Merge branch 'linus' into cpumask-for-linus

Conflicts:
	arch/x86/kernel/cpu/common.c

379 files changed, 20041 insertions, 9451 deletions

diff --git a/sound/Kconfig b/sound/Kconfig
index 200aca1faa71..1eceb85287c5 100644
--- a/sound/Kconfig
+++ b/sound/Kconfig
@@ -60,6 +60,8 @@ source "sound/aoa/Kconfig"
 
 source "sound/arm/Kconfig"
 
+source "sound/atmel/Kconfig"
+
 source "sound/spi/Kconfig"
 
 source "sound/mips/Kconfig"
diff --git a/sound/Makefile b/sound/Makefile
index c76d70716fa5..ec467decfa79 100644
--- a/sound/Makefile
+++ b/sound/Makefile
@@ -6,7 +6,7 @@ obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o
 obj-$(CONFIG_SOUND_PRIME) += oss/
 obj-$(CONFIG_DMASOUND) += oss/
 obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ \
-	sparc/ spi/ parisc/ pcmcia/ mips/ soc/
+	sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/
 obj-$(CONFIG_SND_AOA) += aoa/
 
 # This one must be compilable even if sound is configured out
diff --git a/sound/aoa/aoa-gpio.h b/sound/aoa/aoa-gpio.h
index ee64f5de8966..6065b0344e23 100644
--- a/sound/aoa/aoa-gpio.h
+++ b/sound/aoa/aoa-gpio.h
@@ -34,10 +34,12 @@ struct gpio_methods {
 	void (*set_headphone)(struct gpio_runtime *rt, int on);
 	void (*set_speakers)(struct gpio_runtime *rt, int on);
 	void (*set_lineout)(struct gpio_runtime *rt, int on);
+	void (*set_master)(struct gpio_runtime *rt, int on);
 
 	int (*get_headphone)(struct gpio_runtime *rt);
 	int (*get_speakers)(struct gpio_runtime *rt);
 	int (*get_lineout)(struct gpio_runtime *rt);
+	int (*get_master)(struct gpio_runtime *rt);
 
 	void (*set_hw_reset)(struct gpio_runtime *rt, int on);
 
diff --git a/sound/aoa/core/alsa.c b/sound/aoa/core/alsa.c
index 617850463582..0fa3855b4790 100644
--- a/sound/aoa/core/alsa.c
+++ b/sound/aoa/core/alsa.c
@@ -23,9 +23,10 @@ int aoa_alsa_init(char *name, struct module *mod, struct device *dev)
 		/* cannot be EEXIST due to usage in aoa_fabric_register */
 		return -EBUSY;
 
-	alsa_card = snd_card_new(index, name, mod, sizeof(struct aoa_card));
-	if (!alsa_card)
-		return -ENOMEM;
+	err = snd_card_create(index, name, mod, sizeof(struct aoa_card),
+			      &alsa_card);
+	if (err < 0)
+		return err;
 	aoa_card = alsa_card->private_data;
 	aoa_card->alsa_card = alsa_card;
 	alsa_card->dev = dev;
diff --git a/sound/aoa/core/gpio-feature.c b/sound/aoa/core/gpio-feature.c
index c93ad5dec66b..de8e03afa97b 100644
--- a/sound/aoa/core/gpio-feature.c
+++ b/sound/aoa/core/gpio-feature.c
@@ -14,7 +14,7 @@
 #include <linux/interrupt.h>
 #include "../aoa.h"
 
-/* TODO: these are 20 global variables
+/* TODO: these are lots of global variables
  * that aren't used on most machines...
  * Move them into a dynamically allocated
  * structure and use that.
@@ -23,6 +23,7 @@
 /* these are the GPIO numbers (register addresses as offsets into
  * the GPIO space) */
 static int headphone_mute_gpio;
+static int master_mute_gpio;
 static int amp_mute_gpio;
 static int lineout_mute_gpio;
 static int hw_reset_gpio;
@@ -32,6 +33,7 @@ static int linein_detect_gpio;
 
 /* see the SWITCH_GPIO macro */
 static int headphone_mute_gpio_activestate;
+static int master_mute_gpio_activestate;
 static int amp_mute_gpio_activestate;
 static int lineout_mute_gpio_activestate;
 static int hw_reset_gpio_activestate;
@@ -156,6 +158,7 @@ static int ftr_gpio_get_##name(struct gpio_runtime *rt)		\
 FTR_GPIO(headphone, 0);
 FTR_GPIO(amp, 1);
 FTR_GPIO(lineout, 2);
+FTR_GPIO(master, 3);
 
 static void ftr_gpio_set_hw_reset(struct gpio_runtime *rt, int on)
 {
@@ -172,6 +175,8 @@ static void ftr_gpio_set_hw_reset(struct gpio_runtime *rt, int on)
 			  hw_reset_gpio, v);
 }
 
+static struct gpio_methods methods;
+
 static void ftr_gpio_all_amps_off(struct gpio_runtime *rt)
 {
 	int saved;
@@ -181,6 +186,8 @@ static void ftr_gpio_all_amps_off(struct gpio_runtime *rt)
 	ftr_gpio_set_headphone(rt, 0);
 	ftr_gpio_set_amp(rt, 0);
 	ftr_gpio_set_lineout(rt, 0);
+	if (methods.set_master)
+		ftr_gpio_set_master(rt, 0);
 	rt->implementation_private = saved;
 }
 
@@ -193,6 +200,8 @@ static void ftr_gpio_all_amps_restore(struct gpio_runtime *rt)
 	ftr_gpio_set_headphone(rt, (s>>0)&1);
 	ftr_gpio_set_amp(rt, (s>>1)&1);
 	ftr_gpio_set_lineout(rt, (s>>2)&1);
+	if (methods.set_master)
+		ftr_gpio_set_master(rt, (s>>3)&1);
 }
 
 static void ftr_handle_notify(struct work_struct *work)
@@ -231,6 +240,12 @@ static void ftr_gpio_init(struct gpio_runtime *rt)
 	get_gpio("hw-reset", "audio-hw-reset",
 		 &hw_reset_gpio,
 		 &hw_reset_gpio_activestate);
+	if (get_gpio("master-mute", NULL,
+		     &master_mute_gpio,
+		     &master_mute_gpio_activestate)) {
+		methods.set_master = ftr_gpio_set_master;
+		methods.get_master = ftr_gpio_get_master;
+	}
 
 	headphone_detect_node = get_gpio("headphone-detect", NULL,
 					 &headphone_detect_gpio,
diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c
index ad60f5d10e82..fbf5c933baa4 100644
--- a/sound/aoa/fabrics/layout.c
+++ b/sound/aoa/fabrics/layout.c
@@ -1,16 +1,14 @@
 /*
- * Apple Onboard Audio driver -- layout fabric
+ * Apple Onboard Audio driver -- layout/machine id fabric
  *
- * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ * Copyright 2006-2008 Johannes Berg <johannes@sipsolutions.net>
  *
  * GPL v2, can be found in COPYING.
  *
  *
- * This fabric module looks for sound codecs
- * based on the layout-id property in the device tree.
- *
+ * This fabric module looks for sound codecs based on the
+ * layout-id or device-id property in the device tree.
  */
-
 #include <asm/prom.h>
 #include <linux/list.h>
 #include <linux/module.h>
@@ -63,7 +61,7 @@ struct codec_connect_info {
 #define LAYOUT_FLAG_COMBO_LINEOUT_SPDIF	(1<<0)
 
 struct layout {
-	unsigned int layout_id;
+	unsigned int layout_id, device_id;
 	struct codec_connect_info codecs[MAX_CODECS_PER_BUS];
 	int flags;
 
@@ -111,6 +109,10 @@ MODULE_ALIAS("sound-layout-96");
 MODULE_ALIAS("sound-layout-98");
 MODULE_ALIAS("sound-layout-100");
 
+MODULE_ALIAS("aoa-device-id-14");
+MODULE_ALIAS("aoa-device-id-22");
+MODULE_ALIAS("aoa-device-id-35");
+
 /* onyx with all but microphone connected */
 static struct codec_connection onyx_connections_nomic[] = {
 	{
@@ -518,6 +520,27 @@ static struct layout layouts[] = {
 		.connections = onyx_connections_noheadphones,
 	  },
 	},
+	/* PowerMac3,4 */
+	{ .device_id = 14,
+	  .codecs[0] = {
+		.name = "tas",
+		.connections = tas_connections_noline,
+	  },
+	},
+	/* PowerMac3,6 */
+	{ .device_id = 22,
+	  .codecs[0] = {
+		.name = "tas",
+		.connections = tas_connections_all,
+	  },
+	},
+	/* PowerBook5,2 */
+	{ .device_id = 35,
+	  .codecs[0] = {
+		.name = "tas",
+		.connections = tas_connections_all,
+	  },
+	},
 	{}
 };
 
@@ -526,7 +549,7 @@ static struct layout *find_layout_by_id(unsigned int id)
 	struct layout *l;
 
 	l = layouts;
-	while (l->layout_id) {
+	while (l->codecs[0].name) {
 		if (l->layout_id == id)
 			return l;
 		l++;
@@ -534,6 +557,19 @@ static struct layout *find_layout_by_id(unsigned int id)
 	return NULL;
 }
 
+static struct layout *find_layout_by_device(unsigned int id)
+{
+	struct layout *l;
+
+	l = layouts;
+	while (l->codecs[0].name) {
+		if (l->device_id == id)
+			return l;
+		l++;
+	}
+	return NULL;
+}
+
 static void use_layout(struct layout *l)
 {
 	int i;
@@ -564,6 +600,7 @@ struct layout_dev {
 	struct snd_kcontrol *headphone_ctrl;
 	struct snd_kcontrol *lineout_ctrl;
 	struct snd_kcontrol *speaker_ctrl;
+	struct snd_kcontrol *master_ctrl;
 	struct snd_kcontrol *headphone_detected_ctrl;
 	struct snd_kcontrol *lineout_detected_ctrl;
 
@@ -615,6 +652,7 @@ static struct snd_kcontrol_new n##_ctl = {				\
 AMP_CONTROL(headphone, "Headphone Switch");
 AMP_CONTROL(speakers, "Speakers Switch");
 AMP_CONTROL(lineout, "Line-Out Switch");
+AMP_CONTROL(master, "Master Switch");
 
 static int detect_choice_get(struct snd_kcontrol *kcontrol,
 			     struct snd_ctl_elem_value *ucontrol)
@@ -855,6 +893,11 @@ static void layout_attached_codec(struct aoa_codec *codec)
  	lineout = codec->gpio->methods->get_detect(codec->gpio,
 						   AOA_NOTIFY_LINE_OUT);
 
+	if (codec->gpio->methods->set_master) {
+		ctl = snd_ctl_new1(&master_ctl, codec->gpio);
+		ldev->master_ctrl = ctl;
+		aoa_snd_ctl_add(ctl);
+	}
 	while (cc->connected) {
 		if (cc->connected & CC_SPEAKERS) {
 			if (headphones <= 0 && lineout <= 0)
@@ -938,8 +981,8 @@ static struct aoa_fabric layout_fabric = {
 static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
 {
 	struct device_node *sound = NULL;
-	const unsigned int *layout_id;
-	struct layout *layout;
+	const unsigned int *id;
+	struct layout *layout = NULL;
 	struct layout_dev *ldev = NULL;
 	int err;
 
@@ -952,15 +995,18 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
 		if (sound->type && strcasecmp(sound->type, "soundchip") == 0)
 			break;
 	}
-	if (!sound) return -ENODEV;
+	if (!sound)
+		return -ENODEV;
 
-	layout_id = of_get_property(sound, "layout-id", NULL);
-	if (!layout_id)
-		goto outnodev;
-	printk(KERN_INFO "snd-aoa-fabric-layout: found bus with layout %d\n",
-	       *layout_id);
+	id = of_get_property(sound, "layout-id", NULL);
+	if (id) {
+		layout = find_layout_by_id(*id);
+	} else {
+		id = of_get_property(sound, "device-id", NULL);
+		if (id)
+			layout = find_layout_by_device(*id);
+	}
 
-	layout = find_layout_by_id(*layout_id);
 	if (!layout) {
 		printk(KERN_ERR "snd-aoa-fabric-layout: unknown layout\n");
 		goto outnodev;
@@ -976,6 +1022,7 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
 	ldev->layout = layout;
 	ldev->gpio.node = sound->parent;
 	switch (layout->layout_id) {
+	case 0:  /* anything with device_id, not layout_id */
 	case 41: /* that unknown machine no one seems to have */
 	case 51: /* PowerBook5,4 */
 	case 58: /* Mac Mini */
diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c
index be468edf3ecb..418c84c99d69 100644
--- a/sound/aoa/soundbus/i2sbus/core.c
+++ b/sound/aoa/soundbus/i2sbus/core.c
@@ -1,7 +1,7 @@
 /*
  * i2sbus driver
  *
- * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ * Copyright 2006-2008 Johannes Berg <johannes@sipsolutions.net>
  *
  * GPL v2, can be found in COPYING.
  */
@@ -186,13 +186,25 @@ static int i2sbus_add_dev(struct macio_dev *macio,
 		}
 	}
 	if (i == 1) {
-		const u32 *layout_id =
-			of_get_property(sound, "layout-id", NULL);
-		if (layout_id) {
-			layout = *layout_id;
+		const u32 *id = of_get_property(sound, "layout-id", NULL);
+
+		if (id) {
+			layout = *id;
 			snprintf(dev->sound.modalias, 32,
 				 "sound-layout-%d", layout);
 			ok = 1;
+		} else {
+			id = of_get_property(sound, "device-id", NULL);
+			/*
+			 * We probably cannot handle all device-id machines,
+			 * so restrict to those we do handle for now.
+			 */
+			if (id && (*id == 22 || *id == 14 || *id == 35)) {
+				snprintf(dev->sound.modalias, 32,
+					 "aoa-device-id-%d", *id);
+				ok = 1;
+				layout = -1;
+			}
 		}
 	}
 	/* for the time being, until we can handle non-layout-id
diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig
index f8e6de48d816..885683a3b0bd 100644
--- a/sound/arm/Kconfig
+++ b/sound/arm/Kconfig
@@ -11,17 +11,6 @@ menuconfig SND_ARM
 
 if SND_ARM
 
-config SND_SA11XX_UDA1341
-	tristate "SA11xx UDA1341TS driver (iPaq H3600)"
-	depends on ARCH_SA1100 && L3
-	select SND_PCM
-	help
-	  Say Y here if you have a Compaq iPaq H3x00 handheld computer
-	  and want to use its Philips UDA 1341 audio chip.
-
-	  To compile this driver as a module, choose M here: the module
-	  will be called snd-sa11xx-uda1341.
-
 config SND_ARMAACI
 	tristate "ARM PrimeCell PL041 AC Link support"
 	depends on ARM_AMBA
diff --git a/sound/arm/Makefile b/sound/arm/Makefile
index 2054de11de8a..5a549ed6c8aa 100644
--- a/sound/arm/Makefile
+++ b/sound/arm/Makefile
@@ -2,9 +2,6 @@
 # Makefile for ALSA
 #
 
-obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-sa11xx-uda1341.o 
-snd-sa11xx-uda1341-objs		:= sa11xx-uda1341.o
-
 obj-$(CONFIG_SND_ARMAACI)	+= snd-aaci.o
 snd-aaci-objs			:= aaci.o devdma.o
 
diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c
index 772901e41ecb..7fbd68fab944 100644
--- a/sound/arm/aaci.c
+++ b/sound/arm/aaci.c
@@ -995,10 +995,11 @@ static struct aaci * __devinit aaci_init_card(struct amba_device *dev)
 {
 	struct aaci *aaci;
 	struct snd_card *card;
+	int err;
 
-	card = snd_card_new(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
-			    THIS_MODULE, sizeof(struct aaci));
-	if (card == NULL)
+	err = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
+			      THIS_MODULE, sizeof(struct aaci), &card);
+	if (err < 0)
 		return NULL;
 
 	card->private_free = aaci_free_card;
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c
index 35afd0c33be5..7793d2a511ce 100644
--- a/sound/arm/pxa2xx-ac97-lib.c
+++ b/sound/arm/pxa2xx-ac97-lib.c
@@ -21,7 +21,6 @@
 #include <sound/pxa2xx-lib.h>
 
 #include <asm/irq.h>
-#include <mach/hardware.h>
 #include <mach/regs-ac97.h>
 #include <mach/pxa2xx-gpio.h>
 #include <mach/audio.h>
@@ -31,6 +30,7 @@ static DECLARE_WAIT_QUEUE_HEAD(gsr_wq);
 static volatile long gsr_bits;
 static struct clk *ac97_clk;
 static struct clk *ac97conf_clk;
+static int reset_gpio;
 
 /*
  * Beware PXA27x bugs:
@@ -42,6 +42,45 @@ static struct clk *ac97conf_clk;
  * 1 jiffy timeout if interrupt never comes).
  */
 
+enum {
+	RESETGPIO_FORCE_HIGH,
+	RESETGPIO_FORCE_LOW,
+	RESETGPIO_NORMAL_ALTFUNC
+};
+
+/**
+ * set_resetgpio_mode - computes and sets the AC97_RESET gpio mode on PXA
+ * @mode: chosen action
+ *
+ * As the PXA27x CPUs suffer from a AC97 bug, a manual control of the reset line
+ * must be done to insure proper work of AC97 reset line.  This function
+ * computes the correct gpio_mode for further use by reset functions, and
+ * applied the change through pxa_gpio_mode.
+ */
+static void set_resetgpio_mode(int resetgpio_action)
+{
+	int mode = 0;
+
+	if (reset_gpio)
+		switch (resetgpio_action) {
+		case RESETGPIO_NORMAL_ALTFUNC:
+			if (reset_gpio == 113)
+				mode = 113 | GPIO_OUT | GPIO_DFLT_LOW;
+			if (reset_gpio == 95)
+				mode = 95 | GPIO_ALT_FN_1_OUT;
+			break;
+		case RESETGPIO_FORCE_LOW:
+			mode = reset_gpio | GPIO_OUT | GPIO_DFLT_LOW;
+			break;
+		case RESETGPIO_FORCE_HIGH:
+			mode = reset_gpio | GPIO_OUT | GPIO_DFLT_HIGH;
+			break;
+		};
+
+	if (mode)
+		pxa_gpio_mode(mode);
+}
+
 unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
 {
 	unsigned short val = -1;
@@ -137,10 +176,10 @@ static inline void pxa_ac97_warm_pxa27x(void)
 
 	/* warm reset broken on Bulverde,
 	   so manually keep AC97 reset high */
-	pxa_gpio_mode(113 | GPIO_OUT | GPIO_DFLT_HIGH);
+	set_resetgpio_mode(RESETGPIO_FORCE_HIGH);
 	udelay(10);
 	GCR |= GCR_WARM_RST;
-	pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
+	set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC);
 	udelay(500);
 }
 
@@ -308,8 +347,8 @@ int pxa2xx_ac97_hw_resume(void)
 		pxa_gpio_mode(GPIO29_SDATA_IN_AC97_MD);
 	}
 	if (cpu_is_pxa27x()) {
-		/* Use GPIO 113 as AC97 Reset on Bulverde */
-		pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
+		/* Use GPIO 113 or 95 as AC97 Reset on Bulverde */
+		set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC);
 	}
 	clk_enable(ac97_clk);
 	return 0;
@@ -320,6 +359,27 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_resume);
 int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev)
 {
 	int ret;
+	struct pxa2xx_ac97_platform_data *pdata = dev->dev.platform_data;
+
+	if (pdata) {
+		switch (pdata->reset_gpio) {
+		case 95:
+		case 113:
+			reset_gpio = pdata->reset_gpio;
+			break;
+		case 0:
+			reset_gpio = 113;
+			break;
+		case -1:
+			break;
+		default:
+			dev_err(&dev->dev, "Invalid reset GPIO %d\n",
+				pdata->reset_gpio);
+		}
+	} else {
+		if (cpu_is_pxa27x())
+			reset_gpio = 113;
+	}
 
 	if (cpu_is_pxa25x() || cpu_is_pxa27x()) {
 		pxa_gpio_mode(GPIO31_SYNC_AC97_MD);
@@ -330,7 +390,7 @@ int __devinit pxa2xx_ac97_hw_probe(struct platform_device *dev)
 
 	if (cpu_is_pxa27x()) {
 		/* Use GPIO 113 as AC97 Reset on Bulverde */
-		pxa_gpio_mode(113 | GPIO_ALT_FN_2_OUT);
+		set_resetgpio_mode(RESETGPIO_NORMAL_ALTFUNC);
 		ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK");
 		if (IS_ERR(ac97conf_clk)) {
 			ret = PTR_ERR(ac97conf_clk);
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index 85cf591d4e11..c570ebd9d177 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -20,8 +20,6 @@
 #include <sound/initval.h>
 #include <sound/pxa2xx-lib.h>
 
-#include <mach/hardware.h>
-#include <mach/pxa-regs.h>
 #include <mach/regs-ac97.h>
 #include <mach/audio.h>
 
@@ -173,10 +171,9 @@ static int __devinit pxa2xx_ac97_probe(struct platform_device *dev)
 	struct snd_ac97_template ac97_template;
 	int ret;
 
-	ret = -ENOMEM;
-	card = snd_card_new(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
-			    THIS_MODULE, 0);
-	if (!card)
+	ret = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
+			      THIS_MODULE, 0, &card);
+	if (ret < 0)
 		goto err;
 
 	card->dev = &dev->dev;
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 75a0d746fb60..108b643229ba 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -12,8 +12,7 @@
 #include <sound/pcm_params.h>
 #include <sound/pxa2xx-lib.h>
 
-#include <asm/dma.h>
-#include <mach/pxa-regs.h>
+#include <mach/dma.h>
 
 #include "pxa2xx-pcm.h"
 
diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c
deleted file mode 100644
index 1dcd51d81d10..000000000000
--- a/sound/arm/sa11xx-uda1341.c
+++ /dev/null
@@ -1,983 +0,0 @@
-/*
- *  Driver for Philips UDA1341TS on Compaq iPAQ H3600 soundcard
- *  Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
- *
- *   This program is free software; you can redistribute it and/or modify
- *   it under the terms of the GNU General Public License.
- * 
- * History:
- *
- * 2002-03-13   Tomas Kasparek  initial release - based on h3600-uda1341.c from OSS
- * 2002-03-20   Tomas Kasparek  playback over ALSA is working
- * 2002-03-28   Tomas Kasparek  playback over OSS emulation is working
- * 2002-03-29   Tomas Kasparek  basic capture is working (native ALSA)
- * 2002-03-29   Tomas Kasparek  capture is working (OSS emulation)
- * 2002-04-04   Tomas Kasparek  better rates handling (allow non-standard rates)
- * 2003-02-14   Brian Avery     fixed full duplex mode, other updates
- * 2003-02-20   Tomas Kasparek  merged updates by Brian (except HAL)
- * 2003-04-19   Jaroslav Kysela recoded DMA stuff to follow 2.4.18rmk3-hh24 kernel
- *                              working suspend and resume
- * 2003-04-28   Tomas Kasparek  updated work by Jaroslav to compile it under 2.5.x again
- *                              merged HAL layer (patches from Brian)
- */
-
-/***************************************************************************************************
-*
-* To understand what Alsa Drivers should be doing look at "Writing an Alsa Driver" by Takashi Iwai
-* available in the Alsa doc section on the website		
-* 
-* A few notes to make things clearer. The UDA1341 is hooked up to Serial port 4 on the SA1100.
-* We are using  SSP mode to talk to the UDA1341. The UDA1341 bit & wordselect clocks are generated
-* by this UART. Unfortunately, the clock only runs if the transmit buffer has something in it.
-* So, if we are just recording, we feed the transmit DMA stream a bunch of 0x0000 so that the
-* transmit buffer is full and the clock keeps going. The zeroes come from FLUSH_BASE_PHYS which
-* is a mem loc that always decodes to 0's w/ no off chip access.
-*
-* Some alsa terminology:
-*	frame => num_channels * sample_size  e.g stereo 16 bit is 2 * 16 = 32 bytes
-*	period => the least number of bytes that will generate an interrupt e.g. we have a 1024 byte
-*             buffer and 4 periods in the runtime structure this means we'll get an int every 256
-*             bytes or 4 times per buffer.
-*             A number of the sizes are in frames rather than bytes, use frames_to_bytes and
-*             bytes_to_frames to convert.  The easiest way to tell the units is to look at the
-*             type i.e. runtime-> buffer_size is in frames and its type is snd_pcm_uframes_t
-*             
-*	Notes about the pointer fxn:
-*	The pointer fxn needs to return the offset into the dma buffer in frames.
-*	Interrupts must be blocked before calling the dma_get_pos fxn to avoid race with interrupts.
-*
-*	Notes about pause/resume
-*	Implementing this would be complicated so it's skipped.  The problem case is:
-*	A full duplex connection is going, then play is paused. At this point you need to start xmitting
-*	0's to keep the record active which means you cant just freeze the dma and resume it later you'd
-*	need to	save off the dma info, and restore it properly on a resume.  Yeach!
-*
-*	Notes about transfer methods:
-*	The async write calls fail.  I probably need to implement something else to support them?
-* 
-***************************************************************************************************/
-
-#include <linux/module.h>
-#include <linux/moduleparam.h>
-#include <linux/init.h>
-#include <linux/err.h>
-#include <linux/platform_device.h>
-#include <linux/errno.h>
-#include <linux/ioctl.h>
-#include <linux/delay.h>
-#include <linux/slab.h>
-
-#ifdef CONFIG_PM
-#include <linux/pm.h>
-#endif
-
-#include <mach/hardware.h>
-#include <mach/h3600.h>
-#include <asm/mach-types.h>
-#include <asm/dma.h>
-
-#include <sound/core.h>
-#include <sound/pcm.h>
-#include <sound/initval.h>
-
-#include <linux/l3/l3.h>
-
-#undef DEBUG_MODE
-#undef DEBUG_FUNCTION_NAMES
-#include <sound/uda1341.h>
-
-/*
- * FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels?
- * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this
- * module for Familiar 0.6.1
- */
-
-/* {{{ Type definitions */
-
-MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>");
-MODULE_LICENSE("GPL");
-MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA");
-MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}");
-
-static char *id;	/* ID for this card */
-
-module_param(id, charp, 0444);
-MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard.");
-
-struct audio_stream {
-	char *id;		/* identification string */
-	int stream_id;		/* numeric identification */	
-	dma_device_t dma_dev;	/* device identifier for DMA */
-#ifdef HH_VERSION
-	dmach_t dmach;		/* dma channel identification */
-#else
-	dma_regs_t *dma_regs;	/* points to our DMA registers */
-#endif
-	unsigned int active:1;	/* we are using this stream for transfer now */
-	int period;		/* current transfer period */
-	int periods;		/* current count of periods registerd in the DMA engine */
-	int tx_spin;		/* are we recoding - flag used to do DMA trans. for sync */
-	unsigned int old_offset;
-	spinlock_t dma_lock;	/* for locking in DMA operations (see dma-sa1100.c in the kernel) */
-	struct snd_pcm_substream *stream;
-};
-
-struct sa11xx_uda1341 {
-	struct snd_card *card;
-	struct l3_client *uda1341;
-	struct snd_pcm *pcm;
-	long samplerate;
-	struct audio_stream s[2];	/* playback & capture */
-};
-
-static unsigned int rates[] = {
-	8000,  10666, 10985, 14647,
-	16000, 21970, 22050, 24000,
-	29400, 32000, 44100, 48000,
-};
-
-static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
-	.count	= ARRAY_SIZE(rates),
-	.list	= rates,
-	.mask	= 0,
-};
-
-static struct platform_device *device;
-
-/* }}} */
-
-/* {{{ Clock and sample rate stuff */
-
-/*
- * Stop-gap solution until rest of hh.org HAL stuff is merged.
- */
-#define GPIO_H3600_CLK_SET0		GPIO_GPIO (12)
-#define GPIO_H3600_CLK_SET1		GPIO_GPIO (13)
-
-#ifdef CONFIG_SA1100_H3XXX
-#define	clr_sa11xx_uda1341_egpio(x)	clr_h3600_egpio(x)
-#define set_sa11xx_uda1341_egpio(x)	set_h3600_egpio(x)
-#else
-#error This driver could serve H3x00 handhelds only!
-#endif
-
-static void sa11xx_uda1341_set_audio_clock(long val)
-{
-	switch (val) {
-	case 24000: case 32000: case 48000:	/* 00: 12.288 MHz */
-		GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
-		break;
-
-	case 22050: case 29400: case 44100:	/* 01: 11.2896 MHz */
-		GPSR = GPIO_H3600_CLK_SET0;
-		GPCR = GPIO_H3600_CLK_SET1;
-		break;
-
-	case 8000: case 10666: case 16000:	/* 10: 4.096 MHz */
-		GPCR = GPIO_H3600_CLK_SET0;
-		GPSR = GPIO_H3600_CLK_SET1;
-		break;
-
-	case 10985: case 14647: case 21970:	/* 11: 5.6245 MHz */
-		GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1;
-		break;
-	}
-}
-
-static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341 *sa11xx_uda1341, long rate)
-{
-	int clk_div = 0;
-	int clk=0;
-
-	/* We don't want to mess with clocks when frames are in flight */
-	Ser4SSCR0 &= ~SSCR0_SSE;
-	/* wait for any frame to complete */
-	udelay(125);
-
-	/*
-	 * We have the following clock sources:
-	 * 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz
-	 * Those can be divided either by 256, 384 or 512.
-	 * This makes up 12 combinations for the following samplerates...
-	 */
-	if (rate >= 48000)
-		rate = 48000;
-	else if (rate >= 44100)
-		rate = 44100;
-	else if (rate >= 32000)
-		rate = 32000;
-	else if (rate >= 29400)
-		rate = 29400;
-	else if (rate >= 24000)
-		rate = 24000;
-	else if (rate >= 22050)
-		rate = 22050;
-	else if (rate >= 21970)
-		rate = 21970;
-	else if (rate >= 16000)
-		rate = 16000;
-	else if (rate >= 14647)
-		rate = 14647;
-	else if (rate >= 10985)
-		rate = 10985;
-	else if (rate >= 10666)
-		rate = 10666;
-	else
-		rate = 8000;
-
-	/* Set the external clock generator */
-	
-	sa11xx_uda1341_set_audio_clock(rate);
-
-	/* Select the clock divisor */
-	switch (rate) {
-	case 8000:
-	case 10985:
-	case 22050:
-	case 24000:
-		clk = F512;
-		clk_div = SSCR0_SerClkDiv(16);
-		break;
-	case 16000:
-	case 21970:
-	case 44100:
-	case 48000:
-		clk = F256;
-		clk_div = SSCR0_SerClkDiv(8);
-		break;
-	case 10666:
-	case 14647:
-	case 29400:
-	case 32000:
-		clk = F384;
-		clk_div = SSCR0_SerClkDiv(12);
-		break;
-	}
-
-	/* FMT setting should be moved away when other FMTs are added (FIXME) */
-	l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16);
-	
-	l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk);        
-	Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE;
-	sa11xx_uda1341->samplerate = rate;
-}
-
-/* }}} */
-
-/* {{{ HW init and shutdown */
-
-static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341)
-{
-	unsigned long flags;
-
-	/* Setup DMA stuff */
-	sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out";
-	sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK;
-	sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr;
-
-	sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in";
-	sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE;
-	sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd;
-
-	/* Initialize the UDA1341 internal state */
-       
-	/* Setup the uarts */
-	local_irq_save(flags);
-	GAFR |= (GPIO_SSP_CLK);
-	GPDR &= ~(GPIO_SSP_CLK);
-	Ser4SSCR0 = 0;
-	Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8);
-	Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk;
-	Ser4SSCR0 |= SSCR0_SSE;
-	local_irq_restore(flags);
-
-	/* Enable the audio power */
-
-	clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
-	set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
-	set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
- 
-	/* Wait for the UDA1341 to wake up */
-	mdelay(1); //FIXME - was removed by Perex - Why?
-
-	/* Initialize the UDA1341 internal state */
-	l3_open(sa11xx_uda1341->uda1341);
-	
-	/* external clock configuration (after l3_open - regs must be initialized */
-	sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate);
-
-	/* Wait for the UDA1341 to wake up */
-	set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
-	mdelay(1);	
-
-	/* make the left and right channels unswapped (flip the WS latch) */
-	Ser4SSDR = 0;
-
-	clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
-}
-
-static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341)
-{
-	/* mute on */
-	set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
-	
-	/* disable the audio power and all signals leading to the audio chip */
-	l3_close(sa11xx_uda1341->uda1341);
-	Ser4SSCR0 = 0;
-	clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET);
-
-	/* power off and mute off */
-	/* FIXME - is muting off necesary??? */
-
-	clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON);
-	clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE);
-}
-
-/* }}} */
-
-/* {{{ DMA staff */
-
-/*
- * these are the address and sizes used to fill the xmit buffer
- * so we can get a clock in record only mode
- */
-#define FORCE_CLOCK_ADDR		(dma_addr_t)FLUSH_BASE_PHYS
-#define FORCE_CLOCK_SIZE		4096 // was 2048
-
-// FIXME Why this value exactly - wrote comment
-#define DMA_BUF_SIZE	8176	/* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */
-
-#ifdef HH_VERSION
-
-static int audio_dma_request(struct audio_stream *s, void (*callback)(void *, int))
-{
-	int ret;
-
-	ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev);
-	if (ret < 0) {
-		printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
-		return ret;
-	}
-	sa1100_dma_set_callback(s->dmach, callback);
-	return 0;
-}
-
-static inline void audio_dma_free(struct audio_stream *s)
-{
-	sa1100_free_dma(s->dmach);
-	s->dmach = -1;
-}
-
-#else
-
-static int audio_dma_request(struct audio_stream *s, void (*callback)(void *))
-{
-	int ret;
-
-	ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs);
-	if (ret < 0)
-		printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev);
-	return ret;
-}
-
-static void audio_dma_free(struct audio_stream *s)
-{
-	sa1100_free_dma(s->dma_regs);
-	s->dma_regs = 0;
-}
-
-#endif
-
-static u_int audio_get_dma_pos(struct audio_stream *s)
-{
-	struct snd_pcm_substream *substream = s->stream;
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	unsigned int offset;
-	unsigned long flags;
-	dma_addr_t addr;
-	
-	// this must be called w/ interrupts locked out see dma-sa1100.c in the kernel
-	spin_lock_irqsave(&s->dma_lock, flags);
-#ifdef HH_VERSION	
-	sa1100_dma_get_current(s->dmach, NULL, &addr);
-#else
-	addr = sa1100_get_dma_pos((s)->dma_regs);
-#endif
-	offset = addr - runtime->dma_addr;
-	spin_unlock_irqrestore(&s->dma_lock, flags);
-	
-	offset = bytes_to_frames(runtime,offset);
-	if (offset >= runtime->buffer_size)
-		offset = 0;
-
-	return offset;
-}
-
-/*
- * this stops the dma and clears the dma ptrs
- */
-static void audio_stop_dma(struct audio_stream *s)
-{
-	unsigned long flags;
-
-	spin_lock_irqsave(&s->dma_lock, flags);	
-	s->active = 0;
-	s->period = 0;
-	/* this stops the dma channel and clears the buffer ptrs */
-#ifdef HH_VERSION
-	sa1100_dma_flush_all(s->dmach);
-#else
-	sa1100_clear_dma(s->dma_regs);	
-#endif
-	spin_unlock_irqrestore(&s->dma_lock, flags);
-}
-
-static void audio_process_dma(struct audio_stream *s)
-{
-	struct snd_pcm_substream *substream = s->stream;
-	struct snd_pcm_runtime *runtime;
-	unsigned int dma_size;		
-	unsigned int offset;
-	int ret;
-                
-	/* we are requested to process synchronization DMA transfer */
-	if (s->tx_spin) {
-		if (snd_BUG_ON(s->stream_id != SNDRV_PCM_STREAM_PLAYBACK))
-			return;
-		/* fill the xmit dma buffers and return */
-#ifdef HH_VERSION
-		sa1100_dma_set_spin(s->dmach, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
-#else
-		while (1) {
-			ret = sa1100_start_dma(s->dma_regs, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE);
-			if (ret)
-				return;   
-		}
-#endif
-		return;
-	}
-
-	/* must be set here - only valid for running streams, not for forced_clock dma fills  */
-	runtime = substream->runtime;
-	while (s->active && s->periods < runtime->periods) {
-		dma_size = frames_to_bytes(runtime, runtime->period_size);
-		if (s->old_offset) {
-			/* a little trick, we need resume from old position */
-			offset = frames_to_bytes(runtime, s->old_offset - 1);
-			s->old_offset = 0;
-			s->periods = 0;
-			s->period = offset / dma_size;
-			offset %= dma_size;
-			dma_size = dma_size - offset;
-			if (!dma_size)
-				continue;		/* special case */
-		} else {
-			offset = dma_size * s->period;
-			snd_BUG_ON(dma_size > DMA_BUF_SIZE);
-		}
-#ifdef HH_VERSION
-		ret = sa1100_dma_queue_buffer(s->dmach, s, runtime->dma_addr + offset, dma_size);
-		if (ret)
-			return; //FIXME
-#else
-		ret = sa1100_start_dma((s)->dma_regs, runtime->dma_addr + offset, dma_size);
-		if (ret) {
-			printk(KERN_ERR "audio_process_dma: cannot queue DMA buffer (%i)\n", ret);
-			return;
-		}
-#endif
-
-		s->period++;
-		s->period %= runtime->periods;
-		s->periods++;
-	}
-}
-
-#ifdef HH_VERSION
-static void audio_dma_callback(void *data, int size)
-#else
-static void audio_dma_callback(void *data)
-#endif
-{
-	struct audio_stream *s = data;
-        
-	/* 
-	 * If we are getting a callback for an active stream then we inform
-	 * the PCM middle layer we've finished a period
-	 */
- 	if (s->active)
-		snd_pcm_period_elapsed(s->stream);
-
-	spin_lock(&s->dma_lock);
-	if (!s->tx_spin && s->periods > 0)
-		s->periods--;
-	audio_process_dma(s);
-	spin_unlock(&s->dma_lock);
-}
-
-/* }}} */
-
-/* {{{ PCM setting */
-
-/* {{{ trigger & timer */
-
-static int snd_sa11xx_uda1341_trigger(struct snd_pcm_substream *substream, int cmd)
-{
-	struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
-	int stream_id = substream->pstr->stream;
-	struct audio_stream *s = &chip->s[stream_id];
-	struct audio_stream *s1 = &chip->s[stream_id ^ 1];
-	int err = 0;
-
-	/* note local interrupts are already disabled in the midlevel code */
-	spin_lock(&s->dma_lock);
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-		/* now we need to make sure a record only stream has a clock */
-		if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
-			/* we need to force fill the xmit DMA with zeros */
-			s1->tx_spin = 1;
-			audio_process_dma(s1);
-		}
-		/* this case is when you were recording then you turn on a
-		 * playback stream so we stop (also clears it) the dma first,
-		 * clear the sync flag and then we let it turned on
-		 */		
-		else {
- 			s->tx_spin = 0;
- 		}
-
-		/* requested stream startup */
-		s->active = 1;
-		audio_process_dma(s);
-		break;
-	case SNDRV_PCM_TRIGGER_STOP:
-		/* requested stream shutdown */
-		audio_stop_dma(s);
-		
-		/*
-		 * now we need to make sure a record only stream has a clock
-		 * so if we're stopping a playback with an active capture
-		 * we need to turn the 0 fill dma on for the xmit side
-		 */
-		if (stream_id == SNDRV_PCM_STREAM_PLAYBACK && s1->active) {
-			/* we need to force fill the xmit DMA with zeros */
-			s->tx_spin = 1;
-			audio_process_dma(s);
-		}
-		/*
-		 * we killed a capture only stream, so we should also kill
-		 * the zero fill transmit
-		 */
-		else {
-			if (s1->tx_spin) {
-				s1->tx_spin = 0;
-				audio_stop_dma(s1);
-			}
-		}
-		
-		break;
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-		s->active = 0;
-#ifdef HH_VERSION		
-		sa1100_dma_stop(s->dmach);
-#else
-		//FIXME - DMA API
-#endif		
-		s->old_offset = audio_get_dma_pos(s) + 1;
-#ifdef HH_VERSION		
-		sa1100_dma_flush_all(s->dmach);
-#else
-		//FIXME - DMA API
-#endif		
-		s->periods = 0;
-		break;
-	case SNDRV_PCM_TRIGGER_RESUME:
-		s->active = 1;
-		s->tx_spin = 0;
-		audio_process_dma(s);
-		if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
-			s1->tx_spin = 1;
-			audio_process_dma(s1);
-		}
-		break;
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-#ifdef HH_VERSION		
-		sa1100_dma_stop(s->dmach);
-#else
-		//FIXME - DMA API
-#endif
-		s->active = 0;
-		if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) {
-			if (s1->active) {
-				s->tx_spin = 1;
-				s->old_offset = audio_get_dma_pos(s) + 1;
-#ifdef HH_VERSION				
-				sa1100_dma_flush_all(s->dmach);
-#else
-				//FIXME - DMA API
-#endif				
-				audio_process_dma(s);
-			}
-		} else {
-			if (s1->tx_spin) {
-				s1->tx_spin = 0;
-#ifdef HH_VERSION				
-				sa1100_dma_flush_all(s1->dmach);
-#else
-				//FIXME - DMA API
-#endif				
-			}
-		}
-		break;
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		s->active = 1;
-		if (s->old_offset) {
-			s->tx_spin = 0;
-			audio_process_dma(s);
-			break;
-		}
-		if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) {
-			s1->tx_spin = 1;
-			audio_process_dma(s1);
-		}
-#ifdef HH_VERSION		
-		sa1100_dma_resume(s->dmach);
-#else
-		//FIXME - DMA API
-#endif
-		break;
-	default:
-		err = -EINVAL;
-		break;
-	}
-	spin_unlock(&s->dma_lock);	
-	return err;
-}
-
-static int snd_sa11xx_uda1341_prepare(struct snd_pcm_substream *substream)
-{
-	struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct audio_stream *s = &chip->s[substream->pstr->stream];
-        
-	/* set requested samplerate */
-	sa11xx_uda1341_set_samplerate(chip, runtime->rate);
-
-	/* set requestd format when available */
-	/* set FMT here !!! FIXME */
-
-	s->period = 0;
-	s->periods = 0;
-        
-	return 0;
-}
-
-static snd_pcm_uframes_t snd_sa11xx_uda1341_pointer(struct snd_pcm_substream *substream)
-{
-	struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
-	return audio_get_dma_pos(&chip->s[substream->pstr->stream]);
-}
-
-/* }}} */
-
-static struct snd_pcm_hardware snd_sa11xx_uda1341_capture =
-{
-	.info			= (SNDRV_PCM_INFO_INTERLEAVED |
-				   SNDRV_PCM_INFO_BLOCK_TRANSFER |
-				   SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-				   SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
-	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
-	.rates			= (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
-				   SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
-				   SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
-				   SNDRV_PCM_RATE_KNOT),
-	.rate_min		= 8000,
-	.rate_max		= 48000,
-	.channels_min		= 2,
-	.channels_max		= 2,
-	.buffer_bytes_max	= 64*1024,
-	.period_bytes_min	= 64,
-	.period_bytes_max	= DMA_BUF_SIZE,
-	.periods_min		= 2,
-	.periods_max		= 255,
-	.fifo_size		= 0,
-};
-
-static struct snd_pcm_hardware snd_sa11xx_uda1341_playback =
-{
-	.info			= (SNDRV_PCM_INFO_INTERLEAVED |
-				   SNDRV_PCM_INFO_BLOCK_TRANSFER |
-				   SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-				   SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
-	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
-	.rates			= (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\
-                                   SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\
-				   SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
-				   SNDRV_PCM_RATE_KNOT),
-	.rate_min		= 8000,
-	.rate_max		= 48000,
-	.channels_min		= 2,
-	.channels_max		= 2,
-	.buffer_bytes_max	= 64*1024,
-	.period_bytes_min	= 64,
-	.period_bytes_max	= DMA_BUF_SIZE,
-	.periods_min		= 2,
-	.periods_max		= 255,
-	.fifo_size		= 0,
-};
-
-static int snd_card_sa11xx_uda1341_open(struct snd_pcm_substream *substream)
-{
-	struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	int stream_id = substream->pstr->stream;
-	int err;
-
-	chip->s[stream_id].stream = substream;
-
-	if (stream_id == SNDRV_PCM_STREAM_PLAYBACK)
-		runtime->hw = snd_sa11xx_uda1341_playback;
-	else
-		runtime->hw = snd_sa11xx_uda1341_capture;
-	if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
-		return err;
-	if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0)
-		return err;
-        
-	return 0;
-}
-
-static int snd_card_sa11xx_uda1341_close(struct snd_pcm_substream *substream)
-{
-	struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream);
-
-	chip->s[substream->pstr->stream].stream = NULL;
-	return 0;
-}
-
-/* {{{ HW params & free */
-
-static int snd_sa11xx_uda1341_hw_params(struct snd_pcm_substream *substream,
-					struct snd_pcm_hw_params *hw_params)
-{
-        
-	return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
-}
-
-static int snd_sa11xx_uda1341_hw_free(struct snd_pcm_substream *substream)
-{
-	return snd_pcm_lib_free_pages(substream);
-}
-
-/* }}} */
-
-static struct snd_pcm_ops snd_card_sa11xx_uda1341_playback_ops = {
-	.open			= snd_card_sa11xx_uda1341_open,
-	.close			= snd_card_sa11xx_uda1341_close,
-	.ioctl			= snd_pcm_lib_ioctl,
-	.hw_params	        = snd_sa11xx_uda1341_hw_params,
-	.hw_free	        = snd_sa11xx_uda1341_hw_free,
-	.prepare		= snd_sa11xx_uda1341_prepare,
-	.trigger		= snd_sa11xx_uda1341_trigger,
-	.pointer		= snd_sa11xx_uda1341_pointer,
-};
-
-static struct snd_pcm_ops snd_card_sa11xx_uda1341_capture_ops = {
-	.open			= snd_card_sa11xx_uda1341_open,
-	.close			= snd_card_sa11xx_uda1341_close,
-	.ioctl			= snd_pcm_lib_ioctl,
-	.hw_params	        = snd_sa11xx_uda1341_hw_params,
-	.hw_free	        = snd_sa11xx_uda1341_hw_free,
-	.prepare		= snd_sa11xx_uda1341_prepare,
-	.trigger		= snd_sa11xx_uda1341_trigger,
-	.pointer		= snd_sa11xx_uda1341_pointer,
-};
-
-static int __init snd_card_sa11xx_uda1341_pcm(struct sa11xx_uda1341 *sa11xx_uda1341, int device)
-{
-	struct snd_pcm *pcm;
-	int err;
-
-	if ((err = snd_pcm_new(sa11xx_uda1341->card, "UDA1341 PCM", device, 1, 1, &pcm)) < 0)
-		return err;
-
-	/*
-	 * this sets up our initial buffers and sets the dma_type to isa.
-	 * isa works but I'm not sure why (or if) it's the right choice
-	 * this may be too large, trying it for now
-	 */
-	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, 
-					      snd_dma_isa_data(),
-					      64*1024, 64*1024);
-
-	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_sa11xx_uda1341_playback_ops);
-	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_sa11xx_uda1341_capture_ops);
-	pcm->private_data = sa11xx_uda1341;
-	pcm->info_flags = 0;
-	strcpy(pcm->name, "UDA1341 PCM");
-
-	sa11xx_uda1341_audio_init(sa11xx_uda1341);
-
-	/* setup DMA controller */
-	audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK], audio_dma_callback);
-	audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE], audio_dma_callback);
-
-	sa11xx_uda1341->pcm = pcm;
-
-	return 0;
-}
-
-/* }}} */
-
-/* {{{ module init & exit */
-
-#ifdef CONFIG_PM
-
-static int snd_sa11xx_uda1341_suspend(struct platform_device *devptr,
-				      pm_message_t state)
-{
-	struct snd_card *card = platform_get_drvdata(devptr);
-	struct sa11xx_uda1341 *chip = card->private_data;
-
-	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
-	snd_pcm_suspend_all(chip->pcm);
-#ifdef HH_VERSION
-	sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
-	sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
-#else
-	//FIXME
-#endif
-	l3_command(chip->uda1341, CMD_SUSPEND, NULL);
-	sa11xx_uda1341_audio_shutdown(chip);
-
-	return 0;
-}
-
-static int snd_sa11xx_uda1341_resume(struct platform_device *devptr)
-{
-	struct snd_card *card = platform_get_drvdata(devptr);
-	struct sa11xx_uda1341 *chip = card->private_data;
-
-	sa11xx_uda1341_audio_init(chip);
-	l3_command(chip->uda1341, CMD_RESUME, NULL);
-#ifdef HH_VERSION	
-	sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach);
-	sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach);
-#else
-	//FIXME
-#endif
-	snd_power_change_state(card, SNDRV_CTL_POWER_D0);
-	return 0;
-}
-#endif /* COMFIG_PM */
-
-void snd_sa11xx_uda1341_free(struct snd_card *card)
-{
-	struct sa11xx_uda1341 *chip = card->private_data;
-
-	audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]);
-	audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]);
-}
-
-static int __devinit sa11xx_uda1341_probe(struct platform_device *devptr)
-{
-	int err;
-	struct snd_card *card;
-	struct sa11xx_uda1341 *chip;
-
-	/* register the soundcard */
-	card = snd_card_new(-1, id, THIS_MODULE, sizeof(struct sa11xx_uda1341));
-	if (card == NULL)
-		return -ENOMEM;
-
-	chip = card->private_data;
-	spin_lock_init(&chip->s[0].dma_lock);
-	spin_lock_init(&chip->s[1].dma_lock);
-
-	card->private_free = snd_sa11xx_uda1341_free;
-	chip->card = card;
-	chip->samplerate = AUDIO_RATE_DEFAULT;
-
-	// mixer
-	if ((err = snd_chip_uda1341_mixer_new(card, &chip->uda1341)))
-		goto nodev;
-
-	// PCM
-	if ((err = snd_card_sa11xx_uda1341_pcm(chip, 0)) < 0)
-		goto nodev;
-        
-	strcpy(card->driver, "UDA1341");
-	strcpy(card->shortname, "H3600 UDA1341TS");
-	sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS");
-        
-	snd_card_set_dev(card, &devptr->dev);
-
-	if ((err = snd_card_register(card)) == 0) {
-		printk( KERN_INFO "iPAQ audio support initialized\n" );
-		platform_set_drvdata(devptr, card);
-		return 0;
-	}
-        
- nodev:
-	snd_card_free(card);
-	return err;
-}
-
-static int __devexit sa11xx_uda1341_remove(struct platform_device *devptr)
-{
-	snd_card_free(platform_get_drvdata(devptr));
-	platform_set_drvdata(devptr, NULL);
-	return 0;
-}
-
-#define SA11XX_UDA1341_DRIVER	"sa11xx_uda1341"
-
-static struct platform_driver sa11xx_uda1341_driver = {
-	.probe		= sa11xx_uda1341_probe,
-	.remove		= __devexit_p(sa11xx_uda1341_remove),
-#ifdef CONFIG_PM
-	.suspend	= snd_sa11xx_uda1341_suspend,
-	.resume		= snd_sa11xx_uda1341_resume,
-#endif
-	.driver		= {
-		.name	= SA11XX_UDA1341_DRIVER,
-	},
-};
-
-static int __init sa11xx_uda1341_init(void)
-{
-	int err;
-
-	if (!machine_is_h3xxx())
-		return -ENODEV;
-	if ((err = platform_driver_register(&sa11xx_uda1341_driver)) < 0)
-		return err;
-	device = platform_device_register_simple(SA11XX_UDA1341_DRIVER, -1, NULL, 0);
-	if (!IS_ERR(device)) {
-		if (platform_get_drvdata(device))
-			return 0;
-		platform_device_unregister(device);
-		err = -ENODEV;
-	} else
-		err = PTR_ERR(device);
-	platform_driver_unregister(&sa11xx_uda1341_driver);
-	return err;
-}
-
-static void __exit sa11xx_uda1341_exit(void)
-{
-	platform_device_unregister(device);
-	platform_driver_unregister(&sa11xx_uda1341_driver);
-}
-
-module_init(sa11xx_uda1341_init);
-module_exit(sa11xx_uda1341_exit);
-
-/* }}} */
-
-/*
- * Local variables:
- * indent-tabs-mode: t
- * End:
- */
diff --git a/sound/atmel/Kconfig b/sound/atmel/Kconfig
new file mode 100644
index 000000000000..6c228a91940d
--- /dev/null
+++ b/sound/atmel/Kconfig
@@ -0,0 +1,19 @@
+menu "Atmel devices (AVR32 and AT91)"
+	depends on AVR32 || ARCH_AT91
+
+config SND_ATMEL_ABDAC
+	tristate "Atmel Audio Bitstream DAC (ABDAC) driver"
+	select SND_PCM
+	depends on DW_DMAC && AVR32
+	help
+	  ALSA sound driver for the Atmel Audio Bitstream DAC (ABDAC).
+
+config SND_ATMEL_AC97C
+	tristate "Atmel AC97 Controller (AC97C) driver"
+	select SND_PCM
+	select SND_AC97_CODEC
+	depends on DW_DMAC && AVR32
+	help
+	  ALSA sound driver for the Atmel AC97 controller.
+
+endmenu
diff --git a/sound/atmel/Makefile b/sound/atmel/Makefile
new file mode 100644
index 000000000000..219dcfac6086
--- /dev/null
+++ b/sound/atmel/Makefile
@@ -0,0 +1,5 @@
+snd-atmel-abdac-objs		:= abdac.o
+snd-atmel-ac97c-objs		:= ac97c.o
+
+obj-$(CONFIG_SND_ATMEL_ABDAC)	+= snd-atmel-abdac.o
+obj-$(CONFIG_SND_ATMEL_AC97C)	+= snd-atmel-ac97c.o
diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c
new file mode 100644
index 000000000000..28b3c7f7cfe6
--- /dev/null
+++ b/sound/atmel/abdac.c
@@ -0,0 +1,602 @@
+/*
+ * Driver for the Atmel on-chip Audio Bitstream DAC (ABDAC)
+ *
+ * Copyright (C) 2006-2009 Atmel Corporation
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published by
+ * the Free Software Foundation.
+ */
+#include <linux/clk.h>
+#include <linux/bitmap.h>
+#include <linux/dw_dmac.h>
+#include <linux/dmaengine.h>
+#include <linux/dma-mapping.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/io.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/atmel-abdac.h>
+
+/* DAC register offsets */
+#define DAC_DATA                                0x0000
+#define DAC_CTRL                                0x0008
+#define DAC_INT_MASK                            0x000c
+#define DAC_INT_EN                              0x0010
+#define DAC_INT_DIS                             0x0014
+#define DAC_INT_CLR                             0x0018
+#define DAC_INT_STATUS                          0x001c
+
+/* Bitfields in CTRL */
+#define DAC_SWAP_OFFSET                         30
+#define DAC_SWAP_SIZE                           1
+#define DAC_EN_OFFSET                           31
+#define DAC_EN_SIZE                             1
+
+/* Bitfields in INT_MASK/INT_EN/INT_DIS/INT_STATUS/INT_CLR */
+#define DAC_UNDERRUN_OFFSET                     28
+#define DAC_UNDERRUN_SIZE                       1
+#define DAC_TX_READY_OFFSET                     29
+#define DAC_TX_READY_SIZE                       1
+
+/* Bit manipulation macros */
+#define DAC_BIT(name)					\
+	(1 << DAC_##name##_OFFSET)
+#define DAC_BF(name, value)				\
+	(((value) & ((1 << DAC_##name##_SIZE) - 1))	\
+	 << DAC_##name##_OFFSET)
+#define DAC_BFEXT(name, value)				\
+	(((value) >> DAC_##name##_OFFSET)		\
+	 & ((1 << DAC_##name##_SIZE) - 1))
+#define DAC_BFINS(name, value, old)			\
+	(((old) & ~(((1 << DAC_##name##_SIZE) - 1)	\
+		    << DAC_##name##_OFFSET))		\
+	 | DAC_BF(name, value))
+
+/* Register access macros */
+#define dac_readl(port, reg)				\
+	__raw_readl((port)->regs + DAC_##reg)
+#define dac_writel(port, reg, value)			\
+	__raw_writel((value), (port)->regs + DAC_##reg)
+
+/*
+ * ABDAC supports a maximum of 6 different rates from a generic clock. The
+ * generic clock has a power of two divider, which gives 6 steps from 192 kHz
+ * to 5112 Hz.
+ */
+#define MAX_NUM_RATES	6
+/* ALSA seems to use rates between 192000 Hz and 5112 Hz. */
+#define RATE_MAX	192000
+#define RATE_MIN	5112
+
+enum {
+	DMA_READY = 0,
+};
+
+struct atmel_abdac_dma {
+	struct dma_chan		*chan;
+	struct dw_cyclic_desc	*cdesc;
+};
+
+struct atmel_abdac {
+	struct clk				*pclk;
+	struct clk				*sample_clk;
+	struct platform_device			*pdev;
+	struct atmel_abdac_dma			dma;
+
+	struct snd_pcm_hw_constraint_list	constraints_rates;
+	struct snd_pcm_substream		*substream;
+	struct snd_card				*card;
+	struct snd_pcm				*pcm;
+
+	void __iomem				*regs;
+	unsigned long				flags;
+	unsigned int				rates[MAX_NUM_RATES];
+	unsigned int				rates_num;
+	int					irq;
+};
+
+#define get_dac(card) ((struct atmel_abdac *)(card)->private_data)
+
+/* This function is called by the DMA driver. */
+static void atmel_abdac_dma_period_done(void *arg)
+{
+	struct atmel_abdac *dac = arg;
+	snd_pcm_period_elapsed(dac->substream);
+}
+
+static int atmel_abdac_prepare_dma(struct atmel_abdac *dac,
+		struct snd_pcm_substream *substream,
+		enum dma_data_direction direction)
+{
+	struct dma_chan			*chan = dac->dma.chan;
+	struct dw_cyclic_desc		*cdesc;
+	struct snd_pcm_runtime		*runtime = substream->runtime;
+	unsigned long			buffer_len, period_len;
+
+	/*
+	 * We don't do DMA on "complex" transfers, i.e. with
+	 * non-halfword-aligned buffers or lengths.
+	 */
+	if (runtime->dma_addr & 1 || runtime->buffer_size & 1) {
+		dev_dbg(&dac->pdev->dev, "too complex transfer\n");
+		return -EINVAL;
+	}
+
+	buffer_len = frames_to_bytes(runtime, runtime->buffer_size);
+	period_len = frames_to_bytes(runtime, runtime->period_size);
+
+	cdesc = dw_dma_cyclic_prep(chan, runtime->dma_addr, buffer_len,
+			period_len, DMA_TO_DEVICE);
+	if (IS_ERR(cdesc)) {
+		dev_dbg(&dac->pdev->dev, "could not prepare cyclic DMA\n");
+		return PTR_ERR(cdesc);
+	}
+
+	cdesc->period_callback = atmel_abdac_dma_period_done;
+	cdesc->period_callback_param = dac;
+
+	dac->dma.cdesc = cdesc;
+
+	set_bit(DMA_READY, &dac->flags);
+
+	return 0;
+}
+
+static struct snd_pcm_hardware atmel_abdac_hw = {
+	.info			= (SNDRV_PCM_INFO_MMAP
+				  | SNDRV_PCM_INFO_MMAP_VALID
+				  | SNDRV_PCM_INFO_INTERLEAVED
+				  | SNDRV_PCM_INFO_BLOCK_TRANSFER
+				  | SNDRV_PCM_INFO_RESUME
+				  | SNDRV_PCM_INFO_PAUSE),
+	.formats		= (SNDRV_PCM_FMTBIT_S16_BE),
+	.rates			= (SNDRV_PCM_RATE_KNOT),
+	.rate_min		= RATE_MIN,
+	.rate_max		= RATE_MAX,
+	.channels_min		= 2,
+	.channels_max		= 2,
+	.buffer_bytes_max	= 64 * 4096,
+	.period_bytes_min	= 4096,
+	.period_bytes_max	= 4096,
+	.periods_min		= 4,
+	.periods_max		= 64,
+};
+
+static int atmel_abdac_open(struct snd_pcm_substream *substream)
+{
+	struct atmel_abdac *dac = snd_pcm_substream_chip(substream);
+
+	dac->substream = substream;
+	atmel_abdac_hw.rate_max = dac->rates[dac->rates_num - 1];
+	atmel_abdac_hw.rate_min = dac->rates[0];
+	substream->runtime->hw = atmel_abdac_hw;
+
+	return snd_pcm_hw_constraint_list(substream->runtime, 0,
+			SNDRV_PCM_HW_PARAM_RATE, &dac->constraints_rates);
+}
+
+static int atmel_abdac_close(struct snd_pcm_substream *substream)
+{
+	struct atmel_abdac *dac = snd_pcm_substream_chip(substream);
+	dac->substream = NULL;
+	return 0;
+}
+
+static int atmel_abdac_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *hw_params)
+{
+	struct atmel_abdac *dac = snd_pcm_substream_chip(substream);
+	int retval;
+
+	retval = snd_pcm_lib_malloc_pages(substream,
+			params_buffer_bytes(hw_params));
+	if (retval < 0)
+		return retval;
+	/* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */
+	if (retval == 1)
+		if (test_and_clear_bit(DMA_READY, &dac->flags))
+			dw_dma_cyclic_free(dac->dma.chan);
+
+	return retval;
+}
+
+static int atmel_abdac_hw_free(struct snd_pcm_substream *substream)
+{
+	struct atmel_abdac *dac = snd_pcm_substream_chip(substream);
+	if (test_and_clear_bit(DMA_READY, &dac->flags))
+		dw_dma_cyclic_free(dac->dma.chan);
+	return snd_pcm_lib_free_pages(substream);
+}
+
+static int atmel_abdac_prepare(struct snd_pcm_substream *substream)
+{
+	struct atmel_abdac *dac = snd_pcm_substream_chip(substream);
+	int retval;
+
+	retval = clk_set_rate(dac->sample_clk, 256 * substream->runtime->rate);
+	if (retval)
+		return retval;
+
+	if (!test_bit(DMA_READY, &dac->flags))
+		retval = atmel_abdac_prepare_dma(dac, substream, DMA_TO_DEVICE);
+
+	return retval;
+}
+
+static int atmel_abdac_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct atmel_abdac *dac = snd_pcm_substream_chip(substream);
+	int retval = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */
+	case SNDRV_PCM_TRIGGER_RESUME: /* fall through */
+	case SNDRV_PCM_TRIGGER_START:
+		clk_enable(dac->sample_clk);
+		retval = dw_dma_cyclic_start(dac->dma.chan);
+		if (retval)
+			goto out;
+		dac_writel(dac, CTRL, DAC_BIT(EN));
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */
+	case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */
+	case SNDRV_PCM_TRIGGER_STOP:
+		dw_dma_cyclic_stop(dac->dma.chan);
+		dac_writel(dac, DATA, 0);
+		dac_writel(dac, CTRL, 0);
+		clk_disable(dac->sample_clk);
+		break;
+	default:
+		retval = -EINVAL;
+		break;
+	}
+out:
+	return retval;
+}
+
+static snd_pcm_uframes_t
+atmel_abdac_pointer(struct snd_pcm_substream *substream)
+{
+	struct atmel_abdac	*dac = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime	*runtime = substream->runtime;
+	snd_pcm_uframes_t	frames;
+	unsigned long		bytes;
+
+	bytes = dw_dma_get_src_addr(dac->dma.chan);
+	bytes -= runtime->dma_addr;
+
+	frames = bytes_to_frames(runtime, bytes);
+	if (frames >= runtime->buffer_size)
+		frames -= runtime->buffer_size;
+
+	return frames;
+}
+
+static irqreturn_t abdac_interrupt(int irq, void *dev_id)
+{
+	struct atmel_abdac *dac = dev_id;
+	u32 status;
+
+	status = dac_readl(dac, INT_STATUS);
+	if (status & DAC_BIT(UNDERRUN)) {
+		dev_err(&dac->pdev->dev, "underrun detected\n");
+		dac_writel(dac, INT_CLR, DAC_BIT(UNDERRUN));
+	} else {
+		dev_err(&dac->pdev->dev, "spurious interrupt (status=0x%x)\n",
+			status);
+		dac_writel(dac, INT_CLR, status);
+	}
+
+	return IRQ_HANDLED;
+}
+
+static struct snd_pcm_ops atmel_abdac_ops = {
+	.open		= atmel_abdac_open,
+	.close		= atmel_abdac_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= atmel_abdac_hw_params,
+	.hw_free	= atmel_abdac_hw_free,
+	.prepare	= atmel_abdac_prepare,
+	.trigger	= atmel_abdac_trigger,
+	.pointer	= atmel_abdac_pointer,
+};
+
+static int __devinit atmel_abdac_pcm_new(struct atmel_abdac *dac)
+{
+	struct snd_pcm_hardware hw = atmel_abdac_hw;
+	struct snd_pcm *pcm;
+	int retval;
+
+	retval = snd_pcm_new(dac->card, dac->card->shortname,
+			dac->pdev->id, 1, 0, &pcm);
+	if (retval)
+		return retval;
+
+	strcpy(pcm->name, dac->card->shortname);
+	pcm->private_data = dac;
+	pcm->info_flags = 0;
+	dac->pcm = pcm;
+
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &atmel_abdac_ops);
+
+	retval = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+			&dac->pdev->dev, hw.periods_min * hw.period_bytes_min,
+			hw.buffer_bytes_max);
+
+	return retval;
+}
+
+static bool filter(struct dma_chan *chan, void *slave)
+{
+	struct dw_dma_slave *dws = slave;
+
+	if (dws->dma_dev == chan->device->dev) {
+		chan->private = dws;
+		return true;
+	} else
+		return false;
+}
+
+static int set_sample_rates(struct atmel_abdac *dac)
+{
+	long new_rate = RATE_MAX;
+	int retval = -EINVAL;
+	int index = 0;
+
+	/* we start at 192 kHz and work our way down to 5112 Hz */
+	while (new_rate >= RATE_MIN && index < (MAX_NUM_RATES + 1)) {
+		new_rate = clk_round_rate(dac->sample_clk, 256 * new_rate);
+		if (new_rate < 0)
+			break;
+		/* make sure we are below the ABDAC clock */
+		if (new_rate <= clk_get_rate(dac->pclk)) {
+			dac->rates[index] = new_rate / 256;
+			index++;
+		}
+		/* divide by 256 and then by two to get next rate */
+		new_rate /= 256 * 2;
+	}
+
+	if (index) {
+		int i;
+
+		/* reverse array, smallest go first */
+		for (i = 0; i < (index / 2); i++) {
+			unsigned int tmp = dac->rates[index - 1 - i];
+			dac->rates[index - 1 - i] = dac->rates[i];
+			dac->rates[i] = tmp;
+		}
+
+		dac->constraints_rates.count = index;
+		dac->constraints_rates.list = dac->rates;
+		dac->constraints_rates.mask = 0;
+		dac->rates_num = index;
+
+		retval = 0;
+	}
+
+	return retval;
+}
+
+static int __devinit atmel_abdac_probe(struct platform_device *pdev)
+{
+	struct snd_card		*card;
+	struct atmel_abdac	*dac;
+	struct resource		*regs;
+	struct atmel_abdac_pdata	*pdata;
+	struct clk		*pclk;
+	struct clk		*sample_clk;
+	int			retval;
+	int			irq;
+
+	regs = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!regs) {
+		dev_dbg(&pdev->dev, "no memory resource\n");
+		return -ENXIO;
+	}
+
+	irq = platform_get_irq(pdev, 0);
+	if (irq < 0) {
+		dev_dbg(&pdev->dev, "could not get IRQ number\n");
+		return irq;
+	}
+
+	pdata = pdev->dev.platform_data;
+	if (!pdata) {
+		dev_dbg(&pdev->dev, "no platform data\n");
+		return -ENXIO;
+	}
+
+	pclk = clk_get(&pdev->dev, "pclk");
+	if (IS_ERR(pclk)) {
+		dev_dbg(&pdev->dev, "no peripheral clock\n");
+		return PTR_ERR(pclk);
+	}
+	sample_clk = clk_get(&pdev->dev, "sample_clk");
+	if (IS_ERR(pclk)) {
+		dev_dbg(&pdev->dev, "no sample clock\n");
+		retval = PTR_ERR(pclk);
+		goto out_put_pclk;
+	}
+	clk_enable(pclk);
+
+	retval = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
+			THIS_MODULE, sizeof(struct atmel_abdac), &card);
+	if (retval) {
+		dev_dbg(&pdev->dev, "could not create sound card device\n");
+		goto out_put_sample_clk;
+	}
+
+	dac = get_dac(card);
+
+	dac->irq = irq;
+	dac->card = card;
+	dac->pclk = pclk;
+	dac->sample_clk = sample_clk;
+	dac->pdev = pdev;
+
+	retval = set_sample_rates(dac);
+	if (retval < 0) {
+		dev_dbg(&pdev->dev, "could not set supported rates\n");
+		goto out_free_card;
+	}
+
+	dac->regs = ioremap(regs->start, regs->end - regs->start + 1);
+	if (!dac->regs) {
+		dev_dbg(&pdev->dev, "could not remap register memory\n");
+		goto out_free_card;
+	}
+
+	/* make sure the DAC is silent and disabled */
+	dac_writel(dac, DATA, 0);
+	dac_writel(dac, CTRL, 0);
+
+	retval = request_irq(irq, abdac_interrupt, 0, "abdac", dac);
+	if (retval) {
+		dev_dbg(&pdev->dev, "could not request irq\n");
+		goto out_unmap_regs;
+	}
+
+	snd_card_set_dev(card, &pdev->dev);
+
+	if (pdata->dws.dma_dev) {
+		struct dw_dma_slave *dws = &pdata->dws;
+		dma_cap_mask_t mask;
+
+		dws->tx_reg = regs->start + DAC_DATA;
+
+		dma_cap_zero(mask);
+		dma_cap_set(DMA_SLAVE, mask);
+
+		dac->dma.chan = dma_request_channel(mask, filter, dws);
+	}
+	if (!pdata->dws.dma_dev || !dac->dma.chan) {
+		dev_dbg(&pdev->dev, "DMA not available\n");
+		retval = -ENODEV;
+		goto out_unset_card_dev;
+	}
+
+	strcpy(card->driver, "Atmel ABDAC");
+	strcpy(card->shortname, "Atmel ABDAC");
+	sprintf(card->longname, "Atmel Audio Bitstream DAC");
+
+	retval = atmel_abdac_pcm_new(dac);
+	if (retval) {
+		dev_dbg(&pdev->dev, "could not register ABDAC pcm device\n");
+		goto out_release_dma;
+	}
+
+	retval = snd_card_register(card);
+	if (retval) {
+		dev_dbg(&pdev->dev, "could not register sound card\n");
+		goto out_release_dma;
+	}
+
+	platform_set_drvdata(pdev, card);
+
+	dev_info(&pdev->dev, "Atmel ABDAC at 0x%p using %s\n",
+			dac->regs, dac->dma.chan->dev->device.bus_id);
+
+	return retval;
+
+out_release_dma:
+	dma_release_channel(dac->dma.chan);
+	dac->dma.chan = NULL;
+out_unset_card_dev:
+	snd_card_set_dev(card, NULL);
+	free_irq(irq, dac);
+out_unmap_regs:
+	iounmap(dac->regs);
+out_free_card:
+	snd_card_free(card);
+out_put_sample_clk:
+	clk_put(sample_clk);
+	clk_disable(pclk);
+out_put_pclk:
+	clk_put(pclk);
+	return retval;
+}
+
+#ifdef CONFIG_PM
+static int atmel_abdac_suspend(struct platform_device *pdev, pm_message_t msg)
+{
+	struct snd_card *card = platform_get_drvdata(pdev);
+	struct atmel_abdac *dac = card->private_data;
+
+	dw_dma_cyclic_stop(dac->dma.chan);
+	clk_disable(dac->sample_clk);
+	clk_disable(dac->pclk);
+
+	return 0;
+}
+
+static int atmel_abdac_resume(struct platform_device *pdev)
+{
+	struct snd_card *card = platform_get_drvdata(pdev);
+	struct atmel_abdac *dac = card->private_data;
+
+	clk_enable(dac->pclk);
+	clk_enable(dac->sample_clk);
+	if (test_bit(DMA_READY, &dac->flags))
+		dw_dma_cyclic_start(dac->dma.chan);
+
+	return 0;
+}
+#else
+#define atmel_abdac_suspend NULL
+#define atmel_abdac_resume NULL
+#endif
+
+static int __devexit atmel_abdac_remove(struct platform_device *pdev)
+{
+	struct snd_card *card = platform_get_drvdata(pdev);
+	struct atmel_abdac *dac = get_dac(card);
+
+	clk_put(dac->sample_clk);
+	clk_disable(dac->pclk);
+	clk_put(dac->pclk);
+
+	dma_release_channel(dac->dma.chan);
+	dac->dma.chan = NULL;
+	snd_card_set_dev(card, NULL);
+	iounmap(dac->regs);
+	free_irq(dac->irq, dac);
+	snd_card_free(card);
+
+	platform_set_drvdata(pdev, NULL);
+
+	return 0;
+}
+
+static struct platform_driver atmel_abdac_driver = {
+	.remove		= __devexit_p(atmel_abdac_remove),
+	.driver		= {
+		.name	= "atmel_abdac",
+	},
+	.suspend	= atmel_abdac_suspend,
+	.resume		= atmel_abdac_resume,
+};
+
+static int __init atmel_abdac_init(void)
+{
+	return platform_driver_probe(&atmel_abdac_driver,
+			atmel_abdac_probe);
+}
+module_init(atmel_abdac_init);
+
+static void __exit atmel_abdac_exit(void)
+{
+	platform_driver_unregister(&atmel_abdac_driver);
+}
+module_exit(atmel_abdac_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Driver for Atmel Audio Bitstream DAC (ABDAC)");
+MODULE_AUTHOR("Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>");
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
new file mode 100644
index 000000000000..dd72e00e5ae1
--- /dev/null
+++ b/sound/atmel/ac97c.c
@@ -0,0 +1,932 @@
+/*
+ * Driver for the Atmel AC97C controller
+ *
+ * Copyright (C) 2005-2009 Atmel Corporation
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published by
+ * the Free Software Foundation.
+ */
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/bitmap.h>
+#include <linux/dmaengine.h>
+#include <linux/dma-mapping.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/mutex.h>
+#include <linux/gpio.h>
+#include <linux/io.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/ac97_codec.h>
+#include <sound/atmel-ac97c.h>
+#include <sound/memalloc.h>
+
+#include <linux/dw_dmac.h>
+
+#include "ac97c.h"
+
+enum {
+	DMA_TX_READY = 0,
+	DMA_RX_READY,
+	DMA_TX_CHAN_PRESENT,
+	DMA_RX_CHAN_PRESENT,
+};
+
+/* Serialize access to opened variable */
+static DEFINE_MUTEX(opened_mutex);
+
+struct atmel_ac97c_dma {
+	struct dma_chan			*rx_chan;
+	struct dma_chan			*tx_chan;
+};
+
+struct atmel_ac97c {
+	struct clk			*pclk;
+	struct platform_device		*pdev;
+	struct atmel_ac97c_dma		dma;
+
+	struct snd_pcm_substream	*playback_substream;
+	struct snd_pcm_substream	*capture_substream;
+	struct snd_card			*card;
+	struct snd_pcm			*pcm;
+	struct snd_ac97			*ac97;
+	struct snd_ac97_bus		*ac97_bus;
+
+	u64				cur_format;
+	unsigned int			cur_rate;
+	unsigned long			flags;
+	/* Serialize access to opened variable */
+	spinlock_t			lock;
+	void __iomem			*regs;
+	int				opened;
+	int				reset_pin;
+};
+
+#define get_chip(card) ((struct atmel_ac97c *)(card)->private_data)
+
+#define ac97c_writel(chip, reg, val)			\
+	__raw_writel((val), (chip)->regs + AC97C_##reg)
+#define ac97c_readl(chip, reg)				\
+	__raw_readl((chip)->regs + AC97C_##reg)
+
+/* This function is called by the DMA driver. */
+static void atmel_ac97c_dma_playback_period_done(void *arg)
+{
+	struct atmel_ac97c *chip = arg;
+	snd_pcm_period_elapsed(chip->playback_substream);
+}
+
+static void atmel_ac97c_dma_capture_period_done(void *arg)
+{
+	struct atmel_ac97c *chip = arg;
+	snd_pcm_period_elapsed(chip->capture_substream);
+}
+
+static int atmel_ac97c_prepare_dma(struct atmel_ac97c *chip,
+		struct snd_pcm_substream *substream,
+		enum dma_data_direction direction)
+{
+	struct dma_chan			*chan;
+	struct dw_cyclic_desc		*cdesc;
+	struct snd_pcm_runtime		*runtime = substream->runtime;
+	unsigned long			buffer_len, period_len;
+
+	/*
+	 * We don't do DMA on "complex" transfers, i.e. with
+	 * non-halfword-aligned buffers or lengths.
+	 */
+	if (runtime->dma_addr & 1 || runtime->buffer_size & 1) {
+		dev_dbg(&chip->pdev->dev, "too complex transfer\n");
+		return -EINVAL;
+	}
+
+	if (direction == DMA_TO_DEVICE)
+		chan = chip->dma.tx_chan;
+	else
+		chan = chip->dma.rx_chan;
+
+	buffer_len = frames_to_bytes(runtime, runtime->buffer_size);
+	period_len = frames_to_bytes(runtime, runtime->period_size);
+
+	cdesc = dw_dma_cyclic_prep(chan, runtime->dma_addr, buffer_len,
+			period_len, direction);
+	if (IS_ERR(cdesc)) {
+		dev_dbg(&chip->pdev->dev, "could not prepare cyclic DMA\n");
+		return PTR_ERR(cdesc);
+	}
+
+	if (direction == DMA_TO_DEVICE) {
+		cdesc->period_callback = atmel_ac97c_dma_playback_period_done;
+		set_bit(DMA_TX_READY, &chip->flags);
+	} else {
+		cdesc->period_callback = atmel_ac97c_dma_capture_period_done;
+		set_bit(DMA_RX_READY, &chip->flags);
+	}
+
+	cdesc->period_callback_param = chip;
+
+	return 0;
+}
+
+static struct snd_pcm_hardware atmel_ac97c_hw = {
+	.info			= (SNDRV_PCM_INFO_MMAP
+				  | SNDRV_PCM_INFO_MMAP_VALID
+				  | SNDRV_PCM_INFO_INTERLEAVED
+				  | SNDRV_PCM_INFO_BLOCK_TRANSFER
+				  | SNDRV_PCM_INFO_JOINT_DUPLEX
+				  | SNDRV_PCM_INFO_RESUME
+				  | SNDRV_PCM_INFO_PAUSE),
+	.formats		= (SNDRV_PCM_FMTBIT_S16_BE
+				  | SNDRV_PCM_FMTBIT_S16_LE),
+	.rates			= (SNDRV_PCM_RATE_CONTINUOUS),
+	.rate_min		= 4000,
+	.rate_max		= 48000,
+	.channels_min		= 1,
+	.channels_max		= 2,
+	.buffer_bytes_max	= 64 * 4096,
+	.period_bytes_min	= 4096,
+	.period_bytes_max	= 4096,
+	.periods_min		= 4,
+	.periods_max		= 64,
+};
+
+static int atmel_ac97c_playback_open(struct snd_pcm_substream *substream)
+{
+	struct atmel_ac97c *chip = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	mutex_lock(&opened_mutex);
+	chip->opened++;
+	runtime->hw = atmel_ac97c_hw;
+	if (chip->cur_rate) {
+		runtime->hw.rate_min = chip->cur_rate;
+		runtime->hw.rate_max = chip->cur_rate;
+	}
+	if (chip->cur_format)
+		runtime->hw.formats = (1ULL << chip->cur_format);
+	mutex_unlock(&opened_mutex);
+	chip->playback_substream = substream;
+	return 0;
+}
+
+static int atmel_ac97c_capture_open(struct snd_pcm_substream *substream)
+{
+	struct atmel_ac97c *chip = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	mutex_lock(&opened_mutex);
+	chip->opened++;
+	runtime->hw = atmel_ac97c_hw;
+	if (chip->cur_rate) {
+		runtime->hw.rate_min = chip->cur_rate;
+		runtime->hw.rate_max = chip->cur_rate;
+	}
+	if (chip->cur_format)
+		runtime->hw.formats = (1ULL << chip->cur_format);
+	mutex_unlock(&opened_mutex);
+	chip->capture_substream = substream;
+	return 0;
+}
+
+static int atmel_ac97c_playback_close(struct snd_pcm_substream *substream)
+{
+	struct atmel_ac97c *chip = snd_pcm_substream_chip(substream);
+
+	mutex_lock(&opened_mutex);
+	chip->opened--;
+	if (!chip->opened) {
+		chip->cur_rate = 0;
+		chip->cur_format = 0;
+	}
+	mutex_unlock(&opened_mutex);
+
+	chip->playback_substream = NULL;
+
+	return 0;
+}
+
+static int atmel_ac97c_capture_close(struct snd_pcm_substream *substream)
+{
+	struct atmel_ac97c *chip = snd_pcm_substream_chip(substream);
+
+	mutex_lock(&opened_mutex);
+	chip->opened--;
+	if (!chip->opened) {
+		chip->cur_rate = 0;
+		chip->cur_format = 0;
+	}
+	mutex_unlock(&opened_mutex);
+
+	chip->capture_substream = NULL;
+
+	return 0;
+}
+
+static int atmel_ac97c_playback_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *hw_params)
+{
+	struct atmel_ac97c *chip = snd_pcm_substream_chip(substream);
+	int retval;
+
+	retval = snd_pcm_lib_malloc_pages(substream,
+					params_buffer_bytes(hw_params));
+	if (retval < 0)
+		return retval;
+	/* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */
+	if (retval == 1)
+		if (test_and_clear_bit(DMA_TX_READY, &chip->flags))
+			dw_dma_cyclic_free(chip->dma.tx_chan);
+
+	/* Set restrictions to params. */
+	mutex_lock(&opened_mutex);
+	chip->cur_rate = params_rate(hw_params);
+	chip->cur_format = params_format(hw_params);
+	mutex_unlock(&opened_mutex);
+
+	return retval;
+}
+
+static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream,
+		struct snd_pcm_hw_params *hw_params)
+{
+	struct atmel_ac97c *chip = snd_pcm_substream_chip(substream);
+	int retval;
+
+	retval = snd_pcm_lib_malloc_pages(substream,
+					params_buffer_bytes(hw_params));
+	if (retval < 0)
+		return retval;
+	/* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */
+	if (retval == 1)
+		if (test_and_clear_bit(DMA_RX_READY, &chip->flags))
+			dw_dma_cyclic_free(chip->dma.rx_chan);
+
+	/* Set restrictions to params. */
+	mutex_lock(&opened_mutex);
+	chip->cur_rate = params_rate(hw_params);
+	chip->cur_format = params_format(hw_params);
+	mutex_unlock(&opened_mutex);
+
+	return retval;
+}
+
+static int atmel_ac97c_playback_hw_free(struct snd_pcm_substream *substream)
+{
+	struct atmel_ac97c *chip = snd_pcm_substream_chip(substream);
+	if (test_and_clear_bit(DMA_TX_READY, &chip->flags))
+		dw_dma_cyclic_free(chip->dma.tx_chan);
+	return snd_pcm_lib_free_pages(substream);
+}
+
+static int atmel_ac97c_capture_hw_free(struct snd_pcm_substream *substream)
+{
+	struct atmel_ac97c *chip = snd_pcm_substream_chip(substream);
+	if (test_and_clear_bit(DMA_RX_READY, &chip->flags))
+		dw_dma_cyclic_free(chip->dma.rx_chan);
+	return snd_pcm_lib_free_pages(substream);
+}
+
+static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream)
+{
+	struct atmel_ac97c *chip = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned long word = 0;
+	int retval;
+
+	/* assign channels to AC97C channel A */
+	switch (runtime->channels) {
+	case 1:
+		word |= AC97C_CH_ASSIGN(PCM_LEFT, A);
+		break;
+	case 2:
+		word |= AC97C_CH_ASSIGN(PCM_LEFT, A)
+			| AC97C_CH_ASSIGN(PCM_RIGHT, A);
+		break;
+	default:
+		/* TODO: support more than two channels */
+		return -EINVAL;
+		break;
+	}
+	ac97c_writel(chip, OCA, word);
+
+	/* configure sample format and size */
+	word = AC97C_CMR_DMAEN | AC97C_CMR_SIZE_16;
+
+	switch (runtime->format) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		word |= AC97C_CMR_CEM_LITTLE;
+		break;
+	case SNDRV_PCM_FORMAT_S16_BE: /* fall through */
+	default:
+		word &= ~(AC97C_CMR_CEM_LITTLE);
+		break;
+	}
+
+	ac97c_writel(chip, CAMR, word);
+
+	/* set variable rate if needed */
+	if (runtime->rate != 48000) {
+		word = ac97c_readl(chip, MR);
+		word |= AC97C_MR_VRA;
+		ac97c_writel(chip, MR, word);
+	} else {
+		word = ac97c_readl(chip, MR);
+		word &= ~(AC97C_MR_VRA);
+		ac97c_writel(chip, MR, word);
+	}
+
+	retval = snd_ac97_set_rate(chip->ac97, AC97_PCM_FRONT_DAC_RATE,
+			runtime->rate);
+	if (retval)
+		dev_dbg(&chip->pdev->dev, "could not set rate %d Hz\n",
+				runtime->rate);
+
+	if (!test_bit(DMA_TX_READY, &chip->flags))
+		retval = atmel_ac97c_prepare_dma(chip, substream,
+				DMA_TO_DEVICE);
+
+	return retval;
+}
+
+static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream)
+{
+	struct atmel_ac97c *chip = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned long word = 0;
+	int retval;
+
+	/* assign channels to AC97C channel A */
+	switch (runtime->channels) {
+	case 1:
+		word |= AC97C_CH_ASSIGN(PCM_LEFT, A);
+		break;
+	case 2:
+		word |= AC97C_CH_ASSIGN(PCM_LEFT, A)
+			| AC97C_CH_ASSIGN(PCM_RIGHT, A);
+		break;
+	default:
+		/* TODO: support more than two channels */
+		return -EINVAL;
+		break;
+	}
+	ac97c_writel(chip, ICA, word);
+
+	/* configure sample format and size */
+	word = AC97C_CMR_DMAEN | AC97C_CMR_SIZE_16;
+
+	switch (runtime->format) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		word |= AC97C_CMR_CEM_LITTLE;
+		break;
+	case SNDRV_PCM_FORMAT_S16_BE: /* fall through */
+	default:
+		word &= ~(AC97C_CMR_CEM_LITTLE);
+		break;
+	}
+
+	ac97c_writel(chip, CAMR, word);
+
+	/* set variable rate if needed */
+	if (runtime->rate != 48000) {
+		word = ac97c_readl(chip, MR);
+		word |= AC97C_MR_VRA;
+		ac97c_writel(chip, MR, word);
+	} else {
+		word = ac97c_readl(chip, MR);
+		word &= ~(AC97C_MR_VRA);
+		ac97c_writel(chip, MR, word);
+	}
+
+	retval = snd_ac97_set_rate(chip->ac97, AC97_PCM_LR_ADC_RATE,
+			runtime->rate);
+	if (retval)
+		dev_dbg(&chip->pdev->dev, "could not set rate %d Hz\n",
+				runtime->rate);
+
+	if (!test_bit(DMA_RX_READY, &chip->flags))
+		retval = atmel_ac97c_prepare_dma(chip, substream,
+				DMA_FROM_DEVICE);
+
+	return retval;
+}
+
+static int
+atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct atmel_ac97c *chip = snd_pcm_substream_chip(substream);
+	unsigned long camr;
+	int retval = 0;
+
+	camr = ac97c_readl(chip, CAMR);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */
+	case SNDRV_PCM_TRIGGER_RESUME: /* fall through */
+	case SNDRV_PCM_TRIGGER_START:
+		retval = dw_dma_cyclic_start(chip->dma.tx_chan);
+		if (retval)
+			goto out;
+		camr |= AC97C_CMR_CENA;
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */
+	case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */
+	case SNDRV_PCM_TRIGGER_STOP:
+		dw_dma_cyclic_stop(chip->dma.tx_chan);
+		if (chip->opened <= 1)
+			camr &= ~AC97C_CMR_CENA;
+		break;
+	default:
+		retval = -EINVAL;
+		goto out;
+	}
+
+	ac97c_writel(chip, CAMR, camr);
+out:
+	return retval;
+}
+
+static int
+atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct atmel_ac97c *chip = snd_pcm_substream_chip(substream);
+	unsigned long camr;
+	int retval = 0;
+
+	camr = ac97c_readl(chip, CAMR);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */
+	case SNDRV_PCM_TRIGGER_RESUME: /* fall through */
+	case SNDRV_PCM_TRIGGER_START:
+		retval = dw_dma_cyclic_start(chip->dma.rx_chan);
+		if (retval)
+			goto out;
+		camr |= AC97C_CMR_CENA;
+		break;
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */
+	case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */
+	case SNDRV_PCM_TRIGGER_STOP:
+		dw_dma_cyclic_stop(chip->dma.rx_chan);
+		if (chip->opened <= 1)
+			camr &= ~AC97C_CMR_CENA;
+		break;
+	default:
+		retval = -EINVAL;
+		break;
+	}
+
+	ac97c_writel(chip, CAMR, camr);
+out:
+	return retval;
+}
+
+static snd_pcm_uframes_t
+atmel_ac97c_playback_pointer(struct snd_pcm_substream *substream)
+{
+	struct atmel_ac97c	*chip = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime	*runtime = substream->runtime;
+	snd_pcm_uframes_t	frames;
+	unsigned long		bytes;
+
+	bytes = dw_dma_get_src_addr(chip->dma.tx_chan);
+	bytes -= runtime->dma_addr;
+
+	frames = bytes_to_frames(runtime, bytes);
+	if (frames >= runtime->buffer_size)
+		frames -= runtime->buffer_size;
+	return frames;
+}
+
+static snd_pcm_uframes_t
+atmel_ac97c_capture_pointer(struct snd_pcm_substream *substream)
+{
+	struct atmel_ac97c	*chip = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime	*runtime = substream->runtime;
+	snd_pcm_uframes_t	frames;
+	unsigned long		bytes;
+
+	bytes = dw_dma_get_dst_addr(chip->dma.rx_chan);
+	bytes -= runtime->dma_addr;
+
+	frames = bytes_to_frames(runtime, bytes);
+	if (frames >= runtime->buffer_size)
+		frames -= runtime->buffer_size;
+	return frames;
+}
+
+static struct snd_pcm_ops atmel_ac97_playback_ops = {
+	.open		= atmel_ac97c_playback_open,
+	.close		= atmel_ac97c_playback_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= atmel_ac97c_playback_hw_params,
+	.hw_free	= atmel_ac97c_playback_hw_free,
+	.prepare	= atmel_ac97c_playback_prepare,
+	.trigger	= atmel_ac97c_playback_trigger,
+	.pointer	= atmel_ac97c_playback_pointer,
+};
+
+static struct snd_pcm_ops atmel_ac97_capture_ops = {
+	.open		= atmel_ac97c_capture_open,
+	.close		= atmel_ac97c_capture_close,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= atmel_ac97c_capture_hw_params,
+	.hw_free	= atmel_ac97c_capture_hw_free,
+	.prepare	= atmel_ac97c_capture_prepare,
+	.trigger	= atmel_ac97c_capture_trigger,
+	.pointer	= atmel_ac97c_capture_pointer,
+};
+
+static int __devinit atmel_ac97c_pcm_new(struct atmel_ac97c *chip)
+{
+	struct snd_pcm		*pcm;
+	struct snd_pcm_hardware	hw = atmel_ac97c_hw;
+	int			capture, playback, retval;
+
+	capture = test_bit(DMA_RX_CHAN_PRESENT, &chip->flags);
+	playback = test_bit(DMA_TX_CHAN_PRESENT, &chip->flags);
+
+	retval = snd_pcm_new(chip->card, chip->card->shortname,
+			chip->pdev->id, playback, capture, &pcm);
+	if (retval)
+		return retval;
+
+	if (capture)
+		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+				&atmel_ac97_capture_ops);
+	if (playback)
+		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+				&atmel_ac97_playback_ops);
+
+	retval = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+			&chip->pdev->dev, hw.periods_min * hw.period_bytes_min,
+			hw.buffer_bytes_max);
+	if (retval)
+		return retval;
+
+	pcm->private_data = chip;
+	pcm->info_flags = 0;
+	strcpy(pcm->name, chip->card->shortname);
+	chip->pcm = pcm;
+
+	return 0;
+}
+
+static int atmel_ac97c_mixer_new(struct atmel_ac97c *chip)
+{
+	struct snd_ac97_template template;
+	memset(&template, 0, sizeof(template));
+	template.private_data = chip;
+	return snd_ac97_mixer(chip->ac97_bus, &template, &chip->ac97);
+}
+
+static void atmel_ac97c_write(struct snd_ac97 *ac97, unsigned short reg,
+		unsigned short val)
+{
+	struct atmel_ac97c *chip = get_chip(ac97);
+	unsigned long word;
+	int timeout = 40;
+
+	word = (reg & 0x7f) << 16 | val;
+
+	do {
+		if (ac97c_readl(chip, COSR) & AC97C_CSR_TXRDY) {
+			ac97c_writel(chip, COTHR, word);
+			return;
+		}
+		udelay(1);
+	} while (--timeout);
+
+	dev_dbg(&chip->pdev->dev, "codec write timeout\n");
+}
+
+static unsigned short atmel_ac97c_read(struct snd_ac97 *ac97,
+		unsigned short reg)
+{
+	struct atmel_ac97c *chip = get_chip(ac97);
+	unsigned long word;
+	int timeout = 40;
+	int write = 10;
+
+	word = (0x80 | (reg & 0x7f)) << 16;
+
+	if ((ac97c_readl(chip, COSR) & AC97C_CSR_RXRDY) != 0)
+		ac97c_readl(chip, CORHR);
+
+retry_write:
+	timeout = 40;
+
+	do {
+		if ((ac97c_readl(chip, COSR) & AC97C_CSR_TXRDY) != 0) {
+			ac97c_writel(chip, COTHR, word);
+			goto read_reg;
+		}
+		udelay(10);
+	} while (--timeout);
+
+	if (!--write)
+		goto timed_out;
+	goto retry_write;
+
+read_reg:
+	do {
+		if ((ac97c_readl(chip, COSR) & AC97C_CSR_RXRDY) != 0) {
+			unsigned short val = ac97c_readl(chip, CORHR);
+			return val;
+		}
+		udelay(10);
+	} while (--timeout);
+
+	if (!--write)
+		goto timed_out;
+	goto retry_write;
+
+timed_out:
+	dev_dbg(&chip->pdev->dev, "codec read timeout\n");
+	return 0xffff;
+}
+
+static bool filter(struct dma_chan *chan, void *slave)
+{
+	struct dw_dma_slave *dws = slave;
+
+	if (dws->dma_dev == chan->device->dev) {
+		chan->private = dws;
+		return true;
+	} else
+		return false;
+}
+
+static void atmel_ac97c_reset(struct atmel_ac97c *chip)
+{
+	ac97c_writel(chip, MR, AC97C_MR_WRST);
+
+	if (gpio_is_valid(chip->reset_pin)) {
+		gpio_set_value(chip->reset_pin, 0);
+		/* AC97 v2.2 specifications says minimum 1 us. */
+		udelay(10);
+		gpio_set_value(chip->reset_pin, 1);
+	}
+
+	udelay(1);
+	ac97c_writel(chip, MR, AC97C_MR_ENA);
+}
+
+static int __devinit atmel_ac97c_probe(struct platform_device *pdev)
+{
+	struct snd_card			*card;
+	struct atmel_ac97c		*chip;
+	struct resource			*regs;
+	struct ac97c_platform_data	*pdata;
+	struct clk			*pclk;
+	static struct snd_ac97_bus_ops	ops = {
+		.write	= atmel_ac97c_write,
+		.read	= atmel_ac97c_read,
+	};
+	int				retval;
+
+	regs = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!regs) {
+		dev_dbg(&pdev->dev, "no memory resource\n");
+		return -ENXIO;
+	}
+
+	pdata = pdev->dev.platform_data;
+	if (!pdata) {
+		dev_dbg(&pdev->dev, "no platform data\n");
+		return -ENXIO;
+	}
+
+	pclk = clk_get(&pdev->dev, "pclk");
+	if (IS_ERR(pclk)) {
+		dev_dbg(&pdev->dev, "no peripheral clock\n");
+		return PTR_ERR(pclk);
+	}
+	clk_enable(pclk);
+
+	retval = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1,
+			THIS_MODULE, sizeof(struct atmel_ac97c), &card);
+	if (retval) {
+		dev_dbg(&pdev->dev, "could not create sound card device\n");
+		goto err_snd_card_new;
+	}
+
+	chip = get_chip(card);
+
+	spin_lock_init(&chip->lock);
+
+	strcpy(card->driver, "Atmel AC97C");
+	strcpy(card->shortname, "Atmel AC97C");
+	sprintf(card->longname, "Atmel AC97 controller");
+
+	chip->card = card;
+	chip->pclk = pclk;
+	chip->pdev = pdev;
+	chip->regs = ioremap(regs->start, regs->end - regs->start + 1);
+
+	if (!chip->regs) {
+		dev_dbg(&pdev->dev, "could not remap register memory\n");
+		goto err_ioremap;
+	}
+
+	if (gpio_is_valid(pdata->reset_pin)) {
+		if (gpio_request(pdata->reset_pin, "reset_pin")) {
+			dev_dbg(&pdev->dev, "reset pin not available\n");
+			chip->reset_pin = -ENODEV;
+		} else {
+			gpio_direction_output(pdata->reset_pin, 1);
+			chip->reset_pin = pdata->reset_pin;
+		}
+	}
+
+	snd_card_set_dev(card, &pdev->dev);
+
+	retval = snd_ac97_bus(card, 0, &ops, chip, &chip->ac97_bus);
+	if (retval) {
+		dev_dbg(&pdev->dev, "could not register on ac97 bus\n");
+		goto err_ac97_bus;
+	}
+
+	atmel_ac97c_reset(chip);
+
+	retval = atmel_ac97c_mixer_new(chip);
+	if (retval) {
+		dev_dbg(&pdev->dev, "could not register ac97 mixer\n");
+		goto err_ac97_bus;
+	}
+
+	if (pdata->rx_dws.dma_dev) {
+		struct dw_dma_slave *dws = &pdata->rx_dws;
+		dma_cap_mask_t mask;
+
+		dws->rx_reg = regs->start + AC97C_CARHR + 2;
+
+		dma_cap_zero(mask);
+		dma_cap_set(DMA_SLAVE, mask);
+
+		chip->dma.rx_chan = dma_request_channel(mask, filter, dws);
+
+		dev_info(&chip->pdev->dev, "using %s for DMA RX\n",
+					chip->dma.rx_chan->dev->device.bus_id);
+		set_bit(DMA_RX_CHAN_PRESENT, &chip->flags);
+	}
+
+	if (pdata->tx_dws.dma_dev) {
+		struct dw_dma_slave *dws = &pdata->tx_dws;
+		dma_cap_mask_t mask;
+
+		dws->tx_reg = regs->start + AC97C_CATHR + 2;
+
+		dma_cap_zero(mask);
+		dma_cap_set(DMA_SLAVE, mask);
+
+		chip->dma.tx_chan = dma_request_channel(mask, filter, dws);
+
+		dev_info(&chip->pdev->dev, "using %s for DMA TX\n",
+					chip->dma.tx_chan->dev->device.bus_id);
+		set_bit(DMA_TX_CHAN_PRESENT, &chip->flags);
+	}
+
+	if (!test_bit(DMA_RX_CHAN_PRESENT, &chip->flags) &&
+			!test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) {
+		dev_dbg(&pdev->dev, "DMA not available\n");
+		retval = -ENODEV;
+		goto err_dma;
+	}
+
+	retval = atmel_ac97c_pcm_new(chip);
+	if (retval) {
+		dev_dbg(&pdev->dev, "could not register ac97 pcm device\n");
+		goto err_dma;
+	}
+
+	retval = snd_card_register(card);
+	if (retval) {
+		dev_dbg(&pdev->dev, "could not register sound card\n");
+		goto err_ac97_bus;
+	}
+
+	platform_set_drvdata(pdev, card);
+
+	dev_info(&pdev->dev, "Atmel AC97 controller at 0x%p\n",
+			chip->regs);
+
+	return 0;
+
+err_dma:
+	if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags))
+		dma_release_channel(chip->dma.rx_chan);
+	if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags))
+		dma_release_channel(chip->dma.tx_chan);
+	clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags);
+	clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags);
+	chip->dma.rx_chan = NULL;
+	chip->dma.tx_chan = NULL;
+err_ac97_bus:
+	snd_card_set_dev(card, NULL);
+
+	if (gpio_is_valid(chip->reset_pin))
+		gpio_free(chip->reset_pin);
+
+	iounmap(chip->regs);
+err_ioremap:
+	snd_card_free(card);
+err_snd_card_new:
+	clk_disable(pclk);
+	clk_put(pclk);
+	return retval;
+}
+
+#ifdef CONFIG_PM
+static int atmel_ac97c_suspend(struct platform_device *pdev, pm_message_t msg)
+{
+	struct snd_card *card = platform_get_drvdata(pdev);
+	struct atmel_ac97c *chip = card->private_data;
+
+	if (test_bit(DMA_RX_READY, &chip->flags))
+		dw_dma_cyclic_stop(chip->dma.rx_chan);
+	if (test_bit(DMA_TX_READY, &chip->flags))
+		dw_dma_cyclic_stop(chip->dma.tx_chan);
+	clk_disable(chip->pclk);
+
+	return 0;
+}
+
+static int atmel_ac97c_resume(struct platform_device *pdev)
+{
+	struct snd_card *card = platform_get_drvdata(pdev);
+	struct atmel_ac97c *chip = card->private_data;
+
+	clk_enable(chip->pclk);
+	if (test_bit(DMA_RX_READY, &chip->flags))
+		dw_dma_cyclic_start(chip->dma.rx_chan);
+	if (test_bit(DMA_TX_READY, &chip->flags))
+		dw_dma_cyclic_start(chip->dma.tx_chan);
+
+	return 0;
+}
+#else
+#define atmel_ac97c_suspend NULL
+#define atmel_ac97c_resume NULL
+#endif
+
+static int __devexit atmel_ac97c_remove(struct platform_device *pdev)
+{
+	struct snd_card *card = platform_get_drvdata(pdev);
+	struct atmel_ac97c *chip = get_chip(card);
+
+	if (gpio_is_valid(chip->reset_pin))
+		gpio_free(chip->reset_pin);
+
+	clk_disable(chip->pclk);
+	clk_put(chip->pclk);
+	iounmap(chip->regs);
+
+	if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags))
+		dma_release_channel(chip->dma.rx_chan);
+	if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags))
+		dma_release_channel(chip->dma.tx_chan);
+	clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags);
+	clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags);
+	chip->dma.rx_chan = NULL;
+	chip->dma.tx_chan = NULL;
+
+	snd_card_set_dev(card, NULL);
+	snd_card_free(card);
+
+	platform_set_drvdata(pdev, NULL);
+
+	return 0;
+}
+
+static struct platform_driver atmel_ac97c_driver = {
+	.remove		= __devexit_p(atmel_ac97c_remove),
+	.driver		= {
+		.name	= "atmel_ac97c",
+	},
+	.suspend	= atmel_ac97c_suspend,
+	.resume		= atmel_ac97c_resume,
+};
+
+static int __init atmel_ac97c_init(void)
+{
+	return platform_driver_probe(&atmel_ac97c_driver,
+			atmel_ac97c_probe);
+}
+module_init(atmel_ac97c_init);
+
+static void __exit atmel_ac97c_exit(void)
+{
+	platform_driver_unregister(&atmel_ac97c_driver);
+}
+module_exit(atmel_ac97c_exit);
+
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Driver for Atmel AC97 controller");
+MODULE_AUTHOR("Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>");
diff --git a/sound/atmel/ac97c.h b/sound/atmel/ac97c.h
new file mode 100644
index 000000000000..c17bd5825980
--- /dev/null
+++ b/sound/atmel/ac97c.h
@@ -0,0 +1,71 @@
+/*
+ * Register definitions for the Atmel AC97C controller
+ *
+ * Copyright (C) 2005-2009 Atmel Corporation
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License version 2 as published
+ * by the Free Software Foundation.
+ */
+#ifndef __SOUND_ATMEL_AC97C_H
+#define __SOUND_ATMEL_AC97C_H
+
+#define AC97C_MR		0x08
+#define AC97C_ICA		0x10
+#define AC97C_OCA		0x14
+#define AC97C_CARHR		0x20
+#define AC97C_CATHR		0x24
+#define AC97C_CASR		0x28
+#define AC97C_CAMR		0x2c
+#define AC97C_CBRHR		0x30
+#define AC97C_CBTHR		0x34
+#define AC97C_CBSR		0x38
+#define AC97C_CBMR		0x3c
+#define AC97C_CORHR		0x40
+#define AC97C_COTHR		0x44
+#define AC97C_COSR		0x48
+#define AC97C_COMR		0x4c
+#define AC97C_SR		0x50
+#define AC97C_IER		0x54
+#define AC97C_IDR		0x58
+#define AC97C_IMR		0x5c
+#define AC97C_VERSION		0xfc
+
+#define AC97C_CATPR		PDC_TPR
+#define AC97C_CATCR		PDC_TCR
+#define AC97C_CATNPR		PDC_TNPR
+#define AC97C_CATNCR		PDC_TNCR
+#define AC97C_CARPR		PDC_RPR
+#define AC97C_CARCR		PDC_RCR
+#define AC97C_CARNPR		PDC_RNPR
+#define AC97C_CARNCR		PDC_RNCR
+#define AC97C_PTCR		PDC_PTCR
+
+#define AC97C_MR_ENA		(1 << 0)
+#define AC97C_MR_WRST		(1 << 1)
+#define AC97C_MR_VRA		(1 << 2)
+
+#define AC97C_CSR_TXRDY		(1 << 0)
+#define AC97C_CSR_UNRUN		(1 << 2)
+#define AC97C_CSR_RXRDY		(1 << 4)
+#define AC97C_CSR_ENDTX		(1 << 10)
+#define AC97C_CSR_ENDRX		(1 << 14)
+
+#define AC97C_CMR_SIZE_20	(0 << 16)
+#define AC97C_CMR_SIZE_18	(1 << 16)
+#define AC97C_CMR_SIZE_16	(2 << 16)
+#define AC97C_CMR_SIZE_10	(3 << 16)
+#define AC97C_CMR_CEM_LITTLE	(1 << 18)
+#define AC97C_CMR_CEM_BIG	(0 << 18)
+#define AC97C_CMR_CENA		(1 << 21)
+#define AC97C_CMR_DMAEN		(1 << 22)
+
+#define AC97C_SR_CAEVT		(1 << 3)
+
+#define AC97C_CH_ASSIGN(slot, channel)					\
+	(AC97C_CHANNEL_##channel << (3 * (AC97_SLOT_##slot - 3)))
+#define AC97C_CHANNEL_NONE	0x0
+#define AC97C_CHANNEL_A		0x1
+#define AC97C_CHANNEL_B		0x2
+
+#endif /* __SOUND_ATMEL_AC97C_H */
diff --git a/sound/core/control.c b/sound/core/control.c
index 636b3b52ef8b..4b20fa2b7e6d 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -1373,12 +1373,9 @@ EXPORT_SYMBOL(snd_ctl_unregister_ioctl_compat);
 static int snd_ctl_fasync(int fd, struct file * file, int on)
 {
 	struct snd_ctl_file *ctl;
-	int err;
+
 	ctl = file->private_data;
-	err = fasync_helper(fd, file, on, &ctl->fasync);
-	if (err < 0)
-		return err;
-	return 0;
+	return fasync_helper(fd, file, on, &ctl->fasync);
 }
 
 /*
diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c
index 195cafc5a553..a70ee7f1ed98 100644
--- a/sound/core/hwdep.c
+++ b/sound/core/hwdep.c
@@ -99,9 +99,6 @@ static int snd_hwdep_open(struct inode *inode, struct file * file)
 	if (hw == NULL)
 		return -ENODEV;
 
-	if (!hw->ops.open)
-		return -ENXIO;
-
 	if (!try_module_get(hw->card->module))
 		return -EFAULT;
 
@@ -113,6 +110,10 @@ static int snd_hwdep_open(struct inode *inode, struct file * file)
 			err = -EBUSY;
 			break;
 		}
+		if (!hw->ops.open) {
+			err = 0;
+			break;
+		}
 		err = hw->ops.open(hw, file);
 		if (err >= 0)
 			break;
@@ -151,7 +152,7 @@ static int snd_hwdep_open(struct inode *inode, struct file * file)
 
 static int snd_hwdep_release(struct inode *inode, struct file * file)
 {
-	int err = -ENXIO;
+	int err = 0;
 	struct snd_hwdep *hw = file->private_data;
 	struct module *mod = hw->card->module;
 
diff --git a/sound/core/init.c b/sound/core/init.c
index 0d5520c415d3..fd56afe846ed 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -121,31 +121,44 @@ static inline int init_info_for_card(struct snd_card *card)
 #endif
 
 /**
- *  snd_card_new - create and initialize a soundcard structure
+ *  snd_card_create - create and initialize a soundcard structure
  *  @idx: card index (address) [0 ... (SNDRV_CARDS-1)]
  *  @xid: card identification (ASCII string)
  *  @module: top level module for locking
  *  @extra_size: allocate this extra size after the main soundcard structure
+ *  @card_ret: the pointer to store the created card instance
  *
  *  Creates and initializes a soundcard structure.
  *
- *  Returns kmallocated snd_card structure. Creates the ALSA control interface
- *  (which is blocked until snd_card_register function is called).
+ *  The function allocates snd_card instance via kzalloc with the given
+ *  space for the driver to use freely.  The allocated struct is stored
+ *  in the given card_ret pointer.
+ *
+ *  Returns zero if successful or a negative error code.
  */
-struct snd_card *snd_card_new(int idx, const char *xid,
-			 struct module *module, int extra_size)
+int snd_card_create(int idx, const char *xid,
+		    struct module *module, int extra_size,
+		    struct snd_card **card_ret)
 {
 	struct snd_card *card;
 	int err, idx2;
 
+	if (snd_BUG_ON(!card_ret))
+		return -EINVAL;
+	*card_ret = NULL;
+
 	if (extra_size < 0)
 		extra_size = 0;
 	card = kzalloc(sizeof(*card) + extra_size, GFP_KERNEL);
-	if (card == NULL)
-		return NULL;
+	if (!card)
+		return -ENOMEM;
 	if (xid) {
-		if (!snd_info_check_reserved_words(xid))
+		if (!snd_info_check_reserved_words(xid)) {
+			snd_printk(KERN_ERR
+				   "given id string '%s' is reserved.\n", xid);
+			err = -EBUSY;
 			goto __error;
+		}
 		strlcpy(card->id, xid, sizeof(card->id));
 	}
 	err = 0;
@@ -195,6 +208,7 @@ struct snd_card *snd_card_new(int idx, const char *xid,
 	INIT_LIST_HEAD(&card->controls);
 	INIT_LIST_HEAD(&card->ctl_files);
 	spin_lock_init(&card->files_lock);
+	INIT_LIST_HEAD(&card->files_list);
 	init_waitqueue_head(&card->shutdown_sleep);
 #ifdef CONFIG_PM
 	mutex_init(&card->power_lock);
@@ -202,26 +216,28 @@ struct snd_card *snd_card_new(int idx, const char *xid,
 #endif
 	/* the control interface cannot be accessed from the user space until */
 	/* snd_cards_bitmask and snd_cards are set with snd_card_register */
-	if ((err = snd_ctl_create(card)) < 0) {
-		snd_printd("unable to register control minors\n");
+	err = snd_ctl_create(card);
+	if (err < 0) {
+		snd_printk(KERN_ERR "unable to register control minors\n");
 		goto __error;
 	}
-	if ((err = snd_info_card_create(card)) < 0) {
-		snd_printd("unable to create card info\n");
+	err = snd_info_card_create(card);
+	if (err < 0) {
+		snd_printk(KERN_ERR "unable to create card info\n");
 		goto __error_ctl;
 	}
 	if (extra_size > 0)
 		card->private_data = (char *)card + sizeof(struct snd_card);
-	return card;
+	*card_ret = card;
+	return 0;
 
       __error_ctl:
 	snd_device_free_all(card, SNDRV_DEV_CMD_PRE);
       __error:
 	kfree(card);
-      	return NULL;
+  	return err;
 }
-
-EXPORT_SYMBOL(snd_card_new);
+EXPORT_SYMBOL(snd_card_create);
 
 /* return non-zero if a card is already locked */
 int snd_card_locked(int card)
@@ -259,6 +275,7 @@ static int snd_disconnect_release(struct inode *inode, struct file *file)
 	list_for_each_entry(_df, &shutdown_files, shutdown_list) {
 		if (_df->file == file) {
 			df = _df;
+			list_del_init(&df->shutdown_list);
 			break;
 		}
 	}
@@ -347,8 +364,7 @@ int snd_card_disconnect(struct snd_card *card)
 	/* phase 2: replace file->f_op with special dummy operations */
 	
 	spin_lock(&card->files_lock);
-	mfile = card->files;
-	while (mfile) {
+	list_for_each_entry(mfile, &card->files_list, list) {
 		file = mfile->file;
 
 		/* it's critical part, use endless loop */
@@ -361,8 +377,6 @@ int snd_card_disconnect(struct snd_card *card)
 
 		mfile->file->f_op = &snd_shutdown_f_ops;
 		fops_get(mfile->file->f_op);
-		
-		mfile = mfile->next;
 	}
 	spin_unlock(&card->files_lock);	
 
@@ -442,7 +456,7 @@ int snd_card_free_when_closed(struct snd_card *card)
 		return ret;
 
 	spin_lock(&card->files_lock);
-	if (card->files == NULL)
+	if (list_empty(&card->files_list))
 		free_now = 1;
 	else
 		card->free_on_last_close = 1;
@@ -462,7 +476,7 @@ int snd_card_free(struct snd_card *card)
 		return ret;
 
 	/* wait, until all devices are ready for the free operation */
-	wait_event(card->shutdown_sleep, card->files == NULL);
+	wait_event(card->shutdown_sleep, list_empty(&card->files_list));
 	snd_card_do_free(card);
 	return 0;
 }
@@ -809,15 +823,13 @@ int snd_card_file_add(struct snd_card *card, struct file *file)
 		return -ENOMEM;
 	mfile->file = file;
 	mfile->disconnected_f_op = NULL;
-	mfile->next = NULL;
 	spin_lock(&card->files_lock);
 	if (card->shutdown) {
 		spin_unlock(&card->files_lock);
 		kfree(mfile);
 		return -ENODEV;
 	}
-	mfile->next = card->files;
-	card->files = mfile;
+	list_add(&mfile->list, &card->files_list);
 	spin_unlock(&card->files_lock);
 	return 0;
 }
@@ -839,29 +851,20 @@ EXPORT_SYMBOL(snd_card_file_add);
  */
 int snd_card_file_remove(struct snd_card *card, struct file *file)
 {
-	struct snd_monitor_file *mfile, *pfile = NULL;
+	struct snd_monitor_file *mfile, *found = NULL;
 	int last_close = 0;
 
 	spin_lock(&card->files_lock);
-	mfile = card->files;
-	while (mfile) {
+	list_for_each_entry(mfile, &card->files_list, list) {
 		if (mfile->file == file) {
-			if (pfile)
-				pfile->next = mfile->next;
-			else
-				card->files = mfile->next;
+			list_del(&mfile->list);
+			if (mfile->disconnected_f_op)
+				fops_put(mfile->disconnected_f_op);
+			found = mfile;
 			break;
 		}
-		pfile = mfile;
-		mfile = mfile->next;
-	}
-	if (mfile && mfile->disconnected_f_op) {
-		fops_put(mfile->disconnected_f_op);
-		spin_lock(&shutdown_lock);
-		list_del(&mfile->shutdown_list);
-		spin_unlock(&shutdown_lock);
 	}
-	if (card->files == NULL)
+	if (list_empty(&card->files_list))
 		last_close = 1;
 	spin_unlock(&card->files_lock);
 	if (last_close) {
@@ -869,11 +872,11 @@ int snd_card_file_remove(struct snd_card *card, struct file *file)
 		if (card->free_on_last_close)
 			snd_card_do_free(card);
 	}
-	if (!mfile) {
+	if (!found) {
 		snd_printk(KERN_ERR "ALSA card file remove problem (%p)\n", file);
 		return -ENOENT;
 	}
-	kfree(mfile);
+	kfree(found);
 	return 0;
 }
 
diff --git a/sound/core/jack.c b/sound/core/jack.c
index 077a85262c1c..c8254c667c62 100644
--- a/sound/core/jack.c
+++ b/sound/core/jack.c
@@ -23,6 +23,14 @@
 #include <sound/jack.h>
 #include <sound/core.h>
 
+static int jack_types[] = {
+	SW_HEADPHONE_INSERT,
+	SW_MICROPHONE_INSERT,
+	SW_LINEOUT_INSERT,
+	SW_JACK_PHYSICAL_INSERT,
+	SW_VIDEOOUT_INSERT,
+};
+
 static int snd_jack_dev_free(struct snd_device *device)
 {
 	struct snd_jack *jack = device->device_data;
@@ -79,6 +87,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type,
 {
 	struct snd_jack *jack;
 	int err;
+	int i;
 	static struct snd_device_ops ops = {
 		.dev_free = snd_jack_dev_free,
 		.dev_register = snd_jack_dev_register,
@@ -100,18 +109,10 @@ int snd_jack_new(struct snd_card *card, const char *id, int type,
 
 	jack->type = type;
 
-	if (type & SND_JACK_HEADPHONE)
-		input_set_capability(jack->input_dev, EV_SW,
-				     SW_HEADPHONE_INSERT);
-	if (type & SND_JACK_LINEOUT)
-		input_set_capability(jack->input_dev, EV_SW,
-				     SW_LINEOUT_INSERT);
-	if (type & SND_JACK_MICROPHONE)
-		input_set_capability(jack->input_dev, EV_SW,
-				     SW_MICROPHONE_INSERT);
-	if (type & SND_JACK_MECHANICAL)
-		input_set_capability(jack->input_dev, EV_SW,
-				     SW_JACK_PHYSICAL_INSERT);
+	for (i = 0; i < ARRAY_SIZE(jack_types); i++)
+		if (type & (1 << i))
+			input_set_capability(jack->input_dev, EV_SW,
+					     jack_types[i]);
 
 	err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops);
 	if (err < 0)
@@ -154,21 +155,17 @@ EXPORT_SYMBOL(snd_jack_set_parent);
  */
 void snd_jack_report(struct snd_jack *jack, int status)
 {
+	int i;
+
 	if (!jack)
 		return;
 
-	if (jack->type & SND_JACK_HEADPHONE)
-		input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT,
-				    status & SND_JACK_HEADPHONE);
-	if (jack->type & SND_JACK_LINEOUT)
-		input_report_switch(jack->input_dev, SW_LINEOUT_INSERT,
-				    status & SND_JACK_LINEOUT);
-	if (jack->type & SND_JACK_MICROPHONE)
-		input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT,
-				    status & SND_JACK_MICROPHONE);
-	if (jack->type & SND_JACK_MECHANICAL)
-		input_report_switch(jack->input_dev, SW_JACK_PHYSICAL_INSERT,
-				    status & SND_JACK_MECHANICAL);
+	for (i = 0; i < ARRAY_SIZE(jack_types); i++) {
+		int testbit = 1 << i;
+		if (jack->type & testbit)
+			input_report_switch(jack->input_dev, jack_types[i],
+					    status & testbit);
+	}
 
 	input_sync(jack->input_dev);
 }
diff --git a/sound/core/misc.c b/sound/core/misc.c
index 38524f615d94..a9710e0c97af 100644
--- a/sound/core/misc.c
+++ b/sound/core/misc.c
@@ -95,12 +95,14 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list)
 {
 	const struct snd_pci_quirk *q;
 
-	for (q = list; q->subvendor; q++)
-		if (q->subvendor == pci->subsystem_vendor &&
-		    (!q->subdevice || q->subdevice == pci->subsystem_device))
+	for (q = list; q->subvendor; q++) {
+		if (q->subvendor != pci->subsystem_vendor)
+			continue;
+		if (!q->subdevice ||
+		    (pci->subsystem_device & q->subdevice_mask) == q->subdevice)
 			return q;
+	}
 	return NULL;
 }
-
 EXPORT_SYMBOL(snd_pci_quirk_lookup);
 #endif
diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c
index 4690b8b5681f..e570649184e2 100644
--- a/sound/core/oss/mixer_oss.c
+++ b/sound/core/oss/mixer_oss.c
@@ -692,6 +692,9 @@ static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer,
 		snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right);
 		if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME)
 			snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right);
+	} else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME) {
+		snd_mixer_oss_put_volume1_vol(fmixer, pslot,
+			slot->numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right);
 	} else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GVOLUME) {
 		snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GVOLUME], left, right);
 	} else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GLOBAL) {
diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c
index 0a1798eafb0b..dda000b9684c 100644
--- a/sound/core/oss/pcm_oss.c
+++ b/sound/core/oss/pcm_oss.c
@@ -1160,9 +1160,11 @@ snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const
 		    runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) {
 #ifdef OSS_DEBUG
 			if (runtime->status->state == SNDRV_PCM_STATE_XRUN)
-				printk("pcm_oss: write: recovering from XRUN\n");
+				printk(KERN_DEBUG "pcm_oss: write: "
+				       "recovering from XRUN\n");
 			else
-				printk("pcm_oss: write: recovering from SUSPEND\n");
+				printk(KERN_DEBUG "pcm_oss: write: "
+				       "recovering from SUSPEND\n");
 #endif
 			ret = snd_pcm_oss_prepare(substream);
 			if (ret < 0)
@@ -1196,9 +1198,11 @@ snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream, char *p
 		    runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) {
 #ifdef OSS_DEBUG
 			if (runtime->status->state == SNDRV_PCM_STATE_XRUN)
-				printk("pcm_oss: read: recovering from XRUN\n");
+				printk(KERN_DEBUG "pcm_oss: read: "
+				       "recovering from XRUN\n");
 			else
-				printk("pcm_oss: read: recovering from SUSPEND\n");
+				printk(KERN_DEBUG "pcm_oss: read: "
+				       "recovering from SUSPEND\n");
 #endif
 			ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DRAIN, NULL);
 			if (ret < 0)
@@ -1242,9 +1246,11 @@ snd_pcm_sframes_t snd_pcm_oss_writev3(struct snd_pcm_substream *substream, void
 		    runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) {
 #ifdef OSS_DEBUG
 			if (runtime->status->state == SNDRV_PCM_STATE_XRUN)
-				printk("pcm_oss: writev: recovering from XRUN\n");
+				printk(KERN_DEBUG "pcm_oss: writev: "
+				       "recovering from XRUN\n");
 			else
-				printk("pcm_oss: writev: recovering from SUSPEND\n");
+				printk(KERN_DEBUG "pcm_oss: writev: "
+				       "recovering from SUSPEND\n");
 #endif
 			ret = snd_pcm_oss_prepare(substream);
 			if (ret < 0)
@@ -1278,9 +1284,11 @@ snd_pcm_sframes_t snd_pcm_oss_readv3(struct snd_pcm_substream *substream, void *
 		    runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) {
 #ifdef OSS_DEBUG
 			if (runtime->status->state == SNDRV_PCM_STATE_XRUN)
-				printk("pcm_oss: readv: recovering from XRUN\n");
+				printk(KERN_DEBUG "pcm_oss: readv: "
+				       "recovering from XRUN\n");
 			else
-				printk("pcm_oss: readv: recovering from SUSPEND\n");
+				printk(KERN_DEBUG "pcm_oss: readv: "
+				       "recovering from SUSPEND\n");
 #endif
 			ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DRAIN, NULL);
 			if (ret < 0)
@@ -1533,7 +1541,7 @@ static int snd_pcm_oss_sync1(struct snd_pcm_substream *substream, size_t size)
 	init_waitqueue_entry(&wait, current);
 	add_wait_queue(&runtime->sleep, &wait);
 #ifdef OSS_DEBUG
-	printk("sync1: size = %li\n", size);
+	printk(KERN_DEBUG "sync1: size = %li\n", size);
 #endif
 	while (1) {
 		result = snd_pcm_oss_write2(substream, runtime->oss.buffer, size, 1);
@@ -1590,7 +1598,7 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file)
 		mutex_lock(&runtime->oss.params_lock);
 		if (runtime->oss.buffer_used > 0) {
 #ifdef OSS_DEBUG
-			printk("sync: buffer_used\n");
+			printk(KERN_DEBUG "sync: buffer_used\n");
 #endif
 			size = (8 * (runtime->oss.period_bytes - runtime->oss.buffer_used) + 7) / width;
 			snd_pcm_format_set_silence(format,
@@ -1603,7 +1611,7 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file)
 			}
 		} else if (runtime->oss.period_ptr > 0) {
 #ifdef OSS_DEBUG
-			printk("sync: period_ptr\n");
+			printk(KERN_DEBUG "sync: period_ptr\n");
 #endif
 			size = runtime->oss.period_bytes - runtime->oss.period_ptr;
 			snd_pcm_format_set_silence(format,
@@ -1895,7 +1903,9 @@ static int snd_pcm_oss_set_fragment(struct snd_pcm_oss_file *pcm_oss_file, unsig
 
 static int snd_pcm_oss_nonblock(struct file * file)
 {
+	spin_lock(&file->f_lock);
 	file->f_flags |= O_NONBLOCK;
+	spin_unlock(&file->f_lock);
 	return 0;
 }
 
@@ -1952,7 +1962,7 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr
 	int err, cmd;
 
 #ifdef OSS_DEBUG
-	printk("pcm_oss: trigger = 0x%x\n", trigger);
+	printk(KERN_DEBUG "pcm_oss: trigger = 0x%x\n", trigger);
 #endif
 	
 	psubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK];
@@ -2170,7 +2180,9 @@ static int snd_pcm_oss_get_space(struct snd_pcm_oss_file *pcm_oss_file, int stre
 	}
 
 #ifdef OSS_DEBUG
-	printk("pcm_oss: space: bytes = %i, fragments = %i, fragstotal = %i, fragsize = %i\n", info.bytes, info.fragments, info.fragstotal, info.fragsize);
+	printk(KERN_DEBUG "pcm_oss: space: bytes = %i, fragments = %i, "
+	       "fragstotal = %i, fragsize = %i\n",
+	       info.bytes, info.fragments, info.fragstotal, info.fragsize);
 #endif
 	if (copy_to_user(_info, &info, sizeof(info)))
 		return -EFAULT;
@@ -2473,7 +2485,7 @@ static long snd_pcm_oss_ioctl(struct file *file, unsigned int cmd, unsigned long
 	if (((cmd >> 8) & 0xff) != 'P')
 		return -EINVAL;
 #ifdef OSS_DEBUG
-	printk("pcm_oss: ioctl = 0x%x\n", cmd);
+	printk(KERN_DEBUG "pcm_oss: ioctl = 0x%x\n", cmd);
 #endif
 	switch (cmd) {
 	case SNDCTL_DSP_RESET:
@@ -2627,7 +2639,8 @@ static ssize_t snd_pcm_oss_read(struct file *file, char __user *buf, size_t coun
 #else
 	{
 		ssize_t res = snd_pcm_oss_read1(substream, buf, count);
-		printk("pcm_oss: read %li bytes (returned %li bytes)\n", (long)count, (long)res);
+		printk(KERN_DEBUG "pcm_oss: read %li bytes "
+		       "(returned %li bytes)\n", (long)count, (long)res);
 		return res;
 	}
 #endif
@@ -2646,7 +2659,8 @@ static ssize_t snd_pcm_oss_write(struct file *file, const char __user *buf, size
 	substream->f_flags = file->f_flags & O_NONBLOCK;
 	result = snd_pcm_oss_write1(substream, buf, count);
 #ifdef OSS_DEBUG
-	printk("pcm_oss: write %li bytes (wrote %li bytes)\n", (long)count, (long)result);
+	printk(KERN_DEBUG "pcm_oss: write %li bytes (wrote %li bytes)\n",
+	       (long)count, (long)result);
 #endif
 	return result;
 }
@@ -2720,7 +2734,7 @@ static int snd_pcm_oss_mmap(struct file *file, struct vm_area_struct *area)
 	int err;
 
 #ifdef OSS_DEBUG
-	printk("pcm_oss: mmap begin\n");
+	printk(KERN_DEBUG "pcm_oss: mmap begin\n");
 #endif
 	pcm_oss_file = file->private_data;
 	switch ((area->vm_flags & (VM_READ | VM_WRITE))) {
@@ -2770,7 +2784,8 @@ static int snd_pcm_oss_mmap(struct file *file, struct vm_area_struct *area)
 	runtime->silence_threshold = 0;
 	runtime->silence_size = 0;
 #ifdef OSS_DEBUG
-	printk("pcm_oss: mmap ok, bytes = 0x%x\n", runtime->oss.mmap_bytes);
+	printk(KERN_DEBUG "pcm_oss: mmap ok, bytes = 0x%x\n",
+	       runtime->oss.mmap_bytes);
 #endif
 	/* In mmap mode we never stop */
 	runtime->stop_threshold = runtime->boundary;
@@ -2872,7 +2887,7 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry,
 			setup = kmalloc(sizeof(*setup), GFP_KERNEL);
 			if (! setup) {
 				buffer->error = -ENOMEM;
-				mutex_lock(&pstr->oss.setup_mutex);
+				mutex_unlock(&pstr->oss.setup_mutex);
 				return;
 			}
 			if (pstr->oss.setup_list == NULL)
@@ -2886,7 +2901,7 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry,
 			if (! template.task_name) {
 				kfree(setup);
 				buffer->error = -ENOMEM;
-				mutex_lock(&pstr->oss.setup_mutex);
+				mutex_unlock(&pstr->oss.setup_mutex);
 				return;
 			}
 		}
diff --git a/sound/core/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h
index ca2f4c39be46..b9afab603711 100644
--- a/sound/core/oss/pcm_plugin.h
+++ b/sound/core/oss/pcm_plugin.h
@@ -176,9 +176,9 @@ static inline int snd_pcm_plug_slave_format(int format, struct snd_mask *format_
 #endif
 
 #ifdef PLUGIN_DEBUG
-#define pdprintf( fmt, args... ) printk( "plugin: " fmt, ##args)
+#define pdprintf(fmt, args...) printk(KERN_DEBUG "plugin: " fmt, ##args)
 #else
-#define pdprintf( fmt, args... ) 
+#define pdprintf(fmt, args...)
 #endif
 
 #endif				/* __PCM_PLUGIN_H */
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 192a433a2403..145931a9ff30 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -667,7 +667,6 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count)
 		spin_lock_init(&substream->self_group.lock);
 		INIT_LIST_HEAD(&substream->self_group.substreams);
 		list_add_tail(&substream->link_list, &substream->self_group.substreams);
-		spin_lock_init(&substream->timer_lock);
 		atomic_set(&substream->mmap_count, 0);
 		prev = substream;
 	}
@@ -692,7 +691,7 @@ EXPORT_SYMBOL(snd_pcm_new_stream);
  *
  * Returns zero if successful, or a negative error code on failure.
  */
-int snd_pcm_new(struct snd_card *card, char *id, int device,
+int snd_pcm_new(struct snd_card *card, const char *id, int device,
 		int playback_count, int capture_count,
 	        struct snd_pcm ** rpcm)
 {
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 921691080f35..fbb2e391591e 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -125,23 +125,32 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram
 	}
 }
 
+#ifdef CONFIG_SND_PCM_XRUN_DEBUG
+#define xrun_debug(substream)	((substream)->pstr->xrun_debug)
+#else
+#define xrun_debug(substream)	0
+#endif
+
+#define dump_stack_on_xrun(substream) do {	\
+		if (xrun_debug(substream) > 1)	\
+			dump_stack();		\
+	} while (0)
+
 static void xrun(struct snd_pcm_substream *substream)
 {
 	snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN);
-#ifdef CONFIG_SND_PCM_XRUN_DEBUG
-	if (substream->pstr->xrun_debug) {
+	if (xrun_debug(substream)) {
 		snd_printd(KERN_DEBUG "XRUN: pcmC%dD%d%c\n",
 			   substream->pcm->card->number,
 			   substream->pcm->device,
 			   substream->stream ? 'c' : 'p');
-		if (substream->pstr->xrun_debug > 1)
-			dump_stack();
+		dump_stack_on_xrun(substream);
 	}
-#endif
 }
 
-static inline snd_pcm_uframes_t snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream,
-							  struct snd_pcm_runtime *runtime)
+static snd_pcm_uframes_t
+snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream,
+			  struct snd_pcm_runtime *runtime)
 {
 	snd_pcm_uframes_t pos;
 
@@ -150,17 +159,21 @@ static inline snd_pcm_uframes_t snd_pcm_update_hw_ptr_pos(struct snd_pcm_substre
 	pos = substream->ops->pointer(substream);
 	if (pos == SNDRV_PCM_POS_XRUN)
 		return pos; /* XRUN */
-#ifdef CONFIG_SND_DEBUG
 	if (pos >= runtime->buffer_size) {
-		snd_printk(KERN_ERR  "BUG: stream = %i, pos = 0x%lx, buffer size = 0x%lx, period size = 0x%lx\n", substream->stream, pos, runtime->buffer_size, runtime->period_size);
+		if (printk_ratelimit()) {
+			snd_printd(KERN_ERR  "BUG: stream = %i, pos = 0x%lx, "
+				   "buffer size = 0x%lx, period size = 0x%lx\n",
+				   substream->stream, pos, runtime->buffer_size,
+				   runtime->period_size);
+		}
+		pos = 0;
 	}
-#endif
 	pos -= pos % runtime->min_align;
 	return pos;
 }
 
-static inline int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream,
-					     struct snd_pcm_runtime *runtime)
+static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream,
+				      struct snd_pcm_runtime *runtime)
 {
 	snd_pcm_uframes_t avail;
 
@@ -182,11 +195,21 @@ static inline int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream
 	return 0;
 }
 
-static inline int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
+#define hw_ptr_error(substream, fmt, args...)				\
+	do {								\
+		if (xrun_debug(substream)) {				\
+			if (printk_ratelimit()) {			\
+				snd_printd("PCM: " fmt, ##args);	\
+			}						\
+			dump_stack_on_xrun(substream);			\
+		}							\
+	} while (0)
+
+static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	snd_pcm_uframes_t pos;
-	snd_pcm_uframes_t new_hw_ptr, hw_ptr_interrupt;
+	snd_pcm_uframes_t new_hw_ptr, hw_ptr_interrupt, hw_base;
 	snd_pcm_sframes_t delta;
 
 	pos = snd_pcm_update_hw_ptr_pos(substream, runtime);
@@ -194,36 +217,53 @@ static inline int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *subs
 		xrun(substream);
 		return -EPIPE;
 	}
-	if (runtime->period_size == runtime->buffer_size)
-		goto __next_buf;
-	new_hw_ptr = runtime->hw_ptr_base + pos;
+	hw_base = runtime->hw_ptr_base;
+	new_hw_ptr = hw_base + pos;
 	hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size;
-
-	delta = hw_ptr_interrupt - new_hw_ptr;
-	if (delta > 0) {
-		if ((snd_pcm_uframes_t)delta < runtime->buffer_size / 2) {
-#ifdef CONFIG_SND_PCM_XRUN_DEBUG
-			if (runtime->periods > 1 && substream->pstr->xrun_debug) {
-				snd_printd(KERN_ERR "Unexpected hw_pointer value [1] (stream = %i, delta: -%ld, max jitter = %ld): wrong interrupt acknowledge?\n", substream->stream, (long) delta, runtime->buffer_size / 2);
-				if (substream->pstr->xrun_debug > 1)
-					dump_stack();
-			}
-#endif
-			return 0;
+	delta = new_hw_ptr - hw_ptr_interrupt;
+	if (hw_ptr_interrupt >= runtime->boundary) {
+		hw_ptr_interrupt -= runtime->boundary;
+		if (hw_base < runtime->boundary / 2)
+			/* hw_base was already lapped; recalc delta */
+			delta = new_hw_ptr - hw_ptr_interrupt;
+	}
+	if (delta < 0) {
+		delta += runtime->buffer_size;
+		if (delta < 0) {
+			hw_ptr_error(substream, 
+				     "Unexpected hw_pointer value "
+				     "(stream=%i, pos=%ld, intr_ptr=%ld)\n",
+				     substream->stream, (long)pos,
+				     (long)hw_ptr_interrupt);
+			/* rebase to interrupt position */
+			hw_base = new_hw_ptr = hw_ptr_interrupt;
+			/* align hw_base to buffer_size */
+			hw_base -= hw_base % runtime->buffer_size;
+			delta = 0;
+		} else {
+			hw_base += runtime->buffer_size;
+			if (hw_base >= runtime->boundary)
+				hw_base = 0;
+			new_hw_ptr = hw_base + pos;
 		}
-	      __next_buf:
-		runtime->hw_ptr_base += runtime->buffer_size;
-		if (runtime->hw_ptr_base == runtime->boundary)
-			runtime->hw_ptr_base = 0;
-		new_hw_ptr = runtime->hw_ptr_base + pos;
 	}
-
+	if (delta > runtime->period_size) {
+		hw_ptr_error(substream,
+			     "Lost interrupts? "
+			     "(stream=%i, delta=%ld, intr_ptr=%ld)\n",
+			     substream->stream, (long)delta,
+			     (long)hw_ptr_interrupt);
+		/* rebase hw_ptr_interrupt */
+		hw_ptr_interrupt =
+			new_hw_ptr - new_hw_ptr % runtime->period_size;
+	}
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
 	    runtime->silence_size > 0)
 		snd_pcm_playback_silence(substream, new_hw_ptr);
 
+	runtime->hw_ptr_base = hw_base;
 	runtime->status->hw_ptr = new_hw_ptr;
-	runtime->hw_ptr_interrupt = new_hw_ptr - new_hw_ptr % runtime->period_size;
+	runtime->hw_ptr_interrupt = hw_ptr_interrupt;
 
 	return snd_pcm_update_hw_ptr_post(substream, runtime);
 }
@@ -233,7 +273,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	snd_pcm_uframes_t pos;
-	snd_pcm_uframes_t old_hw_ptr, new_hw_ptr;
+	snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_base;
 	snd_pcm_sframes_t delta;
 
 	old_hw_ptr = runtime->status->hw_ptr;
@@ -242,29 +282,38 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream)
 		xrun(substream);
 		return -EPIPE;
 	}
-	new_hw_ptr = runtime->hw_ptr_base + pos;
-
-	delta = old_hw_ptr - new_hw_ptr;
-	if (delta > 0) {
-		if ((snd_pcm_uframes_t)delta < runtime->buffer_size / 2) {
-#ifdef CONFIG_SND_PCM_XRUN_DEBUG
-			if (runtime->periods > 2 && substream->pstr->xrun_debug) {
-				snd_printd(KERN_ERR "Unexpected hw_pointer value [2] (stream = %i, delta: -%ld, max jitter = %ld): wrong interrupt acknowledge?\n", substream->stream, (long) delta, runtime->buffer_size / 2);
-				if (substream->pstr->xrun_debug > 1)
-					dump_stack();
-			}
-#endif
+	hw_base = runtime->hw_ptr_base;
+	new_hw_ptr = hw_base + pos;
+
+	delta = new_hw_ptr - old_hw_ptr;
+	if (delta < 0) {
+		delta += runtime->buffer_size;
+		if (delta < 0) {
+			hw_ptr_error(substream, 
+				     "Unexpected hw_pointer value [2] "
+				     "(stream=%i, pos=%ld, old_ptr=%ld)\n",
+				     substream->stream, (long)pos,
+				     (long)old_hw_ptr);
 			return 0;
 		}
-		runtime->hw_ptr_base += runtime->buffer_size;
-		if (runtime->hw_ptr_base == runtime->boundary)
-			runtime->hw_ptr_base = 0;
-		new_hw_ptr = runtime->hw_ptr_base + pos;
+		hw_base += runtime->buffer_size;
+		if (hw_base >= runtime->boundary)
+			hw_base = 0;
+		new_hw_ptr = hw_base + pos;
+	}
+	if (delta > runtime->period_size && runtime->periods > 1) {
+		hw_ptr_error(substream,
+			     "hw_ptr skipping! "
+			     "(pos=%ld, delta=%ld, period=%ld)\n",
+			     (long)pos, (long)delta,
+			     (long)runtime->period_size);
+		return 0;
 	}
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
 	    runtime->silence_size > 0)
 		snd_pcm_playback_silence(substream, new_hw_ptr);
 
+	runtime->hw_ptr_base = hw_base;
 	runtime->status->hw_ptr = new_hw_ptr;
 
 	return snd_pcm_update_hw_ptr_post(substream, runtime);
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index a789efc9df39..a151fb01ba82 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -186,7 +186,7 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream,
 		if (!(params->rmask & (1 << k)))
 			continue;
 #ifdef RULES_DEBUG
-		printk("%s = ", snd_pcm_hw_param_names[k]);
+		printk(KERN_DEBUG "%s = ", snd_pcm_hw_param_names[k]);
 		printk("%04x%04x%04x%04x -> ", m->bits[3], m->bits[2], m->bits[1], m->bits[0]);
 #endif
 		changed = snd_mask_refine(m, constrs_mask(constrs, k));
@@ -206,7 +206,7 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream,
 		if (!(params->rmask & (1 << k)))
 			continue;
 #ifdef RULES_DEBUG
-		printk("%s = ", snd_pcm_hw_param_names[k]);
+		printk(KERN_DEBUG "%s = ", snd_pcm_hw_param_names[k]);
 		if (i->empty)
 			printk("empty");
 		else
@@ -251,7 +251,7 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream,
 			if (!doit)
 				continue;
 #ifdef RULES_DEBUG
-			printk("Rule %d [%p]: ", k, r->func);
+			printk(KERN_DEBUG "Rule %d [%p]: ", k, r->func);
 			if (r->var >= 0) {
 				printk("%s = ", snd_pcm_hw_param_names[r->var]);
 				if (hw_is_mask(r->var)) {
@@ -3246,9 +3246,7 @@ static int snd_pcm_fasync(int fd, struct file * file, int on)
 	err = fasync_helper(fd, file, on, &runtime->fasync);
 out:
 	unlock_kernel();
-	if (err < 0)
-		return err;
-	return 0;
+	return err;
 }
 
 /*
diff --git a/sound/core/pcm_timer.c b/sound/core/pcm_timer.c
index 2c89c04f2916..ca8068b63d6c 100644
--- a/sound/core/pcm_timer.c
+++ b/sound/core/pcm_timer.c
@@ -85,25 +85,19 @@ static unsigned long snd_pcm_timer_resolution(struct snd_timer * timer)
 
 static int snd_pcm_timer_start(struct snd_timer * timer)
 {
-	unsigned long flags;
 	struct snd_pcm_substream *substream;
 	
 	substream = snd_timer_chip(timer);
-	spin_lock_irqsave(&substream->timer_lock, flags);
 	substream->timer_running = 1;
-	spin_unlock_irqrestore(&substream->timer_lock, flags);
 	return 0;
 }
 
 static int snd_pcm_timer_stop(struct snd_timer * timer)
 {
-	unsigned long flags;
 	struct snd_pcm_substream *substream;
 	
 	substream = snd_timer_chip(timer);
-	spin_lock_irqsave(&substream->timer_lock, flags);
 	substream->timer_running = 0;
-	spin_unlock_irqrestore(&substream->timer_lock, flags);
 	return 0;
 }
 
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index 002777ba336a..473247c8e6d3 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -224,156 +224,143 @@ int snd_rawmidi_drain_input(struct snd_rawmidi_substream *substream)
 	return 0;
 }
 
-int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice,
-			    int mode, struct snd_rawmidi_file * rfile)
+/* look for an available substream for the given stream direction;
+ * if a specific subdevice is given, try to assign it
+ */
+static int assign_substream(struct snd_rawmidi *rmidi, int subdevice,
+			    int stream, int mode,
+			    struct snd_rawmidi_substream **sub_ret)
+{
+	struct snd_rawmidi_substream *substream;
+	struct snd_rawmidi_str *s = &rmidi->streams[stream];
+	static unsigned int info_flags[2] = {
+		[SNDRV_RAWMIDI_STREAM_OUTPUT] = SNDRV_RAWMIDI_INFO_OUTPUT,
+		[SNDRV_RAWMIDI_STREAM_INPUT] = SNDRV_RAWMIDI_INFO_INPUT,
+	};
+
+	if (!(rmidi->info_flags & info_flags[stream]))
+		return -ENXIO;
+	if (subdevice >= 0 && subdevice >= s->substream_count)
+		return -ENODEV;
+	if (s->substream_opened >= s->substream_count)
+		return -EAGAIN;
+
+	list_for_each_entry(substream, &s->substreams, list) {
+		if (substream->opened) {
+			if (stream == SNDRV_RAWMIDI_STREAM_INPUT ||
+			    !(mode & SNDRV_RAWMIDI_LFLG_APPEND))
+				continue;
+		}
+		if (subdevice < 0 || subdevice == substream->number) {
+			*sub_ret = substream;
+			return 0;
+		}
+	}
+	return -EAGAIN;
+}
+
+/* open and do ref-counting for the given substream */
+static int open_substream(struct snd_rawmidi *rmidi,
+			  struct snd_rawmidi_substream *substream,
+			  int mode)
+{
+	int err;
+
+	err = snd_rawmidi_runtime_create(substream);
+	if (err < 0)
+		return err;
+	err = substream->ops->open(substream);
+	if (err < 0)
+		return err;
+	substream->opened = 1;
+	if (substream->use_count++ == 0)
+		substream->active_sensing = 1;
+	if (mode & SNDRV_RAWMIDI_LFLG_APPEND)
+		substream->append = 1;
+	rmidi->streams[substream->stream].substream_opened++;
+	return 0;
+}
+
+static void close_substream(struct snd_rawmidi *rmidi,
+			    struct snd_rawmidi_substream *substream,
+			    int cleanup);
+
+static int rawmidi_open_priv(struct snd_rawmidi *rmidi, int subdevice, int mode,
+			     struct snd_rawmidi_file *rfile)
 {
-	struct snd_rawmidi *rmidi;
-	struct list_head *list1, *list2;
 	struct snd_rawmidi_substream *sinput = NULL, *soutput = NULL;
-	struct snd_rawmidi_runtime *input = NULL, *output = NULL;
 	int err;
 
-	if (rfile)
-		rfile->input = rfile->output = NULL;
-	mutex_lock(&register_mutex);
-	rmidi = snd_rawmidi_search(card, device);
-	mutex_unlock(&register_mutex);
-	if (rmidi == NULL) {
-		err = -ENODEV;
-		goto __error1;
-	}
-	if (!try_module_get(rmidi->card->module)) {
-		err = -EFAULT;
-		goto __error1;
-	}
-	if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK))
-		mutex_lock(&rmidi->open_mutex);
+	rfile->input = rfile->output = NULL;
 	if (mode & SNDRV_RAWMIDI_LFLG_INPUT) {
-		if (!(rmidi->info_flags & SNDRV_RAWMIDI_INFO_INPUT)) {
-			err = -ENXIO;
-			goto __error;
-		}
-		if (subdevice >= 0 && (unsigned int)subdevice >= rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_count) {
-			err = -ENODEV;
-			goto __error;
-		}
-		if (rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_opened >=
-		    rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_count) {
-			err = -EAGAIN;
+		err = assign_substream(rmidi, subdevice,
+				       SNDRV_RAWMIDI_STREAM_INPUT,
+				       mode, &sinput);
+		if (err < 0)
 			goto __error;
-		}
 	}
 	if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) {
-		if (!(rmidi->info_flags & SNDRV_RAWMIDI_INFO_OUTPUT)) {
-			err = -ENXIO;
-			goto __error;
-		}
-		if (subdevice >= 0 && (unsigned int)subdevice >= rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_count) {
-			err = -ENODEV;
-			goto __error;
-		}
-		if (rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_opened >=
-		    rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_count) {
-			err = -EAGAIN;
+		err = assign_substream(rmidi, subdevice,
+				       SNDRV_RAWMIDI_STREAM_OUTPUT,
+				       mode, &soutput);
+		if (err < 0)
 			goto __error;
-		}
-	}
-	list1 = rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams.next;
-	while (1) {
-		if (list1 == &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams) {
-			sinput = NULL;
-			if (mode & SNDRV_RAWMIDI_LFLG_INPUT) {
-				err = -EAGAIN;
-				goto __error;
-			}
-			break;
-		}
-		sinput = list_entry(list1, struct snd_rawmidi_substream, list);
-		if ((mode & SNDRV_RAWMIDI_LFLG_INPUT) && sinput->opened)
-			goto __nexti;
-		if (subdevice < 0 || (subdevice >= 0 && subdevice == sinput->number))
-			break;
-	      __nexti:
-		list1 = list1->next;
 	}
-	list2 = rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams.next;
-	while (1) {
-		if (list2 == &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams) {
-			soutput = NULL;
-			if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) {
-				err = -EAGAIN;
-				goto __error;
-			}
-			break;
-		}
-		soutput = list_entry(list2, struct snd_rawmidi_substream, list);
-		if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) {
-			if (mode & SNDRV_RAWMIDI_LFLG_APPEND) {
-				if (soutput->opened && !soutput->append)
-					goto __nexto;
-			} else {
-				if (soutput->opened)
-					goto __nexto;
-			}
-		}
-		if (subdevice < 0 || (subdevice >= 0 && subdevice == soutput->number))
-			break;
-	      __nexto:
-		list2 = list2->next;
-	}
-	if (mode & SNDRV_RAWMIDI_LFLG_INPUT) {
-		if ((err = snd_rawmidi_runtime_create(sinput)) < 0)
-			goto __error;
-		input = sinput->runtime;
-		if ((err = sinput->ops->open(sinput)) < 0)
+
+	if (sinput) {
+		err = open_substream(rmidi, sinput, mode);
+		if (err < 0)
 			goto __error;
-		sinput->opened = 1;
-		rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_opened++;
-	} else {
-		sinput = NULL;
 	}
-	if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) {
-		if (soutput->opened)
-			goto __skip_output;
-		if ((err = snd_rawmidi_runtime_create(soutput)) < 0) {
-			if (mode & SNDRV_RAWMIDI_LFLG_INPUT)
-				sinput->ops->close(sinput);
-			goto __error;
-		}
-		output = soutput->runtime;
-		if ((err = soutput->ops->open(soutput)) < 0) {
-			if (mode & SNDRV_RAWMIDI_LFLG_INPUT)
-				sinput->ops->close(sinput);
+	if (soutput) {
+		err = open_substream(rmidi, soutput, mode);
+		if (err < 0) {
+			if (sinput)
+				close_substream(rmidi, sinput, 0);
 			goto __error;
 		}
-	      __skip_output:
-		soutput->opened = 1;
-		if (mode & SNDRV_RAWMIDI_LFLG_APPEND)
-			soutput->append = 1;
-	      	if (soutput->use_count++ == 0)
-			soutput->active_sensing = 1;
-		rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_opened++;
-	} else {
-		soutput = NULL;
-	}
-	if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK))
-		mutex_unlock(&rmidi->open_mutex);
-	if (rfile) {
-		rfile->rmidi = rmidi;
-		rfile->input = sinput;
-		rfile->output = soutput;
 	}
+
+	rfile->rmidi = rmidi;
+	rfile->input = sinput;
+	rfile->output = soutput;
 	return 0;
 
       __error:
-	if (input != NULL)
+	if (sinput && sinput->runtime)
 		snd_rawmidi_runtime_free(sinput);
-	if (output != NULL)
+	if (soutput && soutput->runtime)
 		snd_rawmidi_runtime_free(soutput);
-	module_put(rmidi->card->module);
-	if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK))
-		mutex_unlock(&rmidi->open_mutex);
-      __error1:
+	return err;
+}
+
+/* called from sound/core/seq/seq_midi.c */
+int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice,
+			    int mode, struct snd_rawmidi_file * rfile)
+{
+	struct snd_rawmidi *rmidi;
+	int err;
+
+	if (snd_BUG_ON(!rfile))
+		return -EINVAL;
+
+	mutex_lock(&register_mutex);
+	rmidi = snd_rawmidi_search(card, device);
+	if (rmidi == NULL) {
+		mutex_unlock(&register_mutex);
+		return -ENODEV;
+	}
+	if (!try_module_get(rmidi->card->module)) {
+		mutex_unlock(&register_mutex);
+		return -ENXIO;
+	}
+	mutex_unlock(&register_mutex);
+
+	mutex_lock(&rmidi->open_mutex);
+	err = rawmidi_open_priv(rmidi, subdevice, mode, rfile);
+	mutex_unlock(&rmidi->open_mutex);
+	if (err < 0)
+		module_put(rmidi->card->module);
 	return err;
 }
 
@@ -385,10 +372,13 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file)
 	unsigned short fflags;
 	int err;
 	struct snd_rawmidi *rmidi;
-	struct snd_rawmidi_file *rawmidi_file;
+	struct snd_rawmidi_file *rawmidi_file = NULL;
 	wait_queue_t wait;
 	struct snd_ctl_file *kctl;
 
+	if ((file->f_flags & O_APPEND) && !(file->f_flags & O_NONBLOCK)) 
+		return -EINVAL;		/* invalid combination */
+
 	if (maj == snd_major) {
 		rmidi = snd_lookup_minor_data(iminor(inode),
 					      SNDRV_DEVICE_TYPE_RAWMIDI);
@@ -402,24 +392,25 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file)
 
 	if (rmidi == NULL)
 		return -ENODEV;
-	if ((file->f_flags & O_APPEND) && !(file->f_flags & O_NONBLOCK)) 
-		return -EINVAL;		/* invalid combination */
+
+	if (!try_module_get(rmidi->card->module))
+		return -ENXIO;
+
+	mutex_lock(&rmidi->open_mutex);
 	card = rmidi->card;
 	err = snd_card_file_add(card, file);
 	if (err < 0)
-		return -ENODEV;
+		goto __error_card;
 	fflags = snd_rawmidi_file_flags(file);
 	if ((file->f_flags & O_APPEND) || maj == SOUND_MAJOR) /* OSS emul? */
 		fflags |= SNDRV_RAWMIDI_LFLG_APPEND;
-	fflags |= SNDRV_RAWMIDI_LFLG_NOOPENLOCK;
 	rawmidi_file = kmalloc(sizeof(*rawmidi_file), GFP_KERNEL);
 	if (rawmidi_file == NULL) {
-		snd_card_file_remove(card, file);
-		return -ENOMEM;
+		err = -ENOMEM;
+		goto __error;
 	}
 	init_waitqueue_entry(&wait, current);
 	add_wait_queue(&rmidi->open_wait, &wait);
-	mutex_lock(&rmidi->open_mutex);
 	while (1) {
 		subdevice = -1;
 		read_lock(&card->ctl_files_rwlock);
@@ -431,8 +422,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file)
 			}
 		}
 		read_unlock(&card->ctl_files_rwlock);
-		err = snd_rawmidi_kernel_open(rmidi->card, rmidi->device,
-					      subdevice, fflags, rawmidi_file);
+		err = rawmidi_open_priv(rmidi, subdevice, fflags, rawmidi_file);
 		if (err >= 0)
 			break;
 		if (err == -EAGAIN) {
@@ -451,67 +441,89 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file)
 			break;
 		}
 	}
+	remove_wait_queue(&rmidi->open_wait, &wait);
+	if (err < 0) {
+		kfree(rawmidi_file);
+		goto __error;
+	}
 #ifdef CONFIG_SND_OSSEMUL
 	if (rawmidi_file->input && rawmidi_file->input->runtime)
 		rawmidi_file->input->runtime->oss = (maj == SOUND_MAJOR);
 	if (rawmidi_file->output && rawmidi_file->output->runtime)
 		rawmidi_file->output->runtime->oss = (maj == SOUND_MAJOR);
 #endif
-	remove_wait_queue(&rmidi->open_wait, &wait);
-	if (err >= 0) {
-		file->private_data = rawmidi_file;
-	} else {
-		snd_card_file_remove(card, file);
-		kfree(rawmidi_file);
-	}
+	file->private_data = rawmidi_file;
+	mutex_unlock(&rmidi->open_mutex);
+	return 0;
+
+ __error:
+	snd_card_file_remove(card, file);
+ __error_card:
 	mutex_unlock(&rmidi->open_mutex);
+	module_put(rmidi->card->module);
 	return err;
 }
 
-int snd_rawmidi_kernel_release(struct snd_rawmidi_file * rfile)
+static void close_substream(struct snd_rawmidi *rmidi,
+			    struct snd_rawmidi_substream *substream,
+			    int cleanup)
 {
-	struct snd_rawmidi *rmidi;
-	struct snd_rawmidi_substream *substream;
-	struct snd_rawmidi_runtime *runtime;
+	rmidi->streams[substream->stream].substream_opened--;
+	if (--substream->use_count)
+		return;
 
-	if (snd_BUG_ON(!rfile))
-		return -ENXIO;
-	rmidi = rfile->rmidi;
-	mutex_lock(&rmidi->open_mutex);
-	if (rfile->input != NULL) {
-		substream = rfile->input;
-		rfile->input = NULL;
-		runtime = substream->runtime;
-		snd_rawmidi_input_trigger(substream, 0);
-		substream->ops->close(substream);
-		if (runtime->private_free != NULL)
-			runtime->private_free(substream);
-		snd_rawmidi_runtime_free(substream);
-		substream->opened = 0;
-		rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_opened--;
-	}
-	if (rfile->output != NULL) {
-		substream = rfile->output;
-		rfile->output = NULL;
-		if (--substream->use_count == 0) {
-			runtime = substream->runtime;
+	if (cleanup) {
+		if (substream->stream == SNDRV_RAWMIDI_STREAM_INPUT)
+			snd_rawmidi_input_trigger(substream, 0);
+		else {
 			if (substream->active_sensing) {
 				unsigned char buf = 0xfe;
-				/* sending single active sensing message to shut the device up */
+				/* sending single active sensing message
+				 * to shut the device up
+				 */
 				snd_rawmidi_kernel_write(substream, &buf, 1);
 			}
 			if (snd_rawmidi_drain_output(substream) == -ERESTARTSYS)
 				snd_rawmidi_output_trigger(substream, 0);
-			substream->ops->close(substream);
-			if (runtime->private_free != NULL)
-				runtime->private_free(substream);
-			snd_rawmidi_runtime_free(substream);
-			substream->opened = 0;
-			substream->append = 0;
 		}
-		rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_opened--;
 	}
+	substream->ops->close(substream);
+	if (substream->runtime->private_free)
+		substream->runtime->private_free(substream);
+	snd_rawmidi_runtime_free(substream);
+	substream->opened = 0;
+	substream->append = 0;
+}
+
+static void rawmidi_release_priv(struct snd_rawmidi_file *rfile)
+{
+	struct snd_rawmidi *rmidi;
+
+	rmidi = rfile->rmidi;
+	mutex_lock(&rmidi->open_mutex);
+	if (rfile->input) {
+		close_substream(rmidi, rfile->input, 1);
+		rfile->input = NULL;
+	}
+	if (rfile->output) {
+		close_substream(rmidi, rfile->output, 1);
+		rfile->output = NULL;
+	}
+	rfile->rmidi = NULL;
 	mutex_unlock(&rmidi->open_mutex);
+	wake_up(&rmidi->open_wait);
+}
+
+/* called from sound/core/seq/seq_midi.c */
+int snd_rawmidi_kernel_release(struct snd_rawmidi_file *rfile)
+{
+	struct snd_rawmidi *rmidi;
+
+	if (snd_BUG_ON(!rfile))
+		return -ENXIO;
+	
+	rmidi = rfile->rmidi;
+	rawmidi_release_priv(rfile);
 	module_put(rmidi->card->module);
 	return 0;
 }
@@ -520,15 +532,14 @@ static int snd_rawmidi_release(struct inode *inode, struct file *file)
 {
 	struct snd_rawmidi_file *rfile;
 	struct snd_rawmidi *rmidi;
-	int err;
 
 	rfile = file->private_data;
-	err = snd_rawmidi_kernel_release(rfile);
 	rmidi = rfile->rmidi;
-	wake_up(&rmidi->open_wait);
+	rawmidi_release_priv(rfile);
 	kfree(rfile);
 	snd_card_file_remove(rmidi->card, file);
-	return err;
+	module_put(rmidi->card->module);
+	return 0;
 }
 
 static int snd_rawmidi_info(struct snd_rawmidi_substream *substream,
diff --git a/sound/core/seq/oss/seq_oss_device.h b/sound/core/seq/oss/seq_oss_device.h
index bf8d2b4cb15e..c0154a959d55 100644
--- a/sound/core/seq/oss/seq_oss_device.h
+++ b/sound/core/seq/oss/seq_oss_device.h
@@ -181,7 +181,7 @@ char *enabled_str(int bool);
 /* for debug */
 #ifdef SNDRV_SEQ_OSS_DEBUG
 extern int seq_oss_debug;
-#define debug_printk(x)	do { if (seq_oss_debug > 0) snd_printk x; } while (0)
+#define debug_printk(x)	do { if (seq_oss_debug > 0) snd_printd x; } while (0)
 #else
 #define debug_printk(x)	/**/
 #endif
diff --git a/sound/core/seq/seq_prioq.c b/sound/core/seq/seq_prioq.c
index 0101a8b99b73..29896ab23403 100644
--- a/sound/core/seq/seq_prioq.c
+++ b/sound/core/seq/seq_prioq.c
@@ -321,7 +321,8 @@ void snd_seq_prioq_leave(struct snd_seq_prioq * f, int client, int timestamp)
 			freeprev = cell;
 		} else {
 #if 0
-			printk("type = %i, source = %i, dest = %i, client = %i\n",
+			printk(KERN_DEBUG "type = %i, source = %i, dest = %i, "
+			       "client = %i\n",
 				cell->event.type,
 				cell->event.source.client,
 				cell->event.dest.client,
diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c
index d4564edd61d7..4e7ec2b49873 100644
--- a/sound/core/sgbuf.c
+++ b/sound/core/sgbuf.c
@@ -38,6 +38,10 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab)
 	if (! sgbuf)
 		return -EINVAL;
 
+	if (dmab->area)
+		vunmap(dmab->area);
+	dmab->area = NULL;
+
 	tmpb.dev.type = SNDRV_DMA_TYPE_DEV;
 	tmpb.dev.dev = sgbuf->dev;
 	for (i = 0; i < sgbuf->pages; i++) {
@@ -48,9 +52,6 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab)
 		tmpb.bytes = (sgbuf->table[i].addr & ~PAGE_MASK) << PAGE_SHIFT;
 		snd_dma_free_pages(&tmpb);
 	}
-	if (dmab->area)
-		vunmap(dmab->area);
-	dmab->area = NULL;
 
 	kfree(sgbuf->table);
 	kfree(sgbuf->page_table);
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 796532081e81..3f0050d0b71e 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -1825,13 +1825,9 @@ static long snd_timer_user_ioctl(struct file *file, unsigned int cmd,
 static int snd_timer_user_fasync(int fd, struct file * file, int on)
 {
 	struct snd_timer_user *tu;
-	int err;
 
 	tu = file->private_data;
-	err = fasync_helper(fd, file, on, &tu->fasync);
-        if (err < 0)
-		return err;
-	return 0;
+	return fasync_helper(fd, file, on, &tu->fasync);
 }
 
 static ssize_t snd_timer_user_read(struct file *file, char __user *buffer,
diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c
index 4cc57f902e2c..257624bd1997 100644
--- a/sound/core/vmaster.c
+++ b/sound/core/vmaster.c
@@ -50,18 +50,38 @@ struct link_slave {
 	struct link_master *master;
 	struct link_ctl_info info;
 	int vals[2];		/* current values */
+	unsigned int flags;
 	struct snd_kcontrol slave; /* the copy of original control entry */
 };
 
+static int slave_update(struct link_slave *slave)
+{
+	struct snd_ctl_elem_value *uctl;
+	int err, ch;
+
+	uctl = kmalloc(sizeof(*uctl), GFP_KERNEL);
+	if (!uctl)
+		return -ENOMEM;
+	uctl->id = slave->slave.id;
+	err = slave->slave.get(&slave->slave, uctl);
+	for (ch = 0; ch < slave->info.count; ch++)
+		slave->vals[ch] = uctl->value.integer.value[ch];
+	kfree(uctl);
+	return 0;
+}
+
 /* get the slave ctl info and save the initial values */
 static int slave_init(struct link_slave *slave)
 {
 	struct snd_ctl_elem_info *uinfo;
-	struct snd_ctl_elem_value *uctl;
-	int err, ch;
+	int err;
 
-	if (slave->info.count)
-		return 0; /* already initialized */
+	if (slave->info.count) {
+		/* already initialized */
+		if (slave->flags & SND_CTL_SLAVE_NEED_UPDATE)
+			return slave_update(slave);
+		return 0;
+	}
 
 	uinfo = kmalloc(sizeof(*uinfo), GFP_KERNEL);
 	if (!uinfo)
@@ -85,15 +105,7 @@ static int slave_init(struct link_slave *slave)
 	slave->info.max_val = uinfo->value.integer.max;
 	kfree(uinfo);
 
-	uctl = kmalloc(sizeof(*uctl), GFP_KERNEL);
-	if (!uctl)
-		return -ENOMEM;
-	uctl->id = slave->slave.id;
-	err = slave->slave.get(&slave->slave, uctl);
-	for (ch = 0; ch < slave->info.count; ch++)
-		slave->vals[ch] = uctl->value.integer.value[ch];
-	kfree(uctl);
-	return 0;
+	return slave_update(slave);
 }
 
 /* initialize master volume */
@@ -229,7 +241,8 @@ static void slave_free(struct snd_kcontrol *kcontrol)
  * - logarithmic volume control (dB level), no linear volume
  * - master can only attenuate the volume, no gain
  */
-int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
+int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave,
+		       unsigned int flags)
 {
 	struct link_master *master_link = snd_kcontrol_chip(master);
 	struct link_slave *srec;
@@ -241,6 +254,7 @@ int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
 	srec->slave = *slave;
 	memcpy(srec->slave.vd, slave->vd, slave->count * sizeof(*slave->vd));
 	srec->master = master_link;
+	srec->flags = flags;
 
 	/* override callbacks */
 	slave->info = slave_info;
@@ -254,8 +268,7 @@ int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave)
 	list_add_tail(&srec->list, &master_link->slaves);
 	return 0;
 }
-
-EXPORT_SYMBOL(snd_ctl_add_slave);
+EXPORT_SYMBOL(_snd_ctl_add_slave);
 
 /*
  * ctl callbacks for master controls
@@ -327,8 +340,20 @@ static void master_free(struct snd_kcontrol *kcontrol)
 }
 
 
-/*
- * Create a virtual master control with the given name
+/**
+ * snd_ctl_make_virtual_master - Create a virtual master control
+ * @name: name string of the control element to create
+ * @tlv: optional TLV int array for dB information
+ *
+ * Creates a virtual matster control with the given name string.
+ * Returns the created control element, or NULL for errors (ENOMEM).
+ *
+ * After creating a vmaster element, you can add the slave controls
+ * via snd_ctl_add_slave() or snd_ctl_add_slave_uncached().
+ *
+ * The optional argument @tlv can be used to specify the TLV information
+ * for dB scale of the master control.  It should be a single element
+ * with #SNDRV_CTL_TLVT_DB_SCALE type, and should be the max 0dB.
  */
 struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
 						 const unsigned int *tlv)
@@ -367,5 +392,4 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name,
 
 	return kctl;
 }
-
 EXPORT_SYMBOL(snd_ctl_make_virtual_master);
diff --git a/sound/drivers/dummy.c b/sound/drivers/dummy.c
index 73be7e14a603..54239d2e0997 100644
--- a/sound/drivers/dummy.c
+++ b/sound/drivers/dummy.c
@@ -588,10 +588,10 @@ static int __devinit snd_dummy_probe(struct platform_device *devptr)
 	int idx, err;
 	int dev = devptr->id;
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-			    sizeof(struct snd_dummy));
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct snd_dummy), &card);
+	if (err < 0)
+		return err;
 	dummy = card->private_data;
 	dummy->card = card;
 	for (idx = 0; idx < MAX_PCM_DEVICES && idx < pcm_devs[dev]; idx++) {
diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c
index 7783843ca9ae..1950ffce2b54 100644
--- a/sound/drivers/ml403-ac97cr.c
+++ b/sound/drivers/ml403-ac97cr.c
@@ -1279,9 +1279,9 @@ static int __devinit snd_ml403_ac97cr_probe(struct platform_device *pfdev)
 	if (!enable[dev])
 		return -ENOENT;
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 	err = snd_ml403_ac97cr_create(card, pfdev, &ml403_ac97cr);
 	if (err < 0) {
 		PDEBUG(INIT_FAILURE, "probe(): create failed!\n");
diff --git a/sound/drivers/mpu401/mpu401.c b/sound/drivers/mpu401/mpu401.c
index 5b996f3faba5..149d05a8202d 100644
--- a/sound/drivers/mpu401/mpu401.c
+++ b/sound/drivers/mpu401/mpu401.c
@@ -73,9 +73,9 @@ static int snd_mpu401_create(int dev, struct snd_card **rcard)
 		snd_printk(KERN_ERR "the uart_enter option is obsolete; remove it\n");
 
 	*rcard = NULL;
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 	strcpy(card->driver, "MPU-401 UART");
 	strcpy(card->shortname, card->driver);
 	sprintf(card->longname, "%s at %#lx, ", card->shortname, port[dev]);
diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c
index 48b64e6b2670..2f8f295d6b0c 100644
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -303,8 +303,10 @@ static void snd_mtpav_output_port_write(struct mtpav *mtp_card,
 
 		snd_mtpav_send_byte(mtp_card, 0xf5);
 		snd_mtpav_send_byte(mtp_card, portp->hwport);
-		//snd_printk("new outport: 0x%x\n", (unsigned int) portp->hwport);
-
+		/*
+		snd_printk(KERN_DEBUG "new outport: 0x%x\n",
+			   (unsigned int) portp->hwport);
+		*/
 		if (!(outbyte & 0x80) && portp->running_status)
 			snd_mtpav_send_byte(mtp_card, portp->running_status);
 	}
@@ -540,7 +542,7 @@ static void snd_mtpav_read_bytes(struct mtpav *mcrd)
 
 	u8 sbyt = snd_mtpav_getreg(mcrd, SREG);
 
-	//printk("snd_mtpav_read_bytes() sbyt: 0x%x\n", sbyt);
+	/* printk(KERN_DEBUG "snd_mtpav_read_bytes() sbyt: 0x%x\n", sbyt); */
 
 	if (!(sbyt & SIGS_BYTE))
 		return;
@@ -585,12 +587,12 @@ static irqreturn_t snd_mtpav_irqh(int irq, void *dev_id)
 static int __devinit snd_mtpav_get_ISA(struct mtpav * mcard)
 {
 	if ((mcard->res_port = request_region(port, 3, "MotuMTPAV MIDI")) == NULL) {
-		snd_printk("MTVAP port 0x%lx is busy\n", port);
+		snd_printk(KERN_ERR "MTVAP port 0x%lx is busy\n", port);
 		return -EBUSY;
 	}
 	mcard->port = port;
 	if (request_irq(irq, snd_mtpav_irqh, IRQF_DISABLED, "MOTU MTPAV", mcard)) {
-		snd_printk("MTVAP IRQ %d busy\n", irq);
+		snd_printk(KERN_ERR "MTVAP IRQ %d busy\n", irq);
 		return -EBUSY;
 	}
 	mcard->irq = irq;
@@ -696,9 +698,9 @@ static int __devinit snd_mtpav_probe(struct platform_device *dev)
 	int err;
 	struct mtpav *mtp_card;
 
-	card = snd_card_new(index, id, THIS_MODULE, sizeof(*mtp_card));
-	if (! card)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, sizeof(*mtp_card), &card);
+	if (err < 0)
+		return err;
 
 	mtp_card = card->private_data;
 	spin_lock_init(&mtp_card->spinlock);
diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c
index 87ba1ddc0115..9284829bf927 100644
--- a/sound/drivers/mts64.c
+++ b/sound/drivers/mts64.c
@@ -957,10 +957,10 @@ static int __devinit snd_mts64_probe(struct platform_device *pdev)
 	if ((err = snd_mts64_probe_port(p)) < 0)
 		return err;
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL) {
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0) {
 		snd_printd("Cannot create card\n");
-		return -ENOMEM;
+		return err;
 	}
 	strcpy(card->driver, DRIVER_NAME);
 	strcpy(card->shortname, "ESI " CARD_NAME);
@@ -1015,7 +1015,7 @@ static int __devinit snd_mts64_probe(struct platform_device *pdev)
 		goto __err;
 	}
 
-	snd_printk("ESI Miditerminal 4140 on 0x%lx\n", p->base);
+	snd_printk(KERN_INFO "ESI Miditerminal 4140 on 0x%lx\n", p->base);
 	return 0;
 
 __err:
diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c
index 780582340fef..6e31e46ca393 100644
--- a/sound/drivers/opl3/opl3_lib.c
+++ b/sound/drivers/opl3/opl3_lib.c
@@ -302,7 +302,7 @@ void snd_opl3_interrupt(struct snd_hwdep * hw)
 	opl3 = hw->private_data;
 	status = inb(opl3->l_port);
 #if 0
-	snd_printk("AdLib IRQ status = 0x%x\n", status);
+	snd_printk(KERN_DEBUG "AdLib IRQ status = 0x%x\n", status);
 #endif
 	if (!(status & 0x80))
 		return;
diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c
index 16feafa2c51e..6e7d09ae0e82 100644
--- a/sound/drivers/opl3/opl3_midi.c
+++ b/sound/drivers/opl3/opl3_midi.c
@@ -125,7 +125,7 @@ static void debug_alloc(struct snd_opl3 *opl3, char *s, int voice) {
 	int i;
 	char *str = "x.24";
 
-	printk("time %.5i: %s [%.2i]: ", opl3->use_time, s, voice);
+	printk(KERN_DEBUG "time %.5i: %s [%.2i]: ", opl3->use_time, s, voice);
 	for (i = 0; i < opl3->max_voices; i++)
 		printk("%c", *(str + opl3->voices[i].state + 1));
 	printk("\n");
@@ -218,7 +218,7 @@ static int opl3_get_voice(struct snd_opl3 *opl3, int instr_4op,
 	for (i = 0; i < END; i++) {
 		if (best[i].voice >= 0) {
 #ifdef DEBUG_ALLOC
-			printk("%s %iop allocation on voice %i\n",
+			printk(KERN_DEBUG "%s %iop allocation on voice %i\n",
 			       alloc_type[i], instr_4op ? 4 : 2,
 			       best[i].voice);
 #endif
@@ -317,7 +317,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan)
 	opl3 = p;
 
 #ifdef DEBUG_MIDI
-	snd_printk("Note on, ch %i, inst %i, note %i, vel %i\n",
+	snd_printk(KERN_DEBUG "Note on, ch %i, inst %i, note %i, vel %i\n",
 		   chan->number, chan->midi_program, note, vel);
 #endif
 
@@ -372,7 +372,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan)
 		return;
 	}
 #ifdef DEBUG_MIDI
-	snd_printk("  --> OPL%i instrument: %s\n",
+	snd_printk(KERN_DEBUG "  --> OPL%i instrument: %s\n",
 		   instr_4op ? 3 : 2, patch->name);
 #endif
 	/* in SYNTH mode, application takes care of voices */
@@ -431,7 +431,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan)
 	}
 
 #ifdef DEBUG_MIDI
-	snd_printk("  --> setting OPL3 connection: 0x%x\n",
+	snd_printk(KERN_DEBUG "  --> setting OPL3 connection: 0x%x\n",
 		   opl3->connection_reg);
 #endif
 	/*
@@ -466,7 +466,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan)
 	/* Program the FM voice characteristics */
 	for (i = 0; i < (instr_4op ? 4 : 2); i++) {
 #ifdef DEBUG_MIDI
-		snd_printk("  --> programming operator %i\n", i);
+		snd_printk(KERN_DEBUG "  --> programming operator %i\n", i);
 #endif
 		op_offset = snd_opl3_regmap[voice_offset][i];
 
@@ -546,7 +546,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan)
 	blocknum |= OPL3_KEYON_BIT;
 
 #ifdef DEBUG_MIDI
-	snd_printk("  --> trigger voice %i\n", voice);
+	snd_printk(KERN_DEBUG "  --> trigger voice %i\n", voice);
 #endif
 	/* Set OPL3 KEYON_BLOCK register of requested voice */ 
 	opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset);
@@ -602,7 +602,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan)
 			prg = extra_prg - 1;
 		}
 #ifdef DEBUG_MIDI
-		snd_printk(" *** allocating extra program\n");
+		snd_printk(KERN_DEBUG " *** allocating extra program\n");
 #endif
 		goto __extra_prg;
 	}
@@ -633,7 +633,7 @@ static void snd_opl3_kill_voice(struct snd_opl3 *opl3, int voice)
 
 	/* kill voice */
 #ifdef DEBUG_MIDI
-	snd_printk("  --> kill voice %i\n", voice);
+	snd_printk(KERN_DEBUG "  --> kill voice %i\n", voice);
 #endif
 	opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset);
 	/* clear Key ON bit */
@@ -670,7 +670,7 @@ void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan
 	opl3 = p;
 
 #ifdef DEBUG_MIDI
-	snd_printk("Note off, ch %i, inst %i, note %i\n",
+	snd_printk(KERN_DEBUG "Note off, ch %i, inst %i, note %i\n",
 		   chan->number, chan->midi_program, note);
 #endif
 
@@ -709,7 +709,7 @@ void snd_opl3_key_press(void *p, int note, int vel, struct snd_midi_channel *cha
 
 	opl3 = p;
 #ifdef DEBUG_MIDI
-	snd_printk("Key pressure, ch#: %i, inst#: %i\n",
+	snd_printk(KERN_DEBUG "Key pressure, ch#: %i, inst#: %i\n",
 		   chan->number, chan->midi_program);
 #endif
 }
@@ -723,7 +723,7 @@ void snd_opl3_terminate_note(void *p, int note, struct snd_midi_channel *chan)
 
 	opl3 = p;
 #ifdef DEBUG_MIDI
-	snd_printk("Terminate note, ch#: %i, inst#: %i\n",
+	snd_printk(KERN_DEBUG "Terminate note, ch#: %i, inst#: %i\n",
 		   chan->number, chan->midi_program);
 #endif
 }
@@ -812,7 +812,7 @@ void snd_opl3_control(void *p, int type, struct snd_midi_channel *chan)
 
 	opl3 = p;
 #ifdef DEBUG_MIDI
-	snd_printk("Controller, TYPE = %i, ch#: %i, inst#: %i\n",
+	snd_printk(KERN_DEBUG "Controller, TYPE = %i, ch#: %i, inst#: %i\n",
 		   type, chan->number, chan->midi_program);
 #endif
 
@@ -849,7 +849,7 @@ void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan,
 
 	opl3 = p;
 #ifdef DEBUG_MIDI
-	snd_printk("NRPN, ch#: %i, inst#: %i\n",
+	snd_printk(KERN_DEBUG "NRPN, ch#: %i, inst#: %i\n",
 		   chan->number, chan->midi_program);
 #endif
 }
@@ -864,6 +864,6 @@ void snd_opl3_sysex(void *p, unsigned char *buf, int len,
 
 	opl3 = p;
 #ifdef DEBUG_MIDI
-	snd_printk("SYSEX\n");
+	snd_printk(KERN_DEBUG "SYSEX\n");
 #endif
 }
diff --git a/sound/drivers/opl3/opl3_oss.c b/sound/drivers/opl3/opl3_oss.c
index 9a2271dc046a..a54b1dc5cc78 100644
--- a/sound/drivers/opl3/opl3_oss.c
+++ b/sound/drivers/opl3/opl3_oss.c
@@ -220,14 +220,14 @@ static int snd_opl3_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format,
 		return -EINVAL;
 
 	if (count < (int)sizeof(sbi)) {
-		snd_printk("FM Error: Patch record too short\n");
+		snd_printk(KERN_ERR "FM Error: Patch record too short\n");
 		return -EINVAL;
 	}
 	if (copy_from_user(&sbi, buf, sizeof(sbi)))
 		return -EFAULT;
 
 	if (sbi.channel < 0 || sbi.channel >= SBFM_MAXINSTR) {
-		snd_printk("FM Error: Invalid instrument number %d\n",
+		snd_printk(KERN_ERR "FM Error: Invalid instrument number %d\n",
 			   sbi.channel);
 		return -EINVAL;
 	}
@@ -254,7 +254,9 @@ static int snd_opl3_ioctl_seq_oss(struct snd_seq_oss_arg *arg, unsigned int cmd,
 	opl3 = arg->private_data;
 	switch (cmd) {
 		case SNDCTL_FM_LOAD_INSTR:
-			snd_printk("OPL3: Obsolete ioctl(SNDCTL_FM_LOAD_INSTR) used. Fix the program.\n");
+			snd_printk(KERN_ERR "OPL3: "
+				   "Obsolete ioctl(SNDCTL_FM_LOAD_INSTR) used. "
+				   "Fix the program.\n");
 			return -EINVAL;
 
 		case SNDCTL_SYNTH_MEMAVL:
diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c
index 962bb9c8b9c8..6d57b6441dec 100644
--- a/sound/drivers/opl3/opl3_synth.c
+++ b/sound/drivers/opl3/opl3_synth.c
@@ -168,7 +168,7 @@ int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file,
 
 #ifdef CONFIG_SND_DEBUG
 	default:
-		snd_printk("unknown IOCTL: 0x%x\n", cmd);
+		snd_printk(KERN_WARNING "unknown IOCTL: 0x%x\n", cmd);
 #endif
 	}
 	return -ENOTTY;
diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c
index a4049eb94d35..b60cef257b58 100644
--- a/sound/drivers/pcsp/pcsp.c
+++ b/sound/drivers/pcsp/pcsp.c
@@ -57,7 +57,7 @@ static int __devinit snd_pcsp_create(struct snd_card *card)
 	else
 		min_div = MAX_DIV;
 #if PCSP_DEBUG
-	printk("PCSP: lpj=%li, min_div=%i, res=%li\n",
+	printk(KERN_DEBUG "PCSP: lpj=%li, min_div=%i, res=%li\n",
 	       loops_per_jiffy, min_div, tp.tv_nsec);
 #endif
 
@@ -98,9 +98,9 @@ static int __devinit snd_card_pcsp_probe(int devnum, struct device *dev)
 	hrtimer_init(&pcsp_chip.timer, CLOCK_MONOTONIC, HRTIMER_MODE_REL);
 	pcsp_chip.timer.function = pcsp_do_timer;
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (!card)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	err = snd_pcsp_create(card);
 	if (err < 0) {
diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c
index b1c047ec19af..60158e2e0eaf 100644
--- a/sound/drivers/portman2x4.c
+++ b/sound/drivers/portman2x4.c
@@ -746,10 +746,10 @@ static int __devinit snd_portman_probe(struct platform_device *pdev)
 	if ((err = snd_portman_probe_port(p)) < 0)
 		return err;
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL) {
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0) {
 		snd_printd("Cannot create card\n");
-		return -ENOMEM;
+		return err;
 	}
 	strcpy(card->driver, DRIVER_NAME);
 	strcpy(card->shortname, CARD_NAME);
diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c
index d8aab9da97c2..b2b6d50c9425 100644
--- a/sound/drivers/serial-u16550.c
+++ b/sound/drivers/serial-u16550.c
@@ -241,7 +241,8 @@ static void snd_uart16550_io_loop(struct snd_uart16550 * uart)
 			snd_rawmidi_receive(uart->midi_input[substream], &c, 1);
 
 		if (status & UART_LSR_OE)
-			snd_printk("%s: Overrun on device at 0x%lx\n",
+			snd_printk(KERN_WARNING
+				   "%s: Overrun on device at 0x%lx\n",
 			       uart->rmidi->name, uart->base);
 	}
 
@@ -636,7 +637,8 @@ static int snd_uart16550_output_byte(struct snd_uart16550 *uart,
 		}
 	} else {
 		if (!snd_uart16550_write_buffer(uart, midi_byte)) {
-			snd_printk("%s: Buffer overrun on device at 0x%lx\n",
+			snd_printk(KERN_WARNING
+				   "%s: Buffer overrun on device at 0x%lx\n",
 				   uart->rmidi->name, uart->base);
 			return 0;
 		}
@@ -815,7 +817,8 @@ static int __devinit snd_uart16550_create(struct snd_card *card,
 	if (irq >= 0 && irq != SNDRV_AUTO_IRQ) {
 		if (request_irq(irq, snd_uart16550_interrupt,
 				IRQF_DISABLED, "Serial MIDI", uart)) {
-			snd_printk("irq %d busy. Using Polling.\n", irq);
+			snd_printk(KERN_WARNING
+				   "irq %d busy. Using Polling.\n", irq);
 		} else {
 			uart->irq = irq;
 		}
@@ -919,26 +922,29 @@ static int __devinit snd_serial_probe(struct platform_device *devptr)
 	case SNDRV_SERIAL_GENERIC:
 		break;
 	default:
-		snd_printk("Adaptor type is out of range 0-%d (%d)\n",
+		snd_printk(KERN_ERR
+			   "Adaptor type is out of range 0-%d (%d)\n",
 			   SNDRV_SERIAL_MAX_ADAPTOR, adaptor[dev]);
 		return -ENODEV;
 	}
 
 	if (outs[dev] < 1 || outs[dev] > SNDRV_SERIAL_MAX_OUTS) {
-		snd_printk("Count of outputs is out of range 1-%d (%d)\n",
+		snd_printk(KERN_ERR
+			   "Count of outputs is out of range 1-%d (%d)\n",
 			   SNDRV_SERIAL_MAX_OUTS, outs[dev]);
 		return -ENODEV;
 	}
 
 	if (ins[dev] < 1 || ins[dev] > SNDRV_SERIAL_MAX_INS) {
-		snd_printk("Count of inputs is out of range 1-%d (%d)\n",
+		snd_printk(KERN_ERR
+			   "Count of inputs is out of range 1-%d (%d)\n",
 			   SNDRV_SERIAL_MAX_INS, ins[dev]);
 		return -ENODEV;
 	}
 
-	card  = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err  = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	strcpy(card->driver, "Serial");
 	strcpy(card->shortname, "Serial MIDI (UART16550A)");
diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c
index f79e3614079d..0e631c3221e3 100644
--- a/sound/drivers/virmidi.c
+++ b/sound/drivers/virmidi.c
@@ -90,15 +90,17 @@ static int __devinit snd_virmidi_probe(struct platform_device *devptr)
 	int idx, err;
 	int dev = devptr->id;
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-			    sizeof(struct snd_card_virmidi));
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct snd_card_virmidi), &card);
+	if (err < 0)
+		return err;
 	vmidi = (struct snd_card_virmidi *)card->private_data;
 	vmidi->card = card;
 
 	if (midi_devs[dev] > MAX_MIDI_DEVICES) {
-		snd_printk("too much midi devices for virmidi %d: force to use %d\n", dev, MAX_MIDI_DEVICES);
+		snd_printk(KERN_WARNING
+			   "too much midi devices for virmidi %d: "
+			   "force to use %d\n", dev, MAX_MIDI_DEVICES);
 		midi_devs[dev] = MAX_MIDI_DEVICES;
 	}
 	for (idx = 0; idx < midi_devs[dev]; idx++) {
diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c
index 14e3354be43a..19c6e376c7c7 100644
--- a/sound/drivers/vx/vx_core.c
+++ b/sound/drivers/vx/vx_core.c
@@ -688,7 +688,8 @@ int snd_vx_dsp_load(struct vx_core *chip, const struct firmware *dsp)
 		image = dsp->data + i;
 		/* Wait DSP ready for a new read */
 		if ((err = vx_wait_isr_bit(chip, ISR_TX_EMPTY)) < 0) {
-			printk("dsp loading error at position %d\n", i);
+			printk(KERN_ERR
+			       "dsp loading error at position %d\n", i);
 			return err;
 		}
 		cptr = image;
diff --git a/sound/drivers/vx/vx_hwdep.c b/sound/drivers/vx/vx_hwdep.c
index 8d6362e2d4c9..46df8817c18f 100644
--- a/sound/drivers/vx/vx_hwdep.c
+++ b/sound/drivers/vx/vx_hwdep.c
@@ -119,16 +119,6 @@ void snd_vx_free_firmware(struct vx_core *chip)
 
 #else /* old style firmware loading */
 
-static int vx_hwdep_open(struct snd_hwdep *hw, struct file *file)
-{
-	return 0;
-}
-
-static int vx_hwdep_release(struct snd_hwdep *hw, struct file *file)
-{
-	return 0;
-}
-
 static int vx_hwdep_dsp_status(struct snd_hwdep *hw,
 			       struct snd_hwdep_dsp_status *info)
 {
@@ -243,8 +233,6 @@ int snd_vx_setup_firmware(struct vx_core *chip)
 
 	hw->iface = SNDRV_HWDEP_IFACE_VX;
 	hw->private_data = chip;
-	hw->ops.open = vx_hwdep_open;
-	hw->ops.release = vx_hwdep_release;
 	hw->ops.dsp_status = vx_hwdep_dsp_status;
 	hw->ops.dsp_load = vx_hwdep_dsp_load;
 	hw->exclusive = 1;
diff --git a/sound/drivers/vx/vx_uer.c b/sound/drivers/vx/vx_uer.c
index 0e1ba9b47904..b0560fec6bba 100644
--- a/sound/drivers/vx/vx_uer.c
+++ b/sound/drivers/vx/vx_uer.c
@@ -103,7 +103,7 @@ static void vx_write_one_cbit(struct vx_core *chip, int index, int val)
  * returns the frequency of UER, or 0 if not sync,
  * or a negative error code.
  */
-static int vx_read_uer_status(struct vx_core *chip, int *mode)
+static int vx_read_uer_status(struct vx_core *chip, unsigned int *mode)
 {
 	int val, freq;
 
diff --git a/sound/i2c/Makefile b/sound/i2c/Makefile
index 37970666a453..36879bf88700 100644
--- a/sound/i2c/Makefile
+++ b/sound/i2c/Makefile
@@ -7,8 +7,6 @@ snd-i2c-objs := i2c.o
 snd-cs8427-objs := cs8427.o
 snd-tea6330t-objs := tea6330t.o
 
-obj-$(CONFIG_L3) += l3/
-
 obj-$(CONFIG_SND) += other/
 
 # Toplevel Module Dependency
diff --git a/sound/i2c/l3/Makefile b/sound/i2c/l3/Makefile
deleted file mode 100644
index 49455b8dcc04..000000000000
--- a/sound/i2c/l3/Makefile
+++ /dev/null
@@ -1,8 +0,0 @@
-#
-# Makefile for ALSA
-#
-
-snd-uda1341-objs := uda1341.o
-
-# Module Dependency
-obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-uda1341.o
diff --git a/sound/i2c/l3/uda1341.c b/sound/i2c/l3/uda1341.c
deleted file mode 100644
index 9840eb43648d..000000000000
--- a/sound/i2c/l3/uda1341.c
+++ /dev/null
@@ -1,935 +0,0 @@
-/*
- * Philips UDA1341 mixer device driver
- * Copyright (c) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz>
- *
- * Portions are Copyright (C) 2000 Lernout & Hauspie Speech Products, N.V.
- *
- * This program is free software; you can redistribute it and/or
- * modify it under the terms of the GNU General Public License.
- *
- * History:
- *
- * 2002-03-13   Tomas Kasparek  initial release - based on uda1341.c from OSS
- * 2002-03-28   Tomas Kasparek  basic mixer is working (volume, bass, treble)
- * 2002-03-30   Tomas Kasparek  proc filesystem support, complete mixer and DSP
- *                              features support
- * 2002-04-12	Tomas Kasparek	proc interface update, code cleanup
- * 2002-05-12   Tomas Kasparek  another code cleanup
- */
-
-#include <linux/module.h>
-#include <linux/init.h>
-#include <linux/types.h>
-#include <linux/slab.h>
-#include <linux/errno.h>
-#include <linux/ioctl.h>
-
-#include <asm/uaccess.h>
-
-#include <sound/core.h>
-#include <sound/control.h>
-#include <sound/initval.h>
-#include <sound/info.h>
-
-#include <linux/l3/l3.h>
-
-#include <sound/uda1341.h>
-
-/* {{{ HW regs definition */
-
-#define STAT0                   0x00
-#define STAT1			0x80
-#define STAT_MASK               0x80
-
-#define DATA0_0			0x00
-#define DATA0_1			0x40
-#define DATA0_2			0x80
-#define DATA_MASK               0xc0
-
-#define IS_DATA0(x)     ((x) >= data0_0 && (x) <= data0_2)
-#define IS_DATA1(x)     ((x) == data1)
-#define IS_STATUS(x)    ((x) == stat0 || (x) == stat1)
-#define IS_EXTEND(x)   ((x) >= ext0 && (x) <= ext6)
-
-/* }}} */
-
-
-static const char *peak_names[] = {
-	"before",
-	"after",
-};
-
-static const char *filter_names[] = {
-	"flat",
-	"min",
-	"min",
-	"max",
-};
-
-static const char *mixer_names[] = {
-	"double differential",
-	"input channel 1 (line in)",
-	"input channel 2 (microphone)",
-	"digital mixer",
-};
-
-static const char *deemp_names[] = {
-	"none",
-	"32 kHz",
-	"44.1 kHz",
-	"48 kHz",        
-};
-
-enum uda1341_regs_names {
-	stat0,
-	stat1,
-	data0_0,
-	data0_1,
-	data0_2,
-	data1,
-	ext0,
-	ext1,
-	ext2,
-	empty,
-	ext4,
-	ext5,
-	ext6,
-	uda1341_reg_last,
-};
-
-static const char *uda1341_reg_names[] = {
-	"stat 0 ",
-	"stat 1 ",
-	"data 00",
-	"data 01",
-	"data 02",
-	"data 1 ",
-	"ext 0",
-	"ext 1",
-	"ext 2",
-	"empty",
-	"ext 4",
-	"ext 5",
-	"ext 6",
-};
-
-static const int uda1341_enum_items[] = {
-	0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
-	2, //peak - before/after
-	4, //deemp - none/32/44.1/48
-	0,
-	4, //filter - flat/min/min/max
-	0, 0, 0,
-	4, //mixer - differ/line/mic/mixer
-	0, 0, 0, 0, 0,
-};
-
-static const char ** uda1341_enum_names[] = {
-	NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL,
-	peak_names, //peak - before/after
-	deemp_names, //deemp - none/32/44.1/48
-	NULL,
-	filter_names, //filter - flat/min/min/max
-	NULL, NULL, NULL,
-	mixer_names, //mixer - differ/line/mic/mixer
-	NULL, NULL, NULL, NULL, NULL,
-};
-
-typedef int uda1341_cfg[CMD_LAST];
-
-struct uda1341 {
-	int (*write) (struct l3_client *uda1341, unsigned short reg, unsigned short val);
-	int (*read) (struct l3_client *uda1341, unsigned short reg);        
-	unsigned char regs[uda1341_reg_last];
-	int active;
-	spinlock_t reg_lock;
-	struct snd_card *card;
-	uda1341_cfg cfg;
-#ifdef CONFIG_PM
-	unsigned char suspend_regs[uda1341_reg_last];
-	uda1341_cfg suspend_cfg;
-#endif
-};
-
-/* transfer 8bit integer into string with binary representation */
-static void int2str_bin8(uint8_t val, char *buf)
-{
-	const int size = sizeof(val) * 8;
-	int i;
-
-	for (i= 0; i < size; i++){
-		*(buf++) = (val >> (size - 1)) ? '1' : '0';
-		val <<= 1;
-	}
-	*buf = '\0'; //end the string with zero
-}
-
-/* {{{ HW manipulation routines */
-
-static int snd_uda1341_codec_write(struct l3_client *clnt, unsigned short reg, unsigned short val)
-{
-	struct uda1341 *uda = clnt->driver_data;
-	unsigned char buf[2] = { 0xc0, 0xe0 }; // for EXT addressing
-	int err = 0;
-
-	uda->regs[reg] = val;
-
-	if (uda->active) {
-		if (IS_DATA0(reg)) {
-			err = l3_write(clnt, UDA1341_DATA0, (const unsigned char *)&val, 1);
-		} else if (IS_DATA1(reg)) {
-			err = l3_write(clnt, UDA1341_DATA1, (const unsigned char *)&val, 1);
-		} else if (IS_STATUS(reg)) {
-			err = l3_write(clnt, UDA1341_STATUS, (const unsigned char *)&val, 1);
-		} else if (IS_EXTEND(reg)) {
-			buf[0] |= (reg - ext0) & 0x7;   //EXT address
-			buf[1] |= val;                  //EXT data
-			err = l3_write(clnt, UDA1341_DATA0, (const unsigned char *)buf, 2);
-		}
-	} else
-		printk(KERN_ERR "UDA1341 codec not active!\n");
-	return err;
-}
-
-static int snd_uda1341_codec_read(struct l3_client *clnt, unsigned short reg)
-{
-	unsigned char val;
-	int err;
-
-	err = l3_read(clnt, reg, &val, 1);
-	if (err == 1)
-		// use just 6bits - the rest is address of the reg
-		return val & 63;
-	return err < 0 ? err : -EIO;
-}
-
-static inline int snd_uda1341_valid_reg(struct l3_client *clnt, unsigned short reg)
-{
-	return reg < uda1341_reg_last;
-}
-
-static int snd_uda1341_update_bits(struct l3_client *clnt, unsigned short reg,
-				   unsigned short mask, unsigned short shift,
-				   unsigned short value, int flush)
-{
-	int change;
-	unsigned short old, new;
-	struct uda1341 *uda = clnt->driver_data;
-
-#if 0
-	printk(KERN_DEBUG "update_bits: reg: %s mask: %d shift: %d val: %d\n",
-	       uda1341_reg_names[reg], mask, shift, value);
-#endif
-        
-	if (!snd_uda1341_valid_reg(clnt, reg))
-		return -EINVAL;
-	spin_lock(&uda->reg_lock);
-	old = uda->regs[reg];
-	new = (old & ~(mask << shift)) | (value << shift);
-	change = old != new;
-	if (change) {
-		if (flush) uda->write(clnt, reg, new);
-		uda->regs[reg] = new;
-	}
-	spin_unlock(&uda->reg_lock);
-	return change;
-}
-
-static int snd_uda1341_cfg_write(struct l3_client *clnt, unsigned short what,
-				 unsigned short value, int flush)
-{
-	struct uda1341 *uda = clnt->driver_data;
-	int ret = 0;
-#ifdef CONFIG_PM
-	int reg;
-#endif
-
-#if 0
-	printk(KERN_DEBUG "cfg_write what: %d value: %d\n", what, value);
-#endif
-
-	uda->cfg[what] = value;
-        
-	switch(what) {
-	case CMD_RESET:
-		ret = snd_uda1341_update_bits(clnt, data0_2, 1, 2, 1, flush);	// MUTE
-		ret = snd_uda1341_update_bits(clnt, stat0, 1, 6, 1, flush);	// RESET
-		ret = snd_uda1341_update_bits(clnt, stat0, 1, 6, 0, flush);	// RESTORE
-		uda->cfg[CMD_RESET]=0;
-		break;
-	case CMD_FS:
-		ret = snd_uda1341_update_bits(clnt, stat0, 3, 4, value, flush);
-		break;
-	case CMD_FORMAT:
-		ret = snd_uda1341_update_bits(clnt, stat0, 7, 1, value, flush);
-		break;
-	case CMD_OGAIN:
-		ret = snd_uda1341_update_bits(clnt, stat1, 1, 6, value, flush);
-		break;
-	case CMD_IGAIN:
-		ret = snd_uda1341_update_bits(clnt, stat1, 1, 5, value, flush);
-		break;
-	case CMD_DAC:
-		ret = snd_uda1341_update_bits(clnt, stat1, 1, 0, value, flush);
-		break;
-	case CMD_ADC:
-		ret = snd_uda1341_update_bits(clnt, stat1, 1, 1, value, flush);
-		break;
-	case CMD_VOLUME:
-		ret = snd_uda1341_update_bits(clnt, data0_0, 63, 0, value, flush);
-		break;
-	case CMD_BASS:
-		ret = snd_uda1341_update_bits(clnt, data0_1, 15, 2, value, flush);
-		break;
-	case CMD_TREBBLE:
-		ret = snd_uda1341_update_bits(clnt, data0_1, 3, 0, value, flush);
-		break;
-	case CMD_PEAK:
-		ret = snd_uda1341_update_bits(clnt, data0_2, 1, 5, value, flush);
-		break;
-	case CMD_DEEMP:
-		ret = snd_uda1341_update_bits(clnt, data0_2, 3, 3, value, flush);
-		break;
-	case CMD_MUTE:
-		ret = snd_uda1341_update_bits(clnt, data0_2, 1, 2, value, flush);
-		break;
-	case CMD_FILTER:
-		ret = snd_uda1341_update_bits(clnt, data0_2, 3, 0, value, flush);
-		break;
-	case CMD_CH1:
-		ret = snd_uda1341_update_bits(clnt, ext0, 31, 0, value, flush);
-		break;
-	case CMD_CH2:
-		ret = snd_uda1341_update_bits(clnt, ext1, 31, 0, value, flush);
-		break;
-	case CMD_MIC:
-		ret = snd_uda1341_update_bits(clnt, ext2, 7, 2, value, flush);
-		break;
-	case CMD_MIXER:
-		ret = snd_uda1341_update_bits(clnt, ext2, 3, 0, value, flush);
-		break;
-	case CMD_AGC:
-		ret = snd_uda1341_update_bits(clnt, ext4, 1, 4, value, flush);
-		break;
-	case CMD_IG:
-		ret = snd_uda1341_update_bits(clnt, ext4, 3, 0, value & 0x3, flush);
-		ret = snd_uda1341_update_bits(clnt, ext5, 31, 0, value >> 2, flush);
-		break;
-	case CMD_AGC_TIME:
-		ret = snd_uda1341_update_bits(clnt, ext6, 7, 2, value, flush);
-		break;
-	case CMD_AGC_LEVEL:
-		ret = snd_uda1341_update_bits(clnt, ext6, 3, 0, value, flush);
-		break;
-#ifdef CONFIG_PM		
-	case CMD_SUSPEND:
-		for (reg = stat0; reg < uda1341_reg_last; reg++)
-			uda->suspend_regs[reg] = uda->regs[reg];
-		for (reg = 0; reg < CMD_LAST; reg++)
-			uda->suspend_cfg[reg] = uda->cfg[reg];
-		break;
-	case CMD_RESUME:
-		for (reg = stat0; reg < uda1341_reg_last; reg++)
-			snd_uda1341_codec_write(clnt, reg, uda->suspend_regs[reg]);
-		for (reg = 0; reg < CMD_LAST; reg++)
-			uda->cfg[reg] = uda->suspend_cfg[reg];
-		break;
-#endif
-	default:
-		ret = -EINVAL;
-		break;
-	}
-                
-	if (!uda->active)
-		printk(KERN_ERR "UDA1341 codec not active!\n");                
-	return ret;
-}
-
-/* }}} */
-
-/* {{{ Proc interface */
-#ifdef CONFIG_PROC_FS
-
-static const char *format_names[] = {
-	"I2S-bus",
-	"LSB 16bits",
-	"LSB 18bits",
-	"LSB 20bits",
-	"MSB",
-	"in LSB 16bits/out MSB",
-	"in LSB 18bits/out MSB",
-	"in LSB 20bits/out MSB",        
-};
-
-static const char *fs_names[] = {
-	"512*fs",
-	"384*fs",
-	"256*fs",
-	"Unused - bad value!",
-};
-
-static const char* bass_values[][16] = {
-	{"0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB",
-	 "0 dB", "0 dB", "0 dB", "0 dB", "undefined", }, //flat
-	{"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "18 dB",
-	 "18 dB", "18 dB", "18 dB", "18 dB", "undefined",}, // min
-	{"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "18 dB",
-	 "18 dB", "18 dB", "18 dB", "18 dB", "undefined",}, // min
-	{"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "20 dB",
-	 "22 dB", "24 dB", "24 dB", "24 dB", "undefined",}, // max
-};
-
-static const char *mic_sens_value[] = {
-	"-3 dB", "0 dB", "3 dB", "9 dB", "15 dB", "21 dB", "27 dB", "not used",
-};
-
-static const unsigned short AGC_atime[] = {
-	11, 16, 11, 16, 21, 11, 16, 21,
-};
-
-static const unsigned short AGC_dtime[] = {
-	100, 100, 200, 200, 200, 400, 400, 400,
-};
-
-static const char *AGC_level[] = {
-	"-9.0", "-11.5", "-15.0", "-17.5",
-};
-
-static const char *ig_small_value[] = {
-	"-3.0", "-2.5", "-2.0", "-1.5", "-1.0", "-0.5",
-};
-
-/*
- * this was computed as peak_value[i] = pow((63-i)*1.42,1.013)
- *
- * UDA1341 datasheet on page 21: Peak value (dB) = (Peak level - 63.5)*5*log2
- * There is an table with these values [level]=value: [3]=-90.31, [7]=-84.29
- * [61]=-2.78, [62] = -1.48, [63] = 0.0
- * I tried to compute it, but using but even using logarithm with base either 10 or 2
- * i was'n able to get values in the table from the formula. So I constructed another
- * formula (see above) to interpolate the values as good as possible. If there is some
- * mistake, please contact me on tomas.kasparek@seznam.cz. Thanks.
- * UDA1341TS datasheet is available at:
- *   http://www-us9.semiconductors.com/acrobat/datasheets/UDA1341TS_3.pdf 
- */
-static const char *peak_value[] = {
-	"-INF dB", "N.A.", "N.A", "90.31 dB", "N.A.", "N.A.", "N.A.", "-84.29 dB",
-	"-82.65 dB", "-81.13 dB", "-79.61 dB", "-78.09 dB", "-76.57 dB", "-75.05 dB", "-73.53 dB",
-	"-72.01 dB", "-70.49 dB", "-68.97 dB", "-67.45 dB", "-65.93 dB", "-64.41 dB", "-62.90 dB",
-	"-61.38 dB", "-59.86 dB", "-58.35 dB", "-56.83 dB", "-55.32 dB", "-53.80 dB", "-52.29 dB",
-	"-50.78 dB", "-49.26 dB", "-47.75 dB", "-46.24 dB", "-44.73 dB", "-43.22 dB", "-41.71 dB",
-	"-40.20 dB", "-38.69 dB", "-37.19 dB", "-35.68 dB", "-34.17 dB", "-32.67 dB", "-31.17 dB",
-	"-29.66 dB", "-28.16 dB", "-26.66 dB", "-25.16 dB", "-23.66 dB", "-22.16 dB", "-20.67 dB",
-	"-19.17 dB", "-17.68 dB", "-16.19 dB", "-14.70 dB", "-13.21 dB", "-11.72 dB", "-10.24 dB",
-	"-8.76 dB", "-7.28 dB", "-5.81 dB", "-4.34 dB", "-2.88 dB", "-1.43 dB", "0.00 dB",
-};
-
-static void snd_uda1341_proc_read(struct snd_info_entry *entry, 
-				  struct snd_info_buffer *buffer)
-{
-	struct l3_client *clnt = entry->private_data;
-	struct uda1341 *uda = clnt->driver_data;
-	int peak;
-
-	peak = snd_uda1341_codec_read(clnt, UDA1341_DATA1);
-	if (peak < 0)
-		peak = 0;
-	
-	snd_iprintf(buffer, "%s\n\n", uda->card->longname);
-
-	// for information about computed values see UDA1341TS datasheet pages 15 - 21
-	snd_iprintf(buffer, "DAC power           : %s\n", uda->cfg[CMD_DAC] ? "on" : "off");
-	snd_iprintf(buffer, "ADC power           : %s\n", uda->cfg[CMD_ADC] ? "on" : "off");
- 	snd_iprintf(buffer, "Clock frequency     : %s\n", fs_names[uda->cfg[CMD_FS]]);
-	snd_iprintf(buffer, "Data format         : %s\n\n", format_names[uda->cfg[CMD_FORMAT]]);
-
-	snd_iprintf(buffer, "Filter mode         : %s\n", filter_names[uda->cfg[CMD_FILTER]]);
-	snd_iprintf(buffer, "Mixer mode          : %s\n", mixer_names[uda->cfg[CMD_MIXER]]);
-	snd_iprintf(buffer, "De-emphasis         : %s\n", deemp_names[uda->cfg[CMD_DEEMP]]);	
-	snd_iprintf(buffer, "Peak detection pos. : %s\n", uda->cfg[CMD_PEAK] ? "after" : "before");
-	snd_iprintf(buffer, "Peak value          : %s\n\n", peak_value[peak]);		
-	
-	snd_iprintf(buffer, "Automatic Gain Ctrl : %s\n", uda->cfg[CMD_AGC] ? "on" : "off");
-	snd_iprintf(buffer, "AGC attack time     : %d ms\n", AGC_atime[uda->cfg[CMD_AGC_TIME]]);
-	snd_iprintf(buffer, "AGC decay time      : %d ms\n", AGC_dtime[uda->cfg[CMD_AGC_TIME]]);
-	snd_iprintf(buffer, "AGC output level    : %s dB\n\n", AGC_level[uda->cfg[CMD_AGC_LEVEL]]);
-
-	snd_iprintf(buffer, "Mute                : %s\n", uda->cfg[CMD_MUTE] ? "on" : "off");
-
-	if (uda->cfg[CMD_VOLUME] == 0)
-		snd_iprintf(buffer, "Volume              : 0 dB\n");
-	else if (uda->cfg[CMD_VOLUME] < 62)
-		snd_iprintf(buffer, "Volume              : %d dB\n", -1*uda->cfg[CMD_VOLUME] +1);
-	else
-		snd_iprintf(buffer, "Volume              : -INF dB\n");
-	snd_iprintf(buffer, "Bass                : %s\n", bass_values[uda->cfg[CMD_FILTER]][uda->cfg[CMD_BASS]]);
-	snd_iprintf(buffer, "Trebble             : %d dB\n", uda->cfg[CMD_FILTER] ? 2*uda->cfg[CMD_TREBBLE] : 0);
-	snd_iprintf(buffer, "Input Gain (6dB)    : %s\n", uda->cfg[CMD_IGAIN] ? "on" : "off");
-	snd_iprintf(buffer, "Output Gain (6dB)   : %s\n", uda->cfg[CMD_OGAIN] ? "on" : "off");
-	snd_iprintf(buffer, "Mic sensitivity     : %s\n", mic_sens_value[uda->cfg[CMD_MIC]]);
-
-	
-	if(uda->cfg[CMD_CH1] < 31)
-		snd_iprintf(buffer, "Mixer gain channel 1: -%d.%c dB\n",
-			    ((uda->cfg[CMD_CH1] >> 1) * 3) + (uda->cfg[CMD_CH1] & 1),
-			    uda->cfg[CMD_CH1] & 1 ? '5' : '0');
-	else
-		snd_iprintf(buffer, "Mixer gain channel 1: -INF dB\n");
-	if(uda->cfg[CMD_CH2] < 31)
-		snd_iprintf(buffer, "Mixer gain channel 2: -%d.%c dB\n",
-			    ((uda->cfg[CMD_CH2] >> 1) * 3) + (uda->cfg[CMD_CH2] & 1),
-			    uda->cfg[CMD_CH2] & 1 ? '5' : '0');
-	else
-		snd_iprintf(buffer, "Mixer gain channel 2: -INF dB\n");
-
-	if(uda->cfg[CMD_IG] > 5)
-		snd_iprintf(buffer, "Input Amp. Gain ch 2: %d.%c dB\n",
-			    (uda->cfg[CMD_IG] >> 1) -3, uda->cfg[CMD_IG] & 1 ? '5' : '0');
-	else
-		snd_iprintf(buffer, "Input Amp. Gain ch 2: %s dB\n",  ig_small_value[uda->cfg[CMD_IG]]);
-}
-
-static void snd_uda1341_proc_regs_read(struct snd_info_entry *entry, 
-				       struct snd_info_buffer *buffer)
-{
-	struct l3_client *clnt = entry->private_data;
-	struct uda1341 *uda = clnt->driver_data;		
-	int reg;
-	char buf[12];
-
-	for (reg = 0; reg < uda1341_reg_last; reg ++) {
-		if (reg == empty)
-			continue;
-		int2str_bin8(uda->regs[reg], buf);
-		snd_iprintf(buffer, "%s = %s\n", uda1341_reg_names[reg], buf);
-	}
-
-	int2str_bin8(snd_uda1341_codec_read(clnt, UDA1341_DATA1), buf);
-	snd_iprintf(buffer, "DATA1 = %s\n", buf);
-}
-#endif /* CONFIG_PROC_FS */
-
-static void __devinit snd_uda1341_proc_init(struct snd_card *card, struct l3_client *clnt)
-{
-	struct snd_info_entry *entry;
-
-	if (! snd_card_proc_new(card, "uda1341", &entry))
-		snd_info_set_text_ops(entry, clnt, snd_uda1341_proc_read);
-	if (! snd_card_proc_new(card, "uda1341-regs", &entry))
-		snd_info_set_text_ops(entry, clnt, snd_uda1341_proc_regs_read);
-}
-
-/* }}} */
-
-/* {{{ Mixer controls setting */
-
-/* {{{ UDA1341 single functions */
-
-#define UDA1341_SINGLE(xname, where, reg, shift, mask, invert) \
-{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_uda1341_info_single, \
-  .get = snd_uda1341_get_single, .put = snd_uda1341_put_single, \
-  .private_value = where | (reg << 5) | (shift << 9) | (mask << 12) | (invert << 18) \
-}
-
-static int snd_uda1341_info_single(struct snd_kcontrol *kcontrol,
-				   struct snd_ctl_elem_info *uinfo)
-{
-	int mask = (kcontrol->private_value >> 12) & 63;
-
-	uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
-	uinfo->count = 1;
-	uinfo->value.integer.min = 0;
-	uinfo->value.integer.max = mask;
-	return 0;
-}
-
-static int snd_uda1341_get_single(struct snd_kcontrol *kcontrol,
-				  struct snd_ctl_elem_value *ucontrol)
-{
-	struct l3_client *clnt = snd_kcontrol_chip(kcontrol);
-	struct uda1341 *uda = clnt->driver_data;
-	int where = kcontrol->private_value & 31;        
-	int mask = (kcontrol->private_value >> 12) & 63;
-	int invert = (kcontrol->private_value >> 18) & 1;
-        
-	ucontrol->value.integer.value[0] = uda->cfg[where];
-	if (invert)
-		ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
-
-	return 0;
-}
-
-static int snd_uda1341_put_single(struct snd_kcontrol *kcontrol,
-				  struct snd_ctl_elem_value *ucontrol)
-{
-	struct l3_client *clnt = snd_kcontrol_chip(kcontrol);
-	struct uda1341 *uda = clnt->driver_data;
-	int where = kcontrol->private_value & 31;        
-	int reg = (kcontrol->private_value >> 5) & 15;
-	int shift = (kcontrol->private_value >> 9) & 7;
-	int mask = (kcontrol->private_value >> 12) & 63;
-	int invert = (kcontrol->private_value >> 18) & 1;
-	unsigned short val;
-
-	val = (ucontrol->value.integer.value[0] & mask);
-	if (invert)
-		val = mask - val;
-
-	uda->cfg[where] = val;
-	return snd_uda1341_update_bits(clnt, reg, mask, shift, val, FLUSH);
-}
-
-/* }}} */
-
-/* {{{ UDA1341 enum functions */
-
-#define UDA1341_ENUM(xname, where, reg, shift, mask, invert) \
-{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_uda1341_info_enum, \
-  .get = snd_uda1341_get_enum, .put = snd_uda1341_put_enum, \
-  .private_value = where | (reg << 5) | (shift << 9) | (mask << 12) | (invert << 18) \
-}
-
-static int snd_uda1341_info_enum(struct snd_kcontrol *kcontrol,
-				 struct snd_ctl_elem_info *uinfo)
-{
-	int where = kcontrol->private_value & 31;
-	const char **texts;
-	
-	// this register we don't handle this way
-	if (!uda1341_enum_items[where])
-		return -EINVAL;
-
-	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
-	uinfo->count = 1;
-	uinfo->value.enumerated.items = uda1341_enum_items[where];
-
-	if (uinfo->value.enumerated.item >= uda1341_enum_items[where])
-		uinfo->value.enumerated.item = uda1341_enum_items[where] - 1;
-
-	texts = uda1341_enum_names[where];
-	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
-	return 0;
-}
-
-static int snd_uda1341_get_enum(struct snd_kcontrol *kcontrol,
-				struct snd_ctl_elem_value *ucontrol)
-{
-	struct l3_client *clnt = snd_kcontrol_chip(kcontrol);
-	struct uda1341 *uda = clnt->driver_data;
-	int where = kcontrol->private_value & 31;        
-        
-	ucontrol->value.enumerated.item[0] = uda->cfg[where];	
-	return 0;
-}
-
-static int snd_uda1341_put_enum(struct snd_kcontrol *kcontrol,
-				struct snd_ctl_elem_value *ucontrol)
-{
-	struct l3_client *clnt = snd_kcontrol_chip(kcontrol);
-	struct uda1341 *uda = clnt->driver_data;
-	int where = kcontrol->private_value & 31;        
-	int reg = (kcontrol->private_value >> 5) & 15;
-	int shift = (kcontrol->private_value >> 9) & 7;
-	int mask = (kcontrol->private_value >> 12) & 63;
-
-	uda->cfg[where] = (ucontrol->value.enumerated.item[0] & mask);
-	
-	return snd_uda1341_update_bits(clnt, reg, mask, shift, uda->cfg[where], FLUSH);
-}
-
-/* }}} */
-
-/* {{{ UDA1341 2regs functions */
-
-#define UDA1341_2REGS(xname, where, reg_1, reg_2, shift_1, shift_2, mask_1, mask_2, invert) \
-{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), .info = snd_uda1341_info_2regs, \
-  .get = snd_uda1341_get_2regs, .put = snd_uda1341_put_2regs, \
-  .private_value = where | (reg_1 << 5) | (reg_2 << 9) | (shift_1 << 13) | (shift_2 << 16) | \
-                         (mask_1 << 19) | (mask_2 << 25) | (invert << 31) \
-}
-
-
-static int snd_uda1341_info_2regs(struct snd_kcontrol *kcontrol,
-				  struct snd_ctl_elem_info *uinfo)
-{
-	int mask_1 = (kcontrol->private_value >> 19) & 63;
-	int mask_2 = (kcontrol->private_value >> 25) & 63;
-	int mask;
-        
-	mask = (mask_2 + 1) * (mask_1 + 1) - 1;
-	uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER;
-	uinfo->count = 1;
-	uinfo->value.integer.min = 0;
-	uinfo->value.integer.max = mask;
-	return 0;
-}
-
-static int snd_uda1341_get_2regs(struct snd_kcontrol *kcontrol,
-				 struct snd_ctl_elem_value *ucontrol)
-{
-	struct l3_client *clnt = snd_kcontrol_chip(kcontrol);
-	struct uda1341 *uda = clnt->driver_data;
-	int where = kcontrol->private_value & 31;
-	int mask_1 = (kcontrol->private_value >> 19) & 63;
-	int mask_2 = (kcontrol->private_value >> 25) & 63;        
-	int invert = (kcontrol->private_value >> 31) & 1;
-	int mask;
-
-	mask = (mask_2 + 1) * (mask_1 + 1) - 1;
-
-	ucontrol->value.integer.value[0] = uda->cfg[where];
-	if (invert)
-		ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0];
-	return 0;
-}
-
-static int snd_uda1341_put_2regs(struct snd_kcontrol *kcontrol,
-				 struct snd_ctl_elem_value *ucontrol)
-{
-	struct l3_client *clnt = snd_kcontrol_chip(kcontrol);
-	struct uda1341 *uda = clnt->driver_data;        
-	int where = kcontrol->private_value & 31;        
-	int reg_1 = (kcontrol->private_value >> 5) & 15;
-	int reg_2 = (kcontrol->private_value >> 9) & 15;        
-	int shift_1 = (kcontrol->private_value >> 13) & 7;
-	int shift_2 = (kcontrol->private_value >> 16) & 7;
-	int mask_1 = (kcontrol->private_value >> 19) & 63;
-	int mask_2 = (kcontrol->private_value >> 25) & 63;        
-	int invert = (kcontrol->private_value >> 31) & 1;
-	int mask;
-	unsigned short val1, val2, val;
-
-	val = ucontrol->value.integer.value[0];
-         
-	mask = (mask_2 + 1) * (mask_1 + 1) - 1;
-
-	val1 = val & mask_1;
-	val2 = (val / (mask_1 + 1)) & mask_2;        
-
-	if (invert) {
-		val1 = mask_1 - val1;
-		val2 = mask_2 - val2;
-	}
-
-	uda->cfg[where] = invert ? mask - val : val;
-        
-	//FIXME - return value
-	snd_uda1341_update_bits(clnt, reg_1, mask_1, shift_1, val1, FLUSH);
-	return snd_uda1341_update_bits(clnt, reg_2, mask_2, shift_2, val2, FLUSH);
-}
-
-/* }}} */
-  
-static struct snd_kcontrol_new snd_uda1341_controls[] = {
-	UDA1341_SINGLE("Master Playback Switch", CMD_MUTE, data0_2, 2, 1, 1),
-	UDA1341_SINGLE("Master Playback Volume", CMD_VOLUME, data0_0, 0, 63, 1),
-
-	UDA1341_SINGLE("Bass Playback Volume", CMD_BASS, data0_1, 2, 15, 0),
-	UDA1341_SINGLE("Treble Playback Volume", CMD_TREBBLE, data0_1, 0, 3, 0),
-
-	UDA1341_SINGLE("Input Gain Switch", CMD_IGAIN, stat1, 5, 1, 0),
-	UDA1341_SINGLE("Output Gain Switch", CMD_OGAIN, stat1, 6, 1, 0),
-
-	UDA1341_SINGLE("Mixer Gain Channel 1 Volume", CMD_CH1, ext0, 0, 31, 1),
-	UDA1341_SINGLE("Mixer Gain Channel 2 Volume", CMD_CH2, ext1, 0, 31, 1),
-
-	UDA1341_SINGLE("Mic Sensitivity Volume", CMD_MIC, ext2, 2, 7, 0),
-
-	UDA1341_SINGLE("AGC Output Level", CMD_AGC_LEVEL, ext6, 0, 3, 0),
-	UDA1341_SINGLE("AGC Time Constant", CMD_AGC_TIME, ext6, 2, 7, 0),
-	UDA1341_SINGLE("AGC Time Constant Switch", CMD_AGC, ext4, 4, 1, 0),
-
-	UDA1341_SINGLE("DAC Power", CMD_DAC, stat1, 0, 1, 0),
-	UDA1341_SINGLE("ADC Power", CMD_ADC, stat1, 1, 1, 0),
-
-	UDA1341_ENUM("Peak detection", CMD_PEAK, data0_2, 5, 1, 0),
-	UDA1341_ENUM("De-emphasis", CMD_DEEMP, data0_2, 3, 3, 0),
-	UDA1341_ENUM("Mixer mode", CMD_MIXER, ext2, 0, 3, 0),
-	UDA1341_ENUM("Filter mode", CMD_FILTER, data0_2, 0, 3, 0),
-
-	UDA1341_2REGS("Gain Input Amplifier Gain (channel 2)", CMD_IG, ext4, ext5, 0, 0, 3, 31, 0),
-};
-
-static void uda1341_free(struct l3_client *clnt)
-{
-	l3_detach_client(clnt); // calls kfree for driver_data (struct uda1341)
-	kfree(clnt);
-}
-
-static int uda1341_dev_free(struct snd_device *device)
-{
-	struct l3_client *clnt = device->device_data;
-	uda1341_free(clnt);
-	return 0;
-}
-
-int __init snd_chip_uda1341_mixer_new(struct snd_card *card, struct l3_client **clntp)
-{
-	static struct snd_device_ops ops = {
-		.dev_free =     uda1341_dev_free,
-	};
-	struct l3_client *clnt;
-	int idx, err;
-
-	if (snd_BUG_ON(!card))
-		return -EINVAL;
-
-	clnt = kzalloc(sizeof(*clnt), GFP_KERNEL);
-	if (clnt == NULL)
-		return -ENOMEM;
-         
-	if ((err = l3_attach_client(clnt, "l3-bit-sa1100-gpio", UDA1341_ALSA_NAME))) {
-		kfree(clnt);
-		return err;
-	}
-
-	for (idx = 0; idx < ARRAY_SIZE(snd_uda1341_controls); idx++) {
-		if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_uda1341_controls[idx], clnt))) < 0) {
-			uda1341_free(clnt);
-			return err;
-		}
-	}
-
-	if ((err = snd_device_new(card, SNDRV_DEV_CODEC, clnt, &ops)) < 0) {
-		uda1341_free(clnt);
-		return err;
-	}
-
-	*clntp = clnt;
-	strcpy(card->mixername, "UDA1341TS Mixer");
-	((struct uda1341 *)clnt->driver_data)->card = card;
-        
-	snd_uda1341_proc_init(card, clnt);
-        
-	return 0;
-}
-
-/* }}} */
-
-/* {{{ L3 operations */
-
-static int uda1341_attach(struct l3_client *clnt)
-{
-	struct uda1341 *uda;
-
-	uda = kzalloc(sizeof(*uda), 0, GFP_KERNEL);
-	if (!uda)
-		return -ENOMEM;
-
-	/* init fixed parts of my copy of registers */
-	uda->regs[stat0]   = STAT0;
-	uda->regs[stat1]   = STAT1;
-
-	uda->regs[data0_0] = DATA0_0;
-	uda->regs[data0_1] = DATA0_1;
-	uda->regs[data0_2] = DATA0_2;
-
-	uda->write = snd_uda1341_codec_write;
-	uda->read = snd_uda1341_codec_read;
-  
-	spin_lock_init(&uda->reg_lock);
-        
-	clnt->driver_data = uda;
-	return 0;
-}
-
-static void uda1341_detach(struct l3_client *clnt)
-{
-	kfree(clnt->driver_data);
-}
-
-static int
-uda1341_command(struct l3_client *clnt, int cmd, void *arg)
-{
-	if (cmd != CMD_READ_REG)
-		return snd_uda1341_cfg_write(clnt, cmd, (int) arg, FLUSH);
-
-	return snd_uda1341_codec_read(clnt, (int) arg);
-}
-
-static int uda1341_open(struct l3_client *clnt)
-{
-	struct uda1341 *uda = clnt->driver_data;
-
-	uda->active = 1;
-
-	/* init default configuration */
-	snd_uda1341_cfg_write(clnt, CMD_RESET, 0, REGS_ONLY);
-	snd_uda1341_cfg_write(clnt, CMD_FS, F256, FLUSH);       // unknown state after reset
-	snd_uda1341_cfg_write(clnt, CMD_FORMAT, LSB16, FLUSH);  // unknown state after reset
-	snd_uda1341_cfg_write(clnt, CMD_OGAIN, ON, FLUSH);      // default off after reset
-	snd_uda1341_cfg_write(clnt, CMD_IGAIN, ON, FLUSH);      // default off after reset
-	snd_uda1341_cfg_write(clnt, CMD_DAC, ON, FLUSH);	// ??? default value after reset
-	snd_uda1341_cfg_write(clnt, CMD_ADC, ON, FLUSH);	// ??? default value after reset
-	snd_uda1341_cfg_write(clnt, CMD_VOLUME, 20, FLUSH);     // default 0dB after reset
-	snd_uda1341_cfg_write(clnt, CMD_BASS, 0, REGS_ONLY);    // default value after reset
-	snd_uda1341_cfg_write(clnt, CMD_TREBBLE, 0, REGS_ONLY); // default value after reset
-	snd_uda1341_cfg_write(clnt, CMD_PEAK, AFTER, REGS_ONLY);// default value after reset
-	snd_uda1341_cfg_write(clnt, CMD_DEEMP, NONE, REGS_ONLY);// default value after reset
-	//at this moment should be QMUTED by h3600_audio_init
-	snd_uda1341_cfg_write(clnt, CMD_MUTE, OFF, REGS_ONLY);  // default value after reset
-	snd_uda1341_cfg_write(clnt, CMD_FILTER, MAX, FLUSH);    // defaul flat after reset
-	snd_uda1341_cfg_write(clnt, CMD_CH1, 31, FLUSH);        // default value after reset
-	snd_uda1341_cfg_write(clnt, CMD_CH2, 4, FLUSH);         // default value after reset
-	snd_uda1341_cfg_write(clnt, CMD_MIC, 4, FLUSH);         // default 0dB after reset
-	snd_uda1341_cfg_write(clnt, CMD_MIXER, MIXER, FLUSH);   // default doub.dif.mode          
-	snd_uda1341_cfg_write(clnt, CMD_AGC, OFF, FLUSH);       // default value after reset
-	snd_uda1341_cfg_write(clnt, CMD_IG, 0, FLUSH);          // unknown state after reset
-	snd_uda1341_cfg_write(clnt, CMD_AGC_TIME, 0, FLUSH);    // default value after reset
-	snd_uda1341_cfg_write(clnt, CMD_AGC_LEVEL, 0, FLUSH);   // default value after reset
-
-	return 0;
-}
-
-static void uda1341_close(struct l3_client *clnt)
-{
-	struct uda1341 *uda = clnt->driver_data;
-
-	uda->active = 0;
-}
-
-/* }}} */
-
-/* {{{ Module and L3 initialization */
-
-static struct l3_ops uda1341_ops = {
-	.open =		uda1341_open,
-	.command =	uda1341_command,
-	.close =	uda1341_close,
-};
-
-static struct l3_driver uda1341_driver = {
-	.name =		UDA1341_ALSA_NAME,
-	.attach_client = uda1341_attach,
-	.detach_client = uda1341_detach,
-	.ops =		&uda1341_ops,
-	.owner =	THIS_MODULE,
-};
-
-static int __init uda1341_init(void)
-{
-	return l3_add_driver(&uda1341_driver);
-}
-
-static void __exit uda1341_exit(void)
-{
-	l3_del_driver(&uda1341_driver);
-}
-
-module_init(uda1341_init);
-module_exit(uda1341_exit);
-
-MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>");
-MODULE_LICENSE("GPL");
-MODULE_DESCRIPTION("Philips UDA1341 CODEC driver for ALSA");
-MODULE_SUPPORTED_DEVICE("{{UDA1341,UDA1341TS}}");
-
-EXPORT_SYMBOL(snd_chip_uda1341_mixer_new);
-
-/* }}} */
-
-/*
- * Local variables:
- * indent-tabs-mode: t
- * End:
- */
diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c
index 9d98a6658ac9..d31c373e076d 100644
--- a/sound/i2c/other/tea575x-tuner.c
+++ b/sound/i2c/other/tea575x-tuner.c
@@ -24,6 +24,7 @@
 #include <linux/delay.h>
 #include <linux/interrupt.h>
 #include <linux/init.h>
+#include <linux/version.h>
 #include <sound/core.h>
 #include <sound/tea575x-tuner.h>
 
@@ -31,6 +32,13 @@ MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_DESCRIPTION("Routines for control of TEA5757/5759 Philips AM/FM radio tuner chips");
 MODULE_LICENSE("GPL");
 
+static int radio_nr = -1;
+module_param(radio_nr, int, 0);
+
+#define RADIO_VERSION KERNEL_VERSION(0, 0, 2)
+#define FREQ_LO		 (87 * 16000)
+#define FREQ_HI		(108 * 16000)
+
 /*
  * definitions
  */
@@ -53,6 +61,17 @@ MODULE_LICENSE("GPL");
 #define TEA575X_BIT_DUMMY	(1<<15)		/* buffer */
 #define TEA575X_BIT_FREQ_MASK	0x7fff
 
+static struct v4l2_queryctrl radio_qctrl[] = {
+	{
+		.id            = V4L2_CID_AUDIO_MUTE,
+		.name          = "Mute",
+		.minimum       = 0,
+		.maximum       = 1,
+		.default_value = 1,
+		.type          = V4L2_CTRL_TYPE_BOOLEAN,
+	}
+};
+
 /*
  * lowlevel part
  */
@@ -84,94 +103,146 @@ static void snd_tea575x_set_freq(struct snd_tea575x *tea)
  * Linux Video interface
  */
 
-static long snd_tea575x_ioctl(struct file *file,
-			     unsigned int cmd, unsigned long data)
+static int vidioc_querycap(struct file *file, void  *priv,
+					struct v4l2_capability *v)
 {
 	struct snd_tea575x *tea = video_drvdata(file);
-	void __user *arg = (void __user *)data;
-
-	switch(cmd) {
-		case VIDIOCGCAP:
-		{
-			struct video_capability v;
-			v.type = VID_TYPE_TUNER;
-			v.channels = 1;
-			v.audios = 1;
-			/* No we don't do pictures */
-			v.maxwidth = 0;
-			v.maxheight = 0;
-			v.minwidth = 0;
-			v.minheight = 0;
-			strcpy(v.name, tea->tea5759 ? "TEA5759" : "TEA5757");
-			if (copy_to_user(arg,&v,sizeof(v)))
-				return -EFAULT;
-			return 0;
-		}
-		case VIDIOCGTUNER:
-		{
-			struct video_tuner v;
-			if (copy_from_user(&v, arg,sizeof(v))!=0)
-				return -EFAULT;
-			if (v.tuner)	/* Only 1 tuner */
-				return -EINVAL;
-			v.rangelow = (87*16000);
-			v.rangehigh = (108*16000);
-			v.flags = VIDEO_TUNER_LOW;
-			v.mode = VIDEO_MODE_AUTO;
-			strcpy(v.name, "FM");
-			v.signal = 0xFFFF;
-			if (copy_to_user(arg, &v, sizeof(v)))
-				return -EFAULT;
-			return 0;
-		}
-		case VIDIOCSTUNER:
-		{
-			struct video_tuner v;
-			if(copy_from_user(&v, arg, sizeof(v)))
-				return -EFAULT;
-			if(v.tuner!=0)
-				return -EINVAL;
-			/* Only 1 tuner so no setting needed ! */
+
+	strcpy(v->card, tea->tea5759 ? "TEA5759" : "TEA5757");
+	strlcpy(v->driver, "tea575x-tuner", sizeof(v->driver));
+	strlcpy(v->card, "Maestro Radio", sizeof(v->card));
+	sprintf(v->bus_info, "PCI");
+	v->version = RADIO_VERSION;
+	v->capabilities = V4L2_CAP_TUNER;
+	return 0;
+}
+
+static int vidioc_g_tuner(struct file *file, void *priv,
+					struct v4l2_tuner *v)
+{
+	if (v->index > 0)
+		return -EINVAL;
+
+	strcpy(v->name, "FM");
+	v->type = V4L2_TUNER_RADIO;
+	v->rangelow = FREQ_LO;
+	v->rangehigh = FREQ_HI;
+	v->rxsubchans = V4L2_TUNER_SUB_MONO|V4L2_TUNER_SUB_STEREO;
+	v->capability = V4L2_TUNER_CAP_LOW;
+	v->audmode = V4L2_TUNER_MODE_MONO;
+	v->signal = 0xffff;
+	return 0;
+}
+
+static int vidioc_s_tuner(struct file *file, void *priv,
+					struct v4l2_tuner *v)
+{
+	if (v->index > 0)
+		return -EINVAL;
+	return 0;
+}
+
+static int vidioc_g_frequency(struct file *file, void *priv,
+					struct v4l2_frequency *f)
+{
+	struct snd_tea575x *tea = video_drvdata(file);
+
+	f->type = V4L2_TUNER_RADIO;
+	f->frequency = tea->freq;
+	return 0;
+}
+
+static int vidioc_s_frequency(struct file *file, void *priv,
+					struct v4l2_frequency *f)
+{
+	struct snd_tea575x *tea = video_drvdata(file);
+
+	if (f->frequency < FREQ_LO || f->frequency > FREQ_HI)
+		return -EINVAL;
+
+	tea->freq = f->frequency;
+
+	snd_tea575x_set_freq(tea);
+
+	return 0;
+}
+
+static int vidioc_g_audio(struct file *file, void *priv,
+					struct v4l2_audio *a)
+{
+	if (a->index > 1)
+		return -EINVAL;
+
+	strcpy(a->name, "Radio");
+	a->capability = V4L2_AUDCAP_STEREO;
+	return 0;
+}
+
+static int vidioc_s_audio(struct file *file, void *priv,
+					struct v4l2_audio *a)
+{
+	if (a->index != 0)
+		return -EINVAL;
+	return 0;
+}
+
+static int vidioc_queryctrl(struct file *file, void *priv,
+					struct v4l2_queryctrl *qc)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(radio_qctrl); i++) {
+		if (qc->id && qc->id == radio_qctrl[i].id) {
+			memcpy(qc, &(radio_qctrl[i]),
+						sizeof(*qc));
 			return 0;
 		}
-		case VIDIOCGFREQ:
-			if(copy_to_user(arg, &tea->freq, sizeof(tea->freq)))
-				return -EFAULT;
-			return 0;
-		case VIDIOCSFREQ:
-			if(copy_from_user(&tea->freq, arg, sizeof(tea->freq)))
-				return -EFAULT;
-			snd_tea575x_set_freq(tea);
-			return 0;
-		case VIDIOCGAUDIO:
-		{
-			struct video_audio v;
-			memset(&v, 0, sizeof(v));
-			strcpy(v.name, "Radio");
-			if(copy_to_user(arg,&v, sizeof(v)))
-				return -EFAULT;
+	}
+	return -EINVAL;
+}
+
+static int vidioc_g_ctrl(struct file *file, void *priv,
+					struct v4l2_control *ctrl)
+{
+	struct snd_tea575x *tea = video_drvdata(file);
+
+	switch (ctrl->id) {
+	case V4L2_CID_AUDIO_MUTE:
+		if (tea->ops->mute) {
+			ctrl->value = tea->mute;
 			return 0;
 		}
-		case VIDIOCSAUDIO:
-		{
-			struct video_audio v;
-			if(copy_from_user(&v, arg, sizeof(v)))
-				return -EFAULT;
-			if (tea->ops->mute)
-				tea->ops->mute(tea,
-					       (v.flags &
-						VIDEO_AUDIO_MUTE) ? 1 : 0);
-			if(v.audio)
-				return -EINVAL;
+	}
+	return -EINVAL;
+}
+
+static int vidioc_s_ctrl(struct file *file, void *priv,
+					struct v4l2_control *ctrl)
+{
+	struct snd_tea575x *tea = video_drvdata(file);
+
+	switch (ctrl->id) {
+	case V4L2_CID_AUDIO_MUTE:
+		if (tea->ops->mute) {
+			tea->ops->mute(tea, ctrl->value);
+			tea->mute = 1;
 			return 0;
 		}
-		default:
-			return -ENOIOCTLCMD;
 	}
+	return -EINVAL;
+}
+
+static int vidioc_g_input(struct file *filp, void *priv, unsigned int *i)
+{
+	*i = 0;
+	return 0;
 }
 
-static void snd_tea575x_release(struct video_device *vfd)
+static int vidioc_s_input(struct file *filp, void *priv, unsigned int i)
 {
+	if (i != 0)
+		return -EINVAL;
+	return 0;
 }
 
 static int snd_tea575x_exclusive_open(struct file *file)
@@ -189,50 +260,91 @@ static int snd_tea575x_exclusive_release(struct file *file)
 	return 0;
 }
 
+static const struct v4l2_file_operations tea575x_fops = {
+	.owner		= THIS_MODULE,
+	.open           = snd_tea575x_exclusive_open,
+	.release        = snd_tea575x_exclusive_release,
+	.ioctl		= video_ioctl2,
+};
+
+static const struct v4l2_ioctl_ops tea575x_ioctl_ops = {
+	.vidioc_querycap    = vidioc_querycap,
+	.vidioc_g_tuner     = vidioc_g_tuner,
+	.vidioc_s_tuner     = vidioc_s_tuner,
+	.vidioc_g_audio     = vidioc_g_audio,
+	.vidioc_s_audio     = vidioc_s_audio,
+	.vidioc_g_input     = vidioc_g_input,
+	.vidioc_s_input     = vidioc_s_input,
+	.vidioc_g_frequency = vidioc_g_frequency,
+	.vidioc_s_frequency = vidioc_s_frequency,
+	.vidioc_queryctrl   = vidioc_queryctrl,
+	.vidioc_g_ctrl      = vidioc_g_ctrl,
+	.vidioc_s_ctrl      = vidioc_s_ctrl,
+};
+
+static struct video_device tea575x_radio = {
+	.name           = "tea575x-tuner",
+	.fops           = &tea575x_fops,
+	.ioctl_ops 	= &tea575x_ioctl_ops,
+	.release	= video_device_release,
+};
+
 /*
  * initialize all the tea575x chips
  */
 void snd_tea575x_init(struct snd_tea575x *tea)
 {
+	int retval;
 	unsigned int val;
+	struct video_device *tea575x_radio_inst;
 
 	val = tea->ops->read(tea);
 	if (val == 0x1ffffff || val == 0) {
-		snd_printk(KERN_ERR "Cannot find TEA575x chip\n");
+		snd_printk(KERN_ERR
+			   "tea575x-tuner: Cannot find TEA575x chip\n");
 		return;
 	}
 
-	memset(&tea->vd, 0, sizeof(tea->vd));
-	strcpy(tea->vd.name, tea->tea5759 ? "TEA5759 radio" : "TEA5757 radio");
-	tea->vd.release = snd_tea575x_release;
-	video_set_drvdata(&tea->vd, tea);
-	tea->vd.fops = &tea->fops;
 	tea->in_use = 0;
-	tea->fops.owner = tea->card->module;
-	tea->fops.open = snd_tea575x_exclusive_open;
-	tea->fops.release = snd_tea575x_exclusive_release;
-	tea->fops.ioctl = snd_tea575x_ioctl;
-	if (video_register_device(&tea->vd, VFL_TYPE_RADIO, tea->dev_nr - 1) < 0) {
-		snd_printk(KERN_ERR "unable to register tea575x tuner\n");
+	tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_10_40;
+	tea->freq = 90500 * 16;		/* 90.5Mhz default */
+
+	tea575x_radio_inst = video_device_alloc();
+	if (tea575x_radio_inst == NULL) {
+		printk(KERN_ERR "tea575x-tuner: not enough memory\n");
 		return;
 	}
-	tea->vd_registered = 1;
 
-	tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_10_40;
-	tea->freq = 90500 * 16;		/* 90.5Mhz default */
+	memcpy(tea575x_radio_inst, &tea575x_radio, sizeof(tea575x_radio));
+
+	strcpy(tea575x_radio.name, tea->tea5759 ?
+				   "TEA5759 radio" : "TEA5757 radio");
+
+	video_set_drvdata(tea575x_radio_inst, tea);
+
+	retval = video_register_device(tea575x_radio_inst,
+				       VFL_TYPE_RADIO, radio_nr);
+	if (retval) {
+		printk(KERN_ERR "tea575x-tuner: can't register video device!\n");
+		kfree(tea575x_radio_inst);
+		return;
+	}
 
 	snd_tea575x_set_freq(tea);
 
 	/* mute on init */
-	if (tea->ops->mute)
+	if (tea->ops->mute) {
 		tea->ops->mute(tea, 1);
+		tea->mute = 1;
+	}
+	tea->vd = tea575x_radio_inst;
 }
 
 void snd_tea575x_exit(struct snd_tea575x *tea)
 {
-	if (tea->vd_registered) {
-		video_unregister_device(&tea->vd);
-		tea->vd_registered = 0;
+	if (tea->vd) {
+		video_unregister_device(tea->vd);
+		tea->vd = NULL;
 	}
 }
 
diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig
index ce0aa044e274..c5c9a9218ff6 100644
--- a/sound/isa/Kconfig
+++ b/sound/isa/Kconfig
@@ -56,8 +56,8 @@ config SND_AD1848
 	  Say Y here to include support for AD1848 (Analog Devices) or
 	  CS4248 (Cirrus Logic - Crystal Semiconductors) chips.
 	  
-	  For newer chips from Cirrus Logic, use the CS4231, CS4232 or
-	  CS4236+ drivers.
+	  For newer chips from Cirrus Logic, use the CS4231 or CS4232+
+	  drivers.
 
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-ad1848.
@@ -94,6 +94,8 @@ config SND_CMI8330
 	tristate "C-Media CMI8330"
 	select SND_WSS_LIB
 	select SND_SB16_DSP
+	select SND_OPL3_LIB
+	select SND_MPU401_UART
 	help
 	  Say Y here to include support for soundcards based on the
 	  C-Media CMI8330 chip.
@@ -112,26 +114,15 @@ config SND_CS4231
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-cs4231.
 
-config SND_CS4232
-	tristate "Generic Cirrus Logic CS4232 driver"
-	select SND_OPL3_LIB
-	select SND_MPU401_UART
-	select SND_WSS_LIB
-	help
-	  Say Y here to include support for CS4232 chips from Cirrus
-	  Logic - Crystal Semiconductors.
-
-	  To compile this driver as a module, choose M here: the module
-	  will be called snd-cs4232.
-
 config SND_CS4236
-	tristate "Generic Cirrus Logic CS4236+ driver"
+	tristate "Generic Cirrus Logic CS4232/CS4236+ driver"
 	select SND_OPL3_LIB
 	select SND_MPU401_UART
 	select SND_WSS_LIB
 	help
-	  Say Y to include support for CS4235,CS4236,CS4237B,CS4238B,
-	  CS4239 chips from Cirrus Logic - Crystal Semiconductors.
+	  Say Y to include support for CS4232,CS4235,CS4236,CS4237B,
+	  CS4238B,CS4239 chips from Cirrus Logic - Crystal
+	  Semiconductors.
 
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-cs4236.
@@ -377,14 +368,17 @@ config SND_SGALAXY
 	  will be called snd-sgalaxy.
 
 config SND_SSCAPE
-	tristate "Ensoniq SoundScape PnP driver"
+	tristate "Ensoniq SoundScape driver"
 	select SND_HWDEP
 	select SND_MPU401_UART
 	select SND_WSS_LIB
 	help
-	  Say Y here to include support for Ensoniq SoundScape PnP
+	  Say Y here to include support for Ensoniq SoundScape 
 	  soundcards.
 
+	  The PCM audio is supported on SoundScape Classic, Elite, PnP
+	  and VIVO cards. The MIDI support is very experimental.
+
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-sscape.
 
@@ -411,5 +405,36 @@ config SND_WAVEFRONT_FIRMWARE_IN_KERNEL
 	  you need to install the firmware files from the
 	  alsa-firmware package.
 
+config SND_MSND_PINNACLE
+	tristate "Turtle Beach MultiSound Pinnacle/Fiji driver"
+	depends on X86 && EXPERIMENTAL
+	select FW_LOADER
+	select SND_MPU401_UART
+	select SND_PCM
+	help
+	  Say Y to include support for Turtle Beach MultiSound Pinnacle/
+	  Fiji soundcards.
+
+	  To compile this driver as a module, choose M here: the module
+	  will be called snd-msnd-pinnacle.
+
+config SND_MSND_CLASSIC
+	tristate "Support for Turtle Beach MultiSound Classic, Tahiti, Monterey"
+	depends on X86 && EXPERIMENTAL
+	select FW_LOADER
+	select SND_MPU401_UART
+	select SND_PCM
+	help
+	  Say M here if you have a Turtle Beach MultiSound Classic, Tahiti or
+	  Monterey (not for the Pinnacle or Fiji).
+
+	  See <file:Documentation/sound/oss/MultiSound> for important information
+	  about this driver.  Note that it has been discontinued, but the
+	  Voyetra Turtle Beach knowledge base entry for it is still available
+	  at <http://www.turtlebeach.com/site/kb_ftp/790.asp>.
+
+	  To compile this driver as a module, choose M here: the module
+	  will be called snd-msnd-classic.
+
 endif	# SND_ISA
 
diff --git a/sound/isa/Makefile b/sound/isa/Makefile
index 63af13d901a5..b906b9a1a81e 100644
--- a/sound/isa/Makefile
+++ b/sound/isa/Makefile
@@ -26,5 +26,5 @@ obj-$(CONFIG_SND_SC6000) += snd-sc6000.o
 obj-$(CONFIG_SND_SGALAXY) += snd-sgalaxy.o
 obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o
 
-obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ gus/ opti9xx/ \
+obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ gus/ msnd/ opti9xx/ \
 		     sb/ wavefront/ wss/
diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c
index 77524244a846..bbcbf92a8ebe 100644
--- a/sound/isa/ad1816a/ad1816a.c
+++ b/sound/isa/ad1816a/ad1816a.c
@@ -156,10 +156,12 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard
 	struct snd_card_ad1816a *acard;
 	struct snd_ad1816a *chip;
 	struct snd_opl3 *opl3;
+	struct snd_timer *timer;
 
-	if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-				 sizeof(struct snd_card_ad1816a))) == NULL)
-		return -ENOMEM;
+	error = snd_card_create(index[dev], id[dev], THIS_MODULE,
+				sizeof(struct snd_card_ad1816a), &card);
+	if (error < 0)
+		return error;
 	acard = (struct snd_card_ad1816a *)card->private_data;
 
 	if ((error = snd_card_ad1816a_pnp(dev, acard, pcard, pid))) {
@@ -194,6 +196,12 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard
 		return error;
 	}
 
+	error = snd_ad1816a_timer(chip, 0, &timer);
+	if (error < 0) {
+		snd_card_free(card);
+		return error;
+	}
+
 	if (mpu_port[dev] > 0) {
 		if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
 					mpu_port[dev], 0, mpu_irq[dev], IRQF_DISABLED,
@@ -207,11 +215,8 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard
 				    OPL3_HW_AUTO, 0, &opl3) < 0) {
 			printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx.\n", fm_port[dev], fm_port[dev] + 2);
 		} else {
-			if ((error = snd_opl3_timer_new(opl3, 1, 2)) < 0) {
-				snd_card_free(card);
-				return error;
-			}
-			if ((error = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) {
+			error = snd_opl3_hwdep_new(opl3, 0, 1, NULL);
+			if (error < 0) {
 				snd_card_free(card);
 				return error;
 			}
diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c
index 3bfca7c59baf..05aef8b97e96 100644
--- a/sound/isa/ad1816a/ad1816a_lib.c
+++ b/sound/isa/ad1816a/ad1816a_lib.c
@@ -37,7 +37,7 @@ static inline int snd_ad1816a_busy_wait(struct snd_ad1816a *chip)
 		if (inb(AD1816A_REG(AD1816A_CHIP_STATUS)) & AD1816A_READY)
 			return 0;
 
-	snd_printk("chip busy.\n");
+	snd_printk(KERN_WARNING "chip busy.\n");
 	return -EBUSY;
 }
 
@@ -196,7 +196,7 @@ static int snd_ad1816a_trigger(struct snd_ad1816a *chip, unsigned char what,
 		spin_unlock(&chip->lock);
 		break;
 	default:
-		snd_printk("invalid trigger mode 0x%x.\n", what);
+		snd_printk(KERN_WARNING "invalid trigger mode 0x%x.\n", what);
 		error = -EINVAL;
 	}
 
@@ -377,7 +377,6 @@ static struct snd_pcm_hardware snd_ad1816a_capture = {
 	.fifo_size =		0,
 };
 
-#if 0 /* not used now */
 static int snd_ad1816a_timer_close(struct snd_timer *timer)
 {
 	struct snd_ad1816a *chip = snd_timer_chip(timer);
@@ -442,8 +441,6 @@ static struct snd_timer_hardware snd_ad1816a_timer_table = {
 	.start =	snd_ad1816a_timer_start,
 	.stop =		snd_ad1816a_timer_stop,
 };
-#endif /* not used now */
-
 
 static int snd_ad1816a_playback_open(struct snd_pcm_substream *substream)
 {
@@ -568,7 +565,7 @@ static const char __devinit *snd_ad1816a_chip_id(struct snd_ad1816a *chip)
 	case AD1816A_HW_AD1815:	return "AD1815";
 	case AD1816A_HW_AD18MAX10: return "AD18max10";
 	default:
-		snd_printk("Unknown chip version %d:%d.\n",
+		snd_printk(KERN_WARNING "Unknown chip version %d:%d.\n",
 			chip->version, chip->hardware);
 		return "AD1816A - unknown";
 	}
@@ -687,7 +684,6 @@ int __devinit snd_ad1816a_pcm(struct snd_ad1816a *chip, int device, struct snd_p
 	return 0;
 }
 
-#if 0 /* not used now */
 int __devinit snd_ad1816a_timer(struct snd_ad1816a *chip, int device, struct snd_timer **rtimer)
 {
 	struct snd_timer *timer;
@@ -709,7 +705,6 @@ int __devinit snd_ad1816a_timer(struct snd_ad1816a *chip, int device, struct snd
 		*rtimer = timer;
 	return 0;
 }
-#endif /* not used now */
 
 /*
  *
diff --git a/sound/isa/ad1848/ad1848.c b/sound/isa/ad1848/ad1848.c
index 223a6c038819..4beeb6f98e0e 100644
--- a/sound/isa/ad1848/ad1848.c
+++ b/sound/isa/ad1848/ad1848.c
@@ -91,9 +91,9 @@ static int __devinit snd_ad1848_probe(struct device *dev, unsigned int n)
 	struct snd_pcm *pcm;
 	int error;
 
-	card = snd_card_new(index[n], id[n], THIS_MODULE, 0);
-	if (!card)
-		return -EINVAL;
+	error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card);
+	if (error < 0)
+		return error;
 
 	error = snd_wss_create(card, port[n], -1, irq[n], dma1[n], -1,
 			thinkpad[n] ? WSS_HW_THINKPAD : WSS_HW_DETECT,
diff --git a/sound/isa/adlib.c b/sound/isa/adlib.c
index 374b7177e111..7465ae036e0b 100644
--- a/sound/isa/adlib.c
+++ b/sound/isa/adlib.c
@@ -53,10 +53,10 @@ static int __devinit snd_adlib_probe(struct device *dev, unsigned int n)
 	struct snd_opl3 *opl3;
 	int error;
 
-	card = snd_card_new(index[n], id[n], THIS_MODULE, 0);
-	if (!card) {
+	error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card);
+	if (error < 0) {
 		dev_err(dev, "could not create card\n");
-		return -EINVAL;
+		return error;
 	}
 
 	card->private_data = request_region(port[n], 4, CRD_NAME);
diff --git a/sound/isa/als100.c b/sound/isa/als100.c
index f1ce30f379c9..5fd52e4d7079 100644
--- a/sound/isa/als100.c
+++ b/sound/isa/als100.c
@@ -163,9 +163,10 @@ static int __devinit snd_card_als100_probe(int dev,
 	struct snd_card_als100 *acard;
 	struct snd_opl3 *opl3;
 
-	if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-				 sizeof(struct snd_card_als100))) == NULL)
-		return -ENOMEM;
+	error = snd_card_create(index[dev], id[dev], THIS_MODULE,
+				sizeof(struct snd_card_als100), &card);
+	if (error < 0)
+		return error;
 	acard = card->private_data;
 
 	if ((error = snd_card_als100_pnp(dev, acard, pcard, pid))) {
diff --git a/sound/isa/azt2320.c b/sound/isa/azt2320.c
index 3e74d1a3928e..f7aa637b0d18 100644
--- a/sound/isa/azt2320.c
+++ b/sound/isa/azt2320.c
@@ -184,9 +184,10 @@ static int __devinit snd_card_azt2320_probe(int dev,
 	struct snd_wss *chip;
 	struct snd_opl3 *opl3;
 
-	if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-				 sizeof(struct snd_card_azt2320))) == NULL)
-		return -ENOMEM;
+	error = snd_card_create(index[dev], id[dev], THIS_MODULE,
+				sizeof(struct snd_card_azt2320), &card);
+	if (error < 0)
+		return error;
 	acard = (struct snd_card_azt2320 *)card->private_data;
 
 	if ((error = snd_card_azt2320_pnp(dev, acard, pcard, pid))) {
diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c
index e49aec700a55..de83608719ea 100644
--- a/sound/isa/cmi8330.c
+++ b/sound/isa/cmi8330.c
@@ -31,11 +31,11 @@
  *  To quickly load the module,
  *
  *  modprobe -a snd-cmi8330 sbport=0x220 sbirq=5 sbdma8=1
- *    sbdma16=5 wssport=0x530 wssirq=11 wssdma=0
+ *    sbdma16=5 wssport=0x530 wssirq=11 wssdma=0 fmport=0x388
  *
  *  This card has two mixers and two PCM devices.  I've cheesed it such
  *  that recording and playback can be done through the same device.
- *  The driver "magically" routes the capturing to the AD1848 codec,
+ *  The driver "magically" routes the capturing to the CMI8330 codec,
  *  and playback to the SB16 codec.  This allows for full-duplex mode
  *  to some extent.
  *  The utilities in alsa-utils are aware of both devices, so passing
@@ -51,6 +51,8 @@
 #include <linux/moduleparam.h>
 #include <sound/core.h>
 #include <sound/wss.h>
+#include <sound/opl3.h>
+#include <sound/mpu401.h>
 #include <sound/sb.h>
 #include <sound/initval.h>
 
@@ -79,6 +81,9 @@ static int sbdma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA;
 static long wssport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
 static int wssirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ;
 static int wssdma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA;
+static long fmport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+static long mpuport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+static int mpuirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ;
 
 module_param_array(index, int, NULL, 0444);
 MODULE_PARM_DESC(index, "Index value for CMI8330 soundcard.");
@@ -107,6 +112,12 @@ MODULE_PARM_DESC(wssirq, "IRQ # for CMI8330 WSS driver.");
 module_param_array(wssdma, int, NULL, 0444);
 MODULE_PARM_DESC(wssdma, "DMA for CMI8330 WSS driver.");
 
+module_param_array(fmport, long, NULL, 0444);
+MODULE_PARM_DESC(fmport, "FM port # for CMI8330 driver.");
+module_param_array(mpuport, long, NULL, 0444);
+MODULE_PARM_DESC(mpuport, "MPU-401 port # for CMI8330 driver.");
+module_param_array(mpuirq, int, NULL, 0444);
+MODULE_PARM_DESC(mpuirq, "IRQ # for CMI8330 MPU-401 port.");
 #ifdef CONFIG_PNP
 static int isa_registered;
 static int pnp_registered;
@@ -149,6 +160,7 @@ struct snd_cmi8330 {
 #ifdef CONFIG_PNP
 	struct pnp_dev *cap;
 	struct pnp_dev *play;
+	struct pnp_dev *mpu;
 #endif
 	struct snd_card *card;
 	struct snd_wss *wss;
@@ -165,7 +177,7 @@ struct snd_cmi8330 {
 #ifdef CONFIG_PNP
 
 static struct pnp_card_device_id snd_cmi8330_pnpids[] = {
-	{ .id = "CMI0001", .devs = { { "@@@0001" }, { "@X@0001" } } },
+	{ .id = "CMI0001", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } } },
 	{ .id = "" }
 };
 
@@ -219,8 +231,10 @@ WSS_SINGLE("3D Control - Switch", 0,
 		CMI8330_RMUX3D, 5, 1, 1),
 WSS_SINGLE("PC Speaker Playback Volume", 0,
 		CMI8330_OUTPUTVOL, 3, 3, 0),
-WSS_SINGLE("FM Playback Switch", 0,
-		CMI8330_RECMUX, 3, 1, 1),
+WSS_DOUBLE("FM Playback Switch", 0,
+		CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1),
+WSS_DOUBLE("FM Playback Volume", 0,
+		CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1),
 WSS_SINGLE(SNDRV_CTL_NAME_IEC958("Input ", CAPTURE, SWITCH), 0,
 		CMI8330_RMUX3D, 7, 1, 1),
 WSS_SINGLE(SNDRV_CTL_NAME_IEC958("Input ", PLAYBACK, SWITCH), 0,
@@ -323,16 +337,21 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard,
 	if (acard->play == NULL)
 		return -EBUSY;
 
+	acard->mpu = pnp_request_card_device(card, id->devs[2].id, NULL);
+	if (acard->play == NULL)
+		return -EBUSY;
+
 	pdev = acard->cap;
 
 	err = pnp_activate_dev(pdev);
 	if (err < 0) {
-		snd_printk(KERN_ERR "CMI8330/C3D (AD1848) PnP configure failure\n");
+		snd_printk(KERN_ERR "CMI8330/C3D PnP configure failure\n");
 		return -EBUSY;
 	}
 	wssport[dev] = pnp_port_start(pdev, 0);
 	wssdma[dev] = pnp_dma(pdev, 0);
 	wssirq[dev] = pnp_irq(pdev, 0);
+	fmport[dev] = pnp_port_start(pdev, 1);
 
 	/* allocate SB16 resources */
 	pdev = acard->play;
@@ -347,6 +366,17 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard,
 	sbdma16[dev] = pnp_dma(pdev, 1);
 	sbirq[dev] = pnp_irq(pdev, 0);
 
+	/* allocate MPU-401 resources */
+	pdev = acard->mpu;
+
+	err = pnp_activate_dev(pdev);
+	if (err < 0) {
+		snd_printk(KERN_ERR
+			   "CMI8330/C3D (MPU-401) PnP configure failure\n");
+		return -EBUSY;
+	}
+	mpuport[dev] = pnp_port_start(pdev, 0);
+	mpuirq[dev] = pnp_irq(pdev, 0);
 	return 0;
 }
 #endif
@@ -467,26 +497,29 @@ static int snd_cmi8330_resume(struct snd_card *card)
 
 #define PFX	"cmi8330: "
 
-static struct snd_card *snd_cmi8330_card_new(int dev)
+static int snd_cmi8330_card_new(int dev, struct snd_card **cardp)
 {
 	struct snd_card *card;
 	struct snd_cmi8330 *acard;
+	int err;
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-			    sizeof(struct snd_cmi8330));
-	if (card == NULL) {
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct snd_cmi8330), &card);
+	if (err < 0) {
 		snd_printk(KERN_ERR PFX "could not get a new card\n");
-		return NULL;
+		return err;
 	}
 	acard = card->private_data;
 	acard->card = card;
-	return card;
+	*cardp = card;
+	return 0;
 }
 
 static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev)
 {
 	struct snd_cmi8330 *acard;
 	int i, err;
+	struct snd_opl3 *opl3;
 
 	acard = card->private_data;
 	err = snd_wss_create(card, wssport[dev] + 4, -1,
@@ -494,11 +527,11 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev)
 			     wssdma[dev], -1,
 			     WSS_HW_DETECT, 0, &acard->wss);
 	if (err < 0) {
-		snd_printk(KERN_ERR PFX "(AD1848) device busy??\n");
+		snd_printk(KERN_ERR PFX "(CMI8330) device busy??\n");
 		return err;
 	}
 	if (acard->wss->hardware != WSS_HW_CMI8330) {
-		snd_printk(KERN_ERR PFX "(AD1848) not found during probe\n");
+		snd_printk(KERN_ERR PFX "(CMI8330) not found during probe\n");
 		return -ENODEV;
 	}
 
@@ -530,6 +563,27 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev)
 		snd_printk(KERN_ERR PFX "failed to create pcms\n");
 		return err;
 	}
+	if (fmport[dev] != SNDRV_AUTO_PORT) {
+		if (snd_opl3_create(card,
+				    fmport[dev], fmport[dev] + 2,
+				    OPL3_HW_AUTO, 0, &opl3) < 0) {
+			snd_printk(KERN_ERR PFX
+				   "no OPL device at 0x%lx-0x%lx ?\n",
+				   fmport[dev], fmport[dev] + 2);
+		} else {
+			err = snd_opl3_hwdep_new(opl3, 0, 1, NULL);
+			if (err < 0)
+				return err;
+		}
+	}
+
+	if (mpuport[dev] != SNDRV_AUTO_PORT) {
+		if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
+					mpuport[dev], 0, mpuirq[dev],
+					IRQF_DISABLED, NULL) < 0)
+			printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n",
+				mpuport[dev]);
+	}
 
 	strcpy(card->driver, "CMI8330/C3D");
 	strcpy(card->shortname, "C-Media CMI8330/C3D");
@@ -564,9 +618,9 @@ static int __devinit snd_cmi8330_isa_probe(struct device *pdev,
 	struct snd_card *card;
 	int err;
 
-	card = snd_cmi8330_card_new(dev);
-	if (! card)
-		return -ENOMEM;
+	err = snd_cmi8330_card_new(dev, &card);
+	if (err < 0)
+		return err;
 	snd_card_set_dev(card, pdev);
 	if ((err = snd_cmi8330_probe(card, dev)) < 0) {
 		snd_card_free(card);
@@ -628,9 +682,9 @@ static int __devinit snd_cmi8330_pnp_detect(struct pnp_card_link *pcard,
 	if (dev >= SNDRV_CARDS)
 		return -ENODEV;
 			       
-	card = snd_cmi8330_card_new(dev);
-	if (! card)
-		return -ENOMEM;
+	res = snd_cmi8330_card_new(dev, &card);
+	if (res < 0)
+		return res;
 	if ((res = snd_cmi8330_pnp(dev, card->private_data, pcard, pid)) < 0) {
 		snd_printk(KERN_ERR PFX "PnP detection failed\n");
 		snd_card_free(card);
diff --git a/sound/isa/cs423x/Makefile b/sound/isa/cs423x/Makefile
index 5870ca21ab59..6d397e8d54ac 100644
--- a/sound/isa/cs423x/Makefile
+++ b/sound/isa/cs423x/Makefile
@@ -3,13 +3,11 @@
 # Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz>
 #
 
-snd-cs4236-lib-objs := cs4236_lib.o
 snd-cs4231-objs := cs4231.o
-snd-cs4232-objs := cs4232.o
-snd-cs4236-objs := cs4236.o
+snd-cs4236-objs := cs4236.o cs4236_lib.o
 
 # Toplevel Module Dependency
 obj-$(CONFIG_SND_CS4231) += snd-cs4231.o
-obj-$(CONFIG_SND_CS4232) += snd-cs4232.o
-obj-$(CONFIG_SND_CS4236) += snd-cs4236.o snd-cs4236-lib.o
+obj-$(CONFIG_SND_CS4236) += snd-cs4236.o
+
 
diff --git a/sound/isa/cs423x/cs4231.c b/sound/isa/cs423x/cs4231.c
index f019d449e2d6..cb9153e75b82 100644
--- a/sound/isa/cs423x/cs4231.c
+++ b/sound/isa/cs423x/cs4231.c
@@ -95,9 +95,9 @@ static int __devinit snd_cs4231_probe(struct device *dev, unsigned int n)
 	struct snd_pcm *pcm;
 	int error;
 
-	card = snd_card_new(index[n], id[n], THIS_MODULE, 0);
-	if (!card)
-		return -EINVAL;
+	error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card);
+	if (error < 0)
+		return error;
 
 	error = snd_wss_create(card, port[n], -1, irq[n], dma1[n], dma2[n],
 			WSS_HW_DETECT, 0, &chip);
diff --git a/sound/isa/cs423x/cs4232.c b/sound/isa/cs423x/cs4232.c
deleted file mode 100644
index 9fad2e6c0c2c..000000000000
--- a/sound/isa/cs423x/cs4232.c
+++ /dev/null
@@ -1,2 +0,0 @@
-#define CS4232
-#include "cs4236.c"
diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c
index 019c9401663e..a076a6ce8071 100644
--- a/sound/isa/cs423x/cs4236.c
+++ b/sound/isa/cs423x/cs4236.c
@@ -33,17 +33,14 @@
 
 MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
 MODULE_LICENSE("GPL");
-#ifdef CS4232
-MODULE_DESCRIPTION("Cirrus Logic CS4232");
+MODULE_DESCRIPTION("Cirrus Logic CS4232-9");
 MODULE_SUPPORTED_DEVICE("{{Turtle Beach,TBS-2000},"
 		"{Turtle Beach,Tropez Plus},"
 		"{SIC CrystalWave 32},"
 		"{Hewlett Packard,Omnibook 5500},"
 		"{TerraTec,Maestro 32/96},"
-		"{Philips,PCA70PS}}");
-#else
-MODULE_DESCRIPTION("Cirrus Logic CS4235-9");
-MODULE_SUPPORTED_DEVICE("{{Crystal Semiconductors,CS4235},"
+		"{Philips,PCA70PS}},"
+		"{{Crystal Semiconductors,CS4235},"
 		"{Crystal Semiconductors,CS4236},"
 		"{Crystal Semiconductors,CS4237},"
 		"{Crystal Semiconductors,CS4238},"
@@ -70,15 +67,11 @@ MODULE_SUPPORTED_DEVICE("{{Crystal Semiconductors,CS4235},"
 		"{Typhoon Soundsystem,CS4236B},"
 		"{Turtle Beach,Malibu},"
 		"{Unknown,Digital PC 5000 Onboard}}");
-#endif
 
-#ifdef CS4232
-#define IDENT "CS4232"
-#define DEV_NAME "cs4232"
-#else
-#define IDENT "CS4236+"
-#define DEV_NAME "cs4236"
-#endif
+MODULE_ALIAS("snd_cs4232");
+
+#define IDENT "CS4232+"
+#define DEV_NAME "cs4232+"
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
@@ -128,9 +121,7 @@ MODULE_PARM_DESC(dma2, "DMA2 # for " IDENT " driver.");
 #ifdef CONFIG_PNP
 static int isa_registered;
 static int pnpc_registered;
-#ifdef CS4232
 static int pnp_registered;
-#endif
 #endif /* CONFIG_PNP */
 
 struct snd_card_cs4236 {
@@ -145,11 +136,10 @@ struct snd_card_cs4236 {
 
 #ifdef CONFIG_PNP
 
-#ifdef CS4232
 /*
  * PNP BIOS
  */
-static const struct pnp_device_id snd_cs4232_pnpbiosids[] = {
+static const struct pnp_device_id snd_cs423x_pnpbiosids[] = {
 	{ .id = "CSC0100" },
 	{ .id = "CSC0000" },
 	/* Guillemot Turtlebeach something appears to be cs4232 compatible
@@ -157,10 +147,8 @@ static const struct pnp_device_id snd_cs4232_pnpbiosids[] = {
 	{ .id = "GIM0100" },
 	{ .id = "" }
 };
-MODULE_DEVICE_TABLE(pnp, snd_cs4232_pnpbiosids);
-#endif /* CS4232 */
+MODULE_DEVICE_TABLE(pnp, snd_cs423x_pnpbiosids);
 
-#ifdef CS4232
 #define CS423X_ISAPNP_DRIVER	"cs4232_isapnp"
 static struct pnp_card_device_id snd_cs423x_pnpids[] = {
 	/* Philips PCA70PS */
@@ -179,12 +167,6 @@ static struct pnp_card_device_id snd_cs423x_pnpids[] = {
 	{ .id = "CSCf032", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
 	/* Netfinity 3000 on-board soundcard */
 	{ .id = "CSCe825", .devs = { { "CSC0100" }, { "CSC0110" }, { "CSC010f" } } },
-	/* --- */
-	{ .id = "" }	/* end */
-};
-#else /* CS4236 */
-#define CS423X_ISAPNP_DRIVER	"cs4236_isapnp"
-static struct pnp_card_device_id snd_cs423x_pnpids[] = {
 	/* Intel Marlin Spike Motherboard - CS4235 */
 	{ .id = "CSC0225", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } },
 	/* Intel Marlin Spike Motherboard (#2) - CS4235 */
@@ -266,7 +248,6 @@ static struct pnp_card_device_id snd_cs423x_pnpids[] = {
 	/* --- */
 	{ .id = "" }	/* end */
 };
-#endif
 
 MODULE_DEVICE_TABLE(pnp_card, snd_cs423x_pnpids);
 
@@ -323,17 +304,19 @@ static int __devinit snd_cs423x_pnp_init_mpu(int dev, struct pnp_dev *pdev)
 	return 0;
 }
 
-#ifdef CS4232
-static int __devinit snd_card_cs4232_pnp(int dev, struct snd_card_cs4236 *acard,
-					 struct pnp_dev *pdev)
+static int __devinit snd_card_cs423x_pnp(int dev, struct snd_card_cs4236 *acard,
+					 struct pnp_dev *pdev,
+					 struct pnp_dev *cdev)
 {
 	acard->wss = pdev;
 	if (snd_cs423x_pnp_init_wss(dev, acard->wss) < 0)
 		return -EBUSY;
-	cport[dev] = -1;
+	if (cdev)
+		cport[dev] = pnp_port_start(cdev, 0);
+	else
+		cport[dev] = -1;
 	return 0;
 }
-#endif
 
 static int __devinit snd_card_cs423x_pnpc(int dev, struct snd_card_cs4236 *acard,
 					  struct pnp_card_link *card,
@@ -382,16 +365,18 @@ static void snd_card_cs4236_free(struct snd_card *card)
 	release_and_free_resource(acard->res_sb_port);
 }
 
-static struct snd_card *snd_cs423x_card_new(int dev)
+static int snd_cs423x_card_new(int dev, struct snd_card **cardp)
 {
 	struct snd_card *card;
+	int err;
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-			    sizeof(struct snd_card_cs4236));
-	if (card == NULL)
-		return NULL;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct snd_card_cs4236), &card);
+	if (err < 0)
+		return err;
 	card->private_free = snd_card_cs4236_free;
-	return card;
+	*cardp = card;
+	return 0;
 }
 
 static int __devinit snd_cs423x_probe(struct snd_card *card, int dev)
@@ -409,40 +394,39 @@ static int __devinit snd_cs423x_probe(struct snd_card *card, int dev)
 			return -EBUSY;
 		}
 
-#ifdef CS4232
 	err = snd_wss_create(card, port[dev], cport[dev],
 			     irq[dev],
 			     dma1[dev], dma2[dev],
-			     WSS_HW_DETECT, 0, &chip);
-	if (err < 0)
-		return err;
-	acard->chip = chip;
-
-	err = snd_wss_pcm(chip, 0, &pcm);
-	if (err < 0)
-		return err;
-
-	err = snd_wss_mixer(chip);
+			     WSS_HW_DETECT3, 0, &chip);
 	if (err < 0)
 		return err;
-
-#else /* CS4236 */
-	err = snd_cs4236_create(card,
-				port[dev], cport[dev],
-				irq[dev], dma1[dev], dma2[dev],
-				WSS_HW_DETECT, 0, &chip);
-	if (err < 0)
-		return err;
-	acard->chip = chip;
-
-	err = snd_cs4236_pcm(chip, 0, &pcm);
-	if (err < 0)
-		return err;
-
-	err = snd_cs4236_mixer(chip);
-	if (err < 0)
-		return err;
-#endif
+	if (chip->hardware & WSS_HW_CS4236B_MASK) {
+		snd_wss_free(chip);
+		err = snd_cs4236_create(card,
+					port[dev], cport[dev],
+					irq[dev], dma1[dev], dma2[dev],
+					WSS_HW_DETECT, 0, &chip);
+		if (err < 0)
+			return err;
+		acard->chip = chip;
+
+		err = snd_cs4236_pcm(chip, 0, &pcm);
+		if (err < 0)
+			return err;
+
+		err = snd_cs4236_mixer(chip);
+		if (err < 0)
+			return err;
+	} else {
+		acard->chip = chip;
+		err = snd_wss_pcm(chip, 0, &pcm);
+		if (err < 0)
+			return err;
+
+		err = snd_wss_mixer(chip);
+		if (err < 0)
+			return err;
+	}
 	strcpy(card->driver, pcm->name);
 	strcpy(card->shortname, pcm->name);
 	sprintf(card->longname, "%s at 0x%lx, irq %i, dma %i",
@@ -512,9 +496,9 @@ static int __devinit snd_cs423x_isa_probe(struct device *pdev,
 	struct snd_card *card;
 	int err;
 
-	card = snd_cs423x_card_new(dev);
-	if (! card)
-		return -ENOMEM;
+	err = snd_cs423x_card_new(dev, &card);
+	if (err < 0)
+		return err;
 	snd_card_set_dev(card, pdev);
 	if ((err = snd_cs423x_probe(card, dev)) < 0) {
 		snd_card_free(card);
@@ -577,13 +561,14 @@ static struct isa_driver cs423x_isa_driver = {
 
 
 #ifdef CONFIG_PNP
-#ifdef CS4232
-static int __devinit snd_cs4232_pnpbios_detect(struct pnp_dev *pdev,
+static int __devinit snd_cs423x_pnpbios_detect(struct pnp_dev *pdev,
 					       const struct pnp_device_id *id)
 {
 	static int dev;
 	int err;
 	struct snd_card *card;
+	struct pnp_dev *cdev;
+	char cid[PNP_ID_LEN];
 
 	if (pnp_device_is_isapnp(pdev))
 		return -ENOENT;	/* we have another procedure - card */
@@ -594,10 +579,19 @@ static int __devinit snd_cs4232_pnpbios_detect(struct pnp_dev *pdev,
 	if (dev >= SNDRV_CARDS)
 		return -ENODEV;
 
-	card = snd_cs423x_card_new(dev);
-	if (! card)
-		return -ENOMEM;
-	if ((err = snd_card_cs4232_pnp(dev, card->private_data, pdev)) < 0) {
+	/* prepare second id */
+	strcpy(cid, pdev->id[0].id);
+	cid[5] = '1';
+	cdev = NULL;
+	list_for_each_entry(cdev, &(pdev->protocol->devices), protocol_list) {
+		if (!strcmp(cdev->id[0].id, cid))
+			break;
+	}
+	err = snd_cs423x_card_new(dev, &card);
+	if (err < 0)
+		return err;
+	err = snd_card_cs423x_pnp(dev, card->private_data, pdev, cdev);
+	if (err < 0) {
 		printk(KERN_ERR "PnP BIOS detection failed for " IDENT "\n");
 		snd_card_free(card);
 		return err;
@@ -612,35 +606,34 @@ static int __devinit snd_cs4232_pnpbios_detect(struct pnp_dev *pdev,
 	return 0;
 }
 
-static void __devexit snd_cs4232_pnp_remove(struct pnp_dev * pdev)
+static void __devexit snd_cs423x_pnp_remove(struct pnp_dev *pdev)
 {
 	snd_card_free(pnp_get_drvdata(pdev));
 	pnp_set_drvdata(pdev, NULL);
 }
 
 #ifdef CONFIG_PM
-static int snd_cs4232_pnp_suspend(struct pnp_dev *pdev, pm_message_t state)
+static int snd_cs423x_pnp_suspend(struct pnp_dev *pdev, pm_message_t state)
 {
 	return snd_cs423x_suspend(pnp_get_drvdata(pdev));
 }
 
-static int snd_cs4232_pnp_resume(struct pnp_dev *pdev)
+static int snd_cs423x_pnp_resume(struct pnp_dev *pdev)
 {
 	return snd_cs423x_resume(pnp_get_drvdata(pdev));
 }
 #endif
 
-static struct pnp_driver cs4232_pnp_driver = {
-	.name = "cs4232-pnpbios",
-	.id_table = snd_cs4232_pnpbiosids,
-	.probe = snd_cs4232_pnpbios_detect,
-	.remove = __devexit_p(snd_cs4232_pnp_remove),
+static struct pnp_driver cs423x_pnp_driver = {
+	.name = "cs423x-pnpbios",
+	.id_table = snd_cs423x_pnpbiosids,
+	.probe = snd_cs423x_pnpbios_detect,
+	.remove = __devexit_p(snd_cs423x_pnp_remove),
 #ifdef CONFIG_PM
-	.suspend	= snd_cs4232_pnp_suspend,
-	.resume		= snd_cs4232_pnp_resume,
+	.suspend	= snd_cs423x_pnp_suspend,
+	.resume		= snd_cs423x_pnp_resume,
 #endif
 };
-#endif /* CS4232 */
 
 static int __devinit snd_cs423x_pnpc_detect(struct pnp_card_link *pcard,
 					    const struct pnp_card_device_id *pid)
@@ -656,9 +649,9 @@ static int __devinit snd_cs423x_pnpc_detect(struct pnp_card_link *pcard,
 	if (dev >= SNDRV_CARDS)
 		return -ENODEV;
 
-	card = snd_cs423x_card_new(dev);
-	if (! card)
-		return -ENOMEM;
+	res = snd_cs423x_card_new(dev, &card);
+	if (res < 0)
+		return res;
 	if ((res = snd_card_cs423x_pnpc(dev, card->private_data, pcard, pid)) < 0) {
 		printk(KERN_ERR "isapnp detection failed and probing for " IDENT
 		       " is not supported\n");
@@ -714,18 +707,14 @@ static int __init alsa_card_cs423x_init(void)
 #ifdef CONFIG_PNP
 	if (!err)
 		isa_registered = 1;
-#ifdef CS4232
-	err = pnp_register_driver(&cs4232_pnp_driver);
+	err = pnp_register_driver(&cs423x_pnp_driver);
 	if (!err)
 		pnp_registered = 1;
-#endif
 	err = pnp_register_card_driver(&cs423x_pnpc_driver);
 	if (!err)
 		pnpc_registered = 1;
-#ifdef CS4232
 	if (pnp_registered)
 		err = 0;
-#endif
 	if (isa_registered)
 		err = 0;
 #endif
@@ -737,10 +726,8 @@ static void __exit alsa_card_cs423x_exit(void)
 #ifdef CONFIG_PNP
 	if (pnpc_registered)
 		pnp_unregister_card_driver(&cs423x_pnpc_driver);
-#ifdef CS4232
 	if (pnp_registered)
-		pnp_unregister_driver(&cs4232_pnp_driver);
-#endif
+		pnp_unregister_driver(&cs423x_pnp_driver);
 	if (isa_registered)
 #endif
 		isa_unregister_driver(&cs423x_isa_driver);
diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c
index 6a85fdc53b60..38835f31298b 100644
--- a/sound/isa/cs423x/cs4236_lib.c
+++ b/sound/isa/cs423x/cs4236_lib.c
@@ -88,10 +88,6 @@
 #include <sound/wss.h>
 #include <sound/asoundef.h>
 
-MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>");
-MODULE_DESCRIPTION("Routines for control of CS4235/4236B/4237B/4238B/4239 chips");
-MODULE_LICENSE("GPL");
-
 /*
  *
  */
@@ -286,7 +282,8 @@ int snd_cs4236_create(struct snd_card *card,
 	if (hardware == WSS_HW_DETECT)
 		hardware = WSS_HW_DETECT3;
 	if (cport < 0x100) {
-		snd_printk("please, specify control port for CS4236+ chips\n");
+		snd_printk(KERN_ERR "please, specify control port "
+			   "for CS4236+ chips\n");
 		return -ENODEV;
 	}
 	err = snd_wss_create(card, port, cport,
@@ -295,7 +292,8 @@ int snd_cs4236_create(struct snd_card *card,
 		return err;
 
 	if (!(chip->hardware & WSS_HW_CS4236B_MASK)) {
-	        snd_printk("CS4236+: MODE3 and extended registers not available, hardware=0x%x\n",chip->hardware);
+		snd_printk(KERN_ERR "CS4236+: MODE3 and extended registers "
+			   "not available, hardware=0x%x\n", chip->hardware);
 		snd_device_free(card, chip);
 		return -ENODEV;
 	}
@@ -303,16 +301,19 @@ int snd_cs4236_create(struct snd_card *card,
 	{
 		int idx;
 		for (idx = 0; idx < 8; idx++)
-			snd_printk("CD%i = 0x%x\n", idx, inb(chip->cport + idx));
+			snd_printk(KERN_DEBUG "CD%i = 0x%x\n",
+				   idx, inb(chip->cport + idx));
 		for (idx = 0; idx < 9; idx++)
-			snd_printk("C%i = 0x%x\n", idx, snd_cs4236_ctrl_in(chip, idx));
+			snd_printk(KERN_DEBUG "C%i = 0x%x\n",
+				   idx, snd_cs4236_ctrl_in(chip, idx));
 	}
 #endif
 	ver1 = snd_cs4236_ctrl_in(chip, 1);
 	ver2 = snd_cs4236_ext_in(chip, CS4236_VERSION);
 	snd_printdd("CS4236: [0x%lx] C1 (version) = 0x%x, ext = 0x%x\n", cport, ver1, ver2);
 	if (ver1 != ver2) {
-		snd_printk("CS4236+ chip detected, but control port 0x%lx is not valid\n", cport);
+		snd_printk(KERN_ERR "CS4236+ chip detected, but "
+			   "control port 0x%lx is not valid\n", cport);
 		snd_device_free(card, chip);
 		return -ENODEV;
 	}
@@ -883,7 +884,8 @@ static int snd_cs4236_get_iec958_switch(struct snd_kcontrol *kcontrol, struct sn
 	spin_lock_irqsave(&chip->reg_lock, flags);
 	ucontrol->value.integer.value[0] = chip->image[CS4231_ALT_FEATURE_1] & 0x02 ? 1 : 0;
 #if 0
-	printk("get valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n",
+	printk(KERN_DEBUG "get valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, "
+	       "C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n",
 			snd_wss_in(chip, CS4231_ALT_FEATURE_1),
 			snd_cs4236_ctrl_in(chip, 3),
 			snd_cs4236_ctrl_in(chip, 4),
@@ -920,7 +922,8 @@ static int snd_cs4236_put_iec958_switch(struct snd_kcontrol *kcontrol, struct sn
 	mutex_unlock(&chip->mce_mutex);
 
 #if 0
-	printk("set valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n",
+	printk(KERN_DEBUG "set valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, "
+	       "C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n",
 			snd_wss_in(chip, CS4231_ALT_FEATURE_1),
 			snd_cs4236_ctrl_in(chip, 3),
 			snd_cs4236_ctrl_in(chip, 4),
@@ -1015,23 +1018,3 @@ int snd_cs4236_mixer(struct snd_wss *chip)
 	}
 	return 0;
 }
-
-EXPORT_SYMBOL(snd_cs4236_create);
-EXPORT_SYMBOL(snd_cs4236_pcm);
-EXPORT_SYMBOL(snd_cs4236_mixer);
-
-/*
- *  INIT part
- */
-
-static int __init alsa_cs4236_init(void)
-{
-	return 0;
-}
-
-static void __exit alsa_cs4236_exit(void)
-{
-}
-
-module_init(alsa_cs4236_init)
-module_exit(alsa_cs4236_exit)
diff --git a/sound/isa/dt019x.c b/sound/isa/dt019x.c
index a0242c3b613e..80f5b1af9be8 100644
--- a/sound/isa/dt019x.c
+++ b/sound/isa/dt019x.c
@@ -150,9 +150,10 @@ static int __devinit snd_card_dt019x_probe(int dev, struct pnp_card_link *pcard,
 	struct snd_card_dt019x *acard;
 	struct snd_opl3 *opl3;
 
-	if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-				 sizeof(struct snd_card_dt019x))) == NULL)
-		return -ENOMEM;
+	error = snd_card_create(index[dev], id[dev], THIS_MODULE,
+				sizeof(struct snd_card_dt019x), &card);
+	if (error < 0)
+		return error;
 	acard = card->private_data;
 
 	snd_card_set_dev(card, &pcard->card->dev);
diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c
index b46377139cf8..442b081cafb7 100644
--- a/sound/isa/es1688/es1688.c
+++ b/sound/isa/es1688/es1688.c
@@ -49,6 +49,7 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
 static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
 static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE;	/* Enable this card */
 static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;	/* 0x220,0x240,0x260 */
+static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;	/* Usually 0x388 */
 static long mpu_port[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -1};
 static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ;	/* 5,7,9,10 */
 static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ;	/* 5,7,9,10 */
@@ -65,6 +66,8 @@ MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver.");
 module_param_array(mpu_port, long, NULL, 0444);
 MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver.");
 module_param_array(irq, int, NULL, 0444);
+module_param_array(fm_port, long, NULL, 0444);
+MODULE_PARM_DESC(fm_port, "FM port # for ES1688 driver.");
 MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver.");
 module_param_array(mpu_irq, int, NULL, 0444);
 MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver.");
@@ -122,9 +125,9 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n)
 	struct snd_pcm *pcm;
 	int error;
 
-	card = snd_card_new(index[n], id[n], THIS_MODULE, 0);
-	if (!card)
-		return -EINVAL;
+	error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card);
+	if (error < 0)
+		return error;
 
 	error = snd_es1688_legacy_create(card, dev, n, &chip);
 	if (error < 0)
@@ -143,13 +146,19 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n)
 	sprintf(card->longname, "%s at 0x%lx, irq %i, dma %i", pcm->name,
 		chip->port, chip->irq, chip->dma8);
 
-	if (snd_opl3_create(card, chip->port, chip->port + 2,
-			OPL3_HW_OPL3, 0, &opl3) < 0)
-		dev_warn(dev, "opl3 not detected at 0x%lx\n", chip->port);
-	else {
-		error =	snd_opl3_hwdep_new(opl3, 0, 1, NULL);
-		if (error < 0)
-			goto out;
+	if (fm_port[n] == SNDRV_AUTO_PORT)
+		fm_port[n] = port[n];	/* share the same port */
+
+	if (fm_port[n] > 0) {
+		if (snd_opl3_create(card, fm_port[n], fm_port[n] + 2,
+				OPL3_HW_OPL3, 0, &opl3) < 0)
+			dev_warn(dev,
+				 "opl3 not detected at 0x%lx\n", fm_port[n]);
+		else {
+			error =	snd_opl3_hwdep_new(opl3, 0, 1, NULL);
+			if (error < 0)
+				goto out;
+		}
 	}
 
 	if (mpu_irq[n] >= 0 && mpu_irq[n] != SNDRV_AUTO_IRQ &&
diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c
index 4fbb508a817f..4c6e14f87f2d 100644
--- a/sound/isa/es1688/es1688_lib.c
+++ b/sound/isa/es1688/es1688_lib.c
@@ -45,7 +45,7 @@ static int snd_es1688_dsp_command(struct snd_es1688 *chip, unsigned char val)
 			return 1;
 		}
 #ifdef CONFIG_SND_DEBUG
-	printk("snd_es1688_dsp_command: timeout (0x%x)\n", val);
+	printk(KERN_DEBUG "snd_es1688_dsp_command: timeout (0x%x)\n", val);
 #endif
 	return 0;
 }
@@ -167,13 +167,16 @@ static int snd_es1688_probe(struct snd_es1688 *chip)
 	hw = ES1688_HW_AUTO;
 	switch (chip->version & 0xfff0) {
 	case 0x4880:
-		snd_printk("[0x%lx] ESS: AudioDrive ES488 detected, but driver is in another place\n", chip->port);
+		snd_printk(KERN_ERR "[0x%lx] ESS: AudioDrive ES488 detected, "
+			   "but driver is in another place\n", chip->port);
 		return -ENODEV;
 	case 0x6880:
 		hw = (chip->version & 0x0f) >= 8 ? ES1688_HW_1688 : ES1688_HW_688;
 		break;
 	default:
-		snd_printk("[0x%lx] ESS: unknown AudioDrive chip with version 0x%x (Jazz16 soundcard?)\n", chip->port, chip->version);
+		snd_printk(KERN_ERR "[0x%lx] ESS: unknown AudioDrive chip "
+			   "with version 0x%x (Jazz16 soundcard?)\n",
+			   chip->port, chip->version);
 		return -ENODEV;
 	}
 
@@ -223,7 +226,7 @@ static int snd_es1688_init(struct snd_es1688 * chip, int enable)
 		}
 	}
 #if 0
-	snd_printk("mpu cfg = 0x%x\n", cfg);
+	snd_printk(KERN_DEBUG "mpu cfg = 0x%x\n", cfg);
 #endif
 	spin_lock_irqsave(&chip->reg_lock, flags);
 	snd_es1688_mixer_write(chip, 0x40, cfg);
@@ -237,7 +240,9 @@ static int snd_es1688_init(struct snd_es1688 * chip, int enable)
 		cfg = 0xf0;	/* enable only DMA counter interrupt */
 		irq_bits = irqs[chip->irq & 0x0f];
 		if (irq_bits < 0) {
-			snd_printk("[0x%lx] ESS: bad IRQ %d for ES1688 chip!!\n", chip->port, chip->irq);
+			snd_printk(KERN_ERR "[0x%lx] ESS: bad IRQ %d "
+				   "for ES1688 chip!!\n",
+				   chip->port, chip->irq);
 #if 0
 			irq_bits = 0;
 			cfg = 0x10;
@@ -250,7 +255,8 @@ static int snd_es1688_init(struct snd_es1688 * chip, int enable)
 		cfg = 0xf0;	/* extended mode DMA enable */
 		dma = chip->dma8;
 		if (dma > 3 || dma == 2) {
-			snd_printk("[0x%lx] ESS: bad DMA channel %d for ES1688 chip!!\n", chip->port, dma);
+			snd_printk(KERN_ERR "[0x%lx] ESS: bad DMA channel %d "
+				   "for ES1688 chip!!\n", chip->port, dma);
 #if 0
 			dma_bits = 0;
 			cfg = 0x00;	/* disable all DMA */
@@ -341,8 +347,9 @@ static int snd_es1688_trigger(struct snd_es1688 *chip, int cmd, unsigned char va
 		return -EINVAL;	/* something is wrong */
 	}
 #if 0
-	printk("trigger: val = 0x%x, value = 0x%x\n", val, value);
-	printk("trigger: pointer = 0x%x\n", snd_dma_pointer(chip->dma8, chip->dma_size));
+	printk(KERN_DEBUG "trigger: val = 0x%x, value = 0x%x\n", val, value);
+	printk(KERN_DEBUG "trigger: pointer = 0x%x\n",
+	       snd_dma_pointer(chip->dma8, chip->dma_size));
 #endif
 	snd_es1688_write(chip, 0xb8, (val & 0xf0) | value);
 	spin_unlock(&chip->reg_lock);
diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c
index 90498e4ca260..8cfbff73a835 100644
--- a/sound/isa/es18xx.c
+++ b/sound/isa/es18xx.c
@@ -2125,10 +2125,10 @@ static int __devinit snd_audiodrive_pnpc(int dev, struct snd_audiodrive *acard,
 #define is_isapnp_selected(dev)		0
 #endif
 
-static struct snd_card *snd_es18xx_card_new(int dev)
+static int snd_es18xx_card_new(int dev, struct snd_card **cardp)
 {
-	return snd_card_new(index[dev], id[dev], THIS_MODULE,
-			    sizeof(struct snd_audiodrive));
+	return snd_card_create(index[dev], id[dev], THIS_MODULE,
+			       sizeof(struct snd_audiodrive), cardp);
 }
 
 static int __devinit snd_audiodrive_probe(struct snd_card *card, int dev)
@@ -2197,9 +2197,9 @@ static int __devinit snd_es18xx_isa_probe1(int dev, struct device *devptr)
 	struct snd_card *card;
 	int err;
 
-	card = snd_es18xx_card_new(dev);
-	if (! card)
-		return -ENOMEM;
+	err = snd_es18xx_card_new(dev, &card);
+	if (err < 0)
+		return err;
 	snd_card_set_dev(card, devptr);
 	if ((err = snd_audiodrive_probe(card, dev)) < 0) {
 		snd_card_free(card);
@@ -2303,9 +2303,9 @@ static int __devinit snd_audiodrive_pnp_detect(struct pnp_dev *pdev,
 	if (dev >= SNDRV_CARDS)
 		return -ENODEV;
 
-	card = snd_es18xx_card_new(dev);
-	if (! card)
-		return -ENOMEM;
+	err = snd_es18xx_card_new(dev, &card);
+	if (err < 0)
+		return err;
 	if ((err = snd_audiodrive_pnp(dev, card->private_data, pdev)) < 0) {
 		snd_card_free(card);
 		return err;
@@ -2362,9 +2362,9 @@ static int __devinit snd_audiodrive_pnpc_detect(struct pnp_card_link *pcard,
 	if (dev >= SNDRV_CARDS)
 		return -ENODEV;
 
-	card = snd_es18xx_card_new(dev);
-	if (! card)
-		return -ENOMEM;
+	res = snd_es18xx_card_new(dev, &card);
+	if (res < 0)
+		return res;
 
 	if ((res = snd_audiodrive_pnpc(dev, card->private_data, pcard, pid)) < 0) {
 		snd_card_free(card);
diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c
index f45f6116c77a..36c27c832360 100644
--- a/sound/isa/gus/gus_dma.c
+++ b/sound/isa/gus/gus_dma.c
@@ -45,7 +45,8 @@ static void snd_gf1_dma_program(struct snd_gus_card * gus,
 	unsigned char dma_cmd;
 	unsigned int address_high;
 
-	// snd_printk("dma_transfer: addr=0x%x, buf=0x%lx, count=0x%x\n", addr, (long) buf, count);
+	snd_printdd("dma_transfer: addr=0x%x, buf=0x%lx, count=0x%x\n",
+		    addr, buf_addr, count);
 
 	if (gus->gf1.dma1 > 3) {
 		if (gus->gf1.enh_mode) {
@@ -77,7 +78,8 @@ static void snd_gf1_dma_program(struct snd_gus_card * gus,
 	snd_gf1_dma_ack(gus);
 	snd_dma_program(gus->gf1.dma1, buf_addr, count, dma_cmd & SNDRV_GF1_DMA_READ ? DMA_MODE_READ : DMA_MODE_WRITE);
 #if 0
-	snd_printk("address = 0x%x, count = 0x%x, dma_cmd = 0x%x\n", address << 1, count, dma_cmd);
+	snd_printk(KERN_DEBUG "address = 0x%x, count = 0x%x, dma_cmd = 0x%x\n",
+		   address << 1, count, dma_cmd);
 #endif
 	spin_lock_irqsave(&gus->reg_lock, flags);
 	if (gus->gf1.enh_mode) {
@@ -142,7 +144,9 @@ static void snd_gf1_dma_interrupt(struct snd_gus_card * gus)
 	snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd);
 	kfree(block);
 #if 0
-	printk("program dma (IRQ) - addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", addr, (long) buffer, count, cmd);
+	snd_printd(KERN_DEBUG "program dma (IRQ) - "
+		   "addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n",
+		   block->addr, block->buf_addr, block->count, block->cmd);
 #endif
 }
 
@@ -203,13 +207,16 @@ int snd_gf1_dma_transfer_block(struct snd_gus_card * gus,
 	}
 	*block = *__block;
 	block->next = NULL;
-#if 0
-	printk("addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", block->addr, (long) block->buffer, block->count, block->cmd);
-#endif
-#if 0
-	printk("gus->gf1.dma_data_pcm_last = 0x%lx\n", (long)gus->gf1.dma_data_pcm_last);
-	printk("gus->gf1.dma_data_pcm = 0x%lx\n", (long)gus->gf1.dma_data_pcm);
-#endif
+
+	snd_printdd("addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n",
+		    block->addr, (long) block->buffer, block->count,
+		    block->cmd);
+
+	snd_printdd("gus->gf1.dma_data_pcm_last = 0x%lx\n",
+		    (long)gus->gf1.dma_data_pcm_last);
+	snd_printdd("gus->gf1.dma_data_pcm = 0x%lx\n",
+		    (long)gus->gf1.dma_data_pcm);
+
 	spin_lock_irqsave(&gus->dma_lock, flags);
 	if (synth) {
 		if (gus->gf1.dma_data_synth_last) {
diff --git a/sound/isa/gus/gus_irq.c b/sound/isa/gus/gus_irq.c
index 041894ddd014..2055aff71b50 100644
--- a/sound/isa/gus/gus_irq.c
+++ b/sound/isa/gus/gus_irq.c
@@ -41,7 +41,7 @@ __again:
 	if (status == 0)
 		return IRQ_RETVAL(handled);
 	handled = 1;
-	// snd_printk("IRQ: status = 0x%x\n", status);
+	/* snd_printk(KERN_DEBUG "IRQ: status = 0x%x\n", status); */
 	if (status & 0x02) {
 		STAT_ADD(gus->gf1.interrupt_stat_midi_in);
 		if (gus->gf1.interrupt_handler_midi_in)
@@ -65,7 +65,9 @@ __again:
 				continue;	/* multi request */
 			already |= _current_;	/* mark request */
 #if 0
-			printk("voice = %i, voice_status = 0x%x, voice_verify = %i\n", voice, voice_status, inb(GUSP(gus, GF1PAGE)));
+			printk(KERN_DEBUG "voice = %i, voice_status = 0x%x, "
+			       "voice_verify = %i\n",
+			       voice, voice_status, inb(GUSP(gus, GF1PAGE)));
 #endif
 			pvoice = &gus->gf1.voices[voice]; 
 			if (pvoice->use) {
diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c
index 38510aeb21c6..edb11eefdfe3 100644
--- a/sound/isa/gus/gus_pcm.c
+++ b/sound/isa/gus/gus_pcm.c
@@ -82,7 +82,10 @@ static int snd_gf1_pcm_block_change(struct snd_pcm_substream *substream,
 
 	count += offset & 31;
 	offset &= ~31;
-	// snd_printk("block change - offset = 0x%x, count = 0x%x\n", offset, count);
+	/*
+	snd_printk(KERN_DEBUG "block change - offset = 0x%x, count = 0x%x\n",
+		   offset, count);
+	*/
 	memset(&block, 0, sizeof(block));
 	block.cmd = SNDRV_GF1_DMA_IRQ;
 	if (snd_pcm_format_unsigned(runtime->format))
@@ -135,7 +138,11 @@ static void snd_gf1_pcm_trigger_up(struct snd_pcm_substream *substream)
 		curr = begin + (pcmp->bpos * pcmp->block_size) / runtime->channels;
 		end = curr + (pcmp->block_size / runtime->channels);
 		end -= snd_pcm_format_width(runtime->format) == 16 ? 2 : 1;
-		// snd_printk("init: curr=0x%x, begin=0x%x, end=0x%x, ctrl=0x%x, ramp=0x%x, rate=0x%x\n", curr, begin, end, voice_ctrl, ramp_ctrl, rate);
+		/*
+		snd_printk(KERN_DEBUG "init: curr=0x%x, begin=0x%x, end=0x%x, "
+			   "ctrl=0x%x, ramp=0x%x, rate=0x%x\n",
+			   curr, begin, end, voice_ctrl, ramp_ctrl, rate);
+		*/
 		pan = runtime->channels == 2 ? (!voice ? 1 : 14) : 8;
 		vol = !voice ? gus->gf1.pcm_volume_level_left : gus->gf1.pcm_volume_level_right;
 		spin_lock_irqsave(&gus->reg_lock, flags);
@@ -205,9 +212,11 @@ static void snd_gf1_pcm_interrupt_wave(struct snd_gus_card * gus,
 	ramp_ctrl = (snd_gf1_read8(gus, SNDRV_GF1_VB_VOLUME_CONTROL) & ~0xa4) | 0x03;
 #if 0
 	snd_gf1_select_voice(gus, pvoice->number);
-	printk("position = 0x%x\n", (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4));
+	printk(KERN_DEBUG "position = 0x%x\n",
+	       (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4));
 	snd_gf1_select_voice(gus, pcmp->pvoices[1]->number);
-	printk("position = 0x%x\n", (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4));
+	printk(KERN_DEBUG "position = 0x%x\n",
+	       (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4));
 	snd_gf1_select_voice(gus, pvoice->number);
 #endif
 	pcmp->bpos++;
@@ -299,7 +308,11 @@ static int snd_gf1_pcm_poke_block(struct snd_gus_card *gus, unsigned char *buf,
 	unsigned int len;
 	unsigned long flags;
 
-	// printk("poke block; buf = 0x%x, pos = %i, count = %i, port = 0x%x\n", (int)buf, pos, count, gus->gf1.port);
+	/*
+	printk(KERN_DEBUG
+	       "poke block; buf = 0x%x, pos = %i, count = %i, port = 0x%x\n",
+	       (int)buf, pos, count, gus->gf1.port);
+	*/
 	while (count > 0) {
 		len = count;
 		if (len > 512)		/* limit, to allow IRQ */
@@ -680,7 +693,8 @@ static int snd_gf1_pcm_playback_open(struct snd_pcm_substream *substream)
 	runtime->private_free = snd_gf1_pcm_playback_free;
 
 #if 0
-	printk("playback.buffer = 0x%lx, gf1.pcm_buffer = 0x%lx\n", (long) pcm->playback.buffer, (long) gus->gf1.pcm_buffer);
+	printk(KERN_DEBUG "playback.buffer = 0x%lx, gf1.pcm_buffer = 0x%lx\n",
+	       (long) pcm->playback.buffer, (long) gus->gf1.pcm_buffer);
 #endif
 	if ((err = snd_gf1_dma_init(gus)) < 0)
 		return err;
diff --git a/sound/isa/gus/gus_uart.c b/sound/isa/gus/gus_uart.c
index f0af3f79b08b..21cc42e4c4be 100644
--- a/sound/isa/gus/gus_uart.c
+++ b/sound/isa/gus/gus_uart.c
@@ -129,8 +129,14 @@ static int snd_gf1_uart_input_open(struct snd_rawmidi_substream *substream)
 	}
 	spin_unlock_irqrestore(&gus->uart_cmd_lock, flags);
 #if 0
-	snd_printk("read init - enable = %i, cmd = 0x%x, stat = 0x%x\n", gus->uart_enable, gus->gf1.uart_cmd, snd_gf1_uart_stat(gus));
-	snd_printk("[0x%x] reg (ctrl/status) = 0x%x, reg (data) = 0x%x (page = 0x%x)\n", gus->gf1.port + 0x100, inb(gus->gf1.port + 0x100), inb(gus->gf1.port + 0x101), inb(gus->gf1.port + 0x102));
+	snd_printk(KERN_DEBUG
+		   "read init - enable = %i, cmd = 0x%x, stat = 0x%x\n",
+		   gus->uart_enable, gus->gf1.uart_cmd, snd_gf1_uart_stat(gus));
+	snd_printk(KERN_DEBUG
+		   "[0x%x] reg (ctrl/status) = 0x%x, reg (data) = 0x%x "
+		   "(page = 0x%x)\n",
+		   gus->gf1.port + 0x100, inb(gus->gf1.port + 0x100),
+		   inb(gus->gf1.port + 0x101), inb(gus->gf1.port + 0x102));
 #endif
 	return 0;
 }
diff --git a/sound/isa/gus/gusclassic.c b/sound/isa/gus/gusclassic.c
index 426532a4d730..086b8f0e0f94 100644
--- a/sound/isa/gus/gusclassic.c
+++ b/sound/isa/gus/gusclassic.c
@@ -148,9 +148,9 @@ static int __devinit snd_gusclassic_probe(struct device *dev, unsigned int n)
 	struct snd_gus_card *gus;
 	int error;
 
-	card = snd_card_new(index[n], id[n], THIS_MODULE, 0);
-	if (!card)
-		return -EINVAL;
+	error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card);
+	if (error < 0)
+		return error;
 
 	if (pcm_channels[n] < 2)
 		pcm_channels[n] = 2;
diff --git a/sound/isa/gus/gusextreme.c b/sound/isa/gus/gusextreme.c
index 7ad4c3b41a84..180a8dea6bd9 100644
--- a/sound/isa/gus/gusextreme.c
+++ b/sound/isa/gus/gusextreme.c
@@ -241,9 +241,9 @@ static int __devinit snd_gusextreme_probe(struct device *dev, unsigned int n)
 	struct snd_opl3 *opl3;
 	int error;
 
-	card = snd_card_new(index[n], id[n], THIS_MODULE, 0);
-	if (!card)
-		return -EINVAL;
+	error = snd_card_create(index[n], id[n], THIS_MODULE, 0, &card);
+	if (error < 0)
+		return error;
 
 	if (mpu_port[n] == SNDRV_AUTO_PORT)
 		mpu_port[n] = 0;
diff --git a/sound/isa/gus/gusmax.c b/sound/isa/gus/gusmax.c
index f94c1976e632..f26eac8d8110 100644
--- a/sound/isa/gus/gusmax.c
+++ b/sound/isa/gus/gusmax.c
@@ -214,10 +214,10 @@ static int __devinit snd_gusmax_probe(struct device *pdev, unsigned int dev)
 	struct snd_wss *wss;
 	struct snd_gusmax *maxcard;
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-			    sizeof(struct snd_gusmax));
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct snd_gusmax), &card);
+	if (err < 0)
+		return err;
 	card->private_free = snd_gusmax_free;
 	maxcard = (struct snd_gusmax *)card->private_data;
 	maxcard->card = card;
diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c
index 5faecfb602d3..534a6eced2b8 100644
--- a/sound/isa/gus/interwave.c
+++ b/sound/isa/gus/interwave.c
@@ -170,7 +170,7 @@ static void snd_interwave_i2c_setlines(struct snd_i2c_bus *bus, int ctrl, int da
 	unsigned long port = bus->private_value;
 
 #if 0
-	printk("i2c_setlines - 0x%lx <- %i,%i\n", port, ctrl, data);
+	printk(KERN_DEBUG "i2c_setlines - 0x%lx <- %i,%i\n", port, ctrl, data);
 #endif
 	outb((data << 1) | ctrl, port);
 	udelay(10);
@@ -183,7 +183,7 @@ static int snd_interwave_i2c_getclockline(struct snd_i2c_bus *bus)
 
 	res = inb(port) & 1;
 #if 0
-	printk("i2c_getclockline - 0x%lx -> %i\n", port, res);
+	printk(KERN_DEBUG "i2c_getclockline - 0x%lx -> %i\n", port, res);
 #endif
 	return res;
 }
@@ -197,7 +197,7 @@ static int snd_interwave_i2c_getdataline(struct snd_i2c_bus *bus, int ack)
 		udelay(10);
 	res = (inb(port) & 2) >> 1;
 #if 0
-	printk("i2c_getdataline - 0x%lx -> %i\n", port, res);
+	printk(KERN_DEBUG "i2c_getdataline - 0x%lx -> %i\n", port, res);
 #endif
 	return res;
 }
@@ -342,7 +342,8 @@ static void __devinit snd_interwave_bank_sizes(struct snd_gus_card * gus, int *s
 			snd_gf1_poke(gus, local, d);
 			snd_gf1_poke(gus, local + 1, d + 1);
 #if 0
-			printk("d = 0x%x, local = 0x%x, local + 1 = 0x%x, idx << 22 = 0x%x\n",
+			printk(KERN_DEBUG "d = 0x%x, local = 0x%x, "
+			       "local + 1 = 0x%x, idx << 22 = 0x%x\n",
 			       d,
 			       snd_gf1_peek(gus, local),
 			       snd_gf1_peek(gus, local + 1),
@@ -356,7 +357,8 @@ static void __devinit snd_interwave_bank_sizes(struct snd_gus_card * gus, int *s
 		}
 	}
 #if 0
-	printk("sizes: %i %i %i %i\n", sizes[0], sizes[1], sizes[2], sizes[3]);
+	printk(KERN_DEBUG "sizes: %i %i %i %i\n",
+	       sizes[0], sizes[1], sizes[2], sizes[3]);
 #endif
 }
 
@@ -410,12 +412,12 @@ static void __devinit snd_interwave_detect_memory(struct snd_gus_card * gus)
 		lmct = (psizes[3] << 24) | (psizes[2] << 16) |
 		    (psizes[1] << 8) | psizes[0];
 #if 0
-		printk("lmct = 0x%08x\n", lmct);
+		printk(KERN_DEBUG "lmct = 0x%08x\n", lmct);
 #endif
 		for (i = 0; i < ARRAY_SIZE(lmc); i++)
 			if (lmct == lmc[i]) {
 #if 0
-				printk("found !!! %i\n", i);
+				printk(KERN_DEBUG "found !!! %i\n", i);
 #endif
 				snd_gf1_write16(gus, SNDRV_GF1_GW_MEMORY_CONFIG, (snd_gf1_look16(gus, SNDRV_GF1_GW_MEMORY_CONFIG) & 0xfff0) | i);
 				snd_interwave_bank_sizes(gus, psizes);
@@ -626,20 +628,22 @@ static void snd_interwave_free(struct snd_card *card)
 		free_irq(iwcard->irq, (void *)iwcard);
 }
 
-static struct snd_card *snd_interwave_card_new(int dev)
+static int snd_interwave_card_new(int dev, struct snd_card **cardp)
 {
 	struct snd_card *card;
 	struct snd_interwave *iwcard;
+	int err;
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-			    sizeof(struct snd_interwave));
-	if (card == NULL)
-		return NULL;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct snd_interwave), &card);
+	if (err < 0)
+		return err;
 	iwcard = card->private_data;
 	iwcard->card = card;
 	iwcard->irq = -1;
 	card->private_free = snd_interwave_free;
-	return card;
+	*cardp = card;
+	return 0;
 }
 
 static int __devinit snd_interwave_probe(struct snd_card *card, int dev)
@@ -778,9 +782,9 @@ static int __devinit snd_interwave_isa_probe1(int dev, struct device *devptr)
 	struct snd_card *card;
 	int err;
 
-	card = snd_interwave_card_new(dev);
-	if (! card)
-		return -ENOMEM;
+	err = snd_interwave_card_new(dev, &card);
+	if (err < 0)
+		return err;
 
 	snd_card_set_dev(card, devptr);
 	if ((err = snd_interwave_probe(card, dev)) < 0) {
@@ -876,9 +880,9 @@ static int __devinit snd_interwave_pnp_detect(struct pnp_card_link *pcard,
 	if (dev >= SNDRV_CARDS)
 		return -ENODEV;
 				
-	card = snd_interwave_card_new(dev);
-	if (! card)
-		return -ENOMEM;
+	res = snd_interwave_card_new(dev, &card);
+	if (res < 0)
+		return res;
 
 	if ((res = snd_interwave_pnp(dev, card->private_data, pcard, pid)) < 0) {
 		snd_card_free(card);
diff --git a/sound/isa/msnd/Makefile b/sound/isa/msnd/Makefile
new file mode 100644
index 000000000000..2171c0aa2f62
--- /dev/null
+++ b/sound/isa/msnd/Makefile
@@ -0,0 +1,9 @@
+
+snd-msnd-lib-objs := msnd.o msnd_midi.o msnd_pinnacle_mixer.o
+snd-msnd-pinnacle-objs := msnd_pinnacle.o
+snd-msnd-classic-objs := msnd_classic.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_MSND_PINNACLE) += snd-msnd-pinnacle.o snd-msnd-lib.o
+obj-$(CONFIG_SND_MSND_CLASSIC) += snd-msnd-classic.o snd-msnd-lib.o
+
diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c
new file mode 100644
index 000000000000..906454413ed2
--- /dev/null
+++ b/sound/isa/msnd/msnd.c
@@ -0,0 +1,705 @@
+/*********************************************************************
+ *
+ * 2002/06/30 Karsten Wiese:
+ *	removed kernel-version dependencies.
+ *	ripped from linux kernel 2.4.18 (OSS Implementation) by me.
+ *	In the OSS Version, this file is compiled to a separate MODULE,
+ *	that is used by the pinnacle and the classic driver.
+ *	since there is no classic driver for alsa yet (i dont have a classic
+ *	& writing one blindfold is difficult) this file's object is statically
+ *	linked into the pinnacle-driver-module for now.	look for the string
+ *		"uncomment this to make this a module again"
+ *	to do guess what.
+ *
+ * the following is a copy of the 2.4.18 OSS FREE file-heading comment:
+ *
+ * msnd.c - Driver Base
+ *
+ * Turtle Beach MultiSound Sound Card Driver for Linux
+ *
+ * Copyright (C) 1998 Andrew Veliath
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ ********************************************************************/
+
+#include <linux/kernel.h>
+#include <linux/types.h>
+#include <linux/interrupt.h>
+#include <linux/io.h>
+#include <linux/fs.h>
+#include <linux/delay.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+
+#include "msnd.h"
+
+#define LOGNAME			"msnd"
+
+
+void snd_msnd_init_queue(void *base, int start, int size)
+{
+	writew(PCTODSP_BASED(start), base + JQS_wStart);
+	writew(PCTODSP_OFFSET(size) - 1, base + JQS_wSize);
+	writew(0, base + JQS_wHead);
+	writew(0, base + JQS_wTail);
+}
+EXPORT_SYMBOL(snd_msnd_init_queue);
+
+static int snd_msnd_wait_TXDE(struct snd_msnd *dev)
+{
+	unsigned int io = dev->io;
+	int timeout = 1000;
+
+	while (timeout-- > 0)
+		if (inb(io + HP_ISR) & HPISR_TXDE)
+			return 0;
+
+	return -EIO;
+}
+
+static int snd_msnd_wait_HC0(struct snd_msnd *dev)
+{
+	unsigned int io = dev->io;
+	int timeout = 1000;
+
+	while (timeout-- > 0)
+		if (!(inb(io + HP_CVR) & HPCVR_HC))
+			return 0;
+
+	return -EIO;
+}
+
+int snd_msnd_send_dsp_cmd(struct snd_msnd *dev, u8 cmd)
+{
+	unsigned long flags;
+
+	spin_lock_irqsave(&dev->lock, flags);
+	if (snd_msnd_wait_HC0(dev) == 0) {
+		outb(cmd, dev->io + HP_CVR);
+		spin_unlock_irqrestore(&dev->lock, flags);
+		return 0;
+	}
+	spin_unlock_irqrestore(&dev->lock, flags);
+
+	snd_printd(KERN_ERR LOGNAME ": Send DSP command timeout\n");
+
+	return -EIO;
+}
+EXPORT_SYMBOL(snd_msnd_send_dsp_cmd);
+
+int snd_msnd_send_word(struct snd_msnd *dev, unsigned char high,
+		   unsigned char mid, unsigned char low)
+{
+	unsigned int io = dev->io;
+
+	if (snd_msnd_wait_TXDE(dev) == 0) {
+		outb(high, io + HP_TXH);
+		outb(mid, io + HP_TXM);
+		outb(low, io + HP_TXL);
+		return 0;
+	}
+
+	snd_printd(KERN_ERR LOGNAME ": Send host word timeout\n");
+
+	return -EIO;
+}
+EXPORT_SYMBOL(snd_msnd_send_word);
+
+int snd_msnd_upload_host(struct snd_msnd *dev, const u8 *bin, int len)
+{
+	int i;
+
+	if (len % 3 != 0) {
+		snd_printk(KERN_ERR LOGNAME
+			   ": Upload host data not multiple of 3!\n");
+		return -EINVAL;
+	}
+
+	for (i = 0; i < len; i += 3)
+		if (snd_msnd_send_word(dev, bin[i], bin[i + 1], bin[i + 2]))
+			return -EIO;
+
+	inb(dev->io + HP_RXL);
+	inb(dev->io + HP_CVR);
+
+	return 0;
+}
+EXPORT_SYMBOL(snd_msnd_upload_host);
+
+int snd_msnd_enable_irq(struct snd_msnd *dev)
+{
+	unsigned long flags;
+
+	if (dev->irq_ref++)
+		return 0;
+
+	snd_printdd(LOGNAME ": Enabling IRQ\n");
+
+	spin_lock_irqsave(&dev->lock, flags);
+	if (snd_msnd_wait_TXDE(dev) == 0) {
+		outb(inb(dev->io + HP_ICR) | HPICR_TREQ, dev->io + HP_ICR);
+		if (dev->type == msndClassic)
+			outb(dev->irqid, dev->io + HP_IRQM);
+
+		outb(inb(dev->io + HP_ICR) & ~HPICR_TREQ, dev->io + HP_ICR);
+		outb(inb(dev->io + HP_ICR) | HPICR_RREQ, dev->io + HP_ICR);
+		enable_irq(dev->irq);
+		snd_msnd_init_queue(dev->DSPQ, dev->dspq_data_buff,
+				    dev->dspq_buff_size);
+		spin_unlock_irqrestore(&dev->lock, flags);
+		return 0;
+	}
+	spin_unlock_irqrestore(&dev->lock, flags);
+
+	snd_printd(KERN_ERR LOGNAME ": Enable IRQ failed\n");
+
+	return -EIO;
+}
+EXPORT_SYMBOL(snd_msnd_enable_irq);
+
+int snd_msnd_disable_irq(struct snd_msnd *dev)
+{
+	unsigned long flags;
+
+	if (--dev->irq_ref > 0)
+		return 0;
+
+	if (dev->irq_ref < 0)
+		snd_printd(KERN_WARNING LOGNAME ": IRQ ref count is %d\n",
+			   dev->irq_ref);
+
+	snd_printdd(LOGNAME ": Disabling IRQ\n");
+
+	spin_lock_irqsave(&dev->lock, flags);
+	if (snd_msnd_wait_TXDE(dev) == 0) {
+		outb(inb(dev->io + HP_ICR) & ~HPICR_RREQ, dev->io + HP_ICR);
+		if (dev->type == msndClassic)
+			outb(HPIRQ_NONE, dev->io + HP_IRQM);
+		disable_irq(dev->irq);
+		spin_unlock_irqrestore(&dev->lock, flags);
+		return 0;
+	}
+	spin_unlock_irqrestore(&dev->lock, flags);
+
+	snd_printd(KERN_ERR LOGNAME ": Disable IRQ failed\n");
+
+	return -EIO;
+}
+EXPORT_SYMBOL(snd_msnd_disable_irq);
+
+static inline long get_play_delay_jiffies(struct snd_msnd *chip, long size)
+{
+	long tmp = (size * HZ * chip->play_sample_size) / 8;
+	return tmp / (chip->play_sample_rate * chip->play_channels);
+}
+
+static void snd_msnd_dsp_write_flush(struct snd_msnd *chip)
+{
+	if (!(chip->mode & FMODE_WRITE) || !test_bit(F_WRITING, &chip->flags))
+		return;
+	set_bit(F_WRITEFLUSH, &chip->flags);
+/*	interruptible_sleep_on_timeout(
+		&chip->writeflush,
+		get_play_delay_jiffies(&chip, chip->DAPF.len));*/
+	clear_bit(F_WRITEFLUSH, &chip->flags);
+	if (!signal_pending(current))
+		schedule_timeout_interruptible(
+			get_play_delay_jiffies(chip, chip->play_period_bytes));
+	clear_bit(F_WRITING, &chip->flags);
+}
+
+void snd_msnd_dsp_halt(struct snd_msnd *chip, struct file *file)
+{
+	if ((file ? file->f_mode : chip->mode) & FMODE_READ) {
+		clear_bit(F_READING, &chip->flags);
+		snd_msnd_send_dsp_cmd(chip, HDEX_RECORD_STOP);
+		snd_msnd_disable_irq(chip);
+		if (file) {
+			snd_printd(KERN_INFO LOGNAME
+				   ": Stopping read for %p\n", file);
+			chip->mode &= ~FMODE_READ;
+		}
+		clear_bit(F_AUDIO_READ_INUSE, &chip->flags);
+	}
+	if ((file ? file->f_mode : chip->mode) & FMODE_WRITE) {
+		if (test_bit(F_WRITING, &chip->flags)) {
+			snd_msnd_dsp_write_flush(chip);
+			snd_msnd_send_dsp_cmd(chip, HDEX_PLAY_STOP);
+		}
+		snd_msnd_disable_irq(chip);
+		if (file) {
+			snd_printd(KERN_INFO
+				   LOGNAME ": Stopping write for %p\n", file);
+			chip->mode &= ~FMODE_WRITE;
+		}
+		clear_bit(F_AUDIO_WRITE_INUSE, &chip->flags);
+	}
+}
+EXPORT_SYMBOL(snd_msnd_dsp_halt);
+
+
+int snd_msnd_DARQ(struct snd_msnd *chip, int bank)
+{
+	int /*size, n,*/ timeout = 3;
+	u16 wTmp;
+	/* void *DAQD; */
+
+	/* Increment the tail and check for queue wrap */
+	wTmp = readw(chip->DARQ + JQS_wTail) + PCTODSP_OFFSET(DAQDS__size);
+	if (wTmp > readw(chip->DARQ + JQS_wSize))
+		wTmp = 0;
+	while (wTmp == readw(chip->DARQ + JQS_wHead) && timeout--)
+		udelay(1);
+
+	if (chip->capturePeriods == 2) {
+		void *pDAQ = chip->mappedbase + DARQ_DATA_BUFF +
+			     bank * DAQDS__size + DAQDS_wStart;
+		unsigned short offset = 0x3000 + chip->capturePeriodBytes;
+
+		if (readw(pDAQ) != PCTODSP_BASED(0x3000))
+			offset = 0x3000;
+		writew(PCTODSP_BASED(offset), pDAQ);
+	}
+
+	writew(wTmp, chip->DARQ + JQS_wTail);
+
+#if 0
+	/* Get our digital audio queue struct */
+	DAQD = bank * DAQDS__size + chip->mappedbase + DARQ_DATA_BUFF;
+
+	/* Get length of data */
+	size = readw(DAQD + DAQDS_wSize);
+
+	/* Read data from the head (unprotected bank 1 access okay
+	   since this is only called inside an interrupt) */
+	outb(HPBLKSEL_1, chip->io + HP_BLKS);
+	n = msnd_fifo_write(&chip->DARF,
+			    (char *)(chip->base + bank * DAR_BUFF_SIZE),
+			    size, 0);
+	if (n <= 0) {
+		outb(HPBLKSEL_0, chip->io + HP_BLKS);
+		return n;
+	}
+	outb(HPBLKSEL_0, chip->io + HP_BLKS);
+#endif
+
+	return 1;
+}
+EXPORT_SYMBOL(snd_msnd_DARQ);
+
+int snd_msnd_DAPQ(struct snd_msnd *chip, int start)
+{
+	u16	DAPQ_tail;
+	int	protect = start, nbanks = 0;
+	void	*DAQD;
+	static int play_banks_submitted;
+	/* unsigned long flags;
+	spin_lock_irqsave(&chip->lock, flags); not necessary */
+
+	DAPQ_tail = readw(chip->DAPQ + JQS_wTail);
+	while (DAPQ_tail != readw(chip->DAPQ + JQS_wHead) || start) {
+		int bank_num = DAPQ_tail / PCTODSP_OFFSET(DAQDS__size);
+
+		if (start) {
+			start = 0;
+			play_banks_submitted = 0;
+		}
+
+		/* Get our digital audio queue struct */
+		DAQD = bank_num * DAQDS__size + chip->mappedbase +
+			DAPQ_DATA_BUFF;
+
+		/* Write size of this bank */
+		writew(chip->play_period_bytes, DAQD + DAQDS_wSize);
+		if (play_banks_submitted < 3)
+			++play_banks_submitted;
+		else if (chip->playPeriods == 2) {
+			unsigned short offset = chip->play_period_bytes;
+
+			if (readw(DAQD + DAQDS_wStart) != PCTODSP_BASED(0x0))
+				offset = 0;
+
+			writew(PCTODSP_BASED(offset), DAQD + DAQDS_wStart);
+		}
+		++nbanks;
+
+		/* Then advance the tail */
+		/*
+		if (protect)
+			snd_printd(KERN_INFO "B %X %lX\n",
+				   bank_num, xtime.tv_usec);
+		*/
+
+		DAPQ_tail = (++bank_num % 3) * PCTODSP_OFFSET(DAQDS__size);
+		writew(DAPQ_tail, chip->DAPQ + JQS_wTail);
+		/* Tell the DSP to play the bank */
+		snd_msnd_send_dsp_cmd(chip, HDEX_PLAY_START);
+		if (protect)
+			if (2 == bank_num)
+				break;
+	}
+	/*
+	if (protect)
+		snd_printd(KERN_INFO "%lX\n", xtime.tv_usec);
+	*/
+	/* spin_unlock_irqrestore(&chip->lock, flags); not necessary */
+	return nbanks;
+}
+EXPORT_SYMBOL(snd_msnd_DAPQ);
+
+static void snd_msnd_play_reset_queue(struct snd_msnd *chip,
+				      unsigned int pcm_periods,
+				      unsigned int pcm_count)
+{
+	int	n;
+	void	*pDAQ = chip->mappedbase + DAPQ_DATA_BUFF;
+
+	chip->last_playbank = -1;
+	chip->playLimit = pcm_count * (pcm_periods - 1);
+	chip->playPeriods = pcm_periods;
+	writew(PCTODSP_OFFSET(0 * DAQDS__size), chip->DAPQ + JQS_wHead);
+	writew(PCTODSP_OFFSET(0 * DAQDS__size), chip->DAPQ + JQS_wTail);
+
+	chip->play_period_bytes = pcm_count;
+
+	for (n = 0; n < pcm_periods; ++n, pDAQ += DAQDS__size) {
+		writew(PCTODSP_BASED((u32)(pcm_count * n)),
+			pDAQ + DAQDS_wStart);
+		writew(0, pDAQ + DAQDS_wSize);
+		writew(1, pDAQ + DAQDS_wFormat);
+		writew(chip->play_sample_size, pDAQ + DAQDS_wSampleSize);
+		writew(chip->play_channels, pDAQ + DAQDS_wChannels);
+		writew(chip->play_sample_rate, pDAQ + DAQDS_wSampleRate);
+		writew(HIMT_PLAY_DONE * 0x100 + n, pDAQ + DAQDS_wIntMsg);
+		writew(n, pDAQ + DAQDS_wFlags);
+	}
+}
+
+static void snd_msnd_capture_reset_queue(struct snd_msnd *chip,
+					 unsigned int pcm_periods,
+					 unsigned int pcm_count)
+{
+	int		n;
+	void		*pDAQ;
+	/* unsigned long	flags; */
+
+	/* snd_msnd_init_queue(chip->DARQ, DARQ_DATA_BUFF, DARQ_BUFF_SIZE); */
+
+	chip->last_recbank = 2;
+	chip->captureLimit = pcm_count * (pcm_periods - 1);
+	chip->capturePeriods = pcm_periods;
+	writew(PCTODSP_OFFSET(0 * DAQDS__size), chip->DARQ + JQS_wHead);
+	writew(PCTODSP_OFFSET(chip->last_recbank * DAQDS__size),
+		chip->DARQ + JQS_wTail);
+
+#if 0 /* Critical section: bank 1 access. this is how the OSS driver does it:*/
+	spin_lock_irqsave(&chip->lock, flags);
+	outb(HPBLKSEL_1, chip->io + HP_BLKS);
+	memset_io(chip->mappedbase, 0, DAR_BUFF_SIZE * 3);
+	outb(HPBLKSEL_0, chip->io + HP_BLKS);
+	spin_unlock_irqrestore(&chip->lock, flags);
+#endif
+
+	chip->capturePeriodBytes = pcm_count;
+	snd_printdd("snd_msnd_capture_reset_queue() %i\n", pcm_count);
+
+	pDAQ = chip->mappedbase + DARQ_DATA_BUFF;
+
+	for (n = 0; n < pcm_periods; ++n, pDAQ += DAQDS__size) {
+		u32 tmp = pcm_count * n;
+
+		writew(PCTODSP_BASED(tmp + 0x3000), pDAQ + DAQDS_wStart);
+		writew(pcm_count, pDAQ + DAQDS_wSize);
+		writew(1, pDAQ + DAQDS_wFormat);
+		writew(chip->capture_sample_size, pDAQ + DAQDS_wSampleSize);
+		writew(chip->capture_channels, pDAQ + DAQDS_wChannels);
+		writew(chip->capture_sample_rate, pDAQ + DAQDS_wSampleRate);
+		writew(HIMT_RECORD_DONE * 0x100 + n, pDAQ + DAQDS_wIntMsg);
+		writew(n, pDAQ + DAQDS_wFlags);
+	}
+}
+
+static struct snd_pcm_hardware snd_msnd_playback = {
+	.info =			SNDRV_PCM_INFO_MMAP |
+				SNDRV_PCM_INFO_INTERLEAVED |
+				SNDRV_PCM_INFO_MMAP_VALID,
+	.formats =		SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+	.rates =		SNDRV_PCM_RATE_8000_48000,
+	.rate_min =		8000,
+	.rate_max =		48000,
+	.channels_min =		1,
+	.channels_max =		2,
+	.buffer_bytes_max =	0x3000,
+	.period_bytes_min =	0x40,
+	.period_bytes_max =	0x1800,
+	.periods_min =		2,
+	.periods_max =		3,
+	.fifo_size =		0,
+};
+
+static struct snd_pcm_hardware snd_msnd_capture = {
+	.info =			SNDRV_PCM_INFO_MMAP |
+				SNDRV_PCM_INFO_INTERLEAVED |
+				SNDRV_PCM_INFO_MMAP_VALID,
+	.formats =		SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE,
+	.rates =		SNDRV_PCM_RATE_8000_48000,
+	.rate_min =		8000,
+	.rate_max =		48000,
+	.channels_min =		1,
+	.channels_max =		2,
+	.buffer_bytes_max =	0x3000,
+	.period_bytes_min =	0x40,
+	.period_bytes_max =	0x1800,
+	.periods_min =		2,
+	.periods_max =		3,
+	.fifo_size =		0,
+};
+
+
+static int snd_msnd_playback_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_msnd *chip = snd_pcm_substream_chip(substream);
+
+	set_bit(F_AUDIO_WRITE_INUSE, &chip->flags);
+	clear_bit(F_WRITING, &chip->flags);
+	snd_msnd_enable_irq(chip);
+
+	runtime->dma_area = chip->mappedbase;
+	runtime->dma_bytes = 0x3000;
+
+	chip->playback_substream = substream;
+	runtime->hw = snd_msnd_playback;
+	return 0;
+}
+
+static int snd_msnd_playback_close(struct snd_pcm_substream *substream)
+{
+	struct snd_msnd *chip = snd_pcm_substream_chip(substream);
+
+	snd_msnd_disable_irq(chip);
+	clear_bit(F_AUDIO_WRITE_INUSE, &chip->flags);
+	return 0;
+}
+
+
+static int snd_msnd_playback_hw_params(struct snd_pcm_substream *substream,
+					struct snd_pcm_hw_params *params)
+{
+	int	i;
+	struct snd_msnd *chip = snd_pcm_substream_chip(substream);
+	void	*pDAQ =	chip->mappedbase + DAPQ_DATA_BUFF;
+
+	chip->play_sample_size = snd_pcm_format_width(params_format(params));
+	chip->play_channels = params_channels(params);
+	chip->play_sample_rate = params_rate(params);
+
+	for (i = 0; i < 3; ++i, pDAQ += DAQDS__size) {
+		writew(chip->play_sample_size, pDAQ + DAQDS_wSampleSize);
+		writew(chip->play_channels, pDAQ + DAQDS_wChannels);
+		writew(chip->play_sample_rate, pDAQ + DAQDS_wSampleRate);
+	}
+	/* dont do this here:
+	 * snd_msnd_calibrate_adc(chip->play_sample_rate);
+	 */
+
+	return 0;
+}
+
+static int snd_msnd_playback_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_msnd *chip = snd_pcm_substream_chip(substream);
+	unsigned int pcm_size = snd_pcm_lib_buffer_bytes(substream);
+	unsigned int pcm_count = snd_pcm_lib_period_bytes(substream);
+	unsigned int pcm_periods = pcm_size / pcm_count;
+
+	snd_msnd_play_reset_queue(chip, pcm_periods, pcm_count);
+	chip->playDMAPos = 0;
+	return 0;
+}
+
+static int snd_msnd_playback_trigger(struct snd_pcm_substream *substream,
+				     int cmd)
+{
+	struct snd_msnd *chip = snd_pcm_substream_chip(substream);
+	int	result = 0;
+
+	if (cmd == SNDRV_PCM_TRIGGER_START) {
+		snd_printdd("snd_msnd_playback_trigger(START)\n");
+		chip->banksPlayed = 0;
+		set_bit(F_WRITING, &chip->flags);
+		snd_msnd_DAPQ(chip, 1);
+	} else if (cmd == SNDRV_PCM_TRIGGER_STOP) {
+		snd_printdd("snd_msnd_playback_trigger(STop)\n");
+		/* interrupt diagnostic, comment this out later */
+		clear_bit(F_WRITING, &chip->flags);
+		snd_msnd_send_dsp_cmd(chip, HDEX_PLAY_STOP);
+	} else {
+		snd_printd(KERN_ERR "snd_msnd_playback_trigger(?????)\n");
+		result = -EINVAL;
+	}
+
+	snd_printdd("snd_msnd_playback_trigger() ENDE\n");
+	return result;
+}
+
+static snd_pcm_uframes_t
+snd_msnd_playback_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_msnd *chip = snd_pcm_substream_chip(substream);
+
+	return bytes_to_frames(substream->runtime, chip->playDMAPos);
+}
+
+
+static struct snd_pcm_ops snd_msnd_playback_ops = {
+	.open =		snd_msnd_playback_open,
+	.close =	snd_msnd_playback_close,
+	.ioctl =	snd_pcm_lib_ioctl,
+	.hw_params =	snd_msnd_playback_hw_params,
+	.prepare =	snd_msnd_playback_prepare,
+	.trigger =	snd_msnd_playback_trigger,
+	.pointer =	snd_msnd_playback_pointer,
+};
+
+static int snd_msnd_capture_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_msnd *chip = snd_pcm_substream_chip(substream);
+
+	set_bit(F_AUDIO_READ_INUSE, &chip->flags);
+	snd_msnd_enable_irq(chip);
+	runtime->dma_area = chip->mappedbase + 0x3000;
+	runtime->dma_bytes = 0x3000;
+	memset(runtime->dma_area, 0, runtime->dma_bytes);
+	chip->capture_substream = substream;
+	runtime->hw = snd_msnd_capture;
+	return 0;
+}
+
+static int snd_msnd_capture_close(struct snd_pcm_substream *substream)
+{
+	struct snd_msnd *chip = snd_pcm_substream_chip(substream);
+
+	snd_msnd_disable_irq(chip);
+	clear_bit(F_AUDIO_READ_INUSE, &chip->flags);
+	return 0;
+}
+
+static int snd_msnd_capture_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_msnd *chip = snd_pcm_substream_chip(substream);
+	unsigned int pcm_size = snd_pcm_lib_buffer_bytes(substream);
+	unsigned int pcm_count = snd_pcm_lib_period_bytes(substream);
+	unsigned int pcm_periods = pcm_size / pcm_count;
+
+	snd_msnd_capture_reset_queue(chip, pcm_periods, pcm_count);
+	chip->captureDMAPos = 0;
+	return 0;
+}
+
+static int snd_msnd_capture_trigger(struct snd_pcm_substream *substream,
+				    int cmd)
+{
+	struct snd_msnd *chip = snd_pcm_substream_chip(substream);
+
+	if (cmd == SNDRV_PCM_TRIGGER_START) {
+		chip->last_recbank = -1;
+		set_bit(F_READING, &chip->flags);
+		if (snd_msnd_send_dsp_cmd(chip, HDEX_RECORD_START) == 0)
+			return 0;
+
+		clear_bit(F_READING, &chip->flags);
+	} else if (cmd == SNDRV_PCM_TRIGGER_STOP) {
+		clear_bit(F_READING, &chip->flags);
+		snd_msnd_send_dsp_cmd(chip, HDEX_RECORD_STOP);
+		return 0;
+	}
+	return -EINVAL;
+}
+
+
+static snd_pcm_uframes_t
+snd_msnd_capture_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_msnd *chip = snd_pcm_substream_chip(substream);
+
+	return bytes_to_frames(runtime, chip->captureDMAPos);
+}
+
+
+static int snd_msnd_capture_hw_params(struct snd_pcm_substream *substream,
+					struct snd_pcm_hw_params *params)
+{
+	int		i;
+	struct snd_msnd *chip = snd_pcm_substream_chip(substream);
+	void		*pDAQ = chip->mappedbase + DARQ_DATA_BUFF;
+
+	chip->capture_sample_size = snd_pcm_format_width(params_format(params));
+	chip->capture_channels = params_channels(params);
+	chip->capture_sample_rate = params_rate(params);
+
+	for (i = 0; i < 3; ++i, pDAQ += DAQDS__size) {
+		writew(chip->capture_sample_size, pDAQ + DAQDS_wSampleSize);
+		writew(chip->capture_channels, pDAQ + DAQDS_wChannels);
+		writew(chip->capture_sample_rate, pDAQ + DAQDS_wSampleRate);
+	}
+	return 0;
+}
+
+
+static struct snd_pcm_ops snd_msnd_capture_ops = {
+	.open =		snd_msnd_capture_open,
+	.close =	snd_msnd_capture_close,
+	.ioctl =	snd_pcm_lib_ioctl,
+	.hw_params =	snd_msnd_capture_hw_params,
+	.prepare =	snd_msnd_capture_prepare,
+	.trigger =	snd_msnd_capture_trigger,
+	.pointer =	snd_msnd_capture_pointer,
+};
+
+
+int snd_msnd_pcm(struct snd_card *card, int device,
+			struct snd_pcm **rpcm)
+{
+	struct snd_msnd *chip = card->private_data;
+	struct snd_pcm	*pcm;
+	int err;
+
+	err = snd_pcm_new(card, "MSNDPINNACLE", device, 1, 1, &pcm);
+	if (err < 0)
+		return err;
+
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_msnd_playback_ops);
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_msnd_capture_ops);
+
+	pcm->private_data = chip;
+	strcpy(pcm->name, "Hurricane");
+
+
+	if (rpcm)
+		*rpcm = pcm;
+	return 0;
+}
+EXPORT_SYMBOL(snd_msnd_pcm);
+
+MODULE_DESCRIPTION("Common routines for Turtle Beach Multisound drivers");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/isa/msnd/msnd.h b/sound/isa/msnd/msnd.h
new file mode 100644
index 000000000000..3773e242b58e
--- /dev/null
+++ b/sound/isa/msnd/msnd.h
@@ -0,0 +1,308 @@
+/*********************************************************************
+ *
+ * msnd.h
+ *
+ * Turtle Beach MultiSound Sound Card Driver for Linux
+ *
+ * Some parts of this header file were derived from the Turtle Beach
+ * MultiSound Driver Development Kit.
+ *
+ * Copyright (C) 1998 Andrew Veliath
+ * Copyright (C) 1993 Turtle Beach Systems, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ ********************************************************************/
+#ifndef __MSND_H
+#define __MSND_H
+
+#define DEFSAMPLERATE		44100
+#define DEFSAMPLESIZE		SNDRV_PCM_FORMAT_S16
+#define DEFCHANNELS		1
+
+#define SRAM_BANK_SIZE		0x8000
+#define SRAM_CNTL_START		0x7F00
+#define SMA_STRUCT_START	0x7F40
+
+#define DSP_BASE_ADDR		0x4000
+#define DSP_BANK_BASE		0x4000
+
+#define AGND			0x01
+#define SIGNAL			0x02
+
+#define EXT_DSP_BIT_DCAL	0x0001
+#define EXT_DSP_BIT_MIDI_CON	0x0002
+
+#define BUFFSIZE		0x8000
+#define HOSTQ_SIZE		0x40
+
+#define DAP_BUFF_SIZE		0x2400
+
+#define DAPQ_STRUCT_SIZE	0x10
+#define DARQ_STRUCT_SIZE	0x10
+#define DAPQ_BUFF_SIZE		(3 * 0x10)
+#define DARQ_BUFF_SIZE		(3 * 0x10)
+#define MODQ_BUFF_SIZE		0x400
+
+#define DAPQ_DATA_BUFF		0x6C00
+#define DARQ_DATA_BUFF		0x6C30
+#define MODQ_DATA_BUFF		0x6C60
+#define MIDQ_DATA_BUFF		0x7060
+
+#define DAPQ_OFFSET		SRAM_CNTL_START
+#define DARQ_OFFSET		(SRAM_CNTL_START + 0x08)
+#define MODQ_OFFSET		(SRAM_CNTL_START + 0x10)
+#define MIDQ_OFFSET		(SRAM_CNTL_START + 0x18)
+#define DSPQ_OFFSET		(SRAM_CNTL_START + 0x20)
+
+#define	HP_ICR			0x00
+#define	HP_CVR			0x01
+#define	HP_ISR			0x02
+#define	HP_IVR			0x03
+#define HP_NU			0x04
+#define HP_INFO			0x04
+#define	HP_TXH			0x05
+#define	HP_RXH			0x05
+#define	HP_TXM			0x06
+#define	HP_RXM			0x06
+#define	HP_TXL			0x07
+#define	HP_RXL			0x07
+
+#define HP_ICR_DEF		0x00
+#define HP_CVR_DEF		0x12
+#define HP_ISR_DEF		0x06
+#define HP_IVR_DEF		0x0f
+#define HP_NU_DEF		0x00
+
+#define	HP_IRQM			0x09
+
+#define	HPR_BLRC		0x08
+#define	HPR_SPR1		0x09
+#define	HPR_SPR2		0x0A
+#define	HPR_TCL0		0x0B
+#define	HPR_TCL1		0x0C
+#define	HPR_TCL2		0x0D
+#define	HPR_TCL3		0x0E
+#define	HPR_TCL4		0x0F
+
+#define	HPICR_INIT		0x80
+#define HPICR_HM1		0x40
+#define HPICR_HM0		0x20
+#define HPICR_HF1		0x10
+#define HPICR_HF0		0x08
+#define	HPICR_TREQ		0x02
+#define	HPICR_RREQ		0x01
+
+#define HPCVR_HC		0x80
+
+#define	HPISR_HREQ		0x80
+#define HPISR_DMA		0x40
+#define HPISR_HF3		0x10
+#define HPISR_HF2		0x08
+#define	HPISR_TRDY		0x04
+#define	HPISR_TXDE		0x02
+#define	HPISR_RXDF		0x01
+
+#define	HPIO_290		0
+#define	HPIO_260		1
+#define	HPIO_250		2
+#define	HPIO_240		3
+#define	HPIO_230		4
+#define	HPIO_220		5
+#define	HPIO_210		6
+#define	HPIO_3E0		7
+
+#define	HPMEM_NONE		0
+#define	HPMEM_B000		1
+#define	HPMEM_C800		2
+#define	HPMEM_D000		3
+#define	HPMEM_D400		4
+#define	HPMEM_D800		5
+#define	HPMEM_E000		6
+#define	HPMEM_E800		7
+
+#define	HPIRQ_NONE		0
+#define HPIRQ_5			1
+#define HPIRQ_7			2
+#define HPIRQ_9			3
+#define HPIRQ_10		4
+#define HPIRQ_11		5
+#define HPIRQ_12		6
+#define HPIRQ_15		7
+
+#define	HIMT_PLAY_DONE		0x00
+#define	HIMT_RECORD_DONE	0x01
+#define	HIMT_MIDI_EOS		0x02
+#define	HIMT_MIDI_OUT		0x03
+
+#define	HIMT_MIDI_IN_UCHAR	0x0E
+#define	HIMT_DSP		0x0F
+
+#define	HDEX_BASE	       	0x92
+#define	HDEX_PLAY_START		(0 + HDEX_BASE)
+#define	HDEX_PLAY_STOP		(1 + HDEX_BASE)
+#define	HDEX_PLAY_PAUSE		(2 + HDEX_BASE)
+#define	HDEX_PLAY_RESUME	(3 + HDEX_BASE)
+#define	HDEX_RECORD_START	(4 + HDEX_BASE)
+#define	HDEX_RECORD_STOP	(5 + HDEX_BASE)
+#define	HDEX_MIDI_IN_START 	(6 + HDEX_BASE)
+#define	HDEX_MIDI_IN_STOP	(7 + HDEX_BASE)
+#define	HDEX_MIDI_OUT_START	(8 + HDEX_BASE)
+#define	HDEX_MIDI_OUT_STOP	(9 + HDEX_BASE)
+#define	HDEX_AUX_REQ		(10 + HDEX_BASE)
+
+#define	HDEXAR_CLEAR_PEAKS	1
+#define	HDEXAR_IN_SET_POTS	2
+#define	HDEXAR_AUX_SET_POTS	3
+#define	HDEXAR_CAL_A_TO_D	4
+#define	HDEXAR_RD_EXT_DSP_BITS	5
+
+/* Pinnacle only HDEXAR defs */
+#define	HDEXAR_SET_ANA_IN	0
+#define	HDEXAR_SET_SYNTH_IN	4
+#define	HDEXAR_READ_DAT_IN	5
+#define	HDEXAR_MIC_SET_POTS	6
+#define	HDEXAR_SET_DAT_IN	7
+
+#define HDEXAR_SET_SYNTH_48	8
+#define HDEXAR_SET_SYNTH_44	9
+
+#define HIWORD(l)		((u16)((((u32)(l)) >> 16) & 0xFFFF))
+#define LOWORD(l)		((u16)(u32)(l))
+#define HIBYTE(w)		((u8)(((u16)(w) >> 8) & 0xFF))
+#define LOBYTE(w)		((u8)(w))
+#define MAKELONG(low, hi)	((long)(((u16)(low))|(((u32)((u16)(hi)))<<16)))
+#define MAKEWORD(low, hi)	((u16)(((u8)(low))|(((u16)((u8)(hi)))<<8)))
+
+#define PCTODSP_OFFSET(w)	(u16)((w)/2)
+#define PCTODSP_BASED(w)	(u16)(((w)/2) + DSP_BASE_ADDR)
+#define DSPTOPC_BASED(w)	(((w) - DSP_BASE_ADDR) * 2)
+
+#ifdef SLOWIO
+#  undef outb
+#  undef inb
+#  define outb			outb_p
+#  define inb			inb_p
+#endif
+
+/* JobQueueStruct */
+#define JQS_wStart		0x00
+#define JQS_wSize		0x02
+#define JQS_wHead		0x04
+#define JQS_wTail		0x06
+#define JQS__size		0x08
+
+/* DAQueueDataStruct */
+#define DAQDS_wStart		0x00
+#define DAQDS_wSize		0x02
+#define DAQDS_wFormat		0x04
+#define DAQDS_wSampleSize	0x06
+#define DAQDS_wChannels		0x08
+#define DAQDS_wSampleRate	0x0A
+#define DAQDS_wIntMsg		0x0C
+#define DAQDS_wFlags		0x0E
+#define DAQDS__size		0x10
+
+#include <sound/pcm.h>
+
+struct snd_msnd {
+	void __iomem		*mappedbase;
+	int			play_period_bytes;
+	int			playLimit;
+	int			playPeriods;
+	int 			playDMAPos;
+	int			banksPlayed;
+	int 			captureDMAPos;
+	int			capturePeriodBytes;
+	int			captureLimit;
+	int			capturePeriods;
+	struct snd_card		*card;
+	void			*msndmidi_mpu;
+	struct snd_rawmidi	*rmidi;
+
+	/* Hardware resources */
+	long io;
+	int memid, irqid;
+	int irq, irq_ref;
+	unsigned long base;
+
+	/* Motorola 56k DSP SMA */
+	void __iomem	*SMA;
+	void __iomem	*DAPQ;
+	void __iomem	*DARQ;
+	void __iomem	*MODQ;
+	void __iomem	*MIDQ;
+	void __iomem	*DSPQ;
+	int dspq_data_buff, dspq_buff_size;
+
+	/* State variables */
+	enum { msndClassic, msndPinnacle } type;
+	mode_t mode;
+	unsigned long flags;
+#define F_RESETTING			0
+#define F_HAVEDIGITAL			1
+#define F_AUDIO_WRITE_INUSE		2
+#define F_WRITING			3
+#define F_WRITEBLOCK			4
+#define F_WRITEFLUSH			5
+#define F_AUDIO_READ_INUSE		6
+#define F_READING			7
+#define F_READBLOCK			8
+#define F_EXT_MIDI_INUSE		9
+#define F_HDR_MIDI_INUSE		10
+#define F_DISABLE_WRITE_NDELAY		11
+	spinlock_t lock;
+	spinlock_t mixer_lock;
+	int nresets;
+	unsigned recsrc;
+#define LEVEL_ENTRIES 32
+	int left_levels[LEVEL_ENTRIES];
+	int right_levels[LEVEL_ENTRIES];
+	int calibrate_signal;
+	int play_sample_size, play_sample_rate, play_channels;
+	int play_ndelay;
+	int capture_sample_size, capture_sample_rate, capture_channels;
+	int capture_ndelay;
+	u8 bCurrentMidiPatch;
+
+	int last_playbank, last_recbank;
+	struct snd_pcm_substream *playback_substream;
+	struct snd_pcm_substream *capture_substream;
+
+};
+
+void snd_msnd_init_queue(void *base, int start, int size);
+
+int snd_msnd_send_dsp_cmd(struct snd_msnd *chip, u8 cmd);
+int snd_msnd_send_word(struct snd_msnd *chip,
+			   unsigned char high,
+			   unsigned char mid,
+			   unsigned char low);
+int snd_msnd_upload_host(struct snd_msnd *chip,
+			     const u8 *bin, int len);
+int snd_msnd_enable_irq(struct snd_msnd *chip);
+int snd_msnd_disable_irq(struct snd_msnd *chip);
+void snd_msnd_dsp_halt(struct snd_msnd *chip, struct file *file);
+int snd_msnd_DAPQ(struct snd_msnd *chip, int start);
+int snd_msnd_DARQ(struct snd_msnd *chip, int start);
+int snd_msnd_pcm(struct snd_card *card, int device, struct snd_pcm **rpcm);
+
+int snd_msndmidi_new(struct snd_card *card, int device);
+void snd_msndmidi_input_read(void *mpu);
+
+void snd_msndmix_setup(struct snd_msnd *chip);
+int __devinit snd_msndmix_new(struct snd_card *card);
+int snd_msndmix_force_recsrc(struct snd_msnd *chip, int recsrc);
+#endif /* __MSND_H */
diff --git a/sound/isa/msnd/msnd_classic.c b/sound/isa/msnd/msnd_classic.c
new file mode 100644
index 000000000000..3b23a096fa4e
--- /dev/null
+++ b/sound/isa/msnd/msnd_classic.c
@@ -0,0 +1,3 @@
+/* The work is in msnd_pinnacle.c, just define MSND_CLASSIC before it. */
+#define MSND_CLASSIC
+#include "msnd_pinnacle.c"
diff --git a/sound/isa/msnd/msnd_classic.h b/sound/isa/msnd/msnd_classic.h
new file mode 100644
index 000000000000..f18d5fa5baf4
--- /dev/null
+++ b/sound/isa/msnd/msnd_classic.h
@@ -0,0 +1,129 @@
+/*********************************************************************
+ *
+ * msnd_classic.h
+ *
+ * Turtle Beach MultiSound Sound Card Driver for Linux
+ *
+ * Some parts of this header file were derived from the Turtle Beach
+ * MultiSound Driver Development Kit.
+ *
+ * Copyright (C) 1998 Andrew Veliath
+ * Copyright (C) 1993 Turtle Beach Systems, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ ********************************************************************/
+#ifndef __MSND_CLASSIC_H
+#define __MSND_CLASSIC_H
+
+#define DSP_NUMIO				0x10
+
+#define	HP_MEMM					0x08
+
+#define	HP_BITM					0x0E
+#define	HP_WAIT					0x0D
+#define	HP_DSPR					0x0A
+#define	HP_PROR					0x0B
+#define	HP_BLKS					0x0C
+
+#define	HPPRORESET_OFF				0
+#define HPPRORESET_ON				1
+
+#define HPDSPRESET_OFF				0
+#define HPDSPRESET_ON				1
+
+#define HPBLKSEL_0				0
+#define HPBLKSEL_1				1
+
+#define HPWAITSTATE_0				0
+#define HPWAITSTATE_1				1
+
+#define HPBITMODE_16				0
+#define HPBITMODE_8				1
+
+#define	HIDSP_INT_PLAY_UNDER			0x00
+#define	HIDSP_INT_RECORD_OVER			0x01
+#define	HIDSP_INPUT_CLIPPING			0x02
+#define	HIDSP_MIDI_IN_OVER			0x10
+#define	HIDSP_MIDI_OVERRUN_ERR  0x13
+
+#define TIME_PRO_RESET_DONE			0x028A
+#define TIME_PRO_SYSEX				0x0040
+#define TIME_PRO_RESET				0x0032
+
+#define DAR_BUFF_SIZE				0x2000
+
+#define MIDQ_BUFF_SIZE				0x200
+#define DSPQ_BUFF_SIZE				0x40
+
+#define DSPQ_DATA_BUFF				0x7260
+
+#define MOP_SYNTH				0x10
+#define MOP_EXTOUT				0x32
+#define MOP_EXTTHRU				0x02
+#define MOP_OUTMASK				0x01
+
+#define MIP_EXTIN				0x01
+#define MIP_SYNTH				0x00
+#define MIP_INMASK				0x32
+
+/* Classic SMA Common Data */
+#define SMA_wCurrPlayBytes			0x0000
+#define SMA_wCurrRecordBytes			0x0002
+#define SMA_wCurrPlayVolLeft			0x0004
+#define SMA_wCurrPlayVolRight			0x0006
+#define SMA_wCurrInVolLeft			0x0008
+#define SMA_wCurrInVolRight			0x000a
+#define SMA_wUser_3				0x000c
+#define SMA_wUser_4				0x000e
+#define SMA_dwUser_5				0x0010
+#define SMA_dwUser_6				0x0014
+#define SMA_wUser_7				0x0018
+#define SMA_wReserved_A				0x001a
+#define SMA_wReserved_B				0x001c
+#define SMA_wReserved_C				0x001e
+#define SMA_wReserved_D				0x0020
+#define SMA_wReserved_E				0x0022
+#define SMA_wReserved_F				0x0024
+#define SMA_wReserved_G				0x0026
+#define SMA_wReserved_H				0x0028
+#define SMA_wCurrDSPStatusFlags			0x002a
+#define SMA_wCurrHostStatusFlags		0x002c
+#define SMA_wCurrInputTagBits			0x002e
+#define SMA_wCurrLeftPeak			0x0030
+#define SMA_wCurrRightPeak			0x0032
+#define SMA_wExtDSPbits				0x0034
+#define SMA_bExtHostbits			0x0036
+#define SMA_bBoardLevel				0x0037
+#define SMA_bInPotPosRight			0x0038
+#define SMA_bInPotPosLeft			0x0039
+#define SMA_bAuxPotPosRight			0x003a
+#define SMA_bAuxPotPosLeft			0x003b
+#define SMA_wCurrMastVolLeft			0x003c
+#define SMA_wCurrMastVolRight			0x003e
+#define SMA_bUser_12				0x0040
+#define SMA_bUser_13				0x0041
+#define SMA_wUser_14				0x0042
+#define SMA_wUser_15				0x0044
+#define SMA_wCalFreqAtoD			0x0046
+#define SMA_wUser_16				0x0048
+#define SMA_wUser_17				0x004a
+#define SMA__size				0x004c
+
+#define INITCODEFILE		"turtlebeach/msndinit.bin"
+#define PERMCODEFILE		"turtlebeach/msndperm.bin"
+#define LONGNAME		"MultiSound (Classic/Monterey/Tahiti)"
+
+#endif /* __MSND_CLASSIC_H */
diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c
new file mode 100644
index 000000000000..cb9aa4c4edd0
--- /dev/null
+++ b/sound/isa/msnd/msnd_midi.c
@@ -0,0 +1,180 @@
+/*
+ *  Copyright (c) by Jaroslav Kysela <perex@perex.cz>
+ *  Copyright (c) 2009 by Krzysztof Helt
+ *  Routines for control of MPU-401 in UART mode
+ *
+ *  MPU-401 supports UART mode which is not capable generate transmit
+ *  interrupts thus output is done via polling. Also, if irq < 0, then
+ *  input is done also via polling. Do not expect good performance.
+ *
+ *
+ *   This program is free software; you can redistribute it and/or modify
+ *   it under the terms of the GNU General Public License as published by
+ *   the Free Software Foundation; either version 2 of the License, or
+ *   (at your option) any later version.
+ *
+ *   This program is distributed in the hope that it will be useful,
+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *   GNU General Public License for more details.
+ *
+ *   You should have received a copy of the GNU General Public License
+ *   along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+#include <linux/io.h>
+#include <linux/delay.h>
+#include <linux/ioport.h>
+#include <linux/errno.h>
+#include <sound/core.h>
+#include <sound/rawmidi.h>
+
+#include "msnd.h"
+
+#define MSNDMIDI_MODE_BIT_INPUT		0
+#define MSNDMIDI_MODE_BIT_OUTPUT		1
+#define MSNDMIDI_MODE_BIT_INPUT_TRIGGER	2
+#define MSNDMIDI_MODE_BIT_OUTPUT_TRIGGER	3
+
+struct snd_msndmidi {
+	struct snd_msnd *dev;
+
+	unsigned long mode;		/* MSNDMIDI_MODE_XXXX */
+
+	struct snd_rawmidi_substream *substream_input;
+
+	spinlock_t input_lock;
+};
+
+/*
+ * input/output open/close - protected by open_mutex in rawmidi.c
+ */
+static int snd_msndmidi_input_open(struct snd_rawmidi_substream *substream)
+{
+	struct snd_msndmidi *mpu;
+
+	snd_printdd("snd_msndmidi_input_open()\n");
+
+	mpu = substream->rmidi->private_data;
+
+	mpu->substream_input = substream;
+
+	snd_msnd_enable_irq(mpu->dev);
+
+	snd_msnd_send_dsp_cmd(mpu->dev, HDEX_MIDI_IN_START);
+	set_bit(MSNDMIDI_MODE_BIT_INPUT, &mpu->mode);
+	return 0;
+}
+
+static int snd_msndmidi_input_close(struct snd_rawmidi_substream *substream)
+{
+	struct snd_msndmidi *mpu;
+
+	mpu = substream->rmidi->private_data;
+	snd_msnd_send_dsp_cmd(mpu->dev, HDEX_MIDI_IN_STOP);
+	clear_bit(MSNDMIDI_MODE_BIT_INPUT, &mpu->mode);
+	mpu->substream_input = NULL;
+	snd_msnd_disable_irq(mpu->dev);
+	return 0;
+}
+
+static void snd_msndmidi_input_drop(struct snd_msndmidi *mpu)
+{
+	u16 tail;
+
+	tail = readw(mpu->dev->MIDQ + JQS_wTail);
+	writew(tail, mpu->dev->MIDQ + JQS_wHead);
+}
+
+/*
+ * trigger input
+ */
+static void snd_msndmidi_input_trigger(struct snd_rawmidi_substream *substream,
+					int up)
+{
+	unsigned long flags;
+	struct snd_msndmidi *mpu;
+
+	snd_printdd("snd_msndmidi_input_trigger(, %i)\n", up);
+
+	mpu = substream->rmidi->private_data;
+	spin_lock_irqsave(&mpu->input_lock, flags);
+	if (up) {
+		if (!test_and_set_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER,
+				      &mpu->mode))
+			snd_msndmidi_input_drop(mpu);
+	} else {
+		clear_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, &mpu->mode);
+	}
+	spin_unlock_irqrestore(&mpu->input_lock, flags);
+	if (up)
+		snd_msndmidi_input_read(mpu);
+}
+
+void snd_msndmidi_input_read(void *mpuv)
+{
+	unsigned long flags;
+	struct snd_msndmidi *mpu = mpuv;
+	void *pwMIDQData = mpu->dev->mappedbase + MIDQ_DATA_BUFF;
+
+	spin_lock_irqsave(&mpu->input_lock, flags);
+	while (readw(mpu->dev->MIDQ + JQS_wTail) !=
+	       readw(mpu->dev->MIDQ + JQS_wHead)) {
+		u16 wTmp, val;
+		val = readw(pwMIDQData + 2 * readw(mpu->dev->MIDQ + JQS_wHead));
+
+			if (test_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER,
+				     &mpu->mode))
+				snd_rawmidi_receive(mpu->substream_input,
+						    (unsigned char *)&val, 1);
+
+		wTmp = readw(mpu->dev->MIDQ + JQS_wHead) + 1;
+		if (wTmp > readw(mpu->dev->MIDQ + JQS_wSize))
+			writew(0,  mpu->dev->MIDQ + JQS_wHead);
+		else
+			writew(wTmp,  mpu->dev->MIDQ + JQS_wHead);
+	}
+	spin_unlock_irqrestore(&mpu->input_lock, flags);
+}
+EXPORT_SYMBOL(snd_msndmidi_input_read);
+
+static struct snd_rawmidi_ops snd_msndmidi_input = {
+	.open =		snd_msndmidi_input_open,
+	.close =	snd_msndmidi_input_close,
+	.trigger =	snd_msndmidi_input_trigger,
+};
+
+static void snd_msndmidi_free(struct snd_rawmidi *rmidi)
+{
+	struct snd_msndmidi *mpu = rmidi->private_data;
+	kfree(mpu);
+}
+
+int snd_msndmidi_new(struct snd_card *card, int device)
+{
+	struct snd_msnd *chip = card->private_data;
+	struct snd_msndmidi *mpu;
+	struct snd_rawmidi *rmidi;
+	int err;
+
+	err = snd_rawmidi_new(card, "MSND-MIDI", device, 1, 1, &rmidi);
+	if (err < 0)
+		return err;
+	mpu = kcalloc(1, sizeof(*mpu), GFP_KERNEL);
+	if (mpu == NULL) {
+		snd_device_free(card, rmidi);
+		return -ENOMEM;
+	}
+	mpu->dev = chip;
+	chip->msndmidi_mpu = mpu;
+	rmidi->private_data = mpu;
+	rmidi->private_free = snd_msndmidi_free;
+	spin_lock_init(&mpu->input_lock);
+	strcpy(rmidi->name, "MSND MIDI");
+	snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+			    &snd_msndmidi_input);
+	rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT;
+	return 0;
+}
diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c
new file mode 100644
index 000000000000..60b6abd71612
--- /dev/null
+++ b/sound/isa/msnd/msnd_pinnacle.c
@@ -0,0 +1,1238 @@
+/*********************************************************************
+ *
+ * Linux multisound pinnacle/fiji driver for ALSA.
+ *
+ * 2002/06/30 Karsten Wiese:
+ *	for now this is only used to build a pinnacle / fiji driver.
+ *	the OSS parent of this code is designed to also support
+ *	the multisound classic via the file msnd_classic.c.
+ *	to make it easier for some brave heart to implemt classic
+ *	support in alsa, i left all the MSND_CLASSIC tokens in this file.
+ *	but for now this untested & undone.
+ *
+ *
+ * ripped from linux kernel 2.4.18 by Karsten Wiese.
+ *
+ * the following is a copy of the 2.4.18 OSS FREE file-heading comment:
+ *
+ * Turtle Beach MultiSound Sound Card Driver for Linux
+ * msnd_pinnacle.c / msnd_classic.c
+ *
+ * -- If MSND_CLASSIC is defined:
+ *
+ *     -> driver for Turtle Beach Classic/Monterey/Tahiti
+ *
+ * -- Else
+ *
+ *     -> driver for Turtle Beach Pinnacle/Fiji
+ *
+ * 12-3-2000  Modified IO port validation  Steve Sycamore
+ *
+ * Copyright (C) 1998 Andrew Veliath
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ ********************************************************************/
+
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/types.h>
+#include <linux/delay.h>
+#include <linux/ioport.h>
+#include <linux/firmware.h>
+#include <linux/isa.h>
+#include <linux/isapnp.h>
+#include <linux/irq.h>
+#include <linux/io.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/asound.h>
+#include <sound/pcm.h>
+#include <sound/mpu401.h>
+
+#ifdef MSND_CLASSIC
+# ifndef __alpha__
+#  define SLOWIO
+# endif
+#endif
+#include "msnd.h"
+#ifdef MSND_CLASSIC
+#  include "msnd_classic.h"
+#  define LOGNAME			"msnd_classic"
+#else
+#  include "msnd_pinnacle.h"
+#  define LOGNAME			"snd_msnd_pinnacle"
+#endif
+
+static void __devinit set_default_audio_parameters(struct snd_msnd *chip)
+{
+	chip->play_sample_size = DEFSAMPLESIZE;
+	chip->play_sample_rate = DEFSAMPLERATE;
+	chip->play_channels = DEFCHANNELS;
+	chip->capture_sample_size = DEFSAMPLESIZE;
+	chip->capture_sample_rate = DEFSAMPLERATE;
+	chip->capture_channels = DEFCHANNELS;
+}
+
+static void snd_msnd_eval_dsp_msg(struct snd_msnd *chip, u16 wMessage)
+{
+	switch (HIBYTE(wMessage)) {
+	case HIMT_PLAY_DONE: {
+		if (chip->banksPlayed < 3)
+			snd_printdd("%08X: HIMT_PLAY_DONE: %i\n",
+				(unsigned)jiffies, LOBYTE(wMessage));
+
+		if (chip->last_playbank == LOBYTE(wMessage)) {
+			snd_printdd("chip.last_playbank == LOBYTE(wMessage)\n");
+			break;
+		}
+		chip->banksPlayed++;
+
+		if (test_bit(F_WRITING, &chip->flags))
+			snd_msnd_DAPQ(chip, 0);
+
+		chip->last_playbank = LOBYTE(wMessage);
+		chip->playDMAPos += chip->play_period_bytes;
+		if (chip->playDMAPos > chip->playLimit)
+			chip->playDMAPos = 0;
+		snd_pcm_period_elapsed(chip->playback_substream);
+
+		break;
+	}
+	case HIMT_RECORD_DONE:
+		if (chip->last_recbank == LOBYTE(wMessage))
+			break;
+		chip->last_recbank = LOBYTE(wMessage);
+		chip->captureDMAPos += chip->capturePeriodBytes;
+		if (chip->captureDMAPos > (chip->captureLimit))
+			chip->captureDMAPos = 0;
+
+		if (test_bit(F_READING, &chip->flags))
+			snd_msnd_DARQ(chip, chip->last_recbank);
+
+		snd_pcm_period_elapsed(chip->capture_substream);
+		break;
+
+	case HIMT_DSP:
+		switch (LOBYTE(wMessage)) {
+#ifndef MSND_CLASSIC
+		case HIDSP_PLAY_UNDER:
+#endif
+		case HIDSP_INT_PLAY_UNDER:
+			snd_printd(KERN_WARNING LOGNAME ": Play underflow %i\n",
+				chip->banksPlayed);
+			if (chip->banksPlayed > 2)
+				clear_bit(F_WRITING, &chip->flags);
+			break;
+
+		case HIDSP_INT_RECORD_OVER:
+			snd_printd(KERN_WARNING LOGNAME ": Record overflow\n");
+			clear_bit(F_READING, &chip->flags);
+			break;
+
+		default:
+			snd_printd(KERN_WARNING LOGNAME
+				   ": DSP message %d 0x%02x\n",
+				   LOBYTE(wMessage), LOBYTE(wMessage));
+			break;
+		}
+		break;
+
+	case HIMT_MIDI_IN_UCHAR:
+		if (chip->msndmidi_mpu)
+			snd_msndmidi_input_read(chip->msndmidi_mpu);
+		break;
+
+	default:
+		snd_printd(KERN_WARNING LOGNAME ": HIMT message %d 0x%02x\n",
+			   HIBYTE(wMessage), HIBYTE(wMessage));
+		break;
+	}
+}
+
+static irqreturn_t snd_msnd_interrupt(int irq, void *dev_id)
+{
+	struct snd_msnd *chip = dev_id;
+	void *pwDSPQData = chip->mappedbase + DSPQ_DATA_BUFF;
+
+	/* Send ack to DSP */
+	/* inb(chip->io + HP_RXL); */
+
+	/* Evaluate queued DSP messages */
+	while (readw(chip->DSPQ + JQS_wTail) != readw(chip->DSPQ + JQS_wHead)) {
+		u16 wTmp;
+
+		snd_msnd_eval_dsp_msg(chip,
+			readw(pwDSPQData + 2 * readw(chip->DSPQ + JQS_wHead)));
+
+		wTmp = readw(chip->DSPQ + JQS_wHead) + 1;
+		if (wTmp > readw(chip->DSPQ + JQS_wSize))
+			writew(0, chip->DSPQ + JQS_wHead);
+		else
+			writew(wTmp, chip->DSPQ + JQS_wHead);
+	}
+	/* Send ack to DSP */
+	inb(chip->io + HP_RXL);
+	return IRQ_HANDLED;
+}
+
+
+static int snd_msnd_reset_dsp(long io, unsigned char *info)
+{
+	int timeout = 100;
+
+	outb(HPDSPRESET_ON, io + HP_DSPR);
+	msleep(1);
+#ifndef MSND_CLASSIC
+	if (info)
+		*info = inb(io + HP_INFO);
+#endif
+	outb(HPDSPRESET_OFF, io + HP_DSPR);
+	msleep(1);
+	while (timeout-- > 0) {
+		if (inb(io + HP_CVR) == HP_CVR_DEF)
+			return 0;
+		msleep(1);
+	}
+	snd_printk(KERN_ERR LOGNAME ": Cannot reset DSP\n");
+
+	return -EIO;
+}
+
+static int __devinit snd_msnd_probe(struct snd_card *card)
+{
+	struct snd_msnd *chip = card->private_data;
+	unsigned char info;
+#ifndef MSND_CLASSIC
+	char *xv, *rev = NULL;
+	char *pin = "TB Pinnacle", *fiji = "TB Fiji";
+	char *pinfiji = "TB Pinnacle/Fiji";
+#endif
+
+	if (!request_region(chip->io, DSP_NUMIO, "probing")) {
+		snd_printk(KERN_ERR LOGNAME ": I/O port conflict\n");
+		return -ENODEV;
+	}
+
+	if (snd_msnd_reset_dsp(chip->io, &info) < 0) {
+		release_region(chip->io, DSP_NUMIO);
+		return -ENODEV;
+	}
+
+#ifdef MSND_CLASSIC
+	strcpy(card->shortname, "Classic/Tahiti/Monterey");
+	strcpy(card->longname, "Turtle Beach Multisound");
+	printk(KERN_INFO LOGNAME ": %s, "
+	       "I/O 0x%lx-0x%lx, IRQ %d, memory mapped to 0x%lX-0x%lX\n",
+	       card->shortname,
+	       chip->io, chip->io + DSP_NUMIO - 1,
+	       chip->irq,
+	       chip->base, chip->base + 0x7fff);
+#else
+	switch (info >> 4) {
+	case 0xf:
+		xv = "<= 1.15";
+		break;
+	case 0x1:
+		xv = "1.18/1.2";
+		break;
+	case 0x2:
+		xv = "1.3";
+		break;
+	case 0x3:
+		xv = "1.4";
+		break;
+	default:
+		xv = "unknown";
+		break;
+	}
+
+	switch (info & 0x7) {
+	case 0x0:
+		rev = "I";
+		strcpy(card->shortname, pin);
+		break;
+	case 0x1:
+		rev = "F";
+		strcpy(card->shortname, pin);
+		break;
+	case 0x2:
+		rev = "G";
+		strcpy(card->shortname, pin);
+		break;
+	case 0x3:
+		rev = "H";
+		strcpy(card->shortname, pin);
+		break;
+	case 0x4:
+		rev = "E";
+		strcpy(card->shortname, fiji);
+		break;
+	case 0x5:
+		rev = "C";
+		strcpy(card->shortname, fiji);
+		break;
+	case 0x6:
+		rev = "D";
+		strcpy(card->shortname, fiji);
+		break;
+	case 0x7:
+		rev = "A-B (Fiji) or A-E (Pinnacle)";
+		strcpy(card->shortname, pinfiji);
+		break;
+	}
+	strcpy(card->longname, "Turtle Beach Multisound Pinnacle");
+	printk(KERN_INFO LOGNAME ": %s revision %s, Xilinx version %s, "
+	       "I/O 0x%lx-0x%lx, IRQ %d, memory mapped to 0x%lX-0x%lX\n",
+	       card->shortname,
+	       rev, xv,
+	       chip->io, chip->io + DSP_NUMIO - 1,
+	       chip->irq,
+	       chip->base, chip->base + 0x7fff);
+#endif
+
+	release_region(chip->io, DSP_NUMIO);
+	return 0;
+}
+
+static int snd_msnd_init_sma(struct snd_msnd *chip)
+{
+	static int initted;
+	u16 mastVolLeft, mastVolRight;
+	unsigned long flags;
+
+#ifdef MSND_CLASSIC
+	outb(chip->memid, chip->io + HP_MEMM);
+#endif
+	outb(HPBLKSEL_0, chip->io + HP_BLKS);
+	/* Motorola 56k shared memory base */
+	chip->SMA = chip->mappedbase + SMA_STRUCT_START;
+
+	if (initted) {
+		mastVolLeft = readw(chip->SMA + SMA_wCurrMastVolLeft);
+		mastVolRight = readw(chip->SMA + SMA_wCurrMastVolRight);
+	} else
+		mastVolLeft = mastVolRight = 0;
+	memset_io(chip->mappedbase, 0, 0x8000);
+
+	/* Critical section: bank 1 access */
+	spin_lock_irqsave(&chip->lock, flags);
+	outb(HPBLKSEL_1, chip->io + HP_BLKS);
+	memset_io(chip->mappedbase, 0, 0x8000);
+	outb(HPBLKSEL_0, chip->io + HP_BLKS);
+	spin_unlock_irqrestore(&chip->lock, flags);
+
+	/* Digital audio play queue */
+	chip->DAPQ = chip->mappedbase + DAPQ_OFFSET;
+	snd_msnd_init_queue(chip->DAPQ, DAPQ_DATA_BUFF, DAPQ_BUFF_SIZE);
+
+	/* Digital audio record queue */
+	chip->DARQ = chip->mappedbase + DARQ_OFFSET;
+	snd_msnd_init_queue(chip->DARQ, DARQ_DATA_BUFF, DARQ_BUFF_SIZE);
+
+	/* MIDI out queue */
+	chip->MODQ = chip->mappedbase + MODQ_OFFSET;
+	snd_msnd_init_queue(chip->MODQ, MODQ_DATA_BUFF, MODQ_BUFF_SIZE);
+
+	/* MIDI in queue */
+	chip->MIDQ = chip->mappedbase + MIDQ_OFFSET;
+	snd_msnd_init_queue(chip->MIDQ, MIDQ_DATA_BUFF, MIDQ_BUFF_SIZE);
+
+	/* DSP -> host message queue */
+	chip->DSPQ = chip->mappedbase + DSPQ_OFFSET;
+	snd_msnd_init_queue(chip->DSPQ, DSPQ_DATA_BUFF, DSPQ_BUFF_SIZE);
+
+	/* Setup some DSP values */
+#ifndef MSND_CLASSIC
+	writew(1, chip->SMA + SMA_wCurrPlayFormat);
+	writew(chip->play_sample_size, chip->SMA + SMA_wCurrPlaySampleSize);
+	writew(chip->play_channels, chip->SMA + SMA_wCurrPlayChannels);
+	writew(chip->play_sample_rate, chip->SMA + SMA_wCurrPlaySampleRate);
+#endif
+	writew(chip->play_sample_rate, chip->SMA + SMA_wCalFreqAtoD);
+	writew(mastVolLeft, chip->SMA + SMA_wCurrMastVolLeft);
+	writew(mastVolRight, chip->SMA + SMA_wCurrMastVolRight);
+#ifndef MSND_CLASSIC
+	writel(0x00010000, chip->SMA + SMA_dwCurrPlayPitch);
+	writel(0x00000001, chip->SMA + SMA_dwCurrPlayRate);
+#endif
+	writew(0x303, chip->SMA + SMA_wCurrInputTagBits);
+
+	initted = 1;
+
+	return 0;
+}
+
+
+static int upload_dsp_code(struct snd_card *card)
+{
+	struct snd_msnd *chip = card->private_data;
+	const struct firmware *init_fw = NULL, *perm_fw = NULL;
+	int err;
+
+	outb(HPBLKSEL_0, chip->io + HP_BLKS);
+
+	err = request_firmware(&init_fw, INITCODEFILE, card->dev);
+	if (err < 0) {
+		printk(KERN_ERR LOGNAME ": Error loading " INITCODEFILE);
+		goto cleanup1;
+	}
+	err = request_firmware(&perm_fw, PERMCODEFILE, card->dev);
+	if (err < 0) {
+		printk(KERN_ERR LOGNAME ": Error loading " PERMCODEFILE);
+		goto cleanup;
+	}
+
+	memcpy_toio(chip->mappedbase, perm_fw->data, perm_fw->size);
+	if (snd_msnd_upload_host(chip, init_fw->data, init_fw->size) < 0) {
+		printk(KERN_WARNING LOGNAME ": Error uploading to DSP\n");
+		err = -ENODEV;
+		goto cleanup;
+	}
+	printk(KERN_INFO LOGNAME ": DSP firmware uploaded\n");
+	err = 0;
+
+cleanup:
+	release_firmware(perm_fw);
+cleanup1:
+	release_firmware(init_fw);
+	return err;
+}
+
+#ifdef MSND_CLASSIC
+static void reset_proteus(struct snd_msnd *chip)
+{
+	outb(HPPRORESET_ON, chip->io + HP_PROR);
+	msleep(TIME_PRO_RESET);
+	outb(HPPRORESET_OFF, chip->io + HP_PROR);
+	msleep(TIME_PRO_RESET_DONE);
+}
+#endif
+
+static int snd_msnd_initialize(struct snd_card *card)
+{
+	struct snd_msnd *chip = card->private_data;
+	int err, timeout;
+
+#ifdef MSND_CLASSIC
+	outb(HPWAITSTATE_0, chip->io + HP_WAIT);
+	outb(HPBITMODE_16, chip->io + HP_BITM);
+
+	reset_proteus(chip);
+#endif
+	err = snd_msnd_init_sma(chip);
+	if (err < 0) {
+		printk(KERN_WARNING LOGNAME ": Cannot initialize SMA\n");
+		return err;
+	}
+
+	err = snd_msnd_reset_dsp(chip->io, NULL);
+	if (err < 0)
+		return err;
+
+	err = upload_dsp_code(card);
+	if (err < 0) {
+		printk(KERN_WARNING LOGNAME ": Cannot upload DSP code\n");
+		return err;
+	}
+
+	timeout = 200;
+
+	while (readw(chip->mappedbase)) {
+		msleep(1);
+		if (!timeout--) {
+			snd_printd(KERN_ERR LOGNAME ": DSP reset timeout\n");
+			return -EIO;
+		}
+	}
+
+	snd_msndmix_setup(chip);
+	return 0;
+}
+
+static int snd_msnd_dsp_full_reset(struct snd_card *card)
+{
+	struct snd_msnd *chip = card->private_data;
+	int rv;
+
+	if (test_bit(F_RESETTING, &chip->flags) || ++chip->nresets > 10)
+		return 0;
+
+	set_bit(F_RESETTING, &chip->flags);
+	snd_msnd_dsp_halt(chip, NULL);	/* Unconditionally halt */
+
+	rv = snd_msnd_initialize(card);
+	if (rv)
+		printk(KERN_WARNING LOGNAME ": DSP reset failed\n");
+	snd_msndmix_force_recsrc(chip, 0);
+	clear_bit(F_RESETTING, &chip->flags);
+	return rv;
+}
+
+static int snd_msnd_dev_free(struct snd_device *device)
+{
+	snd_printdd("snd_msnd_chip_free()\n");
+	return 0;
+}
+
+static int snd_msnd_send_dsp_cmd_chk(struct snd_msnd *chip, u8 cmd)
+{
+	if (snd_msnd_send_dsp_cmd(chip, cmd) == 0)
+		return 0;
+	snd_msnd_dsp_full_reset(chip->card);
+	return snd_msnd_send_dsp_cmd(chip, cmd);
+}
+
+static int __devinit snd_msnd_calibrate_adc(struct snd_msnd *chip, u16 srate)
+{
+	snd_printdd("snd_msnd_calibrate_adc(%i)\n", srate);
+	writew(srate, chip->SMA + SMA_wCalFreqAtoD);
+	if (chip->calibrate_signal == 0)
+		writew(readw(chip->SMA + SMA_wCurrHostStatusFlags)
+		       | 0x0001, chip->SMA + SMA_wCurrHostStatusFlags);
+	else
+		writew(readw(chip->SMA + SMA_wCurrHostStatusFlags)
+		       & ~0x0001, chip->SMA + SMA_wCurrHostStatusFlags);
+	if (snd_msnd_send_word(chip, 0, 0, HDEXAR_CAL_A_TO_D) == 0 &&
+	    snd_msnd_send_dsp_cmd_chk(chip, HDEX_AUX_REQ) == 0) {
+		schedule_timeout_interruptible(msecs_to_jiffies(333));
+		return 0;
+	}
+	printk(KERN_WARNING LOGNAME ": ADC calibration failed\n");
+	return -EIO;
+}
+
+/*
+ * ALSA callback function, called when attempting to open the MIDI device.
+ */
+static int snd_msnd_mpu401_open(struct snd_mpu401 *mpu)
+{
+	snd_msnd_enable_irq(mpu->private_data);
+	snd_msnd_send_dsp_cmd(mpu->private_data, HDEX_MIDI_IN_START);
+	return 0;
+}
+
+static void snd_msnd_mpu401_close(struct snd_mpu401 *mpu)
+{
+	snd_msnd_send_dsp_cmd(mpu->private_data, HDEX_MIDI_IN_STOP);
+	snd_msnd_disable_irq(mpu->private_data);
+}
+
+static long mpu_io[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ;
+
+static int __devinit snd_msnd_attach(struct snd_card *card)
+{
+	struct snd_msnd *chip = card->private_data;
+	int err;
+	static struct snd_device_ops ops = {
+		.dev_free =      snd_msnd_dev_free,
+		};
+
+	err = request_irq(chip->irq, snd_msnd_interrupt, 0, card->shortname,
+			  chip);
+	if (err < 0) {
+		printk(KERN_ERR LOGNAME ": Couldn't grab IRQ %d\n", chip->irq);
+		return err;
+	}
+	request_region(chip->io, DSP_NUMIO, card->shortname);
+
+	if (!request_mem_region(chip->base, BUFFSIZE, card->shortname)) {
+		printk(KERN_ERR LOGNAME
+			": unable to grab memory region 0x%lx-0x%lx\n",
+			chip->base, chip->base + BUFFSIZE - 1);
+		release_region(chip->io, DSP_NUMIO);
+		free_irq(chip->irq, chip);
+		return -EBUSY;
+	}
+	chip->mappedbase = ioremap_nocache(chip->base, 0x8000);
+	if (!chip->mappedbase) {
+		printk(KERN_ERR LOGNAME
+			": unable to map memory region 0x%lx-0x%lx\n",
+			chip->base, chip->base + BUFFSIZE - 1);
+		err = -EIO;
+		goto err_release_region;
+	}
+
+	err = snd_msnd_dsp_full_reset(card);
+	if (err < 0)
+		goto err_release_region;
+
+	/* Register device */
+	err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+	if (err < 0)
+		goto err_release_region;
+
+	err = snd_msnd_pcm(card, 0, NULL);
+	if (err < 0) {
+		printk(KERN_ERR LOGNAME ": error creating new PCM device\n");
+		goto err_release_region;
+	}
+
+	err = snd_msndmix_new(card);
+	if (err < 0) {
+		printk(KERN_ERR LOGNAME ": error creating new Mixer device\n");
+		goto err_release_region;
+	}
+
+
+	if (mpu_io[0] != SNDRV_AUTO_PORT) {
+		struct snd_mpu401 *mpu;
+
+		err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401,
+					  mpu_io[0],
+					  MPU401_MODE_INPUT |
+					  MPU401_MODE_OUTPUT,
+					  mpu_irq[0], IRQF_DISABLED,
+					  &chip->rmidi);
+		if (err < 0) {
+			printk(KERN_ERR LOGNAME
+				": error creating new Midi device\n");
+			goto err_release_region;
+		}
+		mpu = chip->rmidi->private_data;
+
+		mpu->open_input = snd_msnd_mpu401_open;
+		mpu->close_input = snd_msnd_mpu401_close;
+		mpu->private_data = chip;
+	}
+
+	disable_irq(chip->irq);
+	snd_msnd_calibrate_adc(chip, chip->play_sample_rate);
+	snd_msndmix_force_recsrc(chip, 0);
+
+	err = snd_card_register(card);
+	if (err < 0)
+		goto err_release_region;
+
+	return 0;
+
+err_release_region:
+	if (chip->mappedbase)
+		iounmap(chip->mappedbase);
+	release_mem_region(chip->base, BUFFSIZE);
+	release_region(chip->io, DSP_NUMIO);
+	free_irq(chip->irq, chip);
+	return err;
+}
+
+
+static void __devexit snd_msnd_unload(struct snd_card *card)
+{
+	struct snd_msnd *chip = card->private_data;
+
+	iounmap(chip->mappedbase);
+	release_mem_region(chip->base, BUFFSIZE);
+	release_region(chip->io, DSP_NUMIO);
+	free_irq(chip->irq, chip);
+	snd_card_free(card);
+}
+
+#ifndef MSND_CLASSIC
+
+/* Pinnacle/Fiji Logical Device Configuration */
+
+static int __devinit snd_msnd_write_cfg(int cfg, int reg, int value)
+{
+	outb(reg, cfg);
+	outb(value, cfg + 1);
+	if (value != inb(cfg + 1)) {
+		printk(KERN_ERR LOGNAME ": snd_msnd_write_cfg: I/O error\n");
+		return -EIO;
+	}
+	return 0;
+}
+
+static int __devinit snd_msnd_write_cfg_io0(int cfg, int num, u16 io)
+{
+	if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+		return -EIO;
+	if (snd_msnd_write_cfg(cfg, IREG_IO0_BASEHI, HIBYTE(io)))
+		return -EIO;
+	if (snd_msnd_write_cfg(cfg, IREG_IO0_BASELO, LOBYTE(io)))
+		return -EIO;
+	return 0;
+}
+
+static int __devinit snd_msnd_write_cfg_io1(int cfg, int num, u16 io)
+{
+	if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+		return -EIO;
+	if (snd_msnd_write_cfg(cfg, IREG_IO1_BASEHI, HIBYTE(io)))
+		return -EIO;
+	if (snd_msnd_write_cfg(cfg, IREG_IO1_BASELO, LOBYTE(io)))
+		return -EIO;
+	return 0;
+}
+
+static int __devinit snd_msnd_write_cfg_irq(int cfg, int num, u16 irq)
+{
+	if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+		return -EIO;
+	if (snd_msnd_write_cfg(cfg, IREG_IRQ_NUMBER, LOBYTE(irq)))
+		return -EIO;
+	if (snd_msnd_write_cfg(cfg, IREG_IRQ_TYPE, IRQTYPE_EDGE))
+		return -EIO;
+	return 0;
+}
+
+static int __devinit snd_msnd_write_cfg_mem(int cfg, int num, int mem)
+{
+	u16 wmem;
+
+	mem >>= 8;
+	wmem = (u16)(mem & 0xfff);
+	if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+		return -EIO;
+	if (snd_msnd_write_cfg(cfg, IREG_MEMBASEHI, HIBYTE(wmem)))
+		return -EIO;
+	if (snd_msnd_write_cfg(cfg, IREG_MEMBASELO, LOBYTE(wmem)))
+		return -EIO;
+	if (wmem && snd_msnd_write_cfg(cfg, IREG_MEMCONTROL,
+				       MEMTYPE_HIADDR | MEMTYPE_16BIT))
+		return -EIO;
+	return 0;
+}
+
+static int __devinit snd_msnd_activate_logical(int cfg, int num)
+{
+	if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+		return -EIO;
+	if (snd_msnd_write_cfg(cfg, IREG_ACTIVATE, LD_ACTIVATE))
+		return -EIO;
+	return 0;
+}
+
+static int __devinit snd_msnd_write_cfg_logical(int cfg, int num, u16 io0,
+						u16 io1, u16 irq, int mem)
+{
+	if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num))
+		return -EIO;
+	if (snd_msnd_write_cfg_io0(cfg, num, io0))
+		return -EIO;
+	if (snd_msnd_write_cfg_io1(cfg, num, io1))
+		return -EIO;
+	if (snd_msnd_write_cfg_irq(cfg, num, irq))
+		return -EIO;
+	if (snd_msnd_write_cfg_mem(cfg, num, mem))
+		return -EIO;
+	if (snd_msnd_activate_logical(cfg, num))
+		return -EIO;
+	return 0;
+}
+
+static int __devinit snd_msnd_pinnacle_cfg_reset(int cfg)
+{
+	int i;
+
+	/* Reset devices if told to */
+	printk(KERN_INFO LOGNAME ": Resetting all devices\n");
+	for (i = 0; i < 4; ++i)
+		if (snd_msnd_write_cfg_logical(cfg, i, 0, 0, 0, 0))
+			return -EIO;
+
+	return 0;
+}
+#endif
+
+static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;	/* Index 0-MAX */
+static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;	/* ID for this card */
+
+module_param_array(index, int, NULL, S_IRUGO);
+MODULE_PARM_DESC(index, "Index value for msnd_pinnacle soundcard.");
+module_param_array(id, charp, NULL, S_IRUGO);
+MODULE_PARM_DESC(id, "ID string for msnd_pinnacle soundcard.");
+
+static long io[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ;
+static long mem[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+
+static long cfg[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+
+#ifndef MSND_CLASSIC
+/* Extra Peripheral Configuration (Default: Disable) */
+static long ide_io0[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+static long ide_io1[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+static int ide_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ;
+
+static long joystick_io[SNDRV_CARDS] = SNDRV_DEFAULT_PORT;
+/* If we have the digital daugherboard... */
+static int digital[SNDRV_CARDS];
+
+/* Extra Peripheral Configuration */
+static int reset[SNDRV_CARDS];
+#endif
+
+static int write_ndelay[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = 1 };
+
+static int calibrate_signal;
+
+#ifdef CONFIG_PNP
+static int isapnp[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+module_param_array(isapnp, bool, NULL, 0444);
+MODULE_PARM_DESC(isapnp, "ISA PnP detection for specified soundcard.");
+#define has_isapnp(x) isapnp[x]
+#else
+#define has_isapnp(x) 0
+#endif
+
+MODULE_AUTHOR("Karsten Wiese <annabellesgarden@yahoo.de>");
+MODULE_DESCRIPTION("Turtle Beach " LONGNAME " Linux Driver");
+MODULE_LICENSE("GPL");
+MODULE_FIRMWARE(INITCODEFILE);
+MODULE_FIRMWARE(PERMCODEFILE);
+
+module_param_array(io, long, NULL, S_IRUGO);
+MODULE_PARM_DESC(io, "IO port #");
+module_param_array(irq, int, NULL, S_IRUGO);
+module_param_array(mem, long, NULL, S_IRUGO);
+module_param_array(write_ndelay, int, NULL, S_IRUGO);
+module_param(calibrate_signal, int, S_IRUGO);
+#ifndef MSND_CLASSIC
+module_param_array(digital, int, NULL, S_IRUGO);
+module_param_array(cfg, long, NULL, S_IRUGO);
+module_param_array(reset, int, 0, S_IRUGO);
+module_param_array(mpu_io, long, NULL, S_IRUGO);
+module_param_array(mpu_irq, int, NULL, S_IRUGO);
+module_param_array(ide_io0, long, NULL, S_IRUGO);
+module_param_array(ide_io1, long, NULL, S_IRUGO);
+module_param_array(ide_irq, int, NULL, S_IRUGO);
+module_param_array(joystick_io, long, NULL, S_IRUGO);
+#endif
+
+
+static int __devinit snd_msnd_isa_match(struct device *pdev, unsigned int i)
+{
+	if (io[i] == SNDRV_AUTO_PORT)
+		return 0;
+
+	if (irq[i] == SNDRV_AUTO_PORT || mem[i] == SNDRV_AUTO_PORT) {
+		printk(KERN_WARNING LOGNAME ": io, irq and mem must be set\n");
+		return 0;
+	}
+
+#ifdef MSND_CLASSIC
+	if (!(io[i] == 0x290 ||
+	      io[i] == 0x260 ||
+	      io[i] == 0x250 ||
+	      io[i] == 0x240 ||
+	      io[i] == 0x230 ||
+	      io[i] == 0x220 ||
+	      io[i] == 0x210 ||
+	      io[i] == 0x3e0)) {
+		printk(KERN_ERR LOGNAME ": \"io\" - DSP I/O base must be set "
+			" to 0x210, 0x220, 0x230, 0x240, 0x250, 0x260, 0x290, "
+			"or 0x3E0\n");
+		return 0;
+	}
+#else
+	if (io[i] < 0x100 || io[i] > 0x3e0 || (io[i] % 0x10) != 0) {
+		printk(KERN_ERR LOGNAME
+			": \"io\" - DSP I/O base must within the range 0x100 "
+			"to 0x3E0 and must be evenly divisible by 0x10\n");
+		return 0;
+	}
+#endif /* MSND_CLASSIC */
+
+	if (!(irq[i] == 5 ||
+	      irq[i] == 7 ||
+	      irq[i] == 9 ||
+	      irq[i] == 10 ||
+	      irq[i] == 11 ||
+	      irq[i] == 12)) {
+		printk(KERN_ERR LOGNAME
+			": \"irq\" - must be set to 5, 7, 9, 10, 11 or 12\n");
+		return 0;
+	}
+
+	if (!(mem[i] == 0xb0000 ||
+	      mem[i] == 0xc8000 ||
+	      mem[i] == 0xd0000 ||
+	      mem[i] == 0xd8000 ||
+	      mem[i] == 0xe0000 ||
+	      mem[i] == 0xe8000)) {
+		printk(KERN_ERR LOGNAME ": \"mem\" - must be set to "
+		       "0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or "
+		       "0xe8000\n");
+		return 0;
+	}
+
+#ifndef MSND_CLASSIC
+	if (cfg[i] == SNDRV_AUTO_PORT) {
+		printk(KERN_INFO LOGNAME ": Assuming PnP mode\n");
+	} else if (cfg[i] != 0x250 && cfg[i] != 0x260 && cfg[i] != 0x270) {
+		printk(KERN_INFO LOGNAME
+			": Config port must be 0x250, 0x260 or 0x270 "
+			"(or unspecified for PnP mode)\n");
+		return 0;
+	}
+#endif /* MSND_CLASSIC */
+
+	return 1;
+}
+
+static int __devinit snd_msnd_isa_probe(struct device *pdev, unsigned int idx)
+{
+	int err;
+	struct snd_card *card;
+	struct snd_msnd *chip;
+
+	if (has_isapnp(idx) || cfg[idx] == SNDRV_AUTO_PORT) {
+		printk(KERN_INFO LOGNAME ": Assuming PnP mode\n");
+		return -ENODEV;
+	}
+
+	err = snd_card_create(index[idx], id[idx], THIS_MODULE,
+			      sizeof(struct snd_msnd), &card);
+	if (err < 0)
+		return err;
+
+	snd_card_set_dev(card, pdev);
+	chip = card->private_data;
+	chip->card = card;
+
+#ifdef MSND_CLASSIC
+	switch (irq[idx]) {
+	case 5:
+		chip->irqid = HPIRQ_5; break;
+	case 7:
+		chip->irqid = HPIRQ_7; break;
+	case 9:
+		chip->irqid = HPIRQ_9; break;
+	case 10:
+		chip->irqid = HPIRQ_10; break;
+	case 11:
+		chip->irqid = HPIRQ_11; break;
+	case 12:
+		chip->irqid = HPIRQ_12; break;
+	}
+
+	switch (mem[idx]) {
+	case 0xb0000:
+		chip->memid = HPMEM_B000; break;
+	case 0xc8000:
+		chip->memid = HPMEM_C800; break;
+	case 0xd0000:
+		chip->memid = HPMEM_D000; break;
+	case 0xd8000:
+		chip->memid = HPMEM_D800; break;
+	case 0xe0000:
+		chip->memid = HPMEM_E000; break;
+	case 0xe8000:
+		chip->memid = HPMEM_E800; break;
+	}
+#else
+	printk(KERN_INFO LOGNAME ": Non-PnP mode: configuring at port 0x%lx\n",
+			cfg[idx]);
+
+	if (!request_region(cfg[idx], 2, "Pinnacle/Fiji Config")) {
+		printk(KERN_ERR LOGNAME ": Config port 0x%lx conflict\n",
+			   cfg[idx]);
+		snd_card_free(card);
+		return -EIO;
+	}
+	if (reset[idx])
+		if (snd_msnd_pinnacle_cfg_reset(cfg[idx])) {
+			err = -EIO;
+			goto cfg_error;
+		}
+
+	/* DSP */
+	err = snd_msnd_write_cfg_logical(cfg[idx], 0,
+					 io[idx], 0,
+					 irq[idx], mem[idx]);
+
+	if (err)
+		goto cfg_error;
+
+	/* The following are Pinnacle specific */
+
+	/* MPU */
+	if (mpu_io[idx] != SNDRV_AUTO_PORT
+	    && mpu_irq[idx] != SNDRV_AUTO_IRQ) {
+		printk(KERN_INFO LOGNAME
+		       ": Configuring MPU to I/O 0x%lx IRQ %d\n",
+		       mpu_io[idx], mpu_irq[idx]);
+		err = snd_msnd_write_cfg_logical(cfg[idx], 1,
+						 mpu_io[idx], 0,
+						 mpu_irq[idx], 0);
+
+		if (err)
+			goto cfg_error;
+	}
+
+	/* IDE */
+	if (ide_io0[idx] != SNDRV_AUTO_PORT
+	    && ide_io1[idx] != SNDRV_AUTO_PORT
+	    && ide_irq[idx] != SNDRV_AUTO_IRQ) {
+		printk(KERN_INFO LOGNAME
+		       ": Configuring IDE to I/O 0x%lx, 0x%lx IRQ %d\n",
+		       ide_io0[idx], ide_io1[idx], ide_irq[idx]);
+		err = snd_msnd_write_cfg_logical(cfg[idx], 2,
+						 ide_io0[idx], ide_io1[idx],
+						 ide_irq[idx], 0);
+
+		if (err)
+			goto cfg_error;
+	}
+
+	/* Joystick */
+	if (joystick_io[idx] != SNDRV_AUTO_PORT) {
+		printk(KERN_INFO LOGNAME
+		       ": Configuring joystick to I/O 0x%lx\n",
+		       joystick_io[idx]);
+		err = snd_msnd_write_cfg_logical(cfg[idx], 3,
+						 joystick_io[idx], 0,
+						 0, 0);
+
+		if (err)
+			goto cfg_error;
+	}
+	release_region(cfg[idx], 2);
+
+#endif /* MSND_CLASSIC */
+
+	set_default_audio_parameters(chip);
+#ifdef MSND_CLASSIC
+	chip->type = msndClassic;
+#else
+	chip->type = msndPinnacle;
+#endif
+	chip->io = io[idx];
+	chip->irq = irq[idx];
+	chip->base = mem[idx];
+
+	chip->calibrate_signal = calibrate_signal ? 1 : 0;
+	chip->recsrc = 0;
+	chip->dspq_data_buff = DSPQ_DATA_BUFF;
+	chip->dspq_buff_size = DSPQ_BUFF_SIZE;
+	if (write_ndelay[idx])
+		clear_bit(F_DISABLE_WRITE_NDELAY, &chip->flags);
+	else
+		set_bit(F_DISABLE_WRITE_NDELAY, &chip->flags);
+#ifndef MSND_CLASSIC
+	if (digital[idx])
+		set_bit(F_HAVEDIGITAL, &chip->flags);
+#endif
+	spin_lock_init(&chip->lock);
+	err = snd_msnd_probe(card);
+	if (err < 0) {
+		printk(KERN_ERR LOGNAME ": Probe failed\n");
+		snd_card_free(card);
+		return err;
+	}
+
+	err = snd_msnd_attach(card);
+	if (err < 0) {
+		printk(KERN_ERR LOGNAME ": Attach failed\n");
+		snd_card_free(card);
+		return err;
+	}
+	dev_set_drvdata(pdev, card);
+
+	return 0;
+
+#ifndef MSND_CLASSIC
+cfg_error:
+	release_region(cfg[idx], 2);
+	snd_card_free(card);
+	return err;
+#endif
+}
+
+static int __devexit snd_msnd_isa_remove(struct device *pdev, unsigned int dev)
+{
+	snd_msnd_unload(dev_get_drvdata(pdev));
+	dev_set_drvdata(pdev, NULL);
+	return 0;
+}
+
+#define DEV_NAME "msnd-pinnacle"
+
+static struct isa_driver snd_msnd_driver = {
+	.match		= snd_msnd_isa_match,
+	.probe		= snd_msnd_isa_probe,
+	.remove		= __devexit_p(snd_msnd_isa_remove),
+	/* FIXME: suspend, resume */
+	.driver		= {
+		.name	= DEV_NAME
+	},
+};
+
+#ifdef CONFIG_PNP
+static int __devinit snd_msnd_pnp_detect(struct pnp_card_link *pcard,
+					 const struct pnp_card_device_id *pid)
+{
+	static int idx;
+	struct pnp_dev *pnp_dev;
+	struct pnp_dev *mpu_dev;
+	struct snd_card *card;
+	struct snd_msnd *chip;
+	int ret;
+
+	for ( ; idx < SNDRV_CARDS; idx++) {
+		if (has_isapnp(idx))
+			break;
+	}
+	if (idx >= SNDRV_CARDS)
+		return -ENODEV;
+
+	/*
+	 * Check that we still have room for another sound card ...
+	 */
+	pnp_dev = pnp_request_card_device(pcard, pid->devs[0].id, NULL);
+	if (!pnp_dev)
+		return -ENODEV;
+
+	mpu_dev = pnp_request_card_device(pcard, pid->devs[1].id, NULL);
+	if (!mpu_dev)
+		return -ENODEV;
+
+	if (!pnp_is_active(pnp_dev) && pnp_activate_dev(pnp_dev) < 0) {
+		printk(KERN_INFO "msnd_pinnacle: device is inactive\n");
+		return -EBUSY;
+	}
+
+	if (!pnp_is_active(mpu_dev) && pnp_activate_dev(mpu_dev) < 0) {
+		printk(KERN_INFO "msnd_pinnacle: MPU device is inactive\n");
+		return -EBUSY;
+	}
+
+	/*
+	 * Create a new ALSA sound card entry, in anticipation
+	 * of detecting our hardware ...
+	 */
+	ret = snd_card_create(index[idx], id[idx], THIS_MODULE,
+			      sizeof(struct snd_msnd), &card);
+	if (ret < 0)
+		return ret;
+
+	chip = card->private_data;
+	chip->card = card;
+	snd_card_set_dev(card, &pcard->card->dev);
+
+	/*
+	 * Read the correct parameters off the ISA PnP bus ...
+	 */
+	io[idx] = pnp_port_start(pnp_dev, 0);
+	irq[idx] = pnp_irq(pnp_dev, 0);
+	mem[idx] = pnp_mem_start(pnp_dev, 0);
+	mpu_io[idx] = pnp_port_start(mpu_dev, 0);
+	mpu_irq[idx] = pnp_irq(mpu_dev, 0);
+
+	set_default_audio_parameters(chip);
+#ifdef MSND_CLASSIC
+	chip->type = msndClassic;
+#else
+	chip->type = msndPinnacle;
+#endif
+	chip->io = io[idx];
+	chip->irq = irq[idx];
+	chip->base = mem[idx];
+
+	chip->calibrate_signal = calibrate_signal ? 1 : 0;
+	chip->recsrc = 0;
+	chip->dspq_data_buff = DSPQ_DATA_BUFF;
+	chip->dspq_buff_size = DSPQ_BUFF_SIZE;
+	if (write_ndelay[idx])
+		clear_bit(F_DISABLE_WRITE_NDELAY, &chip->flags);
+	else
+		set_bit(F_DISABLE_WRITE_NDELAY, &chip->flags);
+#ifndef MSND_CLASSIC
+	if (digital[idx])
+		set_bit(F_HAVEDIGITAL, &chip->flags);
+#endif
+	spin_lock_init(&chip->lock);
+	ret = snd_msnd_probe(card);
+	if (ret < 0) {
+		printk(KERN_ERR LOGNAME ": Probe failed\n");
+		goto _release_card;
+	}
+
+	ret = snd_msnd_attach(card);
+	if (ret < 0) {
+		printk(KERN_ERR LOGNAME ": Attach failed\n");
+		goto _release_card;
+	}
+
+	pnp_set_card_drvdata(pcard, card);
+	++idx;
+	return 0;
+
+_release_card:
+	snd_card_free(card);
+	return ret;
+}
+
+static void __devexit snd_msnd_pnp_remove(struct pnp_card_link *pcard)
+{
+	snd_msnd_unload(pnp_get_card_drvdata(pcard));
+	pnp_set_card_drvdata(pcard, NULL);
+}
+
+static int isa_registered;
+static int pnp_registered;
+
+static struct pnp_card_device_id msnd_pnpids[] = {
+	/* Pinnacle PnP */
+	{ .id = "BVJ0440", .devs = { { "TBS0000" }, { "TBS0001" } } },
+	{ .id = "" }	/* end */
+};
+
+MODULE_DEVICE_TABLE(pnp_card, msnd_pnpids);
+
+static struct pnp_card_driver msnd_pnpc_driver = {
+	.flags = PNP_DRIVER_RES_DO_NOT_CHANGE,
+	.name = "msnd_pinnacle",
+	.id_table = msnd_pnpids,
+	.probe = snd_msnd_pnp_detect,
+	.remove = __devexit_p(snd_msnd_pnp_remove),
+};
+#endif /* CONFIG_PNP */
+
+static int __init snd_msnd_init(void)
+{
+	int err;
+
+	err = isa_register_driver(&snd_msnd_driver, SNDRV_CARDS);
+#ifdef CONFIG_PNP
+	if (!err)
+		isa_registered = 1;
+
+	err = pnp_register_card_driver(&msnd_pnpc_driver);
+	if (!err)
+		pnp_registered = 1;
+
+	if (isa_registered)
+		err = 0;
+#endif
+	return err;
+}
+
+static void __exit snd_msnd_exit(void)
+{
+#ifdef CONFIG_PNP
+	if (pnp_registered)
+		pnp_unregister_card_driver(&msnd_pnpc_driver);
+	if (isa_registered)
+#endif
+		isa_unregister_driver(&snd_msnd_driver);
+}
+
+module_init(snd_msnd_init);
+module_exit(snd_msnd_exit);
+
diff --git a/sound/isa/msnd/msnd_pinnacle.h b/sound/isa/msnd/msnd_pinnacle.h
new file mode 100644
index 000000000000..48318d1ee340
--- /dev/null
+++ b/sound/isa/msnd/msnd_pinnacle.h
@@ -0,0 +1,181 @@
+/*********************************************************************
+ *
+ * msnd_pinnacle.h
+ *
+ * Turtle Beach MultiSound Sound Card Driver for Linux
+ *
+ * Some parts of this header file were derived from the Turtle Beach
+ * MultiSound Driver Development Kit.
+ *
+ * Copyright (C) 1998 Andrew Veliath
+ * Copyright (C) 1993 Turtle Beach Systems, Inc.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ ********************************************************************/
+#ifndef __MSND_PINNACLE_H
+#define __MSND_PINNACLE_H
+
+#define DSP_NUMIO				0x08
+
+#define IREG_LOGDEVICE				0x07
+#define IREG_ACTIVATE				0x30
+#define LD_ACTIVATE				0x01
+#define LD_DISACTIVATE				0x00
+#define IREG_EECONTROL				0x3F
+#define IREG_MEMBASEHI				0x40
+#define IREG_MEMBASELO				0x41
+#define IREG_MEMCONTROL				0x42
+#define IREG_MEMRANGEHI				0x43
+#define IREG_MEMRANGELO				0x44
+#define MEMTYPE_8BIT				0x00
+#define MEMTYPE_16BIT				0x02
+#define MEMTYPE_RANGE				0x00
+#define MEMTYPE_HIADDR				0x01
+#define IREG_IO0_BASEHI				0x60
+#define IREG_IO0_BASELO				0x61
+#define IREG_IO1_BASEHI				0x62
+#define IREG_IO1_BASELO				0x63
+#define IREG_IRQ_NUMBER				0x70
+#define IREG_IRQ_TYPE				0x71
+#define IRQTYPE_HIGH				0x02
+#define IRQTYPE_LOW				0x00
+#define IRQTYPE_LEVEL				0x01
+#define IRQTYPE_EDGE				0x00
+
+#define	HP_DSPR					0x04
+#define	HP_BLKS					0x04
+
+#define HPDSPRESET_OFF				2
+#define HPDSPRESET_ON				0
+
+#define HPBLKSEL_0				2
+#define HPBLKSEL_1				3
+
+#define	HIMT_DAT_OFF				0x03
+
+#define	HIDSP_PLAY_UNDER			0x00
+#define	HIDSP_INT_PLAY_UNDER			0x01
+#define	HIDSP_SSI_TX_UNDER  			0x02
+#define HIDSP_RECQ_OVERFLOW			0x08
+#define HIDSP_INT_RECORD_OVER			0x09
+#define HIDSP_SSI_RX_OVERFLOW			0x0a
+
+#define	HIDSP_MIDI_IN_OVER			0x10
+
+#define	HIDSP_MIDI_FRAME_ERR			0x11
+#define	HIDSP_MIDI_PARITY_ERR			0x12
+#define	HIDSP_MIDI_OVERRUN_ERR			0x13
+
+#define HIDSP_INPUT_CLIPPING			0x20
+#define	HIDSP_MIX_CLIPPING			0x30
+#define HIDSP_DAT_IN_OFF			0x21
+
+#define TIME_PRO_RESET_DONE			0x028A
+#define TIME_PRO_SYSEX				0x001E
+#define TIME_PRO_RESET				0x0032
+
+#define DAR_BUFF_SIZE				0x1000
+
+#define MIDQ_BUFF_SIZE				0x800
+#define DSPQ_BUFF_SIZE				0x5A0
+
+#define DSPQ_DATA_BUFF				0x7860
+
+#define MOP_WAVEHDR				0
+#define MOP_EXTOUT				1
+#define MOP_HWINIT				0xfe
+#define MOP_NONE				0xff
+#define MOP_MAX					1
+
+#define MIP_EXTIN				0
+#define MIP_WAVEHDR				1
+#define MIP_HWINIT				0xfe
+#define MIP_MAX					1
+
+/* Pinnacle/Fiji SMA Common Data */
+#define SMA_wCurrPlayBytes			0x0000
+#define SMA_wCurrRecordBytes			0x0002
+#define SMA_wCurrPlayVolLeft			0x0004
+#define SMA_wCurrPlayVolRight			0x0006
+#define SMA_wCurrInVolLeft			0x0008
+#define SMA_wCurrInVolRight			0x000a
+#define SMA_wCurrMHdrVolLeft			0x000c
+#define SMA_wCurrMHdrVolRight			0x000e
+#define SMA_dwCurrPlayPitch			0x0010
+#define SMA_dwCurrPlayRate			0x0014
+#define SMA_wCurrMIDIIOPatch			0x0018
+#define SMA_wCurrPlayFormat			0x001a
+#define SMA_wCurrPlaySampleSize			0x001c
+#define SMA_wCurrPlayChannels			0x001e
+#define SMA_wCurrPlaySampleRate			0x0020
+#define SMA_wCurrRecordFormat			0x0022
+#define SMA_wCurrRecordSampleSize		0x0024
+#define SMA_wCurrRecordChannels			0x0026
+#define SMA_wCurrRecordSampleRate		0x0028
+#define SMA_wCurrDSPStatusFlags			0x002a
+#define SMA_wCurrHostStatusFlags		0x002c
+#define SMA_wCurrInputTagBits			0x002e
+#define SMA_wCurrLeftPeak			0x0030
+#define SMA_wCurrRightPeak			0x0032
+#define SMA_bMicPotPosLeft			0x0034
+#define SMA_bMicPotPosRight			0x0035
+#define SMA_bMicPotMaxLeft			0x0036
+#define SMA_bMicPotMaxRight			0x0037
+#define SMA_bInPotPosLeft			0x0038
+#define SMA_bInPotPosRight			0x0039
+#define SMA_bAuxPotPosLeft			0x003a
+#define SMA_bAuxPotPosRight			0x003b
+#define SMA_bInPotMaxLeft			0x003c
+#define SMA_bInPotMaxRight			0x003d
+#define SMA_bAuxPotMaxLeft			0x003e
+#define SMA_bAuxPotMaxRight			0x003f
+#define SMA_bInPotMaxMethod			0x0040
+#define SMA_bAuxPotMaxMethod			0x0041
+#define SMA_wCurrMastVolLeft			0x0042
+#define SMA_wCurrMastVolRight			0x0044
+#define SMA_wCalFreqAtoD			0x0046
+#define SMA_wCurrAuxVolLeft			0x0048
+#define SMA_wCurrAuxVolRight			0x004a
+#define SMA_wCurrPlay1VolLeft			0x004c
+#define SMA_wCurrPlay1VolRight			0x004e
+#define SMA_wCurrPlay2VolLeft			0x0050
+#define SMA_wCurrPlay2VolRight			0x0052
+#define SMA_wCurrPlay3VolLeft			0x0054
+#define SMA_wCurrPlay3VolRight			0x0056
+#define SMA_wCurrPlay4VolLeft			0x0058
+#define SMA_wCurrPlay4VolRight			0x005a
+#define SMA_wCurrPlay1PeakLeft			0x005c
+#define SMA_wCurrPlay1PeakRight			0x005e
+#define SMA_wCurrPlay2PeakLeft			0x0060
+#define SMA_wCurrPlay2PeakRight			0x0062
+#define SMA_wCurrPlay3PeakLeft			0x0064
+#define SMA_wCurrPlay3PeakRight			0x0066
+#define SMA_wCurrPlay4PeakLeft			0x0068
+#define SMA_wCurrPlay4PeakRight			0x006a
+#define SMA_wCurrPlayPeakLeft			0x006c
+#define SMA_wCurrPlayPeakRight			0x006e
+#define SMA_wCurrDATSR				0x0070
+#define SMA_wCurrDATRXCHNL			0x0072
+#define SMA_wCurrDATTXCHNL			0x0074
+#define SMA_wCurrDATRXRate			0x0076
+#define SMA_dwDSPPlayCount			0x0078
+#define SMA__size				0x007c
+
+#define INITCODEFILE		"turtlebeach/pndspini.bin"
+#define PERMCODEFILE		"turtlebeach/pndsperm.bin"
+#define LONGNAME		"MultiSound (Pinnacle/Fiji)"
+
+#endif /* __MSND_PINNACLE_H */
diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c
new file mode 100644
index 000000000000..494058a1a502
--- /dev/null
+++ b/sound/isa/msnd/msnd_pinnacle_mixer.c
@@ -0,0 +1,343 @@
+/***************************************************************************
+			  msnd_pinnacle_mixer.c  -  description
+			     -------------------
+    begin		: Fre Jun 7 2002
+    copyright 		: (C) 2002 by karsten wiese
+    email		: annabellesgarden@yahoo.de
+ ***************************************************************************/
+
+/***************************************************************************
+ *							      		   *
+ *   This program is free software; you can redistribute it and/or modify  *
+ *   it under the terms of the GNU General Public License as published by  *
+ *   the Free Software Foundation; either version 2 of the License, or     *
+ *   (at your option) any later version.				   *
+ *									   *
+ ***************************************************************************/
+
+#include <linux/io.h>
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include "msnd.h"
+#include "msnd_pinnacle.h"
+
+
+#define MSND_MIXER_VOLUME	0
+#define MSND_MIXER_PCM		1
+#define MSND_MIXER_AUX		2	/* Input source 1  (aux1) */
+#define MSND_MIXER_IMIX		3	/*  Recording monitor  */
+#define MSND_MIXER_SYNTH	4
+#define MSND_MIXER_SPEAKER	5
+#define MSND_MIXER_LINE		6
+#define MSND_MIXER_MIC		7
+#define MSND_MIXER_RECLEV	11	/* Recording level */
+#define MSND_MIXER_IGAIN	12	/* Input gain */
+#define MSND_MIXER_OGAIN	13	/* Output gain */
+#define MSND_MIXER_DIGITAL	17	/* Digital (input) 1 */
+
+/*	Device mask bits	*/
+
+#define MSND_MASK_VOLUME	(1 << MSND_MIXER_VOLUME)
+#define MSND_MASK_SYNTH		(1 << MSND_MIXER_SYNTH)
+#define MSND_MASK_PCM		(1 << MSND_MIXER_PCM)
+#define MSND_MASK_SPEAKER	(1 << MSND_MIXER_SPEAKER)
+#define MSND_MASK_LINE		(1 << MSND_MIXER_LINE)
+#define MSND_MASK_MIC		(1 << MSND_MIXER_MIC)
+#define MSND_MASK_IMIX		(1 << MSND_MIXER_IMIX)
+#define MSND_MASK_RECLEV	(1 << MSND_MIXER_RECLEV)
+#define MSND_MASK_IGAIN		(1 << MSND_MIXER_IGAIN)
+#define MSND_MASK_OGAIN		(1 << MSND_MIXER_OGAIN)
+#define MSND_MASK_AUX		(1 << MSND_MIXER_AUX)
+#define MSND_MASK_DIGITAL	(1 << MSND_MIXER_DIGITAL)
+
+static int snd_msndmix_info_mux(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_info *uinfo)
+{
+	static char *texts[3] = {
+		"Analog", "MASS", "SPDIF",
+	};
+	struct snd_msnd *chip = snd_kcontrol_chip(kcontrol);
+	unsigned items = test_bit(F_HAVEDIGITAL, &chip->flags) ? 3 : 2;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = items;
+	if (uinfo->value.enumerated.item >= items)
+		uinfo->value.enumerated.item = items - 1;
+	strcpy(uinfo->value.enumerated.name,
+		texts[uinfo->value.enumerated.item]);
+	return 0;
+}
+
+static int snd_msndmix_get_mux(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_msnd *chip = snd_kcontrol_chip(kcontrol);
+	/* MSND_MASK_IMIX is the default */
+	ucontrol->value.enumerated.item[0] = 0;
+
+	if (chip->recsrc & MSND_MASK_SYNTH) {
+		ucontrol->value.enumerated.item[0] = 1;
+	} else if ((chip->recsrc & MSND_MASK_DIGITAL) &&
+		 test_bit(F_HAVEDIGITAL, &chip->flags)) {
+		ucontrol->value.enumerated.item[0] = 2;
+	}
+
+
+	return 0;
+}
+
+static int snd_msndmix_set_mux(struct snd_msnd *chip, int val)
+{
+	unsigned newrecsrc;
+	int change;
+	unsigned char msndbyte;
+
+	switch (val) {
+	case 0:
+		newrecsrc = MSND_MASK_IMIX;
+		msndbyte = HDEXAR_SET_ANA_IN;
+		break;
+	case 1:
+		newrecsrc = MSND_MASK_SYNTH;
+		msndbyte = HDEXAR_SET_SYNTH_IN;
+		break;
+	case 2:
+		newrecsrc = MSND_MASK_DIGITAL;
+		msndbyte = HDEXAR_SET_DAT_IN;
+		break;
+	default:
+		return -EINVAL;
+	}
+	change  = newrecsrc != chip->recsrc;
+	if (change) {
+		change = 0;
+		if (!snd_msnd_send_word(chip, 0, 0, msndbyte))
+			if (!snd_msnd_send_dsp_cmd(chip, HDEX_AUX_REQ)) {
+				chip->recsrc = newrecsrc;
+				change = 1;
+			}
+	}
+	return change;
+}
+
+static int snd_msndmix_put_mux(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_msnd *msnd = snd_kcontrol_chip(kcontrol);
+	return snd_msndmix_set_mux(msnd, ucontrol->value.enumerated.item[0]);
+}
+
+
+static int snd_msndmix_volume_info(struct snd_kcontrol *kcontrol,
+				   struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 2;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 100;
+	return 0;
+}
+
+static int snd_msndmix_volume_get(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_msnd *msnd = snd_kcontrol_chip(kcontrol);
+	int addr = kcontrol->private_value;
+	unsigned long flags;
+
+	spin_lock_irqsave(&msnd->mixer_lock, flags);
+	ucontrol->value.integer.value[0] = msnd->left_levels[addr] * 100;
+	ucontrol->value.integer.value[0] /= 0xFFFF;
+	ucontrol->value.integer.value[1] = msnd->right_levels[addr] * 100;
+	ucontrol->value.integer.value[1] /= 0xFFFF;
+	spin_unlock_irqrestore(&msnd->mixer_lock, flags);
+	return 0;
+}
+
+#define update_volm(a, b)						\
+	do {								\
+		writew((dev->left_levels[a] >> 1) *			\
+		       readw(dev->SMA + SMA_wCurrMastVolLeft) / 0xffff,	\
+		       dev->SMA + SMA_##b##Left);			\
+		writew((dev->right_levels[a] >> 1)  *			\
+		       readw(dev->SMA + SMA_wCurrMastVolRight) / 0xffff, \
+		       dev->SMA + SMA_##b##Right);			\
+	} while (0);
+
+#define update_potm(d, s, ar)						\
+	do {								\
+		writeb((dev->left_levels[d] >> 8) *			\
+		       readw(dev->SMA + SMA_wCurrMastVolLeft) / 0xffff, \
+		       dev->SMA + SMA_##s##Left);			\
+		writeb((dev->right_levels[d] >> 8) *			\
+		       readw(dev->SMA + SMA_wCurrMastVolRight) / 0xffff, \
+		       dev->SMA + SMA_##s##Right);			\
+		if (snd_msnd_send_word(dev, 0, 0, ar) == 0)		\
+			snd_msnd_send_dsp_cmd(dev, HDEX_AUX_REQ);	\
+	} while (0);
+
+#define update_pot(d, s, ar)						\
+	do {								\
+		writeb(dev->left_levels[d] >> 8,			\
+		       dev->SMA + SMA_##s##Left);			\
+		writeb(dev->right_levels[d] >> 8,			\
+		       dev->SMA + SMA_##s##Right);			\
+		if (snd_msnd_send_word(dev, 0, 0, ar) == 0)		\
+			snd_msnd_send_dsp_cmd(dev, HDEX_AUX_REQ);	\
+	} while (0);
+
+
+static int snd_msndmix_set(struct snd_msnd *dev, int d, int left, int right)
+{
+	int bLeft, bRight;
+	int wLeft, wRight;
+	int updatemaster = 0;
+
+	if (d >= LEVEL_ENTRIES)
+		return -EINVAL;
+
+	bLeft = left * 0xff / 100;
+	wLeft = left * 0xffff / 100;
+
+	bRight = right * 0xff / 100;
+	wRight = right * 0xffff / 100;
+
+	dev->left_levels[d] = wLeft;
+	dev->right_levels[d] = wRight;
+
+	switch (d) {
+		/* master volume unscaled controls */
+	case MSND_MIXER_LINE:			/* line pot control */
+		/* scaled by IMIX in digital mix */
+		writeb(bLeft, dev->SMA + SMA_bInPotPosLeft);
+		writeb(bRight, dev->SMA + SMA_bInPotPosRight);
+		if (snd_msnd_send_word(dev, 0, 0, HDEXAR_IN_SET_POTS) == 0)
+			snd_msnd_send_dsp_cmd(dev, HDEX_AUX_REQ);
+		break;
+	case MSND_MIXER_MIC:			/* mic pot control */
+		if (dev->type == msndClassic)
+			return -EINVAL;
+		/* scaled by IMIX in digital mix */
+		writeb(bLeft, dev->SMA + SMA_bMicPotPosLeft);
+		writeb(bRight, dev->SMA + SMA_bMicPotPosRight);
+		if (snd_msnd_send_word(dev, 0, 0, HDEXAR_MIC_SET_POTS) == 0)
+			snd_msnd_send_dsp_cmd(dev, HDEX_AUX_REQ);
+		break;
+	case MSND_MIXER_VOLUME:		/* master volume */
+		writew(wLeft, dev->SMA + SMA_wCurrMastVolLeft);
+		writew(wRight, dev->SMA + SMA_wCurrMastVolRight);
+		/* fall through */
+
+	case MSND_MIXER_AUX:			/* aux pot control */
+		/* scaled by master volume */
+		/* fall through */
+
+		/* digital controls */
+	case MSND_MIXER_SYNTH:			/* synth vol (dsp mix) */
+	case MSND_MIXER_PCM:			/* pcm vol (dsp mix) */
+	case MSND_MIXER_IMIX:			/* input monitor (dsp mix) */
+		/* scaled by master volume */
+		updatemaster = 1;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	if (updatemaster) {
+		/* update master volume scaled controls */
+		update_volm(MSND_MIXER_PCM, wCurrPlayVol);
+		update_volm(MSND_MIXER_IMIX, wCurrInVol);
+		if (dev->type == msndPinnacle)
+			update_volm(MSND_MIXER_SYNTH, wCurrMHdrVol);
+		update_potm(MSND_MIXER_AUX, bAuxPotPos, HDEXAR_AUX_SET_POTS);
+	}
+
+	return 0;
+}
+
+static int snd_msndmix_volume_put(struct snd_kcontrol *kcontrol,
+				  struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_msnd *msnd = snd_kcontrol_chip(kcontrol);
+	int change, addr = kcontrol->private_value;
+	int left, right;
+	unsigned long flags;
+
+	left = ucontrol->value.integer.value[0] % 101;
+	right = ucontrol->value.integer.value[1] % 101;
+	spin_lock_irqsave(&msnd->mixer_lock, flags);
+	change = msnd->left_levels[addr] != left
+		|| msnd->right_levels[addr] != right;
+	snd_msndmix_set(msnd, addr, left, right);
+	spin_unlock_irqrestore(&msnd->mixer_lock, flags);
+	return change;
+}
+
+
+#define DUMMY_VOLUME(xname, xindex, addr) \
+{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \
+  .info = snd_msndmix_volume_info, \
+  .get = snd_msndmix_volume_get, .put = snd_msndmix_volume_put, \
+  .private_value = addr }
+
+
+static struct snd_kcontrol_new snd_msnd_controls[] = {
+DUMMY_VOLUME("Master Volume", 0, MSND_MIXER_VOLUME),
+DUMMY_VOLUME("PCM Volume", 0, MSND_MIXER_PCM),
+DUMMY_VOLUME("Aux Volume", 0, MSND_MIXER_AUX),
+DUMMY_VOLUME("Line Volume", 0, MSND_MIXER_LINE),
+DUMMY_VOLUME("Mic Volume", 0, MSND_MIXER_MIC),
+DUMMY_VOLUME("Monitor",	0, MSND_MIXER_IMIX),
+{
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Capture Source",
+	.info = snd_msndmix_info_mux,
+	.get = snd_msndmix_get_mux,
+	.put = snd_msndmix_put_mux,
+}
+};
+
+
+int __devinit snd_msndmix_new(struct snd_card *card)
+{
+	struct snd_msnd *chip = card->private_data;
+	unsigned int idx;
+	int err;
+
+	if (snd_BUG_ON(!chip))
+		return -EINVAL;
+	spin_lock_init(&chip->mixer_lock);
+	strcpy(card->mixername, "MSND Pinnacle Mixer");
+
+	for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++)
+		err = snd_ctl_add(card,
+				  snd_ctl_new1(snd_msnd_controls + idx, chip));
+		if (err < 0)
+			return err;
+
+	return 0;
+}
+EXPORT_SYMBOL(snd_msndmix_new);
+
+void snd_msndmix_setup(struct snd_msnd *dev)
+{
+	update_pot(MSND_MIXER_LINE, bInPotPos, HDEXAR_IN_SET_POTS);
+	update_potm(MSND_MIXER_AUX, bAuxPotPos, HDEXAR_AUX_SET_POTS);
+	update_volm(MSND_MIXER_PCM, wCurrPlayVol);
+	update_volm(MSND_MIXER_IMIX, wCurrInVol);
+	if (dev->type == msndPinnacle) {
+		update_pot(MSND_MIXER_MIC, bMicPotPos, HDEXAR_MIC_SET_POTS);
+		update_volm(MSND_MIXER_SYNTH, wCurrMHdrVol);
+	}
+}
+EXPORT_SYMBOL(snd_msndmix_setup);
+
+int snd_msndmix_force_recsrc(struct snd_msnd *dev, int recsrc)
+{
+	dev->recsrc = -1;
+	return snd_msndmix_set_mux(dev, recsrc);
+}
+EXPORT_SYMBOL(snd_msndmix_force_recsrc);
diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c
index 58c972b2af03..ef95279da7a3 100644
--- a/sound/isa/opl3sa2.c
+++ b/sound/isa/opl3sa2.c
@@ -179,12 +179,13 @@ static unsigned char __snd_opl3sa2_read(struct snd_opl3sa2 *chip, unsigned char
 	unsigned char result;
 #if 0
 	outb(0x1d, port);	/* password */
-	printk("read [0x%lx] = 0x%x\n", port, inb(port));
+	printk(KERN_DEBUG "read [0x%lx] = 0x%x\n", port, inb(port));
 #endif
 	outb(reg, chip->port);	/* register */
 	result = inb(chip->port + 1);
 #if 0
-	printk("read [0x%lx] = 0x%x [0x%x]\n", port, result, inb(port));
+	printk(KERN_DEBUG "read [0x%lx] = 0x%x [0x%x]\n",
+	       port, result, inb(port));
 #endif
 	return result;
 }
@@ -233,7 +234,10 @@ static int __devinit snd_opl3sa2_detect(struct snd_card *card)
 		snd_printk(KERN_ERR PFX "can't grab port 0x%lx\n", port);
 		return -EBUSY;
 	}
-	// snd_printk("REG 0A = 0x%x\n", snd_opl3sa2_read(chip, 0x0a));
+	/*
+	snd_printk(KERN_DEBUG "REG 0A = 0x%x\n",
+		   snd_opl3sa2_read(chip, 0x0a));
+	*/
 	chip->version = 0;
 	tmp = snd_opl3sa2_read(chip, OPL3SA2_MISC);
 	if (tmp == 0xff) {
@@ -550,21 +554,27 @@ static int __devinit snd_opl3sa2_mixer(struct snd_card *card)
 #ifdef CONFIG_PM
 static int snd_opl3sa2_suspend(struct snd_card *card, pm_message_t state)
 {
-	struct snd_opl3sa2 *chip = card->private_data;
+	if (card) {
+		struct snd_opl3sa2 *chip = card->private_data;
 
-	snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
-	chip->wss->suspend(chip->wss);
-	/* power down */
-	snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D3);
+		snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
+		chip->wss->suspend(chip->wss);
+		/* power down */
+		snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D3);
+	}
 
 	return 0;
 }
 
 static int snd_opl3sa2_resume(struct snd_card *card)
 {
-	struct snd_opl3sa2 *chip = card->private_data;
+	struct snd_opl3sa2 *chip;
 	int i;
 
+	if (!card)
+		return 0;
+
+	chip = card->private_data;
 	/* power up */
 	snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D0);
 
@@ -613,25 +623,28 @@ static void snd_opl3sa2_free(struct snd_card *card)
 {
 	struct snd_opl3sa2 *chip = card->private_data;
 	if (chip->irq >= 0)
-		free_irq(chip->irq, (void *)chip);
+		free_irq(chip->irq, card);
 	release_and_free_resource(chip->res_port);
 }
 
-static struct snd_card *snd_opl3sa2_card_new(int dev)
+static int snd_opl3sa2_card_new(int dev, struct snd_card **cardp)
 {
 	struct snd_card *card;
 	struct snd_opl3sa2 *chip;
+	int err;
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(struct snd_opl3sa2));
-	if (card == NULL)
-		return NULL;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct snd_opl3sa2), &card);
+	if (err < 0)
+		return err;
 	strcpy(card->driver, "OPL3SA2");
-	strcpy(card->shortname, "Yamaha OPL3-SA2");
+	strcpy(card->shortname, "Yamaha OPL3-SA");
 	chip = card->private_data;
 	spin_lock_init(&chip->reg_lock);
 	chip->irq = -1;
 	card->private_free = snd_opl3sa2_free;
-	return card;
+	*cardp = card;
+	return 0;
 }
 
 static int __devinit snd_opl3sa2_probe(struct snd_card *card, int dev)
@@ -723,9 +736,9 @@ static int __devinit snd_opl3sa2_pnp_detect(struct pnp_dev *pdev,
 	if (dev >= SNDRV_CARDS)
 		return -ENODEV;
 
-	card = snd_opl3sa2_card_new(dev);
-	if (! card)
-		return -ENOMEM;
+	err = snd_opl3sa2_card_new(dev, &card);
+	if (err < 0)
+		return err;
 	if ((err = snd_opl3sa2_pnp(dev, card->private_data, pdev)) < 0) {
 		snd_card_free(card);
 		return err;
@@ -789,9 +802,9 @@ static int __devinit snd_opl3sa2_pnp_cdetect(struct pnp_card_link *pcard,
 	if (dev >= SNDRV_CARDS)
 		return -ENODEV;
 
-	card = snd_opl3sa2_card_new(dev);
-	if (! card)
-		return -ENOMEM;
+	err = snd_opl3sa2_card_new(dev, &card);
+	if (err < 0)
+		return err;
 	if ((err = snd_opl3sa2_pnp(dev, card->private_data, pdev)) < 0) {
 		snd_card_free(card);
 		return err;
@@ -870,9 +883,9 @@ static int __devinit snd_opl3sa2_isa_probe(struct device *pdev,
 	struct snd_card *card;
 	int err;
 
-	card = snd_opl3sa2_card_new(dev);
-	if (! card)
-		return -ENOMEM;
+	err = snd_opl3sa2_card_new(dev, &card);
+	if (err < 0)
+		return err;
 	snd_card_set_dev(card, pdev);
 	if ((err = snd_opl3sa2_probe(card, dev)) < 0) {
 		snd_card_free(card);
diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c
index 440755cc0013..02e30d7c6a93 100644
--- a/sound/isa/opti9xx/miro.c
+++ b/sound/isa/opti9xx/miro.c
@@ -1228,9 +1228,10 @@ static int __devinit snd_miro_probe(struct device *devptr, unsigned int n)
 	struct snd_pcm *pcm;
 	struct snd_rawmidi *rmidi;
 
-	if (!(card = snd_card_new(index, id, THIS_MODULE,
-				  sizeof(struct snd_miro))))
-		return -ENOMEM;
+	error = snd_card_create(index, id, THIS_MODULE,
+				sizeof(struct snd_miro), &card);
+	if (error < 0)
+		return error;
 
 	card->private_free = snd_card_miro_free;
 	miro = card->private_data;
diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c
index 19706b0d8497..5cd555325b9d 100644
--- a/sound/isa/opti9xx/opti92x-ad1848.c
+++ b/sound/isa/opti9xx/opti92x-ad1848.c
@@ -252,7 +252,7 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip,
 #endif	/* OPTi93X */
 
 	default:
-		snd_printk("chip %d not supported\n", hardware);
+		snd_printk(KERN_ERR "chip %d not supported\n", hardware);
 		return -ENODEV;
 	}
 	return 0;
@@ -294,7 +294,7 @@ static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip,
 #endif	/* OPTi93X */
 
 	default:
-		snd_printk("chip %d not supported\n", chip->hardware);
+		snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware);
 	}
 
 	spin_unlock_irqrestore(&chip->lock, flags);
@@ -336,7 +336,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg,
 #endif	/* OPTi93X */
 
 	default:
-		snd_printk("chip %d not supported\n", chip->hardware);
+		snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware);
 	}
 
 	spin_unlock_irqrestore(&chip->lock, flags);
@@ -412,7 +412,7 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip)
 #endif	/* OPTi93X */
 
 	default:
-		snd_printk("chip %d not supported\n", chip->hardware);
+		snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware);
 		return -EINVAL;
 	}
 
@@ -430,7 +430,8 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip)
 		wss_base_bits = 0x02;
 		break;
 	default:
-		snd_printk("WSS port 0x%lx not valid\n", chip->wss_base);
+		snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n",
+			   chip->wss_base);
 		goto __skip_base;
 	}
 	snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), wss_base_bits << 4, 0x30);
@@ -455,7 +456,7 @@ __skip_base:
 		irq_bits = 0x04;
 		break;
 	default:
-		snd_printk("WSS irq # %d not valid\n", chip->irq);
+		snd_printk(KERN_WARNING "WSS irq # %d not valid\n", chip->irq);
 		goto __skip_resources;
 	}
 
@@ -470,13 +471,14 @@ __skip_base:
 		dma_bits = 0x03;
 		break;
 	default:
-		snd_printk("WSS dma1 # %d not valid\n", chip->dma1);
+		snd_printk(KERN_WARNING "WSS dma1 # %d not valid\n",
+			   chip->dma1);
 		goto __skip_resources;
 	}
 
 #if defined(CS4231) || defined(OPTi93X)
 	if (chip->dma1 == chip->dma2) {
-		snd_printk("don't want to share dmas\n");
+		snd_printk(KERN_ERR "don't want to share dmas\n");
 		return -EBUSY;
 	}
 
@@ -485,7 +487,8 @@ __skip_base:
 	case 1:
 		break;
 	default:
-		snd_printk("WSS dma2 # %d not valid\n", chip->dma2);
+		snd_printk(KERN_WARNING "WSS dma2 # %d not valid\n",
+			   chip->dma2);
 		goto __skip_resources;
 	}
 	dma_bits |= 0x04;
@@ -516,7 +519,8 @@ __skip_resources:
 			mpu_port_bits = 0x00;
 			break;
 		default:
-			snd_printk("MPU-401 port 0x%lx not valid\n",
+			snd_printk(KERN_WARNING
+				   "MPU-401 port 0x%lx not valid\n",
 				chip->mpu_port);
 			goto __skip_mpu;
 		}
@@ -535,7 +539,7 @@ __skip_resources:
 			mpu_irq_bits = 0x01;
 			break;
 		default:
-			snd_printk("MPU-401 irq # %d not valid\n",
+			snd_printk(KERN_WARNING "MPU-401 irq # %d not valid\n",
 				chip->mpu_irq);
 			goto __skip_mpu;
 		}
@@ -726,7 +730,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card)
 	if (chip->wss_base == SNDRV_AUTO_PORT) {
 		chip->wss_base = snd_legacy_find_free_ioport(possible_ports, 4);
 		if (chip->wss_base < 0) {
-			snd_printk("unable to find a free WSS port\n");
+			snd_printk(KERN_ERR "unable to find a free WSS port\n");
 			return -EBUSY;
 		}
 	}
@@ -815,14 +819,8 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card)
 				   chip->fm_port, chip->fm_port + 4 - 1);
 		}
 		if (opl3) {
-#ifdef CS4231
-			const int t1dev = 1;
-#else
-			const int t1dev = 0;
-#endif
-			if ((error = snd_opl3_timer_new(opl3, t1dev, t1dev+1)) < 0)
-				return error;
-			if ((error = snd_opl3_hwdep_new(opl3, 0, 1, &synth)) < 0)
+			error = snd_opl3_hwdep_new(opl3, 0, 1, &synth);
+			if (error < 0)
 				return error;
 		}
 	}
@@ -830,15 +828,18 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card)
 	return snd_card_register(card);
 }
 
-static struct snd_card *snd_opti9xx_card_new(void)
+static int snd_opti9xx_card_new(struct snd_card **cardp)
 {
 	struct snd_card *card;
+	int err;
 
-	card = snd_card_new(index, id, THIS_MODULE, sizeof(struct snd_opti9xx));
-	if (! card)
-		return NULL;
+	err = snd_card_create(index, id, THIS_MODULE,
+			      sizeof(struct snd_opti9xx), &card);
+	if (err < 0)
+		return err;
 	card->private_free = snd_card_opti9xx_free;
-	return card;
+	*cardp = card;
+	return 0;
 }
 
 static int __devinit snd_opti9xx_isa_match(struct device *devptr,
@@ -897,15 +898,15 @@ static int __devinit snd_opti9xx_isa_probe(struct device *devptr,
 #if defined(CS4231) || defined(OPTi93X)
 	if (dma2 == SNDRV_AUTO_DMA) {
 		if ((dma2 = snd_legacy_find_free_dma(possible_dma2s[dma1 % 4])) < 0) {
-			snd_printk("unable to find a free DMA2\n");
+			snd_printk(KERN_ERR "unable to find a free DMA2\n");
 			return -EBUSY;
 		}
 	}
 #endif
 
-	card = snd_opti9xx_card_new();
-	if (! card)
-		return -ENOMEM;
+	error = snd_opti9xx_card_new(&card);
+	if (error < 0)
+		return error;
 
 	if ((error = snd_card_opti9xx_detect(card, card->private_data)) < 0) {
 		snd_card_free(card);
@@ -950,9 +951,9 @@ static int __devinit snd_opti9xx_pnp_probe(struct pnp_card_link *pcard,
 		return -EBUSY;
 	if (! isapnp)
 		return -ENODEV;
-	card = snd_opti9xx_card_new();
-	if (! card)
-		return -ENOMEM;
+	error = snd_opti9xx_card_new(&card);
+	if (error < 0)
+		return error;
 	chip = card->private_data;
 
 	hw = snd_card_opti9xx_pnp(chip, pcard, pid);
diff --git a/sound/isa/sb/es968.c b/sound/isa/sb/es968.c
index c8c8e214c843..cafc3a7316a8 100644
--- a/sound/isa/sb/es968.c
+++ b/sound/isa/sb/es968.c
@@ -108,9 +108,10 @@ static int __devinit snd_card_es968_probe(int dev,
 	struct snd_card *card;
 	struct snd_card_es968 *acard;
 
-	if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-				 sizeof(struct snd_card_es968))) == NULL)
-		return -ENOMEM;
+	error = snd_card_create(index[dev], id[dev], THIS_MODULE,
+				sizeof(struct snd_card_es968), &card);
+	if (error < 0)
+		return error;
 	acard = card->private_data;
 	if ((error = snd_card_es968_pnp(dev, acard, pcard, pid))) {
 		snd_card_free(card);
diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c
index 2c201f78ce50..519c36346dec 100644
--- a/sound/isa/sb/sb16.c
+++ b/sound/isa/sb/sb16.c
@@ -324,14 +324,18 @@ static void snd_sb16_free(struct snd_card *card)
 #define is_isapnp_selected(dev)		0
 #endif
 
-static struct snd_card *snd_sb16_card_new(int dev)
+static int snd_sb16_card_new(int dev, struct snd_card **cardp)
 {
-	struct snd_card *card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-					sizeof(struct snd_card_sb16));
-	if (card == NULL)
-		return NULL;
+	struct snd_card *card;
+	int err;
+
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct snd_card_sb16), &card);
+	if (err < 0)
+		return err;
 	card->private_free = snd_sb16_free;
-	return card;
+	*cardp = card;
+	return 0;
 }
 
 static int __devinit snd_sb16_probe(struct snd_card *card, int dev)
@@ -489,9 +493,9 @@ static int __devinit snd_sb16_isa_probe1(int dev, struct device *pdev)
 	struct snd_card *card;
 	int err;
 
-	card = snd_sb16_card_new(dev);
-	if (! card)
-		return -ENOMEM;
+	err = snd_sb16_card_new(dev, &card);
+	if (err < 0)
+		return err;
 
 	acard = card->private_data;
 	/* non-PnP FM port address is hardwired with base port address */
@@ -610,9 +614,9 @@ static int __devinit snd_sb16_pnp_detect(struct pnp_card_link *pcard,
 	for ( ; dev < SNDRV_CARDS; dev++) {
 		if (!enable[dev] || !isapnp[dev])
 			continue;
-		card = snd_sb16_card_new(dev);
-		if (! card)
-			return -ENOMEM;
+		res = snd_sb16_card_new(dev, &card);
+		if (res < 0)
+			return res;
 		snd_card_set_dev(card, &pcard->card->dev);
 		if ((res = snd_card_sb16_pnp(dev, card->private_data, pcard, pid)) < 0 ||
 		    (res = snd_sb16_probe(card, dev)) < 0) {
diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c
index ea06877be4b1..3cd57ee54660 100644
--- a/sound/isa/sb/sb8.c
+++ b/sound/isa/sb/sb8.c
@@ -103,10 +103,10 @@ static int __devinit snd_sb8_probe(struct device *pdev, unsigned int dev)
 	struct snd_opl3 *opl3;
 	int err;
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-			    sizeof(struct snd_sb8));
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct snd_sb8), &card);
+	if (err < 0)
+		return err;
 	acard = card->private_data;
 	card->private_free = snd_sb8_free;
 
diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c
index 406a431af91e..475220bbcc96 100644
--- a/sound/isa/sb/sb_mixer.c
+++ b/sound/isa/sb/sb_mixer.c
@@ -182,7 +182,7 @@ static int snd_sbmixer_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_
 
 static int snd_dt019x_input_sw_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[5] = {
+	static const char *texts[5] = {
 		"CD", "Mic", "Line", "Synth", "Master"
 	};
 
@@ -269,12 +269,73 @@ static int snd_dt019x_input_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl
 }
 
 /*
+ * ALS4000 mono recording control switch
+ */
+
+static int snd_als4k_mono_capture_route_info(struct snd_kcontrol *kcontrol,
+					     struct snd_ctl_elem_info *uinfo)
+{
+	static const char *texts[3] = {
+		"L chan only", "R chan only", "L ch/2 + R ch/2"
+	};
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = 3;
+	if (uinfo->value.enumerated.item > 2)
+		uinfo->value.enumerated.item = 2;
+	strcpy(uinfo->value.enumerated.name,
+	       texts[uinfo->value.enumerated.item]);
+	return 0;
+}
+
+static int snd_als4k_mono_capture_route_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_sb *sb = snd_kcontrol_chip(kcontrol);
+	unsigned long flags;
+	unsigned char oval;
+
+	spin_lock_irqsave(&sb->mixer_lock, flags);
+	oval = snd_sbmixer_read(sb, SB_ALS4000_MONO_IO_CTRL);
+	spin_unlock_irqrestore(&sb->mixer_lock, flags);
+	oval >>= 6;
+	if (oval > 2)
+		oval = 2;
+
+	ucontrol->value.enumerated.item[0] = oval;
+	return 0;
+}
+
+static int snd_als4k_mono_capture_route_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_sb *sb = snd_kcontrol_chip(kcontrol);
+	unsigned long flags;
+	int change;
+	unsigned char nval, oval;
+
+	if (ucontrol->value.enumerated.item[0] > 2)
+		return -EINVAL;
+	spin_lock_irqsave(&sb->mixer_lock, flags);
+	oval = snd_sbmixer_read(sb, SB_ALS4000_MONO_IO_CTRL);
+
+	nval = (oval & ~(3 << 6))
+	     | (ucontrol->value.enumerated.item[0] << 6);
+	change = nval != oval;
+	if (change)
+		snd_sbmixer_write(sb, SB_ALS4000_MONO_IO_CTRL, nval);
+	spin_unlock_irqrestore(&sb->mixer_lock, flags);
+	return change;
+}
+
+/*
  * SBPRO input multiplexer
  */
 
 static int snd_sb8mixer_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[3] = {
+	static const char *texts[3] = {
 		"Mic", "CD", "Line"
 	};
 
@@ -442,6 +503,12 @@ int snd_sbmixer_add_ctl(struct snd_sb *chip, const char *name, int index, int ty
 			.get = snd_dt019x_input_sw_get,
 			.put = snd_dt019x_input_sw_put,
 		},
+		[SB_MIX_MONO_CAPTURE_ALS4K] = {
+			.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+			.info = snd_als4k_mono_capture_route_info,
+			.get = snd_als4k_mono_capture_route_get,
+			.put = snd_als4k_mono_capture_route_put,
+		},
 	};
 	struct snd_kcontrol *ctl;
 	int err;
@@ -636,6 +703,8 @@ static struct sbmix_elem snd_dt019x_ctl_capture_source =
 	};
 
 static struct sbmix_elem *snd_dt019x_controls[] = {
+	/* ALS4000 below has some parts which we might be lacking,
+	 * e.g. snd_als4000_ctl_mono_playback_switch - check it! */
 	&snd_dt019x_ctl_master_play_vol,
 	&snd_dt019x_ctl_pcm_play_vol,
 	&snd_dt019x_ctl_synth_play_vol,
@@ -666,18 +735,21 @@ static unsigned char snd_dt019x_init_values[][2] = {
 /*
  * ALS4000 specific mixer elements
  */
-/* FIXME: SB_ALS4000_MONO_IO_CTRL needs output select ctrl! */
 static struct sbmix_elem snd_als4000_ctl_master_mono_playback_switch =
 	SB_SINGLE("Master Mono Playback Switch", SB_ALS4000_MONO_IO_CTRL, 5, 1);
-static struct sbmix_elem snd_als4000_ctl_master_mono_capture_route =
-	SB_SINGLE("Master Mono Capture Route", SB_ALS4000_MONO_IO_CTRL, 6, 0x03);
-/* FIXME: mono playback switch also available on DT019X? */
+static struct sbmix_elem snd_als4k_ctl_master_mono_capture_route = {
+		.name = "Master Mono Capture Route",
+		.type = SB_MIX_MONO_CAPTURE_ALS4K
+	};
 static struct sbmix_elem snd_als4000_ctl_mono_playback_switch =
 	SB_SINGLE("Mono Playback Switch", SB_DT019X_OUTPUT_SW2, 0, 1);
 static struct sbmix_elem snd_als4000_ctl_mic_20db_boost =
 	SB_SINGLE("Mic Boost (+20dB)", SB_ALS4000_MIC_IN_GAIN, 0, 0x03);
-static struct sbmix_elem snd_als4000_ctl_mixer_loopback =
-	SB_SINGLE("Analog Loopback", SB_ALS4000_MIC_IN_GAIN, 7, 0x01);
+static struct sbmix_elem snd_als4000_ctl_mixer_analog_loopback =
+	SB_SINGLE("Analog Loopback Switch", SB_ALS4000_MIC_IN_GAIN, 7, 0x01);
+static struct sbmix_elem snd_als4000_ctl_mixer_digital_loopback =
+	SB_SINGLE("Digital Loopback Switch",
+		  SB_ALS4000_CR3_CONFIGURATION, 7, 0x01);
 /* FIXME: functionality of 3D controls might be swapped, I didn't find
  * a description of how to identify what is supposed to be what */
 static struct sbmix_elem snd_als4000_3d_control_switch =
@@ -694,6 +766,9 @@ static struct sbmix_elem snd_als4000_3d_control_delay =
 	SB_SINGLE("3D Control - Wide", SB_ALS4000_3D_TIME_DELAY, 0, 0x0f);
 static struct sbmix_elem snd_als4000_3d_control_poweroff_switch =
 	SB_SINGLE("3D PowerOff Switch", SB_ALS4000_3D_TIME_DELAY, 4, 0x01);
+static struct sbmix_elem snd_als4000_ctl_3db_freq_control_switch =
+	SB_SINGLE("Master Playback 8kHz / 20kHz LPF Switch",
+		  SB_ALS4000_FMDAC, 5, 0x01);
 #ifdef NOT_AVAILABLE
 static struct sbmix_elem snd_als4000_ctl_fmdac =
 	SB_SINGLE("FMDAC Switch (Option ?)", SB_ALS4000_FMDAC, 0, 0x01);
@@ -702,35 +777,37 @@ static struct sbmix_elem snd_als4000_ctl_qsound =
 #endif
 
 static struct sbmix_elem *snd_als4000_controls[] = {
-	&snd_sb16_ctl_master_play_vol,
-	&snd_dt019x_ctl_pcm_play_switch,
-	&snd_sb16_ctl_pcm_play_vol,
-	&snd_sb16_ctl_synth_capture_route,
-	&snd_dt019x_ctl_synth_play_switch,
-	&snd_sb16_ctl_synth_play_vol,
-	&snd_sb16_ctl_cd_capture_route,
-	&snd_sb16_ctl_cd_play_switch,
-	&snd_sb16_ctl_cd_play_vol,
-	&snd_sb16_ctl_line_capture_route,
-	&snd_sb16_ctl_line_play_switch,
-	&snd_sb16_ctl_line_play_vol,
-	&snd_sb16_ctl_mic_capture_route,
-	&snd_als4000_ctl_mic_20db_boost,
-	&snd_sb16_ctl_auto_mic_gain,
-	&snd_sb16_ctl_mic_play_switch,
-	&snd_sb16_ctl_mic_play_vol,
-	&snd_sb16_ctl_pc_speaker_vol,
-	&snd_sb16_ctl_capture_vol,
-	&snd_sb16_ctl_play_vol,
-	&snd_als4000_ctl_master_mono_playback_switch,
-	&snd_als4000_ctl_master_mono_capture_route,
-	&snd_als4000_ctl_mono_playback_switch,
-	&snd_als4000_ctl_mixer_loopback,
-	&snd_als4000_3d_control_switch,
-	&snd_als4000_3d_control_ratio,
-	&snd_als4000_3d_control_freq,
-	&snd_als4000_3d_control_delay,
-	&snd_als4000_3d_control_poweroff_switch,
+						/* ALS4000a.PDF regs page */
+	&snd_sb16_ctl_master_play_vol,		/* MX30/31 12 */
+	&snd_dt019x_ctl_pcm_play_switch,	/* MX4C    16 */
+	&snd_sb16_ctl_pcm_play_vol,		/* MX32/33 12 */
+	&snd_sb16_ctl_synth_capture_route,	/* MX3D/3E 14 */
+	&snd_dt019x_ctl_synth_play_switch,	/* MX4C    16 */
+	&snd_sb16_ctl_synth_play_vol,		/* MX34/35 12/13 */
+	&snd_sb16_ctl_cd_capture_route,		/* MX3D/3E 14 */
+	&snd_sb16_ctl_cd_play_switch,		/* MX3C    14 */
+	&snd_sb16_ctl_cd_play_vol,		/* MX36/37 13 */
+	&snd_sb16_ctl_line_capture_route,	/* MX3D/3E 14 */
+	&snd_sb16_ctl_line_play_switch,		/* MX3C    14 */
+	&snd_sb16_ctl_line_play_vol,		/* MX38/39 13 */
+	&snd_sb16_ctl_mic_capture_route,	/* MX3D/3E 14 */
+	&snd_als4000_ctl_mic_20db_boost,	/* MX4D    16 */
+	&snd_sb16_ctl_mic_play_switch,		/* MX3C    14 */
+	&snd_sb16_ctl_mic_play_vol,		/* MX3A    13 */
+	&snd_sb16_ctl_pc_speaker_vol,		/* MX3B    14 */
+	&snd_sb16_ctl_capture_vol,		/* MX3F/40 15 */
+	&snd_sb16_ctl_play_vol,			/* MX41/42 15 */
+	&snd_als4000_ctl_master_mono_playback_switch, /* MX4C 16 */
+	&snd_als4k_ctl_master_mono_capture_route, /* MX4B  16 */
+	&snd_als4000_ctl_mono_playback_switch,	/* MX4C    16 */
+	&snd_als4000_ctl_mixer_analog_loopback, /* MX4D    16 */
+	&snd_als4000_ctl_mixer_digital_loopback, /* CR3    21 */
+	&snd_als4000_3d_control_switch,		 /* MX50   17 */
+	&snd_als4000_3d_control_ratio,		 /* MX50   17 */
+	&snd_als4000_3d_control_freq,		 /* MX50   17 */
+	&snd_als4000_3d_control_delay,		 /* MX51   18 */
+	&snd_als4000_3d_control_poweroff_switch,	/* MX51    18 */
+	&snd_als4000_ctl_3db_freq_control_switch,	/* MX4F    17 */
 #ifdef NOT_AVAILABLE
 	&snd_als4000_ctl_fmdac,
 	&snd_als4000_ctl_qsound,
@@ -905,13 +982,14 @@ static unsigned char dt019x_saved_regs[] = {
 };
 
 static unsigned char als4000_saved_regs[] = {
+	/* please verify in dsheet whether regs to be added
+	   are actually real H/W or just dummy */
 	SB_DSP4_MASTER_DEV, SB_DSP4_MASTER_DEV + 1,
 	SB_DSP4_OUTPUT_SW,
 	SB_DSP4_PCM_DEV, SB_DSP4_PCM_DEV + 1,
 	SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT,
 	SB_DSP4_SYNTH_DEV, SB_DSP4_SYNTH_DEV + 1,
 	SB_DSP4_CD_DEV, SB_DSP4_CD_DEV + 1,
-	SB_DSP4_MIC_AGC,
 	SB_DSP4_MIC_DEV,
 	SB_DSP4_SPEAKER_DEV,
 	SB_DSP4_IGAIN_DEV, SB_DSP4_IGAIN_DEV + 1,
@@ -919,8 +997,10 @@ static unsigned char als4000_saved_regs[] = {
 	SB_DT019X_OUTPUT_SW2,
 	SB_ALS4000_MONO_IO_CTRL,
 	SB_ALS4000_MIC_IN_GAIN,
+	SB_ALS4000_FMDAC,
 	SB_ALS4000_3D_SND_FX,
 	SB_ALS4000_3D_TIME_DELAY,
+	SB_ALS4000_CR3_CONFIGURATION,
 };
 
 static void save_mixer(struct snd_sb *chip, unsigned char *regs, int num_regs)
diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c
index ca35924dc3b3..782010608ef4 100644
--- a/sound/isa/sc6000.c
+++ b/sound/isa/sc6000.c
@@ -489,9 +489,9 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev)
 	char __iomem *vmss_port;
 
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (!card)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	if (xirq == SNDRV_AUTO_IRQ) {
 		xirq = snd_legacy_find_free_irq(possible_irqs);
@@ -576,10 +576,6 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev)
 		snd_printk(KERN_ERR PFX "no OPL device at 0x%x-0x%x ?\n",
 			   0x388, 0x388 + 2);
 	} else {
-		err = snd_opl3_timer_new(opl3, 0, 1);
-		if (err < 0)
-			goto err_unmap2;
-
 		err = snd_opl3_hwdep_new(opl3, 0, 1, NULL);
 		if (err < 0)
 			goto err_unmap2;
diff --git a/sound/isa/sgalaxy.c b/sound/isa/sgalaxy.c
index 2c7503bf1271..6fe27b9d9440 100644
--- a/sound/isa/sgalaxy.c
+++ b/sound/isa/sgalaxy.c
@@ -243,9 +243,9 @@ static int __devinit snd_sgalaxy_probe(struct device *devptr, unsigned int dev)
 	struct snd_card *card;
 	struct snd_wss *chip;
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	xirq = irq[dev];
 	if (xirq == SNDRV_AUTO_IRQ) {
diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c
index 48a16d865834..66187122377c 100644
--- a/sound/isa/sscape.c
+++ b/sound/isa/sscape.c
@@ -89,9 +89,6 @@ MODULE_DEVICE_TABLE(pnp_card, sscape_pnpids);
 #endif
 
 
-#define MPU401_IO(i)     ((i) + 0)
-#define MIDI_DATA_IO(i)  ((i) + 0)
-#define MIDI_CTRL_IO(i)  ((i) + 1)
 #define HOST_CTRL_IO(i)  ((i) + 2)
 #define HOST_DATA_IO(i)  ((i) + 3)
 #define ODIE_ADDR_IO(i)  ((i) + 4)
@@ -129,9 +126,6 @@ enum GA_REG {
 #define DMA_8BIT  0x80
 
 
-#define AD1845_FREQ_SEL_MSB    0x16
-#define AD1845_FREQ_SEL_LSB    0x17
-
 enum card_type {
 	SSCAPE,
 	SSCAPE_PNP,
@@ -141,8 +135,6 @@ enum card_type {
 struct soundscape {
 	spinlock_t lock;
 	unsigned io_base;
-	unsigned wss_base;
-	int codec_type;
 	int ic_type;
 	enum card_type type;
 	struct resource *io_res;
@@ -330,7 +322,7 @@ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data,
  */
 static inline int verify_mpu401(const struct snd_mpu401 * mpu)
 {
-	return ((inb(MIDI_CTRL_IO(mpu->port)) & 0xc0) == 0x80);
+	return ((inb(MPU401C(mpu)) & 0xc0) == 0x80);
 }
 
 /*
@@ -338,7 +330,7 @@ static inline int verify_mpu401(const struct snd_mpu401 * mpu)
  */
 static inline void initialise_mpu401(const struct snd_mpu401 * mpu)
 {
-	outb(0, MIDI_DATA_IO(mpu->port));
+	outb(0, MPU401D(mpu));
 }
 
 /*
@@ -396,20 +388,20 @@ static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, unsigned ti
  */
 static int obp_startup_ack(struct soundscape *s, unsigned timeout)
 {
-	while (timeout != 0) {
+	unsigned long end_time = jiffies + msecs_to_jiffies(timeout);
+
+	do {
 		unsigned long flags;
 		unsigned char x;
 
-		schedule_timeout_uninterruptible(1);
-
 		spin_lock_irqsave(&s->lock, flags);
 		x = inb(HOST_DATA_IO(s->io_base));
 		spin_unlock_irqrestore(&s->lock, flags);
 		if ((x & 0xfe) == 0xfe)
 			return 1;
 
-		--timeout;
-	} /* while */
+		msleep(10);
+	} while (time_before(jiffies, end_time));
 
 	return 0;
 }
@@ -423,20 +415,20 @@ static int obp_startup_ack(struct soundscape *s, unsigned timeout)
  */
 static int host_startup_ack(struct soundscape *s, unsigned timeout)
 {
-	while (timeout != 0) {
+	unsigned long end_time = jiffies + msecs_to_jiffies(timeout);
+
+	do {
 		unsigned long flags;
 		unsigned char x;
 
-		schedule_timeout_uninterruptible(1);
-
 		spin_lock_irqsave(&s->lock, flags);
 		x = inb(HOST_DATA_IO(s->io_base));
 		spin_unlock_irqrestore(&s->lock, flags);
 		if (x == 0xfe)
 			return 1;
 
-		--timeout;
-	} /* while */
+		msleep(10);
+	} while (time_before(jiffies, end_time));
 
 	return 0;
 }
@@ -532,10 +524,10 @@ static int upload_dma_data(struct soundscape *s,
 	 * give it 5 seconds (max) ...
 	 */
 	ret = 0;
-	if (!obp_startup_ack(s, 5)) {
+	if (!obp_startup_ack(s, 5000)) {
 		snd_printk(KERN_ERR "sscape: No response from on-board processor after upload\n");
 		ret = -EAGAIN;
-	} else if (!host_startup_ack(s, 5)) {
+	} else if (!host_startup_ack(s, 5000)) {
 		snd_printk(KERN_ERR "sscape: SoundScape failed to initialise\n");
 		ret = -EAGAIN;
 	}
@@ -732,13 +724,7 @@ static int sscape_midi_get(struct snd_kcontrol *kctl,
 	unsigned long flags;
 
 	spin_lock_irqsave(&s->lock, flags);
-	set_host_mode_unsafe(s->io_base);
-
-	if (host_write_ctrl_unsafe(s->io_base, CMD_GET_MIDI_VOL, 100)) {
-		uctl->value.integer.value[0] = host_read_ctrl_unsafe(s->io_base, 100);
-	}
-
-	set_midi_mode_unsafe(s->io_base);
+	uctl->value.integer.value[0] = s->midi_vol;
 	spin_unlock_irqrestore(&s->lock, flags);
 	return 0;
 }
@@ -773,6 +759,7 @@ static int sscape_midi_put(struct snd_kcontrol *kctl,
 	change = (host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100)
 	          && host_write_ctrl_unsafe(s->io_base, ((unsigned char) uctl->value.integer. value[0]) & 127, 100)
 	          && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100));
+	s->midi_vol = (unsigned char) uctl->value.integer.value[0] & 127;
       __skip_change:
 
 	/*
@@ -815,12 +802,11 @@ static unsigned __devinit get_irq_config(int irq)
  * Perform certain arcane port-checks to see whether there
  * is a SoundScape board lurking behind the given ports.
  */
-static int __devinit detect_sscape(struct soundscape *s)
+static int __devinit detect_sscape(struct soundscape *s, long wss_io)
 {
 	unsigned long flags;
 	unsigned d;
 	int retval = 0;
-	int codec = s->wss_base;
 
 	spin_lock_irqsave(&s->lock, flags);
 
@@ -836,13 +822,11 @@ static int __devinit detect_sscape(struct soundscape *s)
 	if ((d & 0x80) != 0)
 		goto _done;
 
-	if (d == 0) {
-		s->codec_type = 1;
+	if (d == 0)
 		s->ic_type = IC_ODIE;
-	} else if ((d & 0x60) != 0) {
-		s->codec_type = 2;
+	else if ((d & 0x60) != 0)
 		s->ic_type = IC_OPUS;
-	} else
+	else
 		goto _done;
 
 	outb(0xfa, ODIE_ADDR_IO(s->io_base));
@@ -862,10 +846,10 @@ static int __devinit detect_sscape(struct soundscape *s)
 	sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0);
 
 	if (s->type == SSCAPE_VIVO)
-		codec += 4;
+		wss_io += 4;
 	/* wait for WSS codec */
 	for (d = 0; d < 500; d++) {
-		if ((inb(codec) & 0x80) == 0)
+		if ((inb(wss_io) & 0x80) == 0)
 			break;
 		spin_unlock_irqrestore(&s->lock, flags);
 		msleep(1);
@@ -955,82 +939,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l
 
 
 /*
- * Override for the CS4231 playback format function.
- * The AD1845 has much simpler format and rate selection.
- */
-static void ad1845_playback_format(struct snd_wss *chip,
-				   struct snd_pcm_hw_params *params,
-				   unsigned char format)
-{
-	unsigned long flags;
-	unsigned rate = params_rate(params);
-
-	/*
-	 * The AD1845 can't handle sample frequencies
-	 * outside of 4 kHZ to 50 kHZ
-	 */
-	if (rate > 50000)
-		rate = 50000;
-	else if (rate < 4000)
-		rate = 4000;
-
-	spin_lock_irqsave(&chip->reg_lock, flags);
-
-	/*
-	 * Program the AD1845 correctly for the playback stream.
-	 * Note that we do NOT need to toggle the MCE bit because
-	 * the PLAYBACK_ENABLE bit of the Interface Configuration
-	 * register is set.
-	 * 
-	 * NOTE: We seem to need to write to the MSB before the LSB
-	 *       to get the correct sample frequency.
-	 */
-	snd_wss_out(chip, CS4231_PLAYBK_FORMAT, (format & 0xf0));
-	snd_wss_out(chip, AD1845_FREQ_SEL_MSB, (unsigned char) (rate >> 8));
-	snd_wss_out(chip, AD1845_FREQ_SEL_LSB, (unsigned char) rate);
-
-	spin_unlock_irqrestore(&chip->reg_lock, flags);
-}
-
-/*
- * Override for the CS4231 capture format function. 
- * The AD1845 has much simpler format and rate selection.
- */
-static void ad1845_capture_format(struct snd_wss *chip,
-				  struct snd_pcm_hw_params *params,
-				  unsigned char format)
-{
-	unsigned long flags;
-	unsigned rate = params_rate(params);
-
-	/*
-	 * The AD1845 can't handle sample frequencies 
-	 * outside of 4 kHZ to 50 kHZ
-	 */
-	if (rate > 50000)
-		rate = 50000;
-	else if (rate < 4000)
-		rate = 4000;
-
-	spin_lock_irqsave(&chip->reg_lock, flags);
-
-	/*
-	 * Program the AD1845 correctly for the playback stream.
-	 * Note that we do NOT need to toggle the MCE bit because
-	 * the CAPTURE_ENABLE bit of the Interface Configuration
-	 * register is set.
-	 *
-	 * NOTE: We seem to need to write to the MSB before the LSB
-	 *       to get the correct sample frequency.
-	 */
-	snd_wss_out(chip, CS4231_REC_FORMAT, (format & 0xf0));
-	snd_wss_out(chip, AD1845_FREQ_SEL_MSB, (unsigned char) (rate >> 8));
-	snd_wss_out(chip, AD1845_FREQ_SEL_LSB, (unsigned char) rate);
-
-	spin_unlock_irqrestore(&chip->reg_lock, flags);
-}
-
-/*
  * Create an AD1845 PCM subdevice on the SoundScape. The AD1845
  * is very much like a CS4231, with a few extra bits. We will
  * try to support at least some of the extra bits by overriding
@@ -1055,11 +963,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port,
 		unsigned long flags;
 		struct snd_pcm *pcm;
 
-#define AD1845_FREQ_SEL_ENABLE  0x08
-
-#define AD1845_PWR_DOWN_CTRL   0x1b
-#define AD1845_CRYS_CLOCK_SEL  0x1d
-
 /*
  * It turns out that the PLAYBACK_ENABLE bit is set
  * by the lowlevel driver ...
@@ -1074,7 +977,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port,
  */
 
 		if (sscape->type != SSCAPE_VIVO) {
-			int val;
 			/*
 			 * The input clock frequency on the SoundScape must
 			 * be 14.31818 MHz, because we must set this register
@@ -1082,22 +984,10 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port,
 			 */
 			snd_wss_mce_up(chip);
 			spin_lock_irqsave(&chip->reg_lock, flags);
-			snd_wss_out(chip, AD1845_CRYS_CLOCK_SEL, 0x20);
+			snd_wss_out(chip, AD1845_CLOCK, 0x20);
 			spin_unlock_irqrestore(&chip->reg_lock, flags);
 			snd_wss_mce_down(chip);
 
-			/*
-			 * More custom configuration:
-			 * a) select "mode 2" and provide a current drive of 8mA
-			 * b) enable frequency selection (for capture/playback)
-			 */
-			spin_lock_irqsave(&chip->reg_lock, flags);
-			snd_wss_out(chip, CS4231_MISC_INFO,
-				    CS4231_MODE2 | 0x10);
-			val = snd_wss_in(chip, AD1845_PWR_DOWN_CTRL);
-			snd_wss_out(chip, AD1845_PWR_DOWN_CTRL,
-				    val | AD1845_FREQ_SEL_ENABLE);
-			spin_unlock_irqrestore(&chip->reg_lock, flags);
 		}
 
 		err = snd_wss_pcm(chip, 0, &pcm);
@@ -1113,11 +1003,13 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port,
 					    "for AD1845 chip\n");
 			goto _error;
 		}
-		err = snd_wss_timer(chip, 0, NULL);
-		if (err < 0) {
-			snd_printk(KERN_ERR "sscape: No timer device "
-					    "for AD1845 chip\n");
-			goto _error;
+		if (chip->hardware != WSS_HW_AD1848) {
+			err = snd_wss_timer(chip, 0, NULL);
+			if (err < 0) {
+				snd_printk(KERN_ERR "sscape: No timer device "
+						    "for AD1845 chip\n");
+				goto _error;
+			}
 		}
 
 		if (sscape->type != SSCAPE_VIVO) {
@@ -1128,8 +1020,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port,
 						    "MIDI mixer control\n");
 				goto _error;
 			}
-			chip->set_playback_format = ad1845_playback_format;
-			chip->set_capture_format = ad1845_capture_format;
 		}
 
 		strcpy(card->driver, "SoundScape");
@@ -1157,7 +1047,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card)
 	unsigned dma_cfg;
 	unsigned irq_cfg;
 	unsigned mpu_irq_cfg;
-	unsigned xport;
 	struct resource *io_res;
 	struct resource *wss_res;
 	unsigned long flags;
@@ -1177,15 +1066,15 @@ static int __devinit create_sscape(int dev, struct snd_card *card)
 		printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]);
 		return -ENXIO;
 	}
-	xport = port[dev];
 
 	/*
 	 * Grab IO ports that we will need to probe so that we
 	 * can detect and control this hardware ...
 	 */
-	io_res = request_region(xport, 8, "SoundScape");
+	io_res = request_region(port[dev], 8, "SoundScape");
 	if (!io_res) {
-		snd_printk(KERN_ERR "sscape: can't grab port 0x%x\n", xport);
+		snd_printk(KERN_ERR
+			   "sscape: can't grab port 0x%lx\n", port[dev]);
 		return -EBUSY;
 	}
 	wss_res = NULL;
@@ -1212,10 +1101,9 @@ static int __devinit create_sscape(int dev, struct snd_card *card)
 	spin_lock_init(&sscape->fwlock);
 	sscape->io_res = io_res;
 	sscape->wss_res = wss_res;
-	sscape->io_base = xport;
-	sscape->wss_base = wss_port[dev];
+	sscape->io_base = port[dev];
 
-	if (!detect_sscape(sscape)) {
+	if (!detect_sscape(sscape, wss_port[dev])) {
 		printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base);
 		err = -ENODEV;
 		goto _release_dma;
@@ -1288,12 +1176,11 @@ static int __devinit create_sscape(int dev, struct snd_card *card)
 	}
 #define MIDI_DEVNUM  0
 	if (sscape->type != SSCAPE_VIVO) {
-		err = create_mpu401(card, MIDI_DEVNUM,
-				    MPU401_IO(xport), mpu_irq[dev]);
+		err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]);
 		if (err < 0) {
 			printk(KERN_ERR "sscape: Failed to create "
-					"MPU-401 device at 0x%x\n",
-					MPU401_IO(xport));
+					"MPU-401 device at 0x%lx\n",
+					port[dev]);
 			goto _release_dma;
 		}
 
@@ -1357,10 +1244,10 @@ static int __devinit snd_sscape_probe(struct device *pdev, unsigned int dev)
 	struct soundscape *sscape;
 	int ret;
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-			    sizeof(struct soundscape));
-	if (!card)
-		return -ENOMEM;
+	ret = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct soundscape), &card);
+	if (ret < 0)
+		return ret;
 
 	sscape = get_card_soundscape(card);
 	sscape->type = SSCAPE;
@@ -1462,10 +1349,10 @@ static int __devinit sscape_pnp_detect(struct pnp_card_link *pcard,
 	 * Create a new ALSA sound card entry, in anticipation
 	 * of detecting our hardware ...
 	 */
-	card = snd_card_new(index[idx], id[idx], THIS_MODULE,
-			    sizeof(struct soundscape));
-	if (!card)
-		return -ENOMEM;
+	ret = snd_card_create(index[idx], id[idx], THIS_MODULE,
+			      sizeof(struct soundscape), &card);
+	if (ret < 0)
+		return ret;
 
 	sscape = get_card_soundscape(card);
 
diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c
index 4c095bc7c729..a34ae7b1f7d0 100644
--- a/sound/isa/wavefront/wavefront.c
+++ b/sound/isa/wavefront/wavefront.c
@@ -338,15 +338,16 @@ snd_wavefront_free(struct snd_card *card)
 	}
 }
 
-static struct snd_card *snd_wavefront_card_new(int dev)
+static int snd_wavefront_card_new(int dev, struct snd_card **cardp)
 {
 	struct snd_card *card;
 	snd_wavefront_card_t *acard;
+	int err;
 
-	card = snd_card_new (index[dev], id[dev], THIS_MODULE,
-			     sizeof(snd_wavefront_card_t));
-	if (card == NULL)
-		return NULL;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(snd_wavefront_card_t), &card);
+	if (err < 0)
+		return err;
 
 	acard = card->private_data;
 	acard->wavefront.irq = -1;
@@ -357,7 +358,8 @@ static struct snd_card *snd_wavefront_card_new(int dev)
 	acard->wavefront.card = card;
 	card->private_free = snd_wavefront_free;
 
-	return card;
+	*cardp = card;
+	return 0;
 }
 
 static int __devinit
@@ -551,11 +553,11 @@ static int __devinit snd_wavefront_isa_match(struct device *pdev,
 		return 0;
 #endif
 	if (cs4232_pcm_port[dev] == SNDRV_AUTO_PORT) {
-		snd_printk("specify CS4232 port\n");
+		snd_printk(KERN_ERR "specify CS4232 port\n");
 		return 0;
 	}
 	if (ics2115_port[dev] == SNDRV_AUTO_PORT) {
-		snd_printk("specify ICS2115 port\n");
+		snd_printk(KERN_ERR "specify ICS2115 port\n");
 		return 0;
 	}
 	return 1;
@@ -567,9 +569,9 @@ static int __devinit snd_wavefront_isa_probe(struct device *pdev,
 	struct snd_card *card;
 	int err;
 
-	card = snd_wavefront_card_new(dev);
-	if (! card)
-		return -ENOMEM;
+	err = snd_wavefront_card_new(dev, &card);
+	if (err < 0)
+		return err;
 	snd_card_set_dev(card, pdev);
 	if ((err = snd_wavefront_probe(card, dev)) < 0) {
 		snd_card_free(card);
@@ -616,9 +618,9 @@ static int __devinit snd_wavefront_pnp_detect(struct pnp_card_link *pcard,
 	if (dev >= SNDRV_CARDS)
 		return -ENODEV;
 
-	card = snd_wavefront_card_new(dev);
-	if (! card)
-		return -ENOMEM;
+	res = snd_wavefront_card_new(dev, &card);
+	if (res < 0)
+		return res;
 
 	if (snd_wavefront_pnp (dev, card->private_data, pcard, pid) < 0) {
 		if (cs4232_pcm_port[dev] == SNDRV_AUTO_PORT) {
diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c
index 4c410820a994..beb312cca75b 100644
--- a/sound/isa/wavefront/wavefront_synth.c
+++ b/sound/isa/wavefront/wavefront_synth.c
@@ -633,7 +633,7 @@ wavefront_get_sample_status (snd_wavefront_t *dev, int assume_rom)
 		wbuf[1] = i >> 7;
 
 		if (snd_wavefront_cmd (dev, WFC_IDENTIFY_SAMPLE_TYPE, rbuf, wbuf)) {
-			snd_printk("cannot identify sample "
+			snd_printk(KERN_WARNING "cannot identify sample "
 				   "type of slot %d\n", i);
 			dev->sample_status[i] = WF_ST_EMPTY;
 			continue;
diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c
index 3d6c5f2838af..5d2ba1b749ab 100644
--- a/sound/isa/wss/wss_lib.c
+++ b/sound/isa/wss/wss_lib.c
@@ -181,25 +181,6 @@ static void snd_wss_wait(struct snd_wss *chip)
 		udelay(100);
 }
 
-static void snd_wss_outm(struct snd_wss *chip, unsigned char reg,
-			    unsigned char mask, unsigned char value)
-{
-	unsigned char tmp = (chip->image[reg] & mask) | value;
-
-	snd_wss_wait(chip);
-#ifdef CONFIG_SND_DEBUG
-	if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT)
-		snd_printk("outm: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value);
-#endif
-	chip->image[reg] = tmp;
-	if (!chip->calibrate_mute) {
-		wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg);
-		wmb();
-		wss_outb(chip, CS4231P(REG), tmp);
-		mb();
-	}
-}
-
 static void snd_wss_dout(struct snd_wss *chip, unsigned char reg,
 			 unsigned char value)
 {
@@ -219,7 +200,8 @@ void snd_wss_out(struct snd_wss *chip, unsigned char reg, unsigned char value)
 	snd_wss_wait(chip);
 #ifdef CONFIG_SND_DEBUG
 	if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT)
-		snd_printk("out: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value);
+		snd_printk(KERN_DEBUG "out: auto calibration time out "
+			   "- reg = 0x%x, value = 0x%x\n", reg, value);
 #endif
 	wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg);
 	wss_outb(chip, CS4231P(REG), value);
@@ -235,7 +217,8 @@ unsigned char snd_wss_in(struct snd_wss *chip, unsigned char reg)
 	snd_wss_wait(chip);
 #ifdef CONFIG_SND_DEBUG
 	if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT)
-		snd_printk("in: auto calibration time out - reg = 0x%x\n", reg);
+		snd_printk(KERN_DEBUG "in: auto calibration time out "
+			   "- reg = 0x%x\n", reg);
 #endif
 	wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg);
 	mb();
@@ -252,7 +235,7 @@ void snd_cs4236_ext_out(struct snd_wss *chip, unsigned char reg,
 	wss_outb(chip, CS4231P(REG), val);
 	chip->eimage[CS4236_REG(reg)] = val;
 #if 0
-	printk("ext out : reg = 0x%x, val = 0x%x\n", reg, val);
+	printk(KERN_DEBUG "ext out : reg = 0x%x, val = 0x%x\n", reg, val);
 #endif
 }
 EXPORT_SYMBOL(snd_cs4236_ext_out);
@@ -268,7 +251,8 @@ unsigned char snd_cs4236_ext_in(struct snd_wss *chip, unsigned char reg)
 	{
 		unsigned char res;
 		res = wss_inb(chip, CS4231P(REG));
-		printk("ext in : reg = 0x%x, val = 0x%x\n", reg, res);
+		printk(KERN_DEBUG "ext in : reg = 0x%x, val = 0x%x\n",
+		       reg, res);
 		return res;
 	}
 #endif
@@ -394,13 +378,16 @@ void snd_wss_mce_up(struct snd_wss *chip)
 	snd_wss_wait(chip);
 #ifdef CONFIG_SND_DEBUG
 	if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT)
-		snd_printk("mce_up - auto calibration time out (0)\n");
+		snd_printk(KERN_DEBUG
+			   "mce_up - auto calibration time out (0)\n");
 #endif
 	spin_lock_irqsave(&chip->reg_lock, flags);
 	chip->mce_bit |= CS4231_MCE;
 	timeout = wss_inb(chip, CS4231P(REGSEL));
 	if (timeout == 0x80)
-		snd_printk("mce_up [0x%lx]: serious init problem - codec still busy\n", chip->port);
+		snd_printk(KERN_DEBUG "mce_up [0x%lx]: "
+			   "serious init problem - codec still busy\n",
+			   chip->port);
 	if (!(timeout & CS4231_MCE))
 		wss_outb(chip, CS4231P(REGSEL),
 			 chip->mce_bit | (timeout & 0x1f));
@@ -419,7 +406,9 @@ void snd_wss_mce_down(struct snd_wss *chip)
 
 #ifdef CONFIG_SND_DEBUG
 	if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT)
-		snd_printk("mce_down [0x%lx] - auto calibration time out (0)\n", (long)CS4231P(REGSEL));
+		snd_printk(KERN_DEBUG "mce_down [0x%lx] - "
+			   "auto calibration time out (0)\n",
+			   (long)CS4231P(REGSEL));
 #endif
 	spin_lock_irqsave(&chip->reg_lock, flags);
 	chip->mce_bit &= ~CS4231_MCE;
@@ -427,7 +416,9 @@ void snd_wss_mce_down(struct snd_wss *chip)
 	wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f));
 	spin_unlock_irqrestore(&chip->reg_lock, flags);
 	if (timeout == 0x80)
-		snd_printk("mce_down [0x%lx]: serious init problem - codec still busy\n", chip->port);
+		snd_printk(KERN_DEBUG "mce_down [0x%lx]: "
+			   "serious init problem - codec still busy\n",
+			   chip->port);
 	if ((timeout & CS4231_MCE) == 0 || !(chip->hardware & hw_mask))
 		return;
 
@@ -565,7 +556,7 @@ static unsigned char snd_wss_get_format(struct snd_wss *chip,
 	if (channels > 1)
 		rformat |= CS4231_STEREO;
 #if 0
-	snd_printk("get_format: 0x%x (mode=0x%x)\n", format, mode);
+	snd_printk(KERN_DEBUG "get_format: 0x%x (mode=0x%x)\n", format, mode);
 #endif
 	return rformat;
 }
@@ -587,7 +578,15 @@ static void snd_wss_calibrate_mute(struct snd_wss *chip, int mute)
 			     chip->image[CS4231_RIGHT_INPUT]);
 		snd_wss_dout(chip, CS4231_LOOPBACK,
 			     chip->image[CS4231_LOOPBACK]);
+	} else {
+		snd_wss_dout(chip, CS4231_LEFT_INPUT,
+			     0);
+		snd_wss_dout(chip, CS4231_RIGHT_INPUT,
+			     0);
+		snd_wss_dout(chip, CS4231_LOOPBACK,
+			     0xfd);
 	}
+
 	snd_wss_dout(chip, CS4231_AUX1_LEFT_INPUT,
 		     mute | chip->image[CS4231_AUX1_LEFT_INPUT]);
 	snd_wss_dout(chip, CS4231_AUX1_RIGHT_INPUT,
@@ -630,7 +629,6 @@ static void snd_wss_playback_format(struct snd_wss *chip,
 	int full_calib = 1;
 
 	mutex_lock(&chip->mce_mutex);
-	snd_wss_calibrate_mute(chip, 1);
 	if (chip->hardware == WSS_HW_CS4231A ||
 	    (chip->hardware & WSS_HW_CS4232_MASK)) {
 		spin_lock_irqsave(&chip->reg_lock, flags);
@@ -646,6 +644,24 @@ static void snd_wss_playback_format(struct snd_wss *chip,
 			full_calib = 0;
 		}
 		spin_unlock_irqrestore(&chip->reg_lock, flags);
+	} else if (chip->hardware == WSS_HW_AD1845) {
+		unsigned rate = params_rate(params);
+
+		/*
+		 * Program the AD1845 correctly for the playback stream.
+		 * Note that we do NOT need to toggle the MCE bit because
+		 * the PLAYBACK_ENABLE bit of the Interface Configuration
+		 * register is set.
+		 *
+		 * NOTE: We seem to need to write to the MSB before the LSB
+		 *       to get the correct sample frequency.
+		 */
+		spin_lock_irqsave(&chip->reg_lock, flags);
+		snd_wss_out(chip, CS4231_PLAYBK_FORMAT, (pdfr & 0xf0));
+		snd_wss_out(chip, AD1845_UPR_FREQ_SEL, (rate >> 8) & 0xff);
+		snd_wss_out(chip, AD1845_LWR_FREQ_SEL, rate & 0xff);
+		full_calib = 0;
+		spin_unlock_irqrestore(&chip->reg_lock, flags);
 	}
 	if (full_calib) {
 		snd_wss_mce_up(chip);
@@ -663,7 +679,6 @@ static void snd_wss_playback_format(struct snd_wss *chip,
 			udelay(100);	/* this seems to help */
 		snd_wss_mce_down(chip);
 	}
-	snd_wss_calibrate_mute(chip, 0);
 	mutex_unlock(&chip->mce_mutex);
 }
 
@@ -675,7 +690,6 @@ static void snd_wss_capture_format(struct snd_wss *chip,
 	int full_calib = 1;
 
 	mutex_lock(&chip->mce_mutex);
-	snd_wss_calibrate_mute(chip, 1);
 	if (chip->hardware == WSS_HW_CS4231A ||
 	    (chip->hardware & WSS_HW_CS4232_MASK)) {
 		spin_lock_irqsave(&chip->reg_lock, flags);
@@ -690,6 +704,24 @@ static void snd_wss_capture_format(struct snd_wss *chip,
 			full_calib = 0;
 		}
 		spin_unlock_irqrestore(&chip->reg_lock, flags);
+	} else if (chip->hardware == WSS_HW_AD1845) {
+		unsigned rate = params_rate(params);
+
+		/*
+		 * Program the AD1845 correctly for the capture stream.
+		 * Note that we do NOT need to toggle the MCE bit because
+		 * the PLAYBACK_ENABLE bit of the Interface Configuration
+		 * register is set.
+		 *
+		 * NOTE: We seem to need to write to the MSB before the LSB
+		 *       to get the correct sample frequency.
+		 */
+		spin_lock_irqsave(&chip->reg_lock, flags);
+		snd_wss_out(chip, CS4231_REC_FORMAT, (cdfr & 0xf0));
+		snd_wss_out(chip, AD1845_UPR_FREQ_SEL, (rate >> 8) & 0xff);
+		snd_wss_out(chip, AD1845_LWR_FREQ_SEL, rate & 0xff);
+		full_calib = 0;
+		spin_unlock_irqrestore(&chip->reg_lock, flags);
 	}
 	if (full_calib) {
 		snd_wss_mce_up(chip);
@@ -714,7 +746,6 @@ static void snd_wss_capture_format(struct snd_wss *chip,
 		spin_unlock_irqrestore(&chip->reg_lock, flags);
 		snd_wss_mce_down(chip);
 	}
-	snd_wss_calibrate_mute(chip, 0);
 	mutex_unlock(&chip->mce_mutex);
 }
 
@@ -771,10 +802,11 @@ static void snd_wss_init(struct snd_wss *chip)
 {
 	unsigned long flags;
 
+	snd_wss_calibrate_mute(chip, 1);
 	snd_wss_mce_down(chip);
 
 #ifdef SNDRV_DEBUG_MCE
-	snd_printk("init: (1)\n");
+	snd_printk(KERN_DEBUG "init: (1)\n");
 #endif
 	snd_wss_mce_up(chip);
 	spin_lock_irqsave(&chip->reg_lock, flags);
@@ -789,18 +821,20 @@ static void snd_wss_init(struct snd_wss *chip)
 	snd_wss_mce_down(chip);
 
 #ifdef SNDRV_DEBUG_MCE
-	snd_printk("init: (2)\n");
+	snd_printk(KERN_DEBUG "init: (2)\n");
 #endif
 
 	snd_wss_mce_up(chip);
 	spin_lock_irqsave(&chip->reg_lock, flags);
+	chip->image[CS4231_IFACE_CTRL] &= ~CS4231_AUTOCALIB;
+	snd_wss_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]);
 	snd_wss_out(chip,
 		    CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1]);
 	spin_unlock_irqrestore(&chip->reg_lock, flags);
 	snd_wss_mce_down(chip);
 
 #ifdef SNDRV_DEBUG_MCE
-	snd_printk("init: (3) - afei = 0x%x\n",
+	snd_printk(KERN_DEBUG "init: (3) - afei = 0x%x\n",
 		   chip->image[CS4231_ALT_FEATURE_1]);
 #endif
 
@@ -817,7 +851,7 @@ static void snd_wss_init(struct snd_wss *chip)
 	snd_wss_mce_down(chip);
 
 #ifdef SNDRV_DEBUG_MCE
-	snd_printk("init: (4)\n");
+	snd_printk(KERN_DEBUG "init: (4)\n");
 #endif
 
 	snd_wss_mce_up(chip);
@@ -827,9 +861,10 @@ static void snd_wss_init(struct snd_wss *chip)
 			    chip->image[CS4231_REC_FORMAT]);
 	spin_unlock_irqrestore(&chip->reg_lock, flags);
 	snd_wss_mce_down(chip);
+	snd_wss_calibrate_mute(chip, 0);
 
 #ifdef SNDRV_DEBUG_MCE
-	snd_printk("init: (5)\n");
+	snd_printk(KERN_DEBUG "init: (5)\n");
 #endif
 }
 
@@ -885,8 +920,6 @@ static void snd_wss_close(struct snd_wss *chip, unsigned int mode)
 		mutex_unlock(&chip->open_mutex);
 		return;
 	}
-	snd_wss_calibrate_mute(chip, 1);
-
 	/* disable IRQ */
 	spin_lock_irqsave(&chip->reg_lock, flags);
 	if (!(chip->hardware & WSS_HW_AD1848_MASK))
@@ -919,8 +952,6 @@ static void snd_wss_close(struct snd_wss *chip, unsigned int mode)
 	wss_outb(chip, CS4231P(STATUS), 0);	/* clear IRQ */
 	spin_unlock_irqrestore(&chip->reg_lock, flags);
 
-	snd_wss_calibrate_mute(chip, 0);
-
 	chip->mode = 0;
 	mutex_unlock(&chip->open_mutex);
 }
@@ -1113,7 +1144,7 @@ irqreturn_t snd_wss_interrupt(int irq, void *dev_id)
 	if (chip->hardware & WSS_HW_AD1848_MASK)
 		wss_outb(chip, CS4231P(STATUS), 0);
 	else
-		snd_wss_outm(chip, CS4231_IRQ_STATUS, status, 0);
+		snd_wss_out(chip, CS4231_IRQ_STATUS, status);
 	spin_unlock(&chip->reg_lock);
 	return IRQ_HANDLED;
 }
@@ -1278,7 +1309,8 @@ static int snd_wss_probe(struct snd_wss *chip)
 		} else if (rev == 0x03) {
 			chip->hardware = WSS_HW_CS4236B;
 		} else {
-			snd_printk("unknown CS chip with version 0x%x\n", rev);
+			snd_printk(KERN_ERR
+				   "unknown CS chip with version 0x%x\n", rev);
 			return -ENODEV;		/* unknown CS4231 chip? */
 		}
 	}
@@ -1314,6 +1346,10 @@ static int snd_wss_probe(struct snd_wss *chip)
 		chip->image[CS4231_ALT_FEATURE_2] =
 			chip->hardware == WSS_HW_INTERWAVE ? 0xc2 : 0x01;
 	}
+	/* enable fine grained frequency selection */
+	if (chip->hardware == WSS_HW_AD1845)
+		chip->image[AD1845_PWR_DOWN] = 8;
+
 	ptr = (unsigned char *) &chip->image;
 	regnum = (chip->hardware & WSS_HW_AD1848_MASK) ? 16 : 32;
 	snd_wss_mce_down(chip);
@@ -1342,7 +1378,10 @@ static int snd_wss_probe(struct snd_wss *chip)
 				case 6:
 					break;
 				default:
-					snd_printk("unknown CS4235 chip (enhanced version = 0x%x)\n", id);
+					snd_printk(KERN_WARNING
+						"unknown CS4235 chip "
+						"(enhanced version = 0x%x)\n",
+						id);
 				}
 			} else if ((id & 0x1f) == 0x0b) {	/* CS4236/B */
 				switch (id >> 5) {
@@ -1353,7 +1392,10 @@ static int snd_wss_probe(struct snd_wss *chip)
 					chip->hardware = WSS_HW_CS4236B;
 					break;
 				default:
-					snd_printk("unknown CS4236 chip (enhanced version = 0x%x)\n", id);
+					snd_printk(KERN_WARNING
+						"unknown CS4236 chip "
+						"(enhanced version = 0x%x)\n",
+						id);
 				}
 			} else if ((id & 0x1f) == 0x08) {	/* CS4237B */
 				chip->hardware = WSS_HW_CS4237B;
@@ -1364,7 +1406,10 @@ static int snd_wss_probe(struct snd_wss *chip)
 				case 7:
 					break;
 				default:
-					snd_printk("unknown CS4237B chip (enhanced version = 0x%x)\n", id);
+					snd_printk(KERN_WARNING
+						"unknown CS4237B chip "
+						"(enhanced version = 0x%x)\n",
+						id);
 				}
 			} else if ((id & 0x1f) == 0x09) {	/* CS4238B */
 				chip->hardware = WSS_HW_CS4238B;
@@ -1374,7 +1419,10 @@ static int snd_wss_probe(struct snd_wss *chip)
 				case 7:
 					break;
 				default:
-					snd_printk("unknown CS4238B chip (enhanced version = 0x%x)\n", id);
+					snd_printk(KERN_WARNING
+						"unknown CS4238B chip "
+						"(enhanced version = 0x%x)\n",
+						id);
 				}
 			} else if ((id & 0x1f) == 0x1e) {	/* CS4239 */
 				chip->hardware = WSS_HW_CS4239;
@@ -1384,10 +1432,15 @@ static int snd_wss_probe(struct snd_wss *chip)
 				case 6:
 					break;
 				default:
-					snd_printk("unknown CS4239 chip (enhanced version = 0x%x)\n", id);
+					snd_printk(KERN_WARNING
+						"unknown CS4239 chip "
+						"(enhanced version = 0x%x)\n",
+						id);
 				}
 			} else {
-				snd_printk("unknown CS4236/CS423xB chip (enhanced version = 0x%x)\n", id);
+				snd_printk(KERN_WARNING
+					   "unknown CS4236/CS423xB chip "
+					   "(enhanced version = 0x%x)\n", id);
 			}
 		}
 	}
@@ -1618,7 +1671,8 @@ static void snd_wss_resume(struct snd_wss *chip)
 	wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f));
 	spin_unlock_irqrestore(&chip->reg_lock, flags);
 	if (timeout == 0x80)
-		snd_printk("down [0x%lx]: serious init problem - codec still busy\n", chip->port);
+		snd_printk(KERN_ERR "down [0x%lx]: serious init problem "
+			   "- codec still busy\n", chip->port);
 	if ((timeout & CS4231_MCE) == 0 ||
 	    !(chip->hardware & (WSS_HW_CS4231_MASK | WSS_HW_CS4232_MASK))) {
 		return;
@@ -1628,7 +1682,7 @@ static void snd_wss_resume(struct snd_wss *chip)
 }
 #endif /* CONFIG_PM */
 
-static int snd_wss_free(struct snd_wss *chip)
+int snd_wss_free(struct snd_wss *chip)
 {
 	release_and_free_resource(chip->res_port);
 	release_and_free_resource(chip->res_cport);
@@ -1651,6 +1705,7 @@ static int snd_wss_free(struct snd_wss *chip)
 	kfree(chip);
 	return 0;
 }
+EXPORT_SYMBOL(snd_wss_free);
 
 static int snd_wss_dev_free(struct snd_device *device)
 {
@@ -1820,7 +1875,8 @@ int snd_wss_create(struct snd_card *card,
 #if 0
 	if (chip->hardware & WSS_HW_CS4232_MASK) {
 		if (chip->res_cport == NULL)
-			snd_printk("CS4232 control port features are not accessible\n");
+			snd_printk(KERN_ERR "CS4232 control port features are "
+				   "not accessible\n");
 	}
 #endif
 
diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c
index 1881cec11e78..3e763d6a5d67 100644
--- a/sound/mips/au1x00.c
+++ b/sound/mips/au1x00.c
@@ -636,9 +636,10 @@ au1000_init(void)
 	struct snd_card *card;
 	struct snd_au1000 *au1000;
 
-	card = snd_card_new(-1, "AC97", THIS_MODULE, sizeof(struct snd_au1000));
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(-1, "AC97", THIS_MODULE,
+			      sizeof(struct snd_au1000), &card);
+	if (err < 0)
+		return err;
 
 	card->private_free = snd_au1000_free;
 	au1000 = card->private_data;
@@ -678,7 +679,7 @@ au1000_init(void)
 		return err;
 	}
 
-	printk( KERN_INFO "ALSA AC97: Driver Initialized\n" );
+	printk(KERN_INFO "ALSA AC97: Driver Initialized\n");
 	au1000_card = card;
 	return 0;
 }
diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c
index db495be01861..c52691c2fc46 100644
--- a/sound/mips/hal2.c
+++ b/sound/mips/hal2.c
@@ -878,9 +878,9 @@ static int __devinit hal2_probe(struct platform_device *pdev)
 	struct snd_hal2 *chip;
 	int err;
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	err = hal2_create(card, &chip);
 	if (err < 0) {
diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c
index 4c63504348dc..66f3b48ceafc 100644
--- a/sound/mips/sgio2audio.c
+++ b/sound/mips/sgio2audio.c
@@ -936,9 +936,9 @@ static int __devinit snd_sgio2audio_probe(struct platform_device *pdev)
 	struct snd_sgio2audio *chip;
 	int err;
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	err = snd_sgio2audio_create(card, &chip);
 	if (err < 0) {
diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c
index 7cf9913a47b2..d12bd98a37ba 100644
--- a/sound/oss/ad1848.c
+++ b/sound/oss/ad1848.c
@@ -280,7 +280,7 @@ static void wait_for_calibration(ad1848_info * devc)
 	while (timeout > 0 && (ad_read(devc, 11) & 0x20))
 		timeout--;
 	if (ad_read(devc, 11) & 0x20)
-		if ( (devc->model != MD_1845) || (devc->model != MD_1845_SSCAPE))
+		if ((devc->model != MD_1845) && (devc->model != MD_1845_SSCAPE))
 			printk(KERN_WARNING "ad1848: Auto calibration timed out(3).\n");
 }
 
@@ -2107,7 +2107,7 @@ int ad1848_control(int cmd, int arg)
 	switch (cmd)
 	{
 		case AD1848_SET_XTAL:	/* Change clock frequency of AD1845 (only ) */
-			if (devc->model != MD_1845 || devc->model != MD_1845_SSCAPE)
+			if (devc->model != MD_1845 && devc->model != MD_1845_SSCAPE)
 				return -EINVAL;
 			spin_lock_irqsave(&devc->lock,flags);
 			ad_enter_MCE(devc);
diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c
index 81e1f443d094..4191acccbcdb 100644
--- a/sound/oss/au1550_ac97.c
+++ b/sound/oss/au1550_ac97.c
@@ -1627,7 +1627,9 @@ au1550_ioctl(struct inode *inode, struct file *file, unsigned int cmd,
 				    sizeof(abinfo)) ? -EFAULT : 0;
 
 	case SNDCTL_DSP_NONBLOCK:
+		spin_lock(&file->f_lock);
 		file->f_flags |= O_NONBLOCK;
+		spin_unlock(&file->f_lock);
 		return 0;
 
 	case SNDCTL_DSP_GETODELAY:
diff --git a/sound/oss/audio.c b/sound/oss/audio.c
index 89bd27a5e865..b69c05b7ea7b 100644
--- a/sound/oss/audio.c
+++ b/sound/oss/audio.c
@@ -433,7 +433,9 @@ int audio_ioctl(int dev, struct file *file, unsigned int cmd, void __user *arg)
 			return dma_ioctl(dev, cmd, arg);
 		
 		case SNDCTL_DSP_NONBLOCK:
+			spin_lock(&file->f_lock);
 			file->f_flags |= O_NONBLOCK;
+			spin_unlock(&file->f_lock);
 			return 0;
 
 		case SNDCTL_DSP_GETCAPS:
diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c
index 1e90d769b62e..1bfcf7e88546 100644
--- a/sound/oss/dmabuf.c
+++ b/sound/oss/dmabuf.c
@@ -439,7 +439,7 @@ int DMAbuf_sync(int dev)
 			DMAbuf_launch_output(dev, dmap);
 		adev->dmap_out->flags |= DMA_SYNCING;
 		adev->dmap_out->underrun_count = 0;
-		while (!signal_pending(current) && n++ <= adev->dmap_out->nbufs && 
+		while (!signal_pending(current) && n++ < adev->dmap_out->nbufs &&
 		       adev->dmap_out->qlen && adev->dmap_out->underrun_count == 0) {
 			long t = dmabuf_timeout(dmap);
 			spin_unlock_irqrestore(&dmap->lock,flags);
diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c
index 38931f2f6967..1f4774123064 100644
--- a/sound/oss/dmasound/dmasound_atari.c
+++ b/sound/oss/dmasound/dmasound_atari.c
@@ -1524,7 +1524,7 @@ static SETTINGS def_soft = {
 	.speed	= 8000
 } ;
 
-static MACHINE machTT = {
+static __initdata MACHINE machTT = {
 	.name		= "Atari",
 	.name2		= "TT",
 	.owner		= THIS_MODULE,
@@ -1553,7 +1553,7 @@ static MACHINE machTT = {
 	.capabilities	=  DSP_CAP_BATCH	/* As per SNDCTL_DSP_GETCAPS */
 };
 
-static MACHINE machFalcon = {
+static __initdata MACHINE machFalcon = {
 	.name		= "Atari",
 	.name2		= "FALCON",
 	.dma_alloc	= AtaAlloc,
diff --git a/sound/oss/pas2_card.c b/sound/oss/pas2_card.c
index 25f3a22c52ee..7f377ec3486d 100644
--- a/sound/oss/pas2_card.c
+++ b/sound/oss/pas2_card.c
@@ -156,9 +156,7 @@ static int __init config_pas_hw(struct address_info *hw_config)
 						 * 0x80
 						 */ , 0xB88);
 
-	pas_write(0x80
-		  | joystick?0x40:0
-		  ,0xF388);
+	pas_write(0x80 | (joystick ? 0x40 : 0), 0xF388);
 
 	if (pas_irq < 0 || pas_irq > 15)
 	{
diff --git a/sound/oss/pss.c b/sound/oss/pss.c
index 16ed06950dc1..16517a5a1301 100644
--- a/sound/oss/pss.c
+++ b/sound/oss/pss.c
@@ -457,10 +457,9 @@ static void pss_mixer_reset(pss_confdata *devc)
 	}
 }
 
-static int set_volume_mono(unsigned __user *p, int *aleft)
+static int set_volume_mono(unsigned __user *p, unsigned int *aleft)
 {
-	int left;
-	unsigned volume;
+	unsigned int left, volume;
 	if (get_user(volume, p))
 		return -EFAULT;
 	
@@ -471,10 +470,11 @@ static int set_volume_mono(unsigned __user *p, int *aleft)
 	return 0;
 }
 
-static int set_volume_stereo(unsigned __user *p, int *aleft, int *aright)
+static int set_volume_stereo(unsigned __user *p,
+			     unsigned int *aleft,
+			     unsigned int *aright)
 {
-	int left, right;
-	unsigned volume;
+	unsigned int left, right, volume;
 	if (get_user(volume, p))
 		return -EFAULT;
 
diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c
index 5c215f787ca9..c79874696bec 100644
--- a/sound/oss/sequencer.c
+++ b/sound/oss/sequencer.c
@@ -212,7 +212,6 @@ int sequencer_write(int dev, struct file *file, const char __user *buf, int coun
 {
 	unsigned char event_rec[EV_SZ], ev_code;
 	int p = 0, c, ev_size;
-	int err;
 	int mode = translate_mode(file);
 
 	dev = dev >> 4;
@@ -285,7 +284,7 @@ int sequencer_write(int dev, struct file *file, const char __user *buf, int coun
 		{
 			if (!midi_opened[event_rec[2]])
 			{
-				int mode;
+				int err, mode;
 				int dev = event_rec[2];
 
 				if (dev >= max_mididev || midi_devs[dev]==NULL)
diff --git a/sound/oss/sh_dac_audio.c b/sound/oss/sh_dac_audio.c
index e5d423994918..78cfb66e4c59 100644
--- a/sound/oss/sh_dac_audio.c
+++ b/sound/oss/sh_dac_audio.c
@@ -135,7 +135,9 @@ static int dac_audio_ioctl(struct inode *inode, struct file *file,
 		return put_user(AFMT_U8, (int *)arg);
 
 	case SNDCTL_DSP_NONBLOCK:
+		spin_lock(&file->f_lock);
 		file->f_flags |= O_NONBLOCK;
+		spin_unlock(&file->f_lock);
 		return 0;
 
 	case SNDCTL_DSP_GETCAPS:
diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c
index 41562ecde5bb..1edab7b4ea83 100644
--- a/sound/oss/swarm_cs4297a.c
+++ b/sound/oss/swarm_cs4297a.c
@@ -2200,7 +2200,9 @@ static int cs4297a_ioctl(struct inode *inode, struct file *file,
 				    sizeof(abinfo)) ? -EFAULT : 0;
 
 	case SNDCTL_DSP_NONBLOCK:
+		spin_lock(&file->f_lock);
 		file->f_flags |= O_NONBLOCK;
+		spin_unlock(&file->f_lock);
 		return 0;
 
 	case SNDCTL_DSP_GETODELAY:
diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c
index 78b8acc7c3b9..187f72750e8f 100644
--- a/sound/oss/vwsnd.c
+++ b/sound/oss/vwsnd.c
@@ -2673,7 +2673,9 @@ static int vwsnd_audio_do_ioctl(struct inode *inode,
 
 	case SNDCTL_DSP_NONBLOCK:	/* _SIO  ('P',14) */
 		DBGX("SNDCTL_DSP_NONBLOCK\n");
+		spin_lock(&file->f_lock);
 		file->f_flags |= O_NONBLOCK;
+		spin_unlock(&file->f_lock);
 		return 0;
 
 	case SNDCTL_DSP_RESET:		/* _SIO  ('P', 0) */
diff --git a/sound/parisc/harmony.c b/sound/parisc/harmony.c
index 41f870f8a11d..6055fd6d3b38 100644
--- a/sound/parisc/harmony.c
+++ b/sound/parisc/harmony.c
@@ -975,9 +975,9 @@ snd_harmony_probe(struct parisc_device *padev)
 	struct snd_card *card;
 	struct snd_harmony *h;
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	err = snd_harmony_create(card, padev, &h);
 	if (err < 0)
diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig
index 82b9bddcdcd6..93422e3a3f0c 100644
--- a/sound/pci/Kconfig
+++ b/sound/pci/Kconfig
@@ -400,6 +400,26 @@ config SND_INDIGODJ
 	  To compile this driver as a module, choose M here: the module
 	  will be called snd-indigodj
 
+config SND_INDIGOIOX
+	tristate "(Echoaudio) Indigo IOx"
+	select FW_LOADER
+	select SND_PCM
+	help
+	  Say 'Y' or 'M' to include support for Echoaudio Indigo IOx.
+
+	  To compile this driver as a module, choose M here: the module
+	  will be called snd-indigoiox
+
+config SND_INDIGODJX
+	tristate "(Echoaudio) Indigo DJx"
+	select FW_LOADER
+	select SND_PCM
+	help
+	  Say 'Y' or 'M' to include support for Echoaudio Indigo DJx.
+
+	  To compile this driver as a module, choose M here: the module
+	  will be called snd-indigodjx
+
 config SND_EMU10K1
 	tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)"
 	select FW_LOADER
@@ -487,7 +507,7 @@ config SND_FM801
 config SND_FM801_TEA575X_BOOL
 	bool "ForteMedia FM801 + TEA5757 tuner"
 	depends on SND_FM801
-	depends on VIDEO_V4L1=y || VIDEO_V4L1=SND_FM801
+	depends on VIDEO_V4L2=y || VIDEO_V4L2=SND_FM801
 	help
 	  Say Y here to include support for soundcards based on the ForteMedia
 	  FM801 chip with a TEA5757 tuner connected to GPIO1-3 pins (Media
@@ -744,7 +764,8 @@ config SND_VIRTUOSO
 	select SND_OXYGEN_LIB
 	help
 	  Say Y here to include support for sound cards based on the
-	  Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2 and D2X.
+	  Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, and
+	  Essence STX.
 	  Support for the HDAV1.3 (Deluxe) is very experimental.
 
 	  To compile this driver as a module, choose M here: the module
diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c
index e2b843b4f9d0..97ee127ac33d 100644
--- a/sound/pci/ac97/ac97_codec.c
+++ b/sound/pci/ac97/ac97_codec.c
@@ -143,6 +143,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = {
 { 0x43525970, 0xfffffff8, "CS4202",		NULL,		NULL },
 { 0x43585421, 0xffffffff, "HSD11246",		NULL,		NULL },	// SmartMC II
 { 0x43585428, 0xfffffff8, "Cx20468",		patch_conexant,	NULL }, // SmartAMC fixme: the mask might be different
+{ 0x43585430, 0xffffffff, "Cx20468-31",		patch_conexant, NULL },
 { 0x43585431, 0xffffffff, "Cx20551",           patch_cx20551,  NULL },
 { 0x44543031, 0xfffffff0, "DT0398",		NULL,		NULL },
 { 0x454d4328, 0xffffffff, "EM28028",		NULL,		NULL },  // same as TR28028?
@@ -383,7 +384,7 @@ int snd_ac97_update_bits(struct snd_ac97 *ac97, unsigned short reg, unsigned sho
 
 EXPORT_SYMBOL(snd_ac97_update_bits);
 
-/* no lock version - see snd_ac97_updat_bits() */
+/* no lock version - see snd_ac97_update_bits() */
 int snd_ac97_update_bits_nolock(struct snd_ac97 *ac97, unsigned short reg,
 				unsigned short mask, unsigned short value)
 {
@@ -1643,7 +1644,10 @@ static int snd_ac97_modem_build(struct snd_card *card, struct snd_ac97 * ac97)
 {
 	int err, idx;
 
-	//printk("AC97_GPIO_CFG = %x\n",snd_ac97_read(ac97,AC97_GPIO_CFG));
+	/*
+	printk(KERN_DEBUG "AC97_GPIO_CFG = %x\n",
+	       snd_ac97_read(ac97,AC97_GPIO_CFG));
+	*/
 	snd_ac97_write(ac97, AC97_GPIO_CFG, 0xffff & ~(AC97_GPIO_LINE1_OH));
 	snd_ac97_write(ac97, AC97_GPIO_POLARITY, 0xffff & ~(AC97_GPIO_LINE1_OH));
 	snd_ac97_write(ac97, AC97_GPIO_STICKY, 0xffff);
diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c
index 060ea59d5f02..73b17d526c8b 100644
--- a/sound/pci/ac97/ac97_proc.c
+++ b/sound/pci/ac97/ac97_proc.c
@@ -125,6 +125,8 @@ static void snd_ac97_proc_read_main(struct snd_ac97 *ac97, struct snd_info_buffe
         snd_iprintf(buffer, "PCI Subsys Device: 0x%04x\n\n",
                     ac97->subsystem_device);
 
+	snd_iprintf(buffer, "Flags: %x\n", ac97->flags);
+
 	if ((ac97->ext_id & AC97_EI_REV_MASK) >= AC97_EI_REV_23) {
 		val = snd_ac97_read(ac97, AC97_INT_PAGING);
 		snd_ac97_update_bits(ac97, AC97_INT_PAGING,
diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c
index a7f38e63303f..d1f242bd0ac5 100644
--- a/sound/pci/ad1889.c
+++ b/sound/pci/ad1889.c
@@ -995,10 +995,10 @@ snd_ad1889_probe(struct pci_dev *pci,
 	}
 
 	/* (2) */
-	card = snd_card_new(index[devno], id[devno], THIS_MODULE, 0);
+	err = snd_card_create(index[devno], id[devno], THIS_MODULE, 0, &card);
 	/* XXX REVISIT: we can probably allocate chip in this call */
-	if (card == NULL)
-		return -ENOMEM;
+	if (err < 0)
+		return err;
 
 	strcpy(card->driver, "AD1889");
 	strcpy(card->shortname, "Analog Devices AD1889");
diff --git a/sound/pci/ak4531_codec.c b/sound/pci/ak4531_codec.c
index 0f819ddb3ebf..fd135e3d8a84 100644
--- a/sound/pci/ak4531_codec.c
+++ b/sound/pci/ak4531_codec.c
@@ -51,7 +51,8 @@ static void snd_ak4531_dump(struct snd_ak4531 *ak4531)
 	int idx;
 	
 	for (idx = 0; idx < 0x19; idx++)
-		printk("ak4531 0x%x: 0x%x\n", idx, ak4531->regs[idx]);
+		printk(KERN_DEBUG "ak4531 0x%x: 0x%x\n",
+		       idx, ak4531->regs[idx]);
 }
 
 #endif
diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c
index 1a0fd65ec280..4edf270a7809 100644
--- a/sound/pci/ali5451/ali5451.c
+++ b/sound/pci/ali5451/ali5451.c
@@ -2142,7 +2142,7 @@ static int __devinit snd_ali_resources(struct snd_ali *codec)
 {
 	int err;
 
-	snd_ali_printk("resouces allocation ...\n");
+	snd_ali_printk("resources allocation ...\n");
 	err = pci_request_regions(codec->pci, "ALI 5451");
 	if (err < 0)
 		return err;
@@ -2154,7 +2154,7 @@ static int __devinit snd_ali_resources(struct snd_ali *codec)
 		return -EBUSY;
 	}
 	codec->irq = codec->pci->irq;
-	snd_ali_printk("resouces allocated.\n");
+	snd_ali_printk("resources allocated.\n");
 	return 0;
 }
 static int snd_ali_dev_free(struct snd_device *device)
@@ -2307,9 +2307,9 @@ static int __devinit snd_ali_probe(struct pci_dev *pci,
 
 	snd_ali_printk("probe ...\n");
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (!card)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	err = snd_ali_create(card, pci, pcm_channels, spdif, &codec);
 	if (err < 0)
diff --git a/sound/pci/als300.c b/sound/pci/als300.c
index 8df6824b51cd..009b4c8225a5 100644
--- a/sound/pci/als300.c
+++ b/sound/pci/als300.c
@@ -91,7 +91,7 @@
 #define DEBUG_PLAY_REC	0
 
 #if DEBUG_CALLS
-#define snd_als300_dbgcalls(format, args...) printk(format, ##args)
+#define snd_als300_dbgcalls(format, args...) printk(KERN_DEBUG format, ##args)
 #define snd_als300_dbgcallenter() printk(KERN_ERR "--> %s\n", __func__)
 #define snd_als300_dbgcallleave() printk(KERN_ERR "<-- %s\n", __func__)
 #else
@@ -812,10 +812,10 @@ static int __devinit snd_als300_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
 
-	if (card == NULL)
-		return -ENOMEM;
+	if (err < 0)
+		return err;
 
 	chip_type = pci_id->driver_data;
 
diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c
index ba570053d4d5..542a0c65a92c 100644
--- a/sound/pci/als4000.c
+++ b/sound/pci/als4000.c
@@ -889,12 +889,13 @@ static int __devinit snd_card_als4000_probe(struct pci_dev *pci,
 	pci_write_config_word(pci, PCI_COMMAND, word | PCI_COMMAND_IO);
 	pci_set_master(pci);
 	
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 
-			    sizeof(*acard) /* private_data: acard */);
-	if (card == NULL) {
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 
+			      sizeof(*acard) /* private_data: acard */,
+			      &card);
+	if (err < 0) {
 		pci_release_regions(pci);
 		pci_disable_device(pci);
-		return -ENOMEM;
+		return err;
 	}
 
 	acard = card->private_data;
diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c
index 226fe8237d31..9ce8548c03e4 100644
--- a/sound/pci/atiixp.c
+++ b/sound/pci/atiixp.c
@@ -1645,9 +1645,9 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci,
 	struct atiixp *chip;
 	int err;
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	strcpy(card->driver, spdif_aclink ? "ATIIXP" : "ATIIXP-SPDMA");
 	strcpy(card->shortname, "ATI IXP");
diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c
index 0e6e5cc1c501..c3136cccc559 100644
--- a/sound/pci/atiixp_modem.c
+++ b/sound/pci/atiixp_modem.c
@@ -1288,9 +1288,9 @@ static int __devinit snd_atiixp_probe(struct pci_dev *pci,
 	struct atiixp_modem *chip;
 	int err;
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	strcpy(card->driver, "ATIIXP-MODEM");
 	strcpy(card->shortname, "ATI IXP Modem");
diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c
index a36d4d1fd419..9ec122383eef 100644
--- a/sound/pci/au88x0/au88x0.c
+++ b/sound/pci/au88x0/au88x0.c
@@ -250,9 +250,9 @@ snd_vortex_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 		return -ENOENT;
 	}
 	// (2)
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	// (3)
 	if ((err = snd_vortex_create(card, pci, &chip)) < 0) {
diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c
index 649849e540d3..f4aa8ff6f5f9 100644
--- a/sound/pci/au88x0/au88x0_a3d.c
+++ b/sound/pci/au88x0/au88x0_a3d.c
@@ -462,9 +462,10 @@ static void a3dsrc_ZeroSliceIO(a3dsrc_t * a)
 /* Reset Single A3D source. */
 static void a3dsrc_ZeroState(a3dsrc_t * a)
 {
-
-	//printk("vortex: ZeroState slice: %d, source %d\n", a->slice, a->source);
-
+	/*
+	printk(KERN_DEBUG "vortex: ZeroState slice: %d, source %d\n",
+	       a->slice, a->source);
+	*/
 	a3dsrc_SetAtmosState(a, 0, 0, 0, 0);
 	a3dsrc_SetHrtfState(a, A3dHrirZeros, A3dHrirZeros);
 	a3dsrc_SetItdDline(a, A3dItdDlineZeros);
diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c
index b070e5714514..3906f5afe27a 100644
--- a/sound/pci/au88x0/au88x0_core.c
+++ b/sound/pci/au88x0/au88x0_core.c
@@ -1135,7 +1135,10 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma,
 			snd_pcm_sgbuf_get_addr(dma->substream, 0));
 		break;
 	}
-	//printk("vortex: cfg0 = 0x%x\nvortex: cfg1=0x%x\n", dma->cfg0, dma->cfg1);
+	/*
+	printk(KERN_DEBUG "vortex: cfg0 = 0x%x\nvortex: cfg1=0x%x\n",
+	       dma->cfg0, dma->cfg1);
+	*/
 	hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFCFG0 + (adbdma << 3), dma->cfg0);
 	hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFCFG1 + (adbdma << 3), dma->cfg1);
 
@@ -1959,7 +1962,7 @@ vortex_connect_codecplay(vortex_t * vortex, int en, unsigned char mixers[])
 					  ADB_CODECOUT(0 + 4));
 		vortex_connection_mix_adb(vortex, en, 0x11, mixers[3],
 					  ADB_CODECOUT(1 + 4));
-		//printk("SDAC detected ");
+		/* printk(KERN_DEBUG "SDAC detected "); */
 	}
 #else
 	// Use plain direct output to codec.
@@ -2013,7 +2016,11 @@ vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype)
 					resmap[restype] |= (1 << i);
 				else
 					vortex->dma_adb[i].resources[restype] |= (1 << i);
-				//printk("vortex: ResManager: type %d out %d\n", restype, i);
+				/*
+				printk(KERN_DEBUG
+				       "vortex: ResManager: type %d out %d\n",
+				       restype, i);
+				*/
 				return i;
 			}
 		}
@@ -2024,7 +2031,11 @@ vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype)
 		for (i = 0; i < qty; i++) {
 			if (resmap[restype] & (1 << i)) {
 				resmap[restype] &= ~(1 << i);
-				//printk("vortex: ResManager: type %d in %d\n",restype, i);
+				/*
+				printk(KERN_DEBUG
+				       "vortex: ResManager: type %d in %d\n",
+				       restype, i);
+				*/
 				return i;
 			}
 		}
@@ -2789,7 +2800,7 @@ vortex_translateformat(vortex_t * vortex, char bits, char nch, int encod)
 {
 	int a, this_194;
 
-	if ((bits != 8) || (bits != 16))
+	if ((bits != 8) && (bits != 16))
 		return -1;
 
 	switch (encod) {
diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c
index 978b856f5621..2805e34bd41d 100644
--- a/sound/pci/au88x0/au88x0_synth.c
+++ b/sound/pci/au88x0/au88x0_synth.c
@@ -213,38 +213,59 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt,
 	switch (reg) {
 		/* Voice specific parameters */
 	case 0:		/* running */
-		//printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_RUN(wt), (int)val);
+		/*
+		printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n",
+		       WT_RUN(wt), (int)val);
+		*/
 		hwwrite(vortex->mmio, WT_RUN(wt), val);
 		return 0xc;
 		break;
 	case 1:		/* param 0 */
-		//printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,0), (int)val);
+		/*
+		printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n",
+		       WT_PARM(wt,0), (int)val);
+		*/
 		hwwrite(vortex->mmio, WT_PARM(wt, 0), val);
 		return 0xc;
 		break;
 	case 2:		/* param 1 */
-		//printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,1), (int)val);
+		/*
+		printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n",
+		       WT_PARM(wt,1), (int)val);
+		*/
 		hwwrite(vortex->mmio, WT_PARM(wt, 1), val);
 		return 0xc;
 		break;
 	case 3:		/* param 2 */
-		//printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,2), (int)val);
+		/*
+		printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n",
+		       WT_PARM(wt,2), (int)val);
+		*/
 		hwwrite(vortex->mmio, WT_PARM(wt, 2), val);
 		return 0xc;
 		break;
 	case 4:		/* param 3 */
-		//printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,3), (int)val);
+		/*
+		printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n",
+		       WT_PARM(wt,3), (int)val);
+		*/
 		hwwrite(vortex->mmio, WT_PARM(wt, 3), val);
 		return 0xc;
 		break;
 	case 6:		/* mute */
-		//printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_MUTE(wt), (int)val);
+		/*
+		printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n",
+		       WT_MUTE(wt), (int)val);
+		*/
 		hwwrite(vortex->mmio, WT_MUTE(wt), val);
 		return 0xc;
 		break;
 	case 0xb:
 		{		/* delay */
-			//printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_DELAY(wt,0), (int)val);
+			/*
+			printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n",
+			       WT_DELAY(wt,0), (int)val);
+			*/
 			hwwrite(vortex->mmio, WT_DELAY(wt, 3), val);
 			hwwrite(vortex->mmio, WT_DELAY(wt, 2), val);
 			hwwrite(vortex->mmio, WT_DELAY(wt, 1), val);
@@ -272,7 +293,9 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt,
 		return 0;
 		break;
 	}
-	//printk("vortex: WT SetReg(0x%x) = 0x%08x\n", ecx, (int)val);
+	/*
+	printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", ecx, (int)val);
+	*/
 	hwwrite(vortex->mmio, ecx, val);
 	return 1;
 }
diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index c7c54e7748e9..8eea29fc42fe 100644
--- a/sound/pci/aw2/aw2-alsa.c
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -368,9 +368,9 @@ static int __devinit snd_aw2_probe(struct pci_dev *pci,
 	}
 
 	/* (2) Create card instance */
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	/* (3) Create main component */
 	err = snd_aw2_create(card, pci, &chip);
diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c
index 333007c523a1..e9e9b5821d41 100644
--- a/sound/pci/azt3328.c
+++ b/sound/pci/azt3328.c
@@ -211,25 +211,25 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}");
 #endif
 
 #if DEBUG_MIXER
-#define snd_azf3328_dbgmixer(format, args...) printk(format, ##args)
+#define snd_azf3328_dbgmixer(format, args...) printk(KERN_DEBUG format, ##args)
 #else
 #define snd_azf3328_dbgmixer(format, args...)
 #endif
 
 #if DEBUG_PLAY_REC
-#define snd_azf3328_dbgplay(format, args...) printk(KERN_ERR format, ##args)
+#define snd_azf3328_dbgplay(format, args...) printk(KERN_DEBUG format, ##args)
 #else
 #define snd_azf3328_dbgplay(format, args...)
 #endif
 
 #if DEBUG_MISC
-#define snd_azf3328_dbgtimer(format, args...) printk(KERN_ERR format, ##args)
+#define snd_azf3328_dbgtimer(format, args...) printk(KERN_DEBUG format, ##args)
 #else
 #define snd_azf3328_dbgtimer(format, args...)
 #endif
 
 #if DEBUG_GAME
-#define snd_azf3328_dbggame(format, args...) printk(KERN_ERR format, ##args)
+#define snd_azf3328_dbggame(format, args...) printk(KERN_DEBUG format, ##args)
 #else
 #define snd_azf3328_dbggame(format, args...)
 #endif
@@ -2216,9 +2216,9 @@ snd_azf3328_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	strcpy(card->driver, "AZF3328");
 	strcpy(card->shortname, "Aztech AZF3328 (PCI168)");
diff --git a/sound/pci/bt87x.c b/sound/pci/bt87x.c
index 1aa1c0402540..a299340519df 100644
--- a/sound/pci/bt87x.c
+++ b/sound/pci/bt87x.c
@@ -888,9 +888,9 @@ static int __devinit snd_bt87x_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (!card)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	err = snd_bt87x_create(card, pci, &chip);
 	if (err < 0)
diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c
index 0e62205d4081..df757575798a 100644
--- a/sound/pci/ca0106/ca0106_main.c
+++ b/sound/pci/ca0106/ca0106_main.c
@@ -255,6 +255,14 @@ static struct snd_ca0106_details ca0106_chip_details[] = {
 	   .gpio_type = 2,
 	   .i2c_adc = 1,
 	   .spi_dac = 1 } ,
+	/* Giga-byte GA-G1975X mobo
+	 * Novell bnc#395807
+	 */
+	/* FIXME: the GPIO and I2C setting aren't tested well */
+	{ .serial = 0x1458a006,
+	  .name = "Giga-byte GA-G1975X",
+	  .gpio_type = 1,
+	  .i2c_adc = 1 },
 	 /* Shuttle XPC SD31P which has an onboard Creative Labs
 	  * Sound Blaster Live! 24-bit EAX
 	  * high-definition 7.1 audio processor".
@@ -404,7 +412,9 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu,
 	}
 
 	tmp = reg << 25 | value << 16;
-	// snd_printk("I2C-write:reg=0x%x, value=0x%x\n", reg, value);
+	/*
+	snd_printk(KERN_DEBUG "I2C-write:reg=0x%x, value=0x%x\n", reg, value);
+	*/
 	/* Not sure what this I2C channel controls. */
 	/* snd_ca0106_ptr_write(emu, I2C_D0, 0, tmp); */
 
@@ -422,7 +432,7 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu,
 		/* Wait till the transaction ends */
 		while (1) {
 			status = snd_ca0106_ptr_read(emu, I2C_A, 0);
-                	//snd_printk("I2C:status=0x%x\n", status);
+			/*snd_printk(KERN_DEBUG "I2C:status=0x%x\n", status);*/
 			timeout++;
 			if ((status & I2C_A_ADC_START) == 0)
 				break;
@@ -521,7 +531,10 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr
         channel->number = channel_id;
 
 	channel->use = 1;
-        //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
+	/*
+	printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n",
+	       channel_id, chip, channel);
+	*/
         //channel->interrupt = snd_ca0106_pcm_channel_interrupt;
 	channel->epcm = epcm;
 	if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
@@ -614,7 +627,10 @@ static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substre
         channel->number = channel_id;
 
 	channel->use = 1;
-        //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
+	/*
+        printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n",
+	       channel_id, chip, channel);
+	*/
         //channel->interrupt = snd_ca0106_pcm_channel_interrupt;
         channel->epcm = epcm;
 	if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
@@ -705,9 +721,20 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream)
 	u32 reg71;
 	int i;
 	
-        //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1));
-        //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base);
-	//snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
+#if 0 /* debug */
+	snd_printk(KERN_DEBUG
+		   "prepare:channel_number=%d, rate=%d, format=0x%x, "
+		   "channels=%d, buffer_size=%ld, period_size=%ld, "
+		   "periods=%u, frames_to_bytes=%d\n",
+		   channel, runtime->rate, runtime->format,
+		   runtime->channels, runtime->buffer_size,
+		   runtime->period_size, runtime->periods,
+		   frames_to_bytes(runtime, 1));
+	snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, table_base=%p\n",
+		   runtime->dma_addr, runtime->dma_area, table_base);
+	snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",
+		   emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
+#endif /* debug */
 	/* Rate can be set per channel. */
 	/* reg40 control host to fifo */
 	/* reg71 controls DAC rate. */
@@ -799,9 +826,20 @@ static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream)
 	u32 reg71_set = 0;
 	u32 reg71;
 	
-        //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1));
-        //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base);
-	//snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
+#if 0 /* debug */
+	snd_printk(KERN_DEBUG
+		   "prepare:channel_number=%d, rate=%d, format=0x%x, "
+		   "channels=%d, buffer_size=%ld, period_size=%ld, "
+		   "periods=%u, frames_to_bytes=%d\n",
+		   channel, runtime->rate, runtime->format,
+		   runtime->channels, runtime->buffer_size,
+		   runtime->period_size, runtime->periods,
+		   frames_to_bytes(runtime, 1));
+        snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, table_base=%p\n",
+		   runtime->dma_addr, runtime->dma_area, table_base);
+	snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",
+		   emu->buffer.addr, emu->buffer.area, emu->buffer.bytes);
+#endif /* debug */
 	/* reg71 controls ADC rate. */
 	switch (runtime->rate) {
 	case 44100:
@@ -846,7 +884,14 @@ static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream)
 	}
 
 
-        //printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size,  frames_to_bytes(runtime, 1));
+	/*
+	printk(KERN_DEBUG
+	       "prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, "
+	       "buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",
+	       channel, runtime->rate, runtime->format, runtime->channels,
+	       runtime->buffer_size, runtime->period_size,
+	       frames_to_bytes(runtime, 1));
+	*/
 	snd_ca0106_ptr_write(emu, 0x13, channel, 0);
 	snd_ca0106_ptr_write(emu, CAPTURE_DMA_ADDR, channel, runtime->dma_addr);
 	snd_ca0106_ptr_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size)<<16); // buffer size in bytes
@@ -888,13 +933,13 @@ static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream,
 		runtime = s->runtime;
 		epcm = runtime->private_data;
 		channel = epcm->channel_id;
-		/* snd_printk("channel=%d\n",channel); */
+		/* snd_printk(KERN_DEBUG "channel=%d\n", channel); */
 		epcm->running = running;
 		basic |= (0x1 << channel);
 		extended |= (0x10 << channel);
                 snd_pcm_trigger_done(s, substream);
         }
-	/* snd_printk("basic=0x%x, extended=0x%x\n",basic, extended); */
+	/* snd_printk(KERN_DEBUG "basic=0x%x, extended=0x%x\n",basic, extended); */
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -972,8 +1017,13 @@ snd_ca0106_pcm_pointer_playback(struct snd_pcm_substream *substream)
 	ptr=ptr2;
         if (ptr >= runtime->buffer_size)
 		ptr -= runtime->buffer_size;
-	//printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate);
-
+	/*
+	printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, "
+	       "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n",
+	       ptr1, ptr2, ptr, (int)runtime->buffer_size,
+	       (int)runtime->period_size, (int)runtime->frame_bits,
+	       (int)runtime->rate);
+	*/
 	return ptr;
 }
 
@@ -995,8 +1045,13 @@ snd_ca0106_pcm_pointer_capture(struct snd_pcm_substream *substream)
 	ptr=ptr2;
         if (ptr >= runtime->buffer_size)
 		ptr -= runtime->buffer_size;
-	//printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate);
-
+	/*
+	printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, "
+	       "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n",
+	       ptr1, ptr2, ptr, (int)runtime->buffer_size,
+	       (int)runtime->period_size, (int)runtime->frame_bits,
+	       (int)runtime->rate);
+	*/
 	return ptr;
 }
 
@@ -1181,8 +1236,12 @@ static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id)
 		return IRQ_NONE;
 
         stat76 = snd_ca0106_ptr_read(chip, EXTENDED_INT, 0);
-	//snd_printk("interrupt status = 0x%08x, stat76=0x%08x\n", status, stat76);
-	//snd_printk("ptr=0x%08x\n",snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0));
+	/*
+	snd_printk(KERN_DEBUG "interrupt status = 0x%08x, stat76=0x%08x\n",
+		   status, stat76);
+	snd_printk(KERN_DEBUG "ptr=0x%08x\n",
+		   snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0));
+	*/
         mask = 0x11; /* 0x1 for one half, 0x10 for the other half period. */
 	for(i = 0; i < 4; i++) {
 		pchannel = &(chip->playback_channels[i]);
@@ -1470,7 +1529,7 @@ static void ca0106_init_chip(struct snd_ca0106 *chip, int resume)
 		int size, n;
 
 		size = ARRAY_SIZE(i2c_adc_init);
-		/* snd_printk("I2C:array size=0x%x\n", size); */
+		/* snd_printk(KERN_DEBUG "I2C:array size=0x%x\n", size); */
 		for (n = 0; n < size; n++)
 			snd_ca0106_i2c_write(chip, i2c_adc_init[n][0],
 					     i2c_adc_init[n][1]);
@@ -1707,9 +1766,9 @@ static int __devinit snd_ca0106_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	err = snd_ca0106_create(dev, card, pci, &chip);
 	if (err < 0)
diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c
index 1a74ca62c314..c7899c32aba1 100644
--- a/sound/pci/cmipci.c
+++ b/sound/pci/cmipci.c
@@ -3272,9 +3272,9 @@ static int __devinit snd_cmipci_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 	
 	switch (pci->device) {
 	case PCI_DEVICE_ID_CMEDIA_CM8738:
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index 192e7842e181..f6286f84a221 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -834,7 +834,11 @@ static snd_pcm_uframes_t snd_cs4281_pointer(struct snd_pcm_substream *substream)
 	struct cs4281_dma *dma = runtime->private_data;
 	struct cs4281 *chip = snd_pcm_substream_chip(substream);
 
-	// printk("DCC = 0x%x, buffer_size = 0x%x, jiffies = %li\n", snd_cs4281_peekBA0(chip, dma->regDCC), runtime->buffer_size, jiffies);
+	/*
+	printk(KERN_DEBUG "DCC = 0x%x, buffer_size = 0x%x, jiffies = %li\n",
+	       snd_cs4281_peekBA0(chip, dma->regDCC), runtime->buffer_size,
+	       jiffies);
+	*/
 	return runtime->buffer_size -
 	       snd_cs4281_peekBA0(chip, dma->regDCC) - 1;
 }
@@ -1925,9 +1929,9 @@ static int __devinit snd_cs4281_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	if ((err = snd_cs4281_create(card, pci, &chip, dual_codec[dev])) < 0) {
 		snd_card_free(card);
diff --git a/sound/pci/cs46xx/cs46xx.c b/sound/pci/cs46xx/cs46xx.c
index e876b3263e46..c9b3e3d48cbc 100644
--- a/sound/pci/cs46xx/cs46xx.c
+++ b/sound/pci/cs46xx/cs46xx.c
@@ -88,9 +88,9 @@ static int __devinit snd_card_cs46xx_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 	if ((err = snd_cs46xx_create(card, pci,
 				     external_amp[dev], thinkpad[dev],
 				     &chip)) < 0) {
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 8ab07aa63652..1be96ead4244 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -194,7 +194,7 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip,
 	 *  ACSDA = Status Data Register = 474h
 	 */
 #if 0
-	printk("e) reg = 0x%x, val = 0x%x, BA0_ACCAD = 0x%x\n", reg,
+	printk(KERN_DEBUG "e) reg = 0x%x, val = 0x%x, BA0_ACCAD = 0x%x\n", reg,
 			snd_cs46xx_peekBA0(chip, BA0_ACSDA),
 			snd_cs46xx_peekBA0(chip, BA0_ACCAD));
 #endif
@@ -428,8 +428,8 @@ static int cs46xx_wait_for_fifo(struct snd_cs46xx * chip,int retry_timeout)
 	}
   
 	if(status & SERBST_WBSY) {
-		snd_printk( KERN_ERR "cs46xx: failure waiting for FIFO command to complete\n");
-
+		snd_printk(KERN_ERR "cs46xx: failure waiting for "
+			   "FIFO command to complete\n");
 		return -EINVAL;
 	}
 
diff --git a/sound/pci/cs46xx/cs46xx_lib.h b/sound/pci/cs46xx/cs46xx_lib.h
index 018a7de56017..4eb55aa33612 100644
--- a/sound/pci/cs46xx/cs46xx_lib.h
+++ b/sound/pci/cs46xx/cs46xx_lib.h
@@ -62,7 +62,11 @@ static inline void snd_cs46xx_poke(struct snd_cs46xx *chip, unsigned long reg, u
 	unsigned int bank = reg >> 16;
 	unsigned int offset = reg & 0xffff;
 
-	/*if (bank == 0) printk("snd_cs46xx_poke: %04X - %08X\n",reg >> 2,val); */
+	/*
+	if (bank == 0)
+		printk(KERN_DEBUG "snd_cs46xx_poke: %04X - %08X\n",
+		       reg >> 2,val);
+	*/
 	writel(val, chip->region.idx[bank+1].remap_addr + offset);
 }
 
diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c
index 6dea5b5cc774..dc464321d0f3 100644
--- a/sound/pci/cs5530.c
+++ b/sound/pci/cs5530.c
@@ -258,10 +258,10 @@ static int __devinit snd_cs5530_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
 
-	if (card == NULL)
-		return -ENOMEM;
+	if (err < 0)
+		return err;
 
 	err = snd_cs5530_create(card, pci, &chip);
 	if (err < 0) {
diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c
index 826e6dec2e97..c89ed1f5bc2b 100644
--- a/sound/pci/cs5535audio/cs5535audio.c
+++ b/sound/pci/cs5535audio/cs5535audio.c
@@ -312,7 +312,7 @@ static int __devinit snd_cs5535audio_create(struct snd_card *card,
 
 	if (request_irq(pci->irq, snd_cs5535audio_interrupt,
 			IRQF_SHARED, "CS5535 Audio", cs5535au)) {
-		snd_printk("unable to grab IRQ %d\n", pci->irq);
+		snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq);
 		err = -EBUSY;
 		goto sndfail;
 	}
@@ -353,9 +353,9 @@ static int __devinit snd_cs5535audio_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	if ((err = snd_cs5535audio_create(card, pci, &cs5535au)) < 0)
 		goto probefail_out;
diff --git a/sound/pci/echoaudio/Makefile b/sound/pci/echoaudio/Makefile
index 7b576aeb3f8d..1361de77e0cd 100644
--- a/sound/pci/echoaudio/Makefile
+++ b/sound/pci/echoaudio/Makefile
@@ -15,6 +15,8 @@ snd-echo3g-objs := echo3g.o
 snd-indigo-objs := indigo.o
 snd-indigoio-objs := indigoio.o
 snd-indigodj-objs := indigodj.o
+snd-indigoiox-objs := indigoiox.o
+snd-indigodjx-objs := indigodjx.o
 
 obj-$(CONFIG_SND_DARLA20) += snd-darla20.o
 obj-$(CONFIG_SND_GINA20) += snd-gina20.o
@@ -28,3 +30,5 @@ obj-$(CONFIG_SND_ECHO3G) += snd-echo3g.o
 obj-$(CONFIG_SND_INDIGO) += snd-indigo.o
 obj-$(CONFIG_SND_INDIGOIO) += snd-indigoio.o
 obj-$(CONFIG_SND_INDIGODJ) += snd-indigodj.o
+obj-$(CONFIG_SND_INDIGOIOX) += snd-indigoiox.o
+obj-$(CONFIG_SND_INDIGODJX) += snd-indigodjx.o
diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c
index 417e25add82b..57967e580571 100644
--- a/sound/pci/echoaudio/echo3g_dsp.c
+++ b/sound/pci/echoaudio/echo3g_dsp.c
@@ -56,7 +56,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 	}
 
 	chip->comm_page->e3g_frq_register =
-		__constant_cpu_to_le32((E3G_MAGIC_NUMBER / 48000) - 2);
+		cpu_to_le32((E3G_MAGIC_NUMBER / 48000) - 2);
 	chip->device_id = device_id;
 	chip->subdevice_id = subdevice_id;
 	chip->bad_board = TRUE;
diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c
index 8dbc5c4ba421..da2065cd2c0d 100644
--- a/sound/pci/echoaudio/echoaudio.c
+++ b/sound/pci/echoaudio/echoaudio.c
@@ -950,6 +950,8 @@ static int __devinit snd_echo_new_pcm(struct echoaudio *chip)
 	Control interface
 ******************************************************************************/
 
+#ifndef ECHOCARD_HAS_VMIXER
+
 /******************* PCM output volume *******************/
 static int snd_echo_output_gain_info(struct snd_kcontrol *kcontrol,
 				     struct snd_ctl_elem_info *uinfo)
@@ -1001,18 +1003,6 @@ static int snd_echo_output_gain_put(struct snd_kcontrol *kcontrol,
 	return changed;
 }
 
-#ifdef ECHOCARD_HAS_VMIXER
-/* On Vmixer cards this one controls the line-out volume */
-static struct snd_kcontrol_new snd_echo_line_output_gain __devinitdata = {
-	.name = "Line Playback Volume",
-	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ,
-	.info = snd_echo_output_gain_info,
-	.get = snd_echo_output_gain_get,
-	.put = snd_echo_output_gain_put,
-	.tlv = {.p = db_scale_output_gain},
-};
-#else
 static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = {
 	.name = "PCM Playback Volume",
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1022,6 +1012,7 @@ static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = {
 	.put = snd_echo_output_gain_put,
 	.tlv = {.p = db_scale_output_gain},
 };
+
 #endif
 
 
@@ -1997,9 +1988,9 @@ static int __devinit snd_echo_probe(struct pci_dev *pci,
 
 	DE_INIT(("Echoaudio driver starting...\n"));
 	i = 0;
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	snd_card_set_dev(card, &pci->dev);
 
@@ -2037,8 +2028,6 @@ static int __devinit snd_echo_probe(struct pci_dev *pci,
 
 #ifdef ECHOCARD_HAS_VMIXER
 	snd_echo_vmixer.count = num_pipes_out(chip) * num_busses_out(chip);
-	if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_output_gain, chip))) < 0)
-		goto ctl_error;
 	if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vmixer, chip))) < 0)
 		goto ctl_error;
 #else
diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h
index 1c88e051abf2..f9490ae36c2e 100644
--- a/sound/pci/echoaudio/echoaudio.h
+++ b/sound/pci/echoaudio/echoaudio.h
@@ -189,6 +189,9 @@
 #define INDIGO			0x0090
 #define INDIGO_IO		0x00a0
 #define INDIGO_DJ		0x00b0
+#define DC8			0x00c0
+#define INDIGO_IOX		0x00d0
+#define INDIGO_DJX		0x00e0
 #define ECHO3G			0x0100
 
 
diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c
index c3736bbd819e..e32a74897921 100644
--- a/sound/pci/echoaudio/echoaudio_3g.c
+++ b/sound/pci/echoaudio/echoaudio_3g.c
@@ -40,8 +40,7 @@ static int check_asic_status(struct echoaudio *chip)
 	if (wait_handshake(chip))
 		return -EIO;
 
-	chip->comm_page->ext_box_status =
-		__constant_cpu_to_le32(E3G_ASIC_NOT_LOADED);
+	chip->comm_page->ext_box_status = cpu_to_le32(E3G_ASIC_NOT_LOADED);
 	chip->asic_loaded = FALSE;
 	clear_handshake(chip);
 	send_vector(chip, DSP_VC_TEST_ASIC);
diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c
index be0e18192de3..4df51ef5e095 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.c
+++ b/sound/pci/echoaudio/echoaudio_dsp.c
@@ -926,11 +926,11 @@ static int init_dsp_comm_page(struct echoaudio *chip)
 
 	/* Init the comm page */
 	chip->comm_page->comm_size =
-		__constant_cpu_to_le32(sizeof(struct comm_page));
+		cpu_to_le32(sizeof(struct comm_page));
 	chip->comm_page->handshake = 0xffffffff;
 	chip->comm_page->midi_out_free_count =
-		__constant_cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE);
-	chip->comm_page->sample_rate = __constant_cpu_to_le32(44100);
+		cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE);
+	chip->comm_page->sample_rate = cpu_to_le32(44100);
 	chip->sample_rate = 44100;
 
 	/* Set line levels so we don't blast any inputs on startup */
diff --git a/sound/pci/echoaudio/echoaudio_dsp.h b/sound/pci/echoaudio/echoaudio_dsp.h
index e352f3ae292c..cb7d75a0a503 100644
--- a/sound/pci/echoaudio/echoaudio_dsp.h
+++ b/sound/pci/echoaudio/echoaudio_dsp.h
@@ -576,8 +576,13 @@ SET_LAYLA24_FREQUENCY_REG command.
 #define E3G_ASIC_NOT_LOADED		0xffff
 #define E3G_BOX_TYPE_MASK		0xf0
 
-#define EXT_3GBOX_NC			0x01
-#define EXT_3GBOX_NOT_SET		0x02
+/* Indigo express control register values */
+#define INDIGO_EXPRESS_32000		0x02
+#define INDIGO_EXPRESS_44100		0x01
+#define INDIGO_EXPRESS_48000		0x00
+#define INDIGO_EXPRESS_DOUBLE_SPEED	0x10
+#define INDIGO_EXPRESS_QUAD_SPEED	0x04
+#define INDIGO_EXPRESS_CLOCK_MASK	0x17
 
 
 /*
diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c
index db6c952e9d7f..3f1e7475faea 100644
--- a/sound/pci/echoaudio/gina20_dsp.c
+++ b/sound/pci/echoaudio/gina20_dsp.c
@@ -208,10 +208,10 @@ static int set_professional_spdif(struct echoaudio *chip, char prof)
 	DE_ACT(("set_professional_spdif %d\n", prof));
 	if (prof)
 		chip->comm_page->flags |=
-			__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+			cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
 	else
 		chip->comm_page->flags &=
-			~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+			~cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
 	chip->professional_spdif = prof;
 	return update_flags(chip);
 }
diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c
index f05e39f7aad9..0b2cd9c86277 100644
--- a/sound/pci/echoaudio/indigo_dsp.c
+++ b/sound/pci/echoaudio/indigo_dsp.c
@@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 	if ((err = init_line_levels(chip)) < 0)
 		return err;
 
-	/* Default routing of the virtual channels: all vchannels are routed
-	to the stereo output */
-	set_vmixer_gain(chip, 0, 0, 0);
-	set_vmixer_gain(chip, 1, 1, 0);
-	set_vmixer_gain(chip, 0, 2, 0);
-	set_vmixer_gain(chip, 1, 3, 0);
-	set_vmixer_gain(chip, 0, 4, 0);
-	set_vmixer_gain(chip, 1, 5, 0);
-	set_vmixer_gain(chip, 0, 6, 0);
-	set_vmixer_gain(chip, 1, 7, 0);
-	err = update_vmixer_level(chip);
-
 	DE_INIT(("init_hw done\n"));
 	return err;
 }
diff --git a/sound/pci/echoaudio/indigo_express_dsp.c b/sound/pci/echoaudio/indigo_express_dsp.c
new file mode 100644
index 000000000000..9ab625e15652
--- /dev/null
+++ b/sound/pci/echoaudio/indigo_express_dsp.c
@@ -0,0 +1,119 @@
+/************************************************************************
+
+This file is part of Echo Digital Audio's generic driver library.
+Copyright Echo Digital Audio Corporation (c) 1998 - 2005
+All rights reserved
+www.echoaudio.com
+
+This library is free software; you can redistribute it and/or
+modify it under the terms of the GNU Lesser General Public
+License as published by the Free Software Foundation; either
+version 2.1 of the License, or (at your option) any later version.
+
+This library is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+Lesser General Public License for more details.
+
+You should have received a copy of the GNU Lesser General Public
+License along with this library; if not, write to the Free Software
+Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+
+*************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+*************************************************************************/
+
+static int set_sample_rate(struct echoaudio *chip, u32 rate)
+{
+	u32 clock, control_reg, old_control_reg;
+
+	if (wait_handshake(chip))
+		return -EIO;
+
+	old_control_reg = le32_to_cpu(chip->comm_page->control_register);
+	control_reg = old_control_reg & ~INDIGO_EXPRESS_CLOCK_MASK;
+
+	switch (rate) {
+	case 32000:
+		clock = INDIGO_EXPRESS_32000;
+		break;
+	case 44100:
+		clock = INDIGO_EXPRESS_44100;
+		break;
+	case 48000:
+		clock = INDIGO_EXPRESS_48000;
+		break;
+	case 64000:
+		clock = INDIGO_EXPRESS_32000|INDIGO_EXPRESS_DOUBLE_SPEED;
+		break;
+	case 88200:
+		clock = INDIGO_EXPRESS_44100|INDIGO_EXPRESS_DOUBLE_SPEED;
+		break;
+	case 96000:
+		clock = INDIGO_EXPRESS_48000|INDIGO_EXPRESS_DOUBLE_SPEED;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	control_reg |= clock;
+	if (control_reg != old_control_reg) {
+		chip->comm_page->control_register = cpu_to_le32(control_reg);
+		chip->sample_rate = rate;
+		clear_handshake(chip);
+		return send_vector(chip, DSP_VC_UPDATE_CLOCKS);
+	}
+	return 0;
+}
+
+
+
+/* This function routes the sound from a virtual channel to a real output */
+static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe,
+			   int gain)
+{
+	int index;
+
+	if (snd_BUG_ON(pipe >= num_pipes_out(chip) ||
+		       output >= num_busses_out(chip)))
+		return -EINVAL;
+
+	if (wait_handshake(chip))
+		return -EIO;
+
+	chip->vmixer_gain[output][pipe] = gain;
+	index = output * num_pipes_out(chip) + pipe;
+	chip->comm_page->vmixer[index] = gain;
+
+	DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain));
+	return 0;
+}
+
+
+
+/* Tell the DSP to read and update virtual mixer levels in comm page. */
+static int update_vmixer_level(struct echoaudio *chip)
+{
+	if (wait_handshake(chip))
+		return -EIO;
+	clear_handshake(chip);
+	return send_vector(chip, DSP_VC_SET_VMIXER_GAIN);
+}
+
+
+
+static u32 detect_input_clocks(const struct echoaudio *chip)
+{
+	return ECHO_CLOCK_BIT_INTERNAL;
+}
+
+
+
+/* The IndigoIO has no ASIC. Just do nothing */
+static int load_asic(struct echoaudio *chip)
+{
+	return 0;
+}
diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c
index 90730a5ecb42..08392916691e 100644
--- a/sound/pci/echoaudio/indigodj_dsp.c
+++ b/sound/pci/echoaudio/indigodj_dsp.c
@@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 	if ((err = init_line_levels(chip)) < 0)
 		return err;
 
-	/* Default routing of the virtual channels: vchannels 0-3 and
-	vchannels 4-7 are routed to real channels 0-4 */
-	set_vmixer_gain(chip, 0, 0, 0);
-	set_vmixer_gain(chip, 1, 1, 0);
-	set_vmixer_gain(chip, 2, 2, 0);
-	set_vmixer_gain(chip, 3, 3, 0);
-	set_vmixer_gain(chip, 0, 4, 0);
-	set_vmixer_gain(chip, 1, 5, 0);
-	set_vmixer_gain(chip, 2, 6, 0);
-	set_vmixer_gain(chip, 3, 7, 0);
-	err = update_vmixer_level(chip);
-
 	DE_INIT(("init_hw done\n"));
 	return err;
 }
diff --git a/sound/pci/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c
new file mode 100644
index 000000000000..3482ef69f491
--- /dev/null
+++ b/sound/pci/echoaudio/indigodjx.c
@@ -0,0 +1,107 @@
+/*
+ *  ALSA driver for Echoaudio soundcards.
+ *  Copyright (C) 2009 Giuliano Pochini <pochini@shiny.it>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define INDIGO_FAMILY
+#define ECHOCARD_INDIGO_DJX
+#define ECHOCARD_NAME "Indigo DJx"
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_VMIXER
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT	0	/* 8 */
+#define PX_DIGITAL_OUT	8	/* 0 */
+#define PX_ANALOG_IN	8	/* 0 */
+#define PX_DIGITAL_IN	8	/* 0 */
+#define PX_NUM		8
+
+/* Bus indexes */
+#define BX_ANALOG_OUT	0	/* 4 */
+#define BX_DIGITAL_OUT	4	/* 0 */
+#define BX_ANALOG_IN	4	/* 0 */
+#define BX_DIGITAL_IN	4	/* 0 */
+#define BX_NUM		4
+
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+MODULE_FIRMWARE("ea/loader_dsp.fw");
+MODULE_FIRMWARE("ea/indigo_djx_dsp.fw");
+
+#define FW_361_LOADER		0
+#define FW_INDIGO_DJX_DSP	1
+
+static const struct firmware card_fw[] = {
+	{0, "loader_dsp.fw"},
+	{0, "indigo_djx_dsp.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+	{0x1057, 0x3410, 0xECC0, 0x00E0, 0, 0, 0},	/* Indigo DJx*/
+	{0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+	.info = SNDRV_PCM_INFO_MMAP |
+		SNDRV_PCM_INFO_INTERLEAVED |
+		SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		SNDRV_PCM_INFO_MMAP_VALID |
+		SNDRV_PCM_INFO_PAUSE |
+		SNDRV_PCM_INFO_SYNC_START,
+	.formats =	SNDRV_PCM_FMTBIT_U8 |
+			SNDRV_PCM_FMTBIT_S16_LE |
+			SNDRV_PCM_FMTBIT_S24_3LE |
+			SNDRV_PCM_FMTBIT_S32_LE |
+			SNDRV_PCM_FMTBIT_S32_BE,
+	.rates = 	SNDRV_PCM_RATE_32000 |
+			SNDRV_PCM_RATE_44100 |
+			SNDRV_PCM_RATE_48000 |
+			SNDRV_PCM_RATE_88200 |
+			SNDRV_PCM_RATE_96000,
+	.rate_min = 32000,
+	.rate_max = 96000,
+	.channels_min = 1,
+	.channels_max = 4,
+	.buffer_bytes_max = 262144,
+	.period_bytes_min = 32,
+	.period_bytes_max = 131072,
+	.periods_min = 2,
+	.periods_max = 220,
+};
+
+#include "indigodjx_dsp.c"
+#include "indigo_express_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c
new file mode 100644
index 000000000000..f591fc2ed960
--- /dev/null
+++ b/sound/pci/echoaudio/indigodjx_dsp.c
@@ -0,0 +1,68 @@
+/************************************************************************
+
+This file is part of Echo Digital Audio's generic driver library.
+Copyright Echo Digital Audio Corporation (c) 1998 - 2005
+All rights reserved
+www.echoaudio.com
+
+This library is free software; you can redistribute it and/or
+modify it under the terms of the GNU Lesser General Public
+License as published by the Free Software Foundation; either
+version 2.1 of the License, or (at your option) any later version.
+
+This library is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+Lesser General Public License for more details.
+
+You should have received a copy of the GNU Lesser General Public
+License along with this library; if not, write to the Free Software
+Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+
+*************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+*************************************************************************/
+
+static int update_vmixer_level(struct echoaudio *chip);
+static int set_vmixer_gain(struct echoaudio *chip, u16 output,
+			   u16 pipe, int gain);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+	int err;
+
+	DE_INIT(("init_hw() - Indigo DJx\n"));
+	if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJX))
+		return -ENODEV;
+
+	err = init_dsp_comm_page(chip);
+	if (err < 0) {
+		DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+		return err;
+	}
+
+	chip->device_id = device_id;
+	chip->subdevice_id = subdevice_id;
+	chip->bad_board = TRUE;
+	chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJX_DSP];
+	/* Since this card has no ASIC, mark it as loaded so everything
+	   works OK */
+	chip->asic_loaded = TRUE;
+	chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL;
+
+	err = load_firmware(chip);
+	if (err < 0)
+		return err;
+	chip->bad_board = FALSE;
+
+	err = init_line_levels(chip);
+	if (err < 0)
+		return err;
+
+	DE_INIT(("init_hw done\n"));
+	return err;
+}
diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c
index a7e09ec21079..0604c8a85223 100644
--- a/sound/pci/echoaudio/indigoio_dsp.c
+++ b/sound/pci/echoaudio/indigoio_dsp.c
@@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 	if ((err = init_line_levels(chip)) < 0)
 		return err;
 
-	/* Default routing of the virtual channels: all vchannels are routed
-	to the stereo output */
-	set_vmixer_gain(chip, 0, 0, 0);
-	set_vmixer_gain(chip, 1, 1, 0);
-	set_vmixer_gain(chip, 0, 2, 0);
-	set_vmixer_gain(chip, 1, 3, 0);
-	set_vmixer_gain(chip, 0, 4, 0);
-	set_vmixer_gain(chip, 1, 5, 0);
-	set_vmixer_gain(chip, 0, 6, 0);
-	set_vmixer_gain(chip, 1, 7, 0);
-	err = update_vmixer_level(chip);
-
 	DE_INIT(("init_hw done\n"));
 	return err;
 }
diff --git a/sound/pci/echoaudio/indigoiox.c b/sound/pci/echoaudio/indigoiox.c
new file mode 100644
index 000000000000..aebee27a40ff
--- /dev/null
+++ b/sound/pci/echoaudio/indigoiox.c
@@ -0,0 +1,109 @@
+/*
+ *  ALSA driver for Echoaudio soundcards.
+ *  Copyright (C) 2009 Giuliano Pochini <pochini@shiny.it>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License as published by
+ *  the Free Software Foundation; version 2 of the License.
+ *
+ *  This program is distributed in the hope that it will be useful,
+ *  but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *  GNU General Public License for more details.
+ *
+ *  You should have received a copy of the GNU General Public License
+ *  along with this program; if not, write to the Free Software
+ *  Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+ */
+
+#define INDIGO_FAMILY
+#define ECHOCARD_INDIGO_IOX
+#define ECHOCARD_NAME "Indigo IOx"
+#define ECHOCARD_HAS_MONITOR
+#define ECHOCARD_HAS_SUPER_INTERLEAVE
+#define ECHOCARD_HAS_VMIXER
+#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32
+
+/* Pipe indexes */
+#define PX_ANALOG_OUT	0	/* 8 */
+#define PX_DIGITAL_OUT	8	/* 0 */
+#define PX_ANALOG_IN	8	/* 2 */
+#define PX_DIGITAL_IN	10	/* 0 */
+#define PX_NUM		10
+
+/* Bus indexes */
+#define BX_ANALOG_OUT	0	/* 2 */
+#define BX_DIGITAL_OUT	2	/* 0 */
+#define BX_ANALOG_IN	2	/* 2 */
+#define BX_DIGITAL_IN	4	/* 0 */
+#define BX_NUM		4
+
+
+#include <linux/delay.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/pci.h>
+#include <linux/slab.h>
+#include <linux/moduleparam.h>
+#include <linux/firmware.h>
+#include <linux/io.h>
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/control.h>
+#include <sound/tlv.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/asoundef.h>
+#include <sound/initval.h>
+#include <asm/atomic.h>
+#include "echoaudio.h"
+
+MODULE_FIRMWARE("ea/loader_dsp.fw");
+MODULE_FIRMWARE("ea/indigo_iox_dsp.fw");
+
+#define FW_361_LOADER		0
+#define FW_INDIGO_IOX_DSP	1
+
+static const struct firmware card_fw[] = {
+	{0, "loader_dsp.fw"},
+	{0, "indigo_iox_dsp.fw"}
+};
+
+static struct pci_device_id snd_echo_ids[] = {
+	{0x1057, 0x3410, 0xECC0, 0x00D0, 0, 0, 0},	/* Indigo IOx */
+	{0,}
+};
+
+static struct snd_pcm_hardware pcm_hardware_skel = {
+	.info = SNDRV_PCM_INFO_MMAP |
+		SNDRV_PCM_INFO_INTERLEAVED |
+		SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		SNDRV_PCM_INFO_MMAP_VALID |
+		SNDRV_PCM_INFO_PAUSE |
+		SNDRV_PCM_INFO_SYNC_START,
+	.formats =	SNDRV_PCM_FMTBIT_U8 |
+			SNDRV_PCM_FMTBIT_S16_LE |
+			SNDRV_PCM_FMTBIT_S24_3LE |
+			SNDRV_PCM_FMTBIT_S32_LE |
+			SNDRV_PCM_FMTBIT_S32_BE,
+	.rates = 	SNDRV_PCM_RATE_32000 |
+			SNDRV_PCM_RATE_44100 |
+			SNDRV_PCM_RATE_48000 |
+			SNDRV_PCM_RATE_88200 |
+			SNDRV_PCM_RATE_96000,
+	.rate_min = 32000,
+	.rate_max = 96000,
+	.channels_min = 1,
+	.channels_max = 8,
+	.buffer_bytes_max = 262144,
+	.period_bytes_min = 32,
+	.period_bytes_max = 131072,
+	.periods_min = 2,
+	.periods_max = 220,
+};
+
+#include "indigoiox_dsp.c"
+#include "indigo_express_dsp.c"
+#include "echoaudio_dsp.c"
+#include "echoaudio.c"
+
diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c
new file mode 100644
index 000000000000..f357521c79e6
--- /dev/null
+++ b/sound/pci/echoaudio/indigoiox_dsp.c
@@ -0,0 +1,68 @@
+/************************************************************************
+
+This file is part of Echo Digital Audio's generic driver library.
+Copyright Echo Digital Audio Corporation (c) 1998 - 2005
+All rights reserved
+www.echoaudio.com
+
+This library is free software; you can redistribute it and/or
+modify it under the terms of the GNU Lesser General Public
+License as published by the Free Software Foundation; either
+version 2.1 of the License, or (at your option) any later version.
+
+This library is distributed in the hope that it will be useful,
+but WITHOUT ANY WARRANTY; without even the implied warranty of
+MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+Lesser General Public License for more details.
+
+You should have received a copy of the GNU Lesser General Public
+License along with this library; if not, write to the Free Software
+Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+
+*************************************************************************
+
+ Translation from C++ and adaptation for use in ALSA-Driver
+ were made by Giuliano Pochini <pochini@shiny.it>
+
+*************************************************************************/
+
+static int update_vmixer_level(struct echoaudio *chip);
+static int set_vmixer_gain(struct echoaudio *chip, u16 output,
+			   u16 pipe, int gain);
+
+
+static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
+{
+	int err;
+
+	DE_INIT(("init_hw() - Indigo IOx\n"));
+	if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IOX))
+		return -ENODEV;
+
+	err = init_dsp_comm_page(chip);
+	if (err < 0) {
+		DE_INIT(("init_hw - could not initialize DSP comm page\n"));
+		return err;
+	}
+
+	chip->device_id = device_id;
+	chip->subdevice_id = subdevice_id;
+	chip->bad_board = TRUE;
+	chip->dsp_code_to_load = &card_fw[FW_INDIGO_IOX_DSP];
+	/* Since this card has no ASIC, mark it as loaded so everything
+	   works OK */
+	chip->asic_loaded = TRUE;
+	chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL;
+
+	err = load_firmware(chip);
+	if (err < 0)
+		return err;
+	chip->bad_board = FALSE;
+
+	err = init_line_levels(chip);
+	if (err < 0)
+		return err;
+
+	DE_INIT(("init_hw done\n"));
+	return err;
+}
diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c
index ede75c6ca0fb..83750e9fd7b4 100644
--- a/sound/pci/echoaudio/layla20_dsp.c
+++ b/sound/pci/echoaudio/layla20_dsp.c
@@ -284,10 +284,10 @@ static int set_professional_spdif(struct echoaudio *chip, char prof)
 	DE_ACT(("set_professional_spdif %d\n", prof));
 	if (prof)
 		chip->comm_page->flags |=
-			__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+			cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
 	else
 		chip->comm_page->flags &=
-			~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+			~cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
 	chip->professional_spdif = prof;
 	return update_flags(chip);
 }
diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c
index 227386602f9b..551405114cbc 100644
--- a/sound/pci/echoaudio/mia_dsp.c
+++ b/sound/pci/echoaudio/mia_dsp.c
@@ -69,18 +69,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id)
 	if ((err = init_line_levels(chip)))
 		return err;
 
-	/* Default routing of the virtual channels: vchannels 0-3 go to analog
-	outputs and vchannels 4-7 go to S/PDIF outputs */
-	set_vmixer_gain(chip, 0, 0, 0);
-	set_vmixer_gain(chip, 1, 1, 0);
-	set_vmixer_gain(chip, 0, 2, 0);
-	set_vmixer_gain(chip, 1, 3, 0);
-	set_vmixer_gain(chip, 2, 4, 0);
-	set_vmixer_gain(chip, 3, 5, 0);
-	set_vmixer_gain(chip, 2, 6, 0);
-	set_vmixer_gain(chip, 3, 7, 0);
-	err = update_vmixer_level(chip);
-
 	DE_INIT(("init_hw done\n"));
 	return err;
 }
@@ -222,10 +210,10 @@ static int set_professional_spdif(struct echoaudio *chip, char prof)
 	DE_ACT(("set_professional_spdif %d\n", prof));
 	if (prof)
 		chip->comm_page->flags |=
-			__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+			cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
 	else
 		chip->comm_page->flags &=
-			~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
+			~cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF);
 	chip->professional_spdif = prof;
 	return update_flags(chip);
 }
diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c
index 77bf2a83d997..a953d142cb4b 100644
--- a/sound/pci/echoaudio/midi.c
+++ b/sound/pci/echoaudio/midi.c
@@ -44,10 +44,10 @@ static int enable_midi_input(struct echoaudio *chip, char enable)
 	if (enable) {
 		chip->mtc_state = MIDI_IN_STATE_NORMAL;
 		chip->comm_page->flags |=
-			__constant_cpu_to_le32(DSP_FLAG_MIDI_INPUT);
+			cpu_to_le32(DSP_FLAG_MIDI_INPUT);
 	} else
 		chip->comm_page->flags &=
-			~__constant_cpu_to_le32(DSP_FLAG_MIDI_INPUT);
+			~cpu_to_le32(DSP_FLAG_MIDI_INPUT);
 
 	clear_handshake(chip);
 	return send_vector(chip, DSP_VC_UPDATE_FLAGS);
diff --git a/sound/pci/emu10k1/emu10k1.c b/sound/pci/emu10k1/emu10k1.c
index 8354c1a83312..c7f3b994101c 100644
--- a/sound/pci/emu10k1/emu10k1.c
+++ b/sound/pci/emu10k1/emu10k1.c
@@ -114,9 +114,9 @@ static int __devinit snd_card_emu10k1_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 	if (max_buffer_size[dev] < 32)
 		max_buffer_size[dev] = 32;
 	else if (max_buffer_size[dev] > 1024)
diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c
index 0e649dcdbf64..7ef949d99a50 100644
--- a/sound/pci/emu10k1/emu10k1_callback.c
+++ b/sound/pci/emu10k1/emu10k1_callback.c
@@ -103,7 +103,10 @@ snd_emu10k1_synth_get_voice(struct snd_emu10k1 *hw)
 			int ch;
 			vp = &emu->voices[best[i].voice];
 			if ((ch = vp->ch) < 0) {
-				//printk("synth_get_voice: ch < 0 (%d) ??", i);
+				/*
+				printk(KERN_WARNING
+				       "synth_get_voice: ch < 0 (%d) ??", i);
+				*/
 				continue;
 			}
 			vp->emu->num_voices--;
@@ -335,7 +338,7 @@ start_voice(struct snd_emux_voice *vp)
 		return -EINVAL;
 	emem->map_locked++;
 	if (snd_emu10k1_memblk_map(hw, emem) < 0) {
-		// printk("emu: cannot map!\n");
+		/* printk(KERN_ERR "emu: cannot map!\n"); */
 		return -ENOMEM;
 	}
 	mapped_offset = snd_emu10k1_memblk_offset(emem) >> 1;
diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c
index 101a1c13a20d..f18bd6207c50 100644
--- a/sound/pci/emu10k1/emu10k1_main.c
+++ b/sound/pci/emu10k1/emu10k1_main.c
@@ -711,8 +711,7 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 *emu, const char *filena
 static int emu1010_firmware_thread(void *data)
 {
 	struct snd_emu10k1 *emu = data;
-	int tmp, tmp2;
-	int reg;
+	u32 tmp, tmp2, reg;
 	int err;
 
 	for (;;) {
@@ -758,7 +757,8 @@ static int emu1010_firmware_thread(void *data)
 			snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n");
 			snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp);
 			snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2);
-			snd_printk("Audio Dock ver:%d.%d\n", tmp, tmp2);
+			snd_printk(KERN_INFO "Audio Dock ver: %u.%u\n",
+				   tmp, tmp2);
 			/* Sync clocking between 1010 and Dock */
 			/* Allow DLL to settle */
 			msleep(10);
@@ -804,8 +804,7 @@ static int emu1010_firmware_thread(void *data)
 static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu)
 {
 	unsigned int i;
-	int tmp, tmp2;
-	int reg;
+	u32 tmp, tmp2, reg;
 	int err;
 	const char *filename = NULL;
 
@@ -887,7 +886,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu)
 	snd_printk(KERN_INFO "emu1010: Hana Firmware loaded\n");
 	snd_emu1010_fpga_read(emu, EMU_HANA_MAJOR_REV, &tmp);
 	snd_emu1010_fpga_read(emu, EMU_HANA_MINOR_REV, &tmp2);
-	snd_printk("emu1010: Hana version: %d.%d\n", tmp, tmp2);
+	snd_printk(KERN_INFO "emu1010: Hana version: %u.%u\n", tmp, tmp2);
 	/* Enable 48Volt power to Audio Dock */
 	snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, EMU_HANA_DOCK_PWR_ON);
 
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index 5ff4dbb62dad..31542adc6b7e 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -1544,9 +1544,9 @@ static int __devinit snd_emu10k1x_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	if ((err = snd_emu10k1x_create(card, pci, &chip)) < 0) {
 		snd_card_free(card);
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index 7dba08f0ab8e..191e1cd9997d 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -1519,7 +1519,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
 	/* A_PUT_STEREO_OUTPUT(A_EXTOUT_FRONT_L, A_EXTOUT_FRONT_R, playback + SND_EMU10K1_PLAYBACK_CHANNELS); */
 	if (emu->card_capabilities->emu_model) {
 		/* EMU1010 Outputs from PCM Front, Rear, Center, LFE, Side */
-		snd_printk("EMU outputs on\n");
+		snd_printk(KERN_INFO "EMU outputs on\n");
 		for (z = 0; z < 8; z++) {
 			if (emu->card_capabilities->ca0108_chip) {
 				A_OP(icode, &ptr, iACC3, A3_EMU32OUT(z), A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_C_00000000, A_C_00000000);
@@ -1567,7 +1567,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
 
 	if (emu->card_capabilities->emu_model) {
 		if (emu->card_capabilities->ca0108_chip) {
-			snd_printk("EMU2 inputs on\n");
+			snd_printk(KERN_INFO "EMU2 inputs on\n");
 			for (z = 0; z < 0x10; z++) {
 				snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, 
 									bit_shifter16,
@@ -1575,10 +1575,13 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input))
 									A_FXBUS2(z*2) );
 			}
 		} else {
-			snd_printk("EMU inputs on\n");
+			snd_printk(KERN_INFO "EMU inputs on\n");
 			/* Capture 16 (originally 8) channels of S32_LE sound */
 
-			/* printk("emufx.c: gpr=0x%x, tmp=0x%x\n",gpr, tmp); */
+			/*
+			printk(KERN_DEBUG "emufx.c: gpr=0x%x, tmp=0x%x\n",
+			       gpr, tmp);
+			*/
 			/* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */
 			/* A_P16VIN(0) is delayed by one sample,
 			 * so all other A_P16VIN channels will need to also be delayed
diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c
index cf9276ddad42..78f62fd404c2 100644
--- a/sound/pci/emu10k1/emupcm.c
+++ b/sound/pci/emu10k1/emupcm.c
@@ -44,7 +44,7 @@ static void snd_emu10k1_pcm_interrupt(struct snd_emu10k1 *emu,
 	if (epcm->substream == NULL)
 		return;
 #if 0
-	printk("IRQ: position = 0x%x, period = 0x%x, size = 0x%x\n",
+	printk(KERN_DEBUG "IRQ: position = 0x%x, period = 0x%x, size = 0x%x\n",
 			epcm->substream->runtime->hw->pointer(emu, epcm->substream),
 			snd_pcm_lib_period_bytes(epcm->substream),
 			snd_pcm_lib_buffer_bytes(epcm->substream));
@@ -146,7 +146,11 @@ static int snd_emu10k1_pcm_channel_alloc(struct snd_emu10k1_pcm * epcm, int voic
 					      1,
 					      &epcm->extra);
 		if (err < 0) {
-			/* printk("pcm_channel_alloc: failed extra: voices=%d, frame=%d\n", voices, frame); */
+			/*
+			printk(KERN_DEBUG "pcm_channel_alloc: "
+			       "failed extra: voices=%d, frame=%d\n",
+			       voices, frame);
+			*/
 			for (i = 0; i < voices; i++) {
 				snd_emu10k1_voice_free(epcm->emu, epcm->voices[i]);
 				epcm->voices[i] = NULL;
@@ -737,7 +741,10 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream,
 	struct snd_emu10k1_pcm_mixer *mix;
 	int result = 0;
 
-	/* printk("trigger - emu10k1 = 0x%x, cmd = %i, pointer = %i\n", (int)emu, cmd, substream->ops->pointer(substream)); */
+	/*
+	printk(KERN_DEBUG "trigger - emu10k1 = 0x%x, cmd = %i, pointer = %i\n",
+	       (int)emu, cmd, substream->ops->pointer(substream))
+	*/
 	spin_lock(&emu->reg_lock);
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -786,7 +793,10 @@ static int snd_emu10k1_capture_trigger(struct snd_pcm_substream *substream,
 		/* hmm this should cause full and half full interrupt to be raised? */
 		outl(epcm->capture_ipr, emu->port + IPR);
 		snd_emu10k1_intr_enable(emu, epcm->capture_inte);
-		/* printk("adccr = 0x%x, adcbs = 0x%x\n", epcm->adccr, epcm->adcbs); */
+		/*
+		printk(KERN_DEBUG "adccr = 0x%x, adcbs = 0x%x\n",
+		       epcm->adccr, epcm->adcbs);
+		*/
 		switch (epcm->type) {
 		case CAPTURE_AC97ADC:
 			snd_emu10k1_ptr_write(emu, ADCCR, 0, epcm->capture_cr_val);
@@ -857,7 +867,11 @@ static snd_pcm_uframes_t snd_emu10k1_playback_pointer(struct snd_pcm_substream *
 			ptr -= runtime->buffer_size;
 	}
 #endif
-	/* printk("ptr = 0x%x, buffer_size = 0x%x, period_size = 0x%x\n", ptr, runtime->buffer_size, runtime->period_size); */
+	/*
+	printk(KERN_DEBUG
+	       "ptr = 0x%x, buffer_size = 0x%x, period_size = 0x%x\n",
+	       ptr, runtime->buffer_size, runtime->period_size);
+	*/
 	return ptr;
 }
 
@@ -1546,7 +1560,11 @@ static void snd_emu10k1_fx8010_playback_tram_poke1(unsigned short *dst_left,
 						   unsigned int count,
 						   unsigned int tram_shift)
 {
-	/* printk("tram_poke1: dst_left = 0x%p, dst_right = 0x%p, src = 0x%p, count = 0x%x\n", dst_left, dst_right, src, count); */
+	/*
+	printk(KERN_DEBUG "tram_poke1: dst_left = 0x%p, dst_right = 0x%p, "
+	       "src = 0x%p, count = 0x%x\n",
+	       dst_left, dst_right, src, count);
+	*/
 	if ((tram_shift & 1) == 0) {
 		while (count--) {
 			*dst_left-- = *src++;
@@ -1623,7 +1641,12 @@ static int snd_emu10k1_fx8010_playback_prepare(struct snd_pcm_substream *substre
 	struct snd_emu10k1_fx8010_pcm *pcm = &emu->fx8010.pcm[substream->number];
 	unsigned int i;
 	
-	/* printk("prepare: etram_pages = 0x%p, dma_area = 0x%x, buffer_size = 0x%x (0x%x)\n", emu->fx8010.etram_pages, runtime->dma_area, runtime->buffer_size, runtime->buffer_size << 2); */
+	/*
+	printk(KERN_DEBUG "prepare: etram_pages = 0x%p, dma_area = 0x%x, "
+	       "buffer_size = 0x%x (0x%x)\n",
+	       emu->fx8010.etram_pages, runtime->dma_area,
+	       runtime->buffer_size, runtime->buffer_size << 2);
+	*/
 	memset(&pcm->pcm_rec, 0, sizeof(pcm->pcm_rec));
 	pcm->pcm_rec.hw_buffer_size = pcm->buffer_size * 2; /* byte size */
 	pcm->pcm_rec.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream);
diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c
index b5a802bdeb7c..4bfc31d1b281 100644
--- a/sound/pci/emu10k1/io.c
+++ b/sound/pci/emu10k1/io.c
@@ -226,7 +226,9 @@ int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu,
 				break;
 
 			if (timeout > 1000) {
-                		snd_printk("emu10k1:I2C:timeout status=0x%x\n", status);
+                		snd_printk(KERN_WARNING
+					   "emu10k1:I2C:timeout status=0x%x\n",
+					   status);
 				break;
 			}
 		}
diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c
index 749a21b6bd06..e617acaf10e3 100644
--- a/sound/pci/emu10k1/p16v.c
+++ b/sound/pci/emu10k1/p16v.c
@@ -168,7 +168,7 @@ static void snd_p16v_pcm_free_substream(struct snd_pcm_runtime *runtime)
 	struct snd_emu10k1_pcm *epcm = runtime->private_data;
   
 	if (epcm) {
-        	//snd_printk("epcm free: %p\n", epcm);
+        	/* snd_printk(KERN_DEBUG "epcm free: %p\n", epcm); */
 		kfree(epcm);
 	}
 }
@@ -183,14 +183,16 @@ static int snd_p16v_pcm_open_playback_channel(struct snd_pcm_substream *substrea
 	int err;
 
 	epcm = kzalloc(sizeof(*epcm), GFP_KERNEL);
-        //snd_printk("epcm kcalloc: %p\n", epcm);
+        /* snd_printk(KERN_DEBUG "epcm kcalloc: %p\n", epcm); */
 
 	if (epcm == NULL)
 		return -ENOMEM;
 	epcm->emu = emu;
 	epcm->substream = substream;
-        //snd_printk("epcm device=%d, channel_id=%d\n", substream->pcm->device, channel_id);
-  
+	/*
+	snd_printk(KERN_DEBUG "epcm device=%d, channel_id=%d\n",
+		   substream->pcm->device, channel_id);
+	*/
 	runtime->private_data = epcm;
 	runtime->private_free = snd_p16v_pcm_free_substream;
   
@@ -200,10 +202,15 @@ static int snd_p16v_pcm_open_playback_channel(struct snd_pcm_substream *substrea
         channel->number = channel_id;
 
         channel->use=1;
-	//snd_printk("p16v: open channel_id=%d, channel=%p, use=0x%x\n", channel_id, channel, channel->use);
-        //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
-        //channel->interrupt = snd_p16v_pcm_channel_interrupt;
-        channel->epcm=epcm;
+#if 0 /* debug */
+	snd_printk(KERN_DEBUG
+		   "p16v: open channel_id=%d, channel=%p, use=0x%x\n",
+		   channel_id, channel, channel->use);
+	printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n",
+	       channel_id, chip, channel);
+#endif /* debug */
+	/* channel->interrupt = snd_p16v_pcm_channel_interrupt; */
+	channel->epcm = epcm;
 	if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
                 return err;
 
@@ -224,14 +231,16 @@ static int snd_p16v_pcm_open_capture_channel(struct snd_pcm_substream *substream
 	int err;
 
 	epcm = kzalloc(sizeof(*epcm), GFP_KERNEL);
-	//snd_printk("epcm kcalloc: %p\n", epcm);
+	/* snd_printk(KERN_DEBUG "epcm kcalloc: %p\n", epcm); */
 
 	if (epcm == NULL)
 		return -ENOMEM;
 	epcm->emu = emu;
 	epcm->substream = substream;
-	//snd_printk("epcm device=%d, channel_id=%d\n", substream->pcm->device, channel_id);
-
+	/*
+	snd_printk(KERN_DEBUG "epcm device=%d, channel_id=%d\n",
+		   substream->pcm->device, channel_id);
+	*/
 	runtime->private_data = epcm;
 	runtime->private_free = snd_p16v_pcm_free_substream;
   
@@ -241,10 +250,15 @@ static int snd_p16v_pcm_open_capture_channel(struct snd_pcm_substream *substream
 	channel->number = channel_id;
 
 	channel->use=1;
-	//snd_printk("p16v: open channel_id=%d, channel=%p, use=0x%x\n", channel_id, channel, channel->use);
-	//printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel);
-	//channel->interrupt = snd_p16v_pcm_channel_interrupt;
-	channel->epcm=epcm;
+#if 0 /* debug */
+	snd_printk(KERN_DEBUG
+		   "p16v: open channel_id=%d, channel=%p, use=0x%x\n",
+		   channel_id, channel, channel->use);
+	printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n",
+	       channel_id, chip, channel);
+#endif /* debug */
+	/* channel->interrupt = snd_p16v_pcm_channel_interrupt; */
+	channel->epcm = epcm;
 	if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0)
 		return err;
 
@@ -334,9 +348,19 @@ static int snd_p16v_pcm_prepare_playback(struct snd_pcm_substream *substream)
 	int i;
 	u32 tmp;
 	
-        //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1));
-        //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base);
-	//snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->p16v_buffer.addr, emu->p16v_buffer.area, emu->p16v_buffer.bytes);
+#if 0 /* debug */
+	snd_printk(KERN_DEBUG "prepare:channel_number=%d, rate=%d, "
+		   "format=0x%x, channels=%d, buffer_size=%ld, "
+		   "period_size=%ld, periods=%u, frames_to_bytes=%d\n",
+		   channel, runtime->rate, runtime->format, runtime->channels,
+		   runtime->buffer_size, runtime->period_size,
+		   runtime->periods, frames_to_bytes(runtime, 1));
+	snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, table_base=%p\n",
+		   runtime->dma_addr, runtime->dma_area, table_base);
+	snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",
+		   emu->p16v_buffer.addr, emu->p16v_buffer.area,
+		   emu->p16v_buffer.bytes);
+#endif /* debug */
 	tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, channel);
         switch (runtime->rate) {
 	case 44100:
@@ -379,7 +403,15 @@ static int snd_p16v_pcm_prepare_capture(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	int channel = substream->pcm->device - emu->p16v_device_offset;
 	u32 tmp;
-	//printk("prepare capture:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size,  frames_to_bytes(runtime, 1));
+
+	/*
+	printk(KERN_DEBUG "prepare capture:channel_number=%d, rate=%d, "
+	       "format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, "
+	       "frames_to_bytes=%d\n",
+	       channel, runtime->rate, runtime->format, runtime->channels,
+	       runtime->buffer_size, runtime->period_size,
+	       frames_to_bytes(runtime, 1));
+	*/
 	tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, channel);
         switch (runtime->rate) {
 	case 44100:
@@ -459,13 +491,13 @@ static int snd_p16v_pcm_trigger_playback(struct snd_pcm_substream *substream,
 		runtime = s->runtime;
 		epcm = runtime->private_data;
 		channel = substream->pcm->device-emu->p16v_device_offset;
-		//snd_printk("p16v channel=%d\n",channel);
+		/* snd_printk(KERN_DEBUG "p16v channel=%d\n", channel); */
 		epcm->running = running;
 		basic |= (0x1<<channel);
 		inte |= (INTE2_PLAYBACK_CH_0_LOOP<<channel);
                 snd_pcm_trigger_done(s, substream);
         }
-	//snd_printk("basic=0x%x, inte=0x%x\n",basic, inte);
+	/* snd_printk(KERN_DEBUG "basic=0x%x, inte=0x%x\n", basic, inte); */
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -558,8 +590,13 @@ snd_p16v_pcm_pointer_capture(struct snd_pcm_substream *substream)
 		ptr -= runtime->buffer_size;
 		printk(KERN_WARNING "buffer capture limited!\n");
 	}
-	//printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate);
-
+	/*
+	printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, "
+	       "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n",
+	       ptr1, ptr2, ptr, (int)runtime->buffer_size,
+	       (int)runtime->period_size, (int)runtime->frame_bits,
+	       (int)runtime->rate);
+	*/
 	return ptr;
 }
 
@@ -592,7 +629,10 @@ int snd_p16v_free(struct snd_emu10k1 *chip)
 	// release the data
 	if (chip->p16v_buffer.area) {
 		snd_dma_free_pages(&chip->p16v_buffer);
-		//snd_printk("period lables free: %p\n", &chip->p16v_buffer);
+		/*
+		snd_printk(KERN_DEBUG "period lables free: %p\n",
+			   &chip->p16v_buffer);
+		*/
 	}
 	return 0;
 }
@@ -604,7 +644,7 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm *
 	int err;
         int capture=1;
   
-	//snd_printk("snd_p16v_pcm called. device=%d\n", device);
+	/* snd_printk("KERN_DEBUG snd_p16v_pcm called. device=%d\n", device); */
 	emu->p16v_device_offset = device;
 	if (rpcm)
 		*rpcm = NULL;
@@ -631,7 +671,10 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm *
 							 snd_dma_pci_data(emu->pci), 
 							 ((65536 - 64) * 8), ((65536 - 64) * 8))) < 0) 
 			return err;
-		//snd_printk("preallocate playback substream: err=%d\n", err);
+		/*
+		snd_printk(KERN_DEBUG
+			   "preallocate playback substream: err=%d\n", err);
+		*/
 	}
 
 	for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; 
@@ -642,7 +685,10 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm *
 	                                           snd_dma_pci_data(emu->pci), 
 	                                           65536 - 64, 65536 - 64)) < 0)
 			return err;
-		//snd_printk("preallocate capture substream: err=%d\n", err);
+		/*
+		snd_printk(KERN_DEBUG
+			   "preallocate capture substream: err=%d\n", err);
+		*/
 	}
   
 	if (rpcm)
diff --git a/sound/pci/emu10k1/voice.c b/sound/pci/emu10k1/voice.c
index d7300a1aa262..20b8da250bd0 100644
--- a/sound/pci/emu10k1/voice.c
+++ b/sound/pci/emu10k1/voice.c
@@ -53,7 +53,10 @@ static int voice_alloc(struct snd_emu10k1 *emu, int type, int number,
 	*rvoice = NULL;
 	first_voice = last_voice = 0;
 	for (i = emu->next_free_voice, j = 0; j < NUM_G ; i += number, j += number) {
-		// printk("i %d j %d next free %d!\n", i, j, emu->next_free_voice);
+		/*
+		printk(KERN_DEBUG "i %d j %d next free %d!\n",
+		       i, j, emu->next_free_voice);
+		*/
 		i %= NUM_G;
 
 		/* stereo voices must be even/odd */
@@ -71,7 +74,7 @@ static int voice_alloc(struct snd_emu10k1 *emu, int type, int number,
 			}
 		}
 		if (!skip) {
-			// printk("allocated voice %d\n", i);
+			/* printk(KERN_DEBUG "allocated voice %d\n", i); */
 			first_voice = i;
 			last_voice = (i + number) % NUM_G;
 			emu->next_free_voice = last_voice;
@@ -84,7 +87,10 @@ static int voice_alloc(struct snd_emu10k1 *emu, int type, int number,
 	
 	for (i = 0; i < number; i++) {
 		voice = &emu->voices[(first_voice + i) % NUM_G];
-		// printk("voice alloc - %i, %i of %i\n", voice->number, idx-first_voice+1, number);
+		/*
+		printk(kERN_DEBUG "voice alloc - %i, %i of %i\n",
+		       voice->number, idx-first_voice+1, number);
+		*/
 		voice->use = 1;
 		switch (type) {
 		case EMU10K1_PCM:
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 9bf95367c882..18f4d1e98c46 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -584,7 +584,8 @@ static void snd_es1370_codec_write(struct snd_ak4531 *ak4531,
 	unsigned long end_time = jiffies + HZ / 10;
 
 #if 0
-	printk("CODEC WRITE: reg = 0x%x, val = 0x%x (0x%x), creg = 0x%x\n",
+	printk(KERN_DEBUG
+	       "CODEC WRITE: reg = 0x%x, val = 0x%x (0x%x), creg = 0x%x\n",
 	       reg, val, ES_1370_CODEC_WRITE(reg, val), ES_REG(ensoniq, 1370_CODEC));
 #endif
 	do {
@@ -2409,9 +2410,9 @@ static int __devinit snd_audiopci_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	if ((err = snd_ensoniq_create(card, pci, &ensoniq)) < 0) {
 		snd_card_free(card);
diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c
index 4cd9a1faaecc..dd63b132fb8e 100644
--- a/sound/pci/es1938.c
+++ b/sound/pci/es1938.c
@@ -1673,18 +1673,22 @@ static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id)
 
 	status = inb(SLIO_REG(chip, IRQCONTROL));
 #if 0
-	printk("Es1938debug - interrupt status: =0x%x\n", status);
+	printk(KERN_DEBUG "Es1938debug - interrupt status: =0x%x\n", status);
 #endif
 	
 	/* AUDIO 1 */
 	if (status & 0x10) {
 #if 0
-                printk("Es1938debug - AUDIO channel 1 interrupt\n");
-		printk("Es1938debug - AUDIO channel 1 DMAC DMA count: %u\n",
+                printk(KERN_DEBUG
+		       "Es1938debug - AUDIO channel 1 interrupt\n");
+		printk(KERN_DEBUG
+		       "Es1938debug - AUDIO channel 1 DMAC DMA count: %u\n",
 		       inw(SLDM_REG(chip, DMACOUNT)));
-		printk("Es1938debug - AUDIO channel 1 DMAC DMA base: %u\n",
+		printk(KERN_DEBUG
+		       "Es1938debug - AUDIO channel 1 DMAC DMA base: %u\n",
 		       inl(SLDM_REG(chip, DMAADDR)));
-		printk("Es1938debug - AUDIO channel 1 DMAC DMA status: 0x%x\n",
+		printk(KERN_DEBUG
+		       "Es1938debug - AUDIO channel 1 DMAC DMA status: 0x%x\n",
 		       inl(SLDM_REG(chip, DMASTATUS)));
 #endif
 		/* clear irq */
@@ -1699,10 +1703,13 @@ static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id)
 	/* AUDIO 2 */
 	if (status & 0x20) {
 #if 0
-                printk("Es1938debug - AUDIO channel 2 interrupt\n");
-		printk("Es1938debug - AUDIO channel 2 DMAC DMA count: %u\n",
+                printk(KERN_DEBUG
+		       "Es1938debug - AUDIO channel 2 interrupt\n");
+		printk(KERN_DEBUG
+		       "Es1938debug - AUDIO channel 2 DMAC DMA count: %u\n",
 		       inw(SLIO_REG(chip, AUDIO2DMACOUNT)));
-		printk("Es1938debug - AUDIO channel 2 DMAC DMA base: %u\n",
+		printk(KERN_DEBUG
+		       "Es1938debug - AUDIO channel 2 DMAC DMA base: %u\n",
 		       inl(SLIO_REG(chip, AUDIO2DMAADDR)));
 
 #endif
@@ -1799,9 +1806,9 @@ static int __devinit snd_es1938_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 	for (idx = 0; idx < 5; idx++) {
 		if (pci_resource_start(pci, idx) == 0 ||
 		    !(pci_resource_flags(pci, idx) & IORESOURCE_IO)) {
diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c
index e9c3794bbcb8..dc97e8116141 100644
--- a/sound/pci/es1968.c
+++ b/sound/pci/es1968.c
@@ -2645,9 +2645,9 @@ static int __devinit snd_es1968_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (!card)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
                 
 	if (total_bufsize[dev] < 128)
 		total_bufsize[dev] = 128;
diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c
index c129f9e2072c..60cdb9e0b68d 100644
--- a/sound/pci/fm801.c
+++ b/sound/pci/fm801.c
@@ -1468,9 +1468,9 @@ static int __devinit snd_card_fm801_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 	if ((err = snd_fm801_create(card, pci, tea575x_tuner[dev], &chip)) < 0) {
 		snd_card_free(card);
 		return err;
diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c
index 960fd7970384..4de5bacd3929 100644
--- a/sound/pci/hda/hda_beep.c
+++ b/sound/pci/hda/hda_beep.c
@@ -138,6 +138,7 @@ void snd_hda_detach_beep_device(struct hda_codec *codec)
 
 		input_unregister_device(beep->dev);
 		kfree(beep);
+		codec->beep = NULL;
 	}
 }
 EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device);
diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h
index b9679f081cae..51bf6a5daf39 100644
--- a/sound/pci/hda/hda_beep.h
+++ b/sound/pci/hda/hda_beep.h
@@ -39,7 +39,7 @@ struct hda_beep {
 int snd_hda_attach_beep_device(struct hda_codec *codec, int nid);
 void snd_hda_detach_beep_device(struct hda_codec *codec);
 #else
-#define snd_hda_attach_beep_device(...)
+#define snd_hda_attach_beep_device(...)		0
 #define snd_hda_detach_beep_device(...)
 #endif
 #endif
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index d03f99298be9..a4e5e5952115 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -647,9 +647,9 @@ static void /*__devinit*/ setup_fg_nodes(struct hda_codec *codec)
 
 	total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid);
 	for (i = 0; i < total_nodes; i++, nid++) {
-		unsigned int func;
-		func = snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE);
-		switch (func & 0xff) {
+		codec->function_id = snd_hda_param_read(codec, nid,
+						AC_PAR_FUNCTION_TYPE) & 0xff;
+		switch (codec->function_id) {
 		case AC_GRP_AUDIO_FUNCTION:
 			codec->afg = nid;
 			break;
@@ -682,11 +682,140 @@ static int read_widget_caps(struct hda_codec *codec, hda_nid_t fg_node)
 	return 0;
 }
 
+/* read all pin default configurations and save codec->init_pins */
+static int read_pin_defaults(struct hda_codec *codec)
+{
+	int i;
+	hda_nid_t nid = codec->start_nid;
+
+	for (i = 0; i < codec->num_nodes; i++, nid++) {
+		struct hda_pincfg *pin;
+		unsigned int wcaps = get_wcaps(codec, nid);
+		unsigned int wid_type = (wcaps & AC_WCAP_TYPE) >>
+				AC_WCAP_TYPE_SHIFT;
+		if (wid_type != AC_WID_PIN)
+			continue;
+		pin = snd_array_new(&codec->init_pins);
+		if (!pin)
+			return -ENOMEM;
+		pin->nid = nid;
+		pin->cfg = snd_hda_codec_read(codec, nid, 0,
+					      AC_VERB_GET_CONFIG_DEFAULT, 0);
+	}
+	return 0;
+}
+
+/* look up the given pin config list and return the item matching with NID */
+static struct hda_pincfg *look_up_pincfg(struct hda_codec *codec,
+					 struct snd_array *array,
+					 hda_nid_t nid)
+{
+	int i;
+	for (i = 0; i < array->used; i++) {
+		struct hda_pincfg *pin = snd_array_elem(array, i);
+		if (pin->nid == nid)
+			return pin;
+	}
+	return NULL;
+}
+
+/* write a config value for the given NID */
+static void set_pincfg(struct hda_codec *codec, hda_nid_t nid,
+		       unsigned int cfg)
+{
+	int i;
+	for (i = 0; i < 4; i++) {
+		snd_hda_codec_write(codec, nid, 0,
+				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 + i,
+				    cfg & 0xff);
+		cfg >>= 8;
+	}
+}
+
+/* set the current pin config value for the given NID.
+ * the value is cached, and read via snd_hda_codec_get_pincfg()
+ */
+int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list,
+		       hda_nid_t nid, unsigned int cfg)
+{
+	struct hda_pincfg *pin;
+	unsigned int oldcfg;
+
+	oldcfg = snd_hda_codec_get_pincfg(codec, nid);
+	pin = look_up_pincfg(codec, list, nid);
+	if (!pin) {
+		pin = snd_array_new(list);
+		if (!pin)
+			return -ENOMEM;
+		pin->nid = nid;
+	}
+	pin->cfg = cfg;
+
+	/* change only when needed; e.g. if the pincfg is already present
+	 * in user_pins[], don't write it
+	 */
+	cfg = snd_hda_codec_get_pincfg(codec, nid);
+	if (oldcfg != cfg)
+		set_pincfg(codec, nid, cfg);
+	return 0;
+}
+
+int snd_hda_codec_set_pincfg(struct hda_codec *codec,
+			     hda_nid_t nid, unsigned int cfg)
+{
+	return snd_hda_add_pincfg(codec, &codec->driver_pins, nid, cfg);
+}
+EXPORT_SYMBOL_HDA(snd_hda_codec_set_pincfg);
+
+/* get the current pin config value of the given pin NID */
+unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid)
+{
+	struct hda_pincfg *pin;
+
+#ifdef CONFIG_SND_HDA_HWDEP
+	pin = look_up_pincfg(codec, &codec->user_pins, nid);
+	if (pin)
+		return pin->cfg;
+#endif
+	pin = look_up_pincfg(codec, &codec->driver_pins, nid);
+	if (pin)
+		return pin->cfg;
+	pin = look_up_pincfg(codec, &codec->init_pins, nid);
+	if (pin)
+		return pin->cfg;
+	return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_codec_get_pincfg);
+
+/* restore all current pin configs */
+static void restore_pincfgs(struct hda_codec *codec)
+{
+	int i;
+	for (i = 0; i < codec->init_pins.used; i++) {
+		struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
+		set_pincfg(codec, pin->nid,
+			   snd_hda_codec_get_pincfg(codec, pin->nid));
+	}
+}
 
 static void init_hda_cache(struct hda_cache_rec *cache,
 			   unsigned int record_size);
 static void free_hda_cache(struct hda_cache_rec *cache);
 
+/* restore the initial pin cfgs and release all pincfg lists */
+static void restore_init_pincfgs(struct hda_codec *codec)
+{
+	/* first free driver_pins and user_pins, then call restore_pincfg
+	 * so that only the values in init_pins are restored
+	 */
+	snd_array_free(&codec->driver_pins);
+#ifdef CONFIG_SND_HDA_HWDEP
+	snd_array_free(&codec->user_pins);
+#endif
+	restore_pincfgs(codec);
+	snd_array_free(&codec->init_pins);
+}
+
 /*
  * codec destructor
  */
@@ -694,6 +823,7 @@ static void snd_hda_codec_free(struct hda_codec *codec)
 {
 	if (!codec)
 		return;
+	restore_init_pincfgs(codec);
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	cancel_delayed_work(&codec->power_work);
 	flush_workqueue(codec->bus->workq);
@@ -712,6 +842,9 @@ static void snd_hda_codec_free(struct hda_codec *codec)
 	kfree(codec);
 }
 
+static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
+				unsigned int power_state);
+
 /**
  * snd_hda_codec_new - create a HDA codec
  * @bus: the bus to assign
@@ -751,6 +884,8 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr
 	init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info));
 	init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head));
 	snd_array_init(&codec->mixers, sizeof(struct snd_kcontrol *), 32);
+	snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16);
+	snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16);
 	if (codec->bus->modelname) {
 		codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL);
 		if (!codec->modelname) {
@@ -787,15 +922,18 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr
 	setup_fg_nodes(codec);
 	if (!codec->afg && !codec->mfg) {
 		snd_printdd("hda_codec: no AFG or MFG node found\n");
-		snd_hda_codec_free(codec);
-		return -ENODEV;
+		err = -ENODEV;
+		goto error;
 	}
 
-	if (read_widget_caps(codec, codec->afg ? codec->afg : codec->mfg) < 0) {
+	err = read_widget_caps(codec, codec->afg ? codec->afg : codec->mfg);
+	if (err < 0) {
 		snd_printk(KERN_ERR "hda_codec: cannot malloc\n");
-		snd_hda_codec_free(codec);
-		return -ENOMEM;
+		goto error;
 	}
+	err = read_pin_defaults(codec);
+	if (err < 0)
+		goto error;
 
 	if (!codec->subsystem_id) {
 		hda_nid_t nid = codec->afg ? codec->afg : codec->mfg;
@@ -806,12 +944,15 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr
 	if (bus->modelname)
 		codec->modelname = kstrdup(bus->modelname, GFP_KERNEL);
 
+	/* power-up all before initialization */
+	hda_set_power_state(codec,
+			    codec->afg ? codec->afg : codec->mfg,
+			    AC_PWRST_D0);
+
 	if (do_init) {
 		err = snd_hda_codec_configure(codec);
-		if (err < 0) {
-			snd_hda_codec_free(codec);
-			return err;
-		}
+		if (err < 0)
+			goto error;
 	}
 	snd_hda_codec_proc_new(codec);
 
@@ -824,6 +965,10 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr
 	if (codecp)
 		*codecp = codec;
 	return 0;
+
+ error:
+	snd_hda_codec_free(codec);
+	return err;
 }
 EXPORT_SYMBOL_HDA(snd_hda_codec_new);
 
@@ -907,6 +1052,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup_stream);
 
 /* FIXME: more better hash key? */
 #define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24))
+#define HDA_HASH_PINCAP_KEY(nid) (u32)((nid) + (0x02 << 24))
 #define INFO_AMP_CAPS	(1<<0)
 #define INFO_AMP_VOL(ch)	(1 << (1 + (ch)))
 
@@ -997,6 +1143,21 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
 }
 EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps);
 
+u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid)
+{
+	struct hda_amp_info *info;
+
+	info = get_alloc_amp_hash(codec, HDA_HASH_PINCAP_KEY(nid));
+	if (!info)
+		return 0;
+	if (!info->head.val) {
+		info->amp_caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+		info->head.val |= INFO_AMP_CAPS;
+	}
+	return info->amp_caps;
+}
+EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps);
+
 /*
  * read the current volume to info
  * if the cache exists, read the cache value.
@@ -1120,6 +1281,7 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol,
 	u16 nid = get_amp_nid(kcontrol);
 	u8 chs = get_amp_channels(kcontrol);
 	int dir = get_amp_direction(kcontrol);
+	unsigned int ofs = get_amp_offset(kcontrol);
 	u32 caps;
 
 	caps = query_amp_caps(codec, nid, dir);
@@ -1131,6 +1293,8 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol,
 		       kcontrol->id.name);
 		return -EINVAL;
 	}
+	if (ofs < caps)
+		caps -= ofs;
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
 	uinfo->count = chs == 3 ? 2 : 1;
 	uinfo->value.integer.min = 0;
@@ -1139,6 +1303,32 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol,
 }
 EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_info);
 
+
+static inline unsigned int
+read_amp_value(struct hda_codec *codec, hda_nid_t nid,
+	       int ch, int dir, int idx, unsigned int ofs)
+{
+	unsigned int val;
+	val = snd_hda_codec_amp_read(codec, nid, ch, dir, idx);
+	val &= HDA_AMP_VOLMASK;
+	if (val >= ofs)
+		val -= ofs;
+	else
+		val = 0;
+	return val;
+}
+
+static inline int
+update_amp_value(struct hda_codec *codec, hda_nid_t nid,
+		 int ch, int dir, int idx, unsigned int ofs,
+		 unsigned int val)
+{
+	if (val > 0)
+		val += ofs;
+	return snd_hda_codec_amp_update(codec, nid, ch, dir, idx,
+					HDA_AMP_VOLMASK, val);
+}
+
 int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_value *ucontrol)
 {
@@ -1147,14 +1337,13 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol,
 	int chs = get_amp_channels(kcontrol);
 	int dir = get_amp_direction(kcontrol);
 	int idx = get_amp_index(kcontrol);
+	unsigned int ofs = get_amp_offset(kcontrol);
 	long *valp = ucontrol->value.integer.value;
 
 	if (chs & 1)
-		*valp++ = snd_hda_codec_amp_read(codec, nid, 0, dir, idx)
-			& HDA_AMP_VOLMASK;
+		*valp++ = read_amp_value(codec, nid, 0, dir, idx, ofs);
 	if (chs & 2)
-		*valp = snd_hda_codec_amp_read(codec, nid, 1, dir, idx)
-			& HDA_AMP_VOLMASK;
+		*valp = read_amp_value(codec, nid, 1, dir, idx, ofs);
 	return 0;
 }
 EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_get);
@@ -1167,18 +1356,17 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol,
 	int chs = get_amp_channels(kcontrol);
 	int dir = get_amp_direction(kcontrol);
 	int idx = get_amp_index(kcontrol);
+	unsigned int ofs = get_amp_offset(kcontrol);
 	long *valp = ucontrol->value.integer.value;
 	int change = 0;
 
 	snd_hda_power_up(codec);
 	if (chs & 1) {
-		change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx,
-						  0x7f, *valp);
+		change = update_amp_value(codec, nid, 0, dir, idx, ofs, *valp);
 		valp++;
 	}
 	if (chs & 2)
-		change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx,
-						   0x7f, *valp);
+		change |= update_amp_value(codec, nid, 1, dir, idx, ofs, *valp);
 	snd_hda_power_down(codec);
 	return change;
 }
@@ -1190,6 +1378,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	hda_nid_t nid = get_amp_nid(kcontrol);
 	int dir = get_amp_direction(kcontrol);
+	unsigned int ofs = get_amp_offset(kcontrol);
 	u32 caps, val1, val2;
 
 	if (size < 4 * sizeof(unsigned int))
@@ -1198,6 +1387,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 	val2 = (caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT;
 	val2 = (val2 + 1) * 25;
 	val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT);
+	val1 += ofs;
 	val1 = ((int)val1) * ((int)val2);
 	if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv))
 		return -EFAULT;
@@ -1268,7 +1458,6 @@ int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl)
 }
 EXPORT_SYMBOL_HDA(snd_hda_ctl_add);
 
-#ifdef CONFIG_SND_HDA_RECONFIG
 /* Clear all controls assigned to the given codec */
 void snd_hda_ctls_clear(struct hda_codec *codec)
 {
@@ -1279,9 +1468,52 @@ void snd_hda_ctls_clear(struct hda_codec *codec)
 	snd_array_free(&codec->mixers);
 }
 
-void snd_hda_codec_reset(struct hda_codec *codec)
+/* pseudo device locking
+ * toggle card->shutdown to allow/disallow the device access (as a hack)
+ */
+static int hda_lock_devices(struct snd_card *card)
 {
-	int i;
+	spin_lock(&card->files_lock);
+	if (card->shutdown) {
+		spin_unlock(&card->files_lock);
+		return -EINVAL;
+	}
+	card->shutdown = 1;
+	spin_unlock(&card->files_lock);
+	return 0;
+}
+
+static void hda_unlock_devices(struct snd_card *card)
+{
+	spin_lock(&card->files_lock);
+	card->shutdown = 0;
+	spin_unlock(&card->files_lock);
+}
+
+int snd_hda_codec_reset(struct hda_codec *codec)
+{
+	struct snd_card *card = codec->bus->card;
+	int i, pcm;
+
+	if (hda_lock_devices(card) < 0)
+		return -EBUSY;
+	/* check whether the codec isn't used by any mixer or PCM streams */
+	if (!list_empty(&card->ctl_files)) {
+		hda_unlock_devices(card);
+		return -EBUSY;
+	}
+	for (pcm = 0; pcm < codec->num_pcms; pcm++) {
+		struct hda_pcm *cpcm = &codec->pcm_info[pcm];
+		if (!cpcm->pcm)
+			continue;
+		if (cpcm->pcm->streams[0].substream_opened ||
+		    cpcm->pcm->streams[1].substream_opened) {
+			hda_unlock_devices(card);
+			return -EBUSY;
+		}
+	}
+
+	/* OK, let it free */
 
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	cancel_delayed_work(&codec->power_work);
@@ -1291,8 +1523,7 @@ void snd_hda_codec_reset(struct hda_codec *codec)
 	/* relase PCMs */
 	for (i = 0; i < codec->num_pcms; i++) {
 		if (codec->pcm_info[i].pcm) {
-			snd_device_free(codec->bus->card,
-					codec->pcm_info[i].pcm);
+			snd_device_free(card, codec->pcm_info[i].pcm);
 			clear_bit(codec->pcm_info[i].device,
 				  codec->bus->pcm_dev_bits);
 		}
@@ -1305,13 +1536,22 @@ void snd_hda_codec_reset(struct hda_codec *codec)
 	free_hda_cache(&codec->cmd_cache);
 	init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info));
 	init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head));
+	/* free only driver_pins so that init_pins + user_pins are restored */
+	snd_array_free(&codec->driver_pins);
+	restore_pincfgs(codec);
 	codec->num_pcms = 0;
 	codec->pcm_info = NULL;
 	codec->preset = NULL;
+	memset(&codec->patch_ops, 0, sizeof(codec->patch_ops));
+	codec->slave_dig_outs = NULL;
+	codec->spdif_status_reset = 0;
 	module_put(codec->owner);
 	codec->owner = NULL;
+
+	/* allow device access again */
+	hda_unlock_devices(card);
+	return 0;
 }
-#endif /* CONFIG_SND_HDA_RECONFIG */
 
 /* create a virtual master control and add slaves */
 int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
@@ -1336,15 +1576,20 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
 	
 	for (s = slaves; *s; s++) {
 		struct snd_kcontrol *sctl;
-
-		sctl = snd_hda_find_mixer_ctl(codec, *s);
-		if (!sctl) {
-			snd_printdd("Cannot find slave %s, skipped\n", *s);
-			continue;
+		int i = 0;
+		for (;;) {
+			sctl = _snd_hda_find_mixer_ctl(codec, *s, i);
+			if (!sctl) {
+				if (!i)
+					snd_printdd("Cannot find slave %s, "
+						    "skipped\n", *s);
+				break;
+			}
+			err = snd_ctl_add_slave(kctl, sctl);
+			if (err < 0)
+				return err;
+			i++;
 		}
-		err = snd_ctl_add_slave(kctl, sctl);
-		if (err < 0)
-			return err;
 	}
 	return 0;
 }
@@ -1955,6 +2200,8 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid)
 	}
 	for (dig_mix = dig_in_ctls; dig_mix->name; dig_mix++) {
 		kctl = snd_ctl_new1(dig_mix, codec);
+		if (!kctl)
+			return -ENOMEM;
 		kctl->private_value = nid;
 		err = snd_hda_ctl_add(codec, kctl);
 		if (err < 0)
@@ -2074,8 +2321,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg,
 				 * don't power down the widget if it controls
 				 * eapd and EAPD_BTLENABLE is set.
 				 */
-				pincap = snd_hda_param_read(codec, nid,
-							    AC_PAR_PIN_CAP);
+				pincap = snd_hda_query_pin_caps(codec, nid);
 				if (pincap & AC_PINCAP_EAPD) {
 					int eapd = snd_hda_codec_read(codec,
 						nid, 0,
@@ -2144,6 +2390,7 @@ static void hda_call_codec_resume(struct hda_codec *codec)
 	hda_set_power_state(codec,
 			    codec->afg ? codec->afg : codec->mfg,
 			    AC_PWRST_D0);
+	restore_pincfgs(codec); /* restore all current pin configs */
 	hda_exec_init_verbs(codec);
 	if (codec->patch_ops.resume)
 		codec->patch_ops.resume(codec);
@@ -2171,8 +2418,16 @@ int /*__devinit*/ snd_hda_build_controls(struct hda_bus *bus)
 
 	list_for_each_entry(codec, &bus->codec_list, list) {
 		int err = snd_hda_codec_build_controls(codec);
-		if (err < 0)
-			return err;
+		if (err < 0) {
+			printk(KERN_ERR "hda_codec: cannot build controls"
+			       "for #%d (error %d)\n", codec->addr, err); 
+			err = snd_hda_codec_reset(codec);
+			if (err < 0) {
+				printk(KERN_ERR
+				       "hda_codec: cannot revert codec\n");
+				return err;
+			}
+		}
 	}
 	return 0;
 }
@@ -2181,19 +2436,12 @@ EXPORT_SYMBOL_HDA(snd_hda_build_controls);
 int snd_hda_codec_build_controls(struct hda_codec *codec)
 {
 	int err = 0;
-	/* fake as if already powered-on */
-	hda_keep_power_on(codec);
-	/* then fire up */
-	hda_set_power_state(codec,
-			    codec->afg ? codec->afg : codec->mfg,
-			    AC_PWRST_D0);
 	hda_exec_init_verbs(codec);
 	/* continue to initialize... */
 	if (codec->patch_ops.init)
 		err = codec->patch_ops.init(codec);
 	if (!err && codec->patch_ops.build_controls)
 		err = codec->patch_ops.build_controls(codec);
-	snd_hda_power_down(codec);
 	if (err < 0)
 		return err;
 	return 0;
@@ -2306,12 +2554,11 @@ EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format);
 static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
 				u32 *ratesp, u64 *formatsp, unsigned int *bpsp)
 {
-	int i;
-	unsigned int val, streams;
+	unsigned int i, val, wcaps;
 
 	val = 0;
-	if (nid != codec->afg &&
-	    (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) {
+	wcaps = get_wcaps(codec, nid);
+	if (nid != codec->afg && (wcaps & AC_WCAP_FORMAT_OVRD)) {
 		val = snd_hda_param_read(codec, nid, AC_PAR_PCM);
 		if (val == -1)
 			return -EIO;
@@ -2325,15 +2572,20 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
 			if (val & (1 << i))
 				rates |= rate_bits[i].alsa_bits;
 		}
+		if (rates == 0) {
+			snd_printk(KERN_ERR "hda_codec: rates == 0 "
+				   "(nid=0x%x, val=0x%x, ovrd=%i)\n",
+					nid, val,
+					(wcaps & AC_WCAP_FORMAT_OVRD) ? 1 : 0);
+			return -EIO;
+		}
 		*ratesp = rates;
 	}
 
 	if (formatsp || bpsp) {
 		u64 formats = 0;
-		unsigned int bps;
-		unsigned int wcaps;
+		unsigned int streams, bps;
 
-		wcaps = get_wcaps(codec, nid);
 		streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM);
 		if (streams == -1)
 			return -EIO;
@@ -2386,6 +2638,15 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid,
 			formats |= SNDRV_PCM_FMTBIT_U8;
 			bps = 8;
 		}
+		if (formats == 0) {
+			snd_printk(KERN_ERR "hda_codec: formats == 0 "
+				   "(nid=0x%x, val=0x%x, ovrd=%i, "
+				   "streams=0x%x)\n",
+					nid, val,
+					(wcaps & AC_WCAP_FORMAT_OVRD) ? 1 : 0,
+					streams);
+			return -EIO;
+		}
 		if (formatsp)
 			*formatsp = formats;
 		if (bpsp)
@@ -2501,12 +2762,16 @@ static int hda_pcm_default_cleanup(struct hda_pcm_stream *hinfo,
 static int set_pcm_default_values(struct hda_codec *codec,
 				  struct hda_pcm_stream *info)
 {
+	int err;
+
 	/* query support PCM information from the given NID */
 	if (info->nid && (!info->rates || !info->formats)) {
-		snd_hda_query_supported_pcm(codec, info->nid,
+		err = snd_hda_query_supported_pcm(codec, info->nid,
 				info->rates ? NULL : &info->rates,
 				info->formats ? NULL : &info->formats,
 				info->maxbps ? NULL : &info->maxbps);
+		if (err < 0)
+			return err;
 	}
 	if (info->ops.open == NULL)
 		info->ops.open = hda_pcm_default_open_close;
@@ -2549,13 +2814,10 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type)
 		for (i = 0; i < ARRAY_SIZE(audio_idx); i++) {
 			dev = audio_idx[i];
 			if (!test_bit(dev, bus->pcm_dev_bits))
-				break;
-		}
-		if (i >= ARRAY_SIZE(audio_idx)) {
-			snd_printk(KERN_WARNING "Too many audio devices\n");
-			return -EAGAIN;
+				goto ok;
 		}
-		break;
+		snd_printk(KERN_WARNING "Too many audio devices\n");
+		return -EAGAIN;
 	case HDA_PCM_TYPE_SPDIF:
 	case HDA_PCM_TYPE_HDMI:
 	case HDA_PCM_TYPE_MODEM:
@@ -2570,6 +2832,7 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type)
 		snd_printk(KERN_WARNING "Invalid PCM type %d\n", type);
 		return -EINVAL;
 	}
+ ok:
 	set_bit(dev, bus->pcm_dev_bits);
 	return dev;
 }
@@ -2606,24 +2869,36 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec)
 		if (!codec->patch_ops.build_pcms)
 			return 0;
 		err = codec->patch_ops.build_pcms(codec);
-		if (err < 0)
-			return err;
+		if (err < 0) {
+			printk(KERN_ERR "hda_codec: cannot build PCMs"
+			       "for #%d (error %d)\n", codec->addr, err); 
+			err = snd_hda_codec_reset(codec);
+			if (err < 0) {
+				printk(KERN_ERR
+				       "hda_codec: cannot revert codec\n");
+				return err;
+			}
+		}
 	}
 	for (pcm = 0; pcm < codec->num_pcms; pcm++) {
 		struct hda_pcm *cpcm = &codec->pcm_info[pcm];
 		int dev;
 
 		if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams)
-			return 0; /* no substreams assigned */
+			continue; /* no substreams assigned */
 
 		if (!cpcm->pcm) {
 			dev = get_empty_pcm_device(codec->bus, cpcm->pcm_type);
 			if (dev < 0)
-				return 0;
+				continue; /* no fatal error */
 			cpcm->device = dev;
 			err = snd_hda_attach_pcm(codec, cpcm);
-			if (err < 0)
-				return err;
+			if (err < 0) {
+				printk(KERN_ERR "hda_codec: cannot attach "
+				       "PCM stream %d for codec #%d\n",
+				       dev, codec->addr);
+				continue; /* no fatal error */
+			}
 		}
 	}
 	return 0;
@@ -3324,8 +3599,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 		if (ignore_nids && is_in_nid_list(nid, ignore_nids))
 			continue;
 
-		def_conf = snd_hda_codec_read(codec, nid, 0,
-					      AC_VERB_GET_CONFIG_DEFAULT, 0);
+		def_conf = snd_hda_codec_get_pincfg(codec, nid);
 		if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
 			continue;
 		loc = get_defcfg_location(def_conf);
@@ -3401,10 +3675,22 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 			cfg->input_pins[AUTO_PIN_AUX] = nid;
 			break;
 		case AC_JACK_SPDIF_OUT:
-			cfg->dig_out_pin = nid;
+		case AC_JACK_DIG_OTHER_OUT:
+			if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins))
+				continue;
+			cfg->dig_out_pins[cfg->dig_outs] = nid;
+			cfg->dig_out_type[cfg->dig_outs] =
+				(loc == AC_JACK_LOC_HDMI) ?
+				HDA_PCM_TYPE_HDMI : HDA_PCM_TYPE_SPDIF;
+			cfg->dig_outs++;
 			break;
 		case AC_JACK_SPDIF_IN:
+		case AC_JACK_DIG_OTHER_IN:
 			cfg->dig_in_pin = nid;
+			if (loc == AC_JACK_LOC_HDMI)
+				cfg->dig_in_type = HDA_PCM_TYPE_HDMI;
+			else
+				cfg->dig_in_type = HDA_PCM_TYPE_SPDIF;
 			break;
 		}
 	}
@@ -3510,6 +3796,9 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 		   cfg->hp_pins[1], cfg->hp_pins[2],
 		   cfg->hp_pins[3], cfg->hp_pins[4]);
 	snd_printd("   mono: mono_out=0x%x\n", cfg->mono_out_pin);
+	if (cfg->dig_outs)
+		snd_printd("   dig-out=0x%x/0x%x\n",
+			   cfg->dig_out_pins[0], cfg->dig_out_pins[1]);
 	snd_printd("   inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x,"
 		   " cd=0x%x, aux=0x%x\n",
 		   cfg->input_pins[AUTO_PIN_MIC],
@@ -3518,6 +3807,8 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec,
 		   cfg->input_pins[AUTO_PIN_FRONT_LINE],
 		   cfg->input_pins[AUTO_PIN_CD],
 		   cfg->input_pins[AUTO_PIN_AUX]);
+	if (cfg->dig_in_pin)
+		snd_printd("   dig-in=0x%x\n", cfg->dig_in_pin);
 
 	return 0;
 }
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 09a332ada0c6..2fdecf4b0eb6 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -739,6 +739,7 @@ struct hda_codec {
 	hda_nid_t mfg;	/* MFG node id */
 
 	/* ids */
+	u32 function_id;
 	u32 vendor_id;
 	u32 subsystem_id;
 	u32 revision_id;
@@ -778,11 +779,14 @@ struct hda_codec {
 	unsigned short spdif_ctls;	/* SPDIF control bits */
 	unsigned int spdif_in_enable;	/* SPDIF input enable? */
 	hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */
+	struct snd_array init_pins;	/* initial (BIOS) pin configurations */
+	struct snd_array driver_pins;	/* pin configs set by codec parser */
 
 #ifdef CONFIG_SND_HDA_HWDEP
 	struct snd_hwdep *hwdep;	/* assigned hwdep device */
 	struct snd_array init_verbs;	/* additional init verbs */
 	struct snd_array hints;		/* additional hints */
+	struct snd_array user_pins;	/* default pin configs to override */
 #endif
 
 	/* misc flags */
@@ -790,6 +794,9 @@ struct hda_codec {
 					     * status change
 					     * (e.g. Realtek codecs)
 					     */
+	unsigned int pin_amp_workaround:1; /* pin out-amp takes index
+					    * (e.g. Conexant codecs)
+					    */
 #ifdef CONFIG_SND_HDA_POWER_SAVE
 	unsigned int power_on :1;	/* current (global) power-state */
 	unsigned int power_transition :1; /* power-state in transition */
@@ -855,6 +862,18 @@ void snd_hda_codec_resume_cache(struct hda_codec *codec);
 #define snd_hda_sequence_write_cache	snd_hda_sequence_write
 #endif
 
+/* the struct for codec->pin_configs */
+struct hda_pincfg {
+	hda_nid_t nid;
+	unsigned int cfg;
+};
+
+unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid);
+int snd_hda_codec_set_pincfg(struct hda_codec *codec, hda_nid_t nid,
+			     unsigned int cfg);
+int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list,
+		       hda_nid_t nid, unsigned int cfg); /* for hwdep */
+
 /*
  * Mixer
  */
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 65745e96dc70..1d5797a96682 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -144,9 +144,9 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid
 	node->type = (node->wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
 
 	if (node->type == AC_WID_PIN) {
-		node->pin_caps = snd_hda_param_read(codec, node->nid, AC_PAR_PIN_CAP);
+		node->pin_caps = snd_hda_query_pin_caps(codec, node->nid);
 		node->pin_ctl = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		node->def_cfg = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+		node->def_cfg = snd_hda_codec_get_pincfg(codec, node->nid);
 	}
 
 	if (node->wid_caps & AC_WCAP_OUT_AMP) {
diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c
index 4ae51dcb81af..1c57505c2874 100644
--- a/sound/pci/hda/hda_hwdep.c
+++ b/sound/pci/hda/hda_hwdep.c
@@ -30,6 +30,12 @@
 #include <sound/hda_hwdep.h>
 #include <sound/minors.h>
 
+/* hint string pair */
+struct hda_hint {
+	const char *key;
+	const char *val;	/* contained in the same alloc as key */
+};
+
 /*
  * write/read an out-of-bound verb
  */
@@ -99,16 +105,17 @@ static int hda_hwdep_open(struct snd_hwdep *hw, struct file *file)
 
 static void clear_hwdep_elements(struct hda_codec *codec)
 {
-	char **head;
 	int i;
 
 	/* clear init verbs */
 	snd_array_free(&codec->init_verbs);
 	/* clear hints */
-	head = codec->hints.list;
-	for (i = 0; i < codec->hints.used; i++, head++)
-		kfree(*head);
+	for (i = 0; i < codec->hints.used; i++) {
+		struct hda_hint *hint = snd_array_elem(&codec->hints, i);
+		kfree(hint->key); /* we don't need to free hint->val */
+	}
 	snd_array_free(&codec->hints);
+	snd_array_free(&codec->user_pins);
 }
 
 static void hwdep_free(struct snd_hwdep *hwdep)
@@ -140,7 +147,8 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec)
 #endif
 
 	snd_array_init(&codec->init_verbs, sizeof(struct hda_verb), 32);
-	snd_array_init(&codec->hints, sizeof(char *), 32);
+	snd_array_init(&codec->hints, sizeof(struct hda_hint), 32);
+	snd_array_init(&codec->user_pins, sizeof(struct hda_pincfg), 16);
 
 	return 0;
 }
@@ -153,7 +161,13 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec)
 
 static int clear_codec(struct hda_codec *codec)
 {
-	snd_hda_codec_reset(codec);
+	int err;
+
+	err = snd_hda_codec_reset(codec);
+	if (err < 0) {
+		snd_printk(KERN_ERR "The codec is being used, can't free.\n");
+		return err;
+	}
 	clear_hwdep_elements(codec);
 	return 0;
 }
@@ -162,20 +176,29 @@ static int reconfig_codec(struct hda_codec *codec)
 {
 	int err;
 
+	snd_hda_power_up(codec);
 	snd_printk(KERN_INFO "hda-codec: reconfiguring\n");
-	snd_hda_codec_reset(codec);
+	err = snd_hda_codec_reset(codec);
+	if (err < 0) {
+		snd_printk(KERN_ERR
+			   "The codec is being used, can't reconfigure.\n");
+		goto error;
+	}
 	err = snd_hda_codec_configure(codec);
 	if (err < 0)
-		return err;
+		goto error;
 	/* rebuild PCMs */
 	err = snd_hda_codec_build_pcms(codec);
 	if (err < 0)
-		return err;
+		goto error;
 	/* rebuild mixers */
 	err = snd_hda_codec_build_controls(codec);
 	if (err < 0)
-		return err;
-	return snd_card_register(codec->bus->card);
+		goto error;
+	err = snd_card_register(codec->bus->card);
+ error:
+	snd_hda_power_down(codec);
+	return err;
 }
 
 /*
@@ -271,6 +294,22 @@ static ssize_t type##_store(struct device *dev,			\
 CODEC_ACTION_STORE(reconfig);
 CODEC_ACTION_STORE(clear);
 
+static ssize_t init_verbs_show(struct device *dev,
+			       struct device_attribute *attr,
+			       char *buf)
+{
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+	struct hda_codec *codec = hwdep->private_data;
+	int i, len = 0;
+	for (i = 0; i < codec->init_verbs.used; i++) {
+		struct hda_verb *v = snd_array_elem(&codec->init_verbs, i);
+		len += snprintf(buf + len, PAGE_SIZE - len,
+				"0x%02x 0x%03x 0x%04x\n",
+				v->nid, v->verb, v->param);
+	}
+	return len;
+}
+
 static ssize_t init_verbs_store(struct device *dev,
 				struct device_attribute *attr,
 				const char *buf, size_t count)
@@ -293,26 +332,157 @@ static ssize_t init_verbs_store(struct device *dev,
 	return count;
 }
 
+static ssize_t hints_show(struct device *dev,
+			  struct device_attribute *attr,
+			  char *buf)
+{
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+	struct hda_codec *codec = hwdep->private_data;
+	int i, len = 0;
+	for (i = 0; i < codec->hints.used; i++) {
+		struct hda_hint *hint = snd_array_elem(&codec->hints, i);
+		len += snprintf(buf + len, PAGE_SIZE - len,
+				"%s = %s\n", hint->key, hint->val);
+	}
+	return len;
+}
+
+static struct hda_hint *get_hint(struct hda_codec *codec, const char *key)
+{
+	int i;
+
+	for (i = 0; i < codec->hints.used; i++) {
+		struct hda_hint *hint = snd_array_elem(&codec->hints, i);
+		if (!strcmp(hint->key, key))
+			return hint;
+	}
+	return NULL;
+}
+
+static void remove_trail_spaces(char *str)
+{
+	char *p;
+	if (!*str)
+		return;
+	p = str + strlen(str) - 1;
+	for (; isspace(*p); p--) {
+		*p = 0;
+		if (p == str)
+			return;
+	}
+}
+
+#define MAX_HINTS	1024
+
 static ssize_t hints_store(struct device *dev,
 			   struct device_attribute *attr,
 			   const char *buf, size_t count)
 {
 	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
 	struct hda_codec *codec = hwdep->private_data;
-	char *p;
-	char **hint;
+	char *key, *val;
+	struct hda_hint *hint;
 
-	if (!*buf || isspace(*buf) || *buf == '#' || *buf == '\n')
+	while (isspace(*buf))
+		buf++;
+	if (!*buf || *buf == '#' || *buf == '\n')
 		return count;
-	p = kstrndup_noeol(buf, 1024);
-	if (!p)
+	if (*buf == '=')
+		return -EINVAL;
+	key = kstrndup_noeol(buf, 1024);
+	if (!key)
 		return -ENOMEM;
-	hint = snd_array_new(&codec->hints);
+	/* extract key and val */
+	val = strchr(key, '=');
+	if (!val) {
+		kfree(key);
+		return -EINVAL;
+	}
+	*val++ = 0;
+	while (isspace(*val))
+		val++;
+	remove_trail_spaces(key);
+	remove_trail_spaces(val);
+	hint = get_hint(codec, key);
+	if (hint) {
+		/* replace */
+		kfree(hint->key);
+		hint->key = key;
+		hint->val = val;
+		return count;
+	}
+	/* allocate a new hint entry */
+	if (codec->hints.used >= MAX_HINTS)
+		hint = NULL;
+	else
+		hint = snd_array_new(&codec->hints);
 	if (!hint) {
-		kfree(p);
+		kfree(key);
 		return -ENOMEM;
 	}
-	*hint = p;
+	hint->key = key;
+	hint->val = val;
+	return count;
+}
+
+static ssize_t pin_configs_show(struct hda_codec *codec,
+				struct snd_array *list,
+				char *buf)
+{
+	int i, len = 0;
+	for (i = 0; i < list->used; i++) {
+		struct hda_pincfg *pin = snd_array_elem(list, i);
+		len += sprintf(buf + len, "0x%02x 0x%08x\n",
+			       pin->nid, pin->cfg);
+	}
+	return len;
+}
+
+static ssize_t init_pin_configs_show(struct device *dev,
+				     struct device_attribute *attr,
+				     char *buf)
+{
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+	struct hda_codec *codec = hwdep->private_data;
+	return pin_configs_show(codec, &codec->init_pins, buf);
+}
+
+static ssize_t user_pin_configs_show(struct device *dev,
+				     struct device_attribute *attr,
+				     char *buf)
+{
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+	struct hda_codec *codec = hwdep->private_data;
+	return pin_configs_show(codec, &codec->user_pins, buf);
+}
+
+static ssize_t driver_pin_configs_show(struct device *dev,
+				       struct device_attribute *attr,
+				       char *buf)
+{
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+	struct hda_codec *codec = hwdep->private_data;
+	return pin_configs_show(codec, &codec->driver_pins, buf);
+}
+
+#define MAX_PIN_CONFIGS		32
+
+static ssize_t user_pin_configs_store(struct device *dev,
+				      struct device_attribute *attr,
+				      const char *buf, size_t count)
+{
+	struct snd_hwdep *hwdep = dev_get_drvdata(dev);
+	struct hda_codec *codec = hwdep->private_data;
+	int nid, cfg;
+	int err;
+
+	if (sscanf(buf, "%i %i", &nid, &cfg) != 2)
+		return -EINVAL;
+	if (!nid)
+		return -EINVAL;
+	err = snd_hda_add_pincfg(codec, &codec->user_pins, nid, cfg);
+	if (err < 0)
+		return err;
 	return count;
 }
 
@@ -331,8 +501,11 @@ static struct device_attribute codec_attrs[] = {
 	CODEC_ATTR_RO(mfg),
 	CODEC_ATTR_RW(name),
 	CODEC_ATTR_RW(modelname),
-	CODEC_ATTR_WO(init_verbs),
-	CODEC_ATTR_WO(hints),
+	CODEC_ATTR_RW(init_verbs),
+	CODEC_ATTR_RW(hints),
+	CODEC_ATTR_RO(init_pin_configs),
+	CODEC_ATTR_RW(user_pin_configs),
+	CODEC_ATTR_RO(driver_pin_configs),
 	CODEC_ATTR_WO(reconfig),
 	CODEC_ATTR_WO(clear),
 };
@@ -351,4 +524,29 @@ int snd_hda_hwdep_add_sysfs(struct hda_codec *codec)
 	return 0;
 }
 
+/*
+ * Look for hint string
+ */
+const char *snd_hda_get_hint(struct hda_codec *codec, const char *key)
+{
+	struct hda_hint *hint = get_hint(codec, key);
+	return hint ? hint->val : NULL;
+}
+EXPORT_SYMBOL_HDA(snd_hda_get_hint);
+
+int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key)
+{
+	const char *p = snd_hda_get_hint(codec, key);
+	if (!p || !*p)
+		return -ENOENT;
+	switch (toupper(*p)) {
+	case 'T': /* true */
+	case 'Y': /* yes */
+	case '1':
+		return 1;
+	}
+	return 0;
+}
+EXPORT_SYMBOL_HDA(snd_hda_get_bool_hint);
+
 #endif /* CONFIG_SND_HDA_RECONFIG */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 5e909e0da04b..30829ee920c3 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -381,6 +381,7 @@ struct azx {
 
 	/* HD codec */
 	unsigned short codec_mask;
+	int  codec_probe_mask; /* copied from probe_mask option */
 	struct hda_bus *bus;
 
 	/* CORB/RIRB */
@@ -858,13 +859,18 @@ static void azx_stream_start(struct azx *chip, struct azx_dev *azx_dev)
 		      SD_CTL_DMA_START | SD_INT_MASK);
 }
 
-/* stop a stream */
-static void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev)
+/* stop DMA */
+static void azx_stream_clear(struct azx *chip, struct azx_dev *azx_dev)
 {
-	/* stop DMA */
 	azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) &
 		      ~(SD_CTL_DMA_START | SD_INT_MASK));
 	azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */
+}
+
+/* stop a stream */
+static void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev)
+{
+	azx_stream_clear(chip, azx_dev);
 	/* disable SIE */
 	azx_writeb(chip, INTCTL,
 		   azx_readb(chip, INTCTL) & ~(1 << azx_dev->index));
@@ -1075,8 +1081,7 @@ static int azx_setup_periods(struct azx *chip,
 	azx_sd_writel(azx_dev, SD_BDLPL, 0);
 	azx_sd_writel(azx_dev, SD_BDLPU, 0);
 
-	period_bytes = snd_pcm_lib_period_bytes(substream);
-	azx_dev->period_bytes = period_bytes;
+	period_bytes = azx_dev->period_bytes;
 	periods = azx_dev->bufsize / period_bytes;
 
 	/* program the initial BDL entries */
@@ -1123,24 +1128,17 @@ static int azx_setup_periods(struct azx *chip,
  error:
 	snd_printk(KERN_ERR "Too many BDL entries: buffer=%d, period=%d\n",
 		   azx_dev->bufsize, period_bytes);
-	/* reset */
-	azx_sd_writel(azx_dev, SD_BDLPL, 0);
-	azx_sd_writel(azx_dev, SD_BDLPU, 0);
 	return -EINVAL;
 }
 
-/*
- * set up the SD for streaming
- */
-static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
+/* reset stream */
+static void azx_stream_reset(struct azx *chip, struct azx_dev *azx_dev)
 {
 	unsigned char val;
 	int timeout;
 
-	/* make sure the run bit is zero for SD */
-	azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) &
-		      ~SD_CTL_DMA_START);
-	/* reset stream */
+	azx_stream_clear(chip, azx_dev);
+
 	azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) |
 		      SD_CTL_STREAM_RESET);
 	udelay(3);
@@ -1157,7 +1155,15 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
 	while (((val = azx_sd_readb(azx_dev, SD_CTL)) & SD_CTL_STREAM_RESET) &&
 	       --timeout)
 		;
+}
 
+/*
+ * set up the SD for streaming
+ */
+static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev)
+{
+	/* make sure the run bit is zero for SD */
+	azx_stream_clear(chip, azx_dev);
 	/* program the stream_tag */
 	azx_sd_writel(azx_dev, SD_CTL,
 		      (azx_sd_readl(azx_dev, SD_CTL) & ~SD_CTL_STREAM_TAG_MASK)|
@@ -1228,7 +1234,6 @@ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] __devinitdata = {
 };
 
 static int __devinit azx_codec_create(struct azx *chip, const char *model,
-				      unsigned int codec_probe_mask,
 				      int no_init)
 {
 	struct hda_bus_template bus_temp;
@@ -1261,7 +1266,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model,
 
 	/* First try to probe all given codec slots */
 	for (c = 0; c < max_slots; c++) {
-		if ((chip->codec_mask & (1 << c)) & codec_probe_mask) {
+		if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) {
 			if (probe_codec(chip, c) < 0) {
 				/* Some BIOSen give you wrong codec addresses
 				 * that don't exist
@@ -1285,7 +1290,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model,
 
 	/* Then create codec instances */
 	for (c = 0; c < max_slots; c++) {
-		if ((chip->codec_mask & (1 << c)) & codec_probe_mask) {
+		if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) {
 			struct hda_codec *codec;
 			err = snd_hda_codec_new(chip->bus, c, !no_init, &codec);
 			if (err < 0)
@@ -1403,6 +1408,8 @@ static int azx_pcm_open(struct snd_pcm_substream *substream)
 	runtime->private_data = azx_dev;
 	snd_pcm_set_sync(substream);
 	mutex_unlock(&chip->open_mutex);
+
+	azx_stream_reset(chip, azx_dev);
 	return 0;
 }
 
@@ -1429,6 +1436,11 @@ static int azx_pcm_close(struct snd_pcm_substream *substream)
 static int azx_pcm_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *hw_params)
 {
+	struct azx_dev *azx_dev = get_azx_dev(substream);
+
+	azx_dev->bufsize = 0;
+	azx_dev->period_bytes = 0;
+	azx_dev->format_val = 0;
 	return snd_pcm_lib_malloc_pages(substream,
 					params_buffer_bytes(hw_params));
 }
@@ -1443,6 +1455,9 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream)
 	azx_sd_writel(azx_dev, SD_BDLPL, 0);
 	azx_sd_writel(azx_dev, SD_BDLPU, 0);
 	azx_sd_writel(azx_dev, SD_CTL, 0);
+	azx_dev->bufsize = 0;
+	azx_dev->period_bytes = 0;
+	azx_dev->format_val = 0;
 
 	hinfo->ops.cleanup(hinfo, apcm->codec, substream);
 
@@ -1456,23 +1471,37 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream)
 	struct azx_dev *azx_dev = get_azx_dev(substream);
 	struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream];
 	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned int bufsize, period_bytes, format_val;
+	int err;
 
-	azx_dev->bufsize = snd_pcm_lib_buffer_bytes(substream);
-	azx_dev->format_val = snd_hda_calc_stream_format(runtime->rate,
-							 runtime->channels,
-							 runtime->format,
-							 hinfo->maxbps);
-	if (!azx_dev->format_val) {
+	format_val = snd_hda_calc_stream_format(runtime->rate,
+						runtime->channels,
+						runtime->format,
+						hinfo->maxbps);
+	if (!format_val) {
 		snd_printk(KERN_ERR SFX
 			   "invalid format_val, rate=%d, ch=%d, format=%d\n",
 			   runtime->rate, runtime->channels, runtime->format);
 		return -EINVAL;
 	}
 
+	bufsize = snd_pcm_lib_buffer_bytes(substream);
+	period_bytes = snd_pcm_lib_period_bytes(substream);
+
 	snd_printdd("azx_pcm_prepare: bufsize=0x%x, format=0x%x\n",
-		    azx_dev->bufsize, azx_dev->format_val);
-	if (azx_setup_periods(chip, substream, azx_dev) < 0)
-		return -EINVAL;
+		    bufsize, format_val);
+
+	if (bufsize != azx_dev->bufsize ||
+	    period_bytes != azx_dev->period_bytes ||
+	    format_val != azx_dev->format_val) {
+		azx_dev->bufsize = bufsize;
+		azx_dev->period_bytes = period_bytes;
+		azx_dev->format_val = format_val;
+		err = azx_setup_periods(chip, substream, azx_dev);
+		if (err < 0)
+			return err;
+	}
+
 	azx_setup_controller(chip, azx_dev);
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		azx_dev->fifo_size = azx_sd_readw(azx_dev, SD_FIFOSIZE) + 1;
@@ -2059,26 +2088,31 @@ static int __devinit check_position_fix(struct azx *chip, int fix)
 {
 	const struct snd_pci_quirk *q;
 
-	/* Check VIA HD Audio Controller exist */
-	if (chip->pci->vendor == PCI_VENDOR_ID_VIA &&
-	    chip->pci->device == VIA_HDAC_DEVICE_ID) {
+	switch (fix) {
+	case POS_FIX_LPIB:
+	case POS_FIX_POSBUF:
+		return fix;
+	}
+
+	/* Check VIA/ATI HD Audio Controller exist */
+	switch (chip->driver_type) {
+	case AZX_DRIVER_VIA:
+	case AZX_DRIVER_ATI:
 		chip->via_dmapos_patch = 1;
 		/* Use link position directly, avoid any transfer problem. */
 		return POS_FIX_LPIB;
 	}
 	chip->via_dmapos_patch = 0;
 
-	if (fix == POS_FIX_AUTO) {
-		q = snd_pci_quirk_lookup(chip->pci, position_fix_list);
-		if (q) {
-			printk(KERN_INFO
-				    "hda_intel: position_fix set to %d "
-				    "for device %04x:%04x\n",
-				    q->value, q->subvendor, q->subdevice);
-			return q->value;
-		}
+	q = snd_pci_quirk_lookup(chip->pci, position_fix_list);
+	if (q) {
+		printk(KERN_INFO
+		       "hda_intel: position_fix set to %d "
+		       "for device %04x:%04x\n",
+		       q->value, q->subvendor, q->subdevice);
+		return q->value;
 	}
-	return fix;
+	return POS_FIX_AUTO;
 }
 
 /*
@@ -2095,25 +2129,36 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1028, 0x20ac, "Dell Studio Desktop", 0x01),
 	/* including bogus ALC268 in slot#2 that conflicts with ALC888 */
 	SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01),
-	/* conflict of ALC268 in slot#3 (digital I/O); a temporary fix */
-	SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba laptop", 0x03),
+	/* forced codec slots */
+	SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103),
 	{}
 };
 
+#define AZX_FORCE_CODEC_MASK	0x100
+
 static void __devinit check_probe_mask(struct azx *chip, int dev)
 {
 	const struct snd_pci_quirk *q;
 
-	if (probe_mask[dev] == -1) {
+	chip->codec_probe_mask = probe_mask[dev];
+	if (chip->codec_probe_mask == -1) {
 		q = snd_pci_quirk_lookup(chip->pci, probe_mask_list);
 		if (q) {
 			printk(KERN_INFO
 			       "hda_intel: probe_mask set to 0x%x "
 			       "for device %04x:%04x\n",
 			       q->value, q->subvendor, q->subdevice);
-			probe_mask[dev] = q->value;
+			chip->codec_probe_mask = q->value;
 		}
 	}
+
+	/* check forced option */
+	if (chip->codec_probe_mask != -1 &&
+	    (chip->codec_probe_mask & AZX_FORCE_CODEC_MASK)) {
+		chip->codec_mask = chip->codec_probe_mask & 0xff;
+		printk(KERN_INFO "hda_intel: codec_mask forced to 0x%x\n",
+		       chip->codec_mask);
+	}
 }
 
 
@@ -2210,9 +2255,17 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci,
 	gcap = azx_readw(chip, GCAP);
 	snd_printdd("chipset global capabilities = 0x%x\n", gcap);
 
+	/* ATI chips seems buggy about 64bit DMA addresses */
+	if (chip->driver_type == AZX_DRIVER_ATI)
+		gcap &= ~0x01;
+
 	/* allow 64bit DMA address if supported by H/W */
 	if ((gcap & 0x01) && !pci_set_dma_mask(pci, DMA_64BIT_MASK))
 		pci_set_consistent_dma_mask(pci, DMA_64BIT_MASK);
+	else {
+		pci_set_dma_mask(pci, DMA_32BIT_MASK);
+		pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK);
+	}
 
 	/* read number of streams from GCAP register instead of using
 	 * hardcoded value
@@ -2334,10 +2387,10 @@ static int __devinit azx_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (!card) {
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0) {
 		snd_printk(KERN_ERR SFX "Error creating card!\n");
-		return -ENOMEM;
+		return err;
 	}
 
 	err = azx_create(card, pci, dev, pci_id->driver_data, &chip);
@@ -2346,8 +2399,7 @@ static int __devinit azx_probe(struct pci_dev *pci,
 	card->private_data = chip;
 
 	/* create codec instances */
-	err = azx_codec_create(chip, model[dev], probe_mask[dev],
-			       probe_only[dev]);
+	err = azx_codec_create(chip, model[dev], probe_only[dev]);
 	if (err < 0)
 		goto out_free;
 
@@ -2444,10 +2496,10 @@ static struct pci_device_id azx_ids[] = {
 	{ PCI_DEVICE(0x10de, 0x0ac1), .driver_data = AZX_DRIVER_NVIDIA },
 	{ PCI_DEVICE(0x10de, 0x0ac2), .driver_data = AZX_DRIVER_NVIDIA },
 	{ PCI_DEVICE(0x10de, 0x0ac3), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0bd4), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0bd5), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0bd6), .driver_data = AZX_DRIVER_NVIDIA },
-	{ PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA },
+	{ PCI_DEVICE(0x10de, 0x0d94), .driver_data = AZX_DRIVER_NVIDIA },
+	{ PCI_DEVICE(0x10de, 0x0d95), .driver_data = AZX_DRIVER_NVIDIA },
+	{ PCI_DEVICE(0x10de, 0x0d96), .driver_data = AZX_DRIVER_NVIDIA },
+	{ PCI_DEVICE(0x10de, 0x0d97), .driver_data = AZX_DRIVER_NVIDIA },
 	/* Teradici */
 	{ PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA },
 	/* AMD Generic, PCI class code and Vendor ID for HD Audio */
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 44f189cb97ae..83349013b4df 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -26,8 +26,10 @@
 /*
  * for mixer controls
  */
+#define HDA_COMPOSE_AMP_VAL_OFS(nid,chs,idx,dir,ofs)		\
+	((nid) | ((chs)<<16) | ((dir)<<18) | ((idx)<<19) | ((ofs)<<23))
 #define HDA_COMPOSE_AMP_VAL(nid,chs,idx,dir) \
-	((nid) | ((chs)<<16) | ((dir)<<18) | ((idx)<<19))
+	HDA_COMPOSE_AMP_VAL_OFS(nid, chs, idx, dir, 0)
 /* mono volume with index (index=0,1,...) (channel=1,2) */
 #define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \
 	{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx,  \
@@ -96,7 +98,7 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec,
 					    const char *name);
 int snd_hda_add_vmaster(struct hda_codec *codec, char *name,
 			unsigned int *tlv, const char **slaves);
-void snd_hda_codec_reset(struct hda_codec *codec);
+int snd_hda_codec_reset(struct hda_codec *codec);
 int snd_hda_codec_configure(struct hda_codec *codec);
 
 /* amp value bits */
@@ -134,7 +136,7 @@ extern struct hda_ctl_ops snd_hda_bind_sw;	/* for bind-switch */
 
 struct hda_bind_ctls {
 	struct hda_ctl_ops *ops;
-	long values[];
+	unsigned long values[];
 };
 
 int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol,
@@ -227,6 +229,7 @@ struct hda_multi_out {
 	hda_nid_t hp_nid;	/* optional DAC for HP, 0 when not exists */
 	hda_nid_t extra_out_nid[3];	/* optional DACs, 0 when not exists */
 	hda_nid_t dig_out_nid;	/* digital out audio widget */
+	hda_nid_t *slave_dig_outs;
 	int max_channels;	/* currently supported analog channels */
 	int dig_out_used;	/* current usage of digital out (HDA_DIG_XXX) */
 	int no_share_stream;	/* don't share a stream with multiple pins */
@@ -354,9 +357,12 @@ struct auto_pin_cfg {
 	int line_out_type;	/* AUTO_PIN_XXX_OUT */
 	hda_nid_t hp_pins[AUTO_CFG_MAX_OUTS];
 	hda_nid_t input_pins[AUTO_PIN_LAST];
-	hda_nid_t dig_out_pin;
+	int dig_outs;
+	hda_nid_t dig_out_pins[2];
 	hda_nid_t dig_in_pin;
 	hda_nid_t mono_out_pin;
+	int dig_out_type[2]; /* HDA_PCM_TYPE_XXX */
+	int dig_in_type; /* HDA_PCM_TYPE_XXX */
 };
 
 #define get_defcfg_connect(cfg) \
@@ -405,6 +411,7 @@ static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid)
 u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction);
 int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir,
 			      unsigned int caps);
+u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid);
 
 int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl);
 void snd_hda_ctls_clear(struct hda_codec *codec);
@@ -427,6 +434,23 @@ static inline int snd_hda_hwdep_add_sysfs(struct hda_codec *codec)
 }
 #endif
 
+#ifdef CONFIG_SND_HDA_RECONFIG
+const char *snd_hda_get_hint(struct hda_codec *codec, const char *key);
+int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key);
+#else
+static inline
+const char *snd_hda_get_hint(struct hda_codec *codec, const char *key)
+{
+	return NULL;
+}
+
+static inline
+int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key)
+{
+	return -ENOENT;
+}
+#endif
+
 /*
  * power-management
  */
@@ -458,6 +482,7 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec,
 #define get_amp_channels(kc)	(((kc)->private_value >> 16) & 0x3)
 #define get_amp_direction(kc)	(((kc)->private_value >> 18) & 0x1)
 #define get_amp_index(kc)	(((kc)->private_value >> 19) & 0xf)
+#define get_amp_offset(kc)	(((kc)->private_value >> 23) & 0x3f)
 
 /*
  * CEA Short Audio Descriptor data
diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c
index 144b85276d5a..93d7499350c6 100644
--- a/sound/pci/hda/hda_proc.c
+++ b/sound/pci/hda/hda_proc.c
@@ -399,8 +399,10 @@ static void print_conn_list(struct snd_info_buffer *buffer,
 {
 	int c, curr = -1;
 
-	if (conn_len > 1 && wid_type != AC_WID_AUD_MIX &&
-	    wid_type != AC_WID_VOL_KNB)
+	if (conn_len > 1 &&
+	    wid_type != AC_WID_AUD_MIX &&
+	    wid_type != AC_WID_VOL_KNB &&
+	    wid_type != AC_WID_POWER)
 		curr = snd_hda_codec_read(codec, nid, 0,
 					  AC_VERB_GET_CONNECT_SEL, 0);
 	snd_iprintf(buffer, "  Connection: %d\n", conn_len);
@@ -467,8 +469,9 @@ static void print_codec_info(struct snd_info_entry *entry,
 	snd_iprintf(buffer, "Codec: %s\n",
 		    codec->name ? codec->name : "Not Set");
 	snd_iprintf(buffer, "Address: %d\n", codec->addr);
-	snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id);
-	snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id);
+	snd_iprintf(buffer, "Function Id: 0x%x\n", codec->function_id);
+	snd_iprintf(buffer, "Vendor Id: 0x%08x\n", codec->vendor_id);
+	snd_iprintf(buffer, "Subsystem Id: 0x%08x\n", codec->subsystem_id);
 	snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id);
 
 	if (codec->mfg)
@@ -554,8 +557,14 @@ static void print_codec_info(struct snd_info_entry *entry,
 			snd_iprintf(buffer, "  Amp-Out caps: ");
 			print_amp_caps(buffer, codec, nid, HDA_OUTPUT);
 			snd_iprintf(buffer, "  Amp-Out vals: ");
-			print_amp_vals(buffer, codec, nid, HDA_OUTPUT,
-				       wid_caps & AC_WCAP_STEREO, 1);
+			if (wid_type == AC_WID_PIN &&
+			    codec->pin_amp_workaround)
+				print_amp_vals(buffer, codec, nid, HDA_OUTPUT,
+					       wid_caps & AC_WCAP_STEREO,
+					       conn_len);
+			else
+				print_amp_vals(buffer, codec, nid, HDA_OUTPUT,
+					       wid_caps & AC_WCAP_STEREO, 1);
 		}
 
 		switch (wid_type) {
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index e48612323aa0..5bb48ee8b6c6 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -27,11 +27,12 @@
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
+#include "hda_beep.h"
 
 struct ad198x_spec {
 	struct snd_kcontrol_new *mixers[5];
 	int num_mixers;
-
+	unsigned int beep_amp;	/* beep amp value, set via set_beep_amp() */
 	const struct hda_verb *init_verbs[5];	/* initialization verbs
 						 * don't forget NULL termination!
 						 */
@@ -154,6 +155,16 @@ static const char *ad_slave_sws[] = {
 
 static void ad198x_free_kctls(struct hda_codec *codec);
 
+/* additional beep mixers; the actual parameters are overwritten at build */
+static struct snd_kcontrol_new ad_beep_mixer[] = {
+	HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_OUTPUT),
+	HDA_CODEC_MUTE("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 */
+
 static int ad198x_build_controls(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec = codec->spec;
@@ -181,6 +192,21 @@ static int ad198x_build_controls(struct hda_codec *codec)
 			return err;
 	}
 
+	/* create beep controls if needed */
+	if (spec->beep_amp) {
+		struct snd_kcontrol_new *knew;
+		for (knew = ad_beep_mixer; knew->name; knew++) {
+			struct snd_kcontrol *kctl;
+			kctl = snd_ctl_new1(knew, codec);
+			if (!kctl)
+				return -ENOMEM;
+			kctl->private_value = spec->beep_amp;
+			err = snd_hda_ctl_add(codec, kctl);
+			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];
@@ -406,7 +432,8 @@ static void ad198x_free(struct hda_codec *codec)
 		return;
 
 	ad198x_free_kctls(codec);
-	kfree(codec->spec);
+	kfree(spec);
+	snd_hda_detach_beep_device(codec);
 }
 
 static struct hda_codec_ops ad198x_patch_ops = {
@@ -545,8 +572,6 @@ static struct snd_kcontrol_new ad1986a_mixers[] = {
 	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", 0x0f, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x18, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x18, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
@@ -610,8 +635,7 @@ static struct snd_kcontrol_new ad1986a_laptop_mixers[] = {
 	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", 0x0f, 0x0, HDA_OUTPUT),
-	/* HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x18, 0x0, HDA_OUTPUT),
-	   HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x18, 0x0, HDA_OUTPUT),
+	/* 
 	   HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT),
 	   HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), */
 	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
@@ -809,8 +833,6 @@ static struct snd_kcontrol_new ad1986a_laptop_automute_mixers[] = {
 	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", 0x0f, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x18, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x18, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT),
 	{
@@ -1002,10 +1024,8 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD),
 	SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK),
 	SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP),
-	SND_PCI_QUIRK(0x144d, 0xc023, "Samsung X60", AD1986A_SAMSUNG),
-	SND_PCI_QUIRK(0x144d, 0xc024, "Samsung R65", AD1986A_SAMSUNG),
-	SND_PCI_QUIRK(0x144d, 0xc026, "Samsung X11", AD1986A_SAMSUNG),
 	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),
@@ -1027,15 +1047,14 @@ static struct hda_amp_list ad1986a_loopbacks[] = {
 
 static int is_jack_available(struct hda_codec *codec, hda_nid_t nid)
 {
-	unsigned int conf = snd_hda_codec_read(codec, nid, 0,
-					       AC_VERB_GET_CONFIG_DEFAULT, 0);
+	unsigned int conf = snd_hda_codec_get_pincfg(codec, nid);
 	return get_defcfg_connect(conf) != AC_JACK_PORT_NONE;
 }
 
 static int patch_ad1986a(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
-	int board_config;
+	int err, board_config;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -1043,6 +1062,13 @@ static int patch_ad1986a(struct hda_codec *codec)
 
 	codec->spec = 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;
@@ -1222,8 +1248,6 @@ static struct snd_kcontrol_new ad1983_mixers[] = {
 	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_MONO("PC Speaker Playback Volume", 0x10, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("PC Speaker Playback Switch", 0x10, 1, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x0c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT),
@@ -1294,6 +1318,7 @@ static struct hda_amp_list ad1983_loopbacks[] = {
 static int patch_ad1983(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
+	int err;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -1301,6 +1326,13 @@ static int patch_ad1983(struct hda_codec *codec)
 
 	codec->spec = 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;
@@ -1370,8 +1402,6 @@ static struct snd_kcontrol_new ad1981_mixers[] = {
 	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_MONO("PC Speaker Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE_MONO("PC Speaker Playback Switch", 0x0d, 1, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x08, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT),
@@ -1416,8 +1446,8 @@ static struct hda_verb ad1981_init_verbs[] = {
 	{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, 0xb000},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000},
+	{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},
@@ -1682,10 +1712,10 @@ static 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(0x103c, 0, "HP nx", AD1981_HP),
+	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(0x17aa, 0, "Lenovo Thinkpad", AD1981_THINKPAD),
+	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),
 	{}
@@ -1694,7 +1724,7 @@ static struct snd_pci_quirk ad1981_cfg_tbl[] = {
 static int patch_ad1981(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
-	int board_config;
+	int err, board_config;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -1702,6 +1732,13 @@ static int patch_ad1981(struct hda_codec *codec)
 
 	codec->spec = 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;
@@ -1988,9 +2025,6 @@ static struct snd_kcontrol_new ad1988_6stack_mixers2[] = {
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT),
 
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT),
-
 	HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
 
@@ -2034,9 +2068,6 @@ static struct snd_kcontrol_new ad1988_3stack_mixers2[] = {
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT),
 
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT),
-
 	HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
 
@@ -2066,9 +2097,6 @@ static struct snd_kcontrol_new ad1988_laptop_mixers[] = {
 	HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT),
 	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT),
 
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT),
-
 	HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT),
 
@@ -2297,10 +2325,6 @@ static struct hda_verb ad1988_capture_init_verbs[] = {
 	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
 	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
 	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* ADCs; muted */
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 
 	{ }
 };
@@ -2408,10 +2432,6 @@ static struct hda_verb ad1988_3stack_init_verbs[] = {
 	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
 	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
 	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* ADCs; muted */
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	/* Analog Mix output amp */
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
 	{ }
@@ -2483,10 +2503,6 @@ static struct hda_verb ad1988_laptop_init_verbs[] = {
 	{0x0c, AC_VERB_SET_CONNECT_SEL, 0x1},
 	{0x0d, AC_VERB_SET_CONNECT_SEL, 0x1},
 	{0x0e, AC_VERB_SET_CONNECT_SEL, 0x1},
-	/* ADCs; muted */
-	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	/* Analog Mix output amp */
 	{0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */
 	{ }
@@ -2890,7 +2906,7 @@ static int ad1988_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = AD1988_SPDIF_OUT;
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = AD1988_SPDIF_IN;
@@ -2940,7 +2956,7 @@ static struct snd_pci_quirk ad1988_cfg_tbl[] = {
 static int patch_ad1988(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
-	int board_config;
+	int err, board_config;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -2960,7 +2976,7 @@ static int patch_ad1988(struct hda_codec *codec)
 
 	if (board_config == AD1988_AUTO) {
 		/* automatic parse from the BIOS config */
-		int err = ad1988_parse_auto_config(codec);
+		err = ad1988_parse_auto_config(codec);
 		if (err < 0) {
 			ad198x_free(codec);
 			return err;
@@ -2970,6 +2986,13 @@ static int patch_ad1988(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x10);
+	if (err < 0) {
+		ad198x_free(codec);
+		return err;
+	}
+	set_beep_amp(spec, 0x10, 0, HDA_OUTPUT);
+
 	switch (board_config) {
 	case AD1988_6STACK:
 	case AD1988_6STACK_DIG:
@@ -3126,12 +3149,6 @@ static struct snd_kcontrol_new ad1884_base_mixers[] = {
 	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("PC Speaker Playback Volume", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Digital Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT),
-	*/
 	HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
@@ -3204,10 +3221,10 @@ static struct hda_verb ad1884_init_verbs[] = {
 	{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_OUT_MUTE},
+	{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_OUT_MUTE},
+	{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)},
@@ -3240,7 +3257,7 @@ static const char *ad1884_slave_vols[] = {
 	"CD Playback Volume",
 	"Internal Mic Playback Volume",
 	"Docking Mic Playback Volume"
-	"Beep Playback Volume",
+	/* "Beep Playback Volume", */
 	"IEC958 Playback Volume",
 	NULL
 };
@@ -3248,6 +3265,7 @@ static const char *ad1884_slave_vols[] = {
 static int patch_ad1884(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
+	int err;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -3255,6 +3273,13 @@ static int patch_ad1884(struct hda_codec *codec)
 
 	codec->spec = 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;
@@ -3321,8 +3346,6 @@ static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = {
 	HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Docking Mic Boost", 0x25, 0x0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, 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),
@@ -3358,7 +3381,7 @@ static struct hda_verb ad1984_thinkpad_init_verbs[] = {
 	{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_MUTE},
+	{0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
 	/* Analog mixer - docking mic; mute as default */
 	{0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
 	/* enable EAPD bit */
@@ -3379,10 +3402,6 @@ static struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = {
 	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("PC Speaker Playback Volume", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x20, 0x03, HDA_INPUT),
-	*/
 	HDA_CODEC_VOLUME("Line-In Boost", 0x15, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
@@ -3468,7 +3487,7 @@ static const char *ad1984_models[AD1984_MODELS] = {
 
 static struct snd_pci_quirk ad1984_cfg_tbl[] = {
 	/* Lenovo Thinkpad T61/X61 */
-	SND_PCI_QUIRK(0x17aa, 0, "Lenovo Thinkpad", AD1984_THINKPAD),
+	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1984_THINKPAD),
 	SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP),
 	{}
 };
@@ -3561,8 +3580,6 @@ static struct snd_kcontrol_new ad1884a_base_mixers[] = {
 	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("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Line Boost", 0x15, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT),
@@ -3622,10 +3639,10 @@ static struct hda_verb ad1884a_init_verbs[] = {
 	{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},
+	{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_OUT_MUTE},
+	{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},
@@ -3695,8 +3712,6 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = {
 	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("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Dock Mic Boost", 0x25, 0x0, HDA_OUTPUT),
@@ -3724,8 +3739,6 @@ static struct snd_kcontrol_new ad1884a_mobile_mixers[] = {
 	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("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, 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),
@@ -3836,8 +3849,6 @@ static struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = {
 	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("Beep Playback Volume", 0x20, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT),
@@ -3911,9 +3922,9 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
 	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(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP),
-	SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP),
-	SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP),
+	SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE),
+	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(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD),
 	{}
 };
@@ -3921,7 +3932,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = {
 static int patch_ad1884a(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
-	int board_config;
+	int err, board_config;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -3929,6 +3940,13 @@ static int patch_ad1884a(struct hda_codec *codec)
 
 	codec->spec = 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;
@@ -3966,6 +3984,14 @@ static int patch_ad1884a(struct hda_codec *codec)
 		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;
@@ -4083,8 +4109,6 @@ static struct snd_kcontrol_new ad1882_loopback_mixers[] = {
 	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -4097,8 +4121,6 @@ static struct snd_kcontrol_new ad1882a_loopback_mixers[] = {
 	HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT),
 	HDA_CODEC_VOLUME("Digital Mic Boost", 0x1f, 0x0, HDA_INPUT),
 	{ } /* end */
 };
@@ -4257,7 +4279,7 @@ static const char *ad1882_models[AD1986A_MODELS] = {
 static int patch_ad1882(struct hda_codec *codec)
 {
 	struct ad198x_spec *spec;
-	int board_config;
+	int err, board_config;
 
 	spec = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -4265,6 +4287,13 @@ static int patch_ad1882(struct hda_codec *codec)
 
 	codec->spec = 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;
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index f3ebe837f2d5..c921264bbd71 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -680,13 +680,13 @@ static int patch_cmi9880(struct hda_codec *codec)
 		struct auto_pin_cfg cfg;
 
 		/* collect pin default configuration */
-		port_e = snd_hda_codec_read(codec, 0x0f, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
-		port_f = snd_hda_codec_read(codec, 0x10, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+		port_e = snd_hda_codec_get_pincfg(codec, 0x0f);
+		port_f = snd_hda_codec_get_pincfg(codec, 0x10);
 		spec->front_panel = 1;
 		if (get_defcfg_connect(port_e) == AC_JACK_PORT_NONE ||
 		    get_defcfg_connect(port_f) == AC_JACK_PORT_NONE) {
-			port_g = snd_hda_codec_read(codec, 0x1f, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
-			port_h = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+			port_g = snd_hda_codec_get_pincfg(codec, 0x1f);
+			port_h = snd_hda_codec_get_pincfg(codec, 0x20);
 			spec->channel_modes = cmi9880_channel_modes;
 			/* no front panel */
 			if (get_defcfg_connect(port_g) == AC_JACK_PORT_NONE ||
@@ -703,8 +703,8 @@ static int patch_cmi9880(struct hda_codec *codec)
 			spec->multiout.max_channels = cmi9880_channel_modes[0].channels;
 		} else {
 			spec->input_mux = &cmi9880_basic_mux;
-			port_spdifi = snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
-			port_spdifo = snd_hda_codec_read(codec, 0x12, 0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+			port_spdifi = snd_hda_codec_get_pincfg(codec, 0x13);
+			port_spdifo = snd_hda_codec_get_pincfg(codec, 0x12);
 			if (get_defcfg_connect(port_spdifo) != AC_JACK_PORT_NONE)
 				spec->multiout.dig_out_nid = CMI_DIG_OUT_NID;
 			if (get_defcfg_connect(port_spdifi) != AC_JACK_PORT_NONE)
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 0177ef8f4c9e..1f2ad76ca94b 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -58,6 +58,7 @@ struct conexant_spec {
 
 	struct snd_kcontrol_new *mixers[5];
 	int num_mixers;
+	hda_nid_t vmaster_nid;
 
 	const struct hda_verb *init_verbs[5];	/* initialization verbs
 						 * don't forget NULL
@@ -72,6 +73,7 @@ struct conexant_spec {
 					 */
 	unsigned int cur_eapd;
 	unsigned int hp_present;
+	unsigned int no_auto_mic;
 	unsigned int need_dac_fix;
 
 	/* capture */
@@ -461,6 +463,29 @@ static void conexant_free(struct hda_codec *codec)
 	kfree(codec->spec);
 }
 
+static struct snd_kcontrol_new cxt_capture_mixers[] = {
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Capture Source",
+		.info = conexant_mux_enum_info,
+		.get = conexant_mux_enum_get,
+		.put = conexant_mux_enum_put
+	},
+	{}
+};
+
+static const char *slave_vols[] = {
+	"Headphone Playback Volume",
+	"Speaker Playback Volume",
+	NULL
+};
+
+static const char *slave_sws[] = {
+	"Headphone Playback Switch",
+	"Speaker Playback Switch",
+	NULL
+};
+
 static int conexant_build_controls(struct hda_codec *codec)
 {
 	struct conexant_spec *spec = codec->spec;
@@ -488,6 +513,32 @@ static int conexant_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 	}
+
+	/* if we have no master control, let's create it */
+	if (spec->vmaster_nid &&
+	    !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, slave_vols);
+		if (err < 0)
+			return err;
+	}
+	if (spec->vmaster_nid &&
+	    !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
+		err = snd_hda_add_vmaster(codec, "Master Playback Switch",
+					  NULL, slave_sws);
+		if (err < 0)
+			return err;
+	}
+
+	if (spec->input_mux) {
+		err = snd_hda_add_new_ctls(codec, cxt_capture_mixers);
+		if (err < 0)
+			return err;
+	}
+
 	return 0;
 }
 
@@ -719,13 +770,6 @@ static void cxt5045_hp_unsol_event(struct hda_codec *codec,
 }
 
 static struct snd_kcontrol_new cxt5045_mixers[] = {
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = conexant_mux_enum_info,
-		.get = conexant_mux_enum_get,
-		.put = conexant_mux_enum_put
-	},
 	HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x01, HDA_INPUT),
 	HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x02, HDA_INPUT),
@@ -759,13 +803,6 @@ static struct snd_kcontrol_new cxt5045_benq_mixers[] = {
 };
 
 static struct snd_kcontrol_new cxt5045_mixers_hp530[] = {
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = conexant_mux_enum_info,
-		.get = conexant_mux_enum_get,
-		.put = conexant_mux_enum_put
-	},
 	HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x02, HDA_INPUT),
 	HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x01, HDA_INPUT),
@@ -1002,15 +1039,9 @@ static const char *cxt5045_models[CXT5045_MODELS] = {
 };
 
 static struct snd_pci_quirk cxt5045_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x103c, 0x30a5, "HP", CXT5045_LAPTOP_HPSENSE),
-	SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV Series", CXT5045_LAPTOP_HPSENSE),
-	SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2120", CXT5045_LAPTOP_HPSENSE),
-	SND_PCI_QUIRK(0x103c, 0x30b7, "HP DV6000Z", CXT5045_LAPTOP_HPSENSE),
-	SND_PCI_QUIRK(0x103c, 0x30bb, "HP DV8000", CXT5045_LAPTOP_HPSENSE),
-	SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP_HPSENSE),
-	SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV9533EG", CXT5045_LAPTOP_HPSENSE),
 	SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HP530),
-	SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP_HPSENSE),
+	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series",
+			   CXT5045_LAPTOP_HPSENSE),
 	SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P105", CXT5045_LAPTOP_MICSENSE),
 	SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ),
 	SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE),
@@ -1020,8 +1051,8 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1509, 0x1e40, "FIC", CXT5045_LAPTOP_HPMICSENSE),
 	SND_PCI_QUIRK(0x1509, 0x2f05, "FIC", CXT5045_LAPTOP_HPMICSENSE),
 	SND_PCI_QUIRK(0x1509, 0x2f06, "FIC", CXT5045_LAPTOP_HPMICSENSE),
-	SND_PCI_QUIRK(0x1631, 0xc106, "Packard Bell", CXT5045_LAPTOP_HPMICSENSE),
-	SND_PCI_QUIRK(0x1631, 0xc107, "Packard Bell", CXT5045_LAPTOP_HPMICSENSE),
+	SND_PCI_QUIRK_MASK(0x1631, 0xff00, 0xc100, "Packard Bell",
+			   CXT5045_LAPTOP_HPMICSENSE),
 	SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP_HPSENSE),
 	{}
 };
@@ -1035,6 +1066,7 @@ static int patch_cxt5045(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 	codec->spec = spec;
+	codec->pin_amp_workaround = 1;
 
 	spec->multiout.max_channels = 2;
 	spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids);
@@ -1134,7 +1166,7 @@ static int patch_cxt5045(struct hda_codec *codec)
 /* Conexant 5047 specific */
 #define CXT5047_SPDIF_OUT	0x11
 
-static hda_nid_t cxt5047_dac_nids[2] = { 0x10, 0x1c };
+static hda_nid_t cxt5047_dac_nids[1] = { 0x10 }; /* 0x1c */
 static hda_nid_t cxt5047_adc_nids[1] = { 0x12 };
 static hda_nid_t cxt5047_capsrc_nids[1] = { 0x1a };
 
@@ -1142,20 +1174,6 @@ static struct hda_channel_mode cxt5047_modes[1] = {
 	{ 2, NULL },
 };
 
-static struct hda_input_mux cxt5047_capture_source = {
-	.num_items = 1,
-	.items = {
-		{ "Mic", 0x2 },
-	}
-};
-
-static struct hda_input_mux cxt5047_hp_capture_source = {
-	.num_items = 1,
-	.items = {
-		{ "ExtMic", 0x2 },
-	}
-};
-
 static struct hda_input_mux cxt5047_toshiba_capture_source = {
 	.num_items = 2,
 	.items = {
@@ -1179,7 +1197,11 @@ static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol,
 	 * the headphone jack
 	 */
 	bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE;
-	snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0,
+	/* NOTE: Conexat codec needs the index for *OUTPUT* amp of
+	 * pin widgets unlike other codecs.  In this case, we need to
+	 * set index 0x01 for the volume from the mixer amp 0x19.
+	 */
+	snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01,
 				 HDA_AMP_MUTE, bits);
 	bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE;
 	snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0,
@@ -1187,16 +1209,6 @@ static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
-/* bind volumes of both NID 0x13 (Headphones) and 0x1d (Speakers) */
-static struct hda_bind_ctls cxt5047_bind_master_vol = {
-	.ops = &snd_hda_bind_vol,
-	.values = {
-		HDA_COMPOSE_AMP_VAL(0x13, 3, 0, HDA_OUTPUT),
-		HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT),
-		0
-	},
-};
-
 /* mute internal speaker if HP is plugged */
 static void cxt5047_hp_automute(struct hda_codec *codec)
 {
@@ -1207,27 +1219,8 @@ static void cxt5047_hp_automute(struct hda_codec *codec)
 				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
 
 	bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0;
-	snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, bits);
-	/* Mute/Unmute PCM 2 for good measure - some systems need this */
-	snd_hda_codec_amp_stereo(codec, 0x1c, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, bits);
-}
-
-/* mute internal speaker if HP is plugged */
-static void cxt5047_hp2_automute(struct hda_codec *codec)
-{
-	struct conexant_spec *spec = codec->spec;
-	unsigned int bits;
-
-	spec->hp_present = snd_hda_codec_read(codec, 0x13, 0,
-				     AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
-
-	bits = spec->hp_present ? HDA_AMP_MUTE : 0;
-	snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0,
-				 HDA_AMP_MUTE, bits);
-	/* Mute/Unmute PCM 2 for good measure - some systems need this */
-	snd_hda_codec_amp_stereo(codec, 0x1c, HDA_OUTPUT, 0,
+	/* See the note in cxt5047_hp_master_sw_put */
+	snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01,
 				 HDA_AMP_MUTE, bits);
 }
 
@@ -1268,55 +1261,14 @@ static void cxt5047_hp_unsol_event(struct hda_codec *codec,
 	}
 }
 
-/* unsolicited event for HP jack sensing - non-EAPD systems */
-static void cxt5047_hp2_unsol_event(struct hda_codec *codec,
-				  unsigned int res)
-{
-	res >>= 26;
-	switch (res) {
-	case CONEXANT_HP_EVENT:
-		cxt5047_hp2_automute(codec);
-		break;
-	case CONEXANT_MIC_EVENT:
-		cxt5047_hp_automic(codec);
-		break;
-	}
-}
-
-static struct snd_kcontrol_new cxt5047_mixers[] = {
-	HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Gain Volume", 0x1a, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Mic Gain Switch", 0x1a, 0x0, HDA_OUTPUT),
+static struct snd_kcontrol_new cxt5047_base_mixers[] = {
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x19, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x19, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost", 0x1a, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT),
 	HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT),
 	HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("PCM-2 Volume", 0x1c, 0x00, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM-2 Switch", 0x1c, 0x00, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x1d, 0x00, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x13, 0x00, HDA_OUTPUT),
-
-	{}
-};
-
-static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = {
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = conexant_mux_enum_info,
-		.get = conexant_mux_enum_get,
-		.put = conexant_mux_enum_put
-	},
-	HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT),
-	HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT),
-	HDA_BIND_VOL("Master Playback Volume", &cxt5047_bind_master_vol),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Master Playback Switch",
@@ -1329,29 +1281,15 @@ static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = {
 	{}
 };
 
-static struct snd_kcontrol_new cxt5047_hp_mixers[] = {
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.info = conexant_mux_enum_info,
-		.get = conexant_mux_enum_get,
-		.put = conexant_mux_enum_put
-	},
-	HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19,0x02,HDA_INPUT),
-	HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT),
-	HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT),
+static struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = {
+	/* See the note in cxt5047_hp_master_sw_put */
+	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x01, HDA_OUTPUT),
+	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT),
+	{}
+};
+
+static struct snd_kcontrol_new cxt5047_hp_only_mixers[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x13, 0x00, HDA_OUTPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Master Playback Switch",
-		.info = cxt_eapd_info,
-		.get = cxt_eapd_get,
-		.put = cxt5047_hp_master_sw_put,
-		.private_value = 0x13,
-	},
 	{ } /* end */
 };
 
@@ -1362,8 +1300,8 @@ static struct hda_verb cxt5047_init_verbs[] = {
 	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_50 },
 	/* HP, Speaker  */
 	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP },
-	{0x13, AC_VERB_SET_CONNECT_SEL,0x1},
-	{0x1d, AC_VERB_SET_CONNECT_SEL,0x0},
+	{0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, /* mixer(0x19) */
+	{0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mixer(0x19) */
 	/* Record selector: Mic */
 	{0x12, AC_VERB_SET_CONNECT_SEL,0x03},
 	{0x19, AC_VERB_SET_AMP_GAIN_MUTE,
@@ -1383,30 +1321,7 @@ static struct hda_verb cxt5047_init_verbs[] = {
 
 /* configuration for Toshiba Laptops */
 static struct hda_verb cxt5047_toshiba_init_verbs[] = {
-	{0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0 }, /* default on */
-	/* pin sensing on HP and Mic jacks */
-	{0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
-	{0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT},
-	/* Speaker routing */
-	{0x1d, AC_VERB_SET_CONNECT_SEL,0x1},
-	{}
-};
-
-/* configuration for HP Laptops */
-static struct hda_verb cxt5047_hp_init_verbs[] = {
-	/* pin sensing on HP jack */
-	{0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT},
-	/* 0x13 is actually shared by both HP and speaker;
-	 * setting the connection to 0 (=0x19) makes the master volume control
-	 * working mysteriouslly...
-	 */
-	{0x13, AC_VERB_SET_CONNECT_SEL, 0x0},
-	/* Record selector: Ext Mic */
-	{0x12, AC_VERB_SET_CONNECT_SEL,0x03},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE,
-	 AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17},
-	/* Speaker routing */
-	{0x1d, AC_VERB_SET_CONNECT_SEL,0x1},
+	{0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0}, /* default off */
 	{}
 };
 
@@ -1571,11 +1486,9 @@ static const char *cxt5047_models[CXT5047_MODELS] = {
 };
 
 static struct snd_pci_quirk cxt5047_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x103c, 0x30a0, "HP DV1000", CXT5047_LAPTOP),
 	SND_PCI_QUIRK(0x103c, 0x30a5, "HP DV5200T/DV8000T", CXT5047_LAPTOP_HP),
-	SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV2000T/DV3000T", CXT5047_LAPTOP),
-	SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2000Z", CXT5047_LAPTOP),
-	SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6700", CXT5047_LAPTOP),
+	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series",
+			   CXT5047_LAPTOP),
 	SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P100", CXT5047_LAPTOP_EAPD),
 	{}
 };
@@ -1589,6 +1502,7 @@ static int patch_cxt5047(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 	codec->spec = spec;
+	codec->pin_amp_workaround = 1;
 
 	spec->multiout.max_channels = 2;
 	spec->multiout.num_dacs = ARRAY_SIZE(cxt5047_dac_nids);
@@ -1597,9 +1511,8 @@ static int patch_cxt5047(struct hda_codec *codec)
 	spec->num_adc_nids = 1;
 	spec->adc_nids = cxt5047_adc_nids;
 	spec->capsrc_nids = cxt5047_capsrc_nids;
-	spec->input_mux = &cxt5047_capture_source;
 	spec->num_mixers = 1;
-	spec->mixers[0] = cxt5047_mixers;
+	spec->mixers[0] = cxt5047_base_mixers;
 	spec->num_init_verbs = 1;
 	spec->init_verbs[0] = cxt5047_init_verbs;
 	spec->spdif_route = 0;
@@ -1613,21 +1526,22 @@ static int patch_cxt5047(struct hda_codec *codec)
 						  cxt5047_cfg_tbl);
 	switch (board_config) {
 	case CXT5047_LAPTOP:
-		codec->patch_ops.unsol_event = cxt5047_hp2_unsol_event;
+		spec->num_mixers = 2;
+		spec->mixers[1] = cxt5047_hp_spk_mixers;
+		codec->patch_ops.unsol_event = cxt5047_hp_unsol_event;
 		break;
 	case CXT5047_LAPTOP_HP:
-		spec->input_mux = &cxt5047_hp_capture_source;
-		spec->num_init_verbs = 2;
-		spec->init_verbs[1] = cxt5047_hp_init_verbs;
-		spec->mixers[0] = cxt5047_hp_mixers;
+		spec->num_mixers = 2;
+		spec->mixers[1] = cxt5047_hp_only_mixers;
 		codec->patch_ops.unsol_event = cxt5047_hp_unsol_event;
 		codec->patch_ops.init = cxt5047_hp_init;
 		break;
 	case CXT5047_LAPTOP_EAPD:
 		spec->input_mux = &cxt5047_toshiba_capture_source;
+		spec->num_mixers = 2;
+		spec->mixers[1] = cxt5047_hp_spk_mixers;
 		spec->num_init_verbs = 2;
 		spec->init_verbs[1] = cxt5047_toshiba_init_verbs;
-		spec->mixers[0] = cxt5047_toshiba_mixers;
 		codec->patch_ops.unsol_event = cxt5047_hp_unsol_event;
 		break;
 #ifdef CONFIG_SND_DEBUG
@@ -1638,6 +1552,7 @@ static int patch_cxt5047(struct hda_codec *codec)
 		codec->patch_ops.unsol_event = cxt5047_hp_unsol_event;
 #endif	
 	}
+	spec->vmaster_nid = 0x13;
 	return 0;
 }
 
@@ -1673,8 +1588,11 @@ static int cxt5051_hp_master_sw_put(struct snd_kcontrol *kcontrol,
 /* toggle input of built-in and mic jack appropriately */
 static void cxt5051_portb_automic(struct hda_codec *codec)
 {
+	struct conexant_spec *spec = codec->spec;
 	unsigned int present;
 
+	if (spec->no_auto_mic)
+		return;
 	present = snd_hda_codec_read(codec, 0x17, 0,
 				     AC_VERB_GET_PIN_SENSE, 0) &
 		AC_PINSENSE_PRESENCE;
@@ -1690,6 +1608,8 @@ static void cxt5051_portc_automic(struct hda_codec *codec)
 	unsigned int present;
 	hda_nid_t new_adc;
 
+	if (spec->no_auto_mic)
+		return;
 	present = snd_hda_codec_read(codec, 0x18, 0,
 				     AC_VERB_GET_PIN_SENSE, 0) &
 		AC_PINSENSE_PRESENCE;
@@ -1776,6 +1696,22 @@ static struct snd_kcontrol_new cxt5051_hp_mixers[] = {
 	{}
 };
 
+static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = {
+	HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x00, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Switch", 0x14, 0x00, HDA_INPUT),
+	HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT),
+	{
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = "Master Playback Switch",
+		.info = cxt_eapd_info,
+		.get = cxt_eapd_get,
+		.put = cxt5051_hp_master_sw_put,
+		.private_value = 0x1a,
+	},
+
+	{}
+};
+
 static struct hda_verb cxt5051_init_verbs[] = {
 	/* Line in, Mic */
 	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
@@ -1806,6 +1742,66 @@ static struct hda_verb cxt5051_init_verbs[] = {
 	{ } /* end */
 };
 
+static struct hda_verb cxt5051_hp_dv6736_init_verbs[] = {
+	/* Line in, Mic */
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0},
+	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0},
+	/* SPK  */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* HP, Amp  */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x16, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* DAC1 */
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Record selector: Int mic */
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44},
+	{0x14, AC_VERB_SET_CONNECT_SEL, 0x1},
+	/* SPDIF route: PCM */
+	{0x1c, AC_VERB_SET_CONNECT_SEL, 0x0},
+	/* EAPD */
+	{0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
+	{0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
+	{0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT},
+	{ } /* end */
+};
+
+static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = {
+	/* Line in, Mic */
+	{0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
+	{0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
+	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03},
+	/* SPK  */
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* HP, Amp  */
+	{0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x16, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* Docking HP */
+	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x19, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* DAC1 */
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Record selector: Int mic */
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44},
+	/* SPDIF route: PCM */
+	{0x1c, AC_VERB_SET_CONNECT_SEL, 0x0},
+	/* EAPD */
+	{0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */
+	{0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
+	{0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT},
+	{0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTC_EVENT},
+	{0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT},
+	{ } /* end */
+};
+
 /* initialize jack-sensing, too */
 static int cxt5051_init(struct hda_codec *codec)
 {
@@ -1823,18 +1819,24 @@ static int cxt5051_init(struct hda_codec *codec)
 enum {
 	CXT5051_LAPTOP,	 /* Laptops w/ EAPD support */
 	CXT5051_HP,	/* no docking */
+	CXT5051_HP_DV6736,	/* HP without mic switch */
+	CXT5051_LENOVO_X200,	/* Lenovo X200 laptop */
 	CXT5051_MODELS
 };
 
 static const char *cxt5051_models[CXT5051_MODELS] = {
 	[CXT5051_LAPTOP]	= "laptop",
 	[CXT5051_HP]		= "hp",
+	[CXT5051_HP_DV6736]	= "hp-dv6736",
+	[CXT5051_LENOVO_X200]	= "lenovo-x200",
 };
 
 static struct snd_pci_quirk cxt5051_cfg_tbl[] = {
+	SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736),
 	SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board",
 		      CXT5051_LAPTOP),
 	SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP),
+	SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200),
 	{}
 };
 
@@ -1847,6 +1849,7 @@ static int patch_cxt5051(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 	codec->spec = spec;
+	codec->pin_amp_workaround = 1;
 
 	codec->patch_ops = conexant_patch_ops;
 	codec->patch_ops.init = cxt5051_init;
@@ -1867,17 +1870,22 @@ static int patch_cxt5051(struct hda_codec *codec)
 	spec->cur_adc = 0;
 	spec->cur_adc_idx = 0;
 
+	codec->patch_ops.unsol_event = cxt5051_hp_unsol_event;
+
 	board_config = snd_hda_check_board_config(codec, CXT5051_MODELS,
 						  cxt5051_models,
 						  cxt5051_cfg_tbl);
 	switch (board_config) {
 	case CXT5051_HP:
-		codec->patch_ops.unsol_event = cxt5051_hp_unsol_event;
 		spec->mixers[0] = cxt5051_hp_mixers;
 		break;
-	default:
-	case CXT5051_LAPTOP:
-		codec->patch_ops.unsol_event = cxt5051_hp_unsol_event;
+	case CXT5051_HP_DV6736:
+		spec->init_verbs[0] = cxt5051_hp_dv6736_init_verbs;
+		spec->mixers[0] = cxt5051_hp_dv6736_mixers;
+		spec->no_auto_mic = 1;
+		break;
+	case CXT5051_LENOVO_X200:
+		spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs;
 		break;
 	}
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 6c26afcb8262..82097790f6f3 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -30,6 +30,7 @@
 #include <sound/core.h>
 #include "hda_codec.h"
 #include "hda_local.h"
+#include "hda_beep.h"
 
 #define ALC880_FRONT_EVENT		0x01
 #define ALC880_DCVOL_EVENT		0x02
@@ -77,6 +78,7 @@ enum {
 	ALC260_ACER,
 	ALC260_WILL,
 	ALC260_REPLACER_672V,
+	ALC260_FAVORIT100,
 #ifdef CONFIG_SND_DEBUG
 	ALC260_TEST,
 #endif
@@ -103,6 +105,7 @@ enum {
 	ALC262_NEC,
 	ALC262_TOSHIBA_S06,
 	ALC262_TOSHIBA_RX1,
+	ALC262_TYAN,
 	ALC262_AUTO,
 	ALC262_MODEL_LAST /* last tag */
 };
@@ -238,6 +241,13 @@ enum {
 	ALC883_MODEL_LAST,
 };
 
+/* styles of capture selection */
+enum {
+	CAPT_MUX = 0,	/* only mux based */
+	CAPT_MIX,	/* only mixer based */
+	CAPT_1MUX_MIX,	/* first mux and other mixers */
+};
+
 /* for GPIO Poll */
 #define GPIO_MASK	0x03
 
@@ -246,6 +256,7 @@ struct alc_spec {
 	struct snd_kcontrol_new *mixers[5];	/* mixer arrays */
 	unsigned int num_mixers;
 	struct snd_kcontrol_new *cap_mixer;	/* capture mixer */
+	unsigned int beep_amp;	/* beep amp value, set via set_beep_amp() */
 
 	const struct hda_verb *init_verbs[5];	/* initialization verbs
 						 * don't forget NULL
@@ -269,13 +280,15 @@ struct alc_spec {
 					 * dig_out_nid and hp_nid are optional
 					 */
 	hda_nid_t alt_dac_nid;
+	hda_nid_t slave_dig_outs[3];	/* optional - for auto-parsing */
+	int dig_out_type;
 
 	/* capture */
 	unsigned int num_adc_nids;
 	hda_nid_t *adc_nids;
 	hda_nid_t *capsrc_nids;
 	hda_nid_t dig_in_nid;		/* digital-in NID; optional */
-	unsigned char is_mix_capture;	/* matrix-style capture (non-mux) */
+	int capture_style;		/* capture style (CAPT_*) */
 
 	/* capture source */
 	unsigned int num_mux_defs;
@@ -293,7 +306,7 @@ struct alc_spec {
 	/* dynamic controls, init_verbs and input_mux */
 	struct auto_pin_cfg autocfg;
 	struct snd_array kctls;
-	struct hda_input_mux private_imux;
+	struct hda_input_mux private_imux[3];
 	hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS];
 
 	/* hooks */
@@ -305,6 +318,9 @@ struct alc_spec {
 	unsigned int jack_present: 1;
 	unsigned int master_sw: 1;
 
+	/* other flags */
+	unsigned int no_analog :1; /* digital I/O only */
+
 	/* for virtual master */
 	hda_nid_t vmaster_nid;
 #ifdef CONFIG_SND_HDA_POWER_SAVE
@@ -314,13 +330,6 @@ struct alc_spec {
 	/* for PLL fix */
 	hda_nid_t pll_nid;
 	unsigned int pll_coef_idx, pll_coef_bit;
-
-#ifdef SND_HDA_NEEDS_RESUME
-#define ALC_MAX_PINS	16
-	unsigned int num_pins;
-	hda_nid_t pin_nids[ALC_MAX_PINS];
-	unsigned int pin_cfgs[ALC_MAX_PINS];
-#endif
 };
 
 /*
@@ -336,6 +345,7 @@ struct alc_config_preset {
 	hda_nid_t *dac_nids;
 	hda_nid_t dig_out_nid;		/* optional */
 	hda_nid_t hp_nid;		/* optional */
+	hda_nid_t *slave_dig_outs;
 	unsigned int num_adc_nids;
 	hda_nid_t *adc_nids;
 	hda_nid_t *capsrc_nids;
@@ -392,7 +402,8 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol,
 	mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx;
 	imux = &spec->input_mux[mux_idx];
 
-	if (spec->is_mix_capture) {
+	if (spec->capture_style &&
+	    !(spec->capture_style == CAPT_1MUX_MIX && !adc_idx)) {
 		/* Matrix-mixer style (e.g. ALC882) */
 		unsigned int *cur_val = &spec->cur_mux[adc_idx];
 		unsigned int i, idx;
@@ -750,6 +761,24 @@ static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol,
 #endif   /* CONFIG_SND_DEBUG */
 
 /*
+ * set up the input pin config (depending on the given auto-pin type)
+ */
+static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid,
+			      int auto_pin_type)
+{
+	unsigned int val = PIN_IN;
+
+	if (auto_pin_type <= AUTO_PIN_FRONT_MIC) {
+		unsigned int pincap;
+		pincap = snd_hda_query_pin_caps(codec, nid);
+		pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
+		if (pincap & AC_PINCAP_VREF_80)
+			val = PIN_VREF80;
+	}
+	snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val);
+}
+
+/*
  */
 static void add_mixer(struct alc_spec *spec, struct snd_kcontrol_new *mix)
 {
@@ -810,6 +839,7 @@ static void setup_preset(struct alc_spec *spec,
 	spec->multiout.num_dacs = preset->num_dacs;
 	spec->multiout.dac_nids = preset->dac_nids;
 	spec->multiout.dig_out_nid = preset->dig_out_nid;
+	spec->multiout.slave_dig_outs = preset->slave_dig_outs;
 	spec->multiout.hp_nid = preset->hp_nid;
 
 	spec->num_mux_defs = preset->num_mux_defs;
@@ -921,7 +951,7 @@ static void alc_mic_automute(struct hda_codec *codec)
 			 HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0);
 }
 #else
-#define alc_mic_automute(codec) /* NOP */
+#define alc_mic_automute(codec) do {} while(0) /* NOP */
 #endif /* disabled */
 
 /* unsolicited event for HP jack sensing */
@@ -952,7 +982,7 @@ static void alc888_coef_init(struct hda_codec *codec)
 	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0);
 	tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0);
 	snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7);
-	if ((tmp & 0xf0) == 2)
+	if ((tmp & 0xf0) == 0x20)
 		/* alc888S-VC */
 		snd_hda_codec_read(codec, 0x20, 0,
 				   AC_VERB_SET_PROC_COEF, 0x830);
@@ -991,8 +1021,7 @@ static void alc_subsystem_id(struct hda_codec *codec,
 	nid = 0x1d;
 	if (codec->vendor_id == 0x10ec0260)
 		nid = 0x17;
-	ass = snd_hda_codec_read(codec, nid, 0,
-				 AC_VERB_GET_CONFIG_DEFAULT, 0);
+	ass = snd_hda_codec_get_pincfg(codec, nid);
 	if (!(ass & 1) && !(ass & 0x100000))
 		return;
 	if ((ass >> 30) != 1)	/* no physical connection */
@@ -1166,16 +1195,8 @@ static void alc_fix_pincfg(struct hda_codec *codec,
 		return;
 
 	cfg = pinfix[quirk->value];
-	for (; cfg->nid; cfg++) {
-		int i;
-		u32 val = cfg->val;
-		for (i = 0; i < 4; i++) {
-			snd_hda_codec_write(codec, cfg->nid, 0,
-				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 + i,
-				    val & 0xff);
-			val >>= 8;
-		}
-	}
+	for (; cfg->nid; cfg++)
+		snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val);
 }
 
 /*
@@ -1375,8 +1396,6 @@ static struct snd_kcontrol_new alc888_base_mixer[] = {
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -1483,8 +1502,6 @@ static struct snd_kcontrol_new alc880_three_stack_mixer[] = {
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -1578,8 +1595,7 @@ static int alc_cap_sw_put(struct snd_kcontrol *kcontrol,
 				     snd_hda_mixer_amp_switch_put);
 }
 
-#define DEFINE_CAPMIX(num) \
-static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \
+#define _DEFINE_CAPMIX(num) \
 	{ \
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
 		.name = "Capture Switch", \
@@ -1600,7 +1616,9 @@ static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \
 		.get = alc_cap_vol_get, \
 		.put = alc_cap_vol_put, \
 		.tlv = { .c = alc_cap_vol_tlv }, \
-	}, \
+	}
+
+#define _DEFINE_CAPSRC(num) \
 	{ \
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
 		/* .name = "Capture Source", */ \
@@ -1609,15 +1627,28 @@ static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \
 		.info = alc_mux_enum_info, \
 		.get = alc_mux_enum_get, \
 		.put = alc_mux_enum_put, \
-	}, \
-	{ } /* end */ \
+	}
+
+#define DEFINE_CAPMIX(num) \
+static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \
+	_DEFINE_CAPMIX(num),				      \
+	_DEFINE_CAPSRC(num),				      \
+	{ } /* end */					      \
+}
+
+#define DEFINE_CAPMIX_NOSRC(num) \
+static struct snd_kcontrol_new alc_capture_mixer_nosrc ## num[] = { \
+	_DEFINE_CAPMIX(num),					    \
+	{ } /* end */						    \
 }
 
 /* up to three ADCs */
 DEFINE_CAPMIX(1);
 DEFINE_CAPMIX(2);
 DEFINE_CAPMIX(3);
-
+DEFINE_CAPMIX_NOSRC(1);
+DEFINE_CAPMIX_NOSRC(2);
+DEFINE_CAPMIX_NOSRC(3);
 
 /*
  * ALC880 5-stack model
@@ -1706,8 +1737,6 @@ static struct snd_kcontrol_new alc880_six_stack_mixer[] = {
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Channel Mode",
@@ -1884,13 +1913,6 @@ static struct snd_kcontrol_new alc880_asus_w1v_mixer[] = {
 	{ } /* end */
 };
 
-/* additional mixers to alc880_asus_mixer */
-static struct snd_kcontrol_new alc880_pcbeep_mixer[] = {
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
-	{ } /* end */
-};
-
 /* TCL S700 */
 static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = {
 	HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
@@ -1923,8 +1945,6 @@ static struct snd_kcontrol_new alc880_uniwill_mixer[] = {
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 		.name = "Channel Mode",
@@ -1999,6 +2019,13 @@ static const char *alc_slave_sws[] = {
 
 static void alc_free_kctls(struct hda_codec *codec);
 
+/* additional beep mixers; the actual parameters are overwritten at build */
+static struct snd_kcontrol_new alc_beep_mixer[] = {
+	HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT),
+	HDA_CODEC_MUTE("Beep Playback Switch", 0, 0, HDA_INPUT),
+	{ } /* end */
+};
+
 static int alc_build_controls(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
@@ -2020,11 +2047,13 @@ static int alc_build_controls(struct hda_codec *codec)
 						    spec->multiout.dig_out_nid);
 		if (err < 0)
 			return err;
-		err = snd_hda_create_spdif_share_sw(codec,
-						    &spec->multiout);
-		if (err < 0)
-			return err;
-		spec->multiout.share_spdif = 1;
+		if (!spec->no_analog) {
+			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);
@@ -2032,8 +2061,24 @@ static int alc_build_controls(struct hda_codec *codec)
 			return err;
 	}
 
+	/* create beep controls if needed */
+	if (spec->beep_amp) {
+		struct snd_kcontrol_new *knew;
+		for (knew = alc_beep_mixer; knew->name; knew++) {
+			struct snd_kcontrol *kctl;
+			kctl = snd_ctl_new1(knew, codec);
+			if (!kctl)
+				return -ENOMEM;
+			kctl->private_value = spec->beep_amp;
+			err = snd_hda_ctl_add(codec, kctl);
+			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")) {
+	if (!spec->no_analog &&
+	    !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);
@@ -2042,7 +2087,8 @@ static int alc_build_controls(struct hda_codec *codec)
 		if (err < 0)
 			return err;
 	}
-	if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
+	if (!spec->no_analog &&
+	    !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) {
 		err = snd_hda_add_vmaster(codec, "Master Playback Switch",
 					  NULL, alc_slave_sws);
 		if (err < 0)
@@ -2951,6 +2997,14 @@ static int alc880_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
 					     stream_tag, format, substream);
 }
 
+static int alc880_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
+					   struct hda_codec *codec,
+					   struct snd_pcm_substream *substream)
+{
+	struct alc_spec *spec = codec->spec;
+	return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout);
+}
+
 static int alc880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo,
 					 struct hda_codec *codec,
 					 struct snd_pcm_substream *substream)
@@ -3034,7 +3088,8 @@ static struct hda_pcm_stream alc880_pcm_digital_playback = {
 	.ops = {
 		.open = alc880_dig_playback_pcm_open,
 		.close = alc880_dig_playback_pcm_close,
-		.prepare = alc880_dig_playback_pcm_prepare
+		.prepare = alc880_dig_playback_pcm_prepare,
+		.cleanup = alc880_dig_playback_pcm_cleanup
 	},
 };
 
@@ -3061,6 +3116,9 @@ static int alc_build_pcms(struct hda_codec *codec)
 	codec->num_pcms = 1;
 	codec->pcm_info = info;
 
+	if (spec->no_analog)
+		goto skip_analog;
+
 	info->name = spec->stream_name_analog;
 	if (spec->stream_analog_playback) {
 		if (snd_BUG_ON(!spec->multiout.dac_nids))
@@ -3084,12 +3142,17 @@ static int alc_build_pcms(struct hda_codec *codec)
 		}
 	}
 
+ skip_analog:
 	/* SPDIF for stream index #1 */
 	if (spec->multiout.dig_out_nid || spec->dig_in_nid) {
 		codec->num_pcms = 2;
+	        codec->slave_dig_outs = spec->multiout.slave_dig_outs;
 		info = spec->pcm_rec + 1;
 		info->name = spec->stream_name_digital;
-		info->pcm_type = HDA_PCM_TYPE_SPDIF;
+		if (spec->dig_out_type)
+			info->pcm_type = spec->dig_out_type;
+		else
+			info->pcm_type = HDA_PCM_TYPE_SPDIF;
 		if (spec->multiout.dig_out_nid &&
 		    spec->stream_digital_playback) {
 			info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback);
@@ -3104,6 +3167,9 @@ static int alc_build_pcms(struct hda_codec *codec)
 		codec->spdif_status_reset = 1;
 	}
 
+	if (spec->no_analog)
+		return 0;
+
 	/* If the use of more than one ADC is requested for the current
 	 * model, configure a second analog capture-only PCM.
 	 */
@@ -3162,65 +3228,17 @@ static void alc_free(struct hda_codec *codec)
 
 	alc_free_kctls(codec);
 	kfree(spec);
-	codec->spec = NULL; /* to be sure */
+	snd_hda_detach_beep_device(codec);
 }
 
 #ifdef SND_HDA_NEEDS_RESUME
-static void store_pin_configs(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	hda_nid_t nid, end_nid;
-
-	end_nid = codec->start_nid + codec->num_nodes;
-	for (nid = codec->start_nid; nid < end_nid; nid++) {
-		unsigned int wid_caps = get_wcaps(codec, nid);
-		unsigned int wid_type =
-			(wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
-		if (wid_type != AC_WID_PIN)
-			continue;
-		if (spec->num_pins >= ARRAY_SIZE(spec->pin_nids))
-			break;
-		spec->pin_nids[spec->num_pins] = nid;
-		spec->pin_cfgs[spec->num_pins] =
-			snd_hda_codec_read(codec, nid, 0,
-					   AC_VERB_GET_CONFIG_DEFAULT, 0);
-		spec->num_pins++;
-	}
-}
-
-static void resume_pin_configs(struct hda_codec *codec)
-{
-	struct alc_spec *spec = codec->spec;
-	int i;
-
-	for (i = 0; i < spec->num_pins; i++) {
-		hda_nid_t pin_nid = spec->pin_nids[i];
-		unsigned int pin_config = spec->pin_cfgs[i];
-		snd_hda_codec_write(codec, pin_nid, 0,
-				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_0,
-				    pin_config & 0x000000ff);
-		snd_hda_codec_write(codec, pin_nid, 0,
-				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_1,
-				    (pin_config & 0x0000ff00) >> 8);
-		snd_hda_codec_write(codec, pin_nid, 0,
-				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_2,
-				    (pin_config & 0x00ff0000) >> 16);
-		snd_hda_codec_write(codec, pin_nid, 0,
-				    AC_VERB_SET_CONFIG_DEFAULT_BYTES_3,
-				    pin_config >> 24);
-	}
-}
-
 static int alc_resume(struct hda_codec *codec)
 {
-	resume_pin_configs(codec);
 	codec->patch_ops.init(codec);
 	snd_hda_codec_resume_amp(codec);
 	snd_hda_codec_resume_cache(codec);
 	return 0;
 }
-#else
-#define store_pin_configs(codec)
 #endif
 
 /*
@@ -3559,7 +3577,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG),
 	SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST),
 	SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST),
-	SND_PCI_QUIRK(0x1043, 0, "ASUS", ALC880_ASUS), /* default ASUS */
+	SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_ASUS), /* default ASUS */
 	SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_3ST),
 	SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST),
 	SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST),
@@ -3602,7 +3620,8 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_5ST_DIG),
 	SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_5ST_DIG),
 	SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_5ST_DIG),
-	SND_PCI_QUIRK(0x8086, 0, "Intel mobo", ALC880_3ST), /* default Intel */
+	/* default Intel */
+	SND_PCI_QUIRK_VENDOR(0x8086, "Intel mobo", ALC880_3ST),
 	SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_5ST_DIG),
 	SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_6ST_DIG),
 	{}
@@ -3782,7 +3801,7 @@ static struct alc_config_preset alc880_presets[] = {
 		.input_mux = &alc880_capture_source,
 	},
 	[ALC880_UNIWILL_DIG] = {
-		.mixers = { alc880_asus_mixer, alc880_pcbeep_mixer },
+		.mixers = { alc880_asus_mixer },
 		.init_verbs = { alc880_volume_init_verbs,
 				alc880_pin_asus_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc880_asus_dac_nids),
@@ -3820,8 +3839,7 @@ static struct alc_config_preset alc880_presets[] = {
 		.init_hook = alc880_uniwill_p53_hp_automute,
 	},
 	[ALC880_FUJITSU] = {
-		.mixers = { alc880_fujitsu_mixer,
-			    alc880_pcbeep_mixer, },
+		.mixers = { alc880_fujitsu_mixer },
 		.init_verbs = { alc880_volume_init_verbs,
 				alc880_uniwill_p53_init_verbs,
 	       			alc880_beep_init_verbs },
@@ -4114,7 +4132,7 @@ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin,
 static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec,
 						const struct auto_pin_cfg *cfg)
 {
-	struct hda_input_mux *imux = &spec->private_imux;
+	struct hda_input_mux *imux = &spec->private_imux[0];
 	int i, err, idx;
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
@@ -4202,11 +4220,9 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec)
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
 		if (alc880_is_input_pin(nid)) {
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_PIN_WIDGET_CONTROL,
-					    i <= AUTO_PIN_FRONT_MIC ?
-					    PIN_VREF80 : PIN_IN);
-			if (nid != ALC880_PIN_CD_NID)
+			alc_set_input_pin(codec, nid, i);
+			if (nid != ALC880_PIN_CD_NID &&
+			    (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
 				snd_hda_codec_write(codec, nid, 0,
 						    AC_VERB_SET_AMP_GAIN_MUTE,
 						    AMP_OUT_MUTE);
@@ -4221,7 +4237,7 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec)
 static int alc880_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	int err;
+	int i, err;
 	static hda_nid_t alc880_ignore[] = { 0x1d, 0 };
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
@@ -4252,8 +4268,23 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
-		spec->multiout.dig_out_nid = ALC880_DIGOUT_NID;
+	/* check multiple SPDIF-out (for recent codecs) */
+	for (i = 0; i < spec->autocfg.dig_outs; i++) {
+		hda_nid_t dig_nid;
+		err = snd_hda_get_connections(codec,
+					      spec->autocfg.dig_out_pins[i],
+					      &dig_nid, 1);
+		if (err < 0)
+			continue;
+		if (!i)
+			spec->multiout.dig_out_nid = dig_nid;
+		else {
+			spec->multiout.slave_dig_outs = spec->slave_dig_outs;
+			spec->slave_dig_outs[i - 1] = dig_nid;
+			if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1)
+				break;
+		}
+	}
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = ALC880_DIGIN_NID;
 
@@ -4263,9 +4294,8 @@ static int alc880_parse_auto_config(struct hda_codec *codec)
 	add_verb(spec, alc880_volume_init_verbs);
 
 	spec->num_mux_defs = 1;
-	spec->input_mux = &spec->private_imux;
+	spec->input_mux = &spec->private_imux[0];
 
-	store_pin_configs(codec);
 	return 1;
 }
 
@@ -4280,21 +4310,33 @@ static void alc880_auto_init(struct hda_codec *codec)
 		alc_inithook(codec);
 }
 
-/*
- * OK, here we have finally the patch for ALC880
- */
-
 static void set_capture_mixer(struct alc_spec *spec)
 {
-	static struct snd_kcontrol_new *caps[3] = {
-		alc_capture_mixer1,
-		alc_capture_mixer2,
-		alc_capture_mixer3,
+	static struct snd_kcontrol_new *caps[2][3] = {
+		{ alc_capture_mixer_nosrc1,
+		  alc_capture_mixer_nosrc2,
+		  alc_capture_mixer_nosrc3 },
+		{ alc_capture_mixer1,
+		  alc_capture_mixer2,
+		  alc_capture_mixer3 },
 	};
-	if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3)
-		spec->cap_mixer = caps[spec->num_adc_nids - 1];
+	if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) {
+		int mux;
+		if (spec->input_mux && spec->input_mux->num_items > 1)
+			mux = 1;
+		else
+			mux = 0;
+		spec->cap_mixer = caps[mux][spec->num_adc_nids - 1];
+	}
 }
 
+#define set_beep_amp(spec, nid, idx, dir) \
+	((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir))
+
+/*
+ * OK, here we have finally the patch for ALC880
+ */
+
 static int patch_alc880(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
@@ -4330,6 +4372,12 @@ static int patch_alc880(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x1);
+	if (err < 0) {
+		alc_free(codec);
+		return err;
+	}
+
 	if (board_config != ALC880_AUTO)
 		setup_preset(spec, &alc880_presets[board_config]);
 
@@ -4356,6 +4404,7 @@ static int patch_alc880(struct hda_codec *codec)
 		}
 	}
 	set_capture_mixer(spec);
+	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
 	spec->vmaster_nid = 0x0c;
 
@@ -4463,6 +4512,26 @@ static struct hda_input_mux alc260_acer_capture_sources[2] = {
 		},
 	},
 };
+
+/* Maxdata Favorit 100XS */
+static struct hda_input_mux alc260_favorit100_capture_sources[2] = {
+	{
+		.num_items = 2,
+		.items = {
+			{ "Line/Mic", 0x0 },
+			{ "CD", 0x4 },
+		},
+	},
+	{
+		.num_items = 3,
+		.items = {
+			{ "Line/Mic", 0x0 },
+			{ "CD", 0x4 },
+			{ "Mixer", 0x5 },
+		},
+	},
+};
+
 /*
  * This is just place-holder, so there's something for alc_build_pcms to look
  * at when it calculates the maximum number of channels. ALC260 has no mixer
@@ -4505,12 +4574,6 @@ static struct snd_kcontrol_new alc260_input_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new alc260_pc_beep_mixer[] = {
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x07, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x07, 0x05, HDA_INPUT),
-	{ } /* end */
-};
-
 /* update HP, line and mono out pins according to the master switch */
 static void alc260_hp_master_update(struct hda_codec *codec,
 				    hda_nid_t hp, hda_nid_t line,
@@ -4702,8 +4765,6 @@ static struct snd_kcontrol_new alc260_fujitsu_mixer[] = {
 	HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT),
 	ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
 	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT),
 	HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT),
 	{ } /* end */
@@ -4748,8 +4809,18 @@ static struct snd_kcontrol_new alc260_acer_mixer[] = {
 	HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT),
 	HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT),
 	ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
+	{ } /* end */
+};
+
+/* Maxdata Favorit 100XS: one output and one input (0x12) jack
+ */
+static struct snd_kcontrol_new alc260_favorit100_mixer[] = {
+	HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT),
+	ALC_PIN_MODE("Output Jack Mode", 0x0f, ALC_PIN_DIR_INOUT),
+	HDA_CODEC_VOLUME("Line/Mic Playback Volume", 0x07, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Line/Mic Playback Switch", 0x07, 0x0, HDA_INPUT),
+	ALC_PIN_MODE("Line/Mic Jack Mode", 0x12, ALC_PIN_DIR_IN),
 	{ } /* end */
 };
 
@@ -4767,8 +4838,6 @@ static struct snd_kcontrol_new alc260_will_mixer[] = {
 	ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -5126,6 +5195,89 @@ static struct hda_verb alc260_acer_init_verbs[] = {
 	{ }
 };
 
+/* Initialisation sequence for Maxdata Favorit 100XS
+ * (adapted from Acer init verbs).
+ */
+static struct hda_verb alc260_favorit100_init_verbs[] = {
+	/* GPIO 0 enables the output jack.
+	 * Turn this on and rely on the standard mute
+	 * methods whenever the user wants to turn these outputs off.
+	 */
+	{0x01, AC_VERB_SET_GPIO_MASK, 0x01},
+	{0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01},
+	{0x01, AC_VERB_SET_GPIO_DATA, 0x01},
+	/* Line/Mic input jack is connected to Mic1 pin */
+	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50},
+	/* Ensure all other unused pins are disabled and muted. */
+	{0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0},
+	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	/* Disable digital (SPDIF) pins */
+	{0x03, AC_VERB_SET_DIGI_CONVERT_1, 0},
+	{0x06, AC_VERB_SET_DIGI_CONVERT_1, 0},
+
+	/* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum
+	 * bus when acting as outputs.
+	 */
+	{0x0b, AC_VERB_SET_CONNECT_SEL, 0},
+	{0x0d, AC_VERB_SET_CONNECT_SEL, 0},
+
+	/* Start with output sum widgets muted and their output gains at min */
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	{0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+
+	/* Unmute Line-out pin widget amp left and right
+	 * (no equiv mixer ctrl)
+	 */
+	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	/* Unmute Mic1 and Line1 pin widget input buffers since they start as
+	 * inputs. If the pin mode is changed by the user the pin mode control
+	 * will take care of enabling the pin's input/output buffers as needed.
+	 * Therefore there's no need to enable the input buffer at this
+	 * stage.
+	 */
+	{0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
+
+	/* Mute capture amp left and right */
+	{0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	/* Set ADC connection select to match default mixer setting - mic
+	 * (on mic1 pin)
+	 */
+	{0x04, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* Do similar with the second ADC: mute capture input amp and
+	 * set ADC connection to mic to match ALSA's default state.
+	 */
+	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x05, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* Mute all inputs to mixer widget (even unconnected ones) */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */
+	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */
+
+	{ }
+};
+
 static struct hda_verb alc260_will_verbs[] = {
 	{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x0b, AC_VERB_SET_CONNECT_SEL, 0x00},
@@ -5272,8 +5424,6 @@ static struct snd_kcontrol_new alc260_test_mixer[] = {
 	HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT),
 	HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT),
 	HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT),
 	HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT),
@@ -5471,7 +5621,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec,
 static int alc260_auto_create_analog_input_ctls(struct alc_spec *spec,
 						const struct auto_pin_cfg *cfg)
 {
-	struct hda_input_mux *imux = &spec->private_imux;
+	struct hda_input_mux *imux = &spec->private_imux[0];
 	int i, err, idx;
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
@@ -5546,11 +5696,9 @@ static void alc260_auto_init_analog_input(struct hda_codec *codec)
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
 		if (nid >= 0x12) {
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_PIN_WIDGET_CONTROL,
-					    i <= AUTO_PIN_FRONT_MIC ?
-					    PIN_VREF80 : PIN_IN);
-			if (nid != ALC260_PIN_CD_NID)
+			alc_set_input_pin(codec, nid, i);
+			if (nid != ALC260_PIN_CD_NID &&
+			    (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
 				snd_hda_codec_write(codec, nid, 0,
 						    AC_VERB_SET_AMP_GAIN_MUTE,
 						    AMP_OUT_MUTE);
@@ -5623,7 +5771,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = ALC260_DIGOUT_NID;
 	if (spec->kctls.list)
 		add_mixer(spec, spec->kctls.list);
@@ -5631,9 +5779,8 @@ static int alc260_parse_auto_config(struct hda_codec *codec)
 	add_verb(spec, alc260_volume_init_verbs);
 
 	spec->num_mux_defs = 1;
-	spec->input_mux = &spec->private_imux;
+	spec->input_mux = &spec->private_imux[0];
 
-	store_pin_configs(codec);
 	return 1;
 }
 
@@ -5670,6 +5817,7 @@ static const char *alc260_models[ALC260_MODEL_LAST] = {
 	[ALC260_ACER]		= "acer",
 	[ALC260_WILL]		= "will",
 	[ALC260_REPLACER_672V]	= "replacer",
+	[ALC260_FAVORIT100]	= "favorit100",
 #ifdef CONFIG_SND_DEBUG
 	[ALC260_TEST]		= "test",
 #endif
@@ -5679,6 +5827,7 @@ static const char *alc260_models[ALC260_MODEL_LAST] = {
 static struct snd_pci_quirk alc260_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER),
 	SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER),
+	SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100),
 	SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013),
 	SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013),
 	SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013),
@@ -5701,8 +5850,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = {
 static struct alc_config_preset alc260_presets[] = {
 	[ALC260_BASIC] = {
 		.mixers = { alc260_base_output_mixer,
-			    alc260_input_mixer,
-			    alc260_pc_beep_mixer },
+			    alc260_input_mixer },
 		.init_verbs = { alc260_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
 		.dac_nids = alc260_dac_nids,
@@ -5781,6 +5929,18 @@ static struct alc_config_preset alc260_presets[] = {
 		.num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources),
 		.input_mux = alc260_acer_capture_sources,
 	},
+	[ALC260_FAVORIT100] = {
+		.mixers = { alc260_favorit100_mixer },
+		.init_verbs = { alc260_favorit100_init_verbs },
+		.num_dacs = ARRAY_SIZE(alc260_dac_nids),
+		.dac_nids = alc260_dac_nids,
+		.num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids),
+		.adc_nids = alc260_dual_adc_nids,
+		.num_channel_mode = ARRAY_SIZE(alc260_modes),
+		.channel_mode = alc260_modes,
+		.num_mux_defs = ARRAY_SIZE(alc260_favorit100_capture_sources),
+		.input_mux = alc260_favorit100_capture_sources,
+	},
 	[ALC260_WILL] = {
 		.mixers = { alc260_will_mixer },
 		.init_verbs = { alc260_init_verbs, alc260_will_verbs },
@@ -5857,6 +6017,12 @@ static int patch_alc260(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x1);
+	if (err < 0) {
+		alc_free(codec);
+		return err;
+	}
+
 	if (board_config != ALC260_AUTO)
 		setup_preset(spec, &alc260_presets[board_config]);
 
@@ -5882,6 +6048,7 @@ static int patch_alc260(struct hda_codec *codec)
 		}
 	}
 	set_capture_mixer(spec);
+	set_beep_amp(spec, 0x07, 0x05, HDA_INPUT);
 
 	spec->vmaster_nid = 0x08;
 
@@ -6053,8 +6220,6 @@ static struct snd_kcontrol_new alc882_base_mixer[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -6081,8 +6246,6 @@ static struct snd_kcontrol_new alc882_w2jc_mixer[] = {
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -6134,8 +6297,6 @@ static struct snd_kcontrol_new alc882_asus_a7m_mixer[] = {
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -6244,8 +6405,10 @@ static struct snd_kcontrol_new alc882_macpro_mixer[] = {
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x18, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	/* FIXME: this looks suspicious...
 	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x02, HDA_INPUT),
 	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	*/
 	{ } /* end */
 };
 
@@ -6877,19 +7040,9 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec)
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
-		unsigned int vref;
 		if (!nid)
 			continue;
-		vref = PIN_IN;
-		if (1 /*i <= AUTO_PIN_FRONT_MIC*/) {
-			unsigned int pincap;
-			pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
-			if ((pincap >> AC_PINCAP_VREF_SHIFT) &
-			    AC_PINCAP_VREF_80)
-				vref = PIN_VREF80;
-		}
-		snd_hda_codec_write(codec, nid, 0,
-				    AC_VERB_SET_PIN_WIDGET_CONTROL, vref);
+		alc_set_input_pin(codec, nid, AUTO_PIN_FRONT_MIC /*i*/);
 		if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)
 			snd_hda_codec_write(codec, nid, 0,
 					    AC_VERB_SET_AMP_GAIN_MUTE,
@@ -6900,18 +7053,21 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec)
 static void alc882_auto_init_input_src(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	const struct hda_input_mux *imux = spec->input_mux;
 	int c;
 
 	for (c = 0; c < spec->num_adc_nids; c++) {
 		hda_nid_t conn_list[HDA_MAX_NUM_INPUTS];
 		hda_nid_t nid = spec->capsrc_nids[c];
+		unsigned int mux_idx;
+		const struct hda_input_mux *imux;
 		int conns, mute, idx, item;
 
 		conns = snd_hda_get_connections(codec, nid, conn_list,
 						ARRAY_SIZE(conn_list));
 		if (conns < 0)
 			continue;
+		mux_idx = c >= spec->num_mux_defs ? 0 : c;
+		imux = &spec->input_mux[mux_idx];
 		for (idx = 0; idx < conns; idx++) {
 			/* if the current connection is the selected one,
 			 * unmute it as default - otherwise mute it
@@ -6924,8 +7080,20 @@ static void alc882_auto_init_input_src(struct hda_codec *codec)
 					break;
 				}
 			}
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_AMP_GAIN_MUTE, mute);
+			/* check if we have a selector or mixer
+			 * we could check for the widget type instead, but
+			 * just check for Amp-In presence (in case of mixer
+			 * without amp-in there is something wrong, this
+			 * function shouldn't be used or capsrc nid is wrong)
+			 */
+			if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP)
+				snd_hda_codec_write(codec, nid, 0,
+						    AC_VERB_SET_AMP_GAIN_MUTE,
+						    mute);
+			else if (mute != AMP_IN_MUTE(idx))
+				snd_hda_codec_write(codec, nid, 0,
+						    AC_VERB_SET_CONNECT_SEL,
+						    idx);
 		}
 	}
 }
@@ -7054,6 +7222,12 @@ static int patch_alc882(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x1);
+	if (err < 0) {
+		alc_free(codec);
+		return err;
+	}
+
 	if (board_config != ALC882_AUTO)
 		setup_preset(spec, &alc882_presets[board_config]);
 
@@ -7074,7 +7248,7 @@ static int patch_alc882(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc882_pcm_digital_playback;
 	spec->stream_digital_capture = &alc882_pcm_digital_capture;
 
-	spec->is_mix_capture = 1; /* matrix-style capture */
+	spec->capture_style = CAPT_MIX; /* matrix-style capture */
 	if (!spec->adc_nids && spec->input_mux) {
 		/* check whether NID 0x07 is valid */
 		unsigned int wcap = get_wcaps(codec, 0x07);
@@ -7091,6 +7265,7 @@ static int patch_alc882(struct hda_codec *codec)
 		}
 	}
 	set_capture_mixer(spec);
+	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
 	spec->vmaster_nid = 0x0c;
 
@@ -7142,10 +7317,14 @@ static hda_nid_t alc883_adc_nids_rev[2] = {
 	0x09, 0x08
 };
 
+#define alc889_adc_nids		alc880_adc_nids
+
 static hda_nid_t alc883_capsrc_nids[2] = { 0x23, 0x22 };
 
 static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 };
 
+#define alc889_capsrc_nids	alc882_capsrc_nids
+
 /* input MUX */
 /* FIXME: should be a matrix-type input source selection */
 
@@ -7363,8 +7542,6 @@ static struct snd_kcontrol_new alc883_base_mixer[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -7427,8 +7604,6 @@ static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -7452,8 +7627,6 @@ static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -7478,8 +7651,6 @@ static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x18, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -7503,8 +7674,6 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -7912,36 +8081,83 @@ static struct hda_verb alc888_lenovo_sky_verbs[] = {
 	{ } /* end */
 };
 
+static struct hda_verb alc888_6st_dell_verbs[] = {
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
+	{ }
+};
+
+static void alc888_3st_hp_front_automute(struct hda_codec *codec)
+{
+	unsigned int present, bits;
+
+	present = snd_hda_codec_read(codec, 0x1b, 0,
+			AC_VERB_GET_PIN_SENSE, 0) & 0x80000000;
+	bits = present ? HDA_AMP_MUTE : 0;
+	snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, bits);
+	snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, bits);
+	snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0,
+				 HDA_AMP_MUTE, bits);
+}
+
+static void alc888_3st_hp_unsol_event(struct hda_codec *codec,
+				      unsigned int res)
+{
+	switch (res >> 26) {
+	case ALC880_HP_EVENT:
+		alc888_3st_hp_front_automute(codec);
+		break;
+	}
+}
+
 static struct hda_verb alc888_3st_hp_verbs[] = {
 	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},	/* Front: output 0 (0x0c) */
 	{0x16, AC_VERB_SET_CONNECT_SEL, 0x01},	/* Rear : output 1 (0x0d) */
 	{0x18, AC_VERB_SET_CONNECT_SEL, 0x02},	/* CLFE : output 2 (0x0e) */
-	{ }
-};
-
-static struct hda_verb alc888_6st_dell_verbs[] = {
 	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-	{ }
+	{ } /* end */
 };
 
+/*
+ * 2ch mode
+ */
 static struct hda_verb alc888_3st_hp_2ch_init[] = {
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
 	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN },
 	{ 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
-	{ }
+	{ } /* end */
+};
+
+/*
+ * 4ch mode
+ */
+static struct hda_verb alc888_3st_hp_4ch_init[] = {
+	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 },
+	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE },
+	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
+	{ 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x16, AC_VERB_SET_CONNECT_SEL, 0x01 },
+	{ } /* end */
 };
 
+/*
+ * 6ch mode
+ */
 static struct hda_verb alc888_3st_hp_6ch_init[] = {
 	{ 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
+	{ 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 },
 	{ 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },
 	{ 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE },
-	{ }
+	{ 0x16, AC_VERB_SET_CONNECT_SEL, 0x01 },
+	{ } /* end */
 };
 
-static struct hda_channel_mode alc888_3st_hp_modes[2] = {
+static struct hda_channel_mode alc888_3st_hp_modes[3] = {
 	{ 2, alc888_3st_hp_2ch_init },
+	{ 4, alc888_3st_hp_4ch_init },
 	{ 6, alc888_3st_hp_6ch_init },
 };
 
@@ -8202,7 +8418,7 @@ static void alc888_6st_dell_unsol_event(struct hda_codec *codec,
 {
 	switch (res >> 26) {
 	case ALC880_HP_EVENT:
-		printk("hp_event\n");
+		/* printk(KERN_DEBUG "hp_event\n"); */
 		alc888_6st_dell_front_automute(codec);
 		break;
 	}
@@ -8461,6 +8677,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_ACER_ASPIRE),
+	SND_PCI_QUIRK(0x1025, 0x010a, "Acer Ferrari 5000", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE),
 	SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE),
@@ -8468,17 +8685,21 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 		ALC888_ACER_ASPIRE_4930G),
 	SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G",
 		ALC888_ACER_ASPIRE_4930G),
+	SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC883_AUTO),
+	SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC883_AUTO),
 	SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G",
 		ALC888_ACER_ASPIRE_4930G),
 	SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G",
 		ALC888_ACER_ASPIRE_4930G),
-	SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */
+	/* default Acer */
+	SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER),
 	SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL),
 	SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP),
+	SND_PCI_QUIRK(0x103c, 0x2a72, "HP Educ.ar", ALC888_3ST_HP),
 	SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V),
 	SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG),
@@ -8518,7 +8739,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG),
 	SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720),
 	SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720),
-	SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD),
+	SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC883_LAPTOP_EAPD),
 	SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch),
 	SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION),
 	SND_PCI_QUIRK(0x1734, 0x1107, "FSC AMILO Xi2550",
@@ -8543,6 +8764,10 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	{}
 };
 
+static hda_nid_t alc1200_slave_dig_outs[] = {
+	ALC883_DIGOUT_NID, 0,
+};
+
 static struct alc_config_preset alc883_presets[] = {
 	[ALC883_3ST_2ch_DIG] = {
 		.mixers = { alc883_3ST_2ch_mixer },
@@ -8778,6 +9003,8 @@ static struct alc_config_preset alc883_presets[] = {
 		.channel_mode = alc888_3st_hp_modes,
 		.need_dac_fix = 1,
 		.input_mux = &alc883_capture_source,
+		.unsol_event = alc888_3st_hp_unsol_event,
+		.init_hook = alc888_3st_hp_front_automute,
 	},
 	[ALC888_6ST_DELL] = {
 		.mixers = { alc883_base_mixer, alc883_chmode_mixer },
@@ -8883,6 +9110,7 @@ static struct alc_config_preset alc883_presets[] = {
 		.dac_nids = alc883_dac_nids,
 		.dig_out_nid = ALC1200_DIGOUT_NID,
 		.dig_in_nid = ALC883_DIGIN_NID,
+		.slave_dig_outs = alc1200_slave_dig_outs,
 		.num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes),
 		.channel_mode = alc883_sixstack_modes,
 		.input_mux = &alc883_capture_source,
@@ -8950,11 +9178,9 @@ static void alc883_auto_init_analog_input(struct hda_codec *codec)
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
 		if (alc883_is_input_pin(nid)) {
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_PIN_WIDGET_CONTROL,
-					    (i <= AUTO_PIN_FRONT_MIC ?
-					     PIN_VREF80 : PIN_IN));
-			if (nid != ALC883_PIN_CD_NID)
+			alc_set_input_pin(codec, nid, i);
+			if (nid != ALC883_PIN_CD_NID &&
+			    (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
 				snd_hda_codec_write(codec, nid, 0,
 						    AC_VERB_SET_AMP_GAIN_MUTE,
 						    AMP_OUT_MUTE);
@@ -8969,6 +9195,8 @@ static int alc883_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
 	int err = alc880_parse_auto_config(codec);
+	struct auto_pin_cfg *cfg = &spec->autocfg;
+	int i;
 
 	if (err < 0)
 		return err;
@@ -8982,6 +9210,26 @@ static int alc883_parse_auto_config(struct hda_codec *codec)
 	/* hack - override the init verbs */
 	spec->init_verbs[0] = alc883_auto_init_verbs;
 
+	/* setup input_mux for ALC889 */
+	if (codec->vendor_id == 0x10ec0889) {
+		/* digital-mic input pin is excluded in alc880_auto_create..()
+		 * because it's under 0x18
+		 */
+		if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 ||
+		    cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) {
+			struct hda_input_mux *imux = &spec->private_imux[0];
+			for (i = 1; i < 3; i++)
+				memcpy(&spec->private_imux[i],
+				       &spec->private_imux[0],
+				       sizeof(spec->private_imux[0]));
+			imux->items[imux->num_items].label = "Int DMic";
+			imux->items[imux->num_items].index = 0x0b;
+			imux->num_items++;
+			spec->num_mux_defs = 3;
+			spec->input_mux = spec->private_imux;
+		}
+	}
+
 	return 1; /* config found */
 }
 
@@ -9033,6 +9281,12 @@ static int patch_alc883(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x1);
+	if (err < 0) {
+		alc_free(codec);
+		return err;
+	}
+
 	if (board_config != ALC883_AUTO)
 		setup_preset(spec, &alc883_presets[board_config]);
 
@@ -9045,14 +9299,36 @@ static int patch_alc883(struct hda_codec *codec)
 			spec->stream_name_analog = "ALC888 Analog";
 			spec->stream_name_digital = "ALC888 Digital";
 		}
+		if (!spec->num_adc_nids) {
+			spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
+			spec->adc_nids = alc883_adc_nids;
+		}
+		if (!spec->capsrc_nids)
+			spec->capsrc_nids = alc883_capsrc_nids;
+		spec->capture_style = CAPT_MIX; /* matrix-style capture */
 		break;
 	case 0x10ec0889:
 		spec->stream_name_analog = "ALC889 Analog";
 		spec->stream_name_digital = "ALC889 Digital";
+		if (!spec->num_adc_nids) {
+			spec->num_adc_nids = ARRAY_SIZE(alc889_adc_nids);
+			spec->adc_nids = alc889_adc_nids;
+		}
+		if (!spec->capsrc_nids)
+			spec->capsrc_nids = alc889_capsrc_nids;
+		spec->capture_style = CAPT_1MUX_MIX; /* 1mux/Nmix-style
+							capture */
 		break;
 	default:
 		spec->stream_name_analog = "ALC883 Analog";
 		spec->stream_name_digital = "ALC883 Digital";
+		if (!spec->num_adc_nids) {
+			spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
+			spec->adc_nids = alc883_adc_nids;
+		}
+		if (!spec->capsrc_nids)
+			spec->capsrc_nids = alc883_capsrc_nids;
+		spec->capture_style = CAPT_MIX; /* matrix-style capture */
 		break;
 	}
 
@@ -9063,15 +9339,9 @@ static int patch_alc883(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc883_pcm_digital_playback;
 	spec->stream_digital_capture = &alc883_pcm_digital_capture;
 
-	if (!spec->num_adc_nids) {
-		spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids);
-		spec->adc_nids = alc883_adc_nids;
-	}
-	if (!spec->capsrc_nids)
-		spec->capsrc_nids = alc883_capsrc_nids;
-	spec->is_mix_capture = 1; /* matrix-style capture */
 	if (!spec->cap_mixer)
 		set_capture_mixer(spec);
+	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
 	spec->vmaster_nid = 0x0c;
 
@@ -9124,8 +9394,6 @@ static struct snd_kcontrol_new alc262_base_mixer[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
-	/* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	   HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */
 	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT),
 	HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT),
@@ -9146,8 +9414,6 @@ static struct snd_kcontrol_new alc262_hippo1_mixer[] = {
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
-	/* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	   HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */
 	/*HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),*/
 	HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT),
 	{ } /* end */
@@ -9256,8 +9522,6 @@ static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = {
 	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	HDA_CODEC_VOLUME("AUX IN Playback Volume", 0x0b, 0x06, HDA_INPUT),
 	HDA_CODEC_MUTE("AUX IN Playback Switch", 0x0b, 0x06, HDA_INPUT),
 	{ } /* end */
@@ -9286,8 +9550,6 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = {
 	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -9435,6 +9697,67 @@ static struct snd_kcontrol_new alc262_benq_t31_mixer[] = {
 	{ } /* end */
 };
 
+static struct snd_kcontrol_new alc262_tyan_mixer[] = {
+	HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT),
+	HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT),
+	HDA_CODEC_VOLUME("Aux Playback Volume", 0x0b, 0x06, HDA_INPUT),
+	HDA_CODEC_MUTE("Aux Playback Switch", 0x0b, 0x06, HDA_INPUT),
+	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
+	HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT),
+	{ } /* end */
+};
+
+static struct hda_verb alc262_tyan_verbs[] = {
+	/* Headphone automute */
+	{0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT},
+	{0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
+	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
+
+	/* P11 AUX_IN, white 4-pin connector */
+	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
+	{0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, 0xe1},
+	{0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, 0x93},
+	{0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, 0x19},
+
+	{}
+};
+
+/* unsolicited event for HP jack sensing */
+static void alc262_tyan_automute(struct hda_codec *codec)
+{
+	unsigned int mute;
+	unsigned int present;
+
+	snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0);
+	present = snd_hda_codec_read(codec, 0x1b, 0,
+				     AC_VERB_GET_PIN_SENSE, 0);
+	present = (present & 0x80000000) != 0;
+	if (present) {
+		/* mute line output on ATX panel */
+		snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+					 HDA_AMP_MUTE, HDA_AMP_MUTE);
+	} else {
+		/* unmute line output if necessary */
+		mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0);
+		snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0,
+					 HDA_AMP_MUTE, mute);
+	}
+}
+
+static void alc262_tyan_unsol_event(struct hda_codec *codec,
+				       unsigned int res)
+{
+	if ((res >> 26) != ALC880_HP_EVENT)
+		return;
+	alc262_tyan_automute(codec);
+}
+
 #define alc262_capture_mixer		alc882_capture_mixer
 #define alc262_capture_alt_mixer	alc882_capture_alt_mixer
 
@@ -9901,8 +10224,6 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = {
 	},
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Switch", 0x0b, 0x05, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
@@ -10474,8 +10795,14 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 					   alc262_ignore);
 	if (err < 0)
 		return err;
-	if (!spec->autocfg.line_outs)
+	if (!spec->autocfg.line_outs) {
+		if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) {
+			spec->multiout.max_channels = 2;
+			spec->no_analog = 1;
+			goto dig_only;
+		}
 		return 0; /* can't find valid BIOS pin config */
+	}
 	err = alc262_auto_create_multi_out_ctls(spec, &spec->autocfg);
 	if (err < 0)
 		return err;
@@ -10485,8 +10812,11 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+ dig_only:
+	if (spec->autocfg.dig_outs) {
 		spec->multiout.dig_out_nid = ALC262_DIGOUT_NID;
+		spec->dig_out_type = spec->autocfg.dig_out_type[0];
+	}
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = ALC262_DIGIN_NID;
 
@@ -10495,13 +10825,12 @@ static int alc262_parse_auto_config(struct hda_codec *codec)
 
 	add_verb(spec, alc262_volume_init_verbs);
 	spec->num_mux_defs = 1;
-	spec->input_mux = &spec->private_imux;
+	spec->input_mux = &spec->private_imux[0];
 
 	err = alc_auto_add_mic_boost(codec);
 	if (err < 0)
 		return err;
 
-	store_pin_configs(codec);
 	return 1;
 }
 
@@ -10543,21 +10872,19 @@ static const char *alc262_models[ALC262_MODEL_LAST] = {
 	[ALC262_ULTRA]		= "ultra",
 	[ALC262_LENOVO_3000]	= "lenovo-3000",
 	[ALC262_NEC]		= "nec",
+	[ALC262_TYAN]		= "tyan",
 	[ALC262_AUTO]		= "auto",
 };
 
 static struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO),
 	SND_PCI_QUIRK(0x1033, 0x8895, "NEC Versa S9100", ALC262_NEC),
-	SND_PCI_QUIRK(0x103c, 0x12fe, "HP xw9400", ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x12ff, "HP xw4550", ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x1306, "HP xw8600", ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x1307, "HP xw6600", ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x1308, "HP xw4600", ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x1309, "HP xw4*00", ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x130a, "HP xw6*00", ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x130b, "HP xw8*00", ALC262_HP_BPC),
-	SND_PCI_QUIRK(0x103c, 0x170b, "HP xw*", ALC262_HP_BPC),
+	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1200, "HP xw series",
+			   ALC262_HP_BPC),
+	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series",
+			   ALC262_HP_BPC),
+	SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series",
+			   ALC262_HP_BPC),
 	SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL),
 	SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF),
 	SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL),
@@ -10575,17 +10902,17 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO),
 	SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD),
-	SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD),
-	SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD),
-	SND_PCI_QUIRK(0x104d, 0x9033, "Sony VAIO VGN-SR19XN",
-		      ALC262_SONY_ASSAMD),
+	SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */
+	SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO",
+			   ALC262_SONY_ASSAMD),
 	SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1",
 		      ALC262_TOSHIBA_RX1),
 	SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06),
 	SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU),
 	SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU),
-	SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA),
-	SND_PCI_QUIRK(0x144d, 0xc039, "Samsung Q1U EL", ALC262_ULTRA),
+	SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_TYAN),
+	SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc032, "Samsung Q1",
+			   ALC262_ULTRA),
 	SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO),
 	SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 y410", ALC262_LENOVO_3000),
 	SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8),
@@ -10802,6 +11129,19 @@ static struct alc_config_preset alc262_presets[] = {
 		.unsol_event = alc262_hippo_unsol_event,
 		.init_hook = alc262_hippo_automute,
 	},
+	[ALC262_TYAN] = {
+		.mixers = { alc262_tyan_mixer },
+		.init_verbs = { alc262_init_verbs, alc262_tyan_verbs},
+		.num_dacs = ARRAY_SIZE(alc262_dac_nids),
+		.dac_nids = alc262_dac_nids,
+		.hp_nid = 0x02,
+		.dig_out_nid = ALC262_DIGOUT_NID,
+		.num_channel_mode = ARRAY_SIZE(alc262_modes),
+		.channel_mode = alc262_modes,
+		.input_mux = &alc262_capture_source,
+		.unsol_event = alc262_tyan_unsol_event,
+		.init_hook = alc262_tyan_automute,
+	},
 };
 
 static int patch_alc262(struct hda_codec *codec)
@@ -10854,6 +11194,14 @@ static int patch_alc262(struct hda_codec *codec)
 		}
 	}
 
+	if (!spec->no_analog) {
+		err = snd_hda_attach_beep_device(codec, 0x1);
+		if (err < 0) {
+			alc_free(codec);
+			return err;
+		}
+	}
+
 	if (board_config != ALC262_AUTO)
 		setup_preset(spec, &alc262_presets[board_config]);
 
@@ -10865,7 +11213,7 @@ static int patch_alc262(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc262_pcm_digital_playback;
 	spec->stream_digital_capture = &alc262_pcm_digital_capture;
 
-	spec->is_mix_capture = 1;
+	spec->capture_style = CAPT_MIX;
 	if (!spec->adc_nids && spec->input_mux) {
 		/* check whether NID 0x07 is valid */
 		unsigned int wcap = get_wcaps(codec, 0x07);
@@ -10882,8 +11230,10 @@ static int patch_alc262(struct hda_codec *codec)
 			spec->capsrc_nids = alc262_capsrc_nids;
 		}
 	}
-	if (!spec->cap_mixer)
+	if (!spec->cap_mixer && !spec->no_analog)
 		set_capture_mixer(spec);
+	if (!spec->no_analog)
+		set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
 	spec->vmaster_nid = 0x0c;
 
@@ -11263,19 +11613,13 @@ static void alc267_quanta_il1_unsol_event(struct hda_codec *codec,
 static struct hda_verb alc268_base_init_verbs[] = {
 	/* Unmute DAC0-1 and set vol = 0 */
 	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 
 	/*
 	 * Set up output mixers (0x0c - 0x0e)
 	 */
 	/* set vol=0 to output mixers */
 	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
         {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00},
 
 	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -11294,9 +11638,7 @@ static struct hda_verb alc268_base_init_verbs[] = {
 	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	{0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 
 	/* set PCBEEP vol = 0, mute connections */
 	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -11318,10 +11660,8 @@ static struct hda_verb alc268_base_init_verbs[] = {
  */
 static struct hda_verb alc268_volume_init_verbs[] = {
 	/* set output DAC */
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
+	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
 
 	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
 	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24},
@@ -11329,16 +11669,12 @@ static struct hda_verb alc268_volume_init_verbs[] = {
 	{0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
 	{0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20},
 
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
 	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
 	{0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 
 	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 
 	/* set PCBEEP vol = 0, mute connections */
 	{0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
@@ -11537,7 +11873,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec,
 static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec,
 						const struct auto_pin_cfg *cfg)
 {
-	struct hda_input_mux *imux = &spec->private_imux;
+	struct hda_input_mux *imux = &spec->private_imux[0];
 	int i, idx1;
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
@@ -11631,9 +11967,14 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
 					   alc268_ignore);
 	if (err < 0)
 		return err;
-	if (!spec->autocfg.line_outs)
+	if (!spec->autocfg.line_outs) {
+		if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) {
+			spec->multiout.max_channels = 2;
+			spec->no_analog = 1;
+			goto dig_only;
+		}
 		return 0; /* can't find valid BIOS pin config */
-
+	}
 	err = alc268_auto_create_multi_out_ctls(spec, &spec->autocfg);
 	if (err < 0)
 		return err;
@@ -11643,25 +11984,26 @@ static int alc268_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = 2;
 
+ dig_only:
 	/* digital only support output */
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs) {
 		spec->multiout.dig_out_nid = ALC268_DIGOUT_NID;
-
+		spec->dig_out_type = spec->autocfg.dig_out_type[0];
+	}
 	if (spec->kctls.list)
 		add_mixer(spec, spec->kctls.list);
 
-	if (spec->autocfg.speaker_pins[0] != 0x1d)
+	if (!spec->no_analog && spec->autocfg.speaker_pins[0] != 0x1d)
 		add_mixer(spec, alc268_beep_mixer);
 
 	add_verb(spec, alc268_volume_init_verbs);
 	spec->num_mux_defs = 1;
-	spec->input_mux = &spec->private_imux;
+	spec->input_mux = &spec->private_imux[0];
 
 	err = alc_auto_add_mic_boost(codec);
 	if (err < 0)
 		return err;
 
-	store_pin_configs(codec);
 	return 1;
 }
 
@@ -11723,7 +12065,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = {
 
 static struct alc_config_preset alc268_presets[] = {
 	[ALC267_QUANTA_IL1] = {
-		.mixers = { alc267_quanta_il1_mixer },
+		.mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer },
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc267_quanta_il1_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -11805,7 +12147,8 @@ static struct alc_config_preset alc268_presets[] = {
 	},
 	[ALC268_ACER_ASPIRE_ONE] = {
 		.mixers = { alc268_acer_aspire_one_mixer,
-				alc268_capture_alt_mixer },
+			    alc268_beep_mixer,
+			    alc268_capture_alt_mixer },
 		.init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs,
 				alc268_acer_aspire_one_verbs },
 		.num_dacs = ARRAY_SIZE(alc268_dac_nids),
@@ -11874,7 +12217,7 @@ static int patch_alc268(struct hda_codec *codec)
 {
 	struct alc_spec *spec;
 	int board_config;
-	int err;
+	int i, has_beep, err;
 
 	spec = kcalloc(1, sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -11923,15 +12266,30 @@ static int patch_alc268(struct hda_codec *codec)
 
 	spec->stream_digital_playback = &alc268_pcm_digital_playback;
 
-	if (!query_amp_caps(codec, 0x1d, HDA_INPUT))
-		/* override the amp caps for beep generator */
-		snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT,
+	has_beep = 0;
+	for (i = 0; i < spec->num_mixers; i++) {
+		if (spec->mixers[i] == alc268_beep_mixer) {
+			has_beep = 1;
+			break;
+		}
+	}
+
+	if (has_beep) {
+		err = snd_hda_attach_beep_device(codec, 0x1);
+		if (err < 0) {
+			alc_free(codec);
+			return err;
+		}
+		if (!query_amp_caps(codec, 0x1d, HDA_INPUT))
+			/* override the amp caps for beep generator */
+			snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT,
 					  (0x0c << AC_AMPCAP_OFFSET_SHIFT) |
 					  (0x0c << AC_AMPCAP_NUM_STEPS_SHIFT) |
 					  (0x07 << AC_AMPCAP_STEP_SIZE_SHIFT) |
 					  (0 << AC_AMPCAP_MUTE_SHIFT));
+	}
 
-	if (!spec->adc_nids && spec->input_mux) {
+	if (!spec->no_analog && !spec->adc_nids && spec->input_mux) {
 		/* check whether NID 0x07 is valid */
 		unsigned int wcap = get_wcaps(codec, 0x07);
 		int i;
@@ -12012,8 +12370,6 @@ static struct snd_kcontrol_new alc269_base_mixer[] = {
 	HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x4, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x4, HDA_INPUT),
 	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
@@ -12040,8 +12396,6 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = {
 	HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
 	HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT),
 	{ }
 };
 
@@ -12065,8 +12419,6 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = {
 	HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x0b, 0x03, HDA_INPUT),
 	HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x0b, 0x03, HDA_INPUT),
 	HDA_CODEC_VOLUME("Dock Mic Boost", 0x1b, 0, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT),
 	{ }
 };
 
@@ -12103,13 +12455,6 @@ static struct snd_kcontrol_new alc269_fujitsu_mixer[] = {
 	{ } /* end */
 };
 
-/* beep control */
-static struct snd_kcontrol_new alc269_beep_mixer[] = {
-	HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x4, HDA_INPUT),
-	HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x4, HDA_INPUT),
-	{ } /* end */
-};
-
 static struct hda_verb alc269_quanta_fl1_verbs[] = {
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x01},
 	{0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
@@ -12509,7 +12854,7 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec,
 	 */
 	if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 ||
 	    cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) {
-		struct hda_input_mux *imux = &spec->private_imux;
+		struct hda_input_mux *imux = &spec->private_imux[0];
 		imux->items[imux->num_items].label = "Int Mic";
 		imux->items[imux->num_items].index = 0x05;
 		imux->num_items++;
@@ -12527,13 +12872,34 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec,
 #define alc269_pcm_digital_playback	alc880_pcm_digital_playback
 #define alc269_pcm_digital_capture	alc880_pcm_digital_capture
 
+static struct hda_pcm_stream alc269_44k_pcm_analog_playback = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 8,
+	.rates = SNDRV_PCM_RATE_44100, /* fixed rate */
+	/* NID is set in alc_build_pcms */
+	.ops = {
+		.open = alc880_playback_pcm_open,
+		.prepare = alc880_playback_pcm_prepare,
+		.cleanup = alc880_playback_pcm_cleanup
+	},
+};
+
+static struct hda_pcm_stream alc269_44k_pcm_analog_capture = {
+	.substreams = 1,
+	.channels_min = 2,
+	.channels_max = 2,
+	.rates = SNDRV_PCM_RATE_44100, /* fixed rate */
+	/* NID is set in alc_build_pcms */
+};
+
 /*
  * BIOS auto configuration
  */
 static int alc269_parse_auto_config(struct hda_codec *codec)
 {
 	struct alc_spec *spec = codec->spec;
-	int i, err;
+	int err;
 	static hda_nid_t alc269_ignore[] = { 0x1d, 0 };
 
 	err = snd_hda_parse_pin_def_config(codec, &spec->autocfg,
@@ -12550,22 +12916,15 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = ALC269_DIGOUT_NID;
 
 	if (spec->kctls.list)
 		add_mixer(spec, spec->kctls.list);
 
-	/* create a beep mixer control if the pin 0x1d isn't assigned */
-	for (i = 0; i < ARRAY_SIZE(spec->autocfg.input_pins); i++)
-		if (spec->autocfg.input_pins[i] == 0x1d)
-			break;
-	if (i >= ARRAY_SIZE(spec->autocfg.input_pins))
-		add_mixer(spec, alc269_beep_mixer);
-
 	add_verb(spec, alc269_init_verbs);
 	spec->num_mux_defs = 1;
-	spec->input_mux = &spec->private_imux;
+	spec->input_mux = &spec->private_imux[0];
 	/* set default input source */
 	snd_hda_codec_write_cache(codec, alc269_capsrc_nids[0],
 				  0, AC_VERB_SET_CONNECT_SEL,
@@ -12575,10 +12934,9 @@ static int alc269_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
-	if (!spec->cap_mixer)
+	if (!spec->cap_mixer && !spec->no_analog)
 		set_capture_mixer(spec);
 
-	store_pin_configs(codec);
 	return 1;
 }
 
@@ -12675,7 +13033,7 @@ static struct alc_config_preset alc269_presets[] = {
 		.init_hook = alc269_eeepc_dmic_inithook,
 	},
 	[ALC269_FUJITSU] = {
-		.mixers = { alc269_fujitsu_mixer, alc269_beep_mixer },
+		.mixers = { alc269_fujitsu_mixer },
 		.cap_mixer = alc269_epc_capture_mixer,
 		.init_verbs = { alc269_init_verbs,
 				alc269_eeepc_dmic_init_verbs },
@@ -12740,13 +13098,26 @@ static int patch_alc269(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x1);
+	if (err < 0) {
+		alc_free(codec);
+		return err;
+	}
+
 	if (board_config != ALC269_AUTO)
 		setup_preset(spec, &alc269_presets[board_config]);
 
 	spec->stream_name_analog = "ALC269 Analog";
-	spec->stream_analog_playback = &alc269_pcm_analog_playback;
-	spec->stream_analog_capture = &alc269_pcm_analog_capture;
-
+	if (codec->subsystem_id == 0x17aa3bf8) {
+		/* Due to a hardware problem on Lenovo Ideadpad, we need to
+		 * fix the sample rate of analog I/O to 44.1kHz
+		 */
+		spec->stream_analog_playback = &alc269_44k_pcm_analog_playback;
+		spec->stream_analog_capture = &alc269_44k_pcm_analog_capture;
+	} else {
+		spec->stream_analog_playback = &alc269_pcm_analog_playback;
+		spec->stream_analog_capture = &alc269_pcm_analog_capture;
+	}
 	spec->stream_name_digital = "ALC269 Digital";
 	spec->stream_digital_playback = &alc269_pcm_digital_playback;
 	spec->stream_digital_capture = &alc269_pcm_digital_capture;
@@ -12756,6 +13127,7 @@ static int patch_alc269(struct hda_codec *codec)
 	spec->capsrc_nids = alc269_capsrc_nids;
 	if (!spec->cap_mixer)
 		set_capture_mixer(spec);
+	set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
 
 	codec->patch_ops = alc_patch_ops;
 	if (board_config == ALC269_AUTO)
@@ -13006,8 +13378,6 @@ static struct snd_kcontrol_new alc861_asus_mixer[] = {
 static struct snd_kcontrol_new alc861_asus_laptop_mixer[] = {
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x23, 0x0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("PC Beep Playback Switch", 0x23, 0x0, HDA_OUTPUT),
 	{ }
 };
 
@@ -13481,7 +13851,7 @@ static int alc861_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin)
 static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec,
 						const struct auto_pin_cfg *cfg)
 {
-	struct hda_input_mux *imux = &spec->private_imux;
+	struct hda_input_mux *imux = &spec->private_imux[0];
 	int i, err, idx, idx1;
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
@@ -13568,12 +13938,8 @@ static void alc861_auto_init_analog_input(struct hda_codec *codec)
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
-		if (nid >= 0x0c && nid <= 0x11) {
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_PIN_WIDGET_CONTROL,
-					    i <= AUTO_PIN_FRONT_MIC ?
-					    PIN_VREF80 : PIN_IN);
-		}
+		if (nid >= 0x0c && nid <= 0x11)
+			alc_set_input_pin(codec, nid, i);
 	}
 }
 
@@ -13609,7 +13975,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = ALC861_DIGOUT_NID;
 
 	if (spec->kctls.list)
@@ -13618,13 +13984,12 @@ static int alc861_parse_auto_config(struct hda_codec *codec)
 	add_verb(spec, alc861_auto_init_verbs);
 
 	spec->num_mux_defs = 1;
-	spec->input_mux = &spec->private_imux;
+	spec->input_mux = &spec->private_imux[0];
 
 	spec->adc_nids = alc861_adc_nids;
 	spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids);
 	set_capture_mixer(spec);
 
-	store_pin_configs(codec);
 	return 1;
 }
 
@@ -13833,6 +14198,12 @@ static int patch_alc861(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x23);
+	if (err < 0) {
+		alc_free(codec);
+		return err;
+	}
+
 	if (board_config != ALC861_AUTO)
 		setup_preset(spec, &alc861_presets[board_config]);
 
@@ -13844,6 +14215,8 @@ static int patch_alc861(struct hda_codec *codec)
 	spec->stream_digital_playback = &alc861_pcm_digital_playback;
 	spec->stream_digital_capture = &alc861_pcm_digital_capture;
 
+	set_beep_amp(spec, 0x23, 0, HDA_OUTPUT);
+
 	spec->vmaster_nid = 0x03;
 
 	codec->patch_ops = alc_patch_ops;
@@ -14000,9 +14373,6 @@ static struct snd_kcontrol_new alc861vd_6st_mixer[] = {
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
 
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
-
 	{ } /* end */
 };
 
@@ -14026,9 +14396,6 @@ static struct snd_kcontrol_new alc861vd_3st_mixer[] = {
 	HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT),
 	HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT),
 
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
-
 	{ } /* end */
 };
 
@@ -14067,8 +14434,6 @@ static struct snd_kcontrol_new alc861vd_dallas_mixer[] = {
 	HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Beep Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Beep Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -14379,9 +14744,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO),
 	SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS),
 	SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG),
-	SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO),
-	SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO),
-	SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 N200", ALC861VD_LENOVO),
+	SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", ALC861VD_LENOVO),
 	SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG),
 	{}
 };
@@ -14543,11 +14906,9 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec)
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
 		if (alc861vd_is_input_pin(nid)) {
-			snd_hda_codec_write(codec, nid, 0,
-					AC_VERB_SET_PIN_WIDGET_CONTROL,
-					i <= AUTO_PIN_FRONT_MIC ?
-							PIN_VREF80 : PIN_IN);
-			if (nid != ALC861VD_PIN_CD_NID)
+			alc_set_input_pin(codec, nid, i);
+			if (nid != ALC861VD_PIN_CD_NID &&
+			    (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
 				snd_hda_codec_write(codec, nid, 0,
 						AC_VERB_SET_AMP_GAIN_MUTE,
 						AMP_OUT_MUTE);
@@ -14713,7 +15074,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = ALC861VD_DIGOUT_NID;
 
 	if (spec->kctls.list)
@@ -14722,13 +15083,12 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec)
 	add_verb(spec, alc861vd_volume_init_verbs);
 
 	spec->num_mux_defs = 1;
-	spec->input_mux = &spec->private_imux;
+	spec->input_mux = &spec->private_imux[0];
 
 	err = alc_auto_add_mic_boost(codec);
 	if (err < 0)
 		return err;
 
-	store_pin_configs(codec);
 	return 1;
 }
 
@@ -14779,6 +15139,12 @@ static int patch_alc861vd(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x23);
+	if (err < 0) {
+		alc_free(codec);
+		return err;
+	}
+
 	if (board_config != ALC861VD_AUTO)
 		setup_preset(spec, &alc861vd_presets[board_config]);
 
@@ -14801,9 +15167,10 @@ static int patch_alc861vd(struct hda_codec *codec)
 	spec->adc_nids = alc861vd_adc_nids;
 	spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids);
 	spec->capsrc_nids = alc861vd_capsrc_nids;
-	spec->is_mix_capture = 1;
+	spec->capture_style = CAPT_MIX;
 
 	set_capture_mixer(spec);
+	set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
 	spec->vmaster_nid = 0x02;
 
@@ -14992,8 +15359,6 @@ static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = {
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -15015,8 +15380,6 @@ static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = {
 	HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT),
 	HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT),
 	HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT),
-	HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT),
-	HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -15992,56 +16355,55 @@ static const char *alc662_models[ALC662_MODEL_LAST] = {
 };
 
 static struct snd_pci_quirk alc662_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA),
-	SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V),
-	SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG),
-	SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701),
-	SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20),
-	SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS),
 	SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3),
 	SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2),
 	SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1),
-	SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2),
 	SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6),
+	SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6),
+	SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2),
 	SND_PCI_QUIRK(0x1043, 0x1813, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5),
+	SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6),
+	SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2),
 	SND_PCI_QUIRK(0x1043, 0x1864, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2),
-	SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA),
+	/*SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1),*/
 	SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC663_ASUS_MODE3),
-	SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3),
-	SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3),
 	SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC663_ASUS_MODE3),
-	SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3),
+	SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V),
+	/*SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1),*/
+	SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1),
+	SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2),
+	SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1),
 	SND_PCI_QUIRK(0x1043, 0x19f3, "ASUS NB", ALC663_ASUS_MODE4),
-	SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5),
-	SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6),
-	SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6),
-	SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6),
+	SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG),
+	SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701),
+	SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20),
+	SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
 	SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K",
 		      ALC662_3ST_6ch_DIG),
-	SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E),
-	SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS),
-	SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS),
 	SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L",
 		      ALC662_3ST_6ch_DIG),
 	SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG),
+	SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E),
 	SND_PCI_QUIRK(0x1849, 0x3662, "ASROCK K10N78FullHD-hSLI R3.0",
 					ALC662_3ST_6ch_DIG),
-	SND_PCI_QUIRK(0x1854, 0x2000, "ASUS H13-2000", ALC663_ASUS_H13),
-	SND_PCI_QUIRK(0x1854, 0x2001, "ASUS H13-2001", ALC663_ASUS_H13),
-	SND_PCI_QUIRK(0x1854, 0x2002, "ASUS H13-2002", ALC663_ASUS_H13),
+	SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x",
+			   ALC663_ASUS_H13),
 	{}
 };
 
@@ -16361,7 +16723,7 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
 
 	if (alc880_is_fixed_pin(pin)) {
 		nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin));
-                /* printk("DAC nid=%x\n",nid); */
+		/* printk(KERN_DEBUG "DAC nid=%x\n",nid); */
 		/* specify the DAC as the extra output */
 		if (!spec->multiout.hp_nid)
 			spec->multiout.hp_nid = nid;
@@ -16391,26 +16753,58 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin,
 	return 0;
 }
 
+/* return the index of the src widget from the connection list of the nid.
+ * return -1 if not found
+ */
+static int alc662_input_pin_idx(struct hda_codec *codec, hda_nid_t nid,
+				hda_nid_t src)
+{
+	hda_nid_t conn_list[HDA_MAX_CONNECTIONS];
+	int i, conns;
+
+	conns = snd_hda_get_connections(codec, nid, conn_list,
+					ARRAY_SIZE(conn_list));
+	if (conns < 0)
+		return -1;
+	for (i = 0; i < conns; i++)
+		if (conn_list[i] == src)
+			return i;
+	return -1;
+}
+
+static int alc662_is_input_pin(struct hda_codec *codec, hda_nid_t nid)
+{
+	unsigned int pincap = snd_hda_query_pin_caps(codec, nid);
+	return (pincap & AC_PINCAP_IN) != 0;
+}
+
 /* create playback/capture controls for input pins */
-static int alc662_auto_create_analog_input_ctls(struct alc_spec *spec,
+static int alc662_auto_create_analog_input_ctls(struct hda_codec *codec,
 						const struct auto_pin_cfg *cfg)
 {
-	struct hda_input_mux *imux = &spec->private_imux;
+	struct alc_spec *spec = codec->spec;
+	struct hda_input_mux *imux = &spec->private_imux[0];
 	int i, err, idx;
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
-		if (alc880_is_input_pin(cfg->input_pins[i])) {
-			idx = alc880_input_pin_idx(cfg->input_pins[i]);
-			err = new_analog_input(spec, cfg->input_pins[i],
-					       auto_pin_cfg_labels[i],
-					       idx, 0x0b);
-			if (err < 0)
-				return err;
-			imux->items[imux->num_items].label =
-				auto_pin_cfg_labels[i];
-			imux->items[imux->num_items].index =
-				alc880_input_pin_idx(cfg->input_pins[i]);
-			imux->num_items++;
+		if (alc662_is_input_pin(codec, cfg->input_pins[i])) {
+			idx = alc662_input_pin_idx(codec, 0x0b,
+						   cfg->input_pins[i]);
+			if (idx >= 0) {
+				err = new_analog_input(spec, cfg->input_pins[i],
+						       auto_pin_cfg_labels[i],
+						       idx, 0x0b);
+				if (err < 0)
+					return err;
+			}
+			idx = alc662_input_pin_idx(codec, 0x22,
+						   cfg->input_pins[i]);
+			if (idx >= 0) {
+				imux->items[imux->num_items].label =
+					auto_pin_cfg_labels[i];
+				imux->items[imux->num_items].index = idx;
+				imux->num_items++;
+			}
 		}
 	}
 	return 0;
@@ -16460,7 +16854,6 @@ static void alc662_auto_init_hp_out(struct hda_codec *codec)
 		alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0);
 }
 
-#define alc662_is_input_pin(nid)	alc880_is_input_pin(nid)
 #define ALC662_PIN_CD_NID		ALC880_PIN_CD_NID
 
 static void alc662_auto_init_analog_input(struct hda_codec *codec)
@@ -16470,12 +16863,10 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec)
 
 	for (i = 0; i < AUTO_PIN_LAST; i++) {
 		hda_nid_t nid = spec->autocfg.input_pins[i];
-		if (alc662_is_input_pin(nid)) {
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_PIN_WIDGET_CONTROL,
-					    (i <= AUTO_PIN_FRONT_MIC ?
-					     PIN_VREF80 : PIN_IN));
-			if (nid != ALC662_PIN_CD_NID)
+		if (alc662_is_input_pin(codec, nid)) {
+			alc_set_input_pin(codec, nid, i);
+			if (nid != ALC662_PIN_CD_NID &&
+			    (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP))
 				snd_hda_codec_write(codec, nid, 0,
 						    AC_VERB_SET_AMP_GAIN_MUTE,
 						    AMP_OUT_MUTE);
@@ -16513,20 +16904,20 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 					   "Headphone");
 	if (err < 0)
 		return err;
-	err = alc662_auto_create_analog_input_ctls(spec, &spec->autocfg);
+	err = alc662_auto_create_analog_input_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = ALC880_DIGOUT_NID;
 
 	if (spec->kctls.list)
 		add_mixer(spec, spec->kctls.list);
 
 	spec->num_mux_defs = 1;
-	spec->input_mux = &spec->private_imux;
+	spec->input_mux = &spec->private_imux[0];
 
 	add_verb(spec, alc662_auto_init_verbs);
 	if (codec->vendor_id == 0x10ec0663)
@@ -16536,7 +16927,6 @@ static int alc662_parse_auto_config(struct hda_codec *codec)
 	if (err < 0)
 		return err;
 
-	store_pin_configs(codec);
 	return 1;
 }
 
@@ -16588,6 +16978,12 @@ static int patch_alc662(struct hda_codec *codec)
 		}
 	}
 
+	err = snd_hda_attach_beep_device(codec, 0x1);
+	if (err < 0) {
+		alc_free(codec);
+		return err;
+	}
+
 	if (board_config != ALC662_AUTO)
 		setup_preset(spec, &alc662_presets[board_config]);
 
@@ -16611,10 +17007,14 @@ static int patch_alc662(struct hda_codec *codec)
 	spec->adc_nids = alc662_adc_nids;
 	spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids);
 	spec->capsrc_nids = alc662_capsrc_nids;
-	spec->is_mix_capture = 1;
+	spec->capture_style = CAPT_MIX;
 
 	if (!spec->cap_mixer)
 		set_capture_mixer(spec);
+	if (codec->vendor_id == 0x10ec0662)
+		set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
+	else
+		set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
 
 	spec->vmaster_nid = 0x02;
 
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 6094344fb223..b5e108aa8f63 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -43,6 +43,7 @@ enum {
 };
 
 enum {
+	STAC_AUTO,
 	STAC_REF,
 	STAC_9200_OQO,
 	STAC_9200_DELL_D21,
@@ -62,14 +63,17 @@ enum {
 };
 
 enum {
+	STAC_9205_AUTO,
 	STAC_9205_REF,
 	STAC_9205_DELL_M42,
 	STAC_9205_DELL_M43,
 	STAC_9205_DELL_M44,
+	STAC_9205_EAPD,
 	STAC_9205_MODELS
 };
 
 enum {
+	STAC_92HD73XX_AUTO,
 	STAC_92HD73XX_NO_JD, /* no jack-detection */
 	STAC_92HD73XX_REF,
 	STAC_DELL_M6_AMIC,
@@ -80,22 +84,27 @@ enum {
 };
 
 enum {
+	STAC_92HD83XXX_AUTO,
 	STAC_92HD83XXX_REF,
 	STAC_92HD83XXX_PWR_REF,
+	STAC_DELL_S14,
 	STAC_92HD83XXX_MODELS
 };
 
 enum {
+	STAC_92HD71BXX_AUTO,
 	STAC_92HD71BXX_REF,
 	STAC_DELL_M4_1,
 	STAC_DELL_M4_2,
 	STAC_DELL_M4_3,
 	STAC_HP_M4,
 	STAC_HP_DV5,
+	STAC_HP_HDX,
 	STAC_92HD71BXX_MODELS
 };
 
 enum {
+	STAC_925x_AUTO,
 	STAC_925x_REF,
 	STAC_M1,
 	STAC_M1_2,
@@ -108,6 +117,7 @@ enum {
 };
 
 enum {
+	STAC_922X_AUTO,
 	STAC_D945_REF,
 	STAC_D945GTP3,
 	STAC_D945GTP5,
@@ -135,6 +145,7 @@ enum {
 };
 
 enum {
+	STAC_927X_AUTO,
 	STAC_D965_REF_NO_JD, /* no jack-detection */
 	STAC_D965_REF,
 	STAC_D965_3ST,
@@ -144,6 +155,12 @@ enum {
 	STAC_927X_MODELS
 };
 
+enum {
+	STAC_9872_AUTO,
+	STAC_9872_VAIO,
+	STAC_9872_MODELS
+};
+
 struct sigmatel_event {
 	hda_nid_t nid;
 	unsigned char type;
@@ -167,6 +184,7 @@ struct sigmatel_spec {
 	unsigned int alt_switch: 1;
 	unsigned int hp_detect: 1;
 	unsigned int spdif_mute: 1;
+	unsigned int check_volume_offset:1;
 
 	/* gpio lines */
 	unsigned int eapd_mask;
@@ -179,6 +197,7 @@ struct sigmatel_spec {
 	unsigned int stream_delay;
 
 	/* analog loopback */
+	struct snd_kcontrol_new *aloopback_ctl;
 	unsigned char aloopback_mask;
 	unsigned char aloopback_shift;
 
@@ -203,6 +222,8 @@ struct sigmatel_spec {
 	hda_nid_t hp_dacs[5];
 	hda_nid_t speaker_dacs[5];
 
+	int volume_offset;
+
 	/* capture */
 	hda_nid_t *adc_nids;
 	unsigned int num_adcs;
@@ -224,7 +245,6 @@ struct sigmatel_spec {
 	/* pin widgets */
 	hda_nid_t *pin_nids;
 	unsigned int num_pins;
-	unsigned int *pin_configs;
 
 	/* codec specific stuff */
 	struct hda_verb *init;
@@ -400,6 +420,10 @@ static hda_nid_t stac922x_mux_nids[2] = {
         0x12, 0x13,
 };
 
+static hda_nid_t stac927x_slave_dig_outs[2] = {
+	0x1f, 0,
+};
+
 static hda_nid_t stac927x_adc_nids[3] = {
         0x07, 0x08, 0x09
 };
@@ -472,15 +496,21 @@ static hda_nid_t stac92hd73xx_pin_nids[13] = {
 	0x14, 0x22, 0x23
 };
 
-static hda_nid_t stac92hd83xxx_pin_nids[14] = {
+static hda_nid_t stac92hd83xxx_pin_nids[10] = {
 	0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
-	0x0f, 0x10, 0x11, 0x12, 0x13,
-	0x1d, 0x1e, 0x1f, 0x20
+	0x0f, 0x10, 0x11, 0x1f, 0x20,
+};
+
+#define STAC92HD71BXX_NUM_PINS 13
+static hda_nid_t stac92hd71bxx_pin_nids_4port[STAC92HD71BXX_NUM_PINS] = {
+	0x0a, 0x0b, 0x0c, 0x0d, 0x00,
+	0x00, 0x14, 0x18, 0x19, 0x1e,
+	0x1f, 0x20, 0x27
 };
-static hda_nid_t stac92hd71bxx_pin_nids[11] = {
+static hda_nid_t stac92hd71bxx_pin_nids_6port[STAC92HD71BXX_NUM_PINS] = {
 	0x0a, 0x0b, 0x0c, 0x0d, 0x0e,
 	0x0f, 0x14, 0x18, 0x19, 0x1e,
-	0x1f,
+	0x1f, 0x20, 0x27
 };
 
 static hda_nid_t stac927x_pin_nids[14] = {
@@ -842,9 +872,9 @@ static struct hda_verb stac92hd73xx_10ch_core_init[] = {
 };
 
 static struct hda_verb stac92hd83xxx_core_init[] = {
-	{ 0xa, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{ 0xb, AC_VERB_SET_CONNECT_SEL, 0x0},
-	{ 0xd, AC_VERB_SET_CONNECT_SEL, 0x1},
+	{ 0xa, AC_VERB_SET_CONNECT_SEL, 0x1},
+	{ 0xb, AC_VERB_SET_CONNECT_SEL, 0x1},
+	{ 0xd, AC_VERB_SET_CONNECT_SEL, 0x0},
 
 	/* power state controls amps */
 	{ 0x01, AC_VERB_SET_EAPD, 1 << 2},
@@ -854,26 +884,25 @@ static struct hda_verb stac92hd83xxx_core_init[] = {
 static struct hda_verb stac92hd71bxx_core_init[] = {
 	/* set master volume and direct control */
 	{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
-	/* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */
-	{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
-	{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{}
 };
 
-#define HD_DISABLE_PORTF 2
+#define HD_DISABLE_PORTF 1
 static struct hda_verb stac92hd71bxx_analog_core_init[] = {
 	/* start of config #1 */
 
 	/* connect port 0f to audio mixer */
 	{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2},
-	/* unmute right and left channels for node 0x0f */
-	{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	/* start of config #2 */
 
 	/* set master volume and direct control */
 	{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
-	/* unmute right and left channels for nodes 0x0a, 0xd */
+	{}
+};
+
+static struct hda_verb stac92hd71bxx_unmute_core_init[] = {
+	/* unmute right and left channels for nodes 0x0f, 0xa, 0x0d */
+	{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{}
@@ -954,16 +983,6 @@ static struct hda_verb stac9205_core_init[] = {
 		.private_value = HDA_COMPOSE_AMP_VAL(nid, chs, idx, dir) \
 	}
 
-#define STAC_INPUT_SOURCE(cnt) \
-	{ \
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
-		.name = "Input Source", \
-		.count = cnt, \
-		.info = stac92xx_mux_enum_info, \
-		.get = stac92xx_mux_enum_get, \
-		.put = stac92xx_mux_enum_put, \
-	}
-
 #define STAC_ANALOG_LOOPBACK(verb_read, verb_write, cnt) \
 	{ \
 		.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
@@ -978,7 +997,6 @@ static struct hda_verb stac9205_core_init[] = {
 static struct snd_kcontrol_new stac9200_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT),
-	STAC_INPUT_SOURCE(1),
 	HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT),
 	{ } /* end */
@@ -1003,8 +1021,6 @@ static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = {
 	HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT),
 	HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT),
 
-	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3),
-
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT),
 
@@ -1014,9 +1030,22 @@ static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = {
+static struct snd_kcontrol_new stac92hd73xx_6ch_loopback[] = {
+	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3),
+	{}
+};
+
+static struct snd_kcontrol_new stac92hd73xx_8ch_loopback[] = {
 	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4),
+	{}
+};
 
+static struct snd_kcontrol_new stac92hd73xx_10ch_loopback[] = {
+	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5),
+	{}
+};
+
+static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = {
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT),
 
@@ -1041,8 +1070,6 @@ static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = {
 };
 
 static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = {
-	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5),
-
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT),
 
@@ -1094,9 +1121,6 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = {
 };
 
 static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
-	STAC_INPUT_SOURCE(2),
-	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2),
-
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT),
 
@@ -1122,10 +1146,11 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
 	{ } /* end */
 };
 
-static struct snd_kcontrol_new stac92hd71bxx_mixer[] = {
-	STAC_INPUT_SOURCE(2),
-	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2),
+static struct snd_kcontrol_new stac92hd71bxx_loopback[] = {
+	STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2)
+};
 
+static struct snd_kcontrol_new stac92hd71bxx_mixer[] = {
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT),
 
@@ -1137,16 +1162,12 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = {
 static struct snd_kcontrol_new stac925x_mixer[] = {
 	HDA_CODEC_VOLUME("Master Playback Volume", 0x0e, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT),
-	STAC_INPUT_SOURCE(1),
 	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT),
 	{ } /* end */
 };
 
 static struct snd_kcontrol_new stac9205_mixer[] = {
-	STAC_INPUT_SOURCE(2),
-	STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1),
-
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT),
 
@@ -1155,9 +1176,13 @@ static struct snd_kcontrol_new stac9205_mixer[] = {
 	{ } /* end */
 };
 
+static struct snd_kcontrol_new stac9205_loopback[] = {
+	STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1),
+	{}
+};
+
 /* This needs to be generated dynamically based on sequence */
 static struct snd_kcontrol_new stac922x_mixer[] = {
-	STAC_INPUT_SOURCE(2),
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_INPUT),
 
@@ -1168,9 +1193,6 @@ static struct snd_kcontrol_new stac922x_mixer[] = {
 
 
 static struct snd_kcontrol_new stac927x_mixer[] = {
-	STAC_INPUT_SOURCE(3),
-	STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1),
-
 	HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT),
 	HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1b, 0x0, HDA_OUTPUT),
 
@@ -1182,6 +1204,11 @@ static struct snd_kcontrol_new stac927x_mixer[] = {
 	{ } /* end */
 };
 
+static struct snd_kcontrol_new stac927x_loopback[] = {
+	STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1),
+	{}
+};
+
 static struct snd_kcontrol_new stac_dmux_mixer = {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
 	.name = "Digital Input Source",
@@ -1207,10 +1234,7 @@ static const char *slave_vols[] = {
 	"LFE Playback Volume",
 	"Side Playback Volume",
 	"Headphone Playback Volume",
-	"Headphone2 Playback Volume",
 	"Speaker Playback Volume",
-	"External Speaker Playback Volume",
-	"Speaker2 Playback Volume",
 	NULL
 };
 
@@ -1221,10 +1245,7 @@ static const char *slave_sws[] = {
 	"LFE Playback Switch",
 	"Side Playback Switch",
 	"Headphone Playback Switch",
-	"Headphone2 Playback Switch",
 	"Speaker Playback Switch",
-	"External Speaker Playback Switch",
-	"Speaker2 Playback Switch",
 	"IEC958 Playback Switch",
 	NULL
 };
@@ -1294,6 +1315,8 @@ static int stac92xx_build_controls(struct hda_codec *codec)
 		unsigned int vmaster_tlv[4];
 		snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0],
 					HDA_OUTPUT, vmaster_tlv);
+		/* correct volume offset */
+		vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset;
 		err = snd_hda_add_vmaster(codec, "Master Playback Volume",
 					  vmaster_tlv, slave_vols);
 		if (err < 0)
@@ -1306,6 +1329,13 @@ static int stac92xx_build_controls(struct hda_codec *codec)
 			return err;
 	}
 
+	if (spec->aloopback_ctl &&
+	    snd_hda_get_bool_hint(codec, "loopback") == 1) {
+		err = snd_hda_add_new_ctls(codec, spec->aloopback_ctl);
+		if (err < 0)
+			return err;
+	}
+
 	stac92xx_free_kctls(codec); /* no longer needed */
 
 	/* create jack input elements */
@@ -1490,6 +1520,7 @@ static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = {
 };
 
 static const char *stac9200_models[STAC_9200_MODELS] = {
+	[STAC_AUTO] = "auto",
 	[STAC_REF] = "ref",
 	[STAC_9200_OQO] = "oqo",
 	[STAC_9200_DELL_D21] = "dell-d21",
@@ -1511,6 +1542,8 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
+		      "DFI LanParty", STAC_REF),
 	/* Dell laptops have BIOS problem */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01a8,
 		      "unknown Dell", STAC_9200_DELL_D21),
@@ -1633,6 +1666,7 @@ static unsigned int *stac925x_brd_tbl[STAC_925x_MODELS] = {
 };
 
 static const char *stac925x_models[STAC_925x_MODELS] = {
+	[STAC_925x_AUTO] = "auto",
 	[STAC_REF] = "ref",
 	[STAC_M1] = "m1",
 	[STAC_M1_2] = "m1-2",
@@ -1660,6 +1694,7 @@ static struct snd_pci_quirk stac925x_codec_id_cfg_tbl[] = {
 static struct snd_pci_quirk stac925x_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_REF),
 	SND_PCI_QUIRK(0x8384, 0x7632, "Stac9202 Reference Board", STAC_REF),
 
 	/* Default table for unknown ID */
@@ -1691,6 +1726,7 @@ static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = {
 };
 
 static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = {
+	[STAC_92HD73XX_AUTO] = "auto",
 	[STAC_92HD73XX_NO_JD] = "no-jd",
 	[STAC_92HD73XX_REF] = "ref",
 	[STAC_DELL_M6_AMIC] = "dell-m6-amic",
@@ -1703,6 +1739,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 				"DFI LanParty", STAC_92HD73XX_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
+				"DFI LanParty", STAC_92HD73XX_REF),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0254,
 				"Dell Studio 1535", STAC_DELL_M6_DMIC),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0255,
@@ -1726,52 +1764,68 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = {
 	{} /* terminator */
 };
 
-static unsigned int ref92hd83xxx_pin_configs[14] = {
+static unsigned int ref92hd83xxx_pin_configs[10] = {
 	0x02214030, 0x02211010, 0x02a19020, 0x02170130,
 	0x01014050, 0x01819040, 0x01014020, 0x90a3014e,
-	0x40f000f0, 0x40f000f0, 0x40f000f0, 0x40f000f0,
 	0x01451160, 0x98560170,
 };
 
+static unsigned int dell_s14_pin_configs[10] = {
+	0x02214030, 0x02211010, 0x02a19020, 0x01014050,
+	0x40f000f0, 0x01819040, 0x40f000f0, 0x90a60160,
+	0x40f000f0, 0x40f000f0,
+};
+
 static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = {
 	[STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs,
 	[STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs,
+	[STAC_DELL_S14] = dell_s14_pin_configs,
 };
 
 static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = {
+	[STAC_92HD83XXX_AUTO] = "auto",
 	[STAC_92HD83XXX_REF] = "ref",
 	[STAC_92HD83XXX_PWR_REF] = "mic-ref",
+	[STAC_DELL_S14] = "dell-s14",
 };
 
 static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_92HD83XXX_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
+		      "DFI LanParty", STAC_92HD83XXX_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ba,
+		      "unknown Dell", STAC_DELL_S14),
 	{} /* terminator */
 };
 
-static unsigned int ref92hd71bxx_pin_configs[11] = {
+static unsigned int ref92hd71bxx_pin_configs[STAC92HD71BXX_NUM_PINS] = {
 	0x02214030, 0x02a19040, 0x01a19020, 0x01014010,
 	0x0181302e, 0x01014010, 0x01019020, 0x90a000f0,
-	0x90a000f0, 0x01452050, 0x01452050,
+	0x90a000f0, 0x01452050, 0x01452050, 0x00000000,
+	0x00000000
 };
 
-static unsigned int dell_m4_1_pin_configs[11] = {
+static unsigned int dell_m4_1_pin_configs[STAC92HD71BXX_NUM_PINS] = {
 	0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110,
 	0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0,
-	0x40f000f0, 0x4f0000f0, 0x4f0000f0,
+	0x40f000f0, 0x4f0000f0, 0x4f0000f0, 0x00000000,
+	0x00000000
 };
 
-static unsigned int dell_m4_2_pin_configs[11] = {
+static unsigned int dell_m4_2_pin_configs[STAC92HD71BXX_NUM_PINS] = {
 	0x0421101f, 0x04a11221, 0x90a70330, 0x90170110,
 	0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0,
-	0x40f000f0, 0x044413b0, 0x044413b0,
+	0x40f000f0, 0x044413b0, 0x044413b0, 0x00000000,
+	0x00000000
 };
 
-static unsigned int dell_m4_3_pin_configs[11] = {
+static unsigned int dell_m4_3_pin_configs[STAC92HD71BXX_NUM_PINS] = {
 	0x0421101f, 0x04a11221, 0x90a70330, 0x90170110,
 	0x40f000f0, 0x40f000f0, 0x40f000f0, 0x90a000f0,
-	0x40f000f0, 0x044413b0, 0x044413b0,
+	0x40f000f0, 0x044413b0, 0x044413b0, 0x00000000,
+	0x00000000
 };
 
 static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = {
@@ -1781,35 +1835,38 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = {
 	[STAC_DELL_M4_3]	= dell_m4_3_pin_configs,
 	[STAC_HP_M4]		= NULL,
 	[STAC_HP_DV5]		= NULL,
+	[STAC_HP_HDX]           = NULL,
 };
 
 static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = {
+	[STAC_92HD71BXX_AUTO] = "auto",
 	[STAC_92HD71BXX_REF] = "ref",
 	[STAC_DELL_M4_1] = "dell-m4-1",
 	[STAC_DELL_M4_2] = "dell-m4-2",
 	[STAC_DELL_M4_3] = "dell-m4-3",
 	[STAC_HP_M4] = "hp-m4",
 	[STAC_HP_DV5] = "hp-dv5",
+	[STAC_HP_HDX] = "hp-hdx",
 };
 
 static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_92HD71BXX_REF),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f2,
-		      "HP dv5", STAC_HP_M4),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4,
-		      "HP dv7", STAC_HP_DV5),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f7,
-		      "HP dv4", STAC_HP_DV5),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc,
-		      "HP dv7", STAC_HP_M4),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3600,
-		      "HP dv5", STAC_HP_DV5),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3603,
-		      "HP dv5", STAC_HP_DV5),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
+		      "DFI LanParty", STAC_92HD71BXX_REF),
+	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080,
+		      "HP", STAC_HP_DV5),
+	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0,
+		      "HP dv4-7", STAC_HP_DV5),
+	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3600,
+		      "HP dv4-7", STAC_HP_DV5),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3610,
+		      "HP HDX", STAC_HP_HDX),  /* HDX18 */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a,
-				"unknown HP", STAC_HP_M4),
+		      "HP mini 1000", STAC_HP_M4),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361b,
+		      "HP HDX", STAC_HP_HDX),  /* HDX16 */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233,
 				"unknown Dell", STAC_DELL_M4_1),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234,
@@ -1961,6 +2018,7 @@ static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = {
 };
 
 static const char *stac922x_models[STAC_922X_MODELS] = {
+	[STAC_922X_AUTO] = "auto",
 	[STAC_D945_REF]	= "ref",
 	[STAC_D945GTP5]	= "5stack",
 	[STAC_D945GTP3]	= "3stack",
@@ -1988,6 +2046,8 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_D945_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
+		      "DFI LanParty", STAC_D945_REF),
 	/* Intel 945G based systems */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0101,
 		      "Intel D945G", STAC_D945GTP3),
@@ -2041,6 +2101,9 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = {
 		      "Intel D945P", STAC_D945GTP3),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0707,
 		      "Intel D945P", STAC_D945GTP5),
+	/* other intel */
+	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0204,
+		      "Intel D945", STAC_D945_REF),
 	/* other systems  */
 	/* Apple Intel Mac (Mac Mini, MacBook, MacBook Pro...) */
 	SND_PCI_QUIRK(0x8384, 0x7680,
@@ -2065,31 +2128,7 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d7,
 		      "Dell XPS M1210", STAC_922X_DELL_M82),
 	/* ECS/PC Chips boards */
-	SND_PCI_QUIRK(0x1019, 0x2144,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2608,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2633,
-		      "ECS/PC chips P17G/1333", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2811,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2812,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2813,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2814,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2815,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2816,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2817,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2818,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2819,
-		      "ECS/PC chips", STAC_ECS_202),
-	SND_PCI_QUIRK(0x1019, 0x2820,
+	SND_PCI_QUIRK_MASK(0x1019, 0xf000, 0x2000,
 		      "ECS/PC chips", STAC_ECS_202),
 	{} /* terminator */
 };
@@ -2132,6 +2171,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = {
 };
 
 static const char *stac927x_models[STAC_927X_MODELS] = {
+	[STAC_927X_AUTO]	= "auto",
 	[STAC_D965_REF_NO_JD]	= "ref-no-jd",
 	[STAC_D965_REF]		= "ref",
 	[STAC_D965_3ST]		= "3stack",
@@ -2144,26 +2184,16 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_D965_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
+		      "DFI LanParty", STAC_D965_REF),
 	 /* Intel 946 based systems */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x3d01, "Intel D946", STAC_D965_3ST),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xa301, "Intel D946", STAC_D965_3ST),
 	/* 965 based 3 stack systems */
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2116, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2115, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2114, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2113, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2112, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2111, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2110, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2009, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2008, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2007, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2006, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2005, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2004, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2003, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2002, "Intel D965", STAC_D965_3ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2001, "Intel D965", STAC_D965_3ST),
+	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2100,
+			   "Intel D965", STAC_D965_3ST),
+	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2000,
+			   "Intel D965", STAC_D965_3ST),
 	/* Dell 3 stack systems */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01f7, "Dell XPS M1730", STAC_DELL_3ST),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01dd, "Dell Dimension E520", STAC_DELL_3ST),
@@ -2179,15 +2209,10 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = {
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x02ff, "Dell     ", STAC_DELL_BIOS),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x0209, "Dell XPS 1330", STAC_DELL_BIOS),
 	/* 965 based 5 stack systems */
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2301, "Intel D965", STAC_D965_5ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2302, "Intel D965", STAC_D965_5ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2303, "Intel D965", STAC_D965_5ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2304, "Intel D965", STAC_D965_5ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2305, "Intel D965", STAC_D965_5ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2501, "Intel D965", STAC_D965_5ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2502, "Intel D965", STAC_D965_5ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2503, "Intel D965", STAC_D965_5ST),
-	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2504, "Intel D965", STAC_D965_5ST),
+	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2300,
+			   "Intel D965", STAC_D965_5ST),
+	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2500,
+			   "Intel D965", STAC_D965_5ST),
 	{} /* terminator */
 };
 
@@ -2240,19 +2265,25 @@ static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = {
 	[STAC_9205_DELL_M42] = dell_9205_m42_pin_configs,
 	[STAC_9205_DELL_M43] = dell_9205_m43_pin_configs,
 	[STAC_9205_DELL_M44] = dell_9205_m44_pin_configs,
+	[STAC_9205_EAPD] = NULL,
 };
 
 static const char *stac9205_models[STAC_9205_MODELS] = {
+	[STAC_9205_AUTO] = "auto",
 	[STAC_9205_REF] = "ref",
 	[STAC_9205_DELL_M42] = "dell-m42",
 	[STAC_9205_DELL_M43] = "dell-m43",
 	[STAC_9205_DELL_M44] = "dell-m44",
+	[STAC_9205_EAPD] = "eapd",
 };
 
 static struct snd_pci_quirk stac9205_cfg_tbl[] = {
 	/* SigmaTel reference board */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668,
 		      "DFI LanParty", STAC_9205_REF),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101,
+		      "DFI LanParty", STAC_9205_REF),
+	/* Dell */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f1,
 		      "unknown Dell", STAC_9205_DELL_M42),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f2,
@@ -2283,101 +2314,24 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = {
 		      "Dell Inspiron", STAC_9205_DELL_M44),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228,
 		      "Dell Vostro 1500", STAC_9205_DELL_M42),
+	/* Gateway */
+	SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD),
 	{} /* terminator */
 };
 
-static int stac92xx_save_bios_config_regs(struct hda_codec *codec)
+static void stac92xx_set_config_regs(struct hda_codec *codec,
+				     unsigned int *pincfgs)
 {
 	int i;
 	struct sigmatel_spec *spec = codec->spec;
-	
-	kfree(spec->pin_configs);
-	spec->pin_configs = kcalloc(spec->num_pins, sizeof(*spec->pin_configs),
-				    GFP_KERNEL);
-	if (!spec->pin_configs)
-		return -ENOMEM;
-	
-	for (i = 0; i < spec->num_pins; i++) {
-		hda_nid_t nid = spec->pin_nids[i];
-		unsigned int pin_cfg;
-		
-		pin_cfg = snd_hda_codec_read(codec, nid, 0, 
-			AC_VERB_GET_CONFIG_DEFAULT, 0x00);	
-		snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x bios pin config %8.8x\n",
-					nid, pin_cfg);
-		spec->pin_configs[i] = pin_cfg;
-	}
-	
-	return 0;
-}
 
-static void stac92xx_set_config_reg(struct hda_codec *codec,
-				    hda_nid_t pin_nid, unsigned int pin_config)
-{
-	int i;
-	snd_hda_codec_write(codec, pin_nid, 0,
-			    AC_VERB_SET_CONFIG_DEFAULT_BYTES_0,
-			    pin_config & 0x000000ff);
-	snd_hda_codec_write(codec, pin_nid, 0,
-			    AC_VERB_SET_CONFIG_DEFAULT_BYTES_1,
-			    (pin_config & 0x0000ff00) >> 8);
-	snd_hda_codec_write(codec, pin_nid, 0,
-			    AC_VERB_SET_CONFIG_DEFAULT_BYTES_2,
-			    (pin_config & 0x00ff0000) >> 16);
-	snd_hda_codec_write(codec, pin_nid, 0,
-			    AC_VERB_SET_CONFIG_DEFAULT_BYTES_3,
-			    pin_config >> 24);
-	i = snd_hda_codec_read(codec, pin_nid, 0,
-			       AC_VERB_GET_CONFIG_DEFAULT,
-			       0x00);	
-	snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n",
-		    pin_nid, i);
-}
-
-static void stac92xx_set_config_regs(struct hda_codec *codec)
-{
-	int i;
-	struct sigmatel_spec *spec = codec->spec;
-
- 	if (!spec->pin_configs)
- 		return;
+	if (!pincfgs)
+		return;
 
 	for (i = 0; i < spec->num_pins; i++)
-		stac92xx_set_config_reg(codec, spec->pin_nids[i],
-					spec->pin_configs[i]);
-}
-
-static int stac_save_pin_cfgs(struct hda_codec *codec, unsigned int *pins)
-{
-	struct sigmatel_spec *spec = codec->spec;
-
-	if (!pins)
-		return stac92xx_save_bios_config_regs(codec);
-
-	kfree(spec->pin_configs);
-	spec->pin_configs = kmemdup(pins,
-				    spec->num_pins * sizeof(*pins),
-				    GFP_KERNEL);
-	if (!spec->pin_configs)
-		return -ENOMEM;
-
-	stac92xx_set_config_regs(codec);
-	return 0;
-}
-
-static void stac_change_pin_config(struct hda_codec *codec, hda_nid_t nid,
-				   unsigned int cfg)
-{
-	struct sigmatel_spec *spec = codec->spec;
-	int i;
-
-	for (i = 0; i < spec->num_pins; i++) {
-		if (spec->pin_nids[i] == nid) {
-			spec->pin_configs[i] = cfg;
-			stac92xx_set_config_reg(codec, nid, cfg);
-			break;
-		}
-	}
+		if (spec->pin_nids[i] && pincfgs[i])
+			snd_hda_codec_set_pincfg(codec, spec->pin_nids[i],
+						 pincfgs[i]);
 }
 
 /*
@@ -2567,7 +2521,7 @@ static int stac92xx_build_pcms(struct hda_codec *codec)
 		codec->num_pcms++;
 		info++;
 		info->name = "STAC92xx Digital";
-		info->pcm_type = HDA_PCM_TYPE_SPDIF;
+		info->pcm_type = spec->autocfg.dig_out_type[0];
 		if (spec->multiout.dig_out_nid) {
 			info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_digital_playback;
 			info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid;
@@ -2583,8 +2537,7 @@ static int stac92xx_build_pcms(struct hda_codec *codec)
 
 static unsigned int stac92xx_get_vref(struct hda_codec *codec, hda_nid_t nid)
 {
-	unsigned int pincap = snd_hda_param_read(codec, nid,
-						 AC_PAR_PIN_CAP);
+	unsigned int pincap = snd_hda_query_pin_caps(codec, nid);
 	pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT;
 	if (pincap & AC_PINCAP_VREF_100)
 		return AC_PINCTL_VREF_100;
@@ -2759,22 +2712,37 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = {
 };
 
 /* add dynamic controls */
-static int stac92xx_add_control_temp(struct sigmatel_spec *spec,
-				     struct snd_kcontrol_new *ktemp,
-				     int idx, const char *name,
-				     unsigned long val)
+static struct snd_kcontrol_new *
+stac_control_new(struct sigmatel_spec *spec,
+		 struct snd_kcontrol_new *ktemp,
+		 const char *name)
 {
 	struct snd_kcontrol_new *knew;
 
 	snd_array_init(&spec->kctls, sizeof(*knew), 32);
 	knew = snd_array_new(&spec->kctls);
 	if (!knew)
-		return -ENOMEM;
+		return NULL;
 	*knew = *ktemp;
-	knew->index = idx;
 	knew->name = kstrdup(name, GFP_KERNEL);
-	if (!knew->name)
+	if (!knew->name) {
+		/* roolback */
+		memset(knew, 0, sizeof(*knew));
+		spec->kctls.alloced--;
+		return NULL;
+	}
+	return knew;
+}
+
+static int stac92xx_add_control_temp(struct sigmatel_spec *spec,
+				     struct snd_kcontrol_new *ktemp,
+				     int idx, const char *name,
+				     unsigned long val)
+{
+	struct snd_kcontrol_new *knew = stac_control_new(spec, ktemp, name);
+	if (!knew)
 		return -ENOMEM;
+	knew->index = idx;
 	knew->private_value = val;
 	return 0;
 }
@@ -2796,6 +2764,29 @@ static inline int stac92xx_add_control(struct sigmatel_spec *spec, int type,
 	return stac92xx_add_control_idx(spec, type, 0, name, val);
 }
 
+static struct snd_kcontrol_new stac_input_src_temp = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Input Source",
+	.info = stac92xx_mux_enum_info,
+	.get = stac92xx_mux_enum_get,
+	.put = stac92xx_mux_enum_put,
+};
+
+static int stac92xx_add_input_source(struct sigmatel_spec *spec)
+{
+	struct snd_kcontrol_new *knew;
+	struct hda_input_mux *imux = &spec->private_imux;
+
+	if (!spec->num_adcs || imux->num_items <= 1)
+		return 0; /* no need for input source control */
+	knew = stac_control_new(spec, &stac_input_src_temp,
+				stac_input_src_temp.name);
+	if (!knew)
+		return -ENOMEM;
+	knew->count = spec->num_adcs;
+	return 0;
+}
+
 /* check whether the line-input can be used as line-out */
 static hda_nid_t check_line_out_switch(struct hda_codec *codec)
 {
@@ -2807,7 +2798,7 @@ static hda_nid_t check_line_out_switch(struct hda_codec *codec)
 	if (cfg->line_out_type != AUTO_PIN_LINE_OUT)
 		return 0;
 	nid = cfg->input_pins[AUTO_PIN_LINE];
-	pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+	pincap = snd_hda_query_pin_caps(codec, nid);
 	if (pincap & AC_PINCAP_OUT)
 		return nid;
 	return 0;
@@ -2826,12 +2817,11 @@ static hda_nid_t check_mic_out_switch(struct hda_codec *codec)
 	mic_pin = AUTO_PIN_MIC;
 	for (;;) {
 		hda_nid_t nid = cfg->input_pins[mic_pin];
-		def_conf = snd_hda_codec_read(codec, nid, 0,
-					      AC_VERB_GET_CONFIG_DEFAULT, 0);
+		def_conf = snd_hda_codec_get_pincfg(codec, nid);
 		/* some laptops have an internal analog microphone
 		 * which can't be used as a output */
 		if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) {
-			pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP);
+			pincap = snd_hda_query_pin_caps(codec, nid);
 			if (pincap & AC_PINCAP_OUT)
 				return nid;
 		}
@@ -2879,8 +2869,7 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid)
 	conn_len = snd_hda_get_connections(codec, nid, conn,
 					   HDA_MAX_CONNECTIONS);
 	for (j = 0; j < conn_len; j++) {
-		wcaps = snd_hda_param_read(codec, conn[j],
-					   AC_PAR_AUDIO_WIDGET_CAP);
+		wcaps = get_wcaps(codec, conn[j]);
 		wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT;
 		/* we check only analog outputs */
 		if (wtype != AC_WID_AUD_OUT || (wcaps & AC_WCAP_DIGITAL))
@@ -2895,6 +2884,16 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid)
 			return conn[j];
 		}
 	}
+	/* if all DACs are already assigned, connect to the primary DAC */
+	if (conn_len > 1) {
+		for (j = 0; j < conn_len; j++) {
+			if (conn[j] == spec->multiout.dac_nids[0]) {
+				snd_hda_codec_write_cache(codec, nid, 0,
+						  AC_VERB_SET_CONNECT_SEL, j);
+				break;
+			}
+		}
+	}
 	return 0;
 }
 
@@ -2935,6 +2934,26 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec)
 		add_spec_dacs(spec, dac);
 	}
 
+	for (i = 0; i < cfg->hp_outs; i++) {
+		nid = cfg->hp_pins[i];
+		dac = get_unassigned_dac(codec, nid);
+		if (dac) {
+			if (!spec->multiout.hp_nid)
+				spec->multiout.hp_nid = dac;
+			else
+				add_spec_extra_dacs(spec, dac);
+		}
+		spec->hp_dacs[i] = dac;
+	}
+
+	for (i = 0; i < cfg->speaker_outs; i++) {
+		nid = cfg->speaker_pins[i];
+		dac = get_unassigned_dac(codec, nid);
+		if (dac)
+			add_spec_extra_dacs(spec, dac);
+		spec->speaker_dacs[i] = dac;
+	}
+
 	/* add line-in as output */
 	nid = check_line_out_switch(codec);
 	if (nid) {
@@ -2962,26 +2981,6 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec)
 		}
 	}
 
-	for (i = 0; i < cfg->hp_outs; i++) {
-		nid = cfg->hp_pins[i];
-		dac = get_unassigned_dac(codec, nid);
-		if (dac) {
-			if (!spec->multiout.hp_nid)
-				spec->multiout.hp_nid = dac;
-			else
-				add_spec_extra_dacs(spec, dac);
-		}
-		spec->hp_dacs[i] = dac;
-	}
-
-	for (i = 0; i < cfg->speaker_outs; i++) {
-		nid = cfg->speaker_pins[i];
-		dac = get_unassigned_dac(codec, nid);
-		if (dac)
-			add_spec_extra_dacs(spec, dac);
-		spec->speaker_dacs[i] = dac;
-	}
-
 	snd_printd("stac92xx: dac_nids=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n",
 		   spec->multiout.num_dacs,
 		   spec->multiout.dac_nids[0],
@@ -2994,24 +2993,47 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec)
 }
 
 /* create volume control/switch for the given prefx type */
-static int create_controls(struct sigmatel_spec *spec, const char *pfx, hda_nid_t nid, int chs)
+static int create_controls_idx(struct hda_codec *codec, const char *pfx,
+			       int idx, hda_nid_t nid, int chs)
 {
+	struct sigmatel_spec *spec = codec->spec;
 	char name[32];
 	int err;
 
+	if (!spec->check_volume_offset) {
+		unsigned int caps, step, nums, db_scale;
+		caps = query_amp_caps(codec, nid, HDA_OUTPUT);
+		step = (caps & AC_AMPCAP_STEP_SIZE) >>
+			AC_AMPCAP_STEP_SIZE_SHIFT;
+		step = (step + 1) * 25; /* in .01dB unit */
+		nums = (caps & AC_AMPCAP_NUM_STEPS) >>
+			AC_AMPCAP_NUM_STEPS_SHIFT;
+		db_scale = nums * step;
+		/* if dB scale is over -64dB, and finer enough,
+		 * let's reduce it to half
+		 */
+		if (db_scale > 6400 && nums >= 0x1f)
+			spec->volume_offset = nums / 2;
+		spec->check_volume_offset = 1;
+	}
+
 	sprintf(name, "%s Playback Volume", pfx);
-	err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name,
-				   HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
+	err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_VOL, idx, name,
+		HDA_COMPOSE_AMP_VAL_OFS(nid, chs, 0, HDA_OUTPUT,
+					spec->volume_offset));
 	if (err < 0)
 		return err;
 	sprintf(name, "%s Playback Switch", pfx);
-	err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name,
+	err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_MUTE, idx, name,
 				   HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT));
 	if (err < 0)
 		return err;
 	return 0;
 }
 
+#define create_controls(codec, pfx, nid, chs) \
+	create_controls_idx(codec, pfx, 0, nid, chs)
+
 static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid)
 {
 	if (spec->multiout.num_dacs > 4) {
@@ -3037,40 +3059,32 @@ static int add_spec_extra_dacs(struct sigmatel_spec *spec, hda_nid_t nid)
 	return 1;
 }
 
-static int is_unique_dac(struct sigmatel_spec *spec, hda_nid_t nid)
-{
-	int i;
-
-	if (spec->autocfg.line_outs != 1)
-		return 0;
-	if (spec->multiout.hp_nid == nid)
-		return 0;
-	for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++)
-		if (spec->multiout.extra_out_nid[i] == nid)
-			return 0;
-	return 1;
-}
-
-/* add playback controls from the parsed DAC table */
-static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
-					       const struct auto_pin_cfg *cfg)
+/* Create output controls
+ * The mixer elements are named depending on the given type (AUTO_PIN_XXX_OUT)
+ */
+static int create_multi_out_ctls(struct hda_codec *codec, int num_outs,
+				 const hda_nid_t *pins,
+				 const hda_nid_t *dac_nids,
+				 int type)
 {
 	struct sigmatel_spec *spec = codec->spec;
 	static const char *chname[4] = {
 		"Front", "Surround", NULL /*CLFE*/, "Side"
 	};
-	hda_nid_t nid = 0;
+	hda_nid_t nid;
 	int i, err;
 	unsigned int wid_caps;
 
-	for (i = 0; i < cfg->line_outs && spec->multiout.dac_nids[i]; i++) {
-		nid = spec->multiout.dac_nids[i];
-		if (i == 2) {
+	for (i = 0; i < num_outs && i < ARRAY_SIZE(chname); i++) {
+		nid = dac_nids[i];
+		if (!nid)
+			continue;
+		if (type != AUTO_PIN_HP_OUT && i == 2) {
 			/* Center/LFE */
-			err = create_controls(spec, "Center", nid, 1);
+			err = create_controls(codec, "Center", nid, 1);
 			if (err < 0)
 				return err;
-			err = create_controls(spec, "LFE", nid, 2);
+			err = create_controls(codec, "LFE", nid, 2);
 			if (err < 0)
 				return err;
 
@@ -3086,23 +3100,47 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
 			}
 
 		} else {
-			const char *name = chname[i];
-			/* if it's a single DAC, assign a better name */
-			if (!i && is_unique_dac(spec, nid)) {
-				switch (cfg->line_out_type) {
-				case AUTO_PIN_HP_OUT:
-					name = "Headphone";
-					break;
-				case AUTO_PIN_SPEAKER_OUT:
-					name = "Speaker";
-					break;
-				}
+			const char *name;
+			int idx;
+			switch (type) {
+			case AUTO_PIN_HP_OUT:
+				name = "Headphone";
+				idx = i;
+				break;
+			case AUTO_PIN_SPEAKER_OUT:
+				name = "Speaker";
+				idx = i;
+				break;
+			default:
+				name = chname[i];
+				idx = 0;
+				break;
 			}
-			err = create_controls(spec, name, nid, 3);
+			err = create_controls_idx(codec, name, idx, nid, 3);
 			if (err < 0)
 				return err;
+			if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) {
+				wid_caps = get_wcaps(codec, pins[i]);
+				if (wid_caps & AC_WCAP_UNSOL_CAP)
+					spec->hp_detect = 1;
+			}
 		}
 	}
+	return 0;
+}
+
+/* add playback controls from the parsed DAC table */
+static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec,
+					       const struct auto_pin_cfg *cfg)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	int err;
+
+	err = create_multi_out_ctls(codec, cfg->line_outs, cfg->line_out_pins,
+				    spec->multiout.dac_nids,
+				    cfg->line_out_type);
+	if (err < 0)
+		return err;
 
 	if (cfg->hp_outs > 1 && cfg->line_out_type == AUTO_PIN_LINE_OUT) {
 		err = stac92xx_add_control(spec,
@@ -3137,40 +3175,18 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec,
 					struct auto_pin_cfg *cfg)
 {
 	struct sigmatel_spec *spec = codec->spec;
-	hda_nid_t nid;
-	int i, err, nums;
+	int err;
+
+	err = create_multi_out_ctls(codec, cfg->hp_outs, cfg->hp_pins,
+				    spec->hp_dacs, AUTO_PIN_HP_OUT);
+	if (err < 0)
+		return err;
+
+	err = create_multi_out_ctls(codec, cfg->speaker_outs, cfg->speaker_pins,
+				    spec->speaker_dacs, AUTO_PIN_SPEAKER_OUT);
+	if (err < 0)
+		return err;
 
-	nums = 0;
-	for (i = 0; i < cfg->hp_outs; i++) {
-		static const char *pfxs[] = {
-			"Headphone", "Headphone2", "Headphone3",
-		};
-		unsigned int wid_caps = get_wcaps(codec, cfg->hp_pins[i]);
-		if (wid_caps & AC_WCAP_UNSOL_CAP)
-			spec->hp_detect = 1;
-		if (nums >= ARRAY_SIZE(pfxs))
-			continue;
-		nid = spec->hp_dacs[i];
-		if (!nid)
-			continue;
-		err = create_controls(spec, pfxs[nums++], nid, 3);
-		if (err < 0)
-			return err;
-	}
-	nums = 0;
-	for (i = 0; i < cfg->speaker_outs; i++) {
-		static const char *pfxs[] = {
-			"Speaker", "External Speaker", "Speaker2",
-		};
-		if (nums >= ARRAY_SIZE(pfxs))
-			continue;
-		nid = spec->speaker_dacs[i];
-		if (!nid)
-			continue;
-		err = create_controls(spec, pfxs[nums++], nid, 3);
-		if (err < 0)
-			return err;
-	}
 	return 0;
 }
 
@@ -3379,11 +3395,7 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec,
 		unsigned int wcaps;
 		unsigned int def_conf;
 
-		def_conf = snd_hda_codec_read(codec,
-					      spec->dmic_nids[i],
-					      0,
-					      AC_VERB_GET_CONFIG_DEFAULT,
-					      0);
+		def_conf = snd_hda_codec_get_pincfg(codec, spec->dmic_nids[i]);
 		if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE)
 			continue;
 
@@ -3507,6 +3519,7 @@ static void stac92xx_auto_init_hp_out(struct hda_codec *codec)
 static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out, hda_nid_t dig_in)
 {
 	struct sigmatel_spec *spec = codec->spec;
+	int hp_swap = 0;
 	int err;
 
 	if ((err = snd_hda_parse_pin_def_config(codec,
@@ -3516,7 +3529,6 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 	if (! spec->autocfg.line_outs)
 		return 0; /* can't find valid pin config */
 
-#if 0 /* FIXME: temporarily disabled */
 	/* If we have no real line-out pin and multiple hp-outs, HPs should
 	 * be set up as multi-channel outputs.
 	 */
@@ -3535,8 +3547,8 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 		spec->autocfg.line_outs = spec->autocfg.hp_outs;
 		spec->autocfg.line_out_type = AUTO_PIN_HP_OUT;
 		spec->autocfg.hp_outs = 0;
+		hp_swap = 1;
 	}
-#endif /* FIXME: temporarily disabled */
 	if (spec->autocfg.mono_out_pin) {
 		int dir = get_wcaps(codec, spec->autocfg.mono_out_pin) &
 			(AC_WCAP_OUT_AMP | AC_WCAP_IN_AMP);
@@ -3629,12 +3641,19 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 #endif
 
 	err = stac92xx_auto_create_hp_ctls(codec, &spec->autocfg);
-
 	if (err < 0)
 		return err;
 
-	err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg);
+	/* All output parsing done, now restore the swapped hp pins */
+	if (hp_swap) {
+		memcpy(spec->autocfg.hp_pins, spec->autocfg.line_out_pins,
+		       sizeof(spec->autocfg.hp_pins));
+		spec->autocfg.hp_outs = spec->autocfg.line_outs;
+		spec->autocfg.line_out_type = AUTO_PIN_HP_OUT;
+		spec->autocfg.line_outs = 0;
+	}
 
+	err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg);
 	if (err < 0)
 		return err;
 
@@ -3663,11 +3682,15 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out
 			return err;
 	}
 
+	err = stac92xx_add_input_source(spec);
+	if (err < 0)
+		return err;
+
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 	if (spec->multiout.max_channels > 2)
 		spec->surr_switch = 1;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = dig_out;
 	if (dig_in && spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = dig_in;
@@ -3730,9 +3753,7 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec,
 		for (i = 0; i < spec->autocfg.line_outs && lfe_pin == 0x0; i++) {
 			hda_nid_t pin = spec->autocfg.line_out_pins[i];
 			unsigned int defcfg;
-			defcfg = snd_hda_codec_read(codec, pin, 0,
-						 AC_VERB_GET_CONFIG_DEFAULT,
-						 0x00);
+			defcfg = snd_hda_codec_get_pincfg(codec, pin);
 			if (get_defcfg_device(defcfg) == AC_JACK_SPEAKER) {
 				unsigned int wcaps = get_wcaps(codec, pin);
 				wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP);
@@ -3745,7 +3766,7 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec,
 	}
 
 	if (lfe_pin) {
-		err = create_controls(spec, "LFE", lfe_pin, 1);
+		err = create_controls(codec, "LFE", lfe_pin, 1);
 		if (err < 0)
 			return err;
 	}
@@ -3776,7 +3797,11 @@ static int stac9200_parse_auto_config(struct hda_codec *codec)
 			return err;
 	}
 
-	if (spec->autocfg.dig_out_pin)
+	err = stac92xx_add_input_source(spec);
+	if (err < 0)
+		return err;
+
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = 0x05;
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = 0x04;
@@ -3832,8 +3857,7 @@ static int stac92xx_add_jack(struct hda_codec *codec,
 #ifdef CONFIG_SND_JACK
 	struct sigmatel_spec *spec = codec->spec;
 	struct sigmatel_jack *jack;
-	int def_conf = snd_hda_codec_read(codec, nid,
-			0, AC_VERB_GET_CONFIG_DEFAULT, 0);
+	int def_conf = snd_hda_codec_get_pincfg(codec, nid);
 	int connectivity = get_defcfg_connect(def_conf);
 	char name[32];
 
@@ -3948,6 +3972,36 @@ static void stac92xx_power_down(struct hda_codec *codec)
 static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid,
 				  int enable);
 
+/* override some hints from the hwdep entry */
+static void stac_store_hints(struct hda_codec *codec)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	const char *p;
+	int val;
+
+	val = snd_hda_get_bool_hint(codec, "hp_detect");
+	if (val >= 0)
+		spec->hp_detect = val;
+	p = snd_hda_get_hint(codec, "gpio_mask");
+	if (p) {
+		spec->gpio_mask = simple_strtoul(p, NULL, 0);
+		spec->eapd_mask = spec->gpio_dir = spec->gpio_data =
+			spec->gpio_mask;
+	}
+	p = snd_hda_get_hint(codec, "gpio_dir");
+	if (p)
+		spec->gpio_dir = simple_strtoul(p, NULL, 0) & spec->gpio_mask;
+	p = snd_hda_get_hint(codec, "gpio_data");
+	if (p)
+		spec->gpio_data = simple_strtoul(p, NULL, 0) & spec->gpio_mask;
+	p = snd_hda_get_hint(codec, "eapd_mask");
+	if (p)
+		spec->eapd_mask = simple_strtoul(p, NULL, 0) & spec->gpio_mask;
+	val = snd_hda_get_bool_hint(codec, "eapd_switch");
+	if (val >= 0)
+		spec->eapd_switch = val;
+}
+
 static int stac92xx_init(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
@@ -3964,6 +4018,9 @@ static int stac92xx_init(struct hda_codec *codec)
 				spec->adc_nids[i], 0,
 				AC_VERB_SET_POWER_STATE, AC_PWRST_D3);
 
+	/* override some hints */
+	stac_store_hints(codec);
+
 	/* set up GPIO */
 	gpio = spec->gpio_data;
 	/* turn on EAPD statically when spec->eapd_switch isn't set.
@@ -4013,8 +4070,7 @@ static int stac92xx_init(struct hda_codec *codec)
 								 pinctl);
 				}
 			}
-			conf = snd_hda_codec_read(codec, nid, 0,
-					      AC_VERB_GET_CONFIG_DEFAULT, 0);
+			conf = snd_hda_codec_get_pincfg(codec, nid);
 			if (get_defcfg_connect(conf) != AC_JACK_PORT_FIXED) {
 				enable_pin_detect(codec, nid,
 						  STAC_INSERT_EVENT);
@@ -4026,8 +4082,8 @@ static int stac92xx_init(struct hda_codec *codec)
 	for (i = 0; i < spec->num_dmics; i++)
 		stac92xx_auto_set_pinctl(codec, spec->dmic_nids[i],
 					AC_PINCTL_IN_EN);
-	if (cfg->dig_out_pin)
-		stac92xx_auto_set_pinctl(codec, cfg->dig_out_pin,
+	if (cfg->dig_out_pins[0])
+		stac92xx_auto_set_pinctl(codec, cfg->dig_out_pins[0],
 					 AC_PINCTL_OUT_EN);
 	if (cfg->dig_in_pin)
 		stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin,
@@ -4055,8 +4111,7 @@ static int stac92xx_init(struct hda_codec *codec)
 			stac_toggle_power_map(codec, nid, 1);
 			continue;
 		}
-		def_conf = snd_hda_codec_read(codec, nid, 0,
-					      AC_VERB_GET_CONFIG_DEFAULT, 0);
+		def_conf = snd_hda_codec_get_pincfg(codec, nid);
 		def_conf = get_defcfg_connect(def_conf);
 		/* skip any ports that don't have jacks since presence
  		 * detection is useless */
@@ -4110,7 +4165,6 @@ static void stac92xx_free(struct hda_codec *codec)
 	if (! spec)
 		return;
 
-	kfree(spec->pin_configs);
 	stac92xx_free_jacks(codec);
 	snd_array_free(&spec->events);
 
@@ -4121,7 +4175,9 @@ static void stac92xx_free(struct hda_codec *codec)
 static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid,
 				unsigned int flag)
 {
-	unsigned int pin_ctl = snd_hda_codec_read(codec, nid,
+	unsigned int old_ctl, pin_ctl;
+
+	pin_ctl = snd_hda_codec_read(codec, nid,
 			0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00);
 
 	if (pin_ctl & AC_PINCTL_IN_EN) {
@@ -4135,14 +4191,17 @@ static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid,
 			return;
 	}
 
+	old_ctl = pin_ctl;
 	/* if setting pin direction bits, clear the current
 	   direction bits first */
 	if (flag & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN))
 		pin_ctl &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN);
 	
-	snd_hda_codec_write_cache(codec, nid, 0,
-			AC_VERB_SET_PIN_WIDGET_CONTROL,
-			pin_ctl | flag);
+	pin_ctl |= flag;
+	if (old_ctl != pin_ctl)
+		snd_hda_codec_write_cache(codec, nid, 0,
+					  AC_VERB_SET_PIN_WIDGET_CONTROL,
+					  pin_ctl);
 }
 
 static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid,
@@ -4150,9 +4209,10 @@ static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid,
 {
 	unsigned int pin_ctl = snd_hda_codec_read(codec, nid,
 			0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00);
-	snd_hda_codec_write_cache(codec, nid, 0,
-			AC_VERB_SET_PIN_WIDGET_CONTROL,
-			pin_ctl & ~flag);
+	if (pin_ctl & flag)
+		snd_hda_codec_write_cache(codec, nid, 0,
+					  AC_VERB_SET_PIN_WIDGET_CONTROL,
+					  pin_ctl & ~flag);
 }
 
 static int get_pin_presence(struct hda_codec *codec, hda_nid_t nid)
@@ -4415,7 +4475,6 @@ static int stac92xx_resume(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec = codec->spec;
 
-	stac92xx_set_config_regs(codec);
 	stac92xx_init(codec);
 	snd_hda_codec_resume_amp(codec);
 	snd_hda_codec_resume_cache(codec);
@@ -4426,6 +4485,37 @@ static int stac92xx_resume(struct hda_codec *codec)
 	return 0;
 }
 
+
+/*
+ * using power check for controlling mute led of HP HDX notebooks
+ * check for mute state only on Speakers (nid = 0x10)
+ *
+ * For this feature CONFIG_SND_HDA_POWER_SAVE is needed, otherwise
+ * the LED is NOT working properly !
+ */
+
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+static int stac92xx_hp_hdx_check_power_status(struct hda_codec *codec,
+					      hda_nid_t nid)
+{
+	struct sigmatel_spec *spec = codec->spec;
+
+	if (nid == 0x10) {
+		if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) &
+		    HDA_AMP_MUTE)
+			spec->gpio_data &= ~0x08;  /* orange */
+		else
+			spec->gpio_data |= 0x08;   /* white */
+
+		stac_gpio_set(codec, spec->gpio_mask,
+			      spec->gpio_dir,
+			      spec->gpio_data);
+	}
+
+	return 0;
+}
+#endif
+
 static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state)
 {
 	struct sigmatel_spec *spec = codec->spec;
@@ -4464,16 +4554,11 @@ static int patch_stac9200(struct hda_codec *codec)
 	spec->board_config = snd_hda_check_board_config(codec, STAC_9200_MODELS,
 							stac9200_models,
 							stac9200_cfg_tbl);
-	if (spec->board_config < 0) {
+	if (spec->board_config < 0)
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9200, using BIOS defaults\n");
-		err = stac92xx_save_bios_config_regs(codec);
-	} else
-		err = stac_save_pin_cfgs(codec,
+	else
+		stac92xx_set_config_regs(codec,
 					 stac9200_brd_tbl[spec->board_config]);
-	if (err < 0) {
-		stac92xx_free(codec);
-		return err;
-	}
 
 	spec->multiout.max_channels = 2;
 	spec->multiout.num_dacs = 1;
@@ -4541,17 +4626,12 @@ static int patch_stac925x(struct hda_codec *codec)
 							stac925x_models,
 							stac925x_cfg_tbl);
  again:
-	if (spec->board_config < 0) {
+	if (spec->board_config < 0)
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC925x,"
 				      "using BIOS defaults\n");
-		err = stac92xx_save_bios_config_regs(codec);
-	} else
-		err = stac_save_pin_cfgs(codec,
+	else
+		stac92xx_set_config_regs(codec,
 					 stac925x_brd_tbl[spec->board_config]);
-	if (err < 0) {
-		stac92xx_free(codec);
-		return err;
-	}
 
 	spec->multiout.max_channels = 2;
 	spec->multiout.num_dacs = 1;
@@ -4629,17 +4709,12 @@ static int patch_stac92hd73xx(struct hda_codec *codec)
 							stac92hd73xx_models,
 							stac92hd73xx_cfg_tbl);
 again:
-	if (spec->board_config < 0) {
+	if (spec->board_config < 0)
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for"
 			" STAC92HD73XX, using BIOS defaults\n");
-		err = stac92xx_save_bios_config_regs(codec);
-	} else
-		err = stac_save_pin_cfgs(codec,
+	else
+		stac92xx_set_config_regs(codec,
 				stac92hd73xx_brd_tbl[spec->board_config]);
-	if (err < 0) {
-		stac92xx_free(codec);
-		return err;
-	}
 
 	num_dacs = snd_hda_get_connections(codec, 0x0a,
 			conn, STAC92HD73_DAC_COUNT + 2) - 1;
@@ -4653,14 +4728,18 @@ again:
 	case 0x3: /* 6 Channel */
 		spec->mixer = stac92hd73xx_6ch_mixer;
 		spec->init = stac92hd73xx_6ch_core_init;
+		spec->aloopback_ctl = stac92hd73xx_6ch_loopback;
 		break;
 	case 0x4: /* 8 Channel */
 		spec->mixer = stac92hd73xx_8ch_mixer;
 		spec->init = stac92hd73xx_8ch_core_init;
+		spec->aloopback_ctl = stac92hd73xx_8ch_loopback;
 		break;
 	case 0x5: /* 10 Channel */
 		spec->mixer = stac92hd73xx_10ch_mixer;
 		spec->init = stac92hd73xx_10ch_core_init;
+		spec->aloopback_ctl = stac92hd73xx_10ch_loopback;
+		break;
 	}
 	spec->multiout.dac_nids = spec->dac_nids;
 
@@ -4699,18 +4778,18 @@ again:
 			spec->init = dell_m6_core_init;
 		switch (spec->board_config) {
 		case STAC_DELL_M6_AMIC: /* Analog Mics */
-			stac92xx_set_config_reg(codec, 0x0b, 0x90A70170);
+			snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170);
 			spec->num_dmics = 0;
 			spec->private_dimux.num_items = 1;
 			break;
 		case STAC_DELL_M6_DMIC: /* Digital Mics */
-			stac92xx_set_config_reg(codec, 0x13, 0x90A60160);
+			snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160);
 			spec->num_dmics = 1;
 			spec->private_dimux.num_items = 2;
 			break;
 		case STAC_DELL_M6_BOTH: /* Both */
-			stac92xx_set_config_reg(codec, 0x0b, 0x90A70170);
-			stac92xx_set_config_reg(codec, 0x13, 0x90A60160);
+			snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170);
+			snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160);
 			spec->num_dmics = 1;
 			spec->private_dimux.num_items = 2;
 			break;
@@ -4773,6 +4852,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 	hda_nid_t conn[STAC92HD83_DAC_COUNT + 1];
 	int err;
 	int num_dacs;
+	hda_nid_t nid;
 
 	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -4791,15 +4871,6 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 	spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids);
 	spec->multiout.dac_nids = spec->dac_nids;
 
-
-	/* set port 0xe to select the last DAC
-	 */
-	num_dacs = snd_hda_get_connections(codec, 0x0e,
-		conn, STAC92HD83_DAC_COUNT + 1) - 1;
-
-	snd_hda_codec_write_cache(codec, 0xe, 0,
-		AC_VERB_SET_CONNECT_SEL, num_dacs);
-
 	spec->init = stac92hd83xxx_core_init;
 	spec->mixer = stac92hd83xxx_mixer;
 	spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids);
@@ -4814,17 +4885,12 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 							stac92hd83xxx_models,
 							stac92hd83xxx_cfg_tbl);
 again:
-	if (spec->board_config < 0) {
+	if (spec->board_config < 0)
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for"
 			" STAC92HD83XXX, using BIOS defaults\n");
-		err = stac92xx_save_bios_config_regs(codec);
-	} else
-		err = stac_save_pin_cfgs(codec,
+	else
+		stac92xx_set_config_regs(codec,
 				stac92hd83xxx_brd_tbl[spec->board_config]);
-	if (err < 0) {
-		stac92xx_free(codec);
-		return err;
-	}
 
 	switch (codec->vendor_id) {
 	case 0x111d7604:
@@ -4851,6 +4917,23 @@ again:
 		return err;
 	}
 
+	switch (spec->board_config) {
+	case STAC_DELL_S14:
+		nid = 0xf;
+		break;
+	default:
+		nid = 0xe;
+		break;
+	}
+
+	num_dacs = snd_hda_get_connections(codec, nid,
+				conn, STAC92HD83_DAC_COUNT + 1) - 1;
+
+	/* set port X to select the last DAC
+	 */
+	snd_hda_codec_write_cache(codec, nid, 0,
+			AC_VERB_SET_CONNECT_SEL, num_dacs);
+
 	codec->patch_ops = stac92xx_patch_ops;
 
 	codec->proc_widget_hook = stac92hd_proc_hook;
@@ -4858,7 +4941,16 @@ again:
 	return 0;
 }
 
-static struct hda_input_mux stac92hd71bxx_dmux = {
+static struct hda_input_mux stac92hd71bxx_dmux_nomixer = {
+	.num_items = 3,
+	.items = {
+		{ "Analog Inputs", 0x00 },
+		{ "Digital Mic 1", 0x02 },
+		{ "Digital Mic 2", 0x03 },
+	}
+};
+
+static struct hda_input_mux stac92hd71bxx_dmux_amixer = {
 	.num_items = 4,
 	.items = {
 		{ "Analog Inputs", 0x00 },
@@ -4868,10 +4960,67 @@ static struct hda_input_mux stac92hd71bxx_dmux = {
 	}
 };
 
+/* get the pin connection (fixed, none, etc) */
+static unsigned int stac_get_defcfg_connect(struct hda_codec *codec, int idx)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	unsigned int cfg;
+
+	cfg = snd_hda_codec_get_pincfg(codec, spec->pin_nids[idx]);
+	return get_defcfg_connect(cfg);
+}
+
+static int stac92hd71bxx_connected_ports(struct hda_codec *codec,
+					 hda_nid_t *nids, int num_nids)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	int idx, num;
+	unsigned int def_conf;
+
+	for (num = 0; num < num_nids; num++) {
+		for (idx = 0; idx < spec->num_pins; idx++)
+			if (spec->pin_nids[idx] == nids[num])
+				break;
+		if (idx >= spec->num_pins)
+			break;
+		def_conf = stac_get_defcfg_connect(codec, idx);
+		if (def_conf == AC_JACK_PORT_NONE)
+			break;
+	}
+	return num;
+}
+
+static int stac92hd71bxx_connected_smuxes(struct hda_codec *codec,
+					  hda_nid_t dig0pin)
+{
+	struct sigmatel_spec *spec = codec->spec;
+	int idx;
+
+	for (idx = 0; idx < spec->num_pins; idx++)
+		if (spec->pin_nids[idx] == dig0pin)
+			break;
+	if ((idx + 2) >= spec->num_pins)
+		return 0;
+
+	/* dig1pin case */
+	if (stac_get_defcfg_connect(codec, idx + 1) != AC_JACK_PORT_NONE)
+		return 2;
+
+	/* dig0pin + dig2pin case */
+	if (stac_get_defcfg_connect(codec, idx + 2) != AC_JACK_PORT_NONE)
+		return 2;
+	if (stac_get_defcfg_connect(codec, idx) != AC_JACK_PORT_NONE)
+		return 1;
+	else
+		return 0;
+}
+
 static int patch_stac92hd71bxx(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
+	struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init;
 	int err = 0;
+	unsigned int ndmic_nids = 0;
 
 	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
@@ -4879,27 +5028,32 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
 
 	codec->spec = spec;
 	codec->patch_ops = stac92xx_patch_ops;
-	spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids);
+	spec->num_pins = STAC92HD71BXX_NUM_PINS;
+	switch (codec->vendor_id) {
+	case 0x111d76b6:
+	case 0x111d76b7:
+		spec->pin_nids = stac92hd71bxx_pin_nids_4port;
+		break;
+	case 0x111d7603:
+	case 0x111d7608:
+		/* On 92HD75Bx 0x27 isn't a pin nid */
+		spec->num_pins--;
+		/* fallthrough */
+	default:
+		spec->pin_nids = stac92hd71bxx_pin_nids_6port;
+	}
 	spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids);
-	spec->pin_nids = stac92hd71bxx_pin_nids;
-	memcpy(&spec->private_dimux, &stac92hd71bxx_dmux,
-			sizeof(stac92hd71bxx_dmux));
 	spec->board_config = snd_hda_check_board_config(codec,
 							STAC_92HD71BXX_MODELS,
 							stac92hd71bxx_models,
 							stac92hd71bxx_cfg_tbl);
 again:
-	if (spec->board_config < 0) {
+	if (spec->board_config < 0)
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for"
 			" STAC92HD71BXX, using BIOS defaults\n");
-		err = stac92xx_save_bios_config_regs(codec);
-	} else
-		err = stac_save_pin_cfgs(codec,
+	else
+		stac92xx_set_config_regs(codec,
 				stac92hd71bxx_brd_tbl[spec->board_config]);
-	if (err < 0) {
-		stac92xx_free(codec);
-		return err;
-	}
 
 	if (spec->board_config > STAC_92HD71BXX_REF) {
 		/* GPIO0 = EAPD */
@@ -4908,16 +5062,34 @@ again:
 		spec->gpio_data = 0x01;
 	}
 
+	spec->dmic_nids = stac92hd71bxx_dmic_nids;
+	spec->dmux_nids = stac92hd71bxx_dmux_nids;
+
 	switch (codec->vendor_id) {
 	case 0x111d76b6: /* 4 Port without Analog Mixer */
 	case 0x111d76b7:
+		unmute_init++;
+		/* fallthru */
 	case 0x111d76b4: /* 6 Port without Analog Mixer */
 	case 0x111d76b5:
+		memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_nomixer,
+		       sizeof(stac92hd71bxx_dmux_nomixer));
 		spec->mixer = stac92hd71bxx_mixer;
 		spec->init = stac92hd71bxx_core_init;
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
+		spec->num_dmics = stac92hd71bxx_connected_ports(codec,
+					stac92hd71bxx_dmic_nids,
+					STAC92HD71BXX_NUM_DMICS);
+		if (spec->num_dmics) {
+			spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
+			spec->dinput_mux = &spec->private_dimux;
+			ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1;
+		}
 		break;
 	case 0x111d7608: /* 5 Port with Analog Mixer */
+		memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer,
+		       sizeof(stac92hd71bxx_dmux_amixer));
+		spec->private_dimux.num_items--;
 		switch (spec->board_config) {
 		case STAC_HP_M4:
 			/* Enable VREF power saving on GPIO1 detect */
@@ -4944,7 +5116,15 @@ again:
 
 		/* disable VSW */
 		spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF];
-		stac_change_pin_config(codec, 0xf, 0x40f000f0);
+		unmute_init++;
+		snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0);
+		snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3);
+		stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS - 1] = 0;
+		spec->num_dmics = stac92hd71bxx_connected_ports(codec,
+					stac92hd71bxx_dmic_nids,
+					STAC92HD71BXX_NUM_DMICS - 1);
+		spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
+		ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 2;
 		break;
 	case 0x111d7603: /* 6 Port with Analog Mixer */
 		if ((codec->revision_id & 0xf) == 1)
@@ -4954,12 +5134,23 @@ again:
 		spec->num_pwrs = 0;
 		/* fallthru */
 	default:
+		memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer,
+		       sizeof(stac92hd71bxx_dmux_amixer));
 		spec->dinput_mux = &spec->private_dimux;
 		spec->mixer = stac92hd71bxx_analog_mixer;
 		spec->init = stac92hd71bxx_analog_core_init;
 		codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
+		spec->num_dmics = stac92hd71bxx_connected_ports(codec,
+					stac92hd71bxx_dmic_nids,
+					STAC92HD71BXX_NUM_DMICS);
+		spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
+		ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1;
 	}
 
+	if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP)
+		snd_hda_sequence_write_cache(codec, unmute_init);
+
+	spec->aloopback_ctl = stac92hd71bxx_loopback;
 	spec->aloopback_mask = 0x50;
 	spec->aloopback_shift = 0;
 
@@ -4967,18 +5158,17 @@ again:
 	spec->digbeep_nid = 0x26;
 	spec->mux_nids = stac92hd71bxx_mux_nids;
 	spec->adc_nids = stac92hd71bxx_adc_nids;
-	spec->dmic_nids = stac92hd71bxx_dmic_nids;
-	spec->dmux_nids = stac92hd71bxx_dmux_nids;
 	spec->smux_nids = stac92hd71bxx_smux_nids;
 	spec->pwr_nids = stac92hd71bxx_pwr_nids;
 
 	spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids);
 	spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids);
+	spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e);
 
 	switch (spec->board_config) {
 	case STAC_HP_M4:
 		/* enable internal microphone */
-		stac_change_pin_config(codec, 0x0e, 0x01813040);
+		snd_hda_codec_set_pincfg(codec, 0x0e, 0x01813040);
 		stac92xx_auto_set_pinctl(codec, 0x0e,
 			AC_PINCTL_IN_EN | AC_PINCTL_VREF_80);
 		/* fallthru */
@@ -4993,19 +5183,36 @@ again:
 		spec->num_smuxes = 0;
 		spec->num_dmuxes = 1;
 		break;
-	default:
-		spec->num_dmics = STAC92HD71BXX_NUM_DMICS;
-		spec->num_smuxes = ARRAY_SIZE(stac92hd71bxx_smux_nids);
-		spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids);
+	case STAC_HP_DV5:
+		snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010);
+		stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN);
+		break;
+	case STAC_HP_HDX:
+		spec->num_dmics = 1;
+		spec->num_dmuxes = 1;
+		spec->num_smuxes = 1;
+		/*
+		 * For controlling MUTE LED on HP HDX16/HDX18 notebooks,
+		 * the CONFIG_SND_HDA_POWER_SAVE is needed to be set.
+		 */
+#ifdef CONFIG_SND_HDA_POWER_SAVE
+		/* orange/white mute led on GPIO3, orange=0, white=1 */
+		spec->gpio_mask |= 0x08;
+		spec->gpio_dir  |= 0x08;
+		spec->gpio_data |= 0x08;  /* set to white */
+
+		/* register check_power_status callback. */
+		codec->patch_ops.check_power_status =
+		    stac92xx_hp_hdx_check_power_status;
+#endif	
+		break;
 	};
 
 	spec->multiout.dac_nids = spec->dac_nids;
 	if (spec->dinput_mux)
-		spec->private_dimux.num_items +=
-			spec->num_dmics -
-				(ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1);
+		spec->private_dimux.num_items += spec->num_dmics - ndmic_nids;
 
-	err = stac92xx_parse_auto_config(codec, 0x21, 0x23);
+	err = stac92xx_parse_auto_config(codec, 0x21, 0);
 	if (!err) {
 		if (spec->board_config < 0) {
 			printk(KERN_WARNING "hda_codec: No auto-config is "
@@ -5080,17 +5287,12 @@ static int patch_stac922x(struct hda_codec *codec)
 	}
 
  again:
-	if (spec->board_config < 0) {
+	if (spec->board_config < 0)
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, "
 			"using BIOS defaults\n");
-		err = stac92xx_save_bios_config_regs(codec);
-	} else
-		err = stac_save_pin_cfgs(codec,
+	else
+		stac92xx_set_config_regs(codec,
 				stac922x_brd_tbl[spec->board_config]);
-	if (err < 0) {
-		stac92xx_free(codec);
-		return err;
-	}
 
 	spec->adc_nids = stac922x_adc_nids;
 	spec->mux_nids = stac922x_mux_nids;
@@ -5141,24 +5343,19 @@ static int patch_stac927x(struct hda_codec *codec)
 		return -ENOMEM;
 
 	codec->spec = spec;
+	codec->slave_dig_outs = stac927x_slave_dig_outs;
 	spec->num_pins = ARRAY_SIZE(stac927x_pin_nids);
 	spec->pin_nids = stac927x_pin_nids;
 	spec->board_config = snd_hda_check_board_config(codec, STAC_927X_MODELS,
 							stac927x_models,
 							stac927x_cfg_tbl);
  again:
-	if (spec->board_config < 0 || !stac927x_brd_tbl[spec->board_config]) {
-		if (spec->board_config < 0)
-			snd_printdd(KERN_INFO "hda_codec: Unknown model for"
-				    "STAC927x, using BIOS defaults\n");
-		err = stac92xx_save_bios_config_regs(codec);
-	} else
-		err = stac_save_pin_cfgs(codec,
+	if (spec->board_config < 0)
+		snd_printdd(KERN_INFO "hda_codec: Unknown model for"
+			    "STAC927x, using BIOS defaults\n");
+	else
+		stac92xx_set_config_regs(codec,
 				stac927x_brd_tbl[spec->board_config]);
-	if (err < 0) {
-		stac92xx_free(codec);
-		return err;
-	}
 
 	spec->digbeep_nid = 0x23;
 	spec->adc_nids = stac927x_adc_nids;
@@ -5187,15 +5384,15 @@ static int patch_stac927x(struct hda_codec *codec)
 		case 0x10280209:
 		case 0x1028022e:
 			/* correct the device field to SPDIF out */
-			stac_change_pin_config(codec, 0x21, 0x01442070);
+			snd_hda_codec_set_pincfg(codec, 0x21, 0x01442070);
 			break;
 		};
 		/* configure the analog microphone on some laptops */
-		stac_change_pin_config(codec, 0x0c, 0x90a79130);
+		snd_hda_codec_set_pincfg(codec, 0x0c, 0x90a79130);
 		/* correct the front output jack as a hp out */
-		stac_change_pin_config(codec, 0x0f, 0x0227011f);
+		snd_hda_codec_set_pincfg(codec, 0x0f, 0x0227011f);
 		/* correct the front input jack as a mic */
-		stac_change_pin_config(codec, 0x0e, 0x02a79130);
+		snd_hda_codec_set_pincfg(codec, 0x0e, 0x02a79130);
 		/* fallthru */
 	case STAC_DELL_3ST:
 		/* GPIO2 High = Enable EAPD */
@@ -5222,6 +5419,7 @@ static int patch_stac927x(struct hda_codec *codec)
 	}
 
 	spec->num_pwrs = 0;
+	spec->aloopback_ctl = stac927x_loopback;
 	spec->aloopback_mask = 0x40;
 	spec->aloopback_shift = 0;
 	spec->eapd_switch = 1;
@@ -5280,16 +5478,11 @@ static int patch_stac9205(struct hda_codec *codec)
 							stac9205_models,
 							stac9205_cfg_tbl);
  again:
-	if (spec->board_config < 0) {
+	if (spec->board_config < 0)
 		snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9205, using BIOS defaults\n");
-		err = stac92xx_save_bios_config_regs(codec);
-	} else
-		err = stac_save_pin_cfgs(codec,
+	else
+		stac92xx_set_config_regs(codec,
 					 stac9205_brd_tbl[spec->board_config]);
-	if (err < 0) {
-		stac92xx_free(codec);
-		return err;
-	}
 
 	spec->digbeep_nid = 0x23;
 	spec->adc_nids = stac9205_adc_nids;
@@ -5306,17 +5499,20 @@ static int patch_stac9205(struct hda_codec *codec)
 
 	spec->init = stac9205_core_init;
 	spec->mixer = stac9205_mixer;
+	spec->aloopback_ctl = stac9205_loopback;
 
 	spec->aloopback_mask = 0x40;
 	spec->aloopback_shift = 0;
-	spec->eapd_switch = 1;
+	/* Turn on/off EAPD per HP plugging */
+	if (spec->board_config != STAC_9205_EAPD)
+		spec->eapd_switch = 1;
 	spec->multiout.dac_nids = spec->dac_nids;
 	
 	switch (spec->board_config){
 	case STAC_9205_DELL_M43:
 		/* Enable SPDIF in/out */
-		stac_change_pin_config(codec, 0x1f, 0x01441030);
-		stac_change_pin_config(codec, 0x20, 0x1c410030);
+		snd_hda_codec_set_pincfg(codec, 0x1f, 0x01441030);
+		snd_hda_codec_set_pincfg(codec, 0x20, 0x1c410030);
 
 		/* Enable unsol response for GPIO4/Dock HP connection */
 		err = stac_add_event(spec, codec->afg, STAC_VREF_EVENT, 0x01);
@@ -5373,223 +5569,87 @@ static int patch_stac9205(struct hda_codec *codec)
  * STAC9872 hack
  */
 
-/* static config for Sony VAIO FE550G and Sony VAIO AR */
-static hda_nid_t vaio_dacs[] = { 0x2 };
-#define VAIO_HP_DAC	0x5
-static hda_nid_t vaio_adcs[] = { 0x8 /*,0x6*/ };
-static hda_nid_t vaio_mux_nids[] = { 0x15 };
-
-static struct hda_input_mux vaio_mux = {
-	.num_items = 3,
-	.items = {
-		/* { "HP", 0x0 }, */
-		{ "Mic Jack", 0x1 },
-		{ "Internal Mic", 0x2 },
-		{ "PCM", 0x3 },
-	}
-};
-
-static struct hda_verb vaio_init[] = {
-	{0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */
-	{0x0a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | STAC_HP_EVENT},
-	{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */
-	{0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */
-	{0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */
+static struct hda_verb stac9872_core_init[] = {
 	{0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */
 	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */
 	{}
 };
 
-static struct hda_verb vaio_ar_init[] = {
-	{0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */
-	{0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */
-	{0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */
-	{0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */
-/*	{0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },*/ /* Optical Out */
-	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */
-	{0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */
-	{0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */
-/*	{0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},*/ /* Optical Out */
-	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */
-	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */
-	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */
-	{}
-};
-
-static struct snd_kcontrol_new vaio_mixer[] = {
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x02, 0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x02, 0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x05, 0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x05, 0, HDA_OUTPUT),
-	/* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */
+static struct snd_kcontrol_new stac9872_mixer[] = {
 	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT),
 	HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT),
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.count = 1,
-		.info = stac92xx_mux_enum_info,
-		.get = stac92xx_mux_enum_get,
-		.put = stac92xx_mux_enum_put,
-	},
-	{}
+	{ } /* end */
 };
 
-static struct snd_kcontrol_new vaio_ar_mixer[] = {
-	HDA_CODEC_VOLUME("Headphone Playback Volume", 0x02, 0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Headphone Playback Switch", 0x02, 0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x05, 0, HDA_OUTPUT),
-	HDA_CODEC_MUTE("Speaker Playback Switch", 0x05, 0, HDA_OUTPUT),
-	/* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */
-	HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT),
-	HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT),
-	/*HDA_CODEC_MUTE("Optical Out Switch", 0x10, 0, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Optical Out Volume", 0x10, 0, HDA_OUTPUT),*/
-	{
-		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
-		.name = "Capture Source",
-		.count = 1,
-		.info = stac92xx_mux_enum_info,
-		.get = stac92xx_mux_enum_get,
-		.put = stac92xx_mux_enum_put,
-	},
-	{}
+static hda_nid_t stac9872_pin_nids[] = {
+	0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f,
+	0x11, 0x13, 0x14,
 };
 
-static struct hda_codec_ops stac9872_patch_ops = {
-	.build_controls = stac92xx_build_controls,
-	.build_pcms = stac92xx_build_pcms,
-	.init = stac92xx_init,
-	.free = stac92xx_free,
-#ifdef SND_HDA_NEEDS_RESUME
-	.resume = stac92xx_resume,
-#endif
+static hda_nid_t stac9872_adc_nids[] = {
+	0x8 /*,0x6*/
 };
 
-static int stac9872_vaio_init(struct hda_codec *codec)
-{
-	int err;
-
-	err = stac92xx_init(codec);
-	if (err < 0)
-		return err;
-	if (codec->patch_ops.unsol_event)
-		codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26);
-	return 0;
-}
-
-static void stac9872_vaio_hp_detect(struct hda_codec *codec, unsigned int res)
-{
-	if (get_pin_presence(codec, 0x0a)) {
-		stac92xx_reset_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN);
-		stac92xx_set_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN);
-	} else {
-		stac92xx_reset_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN);
-		stac92xx_set_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN);
-	}
-} 
-
-static void stac9872_vaio_unsol_event(struct hda_codec *codec, unsigned int res)
-{
-	switch (res >> 26) {
-	case STAC_HP_EVENT:
-		stac9872_vaio_hp_detect(codec, res);
-		break;
-	}
-}
-
-static struct hda_codec_ops stac9872_vaio_patch_ops = {
-	.build_controls = stac92xx_build_controls,
-	.build_pcms = stac92xx_build_pcms,
-	.init = stac9872_vaio_init,
-	.free = stac92xx_free,
-	.unsol_event = stac9872_vaio_unsol_event,
-#ifdef CONFIG_PM
-	.resume = stac92xx_resume,
-#endif
+static hda_nid_t stac9872_mux_nids[] = {
+	0x15
 };
 
-enum { /* FE and SZ series. id=0x83847661 and subsys=0x104D0700 or 104D1000. */
-       CXD9872RD_VAIO,
-       /* Unknown. id=0x83847662 and subsys=0x104D1200 or 104D1000. */
-       STAC9872AK_VAIO, 
-       /* Unknown. id=0x83847661 and subsys=0x104D1200. */
-       STAC9872K_VAIO,
-       /* AR Series. id=0x83847664 and subsys=104D1300 */
-       CXD9872AKD_VAIO,
-       STAC_9872_MODELS,
+static unsigned int stac9872_vaio_pin_configs[9] = {
+	0x03211020, 0x411111f0, 0x411111f0, 0x03a15030,
+	0x411111f0, 0x90170110, 0x411111f0, 0x411111f0,
+	0x90a7013e
 };
 
 static const char *stac9872_models[STAC_9872_MODELS] = {
-	[CXD9872RD_VAIO]	= "vaio",
-	[CXD9872AKD_VAIO]	= "vaio-ar",
+	[STAC_9872_AUTO] = "auto",
+	[STAC_9872_VAIO] = "vaio",
+};
+
+static unsigned int *stac9872_brd_tbl[STAC_9872_MODELS] = {
+	[STAC_9872_VAIO] = stac9872_vaio_pin_configs,
 };
 
 static struct snd_pci_quirk stac9872_cfg_tbl[] = {
-	SND_PCI_QUIRK(0x104d, 0x81e6, "Sony VAIO F/S", CXD9872RD_VAIO),
-	SND_PCI_QUIRK(0x104d, 0x81ef, "Sony VAIO F/S", CXD9872RD_VAIO),
-	SND_PCI_QUIRK(0x104d, 0x81fd, "Sony VAIO AR", CXD9872AKD_VAIO),
-	SND_PCI_QUIRK(0x104d, 0x8205, "Sony VAIO AR", CXD9872AKD_VAIO),
-	{}
+	{} /* terminator */
 };
 
 static int patch_stac9872(struct hda_codec *codec)
 {
 	struct sigmatel_spec *spec;
-	int board_config;
+	int err;
 
-	board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS,
-						  stac9872_models,
-						  stac9872_cfg_tbl);
-	if (board_config < 0)
-		/* unknown config, let generic-parser do its job... */
-		return snd_hda_parse_generic_codec(codec);
-	
 	spec  = kzalloc(sizeof(*spec), GFP_KERNEL);
 	if (spec == NULL)
 		return -ENOMEM;
-
 	codec->spec = spec;
-	switch (board_config) {
-	case CXD9872RD_VAIO:
-	case STAC9872AK_VAIO:
-	case STAC9872K_VAIO:
-		spec->mixer = vaio_mixer;
-		spec->init = vaio_init;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = ARRAY_SIZE(vaio_dacs);
-		spec->multiout.dac_nids = vaio_dacs;
-		spec->multiout.hp_nid = VAIO_HP_DAC;
-		spec->num_adcs = ARRAY_SIZE(vaio_adcs);
-		spec->adc_nids = vaio_adcs;
-		spec->num_pwrs = 0;
-		spec->input_mux = &vaio_mux;
-		spec->mux_nids = vaio_mux_nids;
-		codec->patch_ops = stac9872_vaio_patch_ops;
-		break;
-	
-	case CXD9872AKD_VAIO:
-		spec->mixer = vaio_ar_mixer;
-		spec->init = vaio_ar_init;
-		spec->multiout.max_channels = 2;
-		spec->multiout.num_dacs = ARRAY_SIZE(vaio_dacs);
-		spec->multiout.dac_nids = vaio_dacs;
-		spec->multiout.hp_nid = VAIO_HP_DAC;
-		spec->num_adcs = ARRAY_SIZE(vaio_adcs);
-		spec->num_pwrs = 0;
-		spec->adc_nids = vaio_adcs;
-		spec->input_mux = &vaio_mux;
-		spec->mux_nids = vaio_mux_nids;
-		codec->patch_ops = stac9872_patch_ops;
-		break;
-	}
 
+	spec->board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS,
+							stac9872_models,
+							stac9872_cfg_tbl);
+	if (spec->board_config < 0)
+		snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9872, "
+			    "using BIOS defaults\n");
+	else
+		stac92xx_set_config_regs(codec,
+					 stac9872_brd_tbl[spec->board_config]);
+
+	spec->num_pins = ARRAY_SIZE(stac9872_pin_nids);
+	spec->pin_nids = stac9872_pin_nids;
+	spec->multiout.dac_nids = spec->dac_nids;
+	spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids);
+	spec->adc_nids = stac9872_adc_nids;
+	spec->num_muxes = ARRAY_SIZE(stac9872_mux_nids);
+	spec->mux_nids = stac9872_mux_nids;
+	spec->mixer = stac9872_mixer;
+	spec->init = stac9872_core_init;
+
+	err = stac92xx_parse_auto_config(codec, 0x10, 0x12);
+	if (err < 0) {
+		stac92xx_free(codec);
+		return -EINVAL;
+	}
+	spec->input_mux = &spec->private_imux;
+	codec->patch_ops = stac92xx_patch_ops;
 	return 0;
 }
 
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index c761394cbe84..b25a5cc637d6 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -1308,16 +1308,13 @@ static void vt1708_set_pinconfig_connect(struct hda_codec *codec, hda_nid_t nid)
 	unsigned int def_conf;
 	unsigned char seqassoc;
 
-	def_conf = snd_hda_codec_read(codec, nid, 0,
-				      AC_VERB_GET_CONFIG_DEFAULT, 0);
+	def_conf = snd_hda_codec_get_pincfg(codec, nid);
 	seqassoc = (unsigned char) get_defcfg_association(def_conf);
 	seqassoc = (seqassoc << 4) | get_defcfg_sequence(def_conf);
 	if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) {
 		if (seqassoc == 0xff) {
 			def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30));
-			snd_hda_codec_write(codec, nid, 0,
-					    AC_VERB_SET_CONFIG_DEFAULT_BYTES_3,
-					    def_conf >> 24);
+			snd_hda_codec_set_pincfg(codec, nid, def_conf);
 		}
 	}
 
@@ -1354,7 +1351,7 @@ static int vt1708_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = VT1708_DIGOUT_NID;
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = VT1708_DIGIN_NID;
@@ -1827,7 +1824,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = VT1709_DIGOUT_NID;
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = VT1709_DIGIN_NID;
@@ -2371,7 +2368,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = VT1708B_DIGOUT_NID;
 	if (spec->autocfg.dig_in_pin)
 		spec->dig_in_nid = VT1708B_DIGIN_NID;
@@ -2836,7 +2833,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = VT1708S_DIGOUT_NID;
 
 	spec->extra_dig_out_nid = 0x15;
@@ -3155,7 +3152,7 @@ static int vt1702_parse_auto_config(struct hda_codec *codec)
 
 	spec->multiout.max_channels = spec->multiout.num_dacs * 2;
 
-	if (spec->autocfg.dig_out_pin)
+	if (spec->autocfg.dig_outs)
 		spec->multiout.dig_out_nid = VT1702_DIGOUT_NID;
 
 	spec->extra_dig_out_nid = 0x1B;
diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c
index 58d7cda03de5..3dd63f1cda53 100644
--- a/sound/pci/ice1712/ice1712.c
+++ b/sound/pci/ice1712/ice1712.c
@@ -458,7 +458,7 @@ static irqreturn_t snd_ice1712_interrupt(int irq, void *dev_id)
 			u16 pbkstatus;
 			struct snd_pcm_substream *substream;
 			pbkstatus = inw(ICEDS(ice, INTSTAT));
-			/* printk("pbkstatus = 0x%x\n", pbkstatus); */
+			/* printk(KERN_DEBUG "pbkstatus = 0x%x\n", pbkstatus); */
 			for (idx = 0; idx < 6; idx++) {
 				if ((pbkstatus & (3 << (idx * 2))) == 0)
 					continue;
@@ -2648,9 +2648,9 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	strcpy(card->driver, "ICE1712");
 	strcpy(card->shortname, "ICEnsemble ICE1712");
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index bb8d8c766b9d..128510e77a78 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -241,6 +241,8 @@ get_rawmidi_substream(struct snd_ice1712 *ice, unsigned int stream)
 				struct snd_rawmidi_substream, list);
 }
 
+static void enable_midi_irq(struct snd_ice1712 *ice, u8 flag, int enable);
+
 static void vt1724_midi_write(struct snd_ice1712 *ice)
 {
 	struct snd_rawmidi_substream *s;
@@ -254,6 +256,11 @@ static void vt1724_midi_write(struct snd_ice1712 *ice)
 		for (i = 0; i < count; ++i)
 			outb(buffer[i], ICEREG1724(ice, MPU_DATA));
 	}
+	/* mask irq when all bytes have been transmitted.
+	 * enabled again in output_trigger when the new data comes in.
+	 */
+	enable_midi_irq(ice, VT1724_IRQ_MPU_TX,
+			!snd_rawmidi_transmit_empty(s));
 }
 
 static void vt1724_midi_read(struct snd_ice1712 *ice)
@@ -272,31 +279,34 @@ static void vt1724_midi_read(struct snd_ice1712 *ice)
 	}
 }
 
-static void vt1724_enable_midi_irq(struct snd_rawmidi_substream *substream,
-				   u8 flag, int enable)
+/* call with ice->reg_lock */
+static void enable_midi_irq(struct snd_ice1712 *ice, u8 flag, int enable)
 {
-	struct snd_ice1712 *ice = substream->rmidi->private_data;
-	u8 mask;
-
-	spin_lock_irq(&ice->reg_lock);
-	mask = inb(ICEREG1724(ice, IRQMASK));
+	u8 mask = inb(ICEREG1724(ice, IRQMASK));
 	if (enable)
 		mask &= ~flag;
 	else
 		mask |= flag;
 	outb(mask, ICEREG1724(ice, IRQMASK));
+}
+
+static void vt1724_enable_midi_irq(struct snd_rawmidi_substream *substream,
+				   u8 flag, int enable)
+{
+	struct snd_ice1712 *ice = substream->rmidi->private_data;
+
+	spin_lock_irq(&ice->reg_lock);
+	enable_midi_irq(ice, flag, enable);
 	spin_unlock_irq(&ice->reg_lock);
 }
 
 static int vt1724_midi_output_open(struct snd_rawmidi_substream *s)
 {
-	vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 1);
 	return 0;
 }
 
 static int vt1724_midi_output_close(struct snd_rawmidi_substream *s)
 {
-	vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 0);
 	return 0;
 }
 
@@ -311,6 +321,7 @@ static void vt1724_midi_output_trigger(struct snd_rawmidi_substream *s, int up)
 		vt1724_midi_write(ice);
 	} else {
 		ice->midi_output = 0;
+		enable_midi_irq(ice, VT1724_IRQ_MPU_TX, 0);
 	}
 	spin_unlock_irqrestore(&ice->reg_lock, flags);
 }
@@ -320,6 +331,7 @@ static void vt1724_midi_output_drain(struct snd_rawmidi_substream *s)
 	struct snd_ice1712 *ice = s->rmidi->private_data;
 	unsigned long timeout;
 
+	vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 0);
 	/* 32 bytes should be transmitted in less than about 12 ms */
 	timeout = jiffies + msecs_to_jiffies(15);
 	do {
@@ -389,24 +401,24 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id)
 		status &= status_mask;
 		if (status == 0)
 			break;
+		spin_lock(&ice->reg_lock);
 		if (++timeout > 10) {
 			status = inb(ICEREG1724(ice, IRQSTAT));
 			printk(KERN_ERR "ice1724: Too long irq loop, "
 			       "status = 0x%x\n", status);
 			if (status & VT1724_IRQ_MPU_TX) {
 				printk(KERN_ERR "ice1724: Disabling MPU_TX\n");
-				outb(inb(ICEREG1724(ice, IRQMASK)) |
-				     VT1724_IRQ_MPU_TX,
-				     ICEREG1724(ice, IRQMASK));
+				enable_midi_irq(ice, VT1724_IRQ_MPU_TX, 0);
 			}
+			spin_unlock(&ice->reg_lock);
 			break;
 		}
 		handled = 1;
 		if (status & VT1724_IRQ_MPU_TX) {
-			spin_lock(&ice->reg_lock);
 			if (ice->midi_output)
 				vt1724_midi_write(ice);
-			spin_unlock(&ice->reg_lock);
+			else
+				enable_midi_irq(ice, VT1724_IRQ_MPU_TX, 0);
 			/* Due to mysterical reasons, MPU_TX is always
 			 * generated (and can't be cleared) when a PCM
 			 * playback is going.  So let's ignore at the
@@ -415,15 +427,14 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id)
 			status_mask &= ~VT1724_IRQ_MPU_TX;
 		}
 		if (status & VT1724_IRQ_MPU_RX) {
-			spin_lock(&ice->reg_lock);
 			if (ice->midi_input)
 				vt1724_midi_read(ice);
 			else
 				vt1724_midi_clear_rx(ice);
-			spin_unlock(&ice->reg_lock);
 		}
 		/* ack MPU irq */
 		outb(status, ICEREG1724(ice, IRQSTAT));
+		spin_unlock(&ice->reg_lock);
 		if (status & VT1724_IRQ_MTPCM) {
 			/*
 			 * Multi-track PCM
@@ -745,7 +756,14 @@ static int snd_vt1724_playback_pro_prepare(struct snd_pcm_substream *substream)
 
 	spin_unlock_irq(&ice->reg_lock);
 
-	/* printk("pro prepare: ch = %d, addr = 0x%x, buffer = 0x%x, period = 0x%x\n", substream->runtime->channels, (unsigned int)substream->runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream)); */
+	/*
+	printk(KERN_DEBUG "pro prepare: ch = %d, addr = 0x%x, "
+	       "buffer = 0x%x, period = 0x%x\n",
+	       substream->runtime->channels,
+	       (unsigned int)substream->runtime->dma_addr,
+	       snd_pcm_lib_buffer_bytes(substream),
+	       snd_pcm_lib_period_bytes(substream));
+	*/
 	return 0;
 }
 
@@ -2122,7 +2140,9 @@ unsigned char snd_vt1724_read_i2c(struct snd_ice1712 *ice,
 	wait_i2c_busy(ice);
 	val = inb(ICEREG1724(ice, I2C_DATA));
 	mutex_unlock(&ice->i2c_mutex);
-	/* printk("i2c_read: [0x%x,0x%x] = 0x%x\n", dev, addr, val); */
+	/*
+	printk(KERN_DEBUG "i2c_read: [0x%x,0x%x] = 0x%x\n", dev, addr, val);
+	*/
 	return val;
 }
 
@@ -2131,7 +2151,9 @@ void snd_vt1724_write_i2c(struct snd_ice1712 *ice,
 {
 	mutex_lock(&ice->i2c_mutex);
 	wait_i2c_busy(ice);
-	/* printk("i2c_write: [0x%x,0x%x] = 0x%x\n", dev, addr, data); */
+	/*
+	printk(KERN_DEBUG "i2c_write: [0x%x,0x%x] = 0x%x\n", dev, addr, data);
+	*/
 	outb(addr, ICEREG1724(ice, I2C_BYTE_ADDR));
 	outb(data, ICEREG1724(ice, I2C_DATA));
 	outb(dev | VT1724_I2C_WRITE, ICEREG1724(ice, I2C_DEV_ADDR));
@@ -2456,9 +2478,9 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	strcpy(card->driver, "ICE1724");
 	strcpy(card->shortname, "ICEnsemble ICE1724");
diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c
index c51659b9caf6..fd948bfd9aef 100644
--- a/sound/pci/ice1712/juli.c
+++ b/sound/pci/ice1712/juli.c
@@ -345,8 +345,9 @@ static int juli_mute_put(struct snd_kcontrol *kcontrol,
 			new_gpio =  old_gpio &
 				~((unsigned int) kcontrol->private_value);
 	}
-	/* printk("JULI - mute/unmute: control_value: 0x%x, old_gpio: 0x%x, \
-		new_gpio 0x%x\n",
+	/* printk(KERN_DEBUG
+		"JULI - mute/unmute: control_value: 0x%x, old_gpio: 0x%x, "
+		"new_gpio 0x%x\n",
 		(unsigned int)ucontrol->value.integer.value[0], old_gpio,
 		new_gpio); */
 	if (old_gpio != new_gpio) {
diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c
index 48d3679292a7..2a8e5cd8f2d8 100644
--- a/sound/pci/ice1712/prodigy192.c
+++ b/sound/pci/ice1712/prodigy192.c
@@ -133,8 +133,10 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
 		idx  = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + STAC946X_LF_VOLUME;
 	/* due to possible conflicts with stac9460_set_rate_val, mutexing */
 	mutex_lock(&spec->mute_mutex);
-	/*printk("Mute put: reg 0x%02x, ctrl value: 0x%02x\n", idx,
-		ucontrol->value.integer.value[0]);*/
+	/*
+	printk(KERN_DEBUG "Mute put: reg 0x%02x, ctrl value: 0x%02x\n", idx,
+	       ucontrol->value.integer.value[0]);
+	*/
 	change = stac9460_dac_mute(ice, idx, ucontrol->value.integer.value[0]);
 	mutex_unlock(&spec->mute_mutex);
 	return change;
@@ -185,7 +187,10 @@ static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el
 	change = (ovol != nvol);
 	if (change) {
 		ovol =  (0x7f - nvol) | (tmp & 0x80);
-		/*printk("DAC Volume: reg 0x%02x: 0x%02x\n", idx, ovol);*/
+		/*
+		printk(KERN_DEBUG "DAC Volume: reg 0x%02x: 0x%02x\n",
+		       idx, ovol);
+		*/
 		stac9460_put(ice, idx, (0x7f - nvol) | (tmp & 0x80));
 	}
 	return change;
@@ -344,7 +349,7 @@ static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate)
 	for (idx = 0; idx < 7 ; ++idx)
 		changed[idx] = stac9460_dac_mute(ice,
 				STAC946X_MASTER_VOLUME + idx, 0);
-	/*printk("Rate change: %d, new MC: 0x%02x\n", rate, new);*/
+	/*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/
 	stac9460_put(ice, STAC946X_MASTER_CLOCKING, new);
 	udelay(10);
 	/* unmuting - only originally unmuted dacs -
diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c
index e900cdc84849..57648810eaf1 100644
--- a/sound/pci/intel8x0.c
+++ b/sound/pci/intel8x0.c
@@ -689,7 +689,7 @@ static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ich
 			bdbar[idx + 1] = cpu_to_le32(0x80000000 | /* interrupt on completion */
 						     ichdev->fragsize >> ichdev->pos_shift);
 #if 0
-			printk("bdbar[%i] = 0x%x [0x%x]\n",
+			printk(KERN_DEBUG "bdbar[%i] = 0x%x [0x%x]\n",
 			       idx + 0, bdbar[idx + 0], bdbar[idx + 1]);
 #endif
 		}
@@ -701,8 +701,10 @@ static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ich
 	ichdev->lvi_frag = ICH_REG_LVI_MASK % ichdev->frags;
 	ichdev->position = 0;
 #if 0
-	printk("lvi_frag = %i, frags = %i, period_size = 0x%x, period_size1 = 0x%x\n",
-			ichdev->lvi_frag, ichdev->frags, ichdev->fragsize, ichdev->fragsize1);
+	printk(KERN_DEBUG "lvi_frag = %i, frags = %i, period_size = 0x%x, "
+	       "period_size1 = 0x%x\n",
+	       ichdev->lvi_frag, ichdev->frags, ichdev->fragsize,
+	       ichdev->fragsize1);
 #endif
 	/* clear interrupts */
 	iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI);
@@ -768,7 +770,8 @@ static inline void snd_intel8x0_update(struct intel8x0 *chip, struct ichdev *ich
 		ichdev->lvi_frag %= ichdev->frags;
 		ichdev->bdbar[ichdev->lvi * 2] = cpu_to_le32(ichdev->physbuf + ichdev->lvi_frag * ichdev->fragsize1);
 #if 0
-	printk("new: bdbar[%i] = 0x%x [0x%x], prefetch = %i, all = 0x%x, 0x%x\n",
+	printk(KERN_DEBUG "new: bdbar[%i] = 0x%x [0x%x], prefetch = %i, "
+	       "all = 0x%x, 0x%x\n",
 	       ichdev->lvi * 2, ichdev->bdbar[ichdev->lvi * 2],
 	       ichdev->bdbar[ichdev->lvi * 2 + 1], inb(ICH_REG_OFF_PIV + port),
 	       inl(port + 4), inb(port + ICH_REG_OFF_CR));
@@ -2287,23 +2290,23 @@ static void do_ali_reset(struct intel8x0 *chip)
 	iputdword(chip, ICHREG(ALI_INTERRUPTSR), 0x00000000);
 }
 
-static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing)
-{
-	unsigned long end_time;
-	unsigned int cnt, status, nstatus;
-	
-	/* put logic to right state */
-	/* first clear status bits */
-	status = ICH_RCS | ICH_MCINT | ICH_POINT | ICH_PIINT;
-	if (chip->device_type == DEVICE_NFORCE)
-		status |= ICH_NVSPINT;
-	cnt = igetdword(chip, ICHREG(GLOB_STA));
-	iputdword(chip, ICHREG(GLOB_STA), cnt & status);
+#ifdef CONFIG_SND_AC97_POWER_SAVE
+static struct snd_pci_quirk ich_chip_reset_mode[] = {
+	SND_PCI_QUIRK(0x1014, 0x051f, "Thinkpad R32", 1),
+	{ } /* end */
+};
 
+static int snd_intel8x0_ich_chip_cold_reset(struct intel8x0 *chip)
+{
+	unsigned int cnt;
 	/* ACLink on, 2 channels */
+
+	if (snd_pci_quirk_lookup(chip->pci, ich_chip_reset_mode))
+		return -EIO;
+
 	cnt = igetdword(chip, ICHREG(GLOB_CNT));
 	cnt &= ~(ICH_ACLINK | ICH_PCM_246_MASK);
-#ifdef CONFIG_SND_AC97_POWER_SAVE
+
 	/* do cold reset - the full ac97 powerdown may leave the controller
 	 * in a warm state but actually it cannot communicate with the codec.
 	 */
@@ -2312,22 +2315,58 @@ static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing)
 	udelay(10);
 	iputdword(chip, ICHREG(GLOB_CNT), cnt | ICH_AC97COLD);
 	msleep(1);
+	return 0;
+}
+#define snd_intel8x0_ich_chip_can_cold_reset(chip) \
+	(!snd_pci_quirk_lookup(chip->pci, ich_chip_reset_mode))
 #else
+#define snd_intel8x0_ich_chip_cold_reset(chip)	0
+#define snd_intel8x0_ich_chip_can_cold_reset(chip) (0)
+#endif
+
+static int snd_intel8x0_ich_chip_reset(struct intel8x0 *chip)
+{
+	unsigned long end_time;
+	unsigned int cnt;
+	/* ACLink on, 2 channels */
+	cnt = igetdword(chip, ICHREG(GLOB_CNT));
+	cnt &= ~(ICH_ACLINK | ICH_PCM_246_MASK);
 	/* finish cold or do warm reset */
 	cnt |= (cnt & ICH_AC97COLD) == 0 ? ICH_AC97COLD : ICH_AC97WARM;
 	iputdword(chip, ICHREG(GLOB_CNT), cnt);
 	end_time = (jiffies + (HZ / 4)) + 1;
 	do {
 		if ((igetdword(chip, ICHREG(GLOB_CNT)) & ICH_AC97WARM) == 0)
-			goto __ok;
+			return 0;
 		schedule_timeout_uninterruptible(1);
 	} while (time_after_eq(end_time, jiffies));
 	snd_printk(KERN_ERR "AC'97 warm reset still in progress? [0x%x]\n",
 		   igetdword(chip, ICHREG(GLOB_CNT)));
 	return -EIO;
+}
+
+static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing)
+{
+	unsigned long end_time;
+	unsigned int status, nstatus;
+	unsigned int cnt;
+	int err;
+
+	/* put logic to right state */
+	/* first clear status bits */
+	status = ICH_RCS | ICH_MCINT | ICH_POINT | ICH_PIINT;
+	if (chip->device_type == DEVICE_NFORCE)
+		status |= ICH_NVSPINT;
+	cnt = igetdword(chip, ICHREG(GLOB_STA));
+	iputdword(chip, ICHREG(GLOB_STA), cnt & status);
+
+	if (snd_intel8x0_ich_chip_can_cold_reset(chip))
+		err = snd_intel8x0_ich_chip_cold_reset(chip);
+	else
+		err = snd_intel8x0_ich_chip_reset(chip);
+	if (err < 0)
+		return err;
 
-      __ok:
-#endif
 	if (probing) {
 		/* wait for any codec ready status.
 		 * Once it becomes ready it should remain ready
@@ -3058,9 +3097,9 @@ static int __devinit snd_intel8x0_probe(struct pci_dev *pci,
 	int err;
 	struct shortname_table *name;
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	if (spdif_aclink < 0)
 		spdif_aclink = check_default_spdif_aclink(pci);
diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c
index 93449e464566..6ec0fc50d6be 100644
--- a/sound/pci/intel8x0m.c
+++ b/sound/pci/intel8x0m.c
@@ -411,7 +411,10 @@ static void snd_intel8x0_setup_periods(struct intel8x0m *chip, struct ichdev *ic
 			bdbar[idx + 0] = cpu_to_le32(ichdev->physbuf + (((idx >> 1) * ichdev->fragsize) % ichdev->size));
 			bdbar[idx + 1] = cpu_to_le32(0x80000000 | /* interrupt on completion */
 						     ichdev->fragsize >> chip->pcm_pos_shift);
-			// printk("bdbar[%i] = 0x%x [0x%x]\n", idx + 0, bdbar[idx + 0], bdbar[idx + 1]);
+			/*
+			printk(KERN_DEBUG "bdbar[%i] = 0x%x [0x%x]\n",
+			       idx + 0, bdbar[idx + 0], bdbar[idx + 1]);
+			*/
 		}
 		ichdev->frags = ichdev->size / ichdev->fragsize;
 	}
@@ -421,8 +424,10 @@ static void snd_intel8x0_setup_periods(struct intel8x0m *chip, struct ichdev *ic
 	ichdev->lvi_frag = ICH_REG_LVI_MASK % ichdev->frags;
 	ichdev->position = 0;
 #if 0
-	printk("lvi_frag = %i, frags = %i, period_size = 0x%x, period_size1 = 0x%x\n",
-			ichdev->lvi_frag, ichdev->frags, ichdev->fragsize, ichdev->fragsize1);
+	printk(KERN_DEBUG "lvi_frag = %i, frags = %i, period_size = 0x%x, "
+	       "period_size1 = 0x%x\n",
+	       ichdev->lvi_frag, ichdev->frags, ichdev->fragsize,
+	       ichdev->fragsize1);
 #endif
 	/* clear interrupts */
 	iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI);
@@ -465,7 +470,8 @@ static inline void snd_intel8x0_update(struct intel8x0m *chip, struct ichdev *ic
 							     ichdev->lvi_frag *
 							     ichdev->fragsize1);
 #if 0
-		printk("new: bdbar[%i] = 0x%x [0x%x], prefetch = %i, all = 0x%x, 0x%x\n",
+		printk(KERN_DEBUG "new: bdbar[%i] = 0x%x [0x%x], "
+		       "prefetch = %i, all = 0x%x, 0x%x\n",
 		       ichdev->lvi * 2, ichdev->bdbar[ichdev->lvi * 2],
 		       ichdev->bdbar[ichdev->lvi * 2 + 1], inb(ICH_REG_OFF_PIV + port),
 		       inl(port + 4), inb(port + ICH_REG_OFF_CR));
@@ -1269,9 +1275,9 @@ static int __devinit snd_intel8x0m_probe(struct pci_dev *pci,
 	int err;
 	struct shortname_table *name;
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	strcpy(card->driver, "ICH-MODEM");
 	strcpy(card->shortname, "Intel ICH");
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 5f8006b42750..8b79969034be 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -2443,9 +2443,9 @@ snd_korg1212_probe(struct pci_dev *pci,
 		dev++;
 		return -ENOENT;
 	}
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-        if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
         if ((err = snd_korg1212_create(card, pci, &korg1212)) < 0) {
 		snd_card_free(card);
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index 59bbaf8f3e5b..70141548f251 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -2691,9 +2691,9 @@ snd_m3_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	switch (pci->device) {
 	case PCI_DEVICE_ID_ESS_ALLEGRO:
diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c
index f23a73577c22..c1eb84a14c42 100644
--- a/sound/pci/mixart/mixart.c
+++ b/sound/pci/mixart/mixart.c
@@ -607,6 +607,7 @@ static int snd_mixart_hw_params(struct snd_pcm_substream *subs,
 	/* set the format to the board */
 	err = mixart_set_format(stream, format);
 	if(err < 0) {
+		mutex_unlock(&mgr->setup_mutex);
 		return err;
 	}
 
@@ -1365,12 +1366,12 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci,
 		else
 			idx = index[dev] + i;
 		snprintf(tmpid, sizeof(tmpid), "%s-%d", id[dev] ? id[dev] : "MIXART", i);
-		card = snd_card_new(idx, tmpid, THIS_MODULE, 0);
+		err = snd_card_create(idx, tmpid, THIS_MODULE, 0, &card);
 
-		if (! card) {
+		if (err < 0) {
 			snd_printk(KERN_ERR "cannot allocate the card %d\n", i);
 			snd_mixart_free(mgr);
-			return -ENOMEM;
+			return err;
 		}
 
 		strcpy(card->driver, CARD_NAME);
diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c
index 3782b52bc0e8..4cf4cd8c939c 100644
--- a/sound/pci/mixart/mixart_hwdep.c
+++ b/sound/pci/mixart/mixart_hwdep.c
@@ -345,8 +345,8 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw
 	status_daught = readl_be( MIXART_MEM( mgr,MIXART_PSEUDOREG_DXLX_STATUS_OFFSET ));
 
 	/* motherboard xilinx status 5 will say that the board is performing a reset */
-	if( status_xilinx == 5 ) {
-		snd_printk( KERN_ERR "miXart is resetting !\n");
+	if (status_xilinx == 5) {
+		snd_printk(KERN_ERR "miXart is resetting !\n");
 		return -EAGAIN; /* try again later */
 	}
 
@@ -354,13 +354,14 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw
 	case MIXART_MOTHERBOARD_XLX_INDEX:
 
 		/* xilinx already loaded ? */ 
-		if( status_xilinx == 4 ) {
-			snd_printk( KERN_DEBUG "xilinx is already loaded !\n");
+		if (status_xilinx == 4) {
+			snd_printk(KERN_DEBUG "xilinx is already loaded !\n");
 			return 0;
 		}
 		/* the status should be 0 == "idle" */
-		if( status_xilinx != 0 ) {
-			snd_printk( KERN_ERR "xilinx load error ! status = %d\n", status_xilinx);
+		if (status_xilinx != 0) {
+			snd_printk(KERN_ERR "xilinx load error ! status = %d\n",
+				   status_xilinx);
 			return -EIO; /* modprob -r may help ? */
 		}
 
@@ -389,21 +390,23 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw
 
 	case MIXART_MOTHERBOARD_ELF_INDEX:
 
-		if( status_elf == 4 ) {
-			snd_printk( KERN_DEBUG "elf file already loaded !\n");
+		if (status_elf == 4) {
+			snd_printk(KERN_DEBUG "elf file already loaded !\n");
 			return 0;
 		}
 
 		/* the status should be 0 == "idle" */
-		if( status_elf != 0 ) {
-			snd_printk( KERN_ERR "elf load error ! status = %d\n", status_elf);
+		if (status_elf != 0) {
+			snd_printk(KERN_ERR "elf load error ! status = %d\n",
+				   status_elf);
 			return -EIO; /* modprob -r may help ? */
 		}
 
 		/* wait for xilinx status == 4 */
 		err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_MXLX_STATUS_OFFSET, 1, 4, 500); /* 5sec */
 		if (err < 0) {
-			snd_printk( KERN_ERR "xilinx was not loaded or could not be started\n");
+			snd_printk(KERN_ERR "xilinx was not loaded or "
+				   "could not be started\n");
 			return err;
 		}
 
@@ -424,7 +427,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw
 		/* wait for elf status == 4 */
 		err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_ELF_STATUS_OFFSET, 1, 4, 300); /* 3sec */
 		if (err < 0) {
-			snd_printk( KERN_ERR "elf could not be started\n");
+			snd_printk(KERN_ERR "elf could not be started\n");
 			return err;
 		}
 
@@ -437,15 +440,16 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw
 	default:
 
 		/* elf and xilinx should be loaded */
-		if( (status_elf != 4) || (status_xilinx != 4) ) {
-			printk( KERN_ERR "xilinx or elf not successfully loaded\n");
+		if (status_elf != 4 || status_xilinx != 4) {
+			printk(KERN_ERR "xilinx or elf not "
+			       "successfully loaded\n");
 			return -EIO; /* modprob -r may help ? */
 		}
 
 		/* wait for daughter detection != 0 */
 		err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DBRD_PRESENCE_OFFSET, 0, 0, 30); /* 300msec */
 		if (err < 0) {
-			snd_printk( KERN_ERR "error starting elf file\n");
+			snd_printk(KERN_ERR "error starting elf file\n");
 			return err;
 		}
 
@@ -460,8 +464,9 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw
 			return -EINVAL;
 
 		/* daughter should be idle */
-		if( status_daught != 0 ) {
-			printk( KERN_ERR "daughter load error ! status = %d\n", status_daught);
+		if (status_daught != 0) {
+			printk(KERN_ERR "daughter load error ! status = %d\n",
+			       status_daught);
 			return -EIO; /* modprob -r may help ? */
 		}
  
@@ -480,7 +485,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw
 		/* wait for status == 2 */
 		err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DXLX_STATUS_OFFSET, 1, 2, 30); /* 300msec */
 		if (err < 0) {
-			snd_printk( KERN_ERR "daughter board load error\n");
+			snd_printk(KERN_ERR "daughter board load error\n");
 			return err;
 		}
 
@@ -502,7 +507,8 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw
         /* wait for daughter status == 3 */
         err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DXLX_STATUS_OFFSET, 1, 3, 300); /* 3sec */
         if (err < 0) {
-		snd_printk( KERN_ERR "daughter board could not be initialised\n");
+		snd_printk(KERN_ERR
+			   "daughter board could not be initialised\n");
 		return err;
 	}
 
@@ -512,7 +518,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw
 	/* first communication with embedded */
 	err = mixart_first_init(mgr);
         if (err < 0) {
-		snd_printk( KERN_ERR "miXart could not be set up\n");
+		snd_printk(KERN_ERR "miXart could not be set up\n");
 		return err;
 	}
 
@@ -581,16 +587,6 @@ MODULE_FIRMWARE("mixart/miXart8AES.xlx");
 /* miXart hwdep interface id string */
 #define SND_MIXART_HWDEP_ID       "miXart Loader"
 
-static int mixart_hwdep_open(struct snd_hwdep *hw, struct file *file)
-{
-	return 0;
-}
-
-static int mixart_hwdep_release(struct snd_hwdep *hw, struct file *file)
-{
-	return 0;
-}
-
 static int mixart_hwdep_dsp_status(struct snd_hwdep *hw,
 				   struct snd_hwdep_dsp_status *info)
 {
@@ -643,8 +639,6 @@ int snd_mixart_setup_firmware(struct mixart_mgr *mgr)
 
 	hw->iface = SNDRV_HWDEP_IFACE_MIXART;
 	hw->private_data = mgr;
-	hw->ops.open = mixart_hwdep_open;
-	hw->ops.release = mixart_hwdep_release;
 	hw->ops.dsp_status = mixart_hwdep_dsp_status;
 	hw->ops.dsp_load = mixart_hwdep_dsp_load;
 	hw->exclusive = 1;
diff --git a/sound/pci/nm256/nm256.c b/sound/pci/nm256/nm256.c
index 50c9f8a05082..522a040855d4 100644
--- a/sound/pci/nm256/nm256.c
+++ b/sound/pci/nm256/nm256.c
@@ -1668,9 +1668,9 @@ static int __devinit snd_nm256_probe(struct pci_dev *pci,
 		}
 	}
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	switch (pci->device) {
 	case PCI_DEVICE_ID_NEOMAGIC_NM256AV_AUDIO:
diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c
index 1ab833f843eb..84ef13183419 100644
--- a/sound/pci/oxygen/hifier.c
+++ b/sound/pci/oxygen/hifier.c
@@ -45,6 +45,7 @@ MODULE_PARM_DESC(enable, "enable card");
 static struct pci_device_id hifier_ids[] __devinitdata = {
 	{ OXYGEN_PCI_SUBID(0x14c3, 0x1710) },
 	{ OXYGEN_PCI_SUBID(0x14c3, 0x1711) },
+	{ OXYGEN_PCI_SUBID_BROKEN_EEPROM },
 	{ }
 };
 MODULE_DEVICE_TABLE(pci, hifier_ids);
@@ -151,7 +152,6 @@ static const struct oxygen_model model_hifier = {
 	.shortname = "C-Media CMI8787",
 	.longname = "C-Media Oxygen HD Audio",
 	.chip = "CMI8788",
-	.owner = THIS_MODULE,
 	.init = hifier_init,
 	.control_filter = hifier_control_filter,
 	.cleanup = hifier_cleanup,
@@ -173,6 +173,13 @@ static const struct oxygen_model model_hifier = {
 	.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
 };
 
+static int __devinit get_hifier_model(struct oxygen *chip,
+				      const struct pci_device_id *id)
+{
+	chip->model = model_hifier;
+	return 0;
+}
+
 static int __devinit hifier_probe(struct pci_dev *pci,
 				  const struct pci_device_id *pci_id)
 {
@@ -185,7 +192,8 @@ static int __devinit hifier_probe(struct pci_dev *pci,
 		++dev;
 		return -ENOENT;
 	}
-	err = oxygen_pci_probe(pci, index[dev], id[dev], &model_hifier, 0);
+	err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE,
+			       hifier_ids, get_hifier_model);
 	if (err >= 0)
 		++dev;
 	return err;
diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c
index de999c6d6dd3..72db4c39007f 100644
--- a/sound/pci/oxygen/oxygen.c
+++ b/sound/pci/oxygen/oxygen.c
@@ -1,5 +1,5 @@
 /*
- * C-Media CMI8788 driver for C-Media's reference design and for the X-Meridian
+ * C-Media CMI8788 driver for C-Media's reference design and similar models
  *
  * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
  *
@@ -26,6 +26,7 @@
  *
  * GPIO 0 -> DFS0 of AK5385
  * GPIO 1 -> DFS1 of AK5385
+ * GPIO 8 -> enable headphone amplifier on HT-Omega models
  */
 
 #include <linux/delay.h>
@@ -61,7 +62,8 @@ MODULE_PARM_DESC(enable, "enable card");
 enum {
 	MODEL_CMEDIA_REF,	/* C-Media's reference design */
 	MODEL_MERIDIAN,		/* AuzenTech X-Meridian */
-	MODEL_HALO,		/* HT-Omega Claro halo */
+	MODEL_CLARO,		/* HT-Omega Claro */
+	MODEL_CLARO_HALO,	/* HT-Omega Claro halo */
 };
 
 static struct pci_device_id oxygen_ids[] __devinitdata = {
@@ -74,8 +76,8 @@ static struct pci_device_id oxygen_ids[] __devinitdata = {
 	{ OXYGEN_PCI_SUBID(0x147a, 0xa017), .driver_data = MODEL_CMEDIA_REF },
 	{ OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF },
 	{ OXYGEN_PCI_SUBID(0x415a, 0x5431), .driver_data = MODEL_MERIDIAN },
-	{ OXYGEN_PCI_SUBID(0x7284, 0x9761), .driver_data = MODEL_CMEDIA_REF },
-	{ OXYGEN_PCI_SUBID(0x7284, 0x9781), .driver_data = MODEL_HALO },
+	{ OXYGEN_PCI_SUBID(0x7284, 0x9761), .driver_data = MODEL_CLARO },
+	{ OXYGEN_PCI_SUBID(0x7284, 0x9781), .driver_data = MODEL_CLARO_HALO },
 	{ }
 };
 MODULE_DEVICE_TABLE(pci, oxygen_ids);
@@ -86,6 +88,8 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids);
 #define GPIO_AK5385_DFS_DOUBLE	0x0001
 #define GPIO_AK5385_DFS_QUAD	0x0002
 
+#define GPIO_CLARO_HP		0x0100
+
 struct generic_data {
 	u8 ak4396_ctl2;
 	u16 saved_wm8785_registers[2];
@@ -196,10 +200,46 @@ static void meridian_init(struct oxygen *chip)
 	ak5385_init(chip);
 }
 
+static void claro_enable_hp(struct oxygen *chip)
+{
+	msleep(300);
+	oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_CLARO_HP);
+	oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_CLARO_HP);
+}
+
+static void claro_init(struct oxygen *chip)
+{
+	ak4396_init(chip);
+	wm8785_init(chip);
+	claro_enable_hp(chip);
+}
+
+static void claro_halo_init(struct oxygen *chip)
+{
+	ak4396_init(chip);
+	ak5385_init(chip);
+	claro_enable_hp(chip);
+}
+
 static void generic_cleanup(struct oxygen *chip)
 {
 }
 
+static void claro_disable_hp(struct oxygen *chip)
+{
+	oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_CLARO_HP);
+}
+
+static void claro_cleanup(struct oxygen *chip)
+{
+	claro_disable_hp(chip);
+}
+
+static void claro_suspend(struct oxygen *chip)
+{
+	claro_disable_hp(chip);
+}
+
 static void generic_resume(struct oxygen *chip)
 {
 	ak4396_registers_init(chip);
@@ -211,6 +251,12 @@ static void meridian_resume(struct oxygen *chip)
 	ak4396_registers_init(chip);
 }
 
+static void claro_resume(struct oxygen *chip)
+{
+	ak4396_registers_init(chip);
+	claro_enable_hp(chip);
+}
+
 static void set_ak4396_params(struct oxygen *chip,
 			      struct snd_pcm_hw_params *params)
 {
@@ -293,30 +339,10 @@ static void set_ak5385_params(struct oxygen *chip,
 
 static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0);
 
-static int generic_probe(struct oxygen *chip, unsigned long driver_data)
-{
-	if (driver_data == MODEL_MERIDIAN) {
-		chip->model.init = meridian_init;
-		chip->model.resume = meridian_resume;
-		chip->model.set_adc_params = set_ak5385_params;
-		chip->model.device_config = PLAYBACK_0_TO_I2S |
-					    PLAYBACK_1_TO_SPDIF |
-					    CAPTURE_0_FROM_I2S_2 |
-					    CAPTURE_1_FROM_SPDIF;
-	}
-	if (driver_data == MODEL_MERIDIAN || driver_data == MODEL_HALO) {
-		chip->model.misc_flags = OXYGEN_MISC_MIDI;
-		chip->model.device_config |= MIDI_OUTPUT | MIDI_INPUT;
-	}
-	return 0;
-}
-
 static const struct oxygen_model model_generic = {
 	.shortname = "C-Media CMI8788",
 	.longname = "C-Media Oxygen HD Audio",
 	.chip = "CMI8788",
-	.owner = THIS_MODULE,
-	.probe = generic_probe,
 	.init = generic_init,
 	.cleanup = generic_cleanup,
 	.resume = generic_resume,
@@ -341,6 +367,42 @@ static const struct oxygen_model model_generic = {
 	.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
 };
 
+static int __devinit get_oxygen_model(struct oxygen *chip,
+				      const struct pci_device_id *id)
+{
+	chip->model = model_generic;
+	switch (id->driver_data) {
+	case MODEL_MERIDIAN:
+		chip->model.init = meridian_init;
+		chip->model.resume = meridian_resume;
+		chip->model.set_adc_params = set_ak5385_params;
+		chip->model.device_config = PLAYBACK_0_TO_I2S |
+					    PLAYBACK_1_TO_SPDIF |
+					    CAPTURE_0_FROM_I2S_2 |
+					    CAPTURE_1_FROM_SPDIF;
+		break;
+	case MODEL_CLARO:
+		chip->model.init = claro_init;
+		chip->model.cleanup = claro_cleanup;
+		chip->model.suspend = claro_suspend;
+		chip->model.resume = claro_resume;
+		break;
+	case MODEL_CLARO_HALO:
+		chip->model.init = claro_halo_init;
+		chip->model.cleanup = claro_cleanup;
+		chip->model.suspend = claro_suspend;
+		chip->model.resume = claro_resume;
+		chip->model.set_adc_params = set_ak5385_params;
+		break;
+	}
+	if (id->driver_data == MODEL_MERIDIAN ||
+	    id->driver_data == MODEL_CLARO_HALO) {
+		chip->model.misc_flags = OXYGEN_MISC_MIDI;
+		chip->model.device_config |= MIDI_OUTPUT | MIDI_INPUT;
+	}
+	return 0;
+}
+
 static int __devinit generic_oxygen_probe(struct pci_dev *pci,
 					  const struct pci_device_id *pci_id)
 {
@@ -353,8 +415,8 @@ static int __devinit generic_oxygen_probe(struct pci_dev *pci,
 		++dev;
 		return -ENOENT;
 	}
-	err = oxygen_pci_probe(pci, index[dev], id[dev],
-			       &model_generic, pci_id->driver_data);
+	err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE,
+			       oxygen_ids, get_oxygen_model);
 	if (err >= 0)
 		++dev;
 	return err;
diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h
index 19107c6307e5..bd615dbffadb 100644
--- a/sound/pci/oxygen/oxygen.h
+++ b/sound/pci/oxygen/oxygen.h
@@ -18,6 +18,8 @@
 
 #define OXYGEN_IO_SIZE	0x100
 
+#define OXYGEN_EEPROM_ID	0x434d	/* "CM" */
+
 /* model-specific configuration of outputs/inputs */
 #define PLAYBACK_0_TO_I2S	0x0001
      /* PLAYBACK_0_TO_AC97_0		not implemented */
@@ -49,7 +51,13 @@ enum {
 	.subvendor = sv, \
 	.subdevice = sd
 
+#define BROKEN_EEPROM_DRIVER_DATA ((unsigned long)-1)
+#define OXYGEN_PCI_SUBID_BROKEN_EEPROM \
+	OXYGEN_PCI_SUBID(PCI_VENDOR_ID_CMEDIA, 0x8788), \
+	.driver_data = BROKEN_EEPROM_DRIVER_DATA
+
 struct pci_dev;
+struct pci_device_id;
 struct snd_card;
 struct snd_pcm_substream;
 struct snd_pcm_hardware;
@@ -62,8 +70,6 @@ struct oxygen_model {
 	const char *shortname;
 	const char *longname;
 	const char *chip;
-	struct module *owner;
-	int (*probe)(struct oxygen *chip, unsigned long driver_data);
 	void (*init)(struct oxygen *chip);
 	int (*control_filter)(struct snd_kcontrol_new *template);
 	int (*mixer_init)(struct oxygen *chip);
@@ -83,6 +89,7 @@ struct oxygen_model {
 	void (*ac97_switch)(struct oxygen *chip,
 			    unsigned int reg, unsigned int mute);
 	const unsigned int *dac_tlv;
+	unsigned long private_data;
 	size_t model_data_size;
 	unsigned int device_config;
 	u8 dac_channels;
@@ -134,8 +141,12 @@ struct oxygen {
 /* oxygen_lib.c */
 
 int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
-		     const struct oxygen_model *model,
-		     unsigned long driver_data);
+		     struct module *owner,
+		     const struct pci_device_id *ids,
+		     int (*get_model)(struct oxygen *chip,
+				      const struct pci_device_id *id
+				     )
+		    );
 void oxygen_pci_remove(struct pci_dev *pci);
 #ifdef CONFIG_PM
 int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state);
@@ -180,6 +191,9 @@ void oxygen_write_i2c(struct oxygen *chip, u8 device, u8 map, u8 data);
 void oxygen_reset_uart(struct oxygen *chip);
 void oxygen_write_uart(struct oxygen *chip, u8 data);
 
+u16 oxygen_read_eeprom(struct oxygen *chip, unsigned int index);
+void oxygen_write_eeprom(struct oxygen *chip, unsigned int index, u16 value);
+
 static inline void oxygen_set_bits8(struct oxygen *chip,
 				    unsigned int reg, u8 value)
 {
diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c
index 3126c4b403dd..c1eb923f2ac9 100644
--- a/sound/pci/oxygen/oxygen_io.c
+++ b/sound/pci/oxygen/oxygen_io.c
@@ -254,3 +254,34 @@ void oxygen_write_uart(struct oxygen *chip, u8 data)
 	_write_uart(chip, 0, data);
 }
 EXPORT_SYMBOL(oxygen_write_uart);
+
+u16 oxygen_read_eeprom(struct oxygen *chip, unsigned int index)
+{
+	unsigned int timeout;
+
+	oxygen_write8(chip, OXYGEN_EEPROM_CONTROL,
+		      index | OXYGEN_EEPROM_DIR_READ);
+	for (timeout = 0; timeout < 100; ++timeout) {
+		udelay(1);
+		if (!(oxygen_read8(chip, OXYGEN_EEPROM_STATUS)
+		      & OXYGEN_EEPROM_BUSY))
+			break;
+	}
+	return oxygen_read16(chip, OXYGEN_EEPROM_DATA);
+}
+
+void oxygen_write_eeprom(struct oxygen *chip, unsigned int index, u16 value)
+{
+	unsigned int timeout;
+
+	oxygen_write16(chip, OXYGEN_EEPROM_DATA, value);
+	oxygen_write8(chip, OXYGEN_EEPROM_CONTROL,
+		      index | OXYGEN_EEPROM_DIR_WRITE);
+	for (timeout = 0; timeout < 10; ++timeout) {
+		msleep(1);
+		if (!(oxygen_read8(chip, OXYGEN_EEPROM_STATUS)
+		      & OXYGEN_EEPROM_BUSY))
+			return;
+	}
+	snd_printk(KERN_ERR "EEPROM write timeout\n");
+}
diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c
index 84f481d41efa..312251d39696 100644
--- a/sound/pci/oxygen/oxygen_lib.c
+++ b/sound/pci/oxygen/oxygen_lib.c
@@ -34,6 +34,7 @@ MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
 MODULE_DESCRIPTION("C-Media CMI8788 helper library");
 MODULE_LICENSE("GPL v2");
 
+#define DRIVER "oxygen"
 
 static inline int oxygen_uart_input_ready(struct oxygen *chip)
 {
@@ -243,6 +244,62 @@ static void oxygen_proc_init(struct oxygen *chip)
 #define oxygen_proc_init(chip)
 #endif
 
+static const struct pci_device_id *
+oxygen_search_pci_id(struct oxygen *chip, const struct pci_device_id ids[])
+{
+	u16 subdevice;
+
+	/*
+	 * Make sure the EEPROM pins are available, i.e., not used for SPI.
+	 * (This function is called before we initialize or use SPI.)
+	 */
+	oxygen_clear_bits8(chip, OXYGEN_FUNCTION,
+			   OXYGEN_FUNCTION_ENABLE_SPI_4_5);
+	/*
+	 * Read the subsystem device ID directly from the EEPROM, because the
+	 * chip didn't if the first EEPROM word was overwritten.
+	 */
+	subdevice = oxygen_read_eeprom(chip, 2);
+	/*
+	 * We use only the subsystem device ID for searching because it is
+	 * unique even without the subsystem vendor ID, which may have been
+	 * overwritten in the EEPROM.
+	 */
+	for (; ids->vendor; ++ids)
+		if (ids->subdevice == subdevice &&
+		    ids->driver_data != BROKEN_EEPROM_DRIVER_DATA)
+			return ids;
+	return NULL;
+}
+
+static void oxygen_restore_eeprom(struct oxygen *chip,
+				  const struct pci_device_id *id)
+{
+	if (oxygen_read_eeprom(chip, 0) != OXYGEN_EEPROM_ID) {
+		/*
+		 * This function gets called only when a known card model has
+		 * been detected, i.e., we know there is a valid subsystem
+		 * product ID at index 2 in the EEPROM.  Therefore, we have
+		 * been able to deduce the correct subsystem vendor ID, and
+		 * this is enough information to restore the original EEPROM
+		 * contents.
+		 */
+		oxygen_write_eeprom(chip, 1, id->subvendor);
+		oxygen_write_eeprom(chip, 0, OXYGEN_EEPROM_ID);
+
+		oxygen_set_bits8(chip, OXYGEN_MISC,
+				 OXYGEN_MISC_WRITE_PCI_SUBID);
+		pci_write_config_word(chip->pci, PCI_SUBSYSTEM_VENDOR_ID,
+				      id->subvendor);
+		pci_write_config_word(chip->pci, PCI_SUBSYSTEM_ID,
+				      id->subdevice);
+		oxygen_clear_bits8(chip, OXYGEN_MISC,
+				   OXYGEN_MISC_WRITE_PCI_SUBID);
+
+		snd_printk(KERN_INFO "EEPROM ID restored\n");
+	}
+}
+
 static void oxygen_init(struct oxygen *chip)
 {
 	unsigned int i;
@@ -446,30 +503,33 @@ static void oxygen_card_free(struct snd_card *card)
 		free_irq(chip->irq, chip);
 	flush_scheduled_work();
 	chip->model.cleanup(chip);
+	kfree(chip->model_data);
 	mutex_destroy(&chip->mutex);
 	pci_release_regions(chip->pci);
 	pci_disable_device(chip->pci);
 }
 
 int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
-		     const struct oxygen_model *model,
-		     unsigned long driver_data)
+		     struct module *owner,
+		     const struct pci_device_id *ids,
+		     int (*get_model)(struct oxygen *chip,
+				      const struct pci_device_id *id
+				     )
+		    )
 {
 	struct snd_card *card;
 	struct oxygen *chip;
+	const struct pci_device_id *pci_id;
 	int err;
 
-	card = snd_card_new(index, id, model->owner,
-			    sizeof *chip + model->model_data_size);
-	if (!card)
-		return -ENOMEM;
+	err = snd_card_create(index, id, owner, sizeof(*chip), &card);
+	if (err < 0)
+		return err;
 
 	chip = card->private_data;
 	chip->card = card;
 	chip->pci = pci;
 	chip->irq = -1;
-	chip->model = *model;
-	chip->model_data = chip + 1;
 	spin_lock_init(&chip->reg_lock);
 	mutex_init(&chip->mutex);
 	INIT_WORK(&chip->spdif_input_bits_work,
@@ -481,7 +541,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 	if (err < 0)
 		goto err_card;
 
-	err = pci_request_regions(pci, model->chip);
+	err = pci_request_regions(pci, DRIVER);
 	if (err < 0) {
 		snd_printk(KERN_ERR "cannot reserve PCI resources\n");
 		goto err_pci_enable;
@@ -495,20 +555,34 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id,
 	}
 	chip->addr = pci_resource_start(pci, 0);
 
+	pci_id = oxygen_search_pci_id(chip, ids);
+	if (!pci_id) {
+		err = -ENODEV;
+		goto err_pci_regions;
+	}
+	oxygen_restore_eeprom(chip, pci_id);
+	err = get_model(chip, pci_id);
+	if (err < 0)
+		goto err_pci_regions;
+
+	if (chip->model.model_data_size) {
+		chip->model_data = kzalloc(chip->model.model_data_size,
+					   GFP_KERNEL);
+		if (!chip->model_data) {
+			err = -ENOMEM;
+			goto err_pci_regions;
+		}
+	}
+
 	pci_set_master(pci);
 	snd_card_set_dev(card, &pci->dev);
 	card->private_free = oxygen_card_free;
 
-	if (chip->model.probe) {
-		err = chip->model.probe(chip, driver_data);
-		if (err < 0)
-			goto err_card;
-	}
 	oxygen_init(chip);
 	chip->model.init(chip);
 
 	err = request_irq(pci->irq, oxygen_interrupt, IRQF_SHARED,
-			  chip->model.chip, chip);
+			  DRIVER, chip);
 	if (err < 0) {
 		snd_printk(KERN_ERR "cannot grab interrupt %d\n", pci->irq);
 		goto err_card;
diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c
index 6c870c12a177..bc5ce11c8b14 100644
--- a/sound/pci/oxygen/virtuoso.c
+++ b/sound/pci/oxygen/virtuoso.c
@@ -112,6 +112,34 @@
  * CS4362A: AD0 <- 0
  */
 
+/*
+ * Xonar Essence STX
+ * -----------------
+ *
+ * CMI8788:
+ *
+ * I²C <-> PCM1792A
+ *
+ * GPI 0 <- external power present
+ *
+ * GPIO 0 -> enable output to speakers
+ * GPIO 1 -> route HP to front panel (0) or rear jack (1)
+ * GPIO 2 -> M0 of CS5381
+ * GPIO 3 -> M1 of CS5381
+ * GPIO 7 -> route output to speaker jacks (0) or HP (1)
+ * GPIO 8 -> route input jack to line-in (0) or mic-in (1)
+ *
+ * PCM1792A:
+ *
+ * AD0 <- 0
+ *
+ * H6 daughterboard
+ * ----------------
+ *
+ * GPIO 4 <- 0
+ * GPIO 5 <- 0
+ */
+
 #include <linux/pci.h>
 #include <linux/delay.h>
 #include <linux/mutex.h>
@@ -152,6 +180,7 @@ enum {
 	MODEL_DX,
 	MODEL_HDAV,	/* without daughterboard */
 	MODEL_HDAV_H6,	/* with H6 daughterboard */
+	MODEL_STX,
 };
 
 static struct pci_device_id xonar_ids[] __devinitdata = {
@@ -160,6 +189,8 @@ static struct pci_device_id xonar_ids[] __devinitdata = {
 	{ OXYGEN_PCI_SUBID(0x1043, 0x82b7), .driver_data = MODEL_D2X },
 	{ OXYGEN_PCI_SUBID(0x1043, 0x8314), .driver_data = MODEL_HDAV },
 	{ OXYGEN_PCI_SUBID(0x1043, 0x834f), .driver_data = MODEL_D1 },
+	{ OXYGEN_PCI_SUBID(0x1043, 0x835c), .driver_data = MODEL_STX },
+	{ OXYGEN_PCI_SUBID_BROKEN_EEPROM },
 	{ }
 };
 MODULE_DEVICE_TABLE(pci, xonar_ids);
@@ -183,12 +214,14 @@ MODULE_DEVICE_TABLE(pci, xonar_ids);
 #define GPIO_HDAV_DB_H6		0x0000
 #define GPIO_HDAV_DB_XX		0x0020
 
+#define GPIO_ST_HP_REAR		0x0002
+#define GPIO_ST_HP		0x0080
+
 #define I2C_DEVICE_PCM1796(i)	(0x98 + ((i) << 1))	/* 10011, ADx=i, /W=0 */
 #define I2C_DEVICE_CS4398	0x9e	/* 10011, AD1=1, AD0=1, /W=0 */
 #define I2C_DEVICE_CS4362A	0x30	/* 001100, AD0=0, /W=0 */
 
 struct xonar_data {
-	unsigned int model;
 	unsigned int anti_pop_delay;
 	unsigned int dacs;
 	u16 output_enable_bit;
@@ -334,15 +367,9 @@ static void xonar_d2_init(struct oxygen *chip)
 	struct xonar_data *data = chip->model_data;
 
 	data->anti_pop_delay = 300;
+	data->dacs = 4;
 	data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE;
 	data->pcm1796_oversampling = PCM1796_OS_64;
-	if (data->model == MODEL_D2X) {
-		data->ext_power_reg = OXYGEN_GPIO_DATA;
-		data->ext_power_int_reg = OXYGEN_GPIO_INTERRUPT_MASK;
-		data->ext_power_bit = GPIO_D2X_EXT_POWER;
-		oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL,
-				    GPIO_D2X_EXT_POWER);
-	}
 
 	pcm1796_init(chip);
 
@@ -355,6 +382,18 @@ static void xonar_d2_init(struct oxygen *chip)
 	snd_component_add(chip->card, "CS5381");
 }
 
+static void xonar_d2x_init(struct oxygen *chip)
+{
+	struct xonar_data *data = chip->model_data;
+
+	data->ext_power_reg = OXYGEN_GPIO_DATA;
+	data->ext_power_int_reg = OXYGEN_GPIO_INTERRUPT_MASK;
+	data->ext_power_bit = GPIO_D2X_EXT_POWER;
+	oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2X_EXT_POWER);
+
+	xonar_d2_init(chip);
+}
+
 static void update_cs4362a_volumes(struct oxygen *chip)
 {
 	u8 mute;
@@ -422,11 +461,6 @@ static void xonar_d1_init(struct oxygen *chip)
 	data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST;
 	data->cs4362a_fm = CS4362A_FM_SINGLE |
 		CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L;
-	if (data->model == MODEL_DX) {
-		data->ext_power_reg = OXYGEN_GPI_DATA;
-		data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
-		data->ext_power_bit = GPI_DX_EXT_POWER;
-	}
 
 	oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS,
 		       OXYGEN_2WIRE_LENGTH_8 |
@@ -447,6 +481,17 @@ static void xonar_d1_init(struct oxygen *chip)
 	snd_component_add(chip->card, "CS5361");
 }
 
+static void xonar_dx_init(struct oxygen *chip)
+{
+	struct xonar_data *data = chip->model_data;
+
+	data->ext_power_reg = OXYGEN_GPI_DATA;
+	data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
+	data->ext_power_bit = GPI_DX_EXT_POWER;
+
+	xonar_d1_init(chip);
+}
+
 static void xonar_hdav_init(struct oxygen *chip)
 {
 	struct xonar_data *data = chip->model_data;
@@ -458,6 +503,7 @@ static void xonar_hdav_init(struct oxygen *chip)
 		       OXYGEN_2WIRE_SPEED_FAST);
 
 	data->anti_pop_delay = 100;
+	data->dacs = chip->model.private_data == MODEL_HDAV_H6 ? 4 : 1;
 	data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE;
 	data->ext_power_reg = OXYGEN_GPI_DATA;
 	data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
@@ -484,6 +530,36 @@ static void xonar_hdav_init(struct oxygen *chip)
 	snd_component_add(chip->card, "CS5381");
 }
 
+static void xonar_stx_init(struct oxygen *chip)
+{
+	struct xonar_data *data = chip->model_data;
+
+	oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS,
+		       OXYGEN_2WIRE_LENGTH_8 |
+		       OXYGEN_2WIRE_INTERRUPT_MASK |
+		       OXYGEN_2WIRE_SPEED_FAST);
+
+	data->anti_pop_delay = 100;
+	data->dacs = 1;
+	data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE;
+	data->ext_power_reg = OXYGEN_GPI_DATA;
+	data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK;
+	data->ext_power_bit = GPI_DX_EXT_POWER;
+	data->pcm1796_oversampling = PCM1796_OS_64;
+
+	pcm1796_init(chip);
+
+	oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL,
+			  GPIO_DX_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP);
+	oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA,
+			    GPIO_DX_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP);
+
+	xonar_common_init(chip);
+
+	snd_component_add(chip->card, "PCM1792A");
+	snd_component_add(chip->card, "CS5381");
+}
+
 static void xonar_disable_output(struct oxygen *chip)
 {
 	struct xonar_data *data = chip->model_data;
@@ -511,6 +587,11 @@ static void xonar_hdav_cleanup(struct oxygen *chip)
 	xonar_disable_output(chip);
 }
 
+static void xonar_st_cleanup(struct oxygen *chip)
+{
+	xonar_disable_output(chip);
+}
+
 static void xonar_d2_suspend(struct oxygen *chip)
 {
 	xonar_d2_cleanup(chip);
@@ -527,6 +608,11 @@ static void xonar_hdav_suspend(struct oxygen *chip)
 	msleep(2);
 }
 
+static void xonar_st_suspend(struct oxygen *chip)
+{
+	xonar_st_cleanup(chip);
+}
+
 static void xonar_d2_resume(struct oxygen *chip)
 {
 	pcm1796_init(chip);
@@ -554,6 +640,12 @@ static void xonar_hdav_resume(struct oxygen *chip)
 	xonar_enable_output(chip);
 }
 
+static void xonar_st_resume(struct oxygen *chip)
+{
+	pcm1796_init(chip);
+	xonar_enable_output(chip);
+}
+
 static void xonar_hdav_pcm_hardware_filter(unsigned int channel,
 					   struct snd_pcm_hardware *hardware)
 {
@@ -733,6 +825,72 @@ static const struct snd_kcontrol_new front_panel_switch = {
 	.private_value = GPIO_DX_FRONT_PANEL,
 };
 
+static int st_output_switch_info(struct snd_kcontrol *ctl,
+				 struct snd_ctl_elem_info *info)
+{
+	static const char *const names[3] = {
+		"Speakers", "Headphones", "FP Headphones"
+	};
+
+	info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	info->count = 1;
+	info->value.enumerated.items = 3;
+	if (info->value.enumerated.item >= 3)
+		info->value.enumerated.item = 2;
+	strcpy(info->value.enumerated.name, names[info->value.enumerated.item]);
+	return 0;
+}
+
+static int st_output_switch_get(struct snd_kcontrol *ctl,
+				struct snd_ctl_elem_value *value)
+{
+	struct oxygen *chip = ctl->private_data;
+	u16 gpio;
+
+	gpio = oxygen_read16(chip, OXYGEN_GPIO_DATA);
+	if (!(gpio & GPIO_ST_HP))
+		value->value.enumerated.item[0] = 0;
+	else if (gpio & GPIO_ST_HP_REAR)
+		value->value.enumerated.item[0] = 1;
+	else
+		value->value.enumerated.item[0] = 2;
+	return 0;
+}
+
+
+static int st_output_switch_put(struct snd_kcontrol *ctl,
+				struct snd_ctl_elem_value *value)
+{
+	struct oxygen *chip = ctl->private_data;
+	u16 gpio_old, gpio;
+
+	mutex_lock(&chip->mutex);
+	gpio_old = oxygen_read16(chip, OXYGEN_GPIO_DATA);
+	gpio = gpio_old;
+	switch (value->value.enumerated.item[0]) {
+	case 0:
+		gpio &= ~(GPIO_ST_HP | GPIO_ST_HP_REAR);
+		break;
+	case 1:
+		gpio |= GPIO_ST_HP | GPIO_ST_HP_REAR;
+		break;
+	case 2:
+		gpio = (gpio | GPIO_ST_HP) & ~GPIO_ST_HP_REAR;
+		break;
+	}
+	oxygen_write16(chip, OXYGEN_GPIO_DATA, gpio);
+	mutex_unlock(&chip->mutex);
+	return gpio != gpio_old;
+}
+
+static const struct snd_kcontrol_new st_output_switch = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Analog Output",
+	.info = st_output_switch_info,
+	.get = st_output_switch_get,
+	.put = st_output_switch_put,
+};
+
 static void xonar_line_mic_ac97_switch(struct oxygen *chip,
 				       unsigned int reg, unsigned int mute)
 {
@@ -745,8 +903,8 @@ static void xonar_line_mic_ac97_switch(struct oxygen *chip,
 	}
 }
 
-static const DECLARE_TLV_DB_SCALE(pcm1796_db_scale, -12000, 50, 0);
-static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -12700, 100, 0);
+static const DECLARE_TLV_DB_SCALE(pcm1796_db_scale, -6000, 50, 0);
+static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -6000, 100, 0);
 
 static int xonar_d2_control_filter(struct snd_kcontrol_new *template)
 {
@@ -763,6 +921,15 @@ static int xonar_d1_control_filter(struct snd_kcontrol_new *template)
 	return 0;
 }
 
+static int xonar_st_control_filter(struct snd_kcontrol_new *template)
+{
+	if (!strncmp(template->name, "CD Capture ", 11))
+		return 1; /* no CD input */
+	if (!strcmp(template->name, "Stereo Upmixing"))
+		return 1; /* stereo only - we don't need upmixing */
+	return 0;
+}
+
 static int xonar_d2_mixer_init(struct oxygen *chip)
 {
 	return snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip));
@@ -773,51 +940,14 @@ static int xonar_d1_mixer_init(struct oxygen *chip)
 	return snd_ctl_add(chip->card, snd_ctl_new1(&front_panel_switch, chip));
 }
 
-static int xonar_model_probe(struct oxygen *chip, unsigned long driver_data)
+static int xonar_st_mixer_init(struct oxygen *chip)
 {
-	static const char *const names[] = {
-		[MODEL_D1]	= "Xonar D1",
-		[MODEL_DX]	= "Xonar DX",
-		[MODEL_D2]	= "Xonar D2",
-		[MODEL_D2X]	= "Xonar D2X",
-		[MODEL_HDAV]	= "Xonar HDAV1.3",
-		[MODEL_HDAV_H6]	= "Xonar HDAV1.3+H6",
-	};
-	static const u8 dacs[] = {
-		[MODEL_D1]	= 2,
-		[MODEL_DX]	= 2,
-		[MODEL_D2]	= 4,
-		[MODEL_D2X]	= 4,
-		[MODEL_HDAV]	= 1,
-		[MODEL_HDAV_H6]	= 4,
-	};
-	struct xonar_data *data = chip->model_data;
-
-	data->model = driver_data;
-	if (data->model == MODEL_HDAV) {
-		oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL,
-				    GPIO_HDAV_DB_MASK);
-		switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) &
-			GPIO_HDAV_DB_MASK) {
-		case GPIO_HDAV_DB_H6:
-			data->model = MODEL_HDAV_H6;
-			break;
-		case GPIO_HDAV_DB_XX:
-			snd_printk(KERN_ERR "unknown daughterboard\n");
-			return -ENODEV;
-		}
-	}
-
-	data->dacs = dacs[data->model];
-	chip->model.shortname = names[data->model];
-	return 0;
+	return snd_ctl_add(chip->card, snd_ctl_new1(&st_output_switch, chip));
 }
 
 static const struct oxygen_model model_xonar_d2 = {
 	.longname = "Asus Virtuoso 200",
 	.chip = "AV200",
-	.owner = THIS_MODULE,
-	.probe = xonar_model_probe,
 	.init = xonar_d2_init,
 	.control_filter = xonar_d2_control_filter,
 	.mixer_init = xonar_d2_mixer_init,
@@ -837,8 +967,8 @@ static const struct oxygen_model model_xonar_d2 = {
 			 MIDI_OUTPUT |
 			 MIDI_INPUT,
 	.dac_channels = 8,
-	.dac_volume_min = 0x0f,
-	.dac_volume_max = 0xff,
+	.dac_volume_min = 255 - 2*60,
+	.dac_volume_max = 255,
 	.misc_flags = OXYGEN_MISC_MIDI,
 	.function_flags = OXYGEN_FUNCTION_SPI |
 			  OXYGEN_FUNCTION_ENABLE_SPI_4_5,
@@ -849,8 +979,6 @@ static const struct oxygen_model model_xonar_d2 = {
 static const struct oxygen_model model_xonar_d1 = {
 	.longname = "Asus Virtuoso 100",
 	.chip = "AV200",
-	.owner = THIS_MODULE,
-	.probe = xonar_model_probe,
 	.init = xonar_d1_init,
 	.control_filter = xonar_d1_control_filter,
 	.mixer_init = xonar_d1_mixer_init,
@@ -868,7 +996,7 @@ static const struct oxygen_model model_xonar_d1 = {
 			 PLAYBACK_1_TO_SPDIF |
 			 CAPTURE_0_FROM_I2S_2,
 	.dac_channels = 8,
-	.dac_volume_min = 0,
+	.dac_volume_min = 127 - 60,
 	.dac_volume_max = 127,
 	.function_flags = OXYGEN_FUNCTION_2WIRE,
 	.dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
@@ -878,8 +1006,6 @@ static const struct oxygen_model model_xonar_d1 = {
 static const struct oxygen_model model_xonar_hdav = {
 	.longname = "Asus Virtuoso 200",
 	.chip = "AV200",
-	.owner = THIS_MODULE,
-	.probe = xonar_model_probe,
 	.init = xonar_hdav_init,
 	.cleanup = xonar_hdav_cleanup,
 	.suspend = xonar_hdav_suspend,
@@ -897,16 +1023,43 @@ static const struct oxygen_model model_xonar_hdav = {
 			 PLAYBACK_1_TO_SPDIF |
 			 CAPTURE_0_FROM_I2S_2,
 	.dac_channels = 8,
-	.dac_volume_min = 0x0f,
-	.dac_volume_max = 0xff,
+	.dac_volume_min = 255 - 2*60,
+	.dac_volume_max = 255,
 	.misc_flags = OXYGEN_MISC_MIDI,
 	.function_flags = OXYGEN_FUNCTION_2WIRE,
 	.dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
 	.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
 };
 
-static int __devinit xonar_probe(struct pci_dev *pci,
-				 const struct pci_device_id *pci_id)
+static const struct oxygen_model model_xonar_st = {
+	.longname = "Asus Virtuoso 100",
+	.chip = "AV200",
+	.init = xonar_stx_init,
+	.control_filter = xonar_st_control_filter,
+	.mixer_init = xonar_st_mixer_init,
+	.cleanup = xonar_st_cleanup,
+	.suspend = xonar_st_suspend,
+	.resume = xonar_st_resume,
+	.set_dac_params = set_pcm1796_params,
+	.set_adc_params = set_cs53x1_params,
+	.update_dac_volume = update_pcm1796_volume,
+	.update_dac_mute = update_pcm1796_mute,
+	.ac97_switch = xonar_line_mic_ac97_switch,
+	.dac_tlv = pcm1796_db_scale,
+	.model_data_size = sizeof(struct xonar_data),
+	.device_config = PLAYBACK_0_TO_I2S |
+			 PLAYBACK_1_TO_SPDIF |
+			 CAPTURE_0_FROM_I2S_2,
+	.dac_channels = 2,
+	.dac_volume_min = 255 - 2*60,
+	.dac_volume_max = 255,
+	.function_flags = OXYGEN_FUNCTION_2WIRE,
+	.dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+	.adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST,
+};
+
+static int __devinit get_xonar_model(struct oxygen *chip,
+				     const struct pci_device_id *id)
 {
 	static const struct oxygen_model *const models[] = {
 		[MODEL_D1]	= &model_xonar_d1,
@@ -914,7 +1067,57 @@ static int __devinit xonar_probe(struct pci_dev *pci,
 		[MODEL_D2]	= &model_xonar_d2,
 		[MODEL_D2X]	= &model_xonar_d2,
 		[MODEL_HDAV]	= &model_xonar_hdav,
+		[MODEL_STX]	= &model_xonar_st,
 	};
+	static const char *const names[] = {
+		[MODEL_D1]	= "Xonar D1",
+		[MODEL_DX]	= "Xonar DX",
+		[MODEL_D2]	= "Xonar D2",
+		[MODEL_D2X]	= "Xonar D2X",
+		[MODEL_HDAV]	= "Xonar HDAV1.3",
+		[MODEL_HDAV_H6]	= "Xonar HDAV1.3+H6",
+		[MODEL_STX]	= "Xonar Essence STX",
+	};
+	unsigned int model = id->driver_data;
+
+	if (model >= ARRAY_SIZE(models) || !models[model])
+		return -EINVAL;
+	chip->model = *models[model];
+
+	switch (model) {
+	case MODEL_D2X:
+		chip->model.init = xonar_d2x_init;
+		break;
+	case MODEL_DX:
+		chip->model.init = xonar_dx_init;
+		break;
+	case MODEL_HDAV:
+		oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL,
+				    GPIO_HDAV_DB_MASK);
+		switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) &
+			GPIO_HDAV_DB_MASK) {
+		case GPIO_HDAV_DB_H6:
+			model = MODEL_HDAV_H6;
+			break;
+		case GPIO_HDAV_DB_XX:
+			snd_printk(KERN_ERR "unknown daughterboard\n");
+			return -ENODEV;
+		}
+		break;
+	case MODEL_STX:
+		oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL,
+				    GPIO_HDAV_DB_MASK);
+		break;
+	}
+
+	chip->model.shortname = names[model];
+	chip->model.private_data = model;
+	return 0;
+}
+
+static int __devinit xonar_probe(struct pci_dev *pci,
+				 const struct pci_device_id *pci_id)
+{
 	static int dev;
 	int err;
 
@@ -924,10 +1127,8 @@ static int __devinit xonar_probe(struct pci_dev *pci,
 		++dev;
 		return -ENOENT;
 	}
-	BUG_ON(pci_id->driver_data >= ARRAY_SIZE(models));
-	err = oxygen_pci_probe(pci, index[dev], id[dev],
-			       models[pci_id->driver_data],
-			       pci_id->driver_data);
+	err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE,
+			       xonar_ids, get_xonar_model);
 	if (err >= 0)
 		++dev;
 	return err;
diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c
index 27cf2c28d113..80e064a3efff 100644
--- a/sound/pci/pcxhr/pcxhr.c
+++ b/sound/pci/pcxhr/pcxhr.c
@@ -1334,6 +1334,40 @@ static void pcxhr_proc_sync(struct snd_info_entry *entry,
 	snd_iprintf(buffer, "\n");
 }
 
+static void pcxhr_proc_gpio_read(struct snd_info_entry *entry,
+				 struct snd_info_buffer *buffer)
+{
+	struct snd_pcxhr *chip = entry->private_data;
+	struct pcxhr_mgr *mgr = chip->mgr;
+	/* commands available when embedded DSP is running */
+	if (mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX)) {
+		/* gpio ports on stereo boards only available */
+		int value = 0;
+		hr222_read_gpio(mgr, 1, &value);	/* GPI */
+		snd_iprintf(buffer, "GPI: 0x%x\n", value);
+		hr222_read_gpio(mgr, 0, &value);	/* GP0 */
+		snd_iprintf(buffer, "GPO: 0x%x\n", value);
+	} else
+		snd_iprintf(buffer, "no firmware loaded\n");
+	snd_iprintf(buffer, "\n");
+}
+static void pcxhr_proc_gpo_write(struct snd_info_entry *entry,
+				 struct snd_info_buffer *buffer)
+{
+	struct snd_pcxhr *chip = entry->private_data;
+	struct pcxhr_mgr *mgr = chip->mgr;
+	char line[64];
+	int value;
+	/* commands available when embedded DSP is running */
+	if (!(mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX)))
+		return;
+	while (!snd_info_get_line(buffer, line, sizeof(line))) {
+		if (sscanf(line, "GPO: 0x%x", &value) != 1)
+			continue;
+		hr222_write_gpo(mgr, value);	/* GP0 */
+	}
+}
+
 static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip)
 {
 	struct snd_info_entry *entry;
@@ -1342,6 +1376,13 @@ static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip)
 		snd_info_set_text_ops(entry, chip, pcxhr_proc_info);
 	if (! snd_card_proc_new(chip->card, "sync", &entry))
 		snd_info_set_text_ops(entry, chip, pcxhr_proc_sync);
+	/* gpio available on stereo sound cards only */
+	if (chip->mgr->is_hr_stereo &&
+	    !snd_card_proc_new(chip->card, "gpio", &entry)) {
+		snd_info_set_text_ops(entry, chip, pcxhr_proc_gpio_read);
+		entry->c.text.write = pcxhr_proc_gpo_write;
+		entry->mode |= S_IWUSR;
+	}
 }
 /* end of proc interface */
 
@@ -1510,12 +1551,12 @@ static int __devinit pcxhr_probe(struct pci_dev *pci,
 
 		snprintf(tmpid, sizeof(tmpid), "%s-%d",
 			 id[dev] ? id[dev] : card_name, i);
-		card = snd_card_new(idx, tmpid, THIS_MODULE, 0);
+		err = snd_card_create(idx, tmpid, THIS_MODULE, 0, &card);
 
-		if (! card) {
+		if (err < 0) {
 			snd_printk(KERN_ERR "cannot allocate the card %d\n", i);
 			pcxhr_free(mgr);
-			return -ENOMEM;
+			return err;
 		}
 
 		strcpy(card->driver, DRIVER_NAME);
diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h
index 69d87dee6995..bda776c49884 100644
--- a/sound/pci/pcxhr/pcxhr.h
+++ b/sound/pci/pcxhr/pcxhr.h
@@ -27,8 +27,8 @@
 #include <linux/mutex.h>
 #include <sound/pcm.h>
 
-#define PCXHR_DRIVER_VERSION		0x000905	/* 0.9.5 */
-#define PCXHR_DRIVER_VERSION_STRING	"0.9.5"		/* 0.9.5 */
+#define PCXHR_DRIVER_VERSION		0x000906	/* 0.9.6 */
+#define PCXHR_DRIVER_VERSION_STRING	"0.9.6"		/* 0.9.6 */
 
 
 #define PCXHR_MAX_CARDS		6
@@ -124,6 +124,7 @@ struct pcxhr_mgr {
 
 	unsigned char xlx_cfg;		/* copy of PCXHR_XLX_CFG register */
 	unsigned char xlx_selmic;	/* copy of PCXHR_XLX_SELMIC register */
+	unsigned char dsp_reset;	/* copy of PCXHR_DSP_RESET register */
 };
 
 
diff --git a/sound/pci/pcxhr/pcxhr_core.h b/sound/pci/pcxhr/pcxhr_core.h
index bbbd66d13a64..be0173796cdb 100644
--- a/sound/pci/pcxhr/pcxhr_core.h
+++ b/sound/pci/pcxhr/pcxhr_core.h
@@ -1,7 +1,7 @@
 /*
  * Driver for Digigram pcxhr compatible soundcards
  *
- * low level interface with interrupt ans message handling
+ * low level interface with interrupt and message handling
  *
  * Copyright (c) 2004 by Digigram <alsa@digigram.com>
  *
diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c
index 592743a298b0..17cb1233a903 100644
--- a/sound/pci/pcxhr/pcxhr_hwdep.c
+++ b/sound/pci/pcxhr/pcxhr_hwdep.c
@@ -471,16 +471,6 @@ static int pcxhr_hwdep_dsp_load(struct snd_hwdep *hw,
 	return 0;
 }
 
-static int pcxhr_hwdep_open(struct snd_hwdep *hw, struct file *file)
-{
-	return 0;
-}
-
-static int pcxhr_hwdep_release(struct snd_hwdep *hw, struct file *file)
-{
-	return 0;
-}
-
 int pcxhr_setup_firmware(struct pcxhr_mgr *mgr)
 {
 	int err;
@@ -495,8 +485,6 @@ int pcxhr_setup_firmware(struct pcxhr_mgr *mgr)
 
 	hw->iface = SNDRV_HWDEP_IFACE_PCXHR;
 	hw->private_data = mgr;
-	hw->ops.open = pcxhr_hwdep_open;
-	hw->ops.release = pcxhr_hwdep_release;
 	hw->ops.dsp_status = pcxhr_hwdep_dsp_status;
 	hw->ops.dsp_load = pcxhr_hwdep_dsp_load;
 	hw->exclusive = 1;
diff --git a/sound/pci/pcxhr/pcxhr_mix22.c b/sound/pci/pcxhr/pcxhr_mix22.c
index ff019126b672..1cb82c0a9cb3 100644
--- a/sound/pci/pcxhr/pcxhr_mix22.c
+++ b/sound/pci/pcxhr/pcxhr_mix22.c
@@ -53,6 +53,8 @@
 #define PCXHR_DSP_RESET_DSP	0x01
 #define PCXHR_DSP_RESET_MUTE	0x02
 #define PCXHR_DSP_RESET_CODEC	0x08
+#define PCXHR_DSP_RESET_GPO_OFFSET	5
+#define PCXHR_DSP_RESET_GPO_MASK	0x60
 
 /* values for PCHR_XLX_CFG register */
 #define PCXHR_CFG_SYNCDSP_MASK		0x80
@@ -81,6 +83,8 @@
 /* values for PCHR_XLX_STATUS register - READ */
 #define PCXHR_STAT_SRC_LOCK		0x01
 #define PCXHR_STAT_LEVEL_IN		0x02
+#define PCXHR_STAT_GPI_OFFSET		2
+#define PCXHR_STAT_GPI_MASK		0x0C
 #define PCXHR_STAT_MIC_CAPS		0x10
 /* values for PCHR_XLX_STATUS register - WRITE */
 #define PCXHR_STAT_FREQ_SYNC_MASK	0x01
@@ -291,10 +295,11 @@ int hr222_sub_init(struct pcxhr_mgr *mgr)
 	PCXHR_OUTPB(mgr, PCXHR_DSP_RESET,
 		    PCXHR_DSP_RESET_DSP);
 	msleep(5);
-	PCXHR_OUTPB(mgr, PCXHR_DSP_RESET,
-		    PCXHR_DSP_RESET_DSP  |
-		    PCXHR_DSP_RESET_MUTE |
-		    PCXHR_DSP_RESET_CODEC);
+	mgr->dsp_reset = PCXHR_DSP_RESET_DSP  |
+			 PCXHR_DSP_RESET_MUTE |
+			 PCXHR_DSP_RESET_CODEC;
+	PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, mgr->dsp_reset);
+	/* hr222_write_gpo(mgr, 0); does the same */
 	msleep(5);
 
 	/* config AKM */
@@ -496,6 +501,33 @@ int hr222_get_external_clock(struct pcxhr_mgr *mgr,
 }
 
 
+int hr222_read_gpio(struct pcxhr_mgr *mgr, int is_gpi, int *value)
+{
+	if (is_gpi) {
+		unsigned char reg = PCXHR_INPB(mgr, PCXHR_XLX_STATUS);
+		*value = (int)(reg & PCXHR_STAT_GPI_MASK) >>
+			      PCXHR_STAT_GPI_OFFSET;
+	} else {
+		*value = (int)(mgr->dsp_reset & PCXHR_DSP_RESET_GPO_MASK) >>
+			 PCXHR_DSP_RESET_GPO_OFFSET;
+	}
+	return 0;
+}
+
+
+int hr222_write_gpo(struct pcxhr_mgr *mgr, int value)
+{
+	unsigned char reg = mgr->dsp_reset & ~PCXHR_DSP_RESET_GPO_MASK;
+
+	reg |= (unsigned char)(value << PCXHR_DSP_RESET_GPO_OFFSET) &
+	       PCXHR_DSP_RESET_GPO_MASK;
+
+	PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, reg);
+	mgr->dsp_reset = reg;
+	return 0;
+}
+
+
 int hr222_update_analog_audio_level(struct snd_pcxhr *chip,
 				    int is_capture, int channel)
 {
diff --git a/sound/pci/pcxhr/pcxhr_mix22.h b/sound/pci/pcxhr/pcxhr_mix22.h
index 6b318b2f0100..5a37a0007e8f 100644
--- a/sound/pci/pcxhr/pcxhr_mix22.h
+++ b/sound/pci/pcxhr/pcxhr_mix22.h
@@ -32,6 +32,9 @@ int hr222_get_external_clock(struct pcxhr_mgr *mgr,
 			     enum pcxhr_clock_type clock_type,
 			     int *sample_rate);
 
+int hr222_read_gpio(struct pcxhr_mgr *mgr, int is_gpi, int *value);
+int hr222_write_gpo(struct pcxhr_mgr *mgr, int value);
+
 #define HR222_LINE_PLAYBACK_LEVEL_MIN		0	/* -25.5 dB */
 #define HR222_LINE_PLAYBACK_ZERO_LEVEL		51	/* 0.0 dB */
 #define HR222_LINE_PLAYBACK_LEVEL_MAX		99	/* +24.0 dB */
diff --git a/sound/pci/pcxhr/pcxhr_mixer.c b/sound/pci/pcxhr/pcxhr_mixer.c
index 2436e374586f..fec049344621 100644
--- a/sound/pci/pcxhr/pcxhr_mixer.c
+++ b/sound/pci/pcxhr/pcxhr_mixer.c
@@ -789,11 +789,15 @@ static int pcxhr_clock_type_put(struct snd_kcontrol *kcontrol,
 	if (mgr->use_clock_type != ucontrol->value.enumerated.item[0]) {
 		mutex_lock(&mgr->setup_mutex);
 		mgr->use_clock_type = ucontrol->value.enumerated.item[0];
-		if (mgr->use_clock_type)
+		rate = 0;
+		if (mgr->use_clock_type != PCXHR_CLOCK_TYPE_INTERNAL) {
 			pcxhr_get_external_clock(mgr, mgr->use_clock_type,
 						 &rate);
-		else
+		} else {
 			rate = mgr->sample_rate;
+			if (!rate)
+				rate = 48000;
+		}
 		if (rate) {
 			pcxhr_set_clock(mgr, rate);
 			if (mgr->sample_rate)
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 3caacfb9d8e0..6f1034417a02 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -2102,9 +2102,9 @@ snd_card_riptide_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 	if ((err = snd_riptide_create(card, pci, &chip)) < 0) {
 		snd_card_free(card);
 		return err;
diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c
index e7ef3a1a25a8..d7b966e7c4cf 100644
--- a/sound/pci/rme32.c
+++ b/sound/pci/rme32.c
@@ -1941,9 +1941,10 @@ snd_rme32_probe(struct pci_dev *pci, const struct pci_device_id *pci_id)
 		return -ENOENT;
 	}
 
-	if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-				 sizeof(struct rme32))) == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct rme32), &card);
+	if (err < 0)
+		return err;
 	card->private_free = snd_rme32_card_free;
 	rme32 = (struct rme32 *) card->private_data;
 	rme32->card = card;
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index 3fdd488d0975..55fb1c131f58 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -2348,9 +2348,10 @@ snd_rme96_probe(struct pci_dev *pci,
 		dev++;
 		return -ENOENT;
 	}
-	if ((card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-				 sizeof(struct rme96))) == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct rme96), &card);
+	if (err < 0)
+		return err;
 	card->private_free = snd_rme96_card_free;
 	rme96 = (struct rme96 *)card->private_data;	
 	rme96->card = card;
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 44d0c15e2b71..314e73531bd1 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -113,7 +113,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin");
 
 /* the meters are regular i/o-mapped registers, but offset
    considerably from the rest. the peak registers are reset
-   when read; the least-significant 4 bits are full-scale counters; 
+   when read; the least-significant 4 bits are full-scale counters;
    the actual peak value is in the most-significant 24 bits.
 */
 
@@ -131,7 +131,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin");
    26*3 values are read in ss mode
    14*3 in ds mode, with no gap between values
 */
-#define HDSP_9652_peakBase	7164	
+#define HDSP_9652_peakBase	7164
 #define HDSP_9652_rmsBase	4096
 
 /* c.f. the hdsp_9632_meters_t struct */
@@ -173,12 +173,12 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin");
 #define HDSP_SPDIFEmphasis        (1<<10) /* 0=none, 1=on */
 #define HDSP_SPDIFNonAudio        (1<<11) /* 0=off, 1=on */
 #define HDSP_SPDIFOpticalOut      (1<<12) /* 1=use 1st ADAT connector for SPDIF, 0=do not */
-#define HDSP_SyncRef2             (1<<13) 
-#define HDSP_SPDIFInputSelect0    (1<<14) 
-#define HDSP_SPDIFInputSelect1    (1<<15) 
-#define HDSP_SyncRef0             (1<<16) 
+#define HDSP_SyncRef2             (1<<13)
+#define HDSP_SPDIFInputSelect0    (1<<14)
+#define HDSP_SPDIFInputSelect1    (1<<15)
+#define HDSP_SyncRef0             (1<<16)
 #define HDSP_SyncRef1             (1<<17)
-#define HDSP_AnalogExtensionBoard (1<<18) /* For H9632 cards */ 
+#define HDSP_AnalogExtensionBoard (1<<18) /* For H9632 cards */
 #define HDSP_XLRBreakoutCable     (1<<20) /* For H9632 cards */
 #define HDSP_Midi0InterruptEnable (1<<22)
 #define HDSP_Midi1InterruptEnable (1<<23)
@@ -314,7 +314,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin");
 #define HDSP_TimecodeSync       (1<<27)
 #define HDSP_AEBO          	(1<<28) /* H9632 specific Analog Extension Boards */
 #define HDSP_AEBI		(1<<29) /* 0 = present, 1 = absent */
-#define HDSP_midi0IRQPending    (1<<30) 
+#define HDSP_midi0IRQPending    (1<<30)
 #define HDSP_midi1IRQPending    (1<<31)
 
 #define HDSP_spdifFrequencyMask    (HDSP_spdifFrequency0|HDSP_spdifFrequency1|HDSP_spdifFrequency2)
@@ -391,7 +391,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin");
 #define HDSP_CHANNEL_BUFFER_BYTES    (4*HDSP_CHANNEL_BUFFER_SAMPLES)
 
 /* the size of the area we need to allocate for DMA transfers. the
-   size is the same regardless of the number of channels - the 
+   size is the same regardless of the number of channels - the
    Multiface still uses the same memory area.
 
    Note that we allocate 1 more channel than is apparently needed
@@ -460,7 +460,7 @@ struct hdsp {
 	unsigned char	      qs_in_channels;	     /* quad speed mode for H9632 */
 	unsigned char         ds_in_channels;
 	unsigned char         ss_in_channels;	    /* different for multiface/digiface */
-	unsigned char	      qs_out_channels;	    
+	unsigned char	      qs_out_channels;
 	unsigned char         ds_out_channels;
 	unsigned char         ss_out_channels;
 
@@ -502,9 +502,9 @@ static char channel_map_df_ss[HDSP_MAX_CHANNELS] = {
 
 static char channel_map_mf_ss[HDSP_MAX_CHANNELS] = { /* Multiface */
 	/* Analog */
-	0, 1, 2, 3, 4, 5, 6, 7, 
+	0, 1, 2, 3, 4, 5, 6, 7,
 	/* ADAT 2 */
-	16, 17, 18, 19, 20, 21, 22, 23, 
+	16, 17, 18, 19, 20, 21, 22, 23,
 	/* SPDIF */
 	24, 25,
 	-1, -1, -1, -1, -1, -1, -1, -1
@@ -525,11 +525,11 @@ static char channel_map_H9632_ss[HDSP_MAX_CHANNELS] = {
 	/* SPDIF */
 	8, 9,
 	/* Analog */
-	10, 11, 
+	10, 11,
 	/* AO4S-192 and AI4S-192 extension boards */
 	12, 13, 14, 15,
 	/* others don't exist */
-	-1, -1, -1, -1, -1, -1, -1, -1, 
+	-1, -1, -1, -1, -1, -1, -1, -1,
 	-1, -1
 };
 
@@ -539,7 +539,7 @@ static char channel_map_H9632_ds[HDSP_MAX_CHANNELS] = {
 	/* SPDIF */
 	8, 9,
 	/* Analog */
-	10, 11, 
+	10, 11,
 	/* AO4S-192 and AI4S-192 extension boards */
 	12, 13, 14, 15,
 	/* others don't exist */
@@ -587,7 +587,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d
 static struct pci_device_id snd_hdsp_ids[] = {
 	{
 		.vendor = PCI_VENDOR_ID_XILINX,
-		.device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, 
+		.device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP,
 		.subvendor = PCI_ANY_ID,
 		.subdevice = PCI_ANY_ID,
 	}, /* RME Hammerfall-DSP */
@@ -653,7 +653,6 @@ static unsigned int hdsp_read(struct hdsp *hdsp, int reg)
 
 static int hdsp_check_for_iobox (struct hdsp *hdsp)
 {
-
 	if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0;
 	if (hdsp_read (hdsp, HDSP_statusRegister) & HDSP_ConfigError) {
 		snd_printk ("Hammerfall-DSP: no Digiface or Multiface connected!\n");
@@ -661,7 +660,29 @@ static int hdsp_check_for_iobox (struct hdsp *hdsp)
 		return -EIO;
 	}
 	return 0;
+}
 
+static int hdsp_wait_for_iobox(struct hdsp *hdsp, unsigned int loops,
+			       unsigned int delay)
+{
+	unsigned int i;
+
+	if (hdsp->io_type == H9652 || hdsp->io_type == H9632)
+		return 0;
+
+	for (i = 0; i != loops; ++i) {
+		if (hdsp_read(hdsp, HDSP_statusRegister) & HDSP_ConfigError)
+			msleep(delay);
+		else {
+			snd_printd("Hammerfall-DSP: iobox found after %ums!\n",
+				   i * delay);
+			return 0;
+		}
+	}
+
+	snd_printk("Hammerfall-DSP: no Digiface or Multiface connected!\n");
+	hdsp->state &= ~HDSP_FirmwareLoaded;
+	return -EIO;
 }
 
 static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) {
@@ -670,19 +691,19 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) {
 	unsigned long flags;
 
 	if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) {
-		
+
 		snd_printk ("Hammerfall-DSP: loading firmware\n");
 
 		hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_PROGRAM);
 		hdsp_write (hdsp, HDSP_fifoData, 0);
-		
+
 		if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) {
 			snd_printk ("Hammerfall-DSP: timeout waiting for download preparation\n");
 			return -EIO;
 		}
-		
+
 		hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_LOAD);
-		
+
 		for (i = 0; i < 24413; ++i) {
 			hdsp_write(hdsp, HDSP_fifoData, hdsp->firmware_cache[i]);
 			if (hdsp_fifo_wait (hdsp, 127, HDSP_LONG_WAIT)) {
@@ -692,7 +713,7 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) {
 		}
 
 		ssleep(3);
-		
+
 		if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) {
 			snd_printk ("Hammerfall-DSP: timeout at end of firmware loading\n");
 		    	return -EIO;
@@ -705,15 +726,15 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) {
 #endif
 		hdsp_write (hdsp, HDSP_control2Reg, hdsp->control2_register);
 		snd_printk ("Hammerfall-DSP: finished firmware loading\n");
-		
+
 	}
 	if (hdsp->state & HDSP_InitializationComplete) {
 		snd_printk(KERN_INFO "Hammerfall-DSP: firmware loaded from cache, restoring defaults\n");
 		spin_lock_irqsave(&hdsp->lock, flags);
 		snd_hdsp_set_defaults(hdsp);
-		spin_unlock_irqrestore(&hdsp->lock, flags); 
+		spin_unlock_irqrestore(&hdsp->lock, flags);
 	}
-	
+
 	hdsp->state |= HDSP_FirmwareLoaded;
 
 	return 0;
@@ -722,7 +743,7 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) {
 static int hdsp_get_iobox_version (struct hdsp *hdsp)
 {
 	if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) {
-	
+
 		hdsp_write (hdsp, HDSP_control2Reg, HDSP_PROGRAM);
 		hdsp_write (hdsp, HDSP_fifoData, 0);
 		if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT) < 0)
@@ -738,7 +759,7 @@ static int hdsp_get_iobox_version (struct hdsp *hdsp)
 			hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT);
 		} else {
 			hdsp->io_type = Digiface;
-		} 
+		}
 	} else {
 		/* firmware was already loaded, get iobox type */
 		if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1)
@@ -786,13 +807,13 @@ static int hdsp_check_for_firmware (struct hdsp *hdsp, int load_on_demand)
 
 
 static int hdsp_fifo_wait(struct hdsp *hdsp, int count, int timeout)
-{    
+{
 	int i;
 
 	/* the fifoStatus registers reports on how many words
 	   are available in the command FIFO.
 	*/
-	
+
 	for (i = 0; i < timeout; i++) {
 
 		if ((int)(hdsp_read (hdsp, HDSP_fifoStatus) & 0xff) <= count)
@@ -824,11 +845,11 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short
 
 	if (addr >= HDSP_MATRIX_MIXER_SIZE)
 		return -1;
-	
+
 	if (hdsp->io_type == H9652 || hdsp->io_type == H9632) {
 
 		/* from martin bjornsen:
-		   
+
 		   "You can only write dwords to the
 		   mixer memory which contain two
 		   mixer values in the low and high
@@ -847,7 +868,7 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short
 
 		hdsp->mixer_matrix[addr] = data;
 
-		
+
 		/* `addr' addresses a 16-bit wide address, but
 		   the address space accessed via hdsp_write
 		   uses byte offsets. put another way, addr
@@ -856,17 +877,17 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short
 		   to access 0 to 2703 ...
 		*/
 		ad = addr/2;
-	
-		hdsp_write (hdsp, 4096 + (ad*4), 
-			    (hdsp->mixer_matrix[(addr&0x7fe)+1] << 16) + 
+
+		hdsp_write (hdsp, 4096 + (ad*4),
+			    (hdsp->mixer_matrix[(addr&0x7fe)+1] << 16) +
 			    hdsp->mixer_matrix[addr&0x7fe]);
-		
+
 		return 0;
 
 	} else {
 
 		ad = (addr << 16) + data;
-		
+
 		if (hdsp_fifo_wait(hdsp, 127, HDSP_LONG_WAIT))
 			return -1;
 
@@ -902,7 +923,7 @@ static int hdsp_spdif_sample_rate(struct hdsp *hdsp)
 
 	if (status & HDSP_SPDIFErrorFlag)
 		return 0;
-	
+
 	switch (rate_bits) {
 	case HDSP_spdifFrequency32KHz: return 32000;
 	case HDSP_spdifFrequency44_1KHz: return 44100;
@@ -910,13 +931,13 @@ static int hdsp_spdif_sample_rate(struct hdsp *hdsp)
 	case HDSP_spdifFrequency64KHz: return 64000;
 	case HDSP_spdifFrequency88_2KHz: return 88200;
 	case HDSP_spdifFrequency96KHz: return 96000;
-	case HDSP_spdifFrequency128KHz: 
+	case HDSP_spdifFrequency128KHz:
 		if (hdsp->io_type == H9632) return 128000;
 		break;
-	case HDSP_spdifFrequency176_4KHz: 
+	case HDSP_spdifFrequency176_4KHz:
 		if (hdsp->io_type == H9632) return 176400;
 		break;
-	case HDSP_spdifFrequency192KHz: 
+	case HDSP_spdifFrequency192KHz:
 		if (hdsp->io_type == H9632) return 192000;
 		break;
 	default:
@@ -1027,7 +1048,7 @@ static void hdsp_set_dds_value(struct hdsp *hdsp, int rate)
 {
 	u64 n;
 	u32 r;
-	
+
 	if (rate >= 112000)
 		rate /= 4;
 	else if (rate >= 56000)
@@ -1053,35 +1074,35 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally)
 	   there is no need for it (e.g. during module
 	   initialization).
 	*/
-	
-	if (!(hdsp->control_register & HDSP_ClockModeMaster)) {	
+
+	if (!(hdsp->control_register & HDSP_ClockModeMaster)) {
 		if (called_internally) {
 			/* request from ctl or card initialization */
 			snd_printk(KERN_ERR "Hammerfall-DSP: device is not running as a clock master: cannot set sample rate.\n");
 			return -1;
-		} else {		
+		} else {
 			/* hw_param request while in AutoSync mode */
 			int external_freq = hdsp_external_sample_rate(hdsp);
 			int spdif_freq = hdsp_spdif_sample_rate(hdsp);
-		
+
 			if ((spdif_freq == external_freq*2) && (hdsp_autosync_ref(hdsp) >= HDSP_AUTOSYNC_FROM_ADAT1))
 				snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in double speed mode\n");
 			else if (hdsp->io_type == H9632 && (spdif_freq == external_freq*4) && (hdsp_autosync_ref(hdsp) >= HDSP_AUTOSYNC_FROM_ADAT1))
-				snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in quad speed mode\n");			
+				snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in quad speed mode\n");
 			else if (rate != external_freq) {
 				snd_printk(KERN_INFO "Hammerfall-DSP: No AutoSync source for requested rate\n");
 				return -1;
-			}		
-		}	
+			}
+		}
 	}
 
 	current_rate = hdsp->system_sample_rate;
 
 	/* Changing from a "single speed" to a "double speed" rate is
 	   not allowed if any substreams are open. This is because
-	   such a change causes a shift in the location of 
+	   such a change causes a shift in the location of
 	   the DMA buffers and a reduction in the number of available
-	   buffers. 
+	   buffers.
 
 	   Note that a similar but essentially insoluble problem
 	   exists for externally-driven rate changes. All we can do
@@ -1089,7 +1110,7 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally)
 
 	if (rate > 96000 && hdsp->io_type != H9632)
 		return -EINVAL;
-	
+
 	switch (rate) {
 	case 32000:
 		if (current_rate > 48000)
@@ -1179,7 +1200,7 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally)
 			break;
 		}
 	}
-	
+
 	hdsp->system_sample_rate = rate;
 
 	return 0;
@@ -1245,16 +1266,16 @@ static int snd_hdsp_midi_output_write (struct hdsp_midi *hmidi)
 	unsigned char buf[128];
 
 	/* Output is not interrupt driven */
-		
+
 	spin_lock_irqsave (&hmidi->lock, flags);
 	if (hmidi->output) {
 		if (!snd_rawmidi_transmit_empty (hmidi->output)) {
 			if ((n_pending = snd_hdsp_midi_output_possible (hmidi->hdsp, hmidi->id)) > 0) {
 				if (n_pending > (int)sizeof (buf))
 					n_pending = sizeof (buf);
-				
+
 				if ((to_write = snd_rawmidi_transmit (hmidi->output, buf, n_pending)) > 0) {
-					for (i = 0; i < to_write; ++i) 
+					for (i = 0; i < to_write; ++i)
 						snd_hdsp_midi_write_byte (hmidi->hdsp, hmidi->id, buf[i]);
 				}
 			}
@@ -1325,14 +1346,14 @@ static void snd_hdsp_midi_output_timer(unsigned long data)
 {
 	struct hdsp_midi *hmidi = (struct hdsp_midi *) data;
 	unsigned long flags;
-	
+
 	snd_hdsp_midi_output_write(hmidi);
 	spin_lock_irqsave (&hmidi->lock, flags);
 
 	/* this does not bump hmidi->istimer, because the
 	   kernel automatically removed the timer when it
 	   expired, and we are now adding it back, thus
-	   leaving istimer wherever it was set before.  
+	   leaving istimer wherever it was set before.
 	*/
 
 	if (hmidi->istimer) {
@@ -1501,7 +1522,7 @@ static int snd_hdsp_control_spdif_info(struct snd_kcontrol *kcontrol, struct snd
 static int snd_hdsp_control_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	snd_hdsp_convert_to_aes(&ucontrol->value.iec958, hdsp->creg_spdif);
 	return 0;
 }
@@ -1511,7 +1532,7 @@ static int snd_hdsp_control_spdif_put(struct snd_kcontrol *kcontrol, struct snd_
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	u32 val;
-	
+
 	val = snd_hdsp_convert_from_aes(&ucontrol->value.iec958);
 	spin_lock_irq(&hdsp->lock);
 	change = val != hdsp->creg_spdif;
@@ -1530,7 +1551,7 @@ static int snd_hdsp_control_spdif_stream_info(struct snd_kcontrol *kcontrol, str
 static int snd_hdsp_control_spdif_stream_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	snd_hdsp_convert_to_aes(&ucontrol->value.iec958, hdsp->creg_spdif_stream);
 	return 0;
 }
@@ -1540,7 +1561,7 @@ static int snd_hdsp_control_spdif_stream_put(struct snd_kcontrol *kcontrol, stru
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	u32 val;
-	
+
 	val = snd_hdsp_convert_from_aes(&ucontrol->value.iec958);
 	spin_lock_irq(&hdsp->lock);
 	change = val != hdsp->creg_spdif_stream;
@@ -1602,7 +1623,7 @@ static int snd_hdsp_info_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_
 static int snd_hdsp_get_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_spdif_in(hdsp);
 	return 0;
 }
@@ -1612,7 +1633,7 @@ static int snd_hdsp_put_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_e
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	unsigned int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.enumerated.item[0] % ((hdsp->io_type == H9632) ? 4 : 3);
@@ -1649,7 +1670,7 @@ static int hdsp_set_spdif_output(struct hdsp *hdsp, int out)
 static int snd_hdsp_get_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.integer.value[0] = hdsp_spdif_out(hdsp);
 	return 0;
 }
@@ -1659,7 +1680,7 @@ static int snd_hdsp_put_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	unsigned int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
@@ -1693,7 +1714,7 @@ static int hdsp_set_spdif_professional(struct hdsp *hdsp, int val)
 static int snd_hdsp_get_spdif_professional(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.integer.value[0] = hdsp_spdif_professional(hdsp);
 	return 0;
 }
@@ -1703,7 +1724,7 @@ static int snd_hdsp_put_spdif_professional(struct snd_kcontrol *kcontrol, struct
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	unsigned int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
@@ -1737,7 +1758,7 @@ static int hdsp_set_spdif_emphasis(struct hdsp *hdsp, int val)
 static int snd_hdsp_get_spdif_emphasis(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.integer.value[0] = hdsp_spdif_emphasis(hdsp);
 	return 0;
 }
@@ -1747,7 +1768,7 @@ static int snd_hdsp_put_spdif_emphasis(struct snd_kcontrol *kcontrol, struct snd
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	unsigned int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
@@ -1781,7 +1802,7 @@ static int hdsp_set_spdif_nonaudio(struct hdsp *hdsp, int val)
 static int snd_hdsp_get_spdif_nonaudio(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.integer.value[0] = hdsp_spdif_nonaudio(hdsp);
 	return 0;
 }
@@ -1791,7 +1812,7 @@ static int snd_hdsp_put_spdif_nonaudio(struct snd_kcontrol *kcontrol, struct snd
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	unsigned int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
@@ -1828,7 +1849,7 @@ static int snd_hdsp_info_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct
 static int snd_hdsp_get_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	switch (hdsp_spdif_sample_rate(hdsp)) {
 	case 32000:
 		ucontrol->value.enumerated.item[0] = 0;
@@ -1858,7 +1879,7 @@ static int snd_hdsp_get_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct
 		ucontrol->value.enumerated.item[0] = 9;
 		break;
 	default:
-		ucontrol->value.enumerated.item[0] = 6;		
+		ucontrol->value.enumerated.item[0] = 6;
 	}
 	return 0;
 }
@@ -1882,7 +1903,7 @@ static int snd_hdsp_info_system_sample_rate(struct snd_kcontrol *kcontrol, struc
 static int snd_hdsp_get_system_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp->system_sample_rate;
 	return 0;
 }
@@ -1899,7 +1920,7 @@ static int snd_hdsp_get_system_sample_rate(struct snd_kcontrol *kcontrol, struct
 static int snd_hdsp_info_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"};	
+	static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"};
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 	uinfo->value.enumerated.items = (hdsp->io_type == H9632) ? 10 : 7 ;
@@ -1912,7 +1933,7 @@ static int snd_hdsp_info_autosync_sample_rate(struct snd_kcontrol *kcontrol, str
 static int snd_hdsp_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	switch (hdsp_external_sample_rate(hdsp)) {
 	case 32000:
 		ucontrol->value.enumerated.item[0] = 0;
@@ -1940,9 +1961,9 @@ static int snd_hdsp_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, stru
 		break;
 	case 192000:
 		ucontrol->value.enumerated.item[0] = 9;
-		break;	
+		break;
 	default:
-		ucontrol->value.enumerated.item[0] = 6;		
+		ucontrol->value.enumerated.item[0] = 6;
 	}
 	return 0;
 }
@@ -1968,7 +1989,7 @@ static int hdsp_system_clock_mode(struct hdsp *hdsp)
 static int snd_hdsp_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
 	static char *texts[] = {"Master", "Slave" };
-	
+
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 	uinfo->value.enumerated.items = 2;
@@ -1981,7 +2002,7 @@ static int snd_hdsp_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct
 static int snd_hdsp_get_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_system_clock_mode(hdsp);
 	return 0;
 }
@@ -2018,7 +2039,7 @@ static int hdsp_clock_source(struct hdsp *hdsp)
 		case 192000:
 			return 9;
 		default:
-			return 3;	
+			return 3;
 		}
 	} else {
 		return 0;
@@ -2032,7 +2053,7 @@ static int hdsp_set_clock_source(struct hdsp *hdsp, int mode)
 	case HDSP_CLOCK_SOURCE_AUTOSYNC:
 		if (hdsp_external_sample_rate(hdsp) != 0) {
 		    if (!hdsp_set_rate(hdsp, hdsp_external_sample_rate(hdsp), 1)) {
-			hdsp->control_register &= ~HDSP_ClockModeMaster;		
+			hdsp->control_register &= ~HDSP_ClockModeMaster;
 			hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
 			return 0;
 		    }
@@ -2043,7 +2064,7 @@ static int hdsp_set_clock_source(struct hdsp *hdsp, int mode)
 		break;
 	case HDSP_CLOCK_SOURCE_INTERNAL_44_1KHZ:
 		rate = 44100;
-		break;	    
+		break;
 	case HDSP_CLOCK_SOURCE_INTERNAL_48KHZ:
 		rate = 48000;
 		break;
@@ -2078,13 +2099,13 @@ static int snd_hdsp_info_clock_source(struct snd_kcontrol *kcontrol, struct snd_
 {
 	static char *texts[] = {"AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz", "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz", "Internal 96.0 kHz", "Internal 128 kHz", "Internal 176.4 kHz", "Internal 192.0 KHz" };
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 	if (hdsp->io_type == H9632)
 	    uinfo->value.enumerated.items = 10;
 	else
-	    uinfo->value.enumerated.items = 7;	
+	    uinfo->value.enumerated.items = 7;
 	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]);
@@ -2094,7 +2115,7 @@ static int snd_hdsp_info_clock_source(struct snd_kcontrol *kcontrol, struct snd_
 static int snd_hdsp_get_clock_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_clock_source(hdsp);
 	return 0;
 }
@@ -2104,7 +2125,7 @@ static int snd_hdsp_put_clock_source(struct snd_kcontrol *kcontrol, struct snd_c
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.enumerated.item[0];
@@ -2130,7 +2151,7 @@ static int snd_hdsp_put_clock_source(struct snd_kcontrol *kcontrol, struct snd_c
 static int snd_hdsp_get_clock_source_lock(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.integer.value[0] = hdsp->clock_source_locked;
 	return 0;
 }
@@ -2165,7 +2186,7 @@ static int hdsp_da_gain(struct hdsp *hdsp)
 	case HDSP_DAGainMinus10dBV:
 		return 2;
 	default:
-		return 1;	
+		return 1;
 	}
 }
 
@@ -2180,8 +2201,8 @@ static int hdsp_set_da_gain(struct hdsp *hdsp, int mode)
 		hdsp->control_register |= HDSP_DAGainPlus4dBu;
 		break;
 	case 2:
-		hdsp->control_register |= HDSP_DAGainMinus10dBV;		
-		break;	    
+		hdsp->control_register |= HDSP_DAGainMinus10dBV;
+		break;
 	default:
 		return -1;
 
@@ -2193,7 +2214,7 @@ static int hdsp_set_da_gain(struct hdsp *hdsp, int mode)
 static int snd_hdsp_info_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
 	static char *texts[] = {"Hi Gain", "+4 dBu", "-10 dbV"};
-	
+
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 	uinfo->value.enumerated.items = 3;
@@ -2206,7 +2227,7 @@ static int snd_hdsp_info_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_e
 static int snd_hdsp_get_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_da_gain(hdsp);
 	return 0;
 }
@@ -2216,7 +2237,7 @@ static int snd_hdsp_put_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_el
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.enumerated.item[0];
@@ -2250,7 +2271,7 @@ static int hdsp_ad_gain(struct hdsp *hdsp)
 	case HDSP_ADGainLowGain:
 		return 2;
 	default:
-		return 1;	
+		return 1;
 	}
 }
 
@@ -2262,11 +2283,11 @@ static int hdsp_set_ad_gain(struct hdsp *hdsp, int mode)
 		hdsp->control_register |= HDSP_ADGainMinus10dBV;
 		break;
 	case 1:
-		hdsp->control_register |= HDSP_ADGainPlus4dBu;		
+		hdsp->control_register |= HDSP_ADGainPlus4dBu;
 		break;
 	case 2:
-		hdsp->control_register |= HDSP_ADGainLowGain;		
-		break;	    
+		hdsp->control_register |= HDSP_ADGainLowGain;
+		break;
 	default:
 		return -1;
 
@@ -2278,7 +2299,7 @@ static int hdsp_set_ad_gain(struct hdsp *hdsp, int mode)
 static int snd_hdsp_info_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
 	static char *texts[] = {"-10 dBV", "+4 dBu", "Lo Gain"};
-	
+
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 	uinfo->value.enumerated.items = 3;
@@ -2291,7 +2312,7 @@ static int snd_hdsp_info_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_e
 static int snd_hdsp_get_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_ad_gain(hdsp);
 	return 0;
 }
@@ -2301,7 +2322,7 @@ static int snd_hdsp_put_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_el
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.enumerated.item[0];
@@ -2335,7 +2356,7 @@ static int hdsp_phone_gain(struct hdsp *hdsp)
 	case HDSP_PhoneGainMinus12dB:
 		return 2;
 	default:
-		return 0;	
+		return 0;
 	}
 }
 
@@ -2347,11 +2368,11 @@ static int hdsp_set_phone_gain(struct hdsp *hdsp, int mode)
 		hdsp->control_register |= HDSP_PhoneGain0dB;
 		break;
 	case 1:
-		hdsp->control_register |= HDSP_PhoneGainMinus6dB;		
+		hdsp->control_register |= HDSP_PhoneGainMinus6dB;
 		break;
 	case 2:
-		hdsp->control_register |= HDSP_PhoneGainMinus12dB;		
-		break;	    
+		hdsp->control_register |= HDSP_PhoneGainMinus12dB;
+		break;
 	default:
 		return -1;
 
@@ -2363,7 +2384,7 @@ static int hdsp_set_phone_gain(struct hdsp *hdsp, int mode)
 static int snd_hdsp_info_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
 	static char *texts[] = {"0 dB", "-6 dB", "-12 dB"};
-	
+
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 	uinfo->value.enumerated.items = 3;
@@ -2376,7 +2397,7 @@ static int snd_hdsp_info_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ct
 static int snd_hdsp_get_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_phone_gain(hdsp);
 	return 0;
 }
@@ -2386,7 +2407,7 @@ static int snd_hdsp_put_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.enumerated.item[0];
@@ -2432,7 +2453,7 @@ static int hdsp_set_xlr_breakout_cable(struct hdsp *hdsp, int mode)
 static int snd_hdsp_get_xlr_breakout_cable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_xlr_breakout_cable(hdsp);
 	return 0;
 }
@@ -2442,7 +2463,7 @@ static int snd_hdsp_put_xlr_breakout_cable(struct snd_kcontrol *kcontrol, struct
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
@@ -2488,7 +2509,7 @@ static int hdsp_set_aeb(struct hdsp *hdsp, int mode)
 static int snd_hdsp_get_aeb(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_aeb(hdsp);
 	return 0;
 }
@@ -2498,7 +2519,7 @@ static int snd_hdsp_put_aeb(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
@@ -2576,7 +2597,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd
 {
 	static char *texts[] = {"Word", "IEC958", "ADAT1", "ADAT Sync", "ADAT2", "ADAT3" };
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 
@@ -2595,7 +2616,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd
 		uinfo->value.enumerated.items = 0;
 		break;
 	}
-		
+
 	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]);
@@ -2605,7 +2626,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd
 static int snd_hdsp_get_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_pref_sync_ref(hdsp);
 	return 0;
 }
@@ -2615,7 +2636,7 @@ static int snd_hdsp_put_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change, max;
 	unsigned int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 
@@ -2664,7 +2685,7 @@ static int hdsp_autosync_ref(struct hdsp *hdsp)
 	case HDSP_SelSyncRef_SPDIF:
 		return HDSP_AUTOSYNC_FROM_SPDIF;
 	case HDSP_SelSyncRefMask:
-		return HDSP_AUTOSYNC_FROM_NONE;	
+		return HDSP_AUTOSYNC_FROM_NONE;
 	case HDSP_SelSyncRef_ADAT1:
 		return HDSP_AUTOSYNC_FROM_ADAT1;
 	case HDSP_SelSyncRef_ADAT2:
@@ -2680,7 +2701,7 @@ static int hdsp_autosync_ref(struct hdsp *hdsp)
 static int snd_hdsp_info_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
 	static char *texts[] = {"Word", "ADAT Sync", "IEC958", "None", "ADAT1", "ADAT2", "ADAT3" };
-	
+
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 	uinfo->value.enumerated.items = 7;
@@ -2693,7 +2714,7 @@ static int snd_hdsp_info_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_
 static int snd_hdsp_get_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_autosync_ref(hdsp);
 	return 0;
 }
@@ -2727,7 +2748,7 @@ static int hdsp_set_line_output(struct hdsp *hdsp, int out)
 static int snd_hdsp_get_line_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	spin_lock_irq(&hdsp->lock);
 	ucontrol->value.integer.value[0] = hdsp_line_out(hdsp);
 	spin_unlock_irq(&hdsp->lock);
@@ -2739,7 +2760,7 @@ static int snd_hdsp_put_line_out(struct snd_kcontrol *kcontrol, struct snd_ctl_e
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	unsigned int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
@@ -2773,7 +2794,7 @@ static int hdsp_set_precise_pointer(struct hdsp *hdsp, int precise)
 static int snd_hdsp_get_precise_pointer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	spin_lock_irq(&hdsp->lock);
 	ucontrol->value.integer.value[0] = hdsp->precise_ptr;
 	spin_unlock_irq(&hdsp->lock);
@@ -2785,7 +2806,7 @@ static int snd_hdsp_put_precise_pointer(struct snd_kcontrol *kcontrol, struct sn
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	unsigned int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
@@ -2819,7 +2840,7 @@ static int hdsp_set_use_midi_tasklet(struct hdsp *hdsp, int use_tasklet)
 static int snd_hdsp_get_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	spin_lock_irq(&hdsp->lock);
 	ucontrol->value.integer.value[0] = hdsp->use_midi_tasklet;
 	spin_unlock_irq(&hdsp->lock);
@@ -2831,7 +2852,7 @@ static int snd_hdsp_put_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct s
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	unsigned int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
@@ -2873,12 +2894,12 @@ static int snd_hdsp_get_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
 
 	source = ucontrol->value.integer.value[0];
 	destination = ucontrol->value.integer.value[1];
-	
+
 	if (source >= hdsp->max_channels)
 		addr = hdsp_playback_to_output_key(hdsp,source-hdsp->max_channels,destination);
 	else
 		addr = hdsp_input_to_output_key(hdsp,source, destination);
-	
+
 	spin_lock_irq(&hdsp->lock);
 	ucontrol->value.integer.value[2] = hdsp_read_gain (hdsp, addr);
 	spin_unlock_irq(&hdsp->lock);
@@ -2926,7 +2947,7 @@ static int snd_hdsp_put_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem
 
 static int snd_hdsp_info_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo)
 {
-	static char *texts[] = {"No Lock", "Lock", "Sync" };	
+	static char *texts[] = {"No Lock", "Lock", "Sync" };
 	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
 	uinfo->count = 1;
 	uinfo->value.enumerated.items = 3;
@@ -2971,7 +2992,7 @@ static int hdsp_spdif_sync_check(struct hdsp *hdsp)
 	int status = hdsp_read(hdsp, HDSP_statusRegister);
 	if (status & HDSP_SPDIFErrorFlag)
 		return 0;
-	else {	
+	else {
 		if (status & HDSP_SPDIFSync)
 			return 2;
 		else
@@ -3007,7 +3028,7 @@ static int hdsp_adatsync_sync_check(struct hdsp *hdsp)
 			return 1;
 	} else
 		return 0;
-}	
+}
 
 static int snd_hdsp_get_adatsync_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
@@ -3025,17 +3046,17 @@ static int snd_hdsp_get_adatsync_sync_check(struct snd_kcontrol *kcontrol, struc
 }
 
 static int hdsp_adat_sync_check(struct hdsp *hdsp, int idx)
-{	
+{
 	int status = hdsp_read(hdsp, HDSP_statusRegister);
-	
+
 	if (status & (HDSP_Lock0>>idx)) {
 		if (status & (HDSP_Sync0>>idx))
 			return 2;
 		else
-			return 1;		
+			return 1;
 	} else
 		return 0;
-} 
+}
 
 static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
@@ -3053,7 +3074,7 @@ static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct sn
 		break;
 	case Multiface:
 	case H9632:
-		if (offset >= 1) 
+		if (offset >= 1)
 			return -EINVAL;
 		break;
 	default:
@@ -3115,7 +3136,7 @@ static int snd_hdsp_info_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ct
 static int snd_hdsp_get_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
-	
+
 	ucontrol->value.enumerated.item[0] = hdsp_dds_offset(hdsp);
 	return 0;
 }
@@ -3125,7 +3146,7 @@ static int snd_hdsp_put_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
 	int val;
-	
+
 	if (!snd_hdsp_use_is_exclusive(hdsp))
 		return -EBUSY;
 	val = ucontrol->value.enumerated.item[0];
@@ -3170,7 +3191,7 @@ static struct snd_kcontrol_new snd_hdsp_controls[] = {
 	.get =		snd_hdsp_control_spdif_mask_get,
 	.private_value = IEC958_AES0_NONAUDIO |
   			 IEC958_AES0_PROFESSIONAL |
-			 IEC958_AES0_CON_EMPHASIS,	                                                                                      
+			 IEC958_AES0_CON_EMPHASIS,
 },
 {
 	.access =	SNDRV_CTL_ELEM_ACCESS_READ,
@@ -3188,7 +3209,7 @@ HDSP_SPDIF_OUT("IEC958 Output also on ADAT1", 0),
 HDSP_SPDIF_PROFESSIONAL("IEC958 Professional Bit", 0),
 HDSP_SPDIF_EMPHASIS("IEC958 Emphasis Bit", 0),
 HDSP_SPDIF_NON_AUDIO("IEC958 Non-audio Bit", 0),
-/* 'Sample Clock Source' complies with the alsa control naming scheme */ 
+/* 'Sample Clock Source' complies with the alsa control naming scheme */
 HDSP_CLOCK_SOURCE("Sample Clock Source", 0),
 {
 	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
@@ -3240,7 +3261,7 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp)
 				return err;
 		}
 	}
-	
+
 	/* DA, AD and Phone gain and XLR breakout cable controls for H9632 cards */
 	if (hdsp->io_type == H9632) {
 		for (idx = 0; idx < ARRAY_SIZE(snd_hdsp_9632_controls); idx++) {
@@ -3259,7 +3280,7 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp)
 }
 
 /*------------------------------------------------------------
-   /proc interface 
+   /proc interface
  ------------------------------------------------------------*/
 
 static void
@@ -3298,7 +3319,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 			}
 		}
 	}
-	
+
 	status = hdsp_read(hdsp, HDSP_statusRegister);
 	status2 = hdsp_read(hdsp, HDSP_status2Register);
 
@@ -3362,17 +3383,17 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 		break;
 		case HDSP_CLOCK_SOURCE_INTERNAL_192KHZ:
 		clock_source = "Internal 192 kHz";
-		break;	
+		break;
 	default:
-		clock_source = "Error";		
+		clock_source = "Error";
 	}
 	snd_iprintf (buffer, "Sample Clock Source: %s\n", clock_source);
-			
+
 	if (hdsp_system_clock_mode(hdsp))
 		system_clock_mode = "Slave";
 	else
 		system_clock_mode = "Master";
-	
+
 	switch (hdsp_pref_sync_ref (hdsp)) {
 	case HDSP_SYNC_FROM_WORD:
 		pref_sync_ref = "Word Clock";
@@ -3397,7 +3418,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 		break;
 	}
 	snd_iprintf (buffer, "Preferred Sync Reference: %s\n", pref_sync_ref);
-	
+
 	switch (hdsp_autosync_ref (hdsp)) {
 	case HDSP_AUTOSYNC_FROM_WORD:
 		autosync_ref = "Word Clock";
@@ -3410,7 +3431,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 		break;
 	case HDSP_AUTOSYNC_FROM_NONE:
 		autosync_ref = "None";
-		break;	
+		break;
 	case HDSP_AUTOSYNC_FROM_ADAT1:
 		autosync_ref = "ADAT1";
 		break;
@@ -3425,14 +3446,14 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 		break;
 	}
 	snd_iprintf (buffer, "AutoSync Reference: %s\n", autosync_ref);
-	
+
 	snd_iprintf (buffer, "AutoSync Frequency: %d\n", hdsp_external_sample_rate(hdsp));
-	
+
 	snd_iprintf (buffer, "System Clock Mode: %s\n", system_clock_mode);
 
 	snd_iprintf (buffer, "System Clock Frequency: %d\n", hdsp->system_sample_rate);
 	snd_iprintf (buffer, "System Clock Locked: %s\n", hdsp->clock_source_locked ? "Yes" : "No");
-		
+
 	snd_iprintf(buffer, "\n");
 
 	switch (hdsp_spdif_in(hdsp)) {
@@ -3452,7 +3473,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 		snd_iprintf(buffer, "IEC958 input: ???\n");
 		break;
 	}
-	
+
 	if (hdsp->control_register & HDSP_SPDIFOpticalOut)
 		snd_iprintf(buffer, "IEC958 output: Coaxial & ADAT1\n");
 	else
@@ -3510,13 +3531,13 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 		snd_iprintf (buffer, "SPDIF: No Lock\n");
 	else
 		snd_iprintf (buffer, "SPDIF: %s\n", x ? "Sync" : "Lock");
-	
+
 	x = status2 & HDSP_wc_sync;
 	if (status2 & HDSP_wc_lock)
 		snd_iprintf (buffer, "Word Clock: %s\n", x ? "Sync" : "Lock");
 	else
 		snd_iprintf (buffer, "Word Clock: No Lock\n");
-	
+
 	x = status & HDSP_TimecodeSync;
 	if (status & HDSP_TimecodeLock)
 		snd_iprintf(buffer, "ADAT Sync: %s\n", x ? "Sync" : "Lock");
@@ -3524,11 +3545,11 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 		snd_iprintf(buffer, "ADAT Sync: No Lock\n");
 
 	snd_iprintf(buffer, "\n");
-	
+
 	/* Informations about H9632 specific controls */
 	if (hdsp->io_type == H9632) {
 		char *tmp;
-	
+
 		switch (hdsp_ad_gain(hdsp)) {
 		case 0:
 			tmp = "-10 dBV";
@@ -3554,7 +3575,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 			break;
 		}
 		snd_iprintf(buffer, "DA Gain : %s\n", tmp);
-		
+
 		switch (hdsp_phone_gain(hdsp)) {
 		case 0:
 			tmp = "0 dB";
@@ -3568,8 +3589,8 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 		}
 		snd_iprintf(buffer, "Phones Gain : %s\n", tmp);
 
-		snd_iprintf(buffer, "XLR Breakout Cable : %s\n", hdsp_xlr_breakout_cable(hdsp) ? "yes" : "no");	
-		
+		snd_iprintf(buffer, "XLR Breakout Cable : %s\n", hdsp_xlr_breakout_cable(hdsp) ? "yes" : "no");
+
 		if (hdsp->control_register & HDSP_AnalogExtensionBoard)
 			snd_iprintf(buffer, "AEB : on (ADAT1 internal)\n");
 		else
@@ -3632,18 +3653,18 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp)
 
 	/* set defaults:
 
-	   SPDIF Input via Coax 
+	   SPDIF Input via Coax
 	   Master clock mode
 	   maximum latency (7 => 2^7 = 8192 samples, 64Kbyte buffer,
 	                    which implies 2 4096 sample, 32Kbyte periods).
-           Enable line out.			    
+           Enable line out.
 	 */
 
-	hdsp->control_register = HDSP_ClockModeMaster | 
-		                 HDSP_SPDIFInputCoaxial | 
-		                 hdsp_encode_latency(7) | 
+	hdsp->control_register = HDSP_ClockModeMaster |
+		                 HDSP_SPDIFInputCoaxial |
+		                 hdsp_encode_latency(7) |
 		                 HDSP_LineOut;
-	
+
 
 	hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
 
@@ -3661,7 +3682,7 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp)
 	hdsp_compute_period_size(hdsp);
 
 	/* silence everything */
-	
+
 	for (i = 0; i < HDSP_MATRIX_MIXER_SIZE; ++i)
 		hdsp->mixer_matrix[i] = MINUS_INFINITY_GAIN;
 
@@ -3669,7 +3690,7 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp)
 		if (hdsp_write_gain (hdsp, i, MINUS_INFINITY_GAIN))
 			return -EIO;
 	}
-	
+
 	/* H9632 specific defaults */
 	if (hdsp->io_type == H9632) {
 		hdsp->control_register |= (HDSP_DAGainPlus4dBu | HDSP_ADGainPlus4dBu | HDSP_PhoneGain0dB);
@@ -3687,12 +3708,12 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp)
 static void hdsp_midi_tasklet(unsigned long arg)
 {
 	struct hdsp *hdsp = (struct hdsp *)arg;
-	
+
 	if (hdsp->midi[0].pending)
 		snd_hdsp_midi_input_read (&hdsp->midi[0]);
 	if (hdsp->midi[1].pending)
 		snd_hdsp_midi_input_read (&hdsp->midi[1]);
-} 
+}
 
 static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id)
 {
@@ -3704,7 +3725,7 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id)
 	unsigned int midi0status;
 	unsigned int midi1status;
 	int schedule = 0;
-	
+
 	status = hdsp_read(hdsp, HDSP_statusRegister);
 
 	audio = status & HDSP_audioIRQPending;
@@ -3718,15 +3739,18 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id)
 
 	midi0status = hdsp_read (hdsp, HDSP_midiStatusIn0) & 0xff;
 	midi1status = hdsp_read (hdsp, HDSP_midiStatusIn1) & 0xff;
-	
+
+	if (!(hdsp->state & HDSP_InitializationComplete))
+		return IRQ_HANDLED;
+
 	if (audio) {
 		if (hdsp->capture_substream)
 			snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream);
-		
+
 		if (hdsp->playback_substream)
 			snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream);
 	}
-	
+
 	if (midi0 && midi0status) {
 		if (hdsp->use_midi_tasklet) {
 			/* we disable interrupts for this input until processing is done */
@@ -3769,10 +3793,10 @@ static char *hdsp_channel_buffer_location(struct hdsp *hdsp,
 
         if (snd_BUG_ON(channel < 0 || channel >= hdsp->max_channels))
 		return NULL;
-        
+
 	if ((mapped_channel = hdsp->channel_map[channel]) < 0)
 		return NULL;
-	
+
 	if (stream == SNDRV_PCM_STREAM_CAPTURE)
 		return hdsp->capture_buffer + (mapped_channel * HDSP_CHANNEL_BUFFER_BYTES);
 	else
@@ -3965,7 +3989,7 @@ static int snd_hdsp_trigger(struct snd_pcm_substream *substream, int cmd)
 	struct hdsp *hdsp = snd_pcm_substream_chip(substream);
 	struct snd_pcm_substream *other;
 	int running;
-	
+
 	if (hdsp_check_for_iobox (hdsp))
 		return -EIO;
 
@@ -4059,10 +4083,10 @@ static struct snd_pcm_hardware snd_hdsp_playback_subinfo =
 	.formats =		SNDRV_PCM_FMTBIT_S32_LE,
 #endif
 	.rates =		(SNDRV_PCM_RATE_32000 |
-				 SNDRV_PCM_RATE_44100 | 
-				 SNDRV_PCM_RATE_48000 | 
-				 SNDRV_PCM_RATE_64000 | 
-				 SNDRV_PCM_RATE_88200 | 
+				 SNDRV_PCM_RATE_44100 |
+				 SNDRV_PCM_RATE_48000 |
+				 SNDRV_PCM_RATE_64000 |
+				 SNDRV_PCM_RATE_88200 |
 				 SNDRV_PCM_RATE_96000),
 	.rate_min =		32000,
 	.rate_max =		96000,
@@ -4088,10 +4112,10 @@ static struct snd_pcm_hardware snd_hdsp_capture_subinfo =
 	.formats =		SNDRV_PCM_FMTBIT_S32_LE,
 #endif
 	.rates =		(SNDRV_PCM_RATE_32000 |
-				 SNDRV_PCM_RATE_44100 | 
-				 SNDRV_PCM_RATE_48000 | 
-				 SNDRV_PCM_RATE_64000 | 
-				 SNDRV_PCM_RATE_88200 | 
+				 SNDRV_PCM_RATE_44100 |
+				 SNDRV_PCM_RATE_48000 |
+				 SNDRV_PCM_RATE_64000 |
+				 SNDRV_PCM_RATE_88200 |
 				 SNDRV_PCM_RATE_96000),
 	.rate_min =		32000,
 	.rate_max =		96000,
@@ -4170,7 +4194,7 @@ static int snd_hdsp_hw_rule_in_channels_rate(struct snd_pcm_hw_params *params,
 			.max = hdsp->qs_in_channels,
 			.integer = 1,
 		};
-		return snd_interval_refine(c, &t);	
+		return snd_interval_refine(c, &t);
 	} else if (r->min > 48000 && r->max <= 96000) {
 		struct snd_interval t = {
 			.min = hdsp->ds_in_channels,
@@ -4201,7 +4225,7 @@ static int snd_hdsp_hw_rule_out_channels_rate(struct snd_pcm_hw_params *params,
 			.max = hdsp->qs_out_channels,
 			.integer = 1,
 		};
-		return snd_interval_refine(c, &t);	
+		return snd_interval_refine(c, &t);
 	} else if (r->min > 48000 && r->max <= 96000) {
 		struct snd_interval t = {
 			.min = hdsp->ds_out_channels,
@@ -4318,8 +4342,8 @@ static int snd_hdsp_playback_open(struct snd_pcm_substream *substream)
 	if (hdsp->io_type == H9632) {
 		runtime->hw.channels_min = hdsp->qs_out_channels;
 		runtime->hw.channels_max = hdsp->ss_out_channels;
-	}	
-	
+	}
+
 	snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
 			     snd_hdsp_hw_rule_out_channels, hdsp,
 			     SNDRV_PCM_HW_PARAM_CHANNELS, -1);
@@ -4413,13 +4437,6 @@ static int snd_hdsp_capture_release(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int snd_hdsp_hwdep_dummy_op(struct snd_hwdep *hw, struct file *file)
-{
-	/* we have nothing to initialize but the call is required */
-	return 0;
-}
-
-
 /* helper functions for copying meter values */
 static inline int copy_u32_le(void __user *dest, void __iomem *src)
 {
@@ -4536,7 +4553,7 @@ static int hdsp_get_peak(struct hdsp *hdsp, struct hdsp_peak_rms __user *peak_rm
 				hdsp->iobase + HDSP_playbackRmsLevel + i * 8 + 4,
 				hdsp->iobase + HDSP_playbackRmsLevel + i * 8))
 			return -EFAULT;
-		if (copy_u64_le(&peak_rms->input_rms[i], 
+		if (copy_u64_le(&peak_rms->input_rms[i],
 				hdsp->iobase + HDSP_inputRmsLevel + i * 8 + 4,
 				hdsp->iobase + HDSP_inputRmsLevel + i * 8))
 			return -EFAULT;
@@ -4546,7 +4563,7 @@ static int hdsp_get_peak(struct hdsp *hdsp, struct hdsp_peak_rms __user *peak_rm
 
 static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigned int cmd, unsigned long arg)
 {
-	struct hdsp *hdsp = (struct hdsp *)hw->private_data;	
+	struct hdsp *hdsp = (struct hdsp *)hw->private_data;
 	void __user *argp = (void __user *)arg;
 	int err;
 
@@ -4580,7 +4597,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
 		struct hdsp_config_info info;
 		unsigned long flags;
 		int i;
-		
+
 		err = hdsp_check_for_iobox(hdsp);
 		if (err < 0)
 			return err;
@@ -4614,7 +4631,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
 			info.ad_gain = (unsigned char)hdsp_ad_gain(hdsp);
 			info.phone_gain = (unsigned char)hdsp_phone_gain(hdsp);
 			info.xlr_breakout_cable = (unsigned char)hdsp_xlr_breakout_cable(hdsp);
-		
+
 		}
 		if (hdsp->io_type == H9632 || hdsp->io_type == H9652)
 			info.analog_extension_board = (unsigned char)hdsp_aeb(hdsp);
@@ -4625,7 +4642,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
 	}
 	case SNDRV_HDSP_IOCTL_GET_9632_AEB: {
 		struct hdsp_9632_aeb h9632_aeb;
-		
+
 		if (hdsp->io_type != H9632) return -EINVAL;
 		h9632_aeb.aebi = hdsp->ss_in_channels - H9632_SS_CHANNELS;
 		h9632_aeb.aebo = hdsp->ss_out_channels - H9632_SS_CHANNELS;
@@ -4636,7 +4653,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
 	case SNDRV_HDSP_IOCTL_GET_VERSION: {
 		struct hdsp_version hdsp_version;
 		int err;
-		
+
 		if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return -EINVAL;
 		if (hdsp->io_type == Undefined) {
 			if ((err = hdsp_get_iobox_version(hdsp)) < 0)
@@ -4652,7 +4669,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
 		struct hdsp_firmware __user *firmware;
 		u32 __user *firmware_data;
 		int err;
-		
+
 		if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return -EINVAL;
 		/* SNDRV_HDSP_IOCTL_GET_VERSION must have been called */
 		if (hdsp->io_type == Undefined) return -EINVAL;
@@ -4665,25 +4682,25 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne
 
 		if (get_user(firmware_data, &firmware->firmware_data))
 			return -EFAULT;
-		
+
 		if (hdsp_check_for_iobox (hdsp))
 			return -EIO;
 
 		if (copy_from_user(hdsp->firmware_cache, firmware_data, sizeof(hdsp->firmware_cache)) != 0)
 			return -EFAULT;
-		
+
 		hdsp->state |= HDSP_FirmwareCached;
 
 		if ((err = snd_hdsp_load_firmware_from_cache(hdsp)) < 0)
 			return err;
-		
+
 		if (!(hdsp->state & HDSP_InitializationComplete)) {
 			if ((err = snd_hdsp_enable_io(hdsp)) < 0)
 				return err;
-			
-			snd_hdsp_initialize_channels(hdsp);		
+
+			snd_hdsp_initialize_channels(hdsp);
 			snd_hdsp_initialize_midi_flush(hdsp);
-	    
+
 			if ((err = snd_hdsp_create_alsa_devices(hdsp->card, hdsp)) < 0) {
 				snd_printk(KERN_ERR "Hammerfall-DSP: error creating alsa devices\n");
 				return err;
@@ -4730,18 +4747,16 @@ static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp)
 {
 	struct snd_hwdep *hw;
 	int err;
-	
+
 	if ((err = snd_hwdep_new(card, "HDSP hwdep", 0, &hw)) < 0)
 		return err;
-		
+
 	hdsp->hwdep = hw;
 	hw->private_data = hdsp;
 	strcpy(hw->name, "HDSP hwdep interface");
 
-	hw->ops.open = snd_hdsp_hwdep_dummy_op;
 	hw->ops.ioctl = snd_hdsp_hwdep_ioctl;
-	hw->ops.release = snd_hdsp_hwdep_dummy_op;
-		
+
 	return 0;
 }
 
@@ -4774,24 +4789,24 @@ static void snd_hdsp_9652_enable_mixer (struct hdsp *hdsp)
 static int snd_hdsp_enable_io (struct hdsp *hdsp)
 {
 	int i;
-	
+
 	if (hdsp_fifo_wait (hdsp, 0, 100)) {
 		snd_printk(KERN_ERR "Hammerfall-DSP: enable_io fifo_wait failed\n");
 		return -EIO;
 	}
-	
+
 	for (i = 0; i < hdsp->max_channels; ++i) {
 		hdsp_write (hdsp, HDSP_inputEnable + (4 * i), 1);
 		hdsp_write (hdsp, HDSP_outputEnable + (4 * i), 1);
 	}
-	
+
 	return 0;
 }
 
 static void snd_hdsp_initialize_channels(struct hdsp *hdsp)
 {
 	int status, aebi_channels, aebo_channels;
-	
+
 	switch (hdsp->io_type) {
 	case Digiface:
 		hdsp->card_name = "RME Hammerfall DSP + Digiface";
@@ -4804,7 +4819,7 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp)
 		hdsp->ss_in_channels = hdsp->ss_out_channels = H9652_SS_CHANNELS;
 		hdsp->ds_in_channels = hdsp->ds_out_channels = H9652_DS_CHANNELS;
 		break;
-	
+
 	case H9632:
 		status = hdsp_read(hdsp, HDSP_statusRegister);
 		/* HDSP_AEBx bits are low when AEB are connected */
@@ -4824,7 +4839,7 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp)
 		hdsp->ss_in_channels = hdsp->ss_out_channels = MULTIFACE_SS_CHANNELS;
 		hdsp->ds_in_channels = hdsp->ds_out_channels = MULTIFACE_DS_CHANNELS;
 		break;
-		
+
 	default:
  		/* should never get here */
 		break;
@@ -4840,12 +4855,12 @@ static void snd_hdsp_initialize_midi_flush (struct hdsp *hdsp)
 static int snd_hdsp_create_alsa_devices(struct snd_card *card, struct hdsp *hdsp)
 {
 	int err;
-	
+
 	if ((err = snd_hdsp_create_pcm(card, hdsp)) < 0) {
 		snd_printk(KERN_ERR "Hammerfall-DSP: Error creating pcm interface\n");
 		return err;
 	}
-	
+
 
 	if ((err = snd_hdsp_create_midi(card, hdsp, 0)) < 0) {
 		snd_printk(KERN_ERR "Hammerfall-DSP: Error creating first midi interface\n");
@@ -4876,19 +4891,19 @@ static int snd_hdsp_create_alsa_devices(struct snd_card *card, struct hdsp *hdsp
 		snd_printk(KERN_ERR "Hammerfall-DSP: Error setting default values\n");
 		return err;
 	}
-	
+
 	if (!(hdsp->state & HDSP_InitializationComplete)) {
 		strcpy(card->shortname, "Hammerfall DSP");
-		sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, 
+		sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name,
 			hdsp->port, hdsp->irq);
-	    
+
 		if ((err = snd_card_register(card)) < 0) {
 			snd_printk(KERN_ERR "Hammerfall-DSP: error registering card\n");
 			return err;
 		}
 		hdsp->state |= HDSP_InitializationComplete;
 	}
-	
+
 	return 0;
 }
 
@@ -4899,7 +4914,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp)
 	const char *fwfile;
 	const struct firmware *fw;
 	int err;
-		
+
 	if (hdsp->io_type == H9652 || hdsp->io_type == H9632)
 		return 0;
 	if (hdsp->io_type == Undefined) {
@@ -4908,7 +4923,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp)
 		if (hdsp->io_type == H9652 || hdsp->io_type == H9632)
 			return 0;
 	}
-	
+
 	/* caution: max length of firmware filename is 30! */
 	switch (hdsp->io_type) {
 	case Multiface:
@@ -4942,12 +4957,12 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp)
 	memcpy(hdsp->firmware_cache, fw->data, sizeof(hdsp->firmware_cache));
 
 	release_firmware(fw);
-		
+
 	hdsp->state |= HDSP_FirmwareCached;
 
 	if ((err = snd_hdsp_load_firmware_from_cache(hdsp)) < 0)
 		return err;
-		
+
 	if (!(hdsp->state & HDSP_InitializationComplete)) {
 		if ((err = snd_hdsp_enable_io(hdsp)) < 0)
 			return err;
@@ -4994,14 +5009,14 @@ static int __devinit snd_hdsp_create(struct snd_card *card,
 	hdsp->max_channels = 26;
 
 	hdsp->card = card;
-	
+
 	spin_lock_init(&hdsp->lock);
 
 	tasklet_init(&hdsp->midi_tasklet, hdsp_midi_tasklet, (unsigned long)hdsp);
-	
+
 	pci_read_config_word(hdsp->pci, PCI_CLASS_REVISION, &hdsp->firmware_rev);
 	hdsp->firmware_rev &= 0xff;
-	
+
 	/* From Martin Bjoernsen :
 	    "It is important that the card's latency timer register in
 	    the PCI configuration space is set to a value much larger
@@ -5010,7 +5025,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card,
 	    to its maximum 255 to avoid problems with some computers."
 	*/
 	pci_write_config_byte(hdsp->pci, PCI_LATENCY_TIMER, 0xFF);
-	
+
 	strcpy(card->driver, "H-DSP");
 	strcpy(card->mixername, "Xilinx FPGA");
 
@@ -5024,7 +5039,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card,
 	} else {
 		hdsp->card_name = "RME HDSP 9632";
 		hdsp->max_channels = 16;
-		is_9632 = 1;	
+		is_9632 = 1;
 	}
 
 	if ((err = pci_enable_device(pci)) < 0)
@@ -5053,12 +5068,12 @@ static int __devinit snd_hdsp_create(struct snd_card *card,
 
 	if ((err = snd_hdsp_initialize_memory(hdsp)) < 0)
 		return err;
-	
+
 	if (!is_9652 && !is_9632) {
-		/* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */
-		ssleep(2);
+		/* we wait a maximum of 10 seconds to let freshly
+		 * inserted cardbus cards do their hardware init */
+		err = hdsp_wait_for_iobox(hdsp, 1000, 10);
 
-		err = hdsp_check_for_iobox(hdsp);
 		if (err < 0)
 			return err;
 
@@ -5080,35 +5095,35 @@ static int __devinit snd_hdsp_create(struct snd_card *card,
 				return err;
 			return 0;
 		} else {
-			snd_printk(KERN_INFO "Hammerfall-DSP: Firmware already present, initializing card.\n");	    
+			snd_printk(KERN_INFO "Hammerfall-DSP: Firmware already present, initializing card.\n");
 			if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1)
 				hdsp->io_type = Multiface;
-			else 
+			else
 				hdsp->io_type = Digiface;
 		}
 	}
-	
+
 	if ((err = snd_hdsp_enable_io(hdsp)) != 0)
 		return err;
-	
+
 	if (is_9652)
 	        hdsp->io_type = H9652;
-	
+
 	if (is_9632)
 		hdsp->io_type = H9632;
 
 	if ((err = snd_hdsp_create_hwdep(card, hdsp)) < 0)
 		return err;
-	
+
 	snd_hdsp_initialize_channels(hdsp);
 	snd_hdsp_initialize_midi_flush(hdsp);
 
-	hdsp->state |= HDSP_FirmwareLoaded;	
+	hdsp->state |= HDSP_FirmwareLoaded;
 
 	if ((err = snd_hdsp_create_alsa_devices(card, hdsp)) < 0)
 		return err;
 
-	return 0;	
+	return 0;
 }
 
 static int snd_hdsp_free(struct hdsp *hdsp)
@@ -5124,13 +5139,13 @@ static int snd_hdsp_free(struct hdsp *hdsp)
 		free_irq(hdsp->irq, (void *)hdsp);
 
 	snd_hdsp_free_buffers(hdsp);
-	
+
 	if (hdsp->iobase)
 		iounmap(hdsp->iobase);
 
 	if (hdsp->port)
 		pci_release_regions(hdsp->pci);
-		
+
 	pci_disable_device(hdsp->pci);
 	return 0;
 }
@@ -5158,8 +5173,10 @@ static int __devinit snd_hdsp_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	if (!(card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(struct hdsp))))
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct hdsp), &card);
+	if (err < 0)
+		return err;
 
 	hdsp = (struct hdsp *) card->private_data;
 	card->private_free = snd_hdsp_card_free;
@@ -5173,7 +5190,7 @@ static int __devinit snd_hdsp_probe(struct pci_dev *pci,
 	}
 
 	strcpy(card->shortname, "Hammerfall DSP");
-	sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, 
+	sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name,
 		hdsp->port, hdsp->irq);
 
 	if ((err = snd_card_register(card)) < 0) {
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 71231cf1b2b0..bac2dc0c5d85 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -4100,13 +4100,6 @@ static int snd_hdspm_capture_release(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-static int snd_hdspm_hwdep_dummy_op(struct snd_hwdep * hw, struct file *file)
-{
-	/* we have nothing to initialize but the call is required */
-	return 0;
-}
-
-
 static int snd_hdspm_hwdep_ioctl(struct snd_hwdep * hw, struct file *file,
 				 unsigned int cmd, unsigned long arg)
 {
@@ -4213,9 +4206,7 @@ static int __devinit snd_hdspm_create_hwdep(struct snd_card *card,
 	hw->private_data = hdspm;
 	strcpy(hw->name, "HDSPM hwdep interface");
 
-	hw->ops.open = snd_hdspm_hwdep_dummy_op;
 	hw->ops.ioctl = snd_hdspm_hwdep_ioctl;
-	hw->ops.release = snd_hdspm_hwdep_dummy_op;
 
 	return 0;
 }
@@ -4503,10 +4494,10 @@ static int __devinit snd_hdspm_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev],
-			    THIS_MODULE, sizeof(struct hdspm));
-	if (!card)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev],
+			      THIS_MODULE, sizeof(struct hdspm), &card);
+	if (err < 0)
+		return err;
 
 	hdspm = card->private_data;
 	card->private_free = snd_hdspm_card_free;
diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c
index 2570907134d7..bc539abb2105 100644
--- a/sound/pci/rme9652/rme9652.c
+++ b/sound/pci/rme9652/rme9652.c
@@ -2594,11 +2594,11 @@ static int __devinit snd_rme9652_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-			    sizeof(struct snd_rme9652));
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct snd_rme9652), &card);
 
-	if (!card)
-		return -ENOMEM;
+	if (err < 0)
+		return err;
 
 	rme9652 = (struct snd_rme9652 *) card->private_data;
 	card->private_free = snd_rme9652_card_free;
diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c
index df2007e3be7c..baf6d8e3dabc 100644
--- a/sound/pci/sis7019.c
+++ b/sound/pci/sis7019.c
@@ -1387,9 +1387,8 @@ static int __devinit snd_sis7019_probe(struct pci_dev *pci,
 	if (!enable)
 		goto error_out;
 
-	rc = -ENOMEM;
-	card = snd_card_new(index, id, THIS_MODULE, sizeof(*sis));
-	if (!card)
+	rc = snd_card_create(index, id, THIS_MODULE, sizeof(*sis), &card);
+	if (rc < 0)
 		goto error_out;
 
 	strcpy(card->driver, "SiS7019");
diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c
index cd408b86c839..d989215f3556 100644
--- a/sound/pci/sonicvibes.c
+++ b/sound/pci/sonicvibes.c
@@ -273,7 +273,8 @@ static inline void snd_sonicvibes_setdmaa(struct sonicvibes * sonic,
 	outl(count, sonic->dmaa_port + SV_DMA_COUNT0);
 	outb(0x18, sonic->dmaa_port + SV_DMA_MODE);
 #if 0
-	printk("program dmaa: addr = 0x%x, paddr = 0x%x\n", addr, inl(sonic->dmaa_port + SV_DMA_ADDR0));
+	printk(KERN_DEBUG "program dmaa: addr = 0x%x, paddr = 0x%x\n",
+	       addr, inl(sonic->dmaa_port + SV_DMA_ADDR0));
 #endif
 }
 
@@ -288,7 +289,8 @@ static inline void snd_sonicvibes_setdmac(struct sonicvibes * sonic,
 	outl(count, sonic->dmac_port + SV_DMA_COUNT0);
 	outb(0x14, sonic->dmac_port + SV_DMA_MODE);
 #if 0
-	printk("program dmac: addr = 0x%x, paddr = 0x%x\n", addr, inl(sonic->dmac_port + SV_DMA_ADDR0));
+	printk(KERN_DEBUG "program dmac: addr = 0x%x, paddr = 0x%x\n",
+	       addr, inl(sonic->dmac_port + SV_DMA_ADDR0));
 #endif
 }
 
@@ -355,71 +357,104 @@ static unsigned char snd_sonicvibes_in(struct sonicvibes * sonic, unsigned char
 #if 0
 static void snd_sonicvibes_debug(struct sonicvibes * sonic)
 {
-	printk("SV REGS:          INDEX = 0x%02x  ", inb(SV_REG(sonic, INDEX)));
+	printk(KERN_DEBUG
+	       "SV REGS:          INDEX = 0x%02x  ", inb(SV_REG(sonic, INDEX)));
 	printk("                 STATUS = 0x%02x\n", inb(SV_REG(sonic, STATUS)));
-	printk("  0x00: left input      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x00));
+	printk(KERN_DEBUG
+	       "  0x00: left input      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x00));
 	printk("  0x20: synth rate low  = 0x%02x\n", snd_sonicvibes_in(sonic, 0x20));
-	printk("  0x01: right input     = 0x%02x  ", snd_sonicvibes_in(sonic, 0x01));
+	printk(KERN_DEBUG
+	       "  0x01: right input     = 0x%02x  ", snd_sonicvibes_in(sonic, 0x01));
 	printk("  0x21: synth rate high = 0x%02x\n", snd_sonicvibes_in(sonic, 0x21));
-	printk("  0x02: left AUX1       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x02));
+	printk(KERN_DEBUG
+	       "  0x02: left AUX1       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x02));
 	printk("  0x22: ADC clock       = 0x%02x\n", snd_sonicvibes_in(sonic, 0x22));
-	printk("  0x03: right AUX1      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x03));
+	printk(KERN_DEBUG
+	       "  0x03: right AUX1      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x03));
 	printk("  0x23: ADC alt rate    = 0x%02x\n", snd_sonicvibes_in(sonic, 0x23));
-	printk("  0x04: left CD         = 0x%02x  ", snd_sonicvibes_in(sonic, 0x04));
+	printk(KERN_DEBUG
+	       "  0x04: left CD         = 0x%02x  ", snd_sonicvibes_in(sonic, 0x04));
 	printk("  0x24: ADC pll M       = 0x%02x\n", snd_sonicvibes_in(sonic, 0x24));
-	printk("  0x05: right CD        = 0x%02x  ", snd_sonicvibes_in(sonic, 0x05));
+	printk(KERN_DEBUG
+	       "  0x05: right CD        = 0x%02x  ", snd_sonicvibes_in(sonic, 0x05));
 	printk("  0x25: ADC pll N       = 0x%02x\n", snd_sonicvibes_in(sonic, 0x25));
-	printk("  0x06: left line       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x06));
+	printk(KERN_DEBUG
+	       "  0x06: left line       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x06));
 	printk("  0x26: Synth pll M     = 0x%02x\n", snd_sonicvibes_in(sonic, 0x26));
-	printk("  0x07: right line      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x07));
+	printk(KERN_DEBUG
+	       "  0x07: right line      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x07));
 	printk("  0x27: Synth pll N     = 0x%02x\n", snd_sonicvibes_in(sonic, 0x27));
-	printk("  0x08: MIC             = 0x%02x  ", snd_sonicvibes_in(sonic, 0x08));
+	printk(KERN_DEBUG
+	       "  0x08: MIC             = 0x%02x  ", snd_sonicvibes_in(sonic, 0x08));
 	printk("  0x28: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x28));
-	printk("  0x09: Game port       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x09));
+	printk(KERN_DEBUG
+	       "  0x09: Game port       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x09));
 	printk("  0x29: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x29));
-	printk("  0x0a: left synth      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0a));
+	printk(KERN_DEBUG
+	       "  0x0a: left synth      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0a));
 	printk("  0x2a: MPU401          = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2a));
-	printk("  0x0b: right synth     = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0b));
+	printk(KERN_DEBUG
+	       "  0x0b: right synth     = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0b));
 	printk("  0x2b: drive ctrl      = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2b));
-	printk("  0x0c: left AUX2       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0c));
+	printk(KERN_DEBUG
+	       "  0x0c: left AUX2       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0c));
 	printk("  0x2c: SRS space       = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2c));
-	printk("  0x0d: right AUX2      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0d));
+	printk(KERN_DEBUG
+	       "  0x0d: right AUX2      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0d));
 	printk("  0x2d: SRS center      = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2d));
-	printk("  0x0e: left analog     = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0e));
+	printk(KERN_DEBUG
+	       "  0x0e: left analog     = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0e));
 	printk("  0x2e: wave source     = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2e));
-	printk("  0x0f: right analog    = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0f));
+	printk(KERN_DEBUG
+	       "  0x0f: right analog    = 0x%02x  ", snd_sonicvibes_in(sonic, 0x0f));
 	printk("  0x2f: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2f));
-	printk("  0x10: left PCM        = 0x%02x  ", snd_sonicvibes_in(sonic, 0x10));
+	printk(KERN_DEBUG
+	       "  0x10: left PCM        = 0x%02x  ", snd_sonicvibes_in(sonic, 0x10));
 	printk("  0x30: analog power    = 0x%02x\n", snd_sonicvibes_in(sonic, 0x30));
-	printk("  0x11: right PCM       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x11));
+	printk(KERN_DEBUG
+	       "  0x11: right PCM       = 0x%02x  ", snd_sonicvibes_in(sonic, 0x11));
 	printk("  0x31: analog power    = 0x%02x\n", snd_sonicvibes_in(sonic, 0x31));
-	printk("  0x12: DMA data format = 0x%02x  ", snd_sonicvibes_in(sonic, 0x12));
+	printk(KERN_DEBUG
+	       "  0x12: DMA data format = 0x%02x  ", snd_sonicvibes_in(sonic, 0x12));
 	printk("  0x32: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x32));
-	printk("  0x13: P/C enable      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x13));
+	printk(KERN_DEBUG
+	       "  0x13: P/C enable      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x13));
 	printk("  0x33: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x33));
-	printk("  0x14: U/D button      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x14));
+	printk(KERN_DEBUG
+	       "  0x14: U/D button      = 0x%02x  ", snd_sonicvibes_in(sonic, 0x14));
 	printk("  0x34: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x34));
-	printk("  0x15: revision        = 0x%02x  ", snd_sonicvibes_in(sonic, 0x15));
+	printk(KERN_DEBUG
+	       "  0x15: revision        = 0x%02x  ", snd_sonicvibes_in(sonic, 0x15));
 	printk("  0x35: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x35));
-	printk("  0x16: ADC output ctrl = 0x%02x  ", snd_sonicvibes_in(sonic, 0x16));
+	printk(KERN_DEBUG
+	       "  0x16: ADC output ctrl = 0x%02x  ", snd_sonicvibes_in(sonic, 0x16));
 	printk("  0x36: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x36));
-	printk("  0x17: ---             = 0x%02x  ", snd_sonicvibes_in(sonic, 0x17));
+	printk(KERN_DEBUG
+	       "  0x17: ---             = 0x%02x  ", snd_sonicvibes_in(sonic, 0x17));
 	printk("  0x37: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x37));
-	printk("  0x18: DMA A upper cnt = 0x%02x  ", snd_sonicvibes_in(sonic, 0x18));
+	printk(KERN_DEBUG
+	       "  0x18: DMA A upper cnt = 0x%02x  ", snd_sonicvibes_in(sonic, 0x18));
 	printk("  0x38: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x38));
-	printk("  0x19: DMA A lower cnt = 0x%02x  ", snd_sonicvibes_in(sonic, 0x19));
+	printk(KERN_DEBUG
+	       "  0x19: DMA A lower cnt = 0x%02x  ", snd_sonicvibes_in(sonic, 0x19));
 	printk("  0x39: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x39));
-	printk("  0x1a: ---             = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1a));
+	printk(KERN_DEBUG
+	       "  0x1a: ---             = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1a));
 	printk("  0x3a: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3a));
-	printk("  0x1b: ---             = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1b));
+	printk(KERN_DEBUG
+	       "  0x1b: ---             = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1b));
 	printk("  0x3b: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3b));
-	printk("  0x1c: DMA C upper cnt = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1c));
+	printk(KERN_DEBUG
+	       "  0x1c: DMA C upper cnt = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1c));
 	printk("  0x3c: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3c));
-	printk("  0x1d: DMA C upper cnt = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1d));
+	printk(KERN_DEBUG
+	       "  0x1d: DMA C upper cnt = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1d));
 	printk("  0x3d: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3d));
-	printk("  0x1e: PCM rate low    = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1e));
+	printk(KERN_DEBUG
+	       "  0x1e: PCM rate low    = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1e));
 	printk("  0x3e: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3e));
-	printk("  0x1f: PCM rate high   = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1f));
+	printk(KERN_DEBUG
+	       "  0x1f: PCM rate high   = 0x%02x  ", snd_sonicvibes_in(sonic, 0x1f));
 	printk("  0x3f: ---             = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3f));
 }
 
@@ -476,8 +511,8 @@ static void snd_sonicvibes_pll(unsigned int rate,
 	*res_m = m;
 	*res_n = n;
 #if 0
-	printk("metric = %i, xm = %i, xn = %i\n", metric, xm, xn);
-	printk("pll: m = 0x%x, r = 0x%x, n = 0x%x\n", reg, m, r, n);
+	printk(KERN_DEBUG "metric = %i, xm = %i, xn = %i\n", metric, xm, xn);
+	printk(KERN_DEBUG "pll: m = 0x%x, r = 0x%x, n = 0x%x\n", reg, m, r, n);
 #endif
 }
 
@@ -1423,9 +1458,9 @@ static int __devinit snd_sonic_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
  
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 	for (idx = 0; idx < 5; idx++) {
 		if (pci_resource_start(pci, idx) == 0 ||
 		    !(pci_resource_flags(pci, idx) & IORESOURCE_IO)) {
diff --git a/sound/pci/trident/trident.c b/sound/pci/trident/trident.c
index d94b16ffb385..21cef97d478d 100644
--- a/sound/pci/trident/trident.c
+++ b/sound/pci/trident/trident.c
@@ -89,9 +89,9 @@ static int __devinit snd_trident_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	if ((err = snd_trident_create(card, pci,
 				      pcm_channels[dev],
diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c
index c612b435ca2b..a9da9c184660 100644
--- a/sound/pci/trident/trident_main.c
+++ b/sound/pci/trident/trident_main.c
@@ -68,40 +68,40 @@ static void snd_trident_print_voice_regs(struct snd_trident *trident, int voice)
 {
 	unsigned int val, tmp;
 
-	printk("Trident voice %i:\n", voice);
+	printk(KERN_DEBUG "Trident voice %i:\n", voice);
 	outb(voice, TRID_REG(trident, T4D_LFO_GC_CIR));
 	val = inl(TRID_REG(trident, CH_LBA));
-	printk("LBA: 0x%x\n", val);
+	printk(KERN_DEBUG "LBA: 0x%x\n", val);
 	val = inl(TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC));
-	printk("GVSel: %i\n", val >> 31);
-	printk("Pan: 0x%x\n", (val >> 24) & 0x7f);
-	printk("Vol: 0x%x\n", (val >> 16) & 0xff);
-	printk("CTRL: 0x%x\n", (val >> 12) & 0x0f);
-	printk("EC: 0x%x\n", val & 0x0fff);
+	printk(KERN_DEBUG "GVSel: %i\n", val >> 31);
+	printk(KERN_DEBUG "Pan: 0x%x\n", (val >> 24) & 0x7f);
+	printk(KERN_DEBUG "Vol: 0x%x\n", (val >> 16) & 0xff);
+	printk(KERN_DEBUG "CTRL: 0x%x\n", (val >> 12) & 0x0f);
+	printk(KERN_DEBUG "EC: 0x%x\n", val & 0x0fff);
 	if (trident->device != TRIDENT_DEVICE_ID_NX) {
 		val = inl(TRID_REG(trident, CH_DX_CSO_ALPHA_FMS));
-		printk("CSO: 0x%x\n", val >> 16);
+		printk(KERN_DEBUG "CSO: 0x%x\n", val >> 16);
 		printk("Alpha: 0x%x\n", (val >> 4) & 0x0fff);
-		printk("FMS: 0x%x\n", val & 0x0f);
+		printk(KERN_DEBUG "FMS: 0x%x\n", val & 0x0f);
 		val = inl(TRID_REG(trident, CH_DX_ESO_DELTA));
-		printk("ESO: 0x%x\n", val >> 16);
-		printk("Delta: 0x%x\n", val & 0xffff);
+		printk(KERN_DEBUG "ESO: 0x%x\n", val >> 16);
+		printk(KERN_DEBUG "Delta: 0x%x\n", val & 0xffff);
 		val = inl(TRID_REG(trident, CH_DX_FMC_RVOL_CVOL));
 	} else {		// TRIDENT_DEVICE_ID_NX
 		val = inl(TRID_REG(trident, CH_NX_DELTA_CSO));
 		tmp = (val >> 24) & 0xff;
-		printk("CSO: 0x%x\n", val & 0x00ffffff);
+		printk(KERN_DEBUG "CSO: 0x%x\n", val & 0x00ffffff);
 		val = inl(TRID_REG(trident, CH_NX_DELTA_ESO));
 		tmp |= (val >> 16) & 0xff00;
-		printk("Delta: 0x%x\n", tmp);
-		printk("ESO: 0x%x\n", val & 0x00ffffff);
+		printk(KERN_DEBUG "Delta: 0x%x\n", tmp);
+		printk(KERN_DEBUG "ESO: 0x%x\n", val & 0x00ffffff);
 		val = inl(TRID_REG(trident, CH_NX_ALPHA_FMS_FMC_RVOL_CVOL));
-		printk("Alpha: 0x%x\n", val >> 20);
-		printk("FMS: 0x%x\n", (val >> 16) & 0x0f);
+		printk(KERN_DEBUG "Alpha: 0x%x\n", val >> 20);
+		printk(KERN_DEBUG "FMS: 0x%x\n", (val >> 16) & 0x0f);
 	}
-	printk("FMC: 0x%x\n", (val >> 14) & 3);
-	printk("RVol: 0x%x\n", (val >> 7) & 0x7f);
-	printk("CVol: 0x%x\n", val & 0x7f);
+	printk(KERN_DEBUG "FMC: 0x%x\n", (val >> 14) & 3);
+	printk(KERN_DEBUG "RVol: 0x%x\n", (val >> 7) & 0x7f);
+	printk(KERN_DEBUG "CVol: 0x%x\n", val & 0x7f);
 }
 #endif
 
@@ -496,12 +496,17 @@ void snd_trident_write_voice_regs(struct snd_trident * trident,
 	outl(regs[4], TRID_REG(trident, CH_START + 16));
 
 #if 0
-	printk("written %i channel:\n", voice->number);
-	printk("  regs[0] = 0x%x/0x%x\n", regs[0], inl(TRID_REG(trident, CH_START + 0)));
-	printk("  regs[1] = 0x%x/0x%x\n", regs[1], inl(TRID_REG(trident, CH_START + 4)));
-	printk("  regs[2] = 0x%x/0x%x\n", regs[2], inl(TRID_REG(trident, CH_START + 8)));
-	printk("  regs[3] = 0x%x/0x%x\n", regs[3], inl(TRID_REG(trident, CH_START + 12)));
-	printk("  regs[4] = 0x%x/0x%x\n", regs[4], inl(TRID_REG(trident, CH_START + 16)));
+	printk(KERN_DEBUG "written %i channel:\n", voice->number);
+	printk(KERN_DEBUG "  regs[0] = 0x%x/0x%x\n",
+	       regs[0], inl(TRID_REG(trident, CH_START + 0)));
+	printk(KERN_DEBUG "  regs[1] = 0x%x/0x%x\n",
+	       regs[1], inl(TRID_REG(trident, CH_START + 4)));
+	printk(KERN_DEBUG "  regs[2] = 0x%x/0x%x\n",
+	       regs[2], inl(TRID_REG(trident, CH_START + 8)));
+	printk(KERN_DEBUG "  regs[3] = 0x%x/0x%x\n",
+	       regs[3], inl(TRID_REG(trident, CH_START + 12)));
+	printk(KERN_DEBUG "  regs[4] = 0x%x/0x%x\n",
+	       regs[4], inl(TRID_REG(trident, CH_START + 16)));
 #endif
 }
 
@@ -583,7 +588,7 @@ static void snd_trident_write_vol_reg(struct snd_trident * trident,
 		outb(voice->Vol >> 2, TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC + 2));
 		break;
 	case TRIDENT_DEVICE_ID_SI7018:
-		// printk("voice->Vol = 0x%x\n", voice->Vol);
+		/* printk(KERN_DEBUG "voice->Vol = 0x%x\n", voice->Vol); */
 		outw((voice->CTRL << 12) | voice->Vol,
 		     TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC));
 		break;
diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c
index 1aafe956ee2b..809b233dd4a3 100644
--- a/sound/pci/via82xx.c
+++ b/sound/pci/via82xx.c
@@ -466,7 +466,10 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
 					flag = VIA_TBL_BIT_FLAG; /* period boundary */
 			} else
 				flag = 0; /* period continues to the next */
-			// printk("via: tbl %d: at %d  size %d (rest %d)\n", idx, ofs, r, rest);
+			/*
+			printk(KERN_DEBUG "via: tbl %d: at %d  size %d "
+			       "(rest %d)\n", idx, ofs, r, rest);
+			*/
 			((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag);
 			dev->idx_table[idx].offset = ofs;
 			dev->idx_table[idx].size = r;
@@ -2360,14 +2363,14 @@ static struct snd_pci_quirk dxs_whitelist[] __devinitdata = {
 	SND_PCI_QUIRK(0x1019, 0x0996, "ESC Mobo", VIA_DXS_48K),
 	SND_PCI_QUIRK(0x1019, 0x0a81, "ECS K7VTA3 v8.0", VIA_DXS_NO_VRA),
 	SND_PCI_QUIRK(0x1019, 0x0a85, "ECS L7VMM2", VIA_DXS_NO_VRA),
-	SND_PCI_QUIRK(0x1019, 0, "ESC K8", VIA_DXS_SRC),
+	SND_PCI_QUIRK_VENDOR(0x1019, "ESC K8", VIA_DXS_SRC),
 	SND_PCI_QUIRK(0x1019, 0xaa01, "ESC K8T890-A", VIA_DXS_SRC),
 	SND_PCI_QUIRK(0x1025, 0x0033, "Acer Inspire 1353LM", VIA_DXS_NO_VRA),
 	SND_PCI_QUIRK(0x1025, 0x0046, "Acer Aspire 1524 WLMi", VIA_DXS_SRC),
-	SND_PCI_QUIRK(0x1043, 0, "ASUS A7/A8", VIA_DXS_NO_VRA),
-	SND_PCI_QUIRK(0x1071, 0, "Diverse Notebook", VIA_DXS_NO_VRA),
+	SND_PCI_QUIRK_VENDOR(0x1043, "ASUS A7/A8", VIA_DXS_NO_VRA),
+	SND_PCI_QUIRK_VENDOR(0x1071, "Diverse Notebook", VIA_DXS_NO_VRA),
 	SND_PCI_QUIRK(0x10cf, 0x118e, "FSC Laptop", VIA_DXS_ENABLE),
-	SND_PCI_QUIRK(0x1106, 0, "ASRock", VIA_DXS_SRC),
+	SND_PCI_QUIRK_VENDOR(0x1106, "ASRock", VIA_DXS_SRC),
 	SND_PCI_QUIRK(0x1297, 0xa231, "Shuttle AK31v2", VIA_DXS_SRC),
 	SND_PCI_QUIRK(0x1297, 0xa232, "Shuttle", VIA_DXS_SRC),
 	SND_PCI_QUIRK(0x1297, 0xc160, "Shuttle Sk41G", VIA_DXS_SRC),
@@ -2375,7 +2378,7 @@ static struct snd_pci_quirk dxs_whitelist[] __devinitdata = {
 	SND_PCI_QUIRK(0x1462, 0x3800, "MSI KT266", VIA_DXS_ENABLE),
 	SND_PCI_QUIRK(0x1462, 0x7120, "MSI KT4V", VIA_DXS_ENABLE),
 	SND_PCI_QUIRK(0x1462, 0x7142, "MSI K8MM-V", VIA_DXS_ENABLE),
-	SND_PCI_QUIRK(0x1462, 0, "MSI Mobo", VIA_DXS_SRC),
+	SND_PCI_QUIRK_VENDOR(0x1462, "MSI Mobo", VIA_DXS_SRC),
 	SND_PCI_QUIRK(0x147b, 0x1401, "ABIT KD7(-RAID)", VIA_DXS_ENABLE),
 	SND_PCI_QUIRK(0x147b, 0x1411, "ABIT VA-20", VIA_DXS_ENABLE),
 	SND_PCI_QUIRK(0x147b, 0x1413, "ABIT KV8 Pro", VIA_DXS_ENABLE),
@@ -2389,11 +2392,11 @@ static struct snd_pci_quirk dxs_whitelist[] __devinitdata = {
 	SND_PCI_QUIRK(0x161f, 0x2032, "m680x machines", VIA_DXS_48K),
 	SND_PCI_QUIRK(0x1631, 0xe004, "PB EasyNote 3174", VIA_DXS_ENABLE),
 	SND_PCI_QUIRK(0x1695, 0x3005, "EPoX EP-8K9A", VIA_DXS_ENABLE),
-	SND_PCI_QUIRK(0x1695, 0, "EPoX mobo", VIA_DXS_SRC),
-	SND_PCI_QUIRK(0x16f3, 0, "Jetway K8", VIA_DXS_SRC),
-	SND_PCI_QUIRK(0x1734, 0, "FSC Laptop", VIA_DXS_SRC),
+	SND_PCI_QUIRK_VENDOR(0x1695, "EPoX mobo", VIA_DXS_SRC),
+	SND_PCI_QUIRK_VENDOR(0x16f3, "Jetway K8", VIA_DXS_SRC),
+	SND_PCI_QUIRK_VENDOR(0x1734, "FSC Laptop", VIA_DXS_SRC),
 	SND_PCI_QUIRK(0x1849, 0x3059, "ASRock K7VM2", VIA_DXS_NO_VRA),
-	SND_PCI_QUIRK(0x1849, 0, "ASRock mobo", VIA_DXS_SRC),
+	SND_PCI_QUIRK_VENDOR(0x1849, "ASRock mobo", VIA_DXS_SRC),
 	SND_PCI_QUIRK(0x1919, 0x200a, "Soltek SL-K8",  VIA_DXS_NO_VRA),
 	SND_PCI_QUIRK(0x4005, 0x4710, "MSI K7T266", VIA_DXS_SRC),
 	{ } /* terminator */
@@ -2433,9 +2436,9 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci,
 	unsigned int i;
 	int err;
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	card_type = pci_id->driver_data;
 	switch (card_type) {
diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c
index 5bd79d2a5a15..0d54e3503c1e 100644
--- a/sound/pci/via82xx_modem.c
+++ b/sound/pci/via82xx_modem.c
@@ -328,7 +328,10 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre
 					flag = VIA_TBL_BIT_FLAG; /* period boundary */
 			} else
 				flag = 0; /* period continues to the next */
-			// printk("via: tbl %d: at %d  size %d (rest %d)\n", idx, ofs, r, rest);
+			/*
+			printk(KERN_DEBUG "via: tbl %d: at %d  size %d "
+			       "(rest %d)\n", idx, ofs, r, rest);
+			*/
 			((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag);
 			dev->idx_table[idx].offset = ofs;
 			dev->idx_table[idx].size = r;
@@ -1167,9 +1170,9 @@ static int __devinit snd_via82xx_probe(struct pci_dev *pci,
 	unsigned int i;
 	int err;
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	card_type = pci_id->driver_data;
 	switch (card_type) {
diff --git a/sound/pci/vx222/vx222.c b/sound/pci/vx222/vx222.c
index acc352f4a441..fc9136c3e0d7 100644
--- a/sound/pci/vx222/vx222.c
+++ b/sound/pci/vx222/vx222.c
@@ -204,9 +204,9 @@ static int __devinit snd_vx222_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	switch ((int)pci_id->driver_data) {
 	case VX_PCI_VX222_OLD:
diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c
index 7e87f398ff0b..c0efe4491116 100644
--- a/sound/pci/vx222/vx222_ops.c
+++ b/sound/pci/vx222/vx222_ops.c
@@ -107,7 +107,9 @@ static unsigned char vx2_inb(struct vx_core *chip, int offset)
 static void vx2_outb(struct vx_core *chip, int offset, unsigned char val)
 {
 	outb(val, vx2_reg_addr(chip, offset));
-	//printk("outb: %x -> %x\n", val, vx2_reg_addr(chip, offset));
+	/*
+	printk(KERN_DEBUG "outb: %x -> %x\n", val, vx2_reg_addr(chip, offset));
+	*/
 }
 
 /**
@@ -126,7 +128,9 @@ static unsigned int vx2_inl(struct vx_core *chip, int offset)
  */
 static void vx2_outl(struct vx_core *chip, int offset, unsigned int val)
 {
-	// printk("outl: %x -> %x\n", val, vx2_reg_addr(chip, offset));
+	/*
+	printk(KERN_DEBUG "outl: %x -> %x\n", val, vx2_reg_addr(chip, offset));
+	*/
 	outl(val, vx2_reg_addr(chip, offset));
 }
 
diff --git a/sound/pci/ymfpci/ymfpci.c b/sound/pci/ymfpci/ymfpci.c
index 2631a554845e..4af66661f9b0 100644
--- a/sound/pci/ymfpci/ymfpci.c
+++ b/sound/pci/ymfpci/ymfpci.c
@@ -187,9 +187,9 @@ static int __devinit snd_card_ymfpci_probe(struct pci_dev *pci,
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	switch (pci_id->device) {
 	case 0x0004: str = "YMF724";  model = "DS-1"; break;
diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c
index 90d0d62bd0b4..2f0925236a1b 100644
--- a/sound/pci/ymfpci/ymfpci_main.c
+++ b/sound/pci/ymfpci/ymfpci_main.c
@@ -318,7 +318,12 @@ static void snd_ymfpci_pcm_interrupt(struct snd_ymfpci *chip, struct snd_ymfpci_
 		ypcm->period_pos += delta;
 		ypcm->last_pos = pos;
 		if (ypcm->period_pos >= ypcm->period_size) {
-			// printk("done - active_bank = 0x%x, start = 0x%x\n", chip->active_bank, voice->bank[chip->active_bank].start);
+			/*
+			printk(KERN_DEBUG
+			       "done - active_bank = 0x%x, start = 0x%x\n",
+			       chip->active_bank,
+			       voice->bank[chip->active_bank].start);
+			*/
 			ypcm->period_pos %= ypcm->period_size;
 			spin_unlock(&chip->reg_lock);
 			snd_pcm_period_elapsed(ypcm->substream);
@@ -366,7 +371,12 @@ static void snd_ymfpci_pcm_capture_interrupt(struct snd_pcm_substream *substream
 		ypcm->last_pos = pos;
 		if (ypcm->period_pos >= ypcm->period_size) {
 			ypcm->period_pos %= ypcm->period_size;
-			// printk("done - active_bank = 0x%x, start = 0x%x\n", chip->active_bank, voice->bank[chip->active_bank].start);
+			/*
+			printk(KERN_DEBUG
+			       "done - active_bank = 0x%x, start = 0x%x\n",
+			       chip->active_bank,
+			       voice->bank[chip->active_bank].start);
+			*/
 			spin_unlock(&chip->reg_lock);
 			snd_pcm_period_elapsed(substream);
 			spin_lock(&chip->reg_lock);
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf.c b/sound/pcmcia/pdaudiocf/pdaudiocf.c
index 819aaaac432f..7dea74b71cf1 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf.c
@@ -91,7 +91,7 @@ static int snd_pdacf_dev_free(struct snd_device *device)
  */
 static int snd_pdacf_probe(struct pcmcia_device *link)
 {
-	int i;
+	int i, err;
 	struct snd_pdacf *pdacf;
 	struct snd_card *card;
 	static struct snd_device_ops ops = {
@@ -112,20 +112,23 @@ static int snd_pdacf_probe(struct pcmcia_device *link)
 		return -ENODEV; /* disabled explicitly */
 
 	/* ok, create a card instance */
-	card = snd_card_new(index[i], id[i], THIS_MODULE, 0);
-	if (card == NULL) {
+	err = snd_card_create(index[i], id[i], THIS_MODULE, 0, &card);
+	if (err < 0) {
 		snd_printk(KERN_ERR "pdacf: cannot create a card instance\n");
-		return -ENOMEM;
+		return err;
 	}
 
 	pdacf = snd_pdacf_create(card);
-	if (! pdacf)
-		return -EIO;
+	if (!pdacf) {
+		snd_card_free(card);
+		return -ENOMEM;
+	}
 
-	if (snd_device_new(card, SNDRV_DEV_LOWLEVEL, pdacf, &ops) < 0) {
+	err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, pdacf, &ops);
+	if (err < 0) {
 		kfree(pdacf);
 		snd_card_free(card);
-		return -ENODEV;
+		return err;
 	}
 
 	snd_card_set_dev(card, &handle_to_dev(link));
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c
index dfa40b0ed86d..5d2afa0b0ce4 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c
@@ -82,14 +82,21 @@ static void pdacf_ak4117_write(void *private_data, unsigned char reg, unsigned c
 #if 0
 void pdacf_dump(struct snd_pdacf *chip)
 {
-	printk("PDAUDIOCF DUMP (0x%lx):\n", chip->port);
-	printk("WPD         : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_WDP));
-	printk("RDP         : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_RDP));
-	printk("TCR         : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_TCR));
-	printk("SCR         : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_SCR));
-	printk("ISR         : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_ISR));
-	printk("IER         : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_IER));
-	printk("AK_IFR      : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_AK_IFR));
+	printk(KERN_DEBUG "PDAUDIOCF DUMP (0x%lx):\n", chip->port);
+	printk(KERN_DEBUG "WPD         : 0x%x\n",
+	       inw(chip->port + PDAUDIOCF_REG_WDP));
+	printk(KERN_DEBUG "RDP         : 0x%x\n",
+	       inw(chip->port + PDAUDIOCF_REG_RDP));
+	printk(KERN_DEBUG "TCR         : 0x%x\n",
+	       inw(chip->port + PDAUDIOCF_REG_TCR));
+	printk(KERN_DEBUG "SCR         : 0x%x\n",
+	       inw(chip->port + PDAUDIOCF_REG_SCR));
+	printk(KERN_DEBUG "ISR         : 0x%x\n",
+	       inw(chip->port + PDAUDIOCF_REG_ISR));
+	printk(KERN_DEBUG "IER         : 0x%x\n",
+	       inw(chip->port + PDAUDIOCF_REG_IER));
+	printk(KERN_DEBUG "AK_IFR      : 0x%x\n",
+	       inw(chip->port + PDAUDIOCF_REG_AK_IFR));
 }
 #endif
 
diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c b/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c
index ea903c8e90dd..dcd32201bc8c 100644
--- a/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c
+++ b/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c
@@ -269,7 +269,7 @@ void pdacf_tasklet(unsigned long private_data)
 
 	rdp = inw(chip->port + PDAUDIOCF_REG_RDP);
 	wdp = inw(chip->port + PDAUDIOCF_REG_WDP);
-	// printk("TASKLET: rdp = %x, wdp = %x\n", rdp, wdp);
+	/* printk(KERN_DEBUG "TASKLET: rdp = %x, wdp = %x\n", rdp, wdp); */
 	size = wdp - rdp;
 	if (size < 0)
 		size += 0x10000;
@@ -321,5 +321,5 @@ void pdacf_tasklet(unsigned long private_data)
 		spin_lock(&chip->reg_lock);
 	}
 	spin_unlock(&chip->reg_lock);
-	// printk("TASKLET: end\n");
+	/* printk(KERN_DEBUG "TASKLET: end\n"); */
 }
diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c
index 706602a40600..7445cc8a47d3 100644
--- a/sound/pcmcia/vx/vxpocket.c
+++ b/sound/pcmcia/vx/vxpocket.c
@@ -130,23 +130,26 @@ static struct snd_vx_hardware vxp440_hw = {
 /*
  * create vxpocket instance
  */
-static struct snd_vxpocket *snd_vxpocket_new(struct snd_card *card, int ibl,
-					     struct pcmcia_device *link)
+static int snd_vxpocket_new(struct snd_card *card, int ibl,
+			    struct pcmcia_device *link,
+			    struct snd_vxpocket **chip_ret)
 {
 	struct vx_core *chip;
 	struct snd_vxpocket *vxp;
 	static struct snd_device_ops ops = {
 		.dev_free =	snd_vxpocket_dev_free,
 	};
+	int err;
 
 	chip = snd_vx_create(card, &vxpocket_hw, &snd_vxpocket_ops,
 			     sizeof(struct snd_vxpocket) - sizeof(struct vx_core));
-	if (! chip)
-		return NULL;
+	if (!chip)
+		return -ENOMEM;
 
-	if (snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops) < 0) {
+	err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+	if (err < 0) {
 		kfree(chip);
-		return NULL;
+		return err;
 	}
 	chip->ibl.size = ibl;
 
@@ -169,7 +172,8 @@ static struct snd_vxpocket *snd_vxpocket_new(struct snd_card *card, int ibl,
 	link->conf.ConfigIndex = 1;
 	link->conf.Present = PRESENT_OPTION;
 
-	return vxp;
+	*chip_ret = vxp;
+	return 0;
 }
 
 
@@ -292,7 +296,7 @@ static int vxpocket_probe(struct pcmcia_device *p_dev)
 {
 	struct snd_card *card;
 	struct snd_vxpocket *vxp;
-	int i;
+	int i, err;
 
 	/* find an empty slot from the card list */
 	for (i = 0; i < SNDRV_CARDS; i++) {
@@ -307,16 +311,16 @@ static int vxpocket_probe(struct pcmcia_device *p_dev)
 		return -ENODEV; /* disabled explicitly */
 
 	/* ok, create a card instance */
-	card = snd_card_new(index[i], id[i], THIS_MODULE, 0);
-	if (card == NULL) {
+	err = snd_card_create(index[i], id[i], THIS_MODULE, 0, &card);
+	if (err < 0) {
 		snd_printk(KERN_ERR "vxpocket: cannot create a card instance\n");
-		return -ENOMEM;
+		return err;
 	}
 
-	vxp = snd_vxpocket_new(card, ibl[i], p_dev);
-	if (! vxp) {
+	err = snd_vxpocket_new(card, ibl[i], p_dev, &vxp);
+	if (err < 0) {
 		snd_card_free(card);
-		return -ENODEV;
+		return err;
 	}
 	card->private_data = vxp;
 
diff --git a/sound/ppc/Kconfig b/sound/ppc/Kconfig
index 777de2b17178..bd2338ab2ced 100644
--- a/sound/ppc/Kconfig
+++ b/sound/ppc/Kconfig
@@ -13,6 +13,7 @@ config SND_POWERMAC
 	tristate "PowerMac (AWACS, DACA, Burgundy, Tumbler, Keywest)"
 	depends on I2C && INPUT && PPC_PMAC
 	select SND_PCM
+	select SND_VMASTER
 	help
 	  Say Y here to include support for the integrated sound device.
 
diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c
index 7bd33e6552ab..80df9b1f651e 100644
--- a/sound/ppc/awacs.c
+++ b/sound/ppc/awacs.c
@@ -608,9 +608,12 @@ static struct snd_kcontrol_new snd_pmac_screamer_mixers_beige[] __initdata = {
 	AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_LINE, 0),
 };
 
-static struct snd_kcontrol_new snd_pmac_screamer_mixers_imac[] __initdata = {
+static struct snd_kcontrol_new snd_pmac_screamer_mixers_lo[] __initdata = {
 	AWACS_VOLUME("Line out Playback Volume", 2, 6, 1),
-	AWACS_VOLUME("Master Playback Volume", 5, 6, 1),
+};
+
+static struct snd_kcontrol_new snd_pmac_screamer_mixers_imac[] __initdata = {
+	AWACS_VOLUME("Play-through Playback Volume", 5, 6, 1),
 	AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0),
 };
 
@@ -627,6 +630,10 @@ static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac7500[] __initdata = {
 	AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0),
 };
 
+static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac5500[] __initdata = {
+	AWACS_VOLUME("Headphone Playback Volume", 2, 6, 1),
+};
+
 static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac[] __initdata = {
 	AWACS_VOLUME("Master Playback Volume", 2, 6, 1),
 	AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0),
@@ -645,12 +652,19 @@ static struct snd_kcontrol_new snd_pmac_screamer_mixers2[] __initdata = {
 	AWACS_SWITCH("Mic Capture Switch", 0, SHIFT_MUX_LINE, 0),
 };
 
+static struct snd_kcontrol_new snd_pmac_awacs_mixers2_pmac5500[] __initdata = {
+	AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0),
+};
+
 static struct snd_kcontrol_new snd_pmac_awacs_master_sw __initdata =
 AWACS_SWITCH("Master Playback Switch", 1, SHIFT_HDMUTE, 1);
 
 static struct snd_kcontrol_new snd_pmac_awacs_master_sw_imac __initdata =
 AWACS_SWITCH("Line out Playback Switch", 1, SHIFT_HDMUTE, 1);
 
+static struct snd_kcontrol_new snd_pmac_awacs_master_sw_pmac5500 __initdata =
+AWACS_SWITCH("Headphone Playback Switch", 1, SHIFT_HDMUTE, 1);
+
 static struct snd_kcontrol_new snd_pmac_awacs_mic_boost[] __initdata = {
 	AWACS_SWITCH("Mic Boost Capture Switch", 0, SHIFT_GAINLINE, 0),
 };
@@ -766,12 +780,16 @@ static void snd_pmac_awacs_resume(struct snd_pmac *chip)
 }
 #endif /* CONFIG_PM */
 
-#define IS_PM7500 (machine_is_compatible("AAPL,7500"))
+#define IS_PM7500 (machine_is_compatible("AAPL,7500") \
+		|| machine_is_compatible("AAPL,8500") \
+		|| machine_is_compatible("AAPL,9500"))
+#define IS_PM5500 (machine_is_compatible("AAPL,e411"))
 #define IS_BEIGE (machine_is_compatible("AAPL,Gossamer"))
 #define IS_IMAC1 (machine_is_compatible("PowerMac2,1"))
 #define IS_IMAC2 (machine_is_compatible("PowerMac2,2") \
 		|| machine_is_compatible("PowerMac4,1"))
 #define IS_G4AGP (machine_is_compatible("PowerMac3,1"))
+#define IS_LOMBARD (machine_is_compatible("PowerBook1,1"))
 
 static int imac1, imac2;
 
@@ -858,10 +876,14 @@ int __init
 snd_pmac_awacs_init(struct snd_pmac *chip)
 {
 	int pm7500 = IS_PM7500;
+	int pm5500 = IS_PM5500;
 	int beige = IS_BEIGE;
 	int g4agp = IS_G4AGP;
+	int lombard = IS_LOMBARD;
 	int imac;
 	int err, vol;
+	struct snd_kcontrol *vmaster_sw, *vmaster_vol;
+	struct snd_kcontrol *master_vol, *speaker_vol;
 
 	imac1 = IS_IMAC1;
 	imac2 = IS_IMAC2;
@@ -915,7 +937,7 @@ snd_pmac_awacs_init(struct snd_pmac *chip)
 		/* set headphone-jack detection bit */
 		switch (chip->model) {
 		case PMAC_AWACS:
-			chip->hp_stat_mask = pm7500 ? MASK_HDPCONN
+			chip->hp_stat_mask = pm7500 || pm5500 ? MASK_HDPCONN
 				: MASK_LOCONN;
 			break;
 		case PMAC_SCREAMER:
@@ -954,7 +976,7 @@ snd_pmac_awacs_init(struct snd_pmac *chip)
 		return err;
 	if (beige || g4agp)
 		;
-	else if (chip->model == PMAC_SCREAMER)
+	else if (chip->model == PMAC_SCREAMER || pm5500)
 		err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mixers2),
 				   snd_pmac_screamer_mixers2);
 	else if (!pm7500)
@@ -962,19 +984,35 @@ snd_pmac_awacs_init(struct snd_pmac *chip)
 				   snd_pmac_awacs_mixers2);
 	if (err < 0)
 		return err;
+	if (pm5500) {
+		err = build_mixers(chip,
+				   ARRAY_SIZE(snd_pmac_awacs_mixers2_pmac5500),
+				   snd_pmac_awacs_mixers2_pmac5500);
+		if (err < 0)
+			return err;
+	}
 	if (pm7500)
 		err = build_mixers(chip,
 				   ARRAY_SIZE(snd_pmac_awacs_mixers_pmac7500),
 				   snd_pmac_awacs_mixers_pmac7500);
+	else if (pm5500)
+		err = snd_ctl_add(chip->card,
+		    (master_vol = snd_ctl_new1(snd_pmac_awacs_mixers_pmac5500,
+						chip)));
 	else if (beige)
 		err = build_mixers(chip,
 				   ARRAY_SIZE(snd_pmac_screamer_mixers_beige),
 				   snd_pmac_screamer_mixers_beige);
-	else if (imac)
+	else if (imac || lombard) {
+		err = snd_ctl_add(chip->card,
+		    (master_vol = snd_ctl_new1(snd_pmac_screamer_mixers_lo,
+						chip)));
+		if (err < 0)
+			return err;
 		err = build_mixers(chip,
 				   ARRAY_SIZE(snd_pmac_screamer_mixers_imac),
 				   snd_pmac_screamer_mixers_imac);
-	else if (g4agp)
+	} else if (g4agp)
 		err = build_mixers(chip,
 				   ARRAY_SIZE(snd_pmac_screamer_mixers_g4agp),
 				   snd_pmac_screamer_mixers_g4agp);
@@ -984,8 +1022,10 @@ snd_pmac_awacs_init(struct snd_pmac *chip)
 				   snd_pmac_awacs_mixers_pmac);
 	if (err < 0)
 		return err;
-	chip->master_sw_ctl = snd_ctl_new1((pm7500 || imac || g4agp)
+	chip->master_sw_ctl = snd_ctl_new1((pm7500 || imac || g4agp || lombard)
 			? &snd_pmac_awacs_master_sw_imac
+			: pm5500
+			? &snd_pmac_awacs_master_sw_pmac5500
 			: &snd_pmac_awacs_master_sw, chip);
 	err = snd_ctl_add(chip->card, chip->master_sw_ctl);
 	if (err < 0)
@@ -1017,8 +1057,9 @@ snd_pmac_awacs_init(struct snd_pmac *chip)
 #endif /* PMAC_AMP_AVAIL */
 	{
 		/* route A = headphone, route C = speaker */
-		err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_speaker_vol),
-					snd_pmac_awacs_speaker_vol);
+		err = snd_ctl_add(chip->card,
+		    (speaker_vol = snd_ctl_new1(snd_pmac_awacs_speaker_vol,
+						chip)));
 		if (err < 0)
 			return err;
 		chip->speaker_sw_ctl = snd_ctl_new1(imac1
@@ -1031,6 +1072,33 @@ snd_pmac_awacs_init(struct snd_pmac *chip)
 			return err;
 	}
 
+	if (pm5500 || imac || lombard) {
+		vmaster_sw = snd_ctl_make_virtual_master(
+			"Master Playback Switch", (unsigned int *) NULL);
+		err = snd_ctl_add_slave_uncached(vmaster_sw,
+						 chip->master_sw_ctl);
+		if (err < 0)
+			return err;
+		err = snd_ctl_add_slave_uncached(vmaster_sw,
+						  chip->speaker_sw_ctl);
+		if (err < 0)
+			return err;
+		err = snd_ctl_add(chip->card, vmaster_sw);
+		if (err < 0)
+			return err;
+		vmaster_vol = snd_ctl_make_virtual_master(
+			"Master Playback Volume", (unsigned int *) NULL);
+		err = snd_ctl_add_slave(vmaster_vol, master_vol);
+		if (err < 0)
+			return err;
+		err = snd_ctl_add_slave(vmaster_vol, speaker_vol);
+		if (err < 0)
+			return err;
+		err = snd_ctl_add(chip->card, vmaster_vol);
+		if (err < 0)
+			return err;
+	}
+
 	if (beige || g4agp)
 		err = build_mixers(chip,
 				ARRAY_SIZE(snd_pmac_screamer_mic_boost_beige),
diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c
index f860d39af36b..45a76297c38d 100644
--- a/sound/ppc/burgundy.c
+++ b/sound/ppc/burgundy.c
@@ -35,7 +35,7 @@ snd_pmac_burgundy_busy_wait(struct snd_pmac *chip)
 	int timeout = 50;
 	while ((in_le32(&chip->awacs->codec_ctrl) & MASK_NEWECMD) && timeout--)
 		udelay(1);
-	if (! timeout)
+	if (timeout < 0)
 		printk(KERN_DEBUG "burgundy_busy_wait: timeout\n");
 }
 
diff --git a/sound/ppc/daca.c b/sound/ppc/daca.c
index 8a5b29031933..f8d478c2da62 100644
--- a/sound/ppc/daca.c
+++ b/sound/ppc/daca.c
@@ -82,7 +82,7 @@ static int daca_set_volume(struct pmac_daca *mix)
 	data[1] |= mix->deemphasis ? 0x40 : 0;
 	if (i2c_smbus_write_block_data(mix->i2c.client, DACA_REG_AVOL,
 				       2, data) < 0) {
-		snd_printk("failed to set volume \n");
+		snd_printk(KERN_ERR "failed to set volume \n");
 		return -EINVAL;
 	}
 	return 0;
diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c
index af76ee862d27..9b4e9c316695 100644
--- a/sound/ppc/pmac.c
+++ b/sound/ppc/pmac.c
@@ -299,7 +299,7 @@ static int snd_pmac_pcm_trigger(struct snd_pmac *chip, struct pmac_stream *rec,
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 		spin_lock(&chip->reg_lock);
 		rec->running = 0;
-		/*printk("stopped!!\n");*/
+		/*printk(KERN_DEBUG "stopped!!\n");*/
 		snd_pmac_dma_stop(rec);
 		for (i = 0, cp = rec->cmd.cmds; i < rec->nperiods; i++, cp++)
 			out_le16(&cp->command, DBDMA_STOP);
@@ -334,7 +334,7 @@ static snd_pcm_uframes_t snd_pmac_pcm_pointer(struct snd_pmac *chip,
 	}
 #endif
 	count += rec->cur_period * rec->period_size;
-	/*printk("pointer=%d\n", count);*/
+	/*printk(KERN_DEBUG "pointer=%d\n", count);*/
 	return bytes_to_frames(subs->runtime, count);
 }
 
@@ -486,7 +486,7 @@ static void snd_pmac_pcm_update(struct snd_pmac *chip, struct pmac_stream *rec)
 			if (! (stat & ACTIVE))
 				break;
 
-			/*printk("update frag %d\n", rec->cur_period);*/
+			/*printk(KERN_DEBUG "update frag %d\n", rec->cur_period);*/
 			st_le16(&cp->xfer_status, 0);
 			st_le16(&cp->req_count, rec->period_size);
 			/*st_le16(&cp->res_count, 0);*/
@@ -806,7 +806,7 @@ snd_pmac_ctrl_intr(int irq, void *devid)
 	struct snd_pmac *chip = devid;
 	int ctrl = in_le32(&chip->awacs->control);
 
-	/*printk("pmac: control interrupt.. 0x%x\n", ctrl);*/
+	/*printk(KERN_DEBUG "pmac: control interrupt.. 0x%x\n", ctrl);*/
 	if (ctrl & MASK_PORTCHG) {
 		/* do something when headphone is plugged/unplugged? */
 		if (chip->update_automute)
@@ -1033,7 +1033,8 @@ static int __init snd_pmac_detect(struct snd_pmac *chip)
 	}
 	if (of_device_is_compatible(sound, "tumbler")) {
 		chip->model = PMAC_TUMBLER;
-		chip->can_capture = machine_is_compatible("PowerMac4,2");
+		chip->can_capture = machine_is_compatible("PowerMac4,2")
+				|| machine_is_compatible("PowerBook4,1");
 		chip->can_duplex = 0;
 		// chip->can_byte_swap = 0; /* FIXME: check this */
 		chip->num_freqs = ARRAY_SIZE(tumbler_freqs);
diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c
index c936225771ba..5a929069dce9 100644
--- a/sound/ppc/powermac.c
+++ b/sound/ppc/powermac.c
@@ -58,9 +58,9 @@ static int __init snd_pmac_probe(struct platform_device *devptr)
 	char *name_ext;
 	int err;
 
-	card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
 
 	if ((err = snd_pmac_new(card, &chip)) < 0)
 		goto __error;
@@ -110,7 +110,7 @@ static int __init snd_pmac_probe(struct platform_device *devptr)
 			goto __error;
 		break;
 	default:
-		snd_printk("unsupported hardware %d\n", chip->model);
+		snd_printk(KERN_ERR "unsupported hardware %d\n", chip->model);
 		err = -EINVAL;
 		goto __error;
 	}
diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c
index ff321110ec02..f361c26506aa 100644
--- a/sound/ppc/snd_ps3.c
+++ b/sound/ppc/snd_ps3.c
@@ -969,11 +969,9 @@ static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev)
 	}
 
 	/* create card instance */
-	the_card.card = snd_card_new(index, id, THIS_MODULE, 0);
-	if (!the_card.card) {
-		ret = -ENXIO;
+	ret = snd_card_create(index, id, THIS_MODULE, 0, &the_card.card);
+	if (ret < 0)
 		goto clean_irq;
-	}
 
 	strcpy(the_card.card->driver, "PS3");
 	strcpy(the_card.card->shortname, "PS3");
diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c
index 3eb223385416..40222fcc0878 100644
--- a/sound/ppc/tumbler.c
+++ b/sound/ppc/tumbler.c
@@ -41,7 +41,7 @@
 #undef DEBUG
 
 #ifdef DEBUG
-#define DBG(fmt...) printk(fmt)
+#define DBG(fmt...) printk(KERN_DEBUG fmt)
 #else
 #define DBG(fmt...)
 #endif
@@ -240,7 +240,7 @@ static int tumbler_set_master_volume(struct pmac_tumbler *mix)
   
 	if (i2c_smbus_write_i2c_block_data(mix->i2c.client, TAS_REG_VOL, 6,
 					   block) < 0) {
-		snd_printk("failed to set volume \n");
+		snd_printk(KERN_ERR "failed to set volume \n");
 		return -EINVAL;
 	}
 	return 0;
@@ -350,7 +350,7 @@ static int tumbler_set_drc(struct pmac_tumbler *mix)
 
 	if (i2c_smbus_write_i2c_block_data(mix->i2c.client, TAS_REG_DRC,
 					   2, val) < 0) {
-		snd_printk("failed to set DRC\n");
+		snd_printk(KERN_ERR "failed to set DRC\n");
 		return -EINVAL;
 	}
 	return 0;
@@ -386,7 +386,7 @@ static int snapper_set_drc(struct pmac_tumbler *mix)
 
 	if (i2c_smbus_write_i2c_block_data(mix->i2c.client, TAS_REG_DRC,
 					   6, val) < 0) {
-		snd_printk("failed to set DRC\n");
+		snd_printk(KERN_ERR "failed to set DRC\n");
 		return -EINVAL;
 	}
 	return 0;
@@ -506,7 +506,8 @@ static int tumbler_set_mono_volume(struct pmac_tumbler *mix,
 		block[i] = (vol >> ((info->bytes - i - 1) * 8)) & 0xff;
 	if (i2c_smbus_write_i2c_block_data(mix->i2c.client, info->reg,
 					   info->bytes, block) < 0) {
-		snd_printk("failed to set mono volume %d\n", info->index);
+		snd_printk(KERN_ERR "failed to set mono volume %d\n",
+			   info->index);
 		return -EINVAL;
 	}
 	return 0;
@@ -643,7 +644,7 @@ static int snapper_set_mix_vol1(struct pmac_tumbler *mix, int idx, int ch, int r
 	}
 	if (i2c_smbus_write_i2c_block_data(mix->i2c.client, reg,
 					   9, block) < 0) {
-		snd_printk("failed to set mono volume %d\n", reg);
+		snd_printk(KERN_ERR "failed to set mono volume %d\n", reg);
 		return -EINVAL;
 	}
 	return 0;
diff --git a/sound/sh/Kconfig b/sound/sh/Kconfig
index cfc143985802..aed0f90c3919 100644
--- a/sound/sh/Kconfig
+++ b/sound/sh/Kconfig
@@ -15,6 +15,7 @@ config SND_AICA
 	tristate "Dreamcast Yamaha AICA sound"
 	depends on SH_DREAMCAST
 	select SND_PCM
+	select G2_DMA
 	help
 	  ALSA Sound driver for the SEGA Dreamcast console.
 
diff --git a/sound/sh/aica.c b/sound/sh/aica.c
index 7c920f3e7fe3..f551233c5a08 100644
--- a/sound/sh/aica.c
+++ b/sound/sh/aica.c
@@ -609,11 +609,11 @@ static int __devinit snd_aica_probe(struct platform_device *devptr)
 	dreamcastcard = kmalloc(sizeof(struct snd_card_aica), GFP_KERNEL);
 	if (unlikely(!dreamcastcard))
 		return -ENOMEM;
-	dreamcastcard->card =
-	    snd_card_new(index, SND_AICA_DRIVER, THIS_MODULE, 0);
-	if (unlikely(!dreamcastcard->card)) {
+	err = snd_card_create(index, SND_AICA_DRIVER, THIS_MODULE, 0,
+			      &dreamcastcard->card);
+	if (unlikely(err < 0)) {
 		kfree(dreamcastcard);
-		return -ENODEV;
+		return err;
 	}
 	strcpy(dreamcastcard->card->driver, "snd_aica");
 	strcpy(dreamcastcard->card->shortname, SND_AICA_DRIVER);
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index ef025c66cc66..3d2bb6fc6dcc 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -6,6 +6,7 @@ menuconfig SND_SOC
 	tristate "ALSA for SoC audio support"
 	select SND_PCM
 	select AC97_BUS if SND_SOC_AC97_BUS
+	select SND_JACK if INPUT=y || INPUT=SND
 	---help---
 
 	  If you want ASoC support, you should say Y here and also to the
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 86a9b1f5b0f3..0237879fd412 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,4 +1,4 @@
-snd-soc-core-objs := soc-core.o soc-dapm.o
+snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o
 
 obj-$(CONFIG_SND_SOC)	+= snd-soc-core.o
 obj-$(CONFIG_SND_SOC)	+= codecs/
diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c
index 3dcdc4e3cfa0..9ef6b96373f5 100644
--- a/sound/soc/atmel/atmel-pcm.c
+++ b/sound/soc/atmel/atmel-pcm.c
@@ -347,7 +347,7 @@ static int atmel_pcm_mmap(struct snd_pcm_substream *substream,
 		       vma->vm_end - vma->vm_start, vma->vm_page_prot);
 }
 
-struct snd_pcm_ops atmel_pcm_ops = {
+static struct snd_pcm_ops atmel_pcm_ops = {
 	.open		= atmel_pcm_open,
 	.close		= atmel_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c
index ff0054b76502..e588e63f18d2 100644
--- a/sound/soc/atmel/atmel_ssc_dai.c
+++ b/sound/soc/atmel/atmel_ssc_dai.c
@@ -697,6 +697,15 @@ static int atmel_ssc_resume(struct snd_soc_dai *cpu_dai)
 #define ATMEL_SSC_FORMATS (SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_S16_LE |\
 			  SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops atmel_ssc_dai_ops = {
+	.startup	= atmel_ssc_startup,
+	.shutdown	= atmel_ssc_shutdown,
+	.prepare	= atmel_ssc_prepare,
+	.hw_params	= atmel_ssc_hw_params,
+	.set_fmt	= atmel_ssc_set_dai_fmt,
+	.set_clkdiv	= atmel_ssc_set_dai_clkdiv,
+};
+
 struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 	{	.name = "atmel-ssc0",
 		.id = 0,
@@ -712,13 +721,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 			.channels_max = 2,
 			.rates = ATMEL_SSC_RATES,
 			.formats = ATMEL_SSC_FORMATS,},
-		.ops = {
-			.startup = atmel_ssc_startup,
-			.shutdown = atmel_ssc_shutdown,
-			.prepare = atmel_ssc_prepare,
-			.hw_params = atmel_ssc_hw_params,
-			.set_fmt = atmel_ssc_set_dai_fmt,
-			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+		.ops = &atmel_ssc_dai_ops,
 		.private_data = &ssc_info[0],
 	},
 #if NUM_SSC_DEVICES == 3
@@ -736,13 +739,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 			.channels_max = 2,
 			.rates = ATMEL_SSC_RATES,
 			.formats = ATMEL_SSC_FORMATS,},
-		.ops = {
-			.startup = atmel_ssc_startup,
-			.shutdown = atmel_ssc_shutdown,
-			.prepare = atmel_ssc_prepare,
-			.hw_params = atmel_ssc_hw_params,
-			.set_fmt = atmel_ssc_set_dai_fmt,
-			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+		.ops = &atmel_ssc_dai_ops,
 		.private_data = &ssc_info[1],
 	},
 	{	.name = "atmel-ssc2",
@@ -759,13 +756,7 @@ struct snd_soc_dai atmel_ssc_dai[NUM_SSC_DEVICES] = {
 			.channels_max = 2,
 			.rates = ATMEL_SSC_RATES,
 			.formats = ATMEL_SSC_FORMATS,},
-		.ops = {
-			.startup = atmel_ssc_startup,
-			.shutdown = atmel_ssc_shutdown,
-			.prepare = atmel_ssc_prepare,
-			.hw_params = atmel_ssc_hw_params,
-			.set_fmt = atmel_ssc_set_dai_fmt,
-			.set_clkdiv = atmel_ssc_set_dai_clkdiv,},
+		.ops = &atmel_ssc_dai_ops,
 		.private_data = &ssc_info[2],
 	},
 #endif
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
index 43dd8cee83c6..70657534e6b1 100644
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -164,38 +164,38 @@ static int playpaq_wm8510_hw_params(struct snd_pcm_substream *substream,
 	 */
 	switch (params_rate(params)) {
 	case 48000:
-		pll_out = 12288000;
-		mclk_div = WM8510_MCLKDIV_1;
+		pll_out = 24576000;
+		mclk_div = WM8510_MCLKDIV_2;
 		bclk = WM8510_BCLKDIV_8;
 		break;
 
 	case 44100:
-		pll_out = 11289600;
-		mclk_div = WM8510_MCLKDIV_1;
+		pll_out = 22579200;
+		mclk_div = WM8510_MCLKDIV_2;
 		bclk = WM8510_BCLKDIV_8;
 		break;
 
 	case 22050:
-		pll_out = 11289600;
-		mclk_div = WM8510_MCLKDIV_2;
+		pll_out = 22579200;
+		mclk_div = WM8510_MCLKDIV_4;
 		bclk = WM8510_BCLKDIV_8;
 		break;
 
 	case 16000:
-		pll_out = 12288000;
-		mclk_div = WM8510_MCLKDIV_3;
+		pll_out = 24576000;
+		mclk_div = WM8510_MCLKDIV_6;
 		bclk = WM8510_BCLKDIV_8;
 		break;
 
 	case 11025:
-		pll_out = 11289600;
-		mclk_div = WM8510_MCLKDIV_4;
+		pll_out = 22579200;
+		mclk_div = WM8510_MCLKDIV_8;
 		bclk = WM8510_BCLKDIV_8;
 		break;
 
 	case 8000:
-		pll_out = 12288000;
-		mclk_div = WM8510_MCLKDIV_6;
+		pll_out = 24576000;
+		mclk_div = WM8510_MCLKDIV_12;
 		bclk = WM8510_BCLKDIV_8;
 		break;
 
diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c
index 6ea04be911d0..173a239a541c 100644
--- a/sound/soc/atmel/sam9g20_wm8731.c
+++ b/sound/soc/atmel/sam9g20_wm8731.c
@@ -36,6 +36,7 @@
 #include <linux/timer.h>
 #include <linux/interrupt.h>
 #include <linux/platform_device.h>
+#include <linux/i2c.h>
 
 #include <linux/atmel-ssc.h>
 
@@ -45,6 +46,7 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 
+#include <asm/mach-types.h>
 #include <mach/hardware.h>
 #include <mach/gpio.h>
 
@@ -52,6 +54,9 @@
 #include "atmel-pcm.h"
 #include "atmel_ssc_dai.h"
 
+#define MCLK_RATE 12000000
+
+static struct clk *mclk;
 
 static int at91sam9g20ek_startup(struct snd_pcm_substream *substream)
 {
@@ -59,11 +64,12 @@ static int at91sam9g20ek_startup(struct snd_pcm_substream *substream)
 	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
 	int ret;
 
-	/* codec system clock is supplied by PCK0, set to 12MHz */
 	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
-		12000000, SND_SOC_CLOCK_IN);
-	if (ret < 0)
+		MCLK_RATE, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		clk_disable(mclk);
 		return ret;
+	}
 
 	return 0;
 }
@@ -189,6 +195,31 @@ static struct snd_soc_ops at91sam9g20ek_ops = {
 	.shutdown = at91sam9g20ek_shutdown,
 };
 
+static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card,
+					enum snd_soc_bias_level level)
+{
+	static int mclk_on;
+	int ret = 0;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+		if (!mclk_on)
+			ret = clk_enable(mclk);
+		if (ret == 0)
+			mclk_on = 1;
+		break;
+
+	case SND_SOC_BIAS_OFF:
+	case SND_SOC_BIAS_STANDBY:
+		if (mclk_on)
+			clk_disable(mclk);
+		mclk_on = 0;
+		break;
+	}
+
+	return ret;
+}
 
 static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = {
 	SND_SOC_DAPM_MIC("Int Mic", NULL),
@@ -243,21 +274,48 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = {
 };
 
 static struct snd_soc_card snd_soc_at91sam9g20ek = {
-	.name = "WM8731",
+	.name = "AT91SAMG20-EK",
 	.platform = &atmel_soc_platform,
 	.dai_link = &at91sam9g20ek_dai,
 	.num_links = 1,
+	.set_bias_level = at91sam9g20ek_set_bias_level,
 };
 
-static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = {
-	.i2c_bus = 0,
-	.i2c_address = 0x1b,
-};
+/*
+ * FIXME: This is a temporary bodge to avoid cross-tree merge issues.
+ * New drivers should register the wm8731 I2C device in the machine
+ * setup code (under arch/arm for ARM systems).
+ */
+static int wm8731_i2c_register(void)
+{
+	struct i2c_board_info info;
+	struct i2c_adapter *adapter;
+	struct i2c_client *client;
+
+	memset(&info, 0, sizeof(struct i2c_board_info));
+	info.addr = 0x1b;
+	strlcpy(info.type, "wm8731", I2C_NAME_SIZE);
+
+	adapter = i2c_get_adapter(0);
+	if (!adapter) {
+		printk(KERN_ERR "can't get i2c adapter 0\n");
+		return -ENODEV;
+	}
+
+	client = i2c_new_device(adapter, &info);
+	i2c_put_adapter(adapter);
+	if (!client) {
+		printk(KERN_ERR "can't add i2c device at 0x%x\n",
+			(unsigned int)info.addr);
+		return -ENODEV;
+	}
+
+	return 0;
+}
 
 static struct snd_soc_device at91sam9g20ek_snd_devdata = {
 	.card = &snd_soc_at91sam9g20ek,
 	.codec_dev = &soc_codec_dev_wm8731,
-	.codec_data = &at91sam9g20ek_wm8731_setup,
 };
 
 static struct platform_device *at91sam9g20ek_snd_device;
@@ -266,23 +324,56 @@ static int __init at91sam9g20ek_init(void)
 {
 	struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data;
 	struct ssc_device *ssc = NULL;
+	struct clk *pllb;
 	int ret;
 
+	if (!machine_is_at91sam9g20ek())
+		return -ENODEV;
+
+	/*
+	 * Codec MCLK is supplied by PCK0 - set it up.
+	 */
+	mclk = clk_get(NULL, "pck0");
+	if (IS_ERR(mclk)) {
+		printk(KERN_ERR "ASoC: Failed to get MCLK\n");
+		ret = PTR_ERR(mclk);
+		goto err;
+	}
+
+	pllb = clk_get(NULL, "pllb");
+	if (IS_ERR(mclk)) {
+		printk(KERN_ERR "ASoC: Failed to get PLLB\n");
+		ret = PTR_ERR(mclk);
+		goto err_mclk;
+	}
+	ret = clk_set_parent(mclk, pllb);
+	clk_put(pllb);
+	if (ret != 0) {
+		printk(KERN_ERR "ASoC: Failed to set MCLK parent\n");
+		goto err_mclk;
+	}
+
+	clk_set_rate(mclk, MCLK_RATE);
+
 	/*
 	 * Request SSC device
 	 */
 	ssc = ssc_request(0);
 	if (IS_ERR(ssc)) {
+		printk(KERN_ERR "ASoC: Failed to request SSC 0\n");
 		ret = PTR_ERR(ssc);
 		ssc = NULL;
 		goto err_ssc;
 	}
 	ssc_p->ssc = ssc;
 
+	ret = wm8731_i2c_register();
+	if (ret != 0)
+		goto err_ssc;
+
 	at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!at91sam9g20ek_snd_device) {
-		printk(KERN_DEBUG
-				"platform device allocation failed\n");
+		printk(KERN_ERR "ASoC: Platform device allocation failed\n");
 		ret = -ENOMEM;
 	}
 
@@ -292,14 +383,19 @@ static int __init at91sam9g20ek_init(void)
 
 	ret = platform_device_add(at91sam9g20ek_snd_device);
 	if (ret) {
-		printk(KERN_DEBUG
-				"platform device allocation failed\n");
+		printk(KERN_ERR "ASoC: Platform device allocation failed\n");
 		platform_device_put(at91sam9g20ek_snd_device);
 	}
 
 	return ret;
 
 err_ssc:
+	ssc_free(ssc);
+	ssc_p->ssc = NULL;
+err_mclk:
+	clk_put(mclk);
+	mclk = NULL;
+err:
 	return ret;
 }
 
@@ -317,6 +413,8 @@ static void __exit at91sam9g20ek_exit(void)
 
 	platform_device_unregister(at91sam9g20ek_snd_device);
 	at91sam9g20ek_snd_device = NULL;
+	clk_put(mclk);
+	mclk = NULL;
 }
 
 module_init(at91sam9g20ek_init);
diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index bc8d654576c0..30490a259148 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -305,7 +305,7 @@ static int au1xpsc_pcm_close(struct snd_pcm_substream *substream)
 	return 0;
 }
 
-struct snd_pcm_ops au1xpsc_pcm_ops = {
+static struct snd_pcm_ops au1xpsc_pcm_ops = {
 	.open		= au1xpsc_pcm_open,
 	.close		= au1xpsc_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c
index f0e30aec7f23..479d7bdf1865 100644
--- a/sound/soc/au1x/psc-ac97.c
+++ b/sound/soc/au1x/psc-ac97.c
@@ -342,6 +342,11 @@ static int au1xpsc_ac97_resume(struct snd_soc_dai *dai)
 	return 0;
 }
 
+static struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = {
+	.trigger	= au1xpsc_ac97_trigger,
+	.hw_params	= au1xpsc_ac97_hw_params,
+};
+
 struct snd_soc_dai au1xpsc_ac97_dai = {
 	.name			= "au1xpsc_ac97",
 	.ac97_control		= 1,
@@ -361,10 +366,7 @@ struct snd_soc_dai au1xpsc_ac97_dai = {
 		.channels_min	= 2,
 		.channels_max	= 2,
 	},
-	.ops = {
-		.trigger	= au1xpsc_ac97_trigger,
-		.hw_params	= au1xpsc_ac97_hw_params,
-	},
+	.ops = &au1xpsc_ac97_dai_ops,
 };
 EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai);
 
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index f916de4400ed..bb589327ee32 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -367,6 +367,12 @@ static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai)
 	return 0;
 }
 
+static struct snd_soc_dai_ops au1xpsc_i2s_dai_ops = {
+	.trigger	= au1xpsc_i2s_trigger,
+	.hw_params	= au1xpsc_i2s_hw_params,
+	.set_fmt	= au1xpsc_i2s_set_fmt,
+};
+
 struct snd_soc_dai au1xpsc_i2s_dai = {
 	.name			= "au1xpsc_i2s",
 	.probe			= au1xpsc_i2s_probe,
@@ -385,11 +391,7 @@ struct snd_soc_dai au1xpsc_i2s_dai = {
 		.channels_min	= 2,
 		.channels_max	= 8,	/* 2 without external help */
 	},
-	.ops = {
-		.trigger	= au1xpsc_i2s_trigger,
-		.hw_params	= au1xpsc_i2s_hw_params,
-		.set_fmt	= au1xpsc_i2s_set_fmt,
-	},
+	.ops = &au1xpsc_i2s_dai_ops,
 };
 EXPORT_SYMBOL(au1xpsc_i2s_dai);
 
diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c
index 8067cfafa3a7..8cfed1a5dcbe 100644
--- a/sound/soc/blackfin/bf5xx-ac97-pcm.c
+++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c
@@ -297,7 +297,7 @@ static	int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel,
 }
 #endif
 
-struct snd_pcm_ops bf5xx_pcm_ac97_ops = {
+static struct snd_pcm_ops bf5xx_pcm_ac97_ops = {
 	.open		= bf5xx_pcm_open,
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= bf5xx_pcm_hw_params,
diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c
index 3be2be60576d..8a935f2d1767 100644
--- a/sound/soc/blackfin/bf5xx-ac97.c
+++ b/sound/soc/blackfin/bf5xx-ac97.c
@@ -31,72 +31,46 @@
 #include "bf5xx-sport.h"
 #include "bf5xx-ac97.h"
 
-#if defined(CONFIG_BF54x)
-#define PIN_REQ_SPORT_0 {P_SPORT0_TFS, P_SPORT0_DTPRI, P_SPORT0_TSCLK, \
-		P_SPORT0_RFS, P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}
-
-#define PIN_REQ_SPORT_1 {P_SPORT1_TFS, P_SPORT1_DTPRI, P_SPORT1_TSCLK, \
-		P_SPORT1_RFS, P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}
-
-#define PIN_REQ_SPORT_2 {P_SPORT2_TFS, P_SPORT2_DTPRI, P_SPORT2_TSCLK, \
-		P_SPORT2_RFS, P_SPORT2_DRPRI, P_SPORT2_RSCLK, 0}
-
-#define PIN_REQ_SPORT_3 {P_SPORT3_TFS, P_SPORT3_DTPRI, P_SPORT3_TSCLK, \
-		P_SPORT3_RFS, P_SPORT3_DRPRI, P_SPORT3_RSCLK, 0}
-#else
-#define PIN_REQ_SPORT_0 {P_SPORT0_DTPRI, P_SPORT0_TSCLK, P_SPORT0_RFS, \
-		 P_SPORT0_DRPRI, P_SPORT0_RSCLK, 0}
-
-#define PIN_REQ_SPORT_1 {P_SPORT1_DTPRI, P_SPORT1_TSCLK, P_SPORT1_RFS, \
-		 P_SPORT1_DRPRI, P_SPORT1_RSCLK, 0}
-#endif
-
 static int *cmd_count;
 static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM;
 
+#define SPORT_REQ(x) \
+	[x] = {P_SPORT##x##_TFS, P_SPORT##x##_DTPRI, P_SPORT##x##_TSCLK, \
+	       P_SPORT##x##_RFS, P_SPORT##x##_DRPRI, P_SPORT##x##_RSCLK, 0}
 static u16 sport_req[][7] = {
-		PIN_REQ_SPORT_0,
-#ifdef PIN_REQ_SPORT_1
-		PIN_REQ_SPORT_1,
+#ifdef SPORT0_TCR1
+	SPORT_REQ(0),
+#endif
+#ifdef SPORT1_TCR1
+	SPORT_REQ(1),
 #endif
-#ifdef PIN_REQ_SPORT_2
-		PIN_REQ_SPORT_2,
+#ifdef SPORT2_TCR1
+	SPORT_REQ(2),
 #endif
-#ifdef PIN_REQ_SPORT_3
-		PIN_REQ_SPORT_3,
+#ifdef SPORT3_TCR1
+	SPORT_REQ(3),
 #endif
-	};
+};
 
+#define SPORT_PARAMS(x) \
+	[x] = { \
+		.dma_rx_chan = CH_SPORT##x##_RX, \
+		.dma_tx_chan = CH_SPORT##x##_TX, \
+		.err_irq     = IRQ_SPORT##x##_ERROR, \
+		.regs        = (struct sport_register *)SPORT##x##_TCR1, \
+	}
 static struct sport_param sport_params[4] = {
-	{
-		.dma_rx_chan	= CH_SPORT0_RX,
-		.dma_tx_chan	= CH_SPORT0_TX,
-		.err_irq	= IRQ_SPORT0_ERROR,
-		.regs		= (struct sport_register *)SPORT0_TCR1,
-	},
-#ifdef PIN_REQ_SPORT_1
-	{
-		.dma_rx_chan	= CH_SPORT1_RX,
-		.dma_tx_chan	= CH_SPORT1_TX,
-		.err_irq	= IRQ_SPORT1_ERROR,
-		.regs		= (struct sport_register *)SPORT1_TCR1,
-	},
+#ifdef SPORT0_TCR1
+	SPORT_PARAMS(0),
 #endif
-#ifdef PIN_REQ_SPORT_2
-	{
-		.dma_rx_chan	= CH_SPORT2_RX,
-		.dma_tx_chan	= CH_SPORT2_TX,
-		.err_irq	= IRQ_SPORT2_ERROR,
-		.regs		= (struct sport_register *)SPORT2_TCR1,
-	},
+#ifdef SPORT1_TCR1
+	SPORT_PARAMS(1),
 #endif
-#ifdef PIN_REQ_SPORT_3
-	{
-		.dma_rx_chan	= CH_SPORT3_RX,
-		.dma_tx_chan	= CH_SPORT3_TX,
-		.err_irq	= IRQ_SPORT3_ERROR,
-		.regs		= (struct sport_register *)SPORT3_TCR1,
-	}
+#ifdef SPORT2_TCR1
+	SPORT_PARAMS(2),
+#endif
+#ifdef SPORT3_TCR1
+	SPORT_PARAMS(3),
 #endif
 };
 
@@ -332,11 +306,11 @@ static int bf5xx_ac97_probe(struct platform_device *pdev,
 	if (cmd_count == NULL)
 		return -ENOMEM;
 
-	if (peripheral_request_list(&sport_req[sport_num][0], "soc-audio")) {
+	if (peripheral_request_list(sport_req[sport_num], "soc-audio")) {
 		pr_err("Requesting Peripherals failed\n");
 		ret =  -EFAULT;
 		goto peripheral_err;
-		}
+	}
 
 #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
 	/* Request PB3 as reset pin */
@@ -383,9 +357,9 @@ sport_config_err:
 sport_err:
 #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
 	gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
-#endif
 gpio_err:
-	peripheral_free_list(&sport_req[sport_num][0]);
+#endif
+	peripheral_free_list(sport_req[sport_num]);
 peripheral_err:
 	free_page((unsigned long)cmd_count);
 	cmd_count = NULL;
@@ -398,7 +372,7 @@ static void bf5xx_ac97_remove(struct platform_device *pdev,
 {
 	free_page((unsigned long)cmd_count);
 	cmd_count = NULL;
-	peripheral_free_list(&sport_req[sport_num][0]);
+	peripheral_free_list(sport_req[sport_num]);
 #ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET
 	gpio_free(CONFIG_SND_BF5XX_RESET_GPIO_NUM);
 #endif
diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c
index 7f2a5e199075..edfbdc024e66 100644
--- a/sound/soc/blackfin/bf5xx-ad73311.c
+++ b/sound/soc/blackfin/bf5xx-ad73311.c
@@ -114,7 +114,7 @@ static int snd_ad73311_configure(void)
 	SSYNC();
 
 	/* When TUVF is set, the data is already send out */
-	while (!(status & TUVF) && count++ < 10000) {
+	while (!(status & TUVF) && ++count < 10000) {
 		udelay(1);
 		status = bfin_read_SPORT_STAT();
 		SSYNC();
@@ -123,7 +123,7 @@ static int snd_ad73311_configure(void)
 	SSYNC();
 	local_irq_enable();
 
-	if (count == 10000) {
+	if (count >= 10000) {
 		printk(KERN_ERR "ad73311: failed to configure codec\n");
 		return -1;
 	}
diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c
index 53d290b3ea47..1318c4f627b7 100644
--- a/sound/soc/blackfin/bf5xx-i2s-pcm.c
+++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c
@@ -184,7 +184,7 @@ static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream,
 	return 0 ;
 }
 
-struct snd_pcm_ops bf5xx_pcm_i2s_ops = {
+static struct snd_pcm_ops bf5xx_pcm_i2s_ops = {
 	.open		= bf5xx_pcm_open,
 	.ioctl		= snd_pcm_lib_ioctl,
 	.hw_params	= bf5xx_pcm_hw_params,
diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c
index d1d95d2393fe..964824419678 100644
--- a/sound/soc/blackfin/bf5xx-i2s.c
+++ b/sound/soc/blackfin/bf5xx-i2s.c
@@ -287,6 +287,13 @@ static int bf5xx_i2s_resume(struct platform_device *pdev,
 #define BF5XX_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\
 	SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops bf5xx_i2s_dai_ops = {
+	.startup	= bf5xx_i2s_startup,
+	.shutdown	= bf5xx_i2s_shutdown,
+	.hw_params	= bf5xx_i2s_hw_params,
+	.set_fmt	= bf5xx_i2s_set_dai_fmt,
+};
+
 struct snd_soc_dai bf5xx_i2s_dai = {
 	.name = "bf5xx-i2s",
 	.id = 0,
@@ -304,12 +311,7 @@ struct snd_soc_dai bf5xx_i2s_dai = {
 		.channels_max = 2,
 		.rates = BF5XX_I2S_RATES,
 		.formats = BF5XX_I2S_FORMATS,},
-	.ops = {
-		.startup   = bf5xx_i2s_startup,
-		.shutdown  = bf5xx_i2s_shutdown,
-		.hw_params = bf5xx_i2s_hw_params,
-		.set_fmt = bf5xx_i2s_set_dai_fmt,
-	},
+	.ops = &bf5xx_i2s_dai_ops,
 };
 EXPORT_SYMBOL_GPL(bf5xx_i2s_dai);
 
diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c
index 3b99e484d555..b7953c8cf838 100644
--- a/sound/soc/blackfin/bf5xx-sport.c
+++ b/sound/soc/blackfin/bf5xx-sport.c
@@ -133,7 +133,7 @@ static void setup_desc(struct dmasg *desc, void *buf, int fragcount,
 	int i;
 
 	for (i = 0; i < fragcount; ++i) {
-		desc[i].next_desc_addr  = (unsigned long)&(desc[i + 1]);
+		desc[i].next_desc_addr  = &(desc[i + 1]);
 		desc[i].start_addr = (unsigned long)buf + i*fragsize;
 		desc[i].cfg = cfg;
 		desc[i].x_count = x_count;
@@ -143,12 +143,12 @@ static void setup_desc(struct dmasg *desc, void *buf, int fragcount,
 	}
 
 	/* make circular */
-	desc[fragcount-1].next_desc_addr = (unsigned long)desc;
+	desc[fragcount-1].next_desc_addr = desc;
 
-	pr_debug("setup desc: desc0=%p, next0=%lx, desc1=%p,"
-		"next1=%lx\nx_count=%x,y_count=%x,addr=0x%lx,cfs=0x%x\n",
-		&(desc[0]), desc[0].next_desc_addr,
-		&(desc[1]), desc[1].next_desc_addr,
+	pr_debug("setup desc: desc0=%p, next0=%p, desc1=%p,"
+		"next1=%p\nx_count=%x,y_count=%x,addr=0x%lx,cfs=0x%x\n",
+		desc, desc[0].next_desc_addr,
+		desc+1, desc[1].next_desc_addr,
 		desc[0].x_count, desc[0].y_count,
 		desc[0].start_addr, desc[0].cfg);
 }
@@ -184,22 +184,20 @@ static inline int sport_hook_rx_dummy(struct sport_device *sport)
 	BUG_ON(sport->curr_rx_desc == sport->dummy_rx_desc);
 
 	/* Maybe the dummy buffer descriptor ring is damaged */
-	sport->dummy_rx_desc->next_desc_addr = \
-			(unsigned long)(sport->dummy_rx_desc+1);
+	sport->dummy_rx_desc->next_desc_addr = sport->dummy_rx_desc + 1;
 
 	local_irq_save(flags);
-	desc = (struct dmasg *)get_dma_next_desc_ptr(sport->dma_rx_chan);
+	desc = get_dma_next_desc_ptr(sport->dma_rx_chan);
 	/* Copy the descriptor which will be damaged to backup */
 	temp_desc = *desc;
 	desc->x_count = 0xa;
 	desc->y_count = 0;
-	desc->next_desc_addr = (unsigned long)(sport->dummy_rx_desc);
+	desc->next_desc_addr = sport->dummy_rx_desc;
 	local_irq_restore(flags);
 	/* Waiting for dummy buffer descriptor is already hooked*/
 	while ((get_dma_curr_desc_ptr(sport->dma_rx_chan) -
-			sizeof(struct dmasg)) !=
-			(unsigned long)sport->dummy_rx_desc)
-		;
+			sizeof(struct dmasg)) != sport->dummy_rx_desc)
+		continue;
 	sport->curr_rx_desc = sport->dummy_rx_desc;
 	/* Restore the damaged descriptor */
 	*desc = temp_desc;
@@ -210,14 +208,12 @@ static inline int sport_hook_rx_dummy(struct sport_device *sport)
 static inline int sport_rx_dma_start(struct sport_device *sport, int dummy)
 {
 	if (dummy) {
-		sport->dummy_rx_desc->next_desc_addr = \
-				(unsigned long) sport->dummy_rx_desc;
+		sport->dummy_rx_desc->next_desc_addr = sport->dummy_rx_desc;
 		sport->curr_rx_desc = sport->dummy_rx_desc;
 	} else
 		sport->curr_rx_desc = sport->dma_rx_desc;
 
-	set_dma_next_desc_addr(sport->dma_rx_chan, \
-			(unsigned long)(sport->curr_rx_desc));
+	set_dma_next_desc_addr(sport->dma_rx_chan, sport->curr_rx_desc);
 	set_dma_x_count(sport->dma_rx_chan, 0);
 	set_dma_x_modify(sport->dma_rx_chan, 0);
 	set_dma_config(sport->dma_rx_chan, (DMAFLOW_LARGE | NDSIZE_9 | \
@@ -231,14 +227,12 @@ static inline int sport_rx_dma_start(struct sport_device *sport, int dummy)
 static inline int sport_tx_dma_start(struct sport_device *sport, int dummy)
 {
 	if (dummy) {
-		sport->dummy_tx_desc->next_desc_addr = \
-				(unsigned long) sport->dummy_tx_desc;
+		sport->dummy_tx_desc->next_desc_addr = sport->dummy_tx_desc;
 		sport->curr_tx_desc = sport->dummy_tx_desc;
 	} else
 		sport->curr_tx_desc = sport->dma_tx_desc;
 
-	set_dma_next_desc_addr(sport->dma_tx_chan, \
-			(unsigned long)(sport->curr_tx_desc));
+	set_dma_next_desc_addr(sport->dma_tx_chan, sport->curr_tx_desc);
 	set_dma_x_count(sport->dma_tx_chan, 0);
 	set_dma_x_modify(sport->dma_tx_chan, 0);
 	set_dma_config(sport->dma_tx_chan,
@@ -261,11 +255,9 @@ int sport_rx_start(struct sport_device *sport)
 		BUG_ON(sport->curr_rx_desc != sport->dummy_rx_desc);
 		local_irq_save(flags);
 		while ((get_dma_curr_desc_ptr(sport->dma_rx_chan) -
-			sizeof(struct dmasg)) !=
-			(unsigned long)sport->dummy_rx_desc)
-			;
-		sport->dummy_rx_desc->next_desc_addr =
-				(unsigned long)(sport->dma_rx_desc);
+			sizeof(struct dmasg)) != sport->dummy_rx_desc)
+			continue;
+		sport->dummy_rx_desc->next_desc_addr = sport->dma_rx_desc;
 		local_irq_restore(flags);
 		sport->curr_rx_desc = sport->dma_rx_desc;
 	} else {
@@ -310,23 +302,21 @@ static inline int sport_hook_tx_dummy(struct sport_device *sport)
 	BUG_ON(sport->dummy_tx_desc == NULL);
 	BUG_ON(sport->curr_tx_desc == sport->dummy_tx_desc);
 
-	sport->dummy_tx_desc->next_desc_addr = \
-			(unsigned long)(sport->dummy_tx_desc+1);
+	sport->dummy_tx_desc->next_desc_addr = sport->dummy_tx_desc + 1;
 
 	/* Shorten the time on last normal descriptor */
 	local_irq_save(flags);
-	desc = (struct dmasg *)get_dma_next_desc_ptr(sport->dma_tx_chan);
+	desc = get_dma_next_desc_ptr(sport->dma_tx_chan);
 	/* Store the descriptor which will be damaged */
 	temp_desc = *desc;
 	desc->x_count = 0xa;
 	desc->y_count = 0;
-	desc->next_desc_addr = (unsigned long)(sport->dummy_tx_desc);
+	desc->next_desc_addr = sport->dummy_tx_desc;
 	local_irq_restore(flags);
 	/* Waiting for dummy buffer descriptor is already hooked*/
 	while ((get_dma_curr_desc_ptr(sport->dma_tx_chan) - \
-			sizeof(struct dmasg)) != \
-			(unsigned long)sport->dummy_tx_desc)
-		;
+			sizeof(struct dmasg)) != sport->dummy_tx_desc)
+		continue;
 	sport->curr_tx_desc = sport->dummy_tx_desc;
 	/* Restore the damaged descriptor */
 	*desc = temp_desc;
@@ -347,11 +337,9 @@ int sport_tx_start(struct sport_device *sport)
 		/* Hook the normal buffer descriptor */
 		local_irq_save(flags);
 		while ((get_dma_curr_desc_ptr(sport->dma_tx_chan) -
-			sizeof(struct dmasg)) !=
-			(unsigned long)sport->dummy_tx_desc)
-			;
-		sport->dummy_tx_desc->next_desc_addr =
-				(unsigned long)(sport->dma_tx_desc);
+			sizeof(struct dmasg)) != sport->dummy_tx_desc)
+			continue;
+		sport->dummy_tx_desc->next_desc_addr = sport->dma_tx_desc;
 		local_irq_restore(flags);
 		sport->curr_tx_desc = sport->dma_tx_desc;
 	} else {
@@ -536,19 +524,17 @@ static int sport_config_rx_dummy(struct sport_device *sport)
 	unsigned config;
 
 	pr_debug("%s entered\n", __func__);
-#if L1_DATA_A_LENGTH != 0
-	desc = (struct dmasg *) l1_data_sram_alloc(2 * sizeof(*desc));
-#else
-	{
+	if (L1_DATA_A_LENGTH)
+		desc = l1_data_sram_zalloc(2 * sizeof(*desc));
+	else {
 		dma_addr_t addr;
 		desc = dma_alloc_coherent(NULL, 2 * sizeof(*desc), &addr, 0);
+		memset(desc, 0, 2 * sizeof(*desc));
 	}
-#endif
 	if (desc == NULL) {
 		pr_err("Failed to allocate memory for dummy rx desc\n");
 		return -ENOMEM;
 	}
-	memset(desc, 0, 2 * sizeof(*desc));
 	sport->dummy_rx_desc = desc;
 	desc->start_addr = (unsigned long)sport->dummy_buf;
 	config = DMAFLOW_LARGE | NDSIZE_9 | compute_wdsize(sport->wdsize)
@@ -559,8 +545,8 @@ static int sport_config_rx_dummy(struct sport_device *sport)
 	desc->y_count = 0;
 	desc->y_modify = 0;
 	memcpy(desc+1, desc, sizeof(*desc));
-	desc->next_desc_addr = (unsigned long)(desc+1);
-	desc[1].next_desc_addr = (unsigned long)desc;
+	desc->next_desc_addr = desc + 1;
+	desc[1].next_desc_addr = desc;
 	return 0;
 }
 
@@ -571,19 +557,17 @@ static int sport_config_tx_dummy(struct sport_device *sport)
 
 	pr_debug("%s entered\n", __func__);
 
-#if L1_DATA_A_LENGTH != 0
-	desc = (struct dmasg *) l1_data_sram_alloc(2 * sizeof(*desc));
-#else
-	{
+	if (L1_DATA_A_LENGTH)
+		desc = l1_data_sram_zalloc(2 * sizeof(*desc));
+	else {
 		dma_addr_t addr;
 		desc = dma_alloc_coherent(NULL, 2 * sizeof(*desc), &addr, 0);
+		memset(desc, 0, 2 * sizeof(*desc));
 	}
-#endif
 	if (!desc) {
 		pr_err("Failed to allocate memory for dummy tx desc\n");
 		return -ENOMEM;
 	}
-	memset(desc, 0, 2 * sizeof(*desc));
 	sport->dummy_tx_desc = desc;
 	desc->start_addr = (unsigned long)sport->dummy_buf + \
 		sport->dummy_count;
@@ -595,8 +579,8 @@ static int sport_config_tx_dummy(struct sport_device *sport)
 	desc->y_count = 0;
 	desc->y_modify = 0;
 	memcpy(desc+1, desc, sizeof(*desc));
-	desc->next_desc_addr = (unsigned long)(desc+1);
-	desc[1].next_desc_addr = (unsigned long)desc;
+	desc->next_desc_addr = desc + 1;
+	desc[1].next_desc_addr = desc;
 	return 0;
 }
 
@@ -872,17 +856,15 @@ struct sport_device *sport_init(struct sport_param *param, unsigned wdsize,
 	sport->wdsize = wdsize;
 	sport->dummy_count = dummy_count;
 
-#if L1_DATA_A_LENGTH != 0
-	sport->dummy_buf = l1_data_sram_alloc(dummy_count * 2);
-#else
-	sport->dummy_buf = kmalloc(dummy_count * 2, GFP_KERNEL);
-#endif
+	if (L1_DATA_A_LENGTH)
+		sport->dummy_buf = l1_data_sram_zalloc(dummy_count * 2);
+	else
+		sport->dummy_buf = kzalloc(dummy_count * 2, GFP_KERNEL);
 	if (sport->dummy_buf == NULL) {
 		pr_err("Failed to allocate dummy buffer\n");
 		goto __error;
 	}
 
-	memset(sport->dummy_buf, 0, dummy_count * 2);
 	ret = sport_config_rx_dummy(sport);
 	if (ret) {
 		pr_err("Failed to config rx dummy ring\n");
@@ -939,6 +921,7 @@ void sport_done(struct sport_device *sport)
 		sport = NULL;
 }
 EXPORT_SYMBOL(sport_done);
+
 /*
 * It is only used to send several bytes when dma is not enabled
  * sport controller is configured but not enabled.
@@ -1029,4 +1012,3 @@ EXPORT_SYMBOL(sport_send_and_recv);
 MODULE_AUTHOR("Roy Huang");
 MODULE_DESCRIPTION("SPORT driver for ADI Blackfin");
 MODULE_LICENSE("GPL");
-
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index d0e0d691ae51..b6c7f7a01cb0 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -10,9 +10,11 @@ config SND_SOC_I2C_AND_SPI
 
 config SND_SOC_ALL_CODECS
 	tristate "Build all ASoC CODEC drivers"
+	select SND_SOC_L3
 	select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS
 	select SND_SOC_AD1980 if SND_SOC_AC97_BUS
 	select SND_SOC_AD73311 if I2C
+	select SND_SOC_AK4104 if SPI_MASTER
 	select SND_SOC_AK4535 if I2C
 	select SND_SOC_CS4270 if I2C
 	select SND_SOC_PCM3008
@@ -24,6 +26,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_UDA134X
 	select SND_SOC_UDA1380 if I2C
 	select SND_SOC_WM8350 if MFD_WM8350
+	select SND_SOC_WM8400 if MFD_WM8400
 	select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8580 if I2C
 	select SND_SOC_WM8728 if SND_SOC_I2C_AND_SPI
@@ -34,6 +37,7 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM8903 if I2C
 	select SND_SOC_WM8971 if I2C
 	select SND_SOC_WM8990 if I2C
+	select SND_SOC_WM9705 if SND_SOC_AC97_BUS
 	select SND_SOC_WM9712 if SND_SOC_AC97_BUS
 	select SND_SOC_WM9713 if SND_SOC_AC97_BUS
         help
@@ -58,6 +62,9 @@ config SND_SOC_AD1980
 config SND_SOC_AD73311
 	tristate
 
+config SND_SOC_AK4104
+	tristate
+
 config SND_SOC_AK4535
 	tristate
 
@@ -65,12 +72,6 @@ config SND_SOC_AK4535
 config SND_SOC_CS4270
 	tristate
 
-# Cirrus Logic CS4270 Codec Hardware Mute Support
-# Select if you have external muting circuitry attached to your CS4270.
-config SND_SOC_CS4270_HWMUTE
-	bool
-	depends on SND_SOC_CS4270
-
 # Cirrus Logic CS4270 Codec VD = 3.3V Errata
 # Select if you are affected by the errata where the part will not function
 # if MCLK divide-by-1.5 is selected and VD is set to 3.3V.  The driver will
@@ -90,7 +91,6 @@ config SND_SOC_SSM2602
 
 config SND_SOC_TLV320AIC23
 	tristate
-	depends on I2C
 
 config SND_SOC_TLV320AIC26
 	tristate "TI TLV320AIC26 Codec support" if SND_SOC_OF_SIMPLE
@@ -98,15 +98,12 @@ config SND_SOC_TLV320AIC26
 
 config SND_SOC_TLV320AIC3X
 	tristate
-	depends on I2C
 
 config SND_SOC_TWL4030
 	tristate
-	depends on TWL4030_CORE
 
 config SND_SOC_UDA134X
        tristate
-       select SND_SOC_L3
 
 config SND_SOC_UDA1380
         tristate
@@ -114,6 +111,9 @@ config SND_SOC_UDA1380
 config SND_SOC_WM8350
 	tristate
 
+config SND_SOC_WM8400
+	tristate
+
 config SND_SOC_WM8510
 	tristate
 
@@ -144,6 +144,9 @@ config SND_SOC_WM8971
 config SND_SOC_WM8990
 	tristate
 
+config SND_SOC_WM9705
+	tristate
+
 config SND_SOC_WM9712
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index c4ddc9aa2bbd..030d2454725f 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -1,6 +1,7 @@
 snd-soc-ac97-objs := ac97.o
 snd-soc-ad1980-objs := ad1980.o
 snd-soc-ad73311-objs := ad73311.o
+snd-soc-ak4104-objs := ak4104.o
 snd-soc-ak4535-objs := ak4535.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-l3-objs := l3.o
@@ -13,6 +14,7 @@ snd-soc-twl4030-objs := twl4030.o
 snd-soc-uda134x-objs := uda134x.o
 snd-soc-uda1380-objs := uda1380.o
 snd-soc-wm8350-objs := wm8350.o
+snd-soc-wm8400-objs := wm8400.o
 snd-soc-wm8510-objs := wm8510.o
 snd-soc-wm8580-objs := wm8580.o
 snd-soc-wm8728-objs := wm8728.o
@@ -23,12 +25,14 @@ snd-soc-wm8900-objs := wm8900.o
 snd-soc-wm8903-objs := wm8903.o
 snd-soc-wm8971-objs := wm8971.o
 snd-soc-wm8990-objs := wm8990.o
+snd-soc-wm9705-objs := wm9705.o
 snd-soc-wm9712-objs := wm9712.o
 snd-soc-wm9713-objs := wm9713.o
 
 obj-$(CONFIG_SND_SOC_AC97_CODEC)	+= snd-soc-ac97.o
 obj-$(CONFIG_SND_SOC_AD1980)	+= snd-soc-ad1980.o
 obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o
+obj-$(CONFIG_SND_SOC_AK4104)	+= snd-soc-ak4104.o
 obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
@@ -41,6 +45,7 @@ obj-$(CONFIG_SND_SOC_TWL4030)	+= snd-soc-twl4030.o
 obj-$(CONFIG_SND_SOC_UDA134X)	+= snd-soc-uda134x.o
 obj-$(CONFIG_SND_SOC_UDA1380)	+= snd-soc-uda1380.o
 obj-$(CONFIG_SND_SOC_WM8350)	+= snd-soc-wm8350.o
+obj-$(CONFIG_SND_SOC_WM8400)	+= snd-soc-wm8400.o
 obj-$(CONFIG_SND_SOC_WM8510)	+= snd-soc-wm8510.o
 obj-$(CONFIG_SND_SOC_WM8580)	+= snd-soc-wm8580.o
 obj-$(CONFIG_SND_SOC_WM8728)	+= snd-soc-wm8728.o
@@ -51,5 +56,7 @@ obj-$(CONFIG_SND_SOC_WM8900)	+= snd-soc-wm8900.o
 obj-$(CONFIG_SND_SOC_WM8903)	+= snd-soc-wm8903.o
 obj-$(CONFIG_SND_SOC_WM8971)	+= snd-soc-wm8971.o
 obj-$(CONFIG_SND_SOC_WM8990)	+= snd-soc-wm8990.o
+obj-$(CONFIG_SND_SOC_WM8991)	+= snd-soc-wm8991.o
+obj-$(CONFIG_SND_SOC_WM9705)	+= snd-soc-wm9705.o
 obj-$(CONFIG_SND_SOC_WM9712)	+= snd-soc-wm9712.o
 obj-$(CONFIG_SND_SOC_WM9713)	+= snd-soc-wm9713.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index fb53e6511af2..b0d4af145b87 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -30,7 +30,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	int reg = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
 		  AC97_PCM_FRONT_DAC_RATE : AC97_PCM_LR_ADC_RATE;
@@ -41,6 +41,10 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
 		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
 		SNDRV_PCM_RATE_48000)
 
+static struct snd_soc_dai_ops ac97_dai_ops = {
+	.prepare	= ac97_prepare,
+};
+
 struct snd_soc_dai ac97_dai = {
 	.name = "AC97 HiFi",
 	.ac97_control = 1,
@@ -56,8 +60,7 @@ struct snd_soc_dai ac97_dai = {
 		.channels_max = 2,
 		.rates = STD_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_prepare,},
+	.ops = &ac97_dai_ops,
 };
 EXPORT_SYMBOL_GPL(ac97_dai);
 
@@ -84,10 +87,10 @@ static int ac97_soc_probe(struct platform_device *pdev)
 
 	printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION);
 
-	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (!socdev->codec)
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (!socdev->card->codec)
 		return -ENOMEM;
-	codec = socdev->codec;
+	codec = socdev->card->codec;
 	mutex_init(&codec->mutex);
 
 	codec->name = "AC97";
@@ -123,23 +126,21 @@ bus_err:
 	snd_soc_free_pcms(socdev);
 
 err:
-	kfree(socdev->codec->reg_cache);
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
 	return ret;
 }
 
 static int ac97_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (!codec)
 		return 0;
 
 	snd_soc_free_pcms(socdev);
-	kfree(socdev->codec->reg_cache);
-	kfree(socdev->codec);
+	kfree(socdev->card->codec);
 
 	return 0;
 }
@@ -149,7 +150,7 @@ static int ac97_soc_suspend(struct platform_device *pdev, pm_message_t msg)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 
-	snd_ac97_suspend(socdev->codec->ac97);
+	snd_ac97_suspend(socdev->card->codec->ac97);
 
 	return 0;
 }
@@ -158,7 +159,7 @@ static int ac97_soc_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 
-	snd_ac97_resume(socdev->codec->ac97);
+	snd_ac97_resume(socdev->card->codec->ac97);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index 73fdbb4d4a3d..ddb3b08ac23c 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -93,20 +93,6 @@ SOC_ENUM("Capture Source", ad1980_cap_src),
 SOC_SINGLE("Mic Boost Switch", AC97_MIC, 6, 1, 0),
 };
 
-/* add non dapm controls */
-static int ad1980_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(ad1980_snd_ac97_controls); i++) {
-		err = snd_ctl_add(codec->card, snd_soc_cnew(
-				&ad1980_snd_ac97_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 static unsigned int ac97_read(struct snd_soc_codec *codec,
 	unsigned int reg)
 {
@@ -123,7 +109,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec,
 	default:
 		reg = reg >> 1;
 
-		if (reg >= (ARRAY_SIZE(ad1980_reg)))
+		if (reg >= ARRAY_SIZE(ad1980_reg))
 			return -EINVAL;
 
 		return cache[reg];
@@ -137,7 +123,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
 
 	soc_ac97_ops.write(codec->ac97, reg, val);
 	reg = reg >> 1;
-	if (reg < (ARRAY_SIZE(ad1980_reg)))
+	if (reg < ARRAY_SIZE(ad1980_reg))
 		cache[reg] = val;
 
 	return 0;
@@ -200,10 +186,10 @@ static int ad1980_soc_probe(struct platform_device *pdev)
 
 	printk(KERN_INFO "AD1980 SoC Audio Codec\n");
 
-	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (socdev->codec == NULL)
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (socdev->card->codec == NULL)
 		return -ENOMEM;
-	codec = socdev->codec;
+	codec = socdev->card->codec;
 	mutex_init(&codec->mutex);
 
 	codec->reg_cache =
@@ -269,7 +255,8 @@ static int ad1980_soc_probe(struct platform_device *pdev)
 	ext_status = ac97_read(codec, AC97_EXTENDED_STATUS);
 	ac97_write(codec, AC97_EXTENDED_STATUS, ext_status&~0x3800);
 
-	ad1980_add_controls(codec);
+	snd_soc_add_controls(codec, ad1980_snd_ac97_controls,
+				ARRAY_SIZE(ad1980_snd_ac97_controls));
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
 		printk(KERN_ERR "ad1980: failed to register card\n");
@@ -288,15 +275,15 @@ codec_err:
 	kfree(codec->reg_cache);
 
 cache_err:
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
 	return ret;
 }
 
 static int ad1980_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec == NULL)
 		return 0;
diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c
index b09289a1e55a..e61dac5e7b8f 100644
--- a/sound/soc/codecs/ad73311.c
+++ b/sound/soc/codecs/ad73311.c
@@ -53,7 +53,7 @@ static int ad73311_soc_probe(struct platform_device *pdev)
 	codec->owner = THIS_MODULE;
 	codec->dai = &ad73311_dai;
 	codec->num_dai = 1;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
@@ -75,15 +75,15 @@ static int ad73311_soc_probe(struct platform_device *pdev)
 register_err:
 	snd_soc_free_pcms(socdev);
 pcm_err:
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
 	return ret;
 }
 
 static int ad73311_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec == NULL)
 		return 0;
diff --git a/sound/soc/codecs/ad73311.h b/sound/soc/codecs/ad73311.h
index 507ce0c30edf..569573d2d4d7 100644
--- a/sound/soc/codecs/ad73311.h
+++ b/sound/soc/codecs/ad73311.h
@@ -70,7 +70,7 @@
 #define REGD_IGS(x)		(x & 0x7)
 #define REGD_RMOD		(1 << 3)
 #define REGD_OGS(x)		((x & 0x7) << 4)
-#define REGD_MUTE		(x << 7)
+#define REGD_MUTE		(1 << 7)
 
 /* Control register E */
 #define CTRL_REG_E	(4 << 8)
diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c
new file mode 100644
index 000000000000..4d47bc4f7428
--- /dev/null
+++ b/sound/soc/codecs/ak4104.c
@@ -0,0 +1,365 @@
+/*
+ * AK4104 ALSA SoC (ASoC) driver
+ *
+ * Copyright (c) 2009 Daniel Mack <daniel@caiaq.de>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/initval.h>
+#include <linux/spi/spi.h>
+#include <sound/asoundef.h>
+
+#include "ak4104.h"
+
+/* AK4104 registers addresses */
+#define AK4104_REG_CONTROL1		0x00
+#define AK4104_REG_RESERVED		0x01
+#define AK4104_REG_CONTROL2		0x02
+#define AK4104_REG_TX			0x03
+#define AK4104_REG_CHN_STATUS(x)	((x) + 0x04)
+#define AK4104_NUM_REGS			10
+
+#define AK4104_REG_MASK			0x1f
+#define AK4104_READ			0xc0
+#define AK4104_WRITE			0xe0
+#define AK4104_RESERVED_VAL		0x5b
+
+/* Bit masks for AK4104 registers */
+#define AK4104_CONTROL1_RSTN		(1 << 0)
+#define AK4104_CONTROL1_PW		(1 << 1)
+#define AK4104_CONTROL1_DIF0		(1 << 2)
+#define AK4104_CONTROL1_DIF1		(1 << 3)
+
+#define AK4104_CONTROL2_SEL0		(1 << 0)
+#define AK4104_CONTROL2_SEL1		(1 << 1)
+#define AK4104_CONTROL2_MODE		(1 << 2)
+
+#define AK4104_TX_TXE			(1 << 0)
+#define AK4104_TX_V			(1 << 1)
+
+#define DRV_NAME "ak4104"
+
+struct ak4104_private {
+	struct snd_soc_codec codec;
+	u8 reg_cache[AK4104_NUM_REGS];
+};
+
+static int ak4104_fill_cache(struct snd_soc_codec *codec)
+{
+	int i;
+	u8 *reg_cache = codec->reg_cache;
+	struct spi_device *spi = codec->control_data;
+
+	for (i = 0; i < codec->reg_cache_size; i++) {
+		int ret = spi_w8r8(spi, i | AK4104_READ);
+		if (ret < 0) {
+			dev_err(&spi->dev, "SPI write failure\n");
+			return ret;
+		}
+
+		reg_cache[i] = ret;
+	}
+
+	return 0;
+}
+
+static unsigned int ak4104_read_reg_cache(struct snd_soc_codec *codec,
+					  unsigned int reg)
+{
+	u8 *reg_cache = codec->reg_cache;
+
+	if (reg >= codec->reg_cache_size)
+		return -EINVAL;
+
+	return reg_cache[reg];
+}
+
+static int ak4104_spi_write(struct snd_soc_codec *codec, unsigned int reg,
+			    unsigned int value)
+{
+	u8 *cache = codec->reg_cache;
+	struct spi_device *spi = codec->control_data;
+
+	if (reg >= codec->reg_cache_size)
+		return -EINVAL;
+
+	reg &= AK4104_REG_MASK;
+	reg |= AK4104_WRITE;
+
+	/* only write to the hardware if value has changed */
+	if (cache[reg] != value) {
+		u8 tmp[2] = { reg, value };
+		if (spi_write(spi, tmp, sizeof(tmp))) {
+			dev_err(&spi->dev, "SPI write failed\n");
+			return -EIO;
+		}
+
+		cache[reg] = value;
+	}
+
+	return 0;
+}
+
+static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai,
+			      unsigned int format)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int val = 0;
+
+	val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1);
+	if (val < 0)
+		return val;
+
+	val &= ~(AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1);
+
+	/* set DAI format */
+	switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_RIGHT_J:
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		val |= AK4104_CONTROL1_DIF0;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		val |= AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1;
+		break;
+	default:
+		dev_err(codec->dev, "invalid dai format\n");
+		return -EINVAL;
+	}
+
+	/* This device can only be slave */
+	if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
+		return -EINVAL;
+
+	return ak4104_spi_write(codec, AK4104_REG_CONTROL1, val);
+}
+
+static int ak4104_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params,
+			    struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	int val = 0;
+
+	/* set the IEC958 bits: consumer mode, no copyright bit */
+	val |= IEC958_AES0_CON_NOT_COPYRIGHT;
+	ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(0), val);
+
+	val = 0;
+
+	switch (params_rate(params)) {
+	case 44100:
+		val |= IEC958_AES3_CON_FS_44100;
+		break;
+	case 48000:
+		val |= IEC958_AES3_CON_FS_48000;
+		break;
+	case 32000:
+		val |= IEC958_AES3_CON_FS_32000;
+		break;
+	default:
+		dev_err(codec->dev, "unsupported sampling rate\n");
+		return -EINVAL;
+	}
+
+	return ak4104_spi_write(codec, AK4104_REG_CHN_STATUS(3), val);
+}
+
+static struct snd_soc_dai_ops ak4101_dai_ops = {
+	.hw_params = ak4104_hw_params,
+	.set_fmt = ak4104_set_dai_fmt,
+};
+
+struct snd_soc_dai ak4104_dai = {
+	.name = DRV_NAME,
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = SNDRV_PCM_RATE_44100 |
+			 SNDRV_PCM_RATE_48000 |
+			 SNDRV_PCM_RATE_32000,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE  |
+			   SNDRV_PCM_FMTBIT_S24_3LE |
+			   SNDRV_PCM_FMTBIT_S24_LE
+	},
+	.ops = &ak4101_dai_ops,
+};
+
+static struct snd_soc_codec *ak4104_codec;
+
+static int ak4104_spi_probe(struct spi_device *spi)
+{
+	struct snd_soc_codec *codec;
+	struct ak4104_private *ak4104;
+	int ret, val;
+
+	spi->bits_per_word = 8;
+	spi->mode = SPI_MODE_0;
+	ret = spi_setup(spi);
+	if (ret < 0)
+		return ret;
+
+	ak4104 = kzalloc(sizeof(struct ak4104_private), GFP_KERNEL);
+	if (!ak4104) {
+		dev_err(&spi->dev, "could not allocate codec\n");
+		return -ENOMEM;
+	}
+
+	codec = &ak4104->codec;
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	codec->dev = &spi->dev;
+	codec->name = DRV_NAME;
+	codec->owner = THIS_MODULE;
+	codec->dai = &ak4104_dai;
+	codec->num_dai = 1;
+	codec->private_data = ak4104;
+	codec->control_data = spi;
+	codec->reg_cache = ak4104->reg_cache;
+	codec->reg_cache_size = AK4104_NUM_REGS;
+
+	/* read all regs and fill the cache */
+	ret = ak4104_fill_cache(codec);
+	if (ret < 0) {
+		dev_err(&spi->dev, "failed to fill register cache\n");
+		return ret;
+	}
+
+	/* read the 'reserved' register - according to the datasheet, it
+	 * should contain 0x5b. Not a good way to verify the presence of
+	 * the device, but there is no hardware ID register. */
+	if (ak4104_read_reg_cache(codec, AK4104_REG_RESERVED) !=
+					 AK4104_RESERVED_VAL) {
+		ret = -ENODEV;
+		goto error_free_codec;
+	}
+
+	/* set power-up and non-reset bits */
+	val = ak4104_read_reg_cache(codec, AK4104_REG_CONTROL1);
+	val |= AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN;
+	ret = ak4104_spi_write(codec, AK4104_REG_CONTROL1, val);
+	if (ret < 0)
+		goto error_free_codec;
+
+	/* enable transmitter */
+	val = ak4104_read_reg_cache(codec, AK4104_REG_TX);
+	val |= AK4104_TX_TXE;
+	ret = ak4104_spi_write(codec, AK4104_REG_TX, val);
+	if (ret < 0)
+		goto error_free_codec;
+
+	ak4104_codec = codec;
+	ret = snd_soc_register_dai(&ak4104_dai);
+	if (ret < 0) {
+		dev_err(&spi->dev, "failed to register DAI\n");
+		goto error_free_codec;
+	}
+
+	spi_set_drvdata(spi, ak4104);
+	dev_info(&spi->dev, "SPI device initialized\n");
+	return 0;
+
+error_free_codec:
+	kfree(ak4104);
+	ak4104_dai.dev = NULL;
+	return ret;
+}
+
+static int __devexit ak4104_spi_remove(struct spi_device *spi)
+{
+	int ret, val;
+	struct ak4104_private *ak4104 = spi_get_drvdata(spi);
+
+	val = ak4104_read_reg_cache(&ak4104->codec, AK4104_REG_CONTROL1);
+	if (val < 0)
+		return val;
+
+	/* clear power-up and non-reset bits */
+	val &= ~(AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN);
+	ret = ak4104_spi_write(&ak4104->codec, AK4104_REG_CONTROL1, val);
+	if (ret < 0)
+		return ret;
+
+	ak4104_codec = NULL;
+	kfree(ak4104);
+	return 0;
+}
+
+static int ak4104_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = ak4104_codec;
+	int ret;
+
+	/* Connect the codec to the socdev.  snd_soc_new_pcms() needs this. */
+	socdev->card->codec = codec;
+
+	/* Register PCMs */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to create pcms\n");
+		return ret;
+	}
+
+	/* Register the socdev */
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to register card\n");
+		snd_soc_free_pcms(socdev);
+		return ret;
+	}
+
+	return 0;
+}
+
+static int ak4104_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	snd_soc_free_pcms(socdev);
+	return 0;
+};
+
+struct snd_soc_codec_device soc_codec_device_ak4104 = {
+	.probe = 	ak4104_probe,
+	.remove = 	ak4104_remove
+};
+EXPORT_SYMBOL_GPL(soc_codec_device_ak4104);
+
+static struct spi_driver ak4104_spi_driver = {
+	.driver  = {
+		.name   = DRV_NAME,
+		.owner  = THIS_MODULE,
+	},
+	.probe  = ak4104_spi_probe,
+	.remove = __devexit_p(ak4104_spi_remove),
+};
+
+static int __init ak4104_init(void)
+{
+	pr_info("Asahi Kasei AK4104 ALSA SoC Codec Driver\n");
+	return spi_register_driver(&ak4104_spi_driver);
+}
+module_init(ak4104_init);
+
+static void __exit ak4104_exit(void)
+{
+	spi_unregister_driver(&ak4104_spi_driver);
+}
+module_exit(ak4104_exit);
+
+MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
+MODULE_DESCRIPTION("Asahi Kasei AK4104 ALSA SoC driver");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/codecs/ak4104.h b/sound/soc/codecs/ak4104.h
new file mode 100644
index 000000000000..eb88fe7e4def
--- /dev/null
+++ b/sound/soc/codecs/ak4104.h
@@ -0,0 +1,7 @@
+#ifndef _AK4104_H
+#define _AK4104_H
+
+extern struct snd_soc_dai ak4104_dai;
+extern struct snd_soc_codec_device soc_codec_device_ak4104;
+
+#endif
diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c
index 81300d8d42ca..1f63d387a2f4 100644
--- a/sound/soc/codecs/ak4535.c
+++ b/sound/soc/codecs/ak4535.c
@@ -155,21 +155,6 @@ static const struct snd_kcontrol_new ak4535_snd_controls[] = {
 	SOC_SINGLE("Mic Sidetone Volume", AK4535_VOL, 4, 7, 0),
 };
 
-/* add non dapm controls */
-static int ak4535_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(ak4535_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&ak4535_snd_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /* Mono 1 Mixer */
 static const struct snd_kcontrol_new ak4535_mono1_mixer_controls[] = {
 	SOC_DAPM_SINGLE("Mic Sidetone Switch", AK4535_SIG1, 4, 1, 0),
@@ -344,7 +329,7 @@ static int ak4535_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct ak4535_priv *ak4535 = codec->private_data;
 	u8 mode2 = ak4535_read_reg_cache(codec, AK4535_MODE2) & ~(0x3 << 5);
 	int rate = params_rate(params), fs = 256;
@@ -436,6 +421,13 @@ static int ak4535_set_bias_level(struct snd_soc_codec *codec,
 		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
 		SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
 
+static struct snd_soc_dai_ops ak4535_dai_ops = {
+	.hw_params	= ak4535_hw_params,
+	.set_fmt	= ak4535_set_dai_fmt,
+	.digital_mute	= ak4535_mute,
+	.set_sysclk	= ak4535_set_dai_sysclk,
+};
+
 struct snd_soc_dai ak4535_dai = {
 	.name = "AK4535",
 	.playback = {
@@ -450,19 +442,14 @@ struct snd_soc_dai ak4535_dai = {
 		.channels_max = 2,
 		.rates = AK4535_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = ak4535_hw_params,
-		.set_fmt = ak4535_set_dai_fmt,
-		.digital_mute = ak4535_mute,
-		.set_sysclk = ak4535_set_dai_sysclk,
-	},
+	.ops = &ak4535_dai_ops,
 };
 EXPORT_SYMBOL_GPL(ak4535_dai);
 
 static int ak4535_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -471,7 +458,7 @@ static int ak4535_suspend(struct platform_device *pdev, pm_message_t state)
 static int ak4535_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	ak4535_sync(codec);
 	ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	ak4535_set_bias_level(codec, codec->suspend_bias_level);
@@ -484,7 +471,7 @@ static int ak4535_resume(struct platform_device *pdev)
  */
 static int ak4535_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 
 	codec->name = "AK4535";
@@ -510,7 +497,8 @@ static int ak4535_init(struct snd_soc_device *socdev)
 	/* power on device */
 	ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	ak4535_add_controls(codec);
+	snd_soc_add_controls(codec, ak4535_snd_controls,
+				ARRAY_SIZE(ak4535_snd_controls));
 	ak4535_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -537,7 +525,7 @@ static int ak4535_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = ak4535_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -636,7 +624,7 @@ static int ak4535_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = ak4535;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -663,7 +651,7 @@ static int ak4535_probe(struct platform_device *pdev)
 static int ak4535_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		ak4535_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index f1aa0c34421c..7fa09a387622 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -3,27 +3,22 @@
  *
  * Author: Timur Tabi <timur@freescale.com>
  *
- * Copyright 2007 Freescale Semiconductor, Inc.  This file is licensed under
- * the terms of the GNU General Public License version 2.  This program
- * is licensed "as is" without any warranty of any kind, whether express
- * or implied.
+ * Copyright 2007-2009 Freescale Semiconductor, Inc.  This file is licensed
+ * under the terms of the GNU General Public License version 2.  This
+ * program is licensed "as is" without any warranty of any kind, whether
+ * express or implied.
  *
  * This is an ASoC device driver for the Cirrus Logic CS4270 codec.
  *
  * Current features/limitations:
  *
- * 1) Software mode is supported.  Stand-alone mode is automatically
- *    selected if I2C is disabled or if a CS4270 is not found on the I2C
- *    bus.  However, stand-alone mode is only partially implemented because
- *    there is no mechanism yet for this driver and the machine driver to
- *    communicate the values of the M0, M1, MCLK1, and MCLK2 pins.
- * 2) Only I2C is supported, not SPI
- * 3) Only Master mode is supported, not Slave.
- * 4) The machine driver's 'startup' function must call
- *    cs4270_set_dai_sysclk() with the value of MCLK.
- * 5) Only I2S and left-justified modes are supported
- * 6) Power management is not supported
- * 7) The only supported control is volume and hardware mute (if enabled)
+ * - Software mode is supported.  Stand-alone mode is not supported.
+ * - Only I2C is supported, not SPI
+ * - Support for master and slave mode
+ * - The machine driver's 'startup' function must call
+ *   cs4270_set_dai_sysclk() with the value of MCLK.
+ * - Only I2S and left-justified modes are supported
+ * - Power management is not supported
  */
 
 #include <linux/module.h>
@@ -35,18 +30,6 @@
 
 #include "cs4270.h"
 
-/* If I2C is defined, then we support software mode.  However, if we're
-   not compiled as module but I2C is, then we can't use I2C calls. */
-#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE))
-#define USE_I2C
-#endif
-
-/* Private data for the CS4270 */
-struct cs4270_private {
-	unsigned int mclk; /* Input frequency of the MCLK pin */
-	unsigned int mode; /* The mode (I2S or left-justified) */
-};
-
 /*
  * The codec isn't really big-endian or little-endian, since the I2S
  * interface requires data to be sent serially with the MSbit first.
@@ -60,8 +43,6 @@ struct cs4270_private {
 			SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \
 			SNDRV_PCM_FMTBIT_S24_LE  | SNDRV_PCM_FMTBIT_S24_BE)
 
-#ifdef USE_I2C
-
 /* CS4270 registers addresses */
 #define CS4270_CHIPID	0x01	/* Chip ID */
 #define CS4270_PWRCTL	0x02	/* Power Control */
@@ -121,8 +102,22 @@ struct cs4270_private {
 #define CS4270_MUTE_DAC_A	0x01
 #define CS4270_MUTE_DAC_B	0x02
 
-/*
- * Clock Ratio Selection for Master Mode with I2C enabled
+/* Private data for the CS4270 */
+struct cs4270_private {
+	struct snd_soc_codec codec;
+	u8 reg_cache[CS4270_NUMREGS];
+	unsigned int mclk; /* Input frequency of the MCLK pin */
+	unsigned int mode; /* The mode (I2S or left-justified) */
+	unsigned int slave_mode;
+};
+
+/**
+ * struct cs4270_mode_ratios - clock ratio tables
+ * @ratio: the ratio of MCLK to the sample rate
+ * @speed_mode: the Speed Mode bits to set in the Mode Control register for
+ *              this ratio
+ * @mclk: the Ratio Select bits to set in the Mode Control register for this
+ *        ratio
  *
  * The data for this chart is taken from Table 5 of the CS4270 reference
  * manual.
@@ -131,31 +126,30 @@ struct cs4270_private {
  * It is also used by cs4270_set_dai_sysclk() to tell ALSA which sampling
  * rates the CS4270 currently supports.
  *
- * Each element in this array corresponds to the ratios in mclk_ratios[].
- * These two arrays need to be in sync.
- *
- * 'speed_mode' is the corresponding bit pattern to be written to the
+ * @speed_mode is the corresponding bit pattern to be written to the
  * MODE bits of the Mode Control Register
  *
- * 'mclk' is the corresponding bit pattern to be wirten to the MCLK bits of
+ * @mclk is the corresponding bit pattern to be wirten to the MCLK bits of
  * the Mode Control Register.
  *
  * In situations where a single ratio is represented by multiple speed
  * modes, we favor the slowest speed.  E.g, for a ratio of 128, we pick
  * double-speed instead of quad-speed.  However, the CS4270 errata states
- * that Divide-By-1.5 can cause failures, so we avoid that mode where
+ * that divide-By-1.5 can cause failures, so we avoid that mode where
  * possible.
  *
- * ERRATA: There is an errata for the CS4270 where divide-by-1.5 does not
- * work if VD = 3.3V.  If this effects you, select the
+ * Errata: There is an errata for the CS4270 where divide-by-1.5 does not
+ * work if Vd is 3.3V.  If this effects you, select the
  * CONFIG_SND_SOC_CS4270_VD33_ERRATA Kconfig option, and the driver will
  * never select any sample rates that require divide-by-1.5.
  */
-static struct {
+struct cs4270_mode_ratios {
 	unsigned int ratio;
 	u8 speed_mode;
 	u8 mclk;
-} cs4270_mode_ratios[] = {
+};
+
+static struct cs4270_mode_ratios cs4270_mode_ratios[] = {
 	{64, CS4270_MODE_4X, CS4270_MODE_DIV1},
 #ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA
 	{96, CS4270_MODE_4X, CS4270_MODE_DIV15},
@@ -172,34 +166,27 @@ static struct {
 /* The number of MCLK/LRCK ratios supported by the CS4270 */
 #define NUM_MCLK_RATIOS		ARRAY_SIZE(cs4270_mode_ratios)
 
-/*
- * Determine the CS4270 samples rates.
+/**
+ * cs4270_set_dai_sysclk - determine the CS4270 samples rates.
+ * @codec_dai: the codec DAI
+ * @clk_id: the clock ID (ignored)
+ * @freq: the MCLK input frequency
+ * @dir: the clock direction (ignored)
  *
- * 'freq' is the input frequency to MCLK.  The other parameters are ignored.
+ * This function is used to tell the codec driver what the input MCLK
+ * frequency is.
  *
  * The value of MCLK is used to determine which sample rates are supported
  * by the CS4270.  The ratio of MCLK / Fs must be equal to one of nine
- * support values: 64, 96, 128, 192, 256, 384, 512, 768, and 1024.
+ * supported values - 64, 96, 128, 192, 256, 384, 512, 768, and 1024.
  *
  * This function calculates the nine ratios and determines which ones match
  * a standard sample rate.  If there's a match, then it is added to the list
- * of support sample rates.
+ * of supported sample rates.
  *
  * This function must be called by the machine driver's 'startup' function,
  * otherwise the list of supported sample rates will not be available in
  * time for ALSA.
- *
- * Note that in stand-alone mode, the sample rate is determined by input
- * pins M0, M1, MDIV1, and MDIV2.  Also in stand-alone mode, divide-by-3
- * is not a programmable option.  However, divide-by-3 is not an available
- * option in stand-alone mode.  This cases two problems: a ratio of 768 is
- * not available (it requires divide-by-3) and B) ratios 192 and 384 can
- * only be selected with divide-by-1.5, but there is an errate that make
- * this selection difficult.
- *
- * In addition, there is no mechanism for communicating with the machine
- * driver what the input settings can be.  This would need to be implemented
- * for stand-alone mode to work.
  */
 static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 				 int clk_id, unsigned int freq, int dir)
@@ -225,7 +212,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	rates &= ~SNDRV_PCM_RATE_KNOT;
 
 	if (!rates) {
-		printk(KERN_ERR "cs4270: could not find a valid sample rate\n");
+		dev_err(codec->dev, "could not find a valid sample rate\n");
 		return -EINVAL;
 	}
 
@@ -240,8 +227,10 @@ static int cs4270_set_dai_sysclk(struct snd_soc_dai *codec_dai,
 	return 0;
 }
 
-/*
- * Configure the codec for the selected audio format
+/**
+ * cs4270_set_dai_fmt - configure the codec for the selected audio format
+ * @codec_dai: the codec DAI
+ * @format: a SND_SOC_DAIFMT_x value indicating the data format
  *
  * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the
  * codec accordingly.
@@ -258,32 +247,43 @@ static int cs4270_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	struct cs4270_private *cs4270 = codec->private_data;
 	int ret = 0;
 
+	/* set DAI format */
 	switch (format & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 	case SND_SOC_DAIFMT_LEFT_J:
 		cs4270->mode = format & SND_SOC_DAIFMT_FORMAT_MASK;
 		break;
 	default:
-		printk(KERN_ERR "cs4270: invalid DAI format\n");
+		dev_err(codec->dev, "invalid dai format\n");
+		ret = -EINVAL;
+	}
+
+	/* set master/slave audio interface */
+	switch (format & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		cs4270->slave_mode = 1;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		cs4270->slave_mode = 0;
+		break;
+	default:
+		/* all other modes are unsupported by the hardware */
 		ret = -EINVAL;
 	}
 
 	return ret;
 }
 
-/*
- * A list of addresses on which this CS4270 could use.  I2C addresses are
- * 7 bits.  For the CS4270, the upper four bits are always 1001, and the
- * lower three bits are determined via the AD2, AD1, and AD0 pins
- * (respectively).
- */
-static const unsigned short normal_i2c[] = {
-	0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, I2C_CLIENT_END
-};
-I2C_CLIENT_INSMOD;
-
-/*
- * Pre-fill the CS4270 register cache.
+/**
+ * cs4270_fill_cache - pre-fill the CS4270 register cache.
+ * @codec: the codec for this CS4270
+ *
+ * This function fills in the CS4270 register cache by reading the register
+ * values from the hardware.
+ *
+ * This CS4270 registers are cached to avoid excessive I2C I/O operations.
+ * After the initial read to pre-fill the cache, the CS4270 never updates
+ * the register values, so we won't have a cache coherency problem.
  *
  * We use the auto-increment feature of the CS4270 to read all registers in
  * one shot.
@@ -298,7 +298,7 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec)
 		CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache);
 
 	if (length != CS4270_NUMREGS) {
-		printk(KERN_ERR "cs4270: I2C read failure, addr=0x%x\n",
+		dev_err(codec->dev, "i2c read failure, addr=0x%x\n",
 		       i2c_client->addr);
 		return -EIO;
 	}
@@ -306,12 +306,17 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec)
 	return 0;
 }
 
-/*
- * Read from the CS4270 register cache.
+/**
+ * cs4270_read_reg_cache - read from the CS4270 register cache.
+ * @codec: the codec for this CS4270
+ * @reg: the register to read
+ *
+ * This function returns the value for a given register.  It reads only from
+ * the register cache, not the hardware itself.
  *
  * This CS4270 registers are cached to avoid excessive I2C I/O operations.
  * After the initial read to pre-fill the cache, the CS4270 never updates
- * the register values, so we won't have a cache coherncy problem.
+ * the register values, so we won't have a cache coherency problem.
  */
 static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg)
@@ -324,8 +329,11 @@ static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec,
 	return cache[reg - CS4270_FIRSTREG];
 }
 
-/*
- * Write to a CS4270 register via the I2C bus.
+/**
+ * cs4270_i2c_write - write to a CS4270 register via the I2C bus.
+ * @codec: the codec for this CS4270
+ * @reg: the register to write
+ * @value: the value to write to the register
  *
  * This function writes the given value to the given CS4270 register, and
  * also updates the register cache.
@@ -346,7 +354,7 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg,
 	if (cache[reg - CS4270_FIRSTREG] != value) {
 		struct i2c_client *client = codec->control_data;
 		if (i2c_smbus_write_byte_data(client, reg, value)) {
-			printk(KERN_ERR "cs4270: I2C write failed\n");
+			dev_err(codec->dev, "i2c write failed\n");
 			return -EIO;
 		}
 
@@ -357,11 +365,17 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg,
 	return 0;
 }
 
-/*
- * Program the CS4270 with the given hardware parameters.
+/**
+ * cs4270_hw_params - program the CS4270 with the given hardware parameters.
+ * @substream: the audio stream
+ * @params: the hardware parameters to set
+ * @dai: the SOC DAI (ignored)
+ *
+ * This function programs the hardware with the values provided.
+ * Specifically, the sample rate and the data format.
  *
- * The .ops functions are used to provide board-specific data, like
- * input frequencies, to this driver.  This function takes that information,
+ * The .ops functions are used to provide board-specific data, like input
+ * frequencies, to this driver.  This function takes that information,
  * combines it with the hardware parameters provided, and programs the
  * hardware accordingly.
  */
@@ -371,7 +385,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct cs4270_private *cs4270 = codec->private_data;
 	int ret;
 	unsigned int i;
@@ -391,33 +405,28 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 
 	if (i == NUM_MCLK_RATIOS) {
 		/* We did not find a matching ratio */
-		printk(KERN_ERR "cs4270: could not find matching ratio\n");
+		dev_err(codec->dev, "could not find matching ratio\n");
 		return -EINVAL;
 	}
 
-	/* Freeze and power-down the codec */
-
-	ret = snd_soc_write(codec, CS4270_PWRCTL, CS4270_PWRCTL_FREEZE |
-			    CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC |
-			    CS4270_PWRCTL_PDN);
-	if (ret < 0) {
-		printk(KERN_ERR "cs4270: I2C write failed\n");
-		return ret;
-	}
-
-	/* Program the mode control register */
+	/* Set the sample rate */
 
 	reg = snd_soc_read(codec, CS4270_MODE);
 	reg &= ~(CS4270_MODE_SPEED_MASK | CS4270_MODE_DIV_MASK);
-	reg |= cs4270_mode_ratios[i].speed_mode | cs4270_mode_ratios[i].mclk;
+	reg |= cs4270_mode_ratios[i].mclk;
+
+	if (cs4270->slave_mode)
+		reg |= CS4270_MODE_SLAVE;
+	else
+		reg |= cs4270_mode_ratios[i].speed_mode;
 
 	ret = snd_soc_write(codec, CS4270_MODE, reg);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: I2C write failed\n");
+		dev_err(codec->dev, "i2c write failed\n");
 		return ret;
 	}
 
-	/* Program the format register */
+	/* Set the DAI format */
 
 	reg = snd_soc_read(codec, CS4270_FORMAT);
 	reg &= ~(CS4270_FORMAT_DAC_MASK | CS4270_FORMAT_ADC_MASK);
@@ -430,55 +439,23 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 		reg |= CS4270_FORMAT_DAC_LJ | CS4270_FORMAT_ADC_LJ;
 		break;
 	default:
-		printk(KERN_ERR "cs4270: unknown format\n");
+		dev_err(codec->dev, "unknown dai format\n");
 		return -EINVAL;
 	}
 
 	ret = snd_soc_write(codec, CS4270_FORMAT, reg);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: I2C write failed\n");
-		return ret;
-	}
-
-	/* Disable auto-mute.  This feature appears to be buggy, because in
-	   some situations, auto-mute will not deactivate when it should. */
-
-	reg = snd_soc_read(codec, CS4270_MUTE);
-	reg &= ~CS4270_MUTE_AUTO;
-	ret = snd_soc_write(codec, CS4270_MUTE, reg);
-	if (ret < 0) {
-		printk(KERN_ERR "cs4270: I2C write failed\n");
-		return ret;
-	}
-
-	/* Disable automatic volume control.  It's enabled by default, and
-	 * it causes volume change commands to be delayed, sometimes until
-	 * after playback has started.
-	 */
-
-	reg = cs4270_read_reg_cache(codec, CS4270_TRANS);
-	reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO);
-	ret = cs4270_i2c_write(codec, CS4270_TRANS, reg);
-	if (ret < 0) {
-		printk(KERN_ERR "I2C write failed\n");
-		return ret;
-	}
-
-	/* Thaw and power-up the codec */
-
-	ret = snd_soc_write(codec, CS4270_PWRCTL, 0);
-	if (ret < 0) {
-		printk(KERN_ERR "cs4270: I2C write failed\n");
+		dev_err(codec->dev, "i2c write failed\n");
 		return ret;
 	}
 
 	return ret;
 }
 
-#ifdef CONFIG_SND_SOC_CS4270_HWMUTE
-
-/*
- * Set the CS4270 external mute
+/**
+ * cs4270_mute - enable/disable the CS4270 external mute
+ * @dai: the SOC DAI
+ * @mute: 0 = disable mute, 1 = enable mute
  *
  * This function toggles the mute bits in the MUTE register.  The CS4270's
  * mute capability is intended for external muting circuitry, so if the
@@ -493,276 +470,306 @@ static int cs4270_mute(struct snd_soc_dai *dai, int mute)
 	reg6 = snd_soc_read(codec, CS4270_MUTE);
 
 	if (mute)
-		reg6 |= CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B |
-			CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B;
+		reg6 |= CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B;
 	else
-		reg6 &= ~(CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B |
-			  CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B);
+		reg6 &= ~(CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B);
 
 	return snd_soc_write(codec, CS4270_MUTE, reg6);
 }
 
-#endif
-
-static int cs4270_i2c_probe(struct i2c_client *, const struct i2c_device_id *);
-
 /* A list of non-DAPM controls that the CS4270 supports */
 static const struct snd_kcontrol_new cs4270_snd_controls[] = {
 	SOC_DOUBLE_R("Master Playback Volume",
-		CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1)
-};
-
-static const struct i2c_device_id cs4270_id[] = {
-	{"cs4270", 0},
-	{}
-};
-MODULE_DEVICE_TABLE(i2c, cs4270_id);
-
-static struct i2c_driver cs4270_i2c_driver = {
-	.driver = {
-		.name = "CS4270 I2C",
-		.owner = THIS_MODULE,
-	},
-	.id_table = cs4270_id,
-	.probe = cs4270_i2c_probe,
+		CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1),
+	SOC_SINGLE("Digital Sidetone Switch", CS4270_FORMAT, 5, 1, 0),
+	SOC_SINGLE("Soft Ramp Switch", CS4270_TRANS, 6, 1, 0),
+	SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0),
+	SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1),
+	SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0),
+	SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 0)
 };
 
 /*
- * Global variable to store socdev for i2c probe function.
+ * cs4270_codec - global variable to store codec for the ASoC probe function
  *
  * If struct i2c_driver had a private_data field, we wouldn't need to use
- * cs4270_socdec.  This is the only way to pass the socdev structure to
- * cs4270_i2c_probe().
- *
- * The real solution to cs4270_socdev is to create a mechanism
- * that maps I2C addresses to snd_soc_device structures.  Perhaps the
- * creation of the snd_soc_device object should be moved out of
- * cs4270_probe() and into cs4270_i2c_probe(), but that would make this
- * driver dependent on I2C.  The CS4270 supports "stand-alone" mode, whereby
- * the chip is *not* connected to the I2C bus, but is instead configured via
- * input pins.
+ * cs4270_codec.  This is the only way to pass the codec structure from
+ * cs4270_i2c_probe() to cs4270_probe().  Unfortunately, there is no good
+ * way to synchronize these two functions.  cs4270_i2c_probe() can be called
+ * multiple times before cs4270_probe() is called even once.  So for now, we
+ * also only allow cs4270_i2c_probe() to be run once.  That means that we do
+ * not support more than one cs4270 device in the system, at least for now.
  */
-static struct snd_soc_device *cs4270_socdev;
+static struct snd_soc_codec *cs4270_codec;
 
-/*
- * Initialize the I2C interface of the CS4270
- *
- * This function is called for whenever the I2C subsystem finds a device
- * at a particular address.
+static struct snd_soc_dai_ops cs4270_dai_ops = {
+	.hw_params	= cs4270_hw_params,
+	.set_sysclk	= cs4270_set_dai_sysclk,
+	.set_fmt	= cs4270_set_dai_fmt,
+	.digital_mute	= cs4270_mute,
+};
+
+struct snd_soc_dai cs4270_dai = {
+	.name = "cs4270",
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = 0,
+		.formats = CS4270_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = 0,
+		.formats = CS4270_FORMATS,
+	},
+	.ops = &cs4270_dai_ops,
+};
+EXPORT_SYMBOL_GPL(cs4270_dai);
+
+/**
+ * cs4270_probe - ASoC probe function
+ * @pdev: platform device
  *
- * Note: snd_soc_new_pcms() must be called before this function can be called,
- * because of snd_ctl_add().
+ * This function is called when ASoC has all the pieces it needs to
+ * instantiate a sound driver.
  */
-static int cs4270_i2c_probe(struct i2c_client *i2c_client,
-	const struct i2c_device_id *id)
+static int cs4270_probe(struct platform_device *pdev)
 {
-	struct snd_soc_device *socdev = cs4270_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int i;
-	int ret = 0;
-
-	/* Probing all possible addresses has one drawback: if there are
-	   multiple CS4270s on the bus, then you cannot specify which
-	   socdev is matched with which CS4270.  For now, we just reject
-	   this I2C device if the socdev already has one attached. */
-	if (codec->control_data)
-		return -ENODEV;
-
-	/* Note: codec_dai->codec is NULL here */
-
-	codec->reg_cache = kzalloc(CS4270_NUMREGS, GFP_KERNEL);
-	if (!codec->reg_cache) {
-		printk(KERN_ERR "cs4270: could not allocate register cache\n");
-		ret = -ENOMEM;
-		goto error;
-	}
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = cs4270_codec;
+	int ret;
 
-	/* Verify that we have a CS4270 */
+	/* Connect the codec to the socdev.  snd_soc_new_pcms() needs this. */
+	socdev->card->codec = codec;
 
-	ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID);
+	/* Register PCMs */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: failed to read I2C\n");
-		goto error;
-	}
-	/* The top four bits of the chip ID should be 1100. */
-	if ((ret & 0xF0) != 0xC0) {
-		/* The device at this address is not a CS4270 codec */
-		ret = -ENODEV;
-		goto error;
+		dev_err(codec->dev, "failed to create pcms\n");
+		return ret;
 	}
 
-	printk(KERN_INFO "cs4270: found device at I2C address %X\n",
-		i2c_client->addr);
-	printk(KERN_INFO "cs4270: hardware revision %X\n", ret & 0xF);
-
-	codec->control_data = i2c_client;
-	codec->read = cs4270_read_reg_cache;
-	codec->write = cs4270_i2c_write;
-	codec->reg_cache_size = CS4270_NUMREGS;
-
-	/* The I2C interface is set up, so pre-fill our register cache */
-
-	ret = cs4270_fill_cache(codec);
+	/* Add the non-DAPM controls */
+	ret = snd_soc_add_controls(codec, cs4270_snd_controls,
+				ARRAY_SIZE(cs4270_snd_controls));
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: failed to fill register cache\n");
-		goto error;
+		dev_err(codec->dev, "failed to add controls\n");
+		goto error_free_pcms;
 	}
 
-	/* Add the non-DAPM controls */
-
-	for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) {
-		struct snd_kcontrol *kctrl =
-		snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL);
-
-		ret = snd_ctl_add(codec->card, kctrl);
-		if (ret < 0)
-			goto error;
+	/* And finally, register the socdev */
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		dev_err(codec->dev, "failed to register card\n");
+		goto error_free_pcms;
 	}
 
-	i2c_set_clientdata(i2c_client, codec);
-
 	return 0;
 
-error:
-	codec->control_data = NULL;
-
-	kfree(codec->reg_cache);
-	codec->reg_cache = NULL;
-	codec->reg_cache_size = 0;
+error_free_pcms:
+	snd_soc_free_pcms(socdev);
 
 	return ret;
 }
 
-#endif /* USE_I2C*/
+/**
+ * cs4270_remove - ASoC remove function
+ * @pdev: platform device
+ *
+ * This function is the counterpart to cs4270_probe().
+ */
+static int cs4270_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 
-struct snd_soc_dai cs4270_dai = {
-	.name = "CS4270",
-	.playback = {
-		.stream_name = "Playback",
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = 0,
-		.formats = CS4270_FORMATS,
-	},
-	.capture = {
-		.stream_name = "Capture",
-		.channels_min = 1,
-		.channels_max = 2,
-		.rates = 0,
-		.formats = CS4270_FORMATS,
-	},
+	snd_soc_free_pcms(socdev);
+
+	return 0;
 };
-EXPORT_SYMBOL_GPL(cs4270_dai);
 
-/*
- * ASoC probe function
+/**
+ * cs4270_i2c_probe - initialize the I2C interface of the CS4270
+ * @i2c_client: the I2C client object
+ * @id: the I2C device ID (ignored)
  *
- * This function is called when the machine driver calls
- * platform_device_add().
+ * This function is called whenever the I2C subsystem finds a device that
+ * matches the device ID given via a prior call to i2c_add_driver().
  */
-static int cs4270_probe(struct platform_device *pdev)
+static int cs4270_i2c_probe(struct i2c_client *i2c_client,
+	const struct i2c_device_id *id)
 {
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec;
-	int ret = 0;
+	struct cs4270_private *cs4270;
+	unsigned int reg;
+	int ret;
 
-	printk(KERN_INFO "CS4270 ALSA SoC Codec\n");
+	/* For now, we only support one cs4270 device in the system.  See the
+	 * comment for cs4270_codec.
+	 */
+	if (cs4270_codec) {
+		dev_err(&i2c_client->dev, "ignoring CS4270 at addr %X\n",
+		       i2c_client->addr);
+		dev_err(&i2c_client->dev, "only one per board allowed\n");
+		/* Should we return something other than ENODEV here? */
+		return -ENODEV;
+	}
+
+	/* Verify that we have a CS4270 */
+
+	ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID);
+	if (ret < 0) {
+		dev_err(&i2c_client->dev, "failed to read i2c at addr %X\n",
+		       i2c_client->addr);
+		return ret;
+	}
+	/* The top four bits of the chip ID should be 1100. */
+	if ((ret & 0xF0) != 0xC0) {
+		dev_err(&i2c_client->dev, "device at addr %X is not a CS4270\n",
+		       i2c_client->addr);
+		return -ENODEV;
+	}
+
+	dev_info(&i2c_client->dev, "found device at i2c address %X\n",
+		i2c_client->addr);
+	dev_info(&i2c_client->dev, "hardware revision %X\n", ret & 0xF);
 
 	/* Allocate enough space for the snd_soc_codec structure
 	   and our private data together. */
-	codec = kzalloc(ALIGN(sizeof(struct snd_soc_codec), 4) +
-			sizeof(struct cs4270_private), GFP_KERNEL);
-	if (!codec) {
-		printk(KERN_ERR "cs4270: Could not allocate codec structure\n");
+	cs4270 = kzalloc(sizeof(struct cs4270_private), GFP_KERNEL);
+	if (!cs4270) {
+		dev_err(&i2c_client->dev, "could not allocate codec\n");
 		return -ENOMEM;
 	}
+	codec = &cs4270->codec;
 
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
 
+	codec->dev = &i2c_client->dev;
 	codec->name = "CS4270";
 	codec->owner = THIS_MODULE;
 	codec->dai = &cs4270_dai;
 	codec->num_dai = 1;
-	codec->private_data = (void *) codec +
-		ALIGN(sizeof(struct snd_soc_codec), 4);
-
-	socdev->codec = codec;
+	codec->private_data = cs4270;
+	codec->control_data = i2c_client;
+	codec->read = cs4270_read_reg_cache;
+	codec->write = cs4270_i2c_write;
+	codec->reg_cache = cs4270->reg_cache;
+	codec->reg_cache_size = CS4270_NUMREGS;
 
-	/* Register PCMs */
+	/* The I2C interface is set up, so pre-fill our register cache */
 
-	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	ret = cs4270_fill_cache(codec);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: failed to create PCMs\n");
+		dev_err(&i2c_client->dev, "failed to fill register cache\n");
 		goto error_free_codec;
 	}
 
-#ifdef USE_I2C
-	cs4270_socdev = socdev;
+	/* Disable auto-mute.  This feature appears to be buggy.  In some
+	 * situations, auto-mute will not deactivate when it should, so we want
+	 * this feature disabled by default.  An application (e.g. alsactl) can
+	 * re-enabled it by using the controls.
+	 */
 
-	ret = i2c_add_driver(&cs4270_i2c_driver);
-	if (ret) {
-		printk(KERN_ERR "cs4270: failed to attach driver");
-		goto error_free_pcms;
+	reg = cs4270_read_reg_cache(codec, CS4270_MUTE);
+	reg &= ~CS4270_MUTE_AUTO;
+	ret = cs4270_i2c_write(codec, CS4270_MUTE, reg);
+	if (ret < 0) {
+		dev_err(&i2c_client->dev, "i2c write failed\n");
+		return ret;
 	}
 
-	/* Did we find a CS4270 on the I2C bus? */
-	if (codec->control_data) {
-		/* Initialize codec ops */
-		cs4270_dai.ops.hw_params = cs4270_hw_params;
-		cs4270_dai.ops.set_sysclk = cs4270_set_dai_sysclk;
-		cs4270_dai.ops.set_fmt = cs4270_set_dai_fmt;
-#ifdef CONFIG_SND_SOC_CS4270_HWMUTE
-		cs4270_dai.ops.digital_mute = cs4270_mute;
-#endif
-	} else
-		printk(KERN_INFO "cs4270: no I2C device found, "
-			"using stand-alone mode\n");
-#else
-	printk(KERN_INFO "cs4270: I2C disabled, using stand-alone mode\n");
-#endif
+	/* Disable automatic volume control.  The hardware enables, and it
+	 * causes volume change commands to be delayed, sometimes until after
+	 * playback has started.  An application (e.g. alsactl) can
+	 * re-enabled it by using the controls.
+	 */
 
-	ret = snd_soc_init_card(socdev);
+	reg = cs4270_read_reg_cache(codec, CS4270_TRANS);
+	reg &= ~(CS4270_TRANS_SOFT | CS4270_TRANS_ZERO);
+	ret = cs4270_i2c_write(codec, CS4270_TRANS, reg);
 	if (ret < 0) {
-		printk(KERN_ERR "cs4270: failed to register card\n");
-		goto error_del_driver;
+		dev_err(&i2c_client->dev, "i2c write failed\n");
+		return ret;
 	}
 
-	return 0;
+	/* Initialize the DAI. Normally, we'd prefer to have a kmalloc'd DAI
+	 * structure for each CS4270 device, but the machine driver needs to
+	 * have a pointer to the DAI structure, so for now it must be a global
+	 * variable.
+	 */
+	cs4270_dai.dev = &i2c_client->dev;
 
-error_del_driver:
-#ifdef USE_I2C
-	i2c_del_driver(&cs4270_i2c_driver);
+	/* Register the DAI.  If all the other ASoC driver have already
+	 * registered, then this will call our probe function, so
+	 * cs4270_codec needs to be ready.
+	 */
+	cs4270_codec = codec;
+	ret = snd_soc_register_dai(&cs4270_dai);
+	if (ret < 0) {
+		dev_err(&i2c_client->dev, "failed to register DAIe\n");
+		goto error_free_codec;
+	}
 
-error_free_pcms:
-#endif
-	snd_soc_free_pcms(socdev);
+	i2c_set_clientdata(i2c_client, cs4270);
+
+	return 0;
 
 error_free_codec:
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(cs4270);
+	cs4270_codec = NULL;
+	cs4270_dai.dev = NULL;
 
 	return ret;
 }
 
-static int cs4270_remove(struct platform_device *pdev)
+/**
+ * cs4270_i2c_remove - remove an I2C device
+ * @i2c_client: the I2C client object
+ *
+ * This function is the counterpart to cs4270_i2c_probe().
+ */
+static int cs4270_i2c_remove(struct i2c_client *i2c_client)
 {
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-
-	snd_soc_free_pcms(socdev);
-
-#ifdef USE_I2C
-	i2c_del_driver(&cs4270_i2c_driver);
-#endif
+	struct cs4270_private *cs4270 = i2c_get_clientdata(i2c_client);
 
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(cs4270);
+	cs4270_codec = NULL;
+	cs4270_dai.dev = NULL;
 
 	return 0;
 }
 
 /*
+ * cs4270_id - I2C device IDs supported by this driver
+ */
+static struct i2c_device_id cs4270_id[] = {
+	{"cs4270", 0},
+	{}
+};
+MODULE_DEVICE_TABLE(i2c, cs4270_id);
+
+/*
+ * cs4270_i2c_driver - I2C device identification
+ *
+ * This structure tells the I2C subsystem how to identify and support a
+ * given I2C device type.
+ */
+static struct i2c_driver cs4270_i2c_driver = {
+	.driver = {
+		.name = "cs4270",
+		.owner = THIS_MODULE,
+	},
+	.id_table = cs4270_id,
+	.probe = cs4270_i2c_probe,
+	.remove = cs4270_i2c_remove,
+};
+
+/*
  * ASoC codec device structure
  *
  * Assign this variable to the codec_dev field of the machine driver's
@@ -776,13 +783,15 @@ EXPORT_SYMBOL_GPL(soc_codec_device_cs4270);
 
 static int __init cs4270_init(void)
 {
-	return snd_soc_register_dai(&cs4270_dai);
+	pr_info("Cirrus Logic CS4270 ALSA SoC Codec Driver\n");
+
+	return i2c_add_driver(&cs4270_i2c_driver);
 }
 module_init(cs4270_init);
 
 static void __exit cs4270_exit(void)
 {
-	snd_soc_unregister_dai(&cs4270_dai);
+	i2c_del_driver(&cs4270_i2c_driver);
 }
 module_exit(cs4270_exit);
 
diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c
index 9a3e67e5319c..5cda9e6b5a74 100644
--- a/sound/soc/codecs/pcm3008.c
+++ b/sound/soc/codecs/pcm3008.c
@@ -67,11 +67,11 @@ static int pcm3008_soc_probe(struct platform_device *pdev)
 
 	printk(KERN_INFO "PCM3008 SoC Audio Codec %s\n", PCM3008_VERSION);
 
-	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (!socdev->codec)
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (!socdev->card->codec)
 		return -ENOMEM;
 
-	codec = socdev->codec;
+	codec = socdev->card->codec;
 	mutex_init(&codec->mutex);
 
 	codec->name = "PCM3008";
@@ -139,7 +139,7 @@ gpio_err:
 card_err:
 	snd_soc_free_pcms(socdev);
 pcm_err:
-	kfree(socdev->codec);
+	kfree(socdev->card->codec);
 
 	return ret;
 }
@@ -147,7 +147,7 @@ pcm_err:
 static int pcm3008_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct pcm3008_setup_data *setup = socdev->codec_data;
 
 	if (!codec)
@@ -155,7 +155,7 @@ static int pcm3008_soc_remove(struct platform_device *pdev)
 
 	pcm3008_gpio_free(setup);
 	snd_soc_free_pcms(socdev);
-	kfree(socdev->codec);
+	kfree(socdev->card->codec);
 
 	return 0;
 }
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index cac373616768..87f606c76822 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -151,21 +151,6 @@ SOC_ENUM("Capture Source", ssm2602_enum[0]),
 SOC_ENUM("Playback De-emphasis", ssm2602_enum[1]),
 };
 
-/* add non dapm controls */
-static int ssm2602_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(ssm2602_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&ssm2602_snd_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /* Output Mixer */
 static const struct snd_kcontrol_new ssm2602_output_mixer_controls[] = {
 SOC_DAPM_SINGLE("Line Bypass Switch", SSM2602_APANA, 3, 1, 0),
@@ -291,7 +276,7 @@ static int ssm2602_hw_params(struct snd_pcm_substream *substream,
 	u16 srate;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct ssm2602_priv *ssm2602 = codec->private_data;
 	struct i2c_client *i2c = codec->control_data;
 	u16 iface = ssm2602_read_reg_cache(codec, SSM2602_IFACE) & 0xfff3;
@@ -336,7 +321,7 @@ static int ssm2602_startup(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct ssm2602_priv *ssm2602 = codec->private_data;
 	struct i2c_client *i2c = codec->control_data;
 	struct snd_pcm_runtime *master_runtime;
@@ -373,7 +358,7 @@ static int ssm2602_pcm_prepare(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	/* set active */
 	ssm2602_write(codec, SSM2602_ACTIVE, ACTIVE_ACTIVATE_CODEC);
 
@@ -385,7 +370,7 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct ssm2602_priv *ssm2602 = codec->private_data;
 	/* deactivate */
 	if (!codec->active)
@@ -521,6 +506,16 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
 #define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 		SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops ssm2602_dai_ops = {
+	.startup	= ssm2602_startup,
+	.prepare	= ssm2602_pcm_prepare,
+	.hw_params	= ssm2602_hw_params,
+	.shutdown	= ssm2602_shutdown,
+	.digital_mute	= ssm2602_mute,
+	.set_sysclk	= ssm2602_set_dai_sysclk,
+	.set_fmt	= ssm2602_set_dai_fmt,
+};
+
 struct snd_soc_dai ssm2602_dai = {
 	.name = "SSM2602",
 	.playback = {
@@ -535,22 +530,14 @@ struct snd_soc_dai ssm2602_dai = {
 		.channels_max = 2,
 		.rates = SSM2602_RATES,
 		.formats = SSM2602_FORMATS,},
-	.ops = {
-		.startup = ssm2602_startup,
-		.prepare = ssm2602_pcm_prepare,
-		.hw_params = ssm2602_hw_params,
-		.shutdown = ssm2602_shutdown,
-		.digital_mute = ssm2602_mute,
-		.set_sysclk = ssm2602_set_dai_sysclk,
-		.set_fmt = ssm2602_set_dai_fmt,
-	}
+	.ops = &ssm2602_dai_ops,
 };
 EXPORT_SYMBOL_GPL(ssm2602_dai);
 
 static int ssm2602_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -559,7 +546,7 @@ static int ssm2602_suspend(struct platform_device *pdev, pm_message_t state)
 static int ssm2602_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -581,7 +568,7 @@ static int ssm2602_resume(struct platform_device *pdev)
  */
 static int ssm2602_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int reg, ret = 0;
 
 	codec->name = "SSM2602";
@@ -622,7 +609,8 @@ static int ssm2602_init(struct snd_soc_device *socdev)
 			APANA_ENABLE_MIC_BOOST);
 	ssm2602_write(codec, SSM2602_PWR, 0);
 
-	ssm2602_add_controls(codec);
+	snd_soc_add_controls(codec, ssm2602_snd_controls,
+				ARRAY_SIZE(ssm2602_snd_controls));
 	ssm2602_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -653,7 +641,7 @@ static int ssm2602_i2c_probe(struct i2c_client *i2c,
 			     const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = ssm2602_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -747,7 +735,7 @@ static int ssm2602_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = ssm2602;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -768,7 +756,7 @@ static int ssm2602_probe(struct platform_device *pdev)
 static int ssm2602_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		ssm2602_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c
index cfdea007c4cb..c3f4afb5d017 100644
--- a/sound/soc/codecs/tlv320aic23.c
+++ b/sound/soc/codecs/tlv320aic23.c
@@ -183,24 +183,6 @@ static const struct snd_kcontrol_new tlv320aic23_snd_controls[] = {
 	SOC_ENUM("Playback De-emphasis", tlv320aic23_deemph),
 };
 
-/* add non dapm controls */
-static int tlv320aic23_add_controls(struct snd_soc_codec *codec)
-{
-
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(tlv320aic23_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&tlv320aic23_snd_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-
-}
-
 /* PGA Mixer controls for Line and Mic switch */
 static const struct snd_kcontrol_new tlv320aic23_output_mixer_controls[] = {
 	SOC_DAPM_SINGLE("Line Bypass Switch", TLV320AIC23_ANLG, 3, 1, 0),
@@ -423,7 +405,7 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 iface_reg;
 	int ret;
 	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
@@ -471,7 +453,7 @@ static int tlv320aic23_pcm_prepare(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	/* set active */
 	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0001);
@@ -484,7 +466,7 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
 
 	/* deactivate */
@@ -598,6 +580,15 @@ static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
 #define AIC23_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops tlv320aic23_dai_ops = {
+	.prepare	= tlv320aic23_pcm_prepare,
+	.hw_params	= tlv320aic23_hw_params,
+	.shutdown	= tlv320aic23_shutdown,
+	.digital_mute	= tlv320aic23_mute,
+	.set_fmt	= tlv320aic23_set_dai_fmt,
+	.set_sysclk	= tlv320aic23_set_dai_sysclk,
+};
+
 struct snd_soc_dai tlv320aic23_dai = {
 	.name = "tlv320aic23",
 	.playback = {
@@ -612,14 +603,7 @@ struct snd_soc_dai tlv320aic23_dai = {
 		    .channels_max = 2,
 		    .rates = AIC23_RATES,
 		    .formats = AIC23_FORMATS,},
-	.ops = {
-		.prepare = tlv320aic23_pcm_prepare,
-		.hw_params = tlv320aic23_hw_params,
-		.shutdown = tlv320aic23_shutdown,
-		.digital_mute = tlv320aic23_mute,
-		.set_fmt = tlv320aic23_set_dai_fmt,
-		.set_sysclk = tlv320aic23_set_dai_sysclk,
-	}
+	.ops = &tlv320aic23_dai_ops,
 };
 EXPORT_SYMBOL_GPL(tlv320aic23_dai);
 
@@ -627,7 +611,7 @@ static int tlv320aic23_suspend(struct platform_device *pdev,
 			       pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x0);
 	tlv320aic23_set_bias_level(codec, SND_SOC_BIAS_OFF);
@@ -638,7 +622,7 @@ static int tlv320aic23_suspend(struct platform_device *pdev,
 static int tlv320aic23_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u16 reg;
 
@@ -660,7 +644,7 @@ static int tlv320aic23_resume(struct platform_device *pdev)
  */
 static int tlv320aic23_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 	u16 reg;
 
@@ -718,7 +702,8 @@ static int tlv320aic23_init(struct snd_soc_device *socdev)
 
 	tlv320aic23_write(codec, TLV320AIC23_ACTIVE, 0x1);
 
-	tlv320aic23_add_controls(codec);
+	snd_soc_add_controls(codec, tlv320aic23_snd_controls,
+				ARRAY_SIZE(tlv320aic23_snd_controls));
 	tlv320aic23_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -746,7 +731,7 @@ static int tlv320aic23_codec_probe(struct i2c_client *i2c,
 				   const struct i2c_device_id *i2c_id)
 {
 	struct snd_soc_device *socdev = tlv320aic23_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA))
@@ -804,7 +789,7 @@ static int tlv320aic23_probe(struct platform_device *pdev)
 	if (aic23 == NULL)
 		return -ENOMEM;
 	codec = &aic23->codec;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -823,7 +808,7 @@ static int tlv320aic23_probe(struct platform_device *pdev)
 static int tlv320aic23_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct aic23 *aic23 = container_of(codec, struct aic23, codec);
 
 	if (codec->control_data)
diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c
index 29f2f1a017fd..3387d9e736ea 100644
--- a/sound/soc/codecs/tlv320aic26.c
+++ b/sound/soc/codecs/tlv320aic26.c
@@ -130,7 +130,7 @@ static int aic26_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct aic26 *aic26 = codec->private_data;
 	int fsref, divisor, wlen, pval, jval, dval, qval;
 	u16 reg;
@@ -270,6 +270,13 @@ static int aic26_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
 #define AIC26_FORMATS	(SNDRV_PCM_FMTBIT_S8     | SNDRV_PCM_FMTBIT_S16_BE |\
 			 SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE)
 
+static struct snd_soc_dai_ops aic26_dai_ops = {
+	.hw_params	= aic26_hw_params,
+	.digital_mute	= aic26_mute,
+	.set_sysclk	= aic26_set_sysclk,
+	.set_fmt	= aic26_set_fmt,
+};
+
 struct snd_soc_dai aic26_dai = {
 	.name = "tlv320aic26",
 	.playback = {
@@ -286,12 +293,7 @@ struct snd_soc_dai aic26_dai = {
 		.rates = AIC26_RATES,
 		.formats = AIC26_FORMATS,
 	},
-	.ops = {
-		.hw_params = aic26_hw_params,
-		.digital_mute = aic26_mute,
-		.set_sysclk = aic26_set_sysclk,
-		.set_fmt = aic26_set_fmt,
-	},
+	.ops = &aic26_dai_ops,
 };
 EXPORT_SYMBOL_GPL(aic26_dai);
 
@@ -322,9 +324,8 @@ static int aic26_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec;
-	struct snd_kcontrol *kcontrol;
 	struct aic26 *aic26;
-	int i, ret, err;
+	int ret, err;
 
 	dev_info(&pdev->dev, "Probing AIC26 SoC CODEC driver\n");
 	dev_dbg(&pdev->dev, "socdev=%p\n", socdev);
@@ -338,7 +339,7 @@ static int aic26_probe(struct platform_device *pdev)
 		return -ENODEV;
 	}
 	codec = &aic26->codec;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 
 	dev_dbg(&pdev->dev, "Registering PCMs, dev=%p, socdev->dev=%p\n",
 		&pdev->dev, socdev->dev);
@@ -351,11 +352,9 @@ static int aic26_probe(struct platform_device *pdev)
 
 	/* register controls */
 	dev_dbg(&pdev->dev, "Registering controls\n");
-	for (i = 0; i < ARRAY_SIZE(aic26_snd_controls); i++) {
-		kcontrol = snd_soc_cnew(&aic26_snd_controls[i], codec, NULL);
-		err = snd_ctl_add(codec->card, kcontrol);
-		WARN_ON(err < 0);
-	}
+	err = snd_soc_add_controls(codec, aic26_snd_controls,
+			ARRAY_SIZE(aic26_snd_controls));
+	WARN_ON(err < 0);
 
 	/* CODEC is setup, we can register the card now */
 	dev_dbg(&pdev->dev, "Registering card\n");
diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c
index aea0cb72d80a..ab099f482487 100644
--- a/sound/soc/codecs/tlv320aic3x.c
+++ b/sound/soc/codecs/tlv320aic3x.c
@@ -45,6 +45,7 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 #include <sound/initval.h>
+#include <sound/tlv.h>
 
 #include "tlv320aic3x.h"
 
@@ -250,56 +251,86 @@ static const struct soc_enum aic3x_enum[] = {
 	SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf),
 };
 
+/*
+ * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps
+ */
+static DECLARE_TLV_DB_SCALE(dac_tlv, -6350, 50, 0);
+/* ADC PGA gain volumes. From 0 to 59.5 dB in 0.5 dB steps */
+static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 50, 0);
+/*
+ * Output stage volumes. From -78.3 to 0 dB. Muted below -78.3 dB.
+ * Step size is approximately 0.5 dB over most of the scale but increasing
+ * near the very low levels.
+ * Define dB scale so that it is mostly correct for range about -55 to 0 dB
+ * but having increasing dB difference below that (and where it doesn't count
+ * so much). This setting shows -50 dB (actual is -50.3 dB) for register
+ * value 100 and -58.5 dB (actual is -78.3 dB) for register value 117.
+ */
+static DECLARE_TLV_DB_SCALE(output_stage_tlv, -5900, 50, 1);
+
 static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 	/* Output */
-	SOC_DOUBLE_R("PCM Playback Volume", LDAC_VOL, RDAC_VOL, 0, 0x7f, 1),
+	SOC_DOUBLE_R_TLV("PCM Playback Volume",
+			 LDAC_VOL, RDAC_VOL, 0, 0x7f, 1, dac_tlv),
 
-	SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL,
-		     DACR1_2_RLOPM_VOL, 0, 0x7f, 1),
+	SOC_DOUBLE_R_TLV("Line DAC Playback Volume",
+			 DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
 	SOC_SINGLE("LineL Playback Switch", LLOPM_CTRL, 3, 0x01, 0),
 	SOC_SINGLE("LineR Playback Switch", RLOPM_CTRL, 3, 0x01, 0),
-	SOC_DOUBLE_R("LineL DAC Playback Volume", DACL1_2_LLOPM_VOL,
-		     DACR1_2_LLOPM_VOL, 0, 0x7f, 1),
-	SOC_SINGLE("LineL Left PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL,
-		     0, 0x7f, 1),
-	SOC_SINGLE("LineR Right PGA Bypass Playback Volume", PGAR_2_RLOPM_VOL,
-		     0, 0x7f, 1),
-	SOC_DOUBLE_R("LineL Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL,
-		     LINE2R_2_LLOPM_VOL, 0, 0x7f, 1),
-	SOC_DOUBLE_R("LineR Line2 Bypass Playback Volume", LINE2L_2_RLOPM_VOL,
-		     LINE2R_2_RLOPM_VOL, 0, 0x7f, 1),
-
-	SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL,
-		     DACR1_2_MONOLOPM_VOL, 0, 0x7f, 1),
+	SOC_DOUBLE_R_TLV("LineL DAC Playback Volume",
+			 DACL1_2_LLOPM_VOL, DACR1_2_LLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
+	SOC_SINGLE_TLV("LineL Left PGA Bypass Playback Volume",
+		       PGAL_2_LLOPM_VOL, 0, 118, 1, output_stage_tlv),
+	SOC_SINGLE_TLV("LineR Right PGA Bypass Playback Volume",
+		       PGAR_2_RLOPM_VOL, 0, 118, 1, output_stage_tlv),
+	SOC_DOUBLE_R_TLV("LineL Line2 Bypass Playback Volume",
+			 LINE2L_2_LLOPM_VOL, LINE2R_2_LLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
+	SOC_DOUBLE_R_TLV("LineR Line2 Bypass Playback Volume",
+			 LINE2L_2_RLOPM_VOL, LINE2R_2_RLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
+
+	SOC_DOUBLE_R_TLV("Mono DAC Playback Volume",
+			 DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
 	SOC_SINGLE("Mono DAC Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0),
-	SOC_DOUBLE_R("Mono PGA Bypass Playback Volume", PGAL_2_MONOLOPM_VOL,
-		     PGAR_2_MONOLOPM_VOL, 0, 0x7f, 1),
-	SOC_DOUBLE_R("Mono Line2 Bypass Playback Volume", LINE2L_2_MONOLOPM_VOL,
-		     LINE2R_2_MONOLOPM_VOL, 0, 0x7f, 1),
-
-	SOC_DOUBLE_R("HP DAC Playback Volume", DACL1_2_HPLOUT_VOL,
-		     DACR1_2_HPROUT_VOL, 0, 0x7f, 1),
+	SOC_DOUBLE_R_TLV("Mono PGA Bypass Playback Volume",
+			 PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
+	SOC_DOUBLE_R_TLV("Mono Line2 Bypass Playback Volume",
+			 LINE2L_2_MONOLOPM_VOL, LINE2R_2_MONOLOPM_VOL,
+			 0, 118, 1, output_stage_tlv),
+
+	SOC_DOUBLE_R_TLV("HP DAC Playback Volume",
+			 DACL1_2_HPLOUT_VOL, DACR1_2_HPROUT_VOL,
+			 0, 118, 1, output_stage_tlv),
 	SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3,
 		     0x01, 0),
-	SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL,
-		     PGAR_2_HPROUT_VOL, 0, 0x7f, 1),
-	SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL,
-		     0, 0x7f, 1),
-	SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL,
-		     0, 0x7f, 1),
-	SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL,
-		     LINE2R_2_HPROUT_VOL, 0, 0x7f, 1),
-
-	SOC_DOUBLE_R("HPCOM DAC Playback Volume", DACL1_2_HPLCOM_VOL,
-		     DACR1_2_HPRCOM_VOL, 0, 0x7f, 1),
+	SOC_DOUBLE_R_TLV("HP Right PGA Bypass Playback Volume",
+			 PGAR_2_HPLOUT_VOL, PGAR_2_HPROUT_VOL,
+			 0, 118, 1, output_stage_tlv),
+	SOC_SINGLE_TLV("HPL PGA Bypass Playback Volume",
+		       PGAL_2_HPLOUT_VOL, 0, 118, 1, output_stage_tlv),
+	SOC_SINGLE_TLV("HPR PGA Bypass Playback Volume",
+		       PGAL_2_HPROUT_VOL, 0, 118, 1, output_stage_tlv),
+	SOC_DOUBLE_R_TLV("HP Line2 Bypass Playback Volume",
+			 LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL,
+			 0, 118, 1, output_stage_tlv),
+
+	SOC_DOUBLE_R_TLV("HPCOM DAC Playback Volume",
+			 DACL1_2_HPLCOM_VOL, DACR1_2_HPRCOM_VOL,
+			 0, 118, 1, output_stage_tlv),
 	SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3,
 		     0x01, 0),
-	SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL,
-		     0, 0x7f, 1),
-	SOC_SINGLE("HPRCOM PGA Bypass Playback Volume", PGAL_2_HPRCOM_VOL,
-		     0, 0x7f, 1),
-	SOC_DOUBLE_R("HPCOM Line2 Bypass Playback Volume", LINE2L_2_HPLCOM_VOL,
-		     LINE2R_2_HPRCOM_VOL, 0, 0x7f, 1),
+	SOC_SINGLE_TLV("HPLCOM PGA Bypass Playback Volume",
+		       PGAL_2_HPLCOM_VOL, 0, 118, 1, output_stage_tlv),
+	SOC_SINGLE_TLV("HPRCOM PGA Bypass Playback Volume",
+		       PGAL_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv),
+	SOC_DOUBLE_R_TLV("HPCOM Line2 Bypass Playback Volume",
+			 LINE2L_2_HPLCOM_VOL, LINE2R_2_HPRCOM_VOL,
+			 0, 118, 1, output_stage_tlv),
 
 	/*
 	 * Note: enable Automatic input Gain Controller with care. It can
@@ -308,28 +339,13 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = {
 	SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0),
 
 	/* Input */
-	SOC_DOUBLE_R("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 0x7f, 0),
+	SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL,
+			 0, 119, 0, adc_tlv),
 	SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1),
 
 	SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]),
 };
 
-/* add non dapm controls */
-static int aic3x_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(aic3x_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&aic3x_snd_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /* Left DAC Mux */
 static const struct snd_kcontrol_new aic3x_left_dac_mux_controls =
 SOC_DAPM_ENUM("Route", aic3x_enum[LDAC_ENUM]);
@@ -746,7 +762,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct aic3x_priv *aic3x = codec->private_data;
 	int codec_clk = 0, bypass_pll = 0, fsref, last_clk = 0;
 	u8 data, r, p, pll_q, pll_p = 1, pll_r = 1, pll_j = 1;
@@ -1072,6 +1088,13 @@ EXPORT_SYMBOL_GPL(aic3x_button_pressed);
 #define AIC3X_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
 			 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops aic3x_dai_ops = {
+	.hw_params	= aic3x_hw_params,
+	.digital_mute	= aic3x_mute,
+	.set_sysclk	= aic3x_set_dai_sysclk,
+	.set_fmt	= aic3x_set_dai_fmt,
+};
+
 struct snd_soc_dai aic3x_dai = {
 	.name = "tlv320aic3x",
 	.playback = {
@@ -1086,19 +1109,14 @@ struct snd_soc_dai aic3x_dai = {
 		.channels_max = 2,
 		.rates = AIC3X_RATES,
 		.formats = AIC3X_FORMATS,},
-	.ops = {
-		.hw_params = aic3x_hw_params,
-		.digital_mute = aic3x_mute,
-		.set_sysclk = aic3x_set_dai_sysclk,
-		.set_fmt = aic3x_set_dai_fmt,
-	}
+	.ops = &aic3x_dai_ops,
 };
 EXPORT_SYMBOL_GPL(aic3x_dai);
 
 static int aic3x_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	aic3x_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
@@ -1108,7 +1126,7 @@ static int aic3x_suspend(struct platform_device *pdev, pm_message_t state)
 static int aic3x_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u8 *cache = codec->reg_cache;
@@ -1131,7 +1149,7 @@ static int aic3x_resume(struct platform_device *pdev)
  */
 static int aic3x_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct aic3x_setup_data *setup = socdev->codec_data;
 	int reg, ret = 0;
 
@@ -1227,7 +1245,8 @@ static int aic3x_init(struct snd_soc_device *socdev)
 	aic3x_write(codec, AIC3X_GPIO1_REG, (setup->gpio_func[0] & 0xf) << 4);
 	aic3x_write(codec, AIC3X_GPIO2_REG, (setup->gpio_func[1] & 0xf) << 4);
 
-	aic3x_add_controls(codec);
+	snd_soc_add_controls(codec, aic3x_snd_controls,
+				ARRAY_SIZE(aic3x_snd_controls));
 	aic3x_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -1261,7 +1280,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c,
 			   const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = aic3x_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -1366,7 +1385,7 @@ static int aic3x_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = aic3x;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -1392,7 +1411,7 @@ static int aic3x_probe(struct platform_device *pdev)
 static int aic3x_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	/* power down chip */
 	if (codec->control_data)
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c
index ea370a4f86d5..97738e2ece04 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -42,7 +42,7 @@
  */
 static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
 	0x00, /* this register not used		*/
-	0x93, /* REG_CODEC_MODE		(0x1)	*/
+	0x91, /* REG_CODEC_MODE		(0x1)	*/
 	0xc3, /* REG_OPTION		(0x2)	*/
 	0x00, /* REG_UNKNOWN		(0x3)	*/
 	0x00, /* REG_MICBIAS_CTL	(0x4)	*/
@@ -117,6 +117,13 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = {
 	0x00, /* REG_MISC_SET_2		(0x49)	*/
 };
 
+/* codec private data */
+struct twl4030_priv {
+	unsigned int bypass_state;
+	unsigned int codec_powered;
+	unsigned int codec_muted;
+};
+
 /*
  * read twl4030 register cache
  */
@@ -125,6 +132,9 @@ static inline unsigned int twl4030_read_reg_cache(struct snd_soc_codec *codec,
 {
 	u8 *cache = codec->reg_cache;
 
+	if (reg >= TWL4030_CACHEREGNUM)
+		return -EIO;
+
 	return cache[reg];
 }
 
@@ -151,26 +161,22 @@ static int twl4030_write(struct snd_soc_codec *codec,
 	return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg);
 }
 
-static void twl4030_clear_codecpdz(struct snd_soc_codec *codec)
+static void twl4030_codec_enable(struct snd_soc_codec *codec, int enable)
 {
+	struct twl4030_priv *twl4030 = codec->private_data;
 	u8 mode;
 
-	mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
-	twl4030_write(codec, TWL4030_REG_CODEC_MODE,
-		mode & ~TWL4030_CODECPDZ);
-
-	/* REVISIT: this delay is present in TI sample drivers */
-	/* but there seems to be no TRM requirement for it     */
-	udelay(10);
-}
-
-static void twl4030_set_codecpdz(struct snd_soc_codec *codec)
-{
-	u8 mode;
+	if (enable == twl4030->codec_powered)
+		return;
 
 	mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE);
-	twl4030_write(codec, TWL4030_REG_CODEC_MODE,
-		mode | TWL4030_CODECPDZ);
+	if (enable)
+		mode |= TWL4030_CODECPDZ;
+	else
+		mode &= ~TWL4030_CODECPDZ;
+
+	twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
+	twl4030->codec_powered = enable;
 
 	/* REVISIT: this delay is present in TI sample drivers */
 	/* but there seems to be no TRM requirement for it     */
@@ -182,7 +188,7 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 	int i;
 
 	/* clear CODECPDZ prior to setting register defaults */
-	twl4030_clear_codecpdz(codec);
+	twl4030_codec_enable(codec, 0);
 
 	/* set all audio section registers to reasonable defaults */
 	for (i = TWL4030_REG_OPTION; i <= TWL4030_REG_MISC_SET_2; i++)
@@ -190,6 +196,122 @@ static void twl4030_init_chip(struct snd_soc_codec *codec)
 
 }
 
+static void twl4030_codec_mute(struct snd_soc_codec *codec, int mute)
+{
+	struct twl4030_priv *twl4030 = codec->private_data;
+	u8 reg_val;
+
+	if (mute == twl4030->codec_muted)
+		return;
+
+	if (mute) {
+		/* Bypass the reg_cache and mute the volumes
+		 * Headset mute is done in it's own event handler
+		 * Things to mute:  Earpiece, PreDrivL/R, CarkitL/R
+		 */
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_EAR_CTL);
+		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					reg_val & (~TWL4030_EAR_GAIN),
+					TWL4030_REG_EAR_CTL);
+
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PREDL_CTL);
+		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					reg_val & (~TWL4030_PREDL_GAIN),
+					TWL4030_REG_PREDL_CTL);
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PREDR_CTL);
+		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					reg_val & (~TWL4030_PREDR_GAIN),
+					TWL4030_REG_PREDL_CTL);
+
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKL_CTL);
+		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					reg_val & (~TWL4030_PRECKL_GAIN),
+					TWL4030_REG_PRECKL_CTL);
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL);
+		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					reg_val & (~TWL4030_PRECKL_GAIN),
+					TWL4030_REG_PRECKR_CTL);
+
+		/* Disable PLL */
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL);
+		reg_val &= ~TWL4030_APLL_EN;
+		twl4030_write(codec, TWL4030_REG_APLL_CTL, reg_val);
+	} else {
+		/* Restore the volumes
+		 * Headset mute is done in it's own event handler
+		 * Things to restore:  Earpiece, PreDrivL/R, CarkitL/R
+		 */
+		twl4030_write(codec, TWL4030_REG_EAR_CTL,
+			twl4030_read_reg_cache(codec, TWL4030_REG_EAR_CTL));
+
+		twl4030_write(codec, TWL4030_REG_PREDL_CTL,
+			twl4030_read_reg_cache(codec, TWL4030_REG_PREDL_CTL));
+		twl4030_write(codec, TWL4030_REG_PREDR_CTL,
+			twl4030_read_reg_cache(codec, TWL4030_REG_PREDR_CTL));
+
+		twl4030_write(codec, TWL4030_REG_PRECKL_CTL,
+			twl4030_read_reg_cache(codec, TWL4030_REG_PRECKL_CTL));
+		twl4030_write(codec, TWL4030_REG_PRECKR_CTL,
+			twl4030_read_reg_cache(codec, TWL4030_REG_PRECKR_CTL));
+
+		/* Enable PLL */
+		reg_val = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL);
+		reg_val |= TWL4030_APLL_EN;
+		twl4030_write(codec, TWL4030_REG_APLL_CTL, reg_val);
+	}
+
+	twl4030->codec_muted = mute;
+}
+
+static void twl4030_power_up(struct snd_soc_codec *codec)
+{
+	struct twl4030_priv *twl4030 = codec->private_data;
+	u8 anamicl, regmisc1, byte;
+	int i = 0;
+
+	if (twl4030->codec_powered)
+		return;
+
+	/* set CODECPDZ to turn on codec */
+	twl4030_codec_enable(codec, 1);
+
+	/* initiate offset cancellation */
+	anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
+	twl4030_write(codec, TWL4030_REG_ANAMICL,
+		anamicl | TWL4030_CNCL_OFFSET_START);
+
+	/* wait for offset cancellation to complete */
+	do {
+		/* this takes a little while, so don't slam i2c */
+		udelay(2000);
+		twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
+				    TWL4030_REG_ANAMICL);
+	} while ((i++ < 100) &&
+		 ((byte & TWL4030_CNCL_OFFSET_START) ==
+		  TWL4030_CNCL_OFFSET_START));
+
+	/* Make sure that the reg_cache has the same value as the HW */
+	twl4030_write_reg_cache(codec, TWL4030_REG_ANAMICL, byte);
+
+	/* anti-pop when changing analog gain */
+	regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1);
+	twl4030_write(codec, TWL4030_REG_MISC_SET_1,
+		regmisc1 | TWL4030_SMOOTH_ANAVOL_EN);
+
+	/* toggle CODECPDZ as per TRM */
+	twl4030_codec_enable(codec, 0);
+	twl4030_codec_enable(codec, 1);
+}
+
+/*
+ * Unconditional power down
+ */
+static void twl4030_power_down(struct snd_soc_codec *codec)
+{
+	/* power down */
+	twl4030_codec_enable(codec, 0);
+}
+
 /* Earpiece */
 static const char *twl4030_earpiece_texts[] =
 		{"Off", "DACL1", "DACL2", "DACR1"};
@@ -366,6 +488,41 @@ static const struct soc_enum twl4030_micpathtx2_enum =
 static const struct snd_kcontrol_new twl4030_dapm_micpathtx2_control =
 SOC_DAPM_ENUM("Route", twl4030_micpathtx2_enum);
 
+/* Analog bypass for AudioR1 */
+static const struct snd_kcontrol_new twl4030_dapm_abypassr1_control =
+	SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXR1_APGA_CTL, 2, 1, 0);
+
+/* Analog bypass for AudioL1 */
+static const struct snd_kcontrol_new twl4030_dapm_abypassl1_control =
+	SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL1_APGA_CTL, 2, 1, 0);
+
+/* Analog bypass for AudioR2 */
+static const struct snd_kcontrol_new twl4030_dapm_abypassr2_control =
+	SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXR2_APGA_CTL, 2, 1, 0);
+
+/* Analog bypass for AudioL2 */
+static const struct snd_kcontrol_new twl4030_dapm_abypassl2_control =
+	SOC_DAPM_SINGLE("Switch", TWL4030_REG_ARXL2_APGA_CTL, 2, 1, 0);
+
+/* Digital bypass gain, 0 mutes the bypass */
+static const unsigned int twl4030_dapm_dbypass_tlv[] = {
+	TLV_DB_RANGE_HEAD(2),
+	0, 3, TLV_DB_SCALE_ITEM(-2400, 0, 1),
+	4, 7, TLV_DB_SCALE_ITEM(-1800, 600, 0),
+};
+
+/* Digital bypass left (TX1L -> RX2L) */
+static const struct snd_kcontrol_new twl4030_dapm_dbypassl_control =
+	SOC_DAPM_SINGLE_TLV("Volume",
+			TWL4030_REG_ATX2ARXPGA, 3, 7, 0,
+			twl4030_dapm_dbypass_tlv);
+
+/* Digital bypass right (TX1R -> RX2R) */
+static const struct snd_kcontrol_new twl4030_dapm_dbypassr_control =
+	SOC_DAPM_SINGLE_TLV("Volume",
+			TWL4030_REG_ATX2ARXPGA, 0, 7, 0,
+			twl4030_dapm_dbypass_tlv);
+
 static int micpath_event(struct snd_soc_dapm_widget *w,
 	struct snd_kcontrol *kcontrol, int event)
 {
@@ -420,6 +577,79 @@ static int handsfree_event(struct snd_soc_dapm_widget *w,
 	return 0;
 }
 
+static int headsetl_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	unsigned char hs_gain, hs_pop;
+
+	/* Save the current volume */
+	hs_gain = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_GAIN_SET);
+	hs_pop = twl4030_read_reg_cache(w->codec, TWL4030_REG_HS_POPN_SET);
+
+	switch (event) {
+	case SND_SOC_DAPM_POST_PMU:
+		/* Do the anti-pop/bias ramp enable according to the TRM */
+		hs_pop |= TWL4030_VMID_EN;
+		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+		/* Is this needed? Can we just use whatever gain here? */
+		twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET,
+				(hs_gain & (~0x0f)) | 0x0a);
+		hs_pop |= TWL4030_RAMP_EN;
+		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+
+		/* Restore the original volume */
+		twl4030_write(w->codec, TWL4030_REG_HS_GAIN_SET, hs_gain);
+		break;
+	case SND_SOC_DAPM_POST_PMD:
+		/* Do the anti-pop/bias ramp disable according to the TRM */
+		hs_pop &= ~TWL4030_RAMP_EN;
+		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+		/* Bypass the reg_cache to mute the headset */
+		twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE,
+					hs_gain & (~0x0f),
+					TWL4030_REG_HS_GAIN_SET);
+		hs_pop &= ~TWL4030_VMID_EN;
+		twl4030_write(w->codec, TWL4030_REG_HS_POPN_SET, hs_pop);
+		break;
+	}
+	return 0;
+}
+
+static int bypass_event(struct snd_soc_dapm_widget *w,
+		struct snd_kcontrol *kcontrol, int event)
+{
+	struct soc_mixer_control *m =
+		(struct soc_mixer_control *)w->kcontrols->private_value;
+	struct twl4030_priv *twl4030 = w->codec->private_data;
+	unsigned char reg;
+
+	reg = twl4030_read_reg_cache(w->codec, m->reg);
+
+	if (m->reg <= TWL4030_REG_ARXR2_APGA_CTL) {
+		/* Analog bypass */
+		if (reg & (1 << m->shift))
+			twl4030->bypass_state |=
+				(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL));
+		else
+			twl4030->bypass_state &=
+				~(1 << (m->reg - TWL4030_REG_ARXL1_APGA_CTL));
+	} else {
+		/* Digital bypass */
+		if (reg & (0x7 << m->shift))
+			twl4030->bypass_state |= (1 << (m->shift ? 5 : 4));
+		else
+			twl4030->bypass_state &= ~(1 << (m->shift ? 5 : 4));
+	}
+
+	if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) {
+		if (twl4030->bypass_state)
+			twl4030_codec_mute(w->codec, 0);
+		else
+			twl4030_codec_mute(w->codec, 1);
+	}
+	return 0;
+}
+
 /*
  * Some of the gain controls in TWL (mostly those which are associated with
  * the outputs) are implemented in an interesting way:
@@ -614,6 +844,17 @@ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0);
  */
 static DECLARE_TLV_DB_SCALE(input_gain_tlv, 0, 600, 0);
 
+static const char *twl4030_rampdelay_texts[] = {
+	"27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms",
+	"437/323/218 ms", "874/645/437 ms", "1748/1291/874 ms",
+	"3495/2581/1748 ms"
+};
+
+static const struct soc_enum twl4030_rampdelay_enum =
+	SOC_ENUM_SINGLE(TWL4030_REG_HS_POPN_SET, 2,
+			ARRAY_SIZE(twl4030_rampdelay_texts),
+			twl4030_rampdelay_texts);
+
 static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 	/* Common playback gain controls */
 	SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume",
@@ -668,23 +909,9 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 
 	SOC_DOUBLE_TLV("Analog Capture Volume", TWL4030_REG_ANAMIC_GAIN,
 		0, 3, 5, 0, input_gain_tlv),
-};
-
-/* add non dapm controls */
-static int twl4030_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(twl4030_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&twl4030_snd_controls[i],
-						codec, NULL));
-		if (err < 0)
-			return err;
-	}
 
-	return 0;
-}
+	SOC_ENUM("HS ramp delay", twl4030_rampdelay_enum),
+};
 
 static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	/* Left channel inputs */
@@ -714,13 +941,13 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 
 	/* DACs */
 	SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback",
-			TWL4030_REG_AVDAC_CTL, 0, 0),
+			SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback",
-			TWL4030_REG_AVDAC_CTL, 1, 0),
+			SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_DAC("DAC Right2", "Right Rear Playback",
-			TWL4030_REG_AVDAC_CTL, 2, 0),
+			SND_SOC_NOPM, 0, 0),
 	SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback",
-			TWL4030_REG_AVDAC_CTL, 3, 0),
+			SND_SOC_NOPM, 0, 0),
 
 	/* Analog PGAs */
 	SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL,
@@ -732,6 +959,37 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_PGA("ARXL2_APGA", TWL4030_REG_ARXL2_APGA_CTL,
 			0, 0, NULL, 0),
 
+	/* Analog bypasses */
+	SND_SOC_DAPM_SWITCH_E("Right1 Analog Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_abypassr1_control, bypass_event,
+			SND_SOC_DAPM_POST_REG),
+	SND_SOC_DAPM_SWITCH_E("Left1 Analog Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_abypassl1_control,
+			bypass_event, SND_SOC_DAPM_POST_REG),
+	SND_SOC_DAPM_SWITCH_E("Right2 Analog Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_abypassr2_control,
+			bypass_event, SND_SOC_DAPM_POST_REG),
+	SND_SOC_DAPM_SWITCH_E("Left2 Analog Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_abypassl2_control,
+			bypass_event, SND_SOC_DAPM_POST_REG),
+
+	/* Digital bypasses */
+	SND_SOC_DAPM_SWITCH_E("Left Digital Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_dbypassl_control, bypass_event,
+			SND_SOC_DAPM_POST_REG),
+	SND_SOC_DAPM_SWITCH_E("Right Digital Loopback", SND_SOC_NOPM, 0, 0,
+			&twl4030_dapm_dbypassr_control, bypass_event,
+			SND_SOC_DAPM_POST_REG),
+
+	SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL,
+			0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer", TWL4030_REG_AVDAC_CTL,
+			1, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer", TWL4030_REG_AVDAC_CTL,
+			2, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL,
+			3, 0, NULL, 0),
+
 	/* Output MUX controls */
 	/* Earpiece */
 	SND_SOC_DAPM_VALUE_MUX("Earpiece Mux", SND_SOC_NOPM, 0, 0,
@@ -742,8 +1000,9 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_VALUE_MUX("PredriveR Mux", SND_SOC_NOPM, 0, 0,
 		&twl4030_dapm_predriver_control),
 	/* HeadsetL/R */
-	SND_SOC_DAPM_MUX("HeadsetL Mux", SND_SOC_NOPM, 0, 0,
-		&twl4030_dapm_hsol_control),
+	SND_SOC_DAPM_MUX_E("HeadsetL Mux", SND_SOC_NOPM, 0, 0,
+		&twl4030_dapm_hsol_control, headsetl_event,
+		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD),
 	SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0,
 		&twl4030_dapm_hsor_control),
 	/* CarkitL/R */
@@ -782,16 +1041,16 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD|
 		SND_SOC_DAPM_POST_REG),
 
-	/* Analog input muxes with power switch for the physical ADCL/R */
+	/* Analog input muxes with switch for the capture amplifiers */
 	SND_SOC_DAPM_VALUE_MUX("Analog Left Capture Route",
-		TWL4030_REG_AVADC_CTL, 3, 0, &twl4030_dapm_analoglmic_control),
+		TWL4030_REG_ANAMICL, 4, 0, &twl4030_dapm_analoglmic_control),
 	SND_SOC_DAPM_VALUE_MUX("Analog Right Capture Route",
-		TWL4030_REG_AVADC_CTL, 1, 0, &twl4030_dapm_analogrmic_control),
+		TWL4030_REG_ANAMICR, 4, 0, &twl4030_dapm_analogrmic_control),
 
-	SND_SOC_DAPM_PGA("Analog Left Amplifier",
-		TWL4030_REG_ANAMICL, 4, 0, NULL, 0),
-	SND_SOC_DAPM_PGA("Analog Right Amplifier",
-		TWL4030_REG_ANAMICR, 4, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("ADC Physical Left",
+		TWL4030_REG_AVADC_CTL, 3, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("ADC Physical Right",
+		TWL4030_REG_AVADC_CTL, 1, 0, NULL, 0),
 
 	SND_SOC_DAPM_PGA("Digimic0 Enable",
 		TWL4030_REG_ADCMICSEL, 1, 0, NULL, 0),
@@ -801,13 +1060,19 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 	SND_SOC_DAPM_MICBIAS("Mic Bias 1", TWL4030_REG_MICBIAS_CTL, 0, 0),
 	SND_SOC_DAPM_MICBIAS("Mic Bias 2", TWL4030_REG_MICBIAS_CTL, 1, 0),
 	SND_SOC_DAPM_MICBIAS("Headset Mic Bias", TWL4030_REG_MICBIAS_CTL, 2, 0),
+
 };
 
 static const struct snd_soc_dapm_route intercon[] = {
-	{"ARXL1_APGA", NULL, "DAC Left1"},
-	{"ARXR1_APGA", NULL, "DAC Right1"},
-	{"ARXL2_APGA", NULL, "DAC Left2"},
-	{"ARXR2_APGA", NULL, "DAC Right2"},
+	{"Analog L1 Playback Mixer", NULL, "DAC Left1"},
+	{"Analog R1 Playback Mixer", NULL, "DAC Right1"},
+	{"Analog L2 Playback Mixer", NULL, "DAC Left2"},
+	{"Analog R2 Playback Mixer", NULL, "DAC Right2"},
+
+	{"ARXL1_APGA", NULL, "Analog L1 Playback Mixer"},
+	{"ARXR1_APGA", NULL, "Analog R1 Playback Mixer"},
+	{"ARXL2_APGA", NULL, "Analog L2 Playback Mixer"},
+	{"ARXR2_APGA", NULL, "Analog R2 Playback Mixer"},
 
 	/* Internal playback routings */
 	/* Earpiece */
@@ -865,23 +1130,23 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"Analog Right Capture Route", "Sub mic", "SUBMIC"},
 	{"Analog Right Capture Route", "AUXR", "AUXR"},
 
-	{"Analog Left Amplifier", NULL, "Analog Left Capture Route"},
-	{"Analog Right Amplifier", NULL, "Analog Right Capture Route"},
+	{"ADC Physical Left", NULL, "Analog Left Capture Route"},
+	{"ADC Physical Right", NULL, "Analog Right Capture Route"},
 
 	{"Digimic0 Enable", NULL, "DIGIMIC0"},
 	{"Digimic1 Enable", NULL, "DIGIMIC1"},
 
 	/* TX1 Left capture path */
-	{"TX1 Capture Route", "Analog", "Analog Left Amplifier"},
+	{"TX1 Capture Route", "Analog", "ADC Physical Left"},
 	{"TX1 Capture Route", "Digimic0", "Digimic0 Enable"},
 	/* TX1 Right capture path */
-	{"TX1 Capture Route", "Analog", "Analog Right Amplifier"},
+	{"TX1 Capture Route", "Analog", "ADC Physical Right"},
 	{"TX1 Capture Route", "Digimic0", "Digimic0 Enable"},
 	/* TX2 Left capture path */
-	{"TX2 Capture Route", "Analog", "Analog Left Amplifier"},
+	{"TX2 Capture Route", "Analog", "ADC Physical Left"},
 	{"TX2 Capture Route", "Digimic1", "Digimic1 Enable"},
 	/* TX2 Right capture path */
-	{"TX2 Capture Route", "Analog", "Analog Right Amplifier"},
+	{"TX2 Capture Route", "Analog", "ADC Physical Right"},
 	{"TX2 Capture Route", "Digimic1", "Digimic1 Enable"},
 
 	{"ADC Virtual Left1", NULL, "TX1 Capture Route"},
@@ -889,6 +1154,24 @@ static const struct snd_soc_dapm_route intercon[] = {
 	{"ADC Virtual Left2", NULL, "TX2 Capture Route"},
 	{"ADC Virtual Right2", NULL, "TX2 Capture Route"},
 
+	/* Analog bypass routes */
+	{"Right1 Analog Loopback", "Switch", "Analog Right Capture Route"},
+	{"Left1 Analog Loopback", "Switch", "Analog Left Capture Route"},
+	{"Right2 Analog Loopback", "Switch", "Analog Right Capture Route"},
+	{"Left2 Analog Loopback", "Switch", "Analog Left Capture Route"},
+
+	{"Analog R1 Playback Mixer", NULL, "Right1 Analog Loopback"},
+	{"Analog L1 Playback Mixer", NULL, "Left1 Analog Loopback"},
+	{"Analog R2 Playback Mixer", NULL, "Right2 Analog Loopback"},
+	{"Analog L2 Playback Mixer", NULL, "Left2 Analog Loopback"},
+
+	/* Digital bypass routes */
+	{"Right Digital Loopback", "Volume", "TX1 Capture Route"},
+	{"Left Digital Loopback", "Volume", "TX1 Capture Route"},
+
+	{"Analog R2 Playback Mixer", NULL, "Right Digital Loopback"},
+	{"Analog L2 Playback Mixer", NULL, "Left Digital Loopback"},
+
 };
 
 static int twl4030_add_widgets(struct snd_soc_codec *codec)
@@ -902,82 +1185,28 @@ static int twl4030_add_widgets(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static void twl4030_power_up(struct snd_soc_codec *codec)
-{
-	u8 anamicl, regmisc1, byte, popn;
-	int i = 0;
-
-	/* set CODECPDZ to turn on codec */
-	twl4030_set_codecpdz(codec);
-
-	/* initiate offset cancellation */
-	anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL);
-	twl4030_write(codec, TWL4030_REG_ANAMICL,
-		anamicl | TWL4030_CNCL_OFFSET_START);
-
-
-	/* wait for offset cancellation to complete */
-	do {
-		/* this takes a little while, so don't slam i2c */
-		udelay(2000);
-		twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte,
-				    TWL4030_REG_ANAMICL);
-	} while ((i++ < 100) &&
-		 ((byte & TWL4030_CNCL_OFFSET_START) ==
-		  TWL4030_CNCL_OFFSET_START));
-
-	/* anti-pop when changing analog gain */
-	regmisc1 = twl4030_read_reg_cache(codec, TWL4030_REG_MISC_SET_1);
-	twl4030_write(codec, TWL4030_REG_MISC_SET_1,
-		regmisc1 | TWL4030_SMOOTH_ANAVOL_EN);
-
-	/* toggle CODECPDZ as per TRM */
-	twl4030_clear_codecpdz(codec);
-	twl4030_set_codecpdz(codec);
-
-	/* program anti-pop with bias ramp delay */
-	popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
-	popn &= TWL4030_RAMP_DELAY;
-	popn |=	TWL4030_RAMP_DELAY_645MS;
-	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
-	popn |=	TWL4030_VMID_EN;
-	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
-
-	/* enable anti-pop ramp */
-	popn |= TWL4030_RAMP_EN;
-	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
-}
-
-static void twl4030_power_down(struct snd_soc_codec *codec)
-{
-	u8 popn;
-
-	/* disable anti-pop ramp */
-	popn = twl4030_read_reg_cache(codec, TWL4030_REG_HS_POPN_SET);
-	popn &= ~TWL4030_RAMP_EN;
-	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
-
-	/* disable bias out */
-	popn &= ~TWL4030_VMID_EN;
-	twl4030_write(codec, TWL4030_REG_HS_POPN_SET, popn);
-
-	/* power down */
-	twl4030_clear_codecpdz(codec);
-}
-
 static int twl4030_set_bias_level(struct snd_soc_codec *codec,
 				  enum snd_soc_bias_level level)
 {
+	struct twl4030_priv *twl4030 = codec->private_data;
+
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		twl4030_power_up(codec);
+		twl4030_codec_mute(codec, 0);
 		break;
 	case SND_SOC_BIAS_PREPARE:
-		/* TODO: develop a twl4030_prepare function */
+		twl4030_power_up(codec);
+		if (twl4030->bypass_state)
+			twl4030_codec_mute(codec, 0);
+		else
+			twl4030_codec_mute(codec, 1);
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		/* TODO: develop a twl4030_standby function */
-		twl4030_power_down(codec);
+		twl4030_power_up(codec);
+		if (twl4030->bypass_state)
+			twl4030_codec_mute(codec, 0);
+		else
+			twl4030_codec_mute(codec, 1);
 		break;
 	case SND_SOC_BIAS_OFF:
 		twl4030_power_down(codec);
@@ -994,10 +1223,9 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u8 mode, old_mode, format, old_format;
 
-
 	/* bit rate */
 	old_mode = twl4030_read_reg_cache(codec,
 			TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ;
@@ -1039,8 +1267,9 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 
 	if (mode != old_mode) {
 		/* change rate and set CODECPDZ */
+		twl4030_codec_enable(codec, 0);
 		twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);
-		twl4030_set_codecpdz(codec);
+		twl4030_codec_enable(codec, 1);
 	}
 
 	/* sample size */
@@ -1063,13 +1292,13 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 	if (format != old_format) {
 
 		/* clear CODECPDZ before changing format (codec requirement) */
-		twl4030_clear_codecpdz(codec);
+		twl4030_codec_enable(codec, 0);
 
 		/* change format */
 		twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
 
 		/* set CODECPDZ afterwards */
-		twl4030_set_codecpdz(codec);
+		twl4030_codec_enable(codec, 1);
 	}
 	return 0;
 }
@@ -1139,13 +1368,13 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	if (format != old_format) {
 
 		/* clear CODECPDZ before changing format (codec requirement) */
-		twl4030_clear_codecpdz(codec);
+		twl4030_codec_enable(codec, 0);
 
 		/* change format */
 		twl4030_write(codec, TWL4030_REG_AUDIO_IF, format);
 
 		/* set CODECPDZ afterwards */
-		twl4030_set_codecpdz(codec);
+		twl4030_codec_enable(codec, 1);
 	}
 
 	return 0;
@@ -1154,6 +1383,12 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
 #define TWL4030_RATES	 (SNDRV_PCM_RATE_8000_48000)
 #define TWL4030_FORMATS	 (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
 
+static struct snd_soc_dai_ops twl4030_dai_ops = {
+	.hw_params	= twl4030_hw_params,
+	.set_sysclk	= twl4030_set_dai_sysclk,
+	.set_fmt	= twl4030_set_dai_fmt,
+};
+
 struct snd_soc_dai twl4030_dai = {
 	.name = "twl4030",
 	.playback = {
@@ -1168,18 +1403,14 @@ struct snd_soc_dai twl4030_dai = {
 		.channels_max = 2,
 		.rates = TWL4030_RATES,
 		.formats = TWL4030_FORMATS,},
-	.ops = {
-		.hw_params = twl4030_hw_params,
-		.set_sysclk = twl4030_set_dai_sysclk,
-		.set_fmt = twl4030_set_dai_fmt,
-	}
+	.ops = &twl4030_dai_ops,
 };
 EXPORT_SYMBOL_GPL(twl4030_dai);
 
 static int twl4030_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
@@ -1189,7 +1420,7 @@ static int twl4030_suspend(struct platform_device *pdev, pm_message_t state)
 static int twl4030_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	twl4030_set_bias_level(codec, codec->suspend_bias_level);
@@ -1203,7 +1434,7 @@ static int twl4030_resume(struct platform_device *pdev)
 
 static int twl4030_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 
 	printk(KERN_INFO "TWL4030 Audio Codec init \n");
@@ -1233,7 +1464,8 @@ static int twl4030_init(struct snd_soc_device *socdev)
 	/* power on device */
 	twl4030_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	twl4030_add_controls(codec);
+	snd_soc_add_controls(codec, twl4030_snd_controls,
+				ARRAY_SIZE(twl4030_snd_controls));
 	twl4030_add_widgets(codec);
 
 	ret = snd_soc_init_card(socdev);
@@ -1258,12 +1490,20 @@ static int twl4030_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec;
+	struct twl4030_priv *twl4030;
 
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
 	if (codec == NULL)
 		return -ENOMEM;
 
-	socdev->codec = codec;
+	twl4030 = kzalloc(sizeof(struct twl4030_priv), GFP_KERNEL);
+	if (twl4030 == NULL) {
+		kfree(codec);
+		return -ENOMEM;
+	}
+
+	codec->private_data = twl4030;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -1277,11 +1517,13 @@ static int twl4030_probe(struct platform_device *pdev)
 static int twl4030_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	printk(KERN_INFO "TWL4030 Audio Codec remove\n");
+	twl4030_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
+	kfree(codec->private_data);
 	kfree(codec);
 
 	return 0;
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h
index 442e5a828617..33dbb144dad1 100644
--- a/sound/soc/codecs/twl4030.h
+++ b/sound/soc/codecs/twl4030.h
@@ -170,6 +170,9 @@
 #define TWL4030_CLK256FS_EN		0x02
 #define TWL4030_AIF_EN			0x01
 
+/* EAR_CTL (0x21) */
+#define TWL4030_EAR_GAIN		0x30
+
 /* HS_GAIN_SET (0x23) Fields */
 
 #define TWL4030_HSR_GAIN		0x0C
@@ -198,6 +201,18 @@
 #define TWL4030_RAMP_DELAY_2581MS	0x1C
 #define TWL4030_RAMP_EN			0x02
 
+/* PREDL_CTL (0x25) */
+#define TWL4030_PREDL_GAIN		0x30
+
+/* PREDR_CTL (0x26) */
+#define TWL4030_PREDR_GAIN		0x30
+
+/* PRECKL_CTL (0x27) */
+#define TWL4030_PRECKL_GAIN		0x30
+
+/* PRECKR_CTL (0x28) */
+#define TWL4030_PRECKR_GAIN		0x30
+
 /* HFL_CTL (0x29, 0x2A) Fields */
 #define TWL4030_HF_CTL_HB_EN		0x04
 #define TWL4030_HF_CTL_LOOP_EN		0x08
diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c
index a2c5064a774b..ddefb8f80145 100644
--- a/sound/soc/codecs/uda134x.c
+++ b/sound/soc/codecs/uda134x.c
@@ -173,7 +173,7 @@ static int uda134x_startup(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct uda134x_priv *uda134x = codec->private_data;
 	struct snd_pcm_runtime *master_runtime;
 
@@ -206,7 +206,7 @@ static void uda134x_shutdown(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct uda134x_priv *uda134x = codec->private_data;
 
 	if (uda134x->master_substream == substream)
@@ -221,7 +221,7 @@ static int uda134x_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct uda134x_priv *uda134x = codec->private_data;
 	u8 hw_params;
 
@@ -431,38 +431,14 @@ SOC_ENUM("PCM Playback De-emphasis", uda134x_mixer_enum[1]),
 SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0),
 };
 
-static int uda134x_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i, n;
-	const struct snd_kcontrol_new *ctrls;
-	struct uda134x_platform_data *pd = codec->control_data;
-
-	switch (pd->model) {
-	case UDA134X_UDA1340:
-	case UDA134X_UDA1344:
-		n = ARRAY_SIZE(uda1340_snd_controls);
-		ctrls = uda1340_snd_controls;
-		break;
-	case UDA134X_UDA1341:
-		n = ARRAY_SIZE(uda1341_snd_controls);
-		ctrls = uda1341_snd_controls;
-		break;
-	default:
-		printk(KERN_ERR "%s unkown codec type: %d",
-		       __func__, pd->model);
-		return -EINVAL;
-	}
-
-	for (i = 0; i < n; i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&ctrls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
+static struct snd_soc_dai_ops uda134x_dai_ops = {
+	.startup	= uda134x_startup,
+	.shutdown	= uda134x_shutdown,
+	.hw_params	= uda134x_hw_params,
+	.digital_mute	= uda134x_mute,
+	.set_sysclk	= uda134x_set_dai_sysclk,
+	.set_fmt	= uda134x_set_dai_fmt,
+};
 
 struct snd_soc_dai uda134x_dai = {
 	.name = "UDA134X",
@@ -483,14 +459,7 @@ struct snd_soc_dai uda134x_dai = {
 		.formats = UDA134X_FORMATS,
 	},
 	/* pcm operations */
-	.ops = {
-		.startup = uda134x_startup,
-		.shutdown = uda134x_shutdown,
-		.hw_params = uda134x_hw_params,
-		.digital_mute = uda134x_mute,
-		.set_sysclk = uda134x_set_dai_sysclk,
-		.set_fmt = uda134x_set_dai_fmt,
-	}
+	.ops = &uda134x_dai_ops,
 };
 EXPORT_SYMBOL(uda134x_dai);
 
@@ -525,11 +494,11 @@ static int uda134x_soc_probe(struct platform_device *pdev)
 		return -EINVAL;
 	}
 
-	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (socdev->codec == NULL)
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+	if (socdev->card->codec == NULL)
 		return ret;
 
-	codec = socdev->codec;
+	codec = socdev->card->codec;
 
 	uda134x = kzalloc(sizeof(struct uda134x_priv), GFP_KERNEL);
 	if (uda134x == NULL)
@@ -572,7 +541,22 @@ static int uda134x_soc_probe(struct platform_device *pdev)
 		goto pcm_err;
 	}
 
-	ret = uda134x_add_controls(codec);
+	switch (pd->model) {
+	case UDA134X_UDA1340:
+	case UDA134X_UDA1344:
+		ret = snd_soc_add_controls(codec, uda1340_snd_controls,
+					ARRAY_SIZE(uda1340_snd_controls));
+	break;
+	case UDA134X_UDA1341:
+		ret = snd_soc_add_controls(codec, uda1341_snd_controls,
+					ARRAY_SIZE(uda1341_snd_controls));
+	break;
+	default:
+		printk(KERN_ERR "%s unkown codec type: %d",
+			__func__, pd->model);
+	return -EINVAL;
+	}
+
 	if (ret < 0) {
 		printk(KERN_ERR "UDA134X: failed to register controls\n");
 		goto pcm_err;
@@ -602,7 +586,7 @@ priv_err:
 static int uda134x_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
@@ -622,7 +606,7 @@ static int uda134x_soc_suspend(struct platform_device *pdev,
 						pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_OFF);
@@ -632,7 +616,7 @@ static int uda134x_soc_suspend(struct platform_device *pdev,
 static int uda134x_soc_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
 	uda134x_set_bias_level(codec, SND_SOC_BIAS_ON);
diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c
index e6bf0844fbf3..5b21594e0e58 100644
--- a/sound/soc/codecs/uda1380.c
+++ b/sound/soc/codecs/uda1380.c
@@ -25,6 +25,7 @@
 #include <linux/ioctl.h>
 #include <linux/delay.h>
 #include <linux/i2c.h>
+#include <linux/workqueue.h>
 #include <sound/core.h>
 #include <sound/control.h>
 #include <sound/initval.h>
@@ -35,7 +36,8 @@
 
 #include "uda1380.h"
 
-#define UDA1380_VERSION "0.6"
+static struct work_struct uda1380_work;
+static struct snd_soc_codec *uda1380_codec;
 
 /*
  * uda1380 register cache
@@ -52,6 +54,8 @@ static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = {
 	0x0000, 0x8000, 0x0002, 0x0000,
 };
 
+static unsigned long uda1380_cache_dirty;
+
 /*
  * read uda1380 register cache
  */
@@ -73,8 +77,11 @@ static inline void uda1380_write_reg_cache(struct snd_soc_codec *codec,
 	u16 reg, unsigned int value)
 {
 	u16 *cache = codec->reg_cache;
+
 	if (reg >= UDA1380_CACHEREGNUM)
 		return;
+	if ((reg >= 0x10) && (cache[reg] != value))
+		set_bit(reg - 0x10, &uda1380_cache_dirty);
 	cache[reg] = value;
 }
 
@@ -113,6 +120,8 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg,
 					(data[0]<<8) | data[1]);
 			return -EIO;
 		}
+		if (reg >= 0x10)
+			clear_bit(reg - 0x10, &uda1380_cache_dirty);
 		return 0;
 	} else
 		return -EIO;
@@ -120,6 +129,20 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg,
 
 #define uda1380_reset(c)	uda1380_write(c, UDA1380_RESET, 0)
 
+static void uda1380_flush_work(struct work_struct *work)
+{
+	int bit, reg;
+
+	for_each_bit(bit, &uda1380_cache_dirty, UDA1380_CACHEREGNUM - 0x10) {
+		reg = 0x10 + bit;
+		pr_debug("uda1380: flush reg %x val %x:\n", reg,
+				uda1380_read_reg_cache(uda1380_codec, reg));
+		uda1380_write(uda1380_codec, reg,
+				uda1380_read_reg_cache(uda1380_codec, reg));
+		clear_bit(bit, &uda1380_cache_dirty);
+	}
+}
+
 /* declarations of ALSA reg_elem_REAL controls */
 static const char *uda1380_deemp[] = {
 	"None",
@@ -254,7 +277,6 @@ static const struct snd_kcontrol_new uda1380_snd_controls[] = {
 	SOC_SINGLE("DAC Polarity inverting Switch", UDA1380_MIXER, 15, 1, 0),	/* DA_POL_INV */
 	SOC_ENUM("Noise Shaper", uda1380_sel_ns_enum),				/* SEL_NS */
 	SOC_ENUM("Digital Mixer Signal Control", uda1380_mix_enum),		/* MIX_POS, MIX */
-	SOC_SINGLE("Silence Switch", UDA1380_MIXER, 7, 1, 0),			/* SILENCE, force DAC output to silence */
 	SOC_SINGLE("Silence Detector Switch", UDA1380_MIXER, 6, 1, 0),		/* SDET_ON */
 	SOC_ENUM("Silence Detector Setting", uda1380_sdet_enum),		/* SD_VALUE */
 	SOC_ENUM("Oversampling Input", uda1380_os_enum),			/* OS */
@@ -271,21 +293,6 @@ static const struct snd_kcontrol_new uda1380_snd_controls[] = {
 	SOC_SINGLE("AGC Switch", UDA1380_AGC, 0, 1, 0),
 };
 
-/* add non dapm controls */
-static int uda1380_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(uda1380_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&uda1380_snd_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /* Input mux */
 static const struct snd_kcontrol_new uda1380_input_mux_control =
 	SOC_DAPM_ENUM("Route", uda1380_input_sel_enum);
@@ -371,7 +378,7 @@ static int uda1380_add_widgets(struct snd_soc_codec *codec)
 	return 0;
 }
 
-static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai,
+static int uda1380_set_dai_fmt_both(struct snd_soc_dai *codec_dai,
 		unsigned int fmt)
 {
 	struct snd_soc_codec *codec = codec_dai->codec;
@@ -381,61 +388,107 @@ static int uda1380_set_dai_fmt(struct snd_soc_dai *codec_dai,
 	iface = uda1380_read_reg_cache(codec, UDA1380_IFACE);
 	iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK);
 
-	/* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
 		iface |= R01_SFORI_I2S | R01_SFORO_I2S;
 		break;
 	case SND_SOC_DAIFMT_LSB:
-		iface |= R01_SFORI_LSB16 | R01_SFORO_I2S;
+		iface |= R01_SFORI_LSB16 | R01_SFORO_LSB16;
 		break;
 	case SND_SOC_DAIFMT_MSB:
-		iface |= R01_SFORI_MSB | R01_SFORO_I2S;
+		iface |= R01_SFORI_MSB | R01_SFORO_MSB;
 	}
 
-	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM)
-		iface |= R01_SIM;
+	/* DATAI is slave only, so in single-link mode, this has to be slave */
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
+		return -EINVAL;
 
 	uda1380_write(codec, UDA1380_IFACE, iface);
 
 	return 0;
 }
 
-/*
- * Flush reg cache
- * We can only write the interpolator and decimator registers
- * when the DAI is being clocked by the CPU DAI. It's up to the
- * machine and cpu DAI driver to do this before we are called.
- */
-static int uda1380_pcm_prepare(struct snd_pcm_substream *substream,
-			       struct snd_soc_dai *dai)
+static int uda1380_set_dai_fmt_playback(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
 {
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int reg, reg_start, reg_end, clk;
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		reg_start = UDA1380_MVOL;
-		reg_end = UDA1380_MIXER;
-	} else {
-		reg_start = UDA1380_DEC;
-		reg_end = UDA1380_AGC;
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int iface;
+
+	/* set up DAI based upon fmt */
+	iface = uda1380_read_reg_cache(codec, UDA1380_IFACE);
+	iface &= ~R01_SFORI_MASK;
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= R01_SFORI_I2S;
+		break;
+	case SND_SOC_DAIFMT_LSB:
+		iface |= R01_SFORI_LSB16;
+		break;
+	case SND_SOC_DAIFMT_MSB:
+		iface |= R01_SFORI_MSB;
 	}
 
-	/* FIXME disable DAC_CLK */
-	clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
-	uda1380_write(codec, UDA1380_CLK, clk & ~R00_DAC_CLK);
+	/* DATAI is slave only, so this has to be slave */
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS)
+		return -EINVAL;
+
+	uda1380_write(codec, UDA1380_IFACE, iface);
+
+	return 0;
+}
+
+static int uda1380_set_dai_fmt_capture(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	int iface;
+
+	/* set up DAI based upon fmt */
+	iface = uda1380_read_reg_cache(codec, UDA1380_IFACE);
+	iface &= ~(R01_SIM | R01_SFORO_MASK);
 
-	for (reg = reg_start; reg <= reg_end; reg++) {
-		pr_debug("uda1380: flush reg %x val %x:", reg,
-				uda1380_read_reg_cache(codec, reg));
-		uda1380_write(codec, reg, uda1380_read_reg_cache(codec, reg));
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		iface |= R01_SFORO_I2S;
+		break;
+	case SND_SOC_DAIFMT_LSB:
+		iface |= R01_SFORO_LSB16;
+		break;
+	case SND_SOC_DAIFMT_MSB:
+		iface |= R01_SFORO_MSB;
 	}
 
-	/* FIXME enable DAC_CLK */
-	uda1380_write(codec, UDA1380_CLK, clk | R00_DAC_CLK);
+	if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM)
+		iface |= R01_SIM;
 
+	uda1380_write(codec, UDA1380_IFACE, iface);
+
+	return 0;
+}
+
+static int uda1380_trigger(struct snd_pcm_substream *substream, int cmd,
+		struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	int mixer = uda1380_read_reg_cache(codec, UDA1380_MIXER);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		uda1380_write_reg_cache(codec, UDA1380_MIXER,
+					mixer & ~R14_SILENCE);
+		schedule_work(&uda1380_work);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		uda1380_write_reg_cache(codec, UDA1380_MIXER,
+					mixer | R14_SILENCE);
+		schedule_work(&uda1380_work);
+		break;
+	}
 	return 0;
 }
 
@@ -445,7 +498,7 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
 
 	/* set WSPLL power and divider if running from this clock */
@@ -484,7 +537,7 @@ static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
 
 	/* shut down WSPLL power if running from this clock */
@@ -501,24 +554,6 @@ static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream,
 	uda1380_write(codec, UDA1380_CLK, clk);
 }
 
-static int uda1380_mute(struct snd_soc_dai *codec_dai, int mute)
-{
-	struct snd_soc_codec *codec = codec_dai->codec;
-	u16 mute_reg = uda1380_read_reg_cache(codec, UDA1380_DEEMP) & ~R13_MTM;
-
-	/* FIXME: mute(codec,0) is called when the magician clock is already
-	 * set to WSPLL, but for some unknown reason writing to interpolator
-	 * registers works only when clocked by SYSCLK */
-	u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
-	uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk);
-	if (mute)
-		uda1380_write(codec, UDA1380_DEEMP, mute_reg | R13_MTM);
-	else
-		uda1380_write(codec, UDA1380_DEEMP, mute_reg);
-	uda1380_write(codec, UDA1380_CLK, clk);
-	return 0;
-}
-
 static int uda1380_set_bias_level(struct snd_soc_codec *codec,
 	enum snd_soc_bias_level level)
 {
@@ -544,6 +579,27 @@ static int uda1380_set_bias_level(struct snd_soc_codec *codec,
 		       SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
 		       SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
 
+static struct snd_soc_dai_ops uda1380_dai_ops = {
+	.hw_params	= uda1380_pcm_hw_params,
+	.shutdown	= uda1380_pcm_shutdown,
+	.trigger	= uda1380_trigger,
+	.set_fmt	= uda1380_set_dai_fmt_both,
+};
+
+static struct snd_soc_dai_ops uda1380_dai_ops_playback = {
+	.hw_params	= uda1380_pcm_hw_params,
+	.shutdown	= uda1380_pcm_shutdown,
+	.trigger	= uda1380_trigger,
+	.set_fmt	= uda1380_set_dai_fmt_playback,
+};
+
+static struct snd_soc_dai_ops uda1380_dai_ops_capture = {
+	.hw_params	= uda1380_pcm_hw_params,
+	.shutdown	= uda1380_pcm_shutdown,
+	.trigger	= uda1380_trigger,
+	.set_fmt	= uda1380_set_dai_fmt_capture,
+};
+
 struct snd_soc_dai uda1380_dai[] = {
 {
 	.name = "UDA1380",
@@ -559,13 +615,7 @@ struct snd_soc_dai uda1380_dai[] = {
 		.channels_max = 2,
 		.rates = UDA1380_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = uda1380_pcm_hw_params,
-		.shutdown = uda1380_pcm_shutdown,
-		.prepare = uda1380_pcm_prepare,
-		.digital_mute = uda1380_mute,
-		.set_fmt = uda1380_set_dai_fmt,
-	},
+	.ops = &uda1380_dai_ops,
 },
 { /* playback only - dual interface */
 	.name = "UDA1380",
@@ -576,13 +626,7 @@ struct snd_soc_dai uda1380_dai[] = {
 		.rates = UDA1380_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
-	.ops = {
-		.hw_params = uda1380_pcm_hw_params,
-		.shutdown = uda1380_pcm_shutdown,
-		.prepare = uda1380_pcm_prepare,
-		.digital_mute = uda1380_mute,
-		.set_fmt = uda1380_set_dai_fmt,
-	},
+	.ops = &uda1380_dai_ops_playback,
 },
 { /* capture only - dual interface*/
 	.name = "UDA1380",
@@ -593,12 +637,7 @@ struct snd_soc_dai uda1380_dai[] = {
 		.rates = UDA1380_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,
 	},
-	.ops = {
-		.hw_params = uda1380_pcm_hw_params,
-		.shutdown = uda1380_pcm_shutdown,
-		.prepare = uda1380_pcm_prepare,
-		.set_fmt = uda1380_set_dai_fmt,
-	},
+	.ops = &uda1380_dai_ops_capture,
 },
 };
 EXPORT_SYMBOL_GPL(uda1380_dai);
@@ -606,7 +645,7 @@ EXPORT_SYMBOL_GPL(uda1380_dai);
 static int uda1380_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -615,7 +654,7 @@ static int uda1380_suspend(struct platform_device *pdev, pm_message_t state)
 static int uda1380_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -637,7 +676,7 @@ static int uda1380_resume(struct platform_device *pdev)
  */
 static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 
 	codec->name = "UDA1380";
@@ -655,6 +694,9 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
 	codec->reg_cache_step = 1;
 	uda1380_reset(codec);
 
+	uda1380_codec = codec;
+	INIT_WORK(&uda1380_work, uda1380_flush_work);
+
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
@@ -675,7 +717,8 @@ static int uda1380_init(struct snd_soc_device *socdev, int dac_clk)
 	}
 
 	/* uda1380 init */
-	uda1380_add_controls(codec);
+	snd_soc_add_controls(codec, uda1380_snd_controls,
+				ARRAY_SIZE(uda1380_snd_controls));
 	uda1380_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -702,7 +745,7 @@ static int uda1380_i2c_probe(struct i2c_client *i2c,
 {
 	struct snd_soc_device *socdev = uda1380_socdev;
 	struct uda1380_setup_data *setup = socdev->codec_data;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -786,14 +829,12 @@ static int uda1380_probe(struct platform_device *pdev)
 	struct snd_soc_codec *codec;
 	int ret;
 
-	pr_info("UDA1380 Audio Codec %s", UDA1380_VERSION);
-
 	setup = socdev->codec_data;
 	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
 	if (codec == NULL)
 		return -ENOMEM;
 
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -817,7 +858,7 @@ static int uda1380_probe(struct platform_device *pdev)
 static int uda1380_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		uda1380_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 35d99750c383..3b1d0993bed9 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -51,10 +51,17 @@ struct wm8350_output {
 	u16 mute;
 };
 
+struct wm8350_jack_data {
+	struct snd_soc_jack *jack;
+	int report;
+};
+
 struct wm8350_data {
 	struct snd_soc_codec codec;
 	struct wm8350_output out1;
 	struct wm8350_output out2;
+	struct wm8350_jack_data hpl;
+	struct wm8350_jack_data hpr;
 	struct regulator_bulk_data supplies[ARRAY_SIZE(supply_names)];
 };
 
@@ -775,21 +782,6 @@ static const struct snd_soc_dapm_route audio_map[] = {
 	{"Beep", NULL, "IN3R PGA"},
 };
 
-static int wm8350_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8350_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm8350_snd_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 static int wm8350_add_widgets(struct snd_soc_codec *codec)
 {
 	int ret;
@@ -1309,7 +1301,7 @@ static int wm8350_set_bias_level(struct snd_soc_codec *codec,
 static int wm8350_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8350_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -1318,7 +1310,7 @@ static int wm8350_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8350_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
@@ -1328,6 +1320,95 @@ static int wm8350_resume(struct platform_device *pdev)
 	return 0;
 }
 
+static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data)
+{
+	struct wm8350_data *priv = data;
+	u16 reg;
+	int report;
+	int mask;
+	struct wm8350_jack_data *jack = NULL;
+
+	switch (irq) {
+	case WM8350_IRQ_CODEC_JCK_DET_L:
+		jack = &priv->hpl;
+		mask = WM8350_JACK_L_LVL;
+		break;
+
+	case WM8350_IRQ_CODEC_JCK_DET_R:
+		jack = &priv->hpr;
+		mask = WM8350_JACK_R_LVL;
+		break;
+
+	default:
+		BUG();
+	}
+
+	if (!jack->jack) {
+		dev_warn(wm8350->dev, "Jack interrupt called with no jack\n");
+		return;
+	}
+
+	/* Debounce */
+	msleep(200);
+
+	reg = wm8350_reg_read(wm8350, WM8350_JACK_PIN_STATUS);
+	if (reg & mask)
+		report = jack->report;
+	else
+		report = 0;
+
+	snd_soc_jack_report(jack->jack, report, jack->report);
+}
+
+/**
+ * wm8350_hp_jack_detect - Enable headphone jack detection.
+ *
+ * @codec:  WM8350 codec
+ * @which:  left or right jack detect signal
+ * @jack:   jack to report detection events on
+ * @report: value to report
+ *
+ * Enables the headphone jack detection of the WM8350.
+ */
+int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
+			  struct snd_soc_jack *jack, int report)
+{
+	struct wm8350_data *priv = codec->private_data;
+	struct wm8350 *wm8350 = codec->control_data;
+	int irq;
+	int ena;
+
+	switch (which) {
+	case WM8350_JDL:
+		priv->hpl.jack = jack;
+		priv->hpl.report = report;
+		irq = WM8350_IRQ_CODEC_JCK_DET_L;
+		ena = WM8350_JDL_ENA;
+		break;
+
+	case WM8350_JDR:
+		priv->hpr.jack = jack;
+		priv->hpr.report = report;
+		irq = WM8350_IRQ_CODEC_JCK_DET_R;
+		ena = WM8350_JDR_ENA;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	wm8350_set_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
+	wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena);
+
+	/* Sync status */
+	wm8350_hp_jack_handler(wm8350, irq, priv);
+
+	wm8350_unmask_irq(wm8350, irq);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(wm8350_hp_jack_detect);
+
 static struct snd_soc_codec *wm8350_codec;
 
 static int wm8350_probe(struct platform_device *pdev)
@@ -1342,8 +1423,8 @@ static int wm8350_probe(struct platform_device *pdev)
 
 	BUG_ON(!wm8350_codec);
 
-	socdev->codec = wm8350_codec;
-	codec = socdev->codec;
+	socdev->card->codec = wm8350_codec;
+	codec = socdev->card->codec;
 	wm8350 = codec->control_data;
 	priv = codec->private_data;
 
@@ -1381,13 +1462,21 @@ static int wm8350_probe(struct platform_device *pdev)
 	wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME,
 			WM8350_OUT2_VU | WM8350_OUT2R_MUTE);
 
+	wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
+	wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
+	wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L,
+			    wm8350_hp_jack_handler, priv);
+	wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R,
+			    wm8350_hp_jack_handler, priv);
+
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
 		dev_err(&pdev->dev, "failed to create pcms\n");
 		return ret;
 	}
 
-	wm8350_add_controls(codec);
+	snd_soc_add_controls(codec, wm8350_snd_controls,
+				ARRAY_SIZE(wm8350_snd_controls));
 	wm8350_add_widgets(codec);
 
 	wm8350_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
@@ -1409,10 +1498,23 @@ card_err:
 static int wm8350_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8350 *wm8350 = codec->control_data;
+	struct wm8350_data *priv = codec->private_data;
 	int ret;
 
+	wm8350_clear_bits(wm8350, WM8350_JACK_DETECT,
+			  WM8350_JDL_ENA | WM8350_JDR_ENA);
+	wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA);
+
+	wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
+	wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
+	wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L);
+	wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R);
+
+	priv->hpl.jack = NULL;
+	priv->hpr.jack = NULL;
+
 	/* cancel any work waiting to be queued. */
 	ret = cancel_delayed_work(&codec->delayed_work);
 
@@ -1436,6 +1538,16 @@ static int wm8350_remove(struct platform_device *pdev)
 			SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE)
 
+static struct snd_soc_dai_ops wm8350_dai_ops = {
+	 .hw_params	= wm8350_pcm_hw_params,
+	 .digital_mute	= wm8350_mute,
+	 .trigger	= wm8350_pcm_trigger,
+	 .set_fmt	= wm8350_set_dai_fmt,
+	 .set_sysclk	= wm8350_set_dai_sysclk,
+	 .set_pll	= wm8350_set_fll,
+	 .set_clkdiv	= wm8350_set_clkdiv,
+};
+
 struct snd_soc_dai wm8350_dai = {
 	.name = "WM8350",
 	.playback = {
@@ -1452,15 +1564,7 @@ struct snd_soc_dai wm8350_dai = {
 		 .rates = WM8350_RATES,
 		 .formats = WM8350_FORMATS,
 	 },
-	.ops = {
-		 .hw_params = wm8350_pcm_hw_params,
-		 .digital_mute = wm8350_mute,
-		 .trigger = wm8350_pcm_trigger,
-		 .set_fmt = wm8350_set_dai_fmt,
-		 .set_sysclk = wm8350_set_dai_sysclk,
-		 .set_pll = wm8350_set_fll,
-		 .set_clkdiv = wm8350_set_clkdiv,
-	 },
+	.ops = &wm8350_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8350_dai);
 
@@ -1472,7 +1576,7 @@ struct snd_soc_codec_device soc_codec_dev_wm8350 = {
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm8350);
 
-static int wm8350_codec_probe(struct platform_device *pdev)
+static __devinit int wm8350_codec_probe(struct platform_device *pdev)
 {
 	struct wm8350 *wm8350 = platform_get_drvdata(pdev);
 	struct wm8350_data *priv;
diff --git a/sound/soc/codecs/wm8350.h b/sound/soc/codecs/wm8350.h
index cc2887aa6c38..d11bd9288cf9 100644
--- a/sound/soc/codecs/wm8350.h
+++ b/sound/soc/codecs/wm8350.h
@@ -17,4 +17,12 @@
 extern struct snd_soc_dai wm8350_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8350;
 
+enum wm8350_jack {
+	WM8350_JDL = 1,
+	WM8350_JDR = 2,
+};
+
+int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which,
+			  struct snd_soc_jack *jack, int report);
+
 #endif
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
new file mode 100644
index 000000000000..510efa604008
--- /dev/null
+++ b/sound/soc/codecs/wm8400.c
@@ -0,0 +1,1582 @@
+/*
+ * wm8400.c  --  WM8400 ALSA Soc Audio driver
+ *
+ * Copyright 2008, 2009 Wolfson Microelectronics PLC.
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/platform_device.h>
+#include <linux/regulator/consumer.h>
+#include <linux/mfd/wm8400-audio.h>
+#include <linux/mfd/wm8400-private.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+
+#include "wm8400.h"
+
+/* Fake register for internal state */
+#define WM8400_INTDRIVBITS      (WM8400_REGISTER_COUNT + 1)
+#define WM8400_INMIXL_PWR			0
+#define WM8400_AINLMUX_PWR			1
+#define WM8400_INMIXR_PWR			2
+#define WM8400_AINRMUX_PWR			3
+
+static struct regulator_bulk_data power[] = {
+	{
+		.supply = "I2S1VDD",
+	},
+	{
+		.supply = "I2S2VDD",
+	},
+	{
+		.supply = "DCVDD",
+	},
+	{
+		.supply = "AVDD",
+	},
+	{
+		.supply = "FLLVDD",
+	},
+	{
+		.supply = "HPVDD",
+	},
+	{
+		.supply = "SPKVDD",
+	},
+};
+
+/* codec private data */
+struct wm8400_priv {
+	struct snd_soc_codec codec;
+	struct wm8400 *wm8400;
+	u16 fake_register;
+	unsigned int sysclk;
+	unsigned int pcmclk;
+	struct work_struct work;
+	int fll_in, fll_out;
+};
+
+static inline unsigned int wm8400_read(struct snd_soc_codec *codec,
+				       unsigned int reg)
+{
+	struct wm8400_priv *wm8400 = codec->private_data;
+
+	if (reg == WM8400_INTDRIVBITS)
+		return wm8400->fake_register;
+	else
+		return wm8400_reg_read(wm8400->wm8400, reg);
+}
+
+/*
+ * write to the wm8400 register space
+ */
+static int wm8400_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int value)
+{
+	struct wm8400_priv *wm8400 = codec->private_data;
+
+	if (reg == WM8400_INTDRIVBITS) {
+		wm8400->fake_register = value;
+		return 0;
+	} else
+		return wm8400_set_bits(wm8400->wm8400, reg, 0xffff, value);
+}
+
+static void wm8400_codec_reset(struct snd_soc_codec *codec)
+{
+	struct wm8400_priv *wm8400 = codec->private_data;
+
+	wm8400_reset_codec_reg_cache(wm8400->wm8400);
+}
+
+static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600);
+
+static const DECLARE_TLV_DB_LINEAR(in_pga_tlv, -1650, 3000);
+
+static const DECLARE_TLV_DB_LINEAR(out_mix_tlv, -2100, 0);
+
+static const DECLARE_TLV_DB_LINEAR(out_pga_tlv, -7300, 600);
+
+static const DECLARE_TLV_DB_LINEAR(out_omix_tlv, -600, 0);
+
+static const DECLARE_TLV_DB_LINEAR(out_dac_tlv, -7163, 0);
+
+static const DECLARE_TLV_DB_LINEAR(in_adc_tlv, -7163, 1763);
+
+static const DECLARE_TLV_DB_LINEAR(out_sidetone_tlv, -3600, 0);
+
+static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	int reg = mc->reg;
+        int ret;
+        u16 val;
+
+        ret = snd_soc_put_volsw(kcontrol, ucontrol);
+        if (ret < 0)
+                return ret;
+
+        /* now hit the volume update bits (always bit 8) */
+        val = wm8400_read(codec, reg);
+        return wm8400_write(codec, reg, val | 0x0100);
+}
+
+#define WM8400_OUTPGA_SINGLE_R_TLV(xname, reg, shift, max, invert, tlv_array) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\
+		SNDRV_CTL_ELEM_ACCESS_READWRITE,\
+	.tlv.p = (tlv_array), \
+	.info = snd_soc_info_volsw, \
+	.get = snd_soc_get_volsw, .put = wm8400_outpga_put_volsw_vu, \
+	.private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) }
+
+
+static const char *wm8400_digital_sidetone[] =
+	{"None", "Left ADC", "Right ADC", "Reserved"};
+
+static const struct soc_enum wm8400_left_digital_sidetone_enum =
+SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE,
+		WM8400_ADC_TO_DACL_SHIFT, 2, wm8400_digital_sidetone);
+
+static const struct soc_enum wm8400_right_digital_sidetone_enum =
+SOC_ENUM_SINGLE(WM8400_DIGITAL_SIDE_TONE,
+		WM8400_ADC_TO_DACR_SHIFT, 2, wm8400_digital_sidetone);
+
+static const char *wm8400_adcmode[] =
+	{"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"};
+
+static const struct soc_enum wm8400_right_adcmode_enum =
+SOC_ENUM_SINGLE(WM8400_ADC_CTRL, WM8400_ADC_HPF_CUT_SHIFT, 3, wm8400_adcmode);
+
+static const struct snd_kcontrol_new wm8400_snd_controls[] = {
+/* INMIXL */
+SOC_SINGLE("LIN12 PGA Boost", WM8400_INPUT_MIXER3, WM8400_L12MNBST_SHIFT,
+	   1, 0),
+SOC_SINGLE("LIN34 PGA Boost", WM8400_INPUT_MIXER3, WM8400_L34MNBST_SHIFT,
+	   1, 0),
+/* INMIXR */
+SOC_SINGLE("RIN12 PGA Boost", WM8400_INPUT_MIXER3, WM8400_R12MNBST_SHIFT,
+	   1, 0),
+SOC_SINGLE("RIN34 PGA Boost", WM8400_INPUT_MIXER3, WM8400_R34MNBST_SHIFT,
+	   1, 0),
+
+/* LOMIX */
+SOC_SINGLE_TLV("LOMIX LIN3 Bypass Volume", WM8400_OUTPUT_MIXER3,
+	WM8400_LLI3LOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX RIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER3,
+	WM8400_LR12LOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX LIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER3,
+	WM8400_LL12LOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX RIN3 Bypass Volume", WM8400_OUTPUT_MIXER5,
+	WM8400_LRI3LOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX AINRMUX Bypass Volume", WM8400_OUTPUT_MIXER5,
+	WM8400_LRBLOVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("LOMIX AINLMUX Bypass Volume", WM8400_OUTPUT_MIXER5,
+	WM8400_LRBLOVOL_SHIFT, 7, 0, out_mix_tlv),
+
+/* ROMIX */
+SOC_SINGLE_TLV("ROMIX RIN3 Bypass Volume", WM8400_OUTPUT_MIXER4,
+	WM8400_RRI3ROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX LIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER4,
+	WM8400_RL12ROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX RIN12 PGA Bypass Volume", WM8400_OUTPUT_MIXER4,
+	WM8400_RR12ROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX LIN3 Bypass Volume", WM8400_OUTPUT_MIXER6,
+	WM8400_RLI3ROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX AINLMUX Bypass Volume", WM8400_OUTPUT_MIXER6,
+	WM8400_RLBROVOL_SHIFT, 7, 0, out_mix_tlv),
+SOC_SINGLE_TLV("ROMIX AINRMUX Bypass Volume", WM8400_OUTPUT_MIXER6,
+	WM8400_RRBROVOL_SHIFT, 7, 0, out_mix_tlv),
+
+/* LOUT */
+WM8400_OUTPGA_SINGLE_R_TLV("LOUT Volume", WM8400_LEFT_OUTPUT_VOLUME,
+	WM8400_LOUTVOL_SHIFT, WM8400_LOUTVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("LOUT ZC", WM8400_LEFT_OUTPUT_VOLUME, WM8400_LOZC_SHIFT, 1, 0),
+
+/* ROUT */
+WM8400_OUTPGA_SINGLE_R_TLV("ROUT Volume", WM8400_RIGHT_OUTPUT_VOLUME,
+	WM8400_ROUTVOL_SHIFT, WM8400_ROUTVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("ROUT ZC", WM8400_RIGHT_OUTPUT_VOLUME, WM8400_ROZC_SHIFT, 1, 0),
+
+/* LOPGA */
+WM8400_OUTPGA_SINGLE_R_TLV("LOPGA Volume", WM8400_LEFT_OPGA_VOLUME,
+	WM8400_LOPGAVOL_SHIFT, WM8400_LOPGAVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("LOPGA ZC Switch", WM8400_LEFT_OPGA_VOLUME,
+	WM8400_LOPGAZC_SHIFT, 1, 0),
+
+/* ROPGA */
+WM8400_OUTPGA_SINGLE_R_TLV("ROPGA Volume", WM8400_RIGHT_OPGA_VOLUME,
+	WM8400_ROPGAVOL_SHIFT, WM8400_ROPGAVOL_MASK, 0, out_pga_tlv),
+SOC_SINGLE("ROPGA ZC Switch", WM8400_RIGHT_OPGA_VOLUME,
+	WM8400_ROPGAZC_SHIFT, 1, 0),
+
+SOC_SINGLE("LON Mute Switch", WM8400_LINE_OUTPUTS_VOLUME,
+	WM8400_LONMUTE_SHIFT, 1, 0),
+SOC_SINGLE("LOP Mute Switch", WM8400_LINE_OUTPUTS_VOLUME,
+	WM8400_LOPMUTE_SHIFT, 1, 0),
+SOC_SINGLE("LOP Attenuation Switch", WM8400_LINE_OUTPUTS_VOLUME,
+	WM8400_LOATTN_SHIFT, 1, 0),
+SOC_SINGLE("RON Mute Switch", WM8400_LINE_OUTPUTS_VOLUME,
+	WM8400_RONMUTE_SHIFT, 1, 0),
+SOC_SINGLE("ROP Mute Switch", WM8400_LINE_OUTPUTS_VOLUME,
+	WM8400_ROPMUTE_SHIFT, 1, 0),
+SOC_SINGLE("ROP Attenuation Switch", WM8400_LINE_OUTPUTS_VOLUME,
+	WM8400_ROATTN_SHIFT, 1, 0),
+
+SOC_SINGLE("OUT3 Mute Switch", WM8400_OUT3_4_VOLUME,
+	WM8400_OUT3MUTE_SHIFT, 1, 0),
+SOC_SINGLE("OUT3 Attenuation Switch", WM8400_OUT3_4_VOLUME,
+	WM8400_OUT3ATTN_SHIFT, 1, 0),
+
+SOC_SINGLE("OUT4 Mute Switch", WM8400_OUT3_4_VOLUME,
+	WM8400_OUT4MUTE_SHIFT, 1, 0),
+SOC_SINGLE("OUT4 Attenuation Switch", WM8400_OUT3_4_VOLUME,
+	WM8400_OUT4ATTN_SHIFT, 1, 0),
+
+SOC_SINGLE("Speaker Mode Switch", WM8400_CLASSD1,
+	WM8400_CDMODE_SHIFT, 1, 0),
+
+SOC_SINGLE("Speaker Output Attenuation Volume", WM8400_SPEAKER_VOLUME,
+	WM8400_SPKATTN_SHIFT, WM8400_SPKATTN_MASK, 0),
+SOC_SINGLE("Speaker DC Boost Volume", WM8400_CLASSD3,
+	WM8400_DCGAIN_SHIFT, 6, 0),
+SOC_SINGLE("Speaker AC Boost Volume", WM8400_CLASSD3,
+	WM8400_ACGAIN_SHIFT, 6, 0),
+
+WM8400_OUTPGA_SINGLE_R_TLV("Left DAC Digital Volume",
+	WM8400_LEFT_DAC_DIGITAL_VOLUME, WM8400_DACL_VOL_SHIFT,
+	127, 0, out_dac_tlv),
+
+WM8400_OUTPGA_SINGLE_R_TLV("Right DAC Digital Volume",
+	WM8400_RIGHT_DAC_DIGITAL_VOLUME, WM8400_DACR_VOL_SHIFT,
+	127, 0, out_dac_tlv),
+
+SOC_ENUM("Left Digital Sidetone", wm8400_left_digital_sidetone_enum),
+SOC_ENUM("Right Digital Sidetone", wm8400_right_digital_sidetone_enum),
+
+SOC_SINGLE_TLV("Left Digital Sidetone Volume", WM8400_DIGITAL_SIDE_TONE,
+	WM8400_ADCL_DAC_SVOL_SHIFT, 15, 0, out_sidetone_tlv),
+SOC_SINGLE_TLV("Right Digital Sidetone Volume", WM8400_DIGITAL_SIDE_TONE,
+	WM8400_ADCR_DAC_SVOL_SHIFT, 15, 0, out_sidetone_tlv),
+
+SOC_SINGLE("ADC Digital High Pass Filter Switch", WM8400_ADC_CTRL,
+	WM8400_ADC_HPF_ENA_SHIFT, 1, 0),
+
+SOC_ENUM("ADC HPF Mode", wm8400_right_adcmode_enum),
+
+WM8400_OUTPGA_SINGLE_R_TLV("Left ADC Digital Volume",
+	WM8400_LEFT_ADC_DIGITAL_VOLUME,
+	WM8400_ADCL_VOL_SHIFT,
+	WM8400_ADCL_VOL_MASK,
+	0,
+	in_adc_tlv),
+
+WM8400_OUTPGA_SINGLE_R_TLV("Right ADC Digital Volume",
+	WM8400_RIGHT_ADC_DIGITAL_VOLUME,
+	WM8400_ADCR_VOL_SHIFT,
+	WM8400_ADCR_VOL_MASK,
+	0,
+	in_adc_tlv),
+
+WM8400_OUTPGA_SINGLE_R_TLV("LIN12 Volume",
+	WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
+	WM8400_LIN12VOL_SHIFT,
+	WM8400_LIN12VOL_MASK,
+	0,
+	in_pga_tlv),
+
+SOC_SINGLE("LIN12 ZC Switch", WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
+	WM8400_LI12ZC_SHIFT, 1, 0),
+
+SOC_SINGLE("LIN12 Mute Switch", WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
+	WM8400_LI12MUTE_SHIFT, 1, 0),
+
+WM8400_OUTPGA_SINGLE_R_TLV("LIN34 Volume",
+	WM8400_LEFT_LINE_INPUT_3_4_VOLUME,
+	WM8400_LIN34VOL_SHIFT,
+	WM8400_LIN34VOL_MASK,
+	0,
+	in_pga_tlv),
+
+SOC_SINGLE("LIN34 ZC Switch", WM8400_LEFT_LINE_INPUT_3_4_VOLUME,
+	WM8400_LI34ZC_SHIFT, 1, 0),
+
+SOC_SINGLE("LIN34 Mute Switch", WM8400_LEFT_LINE_INPUT_3_4_VOLUME,
+	WM8400_LI34MUTE_SHIFT, 1, 0),
+
+WM8400_OUTPGA_SINGLE_R_TLV("RIN12 Volume",
+	WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
+	WM8400_RIN12VOL_SHIFT,
+	WM8400_RIN12VOL_MASK,
+	0,
+	in_pga_tlv),
+
+SOC_SINGLE("RIN12 ZC Switch", WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
+	WM8400_RI12ZC_SHIFT, 1, 0),
+
+SOC_SINGLE("RIN12 Mute Switch", WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
+	WM8400_RI12MUTE_SHIFT, 1, 0),
+
+WM8400_OUTPGA_SINGLE_R_TLV("RIN34 Volume",
+	WM8400_RIGHT_LINE_INPUT_3_4_VOLUME,
+	WM8400_RIN34VOL_SHIFT,
+	WM8400_RIN34VOL_MASK,
+	0,
+	in_pga_tlv),
+
+SOC_SINGLE("RIN34 ZC Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME,
+	WM8400_RI34ZC_SHIFT, 1, 0),
+
+SOC_SINGLE("RIN34 Mute Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME,
+	WM8400_RI34MUTE_SHIFT, 1, 0),
+
+};
+
+/* add non dapm controls */
+static int wm8400_add_controls(struct snd_soc_codec *codec)
+{
+	return snd_soc_add_controls(codec, wm8400_snd_controls,
+				ARRAY_SIZE(wm8400_snd_controls));
+}
+
+/*
+ * _DAPM_ Controls
+ */
+
+static int inmixer_event (struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *kcontrol, int event)
+{
+	u16 reg, fakepower;
+
+	reg = wm8400_read(w->codec, WM8400_POWER_MANAGEMENT_2);
+	fakepower = wm8400_read(w->codec, WM8400_INTDRIVBITS);
+
+	if (fakepower & ((1 << WM8400_INMIXL_PWR) |
+		(1 << WM8400_AINLMUX_PWR))) {
+		reg |= WM8400_AINL_ENA;
+	} else {
+		reg &= ~WM8400_AINL_ENA;
+	}
+
+	if (fakepower & ((1 << WM8400_INMIXR_PWR) |
+		(1 << WM8400_AINRMUX_PWR))) {
+		reg |= WM8400_AINR_ENA;
+	} else {
+		reg &= ~WM8400_AINL_ENA;
+	}
+	wm8400_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg);
+
+	return 0;
+}
+
+static int outmixer_event (struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol * kcontrol, int event)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	u32 reg_shift = mc->shift;
+	int ret = 0;
+	u16 reg;
+
+	switch (reg_shift) {
+	case WM8400_SPEAKER_MIXER | (WM8400_LDSPK << 8) :
+		reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER1);
+		if (reg & WM8400_LDLO) {
+			printk(KERN_WARNING
+			"Cannot set as Output Mixer 1 LDLO Set\n");
+			ret = -1;
+		}
+		break;
+	case WM8400_SPEAKER_MIXER | (WM8400_RDSPK << 8):
+		reg = wm8400_read(w->codec, WM8400_OUTPUT_MIXER2);
+		if (reg & WM8400_RDRO) {
+			printk(KERN_WARNING
+			"Cannot set as Output Mixer 2 RDRO Set\n");
+			ret = -1;
+		}
+		break;
+	case WM8400_OUTPUT_MIXER1 | (WM8400_LDLO << 8):
+		reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER);
+		if (reg & WM8400_LDSPK) {
+			printk(KERN_WARNING
+			"Cannot set as Speaker Mixer LDSPK Set\n");
+			ret = -1;
+		}
+		break;
+	case WM8400_OUTPUT_MIXER2 | (WM8400_RDRO << 8):
+		reg = wm8400_read(w->codec, WM8400_SPEAKER_MIXER);
+		if (reg & WM8400_RDSPK) {
+			printk(KERN_WARNING
+			"Cannot set as Speaker Mixer RDSPK Set\n");
+			ret = -1;
+		}
+		break;
+	}
+
+	return ret;
+}
+
+/* INMIX dB values */
+static const unsigned int in_mix_tlv[] = {
+	TLV_DB_RANGE_HEAD(1),
+	0,7, TLV_DB_LINEAR_ITEM(-1200, 600),
+};
+
+/* Left In PGA Connections */
+static const struct snd_kcontrol_new wm8400_dapm_lin12_pga_controls[] = {
+SOC_DAPM_SINGLE("LIN1 Switch", WM8400_INPUT_MIXER2, WM8400_LMN1_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LIN2 Switch", WM8400_INPUT_MIXER2, WM8400_LMP2_SHIFT, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8400_dapm_lin34_pga_controls[] = {
+SOC_DAPM_SINGLE("LIN3 Switch", WM8400_INPUT_MIXER2, WM8400_LMN3_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LIN4 Switch", WM8400_INPUT_MIXER2, WM8400_LMP4_SHIFT, 1, 0),
+};
+
+/* Right In PGA Connections */
+static const struct snd_kcontrol_new wm8400_dapm_rin12_pga_controls[] = {
+SOC_DAPM_SINGLE("RIN1 Switch", WM8400_INPUT_MIXER2, WM8400_RMN1_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("RIN2 Switch", WM8400_INPUT_MIXER2, WM8400_RMP2_SHIFT, 1, 0),
+};
+
+static const struct snd_kcontrol_new wm8400_dapm_rin34_pga_controls[] = {
+SOC_DAPM_SINGLE("RIN3 Switch", WM8400_INPUT_MIXER2, WM8400_RMN3_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("RIN4 Switch", WM8400_INPUT_MIXER2, WM8400_RMP4_SHIFT, 1, 0),
+};
+
+/* INMIXL */
+static const struct snd_kcontrol_new wm8400_dapm_inmixl_controls[] = {
+SOC_DAPM_SINGLE_TLV("Record Left Volume", WM8400_INPUT_MIXER3,
+	WM8400_LDBVOL_SHIFT, WM8400_LDBVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("LIN2 Volume", WM8400_INPUT_MIXER5, WM8400_LI2BVOL_SHIFT,
+	7, 0, in_mix_tlv),
+SOC_DAPM_SINGLE("LINPGA12 Switch", WM8400_INPUT_MIXER3, WM8400_L12MNB_SHIFT,
+		1, 0),
+SOC_DAPM_SINGLE("LINPGA34 Switch", WM8400_INPUT_MIXER3, WM8400_L34MNB_SHIFT,
+		1, 0),
+};
+
+/* INMIXR */
+static const struct snd_kcontrol_new wm8400_dapm_inmixr_controls[] = {
+SOC_DAPM_SINGLE_TLV("Record Right Volume", WM8400_INPUT_MIXER4,
+	WM8400_RDBVOL_SHIFT, WM8400_RDBVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("RIN2 Volume", WM8400_INPUT_MIXER6, WM8400_RI2BVOL_SHIFT,
+	7, 0, in_mix_tlv),
+SOC_DAPM_SINGLE("RINPGA12 Switch", WM8400_INPUT_MIXER3, WM8400_L12MNB_SHIFT,
+	1, 0),
+SOC_DAPM_SINGLE("RINPGA34 Switch", WM8400_INPUT_MIXER3, WM8400_L34MNB_SHIFT,
+	1, 0),
+};
+
+/* AINLMUX */
+static const char *wm8400_ainlmux[] =
+	{"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"};
+
+static const struct soc_enum wm8400_ainlmux_enum =
+SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINLMODE_SHIFT,
+	ARRAY_SIZE(wm8400_ainlmux), wm8400_ainlmux);
+
+static const struct snd_kcontrol_new wm8400_dapm_ainlmux_controls =
+SOC_DAPM_ENUM("Route", wm8400_ainlmux_enum);
+
+/* DIFFINL */
+
+/* AINRMUX */
+static const char *wm8400_ainrmux[] =
+	{"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"};
+
+static const struct soc_enum wm8400_ainrmux_enum =
+SOC_ENUM_SINGLE( WM8400_INPUT_MIXER1, WM8400_AINRMODE_SHIFT,
+	ARRAY_SIZE(wm8400_ainrmux), wm8400_ainrmux);
+
+static const struct snd_kcontrol_new wm8400_dapm_ainrmux_controls =
+SOC_DAPM_ENUM("Route", wm8400_ainrmux_enum);
+
+/* RXVOICE */
+static const struct snd_kcontrol_new wm8400_dapm_rxvoice_controls[] = {
+SOC_DAPM_SINGLE_TLV("LIN4/RXN", WM8400_INPUT_MIXER5, WM8400_LR4BVOL_SHIFT,
+			WM8400_LR4BVOL_MASK, 0, in_mix_tlv),
+SOC_DAPM_SINGLE_TLV("RIN4/RXP", WM8400_INPUT_MIXER6, WM8400_RL4BVOL_SHIFT,
+			WM8400_RL4BVOL_MASK, 0, in_mix_tlv),
+};
+
+/* LOMIX */
+static const struct snd_kcontrol_new wm8400_dapm_lomix_controls[] = {
+SOC_DAPM_SINGLE("LOMIX Right ADC Bypass Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LRBLO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX Left ADC Bypass Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LLBLO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX RIN3 Bypass Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LRI3LO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX LIN3 Bypass Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LLI3LO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX RIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LR12LO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX LIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LL12LO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOMIX Left DAC Switch", WM8400_OUTPUT_MIXER1,
+	WM8400_LDLO_SHIFT, 1, 0),
+};
+
+/* ROMIX */
+static const struct snd_kcontrol_new wm8400_dapm_romix_controls[] = {
+SOC_DAPM_SINGLE("ROMIX Left ADC Bypass Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RLBRO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX Right ADC Bypass Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RRBRO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX LIN3 Bypass Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RLI3RO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX RIN3 Bypass Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RRI3RO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX LIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RL12RO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX RIN12 PGA Bypass Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RR12RO_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROMIX Right DAC Switch", WM8400_OUTPUT_MIXER2,
+	WM8400_RDRO_SHIFT, 1, 0),
+};
+
+/* LONMIX */
+static const struct snd_kcontrol_new wm8400_dapm_lonmix_controls[] = {
+SOC_DAPM_SINGLE("LONMIX Left Mixer PGA Switch", WM8400_LINE_MIXER1,
+	WM8400_LLOPGALON_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LONMIX Right Mixer PGA Switch", WM8400_LINE_MIXER1,
+	WM8400_LROPGALON_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LONMIX Inverted LOP Switch", WM8400_LINE_MIXER1,
+	WM8400_LOPLON_SHIFT, 1, 0),
+};
+
+/* LOPMIX */
+static const struct snd_kcontrol_new wm8400_dapm_lopmix_controls[] = {
+SOC_DAPM_SINGLE("LOPMIX Right Mic Bypass Switch", WM8400_LINE_MIXER1,
+	WM8400_LR12LOP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOPMIX Left Mic Bypass Switch", WM8400_LINE_MIXER1,
+	WM8400_LL12LOP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("LOPMIX Left Mixer PGA Switch", WM8400_LINE_MIXER1,
+	WM8400_LLOPGALOP_SHIFT, 1, 0),
+};
+
+/* RONMIX */
+static const struct snd_kcontrol_new wm8400_dapm_ronmix_controls[] = {
+SOC_DAPM_SINGLE("RONMIX Right Mixer PGA Switch", WM8400_LINE_MIXER2,
+	WM8400_RROPGARON_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("RONMIX Left Mixer PGA Switch", WM8400_LINE_MIXER2,
+	WM8400_RLOPGARON_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("RONMIX Inverted ROP Switch", WM8400_LINE_MIXER2,
+	WM8400_ROPRON_SHIFT, 1, 0),
+};
+
+/* ROPMIX */
+static const struct snd_kcontrol_new wm8400_dapm_ropmix_controls[] = {
+SOC_DAPM_SINGLE("ROPMIX Left Mic Bypass Switch", WM8400_LINE_MIXER2,
+	WM8400_RL12ROP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROPMIX Right Mic Bypass Switch", WM8400_LINE_MIXER2,
+	WM8400_RR12ROP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("ROPMIX Right Mixer PGA Switch", WM8400_LINE_MIXER2,
+	WM8400_RROPGAROP_SHIFT, 1, 0),
+};
+
+/* OUT3MIX */
+static const struct snd_kcontrol_new wm8400_dapm_out3mix_controls[] = {
+SOC_DAPM_SINGLE("OUT3MIX LIN4/RXP Bypass Switch", WM8400_OUT3_4_MIXER,
+	WM8400_LI4O3_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("OUT3MIX Left Out PGA Switch", WM8400_OUT3_4_MIXER,
+	WM8400_LPGAO3_SHIFT, 1, 0),
+};
+
+/* OUT4MIX */
+static const struct snd_kcontrol_new wm8400_dapm_out4mix_controls[] = {
+SOC_DAPM_SINGLE("OUT4MIX Right Out PGA Switch", WM8400_OUT3_4_MIXER,
+	WM8400_RPGAO4_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("OUT4MIX RIN4/RXP Bypass Switch", WM8400_OUT3_4_MIXER,
+	WM8400_RI4O4_SHIFT, 1, 0),
+};
+
+/* SPKMIX */
+static const struct snd_kcontrol_new wm8400_dapm_spkmix_controls[] = {
+SOC_DAPM_SINGLE("SPKMIX LIN2 Bypass Switch", WM8400_SPEAKER_MIXER,
+	WM8400_LI2SPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX LADC Bypass Switch", WM8400_SPEAKER_MIXER,
+	WM8400_LB2SPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Left Mixer PGA Switch", WM8400_SPEAKER_MIXER,
+	WM8400_LOPGASPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Left DAC Switch", WM8400_SPEAKER_MIXER,
+	WM8400_LDSPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Right DAC Switch", WM8400_SPEAKER_MIXER,
+	WM8400_RDSPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX Right Mixer PGA Switch", WM8400_SPEAKER_MIXER,
+	WM8400_ROPGASPK_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX RADC Bypass Switch", WM8400_SPEAKER_MIXER,
+	WM8400_RL12ROP_SHIFT, 1, 0),
+SOC_DAPM_SINGLE("SPKMIX RIN2 Bypass Switch", WM8400_SPEAKER_MIXER,
+	WM8400_RI2SPK_SHIFT, 1, 0),
+};
+
+static const struct snd_soc_dapm_widget wm8400_dapm_widgets[] = {
+/* Input Side */
+/* Input Lines */
+SND_SOC_DAPM_INPUT("LIN1"),
+SND_SOC_DAPM_INPUT("LIN2"),
+SND_SOC_DAPM_INPUT("LIN3"),
+SND_SOC_DAPM_INPUT("LIN4/RXN"),
+SND_SOC_DAPM_INPUT("RIN3"),
+SND_SOC_DAPM_INPUT("RIN4/RXP"),
+SND_SOC_DAPM_INPUT("RIN1"),
+SND_SOC_DAPM_INPUT("RIN2"),
+SND_SOC_DAPM_INPUT("Internal ADC Source"),
+
+/* DACs */
+SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8400_POWER_MANAGEMENT_2,
+	WM8400_ADCL_ENA_SHIFT, 0),
+SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8400_POWER_MANAGEMENT_2,
+	WM8400_ADCR_ENA_SHIFT, 0),
+
+/* Input PGAs */
+SND_SOC_DAPM_MIXER("LIN12 PGA", WM8400_POWER_MANAGEMENT_2,
+		   WM8400_LIN12_ENA_SHIFT,
+		   0, &wm8400_dapm_lin12_pga_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_lin12_pga_controls)),
+SND_SOC_DAPM_MIXER("LIN34 PGA", WM8400_POWER_MANAGEMENT_2,
+		   WM8400_LIN34_ENA_SHIFT,
+		   0, &wm8400_dapm_lin34_pga_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_lin34_pga_controls)),
+SND_SOC_DAPM_MIXER("RIN12 PGA", WM8400_POWER_MANAGEMENT_2,
+		   WM8400_RIN12_ENA_SHIFT,
+		   0, &wm8400_dapm_rin12_pga_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_rin12_pga_controls)),
+SND_SOC_DAPM_MIXER("RIN34 PGA", WM8400_POWER_MANAGEMENT_2,
+		   WM8400_RIN34_ENA_SHIFT,
+		   0, &wm8400_dapm_rin34_pga_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_rin34_pga_controls)),
+
+/* INMIXL */
+SND_SOC_DAPM_MIXER_E("INMIXL", WM8400_INTDRIVBITS, WM8400_INMIXL_PWR, 0,
+	&wm8400_dapm_inmixl_controls[0],
+	ARRAY_SIZE(wm8400_dapm_inmixl_controls),
+	inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* AINLMUX */
+SND_SOC_DAPM_MUX_E("AILNMUX", WM8400_INTDRIVBITS, WM8400_AINLMUX_PWR, 0,
+	&wm8400_dapm_ainlmux_controls, inmixer_event,
+	SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* INMIXR */
+SND_SOC_DAPM_MIXER_E("INMIXR", WM8400_INTDRIVBITS, WM8400_INMIXR_PWR, 0,
+	&wm8400_dapm_inmixr_controls[0],
+	ARRAY_SIZE(wm8400_dapm_inmixr_controls),
+	inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* AINRMUX */
+SND_SOC_DAPM_MUX_E("AIRNMUX", WM8400_INTDRIVBITS, WM8400_AINRMUX_PWR, 0,
+	&wm8400_dapm_ainrmux_controls, inmixer_event,
+	SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD),
+
+/* Output Side */
+/* DACs */
+SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8400_POWER_MANAGEMENT_3,
+	WM8400_DACL_ENA_SHIFT, 0),
+SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8400_POWER_MANAGEMENT_3,
+	WM8400_DACR_ENA_SHIFT, 0),
+
+/* LOMIX */
+SND_SOC_DAPM_MIXER_E("LOMIX", WM8400_POWER_MANAGEMENT_3,
+		     WM8400_LOMIX_ENA_SHIFT,
+		     0, &wm8400_dapm_lomix_controls[0],
+		     ARRAY_SIZE(wm8400_dapm_lomix_controls),
+		     outmixer_event, SND_SOC_DAPM_PRE_REG),
+
+/* LONMIX */
+SND_SOC_DAPM_MIXER("LONMIX", WM8400_POWER_MANAGEMENT_3, WM8400_LON_ENA_SHIFT,
+		   0, &wm8400_dapm_lonmix_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_lonmix_controls)),
+
+/* LOPMIX */
+SND_SOC_DAPM_MIXER("LOPMIX", WM8400_POWER_MANAGEMENT_3, WM8400_LOP_ENA_SHIFT,
+		   0, &wm8400_dapm_lopmix_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_lopmix_controls)),
+
+/* OUT3MIX */
+SND_SOC_DAPM_MIXER("OUT3MIX", WM8400_POWER_MANAGEMENT_1, WM8400_OUT3_ENA_SHIFT,
+		   0, &wm8400_dapm_out3mix_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_out3mix_controls)),
+
+/* SPKMIX */
+SND_SOC_DAPM_MIXER_E("SPKMIX", WM8400_POWER_MANAGEMENT_1, WM8400_SPK_ENA_SHIFT,
+		     0, &wm8400_dapm_spkmix_controls[0],
+		     ARRAY_SIZE(wm8400_dapm_spkmix_controls), outmixer_event,
+		     SND_SOC_DAPM_PRE_REG),
+
+/* OUT4MIX */
+SND_SOC_DAPM_MIXER("OUT4MIX", WM8400_POWER_MANAGEMENT_1, WM8400_OUT4_ENA_SHIFT,
+	0, &wm8400_dapm_out4mix_controls[0],
+	ARRAY_SIZE(wm8400_dapm_out4mix_controls)),
+
+/* ROPMIX */
+SND_SOC_DAPM_MIXER("ROPMIX", WM8400_POWER_MANAGEMENT_3, WM8400_ROP_ENA_SHIFT,
+		   0, &wm8400_dapm_ropmix_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_ropmix_controls)),
+
+/* RONMIX */
+SND_SOC_DAPM_MIXER("RONMIX", WM8400_POWER_MANAGEMENT_3, WM8400_RON_ENA_SHIFT,
+		   0, &wm8400_dapm_ronmix_controls[0],
+		   ARRAY_SIZE(wm8400_dapm_ronmix_controls)),
+
+/* ROMIX */
+SND_SOC_DAPM_MIXER_E("ROMIX", WM8400_POWER_MANAGEMENT_3,
+		     WM8400_ROMIX_ENA_SHIFT,
+		     0, &wm8400_dapm_romix_controls[0],
+		     ARRAY_SIZE(wm8400_dapm_romix_controls),
+		     outmixer_event, SND_SOC_DAPM_PRE_REG),
+
+/* LOUT PGA */
+SND_SOC_DAPM_PGA("LOUT PGA", WM8400_POWER_MANAGEMENT_1, WM8400_LOUT_ENA_SHIFT,
+		 0, NULL, 0),
+
+/* ROUT PGA */
+SND_SOC_DAPM_PGA("ROUT PGA", WM8400_POWER_MANAGEMENT_1, WM8400_ROUT_ENA_SHIFT,
+		 0, NULL, 0),
+
+/* LOPGA */
+SND_SOC_DAPM_PGA("LOPGA", WM8400_POWER_MANAGEMENT_3, WM8400_LOPGA_ENA_SHIFT, 0,
+	NULL, 0),
+
+/* ROPGA */
+SND_SOC_DAPM_PGA("ROPGA", WM8400_POWER_MANAGEMENT_3, WM8400_ROPGA_ENA_SHIFT, 0,
+	NULL, 0),
+
+/* MICBIAS */
+SND_SOC_DAPM_MICBIAS("MICBIAS", WM8400_POWER_MANAGEMENT_1,
+	WM8400_MIC1BIAS_ENA_SHIFT, 0),
+
+SND_SOC_DAPM_OUTPUT("LON"),
+SND_SOC_DAPM_OUTPUT("LOP"),
+SND_SOC_DAPM_OUTPUT("OUT3"),
+SND_SOC_DAPM_OUTPUT("LOUT"),
+SND_SOC_DAPM_OUTPUT("SPKN"),
+SND_SOC_DAPM_OUTPUT("SPKP"),
+SND_SOC_DAPM_OUTPUT("ROUT"),
+SND_SOC_DAPM_OUTPUT("OUT4"),
+SND_SOC_DAPM_OUTPUT("ROP"),
+SND_SOC_DAPM_OUTPUT("RON"),
+
+SND_SOC_DAPM_OUTPUT("Internal DAC Sink"),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Make DACs turn on when playing even if not mixed into any outputs */
+	{"Internal DAC Sink", NULL, "Left DAC"},
+	{"Internal DAC Sink", NULL, "Right DAC"},
+
+	/* Make ADCs turn on when recording
+	 * even if not mixed from any inputs */
+	{"Left ADC", NULL, "Internal ADC Source"},
+	{"Right ADC", NULL, "Internal ADC Source"},
+
+	/* Input Side */
+	/* LIN12 PGA */
+	{"LIN12 PGA", "LIN1 Switch", "LIN1"},
+	{"LIN12 PGA", "LIN2 Switch", "LIN2"},
+	/* LIN34 PGA */
+	{"LIN34 PGA", "LIN3 Switch", "LIN3"},
+	{"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"},
+	/* INMIXL */
+	{"INMIXL", "Record Left Volume", "LOMIX"},
+	{"INMIXL", "LIN2 Volume", "LIN2"},
+	{"INMIXL", "LINPGA12 Switch", "LIN12 PGA"},
+	{"INMIXL", "LINPGA34 Switch", "LIN34 PGA"},
+	/* AILNMUX */
+	{"AILNMUX", "INMIXL Mix", "INMIXL"},
+	{"AILNMUX", "DIFFINL Mix", "LIN12 PGA"},
+	{"AILNMUX", "DIFFINL Mix", "LIN34 PGA"},
+	{"AILNMUX", "RXVOICE Mix", "LIN4/RXN"},
+	{"AILNMUX", "RXVOICE Mix", "RIN4/RXP"},
+	/* ADC */
+	{"Left ADC", NULL, "AILNMUX"},
+
+	/* RIN12 PGA */
+	{"RIN12 PGA", "RIN1 Switch", "RIN1"},
+	{"RIN12 PGA", "RIN2 Switch", "RIN2"},
+	/* RIN34 PGA */
+	{"RIN34 PGA", "RIN3 Switch", "RIN3"},
+	{"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"},
+	/* INMIXL */
+	{"INMIXR", "Record Right Volume", "ROMIX"},
+	{"INMIXR", "RIN2 Volume", "RIN2"},
+	{"INMIXR", "RINPGA12 Switch", "RIN12 PGA"},
+	{"INMIXR", "RINPGA34 Switch", "RIN34 PGA"},
+	/* AIRNMUX */
+	{"AIRNMUX", "INMIXR Mix", "INMIXR"},
+	{"AIRNMUX", "DIFFINR Mix", "RIN12 PGA"},
+	{"AIRNMUX", "DIFFINR Mix", "RIN34 PGA"},
+	{"AIRNMUX", "RXVOICE Mix", "LIN4/RXN"},
+	{"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"},
+	/* ADC */
+	{"Right ADC", NULL, "AIRNMUX"},
+
+	/* LOMIX */
+	{"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"},
+	{"LOMIX", "LOMIX LIN3 Bypass Switch", "LIN3"},
+	{"LOMIX", "LOMIX LIN12 PGA Bypass Switch", "LIN12 PGA"},
+	{"LOMIX", "LOMIX RIN12 PGA Bypass Switch", "RIN12 PGA"},
+	{"LOMIX", "LOMIX Right ADC Bypass Switch", "AIRNMUX"},
+	{"LOMIX", "LOMIX Left ADC Bypass Switch", "AILNMUX"},
+	{"LOMIX", "LOMIX Left DAC Switch", "Left DAC"},
+
+	/* ROMIX */
+	{"ROMIX", "ROMIX RIN3 Bypass Switch", "RIN3"},
+	{"ROMIX", "ROMIX LIN3 Bypass Switch", "LIN3"},
+	{"ROMIX", "ROMIX LIN12 PGA Bypass Switch", "LIN12 PGA"},
+	{"ROMIX", "ROMIX RIN12 PGA Bypass Switch", "RIN12 PGA"},
+	{"ROMIX", "ROMIX Right ADC Bypass Switch", "AIRNMUX"},
+	{"ROMIX", "ROMIX Left ADC Bypass Switch", "AILNMUX"},
+	{"ROMIX", "ROMIX Right DAC Switch", "Right DAC"},
+
+	/* SPKMIX */
+	{"SPKMIX", "SPKMIX LIN2 Bypass Switch", "LIN2"},
+	{"SPKMIX", "SPKMIX RIN2 Bypass Switch", "RIN2"},
+	{"SPKMIX", "SPKMIX LADC Bypass Switch", "AILNMUX"},
+	{"SPKMIX", "SPKMIX RADC Bypass Switch", "AIRNMUX"},
+	{"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"},
+	{"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"},
+	{"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"},
+	{"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"},
+
+	/* LONMIX */
+	{"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"},
+	{"LONMIX", "LONMIX Right Mixer PGA Switch", "ROPGA"},
+	{"LONMIX", "LONMIX Inverted LOP Switch", "LOPMIX"},
+
+	/* LOPMIX */
+	{"LOPMIX", "LOPMIX Right Mic Bypass Switch", "RIN12 PGA"},
+	{"LOPMIX", "LOPMIX Left Mic Bypass Switch", "LIN12 PGA"},
+	{"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"},
+
+	/* OUT3MIX */
+	{"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXN"},
+	{"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"},
+
+	/* OUT4MIX */
+	{"OUT4MIX", "OUT4MIX Right Out PGA Switch", "ROPGA"},
+	{"OUT4MIX", "OUT4MIX RIN4/RXP Bypass Switch", "RIN4/RXP"},
+
+	/* RONMIX */
+	{"RONMIX", "RONMIX Right Mixer PGA Switch", "ROPGA"},
+	{"RONMIX", "RONMIX Left Mixer PGA Switch", "LOPGA"},
+	{"RONMIX", "RONMIX Inverted ROP Switch", "ROPMIX"},
+
+	/* ROPMIX */
+	{"ROPMIX", "ROPMIX Left Mic Bypass Switch", "LIN12 PGA"},
+	{"ROPMIX", "ROPMIX Right Mic Bypass Switch", "RIN12 PGA"},
+	{"ROPMIX", "ROPMIX Right Mixer PGA Switch", "ROPGA"},
+
+	/* Out Mixer PGAs */
+	{"LOPGA", NULL, "LOMIX"},
+	{"ROPGA", NULL, "ROMIX"},
+
+	{"LOUT PGA", NULL, "LOMIX"},
+	{"ROUT PGA", NULL, "ROMIX"},
+
+	/* Output Pins */
+	{"LON", NULL, "LONMIX"},
+	{"LOP", NULL, "LOPMIX"},
+	{"OUT3", NULL, "OUT3MIX"},
+	{"LOUT", NULL, "LOUT PGA"},
+	{"SPKN", NULL, "SPKMIX"},
+	{"ROUT", NULL, "ROUT PGA"},
+	{"OUT4", NULL, "OUT4MIX"},
+	{"ROP", NULL, "ROPMIX"},
+	{"RON", NULL, "RONMIX"},
+};
+
+static int wm8400_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, wm8400_dapm_widgets,
+				  ARRAY_SIZE(wm8400_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_new_widgets(codec);
+	return 0;
+}
+
+/*
+ * Clock after FLL and dividers
+ */
+static int wm8400_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+		int clk_id, unsigned int freq, int dir)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct wm8400_priv *wm8400 = codec->private_data;
+
+	wm8400->sysclk = freq;
+	return 0;
+}
+
+struct fll_factors {
+	u16 n;
+	u16 k;
+	u16 outdiv;
+	u16 fratio;
+	u16 freq_ref;
+};
+
+#define FIXED_FLL_SIZE ((1 << 16) * 10)
+
+static int fll_factors(struct wm8400_priv *wm8400, struct fll_factors *factors,
+		       unsigned int Fref, unsigned int Fout)
+{
+	u64 Kpart;
+	unsigned int K, Nmod, target;
+
+	factors->outdiv = 2;
+	while (Fout * factors->outdiv <  90000000 ||
+	       Fout * factors->outdiv > 100000000) {
+		factors->outdiv *= 2;
+		if (factors->outdiv > 32) {
+			dev_err(wm8400->wm8400->dev,
+				"Unsupported FLL output frequency %dHz\n",
+				Fout);
+			return -EINVAL;
+		}
+	}
+	target = Fout * factors->outdiv;
+	factors->outdiv = factors->outdiv >> 2;
+
+	if (Fref < 48000)
+		factors->freq_ref = 1;
+	else
+		factors->freq_ref = 0;
+
+	if (Fref < 1000000)
+		factors->fratio = 9;
+	else
+		factors->fratio = 0;
+
+	/* Ensure we have a fractional part */
+	do {
+		if (Fref < 1000000)
+			factors->fratio--;
+		else
+			factors->fratio++;
+
+		if (factors->fratio < 1 || factors->fratio > 8) {
+			dev_err(wm8400->wm8400->dev,
+				"Unable to calculate FRATIO\n");
+			return -EINVAL;
+		}
+
+		factors->n = target / (Fref * factors->fratio);
+		Nmod = target % (Fref * factors->fratio);
+	} while (Nmod == 0);
+
+	/* Calculate fractional part - scale up so we can round. */
+	Kpart = FIXED_FLL_SIZE * (long long)Nmod;
+
+	do_div(Kpart, (Fref * factors->fratio));
+
+	K = Kpart & 0xFFFFFFFF;
+
+	if ((K % 10) >= 5)
+		K += 5;
+
+	/* Move down to proper range now rounding is done */
+	factors->k = K / 10;
+
+	dev_dbg(wm8400->wm8400->dev,
+		"FLL: Fref=%d Fout=%d N=%x K=%x, FRATIO=%x OUTDIV=%x\n",
+		Fref, Fout,
+		factors->n, factors->k, factors->fratio, factors->outdiv);
+
+	return 0;
+}
+
+static int wm8400_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+			      unsigned int freq_in, unsigned int freq_out)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	struct wm8400_priv *wm8400 = codec->private_data;
+	struct fll_factors factors;
+	int ret;
+	u16 reg;
+
+	if (freq_in == wm8400->fll_in && freq_out == wm8400->fll_out)
+		return 0;
+
+	if (freq_out != 0) {
+		ret = fll_factors(wm8400, &factors, freq_in, freq_out);
+		if (ret != 0)
+			return ret;
+	}
+
+	wm8400->fll_out = freq_out;
+	wm8400->fll_in = freq_in;
+
+	/* We *must* disable the FLL before any changes */
+	reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_2);
+	reg &= ~WM8400_FLL_ENA;
+	wm8400_write(codec, WM8400_POWER_MANAGEMENT_2, reg);
+
+	reg = wm8400_read(codec, WM8400_FLL_CONTROL_1);
+	reg &= ~WM8400_FLL_OSC_ENA;
+	wm8400_write(codec, WM8400_FLL_CONTROL_1, reg);
+
+	if (freq_out == 0)
+		return 0;
+
+	reg &= ~(WM8400_FLL_REF_FREQ | WM8400_FLL_FRATIO_MASK);
+	reg |= WM8400_FLL_FRAC | factors.fratio;
+	reg |= factors.freq_ref << WM8400_FLL_REF_FREQ_SHIFT;
+	wm8400_write(codec, WM8400_FLL_CONTROL_1, reg);
+
+	wm8400_write(codec, WM8400_FLL_CONTROL_2, factors.k);
+	wm8400_write(codec, WM8400_FLL_CONTROL_3, factors.n);
+
+	reg = wm8400_read(codec, WM8400_FLL_CONTROL_4);
+	reg &= WM8400_FLL_OUTDIV_MASK;
+	reg |= factors.outdiv;
+	wm8400_write(codec, WM8400_FLL_CONTROL_4, reg);
+
+	return 0;
+}
+
+/*
+ * Sets ADC and Voice DAC format.
+ */
+static int wm8400_set_dai_fmt(struct snd_soc_dai *codec_dai,
+		unsigned int fmt)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 audio1, audio3;
+
+	audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1);
+	audio3 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_3);
+
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		audio3 &= ~WM8400_AIF_MSTR1;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		audio3 |= WM8400_AIF_MSTR1;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	audio1 &= ~WM8400_AIF_FMT_MASK;
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		audio1 |= WM8400_AIF_FMT_I2S;
+		audio1 &= ~WM8400_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+		audio1 |= WM8400_AIF_FMT_RIGHTJ;
+		audio1 &= ~WM8400_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		audio1 |= WM8400_AIF_FMT_LEFTJ;
+		audio1 &= ~WM8400_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		audio1 |= WM8400_AIF_FMT_DSP;
+		audio1 &= ~WM8400_AIF_LRCLK_INV;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		audio1 |= WM8400_AIF_FMT_DSP | WM8400_AIF_LRCLK_INV;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1);
+	wm8400_write(codec, WM8400_AUDIO_INTERFACE_3, audio3);
+	return 0;
+}
+
+static int wm8400_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
+		int div_id, int div)
+{
+	struct snd_soc_codec *codec = codec_dai->codec;
+	u16 reg;
+
+	switch (div_id) {
+	case WM8400_MCLK_DIV:
+		reg = wm8400_read(codec, WM8400_CLOCKING_2) &
+			~WM8400_MCLK_DIV_MASK;
+		wm8400_write(codec, WM8400_CLOCKING_2, reg | div);
+		break;
+	case WM8400_DACCLK_DIV:
+		reg = wm8400_read(codec, WM8400_CLOCKING_2) &
+			~WM8400_DAC_CLKDIV_MASK;
+		wm8400_write(codec, WM8400_CLOCKING_2, reg | div);
+		break;
+	case WM8400_ADCCLK_DIV:
+		reg = wm8400_read(codec, WM8400_CLOCKING_2) &
+			~WM8400_ADC_CLKDIV_MASK;
+		wm8400_write(codec, WM8400_CLOCKING_2, reg | div);
+		break;
+	case WM8400_BCLK_DIV:
+		reg = wm8400_read(codec, WM8400_CLOCKING_1) &
+			~WM8400_BCLK_DIV_MASK;
+		wm8400_write(codec, WM8400_CLOCKING_1, reg | div);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+/*
+ * Set PCM DAI bit size and sample rate.
+ */
+static int wm8400_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params,
+	struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	u16 audio1 = wm8400_read(codec, WM8400_AUDIO_INTERFACE_1);
+
+	audio1 &= ~WM8400_AIF_WL_MASK;
+	/* bit size */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		break;
+	case SNDRV_PCM_FORMAT_S20_3LE:
+		audio1 |= WM8400_AIF_WL_20BITS;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		audio1 |= WM8400_AIF_WL_24BITS;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		audio1 |= WM8400_AIF_WL_32BITS;
+		break;
+	}
+
+	wm8400_write(codec, WM8400_AUDIO_INTERFACE_1, audio1);
+	return 0;
+}
+
+static int wm8400_mute(struct snd_soc_dai *dai, int mute)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	u16 val = wm8400_read(codec, WM8400_DAC_CTRL) & ~WM8400_DAC_MUTE;
+
+	if (mute)
+		wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE);
+	else
+		wm8400_write(codec, WM8400_DAC_CTRL, val);
+
+	return 0;
+}
+
+/* TODO: set bias for best performance at standby */
+static int wm8400_set_bias_level(struct snd_soc_codec *codec,
+				 enum snd_soc_bias_level level)
+{
+	struct wm8400_priv *wm8400 = codec->private_data;
+	u16 val;
+	int ret;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+		break;
+
+	case SND_SOC_BIAS_PREPARE:
+		/* VMID=2*50k */
+		val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) &
+			~WM8400_VMID_MODE_MASK;
+		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x2);
+		break;
+
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			ret = regulator_bulk_enable(ARRAY_SIZE(power),
+						    &power[0]);
+			if (ret != 0) {
+				dev_err(wm8400->wm8400->dev,
+					"Failed to enable regulators: %d\n",
+					ret);
+				return ret;
+			}
+
+			wm8400_write(codec, WM8400_POWER_MANAGEMENT_1,
+				     WM8400_CODEC_ENA | WM8400_SYSCLK_ENA);
+
+			/* Enable POBCTRL, SOFT_ST, VMIDTOG and BUFDCOPEN */
+			wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+				     WM8400_BUFDCOPEN | WM8400_POBCTRL);
+
+			msleep(50);
+
+			/* Enable VREF & VMID at 2x50k */
+			val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
+			val |= 0x2 | WM8400_VREF_ENA;
+			wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+			/* Enable BUFIOEN */
+			wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+				     WM8400_BUFDCOPEN | WM8400_POBCTRL |
+				     WM8400_BUFIOEN);
+
+			/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
+			wm8400_write(codec, WM8400_ANTIPOP2, WM8400_BUFIOEN);
+		}
+
+		/* VMID=2*300k */
+		val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1) &
+			~WM8400_VMID_MODE_MASK;
+		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val | 0x4);
+		break;
+
+	case SND_SOC_BIAS_OFF:
+		/* Enable POBCTRL and SOFT_ST */
+		wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+			WM8400_POBCTRL | WM8400_BUFIOEN);
+
+		/* Enable POBCTRL, SOFT_ST and BUFDCOPEN */
+		wm8400_write(codec, WM8400_ANTIPOP2, WM8400_SOFTST |
+			WM8400_BUFDCOPEN | WM8400_POBCTRL |
+			WM8400_BUFIOEN);
+
+		/* mute DAC */
+		val = wm8400_read(codec, WM8400_DAC_CTRL);
+		wm8400_write(codec, WM8400_DAC_CTRL, val | WM8400_DAC_MUTE);
+
+		/* Enable any disabled outputs */
+		val = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
+		val |= WM8400_SPK_ENA | WM8400_OUT3_ENA |
+			WM8400_OUT4_ENA | WM8400_LOUT_ENA |
+			WM8400_ROUT_ENA;
+		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+		/* Disable VMID */
+		val &= ~WM8400_VMID_MODE_MASK;
+		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+		msleep(300);
+
+		/* Enable all output discharge bits */
+		wm8400_write(codec, WM8400_ANTIPOP1, WM8400_DIS_LLINE |
+			WM8400_DIS_RLINE | WM8400_DIS_OUT3 |
+			WM8400_DIS_OUT4 | WM8400_DIS_LOUT |
+			WM8400_DIS_ROUT);
+
+		/* Disable VREF */
+		val &= ~WM8400_VREF_ENA;
+		wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, val);
+
+		/* disable POBCTRL, SOFT_ST and BUFDCOPEN */
+		wm8400_write(codec, WM8400_ANTIPOP2, 0x0);
+
+		ret = regulator_bulk_disable(ARRAY_SIZE(power),
+					     &power[0]);
+		if (ret != 0)
+			return ret;
+
+		break;
+	}
+
+	codec->bias_level = level;
+	return 0;
+}
+
+#define WM8400_RATES SNDRV_PCM_RATE_8000_96000
+
+#define WM8400_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+	SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_ops wm8400_dai_ops = {
+	.hw_params = wm8400_hw_params,
+	.digital_mute = wm8400_mute,
+	.set_fmt = wm8400_set_dai_fmt,
+	.set_clkdiv = wm8400_set_dai_clkdiv,
+	.set_sysclk = wm8400_set_dai_sysclk,
+	.set_pll = wm8400_set_dai_pll,
+};
+
+/*
+ * The WM8400 supports 2 different and mutually exclusive DAI
+ * configurations.
+ *
+ * 1. ADC/DAC on Primary Interface
+ * 2. ADC on Primary Interface/DAC on secondary
+ */
+struct snd_soc_dai wm8400_dai = {
+/* ADC/DAC on primary */
+	.name = "WM8400 ADC/DAC Primary",
+	.id = 1,
+	.playback = {
+		.stream_name = "Playback",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8400_RATES,
+		.formats = WM8400_FORMATS,
+	},
+	.capture = {
+		.stream_name = "Capture",
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = WM8400_RATES,
+		.formats = WM8400_FORMATS,
+	},
+	.ops = &wm8400_dai_ops,
+};
+EXPORT_SYMBOL_GPL(wm8400_dai);
+
+static int wm8400_suspend(struct platform_device *pdev, pm_message_t state)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	wm8400_set_bias_level(codec, SND_SOC_BIAS_OFF);
+
+	return 0;
+}
+
+static int wm8400_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	wm8400_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	return 0;
+}
+
+static struct snd_soc_codec *wm8400_codec;
+
+static int wm8400_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret;
+
+	if (!wm8400_codec) {
+		dev_err(&pdev->dev, "wm8400 not yet discovered\n");
+		return -ENODEV;
+	}
+	codec = wm8400_codec;
+
+	socdev->card->codec = codec;
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	wm8400_add_controls(codec);
+	wm8400_add_widgets(codec);
+
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		dev_err(&pdev->dev, "failed to register card\n");
+		goto card_err;
+	}
+
+	return ret;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+pcm_err:
+	return ret;
+}
+
+/* power down chip */
+static int wm8400_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8400 = {
+	.probe =	wm8400_probe,
+	.remove =	wm8400_remove,
+	.suspend =	wm8400_suspend,
+	.resume =	wm8400_resume,
+};
+
+static void wm8400_probe_deferred(struct work_struct *work)
+{
+	struct wm8400_priv *priv = container_of(work, struct wm8400_priv,
+						work);
+	struct snd_soc_codec *codec = &priv->codec;
+	int ret;
+
+	/* charge output caps */
+	wm8400_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	/* We're done, tell the subsystem. */
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(priv->wm8400->dev,
+			"Failed to register codec: %d\n", ret);
+		goto err;
+	}
+
+	ret = snd_soc_register_dai(&wm8400_dai);
+	if (ret != 0) {
+		dev_err(priv->wm8400->dev,
+			"Failed to register DAI: %d\n", ret);
+		goto err_codec;
+	}
+
+	return;
+
+err_codec:
+	snd_soc_unregister_codec(codec);
+err:
+	wm8400_set_bias_level(codec, SND_SOC_BIAS_OFF);
+}
+
+static int wm8400_codec_probe(struct platform_device *dev)
+{
+	struct wm8400_priv *priv;
+	int ret;
+	u16 reg;
+	struct snd_soc_codec *codec;
+
+	priv = kzalloc(sizeof(struct wm8400_priv), GFP_KERNEL);
+	if (priv == NULL)
+		return -ENOMEM;
+
+	codec = &priv->codec;
+	codec->private_data = priv;
+	codec->control_data = dev->dev.driver_data;
+	priv->wm8400 = dev->dev.driver_data;
+
+	ret = regulator_bulk_get(priv->wm8400->dev,
+				 ARRAY_SIZE(power), &power[0]);
+	if (ret != 0) {
+		dev_err(&dev->dev, "Failed to get regulators: %d\n", ret);
+	        goto err;
+	}
+
+	codec->dev = &dev->dev;
+	wm8400_dai.dev = &dev->dev;
+
+	codec->name = "WM8400";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8400_read;
+	codec->write = wm8400_write;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm8400_set_bias_level;
+	codec->dai = &wm8400_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = WM8400_REGISTER_COUNT;
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+	INIT_WORK(&priv->work, wm8400_probe_deferred);
+
+	wm8400_codec_reset(codec);
+
+	reg = wm8400_read(codec, WM8400_POWER_MANAGEMENT_1);
+	wm8400_write(codec, WM8400_POWER_MANAGEMENT_1, reg | WM8400_CODEC_ENA);
+
+	/* Latch volume update bits */
+	reg = wm8400_read(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME);
+	wm8400_write(codec, WM8400_LEFT_LINE_INPUT_1_2_VOLUME,
+		     reg & WM8400_IPVU);
+	reg = wm8400_read(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME);
+	wm8400_write(codec, WM8400_RIGHT_LINE_INPUT_1_2_VOLUME,
+		     reg & WM8400_IPVU);
+
+	wm8400_write(codec, WM8400_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
+	wm8400_write(codec, WM8400_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
+
+	wm8400_codec = codec;
+
+	if (!schedule_work(&priv->work)) {
+		ret = -EINVAL;
+		goto err_regulator;
+	}
+
+	return 0;
+
+err_regulator:
+	wm8400_codec = NULL;
+	regulator_bulk_free(ARRAY_SIZE(power), power);
+err:
+	kfree(priv);
+	return ret;
+}
+
+static int __exit wm8400_codec_remove(struct platform_device *dev)
+{
+	struct wm8400_priv *priv = wm8400_codec->private_data;
+	u16 reg;
+
+	snd_soc_unregister_dai(&wm8400_dai);
+	snd_soc_unregister_codec(wm8400_codec);
+
+	reg = wm8400_read(wm8400_codec, WM8400_POWER_MANAGEMENT_1);
+	wm8400_write(wm8400_codec, WM8400_POWER_MANAGEMENT_1,
+		     reg & (~WM8400_CODEC_ENA));
+
+	regulator_bulk_free(ARRAY_SIZE(power), power);
+	kfree(priv);
+
+	wm8400_codec = NULL;
+
+	return 0;
+}
+
+static struct platform_driver wm8400_codec_driver = {
+	.driver = {
+		.name = "wm8400-codec",
+		.owner = THIS_MODULE,
+	},
+	.probe = wm8400_codec_probe,
+	.remove	= __exit_p(wm8400_codec_remove),
+};
+
+static int __init wm8400_codec_init(void)
+{
+	return platform_driver_register(&wm8400_codec_driver);
+}
+module_init(wm8400_codec_init);
+
+static void __exit wm8400_codec_exit(void)
+{
+	platform_driver_unregister(&wm8400_codec_driver);
+}
+module_exit(wm8400_codec_exit);
+
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8400);
+
+MODULE_DESCRIPTION("ASoC WM8400 driver");
+MODULE_AUTHOR("Mark Brown");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:wm8400-codec");
diff --git a/sound/soc/codecs/wm8400.h b/sound/soc/codecs/wm8400.h
new file mode 100644
index 000000000000..79c5934d4776
--- /dev/null
+++ b/sound/soc/codecs/wm8400.h
@@ -0,0 +1,62 @@
+/*
+ * wm8400.h  --  audio driver for WM8400
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ *
+ */
+
+#ifndef _WM8400_CODEC_H
+#define _WM8400_CODEC_H
+
+#define WM8400_MCLK_DIV 0
+#define WM8400_DACCLK_DIV 1
+#define WM8400_ADCCLK_DIV 2
+#define WM8400_BCLK_DIV 3
+
+#define WM8400_MCLK_DIV_1 0x400
+#define WM8400_MCLK_DIV_2 0x800
+
+#define WM8400_DAC_CLKDIV_1    0x00
+#define WM8400_DAC_CLKDIV_1_5  0x04
+#define WM8400_DAC_CLKDIV_2    0x08
+#define WM8400_DAC_CLKDIV_3    0x0c
+#define WM8400_DAC_CLKDIV_4    0x10
+#define WM8400_DAC_CLKDIV_5_5  0x14
+#define WM8400_DAC_CLKDIV_6    0x18
+
+#define WM8400_ADC_CLKDIV_1    0x00
+#define WM8400_ADC_CLKDIV_1_5  0x20
+#define WM8400_ADC_CLKDIV_2    0x40
+#define WM8400_ADC_CLKDIV_3    0x60
+#define WM8400_ADC_CLKDIV_4    0x80
+#define WM8400_ADC_CLKDIV_5_5  0xa0
+#define WM8400_ADC_CLKDIV_6    0xc0
+
+
+#define WM8400_BCLK_DIV_1                       (0x0 << 1)
+#define WM8400_BCLK_DIV_1_5                     (0x1 << 1)
+#define WM8400_BCLK_DIV_2                       (0x2 << 1)
+#define WM8400_BCLK_DIV_3                       (0x3 << 1)
+#define WM8400_BCLK_DIV_4                       (0x4 << 1)
+#define WM8400_BCLK_DIV_5_5                     (0x5 << 1)
+#define WM8400_BCLK_DIV_6                       (0x6 << 1)
+#define WM8400_BCLK_DIV_8                       (0x7 << 1)
+#define WM8400_BCLK_DIV_11                      (0x8 << 1)
+#define WM8400_BCLK_DIV_12                      (0x9 << 1)
+#define WM8400_BCLK_DIV_16                      (0xA << 1)
+#define WM8400_BCLK_DIV_22                      (0xB << 1)
+#define WM8400_BCLK_DIV_24                      (0xC << 1)
+#define WM8400_BCLK_DIV_32                      (0xD << 1)
+#define WM8400_BCLK_DIV_44                      (0xE << 1)
+#define WM8400_BCLK_DIV_48                      (0xF << 1)
+
+extern struct snd_soc_dai wm8400_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8400;
+
+#endif
diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c
index 40f8238df717..6a4cea09c45d 100644
--- a/sound/soc/codecs/wm8510.c
+++ b/sound/soc/codecs/wm8510.c
@@ -171,22 +171,6 @@ SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST,  8, 1, 0),
 SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 1),
 };
 
-/* add non dapm controls */
-static int wm8510_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8510_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8510_snd_controls[i], codec,
-					NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /* Speaker Output Mixer */
 static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = {
 SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0),
@@ -352,7 +336,7 @@ static int wm8510_set_dai_pll(struct snd_soc_dai *codec_dai,
 		return 0;
 	}
 
-	pll_factors(freq_out*8, freq_in);
+	pll_factors(freq_out*4, freq_in);
 
 	wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n);
 	wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18);
@@ -383,7 +367,7 @@ static int wm8510_set_dai_clkdiv(struct snd_soc_dai *codec_dai,
 		wm8510_write(codec, WM8510_GPIO, reg | div);
 		break;
 	case WM8510_MCLKDIV:
-		reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1f;
+		reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x11f;
 		wm8510_write(codec, WM8510_CLOCK, reg | div);
 		break;
 	case WM8510_ADCCLK:
@@ -468,7 +452,7 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 iface = wm8510_read_reg_cache(codec, WM8510_IFACE) & 0x19f;
 	u16 adn = wm8510_read_reg_cache(codec, WM8510_ADD) & 0x1f1;
 
@@ -570,6 +554,14 @@ static int wm8510_set_bias_level(struct snd_soc_codec *codec,
 #define WM8510_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops wm8510_dai_ops = {
+	.hw_params	= wm8510_pcm_hw_params,
+	.digital_mute	= wm8510_mute,
+	.set_fmt	= wm8510_set_dai_fmt,
+	.set_clkdiv	= wm8510_set_dai_clkdiv,
+	.set_pll	= wm8510_set_dai_pll,
+};
+
 struct snd_soc_dai wm8510_dai = {
 	.name = "WM8510 HiFi",
 	.playback = {
@@ -584,20 +576,14 @@ struct snd_soc_dai wm8510_dai = {
 		.channels_max = 2,
 		.rates = WM8510_RATES,
 		.formats = WM8510_FORMATS,},
-	.ops = {
-		.hw_params = wm8510_pcm_hw_params,
-		.digital_mute = wm8510_mute,
-		.set_fmt = wm8510_set_dai_fmt,
-		.set_clkdiv = wm8510_set_dai_clkdiv,
-		.set_pll = wm8510_set_dai_pll,
-	},
+	.ops = &wm8510_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8510_dai);
 
 static int wm8510_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -606,7 +592,7 @@ static int wm8510_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8510_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -628,7 +614,7 @@ static int wm8510_resume(struct platform_device *pdev)
  */
 static int wm8510_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 
 	codec->name = "WM8510";
@@ -656,7 +642,8 @@ static int wm8510_init(struct snd_soc_device *socdev)
 	/* power on device */
 	codec->bias_level = SND_SOC_BIAS_OFF;
 	wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	wm8510_add_controls(codec);
+	snd_soc_add_controls(codec, wm8510_snd_controls,
+				ARRAY_SIZE(wm8510_snd_controls));
 	wm8510_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -685,7 +672,7 @@ static int wm8510_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = wm8510_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -766,7 +753,7 @@ err_driver:
 static int __devinit wm8510_spi_probe(struct spi_device *spi)
 {
 	struct snd_soc_device *socdev = wm8510_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	codec->control_data = spi;
@@ -832,7 +819,7 @@ static int wm8510_probe(struct platform_device *pdev)
 	if (codec == NULL)
 		return -ENOMEM;
 
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -862,7 +849,7 @@ static int wm8510_probe(struct platform_device *pdev)
 static int wm8510_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8510_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index d004e5845298..442ea6f160fc 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -1,7 +1,7 @@
 /*
  * wm8580.c  --  WM8580 ALSA Soc Audio driver
  *
- * Copyright 2008 Wolfson Microelectronics PLC.
+ * Copyright 2008, 2009 Wolfson Microelectronics PLC.
  *
  *  This program is free software; you can redistribute  it and/or modify it
  *  under  the terms of  the GNU General  Public License as published by the
@@ -35,19 +35,6 @@
 
 #include "wm8580.h"
 
-#define WM8580_VERSION "0.1"
-
-struct pll_state {
-	unsigned int in;
-	unsigned int out;
-};
-
-/* codec private data */
-struct wm8580_priv {
-	struct pll_state a;
-	struct pll_state b;
-};
-
 /* WM8580 register space */
 #define WM8580_PLLA1                         0x00
 #define WM8580_PLLA2                         0x01
@@ -102,6 +89,8 @@ struct wm8580_priv {
 #define WM8580_READBACK                      0x34
 #define WM8580_RESET                         0x35
 
+#define WM8580_MAX_REGISTER                  0x35
+
 /* PLLB4 (register 7h) */
 #define WM8580_PLLB4_MCLKOUTSRC_MASK   0x60
 #define WM8580_PLLB4_MCLKOUTSRC_PLLA   0x20
@@ -193,6 +182,20 @@ static const u16 wm8580_reg[] = {
 	0x0000, 0x0000 /*R53*/
 };
 
+struct pll_state {
+	unsigned int in;
+	unsigned int out;
+};
+
+/* codec private data */
+struct wm8580_priv {
+	struct snd_soc_codec codec;
+	u16 reg_cache[WM8580_MAX_REGISTER + 1];
+	struct pll_state a;
+	struct pll_state b;
+};
+
+
 /*
  * read wm8580 register cache
  */
@@ -200,7 +203,7 @@ static inline unsigned int wm8580_read_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg)
 {
 	u16 *cache = codec->reg_cache;
-	BUG_ON(reg > ARRAY_SIZE(wm8580_reg));
+	BUG_ON(reg >= ARRAY_SIZE(wm8580_reg));
 	return cache[reg];
 }
 
@@ -223,7 +226,7 @@ static int wm8580_write(struct snd_soc_codec *codec, unsigned int reg,
 {
 	u8 data[2];
 
-	BUG_ON(reg > ARRAY_SIZE(wm8580_reg));
+	BUG_ON(reg >= ARRAY_SIZE(wm8580_reg));
 
 	/* Registers are 9 bits wide */
 	value &= 0x1ff;
@@ -330,20 +333,6 @@ SOC_DOUBLE("ADC Mute Switch", WM8580_ADC_CONTROL1, 0, 1, 1, 0),
 SOC_SINGLE("ADC High-Pass Filter Switch", WM8580_ADC_CONTROL1, 4, 1, 0),
 };
 
-/* Add non-DAPM controls */
-static int wm8580_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8580_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm8580_snd_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
 static const struct snd_soc_dapm_widget wm8580_dapm_widgets[] = {
 SND_SOC_DAPM_DAC("DAC1", "Playback", WM8580_PWRDN1, 2, 1),
 SND_SOC_DAPM_DAC("DAC2", "Playback", WM8580_PWRDN1, 3, 1),
@@ -553,7 +542,7 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 paifb = wm8580_read(codec, WM8580_PAIF3 + dai->id);
 
 	paifb &= ~WM8580_AIF_LENGTH_MASK;
@@ -771,8 +760,22 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec,
 	switch (level) {
 	case SND_SOC_BIAS_ON:
 	case SND_SOC_BIAS_PREPARE:
+		break;
+
 	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF) {
+			/* Power up and get individual control of the DACs */
+			reg = wm8580_read(codec, WM8580_PWRDN1);
+			reg &= ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD);
+			wm8580_write(codec, WM8580_PWRDN1, reg);
+
+			/* Make VMID high impedence */
+			reg = wm8580_read(codec,  WM8580_ADC_CONTROL1);
+			reg &= ~0x100;
+			wm8580_write(codec, WM8580_ADC_CONTROL1, reg);
+		}
 		break;
+
 	case SND_SOC_BIAS_OFF:
 		reg = wm8580_read(codec, WM8580_PWRDN1);
 		wm8580_write(codec, WM8580_PWRDN1, reg | WM8580_PWRDN1_PWDN);
@@ -785,6 +788,21 @@ static int wm8580_set_bias_level(struct snd_soc_codec *codec,
 #define WM8580_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops wm8580_dai_ops_playback = {
+	.hw_params	= wm8580_paif_hw_params,
+	.set_fmt	= wm8580_set_paif_dai_fmt,
+	.set_clkdiv	= wm8580_set_dai_clkdiv,
+	.set_pll	= wm8580_set_dai_pll,
+	.digital_mute	= wm8580_digital_mute,
+};
+
+static struct snd_soc_dai_ops wm8580_dai_ops_capture = {
+	.hw_params	= wm8580_paif_hw_params,
+	.set_fmt	= wm8580_set_paif_dai_fmt,
+	.set_clkdiv	= wm8580_set_dai_clkdiv,
+	.set_pll	= wm8580_set_dai_pll,
+};
+
 struct snd_soc_dai wm8580_dai[] = {
 	{
 		.name = "WM8580 PAIFRX",
@@ -796,13 +814,7 @@ struct snd_soc_dai wm8580_dai[] = {
 			.rates = SNDRV_PCM_RATE_8000_192000,
 			.formats = WM8580_FORMATS,
 		},
-		.ops = {
-			 .hw_params = wm8580_paif_hw_params,
-			 .set_fmt = wm8580_set_paif_dai_fmt,
-			 .set_clkdiv = wm8580_set_dai_clkdiv,
-			 .set_pll = wm8580_set_dai_pll,
-			 .digital_mute = wm8580_digital_mute,
-		 },
+		.ops = &wm8580_dai_ops_playback,
 	},
 	{
 		.name = "WM8580 PAIFTX",
@@ -814,109 +826,168 @@ struct snd_soc_dai wm8580_dai[] = {
 			.rates = SNDRV_PCM_RATE_8000_192000,
 			.formats = WM8580_FORMATS,
 		},
-		.ops = {
-			 .hw_params = wm8580_paif_hw_params,
-			 .set_fmt = wm8580_set_paif_dai_fmt,
-			 .set_clkdiv = wm8580_set_dai_clkdiv,
-			 .set_pll = wm8580_set_dai_pll,
-		 },
+		.ops = &wm8580_dai_ops_capture,
 	},
 };
 EXPORT_SYMBOL_GPL(wm8580_dai);
 
-/*
- * initialise the WM8580 driver
- * register the mixer and dsp interfaces with the kernel
- */
-static int wm8580_init(struct snd_soc_device *socdev)
+static struct snd_soc_codec *wm8580_codec;
+
+static int wm8580_probe(struct platform_device *pdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
 	int ret = 0;
 
-	codec->name = "WM8580";
-	codec->owner = THIS_MODULE;
-	codec->read = wm8580_read_reg_cache;
-	codec->write = wm8580_write;
-	codec->set_bias_level = wm8580_set_bias_level;
-	codec->dai = wm8580_dai;
-	codec->num_dai = ARRAY_SIZE(wm8580_dai);
-	codec->reg_cache_size = ARRAY_SIZE(wm8580_reg);
-	codec->reg_cache = kmemdup(wm8580_reg, sizeof(wm8580_reg),
-				   GFP_KERNEL);
-
-	if (codec->reg_cache == NULL)
-		return -ENOMEM;
-
-	/* Get the codec into a known state */
-	wm8580_write(codec, WM8580_RESET, 0);
-
-	/* Power up and get individual control of the DACs */
-	wm8580_write(codec, WM8580_PWRDN1, wm8580_read(codec, WM8580_PWRDN1) &
-		     ~(WM8580_PWRDN1_PWDN | WM8580_PWRDN1_ALLDACPD));
+	if (wm8580_codec == NULL) {
+		dev_err(&pdev->dev, "Codec device not registered\n");
+		return -ENODEV;
+	}
 
-	/* Make VMID high impedence */
-	wm8580_write(codec, WM8580_ADC_CONTROL1,
-		     wm8580_read(codec,  WM8580_ADC_CONTROL1) & ~0x100);
+	socdev->card->codec = wm8580_codec;
+	codec = wm8580_codec;
 
 	/* register pcms */
-	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1,
-			       SNDRV_DEFAULT_STR1);
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
-		printk(KERN_ERR "wm8580: failed to create pcms\n");
+		dev_err(codec->dev, "failed to create pcms: %d\n", ret);
 		goto pcm_err;
 	}
 
-	wm8580_add_controls(codec);
+	snd_soc_add_controls(codec, wm8580_snd_controls,
+			     ARRAY_SIZE(wm8580_snd_controls));
 	wm8580_add_widgets(codec);
-
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
-		printk(KERN_ERR "wm8580: failed to register card\n");
+		dev_err(codec->dev, "failed to register card: %d\n", ret);
 		goto card_err;
 	}
+
 	return ret;
 
 card_err:
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
 pcm_err:
-	kfree(codec->reg_cache);
 	return ret;
 }
 
-/* If the i2c layer weren't so broken, we could pass this kind of data
-   around */
-static struct snd_soc_device *wm8580_socdev;
+/* power down chip */
+static int wm8580_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	return 0;
+}
 
-/*
- * WM8580 2 wire address is determined by GPIO5
- * state during powerup.
- *    low  = 0x1a
- *    high = 0x1b
- */
+struct snd_soc_codec_device soc_codec_dev_wm8580 = {
+	.probe = 	wm8580_probe,
+	.remove = 	wm8580_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580);
+
+static int wm8580_register(struct wm8580_priv *wm8580)
+{
+	int ret, i;
+	struct snd_soc_codec *codec = &wm8580->codec;
+
+	if (wm8580_codec) {
+		dev_err(codec->dev, "Another WM8580 is registered\n");
+		ret = -EINVAL;
+		goto err;
+	}
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
 
+	codec->private_data = wm8580;
+	codec->name = "WM8580";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8580_read_reg_cache;
+	codec->write = wm8580_write;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm8580_set_bias_level;
+	codec->dai = wm8580_dai;
+	codec->num_dai = ARRAY_SIZE(wm8580_dai);
+	codec->reg_cache_size = ARRAY_SIZE(wm8580->reg_cache);
+	codec->reg_cache = &wm8580->reg_cache;
+
+	memcpy(codec->reg_cache, wm8580_reg, sizeof(wm8580_reg));
+
+	/* Get the codec into a known state */
+	ret = wm8580_write(codec, WM8580_RESET, 0);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to reset codec: %d\n", ret);
+		goto err;
+	}
+
+	for (i = 0; i < ARRAY_SIZE(wm8580_dai); i++)
+		wm8580_dai[i].dev = codec->dev;
+
+	wm8580_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+	wm8580_codec = codec;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		goto err;
+	}
+
+	ret = snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+		goto err_codec;
+	}
+
+	return 0;
+
+err_codec:
+	snd_soc_unregister_codec(codec);
+err:
+	kfree(wm8580);
+	return ret;
+}
+
+static void wm8580_unregister(struct wm8580_priv *wm8580)
+{
+	wm8580_set_bias_level(&wm8580->codec, SND_SOC_BIAS_OFF);
+	snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
+	snd_soc_unregister_codec(&wm8580->codec);
+	kfree(wm8580);
+	wm8580_codec = NULL;
+}
+
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 static int wm8580_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
-	struct snd_soc_device *socdev = wm8580_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int ret;
+	struct wm8580_priv *wm8580;
+	struct snd_soc_codec *codec;
 
-	i2c_set_clientdata(i2c, codec);
+	wm8580 = kzalloc(sizeof(struct wm8580_priv), GFP_KERNEL);
+	if (wm8580 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8580->codec;
+	codec->hw_write = (hw_write_t)i2c_master_send;
+
+	i2c_set_clientdata(i2c, wm8580);
 	codec->control_data = i2c;
 
-	ret = wm8580_init(socdev);
-	if (ret < 0)
-		dev_err(&i2c->dev, "failed to initialise WM8580\n");
-	return ret;
+	codec->dev = &i2c->dev;
+
+	return wm8580_register(wm8580);
 }
 
 static int wm8580_i2c_remove(struct i2c_client *client)
 {
-	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-	kfree(codec->reg_cache);
+	struct wm8580_priv *wm8580 = i2c_get_clientdata(client);
+	wm8580_unregister(wm8580);
 	return 0;
 }
 
@@ -928,129 +999,35 @@ MODULE_DEVICE_TABLE(i2c, wm8580_i2c_id);
 
 static struct i2c_driver wm8580_i2c_driver = {
 	.driver = {
-		.name = "WM8580 I2C Codec",
+		.name = "wm8580",
 		.owner = THIS_MODULE,
 	},
 	.probe =    wm8580_i2c_probe,
 	.remove =   wm8580_i2c_remove,
 	.id_table = wm8580_i2c_id,
 };
+#endif
 
-static int wm8580_add_i2c_device(struct platform_device *pdev,
-				 const struct wm8580_setup_data *setup)
+static int __init wm8580_modinit(void)
 {
-	struct i2c_board_info info;
-	struct i2c_adapter *adapter;
-	struct i2c_client *client;
 	int ret;
 
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 	ret = i2c_add_driver(&wm8580_i2c_driver);
 	if (ret != 0) {
-		dev_err(&pdev->dev, "can't add i2c driver\n");
-		return ret;
-	}
-
-	memset(&info, 0, sizeof(struct i2c_board_info));
-	info.addr = setup->i2c_address;
-	strlcpy(info.type, "wm8580", I2C_NAME_SIZE);
-
-	adapter = i2c_get_adapter(setup->i2c_bus);
-	if (!adapter) {
-		dev_err(&pdev->dev, "can't get i2c adapter %d\n",
-			setup->i2c_bus);
-		goto err_driver;
-	}
-
-	client = i2c_new_device(adapter, &info);
-	i2c_put_adapter(adapter);
-	if (!client) {
-		dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
-			(unsigned int)info.addr);
-		goto err_driver;
+		pr_err("Failed to register WM8580 I2C driver: %d\n", ret);
 	}
-
-	return 0;
-
-err_driver:
-	i2c_del_driver(&wm8580_i2c_driver);
-	return -ENODEV;
-}
 #endif
 
-static int wm8580_probe(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct wm8580_setup_data *setup;
-	struct snd_soc_codec *codec;
-	struct wm8580_priv *wm8580;
-	int ret = 0;
-
-	pr_info("WM8580 Audio Codec %s\n", WM8580_VERSION);
-
-	setup = socdev->codec_data;
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
-		return -ENOMEM;
-
-	wm8580 = kzalloc(sizeof(struct wm8580_priv), GFP_KERNEL);
-	if (wm8580 == NULL) {
-		kfree(codec);
-		return -ENOMEM;
-	}
-
-	codec->private_data = wm8580;
-	socdev->codec = codec;
-	mutex_init(&codec->mutex);
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
-	wm8580_socdev = socdev;
-
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	if (setup->i2c_address) {
-		codec->hw_write = (hw_write_t)i2c_master_send;
-		ret = wm8580_add_i2c_device(pdev, setup);
-	}
-#else
-		/* Add other interfaces here */
-#endif
-	return ret;
-}
-
-/* power down chip */
-static int wm8580_remove(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
-
-	if (codec->control_data)
-		wm8580_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	i2c_unregister_device(codec->control_data);
-	i2c_del_driver(&wm8580_i2c_driver);
-#endif
-	kfree(codec->private_data);
-	kfree(codec);
-
 	return 0;
 }
-
-struct snd_soc_codec_device soc_codec_dev_wm8580 = {
-	.probe = 	wm8580_probe,
-	.remove = 	wm8580_remove,
-};
-EXPORT_SYMBOL_GPL(soc_codec_dev_wm8580);
-
-static int __init wm8580_modinit(void)
-{
-	return snd_soc_register_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
-}
 module_init(wm8580_modinit);
 
 static void __exit wm8580_exit(void)
 {
-	snd_soc_unregister_dais(wm8580_dai, ARRAY_SIZE(wm8580_dai));
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&wm8580_i2c_driver);
+#endif
 }
 module_exit(wm8580_exit);
 
diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h
index 09e4422f6f2f..0dfb5ddde6a2 100644
--- a/sound/soc/codecs/wm8580.h
+++ b/sound/soc/codecs/wm8580.h
@@ -28,11 +28,6 @@
 #define WM8580_CLKSRC_OSC  4
 #define WM8580_CLKSRC_NONE 5
 
-struct wm8580_setup_data {
-	int i2c_bus;
-	unsigned short i2c_address;
-};
-
 #define WM8580_DAI_PAIFRX 0
 #define WM8580_DAI_PAIFTX 1
 
diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c
index 80b11983e137..e7ff2121ede9 100644
--- a/sound/soc/codecs/wm8728.c
+++ b/sound/soc/codecs/wm8728.c
@@ -47,7 +47,7 @@ static inline unsigned int wm8728_read_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg)
 {
 	u16 *cache = codec->reg_cache;
-	BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults));
+	BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults));
 	return cache[reg];
 }
 
@@ -55,7 +55,7 @@ static inline void wm8728_write_reg_cache(struct snd_soc_codec *codec,
 	u16 reg, unsigned int value)
 {
 	u16 *cache = codec->reg_cache;
-	BUG_ON(reg > ARRAY_SIZE(wm8728_reg_defaults));
+	BUG_ON(reg >= ARRAY_SIZE(wm8728_reg_defaults));
 	cache[reg] = value;
 }
 
@@ -92,21 +92,6 @@ SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8728_DACLVOL, WM8728_DACRVOL,
 SOC_SINGLE("Deemphasis", WM8728_DACCTL, 1, 1, 0),
 };
 
-static int wm8728_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8728_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm8728_snd_controls[i],
-						codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /*
  * DAPM controls.
  */
@@ -152,7 +137,7 @@ static int wm8728_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 dac = wm8728_read_reg_cache(codec, WM8728_DACCTL);
 
 	dac &= ~0x18;
@@ -259,6 +244,12 @@ static int wm8728_set_bias_level(struct snd_soc_codec *codec,
 #define WM8728_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
+static struct snd_soc_dai_ops wm8728_dai_ops = {
+	.hw_params	= wm8728_hw_params,
+	.digital_mute	= wm8728_mute,
+	.set_fmt	= wm8728_set_dai_fmt,
+};
+
 struct snd_soc_dai wm8728_dai = {
 	.name = "WM8728",
 	.playback = {
@@ -268,18 +259,14 @@ struct snd_soc_dai wm8728_dai = {
 		.rates = WM8728_RATES,
 		.formats = WM8728_FORMATS,
 	},
-	.ops = {
-		 .hw_params = wm8728_hw_params,
-		 .digital_mute = wm8728_mute,
-		 .set_fmt = wm8728_set_dai_fmt,
-	}
+	.ops = &wm8728_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8728_dai);
 
 static int wm8728_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
@@ -289,7 +276,7 @@ static int wm8728_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8728_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8728_set_bias_level(codec, codec->suspend_bias_level);
 
@@ -302,7 +289,7 @@ static int wm8728_resume(struct platform_device *pdev)
  */
 static int wm8728_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 
 	codec->name = "WM8728";
@@ -330,7 +317,8 @@ static int wm8728_init(struct snd_soc_device *socdev)
 	/* power on device */
 	wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-	wm8728_add_controls(codec);
+	snd_soc_add_controls(codec, wm8728_snd_controls,
+				ARRAY_SIZE(wm8728_snd_controls));
 	wm8728_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -363,7 +351,7 @@ static int wm8728_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = wm8728_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -444,7 +432,7 @@ err_driver:
 static int __devinit wm8728_spi_probe(struct spi_device *spi)
 {
 	struct snd_soc_device *socdev = wm8728_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	codec->control_data = spi;
@@ -508,7 +496,7 @@ static int wm8728_probe(struct platform_device *pdev)
 	if (codec == NULL)
 		return -ENOMEM;
 
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -541,7 +529,7 @@ static int wm8728_probe(struct platform_device *pdev)
 static int wm8728_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8728_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c
index c444b9f2701e..e043e3f60008 100644
--- a/sound/soc/codecs/wm8731.c
+++ b/sound/soc/codecs/wm8731.c
@@ -29,15 +29,20 @@
 
 #include "wm8731.h"
 
-#define WM8731_VERSION "0.13"
-
+static struct snd_soc_codec *wm8731_codec;
 struct snd_soc_codec_device soc_codec_dev_wm8731;
 
 /* codec private data */
 struct wm8731_priv {
+	struct snd_soc_codec codec;
+	u16 reg_cache[WM8731_CACHEREGNUM];
 	unsigned int sysclk;
 };
 
+#ifdef CONFIG_SPI_MASTER
+static int wm8731_spi_write(struct spi_device *spi, const char *data, int len);
+#endif
+
 /*
  * wm8731 register cache
  * We can't read the WM8731 register space when we are
@@ -129,22 +134,6 @@ SOC_SINGLE("Store DC Offset Switch", WM8731_APDIGI, 4, 1, 0),
 SOC_ENUM("Playback De-emphasis", wm8731_enum[1]),
 };
 
-/* add non dapm controls */
-static int wm8731_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8731_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm8731_snd_controls[i],
-						codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 /* Output Mixer */
 static const struct snd_kcontrol_new wm8731_output_mixer_controls[] = {
 SOC_DAPM_SINGLE("Line Bypass Switch", WM8731_APANA, 3, 1, 0),
@@ -269,7 +258,7 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8731_priv *wm8731 = codec->private_data;
 	u16 iface = wm8731_read_reg_cache(codec, WM8731_IFACE) & 0xfff3;
 	int i = get_coeff(wm8731->sysclk, params_rate(params));
@@ -299,7 +288,7 @@ static int wm8731_pcm_prepare(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	/* set active */
 	wm8731_write(codec, WM8731_ACTIVE, 0x0001);
@@ -312,7 +301,7 @@ static void wm8731_shutdown(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	/* deactivate */
 	if (!codec->active) {
@@ -414,21 +403,19 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai,
 static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 				 enum snd_soc_bias_level level)
 {
-	u16 reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f;
+	u16 reg;
 
 	switch (level) {
 	case SND_SOC_BIAS_ON:
-		/* vref/mid, osc on, dac unmute */
-		wm8731_write(codec, WM8731_PWR, reg);
 		break;
 	case SND_SOC_BIAS_PREPARE:
 		break;
 	case SND_SOC_BIAS_STANDBY:
-		/* everything off except vref/vmid, */
+		/* Clear PWROFF, gate CLKOUT, everything else as-is */
+		reg = wm8731_read_reg_cache(codec, WM8731_PWR) & 0xff7f;
 		wm8731_write(codec, WM8731_PWR, reg | 0x0040);
 		break;
 	case SND_SOC_BIAS_OFF:
-		/* everything off, dac mute, inactive */
 		wm8731_write(codec, WM8731_ACTIVE, 0x0);
 		wm8731_write(codec, WM8731_PWR, 0xffff);
 		break;
@@ -446,6 +433,15 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec,
 #define WM8731_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
+static struct snd_soc_dai_ops wm8731_dai_ops = {
+	.prepare	= wm8731_pcm_prepare,
+	.hw_params	= wm8731_hw_params,
+	.shutdown	= wm8731_shutdown,
+	.digital_mute	= wm8731_mute,
+	.set_sysclk	= wm8731_set_dai_sysclk,
+	.set_fmt	= wm8731_set_dai_fmt,
+};
+
 struct snd_soc_dai wm8731_dai = {
 	.name = "WM8731",
 	.playback = {
@@ -460,21 +456,14 @@ struct snd_soc_dai wm8731_dai = {
 		.channels_max = 2,
 		.rates = WM8731_RATES,
 		.formats = WM8731_FORMATS,},
-	.ops = {
-		.prepare = wm8731_pcm_prepare,
-		.hw_params = wm8731_hw_params,
-		.shutdown = wm8731_shutdown,
-		.digital_mute = wm8731_mute,
-		.set_sysclk = wm8731_set_dai_sysclk,
-		.set_fmt = wm8731_set_dai_fmt,
-	}
+	.ops = &wm8731_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8731_dai);
 
 static int wm8731_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8731_write(codec, WM8731_ACTIVE, 0x0);
 	wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
@@ -484,7 +473,7 @@ static int wm8731_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8731_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -500,54 +489,33 @@ static int wm8731_resume(struct platform_device *pdev)
 	return 0;
 }
 
-/*
- * initialise the WM8731 driver
- * register the mixer and dsp interfaces with the kernel
- */
-static int wm8731_init(struct snd_soc_device *socdev)
+static int wm8731_probe(struct platform_device *pdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
-	int reg, ret = 0;
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
 
-	codec->name = "WM8731";
-	codec->owner = THIS_MODULE;
-	codec->read = wm8731_read_reg_cache;
-	codec->write = wm8731_write;
-	codec->set_bias_level = wm8731_set_bias_level;
-	codec->dai = &wm8731_dai;
-	codec->num_dai = 1;
-	codec->reg_cache_size = ARRAY_SIZE(wm8731_reg);
-	codec->reg_cache = kmemdup(wm8731_reg, sizeof(wm8731_reg), GFP_KERNEL);
-	if (codec->reg_cache == NULL)
-		return -ENOMEM;
+	if (wm8731_codec == NULL) {
+		dev_err(&pdev->dev, "Codec device not registered\n");
+		return -ENODEV;
+	}
 
-	wm8731_reset(codec);
+	socdev->card->codec = wm8731_codec;
+	codec = wm8731_codec;
 
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
 	if (ret < 0) {
-		printk(KERN_ERR "wm8731: failed to create pcms\n");
+		dev_err(codec->dev, "failed to create pcms: %d\n", ret);
 		goto pcm_err;
 	}
 
-	/* power on device */
-	wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-
-	/* set the update bits */
-	reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V);
-	wm8731_write(codec, WM8731_LOUT1V, reg & ~0x0100);
-	reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V);
-	wm8731_write(codec, WM8731_ROUT1V, reg & ~0x0100);
-	reg = wm8731_read_reg_cache(codec, WM8731_LINVOL);
-	wm8731_write(codec, WM8731_LINVOL, reg & ~0x0100);
-	reg = wm8731_read_reg_cache(codec, WM8731_RINVOL);
-	wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100);
-
-	wm8731_add_controls(codec);
+	snd_soc_add_controls(codec, wm8731_snd_controls,
+			     ARRAY_SIZE(wm8731_snd_controls));
 	wm8731_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
-		printk(KERN_ERR "wm8731: failed to register card\n");
+		dev_err(codec->dev, "failed to register card: %d\n", ret);
 		goto card_err;
 	}
 
@@ -557,133 +525,109 @@ card_err:
 	snd_soc_free_pcms(socdev);
 	snd_soc_dapm_free(socdev);
 pcm_err:
-	kfree(codec->reg_cache);
 	return ret;
 }
 
-static struct snd_soc_device *wm8731_socdev;
-
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-
-/*
- * WM8731 2 wire address is determined by GPIO5
- * state during powerup.
- *    low  = 0x1a
- *    high = 0x1b
- */
-
-static int wm8731_i2c_probe(struct i2c_client *i2c,
-			    const struct i2c_device_id *id)
+/* power down chip */
+static int wm8731_remove(struct platform_device *pdev)
 {
-	struct snd_soc_device *socdev = wm8731_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int ret;
-
-	i2c_set_clientdata(i2c, codec);
-	codec->control_data = i2c;
-
-	ret = wm8731_init(socdev);
-	if (ret < 0)
-		pr_err("failed to initialise WM8731\n");
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 
-	return ret;
-}
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
 
-static int wm8731_i2c_remove(struct i2c_client *client)
-{
-	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-	kfree(codec->reg_cache);
 	return 0;
 }
 
-static const struct i2c_device_id wm8731_i2c_id[] = {
-	{ "wm8731", 0 },
-	{ }
-};
-MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id);
-
-static struct i2c_driver wm8731_i2c_driver = {
-	.driver = {
-		.name = "WM8731 I2C Codec",
-		.owner = THIS_MODULE,
-	},
-	.probe =    wm8731_i2c_probe,
-	.remove =   wm8731_i2c_remove,
-	.id_table = wm8731_i2c_id,
+struct snd_soc_codec_device soc_codec_dev_wm8731 = {
+	.probe = 	wm8731_probe,
+	.remove = 	wm8731_remove,
+	.suspend = 	wm8731_suspend,
+	.resume =	wm8731_resume,
 };
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731);
 
-static int wm8731_add_i2c_device(struct platform_device *pdev,
-				 const struct wm8731_setup_data *setup)
+static int wm8731_register(struct wm8731_priv *wm8731)
 {
-	struct i2c_board_info info;
-	struct i2c_adapter *adapter;
-	struct i2c_client *client;
 	int ret;
+	struct snd_soc_codec *codec = &wm8731->codec;
+	u16 reg;
 
-	ret = i2c_add_driver(&wm8731_i2c_driver);
-	if (ret != 0) {
-		dev_err(&pdev->dev, "can't add i2c driver\n");
-		return ret;
+	if (wm8731_codec) {
+		dev_err(codec->dev, "Another WM8731 is registered\n");
+		return -EINVAL;
 	}
 
-	memset(&info, 0, sizeof(struct i2c_board_info));
-	info.addr = setup->i2c_address;
-	strlcpy(info.type, "wm8731", I2C_NAME_SIZE);
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
 
-	adapter = i2c_get_adapter(setup->i2c_bus);
-	if (!adapter) {
-		dev_err(&pdev->dev, "can't get i2c adapter %d\n",
-			setup->i2c_bus);
-		goto err_driver;
-	}
+	codec->private_data = wm8731;
+	codec->name = "WM8731";
+	codec->owner = THIS_MODULE;
+	codec->read = wm8731_read_reg_cache;
+	codec->write = wm8731_write;
+	codec->bias_level = SND_SOC_BIAS_OFF;
+	codec->set_bias_level = wm8731_set_bias_level;
+	codec->dai = &wm8731_dai;
+	codec->num_dai = 1;
+	codec->reg_cache_size = WM8731_CACHEREGNUM;
+	codec->reg_cache = &wm8731->reg_cache;
 
-	client = i2c_new_device(adapter, &info);
-	i2c_put_adapter(adapter);
-	if (!client) {
-		dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
-			(unsigned int)info.addr);
-		goto err_driver;
+	memcpy(codec->reg_cache, wm8731_reg, sizeof(wm8731_reg));
+
+	ret = wm8731_reset(codec);
+	if (ret < 0) {
+		dev_err(codec->dev, "Failed to issue reset\n");
+		return ret;
 	}
 
-	return 0;
+	wm8731_dai.dev = codec->dev;
 
-err_driver:
-	i2c_del_driver(&wm8731_i2c_driver);
-	return -ENODEV;
-}
-#endif
+	wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
-#if defined(CONFIG_SPI_MASTER)
-static int __devinit wm8731_spi_probe(struct spi_device *spi)
-{
-	struct snd_soc_device *socdev = wm8731_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int ret;
+	/* Latch the update bits */
+	reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V);
+	wm8731_write(codec, WM8731_LOUT1V, reg & ~0x0100);
+	reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V);
+	wm8731_write(codec, WM8731_ROUT1V, reg & ~0x0100);
+	reg = wm8731_read_reg_cache(codec, WM8731_LINVOL);
+	wm8731_write(codec, WM8731_LINVOL, reg & ~0x0100);
+	reg = wm8731_read_reg_cache(codec, WM8731_RINVOL);
+	wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100);
 
-	codec->control_data = spi;
+	/* Disable bypass path by default */
+	reg = wm8731_read_reg_cache(codec, WM8731_APANA);
+	wm8731_write(codec, WM8731_APANA, reg & ~0x4);
 
-	ret = wm8731_init(socdev);
-	if (ret < 0)
-		dev_err(&spi->dev, "failed to initialise WM8731\n");
+	wm8731_codec = codec;
 
-	return ret;
-}
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		return ret;
+	}
+
+	ret = snd_soc_register_dai(&wm8731_dai);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
+		snd_soc_unregister_codec(codec);
+		return ret;
+	}
 
-static int __devexit wm8731_spi_remove(struct spi_device *spi)
-{
 	return 0;
 }
 
-static struct spi_driver wm8731_spi_driver = {
-	.driver = {
-		.name	= "wm8731",
-		.bus	= &spi_bus_type,
-		.owner	= THIS_MODULE,
-	},
-	.probe		= wm8731_spi_probe,
-	.remove		= __devexit_p(wm8731_spi_remove),
-};
+static void wm8731_unregister(struct wm8731_priv *wm8731)
+{
+	wm8731_set_bias_level(&wm8731->codec, SND_SOC_BIAS_OFF);
+	snd_soc_unregister_dai(&wm8731_dai);
+	snd_soc_unregister_codec(&wm8731->codec);
+	kfree(wm8731);
+	wm8731_codec = NULL;
+}
 
+#if defined(CONFIG_SPI_MASTER)
 static int wm8731_spi_write(struct spi_device *spi, const char *data, int len)
 {
 	struct spi_transfer t;
@@ -707,101 +651,121 @@ static int wm8731_spi_write(struct spi_device *spi, const char *data, int len)
 
 	return len;
 }
-#endif /* CONFIG_SPI_MASTER */
 
-static int wm8731_probe(struct platform_device *pdev)
+static int __devinit wm8731_spi_probe(struct spi_device *spi)
 {
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct wm8731_setup_data *setup;
 	struct snd_soc_codec *codec;
 	struct wm8731_priv *wm8731;
-	int ret = 0;
-
-	pr_info("WM8731 Audio Codec %s", WM8731_VERSION);
-
-	setup = socdev->codec_data;
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
-		return -ENOMEM;
 
 	wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL);
-	if (wm8731 == NULL) {
-		kfree(codec);
+	if (wm8731 == NULL)
 		return -ENOMEM;
-	}
 
-	codec->private_data = wm8731;
-	socdev->codec = codec;
-	mutex_init(&codec->mutex);
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
+	codec = &wm8731->codec;
+	codec->control_data = spi;
+	codec->hw_write = (hw_write_t)wm8731_spi_write;
+	codec->dev = &spi->dev;
 
-	wm8731_socdev = socdev;
-	ret = -ENODEV;
+	spi->dev.driver_data = wm8731;
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	if (setup->i2c_address) {
-		codec->hw_write = (hw_write_t)i2c_master_send;
-		ret = wm8731_add_i2c_device(pdev, setup);
-	}
-#endif
-#if defined(CONFIG_SPI_MASTER)
-	if (setup->spi) {
-		codec->hw_write = (hw_write_t)wm8731_spi_write;
-		ret = spi_register_driver(&wm8731_spi_driver);
-		if (ret != 0)
-			printk(KERN_ERR "can't add spi driver");
-	}
-#endif
-
-	if (ret != 0) {
-		kfree(codec->private_data);
-		kfree(codec);
-	}
-	return ret;
+	return wm8731_register(wm8731);
 }
 
-/* power down chip */
-static int wm8731_remove(struct platform_device *pdev)
+static int __devexit wm8731_spi_remove(struct spi_device *spi)
 {
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct wm8731_priv *wm8731 = spi->dev.driver_data;
 
-	if (codec->control_data)
-		wm8731_set_bias_level(codec, SND_SOC_BIAS_OFF);
+	wm8731_unregister(wm8731);
+
+	return 0;
+}
+
+static struct spi_driver wm8731_spi_driver = {
+	.driver = {
+		.name	= "wm8731",
+		.bus	= &spi_bus_type,
+		.owner	= THIS_MODULE,
+	},
+	.probe		= wm8731_spi_probe,
+	.remove		= __devexit_p(wm8731_spi_remove),
+};
+#endif /* CONFIG_SPI_MASTER */
 
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	i2c_unregister_device(codec->control_data);
-	i2c_del_driver(&wm8731_i2c_driver);
-#endif
-#if defined(CONFIG_SPI_MASTER)
-	spi_unregister_driver(&wm8731_spi_driver);
-#endif
-	kfree(codec->private_data);
-	kfree(codec);
+static __devinit int wm8731_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
+{
+	struct wm8731_priv *wm8731;
+	struct snd_soc_codec *codec;
+
+	wm8731 = kzalloc(sizeof(struct wm8731_priv), GFP_KERNEL);
+	if (wm8731 == NULL)
+		return -ENOMEM;
+
+	codec = &wm8731->codec;
+	codec->hw_write = (hw_write_t)i2c_master_send;
 
+	i2c_set_clientdata(i2c, wm8731);
+	codec->control_data = i2c;
+
+	codec->dev = &i2c->dev;
+
+	return wm8731_register(wm8731);
+}
+
+static __devexit int wm8731_i2c_remove(struct i2c_client *client)
+{
+	struct wm8731_priv *wm8731 = i2c_get_clientdata(client);
+	wm8731_unregister(wm8731);
 	return 0;
 }
 
-struct snd_soc_codec_device soc_codec_dev_wm8731 = {
-	.probe = 	wm8731_probe,
-	.remove = 	wm8731_remove,
-	.suspend = 	wm8731_suspend,
-	.resume =	wm8731_resume,
+static const struct i2c_device_id wm8731_i2c_id[] = {
+	{ "wm8731", 0 },
+	{ }
 };
-EXPORT_SYMBOL_GPL(soc_codec_dev_wm8731);
+MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id);
+
+static struct i2c_driver wm8731_i2c_driver = {
+	.driver = {
+		.name = "WM8731 I2C Codec",
+		.owner = THIS_MODULE,
+	},
+	.probe =    wm8731_i2c_probe,
+	.remove =   __devexit_p(wm8731_i2c_remove),
+	.id_table = wm8731_i2c_id,
+};
+#endif
 
 static int __init wm8731_modinit(void)
 {
-	return snd_soc_register_dai(&wm8731_dai);
+	int ret;
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	ret = i2c_add_driver(&wm8731_i2c_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "Failed to register WM8731 I2C driver: %d\n",
+		       ret);
+	}
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	ret = spi_register_driver(&wm8731_spi_driver);
+	if (ret != 0) {
+		printk(KERN_ERR "Failed to register WM8731 SPI driver: %d\n",
+		       ret);
+	}
+#endif
+	return 0;
 }
 module_init(wm8731_modinit);
 
 static void __exit wm8731_exit(void)
 {
-	snd_soc_unregister_dai(&wm8731_dai);
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&wm8731_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&wm8731_spi_driver);
+#endif
 }
 module_exit(wm8731_exit);
 
diff --git a/sound/soc/codecs/wm8731.h b/sound/soc/codecs/wm8731.h
index 95190e9c0c14..cd7b806e8ad0 100644
--- a/sound/soc/codecs/wm8731.h
+++ b/sound/soc/codecs/wm8731.h
@@ -34,12 +34,6 @@
 #define WM8731_SYSCLK	0
 #define WM8731_DAI		0
 
-struct wm8731_setup_data {
-	int            spi;
-	int            i2c_bus;
-	unsigned short i2c_address;
-};
-
 extern struct snd_soc_dai wm8731_dai;
 extern struct snd_soc_codec_device soc_codec_dev_wm8731;
 
diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c
index 5997fa68e0d5..b64509b01a49 100644
--- a/sound/soc/codecs/wm8750.c
+++ b/sound/soc/codecs/wm8750.c
@@ -231,21 +231,6 @@ SOC_SINGLE("Mono Playback Volume", WM8750_MOUTV, 0, 127, 0),
 
 };
 
-/* add non dapm controls */
-static int wm8750_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8750_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8750_snd_controls[i],
-						codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 /*
  * DAPM Controls
  */
@@ -619,7 +604,7 @@ static int wm8750_pcm_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8750_priv *wm8750 = codec->private_data;
 	u16 iface = wm8750_read_reg_cache(codec, WM8750_IFACE) & 0x1f3;
 	u16 srate = wm8750_read_reg_cache(codec, WM8750_SRATE) & 0x1c0;
@@ -694,6 +679,13 @@ static int wm8750_set_bias_level(struct snd_soc_codec *codec,
 #define WM8750_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
+static struct snd_soc_dai_ops wm8750_dai_ops = {
+	.hw_params	= wm8750_pcm_hw_params,
+	.digital_mute	= wm8750_mute,
+	.set_fmt	= wm8750_set_dai_fmt,
+	.set_sysclk	= wm8750_set_dai_sysclk,
+};
+
 struct snd_soc_dai wm8750_dai = {
 	.name = "WM8750",
 	.playback = {
@@ -708,12 +700,7 @@ struct snd_soc_dai wm8750_dai = {
 		.channels_max = 2,
 		.rates = WM8750_RATES,
 		.formats = WM8750_FORMATS,},
-	.ops = {
-		.hw_params = wm8750_pcm_hw_params,
-		.digital_mute = wm8750_mute,
-		.set_fmt = wm8750_set_dai_fmt,
-		.set_sysclk = wm8750_set_dai_sysclk,
-	},
+	.ops = &wm8750_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8750_dai);
 
@@ -727,7 +714,7 @@ static void wm8750_work(struct work_struct *work)
 static int wm8750_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -736,7 +723,7 @@ static int wm8750_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8750_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -769,7 +756,7 @@ static int wm8750_resume(struct platform_device *pdev)
  */
 static int wm8750_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int reg, ret = 0;
 
 	codec->name = "WM8750";
@@ -816,7 +803,8 @@ static int wm8750_init(struct snd_soc_device *socdev)
 	reg = wm8750_read_reg_cache(codec, WM8750_RINVOL);
 	wm8750_write(codec, WM8750_RINVOL, reg | 0x0100);
 
-	wm8750_add_controls(codec);
+	snd_soc_add_controls(codec, wm8750_snd_controls,
+				ARRAY_SIZE(wm8750_snd_controls));
 	wm8750_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -850,7 +838,7 @@ static int wm8750_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = wm8750_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -931,7 +919,7 @@ err_driver:
 static int __devinit wm8750_spi_probe(struct spi_device *spi)
 {
 	struct snd_soc_device *socdev = wm8750_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	codec->control_data = spi;
@@ -1003,7 +991,7 @@ static int wm8750_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = wm8750;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -1057,7 +1045,7 @@ static int run_delayed_work(struct delayed_work *dwork)
 static int wm8750_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8750_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c
index 77620ab98756..a6e8f3f7f052 100644
--- a/sound/soc/codecs/wm8753.c
+++ b/sound/soc/codecs/wm8753.c
@@ -51,8 +51,6 @@
 
 #include "wm8753.h"
 
-#define WM8753_VERSION "0.16"
-
 static int caps_charge = 2000;
 module_param(caps_charge, int, 0);
 MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
@@ -60,12 +58,6 @@ MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
 static void wm8753_set_dai_mode(struct snd_soc_codec *codec,
 	unsigned int mode);
 
-/* codec private data */
-struct wm8753_priv {
-	unsigned int sysclk;
-	unsigned int pcmclk;
-};
-
 /*
  * wm8753 register cache
  * We can't read the WM8753 register space when we
@@ -90,6 +82,14 @@ static const u16 wm8753_reg[] = {
 	0x0000, 0x0000
 };
 
+/* codec private data */
+struct wm8753_priv {
+	unsigned int sysclk;
+	unsigned int pcmclk;
+	struct snd_soc_codec codec;
+	u16 reg_cache[ARRAY_SIZE(wm8753_reg)];
+};
+
 /*
  * read wm8753 register cache
  */
@@ -97,7 +97,7 @@ static inline unsigned int wm8753_read_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg)
 {
 	u16 *cache = codec->reg_cache;
-	if (reg < 1 || reg > (ARRAY_SIZE(wm8753_reg) + 1))
+	if (reg < 1 || reg >= (ARRAY_SIZE(wm8753_reg) + 1))
 		return -1;
 	return cache[reg - 1];
 }
@@ -109,7 +109,7 @@ static inline void wm8753_write_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg, unsigned int value)
 {
 	u16 *cache = codec->reg_cache;
-	if (reg < 1 || reg > 0x3f)
+	if (reg < 1 || reg >= (ARRAY_SIZE(wm8753_reg) + 1))
 		return;
 	cache[reg - 1] = value;
 }
@@ -339,21 +339,6 @@ SOC_ENUM("ADC Data Select", wm8753_enum[27]),
 SOC_ENUM("ROUT2 Phase", wm8753_enum[28]),
 };
 
-/* add non dapm controls */
-static int wm8753_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8753_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8753_snd_controls[i],
-						codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 /*
  * _DAPM_ Controls
  */
@@ -927,7 +912,7 @@ static int wm8753_pcm_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8753_priv *wm8753 = codec->private_data;
 	u16 voice = wm8753_read_reg_cache(codec, WM8753_PCM) & 0x01f3;
 	u16 srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x017f;
@@ -1161,7 +1146,7 @@ static int wm8753_i2s_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8753_priv *wm8753 = codec->private_data;
 	u16 srate = wm8753_read_reg_cache(codec, WM8753_SRATE1) & 0x01c0;
 	u16 hifi = wm8753_read_reg_cache(codec, WM8753_HIFI) & 0x01f3;
@@ -1316,6 +1301,51 @@ static int wm8753_set_bias_level(struct snd_soc_codec *codec,
  * 3. Voice disabled - HIFI over HIFI
  * 4. Voice disabled - HIFI over HIFI, uses voice DAI LRC for capture
  */
+static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode1 = {
+	.hw_params	= wm8753_i2s_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode1h_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode1 = {
+	.hw_params	= wm8753_pcm_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode1v_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops wm8753_dai_ops_voice_mode2 = {
+	.hw_params	= wm8753_pcm_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode2_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode3	= {
+	.hw_params	= wm8753_i2s_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode3_4_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
+static struct snd_soc_dai_ops wm8753_dai_ops_hifi_mode4	= {
+	.hw_params	= wm8753_i2s_hw_params,
+	.digital_mute	= wm8753_mute,
+	.set_fmt	= wm8753_mode3_4_set_dai_fmt,
+	.set_clkdiv	= wm8753_set_dai_clkdiv,
+	.set_pll	= wm8753_set_dai_pll,
+	.set_sysclk	= wm8753_set_dai_sysclk,
+};
+
 static const struct snd_soc_dai wm8753_all_dai[] = {
 /* DAI HiFi mode 1 */
 {	.name = "WM8753 HiFi",
@@ -1332,14 +1362,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS},
-	.ops = {
-		.hw_params = wm8753_i2s_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode1h_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_hifi_mode1,
 },
 /* DAI Voice mode 1 */
 {	.name = "WM8753 Voice",
@@ -1356,14 +1379,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
-	.ops = {
-		.hw_params = wm8753_pcm_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode1v_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_voice_mode1,
 },
 /* DAI HiFi mode 2 - dummy */
 {	.name = "WM8753 HiFi",
@@ -1384,14 +1400,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
-	.ops = {
-		.hw_params = wm8753_pcm_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode2_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_voice_mode2,
 },
 /* DAI HiFi mode 3 */
 {	.name = "WM8753 HiFi",
@@ -1408,14 +1417,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
-	.ops = {
-		.hw_params = wm8753_i2s_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode3_4_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_hifi_mode3,
 },
 /* DAI Voice mode 3 - dummy */
 {	.name = "WM8753 Voice",
@@ -1436,14 +1438,7 @@ static const struct snd_soc_dai wm8753_all_dai[] = {
 		.channels_max = 2,
 		.rates = WM8753_RATES,
 		.formats = WM8753_FORMATS,},
-	.ops = {
-		.hw_params = wm8753_i2s_hw_params,
-		.digital_mute = wm8753_mute,
-		.set_fmt = wm8753_mode3_4_set_dai_fmt,
-		.set_clkdiv = wm8753_set_dai_clkdiv,
-		.set_pll = wm8753_set_dai_pll,
-		.set_sysclk = wm8753_set_dai_sysclk,
-	},
+	.ops = &wm8753_dai_ops_hifi_mode4,
 },
 /* DAI Voice mode 4 - dummy */
 {	.name = "WM8753 Voice",
@@ -1466,30 +1461,35 @@ static void wm8753_set_dai_mode(struct snd_soc_codec *codec, unsigned int mode)
 	if (mode < 4) {
 		int playback_active, capture_active, codec_active, pop_wait;
 		void *private_data;
+		struct list_head list;
 
 		playback_active = wm8753_dai[0].playback.active;
 		capture_active = wm8753_dai[0].capture.active;
 		codec_active = wm8753_dai[0].active;
 		private_data = wm8753_dai[0].private_data;
 		pop_wait = wm8753_dai[0].pop_wait;
+		list = wm8753_dai[0].list;
 		wm8753_dai[0] = wm8753_all_dai[mode << 1];
 		wm8753_dai[0].playback.active = playback_active;
 		wm8753_dai[0].capture.active = capture_active;
 		wm8753_dai[0].active = codec_active;
 		wm8753_dai[0].private_data = private_data;
 		wm8753_dai[0].pop_wait = pop_wait;
+		wm8753_dai[0].list = list;
 
 		playback_active = wm8753_dai[1].playback.active;
 		capture_active = wm8753_dai[1].capture.active;
 		codec_active = wm8753_dai[1].active;
 		private_data = wm8753_dai[1].private_data;
 		pop_wait = wm8753_dai[1].pop_wait;
+		list = wm8753_dai[1].list;
 		wm8753_dai[1] = wm8753_all_dai[(mode << 1) + 1];
 		wm8753_dai[1].playback.active = playback_active;
 		wm8753_dai[1].capture.active = capture_active;
 		wm8753_dai[1].active = codec_active;
 		wm8753_dai[1].private_data = private_data;
 		wm8753_dai[1].pop_wait = pop_wait;
+		wm8753_dai[1].list = list;
 	}
 	wm8753_dai[0].codec = codec;
 	wm8753_dai[1].codec = codec;
@@ -1505,7 +1505,7 @@ static void wm8753_work(struct work_struct *work)
 static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	/* we only need to suspend if we are a valid card */
 	if (!codec->card)
@@ -1518,7 +1518,7 @@ static int wm8753_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8753_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -1531,6 +1531,11 @@ static int wm8753_resume(struct platform_device *pdev)
 	for (i = 0; i < ARRAY_SIZE(wm8753_reg); i++) {
 		if (i + 1 == WM8753_RESET)
 			continue;
+
+		/* No point in writing hardware default values back */
+		if (cache[i] == wm8753_reg[i])
+			continue;
+
 		data[0] = ((i + 1) << 1) | ((cache[i] >> 8) & 0x0001);
 		data[1] = cache[i] & 0x00ff;
 		codec->hw_write(codec->control_data, data, 2);
@@ -1549,44 +1554,129 @@ static int wm8753_resume(struct platform_device *pdev)
 	return 0;
 }
 
+static struct snd_soc_codec *wm8753_codec;
+
+static int wm8753_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	if (!wm8753_codec) {
+		dev_err(&pdev->dev, "WM8753 codec not yet registered\n");
+		return -EINVAL;
+	}
+
+	socdev->card->codec = wm8753_codec;
+	codec = wm8753_codec;
+
+	wm8753_set_dai_mode(codec, 0);
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0) {
+		printk(KERN_ERR "wm8753: failed to create pcms\n");
+		goto pcm_err;
+	}
+
+	snd_soc_add_controls(codec, wm8753_snd_controls,
+			     ARRAY_SIZE(wm8753_snd_controls));
+	wm8753_add_widgets(codec);
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "wm8753: failed to register card\n");
+		goto card_err;
+	}
+
+	return 0;
+
+card_err:
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+pcm_err:
+	return ret;
+}
+
 /*
- * initialise the WM8753 driver
- * register the mixer and dsp interfaces with the kernel
+ * This function forces any delayed work to be queued and run.
  */
-static int wm8753_init(struct snd_soc_device *socdev)
+static int run_delayed_work(struct delayed_work *dwork)
+{
+	int ret;
+
+	/* cancel any work waiting to be queued. */
+	ret = cancel_delayed_work(dwork);
+
+	/* if there was any work waiting then we run it now and
+	 * wait for it's completion */
+	if (ret) {
+		schedule_delayed_work(dwork, 0);
+		flush_scheduled_work();
+	}
+	return ret;
+}
+
+/* power down chip */
+static int wm8753_remove(struct platform_device *pdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
-	int reg, ret = 0;
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+
+	snd_soc_free_pcms(socdev);
+	snd_soc_dapm_free(socdev);
+
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm8753 = {
+	.probe = 	wm8753_probe,
+	.remove = 	wm8753_remove,
+	.suspend = 	wm8753_suspend,
+	.resume =	wm8753_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753);
+
+static int wm8753_register(struct wm8753_priv *wm8753)
+{
+	int ret, i;
+	struct snd_soc_codec *codec = &wm8753->codec;
+	u16 reg;
+
+	if (wm8753_codec) {
+		dev_err(codec->dev, "Multiple WM8753 devices not supported\n");
+		ret = -EINVAL;
+		goto err;
+	}
+
+	mutex_init(&codec->mutex);
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
 
 	codec->name = "WM8753";
 	codec->owner = THIS_MODULE;
 	codec->read = wm8753_read_reg_cache;
 	codec->write = wm8753_write;
+	codec->bias_level = SND_SOC_BIAS_STANDBY;
 	codec->set_bias_level = wm8753_set_bias_level;
 	codec->dai = wm8753_dai;
 	codec->num_dai = 2;
-	codec->reg_cache_size = ARRAY_SIZE(wm8753_reg);
-	codec->reg_cache = kmemdup(wm8753_reg, sizeof(wm8753_reg), GFP_KERNEL);
-
-	if (codec->reg_cache == NULL)
-		return -ENOMEM;
-
-	wm8753_set_dai_mode(codec, 0);
+	codec->reg_cache_size = ARRAY_SIZE(wm8753->reg_cache);
+	codec->reg_cache = &wm8753->reg_cache;
+	codec->private_data = wm8753;
 
-	wm8753_reset(codec);
+	memcpy(codec->reg_cache, wm8753_reg, sizeof(codec->reg_cache));
+	INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work);
 
-	/* register pcms */
-	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	ret = wm8753_reset(codec);
 	if (ret < 0) {
-		printk(KERN_ERR "wm8753: failed to create pcms\n");
-		goto pcm_err;
+		dev_err(codec->dev, "Failed to issue reset\n");
+		goto err;
 	}
 
 	/* charge output caps */
 	wm8753_set_bias_level(codec, SND_SOC_BIAS_PREPARE);
-	codec->bias_level = SND_SOC_BIAS_STANDBY;
 	schedule_delayed_work(&codec->delayed_work,
-		msecs_to_jiffies(caps_charge));
+			      msecs_to_jiffies(caps_charge));
 
 	/* set the update bits */
 	reg = wm8753_read_reg_cache(codec, WM8753_LDAC);
@@ -1610,59 +1700,70 @@ static int wm8753_init(struct snd_soc_device *socdev)
 	reg = wm8753_read_reg_cache(codec, WM8753_RINVOL);
 	wm8753_write(codec, WM8753_RINVOL, reg | 0x0100);
 
-	wm8753_add_controls(codec);
-	wm8753_add_widgets(codec);
-	ret = snd_soc_init_card(socdev);
-	if (ret < 0) {
-		printk(KERN_ERR "wm8753: failed to register card\n");
-		goto card_err;
+	wm8753_codec = codec;
+
+	for (i = 0; i < ARRAY_SIZE(wm8753_dai); i++)
+		wm8753_dai[i].dev = codec->dev;
+
+	ret = snd_soc_register_codec(codec);
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register codec: %d\n", ret);
+		goto err;
 	}
 
-	return ret;
+	ret = snd_soc_register_dais(&wm8753_dai[0], ARRAY_SIZE(wm8753_dai));
+	if (ret != 0) {
+		dev_err(codec->dev, "Failed to register DAIs: %d\n", ret);
+		goto err_codec;
+	}
 
-card_err:
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
-pcm_err:
-	kfree(codec->reg_cache);
+	return 0;
+
+err_codec:
+	run_delayed_work(&codec->delayed_work);
+	snd_soc_unregister_codec(codec);
+err:
+	kfree(wm8753);
 	return ret;
 }
 
-/* If the i2c layer weren't so broken, we could pass this kind of data
-   around */
-static struct snd_soc_device *wm8753_socdev;
+static void wm8753_unregister(struct wm8753_priv *wm8753)
+{
+	wm8753_set_bias_level(&wm8753->codec, SND_SOC_BIAS_OFF);
+	run_delayed_work(&wm8753->codec.delayed_work);
+	snd_soc_unregister_dais(&wm8753_dai[0], ARRAY_SIZE(wm8753_dai));
+	snd_soc_unregister_codec(&wm8753->codec);
+	kfree(wm8753);
+	wm8753_codec = NULL;
+}
 
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
 
-/*
- * WM8753 2 wire address is determined by GPIO5
- * state during powerup.
- *    low  = 0x1a
- *    high = 0x1b
- */
-
 static int wm8753_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
-	struct snd_soc_device *socdev = wm8753_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int ret;
+	struct snd_soc_codec *codec;
+	struct wm8753_priv *wm8753;
 
-	i2c_set_clientdata(i2c, codec);
-	codec->control_data = i2c;
+	wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL);
+	if (wm8753 == NULL)
+		return -ENOMEM;
 
-	ret = wm8753_init(socdev);
-	if (ret < 0)
-		pr_err("failed to initialise WM8753\n");
+        codec = &wm8753->codec;
+        codec->hw_write = (hw_write_t)i2c_master_send;
+        codec->control_data = i2c;
+        i2c_set_clientdata(i2c, wm8753);
 
-	return ret;
+        codec->dev = &i2c->dev;
+
+	return wm8753_register(wm8753);
 }
 
 static int wm8753_i2c_remove(struct i2c_client *client)
 {
-	struct snd_soc_codec *codec = i2c_get_clientdata(client);
-	kfree(codec->reg_cache);
-	return 0;
+        struct wm8753_priv *wm8753 = i2c_get_clientdata(client);
+        wm8753_unregister(wm8753);
+        return 0;
 }
 
 static const struct i2c_device_id wm8753_i2c_id[] = {
@@ -1673,86 +1774,16 @@ MODULE_DEVICE_TABLE(i2c, wm8753_i2c_id);
 
 static struct i2c_driver wm8753_i2c_driver = {
 	.driver = {
-		.name = "WM8753 I2C Codec",
+		.name = "wm8753",
 		.owner = THIS_MODULE,
 	},
 	.probe =    wm8753_i2c_probe,
 	.remove =   wm8753_i2c_remove,
 	.id_table = wm8753_i2c_id,
 };
-
-static int wm8753_add_i2c_device(struct platform_device *pdev,
-				 const struct wm8753_setup_data *setup)
-{
-	struct i2c_board_info info;
-	struct i2c_adapter *adapter;
-	struct i2c_client *client;
-	int ret;
-
-	ret = i2c_add_driver(&wm8753_i2c_driver);
-	if (ret != 0) {
-		dev_err(&pdev->dev, "can't add i2c driver\n");
-		return ret;
-	}
-
-	memset(&info, 0, sizeof(struct i2c_board_info));
-	info.addr = setup->i2c_address;
-	strlcpy(info.type, "wm8753", I2C_NAME_SIZE);
-
-	adapter = i2c_get_adapter(setup->i2c_bus);
-	if (!adapter) {
-		dev_err(&pdev->dev, "can't get i2c adapter %d\n",
-			setup->i2c_bus);
-		goto err_driver;
-	}
-
-	client = i2c_new_device(adapter, &info);
-	i2c_put_adapter(adapter);
-	if (!client) {
-		dev_err(&pdev->dev, "can't add i2c device at 0x%x\n",
-			(unsigned int)info.addr);
-		goto err_driver;
-	}
-
-	return 0;
-
-err_driver:
-	i2c_del_driver(&wm8753_i2c_driver);
-	return -ENODEV;
-}
 #endif
 
 #if defined(CONFIG_SPI_MASTER)
-static int __devinit wm8753_spi_probe(struct spi_device *spi)
-{
-	struct snd_soc_device *socdev = wm8753_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
-	int ret;
-
-	codec->control_data = spi;
-
-	ret = wm8753_init(socdev);
-	if (ret < 0)
-		dev_err(&spi->dev, "failed to initialise WM8753\n");
-
-	return ret;
-}
-
-static int __devexit wm8753_spi_remove(struct spi_device *spi)
-{
-	return 0;
-}
-
-static struct spi_driver wm8753_spi_driver = {
-	.driver = {
-		.name	= "wm8753",
-		.bus	= &spi_bus_type,
-		.owner	= THIS_MODULE,
-	},
-	.probe		= wm8753_spi_probe,
-	.remove		= __devexit_p(wm8753_spi_remove),
-};
-
 static int wm8753_spi_write(struct spi_device *spi, const char *data, int len)
 {
 	struct spi_transfer t;
@@ -1776,120 +1807,69 @@ static int wm8753_spi_write(struct spi_device *spi, const char *data, int len)
 
 	return len;
 }
-#endif
 
-
-static int wm8753_probe(struct platform_device *pdev)
+static int __devinit wm8753_spi_probe(struct spi_device *spi)
 {
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct wm8753_setup_data *setup;
 	struct snd_soc_codec *codec;
 	struct wm8753_priv *wm8753;
-	int ret = 0;
-
-	pr_info("WM8753 Audio Codec %s", WM8753_VERSION);
-
-	setup = socdev->codec_data;
-	codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (codec == NULL)
-		return -ENOMEM;
 
 	wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL);
-	if (wm8753 == NULL) {
-		kfree(codec);
+	if (wm8753 == NULL)
 		return -ENOMEM;
-	}
 
-	codec->private_data = wm8753;
-	socdev->codec = codec;
-	mutex_init(&codec->mutex);
-	INIT_LIST_HEAD(&codec->dapm_widgets);
-	INIT_LIST_HEAD(&codec->dapm_paths);
-	wm8753_socdev = socdev;
-	INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work);
+	codec = &wm8753->codec;
+	codec->control_data = spi;
+	codec->hw_write = (hw_write_t)wm8753_spi_write;
+	codec->dev = &spi->dev;
 
-#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	if (setup->i2c_address) {
-		codec->hw_write = (hw_write_t)i2c_master_send;
-		ret = wm8753_add_i2c_device(pdev, setup);
-	}
-#endif
-#if defined(CONFIG_SPI_MASTER)
-	if (setup->spi) {
-		codec->hw_write = (hw_write_t)wm8753_spi_write;
-		ret = spi_register_driver(&wm8753_spi_driver);
-		if (ret != 0)
-			printk(KERN_ERR "can't add spi driver");
-	}
-#endif
+	spi->dev.driver_data = wm8753;
 
-	if (ret != 0) {
-		kfree(codec->private_data);
-		kfree(codec);
-	}
-	return ret;
+	return wm8753_register(wm8753);
 }
 
-/*
- * This function forces any delayed work to be queued and run.
- */
-static int run_delayed_work(struct delayed_work *dwork)
+static int __devexit wm8753_spi_remove(struct spi_device *spi)
 {
-	int ret;
-
-	/* cancel any work waiting to be queued. */
-	ret = cancel_delayed_work(dwork);
-
-	/* if there was any work waiting then we run it now and
-	 * wait for it's completion */
-	if (ret) {
-		schedule_delayed_work(dwork, 0);
-		flush_scheduled_work();
-	}
-	return ret;
+	struct wm8753_priv *wm8753 = spi->dev.driver_data;
+	wm8753_unregister(wm8753);
+	return 0;
 }
 
-/* power down chip */
-static int wm8753_remove(struct platform_device *pdev)
-{
-	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+static struct spi_driver wm8753_spi_driver = {
+	.driver = {
+		.name	= "wm8753",
+		.bus	= &spi_bus_type,
+		.owner	= THIS_MODULE,
+	},
+	.probe		= wm8753_spi_probe,
+	.remove		= __devexit_p(wm8753_spi_remove),
+};
+#endif
 
-	if (codec->control_data)
-		wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF);
-	run_delayed_work(&codec->delayed_work);
-	snd_soc_free_pcms(socdev);
-	snd_soc_dapm_free(socdev);
+static int __init wm8753_modinit(void)
+{
+	int ret;
 #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
-	i2c_unregister_device(codec->control_data);
-	i2c_del_driver(&wm8753_i2c_driver);
+	ret = i2c_add_driver(&wm8753_i2c_driver);
+	if (ret != 0)
+		pr_err("Failed to register WM8753 I2C driver: %d\n", ret);
 #endif
 #if defined(CONFIG_SPI_MASTER)
-	spi_unregister_driver(&wm8753_spi_driver);
+	ret = spi_register_driver(&wm8753_spi_driver);
+	if (ret != 0)
+		pr_err("Failed to register WM8753 SPI driver: %d\n", ret);
 #endif
-	kfree(codec->private_data);
-	kfree(codec);
-
 	return 0;
 }
-
-struct snd_soc_codec_device soc_codec_dev_wm8753 = {
-	.probe = 	wm8753_probe,
-	.remove = 	wm8753_remove,
-	.suspend = 	wm8753_suspend,
-	.resume =	wm8753_resume,
-};
-EXPORT_SYMBOL_GPL(soc_codec_dev_wm8753);
-
-static int __init wm8753_modinit(void)
-{
-	return snd_soc_register_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai));
-}
 module_init(wm8753_modinit);
 
 static void __exit wm8753_exit(void)
 {
-	snd_soc_unregister_dais(wm8753_dai, ARRAY_SIZE(wm8753_dai));
+#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE)
+	i2c_del_driver(&wm8753_i2c_driver);
+#endif
+#if defined(CONFIG_SPI_MASTER)
+	spi_unregister_driver(&wm8753_spi_driver);
+#endif
 }
 module_exit(wm8753_exit);
 
diff --git a/sound/soc/codecs/wm8753.h b/sound/soc/codecs/wm8753.h
index f55704ce931b..57b2ba244040 100644
--- a/sound/soc/codecs/wm8753.h
+++ b/sound/soc/codecs/wm8753.h
@@ -77,12 +77,6 @@
 #define WM8753_BIASCTL		0x3d
 #define WM8753_ADCTL2		0x3f
 
-struct wm8753_setup_data {
-	int spi;
-	int i2c_bus;
-	unsigned short i2c_address;
-};
-
 #define WM8753_PLL1			0
 #define WM8753_PLL2			1
 
diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c
index 6767de10ded0..46c5ea1ff921 100644
--- a/sound/soc/codecs/wm8900.c
+++ b/sound/soc/codecs/wm8900.c
@@ -517,22 +517,6 @@ SOC_SINGLE("LINEOUT2 LP -12dB", WM8900_REG_LOUTMIXCTL1,
 
 };
 
-/* add non dapm controls */
-static int wm8900_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8900_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm8900_snd_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 static const struct snd_kcontrol_new wm8900_dapm_loutput2_control =
 SOC_DAPM_SINGLE("LINEOUT2L Switch", WM8900_REG_POWER3, 6, 1, 0);
 
@@ -736,7 +720,7 @@ static int wm8900_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 reg;
 
 	reg = wm8900_read(codec, WM8900_REG_AUDIO1) & ~0x60;
@@ -1104,6 +1088,14 @@ static int wm8900_digital_mute(struct snd_soc_dai *codec_dai, int mute)
 	(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
 	 SNDRV_PCM_FORMAT_S24_LE)
 
+static struct snd_soc_dai_ops wm8900_dai_ops = {
+	.hw_params	= wm8900_hw_params,
+	.set_clkdiv	= wm8900_set_dai_clkdiv,
+	.set_pll	= wm8900_set_dai_pll,
+	.set_fmt	= wm8900_set_dai_fmt,
+	.digital_mute	= wm8900_digital_mute,
+};
+
 struct snd_soc_dai wm8900_dai = {
 	.name = "WM8900 HiFi",
 	.playback = {
@@ -1120,13 +1112,7 @@ struct snd_soc_dai wm8900_dai = {
 		.rates = WM8900_RATES,
 		.formats = WM8900_PCM_FORMATS,
 	 },
-	.ops = {
-		.hw_params = wm8900_hw_params,
-		 .set_clkdiv = wm8900_set_dai_clkdiv,
-		 .set_pll = wm8900_set_dai_pll,
-		 .set_fmt = wm8900_set_dai_fmt,
-		 .digital_mute = wm8900_digital_mute,
-	 },
+	.ops = &wm8900_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8900_dai);
 
@@ -1226,7 +1212,7 @@ static int wm8900_set_bias_level(struct snd_soc_codec *codec,
 static int wm8900_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8900_priv *wm8900 = codec->private_data;
 	int fll_out = wm8900->fll_out;
 	int fll_in  = wm8900->fll_in;
@@ -1250,7 +1236,7 @@ static int wm8900_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8900_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8900_priv *wm8900 = codec->private_data;
 	u16 *cache;
 	int i, ret;
@@ -1288,8 +1274,8 @@ static int wm8900_resume(struct platform_device *pdev)
 
 static struct snd_soc_codec *wm8900_codec;
 
-static int wm8900_i2c_probe(struct i2c_client *i2c,
-			    const struct i2c_device_id *id)
+static __devinit int wm8900_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
 {
 	struct wm8900_priv *wm8900;
 	struct snd_soc_codec *codec;
@@ -1388,7 +1374,7 @@ err:
 	return ret;
 }
 
-static int wm8900_i2c_remove(struct i2c_client *client)
+static __devexit int wm8900_i2c_remove(struct i2c_client *client)
 {
 	snd_soc_unregister_dai(&wm8900_dai);
 	snd_soc_unregister_codec(wm8900_codec);
@@ -1414,7 +1400,7 @@ static struct i2c_driver wm8900_i2c_driver = {
 		.owner = THIS_MODULE,
 	},
 	.probe = wm8900_i2c_probe,
-	.remove = wm8900_i2c_remove,
+	.remove = __devexit_p(wm8900_i2c_remove),
 	.id_table = wm8900_i2c_id,
 };
 
@@ -1430,7 +1416,7 @@ static int wm8900_probe(struct platform_device *pdev)
 	}
 
 	codec = wm8900_codec;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 
 	/* Register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
@@ -1439,7 +1425,8 @@ static int wm8900_probe(struct platform_device *pdev)
 		goto pcm_err;
 	}
 
-	wm8900_add_controls(codec);
+	snd_soc_add_controls(codec, wm8900_snd_controls,
+				ARRAY_SIZE(wm8900_snd_controls));
 	wm8900_add_widgets(codec);
 
 	ret = snd_soc_init_card(socdev);
diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c
index bde74546db4a..8cf571f1a803 100644
--- a/sound/soc/codecs/wm8903.c
+++ b/sound/soc/codecs/wm8903.c
@@ -744,21 +744,6 @@ SOC_DOUBLE_R_TLV("Speaker Volume",
 		 0, 63, 0, out_tlv),
 };
 
-static int wm8903_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8903_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm8903_snd_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-
-	return 0;
-}
-
 static const struct snd_kcontrol_new linput_mode_mux =
 	SOC_DAPM_ENUM("Left Input Mode Mux", linput_mode_enum);
 
@@ -1276,7 +1261,7 @@ static int wm8903_startup(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8903_priv *wm8903 = codec->private_data;
 	struct i2c_client *i2c = codec->control_data;
 	struct snd_pcm_runtime *master_runtime;
@@ -1318,7 +1303,7 @@ static void wm8903_shutdown(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8903_priv *wm8903 = codec->private_data;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
@@ -1338,7 +1323,7 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8903_priv *wm8903 = codec->private_data;
 	struct i2c_client *i2c = codec->control_data;
 	int fs = params_rate(params);
@@ -1512,6 +1497,15 @@ static int wm8903_hw_params(struct snd_pcm_substream *substream,
 			SNDRV_PCM_FMTBIT_S20_3LE |\
 			SNDRV_PCM_FMTBIT_S24_LE)
 
+static struct snd_soc_dai_ops wm8903_dai_ops = {
+	.startup	= wm8903_startup,
+	.shutdown	= wm8903_shutdown,
+	.hw_params	= wm8903_hw_params,
+	.digital_mute	= wm8903_digital_mute,
+	.set_fmt	= wm8903_set_dai_fmt,
+	.set_sysclk	= wm8903_set_dai_sysclk,
+};
+
 struct snd_soc_dai wm8903_dai = {
 	.name = "WM8903",
 	.playback = {
@@ -1528,21 +1522,14 @@ struct snd_soc_dai wm8903_dai = {
 		 .rates = WM8903_CAPTURE_RATES,
 		 .formats = WM8903_FORMATS,
 	 },
-	.ops = {
-		 .startup = wm8903_startup,
-		 .shutdown = wm8903_shutdown,
-		 .hw_params = wm8903_hw_params,
-		 .digital_mute = wm8903_digital_mute,
-		 .set_fmt = wm8903_set_dai_fmt,
-		 .set_sysclk = wm8903_set_dai_sysclk
-	}
+	.ops = &wm8903_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8903_dai);
 
 static int wm8903_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
 
@@ -1552,7 +1539,7 @@ static int wm8903_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8903_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct i2c_client *i2c = codec->control_data;
 	int i;
 	u16 *reg_cache = codec->reg_cache;
@@ -1577,8 +1564,8 @@ static int wm8903_resume(struct platform_device *pdev)
 
 static struct snd_soc_codec *wm8903_codec;
 
-static int wm8903_i2c_probe(struct i2c_client *i2c,
-			    const struct i2c_device_id *id)
+static __devinit int wm8903_i2c_probe(struct i2c_client *i2c,
+				      const struct i2c_device_id *id)
 {
 	struct wm8903_priv *wm8903;
 	struct snd_soc_codec *codec;
@@ -1684,7 +1671,7 @@ err:
 	return ret;
 }
 
-static int wm8903_i2c_remove(struct i2c_client *client)
+static __devexit int wm8903_i2c_remove(struct i2c_client *client)
 {
 	struct snd_soc_codec *codec = i2c_get_clientdata(client);
 
@@ -1714,7 +1701,7 @@ static struct i2c_driver wm8903_i2c_driver = {
 		.owner = THIS_MODULE,
 	},
 	.probe    = wm8903_i2c_probe,
-	.remove   = wm8903_i2c_remove,
+	.remove   = __devexit_p(wm8903_i2c_remove),
 	.id_table = wm8903_i2c_id,
 };
 
@@ -1728,7 +1715,7 @@ static int wm8903_probe(struct platform_device *pdev)
 		goto err;
 	}
 
-	socdev->codec = wm8903_codec;
+	socdev->card->codec = wm8903_codec;
 
 	/* register pcms */
 	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
@@ -1737,8 +1724,9 @@ static int wm8903_probe(struct platform_device *pdev)
 		goto err;
 	}
 
-	wm8903_add_controls(socdev->codec);
-	wm8903_add_widgets(socdev->codec);
+	snd_soc_add_controls(socdev->card->codec, wm8903_snd_controls,
+				ARRAY_SIZE(wm8903_snd_controls));
+	wm8903_add_widgets(socdev->card->codec);
 
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -1759,7 +1747,7 @@ err:
 static int wm8903_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c
index 88ead7f8dd98..032dca22dbd3 100644
--- a/sound/soc/codecs/wm8971.c
+++ b/sound/soc/codecs/wm8971.c
@@ -195,21 +195,6 @@ static const struct snd_kcontrol_new wm8971_snd_controls[] = {
 	SOC_DOUBLE_R("Mic Boost", WM8971_LADCIN, WM8971_RADCIN, 4, 3, 0),
 };
 
-/* add non-DAPM controls */
-static int wm8971_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8971_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8971_snd_controls[i],
-					     codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 /*
  * DAPM Controls
  */
@@ -546,7 +531,7 @@ static int wm8971_pcm_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm8971_priv *wm8971 = codec->private_data;
 	u16 iface = wm8971_read_reg_cache(codec, WM8971_IFACE) & 0x1f3;
 	u16 srate = wm8971_read_reg_cache(codec, WM8971_SRATE) & 0x1c0;
@@ -619,6 +604,13 @@ static int wm8971_set_bias_level(struct snd_soc_codec *codec,
 #define WM8971_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 	SNDRV_PCM_FMTBIT_S24_LE)
 
+static struct snd_soc_dai_ops wm8971_dai_ops = {
+	.hw_params	= wm8971_pcm_hw_params,
+	.digital_mute	= wm8971_mute,
+	.set_fmt	= wm8971_set_dai_fmt,
+	.set_sysclk	= wm8971_set_dai_sysclk,
+};
+
 struct snd_soc_dai wm8971_dai = {
 	.name = "WM8971",
 	.playback = {
@@ -633,12 +625,7 @@ struct snd_soc_dai wm8971_dai = {
 		.channels_max = 2,
 		.rates = WM8971_RATES,
 		.formats = WM8971_FORMATS,},
-	.ops = {
-		.hw_params = wm8971_pcm_hw_params,
-		.digital_mute = wm8971_mute,
-		.set_fmt = wm8971_set_dai_fmt,
-		.set_sysclk = wm8971_set_dai_sysclk,
-	},
+	.ops = &wm8971_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8971_dai);
 
@@ -652,7 +639,7 @@ static void wm8971_work(struct work_struct *work)
 static int wm8971_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -661,7 +648,7 @@ static int wm8971_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8971_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -692,7 +679,7 @@ static int wm8971_resume(struct platform_device *pdev)
 
 static int wm8971_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int reg, ret = 0;
 
 	codec->name = "WM8971";
@@ -745,7 +732,8 @@ static int wm8971_init(struct snd_soc_device *socdev)
 	reg = wm8971_read_reg_cache(codec, WM8971_RINVOL);
 	wm8971_write(codec, WM8971_RINVOL, reg | 0x0100);
 
-	wm8971_add_controls(codec);
+	snd_soc_add_controls(codec, wm8971_snd_controls,
+				ARRAY_SIZE(wm8971_snd_controls));
 	wm8971_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -772,7 +760,7 @@ static int wm8971_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = wm8971_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -873,7 +861,7 @@ static int wm8971_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = wm8971;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -908,7 +896,7 @@ static int wm8971_probe(struct platform_device *pdev)
 static int wm8971_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8971_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index a5731faa150c..c518c3e5aa3f 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -115,7 +115,7 @@ static inline unsigned int wm8990_read_reg_cache(struct snd_soc_codec *codec,
 	unsigned int reg)
 {
 	u16 *cache = codec->reg_cache;
-	BUG_ON(reg > (ARRAY_SIZE(wm8990_reg)) - 1);
+	BUG_ON(reg >= ARRAY_SIZE(wm8990_reg));
 	return cache[reg];
 }
 
@@ -128,7 +128,7 @@ static inline void wm8990_write_reg_cache(struct snd_soc_codec *codec,
 	u16 *cache = codec->reg_cache;
 
 	/* Reset register and reserved registers are uncached */
-	if (reg == 0 || reg > ARRAY_SIZE(wm8990_reg) - 1)
+	if (reg == 0 || reg >= ARRAY_SIZE(wm8990_reg))
 		return;
 
 	cache[reg] = value;
@@ -418,21 +418,6 @@ SOC_SINGLE("RIN34 Mute Switch", WM8990_RIGHT_LINE_INPUT_3_4_VOLUME,
 
 };
 
-/* add non dapm controls */
-static int wm8990_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm8990_snd_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8990_snd_controls[i], codec,
-					NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 /*
  * _DAPM_ Controls
  */
@@ -1178,7 +1163,7 @@ static int wm8990_hw_params(struct snd_pcm_substream *substream,
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 audio1 = wm8990_read_reg_cache(codec, WM8990_AUDIO_INTERFACE_1);
 
 	audio1 &= ~WM8990_AIF_WL_MASK;
@@ -1347,6 +1332,15 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec,
  * 1. ADC/DAC on Primary Interface
  * 2. ADC on Primary Interface/DAC on secondary
  */
+static struct snd_soc_dai_ops wm8990_dai_ops = {
+	.hw_params	= wm8990_hw_params,
+	.digital_mute	= wm8990_mute,
+	.set_fmt	= wm8990_set_dai_fmt,
+	.set_clkdiv	= wm8990_set_dai_clkdiv,
+	.set_pll	= wm8990_set_dai_pll,
+	.set_sysclk	= wm8990_set_dai_sysclk,
+};
+
 struct snd_soc_dai wm8990_dai = {
 /* ADC/DAC on primary */
 	.name = "WM8990 ADC/DAC Primary",
@@ -1363,21 +1357,14 @@ struct snd_soc_dai wm8990_dai = {
 		.channels_max = 2,
 		.rates = WM8990_RATES,
 		.formats = WM8990_FORMATS,},
-	.ops = {
-		.hw_params = wm8990_hw_params,
-		.digital_mute = wm8990_mute,
-		.set_fmt = wm8990_set_dai_fmt,
-		.set_clkdiv = wm8990_set_dai_clkdiv,
-		.set_pll = wm8990_set_dai_pll,
-		.set_sysclk = wm8990_set_dai_sysclk,
-	},
+	.ops = &wm8990_dai_ops,
 };
 EXPORT_SYMBOL_GPL(wm8990_dai);
 
 static int wm8990_suspend(struct platform_device *pdev, pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	/* we only need to suspend if we are a valid card */
 	if (!codec->card)
@@ -1390,7 +1377,7 @@ static int wm8990_suspend(struct platform_device *pdev, pm_message_t state)
 static int wm8990_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i;
 	u8 data[2];
 	u16 *cache = codec->reg_cache;
@@ -1418,7 +1405,7 @@ static int wm8990_resume(struct platform_device *pdev)
  */
 static int wm8990_init(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 reg;
 	int ret = 0;
 
@@ -1461,7 +1448,8 @@ static int wm8990_init(struct snd_soc_device *socdev)
 	wm8990_write(codec, WM8990_LEFT_OUTPUT_VOLUME, 0x50 | (1<<8));
 	wm8990_write(codec, WM8990_RIGHT_OUTPUT_VOLUME, 0x50 | (1<<8));
 
-	wm8990_add_controls(codec);
+	snd_soc_add_controls(codec, wm8990_snd_controls,
+				ARRAY_SIZE(wm8990_snd_controls));
 	wm8990_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -1495,7 +1483,7 @@ static int wm8990_i2c_probe(struct i2c_client *i2c,
 			    const struct i2c_device_id *id)
 {
 	struct snd_soc_device *socdev = wm8990_socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret;
 
 	i2c_set_clientdata(i2c, codec);
@@ -1594,7 +1582,7 @@ static int wm8990_probe(struct platform_device *pdev)
 	}
 
 	codec->private_data = wm8990;
-	socdev->codec = codec;
+	socdev->card->codec = codec;
 	mutex_init(&codec->mutex);
 	INIT_LIST_HEAD(&codec->dapm_widgets);
 	INIT_LIST_HEAD(&codec->dapm_paths);
@@ -1620,7 +1608,7 @@ static int wm8990_probe(struct platform_device *pdev)
 static int wm8990_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec->control_data)
 		wm8990_set_bias_level(codec, SND_SOC_BIAS_OFF);
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
new file mode 100644
index 000000000000..3265817c5c26
--- /dev/null
+++ b/sound/soc/codecs/wm9705.c
@@ -0,0 +1,415 @@
+/*
+ * wm9705.c  --  ALSA Soc WM9705 codec support
+ *
+ * Copyright 2008 Ian Molton <spyro@f2s.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation; Version 2 of the  License only.
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/kernel.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include "wm9705.h"
+
+/*
+ * WM9705 register cache
+ */
+static const u16 wm9705_reg[] = {
+	0x6150, 0x8000, 0x8000, 0x8000, /* 0x0  */
+	0x0000, 0x8000, 0x8008, 0x8008, /* 0x8  */
+	0x8808, 0x8808, 0x8808, 0x8808, /* 0x10 */
+	0x8808, 0x0000, 0x8000, 0x0000, /* 0x18 */
+	0x0000, 0x0000, 0x0000, 0x000f, /* 0x20 */
+	0x0605, 0x0000, 0xbb80, 0x0000, /* 0x28 */
+	0x0000, 0xbb80, 0x0000, 0x0000, /* 0x30 */
+	0x0000, 0x2000, 0x0000, 0x0000, /* 0x38 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 0x40 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 0x48 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 0x50 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 0x58 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 0x60 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 0x68 */
+	0x0000, 0x0808, 0x0000, 0x0006, /* 0x70 */
+	0x0000, 0x0000, 0x574d, 0x4c05, /* 0x78 */
+};
+
+static const struct snd_kcontrol_new wm9705_snd_ac97_controls[] = {
+	SOC_DOUBLE("Master Playback Volume", AC97_MASTER, 8, 0, 31, 1),
+	SOC_SINGLE("Master Playback Switch", AC97_MASTER, 15, 1, 1),
+	SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1),
+	SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1),
+	SOC_DOUBLE("PCM Playback Volume", AC97_PCM, 8, 0, 31, 1),
+	SOC_SINGLE("PCM Playback Switch", AC97_PCM, 15, 1, 1),
+	SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1),
+	SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1),
+	SOC_SINGLE("PCBeep Playback Volume", AC97_PC_BEEP, 1, 15, 1),
+	SOC_SINGLE("Phone Playback Volume", AC97_PHONE, 0, 31, 1),
+	SOC_DOUBLE("Line Playback Volume", AC97_LINE, 8, 0, 31, 1),
+	SOC_DOUBLE("CD Playback Volume", AC97_CD, 8, 0, 31, 1),
+	SOC_SINGLE("Mic Playback Volume", AC97_MIC, 0, 31, 1),
+	SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 6, 1, 0),
+	SOC_DOUBLE("Capture Volume", AC97_REC_GAIN, 8, 0, 15, 0),
+	SOC_SINGLE("Capture Switch", AC97_REC_GAIN, 15, 1, 1),
+};
+
+static const char *wm9705_mic[] = {"Mic 1", "Mic 2"};
+static const char *wm9705_rec_sel[] = {"Mic", "CD", "NC", "NC",
+	"Line", "Stereo Mix", "Mono Mix", "Phone"};
+
+static const struct soc_enum wm9705_enum_mic =
+	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, wm9705_mic);
+static const struct soc_enum wm9705_enum_rec_l =
+	SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9705_rec_sel);
+static const struct soc_enum wm9705_enum_rec_r =
+	SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9705_rec_sel);
+
+/* Headphone Mixer */
+static const struct snd_kcontrol_new wm9705_hp_mixer_controls[] = {
+	SOC_DAPM_SINGLE("PCBeep Playback Switch", AC97_PC_BEEP, 15, 1, 1),
+	SOC_DAPM_SINGLE("CD Playback Switch", AC97_CD, 15, 1, 1),
+	SOC_DAPM_SINGLE("Mic Playback Switch", AC97_MIC, 15, 1, 1),
+	SOC_DAPM_SINGLE("Phone Playback Switch", AC97_PHONE, 15, 1, 1),
+	SOC_DAPM_SINGLE("Line Playback Switch", AC97_LINE, 15, 1, 1),
+};
+
+/* Mic source */
+static const struct snd_kcontrol_new wm9705_mic_src_controls =
+	SOC_DAPM_ENUM("Route", wm9705_enum_mic);
+
+/* Capture source */
+static const struct snd_kcontrol_new wm9705_capture_selectl_controls =
+	SOC_DAPM_ENUM("Route", wm9705_enum_rec_l);
+static const struct snd_kcontrol_new wm9705_capture_selectr_controls =
+	SOC_DAPM_ENUM("Route", wm9705_enum_rec_r);
+
+/* DAPM widgets */
+static const struct snd_soc_dapm_widget wm9705_dapm_widgets[] = {
+	SND_SOC_DAPM_MUX("Mic Source", SND_SOC_NOPM, 0, 0,
+		&wm9705_mic_src_controls),
+	SND_SOC_DAPM_MUX("Left Capture Source", SND_SOC_NOPM, 0, 0,
+		&wm9705_capture_selectl_controls),
+	SND_SOC_DAPM_MUX("Right Capture Source", SND_SOC_NOPM, 0, 0,
+		&wm9705_capture_selectr_controls),
+	SND_SOC_DAPM_DAC("Left DAC", "Left HiFi Playback",
+		SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_DAC("Right DAC", "Right HiFi Playback",
+		SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_MIXER_NAMED_CTL("HP Mixer", SND_SOC_NOPM, 0, 0,
+		&wm9705_hp_mixer_controls[0],
+		ARRAY_SIZE(wm9705_hp_mixer_controls)),
+	SND_SOC_DAPM_MIXER("Mono Mixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_ADC("Left ADC", "Left HiFi Capture", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_ADC("Right ADC", "Right HiFi Capture", SND_SOC_NOPM, 0, 0),
+	SND_SOC_DAPM_PGA("Headphone PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Speaker PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Line PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Line out PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Mono PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Phone PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("Mic PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("PCBEEP PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("CD PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_PGA("ADC PGA", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_OUTPUT("HPOUTL"),
+	SND_SOC_DAPM_OUTPUT("HPOUTR"),
+	SND_SOC_DAPM_OUTPUT("LOUT"),
+	SND_SOC_DAPM_OUTPUT("ROUT"),
+	SND_SOC_DAPM_OUTPUT("MONOOUT"),
+	SND_SOC_DAPM_INPUT("PHONE"),
+	SND_SOC_DAPM_INPUT("LINEINL"),
+	SND_SOC_DAPM_INPUT("LINEINR"),
+	SND_SOC_DAPM_INPUT("CDINL"),
+	SND_SOC_DAPM_INPUT("CDINR"),
+	SND_SOC_DAPM_INPUT("PCBEEP"),
+	SND_SOC_DAPM_INPUT("MIC1"),
+	SND_SOC_DAPM_INPUT("MIC2"),
+};
+
+/* Audio map
+ * WM9705 has no switches to disable the route from the inputs to the HP mixer
+ * so in order to prevent active inputs from forcing the audio outputs to be
+ * constantly enabled, we use the mutes on those inputs to simulate such
+ * controls.
+ */
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* HP mixer */
+	{"HP Mixer", "PCBeep Playback Switch", "PCBEEP PGA"},
+	{"HP Mixer", "CD Playback Switch", "CD PGA"},
+	{"HP Mixer", "Mic Playback Switch", "Mic PGA"},
+	{"HP Mixer", "Phone Playback Switch", "Phone PGA"},
+	{"HP Mixer", "Line Playback Switch", "Line PGA"},
+	{"HP Mixer", NULL, "Left DAC"},
+	{"HP Mixer", NULL, "Right DAC"},
+
+	/* mono mixer */
+	{"Mono Mixer", NULL, "HP Mixer"},
+
+	/* outputs */
+	{"Headphone PGA", NULL, "HP Mixer"},
+	{"HPOUTL", NULL, "Headphone PGA"},
+	{"HPOUTR", NULL, "Headphone PGA"},
+	{"Line out PGA", NULL, "HP Mixer"},
+	{"LOUT", NULL, "Line out PGA"},
+	{"ROUT", NULL, "Line out PGA"},
+	{"Mono PGA", NULL, "Mono Mixer"},
+	{"MONOOUT", NULL, "Mono PGA"},
+
+	/* inputs */
+	{"CD PGA", NULL, "CDINL"},
+	{"CD PGA", NULL, "CDINR"},
+	{"Line PGA", NULL, "LINEINL"},
+	{"Line PGA", NULL, "LINEINR"},
+	{"Phone PGA", NULL, "PHONE"},
+	{"Mic Source", "Mic 1", "MIC1"},
+	{"Mic Source", "Mic 2", "MIC2"},
+	{"Mic PGA", NULL, "Mic Source"},
+	{"PCBEEP PGA", NULL, "PCBEEP"},
+
+	/* Left capture selector */
+	{"Left Capture Source", "Mic", "Mic Source"},
+	{"Left Capture Source", "CD", "CDINL"},
+	{"Left Capture Source", "Line", "LINEINL"},
+	{"Left Capture Source", "Stereo Mix", "HP Mixer"},
+	{"Left Capture Source", "Mono Mix", "HP Mixer"},
+	{"Left Capture Source", "Phone", "PHONE"},
+
+	/* Right capture source */
+	{"Right Capture Source", "Mic", "Mic Source"},
+	{"Right Capture Source", "CD", "CDINR"},
+	{"Right Capture Source", "Line", "LINEINR"},
+	{"Right Capture Source", "Stereo Mix", "HP Mixer"},
+	{"Right Capture Source", "Mono Mix", "HP Mixer"},
+	{"Right Capture Source", "Phone", "PHONE"},
+
+	{"ADC PGA", NULL, "Left Capture Source"},
+	{"ADC PGA", NULL, "Right Capture Source"},
+
+	/* ADC's */
+	{"Left ADC",  NULL, "ADC PGA"},
+	{"Right ADC", NULL, "ADC PGA"},
+};
+
+static int wm9705_add_widgets(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_new_controls(codec, wm9705_dapm_widgets,
+					ARRAY_SIZE(wm9705_dapm_widgets));
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_new_widgets(codec);
+
+	return 0;
+}
+
+/* We use a register cache to enhance read performance. */
+static unsigned int ac97_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+	u16 *cache = codec->reg_cache;
+
+	switch (reg) {
+	case AC97_RESET:
+	case AC97_VENDOR_ID1:
+	case AC97_VENDOR_ID2:
+		return soc_ac97_ops.read(codec->ac97, reg);
+	default:
+		reg = reg >> 1;
+
+		if (reg >= (ARRAY_SIZE(wm9705_reg)))
+			return -EIO;
+
+		return cache[reg];
+	}
+}
+
+static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+	unsigned int val)
+{
+	u16 *cache = codec->reg_cache;
+
+	soc_ac97_ops.write(codec->ac97, reg, val);
+	reg = reg >> 1;
+	if (reg < (ARRAY_SIZE(wm9705_reg)))
+		cache[reg] = val;
+
+	return 0;
+}
+
+static int ac97_prepare(struct snd_pcm_substream *substream,
+			struct snd_soc_dai *dai)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->card->codec;
+	int reg;
+	u16 vra;
+
+	vra = ac97_read(codec, AC97_EXTENDED_STATUS);
+	ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		reg = AC97_PCM_FRONT_DAC_RATE;
+	else
+		reg = AC97_PCM_LR_ADC_RATE;
+
+	return ac97_write(codec, reg, runtime->rate);
+}
+
+#define WM9705_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | \
+			SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
+			SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+			SNDRV_PCM_RATE_48000)
+
+static struct snd_soc_dai_ops wm9705_dai_ops = {
+	.prepare	= ac97_prepare,
+};
+
+struct snd_soc_dai wm9705_dai[] = {
+	{
+		.name = "AC97 HiFi",
+		.ac97_control = 1,
+		.playback = {
+			.stream_name = "HiFi Playback",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM9705_AC97_RATES,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.capture = {
+			.stream_name = "HiFi Capture",
+			.channels_min = 1,
+			.channels_max = 2,
+			.rates = WM9705_AC97_RATES,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+		.ops = &wm9705_dai_ops,
+	},
+	{
+		.name = "AC97 Aux",
+		.playback = {
+			.stream_name = "Aux Playback",
+			.channels_min = 1,
+			.channels_max = 1,
+			.rates = WM9705_AC97_RATES,
+			.formats = SNDRV_PCM_FMTBIT_S16_LE,
+		},
+	}
+};
+EXPORT_SYMBOL_GPL(wm9705_dai);
+
+static int wm9705_reset(struct snd_soc_codec *codec)
+{
+	if (soc_ac97_ops.reset) {
+		soc_ac97_ops.reset(codec->ac97);
+		if (ac97_read(codec, 0) == wm9705_reg[0])
+			return 0; /* Success */
+	}
+
+	return -EIO;
+}
+
+static int wm9705_soc_probe(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec;
+	int ret = 0;
+
+	printk(KERN_INFO "WM9705 SoC Audio Codec\n");
+
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec),
+				      GFP_KERNEL);
+	if (socdev->card->codec == NULL)
+		return -ENOMEM;
+	codec = socdev->card->codec;
+	mutex_init(&codec->mutex);
+
+	codec->reg_cache = kmemdup(wm9705_reg, sizeof(wm9705_reg), GFP_KERNEL);
+	if (codec->reg_cache == NULL) {
+		ret = -ENOMEM;
+		goto cache_err;
+	}
+	codec->reg_cache_size = sizeof(wm9705_reg);
+	codec->reg_cache_step = 2;
+
+	codec->name = "WM9705";
+	codec->owner = THIS_MODULE;
+	codec->dai = wm9705_dai;
+	codec->num_dai = ARRAY_SIZE(wm9705_dai);
+	codec->write = ac97_write;
+	codec->read = ac97_read;
+	INIT_LIST_HEAD(&codec->dapm_widgets);
+	INIT_LIST_HEAD(&codec->dapm_paths);
+
+	ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+	if (ret < 0) {
+		printk(KERN_ERR "wm9705: failed to register AC97 codec\n");
+		goto codec_err;
+	}
+
+	/* register pcms */
+	ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+	if (ret < 0)
+		goto pcm_err;
+
+	ret = wm9705_reset(codec);
+	if (ret)
+		goto reset_err;
+
+	snd_soc_add_controls(codec, wm9705_snd_ac97_controls,
+				ARRAY_SIZE(wm9705_snd_ac97_controls));
+	wm9705_add_widgets(codec);
+
+	ret = snd_soc_init_card(socdev);
+	if (ret < 0) {
+		printk(KERN_ERR "wm9705: failed to register card\n");
+		goto pcm_err;
+	}
+
+	return 0;
+
+reset_err:
+	snd_soc_free_pcms(socdev);
+pcm_err:
+	snd_soc_free_ac97_codec(codec);
+codec_err:
+	kfree(codec->reg_cache);
+cache_err:
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
+	return ret;
+}
+
+static int wm9705_soc_remove(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	if (codec == NULL)
+		return 0;
+
+	snd_soc_dapm_free(socdev);
+	snd_soc_free_pcms(socdev);
+	snd_soc_free_ac97_codec(codec);
+	kfree(codec->reg_cache);
+	kfree(codec);
+	return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_wm9705 = {
+	.probe = 	wm9705_soc_probe,
+	.remove = 	wm9705_soc_remove,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_wm9705);
+
+MODULE_DESCRIPTION("ASoC WM9705 driver");
+MODULE_AUTHOR("Ian Molton");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/wm9705.h b/sound/soc/codecs/wm9705.h
new file mode 100644
index 000000000000..d380f110f9e2
--- /dev/null
+++ b/sound/soc/codecs/wm9705.h
@@ -0,0 +1,14 @@
+/*
+ * wm9705.h  --  WM9705 Soc Audio driver
+ */
+
+#ifndef _WM9705_H
+#define _WM9705_H
+
+#define WM9705_DAI_AC97_HIFI	0
+#define WM9705_DAI_AC97_AUX	1
+
+extern struct snd_soc_dai wm9705_dai[2];
+extern struct snd_soc_codec_device soc_codec_dev_wm9705;
+
+#endif
diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c
index af83d629078a..765cf1e7369e 100644
--- a/sound/soc/codecs/wm9712.c
+++ b/sound/soc/codecs/wm9712.c
@@ -154,21 +154,6 @@ SOC_SINGLE("Mic 2 Volume", AC97_MIC, 0, 31, 1),
 SOC_SINGLE("Mic 20dB Boost Switch", AC97_MIC, 7, 1, 0),
 };
 
-/* add non dapm controls */
-static int wm9712_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm9712_snd_ac97_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				  snd_soc_cnew(&wm9712_snd_ac97_controls[i],
-					       codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 /* We have to create a fake left and right HP mixers because
  * the codec only has a single control that is shared by both channels.
  * This makes it impossible to determine the audio path.
@@ -467,7 +452,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec,
 	else {
 		reg = reg >> 1;
 
-		if (reg > (ARRAY_SIZE(wm9712_reg)))
+		if (reg >= (ARRAY_SIZE(wm9712_reg)))
 			return -EIO;
 
 		return cache[reg];
@@ -481,7 +466,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
 
 	soc_ac97_ops.write(codec->ac97, reg, val);
 	reg = reg >> 1;
-	if (reg <= (ARRAY_SIZE(wm9712_reg)))
+	if (reg < (ARRAY_SIZE(wm9712_reg)))
 		cache[reg] = val;
 
 	return 0;
@@ -493,7 +478,7 @@ static int ac97_prepare(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int reg;
 	u16 vra;
 
@@ -514,7 +499,7 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream,
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 vra, xsle;
 
 	vra = ac97_read(codec, AC97_EXTENDED_STATUS);
@@ -532,6 +517,14 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream,
 		SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 |\
 		SNDRV_PCM_RATE_48000)
 
+static struct snd_soc_dai_ops wm9712_dai_ops_hifi = {
+	.prepare	= ac97_prepare,
+};
+
+static struct snd_soc_dai_ops wm9712_dai_ops_aux = {
+	.prepare	= ac97_aux_prepare,
+};
+
 struct snd_soc_dai wm9712_dai[] = {
 {
 	.name = "AC97 HiFi",
@@ -548,8 +541,7 @@ struct snd_soc_dai wm9712_dai[] = {
 		.channels_max = 2,
 		.rates = WM9712_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_prepare,},
+	.ops = &wm9712_dai_ops_hifi,
 },
 {
 	.name = "AC97 Aux",
@@ -559,8 +551,7 @@ struct snd_soc_dai wm9712_dai[] = {
 		.channels_max = 1,
 		.rates = WM9712_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_aux_prepare,},
+	.ops = &wm9712_dai_ops_aux,
 }
 };
 EXPORT_SYMBOL_GPL(wm9712_dai);
@@ -607,7 +598,7 @@ static int wm9712_soc_suspend(struct platform_device *pdev,
 	pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	wm9712_set_bias_level(codec, SND_SOC_BIAS_OFF);
 	return 0;
@@ -616,7 +607,7 @@ static int wm9712_soc_suspend(struct platform_device *pdev,
 static int wm9712_soc_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int i, ret;
 	u16 *cache = codec->reg_cache;
 
@@ -652,10 +643,11 @@ static int wm9712_soc_probe(struct platform_device *pdev)
 
 	printk(KERN_INFO "WM9711/WM9712 SoC Audio Codec %s\n", WM9712_VERSION);
 
-	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (socdev->codec == NULL)
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec),
+				      GFP_KERNEL);
+	if (socdev->card->codec == NULL)
 		return -ENOMEM;
-	codec = socdev->codec;
+	codec = socdev->card->codec;
 	mutex_init(&codec->mutex);
 
 	codec->reg_cache = kmemdup(wm9712_reg, sizeof(wm9712_reg), GFP_KERNEL);
@@ -698,7 +690,8 @@ static int wm9712_soc_probe(struct platform_device *pdev)
 	ac97_write(codec, AC97_VIDEO, ac97_read(codec, AC97_VIDEO) | 0x3000);
 
 	wm9712_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
-	wm9712_add_controls(codec);
+	snd_soc_add_controls(codec, wm9712_snd_ac97_controls,
+				ARRAY_SIZE(wm9712_snd_ac97_controls));
 	wm9712_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0) {
@@ -718,15 +711,15 @@ codec_err:
 	kfree(codec->reg_cache);
 
 cache_err:
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
 	return ret;
 }
 
 static int wm9712_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec == NULL)
 		return 0;
diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c
index f3ca8aaf0139..523bad077fa0 100644
--- a/sound/soc/codecs/wm9713.c
+++ b/sound/soc/codecs/wm9713.c
@@ -32,7 +32,6 @@
 
 struct wm9713_priv {
 	u32 pll_in; /* PLL input frequency */
-	u32 pll_out; /* PLL output frequency */
 };
 
 static unsigned int ac97_read(struct snd_soc_codec *codec,
@@ -190,21 +189,6 @@ SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0),
 SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1),
 };
 
-/* add non dapm controls */
-static int wm9713_add_controls(struct snd_soc_codec *codec)
-{
-	int err, i;
-
-	for (i = 0; i < ARRAY_SIZE(wm9713_snd_ac97_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm9713_snd_ac97_controls[i],
-					codec, NULL));
-		if (err < 0)
-			return err;
-	}
-	return 0;
-}
-
 /* We have to create a fake left and right HP mixers because
  * the codec only has a single control that is shared by both channels.
  * This makes it impossible to determine the audio path using the current
@@ -636,7 +620,7 @@ static unsigned int ac97_read(struct snd_soc_codec *codec,
 	else {
 		reg = reg >> 1;
 
-		if (reg > (ARRAY_SIZE(wm9713_reg)))
+		if (reg >= (ARRAY_SIZE(wm9713_reg)))
 			return -EIO;
 
 		return cache[reg];
@@ -650,7 +634,7 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg,
 	if (reg < 0x7c)
 		soc_ac97_ops.write(codec->ac97, reg, val);
 	reg = reg >> 1;
-	if (reg <= (ARRAY_SIZE(wm9713_reg)))
+	if (reg < (ARRAY_SIZE(wm9713_reg)))
 		cache[reg] = val;
 
 	return 0;
@@ -738,13 +722,13 @@ static int wm9713_set_pll(struct snd_soc_codec *codec,
 	struct _pll_div pll_div;
 
 	/* turn PLL off ? */
-	if (freq_in == 0 || freq_out == 0) {
+	if (freq_in == 0) {
 		/* disable PLL power and select ext source */
 		reg = ac97_read(codec, AC97_HANDSET_RATE);
 		ac97_write(codec, AC97_HANDSET_RATE, reg | 0x0080);
 		reg = ac97_read(codec, AC97_EXTENDED_MID);
 		ac97_write(codec, AC97_EXTENDED_MID, reg | 0x0200);
-		wm9713->pll_out = 0;
+		wm9713->pll_in = 0;
 		return 0;
 	}
 
@@ -788,7 +772,6 @@ static int wm9713_set_pll(struct snd_soc_codec *codec,
 	ac97_write(codec, AC97_EXTENDED_MID, reg & 0xfdff);
 	reg = ac97_read(codec, AC97_HANDSET_RATE);
 	ac97_write(codec, AC97_HANDSET_RATE, reg & 0xff7f);
-	wm9713->pll_out = freq_out;
 	wm9713->pll_in = freq_in;
 
 	/* wait 10ms AC97 link frames for the link to stabilise */
@@ -957,13 +940,14 @@ static void wm9713_voiceshutdown(struct snd_pcm_substream *substream,
 				 struct snd_soc_dai *dai)
 {
 	struct snd_soc_codec *codec = dai->codec;
-	u16 status;
+	u16 status, rate;
 
 	/* Gracefully shut down the voice interface. */
 	status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000;
-	ac97_write(codec, AC97_HANDSET_RATE, 0x0280);
+	rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF;
+	ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200);
 	schedule_timeout_interruptible(msecs_to_jiffies(1));
-	ac97_write(codec, AC97_HANDSET_RATE, 0x0F80);
+	ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00);
 	ac97_write(codec, AC97_EXTENDED_MID, status);
 }
 
@@ -1021,6 +1005,27 @@ static int ac97_aux_prepare(struct snd_pcm_substream *substream,
 	(SNDRV_PCM_FORMAT_S16_LE | SNDRV_PCM_FORMAT_S20_3LE | \
 	 SNDRV_PCM_FORMAT_S24_LE)
 
+static struct snd_soc_dai_ops wm9713_dai_ops_hifi = {
+	.prepare	= ac97_hifi_prepare,
+	.set_clkdiv	= wm9713_set_dai_clkdiv,
+	.set_pll	= wm9713_set_dai_pll,
+};
+
+static struct snd_soc_dai_ops wm9713_dai_ops_aux = {
+	.prepare	= ac97_aux_prepare,
+	.set_clkdiv	= wm9713_set_dai_clkdiv,
+	.set_pll	= wm9713_set_dai_pll,
+};
+
+static struct snd_soc_dai_ops wm9713_dai_ops_voice = {
+	.hw_params	= wm9713_pcm_hw_params,
+	.shutdown	= wm9713_voiceshutdown,
+	.set_clkdiv	= wm9713_set_dai_clkdiv,
+	.set_pll	= wm9713_set_dai_pll,
+	.set_fmt	= wm9713_set_dai_fmt,
+	.set_tristate	= wm9713_set_dai_tristate,
+};
+
 struct snd_soc_dai wm9713_dai[] = {
 {
 	.name = "AC97 HiFi",
@@ -1037,10 +1042,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.channels_max = 2,
 		.rates = WM9713_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_hifi_prepare,
-		.set_clkdiv = wm9713_set_dai_clkdiv,
-		.set_pll = wm9713_set_dai_pll,},
+	.ops = &wm9713_dai_ops_hifi,
 	},
 	{
 	.name = "AC97 Aux",
@@ -1050,10 +1052,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.channels_max = 1,
 		.rates = WM9713_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.prepare = ac97_aux_prepare,
-		.set_clkdiv = wm9713_set_dai_clkdiv,
-		.set_pll = wm9713_set_dai_pll,},
+	.ops = &wm9713_dai_ops_aux,
 	},
 	{
 	.name = "WM9713 Voice",
@@ -1069,14 +1068,7 @@ struct snd_soc_dai wm9713_dai[] = {
 		.channels_max = 2,
 		.rates = WM9713_PCM_RATES,
 		.formats = WM9713_PCM_FORMATS,},
-	.ops = {
-		.hw_params = wm9713_pcm_hw_params,
-		.shutdown = wm9713_voiceshutdown,
-		.set_clkdiv = wm9713_set_dai_clkdiv,
-		.set_pll = wm9713_set_dai_pll,
-		.set_fmt = wm9713_set_dai_fmt,
-		.set_tristate = wm9713_set_dai_tristate,
-	},
+	.ops = &wm9713_dai_ops_voice,
 	},
 };
 EXPORT_SYMBOL_GPL(wm9713_dai);
@@ -1132,7 +1124,7 @@ static int wm9713_soc_suspend(struct platform_device *pdev,
 	pm_message_t state)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	u16 reg;
 
 	/* Disable everything except touchpanel - that will be handled
@@ -1150,7 +1142,7 @@ static int wm9713_soc_suspend(struct platform_device *pdev,
 static int wm9713_soc_resume(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	struct wm9713_priv *wm9713 = codec->private_data;
 	int i, ret;
 	u16 *cache = codec->reg_cache;
@@ -1164,8 +1156,8 @@ static int wm9713_soc_resume(struct platform_device *pdev)
 	wm9713_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
 
 	/* do we need to re-start the PLL ? */
-	if (wm9713->pll_out)
-		wm9713_set_pll(codec, 0, wm9713->pll_in, wm9713->pll_out);
+	if (wm9713->pll_in)
+		wm9713_set_pll(codec, 0, wm9713->pll_in, 0);
 
 	/* only synchronise the codec if warm reset failed */
 	if (ret == 0) {
@@ -1191,10 +1183,11 @@ static int wm9713_soc_probe(struct platform_device *pdev)
 
 	printk(KERN_INFO "WM9713/WM9714 SoC Audio Codec %s\n", WM9713_VERSION);
 
-	socdev->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
-	if (socdev->codec == NULL)
+	socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec),
+				      GFP_KERNEL);
+	if (socdev->card->codec == NULL)
 		return -ENOMEM;
-	codec = socdev->codec;
+	codec = socdev->card->codec;
 	mutex_init(&codec->mutex);
 
 	codec->reg_cache = kmemdup(wm9713_reg, sizeof(wm9713_reg), GFP_KERNEL);
@@ -1245,7 +1238,8 @@ static int wm9713_soc_probe(struct platform_device *pdev)
 	reg = ac97_read(codec, AC97_CD) & 0x7fff;
 	ac97_write(codec, AC97_CD, reg);
 
-	wm9713_add_controls(codec);
+	snd_soc_add_controls(codec, wm9713_snd_ac97_controls,
+				ARRAY_SIZE(wm9713_snd_ac97_controls));
 	wm9713_add_widgets(codec);
 	ret = snd_soc_init_card(socdev);
 	if (ret < 0)
@@ -1265,15 +1259,15 @@ priv_err:
 	kfree(codec->reg_cache);
 
 cache_err:
-	kfree(socdev->codec);
-	socdev->codec = NULL;
+	kfree(socdev->card->codec);
+	socdev->card->codec = NULL;
 	return ret;
 }
 
 static int wm9713_soc_remove(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	if (codec == NULL)
 		return 0;
diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig
index b502741692d6..bd7392c9657e 100644
--- a/sound/soc/davinci/Kconfig
+++ b/sound/soc/davinci/Kconfig
@@ -20,7 +20,7 @@ config SND_DAVINCI_SOC_EVM
 
 config SND_DAVINCI_SOC_SFFSDR
 	tristate "SoC Audio support for SFFSDR"
-	depends on SND_DAVINCI_SOC && MACH_DAVINCI_SFFSDR
+	depends on SND_DAVINCI_SOC && MACH_SFFSDR
 	select SND_DAVINCI_SOC_I2S
 	select SND_SOC_PCM3008
 	select SFFSDR_FPGA
diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c
index 54851f318568..9b90b347007c 100644
--- a/sound/soc/davinci/davinci-evm.c
+++ b/sound/soc/davinci/davinci-evm.c
@@ -186,7 +186,8 @@ static int __init evm_init(void)
 
 	platform_set_drvdata(evm_snd_device, &evm_snd_devdata);
 	evm_snd_devdata.dev = &evm_snd_device->dev;
-	evm_snd_device->dev.platform_data = &evm_snd_data;
+	platform_device_add_data(evm_snd_device, &evm_snd_data,
+				 sizeof(evm_snd_data));
 
 	ret = platform_device_add_resources(evm_snd_device, evm_snd_resources,
 					    ARRAY_SIZE(evm_snd_resources));
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c
index 0fee779e3c76..ffdb9439d3d8 100644
--- a/sound/soc/davinci/davinci-i2s.c
+++ b/sound/soc/davinci/davinci-i2s.c
@@ -499,6 +499,13 @@ static void davinci_i2s_remove(struct platform_device *pdev,
 
 #define DAVINCI_I2S_RATES	SNDRV_PCM_RATE_8000_96000
 
+static struct snd_soc_dai_ops davinci_i2s_dai_ops = {
+	.startup	= davinci_i2s_startup,
+	.trigger	= davinci_i2s_trigger,
+	.hw_params	= davinci_i2s_hw_params,
+	.set_fmt	= davinci_i2s_set_dai_fmt,
+};
+
 struct snd_soc_dai davinci_i2s_dai = {
 	.name = "davinci-i2s",
 	.id = 0,
@@ -514,12 +521,7 @@ struct snd_soc_dai davinci_i2s_dai = {
 		.channels_max = 2,
 		.rates = DAVINCI_I2S_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.startup = davinci_i2s_startup,
-		.trigger = davinci_i2s_trigger,
-		.hw_params = davinci_i2s_hw_params,
-		.set_fmt = davinci_i2s_set_dai_fmt,
-	},
+	.ops = &davinci_i2s_dai_ops,
 };
 EXPORT_SYMBOL_GPL(davinci_i2s_dai);
 
diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c
index 366049d8578c..7af3b5b3a53d 100644
--- a/sound/soc/davinci/davinci-pcm.c
+++ b/sound/soc/davinci/davinci-pcm.c
@@ -286,7 +286,7 @@ static int davinci_pcm_mmap(struct snd_pcm_substream *substream,
 				     runtime->dma_bytes);
 }
 
-struct snd_pcm_ops davinci_pcm_ops = {
+static struct snd_pcm_ops davinci_pcm_ops = {
 	.open = 	davinci_pcm_open,
 	.close = 	davinci_pcm_close,
 	.ioctl = 	snd_pcm_lib_ioctl,
diff --git a/sound/soc/davinci/davinci-sffsdr.c b/sound/soc/davinci/davinci-sffsdr.c
index 4935d1bcbd8d..40eccfe9e358 100644
--- a/sound/soc/davinci/davinci-sffsdr.c
+++ b/sound/soc/davinci/davinci-sffsdr.c
@@ -25,7 +25,9 @@
 
 #include <asm/dma.h>
 #include <asm/mach-types.h>
+#ifdef CONFIG_SFFSDR_FPGA
 #include <asm/plat-sffsdr/sffsdr-fpga.h>
+#endif
 
 #include <mach/mcbsp.h>
 #include <mach/edma.h>
@@ -34,31 +36,45 @@
 #include "davinci-pcm.h"
 #include "davinci-i2s.h"
 
+/*
+ * CLKX and CLKR are the inputs for the Sample Rate Generator.
+ * FSX and FSR are outputs, driven by the sample Rate Generator.
+ */
+#define AUDIO_FORMAT (SND_SOC_DAIFMT_DSP_B |	\
+		      SND_SOC_DAIFMT_CBM_CFS |	\
+		      SND_SOC_DAIFMT_IB_NF)
+
 static int sffsdr_hw_params(struct snd_pcm_substream *substream,
-			    struct snd_pcm_hw_params *params,
-			    struct snd_soc_dai *dai)
+			    struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	int fs;
 	int ret = 0;
 
-	/* Set cpu DAI configuration:
-	 * CLKX and CLKR are the inputs for the Sample Rate Generator.
-	 * FSX and FSR are outputs, driven by the sample Rate Generator. */
-	ret = snd_soc_dai_set_fmt(cpu_dai,
-				  SND_SOC_DAIFMT_RIGHT_J |
-				  SND_SOC_DAIFMT_CBM_CFS |
-				  SND_SOC_DAIFMT_IB_NF);
-	if (ret < 0)
-		return ret;
-
 	/* Fsref can be 32000, 44100 or 48000. */
 	fs = params_rate(params);
 
+#ifndef CONFIG_SFFSDR_FPGA
+	/* Without the FPGA module, the Fs is fixed at 44100 Hz */
+	if (fs != 44100) {
+		pr_debug("warning: only 44.1 kHz is supported without SFFSDR FPGA module\n");
+		return -EINVAL;
+	}
+#endif
+
+	/* set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, AUDIO_FORMAT);
+	if (ret < 0)
+		return ret;
+
 	pr_debug("sffsdr_hw_params: rate = %d Hz\n", fs);
 
+#ifndef CONFIG_SFFSDR_FPGA
+	return 0;
+#else
 	return sffsdr_fpga_set_codec_fs(fs);
+#endif
 }
 
 static struct snd_soc_ops sffsdr_ops = {
@@ -127,7 +143,8 @@ static int __init sffsdr_init(void)
 
 	platform_set_drvdata(sffsdr_snd_device, &sffsdr_snd_devdata);
 	sffsdr_snd_devdata.dev = &sffsdr_snd_device->dev;
-	sffsdr_snd_device->dev.platform_data = &sffsdr_snd_data;
+	platform_device_add_data(sffsdr_snd_device, &sffsdr_snd_data,
+				 sizeof(sffsdr_snd_data));
 
 	ret = platform_device_add_resources(sffsdr_snd_device,
 					    sffsdr_snd_resources,
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 95c12b26fe37..9fc908283371 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -1,17 +1,18 @@
 config SND_SOC_OF_SIMPLE
 	tristate
 
+# ASoC platform support for the Freescale MPC8610 SOC.  This compiles drivers
+# for the SSI and the Elo DMA controller.  You will still need to select
+# a platform driver and a codec driver.
 config SND_SOC_MPC8610
-	bool "ALSA SoC support for the MPC8610 SOC"
-	depends on MPC8610_HPCD
-	default y if MPC8610
-	help
-	  Say Y if you want to add support for codecs attached to the SSI
-          device on an MPC8610.
+	tristate
+	depends on MPC8610
 
 config SND_SOC_MPC8610_HPCD
-	bool "ALSA SoC support for the Freescale MPC8610 HPCD board"
-	depends on SND_SOC_MPC8610
+	tristate "ALSA SoC support for the Freescale MPC8610 HPCD board"
+	# I2C is necessary for the CS4270 driver
+	depends on MPC8610_HPCD && I2C
+	select SND_SOC_MPC8610
 	select SND_SOC_CS4270
 	select SND_SOC_CS4270_VD33_ERRATA
 	default y if MPC8610_HPCD
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 035da4afec34..f85134c86387 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -2,10 +2,13 @@
 obj-$(CONFIG_SND_SOC_OF_SIMPLE) += soc-of-simple.o
 
 # MPC8610 HPCD Machine Support
-obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += mpc8610_hpcd.o
+snd-soc-mpc8610-hpcd-objs := mpc8610_hpcd.o
+obj-$(CONFIG_SND_SOC_MPC8610_HPCD) += snd-soc-mpc8610-hpcd.o
 
 # MPC8610 Platform Support
-obj-$(CONFIG_SND_SOC_MPC8610) += fsl_ssi.o fsl_dma.o
+snd-soc-fsl-ssi-objs := fsl_ssi.o
+snd-soc-fsl-dma-objs := fsl_dma.o
+obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o
 
 obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
 
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 64993eda5679..b3eb8570cd7b 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -142,7 +142,8 @@ static const struct snd_pcm_hardware fsl_dma_hardware = {
 	.info   		= SNDRV_PCM_INFO_INTERLEAVED |
 				  SNDRV_PCM_INFO_MMAP |
 				  SNDRV_PCM_INFO_MMAP_VALID |
-				  SNDRV_PCM_INFO_JOINT_DUPLEX,
+				  SNDRV_PCM_INFO_JOINT_DUPLEX |
+				  SNDRV_PCM_INFO_PAUSE,
 	.formats		= FSLDMA_PCM_FORMATS,
 	.rates  		= FSLDMA_PCM_RATES,
 	.rate_min       	= 5512,
@@ -464,11 +465,7 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
 		sizeof(struct fsl_dma_link_descriptor);
 
 	for (i = 0; i < NUM_DMA_LINKS; i++) {
-		struct fsl_dma_link_descriptor *link = &dma_private->link[i];
-
-		link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
-		link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
-		link->next = cpu_to_be64(temp_link);
+		dma_private->link[i].next = cpu_to_be64(temp_link);
 
 		temp_link += sizeof(struct fsl_dma_link_descriptor);
 	}
@@ -525,79 +522,9 @@ static int fsl_dma_open(struct snd_pcm_substream *substream)
  * This function obtains hardware parameters about the opened stream and
  * programs the DMA controller accordingly.
  *
- * Note that due to a quirk of the SSI's STX register, the target address
- * for the DMA operations depends on the sample size.  So we don't program
- * the dest_addr (for playback -- source_addr for capture) fields in the
- * link descriptors here.  We do that in fsl_dma_prepare()
- */
-static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
-	struct snd_pcm_hw_params *hw_params)
-{
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct fsl_dma_private *dma_private = runtime->private_data;
-
-	dma_addr_t temp_addr;   /* Pointer to next period */
-
-	unsigned int i;
-
-	/* Get all the parameters we need */
-	size_t buffer_size = params_buffer_bytes(hw_params);
-	size_t period_size = params_period_bytes(hw_params);
-
-	/* Initialize our DMA tracking variables */
-	dma_private->period_size = period_size;
-	dma_private->num_periods = params_periods(hw_params);
-	dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size;
-	dma_private->dma_buf_next = dma_private->dma_buf_phys +
-		(NUM_DMA_LINKS * period_size);
-	if (dma_private->dma_buf_next >= dma_private->dma_buf_end)
-		dma_private->dma_buf_next = dma_private->dma_buf_phys;
-
-	/*
-	 * The actual address in STX0 (destination for playback, source for
-	 * capture) is based on the sample size, but we don't know the sample
-	 * size in this function, so we'll have to adjust that later.  See
-	 * comments in fsl_dma_prepare().
-	 *
-	 * The DMA controller does not have a cache, so the CPU does not
-	 * need to tell it to flush its cache.  However, the DMA
-	 * controller does need to tell the CPU to flush its cache.
-	 * That's what the SNOOP bit does.
-	 *
-	 * Also, even though the DMA controller supports 36-bit addressing, for
-	 * simplicity we currently support only 32-bit addresses for the audio
-	 * buffer itself.
-	 */
-	temp_addr = substream->dma_buffer.addr;
-
-	for (i = 0; i < NUM_DMA_LINKS; i++) {
-		struct fsl_dma_link_descriptor *link = &dma_private->link[i];
-
-		link->count = cpu_to_be32(period_size);
-
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-			link->source_addr = cpu_to_be32(temp_addr);
-		else
-			link->dest_addr = cpu_to_be32(temp_addr);
-
-		temp_addr += period_size;
-	}
-
-	return 0;
-}
-
-/**
- * fsl_dma_prepare - prepare the DMA registers for playback.
- *
- * This function is called after the specifics of the audio data are known,
- * i.e. snd_pcm_runtime is initialized.
- *
- * In this function, we finish programming the registers of the DMA
- * controller that are dependent on the sample size.
- *
- * One of the drawbacks with big-endian is that when copying integers of
- * different sizes to a fixed-sized register, the address to which the
- * integer must be copied is dependent on the size of the integer.
+ * One drawback of big-endian is that when copying integers of different
+ * sizes to a fixed-sized register, the address to which the integer must be
+ * copied is dependent on the size of the integer.
  *
  * For example, if P is the address of a 32-bit register, and X is a 32-bit
  * integer, then X should be copied to address P.  However, if X is a 16-bit
@@ -613,22 +540,58 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
  * and 8 bytes at a time).  So we do not support packed 24-bit samples.
  * 24-bit data must be padded to 32 bits.
  */
-static int fsl_dma_prepare(struct snd_pcm_substream *substream)
+static int fsl_dma_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *hw_params)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct fsl_dma_private *dma_private = runtime->private_data;
+
+	/* Number of bits per sample */
+	unsigned int sample_size =
+		snd_pcm_format_physical_width(params_format(hw_params));
+
+	/* Number of bytes per frame */
+	unsigned int frame_size = 2 * (sample_size / 8);
+
+	/* Bus address of SSI STX register */
+	dma_addr_t ssi_sxx_phys = dma_private->ssi_sxx_phys;
+
+	/* Size of the DMA buffer, in bytes */
+	size_t buffer_size = params_buffer_bytes(hw_params);
+
+	/* Number of bytes per period */
+	size_t period_size = params_period_bytes(hw_params);
+
+	/* Pointer to next period */
+	dma_addr_t temp_addr = substream->dma_buffer.addr;
+
+	/* Pointer to DMA controller */
 	struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel;
-	u32 mr;
+
+	u32 mr; /* DMA Mode Register */
+
 	unsigned int i;
-	dma_addr_t ssi_sxx_phys;	/* Bus address of SSI STX register */
-	unsigned int frame_size;	/* Number of bytes per frame */
 
-	ssi_sxx_phys = dma_private->ssi_sxx_phys;
+	/* Initialize our DMA tracking variables */
+	dma_private->period_size = period_size;
+	dma_private->num_periods = params_periods(hw_params);
+	dma_private->dma_buf_end = dma_private->dma_buf_phys + buffer_size;
+	dma_private->dma_buf_next = dma_private->dma_buf_phys +
+		(NUM_DMA_LINKS * period_size);
+
+	if (dma_private->dma_buf_next >= dma_private->dma_buf_end)
+		/* This happens if the number of periods == NUM_DMA_LINKS */
+		dma_private->dma_buf_next = dma_private->dma_buf_phys;
 
 	mr = in_be32(&dma_channel->mr) & ~(CCSR_DMA_MR_BWC_MASK |
 		  CCSR_DMA_MR_SAHTS_MASK | CCSR_DMA_MR_DAHTS_MASK);
 
-	switch (runtime->sample_bits) {
+	/* Due to a quirk of the SSI's STX register, the target address
+	 * for the DMA operations depends on the sample size.  So we calculate
+	 * that offset here.  While we're at it, also tell the DMA controller
+	 * how much data to transfer per sample.
+	 */
+	switch (sample_size) {
 	case 8:
 		mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1;
 		ssi_sxx_phys += 3;
@@ -641,12 +604,12 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream)
 		mr |= CCSR_DMA_MR_DAHTS_4 | CCSR_DMA_MR_SAHTS_4;
 		break;
 	default:
+		/* We should never get here */
 		dev_err(substream->pcm->card->dev,
-			"unsupported sample size %u\n", runtime->sample_bits);
+			"unsupported sample size %u\n", sample_size);
 		return -EINVAL;
 	}
 
-	frame_size = runtime->frame_bits / 8;
 	/*
 	 * BWC should always be a multiple of the frame size.  BWC determines
 	 * how many bytes are sent/received before the DMA controller checks the
@@ -655,7 +618,6 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream)
 	 * capture, the receive FIFO is triggered when it contains one frame, so
 	 * we want to receive one frame at a time.
 	 */
-
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		mr |= CCSR_DMA_MR_BWC(2 * frame_size);
 	else
@@ -663,16 +625,48 @@ static int fsl_dma_prepare(struct snd_pcm_substream *substream)
 
 	out_be32(&dma_channel->mr, mr);
 
-	/*
-	 * Program the address of the DMA transfer to/from the SSI.
-	 */
 	for (i = 0; i < NUM_DMA_LINKS; i++) {
 		struct fsl_dma_link_descriptor *link = &dma_private->link[i];
 
-		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		link->count = cpu_to_be32(period_size);
+
+		/* Even though the DMA controller supports 36-bit addressing,
+		 * for simplicity we allow only 32-bit addresses for the audio
+		 * buffer itself.  This was enforced in fsl_dma_new() with the
+		 * DMA mask.
+		 *
+		 * The snoop bit tells the DMA controller whether it should tell
+		 * the ECM to snoop during a read or write to an address. For
+		 * audio, we use DMA to transfer data between memory and an I/O
+		 * device (the SSI's STX0 or SRX0 register). Snooping is only
+		 * needed if there is a cache, so we need to snoop memory
+		 * addresses only.  For playback, that means we snoop the source
+		 * but not the destination.  For capture, we snoop the
+		 * destination but not the source.
+		 *
+		 * Note that failing to snoop properly is unlikely to cause
+		 * cache incoherency if the period size is larger than the
+		 * size of L1 cache.  This is because filling in one period will
+		 * flush out the data for the previous period.  So if you
+		 * increased period_bytes_min to a large enough size, you might
+		 * get more performance by not snooping, and you'll still be
+		 * okay.
+		 */
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+			link->source_addr = cpu_to_be32(temp_addr);
+			link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
+
 			link->dest_addr = cpu_to_be32(ssi_sxx_phys);
-		else
+			link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP);
+		} else {
 			link->source_addr = cpu_to_be32(ssi_sxx_phys);
+			link->source_attr = cpu_to_be32(CCSR_DMA_ATR_NOSNOOP);
+
+			link->dest_addr = cpu_to_be32(temp_addr);
+			link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP);
+		}
+
+		temp_addr += period_size;
 	}
 
 	return 0;
@@ -808,7 +802,6 @@ static struct snd_pcm_ops fsl_dma_ops = {
 	.ioctl  	= snd_pcm_lib_ioctl,
 	.hw_params      = fsl_dma_hw_params,
 	.hw_free	= fsl_dma_hw_free,
-	.prepare	= fsl_dma_prepare,
 	.pointer	= fsl_dma_pointer,
 };
 
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index c6d6eb71dc1d..169bca295b78 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -72,6 +72,7 @@
  * @dev: struct device pointer
  * @playback: the number of playback streams opened
  * @capture: the number of capture streams opened
+ * @asynchronous: 0=synchronous mode, 1=asynchronous mode
  * @cpu_dai: the CPU DAI for this device
  * @dev_attr: the sysfs device attribute structure
  * @stats: SSI statistics
@@ -86,6 +87,7 @@ struct fsl_ssi_private {
 	struct device *dev;
 	unsigned int playback;
 	unsigned int capture;
+	int asynchronous;
 	struct snd_soc_dai cpu_dai;
 	struct device_attribute dev_attr;
 
@@ -301,9 +303,10 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 		 *
 		 * FIXME: Little-endian samples require a different shift dir
 		 */
-		clrsetbits_be32(&ssi->scr, CCSR_SSI_SCR_I2S_MODE_MASK,
-			CCSR_SSI_SCR_TFR_CLK_DIS |
-			CCSR_SSI_SCR_I2S_MODE_SLAVE | CCSR_SSI_SCR_SYN);
+		clrsetbits_be32(&ssi->scr,
+			CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_SYN,
+			CCSR_SSI_SCR_TFR_CLK_DIS | CCSR_SSI_SCR_I2S_MODE_SLAVE
+			| (ssi_private->asynchronous ? 0 : CCSR_SSI_SCR_SYN));
 
 		out_be32(&ssi->stcr,
 			 CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFEN0 |
@@ -382,10 +385,15 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 			SNDRV_PCM_HW_PARAM_RATE,
 			first_runtime->rate, first_runtime->rate);
 
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-			SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
-			first_runtime->sample_bits,
-			first_runtime->sample_bits);
+		/* If we're in synchronous mode, then we need to constrain
+		 * the sample size as well.  We don't support independent sample
+		 * rates in asynchronous mode.
+		 */
+		if (!ssi_private->asynchronous)
+			snd_pcm_hw_constraint_minmax(substream->runtime,
+				SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+				first_runtime->sample_bits,
+				first_runtime->sample_bits);
 
 		ssi_private->second_stream = substream;
 	}
@@ -400,7 +408,7 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
 }
 
 /**
- * fsl_ssi_prepare: prepare the SSI.
+ * fsl_ssi_hw_params - program the sample size
  *
  * Most of the SSI registers have been programmed in the startup function,
  * but the word length must be programmed here.  Unfortunately, programming
@@ -412,23 +420,27 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream,
  * Note: The SxCCR.DC and SxCCR.PM bits are only used if the SSI is the
  * clock master.
  */
-static int fsl_ssi_prepare(struct snd_pcm_substream *substream,
-			   struct snd_soc_dai *dai)
+static int fsl_ssi_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *hw_params, struct snd_soc_dai *cpu_dai)
 {
-	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data;
-
-	struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+	struct fsl_ssi_private *ssi_private = cpu_dai->private_data;
 
 	if (substream == ssi_private->first_stream) {
-		u32 wl;
+		struct ccsr_ssi __iomem *ssi = ssi_private->ssi;
+		unsigned int sample_size =
+			snd_pcm_format_width(params_format(hw_params));
+		u32 wl = CCSR_SSI_SxCCR_WL(sample_size);
 
 		/* The SSI should always be disabled at this points (SSIEN=0) */
-		wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format));
 
 		/* In synchronous mode, the SSI uses STCCR for capture */
-		clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl);
+		if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ||
+		    !ssi_private->asynchronous)
+			clrsetbits_be32(&ssi->stccr,
+					CCSR_SSI_SxCCR_WL_MASK, wl);
+		else
+			clrsetbits_be32(&ssi->srccr,
+					CCSR_SSI_SxCCR_WL_MASK, wl);
 	}
 
 	return 0;
@@ -452,28 +464,33 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd,
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
-	case SNDRV_PCM_TRIGGER_RESUME:
+		clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-			clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
 			setbits32(&ssi->scr,
 				CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE);
 		} else {
-			clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN);
+			long timeout = jiffies + 10;
+
 			setbits32(&ssi->scr,
 				CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE);
 
-			/*
-			 * I think we need this delay to allow time for the SSI
-			 * to put data into its FIFO.  Without it, ALSA starts
-			 * to complain about overruns.
+			/* Wait until the SSI has filled its FIFO. Without this
+			 * delay, ALSA complains about overruns.  When the FIFO
+			 * is full, the DMA controller initiates its first
+			 * transfer.  Until then, however, the DMA's DAR
+			 * register is zero, which translates to an
+			 * out-of-bounds pointer.  This makes ALSA think an
+			 * overrun has occurred.
 			 */
-			mdelay(1);
+			while (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0) &&
+			       (jiffies < timeout));
+			if (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0))
+				return -EIO;
 		}
 		break;
 
 	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 			clrbits32(&ssi->scr, CCSR_SSI_SCR_TE);
@@ -563,6 +580,15 @@ static int fsl_ssi_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
 /**
  * fsl_ssi_dai_template: template CPU DAI for the SSI
  */
+static struct snd_soc_dai_ops fsl_ssi_dai_ops = {
+	.startup	= fsl_ssi_startup,
+	.hw_params	= fsl_ssi_hw_params,
+	.shutdown	= fsl_ssi_shutdown,
+	.trigger	= fsl_ssi_trigger,
+	.set_sysclk	= fsl_ssi_set_sysclk,
+	.set_fmt	= fsl_ssi_set_fmt,
+};
+
 static struct snd_soc_dai fsl_ssi_dai_template = {
 	.playback = {
 		/* The SSI does not support monaural audio. */
@@ -577,14 +603,7 @@ static struct snd_soc_dai fsl_ssi_dai_template = {
 		.rates = FSLSSI_I2S_RATES,
 		.formats = FSLSSI_I2S_FORMATS,
 	},
-	.ops = {
-		.startup = fsl_ssi_startup,
-		.prepare = fsl_ssi_prepare,
-		.shutdown = fsl_ssi_shutdown,
-		.trigger = fsl_ssi_trigger,
-		.set_sysclk = fsl_ssi_set_sysclk,
-		.set_fmt = fsl_ssi_set_fmt,
-	},
+	.ops = &fsl_ssi_dai_ops,
 };
 
 /**
@@ -654,6 +673,7 @@ struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info)
 	ssi_private->ssi_phys = ssi_info->ssi_phys;
 	ssi_private->irq = ssi_info->irq;
 	ssi_private->dev = ssi_info->dev;
+	ssi_private->asynchronous = ssi_info->asynchronous;
 
 	ssi_private->dev->driver_data = fsl_ssi_dai;
 
@@ -704,6 +724,14 @@ void fsl_ssi_destroy_dai(struct snd_soc_dai *fsl_ssi_dai)
 }
 EXPORT_SYMBOL_GPL(fsl_ssi_destroy_dai);
 
+static int __init fsl_ssi_init(void)
+{
+	printk(KERN_INFO "Freescale Synchronous Serial Interface (SSI) ASoC Driver\n");
+
+	return 0;
+}
+module_init(fsl_ssi_init);
+
 MODULE_AUTHOR("Timur Tabi <timur@freescale.com>");
 MODULE_DESCRIPTION("Freescale Synchronous Serial Interface (SSI) ASoC Driver");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h
index 83b44d700e33..eade01feaab6 100644
--- a/sound/soc/fsl/fsl_ssi.h
+++ b/sound/soc/fsl/fsl_ssi.h
@@ -208,6 +208,7 @@ struct ccsr_ssi {
  * ssi_phys: physical address of the SSI registers
  * irq: IRQ of this SSI
  * dev: struct device, used to create the sysfs statistics file
+ * asynchronous: 0=synchronous mode, 1=asynchronous mode
 */
 struct fsl_ssi_info {
 	unsigned int id;
@@ -215,6 +216,7 @@ struct fsl_ssi_info {
 	dma_addr_t ssi_phys;
 	unsigned int irq;
 	struct device *dev;
+	int asynchronous;
 };
 
 struct snd_soc_dai *fsl_ssi_create_dai(struct fsl_ssi_info *ssi_info);
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 9eb1ce185bd0..3aa729df27b5 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -468,6 +468,16 @@ static int psc_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int format)
 /**
  * psc_i2s_dai_template: template CPU Digital Audio Interface
  */
+static struct snd_soc_dai_ops psc_i2s_dai_ops = {
+	.startup	= psc_i2s_startup,
+	.hw_params	= psc_i2s_hw_params,
+	.hw_free	= psc_i2s_hw_free,
+	.shutdown	= psc_i2s_shutdown,
+	.trigger	= psc_i2s_trigger,
+	.set_sysclk	= psc_i2s_set_sysclk,
+	.set_fmt	= psc_i2s_set_fmt,
+};
+
 static struct snd_soc_dai psc_i2s_dai_template = {
 	.playback = {
 		.channels_min = 2,
@@ -481,15 +491,7 @@ static struct snd_soc_dai psc_i2s_dai_template = {
 		.rates = PSC_I2S_RATES,
 		.formats = PSC_I2S_FORMATS,
 	},
-	.ops = {
-		.startup = psc_i2s_startup,
-		.hw_params = psc_i2s_hw_params,
-		.hw_free = psc_i2s_hw_free,
-		.shutdown = psc_i2s_shutdown,
-		.trigger = psc_i2s_trigger,
-		.set_sysclk = psc_i2s_set_sysclk,
-		.set_fmt = psc_i2s_set_fmt,
-	},
+	.ops = &psc_i2s_dai_ops,
 };
 
 /* ---------------------------------------------------------------------
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index acf39a646b2f..ef67d1cdffe7 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -353,6 +353,11 @@ static int mpc8610_hpcd_probe(struct of_device *ofdev,
 	}
 	ssi_info.irq = machine_data->ssi_irq;
 
+	/* Do we want to use asynchronous mode? */
+	ssi_info.asynchronous =
+		of_find_property(np, "fsl,ssi-asynchronous", NULL) ? 1 : 0;
+	if (ssi_info.asynchronous)
+		dev_info(&ofdev->dev, "using asynchronous mode\n");
 
 	/* Map the global utilities registers. */
 	guts_np = of_find_compatible_node(NULL, NULL, "fsl,mpc8610-guts");
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 4f7f04014585..675732e724d5 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -8,7 +8,7 @@ config SND_OMAP_SOC_MCBSP
 
 config SND_OMAP_SOC_N810
 	tristate "SoC Audio support for Nokia N810"
-	depends on SND_OMAP_SOC && MACH_NOKIA_N810
+	depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C
 	select SND_OMAP_SOC_MCBSP
 	select OMAP_MUX
 	select SND_SOC_TLV320AIC3X
@@ -17,7 +17,7 @@ config SND_OMAP_SOC_N810
 
 config SND_OMAP_SOC_OSK5912
 	tristate "SoC Audio support for omap osk5912"
-	depends on SND_OMAP_SOC && MACH_OMAP_OSK
+	depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C
 	select SND_OMAP_SOC_MCBSP
 	select SND_SOC_TLV320AIC23
 	help
@@ -55,3 +55,13 @@ config SND_OMAP_SOC_OMAP3_PANDORA
 	select SND_SOC_TWL4030
 	help
 	  Say Y if you want to add support for SoC audio on the OMAP3 Pandora.
+
+config SND_OMAP_SOC_OMAP3_BEAGLE
+	tristate "SoC Audio support for OMAP3 Beagle"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_BEAGLE
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on the Beagleboard.
+
+
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 76fedd96e365..0c9e4ac37660 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -12,6 +12,7 @@ snd-soc-overo-objs := overo.o
 snd-soc-omap2evm-objs := omap2evm.o
 snd-soc-sdp3430-objs := sdp3430.o
 snd-soc-omap3pandora-objs := omap3pandora.o
+snd-soc-omap3beagle-objs := omap3beagle.o
 
 obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
 obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
@@ -19,3 +20,4 @@ obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
 obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o
 obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index 25593fee9121..a6d1178ce128 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -40,6 +40,13 @@
 #define N810_HEADSET_AMP_GPIO	10
 #define N810_SPEAKER_AMP_GPIO	101
 
+enum {
+	N810_JACK_DISABLED,
+	N810_JACK_HP,
+	N810_JACK_HS,
+	N810_JACK_MIC,
+};
+
 static struct clk *sys_clkout2;
 static struct clk *sys_clkout2_src;
 static struct clk *func96m_clk;
@@ -50,15 +57,32 @@ static int n810_dmic_func;
 
 static void n810_ext_control(struct snd_soc_codec *codec)
 {
+	int hp = 0, line1l = 0;
+
+	switch (n810_jack_func) {
+	case N810_JACK_HS:
+		line1l = 1;
+	case N810_JACK_HP:
+		hp = 1;
+		break;
+	case N810_JACK_MIC:
+		line1l = 1;
+		break;
+	}
+
 	if (n810_spk_func)
 		snd_soc_dapm_enable_pin(codec, "Ext Spk");
 	else
 		snd_soc_dapm_disable_pin(codec, "Ext Spk");
 
-	if (n810_jack_func)
+	if (hp)
 		snd_soc_dapm_enable_pin(codec, "Headphone Jack");
 	else
 		snd_soc_dapm_disable_pin(codec, "Headphone Jack");
+	if (line1l)
+		snd_soc_dapm_enable_pin(codec, "LINE1L");
+	else
+		snd_soc_dapm_disable_pin(codec, "LINE1L");
 
 	if (n810_dmic_func)
 		snd_soc_dapm_enable_pin(codec, "DMic");
@@ -72,7 +96,7 @@ static int n810_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->codec;
+	struct snd_soc_codec *codec = rtd->socdev->card->codec;
 
 	snd_pcm_hw_constraint_minmax(runtime,
 				     SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
@@ -229,7 +253,7 @@ static const struct snd_soc_dapm_route audio_map[] = {
 };
 
 static const char *spk_function[] = {"Off", "On"};
-static const char *jack_function[] = {"Off", "Headphone"};
+static const char *jack_function[] = {"Off", "Headphone", "Headset", "Mic"};
 static const char *input_function[] = {"ADC", "Digital Mic"};
 static const struct soc_enum n810_enum[] = {
 	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
@@ -248,20 +272,23 @@ static const struct snd_kcontrol_new aic33_n810_controls[] = {
 
 static int n810_aic33_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
 	/* Not connected */
 	snd_soc_dapm_nc_pin(codec, "MONO_LOUT");
 	snd_soc_dapm_nc_pin(codec, "HPLCOM");
 	snd_soc_dapm_nc_pin(codec, "HPRCOM");
+	snd_soc_dapm_nc_pin(codec, "MIC3L");
+	snd_soc_dapm_nc_pin(codec, "MIC3R");
+	snd_soc_dapm_nc_pin(codec, "LINE1R");
+	snd_soc_dapm_nc_pin(codec, "LINE2L");
+	snd_soc_dapm_nc_pin(codec, "LINE2R");
 
 	/* Add N810 specific controls */
-	for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&aic33_n810_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, aic33_n810_controls,
+				ARRAY_SIZE(aic33_n810_controls));
+	if (err < 0)
+		return err;
 
 	/* Add N810 specific widgets */
 	snd_soc_dapm_new_controls(codec, aic33_dapm_widgets,
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 05dd5abcddf4..d6882be33452 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -461,6 +461,16 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	return err;
 }
 
+static struct snd_soc_dai_ops omap_mcbsp_dai_ops = {
+	.startup	= omap_mcbsp_dai_startup,
+	.shutdown	= omap_mcbsp_dai_shutdown,
+	.trigger	= omap_mcbsp_dai_trigger,
+	.hw_params	= omap_mcbsp_dai_hw_params,
+	.set_fmt	= omap_mcbsp_dai_set_dai_fmt,
+	.set_clkdiv	= omap_mcbsp_dai_set_clkdiv,
+	.set_sysclk	= omap_mcbsp_dai_set_dai_sysclk,
+};
+
 #define OMAP_MCBSP_DAI_BUILDER(link_id)				\
 {								\
 	.name = "omap-mcbsp-dai-"#link_id,			\
@@ -477,15 +487,7 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 		.rates = OMAP_MCBSP_RATES,			\
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
 	},							\
-	.ops = {						\
-		.startup = omap_mcbsp_dai_startup,		\
-		.shutdown = omap_mcbsp_dai_shutdown,		\
-		.trigger = omap_mcbsp_dai_trigger,		\
-		.hw_params = omap_mcbsp_dai_hw_params,		\
-		.set_fmt = omap_mcbsp_dai_set_dai_fmt,		\
-		.set_clkdiv = omap_mcbsp_dai_set_clkdiv,	\
-		.set_sysclk = omap_mcbsp_dai_set_dai_sysclk,	\
-	},							\
+	.ops = &omap_mcbsp_dai_ops,				\
 	.private_data = &mcbsp_data[(link_id)].bus_id,		\
 }
 
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index dd3bb2933762..8e1431cb46bb 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -265,7 +265,7 @@ static int omap_pcm_mmap(struct snd_pcm_substream *substream,
 				     runtime->dma_bytes);
 }
 
-struct snd_pcm_ops omap_pcm_ops = {
+static struct snd_pcm_ops omap_pcm_ops = {
 	.open		= omap_pcm_open,
 	.close		= omap_pcm_close,
 	.ioctl		= snd_pcm_lib_ioctl,
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index fcc2f5d9a878..fe282d4ef422 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -143,7 +143,7 @@ static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = {
 };
 
 static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = {
-	SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+	SND_SOC_DAPM_MIC("Mic (internal)", NULL),
 	SND_SOC_DAPM_MIC("Mic (external)", NULL),
 	SND_SOC_DAPM_LINE("Line In", NULL),
 };
@@ -155,16 +155,33 @@ static const struct snd_soc_dapm_route omap3pandora_out_map[] = {
 };
 
 static const struct snd_soc_dapm_route omap3pandora_in_map[] = {
-	{"INL", NULL, "Line In"},
-	{"INR", NULL, "Line In"},
-	{"INL", NULL, "Mic (Internal)"},
-	{"INR", NULL, "Mic (external)"},
+	{"AUXL", NULL, "Line In"},
+	{"AUXR", NULL, "Line In"},
+
+	{"MAINMIC", NULL, "Mic Bias 1"},
+	{"Mic Bias 1", NULL, "Mic (internal)"},
+
+	{"SUBMIC", NULL, "Mic Bias 2"},
+	{"Mic Bias 2", NULL, "Mic (external)"},
 };
 
 static int omap3pandora_out_init(struct snd_soc_codec *codec)
 {
 	int ret;
 
+	/* All TWL4030 output pins are floating */
+	snd_soc_dapm_nc_pin(codec, "OUTL");
+	snd_soc_dapm_nc_pin(codec, "OUTR");
+	snd_soc_dapm_nc_pin(codec, "EARPIECE");
+	snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
+	snd_soc_dapm_nc_pin(codec, "PREDRIVER");
+	snd_soc_dapm_nc_pin(codec, "HSOL");
+	snd_soc_dapm_nc_pin(codec, "HSOR");
+	snd_soc_dapm_nc_pin(codec, "CARKITL");
+	snd_soc_dapm_nc_pin(codec, "CARKITR");
+	snd_soc_dapm_nc_pin(codec, "HFL");
+	snd_soc_dapm_nc_pin(codec, "HFR");
+
 	ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets,
 				ARRAY_SIZE(omap3pandora_out_dapm_widgets));
 	if (ret < 0)
@@ -180,18 +197,11 @@ static int omap3pandora_in_init(struct snd_soc_codec *codec)
 {
 	int ret;
 
-	/* All TWL4030 output pins are floating */
-	snd_soc_dapm_nc_pin(codec, "OUTL"),
-	snd_soc_dapm_nc_pin(codec, "OUTR"),
-	snd_soc_dapm_nc_pin(codec, "EARPIECE"),
-	snd_soc_dapm_nc_pin(codec, "PREDRIVEL"),
-	snd_soc_dapm_nc_pin(codec, "PREDRIVER"),
-	snd_soc_dapm_nc_pin(codec, "HSOL"),
-	snd_soc_dapm_nc_pin(codec, "HSOR"),
-	snd_soc_dapm_nc_pin(codec, "CARKITL"),
-	snd_soc_dapm_nc_pin(codec, "CARKITR"),
-	snd_soc_dapm_nc_pin(codec, "HFL"),
-	snd_soc_dapm_nc_pin(codec, "HFR"),
+	/* Not comnnected */
+	snd_soc_dapm_nc_pin(codec, "HSMIC");
+	snd_soc_dapm_nc_pin(codec, "CARKITMIC");
+	snd_soc_dapm_nc_pin(codec, "DIGIMIC0");
+	snd_soc_dapm_nc_pin(codec, "DIGIMIC1");
 
 	ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets,
 				ARRAY_SIZE(omap3pandora_in_dapm_widgets));
@@ -251,10 +261,9 @@ static int __init omap3pandora_soc_init(void)
 {
 	int ret;
 
-	if (!machine_is_omap3_pandora()) {
-		pr_debug(PREFIX "Not OMAP3 Pandora\n");
+	if (!machine_is_omap3_pandora())
 		return -ENODEV;
-	}
+
 	pr_info("OMAP3 Pandora SoC init\n");
 
 	ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power");
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index cd41a948df7b..a952a4eb3361 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -186,13 +186,6 @@ static int __init osk_soc_init(void)
 		return -ENODEV;
 	}
 
-	if (clk_get_usecount(tlv320aic23_mclk) > 0) {
-		/* MCLK is already in use */
-		printk(KERN_WARNING
-		       "MCLK in use at %d Hz. We change it to %d Hz\n",
-		       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
-	}
-
 	/*
 	 * Configure 12 MHz output on MCLK.
 	 */
@@ -205,9 +198,8 @@ static int __init osk_soc_init(void)
 		}
 	}
 
-	printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n",
-	       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK,
-	       clk_get_usecount(tlv320aic23_mclk));
+	printk(KERN_INFO "MCLK = %d [%d]\n",
+	       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
 
 	return 0;
 err1:
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index e226fa75669c..10f1c867f11d 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -28,6 +28,7 @@
 #include <sound/pcm.h>
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
+#include <sound/jack.h>
 
 #include <asm/mach-types.h>
 #include <mach/hardware.h>
@@ -38,6 +39,8 @@
 #include "omap-pcm.h"
 #include "../codecs/twl4030.h"
 
+static struct snd_soc_card snd_soc_sdp3430;
+
 static int sdp3430_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
@@ -81,12 +84,121 @@ static struct snd_soc_ops sdp3430_ops = {
 	.hw_params = sdp3430_hw_params,
 };
 
+/* Headset jack */
+static struct snd_soc_jack hs_jack;
+
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{
+		.pin = "Headset Mic",
+		.mask = SND_JACK_MICROPHONE,
+	},
+	{
+		.pin = "Headset Stereophone",
+		.mask = SND_JACK_HEADPHONE,
+	},
+};
+
+/* Headset jack detection gpios */
+static struct snd_soc_jack_gpio hs_jack_gpios[] = {
+	{
+		.gpio = (OMAP_MAX_GPIO_LINES + 2),
+		.name = "hsdet-gpio",
+		.report = SND_JACK_HEADSET,
+		.debounce_time = 200,
+	},
+};
+
+/* SDP3430 machine DAPM */
+static const struct snd_soc_dapm_widget sdp3430_twl4030_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Ext Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* External Mics: MAINMIC, SUBMIC with bias*/
+	{"MAINMIC", NULL, "Mic Bias 1"},
+	{"SUBMIC", NULL, "Mic Bias 2"},
+	{"Mic Bias 1", NULL, "Ext Mic"},
+	{"Mic Bias 2", NULL, "Ext Mic"},
+
+	/* External Speakers: HFL, HFR */
+	{"Ext Spk", NULL, "HFL"},
+	{"Ext Spk", NULL, "HFR"},
+
+	/* Headset Mic: HSMIC with bias */
+	{"HSMIC", NULL, "Headset Mic Bias"},
+	{"Headset Mic Bias", NULL, "Headset Mic"},
+
+	/* Headset Stereophone (Headphone): HSOL, HSOR */
+	{"Headset Stereophone", NULL, "HSOL"},
+	{"Headset Stereophone", NULL, "HSOR"},
+};
+
+static int sdp3430_twl4030_init(struct snd_soc_codec *codec)
+{
+	int ret;
+
+	/* Add SDP3430 specific widgets */
+	ret = snd_soc_dapm_new_controls(codec, sdp3430_twl4030_dapm_widgets,
+				ARRAY_SIZE(sdp3430_twl4030_dapm_widgets));
+	if (ret)
+		return ret;
+
+	/* Set up SDP3430 specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	/* SDP3430 connected pins */
+	snd_soc_dapm_enable_pin(codec, "Ext Mic");
+	snd_soc_dapm_enable_pin(codec, "Ext Spk");
+	snd_soc_dapm_disable_pin(codec, "Headset Mic");
+	snd_soc_dapm_disable_pin(codec, "Headset Stereophone");
+
+	/* TWL4030 not connected pins */
+	snd_soc_dapm_nc_pin(codec, "AUXL");
+	snd_soc_dapm_nc_pin(codec, "AUXR");
+	snd_soc_dapm_nc_pin(codec, "CARKITMIC");
+	snd_soc_dapm_nc_pin(codec, "DIGIMIC0");
+	snd_soc_dapm_nc_pin(codec, "DIGIMIC1");
+
+	snd_soc_dapm_nc_pin(codec, "OUTL");
+	snd_soc_dapm_nc_pin(codec, "OUTR");
+	snd_soc_dapm_nc_pin(codec, "EARPIECE");
+	snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
+	snd_soc_dapm_nc_pin(codec, "PREDRIVER");
+	snd_soc_dapm_nc_pin(codec, "CARKITL");
+	snd_soc_dapm_nc_pin(codec, "CARKITR");
+
+	ret = snd_soc_dapm_sync(codec);
+	if (ret)
+		return ret;
+
+	/* Headset jack detection */
+	ret = snd_soc_jack_new(&snd_soc_sdp3430, "Headset Jack",
+				SND_JACK_HEADSET, &hs_jack);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+				hs_jack_pins);
+	if (ret)
+		return ret;
+
+	ret = snd_soc_jack_add_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
+				hs_jack_gpios);
+
+	return ret;
+}
+
 /* Digital audio interface glue - connects codec <--> CPU */
 static struct snd_soc_dai_link sdp3430_dai = {
 	.name = "TWL4030",
 	.stream_name = "TWL4030",
 	.cpu_dai = &omap_mcbsp_dai[0],
 	.codec_dai = &twl4030_dai,
+	.init = sdp3430_twl4030_init,
 	.ops = &sdp3430_ops,
 };
 
@@ -142,6 +254,9 @@ module_init(sdp3430_soc_init);
 
 static void __exit sdp3430_soc_exit(void)
 {
+	snd_soc_jack_free_gpios(&hs_jack, ARRAY_SIZE(hs_jack_gpios),
+				hs_jack_gpios);
+
 	platform_device_unregister(sdp3430_snd_device);
 }
 module_exit(sdp3430_soc_exit);
diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig
index f82e10699471..5998ab366e83 100644
--- a/sound/soc/pxa/Kconfig
+++ b/sound/soc/pxa/Kconfig
@@ -61,6 +61,24 @@ config SND_PXA2XX_SOC_TOSA
 	  Say Y if you want to add support for SoC audio on Sharp
 	  Zaurus SL-C6000x models (Tosa).
 
+config SND_PXA2XX_SOC_E740
+	tristate "SoC AC97 Audio support for e740"
+	depends on SND_PXA2XX_SOC && MACH_E740
+	select SND_SOC_WM9705
+	select SND_PXA2XX_SOC_AC97
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  toshiba e740 PDA
+
+config SND_PXA2XX_SOC_E750
+	tristate "SoC AC97 Audio support for e750"
+	depends on SND_PXA2XX_SOC && MACH_E750
+	select SND_SOC_WM9705
+	select SND_PXA2XX_SOC_AC97
+	help
+	  Say Y if you want to add support for SoC audio on the
+	  toshiba e750 PDA
+
 config SND_PXA2XX_SOC_E800
 	tristate "SoC AC97 Audio support for e800"
 	depends on SND_PXA2XX_SOC && MACH_E800
@@ -97,3 +115,12 @@ config SND_SOC_ZYLONITE
 	help
 	  Say Y if you want to add support for SoC audio on the
 	  Marvell Zylonite reference platform.
+
+config SND_PXA2XX_SOC_MIOA701
+        tristate "SoC Audio support for MIO A701"
+        depends on SND_PXA2XX_SOC && MACH_MIOA701
+        select SND_PXA2XX_SOC_AC97
+        select SND_SOC_WM9713
+        help
+          Say Y if you want to add support for SoC audio on the
+          MIO A701.
diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile
index 08a9f2797729..8ed881c5e5cc 100644
--- a/sound/soc/pxa/Makefile
+++ b/sound/soc/pxa/Makefile
@@ -13,17 +13,23 @@ obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o
 snd-soc-corgi-objs := corgi.o
 snd-soc-poodle-objs := poodle.o
 snd-soc-tosa-objs := tosa.o
+snd-soc-e740-objs := e740_wm9705.o
+snd-soc-e750-objs := e750_wm9705.o
 snd-soc-e800-objs := e800_wm9712.o
 snd-soc-spitz-objs := spitz.o
 snd-soc-em-x270-objs := em-x270.o
 snd-soc-palm27x-objs := palm27x.o
 snd-soc-zylonite-objs := zylonite.o
+snd-soc-mioa701-objs := mioa701_wm9713.o
 
 obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o
 obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
 obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o
+obj-$(CONFIG_SND_PXA2XX_SOC_E740) += snd-soc-e740.o
+obj-$(CONFIG_SND_PXA2XX_SOC_E750) += snd-soc-e750.o
 obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o
 obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o
 obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o
 obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
+obj-$(CONFIG_SND_PXA2XX_SOC_MIOA701) += snd-soc-mioa701.o
 obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o
diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c
index 1ba25a559524..d5be2b30cda5 100644
--- a/sound/soc/pxa/corgi.c
+++ b/sound/soc/pxa/corgi.c
@@ -16,6 +16,7 @@
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <linux/timer.h>
+#include <linux/i2c.h>
 #include <linux/interrupt.h>
 #include <linux/platform_device.h>
 #include <linux/gpio.h>
@@ -25,8 +26,6 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/mach-types.h>
-#include <mach/pxa-regs.h>
-#include <mach/hardware.h>
 #include <mach/corgi.h>
 #include <mach/audio.h>
 
@@ -100,7 +99,7 @@ static void corgi_ext_control(struct snd_soc_codec *codec)
 static int corgi_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->codec;
+	struct snd_soc_codec *codec = rtd->socdev->card->codec;
 
 	/* check the jack status at stream startup */
 	corgi_ext_control(codec);
@@ -275,18 +274,16 @@ static const struct snd_kcontrol_new wm8731_corgi_controls[] = {
  */
 static int corgi_wm8731_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
 	snd_soc_dapm_nc_pin(codec, "LLINEIN");
 	snd_soc_dapm_nc_pin(codec, "RLINEIN");
 
 	/* Add corgi specific controls */
-	for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&wm8731_corgi_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, wm8731_corgi_controls,
+				ARRAY_SIZE(wm8731_corgi_controls));
+	if (err < 0)
+		return err;
 
 	/* Add corgi specific widgets */
 	snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
@@ -317,19 +314,44 @@ static struct snd_soc_card snd_soc_corgi = {
 	.num_links = 1,
 };
 
-/* corgi audio private data */
-static struct wm8731_setup_data corgi_wm8731_setup = {
-	.i2c_bus = 0,
-	.i2c_address = 0x1b,
-};
-
 /* corgi audio subsystem */
 static struct snd_soc_device corgi_snd_devdata = {
 	.card = &snd_soc_corgi,
 	.codec_dev = &soc_codec_dev_wm8731,
-	.codec_data = &corgi_wm8731_setup,
 };
 
+/*
+ * FIXME: This is a temporary bodge to avoid cross-tree merge issues.
+ * New drivers should register the wm8731 I2C device in the machine
+ * setup code (under arch/arm for ARM systems).
+ */
+static int wm8731_i2c_register(void)
+{
+	struct i2c_board_info info;
+	struct i2c_adapter *adapter;
+	struct i2c_client *client;
+
+	memset(&info, 0, sizeof(struct i2c_board_info));
+	info.addr = 0x1b;
+	strlcpy(info.type, "wm8731", I2C_NAME_SIZE);
+
+	adapter = i2c_get_adapter(0);
+	if (!adapter) {
+		printk(KERN_ERR "can't get i2c adapter 0\n");
+		return -ENODEV;
+	}
+
+	client = i2c_new_device(adapter, &info);
+	i2c_put_adapter(adapter);
+	if (!client) {
+		printk(KERN_ERR "can't add i2c device at 0x%x\n",
+			(unsigned int)info.addr);
+		return -ENODEV;
+	}
+
+	return 0;
+}
+
 static struct platform_device *corgi_snd_device;
 
 static int __init corgi_init(void)
@@ -340,6 +362,10 @@ static int __init corgi_init(void)
 	      machine_is_husky()))
 		return -ENODEV;
 
+	ret = wm8731_i2c_register();
+	if (ret != 0)
+		return ret;
+
 	corgi_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!corgi_snd_device)
 		return -ENOMEM;
diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c
new file mode 100644
index 000000000000..7cd2f89d7b10
--- /dev/null
+++ b/sound/soc/pxa/e740_wm9705.c
@@ -0,0 +1,211 @@
+/*
+ * e740-wm9705.c  --  SoC audio for e740
+ *
+ * Copyright 2007 (c) Ian Molton <spyro@f2s.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation; version 2 ONLY.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/audio.h>
+#include <mach/eseries-gpio.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/wm9705.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+
+#define E740_AUDIO_OUT 1
+#define E740_AUDIO_IN  2
+
+static int e740_audio_power;
+
+static void e740_sync_audio_power(int status)
+{
+	gpio_set_value(GPIO_E740_WM9705_nAVDD2, !status);
+	gpio_set_value(GPIO_E740_AMP_ON, (status & E740_AUDIO_OUT) ? 1 : 0);
+	gpio_set_value(GPIO_E740_MIC_ON, (status & E740_AUDIO_IN) ? 1 : 0);
+}
+
+static int e740_mic_amp_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	if (event & SND_SOC_DAPM_PRE_PMU)
+		e740_audio_power |= E740_AUDIO_IN;
+	else if (event & SND_SOC_DAPM_POST_PMD)
+		e740_audio_power &= ~E740_AUDIO_IN;
+
+	e740_sync_audio_power(e740_audio_power);
+
+	return 0;
+}
+
+static int e740_output_amp_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	if (event & SND_SOC_DAPM_PRE_PMU)
+		e740_audio_power |= E740_AUDIO_OUT;
+	else if (event & SND_SOC_DAPM_POST_PMD)
+		e740_audio_power &= ~E740_AUDIO_OUT;
+
+	e740_sync_audio_power(e740_audio_power);
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget e740_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+	SND_SOC_DAPM_PGA_E("Output Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			e740_output_amp_event, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_PGA_E("Mic Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			e740_mic_amp_event, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Output Amp", NULL, "LOUT"},
+	{"Output Amp", NULL, "ROUT"},
+	{"Output Amp", NULL, "MONOOUT"},
+
+	{"Speaker", NULL, "Output Amp"},
+	{"Headphone Jack", NULL, "Output Amp"},
+
+	{"MIC1", NULL, "Mic Amp"},
+	{"Mic Amp", NULL, "Mic (Internal)"},
+};
+
+static int e740_ac97_init(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_nc_pin(codec, "HPOUTL");
+	snd_soc_dapm_nc_pin(codec, "HPOUTR");
+	snd_soc_dapm_nc_pin(codec, "PHONE");
+	snd_soc_dapm_nc_pin(codec, "LINEINL");
+	snd_soc_dapm_nc_pin(codec, "LINEINR");
+	snd_soc_dapm_nc_pin(codec, "CDINL");
+	snd_soc_dapm_nc_pin(codec, "CDINR");
+	snd_soc_dapm_nc_pin(codec, "PCBEEP");
+	snd_soc_dapm_nc_pin(codec, "MIC2");
+
+	snd_soc_dapm_new_controls(codec, e740_dapm_widgets,
+					ARRAY_SIZE(e740_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link e740_dai[] = {
+	{
+		.name = "AC97",
+		.stream_name = "AC97 HiFi",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+		.codec_dai = &wm9705_dai[WM9705_DAI_AC97_HIFI],
+		.init = e740_ac97_init,
+	},
+	{
+		.name = "AC97 Aux",
+		.stream_name = "AC97 Aux",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+		.codec_dai = &wm9705_dai[WM9705_DAI_AC97_AUX],
+	},
+};
+
+static struct snd_soc_card e740 = {
+	.name = "Toshiba e740",
+	.platform = &pxa2xx_soc_platform,
+	.dai_link = e740_dai,
+	.num_links = ARRAY_SIZE(e740_dai),
+};
+
+static struct snd_soc_device e740_snd_devdata = {
+	.card = &e740,
+	.codec_dev = &soc_codec_dev_wm9705,
+};
+
+static struct platform_device *e740_snd_device;
+
+static int __init e740_init(void)
+{
+	int ret;
+
+	if (!machine_is_e740())
+		return -ENODEV;
+
+	ret = gpio_request(GPIO_E740_MIC_ON,  "Mic amp");
+	if (ret)
+		return ret;
+
+	ret = gpio_request(GPIO_E740_AMP_ON, "Output amp");
+	if (ret)
+		goto free_mic_amp_gpio;
+
+	ret = gpio_request(GPIO_E740_WM9705_nAVDD2, "Audio power");
+	if (ret)
+		goto free_op_amp_gpio;
+
+	/* Disable audio */
+	ret = gpio_direction_output(GPIO_E740_MIC_ON, 0);
+	if (ret)
+		goto free_apwr_gpio;
+	ret = gpio_direction_output(GPIO_E740_AMP_ON, 0);
+	if (ret)
+		goto free_apwr_gpio;
+	ret = gpio_direction_output(GPIO_E740_WM9705_nAVDD2, 1);
+	if (ret)
+		goto free_apwr_gpio;
+
+	e740_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!e740_snd_device) {
+		ret = -ENOMEM;
+		goto free_apwr_gpio;
+	}
+
+	platform_set_drvdata(e740_snd_device, &e740_snd_devdata);
+	e740_snd_devdata.dev = &e740_snd_device->dev;
+	ret = platform_device_add(e740_snd_device);
+
+	if (!ret)
+		return 0;
+
+/* Fail gracefully */
+	platform_device_put(e740_snd_device);
+free_apwr_gpio:
+	gpio_free(GPIO_E740_WM9705_nAVDD2);
+free_op_amp_gpio:
+	gpio_free(GPIO_E740_AMP_ON);
+free_mic_amp_gpio:
+	gpio_free(GPIO_E740_MIC_ON);
+
+	return ret;
+}
+
+static void __exit e740_exit(void)
+{
+	platform_device_unregister(e740_snd_device);
+}
+
+module_init(e740_init);
+module_exit(e740_exit);
+
+/* Module information */
+MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
+MODULE_DESCRIPTION("ALSA SoC driver for e740");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c
new file mode 100644
index 000000000000..8dceccc5e059
--- /dev/null
+++ b/sound/soc/pxa/e750_wm9705.c
@@ -0,0 +1,187 @@
+/*
+ * e750-wm9705.c  --  SoC audio for e750
+ *
+ * Copyright 2007 (c) Ian Molton <spyro@f2s.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation; version 2 ONLY.
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/audio.h>
+#include <mach/eseries-gpio.h>
+
+#include <asm/mach-types.h>
+
+#include "../codecs/wm9705.h"
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+
+static int e750_spk_amp_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	if (event & SND_SOC_DAPM_PRE_PMU)
+		gpio_set_value(GPIO_E750_SPK_AMP_OFF, 0);
+	else if (event & SND_SOC_DAPM_POST_PMD)
+		gpio_set_value(GPIO_E750_SPK_AMP_OFF, 1);
+
+	return 0;
+}
+
+static int e750_hp_amp_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	if (event & SND_SOC_DAPM_PRE_PMU)
+		gpio_set_value(GPIO_E750_HP_AMP_OFF, 0);
+	else if (event & SND_SOC_DAPM_POST_PMD)
+		gpio_set_value(GPIO_E750_HP_AMP_OFF, 1);
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget e750_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_MIC("Mic (Internal)", NULL),
+	SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			e750_hp_amp_event, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			e750_spk_amp_event, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headphone Amp", NULL, "HPOUTL"},
+	{"Headphone Amp", NULL, "HPOUTR"},
+	{"Headphone Jack", NULL, "Headphone Amp"},
+
+	{"Speaker Amp", NULL, "MONOOUT"},
+	{"Speaker", NULL, "Speaker Amp"},
+
+	{"MIC1", NULL, "Mic (Internal)"},
+};
+
+static int e750_ac97_init(struct snd_soc_codec *codec)
+{
+	snd_soc_dapm_nc_pin(codec, "LOUT");
+	snd_soc_dapm_nc_pin(codec, "ROUT");
+	snd_soc_dapm_nc_pin(codec, "PHONE");
+	snd_soc_dapm_nc_pin(codec, "LINEINL");
+	snd_soc_dapm_nc_pin(codec, "LINEINR");
+	snd_soc_dapm_nc_pin(codec, "CDINL");
+	snd_soc_dapm_nc_pin(codec, "CDINR");
+	snd_soc_dapm_nc_pin(codec, "PCBEEP");
+	snd_soc_dapm_nc_pin(codec, "MIC2");
+
+	snd_soc_dapm_new_controls(codec, e750_dapm_widgets,
+					ARRAY_SIZE(e750_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link e750_dai[] = {
+	{
+		.name = "AC97",
+		.stream_name = "AC97 HiFi",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+		.codec_dai = &wm9705_dai[WM9705_DAI_AC97_HIFI],
+		.init = e750_ac97_init,
+		/* use ops to check startup state */
+	},
+	{
+		.name = "AC97 Aux",
+		.stream_name = "AC97 Aux",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+		.codec_dai = &wm9705_dai[WM9705_DAI_AC97_AUX],
+	},
+};
+
+static struct snd_soc_card e750 = {
+	.name = "Toshiba e750",
+	.platform = &pxa2xx_soc_platform,
+	.dai_link = e750_dai,
+	.num_links = ARRAY_SIZE(e750_dai),
+};
+
+static struct snd_soc_device e750_snd_devdata = {
+	.card = &e750,
+	.codec_dev = &soc_codec_dev_wm9705,
+};
+
+static struct platform_device *e750_snd_device;
+
+static int __init e750_init(void)
+{
+	int ret;
+
+	if (!machine_is_e750())
+		return -ENODEV;
+
+	ret = gpio_request(GPIO_E750_HP_AMP_OFF,  "Headphone amp");
+	if (ret)
+		return ret;
+
+	ret = gpio_request(GPIO_E750_SPK_AMP_OFF, "Speaker amp");
+	if (ret)
+		goto free_hp_amp_gpio;
+
+	ret = gpio_direction_output(GPIO_E750_HP_AMP_OFF, 1);
+	if (ret)
+		goto free_spk_amp_gpio;
+
+	ret = gpio_direction_output(GPIO_E750_SPK_AMP_OFF, 1);
+	if (ret)
+		goto free_spk_amp_gpio;
+
+	e750_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!e750_snd_device) {
+		ret = -ENOMEM;
+		goto free_spk_amp_gpio;
+	}
+
+	platform_set_drvdata(e750_snd_device, &e750_snd_devdata);
+	e750_snd_devdata.dev = &e750_snd_device->dev;
+	ret = platform_device_add(e750_snd_device);
+
+	if (!ret)
+		return 0;
+
+/* Fail gracefully */
+	platform_device_put(e750_snd_device);
+free_spk_amp_gpio:
+	gpio_free(GPIO_E750_SPK_AMP_OFF);
+free_hp_amp_gpio:
+	gpio_free(GPIO_E750_HP_AMP_OFF);
+
+	return ret;
+}
+
+static void __exit e750_exit(void)
+{
+	platform_device_unregister(e750_snd_device);
+	gpio_free(GPIO_E750_SPK_AMP_OFF);
+	gpio_free(GPIO_E750_HP_AMP_OFF);
+}
+
+module_init(e750_init);
+module_exit(e750_exit);
+
+/* Module information */
+MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
+MODULE_DESCRIPTION("ALSA SoC driver for e750");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c
index 2e3386dfa0f0..bc019cdce429 100644
--- a/sound/soc/pxa/e800_wm9712.c
+++ b/sound/soc/pxa/e800_wm9712.c
@@ -1,8 +1,6 @@
 /*
  * e800-wm9712.c  --  SoC audio for e800
  *
- * Based on tosa.c
- *
  * Copyright 2007 (c) Ian Molton <spyro@f2s.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
@@ -13,7 +11,7 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/device.h>
+#include <linux/gpio.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -21,23 +19,85 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/mach-types.h>
-#include <mach/pxa-regs.h>
-#include <mach/hardware.h>
 #include <mach/audio.h>
+#include <mach/eseries-gpio.h>
 
 #include "../codecs/wm9712.h"
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
 
-static struct snd_soc_card e800;
+static int e800_spk_amp_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	if (event & SND_SOC_DAPM_PRE_PMU)
+		gpio_set_value(GPIO_E800_SPK_AMP_ON, 1);
+	else if (event & SND_SOC_DAPM_POST_PMD)
+		gpio_set_value(GPIO_E800_SPK_AMP_ON, 0);
 
-static struct snd_soc_dai_link e800_dai[] = {
+	return 0;
+}
+
+static int e800_hp_amp_event(struct snd_soc_dapm_widget *w,
+				struct snd_kcontrol *kcontrol, int event)
+{
+	if (event & SND_SOC_DAPM_PRE_PMU)
+		gpio_set_value(GPIO_E800_HP_AMP_OFF, 0);
+	else if (event & SND_SOC_DAPM_POST_PMD)
+		gpio_set_value(GPIO_E800_HP_AMP_OFF, 1);
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget e800_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_MIC("Mic (Internal1)", NULL),
+	SND_SOC_DAPM_MIC("Mic (Internal2)", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+	SND_SOC_DAPM_PGA_E("Headphone Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			e800_hp_amp_event, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+	SND_SOC_DAPM_PGA_E("Speaker Amp", SND_SOC_NOPM, 0, 0, NULL, 0,
+			e800_spk_amp_event, SND_SOC_DAPM_PRE_PMU |
+			SND_SOC_DAPM_POST_PMD),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headphone Jack", NULL, "HPOUTL"},
+	{"Headphone Jack", NULL, "HPOUTR"},
+	{"Headphone Jack", NULL, "Headphone Amp"},
+
+	{"Speaker Amp", NULL, "MONOOUT"},
+	{"Speaker", NULL, "Speaker Amp"},
+
+	{"MIC1", NULL, "Mic (Internal1)"},
+	{"MIC2", NULL, "Mic (Internal2)"},
+};
+
+static int e800_ac97_init(struct snd_soc_codec *codec)
 {
-	.name = "AC97 Aux",
-	.stream_name = "AC97 Aux",
-	.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
-	.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
-},
+	snd_soc_dapm_new_controls(codec, e800_dapm_widgets,
+					ARRAY_SIZE(e800_dapm_widgets));
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link e800_dai[] = {
+	{
+		.name = "AC97",
+		.stream_name = "AC97 HiFi",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+		.codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI],
+		.init = e800_ac97_init,
+	},
+	{
+		.name = "AC97 Aux",
+		.stream_name = "AC97 Aux",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+		.codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX],
+	},
 };
 
 static struct snd_soc_card e800 = {
@@ -61,6 +121,22 @@ static int __init e800_init(void)
 	if (!machine_is_e800())
 		return -ENODEV;
 
+	ret = gpio_request(GPIO_E800_HP_AMP_OFF,  "Headphone amp");
+	if (ret)
+		return ret;
+
+	ret = gpio_request(GPIO_E800_SPK_AMP_ON, "Speaker amp");
+	if (ret)
+		goto free_hp_amp_gpio;
+
+	ret = gpio_direction_output(GPIO_E800_HP_AMP_OFF, 1);
+	if (ret)
+		goto free_spk_amp_gpio;
+
+	ret = gpio_direction_output(GPIO_E800_SPK_AMP_ON, 1);
+	if (ret)
+		goto free_spk_amp_gpio;
+
 	e800_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!e800_snd_device)
 		return -ENOMEM;
@@ -69,8 +145,15 @@ static int __init e800_init(void)
 	e800_snd_devdata.dev = &e800_snd_device->dev;
 	ret = platform_device_add(e800_snd_device);
 
-	if (ret)
-		platform_device_put(e800_snd_device);
+	if (!ret)
+		return 0;
+
+/* Fail gracefully */
+	platform_device_put(e800_snd_device);
+free_spk_amp_gpio:
+	gpio_free(GPIO_E800_SPK_AMP_ON);
+free_hp_amp_gpio:
+	gpio_free(GPIO_E800_HP_AMP_OFF);
 
 	return ret;
 }
@@ -78,6 +161,8 @@ static int __init e800_init(void)
 static void __exit e800_exit(void)
 {
 	platform_device_unregister(e800_snd_device);
+	gpio_free(GPIO_E800_SPK_AMP_ON);
+	gpio_free(GPIO_E800_HP_AMP_OFF);
 }
 
 module_init(e800_init);
@@ -86,4 +171,4 @@ module_exit(e800_exit);
 /* Module information */
 MODULE_AUTHOR("Ian Molton <spyro@f2s.com>");
 MODULE_DESCRIPTION("ALSA SoC driver for e800");
-MODULE_LICENSE("GPL");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c
index fe4a729ea648..949be9c2a01b 100644
--- a/sound/soc/pxa/em-x270.c
+++ b/sound/soc/pxa/em-x270.c
@@ -29,8 +29,6 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/mach-types.h>
-#include <mach/pxa-regs.h>
-#include <mach/hardware.h>
 #include <mach/audio.h>
 
 #include "../codecs/wm9712.h"
diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c
new file mode 100644
index 000000000000..19eda8bbfdaf
--- /dev/null
+++ b/sound/soc/pxa/mioa701_wm9713.c
@@ -0,0 +1,250 @@
+/*
+ * Handles the Mitac mioa701 SoC system
+ *
+ * Copyright (C) 2008 Robert Jarzmik
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation in version 2 of the License.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ *
+ * This is a little schema of the sound interconnections :
+ *
+ *    Sagem X200                 Wolfson WM9713
+ *    +--------+             +-------------------+      Rear Speaker
+ *    |        |             |                   |           /-+
+ *    |        +--->----->---+MONOIN         SPKL+--->----+-+  |
+ *    |  GSM   |             |                   |        | |  |
+ *    |        +--->----->---+PCBEEP         SPKR+--->----+-+  |
+ *    |  CHIP  |             |                   |           \-+
+ *    |        +---<-----<---+MONO               |
+ *    |        |             |                   |      Front Speaker
+ *    +--------+             |                   |           /-+
+ *                           |                HPL+--->----+-+  |
+ *                           |                   |        | |  |
+ *                           |               OUT3+--->----+-+  |
+ *                           |                   |           \-+
+ *                           |                   |
+ *                           |                   |     Front Micro
+ *                           |                   |         +
+ *                           |               MIC1+-----<--+o+
+ *                           |                   |         +
+ *                           +-------------------+        ---
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/platform_device.h>
+
+#include <asm/mach-types.h>
+#include <mach/audio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/ac97_codec.h>
+
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ac97.h"
+#include "../codecs/wm9713.h"
+
+#define ARRAY_AND_SIZE(x)	(x), ARRAY_SIZE(x)
+
+#define AC97_GPIO_PULL		0x58
+
+/* Use GPIO8 for rear speaker amplifier */
+static int rear_amp_power(struct snd_soc_codec *codec, int power)
+{
+	unsigned short reg;
+
+	if (power) {
+		reg = snd_soc_read(codec, AC97_GPIO_CFG);
+		snd_soc_write(codec, AC97_GPIO_CFG, reg | 0x0100);
+		reg = snd_soc_read(codec, AC97_GPIO_PULL);
+		snd_soc_write(codec, AC97_GPIO_PULL, reg | (1<<15));
+	} else {
+		reg = snd_soc_read(codec, AC97_GPIO_CFG);
+		snd_soc_write(codec, AC97_GPIO_CFG, reg & ~0x0100);
+		reg = snd_soc_read(codec, AC97_GPIO_PULL);
+		snd_soc_write(codec, AC97_GPIO_PULL, reg & ~(1<<15));
+	}
+
+	return 0;
+}
+
+static int rear_amp_event(struct snd_soc_dapm_widget *widget,
+			  struct snd_kcontrol *kctl, int event)
+{
+	struct snd_soc_codec *codec = widget->codec;
+
+	return rear_amp_power(codec, SND_SOC_DAPM_EVENT_ON(event));
+}
+
+/* mioa701 machine dapm widgets */
+static const struct snd_soc_dapm_widget mioa701_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Front Speaker", NULL),
+	SND_SOC_DAPM_SPK("Rear Speaker", rear_amp_event),
+	SND_SOC_DAPM_MIC("Headset", NULL),
+	SND_SOC_DAPM_LINE("GSM Line Out", NULL),
+	SND_SOC_DAPM_LINE("GSM Line In", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Front Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Call Mic */
+	{"Mic Bias", NULL, "Front Mic"},
+	{"MIC1", NULL, "Mic Bias"},
+
+	/* Headset Mic */
+	{"LINEL", NULL, "Headset Mic"},
+	{"LINER", NULL, "Headset Mic"},
+
+	/* GSM Module */
+	{"MONOIN", NULL, "GSM Line Out"},
+	{"PCBEEP", NULL, "GSM Line Out"},
+	{"GSM Line In", NULL, "MONO"},
+
+	/* headphone connected to HPL, HPR */
+	{"Headset", NULL, "HPL"},
+	{"Headset", NULL, "HPR"},
+
+	/* front speaker connected to HPL, OUT3 */
+	{"Front Speaker", NULL, "HPL"},
+	{"Front Speaker", NULL, "OUT3"},
+
+	/* rear speaker connected to SPKL, SPKR */
+	{"Rear Speaker", NULL, "SPKL"},
+	{"Rear Speaker", NULL, "SPKR"},
+};
+
+static int mioa701_wm9713_init(struct snd_soc_codec *codec)
+{
+	unsigned short reg;
+
+	/* Add mioa701 specific widgets */
+	snd_soc_dapm_new_controls(codec, ARRAY_AND_SIZE(mioa701_dapm_widgets));
+
+	/* Set up mioa701 specific audio path audio_mapnects */
+	snd_soc_dapm_add_routes(codec, ARRAY_AND_SIZE(audio_map));
+
+	/* Prepare GPIO8 for rear speaker amplifier */
+	reg = codec->read(codec, AC97_GPIO_CFG);
+	codec->write(codec, AC97_GPIO_CFG, reg | 0x0100);
+
+	/* Prepare MIC input */
+	reg = codec->read(codec, AC97_3D_CONTROL);
+	codec->write(codec, AC97_3D_CONTROL, reg | 0xc000);
+
+	snd_soc_dapm_enable_pin(codec, "Front Speaker");
+	snd_soc_dapm_enable_pin(codec, "Rear Speaker");
+	snd_soc_dapm_enable_pin(codec, "Front Mic");
+	snd_soc_dapm_enable_pin(codec, "GSM Line In");
+	snd_soc_dapm_enable_pin(codec, "GSM Line Out");
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static struct snd_soc_ops mioa701_ops;
+
+static struct snd_soc_dai_link mioa701_dai[] = {
+	{
+		.name = "AC97",
+		.stream_name = "AC97 HiFi",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI],
+		.codec_dai = &wm9713_dai[WM9713_DAI_AC97_HIFI],
+		.init = mioa701_wm9713_init,
+		.ops = &mioa701_ops,
+	},
+	{
+		.name = "AC97 Aux",
+		.stream_name = "AC97 Aux",
+		.cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX],
+		.codec_dai = &wm9713_dai[WM9713_DAI_AC97_AUX],
+		.ops = &mioa701_ops,
+	},
+};
+
+static struct snd_soc_card mioa701 = {
+	.name = "MioA701",
+	.platform = &pxa2xx_soc_platform,
+	.dai_link = mioa701_dai,
+	.num_links = ARRAY_SIZE(mioa701_dai),
+};
+
+static struct snd_soc_device mioa701_snd_devdata = {
+	.card = &mioa701,
+	.codec_dev = &soc_codec_dev_wm9713,
+};
+
+static struct platform_device *mioa701_snd_device;
+
+static int mioa701_wm9713_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	if (!machine_is_mioa701())
+		return -ENODEV;
+
+	dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will"
+		 "lead to overheating and possible destruction of your device."
+		 "Do not use without a good knowledge of mio's board design!\n");
+
+	mioa701_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!mioa701_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(mioa701_snd_device, &mioa701_snd_devdata);
+	mioa701_snd_devdata.dev = &mioa701_snd_device->dev;
+
+	ret = platform_device_add(mioa701_snd_device);
+	if (!ret)
+		return 0;
+
+	platform_device_put(mioa701_snd_device);
+	return ret;
+}
+
+static int __devexit mioa701_wm9713_remove(struct platform_device *pdev)
+{
+	platform_device_unregister(mioa701_snd_device);
+	return 0;
+}
+
+static struct platform_driver mioa701_wm9713_driver = {
+	.probe		= mioa701_wm9713_probe,
+	.remove		= __devexit_p(mioa701_wm9713_remove),
+	.driver		= {
+		.name		= "mioa701-wm9713",
+		.owner		= THIS_MODULE,
+	},
+};
+
+static int __init mioa701_asoc_init(void)
+{
+	return platform_driver_register(&mioa701_wm9713_driver);
+}
+
+static void __exit mioa701_asoc_exit(void)
+{
+	platform_driver_unregister(&mioa701_wm9713_driver);
+}
+
+module_init(mioa701_asoc_init);
+module_exit(mioa701_asoc_exit);
+
+/* Module information */
+MODULE_AUTHOR("Robert Jarzmik (rjarzmik@free.fr)");
+MODULE_DESCRIPTION("ALSA SoC WM9713 MIO A701");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 4a9cf3083af0..48a73f64500b 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -55,7 +55,7 @@ static void palm27x_ext_control(struct snd_soc_codec *codec)
 static int palm27x_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->codec;
+	struct snd_soc_codec *codec = rtd->socdev->card->codec;
 
 	/* check the jack status at stream startup */
 	palm27x_ext_control(codec);
@@ -146,19 +146,16 @@ static const struct snd_kcontrol_new palm27x_controls[] = {
 
 static int palm27x_ac97_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
 	snd_soc_dapm_nc_pin(codec, "OUT3");
 	snd_soc_dapm_nc_pin(codec, "MONOOUT");
 
 	/* add palm27x specific controls */
-	for (i = 0; i < ARRAY_SIZE(palm27x_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&palm27x_controls[i],
-						codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, palm27x_controls,
+				ARRAY_SIZE(palm27x_controls));
+	if (err < 0)
+		return err;
 
 	/* add palm27x specific widgets */
 	snd_soc_dapm_new_controls(codec, palm27x_dapm_widgets,
diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c
index 6e9827189fff..a51058f66747 100644
--- a/sound/soc/pxa/poodle.c
+++ b/sound/soc/pxa/poodle.c
@@ -17,6 +17,7 @@
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <linux/timer.h>
+#include <linux/i2c.h>
 #include <linux/interrupt.h>
 #include <linux/platform_device.h>
 #include <sound/core.h>
@@ -26,8 +27,6 @@
 
 #include <asm/mach-types.h>
 #include <asm/hardware/locomo.h>
-#include <mach/pxa-regs.h>
-#include <mach/hardware.h>
 #include <mach/poodle.h>
 #include <mach/audio.h>
 
@@ -77,7 +76,7 @@ static void poodle_ext_control(struct snd_soc_codec *codec)
 static int poodle_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->codec;
+	struct snd_soc_codec *codec = rtd->socdev->card->codec;
 
 	/* check the jack status at stream startup */
 	poodle_ext_control(codec);
@@ -240,19 +239,17 @@ static const struct snd_kcontrol_new wm8731_poodle_controls[] = {
  */
 static int poodle_wm8731_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
 	snd_soc_dapm_nc_pin(codec, "LLINEIN");
 	snd_soc_dapm_nc_pin(codec, "RLINEIN");
 	snd_soc_dapm_enable_pin(codec, "MICIN");
 
 	/* Add poodle specific controls */
-	for (i = 0; i < ARRAY_SIZE(wm8731_poodle_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&wm8731_poodle_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, wm8731_poodle_controls,
+				ARRAY_SIZE(wm8731_poodle_controls));
+	if (err < 0)
+		return err;
 
 	/* Add poodle specific widgets */
 	snd_soc_dapm_new_controls(codec, wm8731_dapm_widgets,
@@ -283,17 +280,42 @@ static struct snd_soc_card snd_soc_poodle = {
 	.num_links = 1,
 };
 
-/* poodle audio private data */
-static struct wm8731_setup_data poodle_wm8731_setup = {
-	.i2c_bus = 0,
-	.i2c_address = 0x1b,
-};
+/*
+ * FIXME: This is a temporary bodge to avoid cross-tree merge issues.
+ * New drivers should register the wm8731 I2C device in the machine
+ * setup code (under arch/arm for ARM systems).
+ */
+static int wm8731_i2c_register(void)
+{
+	struct i2c_board_info info;
+	struct i2c_adapter *adapter;
+	struct i2c_client *client;
+
+	memset(&info, 0, sizeof(struct i2c_board_info));
+	info.addr = 0x1b;
+	strlcpy(info.type, "wm8731", I2C_NAME_SIZE);
+
+	adapter = i2c_get_adapter(0);
+	if (!adapter) {
+		printk(KERN_ERR "can't get i2c adapter 0\n");
+		return -ENODEV;
+	}
+
+	client = i2c_new_device(adapter, &info);
+	i2c_put_adapter(adapter);
+	if (!client) {
+		printk(KERN_ERR "can't add i2c device at 0x%x\n",
+			(unsigned int)info.addr);
+		return -ENODEV;
+	}
+
+	return 0;
+}
 
 /* poodle audio subsystem */
 static struct snd_soc_device poodle_snd_devdata = {
 	.card = &snd_soc_poodle,
 	.codec_dev = &soc_codec_dev_wm8731,
-	.codec_data = &poodle_wm8731_setup,
 };
 
 static struct platform_device *poodle_snd_device;
@@ -305,6 +327,10 @@ static int __init poodle_init(void)
 	if (!machine_is_poodle())
 		return -ENODEV;
 
+	ret = wm8731_i2c_register();
+	if (ret != 0)
+		return ret;
+
 	locomo_gpio_set_dir(&poodle_locomo_device.dev,
 		POODLE_LOCOMO_GPIO_AMP_ON, 0);
 	/* should we mute HP at startup - burning power ?*/
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 73cb6b4c2f2d..7acd3febf8b0 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -1,4 +1,3 @@
-#define DEBUG
 /*
  * pxa-ssp.c  --  ALSA Soc Audio Layer
  *
@@ -21,6 +20,8 @@
 #include <linux/clk.h>
 #include <linux/io.h>
 
+#include <asm/irq.h>
+
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/initval.h>
@@ -29,7 +30,7 @@
 #include <sound/pxa2xx-lib.h>
 
 #include <mach/hardware.h>
-#include <mach/pxa-regs.h>
+#include <mach/dma.h>
 #include <mach/regs-ssp.h>
 #include <mach/audio.h>
 #include <mach/ssp.h>
@@ -221,9 +222,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream,
 	int ret = 0;
 
 	if (!cpu_dai->active) {
-		ret = ssp_init(&priv->dev, cpu_dai->id + 1, SSP_NO_IRQ);
-		if (ret < 0)
-			return ret;
+		priv->dev.port = cpu_dai->id + 1;
+		priv->dev.irq = NO_IRQ;
+		clk_enable(priv->dev.ssp->clk);
 		ssp_disable(&priv->dev);
 	}
 	return ret;
@@ -238,7 +239,7 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream,
 
 	if (!cpu_dai->active) {
 		ssp_disable(&priv->dev);
-		ssp_exit(&priv->dev);
+		clk_disable(priv->dev.ssp->clk);
 	}
 }
 
@@ -298,7 +299,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	int val;
 
 	u32 sscr0 = ssp_read_reg(ssp, SSCR0) &
-		~(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ADC);
+		~(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
 
 	dev_dbg(&ssp->pdev->dev,
 		"pxa_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d\n",
@@ -326,7 +327,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	case PXA_SSP_CLK_AUDIO:
 		priv->sysclk = 0;
 		ssp_set_scr(&priv->dev, 1);
-		sscr0 |= SSCR0_ADC;
+		sscr0 |= SSCR0_ACS;
 		break;
 	default:
 		return -ENODEV;
@@ -520,9 +521,20 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	u32 sscr1;
 	u32 sspsp;
 
+	/* check if we need to change anything at all */
+	if (priv->dai_fmt == fmt)
+		return 0;
+
+	/* we can only change the settings if the port is not in use */
+	if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) {
+		dev_err(&ssp->pdev->dev,
+			"can't change hardware dai format: stream is in use");
+		return -EINVAL;
+	}
+
 	/* reset port settings */
 	sscr0 = ssp_read_reg(ssp, SSCR0) &
-		(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ADC);
+		(SSCR0_ECS |  SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
 	sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7);
 	sspsp = 0;
 
@@ -545,18 +557,18 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
-		sscr0 |= SSCR0_MOD | SSCR0_PSP;
+		sscr0 |= SSCR0_PSP;
 		sscr1 |= SSCR1_RWOT | SSCR1_TRAIL;
 
+		/* See hw_params() */
 		switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
 		case SND_SOC_DAIFMT_NB_NF:
-			sspsp |= SSPSP_FSRT;
+			sspsp |= SSPSP_SFRMP;
 			break;
 		case SND_SOC_DAIFMT_NB_IF:
-			sspsp |= SSPSP_SFRMP | SSPSP_FSRT;
 			break;
 		case SND_SOC_DAIFMT_IB_IF:
-			sspsp |= SSPSP_SFRMP;
+			sspsp |= SSPSP_SCMODE(3);
 			break;
 		default:
 			return -EINVAL;
@@ -642,34 +654,65 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 			sscr0 |= SSCR0_FPCKE;
 #endif
 		sscr0 |= SSCR0_DataSize(16);
-		if (params_channels(params) > 1)
-			sscr0 |= SSCR0_EDSS;
 		break;
 	case SNDRV_PCM_FORMAT_S24_LE:
 		sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(8));
-		/* we must be in network mode (2 slots) for 24 bit stereo */
 		break;
 	case SNDRV_PCM_FORMAT_S32_LE:
 		sscr0 |= (SSCR0_EDSS | SSCR0_DataSize(16));
-		/* we must be in network mode (2 slots) for 32 bit stereo */
 		break;
 	}
 	ssp_write_reg(ssp, SSCR0, sscr0);
 
 	switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
 	case SND_SOC_DAIFMT_I2S:
-		/* Cleared when the DAI format is set */
-		sspsp = ssp_read_reg(ssp, SSPSP) | SSPSP_SFRMWDTH(width);
+	       sspsp = ssp_read_reg(ssp, SSPSP);
+
+		if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) &&
+		     (width == 16)) {
+			/* This is a special case where the bitclk is 64fs
+			* and we're not dealing with 2*32 bits of audio
+			* samples.
+			*
+			* The SSP values used for that are all found out by
+			* trying and failing a lot; some of the registers
+			* needed for that mode are only available on PXA3xx.
+			*/
+
+#ifdef CONFIG_PXA3xx
+			if (!cpu_is_pxa3xx())
+				return -EINVAL;
+
+			sspsp |= SSPSP_SFRMWDTH(width * 2);
+			sspsp |= SSPSP_SFRMDLY(width * 4);
+			sspsp |= SSPSP_EDMYSTOP(3);
+			sspsp |= SSPSP_DMYSTOP(3);
+			sspsp |= SSPSP_DMYSTRT(1);
+#else
+			return -EINVAL;
+#endif
+		} else {
+			/* The frame width is the width the LRCLK is
+			 * asserted for; the delay is expressed in
+			 * half cycle units.  We need the extra cycle
+			 * because the data starts clocking out one BCLK
+			 * after LRCLK changes polarity.
+			 */
+			sspsp |= SSPSP_SFRMWDTH(width + 1);
+			sspsp |= SSPSP_SFRMDLY((width + 1) * 2);
+			sspsp |= SSPSP_DMYSTRT(1);
+		}
+
 		ssp_write_reg(ssp, SSPSP, sspsp);
 		break;
 	default:
 		break;
 	}
 
-	/* We always use a network mode so we always require TDM slots
+	/* When we use a network mode, we always require TDM slots
 	 * - complain loudly and fail if they've not been set up yet.
 	 */
-	if (!(ssp_read_reg(ssp, SSTSA) & 0xf)) {
+	if ((sscr0 & SSCR0_MOD) && !(ssp_read_reg(ssp, SSTSA) & 0xf)) {
 		dev_err(&ssp->pdev->dev, "No TDM timeslot configured\n");
 		return -EINVAL;
 	}
@@ -751,7 +794,7 @@ static int pxa_ssp_probe(struct platform_device *pdev,
 	if (!priv)
 		return -ENOMEM;
 
-	priv->dev.ssp = ssp_request(dai->id, "SoC audio");
+	priv->dev.ssp = ssp_request(dai->id + 1, "SoC audio");
 	if (priv->dev.ssp == NULL) {
 		ret = -ENODEV;
 		goto err_priv;
@@ -782,6 +825,19 @@ static void pxa_ssp_remove(struct platform_device *pdev,
 			    SNDRV_PCM_FMTBIT_S24_LE |	\
 			    SNDRV_PCM_FMTBIT_S32_LE)
 
+static struct snd_soc_dai_ops pxa_ssp_dai_ops = {
+	.startup	= pxa_ssp_startup,
+	.shutdown	= pxa_ssp_shutdown,
+	.trigger	= pxa_ssp_trigger,
+	.hw_params	= pxa_ssp_hw_params,
+	.set_sysclk	= pxa_ssp_set_dai_sysclk,
+	.set_clkdiv	= pxa_ssp_set_dai_clkdiv,
+	.set_pll	= pxa_ssp_set_dai_pll,
+	.set_fmt	= pxa_ssp_set_dai_fmt,
+	.set_tdm_slot	= pxa_ssp_set_dai_tdm_slot,
+	.set_tristate	= pxa_ssp_set_dai_tristate,
+};
+
 struct snd_soc_dai pxa_ssp_dai[] = {
 	{
 		.name = "pxa2xx-ssp1",
@@ -802,18 +858,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
-		.ops = {
-			.startup = pxa_ssp_startup,
-			.shutdown = pxa_ssp_shutdown,
-			.trigger = pxa_ssp_trigger,
-			.hw_params = pxa_ssp_hw_params,
-			.set_sysclk = pxa_ssp_set_dai_sysclk,
-			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
-			.set_pll = pxa_ssp_set_dai_pll,
-			.set_fmt = pxa_ssp_set_dai_fmt,
-			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
-			.set_tristate = pxa_ssp_set_dai_tristate,
-		},
+		.ops = &pxa_ssp_dai_ops,
 	},
 	{	.name = "pxa2xx-ssp2",
 		.id = 1,
@@ -833,18 +878,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
-		.ops = {
-			.startup = pxa_ssp_startup,
-			.shutdown = pxa_ssp_shutdown,
-			.trigger = pxa_ssp_trigger,
-			.hw_params = pxa_ssp_hw_params,
-			.set_sysclk = pxa_ssp_set_dai_sysclk,
-			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
-			.set_pll = pxa_ssp_set_dai_pll,
-			.set_fmt = pxa_ssp_set_dai_fmt,
-			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
-			.set_tristate = pxa_ssp_set_dai_tristate,
-		},
+		.ops = &pxa_ssp_dai_ops,
 	},
 	{
 		.name = "pxa2xx-ssp3",
@@ -865,18 +899,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
-		.ops = {
-			.startup = pxa_ssp_startup,
-			.shutdown = pxa_ssp_shutdown,
-			.trigger = pxa_ssp_trigger,
-			.hw_params = pxa_ssp_hw_params,
-			.set_sysclk = pxa_ssp_set_dai_sysclk,
-			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
-			.set_pll = pxa_ssp_set_dai_pll,
-			.set_fmt = pxa_ssp_set_dai_fmt,
-			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
-			.set_tristate = pxa_ssp_set_dai_tristate,
-		},
+		.ops = &pxa_ssp_dai_ops,
 	},
 	{
 		.name = "pxa2xx-ssp4",
@@ -897,18 +920,7 @@ struct snd_soc_dai pxa_ssp_dai[] = {
 			.rates = PXA_SSP_RATES,
 			.formats = PXA_SSP_FORMATS,
 		 },
-		.ops = {
-			.startup = pxa_ssp_startup,
-			.shutdown = pxa_ssp_shutdown,
-			.trigger = pxa_ssp_trigger,
-			.hw_params = pxa_ssp_hw_params,
-			.set_sysclk = pxa_ssp_set_dai_sysclk,
-			.set_clkdiv = pxa_ssp_set_dai_clkdiv,
-			.set_pll = pxa_ssp_set_dai_pll,
-			.set_fmt = pxa_ssp_set_dai_fmt,
-			.set_tdm_slot = pxa_ssp_set_dai_tdm_slot,
-			.set_tristate = pxa_ssp_set_dai_tristate,
-		},
+		.ops = &pxa_ssp_dai_ops,
 	},
 };
 EXPORT_SYMBOL_GPL(pxa_ssp_dai);
diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c
index 812c2b4d3e07..d9c94d71fa61 100644
--- a/sound/soc/pxa/pxa2xx-ac97.c
+++ b/sound/soc/pxa/pxa2xx-ac97.c
@@ -20,8 +20,8 @@
 #include <sound/pxa2xx-lib.h>
 
 #include <mach/hardware.h>
-#include <mach/pxa-regs.h>
 #include <mach/regs-ac97.h>
+#include <mach/dma.h>
 
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-ac97.h"
@@ -106,13 +106,13 @@ static int pxa2xx_ac97_resume(struct snd_soc_dai *dai)
 static int pxa2xx_ac97_probe(struct platform_device *pdev,
 			     struct snd_soc_dai *dai)
 {
-	return pxa2xx_ac97_hw_probe(pdev);
+	return pxa2xx_ac97_hw_probe(to_platform_device(dai->dev));
 }
 
 static void pxa2xx_ac97_remove(struct platform_device *pdev,
 			       struct snd_soc_dai *dai)
 {
-	pxa2xx_ac97_hw_remove(pdev);
+	pxa2xx_ac97_hw_remove(to_platform_device(dai->dev));
 }
 
 static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream,
@@ -164,6 +164,18 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream,
 		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
 		SNDRV_PCM_RATE_48000)
 
+static struct snd_soc_dai_ops pxa_ac97_hifi_dai_ops = {
+	.hw_params	= pxa2xx_ac97_hw_params,
+};
+
+static struct snd_soc_dai_ops pxa_ac97_aux_dai_ops = {
+	.hw_params	= pxa2xx_ac97_hw_aux_params,
+};
+
+static struct snd_soc_dai_ops pxa_ac97_mic_dai_ops = {
+	.hw_params	= pxa2xx_ac97_hw_mic_params,
+};
+
 /*
  * There is only 1 physical AC97 interface for pxa2xx, but it
  * has extra fifo's that can be used for aux DACs and ADCs.
@@ -189,8 +201,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 		.channels_max = 2,
 		.rates = PXA2XX_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = pxa2xx_ac97_hw_params,},
+	.ops = &pxa_ac97_hifi_dai_ops,
 },
 {
 	.name = "pxa2xx-ac97-aux",
@@ -208,8 +219,7 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 		.channels_max = 1,
 		.rates = PXA2XX_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = pxa2xx_ac97_hw_aux_params,},
+	.ops = &pxa_ac97_aux_dai_ops,
 },
 {
 	.name = "pxa2xx-ac97-mic",
@@ -221,23 +231,52 @@ struct snd_soc_dai pxa_ac97_dai[] = {
 		.channels_max = 1,
 		.rates = PXA2XX_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = pxa2xx_ac97_hw_mic_params,},
+	.ops = &pxa_ac97_mic_dai_ops,
 },
 };
 
 EXPORT_SYMBOL_GPL(pxa_ac97_dai);
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
-static int __init pxa_ac97_init(void)
+static int __devinit pxa2xx_ac97_dev_probe(struct platform_device *pdev)
 {
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(pxa_ac97_dai); i++)
+		pxa_ac97_dai[i].dev = &pdev->dev;
+
+	/* Punt most of the init to the SoC probe; we may need the machine
+	 * driver to do interesting things with the clocking to get us up
+	 * and running.
+	 */
 	return snd_soc_register_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
 }
+
+static int __devexit pxa2xx_ac97_dev_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+
+	return 0;
+}
+
+static struct platform_driver pxa2xx_ac97_driver = {
+	.probe		= pxa2xx_ac97_dev_probe,
+	.remove		= __devexit_p(pxa2xx_ac97_dev_remove),
+	.driver		= {
+		.name	= "pxa2xx-ac97",
+		.owner	= THIS_MODULE,
+	},
+};
+
+static int __init pxa_ac97_init(void)
+{
+	return platform_driver_register(&pxa2xx_ac97_driver);
+}
 module_init(pxa_ac97_init);
 
 static void __exit pxa_ac97_exit(void)
 {
-	snd_soc_unregister_dais(pxa_ac97_dai, ARRAY_SIZE(pxa_ac97_dai));
+	platform_driver_unregister(&pxa2xx_ac97_driver);
 }
 module_exit(pxa_ac97_exit);
 
diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c
index 517991fb1099..2f4b6e489b78 100644
--- a/sound/soc/pxa/pxa2xx-i2s.c
+++ b/sound/soc/pxa/pxa2xx-i2s.c
@@ -24,21 +24,12 @@
 #include <sound/pxa2xx-lib.h>
 
 #include <mach/hardware.h>
-#include <mach/pxa-regs.h>
-#include <mach/pxa2xx-gpio.h>
+#include <mach/dma.h>
 #include <mach/audio.h>
 
 #include "pxa2xx-pcm.h"
 #include "pxa2xx-i2s.h"
 
-struct pxa2xx_gpio {
-	u32 sys;
-	u32	rx;
-	u32 tx;
-	u32 clk;
-	u32 frm;
-};
-
 /*
  * I2S Controller Register and Bit Definitions
  */
@@ -106,21 +97,6 @@ static struct pxa2xx_pcm_dma_params pxa2xx_i2s_pcm_stereo_in = {
 				  DCMD_BURST32 | DCMD_WIDTH4,
 };
 
-static struct pxa2xx_gpio gpio_bus[] = {
-	{ /* I2S SoC Slave */
-		.rx = GPIO29_SDATA_IN_I2S_MD,
-		.tx = GPIO30_SDATA_OUT_I2S_MD,
-		.clk = GPIO28_BITCLK_IN_I2S_MD,
-		.frm = GPIO31_SYNC_I2S_MD,
-	},
-	{ /* I2S SoC Master */
-		.rx = GPIO29_SDATA_IN_I2S_MD,
-		.tx = GPIO30_SDATA_OUT_I2S_MD,
-		.clk = GPIO28_BITCLK_OUT_I2S_MD,
-		.frm = GPIO31_SYNC_I2S_MD,
-	},
-};
-
 static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream,
 			      struct snd_soc_dai *dai)
 {
@@ -181,9 +157,6 @@ static int pxa2xx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	if (clk_id != PXA2XX_I2S_SYSCLK)
 		return -ENODEV;
 
-	if (pxa_i2s.master && dir == SND_SOC_CLOCK_OUT)
-		pxa_gpio_mode(gpio_bus[pxa_i2s.master].sys);
-
 	return 0;
 }
 
@@ -194,10 +167,6 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 
-	pxa_gpio_mode(gpio_bus[pxa_i2s.master].rx);
-	pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx);
-	pxa_gpio_mode(gpio_bus[pxa_i2s.master].frm);
-	pxa_gpio_mode(gpio_bus[pxa_i2s.master].clk);
 	BUG_ON(IS_ERR(clk_i2s));
 	clk_enable(clk_i2s);
 	pxa_i2s_wait();
@@ -335,6 +304,15 @@ static int pxa2xx_i2s_resume(struct snd_soc_dai *dai)
 		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
 		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000)
 
+static struct snd_soc_dai_ops pxa_i2s_dai_ops = {
+	.startup	= pxa2xx_i2s_startup,
+	.shutdown	= pxa2xx_i2s_shutdown,
+	.trigger	= pxa2xx_i2s_trigger,
+	.hw_params	= pxa2xx_i2s_hw_params,
+	.set_fmt	= pxa2xx_i2s_set_dai_fmt,
+	.set_sysclk	= pxa2xx_i2s_set_dai_sysclk,
+};
+
 struct snd_soc_dai pxa_i2s_dai = {
 	.name = "pxa2xx-i2s",
 	.id = 0,
@@ -350,14 +328,7 @@ struct snd_soc_dai pxa_i2s_dai = {
 		.channels_max = 2,
 		.rates = PXA2XX_I2S_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.startup = pxa2xx_i2s_startup,
-		.shutdown = pxa2xx_i2s_shutdown,
-		.trigger = pxa2xx_i2s_trigger,
-		.hw_params = pxa2xx_i2s_hw_params,
-		.set_fmt = pxa2xx_i2s_set_dai_fmt,
-		.set_sysclk = pxa2xx_i2s_set_dai_sysclk,
-	},
+	.ops = &pxa_i2s_dai_ops,
 };
 
 EXPORT_SYMBOL_GPL(pxa_i2s_dai);
@@ -398,11 +369,6 @@ static struct platform_driver pxa2xx_i2s_driver = {
 
 static int __init pxa2xx_i2s_init(void)
 {
-	if (cpu_is_pxa27x())
-		gpio_bus[1].sys = GPIO113_I2S_SYSCLK_MD;
-	else
-		gpio_bus[1].sys = GPIO32_SYSCLK_I2S_MD;
-
 	clk_i2s = ERR_PTR(-ENOENT);
 	return platform_driver_register(&pxa2xx_i2s_driver);
 }
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index a3b9e6bdf979..c4cd2acaacb4 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -26,8 +26,6 @@
 #include <sound/soc-dapm.h>
 
 #include <asm/mach-types.h>
-#include <mach/pxa-regs.h>
-#include <mach/hardware.h>
 #include <mach/spitz.h>
 #include "../codecs/wm8750.h"
 #include "pxa2xx-pcm.h"
@@ -109,7 +107,7 @@ static void spitz_ext_control(struct snd_soc_codec *codec)
 static int spitz_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->codec;
+	struct snd_soc_codec *codec = rtd->socdev->card->codec;
 
 	/* check the jack status at stream startup */
 	spitz_ext_control(codec);
@@ -278,7 +276,7 @@ static const struct snd_kcontrol_new wm8750_spitz_controls[] = {
  */
 static int spitz_wm8750_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
 	/* NC codec pins */
 	snd_soc_dapm_nc_pin(codec, "RINPUT1");
@@ -290,12 +288,10 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
 	snd_soc_dapm_nc_pin(codec, "MONO1");
 
 	/* Add spitz specific controls */
-	for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) {
-		err = snd_ctl_add(codec->card,
-			snd_soc_cnew(&wm8750_spitz_controls[i], codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, wm8750_spitz_controls,
+				ARRAY_SIZE(wm8750_spitz_controls));
+	if (err < 0)
+		return err;
 
 	/* Add spitz specific widgets */
 	snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c
index c77194f74c9b..dbbd3e9d1637 100644
--- a/sound/soc/pxa/tosa.c
+++ b/sound/soc/pxa/tosa.c
@@ -30,8 +30,6 @@
 
 #include <asm/mach-types.h>
 #include <mach/tosa.h>
-#include <mach/pxa-regs.h>
-#include <mach/hardware.h>
 #include <mach/audio.h>
 
 #include "../codecs/wm9712.h"
@@ -82,7 +80,7 @@ static void tosa_ext_control(struct snd_soc_codec *codec)
 static int tosa_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct snd_soc_codec *codec = rtd->socdev->codec;
+	struct snd_soc_codec *codec = rtd->socdev->card->codec;
 
 	/* check the jack status at stream startup */
 	tosa_ext_control(codec);
@@ -188,18 +186,16 @@ static const struct snd_kcontrol_new tosa_controls[] = {
 
 static int tosa_ac97_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
 	snd_soc_dapm_nc_pin(codec, "OUT3");
 	snd_soc_dapm_nc_pin(codec, "MONOOUT");
 
 	/* add tosa specific controls */
-	for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&tosa_controls[i],codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, tosa_controls,
+				ARRAY_SIZE(tosa_controls));
+	if (err < 0)
+		return err;
 
 	/* add tosa specific widgets */
 	snd_soc_dapm_new_controls(codec, tosa_dapm_widgets,
diff --git a/sound/soc/pxa/zylonite.c b/sound/soc/pxa/zylonite.c
index f8e9ecd589d3..9a386b4c4ed1 100644
--- a/sound/soc/pxa/zylonite.c
+++ b/sound/soc/pxa/zylonite.c
@@ -14,6 +14,7 @@
 #include <linux/module.h>
 #include <linux/moduleparam.h>
 #include <linux/device.h>
+#include <linux/clk.h>
 #include <linux/i2c.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -26,6 +27,17 @@
 #include "pxa2xx-ac97.h"
 #include "pxa-ssp.h"
 
+/*
+ * There is a physical switch SW15 on the board which changes the MCLK
+ * for the WM9713 between the standard AC97 master clock and the
+ * output of the CLK_POUT signal from the PXA.
+ */
+static int clk_pout;
+module_param(clk_pout, int, 0);
+MODULE_PARM_DESC(clk_pout, "Use CLK_POUT as WM9713 MCLK (SW15 on board).");
+
+static struct clk *pout;
+
 static struct snd_soc_card zylonite;
 
 static const struct snd_soc_dapm_widget zylonite_dapm_widgets[] = {
@@ -61,10 +73,8 @@ static const struct snd_soc_dapm_route audio_map[] = {
 
 static int zylonite_wm9713_init(struct snd_soc_codec *codec)
 {
-	/* Currently we only support use of the AC97 clock here.  If
-	 * CLK_POUT is selected by SW15 then the clock API will need
-	 * to be used to request and enable it here.
-	 */
+	if (clk_pout)
+		snd_soc_dai_set_pll(&codec->dai[0], 0, clk_get_rate(pout), 0);
 
 	snd_soc_dapm_new_controls(codec, zylonite_dapm_widgets,
 				  ARRAY_SIZE(zylonite_dapm_widgets));
@@ -86,40 +96,35 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	unsigned int pll_out = 0;
-	unsigned int acds = 0;
 	unsigned int wm9713_div = 0;
 	int ret = 0;
+	int rate = params_rate(params);
+	int width = snd_pcm_format_physical_width(params_format(params));
 
-	switch (params_rate(params)) {
+	/* Only support ratios that we can generate neatly from the AC97
+	 * based master clock - in particular, this excludes 44.1kHz.
+	 * In most applications the voice DAC will be used for telephony
+	 * data so multiples of 8kHz will be the common case.
+	 */
+	switch (rate) {
 	case 8000:
 		wm9713_div = 12;
-		pll_out = 2048000;
 		break;
 	case 16000:
 		wm9713_div = 6;
-		pll_out = 4096000;
 		break;
 	case 48000:
-	default:
 		wm9713_div = 2;
-		pll_out = 12288000;
-		acds = 1;
 		break;
+	default:
+		/* Don't support OSS emulation */
+		return -EINVAL;
 	}
 
-	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
-		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-	if (ret < 0)
-		return ret;
-
-	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
-		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
-	if (ret < 0)
-		return ret;
+	/* Add 1 to the width for the leading clock cycle */
+	pll_out = rate * (width + 1) * 8;
 
-	ret = snd_soc_dai_set_tdm_slot(cpu_dai,
-				       params_channels(params),
-				       params_channels(params));
+	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
 	if (ret < 0)
 		return ret;
 
@@ -127,19 +132,22 @@ static int zylonite_voice_hw_params(struct snd_pcm_substream *substream,
 	if (ret < 0)
 		return ret;
 
-	ret = snd_soc_dai_set_clkdiv(cpu_dai, PXA_SSP_AUDIO_DIV_ACDS, acds);
+	if (clk_pout)
+		ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_PLL_DIV,
+					     WM9713_PCMDIV(wm9713_div));
+	else
+		ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV,
+					     WM9713_PCMDIV(wm9713_div));
 	if (ret < 0)
 		return ret;
 
-	ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_AUDIO, 0, 1);
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
 	if (ret < 0)
 		return ret;
 
-	/* Note that if the PLL is in use the WM9713_PCMCLK_PLL_DIV needs
-	 * to be set instead.
-	 */
-	ret = snd_soc_dai_set_clkdiv(codec_dai, WM9713_PCMCLK_DIV,
-				     WM9713_PCMDIV(wm9713_div));
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
 	if (ret < 0)
 		return ret;
 
@@ -173,8 +181,72 @@ static struct snd_soc_dai_link zylonite_dai[] = {
 },
 };
 
+static int zylonite_probe(struct platform_device *pdev)
+{
+	int ret;
+
+	if (clk_pout) {
+		pout = clk_get(NULL, "CLK_POUT");
+		if (IS_ERR(pout)) {
+			dev_err(&pdev->dev, "Unable to obtain CLK_POUT: %ld\n",
+				PTR_ERR(pout));
+			return PTR_ERR(pout);
+		}
+
+		ret = clk_enable(pout);
+		if (ret != 0) {
+			dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n",
+				ret);
+			clk_put(pout);
+			return ret;
+		}
+
+		dev_dbg(&pdev->dev, "MCLK enabled at %luHz\n",
+			clk_get_rate(pout));
+	}
+
+	return 0;
+}
+
+static int zylonite_remove(struct platform_device *pdev)
+{
+	if (clk_pout) {
+		clk_disable(pout);
+		clk_put(pout);
+	}
+
+	return 0;
+}
+
+static int zylonite_suspend_post(struct platform_device *pdev,
+				 pm_message_t state)
+{
+	if (clk_pout)
+		clk_disable(pout);
+
+	return 0;
+}
+
+static int zylonite_resume_pre(struct platform_device *pdev)
+{
+	int ret = 0;
+
+	if (clk_pout) {
+		ret = clk_enable(pout);
+		if (ret != 0)
+			dev_err(&pdev->dev, "Unable to enable CLK_POUT: %d\n",
+				ret);
+	}
+
+	return ret;
+}
+
 static struct snd_soc_card zylonite = {
 	.name = "Zylonite",
+	.probe = &zylonite_probe,
+	.remove = &zylonite_remove,
+	.suspend_post = &zylonite_suspend_post,
+	.resume_pre = &zylonite_resume_pre,
 	.platform = &pxa2xx_soc_platform,
 	.dai_link = zylonite_dai,
 	.num_links = ARRAY_SIZE(zylonite_dai),
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index fcd03acf10f6..2f3a21eee051 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -1,19 +1,31 @@
 config SND_S3C24XX_SOC
-	tristate "SoC Audio for the Samsung S3C24XX chips"
-	depends on ARCH_S3C2410
+	tristate "SoC Audio for the Samsung S3CXXXX chips"
+	depends on ARCH_S3C2410 || ARCH_S3C64XX
 	help
 	  Say Y or M if you want to add support for codecs attached to
-	  the S3C24XX AC97, I2S or SSP interface. You will also need
-	  to select the audio interfaces to support below.
+	  the S3C24XX and S3C64XX AC97, I2S or SSP interface. You will
+	  also need to select the audio interfaces to support below.
 
 config SND_S3C24XX_SOC_I2S
 	tristate
+	select S3C2410_DMA
+
+config SND_S3C_I2SV2_SOC
+	tristate
 
 config SND_S3C2412_SOC_I2S
 	tristate
+	select SND_S3C_I2SV2_SOC
+	select S3C2410_DMA
+
+config SND_S3C64XX_SOC_I2S
+	tristate
+	select SND_S3C_I2SV2_SOC
+	select S3C64XX_DMA
 
 config SND_S3C2443_SOC_AC97
 	tristate
+	select S3C2410_DMA
 	select AC97_BUS
 	select SND_SOC_AC97_BUS
 	
@@ -26,6 +38,14 @@ config SND_S3C24XX_SOC_NEO1973_WM8753
 	  Say Y if you want to add support for SoC audio on smdk2440
 	  with the WM8753.
 
+config SND_S3C24XX_SOC_JIVE_WM8750
+	tristate "SoC I2S Audio support for Jive"
+	depends on SND_S3C24XX_SOC && MACH_JIVE
+	select SND_SOC_WM8750
+	select SND_S3C2412_SOC_I2S
+	help
+	  Sat Y if you want to add support for SoC audio on the Jive.
+
 config SND_S3C24XX_SOC_SMDK2443_WM9710
 	tristate "SoC AC97 Audio support for SMDK2443 - WM9710"
 	depends on SND_S3C24XX_SOC && MACH_SMDK2443
@@ -48,4 +68,5 @@ config SND_S3C24XX_SOC_S3C24XX_UDA134X
 	tristate "SoC I2S Audio support UDA134X wired to a S3C24XX"
        	depends on SND_S3C24XX_SOC
        	select SND_S3C24XX_SOC_I2S
+	select SND_SOC_L3
        	select SND_SOC_UDA134X
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index 96b3f3f617d4..07a93a2ebe5f 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -2,19 +2,25 @@
 snd-soc-s3c24xx-objs := s3c24xx-pcm.o
 snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o
 snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o
+snd-soc-s3c64xx-i2s-objs := s3c64xx-i2s.o
 snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o
+snd-soc-s3c-i2s-v2-objs := s3c-i2s-v2.o
 
 obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o
 obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o
 obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o
 obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o
+obj-$(CONFIG_SND_S3C64XX_SOC_I2S) += snd-soc-s3c64xx-i2s.o
+obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o
 
 # S3C24XX Machine Support
+snd-soc-jive-wm8750-objs := jive_wm8750.o
 snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
 snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
 snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
 snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o
 
+obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o
 obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
 obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
 obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o
diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c
new file mode 100644
index 000000000000..32063790d95b
--- /dev/null
+++ b/sound/soc/s3c24xx/jive_wm8750.c
@@ -0,0 +1,201 @@
+/* sound/soc/s3c24xx/jive_wm8750.c
+ *
+ * Copyright 2007,2008 Simtec Electronics
+ *
+ * Based on sound/soc/pxa/spitz.c
+ *	Copyright 2005 Wolfson Microelectronics PLC.
+ *	Copyright 2005 Openedhand Ltd.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+*/
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/clk.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c2412-i2s.h"
+
+#include "../codecs/wm8750.h"
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{ "Headphone Jack", NULL, "LOUT1" },
+	{ "Headphone Jack", NULL, "ROUT1" },
+	{ "Internal Speaker", NULL, "LOUT2" },
+	{ "Internal Speaker", NULL, "ROUT2" },
+	{ "LINPUT1", NULL, "Line Input" },
+	{ "RINPUT1", NULL, "Line Input" },
+};
+
+static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_SPK("Internal Speaker", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static int jive_hw_params(struct snd_pcm_substream *substream,
+			  struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct s3c_i2sv2_rate_calc div;
+	unsigned int clk = 0;
+	int ret = 0;
+
+	switch (params_rate(params)) {
+	case 8000:
+	case 16000:
+	case 48000:
+	case 96000:
+		clk = 12288000;
+		break;
+	case 11025:
+	case 22050:
+	case 44100:
+		clk = 11289600;
+		break;
+	}
+
+	s3c_i2sv2_calc_rate(&div, NULL, params_rate(params),
+			    s3c2412_get_iisclk());
+
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8750_SYSCLK, clk,
+				     SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_RCLK, div.fs_div);
+	if (ret < 0)
+		return ret;
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C2412_DIV_PRESCALER,
+				     div.clk_div - 1);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops jive_ops = {
+	.hw_params	= jive_hw_params,
+};
+
+static int jive_wm8750_init(struct snd_soc_codec *codec)
+{
+	int err;
+
+	/* These endpoints are not being used. */
+	snd_soc_dapm_nc_pin(codec, "LINPUT2");
+	snd_soc_dapm_nc_pin(codec, "RINPUT2");
+	snd_soc_dapm_nc_pin(codec, "LINPUT3");
+	snd_soc_dapm_nc_pin(codec, "RINPUT3");
+	snd_soc_dapm_nc_pin(codec, "OUT3");
+	snd_soc_dapm_nc_pin(codec, "MONO");
+
+	/* Add jive specific widgets */
+	err = snd_soc_dapm_new_controls(codec, wm8750_dapm_widgets,
+					ARRAY_SIZE(wm8750_dapm_widgets));
+	if (err) {
+		printk(KERN_ERR "%s: failed to add widgets (%d)\n",
+		       __func__, err);
+		return err;
+	}
+
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link jive_dai = {
+	.name		= "wm8750",
+	.stream_name	= "WM8750",
+	.cpu_dai	= &s3c2412_i2s_dai,
+	.codec_dai	= &wm8750_dai,
+	.init		= jive_wm8750_init,
+	.ops		= &jive_ops,
+};
+
+/* jive audio machine driver */
+static struct snd_soc_machine snd_soc_machine_jive = {
+	.name		= "Jive",
+	.dai_link	= &jive_dai,
+	.num_links	= 1,
+};
+
+/* jive audio private data */
+static struct wm8750_setup_data jive_wm8750_setup = {
+};
+
+/* jive audio subsystem */
+static struct snd_soc_device jive_snd_devdata = {
+	.machine	= &snd_soc_machine_jive,
+	.platform	= &s3c24xx_soc_platform,
+	.codec_dev	= &soc_codec_dev_wm8750_spi,
+	.codec_data	= &jive_wm8750_setup,
+};
+
+static struct platform_device *jive_snd_device;
+
+static int __init jive_init(void)
+{
+	int ret;
+
+	if (!machine_is_jive())
+		return 0;
+
+	printk("JIVE WM8750 Audio support\n");
+
+	jive_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!jive_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(jive_snd_device, &jive_snd_devdata);
+	jive_snd_devdata.dev = &jive_snd_device->dev;
+	ret = platform_device_add(jive_snd_device);
+
+	if (ret)
+		platform_device_put(jive_snd_device);
+
+	return ret;
+}
+
+static void __exit jive_exit(void)
+{
+	platform_device_unregister(jive_snd_device);
+}
+
+module_init(jive_init);
+module_exit(jive_exit);
+
+MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("ALSA SoC Jive Audio support");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c
index 45bb12e8ea44..289fadf60b10 100644
--- a/sound/soc/s3c24xx/neo1973_wm8753.c
+++ b/sound/soc/s3c24xx/neo1973_wm8753.c
@@ -29,25 +29,17 @@
 #include <mach/regs-clock.h>
 #include <mach/regs-gpio.h>
 #include <mach/hardware.h>
-#include <mach/audio.h>
+#include <plat/audio.h>
 #include <linux/io.h>
 #include <mach/spi-gpio.h>
 
-#include <asm/plat-s3c24xx/regs-iis.h>
+#include <plat/regs-iis.h>
 
 #include "../codecs/wm8753.h"
 #include "lm4857.h"
 #include "s3c24xx-pcm.h"
 #include "s3c24xx-i2s.h"
 
-/* Debugging stuff */
-#define S3C24XX_SOC_NEO1973_WM8753_DEBUG 0
-#if S3C24XX_SOC_NEO1973_WM8753_DEBUG
-#define DBG(x...) printk(KERN_DEBUG "s3c24xx-soc-neo1973-wm8753: " x)
-#else
-#define DBG(x...)
-#endif
-
 /* define the scenarios */
 #define NEO_AUDIO_OFF			0
 #define NEO_GSM_CALL_AUDIO_HANDSET	1
@@ -72,7 +64,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
 	int ret = 0;
 	unsigned long iis_clkrate;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	iis_clkrate = s3c24xx_i2s_get_clockrate();
 
@@ -158,7 +150,7 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	/* disable the PLL */
 	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0);
@@ -181,7 +173,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream,
 	int ret = 0;
 	unsigned long iis_clkrate;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	iis_clkrate = s3c24xx_i2s_get_clockrate();
 
@@ -224,7 +216,7 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	/* disable the PLL */
 	return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0);
@@ -246,7 +238,7 @@ static int neo1973_get_scenario(struct snd_kcontrol *kcontrol,
 
 static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	switch (neo1973_scenario) {
 	case NEO_AUDIO_OFF:
@@ -330,7 +322,7 @@ static int neo1973_set_scenario(struct snd_kcontrol *kcontrol,
 {
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (neo1973_scenario == ucontrol->value.integer.value[0])
 		return 0;
@@ -344,7 +336,7 @@ static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0};
 
 static void lm4857_write_regs(void)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (i2c_master_send(i2c, lm4857_regs, 4) != 4)
 		printk(KERN_ERR "lm4857: i2c write failed\n");
@@ -357,7 +349,7 @@ static int lm4857_get_reg(struct snd_kcontrol *kcontrol,
 	int shift = (kcontrol->private_value >> 8) & 0x0F;
 	int mask = (kcontrol->private_value >> 16) & 0xFF;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask;
 	return 0;
@@ -385,7 +377,7 @@ static int lm4857_get_mode(struct snd_kcontrol *kcontrol,
 {
 	u8 value = lm4857_regs[LM4857_CTRL] & 0x0F;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (value)
 		value -= 5;
@@ -399,7 +391,7 @@ static int lm4857_set_mode(struct snd_kcontrol *kcontrol,
 {
 	u8 value = ucontrol->value.integer.value[0];
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (value)
 		value += 5;
@@ -506,9 +498,9 @@ static const struct snd_kcontrol_new wm8753_neo1973_controls[] = {
  */
 static int neo1973_wm8753_init(struct snd_soc_codec *codec)
 {
-	int i, err;
+	int err;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	/* set up NC codec pins */
 	snd_soc_dapm_nc_pin(codec, "LOUT2");
@@ -526,13 +518,10 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
 	set_scenario_endpoints(codec, NEO_AUDIO_OFF);
 
 	/* add neo1973 specific controls */
-	for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) {
-		err = snd_ctl_add(codec->card,
-				snd_soc_cnew(&wm8753_neo1973_controls[i],
-				codec, NULL));
-		if (err < 0)
-			return err;
-	}
+	err = snd_soc_add_controls(codec, wm8753_neo1973_controls,
+				ARRAY_SIZE(8753_neo1973_controls));
+	if (err < 0)
+		return err;
 
 	/* set up neo1973 specific audio routes */
 	err = snd_soc_dapm_add_routes(codec, dapm_routes,
@@ -585,21 +574,15 @@ static struct snd_soc_card neo1973 = {
 	.num_links = ARRAY_SIZE(neo1973_dai),
 };
 
-static struct wm8753_setup_data neo1973_wm8753_setup = {
-	.i2c_bus = 0,
-	.i2c_address = 0x1a,
-};
-
 static struct snd_soc_device neo1973_snd_devdata = {
 	.card = &neo1973,
 	.codec_dev = &soc_codec_dev_wm8753,
-	.codec_data = &neo1973_wm8753_setup,
 };
 
 static int lm4857_i2c_probe(struct i2c_client *client,
 			    const struct i2c_device_id *id)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	i2c = client;
 
@@ -609,7 +592,7 @@ static int lm4857_i2c_probe(struct i2c_client *client,
 
 static int lm4857_i2c_remove(struct i2c_client *client)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	i2c = NULL;
 
@@ -620,7 +603,7 @@ static u8 lm4857_state;
 
 static int lm4857_suspend(struct i2c_client *dev, pm_message_t state)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	dev_dbg(&dev->dev, "lm4857_suspend\n");
 	lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf;
@@ -633,7 +616,7 @@ static int lm4857_suspend(struct i2c_client *dev, pm_message_t state)
 
 static int lm4857_resume(struct i2c_client *dev)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (lm4857_state) {
 		lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f);
@@ -644,7 +627,7 @@ static int lm4857_resume(struct i2c_client *dev)
 
 static void lm4857_shutdown(struct i2c_client *dev)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	dev_dbg(&dev->dev, "lm4857_shutdown\n");
 	lm4857_regs[LM4857_CTRL] &= 0xf0;
@@ -675,7 +658,7 @@ static int __init neo1973_init(void)
 {
 	int ret;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (!machine_is_neo1973_gta01()) {
 		printk(KERN_INFO
@@ -706,7 +689,7 @@ static int __init neo1973_init(void)
 
 static void __exit neo1973_exit(void)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	i2c_del_driver(&lm4857_i2c_driver);
 	platform_device_unregister(neo1973_snd_device);
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
new file mode 100644
index 000000000000..295a4c910262
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -0,0 +1,638 @@
+/* sound/soc/s3c24xx/s3c-i2c-v2.c
+ *
+ * ALSA Soc Audio Layer - I2S core for newer Samsung SoCs.
+ *
+ * Copyright (c) 2006 Wolfson Microelectronics PLC.
+ *	Graeme Gregory graeme.gregory@wolfsonmicro.com
+ *	linux@wolfsonmicro.com
+ *
+ * Copyright (c) 2008, 2007, 2004-2005 Simtec Electronics
+ *	http://armlinux.simtec.co.uk/
+ *	Ben Dooks <ben@simtec.co.uk>
+ *
+ * This program is free software; you can redistribute  it and/or modify it
+ * under  the terms of  the GNU General  Public License as published by the
+ * Free Software Foundation;  either version 2 of the  License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/kernel.h>
+#include <linux/io.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <plat/regs-s3c2412-iis.h>
+
+#include <plat/audio.h>
+#include <mach/dma.h>
+
+#include "s3c-i2s-v2.h"
+
+#define S3C2412_I2S_DEBUG_CON 0
+
+static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai)
+{
+	return cpu_dai->private_data;
+}
+
+#define bit_set(v, b) (((v) & (b)) ? 1 : 0)
+
+#if S3C2412_I2S_DEBUG_CON
+static void dbg_showcon(const char *fn, u32 con)
+{
+	printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn,
+	       bit_set(con, S3C2412_IISCON_LRINDEX),
+	       bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY),
+	       bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY),
+	       bit_set(con, S3C2412_IISCON_TXFIFO_FULL),
+	       bit_set(con, S3C2412_IISCON_RXFIFO_FULL));
+
+	printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n",
+	       fn,
+	       bit_set(con, S3C2412_IISCON_TXDMA_PAUSE),
+	       bit_set(con, S3C2412_IISCON_RXDMA_PAUSE),
+	       bit_set(con, S3C2412_IISCON_TXCH_PAUSE),
+	       bit_set(con, S3C2412_IISCON_RXCH_PAUSE));
+	printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn,
+	       bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE),
+	       bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE),
+	       bit_set(con, S3C2412_IISCON_IIS_ACTIVE));
+}
+#else
+static inline void dbg_showcon(const char *fn, u32 con)
+{
+}
+#endif
+
+
+/* Turn on or off the transmission path. */
+void s3c2412_snd_txctrl(struct s3c_i2sv2_info *i2s, int on)
+{
+	void __iomem *regs = i2s->regs;
+	u32 fic, con, mod;
+
+	pr_debug("%s(%d)\n", __func__, on);
+
+	fic = readl(regs + S3C2412_IISFIC);
+	con = readl(regs + S3C2412_IISCON);
+	mod = readl(regs + S3C2412_IISMOD);
+
+	pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
+
+	if (on) {
+		con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE;
+		con &= ~S3C2412_IISCON_TXDMA_PAUSE;
+		con &= ~S3C2412_IISCON_TXCH_PAUSE;
+
+		switch (mod & S3C2412_IISMOD_MODE_MASK) {
+		case S3C2412_IISMOD_MODE_TXONLY:
+		case S3C2412_IISMOD_MODE_TXRX:
+			/* do nothing, we are in the right mode */
+			break;
+
+		case S3C2412_IISMOD_MODE_RXONLY:
+			mod &= ~S3C2412_IISMOD_MODE_MASK;
+			mod |= S3C2412_IISMOD_MODE_TXRX;
+			break;
+
+		default:
+			dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n");
+		}
+
+		writel(con, regs + S3C2412_IISCON);
+		writel(mod, regs + S3C2412_IISMOD);
+	} else {
+		/* Note, we do not have any indication that the FIFO problems
+		 * tha the S3C2410/2440 had apply here, so we should be able
+		 * to disable the DMA and TX without resetting the FIFOS.
+		 */
+
+		con |=  S3C2412_IISCON_TXDMA_PAUSE;
+		con |=  S3C2412_IISCON_TXCH_PAUSE;
+		con &= ~S3C2412_IISCON_TXDMA_ACTIVE;
+
+		switch (mod & S3C2412_IISMOD_MODE_MASK) {
+		case S3C2412_IISMOD_MODE_TXRX:
+			mod &= ~S3C2412_IISMOD_MODE_MASK;
+			mod |= S3C2412_IISMOD_MODE_RXONLY;
+			break;
+
+		case S3C2412_IISMOD_MODE_TXONLY:
+			mod &= ~S3C2412_IISMOD_MODE_MASK;
+			con &= ~S3C2412_IISCON_IIS_ACTIVE;
+			break;
+
+		default:
+			dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n");
+		}
+
+		writel(mod, regs + S3C2412_IISMOD);
+		writel(con, regs + S3C2412_IISCON);
+	}
+
+	fic = readl(regs + S3C2412_IISFIC);
+	dbg_showcon(__func__, con);
+	pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
+}
+EXPORT_SYMBOL_GPL(s3c2412_snd_txctrl);
+
+void s3c2412_snd_rxctrl(struct s3c_i2sv2_info *i2s, int on)
+{
+	void __iomem *regs = i2s->regs;
+	u32 fic, con, mod;
+
+	pr_debug("%s(%d)\n", __func__, on);
+
+	fic = readl(regs + S3C2412_IISFIC);
+	con = readl(regs + S3C2412_IISCON);
+	mod = readl(regs + S3C2412_IISMOD);
+
+	pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
+
+	if (on) {
+		con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE;
+		con &= ~S3C2412_IISCON_RXDMA_PAUSE;
+		con &= ~S3C2412_IISCON_RXCH_PAUSE;
+
+		switch (mod & S3C2412_IISMOD_MODE_MASK) {
+		case S3C2412_IISMOD_MODE_TXRX:
+		case S3C2412_IISMOD_MODE_RXONLY:
+			/* do nothing, we are in the right mode */
+			break;
+
+		case S3C2412_IISMOD_MODE_TXONLY:
+			mod &= ~S3C2412_IISMOD_MODE_MASK;
+			mod |= S3C2412_IISMOD_MODE_TXRX;
+			break;
+
+		default:
+			dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
+		}
+
+		writel(mod, regs + S3C2412_IISMOD);
+		writel(con, regs + S3C2412_IISCON);
+	} else {
+		/* See txctrl notes on FIFOs. */
+
+		con &= ~S3C2412_IISCON_RXDMA_ACTIVE;
+		con |=  S3C2412_IISCON_RXDMA_PAUSE;
+		con |=  S3C2412_IISCON_RXCH_PAUSE;
+
+		switch (mod & S3C2412_IISMOD_MODE_MASK) {
+		case S3C2412_IISMOD_MODE_RXONLY:
+			con &= ~S3C2412_IISCON_IIS_ACTIVE;
+			mod &= ~S3C2412_IISMOD_MODE_MASK;
+			break;
+
+		case S3C2412_IISMOD_MODE_TXRX:
+			mod &= ~S3C2412_IISMOD_MODE_MASK;
+			mod |= S3C2412_IISMOD_MODE_TXONLY;
+			break;
+
+		default:
+			dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
+		}
+
+		writel(con, regs + S3C2412_IISCON);
+		writel(mod, regs + S3C2412_IISMOD);
+	}
+
+	fic = readl(regs + S3C2412_IISFIC);
+	pr_debug("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
+}
+EXPORT_SYMBOL_GPL(s3c2412_snd_rxctrl);
+
+/*
+ * Wait for the LR signal to allow synchronisation to the L/R clock
+ * from the codec. May only be needed for slave mode.
+ */
+static int s3c2412_snd_lrsync(struct s3c_i2sv2_info *i2s)
+{
+	u32 iiscon;
+	unsigned long timeout = jiffies + msecs_to_jiffies(5);
+
+	pr_debug("Entered %s\n", __func__);
+
+	while (1) {
+		iiscon = readl(i2s->regs + S3C2412_IISCON);
+		if (iiscon & S3C2412_IISCON_LRINDEX)
+			break;
+
+		if (timeout < jiffies) {
+			printk(KERN_ERR "%s: timeout\n", __func__);
+			return -ETIMEDOUT;
+		}
+	}
+
+	return 0;
+}
+
+/*
+ * Set S3C2412 I2S DAI format
+ */
+static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
+			       unsigned int fmt)
+{
+	struct s3c_i2sv2_info *i2s = to_info(cpu_dai);
+	u32 iismod;
+
+	pr_debug("Entered %s\n", __func__);
+
+	iismod = readl(i2s->regs + S3C2412_IISMOD);
+	pr_debug("hw_params r: IISMOD: %x \n", iismod);
+
+#if defined(CONFIG_CPU_S3C2412) || defined(CONFIG_CPU_S3C2413)
+#define IISMOD_MASTER_MASK S3C2412_IISMOD_MASTER_MASK
+#define IISMOD_SLAVE S3C2412_IISMOD_SLAVE
+#define IISMOD_MASTER S3C2412_IISMOD_MASTER_INTERNAL
+#endif
+
+#if defined(CONFIG_PLAT_S3C64XX)
+/* From Rev1.1 datasheet, we have two master and two slave modes:
+ * IMS[11:10]:
+ *	00 = master mode, fed from PCLK
+ *	01 = master mode, fed from CLKAUDIO
+ *	10 = slave mode, using PCLK
+ *	11 = slave mode, using I2SCLK
+ */
+#define IISMOD_MASTER_MASK (1 << 11)
+#define IISMOD_SLAVE (1 << 11)
+#define IISMOD_MASTER (0x0)
+#endif
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBM_CFM:
+		i2s->master = 0;
+		iismod &= ~IISMOD_MASTER_MASK;
+		iismod |= IISMOD_SLAVE;
+		break;
+	case SND_SOC_DAIFMT_CBS_CFS:
+		i2s->master = 1;
+		iismod &= ~IISMOD_MASTER_MASK;
+		iismod |= IISMOD_MASTER;
+		break;
+	default:
+		pr_debug("unknwon master/slave format\n");
+		return -EINVAL;
+	}
+
+	iismod &= ~S3C2412_IISMOD_SDF_MASK;
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_RIGHT_J:
+		iismod |= S3C2412_IISMOD_SDF_MSB;
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		iismod |= S3C2412_IISMOD_SDF_LSB;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		iismod |= S3C2412_IISMOD_SDF_IIS;
+		break;
+	default:
+		pr_debug("Unknown data format\n");
+		return -EINVAL;
+	}
+
+	writel(iismod, i2s->regs + S3C2412_IISMOD);
+	pr_debug("hw_params w: IISMOD: %x \n", iismod);
+	return 0;
+}
+
+static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *socdai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai_link *dai = rtd->dai;
+	struct s3c_i2sv2_info *i2s = to_info(dai->cpu_dai);
+	u32 iismod;
+
+	pr_debug("Entered %s\n", __func__);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dai->cpu_dai->dma_data = i2s->dma_playback;
+	else
+		dai->cpu_dai->dma_data = i2s->dma_capture;
+
+	/* Working copies of register */
+	iismod = readl(i2s->regs + S3C2412_IISMOD);
+	pr_debug("%s: r: IISMOD: %x\n", __func__, iismod);
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S8:
+		iismod |= S3C2412_IISMOD_8BIT;
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		iismod &= ~S3C2412_IISMOD_8BIT;
+		break;
+	}
+
+	writel(iismod, i2s->regs + S3C2412_IISMOD);
+	pr_debug("%s: w: IISMOD: %x\n", __func__, iismod);
+	return 0;
+}
+
+static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct s3c_i2sv2_info *i2s = to_info(rtd->dai->cpu_dai);
+	int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
+	unsigned long irqs;
+	int ret = 0;
+
+	pr_debug("Entered %s\n", __func__);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		/* On start, ensure that the FIFOs are cleared and reset. */
+
+		writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH,
+		       i2s->regs + S3C2412_IISFIC);
+
+		/* clear again, just in case */
+		writel(0x0, i2s->regs + S3C2412_IISFIC);
+
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		if (!i2s->master) {
+			ret = s3c2412_snd_lrsync(i2s);
+			if (ret)
+				goto exit_err;
+		}
+
+		local_irq_save(irqs);
+
+		if (capture)
+			s3c2412_snd_rxctrl(i2s, 1);
+		else
+			s3c2412_snd_txctrl(i2s, 1);
+
+		local_irq_restore(irqs);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		local_irq_save(irqs);
+
+		if (capture)
+			s3c2412_snd_rxctrl(i2s, 0);
+		else
+			s3c2412_snd_txctrl(i2s, 0);
+
+		local_irq_restore(irqs);
+		break;
+	default:
+		ret = -EINVAL;
+		break;
+	}
+
+exit_err:
+	return ret;
+}
+
+/*
+ * Set S3C2412 Clock dividers
+ */
+static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
+				  int div_id, int div)
+{
+	struct s3c_i2sv2_info *i2s = to_info(cpu_dai);
+	u32 reg;
+
+	pr_debug("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div);
+
+	switch (div_id) {
+	case S3C_I2SV2_DIV_BCLK:
+		reg = readl(i2s->regs + S3C2412_IISMOD);
+		reg &= ~S3C2412_IISMOD_BCLK_MASK;
+		writel(reg | div, i2s->regs + S3C2412_IISMOD);
+
+		pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD));
+		break;
+
+	case S3C_I2SV2_DIV_RCLK:
+		if (div > 3) {
+			/* convert value to bit field */
+
+			switch (div) {
+			case 256:
+				div = S3C2412_IISMOD_RCLK_256FS;
+				break;
+
+			case 384:
+				div = S3C2412_IISMOD_RCLK_384FS;
+				break;
+
+			case 512:
+				div = S3C2412_IISMOD_RCLK_512FS;
+				break;
+
+			case 768:
+				div = S3C2412_IISMOD_RCLK_768FS;
+				break;
+
+			default:
+				return -EINVAL;
+			}
+		}
+
+		reg = readl(i2s->regs + S3C2412_IISMOD);
+		reg &= ~S3C2412_IISMOD_RCLK_MASK;
+		writel(reg | div, i2s->regs + S3C2412_IISMOD);
+		pr_debug("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD));
+		break;
+
+	case S3C_I2SV2_DIV_PRESCALER:
+		if (div >= 0) {
+			writel((div << 8) | S3C2412_IISPSR_PSREN,
+			       i2s->regs + S3C2412_IISPSR);
+		} else {
+			writel(0x0, i2s->regs + S3C2412_IISPSR);
+		}
+		pr_debug("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR));
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+/* default table of all avaialable root fs divisors */
+static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 };
+
+int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
+			  unsigned int *fstab,
+			  unsigned int rate, struct clk *clk)
+{
+	unsigned long clkrate = clk_get_rate(clk);
+	unsigned int div;
+	unsigned int fsclk;
+	unsigned int actual;
+	unsigned int fs;
+	unsigned int fsdiv;
+	signed int deviation = 0;
+	unsigned int best_fs = 0;
+	unsigned int best_div = 0;
+	unsigned int best_rate = 0;
+	unsigned int best_deviation = INT_MAX;
+
+	if (fstab == NULL)
+		fstab = iis_fs_tab;
+
+	for (fs = 0; fs < ARRAY_SIZE(iis_fs_tab); fs++) {
+		fsdiv = iis_fs_tab[fs];
+
+		fsclk = clkrate / fsdiv;
+		div = fsclk / rate;
+
+		if ((fsclk % rate) > (rate / 2))
+			div++;
+
+		if (div <= 1)
+			continue;
+
+		actual = clkrate / (fsdiv * div);
+		deviation = actual - rate;
+
+		printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n",
+		       fsdiv, div, actual, deviation);
+
+		deviation = abs(deviation);
+
+		if (deviation < best_deviation) {
+			best_fs = fsdiv;
+			best_div = div;
+			best_rate = actual;
+			best_deviation = deviation;
+		}
+
+		if (deviation == 0)
+			break;
+	}
+
+	printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n",
+	       best_fs, best_div, best_rate);
+
+	info->fs_div = best_fs;
+	info->clk_div = best_div;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate);
+
+int s3c_i2sv2_probe(struct platform_device *pdev,
+		    struct snd_soc_dai *dai,
+		    struct s3c_i2sv2_info *i2s,
+		    unsigned long base)
+{
+	struct device *dev = &pdev->dev;
+
+	i2s->dev = dev;
+
+	/* record our i2s structure for later use in the callbacks */
+	dai->private_data = i2s;
+
+	i2s->regs = ioremap(base, 0x100);
+	if (i2s->regs == NULL) {
+		dev_err(dev, "cannot ioremap registers\n");
+		return -ENXIO;
+	}
+
+	i2s->iis_pclk = clk_get(dev, "iis");
+	if (i2s->iis_pclk == NULL) {
+		dev_err(dev, "failed to get iis_clock\n");
+		iounmap(i2s->regs);
+		return -ENOENT;
+	}
+
+	clk_enable(i2s->iis_pclk);
+
+	s3c2412_snd_txctrl(i2s, 0);
+	s3c2412_snd_rxctrl(i2s, 0);
+
+	return 0;
+}
+
+EXPORT_SYMBOL_GPL(s3c_i2sv2_probe);
+
+#ifdef CONFIG_PM
+static int s3c2412_i2s_suspend(struct snd_soc_dai *dai)
+{
+	struct s3c_i2sv2_info *i2s = to_info(dai);
+	u32 iismod;
+
+	if (dai->active) {
+		i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD);
+		i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON);
+		i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR);
+
+		/* some basic suspend checks */
+
+		iismod = readl(i2s->regs + S3C2412_IISMOD);
+
+		if (iismod & S3C2412_IISCON_RXDMA_ACTIVE)
+			pr_warning("%s: RXDMA active?\n", __func__);
+
+		if (iismod & S3C2412_IISCON_TXDMA_ACTIVE)
+			pr_warning("%s: TXDMA active?\n", __func__);
+
+		if (iismod & S3C2412_IISCON_IIS_ACTIVE)
+			pr_warning("%s: IIS active\n", __func__);
+	}
+
+	return 0;
+}
+
+static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
+{
+	struct s3c_i2sv2_info *i2s = to_info(dai);
+
+	pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n",
+		dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
+
+	if (dai->active) {
+		writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON);
+		writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD);
+		writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR);
+
+		writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH,
+		       i2s->regs + S3C2412_IISFIC);
+
+		ndelay(250);
+		writel(0x0, i2s->regs + S3C2412_IISFIC);
+	}
+
+	return 0;
+}
+#else
+#define s3c2412_i2s_suspend NULL
+#define s3c2412_i2s_resume  NULL
+#endif
+
+int s3c_i2sv2_register_dai(struct snd_soc_dai *dai)
+{
+	dai->ops.trigger = s3c2412_i2s_trigger;
+	dai->ops.hw_params = s3c2412_i2s_hw_params;
+	dai->ops.set_fmt = s3c2412_i2s_set_fmt;
+	dai->ops.set_clkdiv = s3c2412_i2s_set_clkdiv;
+
+	dai->suspend = s3c2412_i2s_suspend;
+	dai->resume = s3c2412_i2s_resume;
+
+	return snd_soc_register_dai(dai);
+}
+
+EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai);
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.h b/sound/soc/s3c24xx/s3c-i2s-v2.h
new file mode 100644
index 000000000000..f66854a77fb2
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.h
@@ -0,0 +1,90 @@
+/* sound/soc/s3c24xx/s3c-i2s-v2.h
+ *
+ * ALSA Soc Audio Layer - S3C_I2SV2 I2S driver
+ *
+ * Copyright (c) 2007 Simtec Electronics
+ *	http://armlinux.simtec.co.uk/
+ *	Ben Dooks <ben@simtec.co.uk>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+*/
+
+/* This code is the core support for the I2S block found in a number of
+ * Samsung SoC devices which is unofficially named I2S-V2. Currently the
+ * S3C2412 and the S3C64XX series use this block to provide 1 or 2 I2S
+ * channels via configurable GPIO.
+ */
+
+#ifndef __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H
+#define __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H __FILE__
+
+#define S3C_I2SV2_DIV_BCLK	(1)
+#define S3C_I2SV2_DIV_RCLK	(2)
+#define S3C_I2SV2_DIV_PRESCALER	(3)
+
+/**
+ * struct s3c_i2sv2_info - S3C I2S-V2 information
+ * @dev: The parent device passed to use from the probe.
+ * @regs: The pointer to the device registe block.
+ * @master: True if the I2S core is the I2S bit clock master.
+ * @dma_playback: DMA information for playback channel.
+ * @dma_capture: DMA information for capture channel.
+ * @suspend_iismod: PM save for the IISMOD register.
+ * @suspend_iiscon: PM save for the IISCON register.
+ * @suspend_iispsr: PM save for the IISPSR register.
+ *
+ * This is the private codec state for the hardware associated with an
+ * I2S channel such as the register mappings and clock sources.
+ */
+struct s3c_i2sv2_info {
+	struct device	*dev;
+	void __iomem	*regs;
+
+	struct clk	*iis_pclk;
+	struct clk	*iis_cclk;
+	struct clk	*iis_clk;
+
+	unsigned char	 master;
+
+	struct s3c24xx_pcm_dma_params	*dma_playback;
+	struct s3c24xx_pcm_dma_params	*dma_capture;
+
+	u32		 suspend_iismod;
+	u32		 suspend_iiscon;
+	u32		 suspend_iispsr;
+};
+
+struct s3c_i2sv2_rate_calc {
+	unsigned int	clk_div;	/* for prescaler */
+	unsigned int	fs_div;		/* for root frame clock */
+};
+
+extern int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
+				   unsigned int *fstab,
+				   unsigned int rate, struct clk *clk);
+
+/**
+ * s3c_i2sv2_probe - probe for i2s device helper
+ * @pdev: The platform device supplied to the original probe.
+ * @dai: The ASoC DAI structure supplied to the original probe.
+ * @i2s: Our local i2s structure to fill in.
+ * @base: The base address for the registers.
+ */
+extern int s3c_i2sv2_probe(struct platform_device *pdev,
+			   struct snd_soc_dai *dai,
+			   struct s3c_i2sv2_info *i2s,
+			   unsigned long base);
+
+/**
+ * s3c_i2sv2_register_dai - register dai with soc core
+ * @dai: The snd_soc_dai structure to register
+ *
+ * Fill in any missing fields and then register the given dai with the
+ * soc core.
+ */
+extern int s3c_i2sv2_register_dai(struct snd_soc_dai *dai);
+
+#endif /* __SND_SOC_S3C24XX_S3C_I2SV2_I2S_H */
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index f3fc0aba0aaf..1ca3cdaa8213 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -22,6 +22,7 @@
 #include <linux/delay.h>
 #include <linux/clk.h>
 #include <linux/kernel.h>
+#include <linux/io.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -30,26 +31,16 @@
 #include <sound/soc.h>
 #include <mach/hardware.h>
 
-#include <linux/io.h>
-#include <asm/dma.h>
-
-#include <asm/plat-s3c24xx/regs-s3c2412-iis.h>
+#include <plat/regs-s3c2412-iis.h>
 
-#include <mach/regs-gpio.h>
-#include <mach/audio.h>
+#include <plat/regs-gpio.h>
+#include <plat/audio.h>
 #include <mach/dma.h>
 
 #include "s3c24xx-pcm.h"
 #include "s3c2412-i2s.h"
 
 #define S3C2412_I2S_DEBUG 0
-#define S3C2412_I2S_DEBUG_CON 0
-
-#if S3C2412_I2S_DEBUG
-#define DBG(x...) printk(KERN_INFO x)
-#else
-#define DBG(x...) do { } while (0)
-#endif
 
 static struct s3c2410_dma_client s3c2412_dma_client_out = {
 	.name		= "I2S PCM Stereo out"
@@ -73,431 +64,7 @@ static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_in = {
 	.dma_size	= 4,
 };
 
-struct s3c2412_i2s_info {
-	struct device	*dev;
-	void __iomem	*regs;
-	struct clk	*iis_clk;
-	struct clk	*iis_pclk;
-	struct clk	*iis_cclk;
-
-	u32		 suspend_iismod;
-	u32		 suspend_iiscon;
-	u32		 suspend_iispsr;
-};
-
-static struct s3c2412_i2s_info s3c2412_i2s;
-
-#define bit_set(v, b) (((v) & (b)) ? 1 : 0)
-
-#if S3C2412_I2S_DEBUG_CON
-static void dbg_showcon(const char *fn, u32 con)
-{
-	printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn,
-	       bit_set(con, S3C2412_IISCON_LRINDEX),
-	       bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY),
-	       bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY),
-	       bit_set(con, S3C2412_IISCON_TXFIFO_FULL),
-	       bit_set(con, S3C2412_IISCON_RXFIFO_FULL));
-
-	printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n",
-	       fn,
-	       bit_set(con, S3C2412_IISCON_TXDMA_PAUSE),
-	       bit_set(con, S3C2412_IISCON_RXDMA_PAUSE),
-	       bit_set(con, S3C2412_IISCON_TXCH_PAUSE),
-	       bit_set(con, S3C2412_IISCON_RXCH_PAUSE));
-	printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn,
-	       bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE),
-	       bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE),
-	       bit_set(con, S3C2412_IISCON_IIS_ACTIVE));
-}
-#else
-static inline void dbg_showcon(const char *fn, u32 con)
-{
-}
-#endif
-
-/* Turn on or off the transmission path. */
-static void s3c2412_snd_txctrl(int on)
-{
-	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
-	void __iomem *regs = i2s->regs;
-	u32 fic, con, mod;
-
-	DBG("%s(%d)\n", __func__, on);
-
-	fic = readl(regs + S3C2412_IISFIC);
-	con = readl(regs + S3C2412_IISCON);
-	mod = readl(regs + S3C2412_IISMOD);
-
-	DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
-
-	if (on) {
-		con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE;
-		con &= ~S3C2412_IISCON_TXDMA_PAUSE;
-		con &= ~S3C2412_IISCON_TXCH_PAUSE;
-
-		switch (mod & S3C2412_IISMOD_MODE_MASK) {
-		case S3C2412_IISMOD_MODE_TXONLY:
-		case S3C2412_IISMOD_MODE_TXRX:
-			/* do nothing, we are in the right mode */
-			break;
-
-		case S3C2412_IISMOD_MODE_RXONLY:
-			mod &= ~S3C2412_IISMOD_MODE_MASK;
-			mod |= S3C2412_IISMOD_MODE_TXRX;
-			break;
-
-		default:
-			dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n");
-		}
-
-		writel(con, regs + S3C2412_IISCON);
-		writel(mod, regs + S3C2412_IISMOD);
-	} else {
-		/* Note, we do not have any indication that the FIFO problems
-		 * tha the S3C2410/2440 had apply here, so we should be able
-		 * to disable the DMA and TX without resetting the FIFOS.
-		 */
-
-		con |=  S3C2412_IISCON_TXDMA_PAUSE;
-		con |=  S3C2412_IISCON_TXCH_PAUSE;
-		con &= ~S3C2412_IISCON_TXDMA_ACTIVE;
-
-		switch (mod & S3C2412_IISMOD_MODE_MASK) {
-		case S3C2412_IISMOD_MODE_TXRX:
-			mod &= ~S3C2412_IISMOD_MODE_MASK;
-			mod |= S3C2412_IISMOD_MODE_RXONLY;
-			break;
-
-		case S3C2412_IISMOD_MODE_TXONLY:
-			mod &= ~S3C2412_IISMOD_MODE_MASK;
-			con &= ~S3C2412_IISCON_IIS_ACTIVE;
-			break;
-
-		default:
-			dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n");
-		}
-
-		writel(mod, regs + S3C2412_IISMOD);
-		writel(con, regs + S3C2412_IISCON);
-	}
-
-	fic = readl(regs + S3C2412_IISFIC);
-	dbg_showcon(__func__, con);
-	DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
-}
-
-static void s3c2412_snd_rxctrl(int on)
-{
-	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
-	void __iomem *regs = i2s->regs;
-	u32 fic, con, mod;
-
-	DBG("%s(%d)\n", __func__, on);
-
-	fic = readl(regs + S3C2412_IISFIC);
-	con = readl(regs + S3C2412_IISCON);
-	mod = readl(regs + S3C2412_IISMOD);
-
-	DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
-
-	if (on) {
-		con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE;
-		con &= ~S3C2412_IISCON_RXDMA_PAUSE;
-		con &= ~S3C2412_IISCON_RXCH_PAUSE;
-
-		switch (mod & S3C2412_IISMOD_MODE_MASK) {
-		case S3C2412_IISMOD_MODE_TXRX:
-		case S3C2412_IISMOD_MODE_RXONLY:
-			/* do nothing, we are in the right mode */
-			break;
-
-		case S3C2412_IISMOD_MODE_TXONLY:
-			mod &= ~S3C2412_IISMOD_MODE_MASK;
-			mod |= S3C2412_IISMOD_MODE_TXRX;
-			break;
-
-		default:
-			dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
-		}
-
-		writel(mod, regs + S3C2412_IISMOD);
-		writel(con, regs + S3C2412_IISCON);
-	} else {
-		/* See txctrl notes on FIFOs. */
-
-		con &= ~S3C2412_IISCON_RXDMA_ACTIVE;
-		con |=  S3C2412_IISCON_RXDMA_PAUSE;
-		con |=  S3C2412_IISCON_RXCH_PAUSE;
-
-		switch (mod & S3C2412_IISMOD_MODE_MASK) {
-		case S3C2412_IISMOD_MODE_RXONLY:
-			con &= ~S3C2412_IISCON_IIS_ACTIVE;
-			mod &= ~S3C2412_IISMOD_MODE_MASK;
-			break;
-
-		case S3C2412_IISMOD_MODE_TXRX:
-			mod &= ~S3C2412_IISMOD_MODE_MASK;
-			mod |= S3C2412_IISMOD_MODE_TXONLY;
-			break;
-
-		default:
-			dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n");
-		}
-
-		writel(con, regs + S3C2412_IISCON);
-		writel(mod, regs + S3C2412_IISMOD);
-	}
-
-	fic = readl(regs + S3C2412_IISFIC);
-	DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic);
-}
-
-
-/*
- * Wait for the LR signal to allow synchronisation to the L/R clock
- * from the codec. May only be needed for slave mode.
- */
-static int s3c2412_snd_lrsync(void)
-{
-	u32 iiscon;
-	unsigned long timeout = jiffies + msecs_to_jiffies(5);
-
-	DBG("Entered %s\n", __func__);
-
-	while (1) {
-		iiscon = readl(s3c2412_i2s.regs + S3C2412_IISCON);
-		if (iiscon & S3C2412_IISCON_LRINDEX)
-			break;
-
-		if (timeout < jiffies) {
-			printk(KERN_ERR "%s: timeout\n", __func__);
-			return -ETIMEDOUT;
-		}
-	}
-
-	return 0;
-}
-
-/*
- * Check whether CPU is the master or slave
- */
-static inline int s3c2412_snd_is_clkmaster(void)
-{
-	u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
-
-	DBG("Entered %s\n", __func__);
-
-	iismod &= S3C2412_IISMOD_MASTER_MASK;
-	return !(iismod == S3C2412_IISMOD_SLAVE);
-}
-
-/*
- * Set S3C2412 I2S DAI format
- */
-static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
-			       unsigned int fmt)
-{
-	u32 iismod;
-
-
-	DBG("Entered %s\n", __func__);
-
-	iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
-	DBG("hw_params r: IISMOD: %x \n", iismod);
-
-	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
-	case SND_SOC_DAIFMT_CBM_CFM:
-		iismod &= ~S3C2412_IISMOD_MASTER_MASK;
-		iismod |= S3C2412_IISMOD_SLAVE;
-		break;
-	case SND_SOC_DAIFMT_CBS_CFS:
-		iismod &= ~S3C2412_IISMOD_MASTER_MASK;
-		iismod |= S3C2412_IISMOD_MASTER_INTERNAL;
-		break;
-	default:
-		DBG("unknwon master/slave format\n");
-		return -EINVAL;
-	}
-
-	iismod &= ~S3C2412_IISMOD_SDF_MASK;
-
-	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
-	case SND_SOC_DAIFMT_RIGHT_J:
-		iismod |= S3C2412_IISMOD_SDF_MSB;
-		break;
-	case SND_SOC_DAIFMT_LEFT_J:
-		iismod |= S3C2412_IISMOD_SDF_LSB;
-		break;
-	case SND_SOC_DAIFMT_I2S:
-		iismod |= S3C2412_IISMOD_SDF_IIS;
-		break;
-	default:
-		DBG("Unknown data format\n");
-		return -EINVAL;
-	}
-
-	writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD);
-	DBG("hw_params w: IISMOD: %x \n", iismod);
-	return 0;
-}
-
-static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params,
-				 struct snd_soc_dai *dai)
-{
-	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	u32 iismod;
-
-	DBG("Entered %s\n", __func__);
-
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
-		rtd->dai->cpu_dai->dma_data = &s3c2412_i2s_pcm_stereo_out;
-	else
-		rtd->dai->cpu_dai->dma_data = &s3c2412_i2s_pcm_stereo_in;
-
-	/* Working copies of register */
-	iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
-	DBG("%s: r: IISMOD: %x\n", __func__, iismod);
-
-	switch (params_format(params)) {
-	case SNDRV_PCM_FORMAT_S8:
-		iismod |= S3C2412_IISMOD_8BIT;
-		break;
-	case SNDRV_PCM_FORMAT_S16_LE:
-		iismod &= ~S3C2412_IISMOD_8BIT;
-		break;
-	}
-
-	writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD);
-	DBG("%s: w: IISMOD: %x\n", __func__, iismod);
-	return 0;
-}
-
-static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
-			       struct snd_soc_dai *dai)
-{
-	int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
-	unsigned long irqs;
-	int ret = 0;
-
-	DBG("Entered %s\n", __func__);
-
-	switch (cmd) {
-	case SNDRV_PCM_TRIGGER_START:
-		/* On start, ensure that the FIFOs are cleared and reset. */
-
-		writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH,
-		       s3c2412_i2s.regs + S3C2412_IISFIC);
-
-		/* clear again, just in case */
-		writel(0x0, s3c2412_i2s.regs + S3C2412_IISFIC);
-
-	case SNDRV_PCM_TRIGGER_RESUME:
-	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		if (!s3c2412_snd_is_clkmaster()) {
-			ret = s3c2412_snd_lrsync();
-			if (ret)
-				goto exit_err;
-		}
-
-		local_irq_save(irqs);
-
-		if (capture)
-			s3c2412_snd_rxctrl(1);
-		else
-			s3c2412_snd_txctrl(1);
-
-		local_irq_restore(irqs);
-		break;
-
-	case SNDRV_PCM_TRIGGER_STOP:
-	case SNDRV_PCM_TRIGGER_SUSPEND:
-	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		local_irq_save(irqs);
-
-		if (capture)
-			s3c2412_snd_rxctrl(0);
-		else
-			s3c2412_snd_txctrl(0);
-
-		local_irq_restore(irqs);
-		break;
-	default:
-		ret = -EINVAL;
-		break;
-	}
-
-exit_err:
-	return ret;
-}
-
-/* default table of all avaialable root fs divisors */
-static unsigned int s3c2412_iis_fs[] = { 256, 512, 384, 768, 0 };
-
-int s3c2412_iis_calc_rate(struct s3c2412_rate_calc *info,
-			  unsigned int *fstab,
-			  unsigned int rate, struct clk *clk)
-{
-	unsigned long clkrate = clk_get_rate(clk);
-	unsigned int div;
-	unsigned int fsclk;
-	unsigned int actual;
-	unsigned int fs;
-	unsigned int fsdiv;
-	signed int deviation = 0;
-	unsigned int best_fs = 0;
-	unsigned int best_div = 0;
-	unsigned int best_rate = 0;
-	unsigned int best_deviation = INT_MAX;
-
-
-	if (fstab == NULL)
-		fstab = s3c2412_iis_fs;
-
-	for (fs = 0;; fs++) {
-		fsdiv = s3c2412_iis_fs[fs];
-
-		if (fsdiv == 0)
-			break;
-
-		fsclk = clkrate / fsdiv;
-		div = fsclk / rate;
-
-		if ((fsclk % rate) > (rate / 2))
-			div++;
-
-		if (div <= 1)
-			continue;
-
-		actual = clkrate / (fsdiv * div);
-		deviation = actual - rate;
-
-		printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n",
-		       fsdiv, div, actual, deviation);
-
-		deviation = abs(deviation);
-
-		if (deviation < best_deviation) {
-			best_fs = fsdiv;
-			best_div = div;
-			best_rate = actual;
-			best_deviation = deviation;
-		}
-
-		if (deviation == 0)
-			break;
-	}
-
-	printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n",
-	       best_fs, best_div, best_rate);
-
-	info->fs_div = best_fs;
-	info->clk_div = best_div;
-
-	return 0;
-}
-EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate);
+static struct s3c_i2sv2_info s3c2412_i2s;
 
 /*
  * Set S3C2412 Clock source
@@ -507,15 +74,17 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
 {
 	u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
 
-	DBG("%s(%p, %d, %u, %d)\n", __func__, cpu_dai, clk_id,
+	pr_debug("%s(%p, %d, %u, %d)\n", __func__, cpu_dai, clk_id,
 	    freq, dir);
 
 	switch (clk_id) {
 	case S3C2412_CLKSRC_PCLK:
+		s3c2412_i2s.master = 1;
 		iismod &= ~S3C2412_IISMOD_MASTER_MASK;
 		iismod |= S3C2412_IISMOD_MASTER_INTERNAL;
 		break;
 	case S3C2412_CLKSRC_I2SCLK:
+		s3c2412_i2s.master = 0;
 		iismod &= ~S3C2412_IISMOD_MASTER_MASK;
 		iismod |= S3C2412_IISMOD_MASTER_EXTERNAL;
 		break;
@@ -527,74 +96,6 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
 	return 0;
 }
 
-/*
- * Set S3C2412 Clock dividers
- */
-static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
-				  int div_id, int div)
-{
-	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
-	u32 reg;
-
-	DBG("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div);
-
-	switch (div_id) {
-	case S3C2412_DIV_BCLK:
-		reg = readl(i2s->regs + S3C2412_IISMOD);
-		reg &= ~S3C2412_IISMOD_BCLK_MASK;
-		writel(reg | div, i2s->regs + S3C2412_IISMOD);
-
-		DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD));
-		break;
-
-	case S3C2412_DIV_RCLK:
-		if (div > 3) {
-			/* convert value to bit field */
-
-			switch (div) {
-			case 256:
-				div = S3C2412_IISMOD_RCLK_256FS;
-				break;
-
-			case 384:
-				div = S3C2412_IISMOD_RCLK_384FS;
-				break;
-
-			case 512:
-				div = S3C2412_IISMOD_RCLK_512FS;
-				break;
-
-			case 768:
-				div = S3C2412_IISMOD_RCLK_768FS;
-				break;
-
-			default:
-				return -EINVAL;
-			}
-		}
-
-		reg = readl(s3c2412_i2s.regs + S3C2412_IISMOD);
-		reg &= ~S3C2412_IISMOD_RCLK_MASK;
-		writel(reg | div, i2s->regs + S3C2412_IISMOD);
-		DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD));
-		break;
-
-	case S3C2412_DIV_PRESCALER:
-		if (div >= 0) {
-			writel((div << 8) | S3C2412_IISPSR_PSREN,
-			       i2s->regs + S3C2412_IISPSR);
-		} else {
-			writel(0x0, i2s->regs + S3C2412_IISPSR);
-		}
-		DBG("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR));
-		break;
-
-	default:
-		return -EINVAL;
-	}
-
-	return 0;
-}
 
 struct clk *s3c2412_get_iisclk(void)
 {
@@ -606,34 +107,30 @@ EXPORT_SYMBOL_GPL(s3c2412_get_iisclk);
 static int s3c2412_i2s_probe(struct platform_device *pdev,
 			     struct snd_soc_dai *dai)
 {
-	DBG("Entered %s\n", __func__);
+	int ret;
 
-	s3c2412_i2s.dev = &pdev->dev;
+	pr_debug("Entered %s\n", __func__);
 
-	s3c2412_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100);
-	if (s3c2412_i2s.regs == NULL)
-		return -ENXIO;
+	ret = s3c_i2sv2_probe(pdev, dai, &s3c2412_i2s, S3C2410_PA_IIS);
+	if (ret)
+		return ret;
 
-	s3c2412_i2s.iis_pclk = clk_get(&pdev->dev, "iis");
-	if (s3c2412_i2s.iis_pclk == NULL) {
-		DBG("failed to get iis_clock\n");
-		iounmap(s3c2412_i2s.regs);
-		return -ENODEV;
-	}
+	s3c2412_i2s.dma_capture = &s3c2412_i2s_pcm_stereo_in;
+	s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out;
 
 	s3c2412_i2s.iis_cclk = clk_get(&pdev->dev, "i2sclk");
 	if (s3c2412_i2s.iis_cclk == NULL) {
-		DBG("failed to get i2sclk clock\n");
+		pr_debug("failed to get i2sclk clock\n");
 		iounmap(s3c2412_i2s.regs);
 		return -ENODEV;
 	}
 
-	clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll"));
+	/* Set MPLL as the source for IIS CLK */
 
-	clk_enable(s3c2412_i2s.iis_pclk);
+	clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll"));
 	clk_enable(s3c2412_i2s.iis_cclk);
 
-	s3c2412_i2s.iis_clk = s3c2412_i2s.iis_pclk;
+	s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk;
 
 	/* Configure the I2S pins in correct mode */
 	s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK);
@@ -642,78 +139,22 @@ static int s3c2412_i2s_probe(struct platform_device *pdev,
 	s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI);
 	s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO);
 
-	s3c2412_snd_txctrl(0);
-	s3c2412_snd_rxctrl(0);
-
 	return 0;
 }
 
-#ifdef CONFIG_PM
-static int s3c2412_i2s_suspend(struct snd_soc_dai *dai)
-{
-	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
-	u32 iismod;
-
-	if (dai->active) {
-		i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD);
-		i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON);
-		i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR);
-
-		/* some basic suspend checks */
-
-		iismod = readl(i2s->regs + S3C2412_IISMOD);
-
-		if (iismod & S3C2412_IISCON_RXDMA_ACTIVE)
-			pr_warning("%s: RXDMA active?\n", __func__);
-
-		if (iismod & S3C2412_IISCON_TXDMA_ACTIVE)
-			pr_warning("%s: TXDMA active?\n", __func__);
-
-		if (iismod & S3C2412_IISCON_IIS_ACTIVE)
-			pr_warning("%s: IIS active\n", __func__);
-	}
-
-	return 0;
-}
-
-static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
-{
-	struct s3c2412_i2s_info *i2s = &s3c2412_i2s;
-
-	pr_info("dai_active %d, IISMOD %08x, IISCON %08x\n",
-		dai->active, i2s->suspend_iismod, i2s->suspend_iiscon);
-
-	if (dai->active) {
-		writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON);
-		writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD);
-		writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR);
-
-		writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH,
-		       i2s->regs + S3C2412_IISFIC);
-
-		ndelay(250);
-		writel(0x0, i2s->regs + S3C2412_IISFIC);
-
-	}
-
-	return 0;
-}
-#else
-#define s3c2412_i2s_suspend NULL
-#define s3c2412_i2s_resume  NULL
-#endif /* CONFIG_PM */
-
 #define S3C2412_I2S_RATES \
 	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
 	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
 	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
+static struct snd_soc_dai_ops s3c2412_i2s_dai_ops = {
+	.set_sysclk	= s3c2412_i2s_set_sysclk,
+};
+
 struct snd_soc_dai s3c2412_i2s_dai = {
-	.name	= "s3c2412-i2s",
-	.id	= 0,
-	.probe	= s3c2412_i2s_probe,
-	.suspend = s3c2412_i2s_suspend,
-	.resume = s3c2412_i2s_resume,
+	.name		= "s3c2412-i2s",
+	.id		= 0,
+	.probe		= s3c2412_i2s_probe,
 	.playback = {
 		.channels_min	= 2,
 		.channels_max	= 2,
@@ -726,19 +167,13 @@ struct snd_soc_dai s3c2412_i2s_dai = {
 		.rates		= S3C2412_I2S_RATES,
 		.formats	= SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,
 	},
-	.ops = {
-		.trigger	= s3c2412_i2s_trigger,
-		.hw_params	= s3c2412_i2s_hw_params,
-		.set_fmt	= s3c2412_i2s_set_fmt,
-		.set_clkdiv	= s3c2412_i2s_set_clkdiv,
-		.set_sysclk	= s3c2412_i2s_set_sysclk,
-	},
+	.ops = &s3c2412_i2s_dai_ops,
 };
 EXPORT_SYMBOL_GPL(s3c2412_i2s_dai);
 
 static int __init s3c2412_i2s_init(void)
 {
-	return snd_soc_register_dai(&s3c2412_i2s_dai);
+	return  s3c_i2sv2_register_dai(&s3c2412_i2s_dai);
 }
 module_init(s3c2412_i2s_init);
 
@@ -748,7 +183,6 @@ static void __exit s3c2412_i2s_exit(void)
 }
 module_exit(s3c2412_i2s_exit);
 
-
 /* Module information */
 MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
 MODULE_DESCRIPTION("S3C2412 I2S SoC Interface");
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h
index aac08a25e541..92848e54be16 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.h
+++ b/sound/soc/s3c24xx/s3c2412-i2s.h
@@ -15,9 +15,11 @@
 #ifndef __SND_SOC_S3C24XX_S3C2412_I2S_H
 #define __SND_SOC_S3C24XX_S3C2412_I2S_H __FILE__
 
-#define S3C2412_DIV_BCLK	(1)
-#define S3C2412_DIV_RCLK	(2)
-#define S3C2412_DIV_PRESCALER	(3)
+#include "s3c-i2s-v2.h"
+
+#define S3C2412_DIV_BCLK	S3C_I2SV2_DIV_BCLK
+#define S3C2412_DIV_RCLK	S3C_I2SV2_DIV_RCLK
+#define S3C2412_DIV_PRESCALER	S3C_I2SV2_DIV_PRESCALER
 
 #define S3C2412_CLKSRC_PCLK	(0)
 #define S3C2412_CLKSRC_I2SCLK	(1)
@@ -26,13 +28,4 @@ extern struct clk *s3c2412_get_iisclk(void);
 
 extern struct snd_soc_dai s3c2412_i2s_dai;
 
-struct s3c2412_rate_calc {
-	unsigned int	clk_div;	/* for prescaler */
-	unsigned int	fs_div;		/* for root frame clock */
-};
-
-extern int s3c2412_iis_calc_rate(struct s3c2412_rate_calc *info,
-				 unsigned int *fstab,
-				 unsigned int rate, struct clk *clk);
-
 #endif /* __SND_SOC_S3C24XX_S3C2412_I2S_H */
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 5822d2dd49ba..3698f707c44d 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -31,7 +31,7 @@
 #include <plat/regs-ac97.h>
 #include <mach/regs-gpio.h>
 #include <mach/regs-clock.h>
-#include <mach/audio.h>
+#include <plat/audio.h>
 #include <asm/dma.h>
 #include <mach/dma.h>
 
@@ -355,6 +355,16 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
 		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \
 		SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
 
+static struct snd_soc_dai_ops s3c2443_ac97_dai_ops = {
+	.hw_params	= s3c2443_ac97_hw_params,
+	.trigger	= s3c2443_ac97_trigger,
+};
+
+static struct snd_soc_dai_ops s3c2443_ac97_mic_dai_ops = {
+	.hw_params	= s3c2443_ac97_hw_mic_params,
+	.trigger	= s3c2443_ac97_mic_trigger,
+};
+
 struct snd_soc_dai s3c2443_ac97_dai[] = {
 {
 	.name = "s3c2443-ac97",
@@ -374,9 +384,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = {
 		.channels_max = 2,
 		.rates = s3c2443_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = s3c2443_ac97_hw_params,
-		.trigger = s3c2443_ac97_trigger},
+	.ops = &s3c2443_ac97_dai_ops,
 },
 {
 	.name = "pxa2xx-ac97-mic",
@@ -388,9 +396,7 @@ struct snd_soc_dai s3c2443_ac97_dai[] = {
 		.channels_max = 1,
 		.rates = s3c2443_AC97_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.hw_params = s3c2443_ac97_hw_mic_params,
-		.trigger = s3c2443_ac97_mic_trigger,},
+	.ops = &s3c2443_ac97_mic_dai_ops,
 },
 };
 EXPORT_SYMBOL_GPL(s3c2443_ac97_dai);
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 6f4d439b57aa..cc066964dad6 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -4,7 +4,7 @@
  * (c) 2006 Wolfson Microelectronics PLC.
  * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
  *
- * (c) 2004-2005 Simtec Electronics
+ * Copyright 2004-2005 Simtec Electronics
  *	http://armlinux.simtec.co.uk/
  *	Ben Dooks <ben@simtec.co.uk>
  *
@@ -30,22 +30,15 @@
 #include <mach/hardware.h>
 #include <mach/regs-gpio.h>
 #include <mach/regs-clock.h>
-#include <mach/audio.h>
+#include <plat/audio.h>
 #include <asm/dma.h>
 #include <mach/dma.h>
 
-#include <asm/plat-s3c24xx/regs-iis.h>
+#include <plat/regs-iis.h>
 
 #include "s3c24xx-pcm.h"
 #include "s3c24xx-i2s.h"
 
-#define S3C24XX_I2S_DEBUG 0
-#if S3C24XX_I2S_DEBUG
-#define DBG(x...) printk(KERN_DEBUG "s3c24xx-i2s: " x)
-#else
-#define DBG(x...)
-#endif
-
 static struct s3c2410_dma_client s3c24xx_dma_client_out = {
 	.name = "I2S PCM Stereo out"
 };
@@ -84,13 +77,13 @@ static void s3c24xx_snd_txctrl(int on)
 	u32 iiscon;
 	u32 iismod;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON);
 	iiscon  = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
 	iismod  = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
 
-	DBG("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
+	pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon);
 
 	if (on) {
 		iisfcon |= S3C2410_IISFCON_TXDMA | S3C2410_IISFCON_TXENABLE;
@@ -120,7 +113,7 @@ static void s3c24xx_snd_txctrl(int on)
 		writel(iismod,  s3c24xx_i2s.regs + S3C2410_IISMOD);
 	}
 
-	DBG("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
+	pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon);
 }
 
 static void s3c24xx_snd_rxctrl(int on)
@@ -129,13 +122,13 @@ static void s3c24xx_snd_rxctrl(int on)
 	u32 iiscon;
 	u32 iismod;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	iisfcon = readl(s3c24xx_i2s.regs + S3C2410_IISFCON);
 	iiscon  = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
 	iismod  = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
 
-	DBG("r: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
+	pr_debug("r: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon);
 
 	if (on) {
 		iisfcon |= S3C2410_IISFCON_RXDMA | S3C2410_IISFCON_RXENABLE;
@@ -165,7 +158,7 @@ static void s3c24xx_snd_rxctrl(int on)
 		writel(iismod,  s3c24xx_i2s.regs + S3C2410_IISMOD);
 	}
 
-	DBG("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
+	pr_debug("w: IISCON: %x IISMOD: %x IISFCON: %x\n", iiscon, iismod, iisfcon);
 }
 
 /*
@@ -177,7 +170,7 @@ static int s3c24xx_snd_lrsync(void)
 	u32 iiscon;
 	int timeout = 50; /* 5ms */
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	while (1) {
 		iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
@@ -197,7 +190,7 @@ static int s3c24xx_snd_lrsync(void)
  */
 static inline int s3c24xx_snd_is_clkmaster(void)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1;
 }
@@ -210,10 +203,10 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 {
 	u32 iismod;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
-	DBG("hw_params r: IISMOD: %lx \n", iismod);
+	pr_debug("hw_params r: IISMOD: %x \n", iismod);
 
 	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
 	case SND_SOC_DAIFMT_CBM_CFM:
@@ -238,7 +231,7 @@ static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 	}
 
 	writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
-	DBG("hw_params w: IISMOD: %lx \n", iismod);
+	pr_debug("hw_params w: IISMOD: %x \n", iismod);
 	return 0;
 }
 
@@ -249,7 +242,7 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	u32 iismod;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out;
@@ -258,7 +251,7 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
 
 	/* Working copies of register */
 	iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
-	DBG("hw_params r: IISMOD: %lx\n", iismod);
+	pr_debug("hw_params r: IISMOD: %x\n", iismod);
 
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S8:
@@ -276,7 +269,7 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	}
 
 	writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
-	DBG("hw_params w: IISMOD: %lx\n", iismod);
+	pr_debug("hw_params w: IISMOD: %x\n", iismod);
 	return 0;
 }
 
@@ -285,7 +278,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
 {
 	int ret = 0;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
@@ -327,7 +320,7 @@ static int s3c24xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
 {
 	u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	iismod &= ~S3C2440_IISMOD_MPLL;
 
@@ -353,7 +346,7 @@ static int s3c24xx_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
 {
 	u32 reg;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	switch (div_id) {
 	case S3C24XX_DIV_BCLK:
@@ -389,7 +382,7 @@ EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate);
 static int s3c24xx_i2s_probe(struct platform_device *pdev,
 			     struct snd_soc_dai *dai)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100);
 	if (s3c24xx_i2s.regs == NULL)
@@ -397,7 +390,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev,
 
 	s3c24xx_i2s.iis_clk = clk_get(&pdev->dev, "iis");
 	if (s3c24xx_i2s.iis_clk == NULL) {
-		DBG("failed to get iis_clock\n");
+		pr_err("failed to get iis_clock\n");
 		iounmap(s3c24xx_i2s.regs);
 		return -ENODEV;
 	}
@@ -421,7 +414,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev,
 #ifdef CONFIG_PM
 static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	s3c24xx_i2s.iiscon = readl(s3c24xx_i2s.regs + S3C2410_IISCON);
 	s3c24xx_i2s.iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
@@ -435,7 +428,7 @@ static int s3c24xx_i2s_suspend(struct snd_soc_dai *cpu_dai)
 
 static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai)
 {
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 	clk_enable(s3c24xx_i2s.iis_clk);
 
 	writel(s3c24xx_i2s.iiscon, s3c24xx_i2s.regs + S3C2410_IISCON);
@@ -456,6 +449,14 @@ static int s3c24xx_i2s_resume(struct snd_soc_dai *cpu_dai)
 	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
 	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
+static struct snd_soc_dai_ops s3c24xx_i2s_dai_ops = {
+	.trigger	= s3c24xx_i2s_trigger,
+	.hw_params	= s3c24xx_i2s_hw_params,
+	.set_fmt	= s3c24xx_i2s_set_fmt,
+	.set_clkdiv	= s3c24xx_i2s_set_clkdiv,
+	.set_sysclk	= s3c24xx_i2s_set_sysclk,
+};
+
 struct snd_soc_dai s3c24xx_i2s_dai = {
 	.name = "s3c24xx-i2s",
 	.id = 0,
@@ -472,13 +473,7 @@ struct snd_soc_dai s3c24xx_i2s_dai = {
 		.channels_max = 2,
 		.rates = S3C24XX_I2S_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,},
-	.ops = {
-		.trigger = s3c24xx_i2s_trigger,
-		.hw_params = s3c24xx_i2s_hw_params,
-		.set_fmt = s3c24xx_i2s_set_fmt,
-		.set_clkdiv = s3c24xx_i2s_set_clkdiv,
-		.set_sysclk = s3c24xx_i2s_set_sysclk,
-	},
+	.ops = &s3c24xx_i2s_dai_ops,
 };
 EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai);
 
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index 7c64d31d067e..a9d68fa2b34a 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -4,7 +4,7 @@
  * (c) 2006 Wolfson Microelectronics PLC.
  * Graeme Gregory graeme.gregory@wolfsonmicro.com or linux@wolfsonmicro.com
  *
- * (c) 2004-2005 Simtec Electronics
+ * Copyright 2004-2005 Simtec Electronics
  *	http://armlinux.simtec.co.uk/
  *	Ben Dooks <ben@simtec.co.uk>
  *
@@ -29,17 +29,10 @@
 #include <asm/dma.h>
 #include <mach/hardware.h>
 #include <mach/dma.h>
-#include <mach/audio.h>
+#include <plat/audio.h>
 
 #include "s3c24xx-pcm.h"
 
-#define S3C24XX_PCM_DEBUG 0
-#if S3C24XX_PCM_DEBUG
-#define DBG(x...) printk(KERN_DEBUG "s3c24xx-pcm: " x)
-#else
-#define DBG(x...)
-#endif
-
 static const struct snd_pcm_hardware s3c24xx_pcm_hardware = {
 	.info			= SNDRV_PCM_INFO_INTERLEAVED |
 				    SNDRV_PCM_INFO_BLOCK_TRANSFER |
@@ -84,16 +77,16 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream)
 	dma_addr_t pos = prtd->dma_pos;
 	int ret;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	while (prtd->dma_loaded < prtd->dma_limit) {
 		unsigned long len = prtd->dma_period;
 
-		DBG("dma_loaded: %d\n", prtd->dma_loaded);
+		pr_debug("dma_loaded: %d\n", prtd->dma_loaded);
 
 		if ((pos + len) > prtd->dma_end) {
 			len  = prtd->dma_end - pos;
-			DBG(KERN_DEBUG "%s: corrected dma len %ld\n",
+			pr_debug(KERN_DEBUG "%s: corrected dma len %ld\n",
 			       __func__, len);
 		}
 
@@ -119,7 +112,7 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel,
 	struct snd_pcm_substream *substream = dev_id;
 	struct s3c24xx_runtime_data *prtd;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (result == S3C2410_RES_ABORT || result == S3C2410_RES_ERR)
 		return;
@@ -148,7 +141,7 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream,
 	unsigned long totbytes = params_buffer_bytes(params);
 	int ret = 0;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	/* return if this is a bufferless transfer e.g.
 	 * codec <--> BT codec or GSM modem -- lg FIXME */
@@ -161,14 +154,14 @@ static int s3c24xx_pcm_hw_params(struct snd_pcm_substream *substream,
 		/* prepare DMA */
 		prtd->params = dma;
 
-		DBG("params %p, client %p, channel %d\n", prtd->params,
+		pr_debug("params %p, client %p, channel %d\n", prtd->params,
 			prtd->params->client, prtd->params->channel);
 
 		ret = s3c2410_dma_request(prtd->params->channel,
 					  prtd->params->client, NULL);
 
 		if (ret < 0) {
-			DBG(KERN_ERR "failed to get dma channel\n");
+			printk(KERN_ERR "failed to get dma channel\n");
 			return ret;
 		}
 	}
@@ -196,7 +189,7 @@ static int s3c24xx_pcm_hw_free(struct snd_pcm_substream *substream)
 {
 	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	/* TODO - do we need to ensure DMA flushed */
 	snd_pcm_set_runtime_buffer(substream, NULL);
@@ -214,7 +207,7 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream)
 	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
 	int ret = 0;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	/* return if this is a bufferless transfer e.g.
 	 * codec <--> BT codec or GSM modem -- lg FIXME */
@@ -259,7 +252,7 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
 	int ret = 0;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	spin_lock(&prtd->lock);
 
@@ -297,7 +290,7 @@ s3c24xx_pcm_pointer(struct snd_pcm_substream *substream)
 	unsigned long res;
 	dma_addr_t src, dst;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	spin_lock(&prtd->lock);
 	s3c2410_dma_getposition(prtd->params->channel, &src, &dst);
@@ -309,7 +302,7 @@ s3c24xx_pcm_pointer(struct snd_pcm_substream *substream)
 
 	spin_unlock(&prtd->lock);
 
-	DBG("Pointer %x %x\n", src, dst);
+	pr_debug("Pointer %x %x\n", src, dst);
 
 	/* we seem to be getting the odd error from the pcm library due
 	 * to out-of-bounds pointers. this is maybe due to the dma engine
@@ -330,7 +323,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct s3c24xx_runtime_data *prtd;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware);
 
@@ -349,10 +342,10 @@ static int s3c24xx_pcm_close(struct snd_pcm_substream *substream)
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct s3c24xx_runtime_data *prtd = runtime->private_data;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (!prtd)
-		DBG("s3c24xx_pcm_close called with prtd == NULL\n");
+		pr_debug("s3c24xx_pcm_close called with prtd == NULL\n");
 
 	kfree(prtd);
 
@@ -364,7 +357,7 @@ static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream,
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
 				     runtime->dma_area,
@@ -390,7 +383,7 @@ static int s3c24xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
 	struct snd_dma_buffer *buf = &substream->dma_buffer;
 	size_t size = s3c24xx_pcm_hardware.buffer_bytes_max;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	buf->dev.type = SNDRV_DMA_TYPE_DEV;
 	buf->dev.dev = pcm->card->dev;
@@ -409,7 +402,7 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm)
 	struct snd_dma_buffer *buf;
 	int stream;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	for (stream = 0; stream < 2; stream++) {
 		substream = pcm->streams[stream].substream;
@@ -433,7 +426,7 @@ static int s3c24xx_pcm_new(struct snd_card *card,
 {
 	int ret = 0;
 
-	DBG("Entered %s\n", __func__);
+	pr_debug("Entered %s\n", __func__);
 
 	if (!card->dev->dma_mask)
 		card->dev->dma_mask = &s3c24xx_pcm_dmamask;
diff --git a/sound/soc/s3c24xx/s3c24xx_uda134x.c b/sound/soc/s3c24xx/s3c24xx_uda134x.c
index a0a4d1832a14..8e79a416db57 100644
--- a/sound/soc/s3c24xx/s3c24xx_uda134x.c
+++ b/sound/soc/s3c24xx/s3c24xx_uda134x.c
@@ -22,7 +22,7 @@
 #include <sound/s3c24xx_uda134x.h>
 #include <sound/uda134x.h>
 
-#include <asm/plat-s3c24xx/regs-iis.h>
+#include <plat/regs-iis.h>
 
 #include "s3c24xx-pcm.h"
 #include "s3c24xx-i2s.h"
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.c b/sound/soc/s3c24xx/s3c64xx-i2s.c
new file mode 100644
index 000000000000..33c5de7e255f
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.c
@@ -0,0 +1,222 @@
+/* sound/soc/s3c24xx/s3c64xx-i2s.c
+ *
+ * ALSA SoC Audio Layer - S3C64XX I2S driver
+ *
+ * Copyright 2008 Openmoko, Inc.
+ * Copyright 2008 Simtec Electronics
+ *      Ben Dooks <ben@simtec.co.uk>
+ *      http://armlinux.simtec.co.uk/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/clk.h>
+#include <linux/kernel.h>
+#include <linux/gpio.h>
+#include <linux/io.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+
+#include <plat/regs-s3c2412-iis.h>
+#include <plat/gpio-bank-d.h>
+#include <plat/gpio-bank-e.h>
+#include <plat/gpio-cfg.h>
+#include <plat/audio.h>
+
+#include <mach/map.h>
+#include <mach/dma.h>
+
+#include "s3c24xx-pcm.h"
+#include "s3c64xx-i2s.h"
+
+static struct s3c2410_dma_client s3c64xx_dma_client_out = {
+	.name		= "I2S PCM Stereo out"
+};
+
+static struct s3c2410_dma_client s3c64xx_dma_client_in = {
+	.name		= "I2S PCM Stereo in"
+};
+
+static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_out[2] = {
+	[0] = {
+		.channel	= DMACH_I2S0_OUT,
+		.client		= &s3c64xx_dma_client_out,
+		.dma_addr	= S3C64XX_PA_IIS0 + S3C2412_IISTXD,
+		.dma_size	= 4,
+	},
+	[1] = {
+		.channel	= DMACH_I2S1_OUT,
+		.client		= &s3c64xx_dma_client_out,
+		.dma_addr	= S3C64XX_PA_IIS1 + S3C2412_IISTXD,
+		.dma_size	= 4,
+	},
+};
+
+static struct s3c24xx_pcm_dma_params s3c64xx_i2s_pcm_stereo_in[2] = {
+	[0] = {
+		.channel	= DMACH_I2S0_IN,
+		.client		= &s3c64xx_dma_client_in,
+		.dma_addr	= S3C64XX_PA_IIS0 + S3C2412_IISRXD,
+		.dma_size	= 4,
+	},
+	[1] = {
+		.channel	= DMACH_I2S1_IN,
+		.client		= &s3c64xx_dma_client_in,
+		.dma_addr	= S3C64XX_PA_IIS1 + S3C2412_IISRXD,
+		.dma_size	= 4,
+	},
+};
+
+static struct s3c_i2sv2_info s3c64xx_i2s[2];
+
+static inline struct s3c_i2sv2_info *to_info(struct snd_soc_dai *cpu_dai)
+{
+	return cpu_dai->private_data;
+}
+
+static int s3c64xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai,
+				  int clk_id, unsigned int freq, int dir)
+{
+	struct s3c_i2sv2_info *i2s = to_info(cpu_dai);
+	u32 iismod = readl(i2s->regs + S3C2412_IISMOD);
+
+	switch (clk_id) {
+	case S3C64XX_CLKSRC_PCLK:
+		iismod &= ~S3C64XX_IISMOD_IMS_SYSMUX;
+		break;
+
+	case S3C64XX_CLKSRC_MUX:
+		iismod |= S3C64XX_IISMOD_IMS_SYSMUX;
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	writel(iismod, i2s->regs + S3C2412_IISMOD);
+
+	return 0;
+}
+
+
+unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *dai)
+{
+	struct s3c_i2sv2_info *i2s = to_info(dai);
+
+	return clk_get_rate(i2s->iis_cclk);
+}
+EXPORT_SYMBOL_GPL(s3c64xx_i2s_get_clockrate);
+
+static int s3c64xx_i2s_probe(struct platform_device *pdev,
+			     struct snd_soc_dai *dai)
+{
+	struct device *dev = &pdev->dev;
+	struct s3c_i2sv2_info *i2s;
+	int ret;
+
+	dev_dbg(dev, "%s: probing dai %d\n", __func__, pdev->id);
+
+	if (pdev->id < 0 || pdev->id > ARRAY_SIZE(s3c64xx_i2s)) {
+		dev_err(dev, "id %d out of range\n", pdev->id);
+		return -EINVAL;
+	}
+
+	i2s = &s3c64xx_i2s[pdev->id];
+
+	ret = s3c_i2sv2_probe(pdev, dai, i2s,
+			      pdev->id ? S3C64XX_PA_IIS1 : S3C64XX_PA_IIS0);
+	if (ret)
+		return ret;
+
+	i2s->dma_capture = &s3c64xx_i2s_pcm_stereo_in[pdev->id];
+	i2s->dma_playback = &s3c64xx_i2s_pcm_stereo_out[pdev->id];
+
+	i2s->iis_cclk = clk_get(dev, "audio-bus");
+	if (IS_ERR(i2s->iis_cclk)) {
+		dev_err(dev, "failed to get audio-bus");
+		iounmap(i2s->regs);
+		return -ENODEV;
+	}
+
+	/* configure GPIO for i2s port */
+	switch (pdev->id) {
+	case 0:
+		s3c_gpio_cfgpin(S3C64XX_GPD(0), S3C64XX_GPD0_I2S0_CLK);
+		s3c_gpio_cfgpin(S3C64XX_GPD(1), S3C64XX_GPD1_I2S0_CDCLK);
+		s3c_gpio_cfgpin(S3C64XX_GPD(2), S3C64XX_GPD2_I2S0_LRCLK);
+		s3c_gpio_cfgpin(S3C64XX_GPD(3), S3C64XX_GPD3_I2S0_DI);
+		s3c_gpio_cfgpin(S3C64XX_GPD(4), S3C64XX_GPD4_I2S0_D0);
+		break;
+	case 1:
+		s3c_gpio_cfgpin(S3C64XX_GPE(0), S3C64XX_GPE0_I2S1_CLK);
+		s3c_gpio_cfgpin(S3C64XX_GPE(1), S3C64XX_GPE1_I2S1_CDCLK);
+		s3c_gpio_cfgpin(S3C64XX_GPE(2), S3C64XX_GPE2_I2S1_LRCLK);
+		s3c_gpio_cfgpin(S3C64XX_GPE(3), S3C64XX_GPE3_I2S1_DI);
+		s3c_gpio_cfgpin(S3C64XX_GPE(4), S3C64XX_GPE4_I2S1_D0);
+	}
+
+	return 0;
+}
+
+
+#define S3C64XX_I2S_RATES \
+	(SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
+	SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+	SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+
+#define S3C64XX_I2S_FMTS \
+	(SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE)
+
+static struct snd_soc_dai_ops s3c64xx_i2s_dai_ops = {
+	.set_sysclk	= s3c64xx_i2s_set_sysclk,	
+};
+
+struct snd_soc_dai s3c64xx_i2s_dai = {
+	.name		= "s3c64xx-i2s",
+	.id		= 0,
+	.probe		= s3c64xx_i2s_probe,
+	.playback = {
+		.channels_min	= 2,
+		.channels_max	= 2,
+		.rates		= S3C64XX_I2S_RATES,
+		.formats	= S3C64XX_I2S_FMTS,
+	},
+	.capture = {
+		.channels_min	= 2,
+		.channels_max	= 2,
+		.rates		= S3C64XX_I2S_RATES,
+		.formats	= S3C64XX_I2S_FMTS,
+	},
+	.ops = &s3c64xx_i2s_dai_ops,
+};
+EXPORT_SYMBOL_GPL(s3c64xx_i2s_dai);
+
+static int __init s3c64xx_i2s_init(void)
+{
+	return  s3c_i2sv2_register_dai(&s3c64xx_i2s_dai);
+}
+module_init(s3c64xx_i2s_init);
+
+static void __exit s3c64xx_i2s_exit(void)
+{
+	snd_soc_unregister_dai(&s3c64xx_i2s_dai);
+}
+module_exit(s3c64xx_i2s_exit);
+
+/* Module information */
+MODULE_AUTHOR("Ben Dooks, <ben@simtec.co.uk>");
+MODULE_DESCRIPTION("S3C64XX I2S SoC Interface");
+MODULE_LICENSE("GPL");
+
+
+
diff --git a/sound/soc/s3c24xx/s3c64xx-i2s.h b/sound/soc/s3c24xx/s3c64xx-i2s.h
new file mode 100644
index 000000000000..b7ffe3c38b66
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c64xx-i2s.h
@@ -0,0 +1,31 @@
+/* sound/soc/s3c24xx/s3c64xx-i2s.h
+ *
+ * ALSA SoC Audio Layer - S3C64XX I2S driver
+ *
+ * Copyright 2008 Openmoko, Inc.
+ * Copyright 2008 Simtec Electronics
+ *      Ben Dooks <ben@simtec.co.uk>
+ *      http://armlinux.simtec.co.uk/
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef __SND_SOC_S3C24XX_S3C64XX_I2S_H
+#define __SND_SOC_S3C24XX_S3C64XX_I2S_H __FILE__
+
+#include "s3c-i2s-v2.h"
+
+#define S3C64XX_DIV_BCLK	S3C_I2SV2_DIV_BCLK
+#define S3C64XX_DIV_RCLK	S3C_I2SV2_DIV_RCLK
+#define S3C64XX_DIV_PRESCALER	S3C_I2SV2_DIV_PRESCALER
+
+#define S3C64XX_CLKSRC_PCLK	(0)
+#define S3C64XX_CLKSRC_MUX	(1)
+
+extern struct snd_soc_dai s3c64xx_i2s_dai;
+
+extern unsigned long s3c64xx_i2s_get_clockrate(struct snd_soc_dai *cpu_dai);
+
+#endif /* __SND_SOC_S3C24XX_S3C64XX_I2S_H */
diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c
index eab31838badf..41db75af3c69 100644
--- a/sound/soc/sh/hac.c
+++ b/sound/soc/sh/hac.c
@@ -267,6 +267,10 @@ static int hac_hw_params(struct snd_pcm_substream *substream,
 #define AC97_FMTS	\
 	SNDRV_PCM_FMTBIT_S16_LE
 
+static struct snd_soc_dai_ops hac_dai_ops = {
+	.hw_params	= hac_hw_params,
+};
+
 struct snd_soc_dai sh4_hac_dai[] = {
 {
 	.name			= "HAC0",
@@ -284,9 +288,7 @@ struct snd_soc_dai sh4_hac_dai[] = {
 		.channels_min	= 2,
 		.channels_max	= 2,
 	},
-	.ops = {
-		.hw_params	= hac_hw_params,
-	},
+	.ops = &hac_dai_ops,
 },
 #ifdef CONFIG_CPU_SUBTYPE_SH7760
 {
@@ -305,9 +307,7 @@ struct snd_soc_dai sh4_hac_dai[] = {
 		.channels_min	= 2,
 		.channels_max	= 2,
 	},
-	.ops = {
-		.hw_params	= hac_hw_params,
-	},
+	.ops = &hac_dai_ops,
 
 },
 #endif
diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c
index d1e5390fddeb..56fa0872abbb 100644
--- a/sound/soc/sh/ssi.c
+++ b/sound/soc/sh/ssi.c
@@ -336,6 +336,16 @@ static int ssi_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 	 SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE |	\
 	 SNDRV_PCM_FMTBIT_S32_LE  | SNDRV_PCM_FMTBIT_U32_LE)
 
+static struct snd_soc_dai_ops ssi_dai_ops = {
+	.startup	= ssi_startup,
+	.shutdown	= ssi_shutdown,
+	.trigger	= ssi_trigger,
+	.hw_params	= ssi_hw_params,
+	.set_sysclk	= ssi_set_sysclk,
+	.set_clkdiv	= ssi_set_clkdiv,
+	.set_fmt	= ssi_set_fmt,
+};
+
 struct snd_soc_dai sh4_ssi_dai[] = {
 {
 	.name			= "SSI0",
@@ -352,15 +362,7 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 		.channels_min	= 2,
 		.channels_max	= 8,
 	},
-	.ops = {
-		.startup	= ssi_startup,
-		.shutdown	= ssi_shutdown,
-		.trigger	= ssi_trigger,
-		.hw_params	= ssi_hw_params,
-		.set_sysclk	= ssi_set_sysclk,
-		.set_clkdiv	= ssi_set_clkdiv,
-		.set_fmt	= ssi_set_fmt,
-	},
+	.ops = &ssi_dai_ops,
 },
 #ifdef CONFIG_CPU_SUBTYPE_SH7760
 {
@@ -378,15 +380,7 @@ struct snd_soc_dai sh4_ssi_dai[] = {
 		.channels_min	= 2,
 		.channels_max	= 8,
 	},
-	.ops = {
-		.startup	= ssi_startup,
-		.shutdown	= ssi_shutdown,
-		.trigger	= ssi_trigger,
-		.hw_params	= ssi_hw_params,
-		.set_sysclk	= ssi_set_sysclk,
-		.set_clkdiv	= ssi_set_clkdiv,
-		.set_fmt	= ssi_set_fmt,
-	},
+	.ops = &ssi_dai_ops,
 },
 #endif
 };
diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c
index ec3f8bb4b51d..6e710f705a74 100644
--- a/sound/soc/soc-core.c
+++ b/sound/soc/soc-core.c
@@ -133,8 +133,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 	mutex_lock(&pcm_mutex);
 
 	/* startup the audio subsystem */
-	if (cpu_dai->ops.startup) {
-		ret = cpu_dai->ops.startup(substream, cpu_dai);
+	if (cpu_dai->ops->startup) {
+		ret = cpu_dai->ops->startup(substream, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't open interface %s\n",
 				cpu_dai->name);
@@ -150,8 +150,8 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 		}
 	}
 
-	if (codec_dai->ops.startup) {
-		ret = codec_dai->ops.startup(substream, codec_dai);
+	if (codec_dai->ops->startup) {
+		ret = codec_dai->ops->startup(substream, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't open codec %s\n",
 				codec_dai->name);
@@ -234,7 +234,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 		cpu_dai->capture.active = codec_dai->capture.active = 1;
 	cpu_dai->active = codec_dai->active = 1;
 	cpu_dai->runtime = runtime;
-	socdev->codec->active++;
+	card->codec->active++;
 	mutex_unlock(&pcm_mutex);
 	return 0;
 
@@ -247,8 +247,8 @@ codec_dai_err:
 		platform->pcm_ops->close(substream);
 
 platform_err:
-	if (cpu_dai->ops.shutdown)
-		cpu_dai->ops.shutdown(substream, cpu_dai);
+	if (cpu_dai->ops->shutdown)
+		cpu_dai->ops->shutdown(substream, cpu_dai);
 out:
 	mutex_unlock(&pcm_mutex);
 	return ret;
@@ -264,7 +264,7 @@ static void close_delayed_work(struct work_struct *work)
 	struct snd_soc_card *card = container_of(work, struct snd_soc_card,
 						 delayed_work.work);
 	struct snd_soc_device *socdev = card->socdev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = card->codec;
 	struct snd_soc_dai *codec_dai;
 	int i;
 
@@ -319,7 +319,7 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = card->codec;
 
 	mutex_lock(&pcm_mutex);
 
@@ -340,11 +340,11 @@ static int soc_codec_close(struct snd_pcm_substream *substream)
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
 		snd_soc_dai_digital_mute(codec_dai, 1);
 
-	if (cpu_dai->ops.shutdown)
-		cpu_dai->ops.shutdown(substream, cpu_dai);
+	if (cpu_dai->ops->shutdown)
+		cpu_dai->ops->shutdown(substream, cpu_dai);
 
-	if (codec_dai->ops.shutdown)
-		codec_dai->ops.shutdown(substream, codec_dai);
+	if (codec_dai->ops->shutdown)
+		codec_dai->ops->shutdown(substream, codec_dai);
 
 	if (machine->ops && machine->ops->shutdown)
 		machine->ops->shutdown(substream);
@@ -387,7 +387,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = card->codec;
 	int ret = 0;
 
 	mutex_lock(&pcm_mutex);
@@ -408,16 +408,16 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream)
 		}
 	}
 
-	if (codec_dai->ops.prepare) {
-		ret = codec_dai->ops.prepare(substream, codec_dai);
+	if (codec_dai->ops->prepare) {
+		ret = codec_dai->ops->prepare(substream, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: codec DAI prepare error\n");
 			goto out;
 		}
 	}
 
-	if (cpu_dai->ops.prepare) {
-		ret = cpu_dai->ops.prepare(substream, cpu_dai);
+	if (cpu_dai->ops->prepare) {
+		ret = cpu_dai->ops->prepare(substream, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: cpu DAI prepare error\n");
 			goto out;
@@ -494,8 +494,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
-	if (codec_dai->ops.hw_params) {
-		ret = codec_dai->ops.hw_params(substream, params, codec_dai);
+	if (codec_dai->ops->hw_params) {
+		ret = codec_dai->ops->hw_params(substream, params, codec_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: can't set codec %s hw params\n",
 				codec_dai->name);
@@ -503,8 +503,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream,
 		}
 	}
 
-	if (cpu_dai->ops.hw_params) {
-		ret = cpu_dai->ops.hw_params(substream, params, cpu_dai);
+	if (cpu_dai->ops->hw_params) {
+		ret = cpu_dai->ops->hw_params(substream, params, cpu_dai);
 		if (ret < 0) {
 			printk(KERN_ERR "asoc: interface %s hw params failed\n",
 				cpu_dai->name);
@@ -526,12 +526,12 @@ out:
 	return ret;
 
 platform_err:
-	if (cpu_dai->ops.hw_free)
-		cpu_dai->ops.hw_free(substream, cpu_dai);
+	if (cpu_dai->ops->hw_free)
+		cpu_dai->ops->hw_free(substream, cpu_dai);
 
 interface_err:
-	if (codec_dai->ops.hw_free)
-		codec_dai->ops.hw_free(substream, codec_dai);
+	if (codec_dai->ops->hw_free)
+		codec_dai->ops->hw_free(substream, codec_dai);
 
 codec_err:
 	if (machine->ops && machine->ops->hw_free)
@@ -553,7 +553,7 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *cpu_dai = machine->cpu_dai;
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = card->codec;
 
 	mutex_lock(&pcm_mutex);
 
@@ -570,11 +570,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream)
 		platform->pcm_ops->hw_free(substream);
 
 	/* now free hw params for the DAI's  */
-	if (codec_dai->ops.hw_free)
-		codec_dai->ops.hw_free(substream, codec_dai);
+	if (codec_dai->ops->hw_free)
+		codec_dai->ops->hw_free(substream, codec_dai);
 
-	if (cpu_dai->ops.hw_free)
-		cpu_dai->ops.hw_free(substream, cpu_dai);
+	if (cpu_dai->ops->hw_free)
+		cpu_dai->ops->hw_free(substream, cpu_dai);
 
 	mutex_unlock(&pcm_mutex);
 	return 0;
@@ -591,8 +591,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	struct snd_soc_dai *codec_dai = machine->codec_dai;
 	int ret;
 
-	if (codec_dai->ops.trigger) {
-		ret = codec_dai->ops.trigger(substream, cmd, codec_dai);
+	if (codec_dai->ops->trigger) {
+		ret = codec_dai->ops->trigger(substream, cmd, codec_dai);
 		if (ret < 0)
 			return ret;
 	}
@@ -603,8 +603,8 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 			return ret;
 	}
 
-	if (cpu_dai->ops.trigger) {
-		ret = cpu_dai->ops.trigger(substream, cmd, cpu_dai);
+	if (cpu_dai->ops->trigger) {
+		ret = cpu_dai->ops->trigger(substream, cmd, cpu_dai);
 		if (ret < 0)
 			return ret;
 	}
@@ -629,7 +629,7 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	struct snd_soc_card *card = socdev->card;
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = card->codec;
 	int i;
 
 	/* Due to the resume being scheduled into a workqueue we could
@@ -645,8 +645,8 @@ static int soc_suspend(struct platform_device *pdev, pm_message_t state)
 	/* mute any active DAC's */
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
-		if (dai->ops.digital_mute && dai->playback.active)
-			dai->ops.digital_mute(dai, 1);
+		if (dai->ops->digital_mute && dai->playback.active)
+			dai->ops->digital_mute(dai, 1);
 	}
 
 	/* suspend all pcms */
@@ -705,7 +705,7 @@ static void soc_resume_deferred(struct work_struct *work)
 	struct snd_soc_device *socdev = card->socdev;
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = card->codec;
 	struct platform_device *pdev = to_platform_device(socdev->dev);
 	int i;
 
@@ -741,8 +741,8 @@ static void soc_resume_deferred(struct work_struct *work)
 	/* unmute any active DACs */
 	for (i = 0; i < card->num_links; i++) {
 		struct snd_soc_dai *dai = card->dai_link[i].codec_dai;
-		if (dai->ops.digital_mute && dai->playback.active)
-			dai->ops.digital_mute(dai, 0);
+		if (dai->ops->digital_mute && dai->playback.active)
+			dai->ops->digital_mute(dai, 0);
 	}
 
 	for (i = 0; i < card->num_links; i++) {
@@ -982,8 +982,8 @@ static struct platform_driver soc_driver = {
 static int soc_new_pcm(struct snd_soc_device *socdev,
 	struct snd_soc_dai_link *dai_link, int num)
 {
-	struct snd_soc_codec *codec = socdev->codec;
 	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_codec *codec = card->codec;
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_dai *codec_dai = dai_link->codec_dai;
 	struct snd_soc_dai *cpu_dai = dai_link->cpu_dai;
@@ -998,7 +998,7 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
 
 	rtd->dai = dai_link;
 	rtd->socdev = socdev;
-	codec_dai->codec = socdev->codec;
+	codec_dai->codec = card->codec;
 
 	/* check client and interface hw capabilities */
 	sprintf(new_name, "%s %s-%d", dai_link->stream_name, codec_dai->name,
@@ -1048,9 +1048,8 @@ static int soc_new_pcm(struct snd_soc_device *socdev,
 }
 
 /* codec register dump */
-static ssize_t soc_codec_reg_show(struct snd_soc_device *devdata, char *buf)
+static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf)
 {
-	struct snd_soc_codec *codec = devdata->codec;
 	int i, step = 1, count = 0;
 
 	if (!codec->reg_cache_size)
@@ -1090,7 +1089,7 @@ static ssize_t codec_reg_show(struct device *dev,
 	struct device_attribute *attr, char *buf)
 {
 	struct snd_soc_device *devdata = dev_get_drvdata(dev);
-	return soc_codec_reg_show(devdata, buf);
+	return soc_codec_reg_show(devdata->card->codec, buf);
 }
 
 static DEVICE_ATTR(codec_reg, 0444, codec_reg_show, NULL);
@@ -1107,12 +1106,10 @@ static ssize_t codec_reg_read_file(struct file *file, char __user *user_buf,
 {
 	ssize_t ret;
 	struct snd_soc_codec *codec = file->private_data;
-	struct device *card_dev = codec->card->dev;
-	struct snd_soc_device *devdata = card_dev->driver_data;
 	char *buf = kmalloc(PAGE_SIZE, GFP_KERNEL);
 	if (!buf)
 		return -ENOMEM;
-	ret = soc_codec_reg_show(devdata, buf);
+	ret = soc_codec_reg_show(codec, buf);
 	if (ret >= 0)
 		ret = simple_read_from_buffer(user_buf, count, ppos, buf, ret);
 	kfree(buf);
@@ -1309,19 +1306,19 @@ EXPORT_SYMBOL_GPL(snd_soc_test_bits);
  */
 int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid)
 {
-	struct snd_soc_codec *codec = socdev->codec;
 	struct snd_soc_card *card = socdev->card;
-	int ret = 0, i;
+	struct snd_soc_codec *codec = card->codec;
+	int ret, i;
 
 	mutex_lock(&codec->mutex);
 
 	/* register a sound card */
-	codec->card = snd_card_new(idx, xid, codec->owner, 0);
-	if (!codec->card) {
+	ret = snd_card_create(idx, xid, codec->owner, 0, &codec->card);
+	if (ret < 0) {
 		printk(KERN_ERR "asoc: can't create sound card for codec %s\n",
 			codec->name);
 		mutex_unlock(&codec->mutex);
-		return -ENODEV;
+		return ret;
 	}
 
 	codec->card->dev = socdev->dev;
@@ -1355,8 +1352,8 @@ EXPORT_SYMBOL_GPL(snd_soc_new_pcms);
  */
 int snd_soc_init_card(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
 	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_codec *codec = card->codec;
 	int ret = 0, i, ac97 = 0, err = 0;
 
 	for (i = 0; i < card->num_links; i++) {
@@ -1407,7 +1404,7 @@ int snd_soc_init_card(struct snd_soc_device *socdev)
 	if (err < 0)
 		printk(KERN_WARNING "asoc: failed to add codec sysfs files\n");
 
-	soc_init_codec_debugfs(socdev->codec);
+	soc_init_codec_debugfs(codec);
 	mutex_unlock(&codec->mutex);
 
 out:
@@ -1424,18 +1421,19 @@ EXPORT_SYMBOL_GPL(snd_soc_init_card);
  */
 void snd_soc_free_pcms(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 #ifdef CONFIG_SND_SOC_AC97_BUS
 	struct snd_soc_dai *codec_dai;
 	int i;
 #endif
 
 	mutex_lock(&codec->mutex);
-	soc_cleanup_codec_debugfs(socdev->codec);
+	soc_cleanup_codec_debugfs(codec);
 #ifdef CONFIG_SND_SOC_AC97_BUS
 	for (i = 0; i < codec->num_dai; i++) {
 		codec_dai = &codec->dai[i];
-		if (codec_dai->ac97_control && codec->ac97) {
+		if (codec_dai->ac97_control && codec->ac97 &&
+		    strcmp(codec->name, "AC97") != 0) {
 			soc_ac97_dev_unregister(codec);
 			goto free_card;
 		}
@@ -1498,6 +1496,37 @@ struct snd_kcontrol *snd_soc_cnew(const struct snd_kcontrol_new *_template,
 EXPORT_SYMBOL_GPL(snd_soc_cnew);
 
 /**
+ * snd_soc_add_controls - add an array of controls to a codec.
+ * Convienience function to add a list of controls. Many codecs were
+ * duplicating this code.
+ *
+ * @codec: codec to add controls to
+ * @controls: array of controls to add
+ * @num_controls: number of elements in the array
+ *
+ * Return 0 for success, else error.
+ */
+int snd_soc_add_controls(struct snd_soc_codec *codec,
+	const struct snd_kcontrol_new *controls, int num_controls)
+{
+	struct snd_card *card = codec->card;
+	int err, i;
+
+	for (i = 0; i < num_controls; i++) {
+		const struct snd_kcontrol_new *control = &controls[i];
+		err = snd_ctl_add(card, snd_soc_cnew(control, codec, NULL));
+		if (err < 0) {
+			dev_err(codec->dev, "%s: Failed to add %s\n",
+				codec->name, control->name);
+			return err;
+		}
+	}
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_add_controls);
+
+/**
  * snd_soc_info_enum_double - enumerated double mixer info callback
  * @kcontrol: mixer control
  * @uinfo: control element information
@@ -2023,8 +2052,8 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8);
 int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 	unsigned int freq, int dir)
 {
-	if (dai->ops.set_sysclk)
-		return dai->ops.set_sysclk(dai, clk_id, freq, dir);
+	if (dai->ops->set_sysclk)
+		return dai->ops->set_sysclk(dai, clk_id, freq, dir);
 	else
 		return -EINVAL;
 }
@@ -2043,8 +2072,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk);
 int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
 	int div_id, int div)
 {
-	if (dai->ops.set_clkdiv)
-		return dai->ops.set_clkdiv(dai, div_id, div);
+	if (dai->ops->set_clkdiv)
+		return dai->ops->set_clkdiv(dai, div_id, div);
 	else
 		return -EINVAL;
 }
@@ -2062,8 +2091,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv);
 int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
 	int pll_id, unsigned int freq_in, unsigned int freq_out)
 {
-	if (dai->ops.set_pll)
-		return dai->ops.set_pll(dai, pll_id, freq_in, freq_out);
+	if (dai->ops->set_pll)
+		return dai->ops->set_pll(dai, pll_id, freq_in, freq_out);
 	else
 		return -EINVAL;
 }
@@ -2078,8 +2107,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll);
  */
 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
 {
-	if (dai->ops.set_fmt)
-		return dai->ops.set_fmt(dai, fmt);
+	if (dai->ops->set_fmt)
+		return dai->ops->set_fmt(dai, fmt);
 	else
 		return -EINVAL;
 }
@@ -2097,8 +2126,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt);
 int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
 	unsigned int mask, int slots)
 {
-	if (dai->ops.set_sysclk)
-		return dai->ops.set_tdm_slot(dai, mask, slots);
+	if (dai->ops->set_sysclk)
+		return dai->ops->set_tdm_slot(dai, mask, slots);
 	else
 		return -EINVAL;
 }
@@ -2113,8 +2142,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot);
  */
 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate)
 {
-	if (dai->ops.set_sysclk)
-		return dai->ops.set_tristate(dai, tristate);
+	if (dai->ops->set_sysclk)
+		return dai->ops->set_tristate(dai, tristate);
 	else
 		return -EINVAL;
 }
@@ -2129,8 +2158,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate);
  */
 int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute)
 {
-	if (dai->ops.digital_mute)
-		return dai->ops.digital_mute(dai, mute);
+	if (dai->ops->digital_mute)
+		return dai->ops->digital_mute(dai, mute);
 	else
 		return -EINVAL;
 }
@@ -2183,6 +2212,9 @@ static int snd_soc_unregister_card(struct snd_soc_card *card)
 	return 0;
 }
 
+static struct snd_soc_dai_ops null_dai_ops = {
+};
+
 /**
  * snd_soc_register_dai - Register a DAI with the ASoC core
  *
@@ -2197,6 +2229,9 @@ int snd_soc_register_dai(struct snd_soc_dai *dai)
 	if (!dai->dev)
 		printk(KERN_WARNING "No device for DAI %s\n", dai->name);
 
+	if (!dai->ops)
+		dai->ops = &null_dai_ops;
+
 	INIT_LIST_HEAD(&dai->list);
 
 	mutex_lock(&client_mutex);
diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c
index a2f1da8b4646..735903a74675 100644
--- a/sound/soc/soc-dapm.c
+++ b/sound/soc/soc-dapm.c
@@ -54,14 +54,15 @@
 static int dapm_up_seq[] = {
 	snd_soc_dapm_pre, snd_soc_dapm_micbias, snd_soc_dapm_mic,
 	snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_dac,
-	snd_soc_dapm_mixer, snd_soc_dapm_pga, snd_soc_dapm_adc, snd_soc_dapm_hp,
-	snd_soc_dapm_spk, snd_soc_dapm_post
+	snd_soc_dapm_mixer, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_pga,
+	snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk, snd_soc_dapm_post
 };
+
 static int dapm_down_seq[] = {
 	snd_soc_dapm_pre, snd_soc_dapm_adc, snd_soc_dapm_hp, snd_soc_dapm_spk,
-	snd_soc_dapm_pga, snd_soc_dapm_mixer, snd_soc_dapm_dac, snd_soc_dapm_mic,
-	snd_soc_dapm_micbias, snd_soc_dapm_mux, snd_soc_dapm_value_mux,
-	snd_soc_dapm_post
+	snd_soc_dapm_pga, snd_soc_dapm_mixer_named_ctl, snd_soc_dapm_mixer,
+	snd_soc_dapm_dac, snd_soc_dapm_mic, snd_soc_dapm_micbias,
+	snd_soc_dapm_mux, snd_soc_dapm_value_mux, snd_soc_dapm_post
 };
 
 static int dapm_status = 1;
@@ -101,7 +102,8 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w,
 {
 	switch (w->id) {
 	case snd_soc_dapm_switch:
-	case snd_soc_dapm_mixer: {
+	case snd_soc_dapm_mixer:
+	case snd_soc_dapm_mixer_named_ctl: {
 		int val;
 		struct soc_mixer_control *mc = (struct soc_mixer_control *)
 			w->kcontrols[i].private_value;
@@ -323,15 +325,32 @@ static int dapm_new_mixer(struct snd_soc_codec *codec,
 			if (path->name != (char*)w->kcontrols[i].name)
 				continue;
 
-			/* add dapm control with long name */
-			name_len = 2 + strlen(w->name)
-				+ strlen(w->kcontrols[i].name);
+			/* add dapm control with long name.
+			 * for dapm_mixer this is the concatenation of the
+			 * mixer and kcontrol name.
+			 * for dapm_mixer_named_ctl this is simply the
+			 * kcontrol name.
+			 */
+			name_len = strlen(w->kcontrols[i].name) + 1;
+			if (w->id != snd_soc_dapm_mixer_named_ctl)
+				name_len += 1 + strlen(w->name);
+
 			path->long_name = kmalloc(name_len, GFP_KERNEL);
+
 			if (path->long_name == NULL)
 				return -ENOMEM;
 
-			snprintf(path->long_name, name_len, "%s %s",
-				 w->name, w->kcontrols[i].name);
+			switch (w->id) {
+			default:
+				snprintf(path->long_name, name_len, "%s %s",
+					 w->name, w->kcontrols[i].name);
+				break;
+			case snd_soc_dapm_mixer_named_ctl:
+				snprintf(path->long_name, name_len, "%s",
+					 w->kcontrols[i].name);
+				break;
+			}
+
 			path->long_name[name_len - 1] = '\0';
 
 			path->kcontrol = snd_soc_cnew(&w->kcontrols[i], w,
@@ -503,6 +522,137 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w,
 EXPORT_SYMBOL_GPL(dapm_reg_event);
 
 /*
+ * Scan a single DAPM widget for a complete audio path and update the
+ * power status appropriately.
+ */
+static int dapm_power_widget(struct snd_soc_codec *codec, int event,
+			     struct snd_soc_dapm_widget *w)
+{
+	int in, out, power_change, power, ret;
+
+	/* vmid - no action */
+	if (w->id == snd_soc_dapm_vmid)
+		return 0;
+
+	/* active ADC */
+	if (w->id == snd_soc_dapm_adc && w->active) {
+		in = is_connected_input_ep(w);
+		dapm_clear_walk(w->codec);
+		w->power = (in != 0) ? 1 : 0;
+		dapm_update_bits(w);
+		return 0;
+	}
+
+	/* active DAC */
+	if (w->id == snd_soc_dapm_dac && w->active) {
+		out = is_connected_output_ep(w);
+		dapm_clear_walk(w->codec);
+		w->power = (out != 0) ? 1 : 0;
+		dapm_update_bits(w);
+		return 0;
+	}
+
+	/* pre and post event widgets */
+	if (w->id == snd_soc_dapm_pre) {
+		if (!w->event)
+			return 0;
+
+		if (event == SND_SOC_DAPM_STREAM_START) {
+			ret = w->event(w,
+				       NULL, SND_SOC_DAPM_PRE_PMU);
+			if (ret < 0)
+				return ret;
+		} else if (event == SND_SOC_DAPM_STREAM_STOP) {
+			ret = w->event(w,
+				       NULL, SND_SOC_DAPM_PRE_PMD);
+			if (ret < 0)
+				return ret;
+		}
+		return 0;
+	}
+	if (w->id == snd_soc_dapm_post) {
+		if (!w->event)
+			return 0;
+
+		if (event == SND_SOC_DAPM_STREAM_START) {
+			ret = w->event(w,
+				       NULL, SND_SOC_DAPM_POST_PMU);
+			if (ret < 0)
+				return ret;
+		} else if (event == SND_SOC_DAPM_STREAM_STOP) {
+			ret = w->event(w,
+				       NULL, SND_SOC_DAPM_POST_PMD);
+			if (ret < 0)
+				return ret;
+		}
+		return 0;
+	}
+
+	/* all other widgets */
+	in = is_connected_input_ep(w);
+	dapm_clear_walk(w->codec);
+	out = is_connected_output_ep(w);
+	dapm_clear_walk(w->codec);
+	power = (out != 0 && in != 0) ? 1 : 0;
+	power_change = (w->power == power) ? 0 : 1;
+	w->power = power;
+
+	if (!power_change)
+		return 0;
+
+	/* call any power change event handlers */
+	if (w->event)
+		pr_debug("power %s event for %s flags %x\n",
+			 w->power ? "on" : "off",
+			 w->name, w->event_flags);
+
+	/* power up pre event */
+	if (power && w->event &&
+	    (w->event_flags & SND_SOC_DAPM_PRE_PMU)) {
+		ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU);
+		if (ret < 0)
+			return ret;
+	}
+
+	/* power down pre event */
+	if (!power && w->event &&
+	    (w->event_flags & SND_SOC_DAPM_PRE_PMD)) {
+		ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD);
+		if (ret < 0)
+			return ret;
+	}
+
+	/* Lower PGA volume to reduce pops */
+	if (w->id == snd_soc_dapm_pga && !power)
+		dapm_set_pga(w, power);
+
+	dapm_update_bits(w);
+
+	/* Raise PGA volume to reduce pops */
+	if (w->id == snd_soc_dapm_pga && power)
+		dapm_set_pga(w, power);
+
+	/* power up post event */
+	if (power && w->event &&
+	    (w->event_flags & SND_SOC_DAPM_POST_PMU)) {
+		ret = w->event(w,
+			       NULL, SND_SOC_DAPM_POST_PMU);
+		if (ret < 0)
+			return ret;
+	}
+
+	/* power down post event */
+	if (!power && w->event &&
+	    (w->event_flags & SND_SOC_DAPM_POST_PMD)) {
+		ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD);
+		if (ret < 0)
+			return ret;
+	}
+
+	return 0;
+}
+
+/*
  * Scan each dapm widget for complete audio path.
  * A complete path is a route that has valid endpoints i.e.:-
  *
@@ -514,7 +664,7 @@ EXPORT_SYMBOL_GPL(dapm_reg_event);
 static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
 {
 	struct snd_soc_dapm_widget *w;
-	int in, out, i, c = 1, *seq = NULL, ret = 0, power_change, power;
+	int i, c = 1, *seq = NULL, ret = 0;
 
 	/* do we have a sequenced stream event */
 	if (event == SND_SOC_DAPM_STREAM_START) {
@@ -525,135 +675,20 @@ static int dapm_power_widgets(struct snd_soc_codec *codec, int event)
 		seq = dapm_down_seq;
 	}
 
-	for(i = 0; i < c; i++) {
+	for (i = 0; i < c; i++) {
 		list_for_each_entry(w, &codec->dapm_widgets, list) {
 
 			/* is widget in stream order */
 			if (seq && seq[i] && w->id != seq[i])
 				continue;
 
-			/* vmid - no action */
-			if (w->id == snd_soc_dapm_vmid)
-				continue;
-
-			/* active ADC */
-			if (w->id == snd_soc_dapm_adc && w->active) {
-				in = is_connected_input_ep(w);
-				dapm_clear_walk(w->codec);
-				w->power = (in != 0) ? 1 : 0;
-				dapm_update_bits(w);
-				continue;
-			}
-
-			/* active DAC */
-			if (w->id == snd_soc_dapm_dac && w->active) {
-				out = is_connected_output_ep(w);
-				dapm_clear_walk(w->codec);
-				w->power = (out != 0) ? 1 : 0;
-				dapm_update_bits(w);
-				continue;
-			}
-
-			/* pre and post event widgets */
-			if (w->id == snd_soc_dapm_pre) {
-				if (!w->event)
-					continue;
-
-				if (event == SND_SOC_DAPM_STREAM_START) {
-					ret = w->event(w,
-						NULL, SND_SOC_DAPM_PRE_PMU);
-					if (ret < 0)
-						return ret;
-				} else if (event == SND_SOC_DAPM_STREAM_STOP) {
-					ret = w->event(w,
-						NULL, SND_SOC_DAPM_PRE_PMD);
-					if (ret < 0)
-						return ret;
-				}
-				continue;
-			}
-			if (w->id == snd_soc_dapm_post) {
-				if (!w->event)
-					continue;
-
-				if (event == SND_SOC_DAPM_STREAM_START) {
-					ret = w->event(w,
-						NULL, SND_SOC_DAPM_POST_PMU);
-					if (ret < 0)
-						return ret;
-				} else if (event == SND_SOC_DAPM_STREAM_STOP) {
-					ret = w->event(w,
-						NULL, SND_SOC_DAPM_POST_PMD);
-					if (ret < 0)
-						return ret;
-				}
-				continue;
-			}
-
-			/* all other widgets */
-			in = is_connected_input_ep(w);
-			dapm_clear_walk(w->codec);
-			out = is_connected_output_ep(w);
-			dapm_clear_walk(w->codec);
-			power = (out != 0 && in != 0) ? 1 : 0;
-			power_change = (w->power == power) ? 0: 1;
-			w->power = power;
-
-			if (!power_change)
-				continue;
-
-			/* call any power change event handlers */
-			if (w->event)
-				pr_debug("power %s event for %s flags %x\n",
-					 w->power ? "on" : "off",
-					 w->name, w->event_flags);
-
-			/* power up pre event */
-			if (power && w->event &&
-			    (w->event_flags & SND_SOC_DAPM_PRE_PMU)) {
-				ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU);
-				if (ret < 0)
-					return ret;
-			}
-
-			/* power down pre event */
-			if (!power && w->event &&
-			    (w->event_flags & SND_SOC_DAPM_PRE_PMD)) {
-				ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD);
-				if (ret < 0)
-					return ret;
-			}
-
-			/* Lower PGA volume to reduce pops */
-			if (w->id == snd_soc_dapm_pga && !power)
-				dapm_set_pga(w, power);
-
-			dapm_update_bits(w);
-
-			/* Raise PGA volume to reduce pops */
-			if (w->id == snd_soc_dapm_pga && power)
-				dapm_set_pga(w, power);
-
-			/* power up post event */
-			if (power && w->event &&
-			    (w->event_flags & SND_SOC_DAPM_POST_PMU)) {
-				ret = w->event(w,
-					       NULL, SND_SOC_DAPM_POST_PMU);
-				if (ret < 0)
-					return ret;
-			}
-
-			/* power down post event */
-			if (!power && w->event &&
-			    (w->event_flags & SND_SOC_DAPM_POST_PMD)) {
-				ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD);
-				if (ret < 0)
-					return ret;
-			}
+			ret = dapm_power_widget(codec, event, w);
+			if (ret != 0)
+				return ret;
 		}
 	}
 
-	return ret;
+	return 0;
 }
 
 #ifdef DEBUG
@@ -687,6 +722,7 @@ static void dbg_dump_dapm(struct snd_soc_codec* codec, const char *action)
 		case snd_soc_dapm_adc:
 		case snd_soc_dapm_pga:
 		case snd_soc_dapm_mixer:
+		case snd_soc_dapm_mixer_named_ctl:
 			if (w->name) {
 				in = is_connected_input_ep(w);
 				dapm_clear_walk(w->codec);
@@ -760,6 +796,7 @@ static int dapm_mixer_update_power(struct snd_soc_dapm_widget *widget,
 	int found = 0;
 
 	if (widget->id != snd_soc_dapm_mixer &&
+	    widget->id != snd_soc_dapm_mixer_named_ctl &&
 	    widget->id != snd_soc_dapm_switch)
 		return -ENODEV;
 
@@ -795,7 +832,7 @@ static ssize_t dapm_widget_show(struct device *dev,
 	struct device_attribute *attr, char *buf)
 {
 	struct snd_soc_device *devdata = dev_get_drvdata(dev);
-	struct snd_soc_codec *codec = devdata->codec;
+	struct snd_soc_codec *codec = devdata->card->codec;
 	struct snd_soc_dapm_widget *w;
 	int count = 0;
 	char *state = "not set";
@@ -813,6 +850,7 @@ static ssize_t dapm_widget_show(struct device *dev,
 		case snd_soc_dapm_adc:
 		case snd_soc_dapm_pga:
 		case snd_soc_dapm_mixer:
+		case snd_soc_dapm_mixer_named_ctl:
 			if (w->name)
 				count += sprintf(buf + count, "%s: %s\n",
 					w->name, w->power ? "On":"Off");
@@ -876,7 +914,7 @@ static void dapm_free_widgets(struct snd_soc_codec *codec)
 }
 
 static int snd_soc_dapm_set_pin(struct snd_soc_codec *codec,
-	char *pin, int status)
+				const char *pin, int status)
 {
 	struct snd_soc_dapm_widget *w;
 
@@ -991,6 +1029,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_codec *codec,
 		break;
 	case snd_soc_dapm_switch:
 	case snd_soc_dapm_mixer:
+	case snd_soc_dapm_mixer_named_ctl:
 		ret = dapm_connect_mixer(codec, wsource, wsink, path, control);
 		if (ret != 0)
 			goto err;
@@ -1068,6 +1107,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec)
 		switch(w->id) {
 		case snd_soc_dapm_switch:
 		case snd_soc_dapm_mixer:
+		case snd_soc_dapm_mixer_named_ctl:
 			dapm_new_mixer(codec, w);
 			break;
 		case snd_soc_dapm_mux:
@@ -1396,6 +1436,76 @@ out:
 EXPORT_SYMBOL_GPL(snd_soc_dapm_put_value_enum_double);
 
 /**
+ * snd_soc_dapm_info_pin_switch - Info for a pin switch
+ *
+ * @kcontrol: mixer control
+ * @uinfo: control element information
+ *
+ * Callback to provide information about a pin switch control.
+ */
+int snd_soc_dapm_info_pin_switch(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = 1;
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_info_pin_switch);
+
+/**
+ * snd_soc_dapm_get_pin_switch - Get information for a pin switch
+ *
+ * @kcontrol: mixer control
+ * @ucontrol: Value
+ */
+int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	const char *pin = (const char *)kcontrol->private_value;
+
+	mutex_lock(&codec->mutex);
+
+	ucontrol->value.integer.value[0] =
+		snd_soc_dapm_get_pin_status(codec, pin);
+
+	mutex_unlock(&codec->mutex);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_switch);
+
+/**
+ * snd_soc_dapm_put_pin_switch - Set information for a pin switch
+ *
+ * @kcontrol: mixer control
+ * @ucontrol: Value
+ */
+int snd_soc_dapm_put_pin_switch(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	const char *pin = (const char *)kcontrol->private_value;
+
+	mutex_lock(&codec->mutex);
+
+	if (ucontrol->value.integer.value[0])
+		snd_soc_dapm_enable_pin(codec, pin);
+	else
+		snd_soc_dapm_disable_pin(codec, pin);
+
+	snd_soc_dapm_sync(codec);
+
+	mutex_unlock(&codec->mutex);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_put_pin_switch);
+
+/**
  * snd_soc_dapm_new_control - create new dapm control
  * @codec: audio codec
  * @widget: widget template
@@ -1527,8 +1637,8 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_stream_event);
 int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
 				enum snd_soc_bias_level level)
 {
-	struct snd_soc_codec *codec = socdev->codec;
 	struct snd_soc_card *card = socdev->card;
+	struct snd_soc_codec *codec = socdev->card->codec;
 	int ret = 0;
 
 	if (card->set_bias_level)
@@ -1549,7 +1659,7 @@ int snd_soc_dapm_set_bias_level(struct snd_soc_device *socdev,
  * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
  * do any widget power switching.
  */
-int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin)
+int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin)
 {
 	return snd_soc_dapm_set_pin(codec, pin, 1);
 }
@@ -1564,7 +1674,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_enable_pin);
  * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
  * do any widget power switching.
  */
-int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin)
+int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, const char *pin)
 {
 	return snd_soc_dapm_set_pin(codec, pin, 0);
 }
@@ -1584,7 +1694,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin);
  * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
  * do any widget power switching.
  */
-int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin)
+int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, const char *pin)
 {
 	return snd_soc_dapm_set_pin(codec, pin, 0);
 }
@@ -1599,7 +1709,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin);
  *
  * Returns 1 for connected otherwise 0.
  */
-int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin)
+int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, const char *pin)
 {
 	struct snd_soc_dapm_widget *w;
 
@@ -1620,7 +1730,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_get_pin_status);
  */
 void snd_soc_dapm_free(struct snd_soc_device *socdev)
 {
-	struct snd_soc_codec *codec = socdev->codec;
+	struct snd_soc_codec *codec = socdev->card->codec;
 
 	snd_soc_dapm_sys_remove(socdev->dev);
 	dapm_free_widgets(codec);
diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c
new file mode 100644
index 000000000000..28346fb2e70c
--- /dev/null
+++ b/sound/soc/soc-jack.c
@@ -0,0 +1,267 @@
+/*
+ * soc-jack.c  --  ALSA SoC jack handling
+ *
+ * Copyright 2008 Wolfson Microelectronics PLC.
+ *
+ * Author: Mark Brown <broonie@opensource.wolfsonmicro.com>
+ *
+ *  This program is free software; you can redistribute  it and/or modify it
+ *  under  the terms of  the GNU General  Public License as published by the
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#include <sound/jack.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <linux/gpio.h>
+#include <linux/interrupt.h>
+#include <linux/workqueue.h>
+#include <linux/delay.h>
+
+/**
+ * snd_soc_jack_new - Create a new jack
+ * @card:  ASoC card
+ * @id:    an identifying string for this jack
+ * @type:  a bitmask of enum snd_jack_type values that can be detected by
+ *         this jack
+ * @jack:  structure to use for the jack
+ *
+ * Creates a new jack object.
+ *
+ * Returns zero if successful, or a negative error code on failure.
+ * On success jack will be initialised.
+ */
+int snd_soc_jack_new(struct snd_soc_card *card, const char *id, int type,
+		     struct snd_soc_jack *jack)
+{
+	jack->card = card;
+	INIT_LIST_HEAD(&jack->pins);
+
+	return snd_jack_new(card->codec->card, id, type, &jack->jack);
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_new);
+
+/**
+ * snd_soc_jack_report - Report the current status for a jack
+ *
+ * @jack:   the jack
+ * @status: a bitmask of enum snd_jack_type values that are currently detected.
+ * @mask:   a bitmask of enum snd_jack_type values that being reported.
+ *
+ * If configured using snd_soc_jack_add_pins() then the associated
+ * DAPM pins will be enabled or disabled as appropriate and DAPM
+ * synchronised.
+ *
+ * Note: This function uses mutexes and should be called from a
+ * context which can sleep (such as a workqueue).
+ */
+void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask)
+{
+	struct snd_soc_codec *codec = jack->card->codec;
+	struct snd_soc_jack_pin *pin;
+	int enable;
+	int oldstatus;
+
+	if (!jack) {
+		WARN_ON_ONCE(!jack);
+		return;
+	}
+
+	mutex_lock(&codec->mutex);
+
+	oldstatus = jack->status;
+
+	jack->status &= ~mask;
+	jack->status |= status;
+
+	/* The DAPM sync is expensive enough to be worth skipping */
+	if (jack->status == oldstatus)
+		goto out;
+
+	list_for_each_entry(pin, &jack->pins, list) {
+		enable = pin->mask & status;
+
+		if (pin->invert)
+			enable = !enable;
+
+		if (enable)
+			snd_soc_dapm_enable_pin(codec, pin->pin);
+		else
+			snd_soc_dapm_disable_pin(codec, pin->pin);
+	}
+
+	snd_soc_dapm_sync(codec);
+
+	snd_jack_report(jack->jack, status);
+
+out:
+	mutex_unlock(&codec->mutex);
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_report);
+
+/**
+ * snd_soc_jack_add_pins - Associate DAPM pins with an ASoC jack
+ *
+ * @jack:  ASoC jack
+ * @count: Number of pins
+ * @pins:  Array of pins
+ *
+ * After this function has been called the DAPM pins specified in the
+ * pins array will have their status updated to reflect the current
+ * state of the jack whenever the jack status is updated.
+ */
+int snd_soc_jack_add_pins(struct snd_soc_jack *jack, int count,
+			  struct snd_soc_jack_pin *pins)
+{
+	int i;
+
+	for (i = 0; i < count; i++) {
+		if (!pins[i].pin) {
+			printk(KERN_ERR "No name for pin %d\n", i);
+			return -EINVAL;
+		}
+		if (!pins[i].mask) {
+			printk(KERN_ERR "No mask for pin %d (%s)\n", i,
+			       pins[i].pin);
+			return -EINVAL;
+		}
+
+		INIT_LIST_HEAD(&pins[i].list);
+		list_add(&(pins[i].list), &jack->pins);
+	}
+
+	/* 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.
+	 */
+	snd_soc_jack_report(jack, 0, 0);
+
+	return 0;
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_add_pins);
+
+#ifdef CONFIG_GPIOLIB
+/* gpio detect */
+static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio)
+{
+	struct snd_soc_jack *jack = gpio->jack;
+	int enable;
+	int report;
+
+	if (gpio->debounce_time > 0)
+		mdelay(gpio->debounce_time);
+
+	enable = gpio_get_value(gpio->gpio);
+	if (gpio->invert)
+		enable = !enable;
+
+	if (enable)
+		report = gpio->report;
+	else
+		report = 0;
+
+	snd_soc_jack_report(jack, report, gpio->report);
+}
+
+/* irq handler for gpio pin */
+static irqreturn_t gpio_handler(int irq, void *data)
+{
+	struct snd_soc_jack_gpio *gpio = data;
+
+	schedule_work(&gpio->work);
+
+	return IRQ_HANDLED;
+}
+
+/* gpio work */
+static void gpio_work(struct work_struct *work)
+{
+	struct snd_soc_jack_gpio *gpio;
+
+	gpio = container_of(work, struct snd_soc_jack_gpio, work);
+	snd_soc_jack_gpio_detect(gpio);
+}
+
+/**
+ * snd_soc_jack_add_gpios - Associate GPIO pins with an ASoC jack
+ *
+ * @jack:  ASoC jack
+ * @count: number of pins
+ * @gpios: array of gpio pins
+ *
+ * This function will request gpio, set data direction and request irq
+ * for each gpio in the array.
+ */
+int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count,
+			struct snd_soc_jack_gpio *gpios)
+{
+	int i, ret;
+
+	for (i = 0; i < count; i++) {
+		if (!gpio_is_valid(gpios[i].gpio)) {
+			printk(KERN_ERR "Invalid gpio %d\n",
+				gpios[i].gpio);
+			ret = -EINVAL;
+			goto undo;
+		}
+		if (!gpios[i].name) {
+			printk(KERN_ERR "No name for gpio %d\n",
+				gpios[i].gpio);
+			ret = -EINVAL;
+			goto undo;
+		}
+
+		ret = gpio_request(gpios[i].gpio, gpios[i].name);
+		if (ret)
+			goto undo;
+
+		ret = gpio_direction_input(gpios[i].gpio);
+		if (ret)
+			goto err;
+
+		ret = request_irq(gpio_to_irq(gpios[i].gpio),
+				gpio_handler,
+				IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING,
+				jack->card->dev->driver->name,
+				&gpios[i]);
+		if (ret)
+			goto err;
+
+		INIT_WORK(&gpios[i].work, gpio_work);
+		gpios[i].jack = jack;
+	}
+
+	return 0;
+
+err:
+	gpio_free(gpios[i].gpio);
+undo:
+	snd_soc_jack_free_gpios(jack, i, gpios);
+
+	return ret;
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_add_gpios);
+
+/**
+ * snd_soc_jack_free_gpios - Release GPIO pins' resources of an ASoC jack
+ *
+ * @jack:  ASoC jack
+ * @count: number of pins
+ * @gpios: array of gpio pins
+ *
+ * Release gpio and irq resources for gpio pins associated with an ASoC jack.
+ */
+void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count,
+			struct snd_soc_jack_gpio *gpios)
+{
+	int i;
+
+	for (i = 0; i < count; i++) {
+		free_irq(gpio_to_irq(gpios[i].gpio), &gpios[i]);
+		gpio_free(gpios[i].gpio);
+		gpios[i].jack = NULL;
+	}
+}
+EXPORT_SYMBOL_GPL(snd_soc_jack_free_gpios);
+#endif	/* CONFIG_GPIOLIB */
diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c
index f87933e48812..574af56ba8a6 100644
--- a/sound/sparc/amd7930.c
+++ b/sound/sparc/amd7930.c
@@ -954,7 +954,8 @@ static int __devinit snd_amd7930_create(struct snd_card *card,
 	amd->regs = of_ioremap(&op->resource[0], 0,
 			       resource_size(&op->resource[0]), "amd7930");
 	if (!amd->regs) {
-		snd_printk("amd7930-%d: Unable to map chip registers.\n", dev);
+		snd_printk(KERN_ERR
+			   "amd7930-%d: Unable to map chip registers.\n", dev);
 		return -EIO;
 	}
 
@@ -962,7 +963,7 @@ static int __devinit snd_amd7930_create(struct snd_card *card,
 
 	if (request_irq(irq, snd_amd7930_interrupt,
 			IRQF_DISABLED | IRQF_SHARED, "amd7930", amd)) {
-		snd_printk("amd7930-%d: Unable to grab IRQ %d\n",
+		snd_printk(KERN_ERR "amd7930-%d: Unable to grab IRQ %d\n",
 			   dev, irq);
 		snd_amd7930_free(amd);
 		return -EBUSY;
@@ -1018,9 +1019,10 @@ static int __devinit amd7930_sbus_probe(struct of_device *op, const struct of_de
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev_num], id[dev_num], THIS_MODULE, 0);
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev_num], id[dev_num], THIS_MODULE, 0,
+			      &card);
+	if (err < 0)
+		return err;
 
 	strcpy(card->driver, "AMD7930");
 	strcpy(card->shortname, "Sun AMD7930");
diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c
index 41c387587474..7d93fa705ccf 100644
--- a/sound/sparc/cs4231.c
+++ b/sound/sparc/cs4231.c
@@ -1563,6 +1563,7 @@ static int __init cs4231_attach_begin(struct snd_card **rcard)
 {
 	struct snd_card *card;
 	struct snd_cs4231 *chip;
+	int err;
 
 	*rcard = NULL;
 
@@ -1574,10 +1575,10 @@ static int __init cs4231_attach_begin(struct snd_card **rcard)
 		return -ENOENT;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-			    sizeof(struct snd_cs4231));
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct snd_cs4231), &card);
+	if (err < 0)
+		return err;
 
 	strcpy(card->driver, "CS4231");
 	strcpy(card->shortname, "Sun CS4231");
diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c
index 23ed6f04a718..af95ff1e126c 100644
--- a/sound/sparc/dbri.c
+++ b/sound/sparc/dbri.c
@@ -2612,10 +2612,10 @@ static int __devinit dbri_probe(struct of_device *op, const struct of_device_id
 		return -ENODEV;
 	}
 
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-			    sizeof(struct snd_dbri));
-	if (card == NULL)
-		return -ENOMEM;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct snd_dbri), &card);
+	if (err < 0)
+		return err;
 
 	strcpy(card->driver, "DBRI");
 	strcpy(card->shortname, "Sun DBRI");
diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c
index 09802e8a6fb8..4c7b051f9d17 100644
--- a/sound/spi/at73c213.c
+++ b/sound/spi/at73c213.c
@@ -965,12 +965,11 @@ static int __devinit snd_at73c213_probe(struct spi_device *spi)
 		return PTR_ERR(board->dac_clk);
 	}
 
-	retval = -ENOMEM;
-
 	/* Allocate "card" using some unused identifiers. */
 	snprintf(id, sizeof id, "at73c213_%d", board->ssc_id);
-	card = snd_card_new(-1, id, THIS_MODULE, sizeof(struct snd_at73c213));
-	if (!card)
+	retval = snd_card_create(-1, id, THIS_MODULE,
+				 sizeof(struct snd_at73c213), &card);
+	if (retval < 0)
 		goto out;
 
 	chip = card->private_data;
diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c
index 0a5391436add..ff0b2a8fd25b 100644
--- a/sound/synth/emux/emux_hwdep.c
+++ b/sound/synth/emux/emux_hwdep.c
@@ -24,25 +24,6 @@
 #include <asm/uaccess.h>
 #include "emux_voice.h"
 
-/*
- * open the hwdep device
- */
-static int
-snd_emux_hwdep_open(struct snd_hwdep *hw, struct file *file)
-{
-	return 0;
-}
-
-
-/*
- * close the device
- */
-static int
-snd_emux_hwdep_release(struct snd_hwdep *hw, struct file *file)
-{
-	return 0;
-}
-
 
 #define TMP_CLIENT_ID	0x1001
 
@@ -146,8 +127,6 @@ snd_emux_init_hwdep(struct snd_emux *emu)
 	emu->hwdep = hw;
 	strcpy(hw->name, SNDRV_EMUX_HWDEP_NAME);
 	hw->iface = SNDRV_HWDEP_IFACE_EMUX_WAVETABLE;
-	hw->ops.open = snd_emux_hwdep_open;
-	hw->ops.release = snd_emux_hwdep_release;
 	hw->ops.ioctl = snd_emux_hwdep_ioctl;
 	hw->exclusive = 1;
 	hw->private_data = emu;
diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c
index 5c47b6c09264..87e42206c4ef 100644
--- a/sound/synth/emux/emux_oss.c
+++ b/sound/synth/emux/emux_oss.c
@@ -132,7 +132,7 @@ snd_emux_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure)
 	p = snd_emux_create_port(emu, tmpname, 32,
 				 1, &callback);
 	if (p == NULL) {
-		snd_printk("can't create port\n");
+		snd_printk(KERN_ERR "can't create port\n");
 		snd_emux_dec_count(emu);
 		mutex_unlock(&emu->register_mutex);
 		return -ENOMEM;
diff --git a/sound/synth/emux/emux_seq.c b/sound/synth/emux/emux_seq.c
index 335aa2ce2574..ca5f7effb4df 100644
--- a/sound/synth/emux/emux_seq.c
+++ b/sound/synth/emux/emux_seq.c
@@ -74,15 +74,15 @@ snd_emux_init_seq(struct snd_emux *emu, struct snd_card *card, int index)
 	emu->client = snd_seq_create_kernel_client(card, index,
 						   "%s WaveTable", emu->name);
 	if (emu->client < 0) {
-		snd_printk("can't create client\n");
+		snd_printk(KERN_ERR "can't create client\n");
 		return -ENODEV;
 	}
 
 	if (emu->num_ports < 0) {
-		snd_printk("seqports must be greater than zero\n");
+		snd_printk(KERN_WARNING "seqports must be greater than zero\n");
 		emu->num_ports = 1;
 	} else if (emu->num_ports >= SNDRV_EMUX_MAX_PORTS) {
-		snd_printk("too many ports."
+		snd_printk(KERN_WARNING "too many ports."
 			   "limited max. ports %d\n", SNDRV_EMUX_MAX_PORTS);
 		emu->num_ports = SNDRV_EMUX_MAX_PORTS;
 	}
@@ -100,7 +100,7 @@ snd_emux_init_seq(struct snd_emux *emu, struct snd_card *card, int index)
 		p = snd_emux_create_port(emu, tmpname, MIDI_CHANNELS,
 					 0, &pinfo);
 		if (p == NULL) {
-			snd_printk("can't create port\n");
+			snd_printk(KERN_ERR "can't create port\n");
 			return -ENOMEM;
 		}
 
@@ -147,12 +147,12 @@ snd_emux_create_port(struct snd_emux *emu, char *name,
 
 	/* Allocate structures for this channel */
 	if ((p = kzalloc(sizeof(*p), GFP_KERNEL)) == NULL) {
-		snd_printk("no memory\n");
+		snd_printk(KERN_ERR "no memory\n");
 		return NULL;
 	}
 	p->chset.channels = kcalloc(max_channels, sizeof(struct snd_midi_channel), GFP_KERNEL);
 	if (p->chset.channels == NULL) {
-		snd_printk("no memory\n");
+		snd_printk(KERN_ERR "no memory\n");
 		kfree(p);
 		return NULL;
 	}
@@ -376,12 +376,12 @@ int snd_emux_init_virmidi(struct snd_emux *emu, struct snd_card *card)
 			goto __error;
 		}
 		emu->vmidi[i] = rmidi;
-		//snd_printk("virmidi %d ok\n", i);
+		/* snd_printk(KERN_DEBUG "virmidi %d ok\n", i); */
 	}
 	return 0;
 
 __error:
-	//snd_printk("error init..\n");
+	/* snd_printk(KERN_DEBUG "error init..\n"); */
 	snd_emux_delete_virmidi(emu);
 	return -ENOMEM;
 }
diff --git a/sound/synth/emux/emux_synth.c b/sound/synth/emux/emux_synth.c
index 2cc6f6f79065..3e921b386fd5 100644
--- a/sound/synth/emux/emux_synth.c
+++ b/sound/synth/emux/emux_synth.c
@@ -956,7 +956,8 @@ void snd_emux_lock_voice(struct snd_emux *emu, int voice)
 	if (emu->voices[voice].state == SNDRV_EMUX_ST_OFF)
 		emu->voices[voice].state = SNDRV_EMUX_ST_LOCKED;
 	else
-		snd_printk("invalid voice for lock %d (state = %x)\n",
+		snd_printk(KERN_WARNING
+			   "invalid voice for lock %d (state = %x)\n",
 			   voice, emu->voices[voice].state);
 	spin_unlock_irqrestore(&emu->voice_lock, flags);
 }
@@ -973,7 +974,8 @@ void snd_emux_unlock_voice(struct snd_emux *emu, int voice)
 	if (emu->voices[voice].state == SNDRV_EMUX_ST_LOCKED)
 		emu->voices[voice].state = SNDRV_EMUX_ST_OFF;
 	else
-		snd_printk("invalid voice for unlock %d (state = %x)\n",
+		snd_printk(KERN_WARNING
+			   "invalid voice for unlock %d (state = %x)\n",
 			   voice, emu->voices[voice].state);
 	spin_unlock_irqrestore(&emu->voice_lock, flags);
 }
diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c
index 36d53bd317ed..63c8f45c0c22 100644
--- a/sound/synth/emux/soundfont.c
+++ b/sound/synth/emux/soundfont.c
@@ -133,7 +133,7 @@ snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data,
 	int  rc;
 
 	if (count < (long)sizeof(patch)) {
-		snd_printk("patch record too small %ld\n", count);
+		snd_printk(KERN_ERR "patch record too small %ld\n", count);
 		return -EINVAL;
 	}
 	if (copy_from_user(&patch, data, sizeof(patch)))
@@ -143,15 +143,16 @@ snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data,
 	data += sizeof(patch);
 
 	if (patch.key != SNDRV_OSS_SOUNDFONT_PATCH) {
-		snd_printk("'The wrong kind of patch' %x\n", patch.key);
+		snd_printk(KERN_ERR "The wrong kind of patch %x\n", patch.key);
 		return -EINVAL;
 	}
 	if (count < patch.len) {
-		snd_printk("Patch too short %ld, need %d\n", count, patch.len);
+		snd_printk(KERN_ERR "Patch too short %ld, need %d\n",
+			   count, patch.len);
 		return -EINVAL;
 	}
 	if (patch.len < 0) {
-		snd_printk("poor length %d\n", patch.len);
+		snd_printk(KERN_ERR "poor length %d\n", patch.len);
 		return -EINVAL;
 	}
 
@@ -195,7 +196,8 @@ snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data,
 	case SNDRV_SFNT_REMOVE_INFO:
 		/* patch must be opened */
 		if (!sflist->currsf) {
-			snd_printk("soundfont: remove_info: patch not opened\n");
+			snd_printk(KERN_ERR "soundfont: remove_info: "
+				   "patch not opened\n");
 			rc = -EINVAL;
 		} else {
 			int bank, instr;
@@ -531,7 +533,7 @@ load_info(struct snd_sf_list *sflist, const void __user *data, long count)
 		return -EINVAL;
 
 	if (count < (long)sizeof(hdr)) {
-		printk("Soundfont error: invalid patch zone length\n");
+		printk(KERN_ERR "Soundfont error: invalid patch zone length\n");
 		return -EINVAL;
 	}
 	if (copy_from_user((char*)&hdr, data, sizeof(hdr)))
@@ -541,12 +543,14 @@ load_info(struct snd_sf_list *sflist, const void __user *data, long count)
 	count -= sizeof(hdr);
 
 	if (hdr.nvoices <= 0 || hdr.nvoices >= 100) {
-		printk("Soundfont error: Illegal voice number %d\n", hdr.nvoices);
+		printk(KERN_ERR "Soundfont error: Illegal voice number %d\n",
+		       hdr.nvoices);
 		return -EINVAL;
 	}
 
 	if (count < (long)sizeof(struct soundfont_voice_info) * hdr.nvoices) {
-		printk("Soundfont Error: patch length(%ld) is smaller than nvoices(%d)\n",
+		printk(KERN_ERR "Soundfont Error: "
+		       "patch length(%ld) is smaller than nvoices(%d)\n",
 		       count, hdr.nvoices);
 		return -EINVAL;
 	}
@@ -952,7 +956,7 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data,
 	int rc;
 
 	if (count < (long)sizeof(patch)) {
-		snd_printk("patch record too small %ld\n", count);
+		snd_printk(KERN_ERR "patch record too small %ld\n", count);
 		return -EINVAL;
 	}
 	if (copy_from_user(&patch, data, sizeof(patch)))
@@ -1034,7 +1038,8 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data,
 	/* panning position; -128 - 127 => 0-127 */
 	zone->v.pan = (patch.panning + 128) / 2;
 #if 0
-	snd_printk("gus: basefrq=%d (ofs=%d) root=%d,tune=%d, range:%d-%d\n",
+	snd_printk(KERN_DEBUG
+		   "gus: basefrq=%d (ofs=%d) root=%d,tune=%d, range:%d-%d\n",
 		   (int)patch.base_freq, zone->v.rate_offset,
 		   zone->v.root, zone->v.tune, zone->v.low, zone->v.high);
 #endif
@@ -1068,7 +1073,8 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data,
 		zone->v.parm.volrelease = 0x8000 | snd_sf_calc_parm_decay(release);
 		zone->v.attenuation = calc_gus_attenuation(patch.env_offset[0]);
 #if 0
-		snd_printk("gus: atkhld=%x, dcysus=%x, volrel=%x, att=%d\n",
+		snd_printk(KERN_DEBUG
+			   "gus: atkhld=%x, dcysus=%x, volrel=%x, att=%d\n",
 			   zone->v.parm.volatkhld,
 			   zone->v.parm.voldcysus,
 			   zone->v.parm.volrelease,
diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig
index 4f0eac9bff1e..523aec188ccf 100644
--- a/sound/usb/Kconfig
+++ b/sound/usb/Kconfig
@@ -48,7 +48,10 @@ config SND_USB_CAIAQ
 	    * Native Instruments Kore Controller
 	    * Native Instruments Kore Controller 2
 	    * Native Instruments Audio Kontrol 1
+	    * Native Instruments Audio 4 DJ
 	    * Native Instruments Audio 8 DJ
+	    * Native Instruments Guitar Rig Session I/O
+	    * Native Instruments Guitar Rig mobile
 
 	   To compile this driver as a module, choose M here: the module
 	   will be called snd-usb-caiaq.
diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c
index b3a603325835..08d51e0c9fea 100644
--- a/sound/usb/caiaq/caiaq-audio.c
+++ b/sound/usb/caiaq/caiaq-audio.c
@@ -114,6 +114,7 @@ static int stream_start(struct snd_usb_caiaqdev *dev)
 	dev->output_panic = 0;
 	dev->first_packet = 1;
 	dev->streaming = 1;
+	dev->warned = 0;
 
 	for (i = 0; i < N_URBS; i++) {
 		ret = usb_submit_urb(dev->data_urbs_in[i], GFP_ATOMIC);
@@ -376,6 +377,9 @@ static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev,
 
 		for (stream = 0; stream < dev->n_streams; stream++, i++) {
 			sub = dev->sub_capture[stream];
+			if (dev->input_panic)
+				usb_buf[i] = 0;
+
 			if (sub) {
 				struct snd_pcm_runtime *rt = sub->runtime;
 				char *audio_buf = rt->dma_area;
@@ -397,6 +401,9 @@ static void read_in_urb(struct snd_usb_caiaqdev *dev,
 	if (!dev->streaming)
 		return;
 
+	if (iso->actual_length < dev->bpp)
+		return;
+
 	switch (dev->spec.data_alignment) {
 	case 0:
 		read_in_urb_mode0(dev, urb, iso);
@@ -406,10 +413,11 @@ static void read_in_urb(struct snd_usb_caiaqdev *dev,
 		break;
 	}
 
-	if (dev->input_panic || dev->output_panic) {
+	if ((dev->input_panic || dev->output_panic) && !dev->warned) {
 		debug("streaming error detected %s %s\n", 
 				dev->input_panic ? "(input)" : "",
 				dev->output_panic ? "(output)" : "");
+		dev->warned = 1;
 	}
 }
 
@@ -638,9 +646,10 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev)
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1):
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3):
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_SESSIONIO):
-		dev->samplerates |= SNDRV_PCM_RATE_88200;
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_GUITARRIGMOBILE):
 		dev->samplerates |= SNDRV_PCM_RATE_192000;
-		break;
+		/* fall thru */
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ):
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ):
 		dev->samplerates |= SNDRV_PCM_RATE_88200;
 		break;
diff --git a/sound/usb/caiaq/caiaq-control.c b/sound/usb/caiaq/caiaq-control.c
index ccd763dd7167..e92c2bbf4fe9 100644
--- a/sound/usb/caiaq/caiaq-control.c
+++ b/sound/usb/caiaq/caiaq-control.c
@@ -39,12 +39,12 @@ static int control_info(struct snd_kcontrol *kcontrol,
 	struct snd_usb_caiaqdev *dev = caiaqdev(chip->card);
 	int pos = kcontrol->private_value;
 	int is_intval = pos & CNT_INTVAL;
+	unsigned int id = dev->chip.usb_id;
 
 	uinfo->count = 1;
 	pos &= ~CNT_INTVAL;
 
-	if (dev->chip.usb_id ==
-		USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ)
+	if (id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ)
 		&& (pos == 0)) {
 		/* current input mode of A8DJ */
 		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
@@ -53,6 +53,15 @@ static int control_info(struct snd_kcontrol *kcontrol,
 		return 0;
 	}
 
+	if (id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ)
+		&& (pos == 0)) {
+		/* current input mode of A4DJ */
+		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+		uinfo->value.integer.min = 0;
+		uinfo->value.integer.max = 1;
+		return 0;
+	}
+
 	if (is_intval) {
 		uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
 		uinfo->value.integer.min = 0;
@@ -73,6 +82,14 @@ static int control_get(struct snd_kcontrol *kcontrol,
 	struct snd_usb_caiaqdev *dev = caiaqdev(chip->card);
 	int pos = kcontrol->private_value;
 
+	if (dev->chip.usb_id ==
+		USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ)) {
+		/* A4DJ has only one control */
+		/* do not expose hardware input mode 0 */
+		ucontrol->value.integer.value[0] = dev->control_state[0] - 1;
+		return 0;
+	}
+
 	if (pos & CNT_INTVAL)
 		ucontrol->value.integer.value[0]
 			= dev->control_state[pos & ~CNT_INTVAL];
@@ -90,10 +107,20 @@ static int control_put(struct snd_kcontrol *kcontrol,
 	struct snd_usb_caiaqdev *dev = caiaqdev(chip->card);
 	int pos = kcontrol->private_value;
 
+	if (dev->chip.usb_id ==
+		USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ)) {
+		/* A4DJ has only one control */
+		/* do not expose hardware input mode 0 */
+		dev->control_state[0] = ucontrol->value.integer.value[0] + 1;
+		snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO,
+				dev->control_state, sizeof(dev->control_state));
+		return 1;
+	}
+
 	if (pos & CNT_INTVAL) {
 		dev->control_state[pos & ~CNT_INTVAL]
 			= ucontrol->value.integer.value[0];
-		snd_usb_caiaq_send_command(dev, EP1_CMD_DIMM_LEDS,
+		snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO,
 				dev->control_state, sizeof(dev->control_state));
 	} else {
 		if (ucontrol->value.integer.value[0])
@@ -243,10 +270,13 @@ static struct caiaq_controller a8dj_controller[] = {
 	{ "GND lift for TC Vinyl mode", 	24 + 0 		},
 	{ "GND lift for TC CD/Line mode", 	24 + 1 		},
 	{ "GND lift for phono mode", 		24 + 2 		},
-	{ "GND lift for TC Vinyl mode", 	24 + 3 		},
 	{ "Software lock", 			40 		}
 };
 
+static struct caiaq_controller a4dj_controller[] = {
+	{ "Current input mode",	0 | CNT_INTVAL 	}
+};
+
 static int __devinit add_controls(struct caiaq_controller *c, int num,
 				  struct snd_usb_caiaqdev *dev)
 {
@@ -295,6 +325,10 @@ int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev)
 		ret = add_controls(a8dj_controller,
 			ARRAY_SIZE(a8dj_controller), dev);
 		break;
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ):
+		ret = add_controls(a4dj_controller,
+			ARRAY_SIZE(a4dj_controller), dev);
+		break;
 	}
 
 	return ret;
diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c
index 41c36b055f6b..cf573a982fdc 100644
--- a/sound/usb/caiaq/caiaq-device.c
+++ b/sound/usb/caiaq/caiaq-device.c
@@ -42,15 +42,17 @@
 #endif
 
 MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.10");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.13");
 MODULE_LICENSE("GPL");
 MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
 			 "{Native Instruments, RigKontrol3},"
 			 "{Native Instruments, Kore Controller},"
 			 "{Native Instruments, Kore Controller 2},"
 			 "{Native Instruments, Audio Kontrol 1},"
+			 "{Native Instruments, Audio 4 DJ},"
 			 "{Native Instruments, Audio 8 DJ},"
-			 "{Native Instruments, Session I/O}}");
+			 "{Native Instruments, Session I/O},"
+			 "{Native Instruments, GuitarRig mobile}");
 
 static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */
 static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */
@@ -116,6 +118,16 @@ static struct usb_device_id snd_usb_id_table[] = {
 		.idVendor =     USB_VID_NATIVEINSTRUMENTS,
 		.idProduct =    USB_PID_SESSIONIO
 	},
+	{
+		.match_flags =  USB_DEVICE_ID_MATCH_DEVICE,
+		.idVendor =     USB_VID_NATIVEINSTRUMENTS,
+		.idProduct =    USB_PID_GUITARRIGMOBILE
+	},
+	{
+		.match_flags =  USB_DEVICE_ID_MATCH_DEVICE,
+		.idVendor =     USB_VID_NATIVEINSTRUMENTS,
+		.idProduct =    USB_PID_AUDIO4DJ
+	},
 	{ /* terminator */ }
 };
 
@@ -239,6 +251,8 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev,
 		
 	if (dev->audio_parm_answer != 1) 
 		debug("unable to set the device's audio params\n");
+	else
+		dev->bpp = bpp;
 
 	return dev->audio_parm_answer == 1 ? 0 : -EINVAL;
 }
@@ -300,6 +314,12 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev)
 		}
 
 		break;
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ):
+		/* Audio 4 DJ - default input mode to phono */
+		dev->control_state[0] = 2;
+		snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO,
+			dev->control_state, 1);
+		break;
 	}
 	
 	if (dev->spec.num_analog_audio_out +
@@ -336,9 +356,10 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev)
 		log("Unable to set up control system (ret=%d)\n", ret);
 }
 
-static struct snd_card* create_card(struct usb_device* usb_dev)
+static int create_card(struct usb_device* usb_dev, struct snd_card **cardp)
 {
 	int devnum;
+	int err;
 	struct snd_card *card;
 	struct snd_usb_caiaqdev *dev;
 
@@ -347,12 +368,12 @@ static struct snd_card* create_card(struct usb_device* usb_dev)
 			break;
 
 	if (devnum >= SNDRV_CARDS)
-		return NULL;
+		return -ENODEV;
 
-	card = snd_card_new(index[devnum], id[devnum], THIS_MODULE, 
-					sizeof(struct snd_usb_caiaqdev));
-	if (!card)
-		return NULL;
+	err = snd_card_create(index[devnum], id[devnum], THIS_MODULE, 
+			      sizeof(struct snd_usb_caiaqdev), &card);
+	if (err < 0)
+		return err;
 
 	dev = caiaqdev(card);
 	dev->chip.dev = usb_dev;
@@ -362,7 +383,8 @@ static struct snd_card* create_card(struct usb_device* usb_dev)
 	spin_lock_init(&dev->spinlock);
 	snd_card_set_dev(card, &usb_dev->dev);
 
-	return card;
+	*cardp = card;
+	return 0;
 }
 
 static int __devinit init_card(struct snd_usb_caiaqdev *dev)
@@ -441,10 +463,10 @@ static int __devinit snd_probe(struct usb_interface *intf,
 	struct snd_card *card;
 	struct usb_device *device = interface_to_usbdev(intf);
 	
-	card = create_card(device);
+	ret = create_card(device, &card);
 	
-	if (!card)
-		return -ENOMEM;
+	if (ret < 0)
+		return ret;
 			
 	usb_set_intfdata(intf, card);
 	ret = init_card(caiaqdev(card));
diff --git a/sound/usb/caiaq/caiaq-device.h b/sound/usb/caiaq/caiaq-device.h
index ab56e738c5fc..4cce1ad7493d 100644
--- a/sound/usb/caiaq/caiaq-device.h
+++ b/sound/usb/caiaq/caiaq-device.h
@@ -10,8 +10,10 @@
 #define USB_PID_KORECONTROLLER	0x4711
 #define USB_PID_KORECONTROLLER2	0x4712
 #define USB_PID_AK1		0x0815
+#define USB_PID_AUDIO4DJ	0x0839
 #define USB_PID_AUDIO8DJ	0x1978
 #define USB_PID_SESSIONIO	0x1915
+#define USB_PID_GUITARRIGMOBILE	0x0d8d
 
 #define EP1_BUFSIZE 64
 #define CAIAQ_USB_STR_LEN 0xff
@@ -87,9 +89,9 @@ struct snd_usb_caiaqdev {
 	int audio_out_buf_pos[MAX_STREAMS];
 	int period_in_count[MAX_STREAMS];
 	int period_out_count[MAX_STREAMS];
-	int input_panic, output_panic;
+	int input_panic, output_panic, warned;
 	char *audio_in_buf, *audio_out_buf;
-	unsigned int samplerates;
+	unsigned int samplerates, bpp;
 
 	struct snd_pcm_substream *sub_playback[MAX_STREAMS];
 	struct snd_pcm_substream *sub_capture[MAX_STREAMS];
diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c
index 19e37451c216..c2db0f959681 100644
--- a/sound/usb/usbaudio.c
+++ b/sound/usb/usbaudio.c
@@ -107,7 +107,7 @@ MODULE_PARM_DESC(ignore_ctl_error,
 #define MAX_PACKS_HS	(MAX_PACKS * 8)	/* in high speed mode */
 #define MAX_URBS	8
 #define SYNC_URBS	4	/* always four urbs for sync */
-#define MIN_PACKS_URB	1	/* minimum 1 packet per urb */
+#define MAX_QUEUE	24	/* try not to exceed this queue length, in ms */
 
 struct audioformat {
 	struct list_head list;
@@ -525,7 +525,7 @@ static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs)
 /*
  * Prepare urb for streaming before playback starts or when paused.
  *
- * We don't have any data, so we send a frame of silence.
+ * We don't have any data, so we send silence.
  */
 static int prepare_nodata_playback_urb(struct snd_usb_substream *subs,
 				       struct snd_pcm_runtime *runtime,
@@ -537,13 +537,13 @@ static int prepare_nodata_playback_urb(struct snd_usb_substream *subs,
 
 	offs = 0;
 	urb->dev = ctx->subs->dev;
-	urb->number_of_packets = subs->packs_per_ms;
-	for (i = 0; i < subs->packs_per_ms; ++i) {
+	for (i = 0; i < ctx->packets; ++i) {
 		counts = snd_usb_audio_next_packet_size(subs);
 		urb->iso_frame_desc[i].offset = offs * stride;
 		urb->iso_frame_desc[i].length = counts * stride;
 		offs += counts;
 	}
+	urb->number_of_packets = ctx->packets;
 	urb->transfer_buffer_length = offs * stride;
 	memset(urb->transfer_buffer,
 	       subs->cur_audiofmt->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0,
@@ -1034,9 +1034,9 @@ static void release_substream_urbs(struct snd_usb_substream *subs, int force)
 static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int period_bytes,
 			       unsigned int rate, unsigned int frame_bits)
 {
-	unsigned int maxsize, n, i;
+	unsigned int maxsize, i;
 	int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK;
-	unsigned int npacks[MAX_URBS], urb_packs, total_packs, packs_per_ms;
+	unsigned int urb_packs, total_packs, packs_per_ms;
 
 	/* calculate the frequency in 16.16 format */
 	if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL)
@@ -1070,8 +1070,7 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri
 	subs->packs_per_ms = packs_per_ms;
 
 	if (is_playback) {
-		urb_packs = nrpacks;
-		urb_packs = max(urb_packs, (unsigned int)MIN_PACKS_URB);
+		urb_packs = max(nrpacks, 1);
 		urb_packs = min(urb_packs, (unsigned int)MAX_PACKS);
 	} else
 		urb_packs = 1;
@@ -1079,7 +1078,7 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri
 
 	/* decide how many packets to be used */
 	if (is_playback) {
-		unsigned int minsize;
+		unsigned int minsize, maxpacks;
 		/* determine how small a packet can be */
 		minsize = (subs->freqn >> (16 - subs->datainterval))
 			  * (frame_bits >> 3);
@@ -1092,8 +1091,13 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri
 		total_packs = (total_packs + packs_per_ms - 1)
 				& ~(packs_per_ms - 1);
 		/* we need at least two URBs for queueing */
-		if (total_packs < 2 * MIN_PACKS_URB * packs_per_ms)
-			total_packs = 2 * MIN_PACKS_URB * packs_per_ms;
+		if (total_packs < 2 * packs_per_ms) {
+			total_packs = 2 * packs_per_ms;
+		} else {
+			/* and we don't want too long a queue either */
+			maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2);
+			total_packs = min(total_packs, maxpacks);
+		}
 	} else {
 		total_packs = MAX_URBS * urb_packs;
 	}
@@ -1102,31 +1106,11 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri
 		/* too much... */
 		subs->nurbs = MAX_URBS;
 		total_packs = MAX_URBS * urb_packs;
-	}
-	n = total_packs;
-	for (i = 0; i < subs->nurbs; i++) {
-		npacks[i] = n > urb_packs ? urb_packs : n;
-		n -= urb_packs;
-	}
-	if (subs->nurbs <= 1) {
+	} else if (subs->nurbs < 2) {
 		/* too little - we need at least two packets
 		 * to ensure contiguous playback/capture
 		 */
 		subs->nurbs = 2;
-		npacks[0] = (total_packs + 1) / 2;
-		npacks[1] = total_packs - npacks[0];
-	} else if (npacks[subs->nurbs-1] < MIN_PACKS_URB * packs_per_ms) {
-		/* the last packet is too small.. */
-		if (subs->nurbs > 2) {
-			/* merge to the first one */
-			npacks[0] += npacks[subs->nurbs - 1];
-			subs->nurbs--;
-		} else {
-			/* divide to two */
-			subs->nurbs = 2;
-			npacks[0] = (total_packs + 1) / 2;
-			npacks[1] = total_packs - npacks[0];
-		}
 	}
 
 	/* allocate and initialize data urbs */
@@ -1134,7 +1118,8 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri
 		struct snd_urb_ctx *u = &subs->dataurb[i];
 		u->index = i;
 		u->subs = subs;
-		u->packets = npacks[i];
+		u->packets = (i + 1) * total_packs / subs->nurbs
+			- i * total_packs / subs->nurbs;
 		u->buffer_size = maxsize * u->packets;
 		if (subs->fmt_type == USB_FORMAT_TYPE_II)
 			u->packets++; /* for transfer delimiter */
@@ -1292,14 +1277,14 @@ static int init_usb_sample_rate(struct usb_device *dev, int iface,
 		if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR,
 					   USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT,
 					   SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) {
-			snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep 0x%x\n",
+			snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n",
 				   dev->devnum, iface, fmt->altsetting, rate, ep);
 			return err;
 		}
 		if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), GET_CUR,
 					   USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_IN,
 					   SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) {
-			snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep 0x%x\n",
+			snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n",
 				   dev->devnum, iface, fmt->altsetting, ep);
 			return 0; /* some devices don't support reading */
 		}
@@ -1431,9 +1416,11 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt)
 	subs->cur_audiofmt = fmt;
 
 #if 0
-	printk("setting done: format = %d, rate = %d..%d, channels = %d\n",
+	printk(KERN_DEBUG
+	       "setting done: format = %d, rate = %d..%d, channels = %d\n",
 	       fmt->format, fmt->rate_min, fmt->rate_max, fmt->channels);
-	printk("  datapipe = 0x%0x, syncpipe = 0x%0x\n",
+	printk(KERN_DEBUG
+	       "  datapipe = 0x%0x, syncpipe = 0x%0x\n",
 	       subs->datapipe, subs->syncpipe);
 #endif
 
@@ -1468,7 +1455,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream,
 	channels = params_channels(hw_params);
 	fmt = find_format(subs, format, rate, channels);
 	if (!fmt) {
-		snd_printd(KERN_DEBUG "cannot set format: format = 0x%x, rate = %d, channels = %d\n",
+		snd_printd(KERN_DEBUG "cannot set format: format = %#x, rate = %d, channels = %d\n",
 			   format, rate, channels);
 		return -EINVAL;
 	}
@@ -1795,7 +1782,7 @@ static int check_hw_params_convention(struct snd_usb_substream *subs)
 			if (rates[f->format] && rates[f->format] != f->rates)
 				goto __out;
 		}
-		channels[f->format] |= (1 << f->channels);
+		channels[f->format] |= 1 << (f->channels - 1);
 		rates[f->format] |= f->rates;
 		/* needs knot? */
 		if (f->rates & SNDRV_PCM_RATE_KNOT)
@@ -1822,7 +1809,7 @@ static int check_hw_params_convention(struct snd_usb_substream *subs)
 			continue;
 		for (i = 0; i < 32; i++) {
 			if (f->rates & (1 << i))
-				channels[i] |= (1 << f->channels);
+				channels[i] |= 1 << (f->channels - 1);
 		}
 	}
 	cmaster = 0;
@@ -1919,7 +1906,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre
 	 * in the current code assume the 1ms period.
 	 */
 	snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME,
-				     1000 * MIN_PACKS_URB,
+				     1000,
 				     /*(nrpacks * MAX_URBS) * 1000*/ UINT_MAX);
 
 	err = check_hw_params_convention(subs);
@@ -2160,7 +2147,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s
 		fp = list_entry(p, struct audioformat, list);
 		snd_iprintf(buffer, "  Interface %d\n", fp->iface);
 		snd_iprintf(buffer, "    Altset %d\n", fp->altsetting);
-		snd_iprintf(buffer, "    Format: 0x%x\n", fp->format);
+		snd_iprintf(buffer, "    Format: %#x\n", fp->format);
 		snd_iprintf(buffer, "    Channels: %d\n", fp->channels);
 		snd_iprintf(buffer, "    Endpoint: %d %s (%s)\n",
 			    fp->endpoint & USB_ENDPOINT_NUMBER_MASK,
@@ -2180,7 +2167,7 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s
 			snd_iprintf(buffer, "\n");
 		}
 		// snd_iprintf(buffer, "    Max Packet Size = %d\n", fp->maxpacksize);
-		// snd_iprintf(buffer, "    EP Attribute = 0x%x\n", fp->attributes);
+		// snd_iprintf(buffer, "    EP Attribute = %#x\n", fp->attributes);
 	}
 }
 
@@ -2621,7 +2608,7 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat
 		fp->format = SNDRV_PCM_FORMAT_MPEG;
 		break;
 	default:
-		snd_printd(KERN_INFO "%d:%u:%d : unknown format tag 0x%x is detected.  processed as MPEG.\n",
+		snd_printd(KERN_INFO "%d:%u:%d : unknown format tag %#x is detected.  processed as MPEG.\n",
 			   chip->dev->devnum, fp->iface, fp->altsetting, format);
 		fp->format = SNDRV_PCM_FORMAT_MPEG;
 		break;
@@ -2819,7 +2806,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 			continue;
 		}
 
-		snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint 0x%x\n", dev->devnum, iface_no, altno, fp->endpoint);
+		snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint %#x\n", dev->devnum, iface_no, altno, fp->endpoint);
 		err = add_audio_endpoint(chip, stream, fp);
 		if (err < 0) {
 			kfree(fp->rate_table);
@@ -3466,10 +3453,10 @@ static int snd_usb_audio_create(struct usb_device *dev, int idx,
 		return -ENXIO;
 	}
 
-	card = snd_card_new(index[idx], id[idx], THIS_MODULE, 0);
-	if (card == NULL) {
+	err = snd_card_create(index[idx], id[idx], THIS_MODULE, 0, &card);
+	if (err < 0) {
 		snd_printk(KERN_ERR "cannot create card instance %d\n", idx);
-		return -ENOMEM;
+		return err;
 	}
 
 	chip = kzalloc(sizeof(*chip), GFP_KERNEL);
@@ -3766,7 +3753,7 @@ static int usb_audio_resume(struct usb_interface *intf)
 
 static int __init snd_usb_audio_init(void)
 {
-	if (nrpacks < MIN_PACKS_URB || nrpacks > MAX_PACKS) {
+	if (nrpacks < 1 || nrpacks > MAX_PACKS) {
 		printk(KERN_WARNING "invalid nrpacks value.\n");
 		return -EINVAL;
 	}
diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c
index 00397c8a765b..ecb58e7a6245 100644
--- a/sound/usb/usbmixer.c
+++ b/sound/usb/usbmixer.c
@@ -66,6 +66,7 @@ static const struct rc_config {
 	{ USB_ID(0x041e, 0x3000), 0, 1, 2, 1,  18, 0x0013 }, /* Extigy       */
 	{ USB_ID(0x041e, 0x3020), 2, 1, 6, 6,  18, 0x0013 }, /* Audigy 2 NX  */
 	{ USB_ID(0x041e, 0x3040), 2, 2, 6, 6,  2,  0x6e91 }, /* Live! 24-bit */
+	{ USB_ID(0x041e, 0x3048), 2, 2, 6, 6,  2,  0x6e91 }, /* Toshiba SB0500 */
 };
 
 struct usb_mixer_interface {
@@ -78,7 +79,6 @@ struct usb_mixer_interface {
 
 	/* Sound Blaster remote control stuff */
 	const struct rc_config *rc_cfg;
-	unsigned long rc_hwdep_open;
 	u32 rc_code;
 	wait_queue_head_t rc_waitq;
 	struct urb *rc_urb;
@@ -110,6 +110,8 @@ struct mixer_build {
 	const struct usbmix_selector_map *selector_map;
 };
 
+#define MAX_CHANNELS	10	/* max logical channels */
+
 struct usb_mixer_elem_info {
 	struct usb_mixer_interface *mixer;
 	struct usb_mixer_elem_info *next_id_elem; /* list of controls with same id */
@@ -120,6 +122,8 @@ struct usb_mixer_elem_info {
 	int channels;
 	int val_type;
 	int min, max, res;
+	int cached;
+	int cache_val[MAX_CHANNELS];
 	u8 initialized;
 };
 
@@ -181,8 +185,6 @@ enum {
 	USB_PROC_DCR_RELEASE = 6,
 };
 
-#define MAX_CHANNELS	10	/* max logical channels */
-
 
 /*
  * manual mapping of mixer names
@@ -219,7 +221,10 @@ static int check_ignored_ctl(struct mixer_build *state, int unitid, int control)
 	for (p = state->map; p->id; p++) {
 		if (p->id == unitid && ! p->name &&
 		    (! control || ! p->control || control == p->control)) {
-			// printk("ignored control %d:%d\n", unitid, control);
+			/*
+			printk(KERN_DEBUG "ignored control %d:%d\n",
+			       unitid, control);
+			*/
 			return 1;
 		}
 	}
@@ -376,11 +381,35 @@ static int get_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int *
 }
 
 /* channel = 0: master, 1 = first channel */
-static inline int get_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int *value)
+static inline int get_cur_mix_raw(struct usb_mixer_elem_info *cval,
+				  int channel, int *value)
 {
 	return get_ctl_value(cval, GET_CUR, (cval->control << 8) | channel, value);
 }
 
+static int get_cur_mix_value(struct usb_mixer_elem_info *cval,
+			     int channel, int index, int *value)
+{
+	int err;
+
+	if (cval->cached & (1 << channel)) {
+		*value = cval->cache_val[index];
+		return 0;
+	}
+	err = get_cur_mix_raw(cval, channel, value);
+	if (err < 0) {
+		if (!cval->mixer->ignore_ctl_error)
+			snd_printd(KERN_ERR "cannot get current value for "
+				   "control %d ch %d: err = %d\n",
+				   cval->control, channel, err);
+		return err;
+	}
+	cval->cached |= 1 << channel;
+	cval->cache_val[index] = *value;
+	return 0;
+}
+
+
 /*
  * set a mixer value
  */
@@ -412,9 +441,17 @@ static int set_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int v
 	return set_ctl_value(cval, SET_CUR, validx, value);
 }
 
-static inline int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int value)
+static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel,
+			     int index, int value)
 {
-	return set_ctl_value(cval, SET_CUR, (cval->control << 8) | channel, value);
+	int err;
+	err = set_ctl_value(cval, SET_CUR, (cval->control << 8) | channel,
+			    value);
+	if (err < 0)
+		return err;
+	cval->cached |= 1 << channel;
+	cval->cache_val[index] = value;
+	return 0;
 }
 
 /*
@@ -718,7 +755,7 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min)
 		if (cval->min + cval->res < cval->max) {
 			int last_valid_res = cval->res;
 			int saved, test, check;
-			get_cur_mix_value(cval, minchn, &saved);
+			get_cur_mix_raw(cval, minchn, &saved);
 			for (;;) {
 				test = saved;
 				if (test < cval->max)
@@ -726,8 +763,8 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min)
 				else
 					test -= cval->res;
 				if (test < cval->min || test > cval->max ||
-				    set_cur_mix_value(cval, minchn, test) ||
-				    get_cur_mix_value(cval, minchn, &check)) {
+				    set_cur_mix_value(cval, minchn, 0, test) ||
+				    get_cur_mix_raw(cval, minchn, &check)) {
 					cval->res = last_valid_res;
 					break;
 				}
@@ -735,7 +772,7 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min)
 					break;
 				cval->res *= 2;
 			}
-			set_cur_mix_value(cval, minchn, saved);
+			set_cur_mix_value(cval, minchn, 0, saved);
 		}
 
 		cval->initialized = 1;
@@ -775,35 +812,25 @@ static int mixer_ctl_feature_get(struct snd_kcontrol *kcontrol, struct snd_ctl_e
 	struct usb_mixer_elem_info *cval = kcontrol->private_data;
 	int c, cnt, val, err;
 
+	ucontrol->value.integer.value[0] = cval->min;
 	if (cval->cmask) {
 		cnt = 0;
 		for (c = 0; c < MAX_CHANNELS; c++) {
-			if (cval->cmask & (1 << c)) {
-				err = get_cur_mix_value(cval, c + 1, &val);
-				if (err < 0) {
-					if (cval->mixer->ignore_ctl_error) {
-						ucontrol->value.integer.value[0] = cval->min;
-						return 0;
-					}
-					snd_printd(KERN_ERR "cannot get current value for control %d ch %d: err = %d\n", cval->control, c + 1, err);
-					return err;
-				}
-				val = get_relative_value(cval, val);
-				ucontrol->value.integer.value[cnt] = val;
-				cnt++;
-			}
+			if (!(cval->cmask & (1 << c)))
+				continue;
+			err = get_cur_mix_value(cval, c + 1, cnt, &val);
+			if (err < 0)
+				return cval->mixer->ignore_ctl_error ? 0 : err;
+			val = get_relative_value(cval, val);
+			ucontrol->value.integer.value[cnt] = val;
+			cnt++;
 		}
+		return 0;
 	} else {
 		/* master channel */
-		err = get_cur_mix_value(cval, 0, &val);
-		if (err < 0) {
-			if (cval->mixer->ignore_ctl_error) {
-				ucontrol->value.integer.value[0] = cval->min;
-				return 0;
-			}
-			snd_printd(KERN_ERR "cannot get current value for control %d master ch: err = %d\n", cval->control, err);
-			return err;
-		}
+		err = get_cur_mix_value(cval, 0, 0, &val);
+		if (err < 0)
+			return cval->mixer->ignore_ctl_error ? 0 : err;
 		val = get_relative_value(cval, val);
 		ucontrol->value.integer.value[0] = val;
 	}
@@ -820,34 +847,28 @@ static int mixer_ctl_feature_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e
 	if (cval->cmask) {
 		cnt = 0;
 		for (c = 0; c < MAX_CHANNELS; c++) {
-			if (cval->cmask & (1 << c)) {
-				err = get_cur_mix_value(cval, c + 1, &oval);
-				if (err < 0) {
-					if (cval->mixer->ignore_ctl_error)
-						return 0;
-					return err;
-				}
-				val = ucontrol->value.integer.value[cnt];
-				val = get_abs_value(cval, val);
-				if (oval != val) {
-					set_cur_mix_value(cval, c + 1, val);
-					changed = 1;
-				}
-				get_cur_mix_value(cval, c + 1, &val);
-				cnt++;
+			if (!(cval->cmask & (1 << c)))
+				continue;
+			err = get_cur_mix_value(cval, c + 1, cnt, &oval);
+			if (err < 0)
+				return cval->mixer->ignore_ctl_error ? 0 : err;
+			val = ucontrol->value.integer.value[cnt];
+			val = get_abs_value(cval, val);
+			if (oval != val) {
+				set_cur_mix_value(cval, c + 1, cnt, val);
+				changed = 1;
 			}
+			cnt++;
 		}
 	} else {
 		/* master channel */
-		err = get_cur_mix_value(cval, 0, &oval);
-		if (err < 0 && cval->mixer->ignore_ctl_error)
-			return 0;
+		err = get_cur_mix_value(cval, 0, 0, &oval);
 		if (err < 0)
-			return err;
+			return cval->mixer->ignore_ctl_error ? 0 : err;
 		val = ucontrol->value.integer.value[0];
 		val = get_abs_value(cval, val);
 		if (val != oval) {
-			set_cur_mix_value(cval, 0, val);
+			set_cur_mix_value(cval, 0, 0, val);
 			changed = 1;
 		}
 	}
@@ -1706,7 +1727,8 @@ static void snd_usb_mixer_memory_change(struct usb_mixer_interface *mixer,
 		break;
 	/* live24ext: 4 = line-in jack */
 	case 3:	/* hp-out jack (may actuate Mute) */
-		if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040))
+		if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) ||
+		    mixer->chip->usb_id == USB_ID(0x041e, 0x3048))
 			snd_usb_mixer_notify_id(mixer, mixer->rc_cfg->mute_mixer_id);
 		break;
 	default:
@@ -1797,24 +1819,6 @@ static void snd_usb_soundblaster_remote_complete(struct urb *urb)
 	wake_up(&mixer->rc_waitq);
 }
 
-static int snd_usb_sbrc_hwdep_open(struct snd_hwdep *hw, struct file *file)
-{
-	struct usb_mixer_interface *mixer = hw->private_data;
-
-	if (test_and_set_bit(0, &mixer->rc_hwdep_open))
-		return -EBUSY;
-	return 0;
-}
-
-static int snd_usb_sbrc_hwdep_release(struct snd_hwdep *hw, struct file *file)
-{
-	struct usb_mixer_interface *mixer = hw->private_data;
-
-	clear_bit(0, &mixer->rc_hwdep_open);
-	smp_mb__after_clear_bit();
-	return 0;
-}
-
 static long snd_usb_sbrc_hwdep_read(struct snd_hwdep *hw, char __user *buf,
 				     long count, loff_t *offset)
 {
@@ -1867,9 +1871,8 @@ static int snd_usb_soundblaster_remote_init(struct usb_mixer_interface *mixer)
 	hwdep->iface = SNDRV_HWDEP_IFACE_SB_RC;
 	hwdep->private_data = mixer;
 	hwdep->ops.read = snd_usb_sbrc_hwdep_read;
-	hwdep->ops.open = snd_usb_sbrc_hwdep_open;
-	hwdep->ops.release = snd_usb_sbrc_hwdep_release;
 	hwdep->ops.poll = snd_usb_sbrc_hwdep_poll;
+	hwdep->exclusive = 1;
 
 	mixer->rc_urb = usb_alloc_urb(0, GFP_KERNEL);
 	if (!mixer->rc_urb)
@@ -1956,8 +1959,9 @@ static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer)
 	int i, err;
 
 	for (i = 0; i < ARRAY_SIZE(snd_audigy2nx_controls); ++i) {
-		if (i > 1 &&  /* Live24ext has 2 LEDs only */
-			mixer->chip->usb_id == USB_ID(0x041e, 0x3040))
+		if (i > 1 && /* Live24ext has 2 LEDs only */
+			(mixer->chip->usb_id == USB_ID(0x041e, 0x3040) ||
+			 mixer->chip->usb_id == USB_ID(0x041e, 0x3048)))
 			break; 
 		err = snd_ctl_add(mixer->chip->card,
 				  snd_ctl_new1(&snd_audigy2nx_controls[i], mixer));
@@ -1994,7 +1998,8 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry,
 	snd_iprintf(buffer, "%s jacks\n\n", mixer->chip->card->shortname);
 	if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020))
 		jacks = jacks_audigy2nx;
-	else if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040))
+	else if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) ||
+		 mixer->chip->usb_id == USB_ID(0x041e, 0x3048))
 		jacks = jacks_live24ext;
 	else
 		return;
@@ -2044,7 +2049,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif,
 		goto _error;
 
 	if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020) ||
-	    mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) {
+	    mixer->chip->usb_id == USB_ID(0x041e, 0x3040) ||
+	    mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) {
 		struct snd_info_entry *entry;
 
 		if ((err = snd_audigy2nx_controls_create(mixer)) < 0)
diff --git a/sound/usb/usbmixer_maps.c b/sound/usb/usbmixer_maps.c
index d755be0ad811..3e5d66cf1f5a 100644
--- a/sound/usb/usbmixer_maps.c
+++ b/sound/usb/usbmixer_maps.c
@@ -261,6 +261,22 @@ static struct usbmix_name_map aureon_51_2_map[] = {
 	{} /* terminator */
 };
 
+static struct usbmix_name_map scratch_live_map[] = {
+	/* 1: IT Line 1 (USB streaming) */
+	/* 2: OT Line 1 (Speaker) */
+	/* 3: IT Line 1 (Line connector) */
+	{ 4, "Line 1 In" }, /* FU */
+	/* 5: OT Line 1 (USB streaming) */
+	/* 6: IT Line 2 (USB streaming) */
+	/* 7: OT Line 2 (Speaker) */
+	/* 8: IT Line 2 (Line connector) */
+	{ 9, "Line 2 In" }, /* FU */
+	/* 10: OT Line 2 (USB streaming) */
+	/* 11: IT Mic (Line connector) */
+	/* 12: OT Mic (USB streaming) */
+	{ 0 } /* terminator */
+};
+
 /*
  * Control map entries
  */
@@ -285,6 +301,11 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = {
 		.map = live24ext_map,
 	},
 	{
+		.id = USB_ID(0x041e, 0x3048),
+		.map = audigy2nx_map,
+		.selector_map = audigy2nx_selectors,
+	},
+	{
 		/* Hercules DJ Console (Windows Edition) */
 		.id = USB_ID(0x06f8, 0xb000),
 		.ignore_ctl_error = 1,
@@ -311,6 +332,11 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = {
 		.id = USB_ID(0x0ccd, 0x0028),
 		.map = aureon_51_2_map,
 	},
+	{
+		.id = USB_ID(0x13e5, 0x0001),
+		.map = scratch_live_map,
+		.ignore_ctl_error = 1,
+	},
 	{ 0 } /* terminator */
 };
 
diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h
index 5d8ef09b9dcc..647ef5029651 100644
--- a/sound/usb/usbquirks.h
+++ b/sound/usb/usbquirks.h
@@ -39,6 +39,16 @@
 	.idProduct = prod, \
 	.bInterfaceClass = USB_CLASS_VENDOR_SPEC
 
+/* Creative/Toshiba Multimedia Center SB-0500 */
+{
+	USB_DEVICE(0x041e, 0x3048),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		.vendor_name = "Toshiba",
+		.product_name = "SB-0500",
+		.ifnum = QUIRK_NO_INTERFACE
+	}
+},
+
 /* Creative/E-Mu devices */
 {
 	USB_DEVICE(0x041e, 0x3010),
diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c
index 73e59f4403a4..98276aafefe6 100644
--- a/sound/usb/usx2y/us122l.c
+++ b/sound/usb/usx2y/us122l.c
@@ -478,19 +478,21 @@ static bool us122l_create_card(struct snd_card *card)
 	return true;
 }
 
-static struct snd_card *usx2y_create_card(struct usb_device *device)
+static int usx2y_create_card(struct usb_device *device, struct snd_card **cardp)
 {
 	int		dev;
 	struct snd_card *card;
+	int err;
+
 	for (dev = 0; dev < SNDRV_CARDS; ++dev)
 		if (enable[dev] && !snd_us122l_card_used[dev])
 			break;
 	if (dev >= SNDRV_CARDS)
-		return NULL;
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE,
-			    sizeof(struct us122l));
-	if (!card)
-		return NULL;
+		return -ENODEV;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct us122l), &card);
+	if (err < 0)
+		return err;
 	snd_us122l_card_used[US122L(card)->chip.index = dev] = 1;
 
 	US122L(card)->chip.dev = device;
@@ -509,46 +511,57 @@ static struct snd_card *usx2y_create_card(struct usb_device *device)
 		US122L(card)->chip.dev->devnum
 		);
 	snd_card_set_dev(card, &device->dev);
-	return card;
+	*cardp = card;
+	return 0;
 }
 
-static void *us122l_usb_probe(struct usb_interface *intf,
-			      const struct usb_device_id *device_id)
+static int us122l_usb_probe(struct usb_interface *intf,
+			    const struct usb_device_id *device_id,
+			    struct snd_card **cardp)
 {
 	struct usb_device *device = interface_to_usbdev(intf);
-	struct snd_card *card = usx2y_create_card(device);
+	struct snd_card *card;
+	int err;
 
-	if (!card)
-		return NULL;
+	err = usx2y_create_card(device, &card);
+	if (err < 0)
+		return err;
 
-	if (!us122l_create_card(card) ||
-	    snd_card_register(card) < 0) {
+	if (!us122l_create_card(card)) {
 		snd_card_free(card);
-		return NULL;
+		return -EINVAL;
+	}
+
+	err = snd_card_register(card);
+	if (err < 0) {
+		snd_card_free(card);
+		return err;
 	}
 
 	usb_get_dev(device);
-	return card;
+	*cardp = card;
+	return 0;
 }
 
 static int snd_us122l_probe(struct usb_interface *intf,
 			    const struct usb_device_id *id)
 {
 	struct snd_card *card;
+	int err;
+
 	snd_printdd(KERN_DEBUG"%p:%i\n",
 		    intf, intf->cur_altsetting->desc.bInterfaceNumber);
 	if (intf->cur_altsetting->desc.bInterfaceNumber != 1)
 		return 0;
 
-	card = us122l_usb_probe(usb_get_intf(intf), id);
-
-	if (card) {
-		usb_set_intfdata(intf, card);
-		return 0;
+	err = us122l_usb_probe(usb_get_intf(intf), id, &card);
+	if (err < 0) {
+		usb_put_intf(intf);
+		return err;
 	}
 
-	usb_put_intf(intf);
-	return -EIO;
+	usb_set_intfdata(intf, card);
+	return 0;
 }
 
 static void snd_us122l_disconnect(struct usb_interface *intf)
diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c
index 1558a5c4094f..4af8740db717 100644
--- a/sound/usb/usx2y/usX2Yhwdep.c
+++ b/sound/usb/usx2y/usX2Yhwdep.c
@@ -30,9 +30,6 @@
 #include "usbusx2y.h"
 #include "usX2Yhwdep.h"
 
-int usX2Y_hwdep_pcm_new(struct snd_card *card);
-
-
 static int snd_us428ctls_vm_fault(struct vm_area_struct *area,
 				  struct vm_fault *vmf)
 {
@@ -106,16 +103,6 @@ static unsigned int snd_us428ctls_poll(struct snd_hwdep *hw, struct file *file,
 }
 
 
-static int snd_usX2Y_hwdep_open(struct snd_hwdep *hw, struct file *file)
-{
-	return 0;
-}
-
-static int snd_usX2Y_hwdep_release(struct snd_hwdep *hw, struct file *file)
-{
-	return 0;
-}
-
 static int snd_usX2Y_hwdep_dsp_status(struct snd_hwdep *hw,
 				      struct snd_hwdep_dsp_status *info)
 {
@@ -267,8 +254,6 @@ int usX2Y_hwdep_new(struct snd_card *card, struct usb_device* device)
 
 	hw->iface = SNDRV_HWDEP_IFACE_USX2Y;
 	hw->private_data = usX2Y(card);
-	hw->ops.open = snd_usX2Y_hwdep_open;
-	hw->ops.release = snd_usX2Y_hwdep_release;
 	hw->ops.dsp_status = snd_usX2Y_hwdep_dsp_status;
 	hw->ops.dsp_load = snd_usX2Y_hwdep_dsp_load;
 	hw->ops.mmap = snd_us428ctls_mmap;
diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c
index 70b96355ca4c..24393dafcb6e 100644
--- a/sound/usb/usx2y/usb_stream.c
+++ b/sound/usb/usx2y/usb_stream.c
@@ -557,7 +557,7 @@ static void stream_start(struct usb_stream_kernel *sk,
 		s->idle_insize -= max_diff - max_diff_0;
 		s->idle_insize += urb_size - s->period_size;
 		if (s->idle_insize < 0) {
-			snd_printk("%i %i %i\n",
+			snd_printk(KERN_WARNING "%i %i %i\n",
 				   s->idle_insize, urb_size, s->period_size);
 			return;
 		} else if (s->idle_insize == 0) {
diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c
index 11639bd72a51..5ce0da23ee96 100644
--- a/sound/usb/usx2y/usbusx2y.c
+++ b/sound/usb/usx2y/usbusx2y.c
@@ -227,9 +227,9 @@ static void i_usX2Y_In04Int(struct urb *urb)
 	
 	if (usX2Y->US04) {
 		if (0 == usX2Y->US04->submitted)
-			do
+			do {
 				err = usb_submit_urb(usX2Y->US04->urb[usX2Y->US04->submitted++], GFP_ATOMIC);
-			while (!err && usX2Y->US04->submitted < usX2Y->US04->len);
+			} while (!err && usX2Y->US04->submitted < usX2Y->US04->len);
 	} else
 		if (us428ctls && us428ctls->p4outLast >= 0 && us428ctls->p4outLast < N_us428_p4out_BUFS) {
 			if (us428ctls->p4outLast != us428ctls->p4outSent) {
@@ -333,18 +333,21 @@ static struct usb_device_id snd_usX2Y_usb_id_table[] = {
 	{ /* terminator */ }
 };
 
-static struct snd_card *usX2Y_create_card(struct usb_device *device)
+static int usX2Y_create_card(struct usb_device *device, struct snd_card **cardp)
 {
 	int		dev;
 	struct snd_card *	card;
+	int err;
+
 	for (dev = 0; dev < SNDRV_CARDS; ++dev)
 		if (enable[dev] && !snd_usX2Y_card_used[dev])
 			break;
 	if (dev >= SNDRV_CARDS)
-		return NULL;
-	card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(struct usX2Ydev));
-	if (!card)
-		return NULL;
+		return -ENODEV;
+	err = snd_card_create(index[dev], id[dev], THIS_MODULE,
+			      sizeof(struct usX2Ydev), &card);
+	if (err < 0)
+		return err;
 	snd_usX2Y_card_used[usX2Y(card)->chip.index = dev] = 1;
 	card->private_free = snd_usX2Y_card_private_free;
 	usX2Y(card)->chip.dev = device;
@@ -362,26 +365,36 @@ static struct snd_card *usX2Y_create_card(struct usb_device *device)
 		usX2Y(card)->chip.dev->bus->busnum, usX2Y(card)->chip.dev->devnum
 		);
 	snd_card_set_dev(card, &device->dev);
-	return card;
+	*cardp = card;
+	return 0;
 }
 
 
-static void *usX2Y_usb_probe(struct usb_device *device, struct usb_interface *intf, const struct usb_device_id *device_id)
+static int usX2Y_usb_probe(struct usb_device *device,
+			   struct usb_interface *intf,
+			   const struct usb_device_id *device_id,
+			   struct snd_card **cardp)
 {
 	int		err;
 	struct snd_card *	card;
+
+	*cardp = NULL;
 	if (le16_to_cpu(device->descriptor.idVendor) != 0x1604 ||
 	    (le16_to_cpu(device->descriptor.idProduct) != USB_ID_US122 &&
 	     le16_to_cpu(device->descriptor.idProduct) != USB_ID_US224 &&
-	     le16_to_cpu(device->descriptor.idProduct) != USB_ID_US428) ||
-	    !(card = usX2Y_create_card(device)))
-		return NULL;
+	     le16_to_cpu(device->descriptor.idProduct) != USB_ID_US428))
+		return -EINVAL;
+
+	err = usX2Y_create_card(device, &card);
+	if (err < 0)
+		return err;
 	if ((err = usX2Y_hwdep_new(card, device)) < 0  ||
 	    (err = snd_card_register(card)) < 0) {
 		snd_card_free(card);
-		return NULL;
+		return err;
 	}
-	return card;
+	*cardp = card;
+	return 0;
 }
 
 /*
@@ -389,13 +402,14 @@ static void *usX2Y_usb_probe(struct usb_device *device, struct usb_interface *in
  */
 static int snd_usX2Y_probe(struct usb_interface *intf, const struct usb_device_id *id)
 {
-	void *chip;
-	chip = usX2Y_usb_probe(interface_to_usbdev(intf), intf, id);
-	if (chip) {
-		usb_set_intfdata(intf, chip);
-		return 0;
-	} else
-		return -EIO;
+	struct snd_card *card;
+	int err;
+
+	err = usX2Y_usb_probe(interface_to_usbdev(intf), intf, id, &card);
+	if (err < 0)
+		return err;
+	dev_set_drvdata(&intf->dev, card);
+	return 0;
 }
 
 static void snd_usX2Y_disconnect(struct usb_interface *intf)
diff --git a/sound/usb/usx2y/usx2yhwdeppcm.h b/sound/usb/usx2y/usx2yhwdeppcm.h
index c3382fdc386b..9c4fb84b2aa0 100644
--- a/sound/usb/usx2y/usx2yhwdeppcm.h
+++ b/sound/usb/usx2y/usx2yhwdeppcm.h
@@ -18,3 +18,5 @@ struct snd_usX2Y_hwdep_pcm_shm {
 	volatile unsigned captured_iso_frames;
 	int capture_iso_start;
 };
+
+int usX2Y_hwdep_pcm_new(struct snd_card *card);