_AD_GAIN0			0x100000
+#define HDSPM_c0_AD_GAIN1			0x200000
+#define HDSPM_c0_DA_GAIN0			0x400000
+#define HDSPM_c0_DA_GAIN1			0x800000
+#define HDSPM_c0_PH_GAIN0			0x1000000
+#define HDSPM_c0_PH_GAIN1			0x2000000
+#define HDSPM_c0_Sym6db				0x4000000
+
+
 /* --- bit helper defines */
 #define HDSPM_LatencyMask    (HDSPM_Latency0|HDSPM_Latency1|HDSPM_Latency2)
 #define HDSPM_FrequencyMask  (HDSPM_Frequency0|HDSPM_Frequency1|\
@@ -341,11 +451,11 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 #define HDSPM_madiLock           (1<<3)	/* MADI Locked =1, no=0 */
 #define HDSPM_madiSync          (1<<18) /* MADI is in sync */
 
-#define HDSPM_tcoLock    0x00000020 /* Optional TCO locked status FOR HDSPe MADI! */
-#define HDSPM_tcoSync    0x10000000 /* Optional TCO sync status */
+#define HDSPM_tcoLockMadi    0x00000020 /* Optional TCO locked status for HDSPe MADI*/
+#define HDSPM_tcoSync    0x10000000 /* Optional TCO sync status for HDSPe MADI and AES32!*/
 
-#define HDSPM_syncInLock 0x00010000 /* Sync In lock status FOR HDSPe MADI! */
-#define HDSPM_syncInSync 0x00020000 /* Sync In sync status FOR HDSPe MADI! */
+#define HDSPM_syncInLock 0x00010000 /* Sync In lock status for HDSPe MADI! */
+#define HDSPM_syncInSync 0x00020000 /* Sync In sync status for HDSPe MADI! */
 
 #define HDSPM_BufferPositionMask 0x000FFC0 /* Bit 6..15 : h/w buffer pointer */
 			/* since 64byte accurate, last 6 bits are not used */
@@ -363,7 +473,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 					 * Interrupt
 					 */
 #define HDSPM_tco_detect         0x08000000
-#define HDSPM_tco_lock	         0x20000000
+#define HDSPM_tcoLockAes         0x20000000 /* Optional TCO locked status for HDSPe AES */
 
 #define HDSPM_s2_tco_detect      0x00000040
 #define HDSPM_s2_AEBO_D          0x00000080
@@ -461,7 +571,9 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 #define HDSPM_AES32_AUTOSYNC_FROM_AES6 6
 #define HDSPM_AES32_AUTOSYNC_FROM_AES7 7
 #define HDSPM_AES32_AUTOSYNC_FROM_AES8 8
-#define HDSPM_AES32_AUTOSYNC_FROM_NONE 9
+#define HDSPM_AES32_AUTOSYNC_FROM_TCO 9
+#define HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN 10
+#define HDSPM_AES32_AUTOSYNC_FROM_NONE 11
 
 /*  status2 */
 /* HDSPM_LockAES_bit is given by HDSPM_LockAES >> (AES# - 1) */
@@ -537,36 +649,39 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}");
 /* names for speed modes */
 static char *hdspm_speed_names[] = { "single", "double", "quad" };
 
-static char *texts_autosync_aes_tco[] = { "Word Clock",
+static const char *const texts_autosync_aes_tco[] = { "Word Clock",
 					  "AES1", "AES2", "AES3", "AES4",
 					  "AES5", "AES6", "AES7", "AES8",
-					  "TCO" };
-static char *texts_autosync_aes[] = { "Word Clock",
+					  "TCO", "Sync In"
+};
+static const char *const texts_autosync_aes[] = { "Word Clock",
 				      "AES1", "AES2", "AES3", "AES4",
-				      "AES5", "AES6", "AES7", "AES8" };
-static char *texts_autosync_madi_tco[] = { "Word Clock",
+				      "AES5", "AES6", "AES7", "AES8",
+				      "Sync In"
+};
+static const char *const texts_autosync_madi_tco[] = { "Word Clock",
 					   "MADI", "TCO", "Sync In" };
-static char *texts_autosync_madi[] = { "Word Clock",
+static const char *const texts_autosync_madi[] = { "Word Clock",
 				       "MADI", "Sync In" };
 
-static char *texts_autosync_raydat_tco[] = {
+static const char *const texts_autosync_raydat_tco[] = {
 	"Word Clock",
 	"ADAT 1", "ADAT 2", "ADAT 3", "ADAT 4",
 	"AES", "SPDIF", "TCO", "Sync In"
 };
-static char *texts_autosync_raydat[] = {
+static const char *const texts_autosync_raydat[] = {
 	"Word Clock",
 	"ADAT 1", "ADAT 2", "ADAT 3", "ADAT 4",
 	"AES", "SPDIF", "Sync In"
 };
-static char *texts_autosync_aio_tco[] = {
+static const char *const texts_autosync_aio_tco[] = {
 	"Word Clock",
 	"ADAT", "AES", "SPDIF", "TCO", "Sync In"
 };
-static char *texts_autosync_aio[] = { "Word Clock",
+static const char *const texts_autosync_aio[] = { "Word Clock",
 				      "ADAT", "AES", "SPDIF", "Sync In" };
 
-static char *texts_freq[] = {
+static const char *const texts_freq[] = {
 	"No Lock",
 	"32 kHz",
 	"44.1 kHz",
@@ -629,7 +744,8 @@ static char *texts_ports_aio_in_ss[] = {
 	"AES.L", "AES.R",
 	"SPDIF.L", "SPDIF.R",
 	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6",
-	"ADAT.7", "ADAT.8"
+	"ADAT.7", "ADAT.8",
+	"AEB.1", "AEB.2", "AEB.3", "AEB.4"
 };
 
 static char *texts_ports_aio_out_ss[] = {
@@ -638,14 +754,16 @@ static char *texts_ports_aio_out_ss[] = {
 	"SPDIF.L", "SPDIF.R",
 	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4", "ADAT.5", "ADAT.6",
 	"ADAT.7", "ADAT.8",
-	"Phone.L", "Phone.R"
+	"Phone.L", "Phone.R",
+	"AEB.1", "AEB.2", "AEB.3", "AEB.4"
 };
 
 static char *texts_ports_aio_in_ds[] = {
 	"Analogue.L", "Analogue.R",
 	"AES.L", "AES.R",
 	"SPDIF.L", "SPDIF.R",
-	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4"
+	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
+	"AEB.1", "AEB.2", "AEB.3", "AEB.4"
 };
 
 static char *texts_ports_aio_out_ds[] = {
@@ -653,14 +771,16 @@ static char *texts_ports_aio_out_ds[] = {
 	"AES.L", "AES.R",
 	"SPDIF.L", "SPDIF.R",
 	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
-	"Phone.L", "Phone.R"
+	"Phone.L", "Phone.R",
+	"AEB.1", "AEB.2", "AEB.3", "AEB.4"
 };
 
 static char *texts_ports_aio_in_qs[] = {
 	"Analogue.L", "Analogue.R",
 	"AES.L", "AES.R",
 	"SPDIF.L", "SPDIF.R",
-	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4"
+	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
+	"AEB.1", "AEB.2", "AEB.3", "AEB.4"
 };
 
 static char *texts_ports_aio_out_qs[] = {
@@ -668,7 +788,8 @@ static char *texts_ports_aio_out_qs[] = {
 	"AES.L", "AES.R",
 	"SPDIF.L", "SPDIF.R",
 	"ADAT.1", "ADAT.2", "ADAT.3", "ADAT.4",
-	"Phone.L", "Phone.R"
+	"Phone.L", "Phone.R",
+	"AEB.1", "AEB.2", "AEB.3", "AEB.4"
 };
 
 static char *texts_ports_aes32[] = {
@@ -745,8 +866,8 @@ static char channel_map_aio_in_ss[HDSPM_MAX_CHANNELS] = {
 	8, 9,			/* aes in, */
 	10, 11,			/* spdif in */
 	12, 13, 14, 15, 16, 17, 18, 19,	/* ADAT in */
-	-1, -1,
-	-1, -1, -1, -1, -1, -1, -1, -1,
+	2, 3, 4, 5,		/* AEB */
+	-1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
@@ -760,7 +881,8 @@ static char channel_map_aio_out_ss[HDSPM_MAX_CHANNELS] = {
 	10, 11,			/* spdif out */
 	12, 13, 14, 15, 16, 17, 18, 19,	/* ADAT out */
 	6, 7,			/* phone out */
-	-1, -1, -1, -1, -1, -1, -1, -1,
+	2, 3, 4, 5,		/* AEB */
+	-1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
@@ -773,7 +895,8 @@ static char channel_map_aio_in_ds[HDSPM_MAX_CHANNELS] = {
 	8, 9,			/* aes in */
 	10, 11,			/* spdif in */
 	12, 14, 16, 18,		/* adat in */
-	-1, -1, -1, -1, -1, -1,
+	2, 3, 4, 5,		/* AEB */
+	-1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
@@ -788,7 +911,7 @@ static char channel_map_aio_out_ds[HDSPM_MAX_CHANNELS] = {
 	10, 11,			/* spdif out */
 	12, 14, 16, 18,		/* adat out */
 	6, 7,			/* phone out */
-	-1, -1, -1, -1,
+	2, 3, 4, 5,		/* AEB */
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
@@ -802,7 +925,8 @@ static char channel_map_aio_in_qs[HDSPM_MAX_CHANNELS] = {
 	8, 9,			/* aes in */
 	10, 11,			/* spdif in */
 	12, 16,			/* adat in */
-	-1, -1, -1, -1, -1, -1, -1, -1,
+	2, 3, 4, 5,		/* AEB */
+	-1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
@@ -817,7 +941,8 @@ static char channel_map_aio_out_qs[HDSPM_MAX_CHANNELS] = {
 	10, 11,			/* spdif out */
 	12, 16,			/* adat out */
 	6, 7,			/* phone out */
-	-1, -1, -1, -1, -1, -1,
+	2, 3, 4, 5,		/* AEB */
+	-1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1, -1, -1, -1, -1, -1, -1,
@@ -856,11 +981,11 @@ struct hdspm_midi {
 };
 
 struct hdspm_tco {
-	int input;
-	int framerate;
-	int wordclock;
-	int samplerate;
-	int pull;
+	int input; /* 0: LTC, 1:Video, 2: WC*/
+	int framerate; /* 0=24, 1=25, 2=29.97, 3=29.97d, 4=30, 5=30d */
+	int wordclock; /* 0=1:1, 1=44.1->48, 2=48->44.1 */
+	int samplerate; /* 0=44.1, 1=48, 2= freq from app */
+	int pull; /*   0=0, 1=+0.1%, 2=-0.1%, 3=+4%, 4=-4%*/
 	int term; /* 0 = off, 1 = on */
 };
 
@@ -879,7 +1004,7 @@ struct hdspm {
 
 	u32 control_register;	/* cached value */
 	u32 control2_register;	/* cached value */
-	u32 settings_register;
+	u32 settings_register;  /* cached value for AIO / RayDat (sync reference, master/slave) */
 
 	struct hdspm_midi midi[4];
 	struct tasklet_struct midi_tasklet;
@@ -941,7 +1066,7 @@ struct hdspm {
 
 	struct hdspm_tco *tco;  /* NULL if no TCO detected */
 
-	char **texts_autosync;
+	const char *const *texts_autosync;
 	int texts_autosync_items;
 
 	cycles_t last_interrupt;
@@ -976,12 +1101,24 @@ static inline void snd_hdspm_initialize_midi_flush(struct hdspm *hdspm);
 static inline int hdspm_get_pll_freq(struct hdspm *hdspm);
 static int hdspm_update_simple_mixer_controls(struct hdspm *hdspm);
 static int hdspm_autosync_ref(struct hdspm *hdspm);
+static int hdspm_set_toggle_setting(struct hdspm *hdspm, u32 regmask, int out);
 static int snd_hdspm_set_defaults(struct hdspm *hdspm);
 static int hdspm_system_clock_mode(struct hdspm *hdspm);
 static void hdspm_set_sgbuf(struct hdspm *hdspm,
 			    struct snd_pcm_substream *substream,
 			     unsigned int reg, int channels);
 
+static int hdspm_aes_sync_check(struct hdspm *hdspm, int idx);
+static int hdspm_wc_sync_check(struct hdspm *hdspm);
+static int hdspm_tco_sync_check(struct hdspm *hdspm);
+static int hdspm_sync_in_sync_check(struct hdspm *hdspm);
+
+static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index);
+static int hdspm_get_tco_sample_rate(struct hdspm *hdspm);
+static int hdspm_get_wc_sample_rate(struct hdspm *hdspm);
+
+
+
 static inline int HDSPM_bit2freq(int n)
 {
 	static const int bit2freq_tab[] = {
@@ -992,6 +1129,12 @@ static inline int HDSPM_bit2freq(int n)
 	return bit2freq_tab[n];
 }
 
+static bool hdspm_is_raydat_or_aio(struct hdspm *hdspm)
+{
+	return ((AIO == hdspm->io_type) || (RayDAT == hdspm->io_type));
+}
+
+
 /* Write/read to/from HDSPM with Adresses in Bytes
    not words but only 32Bit writes are allowed */
 
@@ -1107,14 +1250,11 @@ static int hdspm_rate_multiplier(struct hdspm *hdspm, int rate)
 		else if (hdspm->control_register &
 				HDSPM_DoubleSpeed)
 			return rate * 2;
-	};
+	}
 	return rate;
 }
 
-static int hdspm_tco_sync_check(struct hdspm *hdspm);
-static int hdspm_sync_in_sync_check(struct hdspm *hdspm);
-
-/* check for external sample rate */
+/* check for external sample rate, returns the sample rate in Hz*/
 static int hdspm_external_sample_rate(struct hdspm *hdspm)
 {
 	unsigned int status, status2, timecode;
@@ -1127,17 +1267,36 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm)
 		timecode = hdspm_read(hdspm, HDSPM_timecodeRegister);
 
 		syncref = hdspm_autosync_ref(hdspm);
+		switch (syncref) {
+		case HDSPM_AES32_AUTOSYNC_FROM_WORD:
+		/* Check WC sync and get sample rate */
+			if (hdspm_wc_sync_check(hdspm))
+				return HDSPM_bit2freq(hdspm_get_wc_sample_rate(hdspm));
+			break;
 
-		if (syncref == HDSPM_AES32_AUTOSYNC_FROM_WORD &&
-				status & HDSPM_AES32_wcLock)
-			return HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF);
+		case HDSPM_AES32_AUTOSYNC_FROM_AES1:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES2:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES3:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES4:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES5:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES6:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES7:
+		case HDSPM_AES32_AUTOSYNC_FROM_AES8:
+		/* Check AES sync and get sample rate */
+			if (hdspm_aes_sync_check(hdspm, syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1))
+				return HDSPM_bit2freq(hdspm_get_aes_sample_rate(hdspm,
+							syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1));
+			break;
 
-		if (syncref >= HDSPM_AES32_AUTOSYNC_FROM_AES1 &&
-				syncref <= HDSPM_AES32_AUTOSYNC_FROM_AES8 &&
-				status2 & (HDSPM_LockAES >>
-				(syncref - HDSPM_AES32_AUTOSYNC_FROM_AES1)))
-			return HDSPM_bit2freq((timecode >> (4*(syncref-HDSPM_AES32_AUTOSYNC_FROM_AES1))) & 0xF);
-		return 0;
+
+		case HDSPM_AES32_AUTOSYNC_FROM_TCO:
+		/* Check TCO sync and get sample rate */
+			if (hdspm_tco_sync_check(hdspm))
+				return HDSPM_bit2freq(hdspm_get_tco_sample_rate(hdspm));
+			break;
+		default:
+			return 0;
+		} /* end switch(syncref) */
 		break;
 
 	case MADIface:
@@ -2129,6 +2288,9 @@ static int hdspm_get_wc_sample_rate(struct hdspm *hdspm)
 		status = hdspm_read(hdspm, HDSPM_RD_STATUS_1);
 		return (status >> 16) & 0xF;
 		break;
+	case AES32:
+		status = hdspm_read(hdspm, HDSPM_statusRegister);
+		return (status >> HDSPM_AES32_wcFreq_bit) & 0xF;
 	default:
 		break;
 	}
@@ -2152,6 +2314,9 @@ static int hdspm_get_tco_sample_rate(struct hdspm *hdspm)
 			status = hdspm_read(hdspm, HDSPM_RD_STATUS_1);
 			return (status >> 20) & 0xF;
 			break;
+		case AES32:
+			status = hdspm_read(hdspm, HDSPM_statusRegister);
+			return (status >> 1) & 0xF;
 		default:
 			break;
 		}
@@ -2183,6 +2348,23 @@ static int hdspm_get_sync_in_sample_rate(struct hdspm *hdspm)
 	return 0;
 }
 
+/**
+ * Returns the AES sample rate class for the given card.
+ **/
+static int hdspm_get_aes_sample_rate(struct hdspm *hdspm, int index)
+{
+	int timecode;
+
+	switch (hdspm->io_type) {
+	case AES32:
+		timecode = hdspm_read(hdspm, HDSPM_timecodeRegister);
+		return (timecode >> (4*index)) & 0xF;
+		break;
+	default:
+		break;
+	}
+	return 0;
+}
 
 /**
  * Returns the sample rate class for input source <idx> for
@@ -2196,15 +2378,23 @@ static int hdspm_get_s1_sample_rate(struct hdspm *hdspm, unsigned int idx)
 }
 
 #define ENUMERATED_CTL_INFO(info, texts) \
-{ \
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; \
-	uinfo->count = 1; \
-	uinfo->value.enumerated.items = ARRAY_SIZE(texts); \
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) \
-		uinfo->value.enumerated.item =	uinfo->value.enumerated.items - 1; \
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); \
-}
+	snd_ctl_enum_info(info, 1, ARRAY_SIZE(texts), texts)
+
 
+/* Helper function to query the external sample rate and return the
+ * corresponding enum to be returned to userspace.
+ */
+static int hdspm_external_rate_to_enum(struct hdspm *hdspm)
+{
+	int rate = hdspm_external_sample_rate(hdspm);
+	int i, selected_rate = 0;
+	for (i = 1; i < 10; i++)
+		if (HDSPM_bit2freq(i) == rate) {
+			selected_rate = i;
+			break;
+		}
+	return selected_rate;
+}
 
 
 #define HDSPM_AUTOSYNC_SAMPLE_RATE(xname, xindex) \
@@ -2270,7 +2460,7 @@ static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol,
 		default:
 			ucontrol->value.enumerated.item[0] =
 				hdspm_get_s1_sample_rate(hdspm,
-						ucontrol->id.index-1);
+						kcontrol->private_value-1);
 		}
 		break;
 
@@ -2289,28 +2479,24 @@ static int snd_hdspm_get_autosync_sample_rate(struct snd_kcontrol *kcontrol,
 			ucontrol->value.enumerated.item[0] =
 				hdspm_get_sync_in_sample_rate(hdspm);
 			break;
+		case 11: /* External Rate */
+			ucontrol->value.enumerated.item[0] =
+				hdspm_external_rate_to_enum(hdspm);
+			break;
 		default: /* AES1 to AES8 */
 			ucontrol->value.enumerated.item[0] =
-				hdspm_get_s1_sample_rate(hdspm,
-						kcontrol->private_value-1);
+				hdspm_get_aes_sample_rate(hdspm,
+						kcontrol->private_value -
+						HDSPM_AES32_AUTOSYNC_FROM_AES1);
 			break;
 		}
 		break;
 
 	case MADI:
 	case MADIface:
-		{
-			int rate = hdspm_external_sample_rate(hdspm);
-			int i, selected_rate = 0;
-			for (i = 1; i < 10; i++)
-				if (HDSPM_bit2freq(i) == rate) {
-					selected_rate = i;
-					break;
-				}
-			ucontrol->value.enumerated.item[0] = selected_rate;
-		}
+		ucontrol->value.enumerated.item[0] =
+			hdspm_external_rate_to_enum(hdspm);
 		break;
-
 	default:
 		break;
 	}
@@ -2359,33 +2545,17 @@ static int hdspm_system_clock_mode(struct hdspm *hdspm)
  **/
 static void hdspm_set_system_clock_mode(struct hdspm *hdspm, int mode)
 {
-	switch (hdspm->io_type) {
-	case AIO:
-	case RayDAT:
-		if (0 == mode)
-			hdspm->settings_register |= HDSPM_c0Master;
-		else
-			hdspm->settings_register &= ~HDSPM_c0Master;
-
-		hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register);
-		break;
-
-	default:
-		if (0 == mode)
-			hdspm->control_register |= HDSPM_ClockModeMaster;
-		else
-			hdspm->control_register &= ~HDSPM_ClockModeMaster;
-
-		hdspm_write(hdspm, HDSPM_controlRegister,
-				hdspm->control_register);
-	}
+	hdspm_set_toggle_setting(hdspm,
+			(hdspm_is_raydat_or_aio(hdspm)) ?
+			HDSPM_c0Master : HDSPM_ClockModeMaster,
+			(0 == mode));
 }
 
 
 static int snd_hdspm_info_system_clock_mode(struct snd_kcontrol *kcontrol,
 					    struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "Master", "AutoSync" };
+	static const char *const texts[] = { "Master", "AutoSync" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -2809,16 +2979,7 @@ static int snd_hdspm_info_pref_sync_ref(struct snd_kcontrol *kcontrol,
 {
 	struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
 
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = hdspm->texts_autosync_items;
-
-	if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items)
-		uinfo->value.enumerated.item =
-			uinfo->value.enumerated.items - 1;
-
-	strcpy(uinfo->value.enumerated.name,
-			hdspm->texts_autosync[uinfo->value.enumerated.item]);
+	snd_ctl_enum_info(uinfo, 1, hdspm->texts_autosync_items, hdspm->texts_autosync);
 
 	return 0;
 }
@@ -2873,19 +3034,20 @@ static int snd_hdspm_put_pref_sync_ref(struct snd_kcontrol *kcontrol,
 
 static int hdspm_autosync_ref(struct hdspm *hdspm)
 {
+	/* This looks at the autosync selected sync reference */
 	if (AES32 == hdspm->io_type) {
+
 		unsigned int status = hdspm_read(hdspm, HDSPM_statusRegister);
-		unsigned int syncref =
-			(status >> HDSPM_AES32_syncref_bit) & 0xF;
-		if (syncref == 0)
-			return HDSPM_AES32_AUTOSYNC_FROM_WORD;
-		if (syncref <= 8)
+		unsigned int syncref = (status >> HDSPM_AES32_syncref_bit) & 0xF;
+		if ((syncref >= HDSPM_AES32_AUTOSYNC_FROM_WORD) &&
+				(syncref <= HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN)) {
 			return syncref;
+		}
 		return HDSPM_AES32_AUTOSYNC_FROM_NONE;
+
 	} else if (MADI == hdspm->io_type) {
-		/* This looks at the autosync selected sync reference */
-		unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
 
+		unsigned int status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
 		switch (status2 & HDSPM_SelSyncRefMask) {
 		case HDSPM_SelSyncRef_WORD:
 			return HDSPM_AUTOSYNC_FROM_WORD;
@@ -2898,7 +3060,7 @@ static int hdspm_autosync_ref(struct hdspm *hdspm)
 		case HDSPM_SelSyncRef_NVALID:
 			return HDSPM_AUTOSYNC_FROM_NONE;
 		default:
-			return 0;
+			return HDSPM_AUTOSYNC_FROM_NONE;
 		}
 
 	}
@@ -2912,31 +3074,15 @@ static int snd_hdspm_info_autosync_ref(struct snd_kcontrol *kcontrol,
 	struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
 
 	if (AES32 == hdspm->io_type) {
-		static char *texts[] = { "WordClock", "AES1", "AES2", "AES3",
-			"AES4",	"AES5", "AES6", "AES7", "AES8", "None"};
-
-		uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-		uinfo->count = 1;
-		uinfo->value.enumerated.items = 10;
-		if (uinfo->value.enumerated.item >=
-		    uinfo->value.enumerated.items)
-			uinfo->value.enumerated.item =
-				uinfo->value.enumerated.items - 1;
-		strcpy(uinfo->value.enumerated.name,
-				texts[uinfo->value.enumerated.item]);
+		static const char *const texts[] = { "WordClock", "AES1", "AES2", "AES3",
+			"AES4",	"AES5", "AES6", "AES7", "AES8", "TCO", "Sync In", "None"};
+
+		ENUMERATED_CTL_INFO(uinfo, texts);
 	} else if (MADI == hdspm->io_type) {
-		static char *texts[] = {"Word Clock", "MADI", "TCO",
+		static const char *const texts[] = {"Word Clock", "MADI", "TCO",
 			"Sync In", "None" };
 
-		uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-		uinfo->count = 1;
-		uinfo->value.enumerated.items = 5;
-		if (uinfo->value.enumerated.item >=
-				uinfo->value.enumerated.items)
-			uinfo->value.enumerated.item =
-				uinfo->value.enumerated.items - 1;
-		strcpy(uinfo->value.enumerated.name,
-				texts[uinfo->value.enumerated.item]);
+		ENUMERATED_CTL_INFO(uinfo, texts);
 	}
 	return 0;
 }
@@ -2964,7 +3110,7 @@ static int snd_hdspm_get_autosync_ref(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_tco_video_input_format(struct snd_kcontrol *kcontrol,
 				       struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"No video", "NTSC", "PAL"};
+	static const char *const texts[] = {"No video", "NTSC", "PAL"};
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3010,7 +3156,7 @@ static int snd_hdspm_get_tco_video_input_format(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_tco_ltc_frames(struct snd_kcontrol *kcontrol,
 				       struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"No lock", "24 fps", "25 fps", "29.97 fps",
+	static const char *const texts[] = {"No lock", "24 fps", "25 fps", "29.97 fps",
 				"30 fps"};
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
@@ -3027,19 +3173,19 @@ static int hdspm_tco_ltc_frames(struct hdspm *hdspm)
 					HDSPM_TCO1_LTC_Format_MSB)) {
 		case 0:
 			/* 24 fps */
-			ret = 1;
+			ret = fps_24;
 			break;
 		case HDSPM_TCO1_LTC_Format_LSB:
 			/* 25 fps */
-			ret = 2;
+			ret = fps_25;
 			break;
 		case HDSPM_TCO1_LTC_Format_MSB:
-			/* 25 fps */
-			ret = 3;
+			/* 29.97 fps */
+			ret = fps_2997;
 			break;
 		default:
 			/* 30 fps */
-			ret = 4;
+			ret = fps_30;
 			break;
 		}
 	}
@@ -3067,16 +3213,35 @@ static int snd_hdspm_get_tco_ltc_frames(struct snd_kcontrol *kcontrol,
 
 static int hdspm_toggle_setting(struct hdspm *hdspm, u32 regmask)
 {
-	return (hdspm->control_register & regmask) ? 1 : 0;
+	u32 reg;
+
+	if (hdspm_is_raydat_or_aio(hdspm))
+		reg = hdspm->settings_register;
+	else
+		reg = hdspm->control_register;
+
+	return (reg & regmask) ? 1 : 0;
 }
 
 static int hdspm_set_toggle_setting(struct hdspm *hdspm, u32 regmask, int out)
 {
+	u32 *reg;
+	u32 target_reg;
+
+	if (hdspm_is_raydat_or_aio(hdspm)) {
+		reg = &(hdspm->settings_register);
+		target_reg = HDSPM_WR_SETTINGS;
+	} else {
+		reg = &(hdspm->control_register);
+		target_reg = HDSPM_controlRegister;
+	}
+
 	if (out)
-		hdspm->control_register |= regmask;
+		*reg |= regmask;
 	else
-		hdspm->control_register &= ~regmask;
-	hdspm_write(hdspm, HDSPM_controlRegister, hdspm->control_register);
+		*reg &= ~regmask;
+
+	hdspm_write(hdspm, target_reg, *reg);
 
 	return 0;
 }
@@ -3141,7 +3306,7 @@ static int hdspm_set_input_select(struct hdspm * hdspm, int out)
 static int snd_hdspm_info_input_select(struct snd_kcontrol *kcontrol,
 				       struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "optical", "coaxial" };
+	static const char *const texts[] = { "optical", "coaxial" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3203,7 +3368,7 @@ static int hdspm_set_ds_wire(struct hdspm * hdspm, int ds)
 static int snd_hdspm_info_ds_wire(struct snd_kcontrol *kcontrol,
 				  struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "Single", "Double" };
+	static const char *const texts[] = { "Single", "Double" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3276,7 +3441,7 @@ static int hdspm_set_qs_wire(struct hdspm * hdspm, int mode)
 static int snd_hdspm_info_qs_wire(struct snd_kcontrol *kcontrol,
 				       struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "Single", "Double", "Quad" };
+	static const char *const texts[] = { "Single", "Double", "Quad" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3313,6 +3478,84 @@ static int snd_hdspm_put_qs_wire(struct snd_kcontrol *kcontrol,
 	return change;
 }
 
+#define HDSPM_CONTROL_TRISTATE(xname, xindex) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
+	.name = xname, \
+	.private_value = xindex, \
+	.info = snd_hdspm_info_tristate, \
+	.get = snd_hdspm_get_tristate, \
+	.put = snd_hdspm_put_tristate \
+}
+
+static int hdspm_tristate(struct hdspm *hdspm, u32 regmask)
+{
+	u32 reg = hdspm->settings_register & (regmask * 3);
+	return reg / regmask;
+}
+
+static int hdspm_set_tristate(struct hdspm *hdspm, int mode, u32 regmask)
+{
+	hdspm->settings_register &= ~(regmask * 3);
+	hdspm->settings_register |= (regmask * mode);
+	hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register);
+
+	return 0;
+}
+
+static int snd_hdspm_info_tristate(struct snd_kcontrol *kcontrol,
+				       struct snd_ctl_elem_info *uinfo)
+{
+	u32 regmask = kcontrol->private_value;
+
+	static const char *const texts_spdif[] = { "Optical", "Coaxial", "Internal" };
+	static const char *const texts_levels[] = { "Hi Gain", "+4 dBu", "-10 dBV" };
+
+	switch (regmask) {
+	case HDSPM_c0_Input0:
+		ENUMERATED_CTL_INFO(uinfo, texts_spdif);
+		break;
+	default:
+		ENUMERATED_CTL_INFO(uinfo, texts_levels);
+		break;
+	}
+	return 0;
+}
+
+static int snd_hdspm_get_tristate(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+	u32 regmask = kcontrol->private_value;
+
+	spin_lock_irq(&hdspm->lock);
+	ucontrol->value.enumerated.item[0] = hdspm_tristate(hdspm, regmask);
+	spin_unlock_irq(&hdspm->lock);
+	return 0;
+}
+
+static int snd_hdspm_put_tristate(struct snd_kcontrol *kcontrol,
+				      struct snd_ctl_elem_value *ucontrol)
+{
+	struct hdspm *hdspm = snd_kcontrol_chip(kcontrol);
+	u32 regmask = kcontrol->private_value;
+	int change;
+	int val;
+
+	if (!snd_hdspm_use_is_exclusive(hdspm))
+		return -EBUSY;
+	val = ucontrol->value.integer.value[0];
+	if (val < 0)
+		val = 0;
+	if (val > 2)
+		val = 2;
+
+	spin_lock_irq(&hdspm->lock);
+	change = val != hdspm_tristate(hdspm, regmask);
+	hdspm_set_tristate(hdspm, val, regmask);
+	spin_unlock_irq(&hdspm->lock);
+	return change;
+}
+
 #define HDSPM_MADI_SPEEDMODE(xname, xindex) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
 	.name = xname, \
@@ -3352,7 +3595,7 @@ static int hdspm_set_madi_speedmode(struct hdspm *hdspm, int mode)
 static int snd_hdspm_info_madi_speedmode(struct snd_kcontrol *kcontrol,
 				       struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "Single", "Double", "Quad" };
+	static const char *const texts[] = { "Single", "Double", "Quad" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3587,7 +3830,7 @@ static int snd_hdspm_put_playback_mixer(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_sync_check(struct snd_kcontrol *kcontrol,
 				     struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "No Lock", "Lock", "Sync", "N/A" };
+	static const char *const texts[] = { "No Lock", "Lock", "Sync", "N/A" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3595,7 +3838,7 @@ static int snd_hdspm_info_sync_check(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_tco_info_lock_check(struct snd_kcontrol *kcontrol,
 				     struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "No Lock", "Lock" };
+	static const char *const texts[] = { "No Lock", "Lock" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -3745,9 +3988,18 @@ static int hdspm_tco_sync_check(struct hdspm *hdspm)
 	if (hdspm->tco) {
 		switch (hdspm->io_type) {
 		case MADI:
+			status = hdspm_read(hdspm, HDSPM_statusRegister);
+			if (status & HDSPM_tcoLockMadi) {
+				if (status & HDSPM_tcoSync)
+					return 2;
+				else
+					return 1;
+			}
+			return 0;
+			break;
 		case AES32:
 			status = hdspm_read(hdspm, HDSPM_statusRegister);
-			if (status & HDSPM_tcoLock) {
+			if (status & HDSPM_tcoLockAes) {
 				if (status & HDSPM_tcoSync)
 					return 2;
 				else
@@ -3807,7 +4059,8 @@ static int snd_hdspm_get_sync_check(struct snd_kcontrol *kcontrol,
 		case 5: /* SYNC IN */
 			val = hdspm_sync_in_sync_check(hdspm); break;
 		default:
-			val = hdspm_s1_sync_check(hdspm, ucontrol->id.index-1);
+			val = hdspm_s1_sync_check(hdspm,
+					kcontrol->private_value-1);
 		}
 		break;
 
@@ -3975,7 +4228,8 @@ static void hdspm_tco_write(struct hdspm *hdspm)
 static int snd_hdspm_info_tco_sample_rate(struct snd_kcontrol *kcontrol,
 					  struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "44.1 kHz", "48 kHz" };
+	/* TODO freq from app could be supported here, see tco->samplerate */
+	static const char *const texts[] = { "44.1 kHz", "48 kHz" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -4021,7 +4275,8 @@ static int snd_hdspm_put_tco_sample_rate(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_tco_pull(struct snd_kcontrol *kcontrol,
 				   struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "0", "+ 0.1 %", "- 0.1 %", "+ 4 %", "- 4 %" };
+	static const char *const texts[] = { "0", "+ 0.1 %", "- 0.1 %",
+		"+ 4 %", "- 4 %" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -4066,7 +4321,7 @@ static int snd_hdspm_put_tco_pull(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_tco_wck_conversion(struct snd_kcontrol *kcontrol,
 					     struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "1:1", "44.1 -> 48", "48 -> 44.1" };
+	static const char *const texts[] = { "1:1", "44.1 -> 48", "48 -> 44.1" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -4112,7 +4367,7 @@ static int snd_hdspm_put_tco_wck_conversion(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_tco_frame_rate(struct snd_kcontrol *kcontrol,
 					  struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "24 fps", "25 fps", "29.97fps",
+	static const char *const texts[] = { "24 fps", "25 fps", "29.97fps",
 		"29.97 dfps", "30 fps", "30 dfps" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
@@ -4159,7 +4414,7 @@ static int snd_hdspm_put_tco_frame_rate(struct snd_kcontrol *kcontrol,
 static int snd_hdspm_info_tco_sync_source(struct snd_kcontrol *kcontrol,
 					  struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = { "LTC", "Video", "WCK" };
+	static const char *const texts[] = { "LTC", "Video", "WCK" };
 	ENUMERATED_CTL_INFO(uinfo, texts);
 	return 0;
 }
@@ -4284,7 +4539,6 @@ static struct snd_kcontrol_new snd_hdspm_controls_aio[] = {
 	HDSPM_INTERNAL_CLOCK("Internal Clock", 0),
 	HDSPM_SYSTEM_CLOCK_MODE("System Clock Mode", 0),
 	HDSPM_PREF_SYNC_REF("Preferred Sync Reference", 0),
-	HDSPM_AUTOSYNC_REF("AutoSync Reference", 0),
 	HDSPM_SYSTEM_SAMPLE_RATE("System Sample Rate", 0),
 	HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 0),
 	HDSPM_SYNC_CHECK("WC SyncCheck", 0),
@@ -4298,7 +4552,16 @@ static struct snd_kcontrol_new snd_hdspm_controls_aio[] = {
 	HDSPM_AUTOSYNC_SAMPLE_RATE("SPDIF Frequency", 2),
 	HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT Frequency", 3),
 	HDSPM_AUTOSYNC_SAMPLE_RATE("TCO Frequency", 4),
-	HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 5)
+	HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 5),
+	HDSPM_CONTROL_TRISTATE("S/PDIF Input", HDSPM_c0_Input0),
+	HDSPM_TOGGLE_SETTING("S/PDIF Out Optical", HDSPM_c0_Spdif_Opt),
+	HDSPM_TOGGLE_SETTING("S/PDIF Out Professional", HDSPM_c0_Pro),
+	HDSPM_TOGGLE_SETTING("ADAT internal (AEB/TEB)", HDSPM_c0_AEB1),
+	HDSPM_TOGGLE_SETTING("XLR Breakout Cable", HDSPM_c0_Sym6db),
+	HDSPM_TOGGLE_SETTING("Single Speed WordClock Out", HDSPM_c0_Wck48),
+	HDSPM_CONTROL_TRISTATE("Input Level", HDSPM_c0_AD_GAIN0),
+	HDSPM_CONTROL_TRISTATE("Output Level", HDSPM_c0_DA_GAIN0),
+	HDSPM_CONTROL_TRISTATE("Phones Level", HDSPM_c0_PH_GAIN0)
 
 		/*
 		   HDSPM_INPUT_SELECT("Input Select", 0),
@@ -4335,7 +4598,9 @@ static struct snd_kcontrol_new snd_hdspm_controls_raydat[] = {
 	HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT3 Frequency", 5),
 	HDSPM_AUTOSYNC_SAMPLE_RATE("ADAT4 Frequency", 6),
 	HDSPM_AUTOSYNC_SAMPLE_RATE("TCO Frequency", 7),
-	HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 8)
+	HDSPM_AUTOSYNC_SAMPLE_RATE("SYNC IN Frequency", 8),
+	HDSPM_TOGGLE_SETTING("S/PDIF Out Professional", HDSPM_c0_Pro),
+	HDSPM_TOGGLE_SETTING("Single Speed WordClock Out", HDSPM_c0_Wck48)
 };
 
 static struct snd_kcontrol_new snd_hdspm_controls_aes32[] = {
@@ -4345,7 +4610,7 @@ static struct snd_kcontrol_new snd_hdspm_controls_aes32[] = {
 	HDSPM_PREF_SYNC_REF("Preferred Sync Reference", 0),
 	HDSPM_AUTOSYNC_REF("AutoSync Reference", 0),
 	HDSPM_SYSTEM_SAMPLE_RATE("System Sample Rate", 0),
-	HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 0),
+	HDSPM_AUTOSYNC_SAMPLE_RATE("External Rate", 11),
 	HDSPM_SYNC_CHECK("WC Sync Check", 0),
 	HDSPM_SYNC_CHECK("AES1 Sync Check", 1),
 	HDSPM_SYNC_CHECK("AES2 Sync Check", 2),
@@ -4501,77 +4766,22 @@ static int snd_hdspm_create_controls(struct snd_card *card,
  ------------------------------------------------------------*/
 
 static void
-snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
-			 struct snd_info_buffer *buffer)
+snd_hdspm_proc_read_tco(struct snd_info_entry *entry,
+					struct snd_info_buffer *buffer)
 {
 	struct hdspm *hdspm = entry->private_data;
-	unsigned int status, status2, control, freq;
-
-	char *pref_sync_ref;
-	char *autosync_ref;
-	char *system_clock_mode;
-	char *insel;
-	int x, x2;
-
-	/* TCO stuff */
+	unsigned int status, control;
 	int a, ltc, frames, seconds, minutes, hours;
 	unsigned int period;
 	u64 freq_const = 0;
 	u32 rate;
 
+	snd_iprintf(buffer, "--- TCO ---\n");
+
 	status = hdspm_read(hdspm, HDSPM_statusRegister);
-	status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
 	control = hdspm->control_register;
-	freq = hdspm_read(hdspm, HDSPM_timecodeRegister);
 
-	snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n",
-			hdspm->card_name, hdspm->card->number + 1,
-			hdspm->firmware_rev,
-			(status2 & HDSPM_version0) |
-			(status2 & HDSPM_version1) | (status2 &
-				HDSPM_version2));
 
-	snd_iprintf(buffer, "HW Serial: 0x%06x%06x\n",
-			(hdspm_read(hdspm, HDSPM_midiStatusIn1)>>8) & 0xFFFFFF,
-			hdspm->serial);
-
-	snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n",
-			hdspm->irq, hdspm->port, (unsigned long)hdspm->iobase);
-
-	snd_iprintf(buffer, "--- System ---\n");
-
-	snd_iprintf(buffer,
-		"IRQ Pending: Audio=%d, MIDI0=%d, MIDI1=%d, IRQcount=%d\n",
-		status & HDSPM_audioIRQPending,
-		(status & HDSPM_midi0IRQPending) ? 1 : 0,
-		(status & HDSPM_midi1IRQPending) ? 1 : 0,
-		hdspm->irq_count);
-	snd_iprintf(buffer,
-		"HW pointer: id = %d, rawptr = %d (%d->%d) "
-		"estimated= %ld (bytes)\n",
-		((status & HDSPM_BufferID) ? 1 : 0),
-		(status & HDSPM_BufferPositionMask),
-		(status & HDSPM_BufferPositionMask) %
-		(2 * (int)hdspm->period_bytes),
-		((status & HDSPM_BufferPositionMask) - 64) %
-		(2 * (int)hdspm->period_bytes),
-		(long) hdspm_hw_pointer(hdspm) * 4);
-
-	snd_iprintf(buffer,
-		"MIDI FIFO: Out1=0x%x, Out2=0x%x, In1=0x%x, In2=0x%x \n",
-		hdspm_read(hdspm, HDSPM_midiStatusOut0) & 0xFF,
-		hdspm_read(hdspm, HDSPM_midiStatusOut1) & 0xFF,
-		hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xFF,
-		hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xFF);
-	snd_iprintf(buffer,
-		"MIDIoverMADI FIFO: In=0x%x, Out=0x%x \n",
-		hdspm_read(hdspm, HDSPM_midiStatusIn2) & 0xFF,
-		hdspm_read(hdspm, HDSPM_midiStatusOut2) & 0xFF);
-	snd_iprintf(buffer,
-		"Register: ctrl1=0x%x, ctrl2=0x%x, status1=0x%x, "
-		"status2=0x%x\n",
-		hdspm->control_register, hdspm->control2_register,
-		status, status2);
 	if (status & HDSPM_tco_detect) {
 		snd_iprintf(buffer, "TCO module detected.\n");
 		a = hdspm_read(hdspm, HDSPM_RD_TCO+4);
@@ -4665,6 +4875,75 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
 	} else {
 		snd_iprintf(buffer, "No TCO module detected.\n");
 	}
+}
+
+static void
+snd_hdspm_proc_read_madi(struct snd_info_entry *entry,
+			 struct snd_info_buffer *buffer)
+{
+	struct hdspm *hdspm = entry->private_data;
+	unsigned int status, status2, control, freq;
+
+	char *pref_sync_ref;
+	char *autosync_ref;
+	char *system_clock_mode;
+	char *insel;
+	int x, x2;
+
+	status = hdspm_read(hdspm, HDSPM_statusRegister);
+	status2 = hdspm_read(hdspm, HDSPM_statusRegister2);
+	control = hdspm->control_register;
+	freq = hdspm_read(hdspm, HDSPM_timecodeRegister);
+
+	snd_iprintf(buffer, "%s (Card #%d) Rev.%x Status2first3bits: %x\n",
+			hdspm->card_name, hdspm->card->number + 1,
+			hdspm->firmware_rev,
+			(status2 & HDSPM_version0) |
+			(status2 & HDSPM_version1) | (status2 &
+				HDSPM_version2));
+
+	snd_iprintf(buffer, "HW Serial: 0x%06x%06x\n",
+			(hdspm_read(hdspm, HDSPM_midiStatusIn1)>>8) & 0xFFFFFF,
+			hdspm->serial);
+
+	snd_iprintf(buffer, "IRQ: %d Registers bus: 0x%lx VM: 0x%lx\n",
+			hdspm->irq, hdspm->port, (unsigned long)hdspm->iobase);
+
+	snd_iprintf(buffer, "--- System ---\n");
+
+	snd_iprintf(buffer,
+		"IRQ Pending: Audio=%d, MIDI0=%d, MIDI1=%d, IRQcount=%d\n",
+		status & HDSPM_audioIRQPending,
+		(status & HDSPM_midi0IRQPending) ? 1 : 0,
+		(status & HDSPM_midi1IRQPending) ? 1 : 0,
+		hdspm->irq_count);
+	snd_iprintf(buffer,
+		"HW pointer: id = %d, rawptr = %d (%d->%d) "
+		"estimated= %ld (bytes)\n",
+		((status & HDSPM_BufferID) ? 1 : 0),
+		(status & HDSPM_BufferPositionMask),
+		(status & HDSPM_BufferPositionMask) %
+		(2 * (int)hdspm->period_bytes),
+		((status & HDSPM_BufferPositionMask) - 64) %
+		(2 * (int)hdspm->period_bytes),
+		(long) hdspm_hw_pointer(hdspm) * 4);
+
+	snd_iprintf(buffer,
+		"MIDI FIFO: Out1=0x%x, Out2=0x%x, In1=0x%x, In2=0x%x \n",
+		hdspm_read(hdspm, HDSPM_midiStatusOut0) & 0xFF,
+		hdspm_read(hdspm, HDSPM_midiStatusOut1) & 0xFF,
+		hdspm_read(hdspm, HDSPM_midiStatusIn0) & 0xFF,
+		hdspm_read(hdspm, HDSPM_midiStatusIn1) & 0xFF);
+	snd_iprintf(buffer,
+		"MIDIoverMADI FIFO: In=0x%x, Out=0x%x \n",
+		hdspm_read(hdspm, HDSPM_midiStatusIn2) & 0xFF,
+		hdspm_read(hdspm, HDSPM_midiStatusOut2) & 0xFF);
+	snd_iprintf(buffer,
+		"Register: ctrl1=0x%x, ctrl2=0x%x, status1=0x%x, "
+		"status2=0x%x\n",
+		hdspm->control_register, hdspm->control2_register,
+		status, status2);
+
 
 	snd_iprintf(buffer, "--- Settings ---\n");
 
@@ -4768,6 +5047,9 @@ snd_hdspm_proc_read_madi(struct snd_info_entry * entry,
 		(status & HDSPM_RX_64ch) ? "64 channels" :
 		"56 channels");
 
+	/* call readout function for TCO specific status */
+	snd_hdspm_proc_read_tco(entry, buffer);
+
 	snd_iprintf(buffer, "\n");
 }
 
@@ -4909,11 +5191,18 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry,
 		autosync_ref = "AES7"; break;
 	case HDSPM_AES32_AUTOSYNC_FROM_AES8:
 		autosync_ref = "AES8"; break;
+	case HDSPM_AES32_AUTOSYNC_FROM_TCO:
+		autosync_ref = "TCO"; break;
+	case HDSPM_AES32_AUTOSYNC_FROM_SYNC_IN:
+		autosync_ref = "Sync In"; break;
 	default:
 		autosync_ref = "---"; break;
 	}
 	snd_iprintf(buffer, "AutoSync ref = %s\n", autosync_ref);
 
+	/* call readout function for TCO specific status */
+	snd_hdspm_proc_read_tco(entry, buffer);
+
 	snd_iprintf(buffer, "\n");
 }
 
@@ -5097,7 +5386,7 @@ static int snd_hdspm_set_defaults(struct hdspm * hdspm)
 
 	case AES32:
 		hdspm->control_register =
-			HDSPM_ClockModeMaster |	/* Master Cloack Mode on */
+			HDSPM_ClockModeMaster |	/* Master Clock Mode on */
 			hdspm_encode_latency(7) | /* latency max=8192samples */
 			HDSPM_SyncRef0 |	/* AES1 is syncclock */
 			HDSPM_LineOut |	/* Analog output in */
@@ -5123,9 +5412,8 @@ static int snd_hdspm_set_defaults(struct hdspm * hdspm)
 
 	all_in_all_mixer(hdspm, 0 * UNITY_GAIN);
 
-	if (hdspm->io_type == AIO || hdspm->io_type == RayDAT) {
+	if (hdspm_is_raydat_or_aio(hdspm))
 		hdspm_write(hdspm, HDSPM_WR_SETTINGS, hdspm->settings_register);
-	}
 
 	/* set a default rate so that the channel map is set up. */
 	hdspm_set_rate(hdspm, 48000, 1);
@@ -5371,6 +5659,16 @@ static int snd_hdspm_hw_params(struct snd_pcm_substream *substream,
 	   */
 
 
+	/*  For AES cards, the float format bit is the same as the
+	 *  preferred sync reference. Since we don't want to break
+	 *  sync settings, we have to skip the remaining part of this
+	 *  function.
+	 */
+	if (hdspm->io_type == AES32) {
+		return 0;
+	}
+
+
 	/* Switch to native float format if requested */
 	if (SNDRV_PCM_FORMAT_FLOAT_LE == params_format(params)) {
 		if (!(hdspm->control_register & HDSPe_FLOAT_FORMAT))
@@ -6013,7 +6311,7 @@ static int snd_hdspm_hwdep_ioctl(struct snd_hwdep *hw, struct file *file,
 				ltc.format = fps_2997;
 				break;
 			default:
-				ltc.format = 30;
+				ltc.format = fps_30;
 				break;
 			}
 			if (i & HDSPM_TCO1_set_drop_frame_flag) {
@@ -6479,10 +6777,6 @@ static int snd_hdspm_create(struct snd_card *card,
 		break;
 
 	case AIO:
-		if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) {
-			snd_printk(KERN_INFO "HDSPM: AEB input board found, but not supported\n");
-		}
-
 		hdspm->ss_in_channels = AIO_IN_SS_CHANNELS;
 		hdspm->ds_in_channels = AIO_IN_DS_CHANNELS;
 		hdspm->qs_in_channels = AIO_IN_QS_CHANNELS;
@@ -6490,6 +6784,20 @@ static int snd_hdspm_create(struct snd_card *card,
 		hdspm->ds_out_channels = AIO_OUT_DS_CHANNELS;
 		hdspm->qs_out_channels = AIO_OUT_QS_CHANNELS;
 
+		if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBI_D)) {
+			snd_printk(KERN_INFO "HDSPM: AEB input board found\n");
+			hdspm->ss_in_channels += 4;
+			hdspm->ds_in_channels += 4;
+			hdspm->qs_in_channels += 4;
+		}
+
+		if (0 == (hdspm_read(hdspm, HDSPM_statusRegister2) & HDSPM_s2_AEBO_D)) {
+			snd_printk(KERN_INFO "HDSPM: AEB output board found\n");
+			hdspm->ss_out_channels += 4;
+			hdspm->ds_out_channels += 4;
+			hdspm->qs_out_channels += 4;
+		}
+
 		hdspm->channel_map_out_ss = channel_map_aio_out_ss;
 		hdspm->channel_map_out_ds = channel_map_aio_out_ds;
 		hdspm->channel_map_out_qs = channel_map_aio_out_qs;
@@ -6558,6 +6866,7 @@ static int snd_hdspm_create(struct snd_card *card,
 		break;
 
 	case MADI:
+	case AES32:
 		if (hdspm_read(hdspm, HDSPM_statusRegister) & HDSPM_tco_detect) {
 			hdspm->midiPorts++;
 			hdspm->tco = kzalloc(sizeof(struct hdspm_tco),
@@ -6565,7 +6874,7 @@ static int snd_hdspm_create(struct snd_card *card,
 			if (NULL != hdspm->tco) {
 				hdspm_tco_write(hdspm);
 			}
-			snd_printk(KERN_INFO "HDSPM: MADI TCO module found\n");
+			snd_printk(KERN_INFO "HDSPM: MADI/AES TCO module found\n");
 		} else {
 			hdspm->tco = NULL;
 		}
@@ -6580,10 +6889,12 @@ static int snd_hdspm_create(struct snd_card *card,
 	case AES32:
 		if (hdspm->tco) {
 			hdspm->texts_autosync = texts_autosync_aes_tco;
-			hdspm->texts_autosync_items = 10;
+			hdspm->texts_autosync_items =
+				ARRAY_SIZE(texts_autosync_aes_tco);
 		} else {
 			hdspm->texts_autosync = texts_autosync_aes;
-			hdspm->texts_autosync_items = 9;
+			hdspm->texts_autosync_items =
+				ARRAY_SIZE(texts_autosync_aes);
 		}
 		break;
 
diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c
index f23f331e9a97..a57643d6402f 100644
--- a/sound/soc/cirrus/ep93xx-i2s.c
+++ b/sound/soc/cirrus/ep93xx-i2s.c
@@ -408,7 +408,6 @@ static int ep93xx_i2s_probe(struct platform_device *pdev)
 	return 0;
 
 fail_put_lrclk:
-	dev_set_drvdata(&pdev->dev, NULL);
 	clk_put(info->lrclk);
 fail_put_sclk:
 	clk_put(info->sclk);
@@ -423,7 +422,6 @@ static int ep93xx_i2s_remove(struct platform_device *pdev)
 	struct ep93xx_i2s_info *info = dev_get_drvdata(&pdev->dev);
 
 	snd_soc_unregister_component(&pdev->dev);
-	dev_set_drvdata(&pdev->dev, NULL);
 	clk_put(info->lrclk);
 	clk_put(info->sclk);
 	clk_put(info->mclk);
diff --git a/sound/soc/codecs/dmic.c b/sound/soc/codecs/dmic.c
index 66967ba6f757..b2090b2a5e2d 100644
--- a/sound/soc/codecs/dmic.c
+++ b/sound/soc/codecs/dmic.c
@@ -50,20 +50,11 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"DMIC AIF", NULL, "DMic"},
 };
 
-static int dmic_probe(struct snd_soc_codec *codec)
-{
-	struct snd_soc_dapm_context *dapm = &codec->dapm;
-
-	snd_soc_dapm_new_controls(dapm, dmic_dapm_widgets,
-				  ARRAY_SIZE(dmic_dapm_widgets));
-        snd_soc_dapm_add_routes(dapm, intercon, ARRAY_SIZE(intercon));
-	snd_soc_dapm_new_widgets(dapm);
-
-	return 0;
-}
-
 static struct snd_soc_codec_driver soc_dmic = {
-	.probe	= dmic_probe,
+	.dapm_widgets = dmic_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(dmic_dapm_widgets),
+	.dapm_routes = intercon,
+	.num_dapm_routes = ARRAY_SIZE(intercon),
 };
 
 static int dmic_dev_probe(struct platform_device *pdev)
diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c
index 4db7314baabc..c26a8f814b18 100644
--- a/sound/soc/codecs/rt5640.c
+++ b/sound/soc/codecs/rt5640.c
@@ -50,8 +50,6 @@ static const struct regmap_range_cfg rt5640_ranges[] = {
 
 static struct reg_default init_list[] = {
 	{RT5640_PR_BASE + 0x3d,	0x3600},
-	{RT5640_PR_BASE + 0x1c,	0x0D21},
-	{RT5640_PR_BASE + 0x1b,	0x0000},
 	{RT5640_PR_BASE + 0x12,	0x0aa8},
 	{RT5640_PR_BASE + 0x14,	0x0aaa},
 	{RT5640_PR_BASE + 0x20,	0x6110},
@@ -384,15 +382,11 @@ static const SOC_ENUM_SINGLE_DECL(
 
 static const struct snd_kcontrol_new rt5640_snd_controls[] = {
 	/* Speaker Output Volume */
-	SOC_DOUBLE("Speaker Playback Switch", RT5640_SPK_VOL,
-		RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1),
 	SOC_DOUBLE("Speaker Channel Switch", RT5640_SPK_VOL,
 		RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1),
 	SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL,
 		RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv),
 	/* Headphone Output Volume */
-	SOC_DOUBLE("HP Playback Switch", RT5640_HP_VOL,
-		RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1),
 	SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL,
 		RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1),
 	SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL,
@@ -737,6 +731,22 @@ static const struct snd_kcontrol_new rt5640_mono_mix[] = {
 			RT5640_M_BST1_MM_SFT, 1, 1),
 };
 
+static const struct snd_kcontrol_new spk_l_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL,
+		RT5640_L_MUTE_SFT, 1, 1);
+
+static const struct snd_kcontrol_new spk_r_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_SPK_VOL,
+		RT5640_R_MUTE_SFT, 1, 1);
+
+static const struct snd_kcontrol_new hp_l_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL,
+		RT5640_L_MUTE_SFT, 1, 1);
+
+static const struct snd_kcontrol_new hp_r_enable_control =
+	SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5640_HP_VOL,
+		RT5640_R_MUTE_SFT, 1, 1);
+
 /* Stereo ADC source */
 static const char * const rt5640_stereo_adc1_src[] = {
 	"DIG MIX", "ADC"
@@ -868,33 +878,6 @@ static const SOC_ENUM_SINGLE_DECL(
 static const struct snd_kcontrol_new rt5640_sdi_mux =
 	SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum);
 
-static int spk_event(struct snd_soc_dapm_widget *w,
-	struct snd_kcontrol *kcontrol, int event)
-{
-	struct snd_soc_codec *codec = w->codec;
-	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
-
-	switch (event) {
-	case SND_SOC_DAPM_POST_PMU:
-		regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1,
-					0x0001, 0x0001);
-		regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c,
-					0xf000, 0xf000);
-		break;
-
-	case SND_SOC_DAPM_PRE_PMD:
-		regmap_update_bits(rt5640->regmap, RT5640_PR_BASE + 0x1c,
-					0xf000, 0x0000);
-		regmap_update_bits(rt5640->regmap, RT5640_PWR_DIG1,
-					0x0001, 0x0000);
-		break;
-
-	default:
-		return 0;
-	}
-	return 0;
-}
-
 static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -943,6 +926,117 @@ static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+void hp_amp_power_on(struct snd_soc_codec *codec)
+{
+	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+	/* depop parameters */
+	regmap_update_bits(rt5640->regmap, RT5640_PR_BASE +
+		RT5640_CHPUMP_INT_REG1, 0x0700, 0x0200);
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2,
+		RT5640_DEPOP_MASK, RT5640_DEPOP_MAN);
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1,
+		RT5640_HP_CP_MASK | RT5640_HP_SG_MASK | RT5640_HP_CB_MASK,
+		RT5640_HP_CP_PU | RT5640_HP_SG_DIS | RT5640_HP_CB_PU);
+	regmap_write(rt5640->regmap, RT5640_PR_BASE + RT5640_HP_DCC_INT1,
+			   0x9f00);
+	/* headphone amp power on */
+	regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1,
+		RT5640_PWR_FV1 | RT5640_PWR_FV2, 0);
+	regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1,
+		RT5640_PWR_HA,
+		RT5640_PWR_HA);
+	usleep_range(10000, 15000);
+	regmap_update_bits(rt5640->regmap, RT5640_PWR_ANLG1,
+		RT5640_PWR_FV1 | RT5640_PWR_FV2 ,
+		RT5640_PWR_FV1 | RT5640_PWR_FV2);
+}
+
+static void rt5640_pmu_depop(struct snd_soc_codec *codec)
+{
+	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M2,
+		RT5640_DEPOP_MASK | RT5640_DIG_DP_MASK,
+		RT5640_DEPOP_AUTO | RT5640_DIG_DP_EN);
+	regmap_update_bits(rt5640->regmap, RT5640_CHARGE_PUMP,
+		RT5640_PM_HP_MASK, RT5640_PM_HP_HV);
+
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M3,
+		RT5640_CP_FQ1_MASK | RT5640_CP_FQ2_MASK | RT5640_CP_FQ3_MASK,
+		(RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ1_SFT) |
+		(RT5640_CP_FQ_12_KHZ << RT5640_CP_FQ2_SFT) |
+		(RT5640_CP_FQ_192_KHZ << RT5640_CP_FQ3_SFT));
+
+	regmap_write(rt5640->regmap, RT5640_PR_BASE +
+		RT5640_MAMP_INT_REG2, 0x1c00);
+	regmap_update_bits(rt5640->regmap, RT5640_DEPOP_M1,
+		RT5640_HP_CP_MASK | RT5640_HP_SG_MASK,
+		RT5640_HP_CP_PD | RT5640_HP_SG_EN);
+	regmap_update_bits(rt5640->regmap, RT5640_PR_BASE +
+		RT5640_CHPUMP_INT_REG1, 0x0700, 0x0400);
+}
+
+static int rt5640_hp_event(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		rt5640_pmu_depop(codec);
+		rt5640->hp_mute = 0;
+		break;
+
+	case SND_SOC_DAPM_PRE_PMD:
+		rt5640->hp_mute = 1;
+		usleep_range(70000, 75000);
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
+static int rt5640_hp_power_event(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		hp_amp_power_on(codec);
+		break;
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
+static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *kcontrol, int event)
+{
+	struct snd_soc_codec *codec = w->codec;
+	struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		if (!rt5640->hp_mute)
+			usleep_range(80000, 85000);
+
+		break;
+
+	default:
+		return 0;
+	}
+
+	return 0;
+}
+
 static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = {
 	SND_SOC_DAPM_SUPPLY("PLL1", RT5640_PWR_ANLG2,
 			RT5640_PWR_PLL_BIT, 0, NULL, 0),
@@ -1132,15 +1226,28 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = {
 		rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)),
 	SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1,
 		RT5640_PWR_MA_BIT, 0, NULL, 0),
-	SND_SOC_DAPM_SUPPLY("Improve HP Amp Drv", RT5640_PWR_ANLG1,
-		SND_SOC_NOPM, 0, NULL, 0),
-	SND_SOC_DAPM_PGA("HP L Amp", RT5640_PWR_ANLG1,
+	SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM,
+		0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0,
+		rt5640_hp_event,
+		SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+	SND_SOC_DAPM_SUPPLY("HP L Amp", RT5640_PWR_ANLG1,
 		RT5640_PWR_HP_L_BIT, 0, NULL, 0),
-	SND_SOC_DAPM_PGA("HP R Amp", RT5640_PWR_ANLG1,
+	SND_SOC_DAPM_SUPPLY("HP R Amp", RT5640_PWR_ANLG1,
 		RT5640_PWR_HP_R_BIT, 0, NULL, 0),
 	SND_SOC_DAPM_SUPPLY("Improve SPK Amp Drv", RT5640_PWR_DIG1,
-		SND_SOC_NOPM, 0, spk_event,
-		SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU),
+		RT5640_PWR_CLS_D_BIT, 0, NULL, 0),
+
+	/* Output Switch */
+	SND_SOC_DAPM_SWITCH("Speaker L Playback", SND_SOC_NOPM, 0, 0,
+			&spk_l_enable_control),
+	SND_SOC_DAPM_SWITCH("Speaker R Playback", SND_SOC_NOPM, 0, 0,
+			&spk_r_enable_control),
+	SND_SOC_DAPM_SWITCH("HP L Playback", SND_SOC_NOPM, 0, 0,
+			&hp_l_enable_control),
+	SND_SOC_DAPM_SWITCH("HP R Playback", SND_SOC_NOPM, 0, 0,
+			&hp_r_enable_control),
+	SND_SOC_DAPM_POST("HP Post", rt5640_hp_post_event),
 	/* Output Lines */
 	SND_SOC_DAPM_OUTPUT("SPOLP"),
 	SND_SOC_DAPM_OUTPUT("SPOLN"),
@@ -1381,9 +1488,11 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
 	{"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"},
 	{"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"},
 	{"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"},
+	{"HPO MIX L", NULL, "HP L Amp"},
 	{"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"},
 	{"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"},
 	{"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"},
+	{"HPO MIX R", NULL, "HP R Amp"},
 
 	{"LOUT MIX", "DAC L1 Switch", "DAC L1"},
 	{"LOUT MIX", "DAC R1 Switch", "DAC R1"},
@@ -1396,13 +1505,15 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
 	{"Mono MIX", "OUTVOL L Switch", "OUTVOL L"},
 	{"Mono MIX", "BST1 Switch", "BST1"},
 
-	{"HP L Amp", NULL, "HPO MIX L"},
-	{"HP R Amp", NULL, "HPO MIX R"},
+	{"HP Amp", NULL, "HPO MIX L"},
+	{"HP Amp", NULL, "HPO MIX R"},
 
-	{"SPOLP", NULL, "SPOL MIX"},
-	{"SPOLN", NULL, "SPOL MIX"},
-	{"SPORP", NULL, "SPOR MIX"},
-	{"SPORN", NULL, "SPOR MIX"},
+	{"Speaker L Playback", "Switch", "SPOL MIX"},
+	{"Speaker R Playback", "Switch", "SPOR MIX"},
+	{"SPOLP", NULL, "Speaker L Playback"},
+	{"SPOLN", NULL, "Speaker L Playback"},
+	{"SPORP", NULL, "Speaker R Playback"},
+	{"SPORN", NULL, "Speaker R Playback"},
 
 	{"SPOLP", NULL, "Improve SPK Amp Drv"},
 	{"SPOLN", NULL, "Improve SPK Amp Drv"},
@@ -1412,8 +1523,10 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = {
 	{"HPOL", NULL, "Improve HP Amp Drv"},
 	{"HPOR", NULL, "Improve HP Amp Drv"},
 
-	{"HPOL", NULL, "HP L Amp"},
-	{"HPOR", NULL, "HP R Amp"},
+	{"HP L Playback", "Switch", "HP Amp"},
+	{"HP R Playback", "Switch", "HP Amp"},
+	{"HPOL", NULL, "HP L Playback"},
+	{"HPOR", NULL, "HP R Playback"},
 	{"LOUTL", NULL, "LOUT MIX"},
 	{"LOUTR", NULL, "LOUT MIX"},
 	{"MONOP", NULL, "Mono MIX"},
@@ -1792,17 +1905,13 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec,
 				RT5640_PWR_BG | RT5640_PWR_VREF2,
 				RT5640_PWR_VREF1 | RT5640_PWR_MB |
 				RT5640_PWR_BG | RT5640_PWR_VREF2);
-			mdelay(10);
+			usleep_range(10000, 15000);
 			snd_soc_update_bits(codec, RT5640_PWR_ANLG1,
 				RT5640_PWR_FV1 | RT5640_PWR_FV2,
 				RT5640_PWR_FV1 | RT5640_PWR_FV2);
 			regcache_sync(rt5640->regmap);
 			snd_soc_update_bits(codec, RT5640_DUMMY1,
 						0x0301, 0x0301);
-			snd_soc_update_bits(codec, RT5640_DEPOP_M1,
-						0x001d, 0x0019);
-			snd_soc_update_bits(codec, RT5640_DEPOP_M2,
-						0x2000, 0x2000);
 			snd_soc_update_bits(codec, RT5640_MICBIAS,
 						0x0030, 0x0030);
 		}
@@ -1846,8 +1955,6 @@ static int rt5640_probe(struct snd_soc_codec *codec)
 	rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
 	snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301);
-	snd_soc_update_bits(codec, RT5640_DEPOP_M1, 0x001d, 0x0019);
-	snd_soc_update_bits(codec, RT5640_DEPOP_M2, 0x2000, 0x2000);
 	snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030);
 	snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00);
 
@@ -2069,6 +2176,8 @@ static int rt5640_i2c_probe(struct i2c_client *i2c,
 		regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4,
 					RT5640_IN_DF2, RT5640_IN_DF2);
 
+	rt5640->hp_mute = 1;
+
 	ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640,
 			rt5640_dai, ARRAY_SIZE(rt5640_dai));
 	if (ret < 0)
diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h
index c48286d7118f..5e8df25a13f3 100644
--- a/sound/soc/codecs/rt5640.h
+++ b/sound/soc/codecs/rt5640.h
@@ -145,6 +145,8 @@
 
 
 /* Index of Codec Private Register definition */
+#define RT5640_CHPUMP_INT_REG1			0x24
+#define RT5640_MAMP_INT_REG2			0x37
 #define RT5640_3D_SPK				0x63
 #define RT5640_WND_1				0x6c
 #define RT5640_WND_2				0x6d
@@ -153,6 +155,7 @@
 #define RT5640_WND_5				0x70
 #define RT5640_WND_8				0x73
 #define RT5640_DIP_SPK_INF			0x75
+#define RT5640_HP_DCC_INT1			0x77
 #define RT5640_EQ_BW_LOP			0xa0
 #define RT5640_EQ_GN_LOP			0xa1
 #define RT5640_EQ_FC_BP1			0xa2
@@ -1201,6 +1204,14 @@
 #define RT5640_CP_FQ2_SFT			4
 #define RT5640_CP_FQ3_MASK			(0x7)
 #define RT5640_CP_FQ3_SFT			0
+#define RT5640_CP_FQ_1_5_KHZ			0
+#define RT5640_CP_FQ_3_KHZ			1
+#define RT5640_CP_FQ_6_KHZ			2
+#define RT5640_CP_FQ_12_KHZ			3
+#define RT5640_CP_FQ_24_KHZ			4
+#define RT5640_CP_FQ_48_KHZ			5
+#define RT5640_CP_FQ_96_KHZ			6
+#define RT5640_CP_FQ_192_KHZ			7
 
 /* HPOUT charge pump (0x91) */
 #define RT5640_OSW_L_MASK			(0x1 << 11)
@@ -2087,6 +2098,7 @@ struct rt5640_priv {
 	int pll_out;
 
 	int dmic_en;
+	bool hp_mute;
 };
 
 #endif
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index f8d30e5f6371..492644e67ace 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -561,8 +561,9 @@ static int ssm2602_suspend(struct snd_soc_codec *codec)
 
 static int ssm2602_resume(struct snd_soc_codec *codec)
 {
-	snd_soc_cache_sync(codec);
+	struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec);
 
+	regcache_sync(ssm2602->regmap);
 	ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	return 0;
diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c
index 17df4e32feac..2ed57d4aa445 100644
--- a/sound/soc/codecs/tlv320aic32x4.c
+++ b/sound/soc/codecs/tlv320aic32x4.c
@@ -338,18 +338,6 @@ static inline int aic32x4_get_divs(int mclk, int rate)
 	return -EINVAL;
 }
 
-static int aic32x4_add_widgets(struct snd_soc_codec *codec)
-{
-	snd_soc_dapm_new_controls(&codec->dapm, aic32x4_dapm_widgets,
-				  ARRAY_SIZE(aic32x4_dapm_widgets));
-
-	snd_soc_dapm_add_routes(&codec->dapm, aic32x4_dapm_routes,
-				ARRAY_SIZE(aic32x4_dapm_routes));
-
-	snd_soc_dapm_new_widgets(&codec->dapm);
-	return 0;
-}
-
 static int aic32x4_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 				  int clk_id, unsigned int freq, int dir)
 {
@@ -683,9 +671,6 @@ static int aic32x4_probe(struct snd_soc_codec *codec)
 	}
 
 	aic32x4_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	snd_soc_add_codec_controls(codec, aic32x4_snd_controls,
-			     ARRAY_SIZE(aic32x4_snd_controls));
-	aic32x4_add_widgets(codec);
 
 	/*
 	 * Workaround: for an unknown reason, the ADC needs to be powered up
@@ -714,6 +699,13 @@ static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = {
 	.suspend = aic32x4_suspend,
 	.resume = aic32x4_resume,
 	.set_bias_level = aic32x4_set_bias_level,
+
+	.controls = aic32x4_snd_controls,
+	.num_controls = ARRAY_SIZE(aic32x4_snd_controls),
+	.dapm_widgets = aic32x4_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets),
+	.dapm_routes = aic32x4_dapm_routes,
+	.num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes),
 };
 
 static int aic32x4_i2c_probe(struct i2c_client *i2c,
diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c
index 91dfbfeda6f8..4dfa8dceeabf 100644
--- a/sound/soc/codecs/wm8904.c
+++ b/sound/soc/codecs/wm8904.c
@@ -1202,7 +1202,6 @@ static int wm8904_add_widgets(struct snd_soc_codec *codec)
 		break;
 	}
 
-	snd_soc_dapm_new_widgets(dapm);
 	return 0;
 }
 
diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c
index 36782f067cc5..11d80f3b6137 100644
--- a/sound/soc/codecs/wm8962.c
+++ b/sound/soc/codecs/wm8962.c
@@ -3174,7 +3174,7 @@ static ssize_t wm8962_beep_set(struct device *dev,
 	long int time;
 	int ret;
 
-	ret = strict_strtol(buf, 10, &time);
+	ret = kstrtol(buf, 10, &time);
 	if (ret != 0)
 		return ret;
 
diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c
index 70eb37a5dd16..25c31f1655f6 100644
--- a/sound/soc/dwc/designware_i2s.c
+++ b/sound/soc/dwc/designware_i2s.c
@@ -421,13 +421,11 @@ static int dw_i2s_probe(struct platform_device *pdev)
 					 dw_i2s_dai, 1);
 	if (ret != 0) {
 		dev_err(&pdev->dev, "not able to register dai\n");
-		goto err_set_drvdata;
+		goto err_clk_disable;
 	}
 
 	return 0;
 
-err_set_drvdata:
-	dev_set_drvdata(&pdev->dev, NULL);
 err_clk_disable:
 	clk_disable(dev->clk);
 err_clk_put:
@@ -440,7 +438,6 @@ static int dw_i2s_remove(struct platform_device *pdev)
 	struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev);
 
 	snd_soc_unregister_component(&pdev->dev);
-	dev_set_drvdata(&pdev->dev, NULL);
 
 	clk_put(dev->clk);
 
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index cd088cc8c866..b7ab71f2ccc1 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -193,6 +193,17 @@ config SND_SOC_IMX_SGTL5000
 	  Say Y if you want to add support for SoC audio on an i.MX board with
 	  a sgtl5000 codec.
 
+config SND_SOC_IMX_SPDIF
+	tristate "SoC Audio support for i.MX boards with S/PDIF"
+	select SND_SOC_IMX_PCM_DMA
+	select SND_SOC_FSL_SPDIF
+	select SND_SOC_SPDIF
+	select REGMAP_MMIO
+	help
+	  SoC Audio support for i.MX boards with S/PDIF
+	  Say Y if you want to add support for SoC audio on an i.MX board with
+	  a S/DPDIF.
+
 config SND_SOC_IMX_MC13783
 	tristate "SoC Audio support for I.MX boards with mc13783"
 	depends on MFD_MC13783 && ARM
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 4b5970e014dd..8db705b0fdf9 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -45,6 +45,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o
 snd-soc-wm1133-ev1-objs := wm1133-ev1.o
 snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o
 snd-soc-imx-wm8962-objs := imx-wm8962.o
+snd-soc-imx-spdif-objs := imx-spdif.o
 snd-soc-imx-mc13783-objs := imx-mc13783.o
 
 obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o
@@ -53,4 +54,5 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o
 obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o
 obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o
 obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o
+obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o
 obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 42a43820d993..3920c3e849ce 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -411,8 +411,8 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-int fsl_spdif_startup(struct snd_pcm_substream *substream,
-			struct snd_soc_dai *cpu_dai)
+static int fsl_spdif_startup(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *cpu_dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai);
@@ -546,7 +546,7 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream,
 	return 0;
 }
 
-struct snd_soc_dai_ops fsl_spdif_dai_ops = {
+static struct snd_soc_dai_ops fsl_spdif_dai_ops = {
 	.startup = fsl_spdif_startup,
 	.hw_params = fsl_spdif_hw_params,
 	.trigger = fsl_spdif_trigger,
@@ -555,7 +555,6 @@ struct snd_soc_dai_ops fsl_spdif_dai_ops = {
 
 
 /*
- * ============================================
  * FSL SPDIF IEC958 controller(mixer) functions
  *
  *	Channel status get/put control
@@ -563,7 +562,6 @@ struct snd_soc_dai_ops fsl_spdif_dai_ops = {
  *	Valid bit value get control
  *	DPLL lock status get control
  *	User bit sync mode selection control
- * ============================================
  */
 
 static int fsl_spdif_info(struct snd_kcontrol *kcontrol,
@@ -921,7 +919,7 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai)
 	return 0;
 }
 
-struct snd_soc_dai_driver fsl_spdif_dai = {
+static struct snd_soc_dai_driver fsl_spdif_dai = {
 	.probe = &fsl_spdif_dai_probe,
 	.playback = {
 		.channels_min = 2,
@@ -942,11 +940,7 @@ static const struct snd_soc_component_driver fsl_spdif_component = {
 	.name		= "fsl-spdif",
 };
 
-/*
- * ================
- * FSL SPDIF REGMAP
- * ================
- */
+/* FSL SPDIF REGMAP */
 
 static bool fsl_spdif_readable_reg(struct device *dev, unsigned int reg)
 {
@@ -1077,9 +1071,9 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv,
 			break;
 	}
 
-	dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate",
+	dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate\n",
 			spdif_priv->txclk_src[index], rate[index]);
-	dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate",
+	dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate\n",
 			spdif_priv->txclk_div[index], rate[index]);
 
 	return 0;
@@ -1119,10 +1113,8 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 	}
 
 	regs = devm_ioremap_resource(&pdev->dev, res);
-	if (IS_ERR(regs)) {
-		dev_err(&pdev->dev, "could not map device resources\n");
+	if (IS_ERR(regs))
 		return PTR_ERR(regs);
-	}
 
 	spdif_priv->regmap = devm_regmap_init_mmio_clk(&pdev->dev,
 			"core", regs, &fsl_spdif_regmap_config);
@@ -1184,7 +1176,7 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 					 &spdif_priv->cpu_dai_drv, 1);
 	if (ret) {
 		dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
-		goto error_dev;
+		return ret;
 	}
 
 	ret = imx_pcm_dma_init(pdev);
@@ -1197,8 +1189,6 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 
 error_component:
 	snd_soc_unregister_component(&pdev->dev);
-error_dev:
-	dev_set_drvdata(&pdev->dev, NULL);
 
 	return ret;
 }
@@ -1207,7 +1197,6 @@ static int fsl_spdif_remove(struct platform_device *pdev)
 {
 	imx_pcm_dma_exit(pdev);
 	snd_soc_unregister_component(&pdev->dev);
-	dev_set_drvdata(&pdev->dev, NULL);
 
 	return 0;
 }
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 5cf626c4dc96..c6b743978d5e 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -1114,7 +1114,6 @@ error_dai:
 	snd_soc_unregister_component(&pdev->dev);
 
 error_dev:
-	dev_set_drvdata(&pdev->dev, NULL);
 	device_remove_file(&pdev->dev, dev_attr);
 
 error_clk:
diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c
index ab17381cc981..d3bf71a0ec56 100644
--- a/sound/soc/fsl/imx-audmux.c
+++ b/sound/soc/fsl/imx-audmux.c
@@ -335,7 +335,8 @@ static int imx_audmux_probe(struct platform_device *pdev)
 	if (audmux_type == IMX31_AUDMUX)
 		audmux_debugfs_init();
 
-	imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node);
+	if (of_id)
+		imx_audmux_parse_dt_defaults(pdev, pdev->dev.of_node);
 
 	return 0;
 }
diff --git a/sound/soc/fsl/imx-spdif.c b/sound/soc/fsl/imx-spdif.c
new file mode 100644
index 000000000000..816013b0ebba
--- /dev/null
+++ b/sound/soc/fsl/imx-spdif.c
@@ -0,0 +1,148 @@
+/*
+ * Copyright (C) 2013 Freescale Semiconductor, Inc.
+ *
+ * The code contained herein is licensed under the GNU General Public
+ * License. You may obtain a copy of the GNU General Public License
+ * Version 2 or later at the following locations:
+ *
+ * http://www.opensource.org/licenses/gpl-license.html
+ * http://www.gnu.org/copyleft/gpl.html
+ */
+
+#include <linux/module.h>
+#include <linux/of_platform.h>
+#include <sound/soc.h>
+
+struct imx_spdif_data {
+	struct snd_soc_dai_link dai[2];
+	struct snd_soc_card card;
+	struct platform_device *txdev;
+	struct platform_device *rxdev;
+};
+
+static int imx_spdif_audio_probe(struct platform_device *pdev)
+{
+	struct device_node *spdif_np, *np = pdev->dev.of_node;
+	struct imx_spdif_data *data;
+	int ret = 0, num_links = 0;
+
+	spdif_np = of_parse_phandle(np, "spdif-controller", 0);
+	if (!spdif_np) {
+		dev_err(&pdev->dev, "failed to find spdif-controller\n");
+		ret = -EINVAL;
+		goto end;
+	}
+
+	data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL);
+	if (!data) {
+		dev_err(&pdev->dev, "failed to allocate memory\n");
+		ret = -ENOMEM;
+		goto end;
+	}
+
+	if (of_property_read_bool(np, "spdif-out")) {
+		data->dai[num_links].name = "S/PDIF TX";
+		data->dai[num_links].stream_name = "S/PDIF PCM Playback";
+		data->dai[num_links].codec_dai_name = "dit-hifi";
+		data->dai[num_links].codec_name = "spdif-dit";
+		data->dai[num_links].cpu_of_node = spdif_np;
+		data->dai[num_links].platform_of_node = spdif_np;
+		num_links++;
+
+		data->txdev = platform_device_register_simple("spdif-dit", -1, NULL, 0);
+		if (IS_ERR(data->txdev)) {
+			ret = PTR_ERR(data->txdev);
+			dev_err(&pdev->dev, "register dit failed: %d\n", ret);
+			goto end;
+		}
+	}
+
+	if (of_property_read_bool(np, "spdif-in")) {
+		data->dai[num_links].name = "S/PDIF RX";
+		data->dai[num_links].stream_name = "S/PDIF PCM Capture";
+		data->dai[num_links].codec_dai_name = "dir-hifi";
+		data->dai[num_links].codec_name = "spdif-dir";
+		data->dai[num_links].cpu_of_node = spdif_np;
+		data->dai[num_links].platform_of_node = spdif_np;
+		num_links++;
+
+		data->rxdev = platform_device_register_simple("spdif-dir", -1, NULL, 0);
+		if (IS_ERR(data->rxdev)) {
+			ret = PTR_ERR(data->rxdev);
+			dev_err(&pdev->dev, "register dir failed: %d\n", ret);
+			goto error_dit;
+		}
+	}
+
+	if (!num_links) {
+		dev_err(&pdev->dev, "no enabled S/PDIF DAI link\n");
+		goto error_dir;
+	}
+
+	data->card.dev = &pdev->dev;
+	data->card.num_links = num_links;
+	data->card.dai_link = data->dai;
+
+	ret = snd_soc_of_parse_card_name(&data->card, "model");
+	if (ret)
+		goto error_dir;
+
+	ret = snd_soc_register_card(&data->card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed: %d\n", ret);
+		goto error_dir;
+	}
+
+	platform_set_drvdata(pdev, data);
+
+	goto end;
+
+error_dir:
+	if (data->rxdev)
+		platform_device_unregister(data->rxdev);
+error_dit:
+	if (data->txdev)
+		platform_device_unregister(data->txdev);
+end:
+	if (spdif_np)
+		of_node_put(spdif_np);
+
+	return ret;
+}
+
+static int imx_spdif_audio_remove(struct platform_device *pdev)
+{
+	struct imx_spdif_data *data = platform_get_drvdata(pdev);
+
+	if (data->rxdev)
+		platform_device_unregister(data->rxdev);
+	if (data->txdev)
+		platform_device_unregister(data->txdev);
+
+	snd_soc_unregister_card(&data->card);
+
+	return 0;
+}
+
+static const struct of_device_id imx_spdif_dt_ids[] = {
+	{ .compatible = "fsl,imx-audio-spdif", },
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, imx_spdif_dt_ids);
+
+static struct platform_driver imx_spdif_driver = {
+	.driver = {
+		.name = "imx-spdif",
+		.owner = THIS_MODULE,
+		.of_match_table = imx_spdif_dt_ids,
+	},
+	.probe = imx_spdif_audio_probe,
+	.remove = imx_spdif_audio_remove,
+};
+
+module_platform_driver(imx_spdif_driver);
+
+MODULE_AUTHOR("Freescale Semiconductor, Inc.");
+MODULE_DESCRIPTION("Freescale i.MX S/PDIF machine driver");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:imx-spdif");
diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c
index 6cf8355a8542..8c49147db84c 100644
--- a/sound/soc/generic/simple-card.c
+++ b/sound/soc/generic/simple-card.c
@@ -105,6 +105,7 @@ static int asoc_simple_card_remove(struct platform_device *pdev)
 static struct platform_driver asoc_simple_card = {
 	.driver = {
 		.name	= "asoc-simple-card",
+		.owner = THIS_MODULE,
 	},
 	.probe		= asoc_simple_card_probe,
 	.remove		= asoc_simple_card_remove,
@@ -112,6 +113,7 @@ static struct platform_driver asoc_simple_card = {
 
 module_platform_driver(asoc_simple_card);
 
+MODULE_ALIAS("platform:asoc-simple-card");
 MODULE_LICENSE("GPL");
 MODULE_DESCRIPTION("ASoC Simple Sound Card");
 MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
diff --git a/sound/soc/kirkwood/Kconfig b/sound/soc/kirkwood/Kconfig
index 9e1970c44e86..78ed4a42ad21 100644
--- a/sound/soc/kirkwood/Kconfig
+++ b/sound/soc/kirkwood/Kconfig
@@ -1,6 +1,6 @@
 config SND_KIRKWOOD_SOC
-	tristate "SoC Audio for the Marvell Kirkwood chip"
-	depends on ARCH_KIRKWOOD || COMPILE_TEST
+	tristate "SoC Audio for the Marvell Kirkwood and Dove chips"
+	depends on ARCH_KIRKWOOD || ARCH_DOVE || COMPILE_TEST
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Kirkwood I2S interface. You will also need to select the
diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c
index e5f3f7a9ea26..7fce340ab3ef 100644
--- a/sound/soc/kirkwood/kirkwood-i2s.c
+++ b/sound/soc/kirkwood/kirkwood-i2s.c
@@ -22,6 +22,8 @@
 #include <sound/pcm_params.h>
 #include <sound/soc.h>
 #include <linux/platform_data/asoc-kirkwood.h>
+#include <linux/of.h>
+
 #include "kirkwood.h"
 
 #define DRV_NAME	"mvebu-audio"
@@ -453,6 +455,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 	struct snd_soc_dai_driver *soc_dai = &kirkwood_i2s_dai;
 	struct kirkwood_dma_data *priv;
 	struct resource *mem;
+	struct device_node *np = pdev->dev.of_node;
 	int err;
 
 	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
@@ -473,14 +476,16 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 		return -ENXIO;
 	}
 
-	if (!data) {
-		dev_err(&pdev->dev, "no platform data ?!\n");
+	if (np) {
+		priv->burst = 128;		/* might be 32 or 128 */
+	} else if (data) {
+		priv->burst = data->burst;
+	} else {
+		dev_err(&pdev->dev, "no DT nor platform data ?!\n");
 		return -EINVAL;
 	}
 
-	priv->burst = data->burst;
-
-	priv->clk = devm_clk_get(&pdev->dev, NULL);
+	priv->clk = devm_clk_get(&pdev->dev, np ? "internal" : NULL);
 	if (IS_ERR(priv->clk)) {
 		dev_err(&pdev->dev, "no clock\n");
 		return PTR_ERR(priv->clk);
@@ -507,7 +512,7 @@ static int kirkwood_i2s_dev_probe(struct platform_device *pdev)
 	priv->ctl_rec = KIRKWOOD_RECCTL_SIZE_24;
 
 	/* Select the burst size */
-	if (data->burst == 32) {
+	if (priv->burst == 32) {
 		priv->ctl_play |= KIRKWOOD_PLAYCTL_BURST_32;
 		priv->ctl_rec |= KIRKWOOD_RECCTL_BURST_32;
 	} else {
@@ -552,12 +557,21 @@ static int kirkwood_i2s_dev_remove(struct platform_device *pdev)
 	return 0;
 }
 
+#ifdef CONFIG_OF
+static struct of_device_id mvebu_audio_of_match[] = {
+	{ .compatible = "marvell,mvebu-audio" },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, mvebu_audio_of_match);
+#endif
+
 static struct platform_driver kirkwood_i2s_driver = {
 	.probe  = kirkwood_i2s_dev_probe,
 	.remove = kirkwood_i2s_dev_remove,
 	.driver = {
 		.name = DRV_NAME,
 		.owner = THIS_MODULE,
+		.of_match_table = of_match_ptr(mvebu_audio_of_match),
 	},
 };
 
diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c
index ce084eb10c49..4bb273786ff3 100644
--- a/sound/soc/mxs/mxs-sgtl5000.c
+++ b/sound/soc/mxs/mxs-sgtl5000.c
@@ -105,11 +105,13 @@ static struct snd_soc_dai_link mxs_sgtl5000_dai[] = {
 		.stream_name	= "HiFi Playback",
 		.codec_dai_name	= "sgtl5000",
 		.ops		= &mxs_sgtl5000_hifi_ops,
+		.playback_only	= true,
 	}, {
 		.name		= "HiFi Rx",
 		.stream_name	= "HiFi Capture",
 		.codec_dai_name	= "sgtl5000",
 		.ops		= &mxs_sgtl5000_hifi_ops,
+		.capture_only	= true,
 	},
 };
 
diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c
index 361e4c03646e..83433fdea32a 100644
--- a/sound/soc/omap/mcbsp.c
+++ b/sound/soc/omap/mcbsp.c
@@ -781,7 +781,7 @@ static ssize_t prop##_store(struct device *dev,				\
 	unsigned long val;						\
 	int status;							\
 									\
-	status = strict_strtoul(buf, 0, &val);				\
+	status = kstrtoul(buf, 0, &val);				\
 	if (status)							\
 		return status;						\
 									\
diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c
index a0c67f60f594..9338d11e9216 100644
--- a/sound/soc/samsung/dma.c
+++ b/sound/soc/samsung/dma.c
@@ -90,6 +90,13 @@ static void dma_enqueue(struct snd_pcm_substream *substream)
 	dma_info.period = prtd->dma_period;
 	dma_info.len = prtd->dma_period*limit;
 
+	if (dma_info.cap == DMA_CYCLIC) {
+		dma_info.buf = pos;
+		prtd->params->ops->prepare(prtd->params->ch, &dma_info);
+		prtd->dma_loaded += limit;
+		return;
+	}
+
 	while (prtd->dma_loaded < limit) {
 		pr_debug("dma_loaded: %d\n", prtd->dma_loaded);
 
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 30390260bb67..b33ca7cd085b 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -235,6 +235,8 @@ struct fsi_stream {
 	struct sh_dmae_slave	slave; /* see fsi_handler_init() */
 	struct work_struct	work;
 	dma_addr_t		dma;
+	int			loop_cnt;
+	int			additional_pos;
 };
 
 struct fsi_clk {
@@ -1289,6 +1291,8 @@ static int fsi_dma_init(struct fsi_priv *fsi, struct fsi_stream *io)
 	io->bus_option = BUSOP_SET(24, PACKAGE_24BITBUS_BACK) |
 			 BUSOP_SET(16, PACKAGE_16BITBUS_STREAM);
 
+	io->loop_cnt = 2; /* push 1st, 2nd period first, then 3rd, 4th... */
+	io->additional_pos = 0;
 	io->dma = dma_map_single(dai->dev, runtime->dma_area,
 				 snd_pcm_lib_buffer_bytes(io->substream), dir);
 	return 0;
@@ -1305,11 +1309,15 @@ static int fsi_dma_quit(struct fsi_priv *fsi, struct fsi_stream *io)
 	return 0;
 }
 
-static dma_addr_t fsi_dma_get_area(struct fsi_stream *io)
+static dma_addr_t fsi_dma_get_area(struct fsi_stream *io, int additional)
 {
 	struct snd_pcm_runtime *runtime = io->substream->runtime;
+	int period = io->period_pos + additional;
 
-	return io->dma + samples_to_bytes(runtime, io->buff_sample_pos);
+	if (period >= runtime->periods)
+		period = 0;
+
+	return io->dma + samples_to_bytes(runtime, period * io->period_samples);
 }
 
 static void fsi_dma_complete(void *data)
@@ -1321,7 +1329,7 @@ static void fsi_dma_complete(void *data)
 	enum dma_data_direction dir = fsi_stream_is_play(fsi, io) ?
 		DMA_TO_DEVICE : DMA_FROM_DEVICE;
 
-	dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io),
+	dma_sync_single_for_cpu(dai->dev, fsi_dma_get_area(io, 0),
 			samples_to_bytes(runtime, io->period_samples), dir);
 
 	io->buff_sample_pos += io->period_samples;
@@ -1347,7 +1355,7 @@ static void fsi_dma_do_work(struct work_struct *work)
 	struct snd_pcm_runtime *runtime;
 	enum dma_data_direction dir;
 	int is_play = fsi_stream_is_play(fsi, io);
-	int len;
+	int len, i;
 	dma_addr_t buf;
 
 	if (!fsi_stream_is_working(fsi, io))
@@ -1357,26 +1365,33 @@ static void fsi_dma_do_work(struct work_struct *work)
 	runtime	= io->substream->runtime;
 	dir	= is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE;
 	len	= samples_to_bytes(runtime, io->period_samples);
-	buf	= fsi_dma_get_area(io);
 
-	dma_sync_single_for_device(dai->dev, buf, len, dir);
+	for (i = 0; i < io->loop_cnt; i++) {
+		buf	= fsi_dma_get_area(io, io->additional_pos);
 
-	desc = dmaengine_prep_slave_single(io->chan, buf, len, dir,
-					   DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
-	if (!desc) {
-		dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
-		return;
-	}
+		dma_sync_single_for_device(dai->dev, buf, len, dir);
 
-	desc->callback		= fsi_dma_complete;
-	desc->callback_param	= io;
+		desc = dmaengine_prep_slave_single(io->chan, buf, len, dir,
+					DMA_PREP_INTERRUPT | DMA_CTRL_ACK);
+		if (!desc) {
+			dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n");
+			return;
+		}
 
-	if (dmaengine_submit(desc) < 0) {
-		dev_err(dai->dev, "tx_submit() fail\n");
-		return;
+		desc->callback		= fsi_dma_complete;
+		desc->callback_param	= io;
+
+		if (dmaengine_submit(desc) < 0) {
+			dev_err(dai->dev, "tx_submit() fail\n");
+			return;
+		}
+
+		dma_async_issue_pending(io->chan);
+
+		io->additional_pos = 1;
 	}
 
-	dma_async_issue_pending(io->chan);
+	io->loop_cnt = 1;
 
 	/*
 	 * FIXME
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index 528f8708221d..4d0561312f3b 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -203,7 +203,7 @@ static ssize_t pmdown_time_set(struct device *dev,
 	struct snd_soc_pcm_runtime *rtd = dev_get_drvdata(dev);
 	int ret;
 
-	ret = strict_strtol(buf, 10, &rtd->pmdown_time);
+	ret = kstrtol(buf, 10, &rtd->pmdown_time);
 	if (ret)
 		return ret;
 
@@ -248,6 +248,7 @@ static ssize_t codec_reg_write_file(struct file *file,
 	char *start = buf;
 	unsigned long reg, value;
 	struct snd_soc_codec *codec = file->private_data;
+	int ret;
 
 	buf_size = min(count, (sizeof(buf)-1));
 	if (copy_from_user(buf, user_buf, buf_size))
@@ -259,8 +260,9 @@ static ssize_t codec_reg_write_file(struct file *file,
 	reg = simple_strtoul(start, &start, 16);
 	while (*start == ' ')
 		start++;
-	if (strict_strtoul(start, 16, &value))
-		return -EINVAL;
+	ret = kstrtoul(start, 16, &value);
+	if (ret)
+		return ret;
 
 	/* Userspace has been fiddling around behind the kernel's back */
 	add_taint(TAINT_USER, LOCKDEP_NOW_UNRELIABLE);
@@ -1243,9 +1245,6 @@ static int soc_post_component_init(struct snd_soc_card *card,
 	}
 	rtd->card = card;
 
-	/* Make sure all DAPM widgets are instantiated */
-	snd_soc_dapm_new_widgets(&codec->dapm);
-
 	/* machine controls, routes and widgets are not prefixed */
 	temp = codec->name_prefix;
 	codec->name_prefix = NULL;
@@ -1741,8 +1740,6 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 		snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes,
 					card->num_dapm_routes);
 
-	snd_soc_dapm_new_widgets(&card->dapm);
-
 	for (i = 0; i < card->num_links; i++) {
 		dai_link = &card->dai_link[i];
 		dai_fmt = dai_link->dai_fmt;
@@ -1821,12 +1818,12 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card)
 		}
 	}
 
-	snd_soc_dapm_new_widgets(&card->dapm);
-
 	if (card->fully_routed)
 		list_for_each_entry(codec, &card->codec_dev_list, card_list)
 			snd_soc_dapm_auto_nc_codec_pins(codec);
 
+	snd_soc_dapm_new_widgets(card);
+
 	ret = snd_card_register(card->snd_card);
 	if (ret < 0) {
 		dev_err(card->dev, "ASoC: failed to register soundcard %d\n",
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index d84bd0f167b6..c17c14c394df 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -229,6 +229,8 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget,
 			template.id = snd_soc_dapm_kcontrol;
 			template.name = kcontrol->id.name;
 
+			data->value = template.on_val;
+
 			data->widget = snd_soc_dapm_new_control(widget->dapm,
 				&template);
 			if (!data->widget) {
@@ -2374,6 +2376,9 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
 			wsource->ext = 1;
 	}
 
+	dapm_mark_dirty(wsource, "Route added");
+	dapm_mark_dirty(wsink, "Route added");
+
 	/* connect static paths */
 	if (control == NULL) {
 		list_add(&path->list, &dapm->card->paths);
@@ -2436,9 +2441,6 @@ static int snd_soc_dapm_add_path(struct snd_soc_dapm_context *dapm,
 		return 0;
 	}
 
-	dapm_mark_dirty(wsource, "Route added");
-	dapm_mark_dirty(wsink, "Route added");
-
 	return 0;
 err:
 	kfree(path);
@@ -2712,9 +2714,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes);
  *
  * Returns 0 for success.
  */
-int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm)
+int snd_soc_dapm_new_widgets(struct snd_soc_card *card)
 {
-	struct snd_soc_card *card = dapm->card;
 	struct snd_soc_dapm_widget *w;
 	unsigned int val;
 
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
index 7aa26b5178aa..71358e3b54d9 100644
--- a/sound/soc/soc-jack.c
+++ b/sound/soc/soc-jack.c
@@ -183,8 +183,6 @@ int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count,
 		list_add(&(pins[i].list), &jack->pins);
 	}
 
-	snd_soc_dapm_new_widgets(&jack->codec->card->dapm);
-
 	/* Update to reflect the last reported status; canned jack
 	 * implementations are likely to set their state before the
 	 * card has an opportunity to associate pins.
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index fb70fbe26862..330c9a6b5cb5 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -2020,6 +2020,16 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num)
 			capture = 1;
 	}
 
+	if (rtd->dai_link->playback_only) {
+		playback = 1;
+		capture = 0;
+	}
+
+	if (rtd->dai_link->capture_only) {
+		playback = 0;
+		capture = 1;
+	}
+
 	/* create the PCM */
 	if (rtd->dai_link->no_pcm) {
 		snprintf(new_name, sizeof(new_name), "(%s)",
diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c
index b9defcdeb7ef..780bf3f62d28 100644
--- a/sound/usb/6fire/firmware.c
+++ b/sound/usb/6fire/firmware.c
@@ -346,10 +346,10 @@ static int usb6fire_fw_check(u8 *version)
 		if (!memcmp(version, known_fw_versions + i, 2))
 			return 0;
 
-	snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. "
+	snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %4ph. "
 			"please reconnect to power. if this failure "
 			"still happens, check your firmware installation.",
-			4, version);
+			version);
 	return -EINVAL;
 }
 
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 659950e5b94f..93e970f2b3c0 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -418,6 +418,9 @@ struct snd_usb_endpoint *snd_usb_add_endpoint(struct snd_usb_audio *chip,
 	struct snd_usb_endpoint *ep;
 	int is_playback = direction == SNDRV_PCM_STREAM_PLAYBACK;
 
+	if (WARN_ON(!alts))
+		return NULL;
+
 	mutex_lock(&chip->mutex);
 
 	list_for_each_entry(ep, &chip->ep_list, list) {
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 15b151ed4899..b375d58871e7 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -327,6 +327,137 @@ static int search_roland_implicit_fb(struct usb_device *dev, int ifnum,
 	return 0;
 }
 
+static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs,
+					 struct usb_device *dev,
+					 struct usb_interface_descriptor *altsd,
+					 unsigned int attr)
+{
+	struct usb_host_interface *alts;
+	struct usb_interface *iface;
+	unsigned int ep;
+
+	/* Implicit feedback sync EPs consumers are always playback EPs */
+	if (subs->direction != SNDRV_PCM_STREAM_PLAYBACK)
+		return 0;
+
+	switch (subs->stream->chip->usb_id) {
+	case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */
+	case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */
+		ep = 0x81;
+		iface = usb_ifnum_to_if(dev, 3);
+
+		if (!iface || iface->num_altsetting == 0)
+			return -EINVAL;
+
+		alts = &iface->altsetting[1];
+		goto add_sync_ep;
+		break;
+	case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */
+	case USB_ID(0x0763, 0x2081):
+		ep = 0x81;
+		iface = usb_ifnum_to_if(dev, 2);
+
+		if (!iface || iface->num_altsetting == 0)
+			return -EINVAL;
+
+		alts = &iface->altsetting[1];
+		goto add_sync_ep;
+	}
+	if (attr == USB_ENDPOINT_SYNC_ASYNC &&
+	    altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
+	    altsd->bInterfaceProtocol == 2 &&
+	    altsd->bNumEndpoints == 1 &&
+	    USB_ID_VENDOR(subs->stream->chip->usb_id) == 0x0582 /* Roland */ &&
+	    search_roland_implicit_fb(dev, altsd->bInterfaceNumber + 1,
+				      altsd->bAlternateSetting,
+				      &alts, &ep) >= 0) {
+		goto add_sync_ep;
+	}
+
+	/* No quirk */
+	return 0;
+
+add_sync_ep:
+	subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
+						   alts, ep, !subs->direction,
+						   SND_USB_ENDPOINT_TYPE_DATA);
+	if (!subs->sync_endpoint)
+		return -EINVAL;
+
+	subs->data_endpoint->sync_master = subs->sync_endpoint;
+
+	return 0;
+}
+
+static int set_sync_endpoint(struct snd_usb_substream *subs,
+			     struct audioformat *fmt,
+			     struct usb_device *dev,
+			     struct usb_host_interface *alts,
+			     struct usb_interface_descriptor *altsd)
+{
+	int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
+	unsigned int ep, attr;
+	bool implicit_fb;
+	int err;
+
+	/* we need a sync pipe in async OUT or adaptive IN mode */
+	/* check the number of EP, since some devices have broken
+	 * descriptors which fool us.  if it has only one EP,
+	 * assume it as adaptive-out or sync-in.
+	 */
+	attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE;
+
+	err = set_sync_ep_implicit_fb_quirk(subs, dev, altsd, attr);
+	if (err < 0)
+		return err;
+
+	if (altsd->bNumEndpoints < 2)
+		return 0;
+
+	if ((is_playback && attr != USB_ENDPOINT_SYNC_ASYNC) ||
+	    (!is_playback && attr != USB_ENDPOINT_SYNC_ADAPTIVE))
+		return 0;
+
+	/* check sync-pipe endpoint */
+	/* ... and check descriptor size before accessing bSynchAddress
+	   because there is a version of the SB Audigy 2 NX firmware lacking
+	   the audio fields in the endpoint descriptors */
+	if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC ||
+	    (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+	     get_endpoint(alts, 1)->bSynchAddress != 0)) {
+		snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n",
+			   dev->devnum, fmt->iface, fmt->altsetting,
+			   get_endpoint(alts, 1)->bmAttributes,
+			   get_endpoint(alts, 1)->bLength,
+			   get_endpoint(alts, 1)->bSynchAddress);
+		return -EINVAL;
+	}
+	ep = get_endpoint(alts, 1)->bEndpointAddress;
+	if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
+	    ((is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
+	     (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
+		snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n",
+			   dev->devnum, fmt->iface, fmt->altsetting,
+			   is_playback, ep, get_endpoint(alts, 0)->bSynchAddress);
+		return -EINVAL;
+	}
+
+	implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK)
+			== USB_ENDPOINT_USAGE_IMPLICIT_FB;
+
+	subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
+						   alts, ep, !subs->direction,
+						   implicit_fb ?
+							SND_USB_ENDPOINT_TYPE_DATA :
+							SND_USB_ENDPOINT_TYPE_SYNC);
+	if (!subs->sync_endpoint)
+		return -EINVAL;
+
+	subs->data_endpoint->sync_master = subs->sync_endpoint;
+
+	return 0;
+}
+
 /*
  * find a matching format and set up the interface
  */
@@ -336,9 +467,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
 	struct usb_host_interface *alts;
 	struct usb_interface_descriptor *altsd;
 	struct usb_interface *iface;
-	unsigned int ep, attr;
-	int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
-	int err, implicit_fb = 0;
+	int err;
 
 	iface = usb_ifnum_to_if(dev, fmt->iface);
 	if (WARN_ON(!iface))
@@ -383,118 +512,22 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
 	subs->data_endpoint = snd_usb_add_endpoint(subs->stream->chip,
 						   alts, fmt->endpoint, subs->direction,
 						   SND_USB_ENDPOINT_TYPE_DATA);
+
 	if (!subs->data_endpoint)
 		return -EINVAL;
 
-	/* we need a sync pipe in async OUT or adaptive IN mode */
-	/* check the number of EP, since some devices have broken
-	 * descriptors which fool us.  if it has only one EP,
-	 * assume it as adaptive-out or sync-in.
-	 */
-	attr = fmt->ep_attr & USB_ENDPOINT_SYNCTYPE;
-
-	switch (subs->stream->chip->usb_id) {
-	case USB_ID(0x0763, 0x2030): /* M-Audio Fast Track C400 */
-	case USB_ID(0x0763, 0x2031): /* M-Audio Fast Track C600 */
-		if (is_playback) {
-			implicit_fb = 1;
-			ep = 0x81;
-			iface = usb_ifnum_to_if(dev, 3);
-
-			if (!iface || iface->num_altsetting == 0)
-				return -EINVAL;
-
-			alts = &iface->altsetting[1];
-			goto add_sync_ep;
-		}
-		break;
-	case USB_ID(0x0763, 0x2080): /* M-Audio FastTrack Ultra */
-	case USB_ID(0x0763, 0x2081):
-		if (is_playback) {
-			implicit_fb = 1;
-			ep = 0x81;
-			iface = usb_ifnum_to_if(dev, 2);
-
-			if (!iface || iface->num_altsetting == 0)
-				return -EINVAL;
-
-			alts = &iface->altsetting[1];
-			goto add_sync_ep;
-		}
-	}
-	if (is_playback &&
-	    attr == USB_ENDPOINT_SYNC_ASYNC &&
-	    altsd->bInterfaceClass == USB_CLASS_VENDOR_SPEC &&
-	    altsd->bInterfaceProtocol == 2 &&
-	    altsd->bNumEndpoints == 1 &&
-	    USB_ID_VENDOR(subs->stream->chip->usb_id) == 0x0582 /* Roland */ &&
-	    search_roland_implicit_fb(dev, altsd->bInterfaceNumber + 1,
-				      altsd->bAlternateSetting,
-				      &alts, &ep) >= 0) {
-		implicit_fb = 1;
-		goto add_sync_ep;
-	}
-
-	if (((is_playback && attr == USB_ENDPOINT_SYNC_ASYNC) ||
-	     (!is_playback && attr == USB_ENDPOINT_SYNC_ADAPTIVE)) &&
-	    altsd->bNumEndpoints >= 2) {
-		/* check sync-pipe endpoint */
-		/* ... and check descriptor size before accessing bSynchAddress
-		   because there is a version of the SB Audigy 2 NX firmware lacking
-		   the audio fields in the endpoint descriptors */
-		if ((get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_XFERTYPE_MASK) != USB_ENDPOINT_XFER_ISOC ||
-		    (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
-		     get_endpoint(alts, 1)->bSynchAddress != 0 &&
-		     !implicit_fb)) {
-			snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n",
-				   dev->devnum, fmt->iface, fmt->altsetting,
-				   get_endpoint(alts, 1)->bmAttributes,
-				   get_endpoint(alts, 1)->bLength,
-				   get_endpoint(alts, 1)->bSynchAddress);
-			return -EINVAL;
-		}
-		ep = get_endpoint(alts, 1)->bEndpointAddress;
-		if (!implicit_fb &&
-		    get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE &&
-		    (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) ||
-		     (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) {
-			snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n",
-				   dev->devnum, fmt->iface, fmt->altsetting,
-				   is_playback, ep, get_endpoint(alts, 0)->bSynchAddress);
-			return -EINVAL;
-		}
-
-		implicit_fb = (get_endpoint(alts, 1)->bmAttributes & USB_ENDPOINT_USAGE_MASK)
-				== USB_ENDPOINT_USAGE_IMPLICIT_FB;
-
-add_sync_ep:
-		subs->sync_endpoint = snd_usb_add_endpoint(subs->stream->chip,
-							   alts, ep, !subs->direction,
-							   implicit_fb ?
-								SND_USB_ENDPOINT_TYPE_DATA :
-								SND_USB_ENDPOINT_TYPE_SYNC);
-		if (!subs->sync_endpoint)
-			return -EINVAL;
-
-		subs->data_endpoint->sync_master = subs->sync_endpoint;
-	}
+	err = set_sync_endpoint(subs, fmt, dev, alts, altsd);
+	if (err < 0)
+		return err;
 
-	if ((err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt)) < 0)
+	err = snd_usb_init_pitch(subs->stream->chip, fmt->iface, alts, fmt);
+	if (err < 0)
 		return err;
 
 	subs->cur_audiofmt = fmt;
 
 	snd_usb_set_format_quirk(subs, fmt);
 
-#if 0
-	printk(KERN_DEBUG
-	       "setting done: format = %d, rate = %d..%d, channels = %d\n",
-	       fmt->format, fmt->rate_min, fmt->rate_max, fmt->channels);
-	printk(KERN_DEBUG
-	       "  datapipe = 0x%0x, syncpipe = 0x%0x\n",
-	       subs->datapipe, subs->syncpipe);
-#endif
-
 	return 0;
 }
 
diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c
index 1f9bbd55553f..5a51b18c50fe 100644
--- a/sound/usb/usx2y/usbusx2y.c
+++ b/sound/usb/usx2y/usbusx2y.c
@@ -305,11 +305,9 @@ static void usX2Y_unlinkSeq(struct snd_usX2Y_AsyncSeq *S)
 {
 	int	i;
 	for (i = 0; i < URBS_AsyncSeq; ++i) {
-		if (S[i].urb) {
-			usb_kill_urb(S->urb[i]);
-			usb_free_urb(S->urb[i]);
-			S->urb[i] = NULL;
-		}
+		usb_kill_urb(S->urb[i]);
+		usb_free_urb(S->urb[i]);
+		S->urb[i] = NULL;
 	}
 	kfree(S->buffer);
 }