init android origin source code

33 files changed, 7711 insertions, 0 deletions

diff --git a/ANDROID_3.4.5/sound/aoa/Kconfig b/ANDROID_3.4.5/sound/aoa/Kconfig
new file mode 100644
index 00000000..c081e18b
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/Kconfig
@@ -0,0 +1,17 @@
+menuconfig SND_AOA
+	tristate "Apple Onboard Audio driver"
+	depends on PPC_PMAC
+	select SND_PCM
+	---help---
+	This option enables the new driver for the various
+	Apple Onboard Audio components.
+
+if SND_AOA
+
+source "sound/aoa/fabrics/Kconfig"
+
+source "sound/aoa/codecs/Kconfig"
+
+source "sound/aoa/soundbus/Kconfig"
+
+endif	# SND_AOA
diff --git a/ANDROID_3.4.5/sound/aoa/Makefile b/ANDROID_3.4.5/sound/aoa/Makefile
new file mode 100644
index 00000000..a8c037f9
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/Makefile
@@ -0,0 +1,4 @@
+obj-$(CONFIG_SND_AOA) += core/
+obj-$(CONFIG_SND_AOA_SOUNDBUS) += soundbus/
+obj-$(CONFIG_SND_AOA) += fabrics/
+obj-$(CONFIG_SND_AOA) += codecs/
diff --git a/ANDROID_3.4.5/sound/aoa/aoa-gpio.h b/ANDROID_3.4.5/sound/aoa/aoa-gpio.h
new file mode 100644
index 00000000..6065b034
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/aoa-gpio.h
@@ -0,0 +1,83 @@
+/*
+ * Apple Onboard Audio GPIO definitions
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+
+#ifndef __AOA_GPIO_H
+#define __AOA_GPIO_H
+#include <linux/workqueue.h>
+#include <linux/mutex.h>
+#include <asm/prom.h>
+
+typedef void (*notify_func_t)(void *data);
+
+enum notify_type {
+	AOA_NOTIFY_HEADPHONE,
+	AOA_NOTIFY_LINE_IN,
+	AOA_NOTIFY_LINE_OUT,
+};
+
+struct gpio_runtime;
+struct gpio_methods {
+	/* for initialisation/de-initialisation of the GPIO layer */
+	void (*init)(struct gpio_runtime *rt);
+	void (*exit)(struct gpio_runtime *rt);
+
+	/* turn off headphone, speakers, lineout */
+	void (*all_amps_off)(struct gpio_runtime *rt);
+	/* turn headphone, speakers, lineout back to previous setting */
+	void (*all_amps_restore)(struct gpio_runtime *rt);
+
+	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);
+
+	/* use this to be notified of any events. The notification
+	 * function is passed the data, and is called in process
+	 * context by the use of schedule_work.
+	 * The interface for it is that setting a function to NULL
+	 * removes it, and they return 0 if the operation succeeded,
+	 * and -EBUSY if the notification is already assigned by
+	 * someone else. */
+	int (*set_notify)(struct gpio_runtime *rt,
+			  enum notify_type type,
+			  notify_func_t notify,
+			  void *data);
+	/* returns 0 if not plugged in, 1 if plugged in
+	 * or a negative error code */
+	int (*get_detect)(struct gpio_runtime *rt,
+			  enum notify_type type);
+};
+
+struct gpio_notification {
+	struct delayed_work work;
+	notify_func_t notify;
+	void *data;
+	void *gpio_private;
+	struct mutex mutex;
+};
+
+struct gpio_runtime {
+	/* to be assigned by fabric */
+	struct device_node *node;
+	/* since everyone needs this pointer anyway... */
+	struct gpio_methods *methods;
+	/* to be used by the gpio implementation */
+	int implementation_private;
+	struct gpio_notification headphone_notify;
+	struct gpio_notification line_in_notify;
+	struct gpio_notification line_out_notify;
+};
+
+#endif /* __AOA_GPIO_H */
diff --git a/ANDROID_3.4.5/sound/aoa/aoa.h b/ANDROID_3.4.5/sound/aoa/aoa.h
new file mode 100644
index 00000000..e0878948
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/aoa.h
@@ -0,0 +1,129 @@
+/*
+ * Apple Onboard Audio definitions
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+
+#ifndef __AOA_H
+#define __AOA_H
+#include <asm/prom.h>
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/asound.h>
+#include <sound/control.h>
+#include "aoa-gpio.h"
+#include "soundbus/soundbus.h"
+
+#define MAX_CODEC_NAME_LEN	32
+
+struct aoa_codec {
+	char	name[MAX_CODEC_NAME_LEN];
+
+	struct module *owner;
+
+	/* called when the fabric wants to init this codec.
+	 * Do alsa card manipulations from here. */
+	int (*init)(struct aoa_codec *codec);
+
+	/* called when the fabric is done with the codec.
+	 * The alsa card will be cleaned up so don't bother. */
+	void (*exit)(struct aoa_codec *codec);
+
+	/* May be NULL, but can be used by the fabric.
+	 * Refcounting is the codec driver's responsibility */
+	struct device_node *node;
+
+	/* assigned by fabric before init() is called, points
+	 * to the soundbus device. Cannot be NULL. */
+	struct soundbus_dev *soundbus_dev;
+
+	/* assigned by the fabric before init() is called, points
+	 * to the fabric's gpio runtime record for the relevant
+	 * device. */
+	struct gpio_runtime *gpio;
+
+	/* assigned by the fabric before init() is called, contains
+	 * a codec specific bitmask of what outputs and inputs are
+	 * actually connected */
+	u32 connected;
+
+	/* data the fabric can associate with this structure */
+	void *fabric_data;
+
+	/* private! */
+	struct list_head list;
+	struct aoa_fabric *fabric;
+};
+
+/* return 0 on success */
+extern int
+aoa_codec_register(struct aoa_codec *codec);
+extern void
+aoa_codec_unregister(struct aoa_codec *codec);
+
+#define MAX_LAYOUT_NAME_LEN	32
+
+struct aoa_fabric {
+	char	name[MAX_LAYOUT_NAME_LEN];
+
+	struct module *owner;
+
+	/* once codecs register, they are passed here after.
+	 * They are of course not initialised, since the
+	 * fabric is responsible for initialising some fields
+	 * in the codec structure! */
+	int (*found_codec)(struct aoa_codec *codec);
+	/* called for each codec when it is removed,
+	 * also in the case that aoa_fabric_unregister
+	 * is called and all codecs are removed
+	 * from this fabric.
+	 * Also called if found_codec returned 0 but
+	 * the codec couldn't initialise. */
+	void (*remove_codec)(struct aoa_codec *codec);
+	/* If found_codec returned 0, and the codec
+	 * could be initialised, this is called. */
+	void (*attached_codec)(struct aoa_codec *codec);
+};
+
+/* return 0 on success, -EEXIST if another fabric is
+ * registered, -EALREADY if the same fabric is registered.
+ * Passing NULL can be used to test for the presence
+ * of another fabric, if -EALREADY is returned there is
+ * no other fabric present.
+ * In the case that the function returns -EALREADY
+ * and the fabric passed is not NULL, all codecs
+ * that are not assigned yet are passed to the fabric
+ * again for reconsideration. */
+extern int
+aoa_fabric_register(struct aoa_fabric *fabric, struct device *dev);
+
+/* it is vital to call this when the fabric exits!
+ * When calling, the remove_codec will be called
+ * for all codecs, unless it is NULL. */
+extern void
+aoa_fabric_unregister(struct aoa_fabric *fabric);
+
+/* if for some reason you want to get rid of a codec
+ * before the fabric is removed, use this.
+ * Note that remove_codec is called for it! */
+extern void
+aoa_fabric_unlink_codec(struct aoa_codec *codec);
+
+/* alsa help methods */
+struct aoa_card {
+	struct snd_card *alsa_card;
+};
+        
+extern int aoa_snd_device_new(snd_device_type_t type,
+	void * device_data, struct snd_device_ops * ops);
+extern struct snd_card *aoa_get_card(void);
+extern int aoa_snd_ctl_add(struct snd_kcontrol* control);
+
+/* GPIO stuff */
+extern struct gpio_methods *pmf_gpio_methods;
+extern struct gpio_methods *ftr_gpio_methods;
+/* extern struct gpio_methods *map_gpio_methods; */
+
+#endif /* __AOA_H */
diff --git a/ANDROID_3.4.5/sound/aoa/codecs/Kconfig b/ANDROID_3.4.5/sound/aoa/codecs/Kconfig
new file mode 100644
index 00000000..0c68e328
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/codecs/Kconfig
@@ -0,0 +1,24 @@
+config SND_AOA_ONYX
+	tristate "support Onyx chip"
+	select I2C
+	select I2C_POWERMAC
+	---help---
+	This option enables support for the Onyx (pcm3052)
+	codec chip found in the latest Apple machines
+	(most of those with digital audio output).
+
+config SND_AOA_TAS
+	tristate "support TAS chips"
+	select I2C
+	select I2C_POWERMAC
+	---help---
+	This option enables support for the tas chips
+	found in a lot of Apple Machines, especially
+	iBooks and PowerBooks without digital.
+
+config SND_AOA_TOONIE
+	tristate "support Toonie chip"
+	---help---
+	This option enables support for the toonie codec
+	found in the Mac Mini. If you have a Mac Mini and
+	want to hear sound, select this option.
diff --git a/ANDROID_3.4.5/sound/aoa/codecs/Makefile b/ANDROID_3.4.5/sound/aoa/codecs/Makefile
new file mode 100644
index 00000000..c3ee77fc
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/codecs/Makefile
@@ -0,0 +1,7 @@
+snd-aoa-codec-onyx-objs := onyx.o
+snd-aoa-codec-tas-objs := tas.o
+snd-aoa-codec-toonie-objs := toonie.o
+
+obj-$(CONFIG_SND_AOA_ONYX) += snd-aoa-codec-onyx.o
+obj-$(CONFIG_SND_AOA_TAS) += snd-aoa-codec-tas.o
+obj-$(CONFIG_SND_AOA_TOONIE) += snd-aoa-codec-toonie.o
diff --git a/ANDROID_3.4.5/sound/aoa/codecs/onyx.c b/ANDROID_3.4.5/sound/aoa/codecs/onyx.c
new file mode 100644
index 00000000..270790d3
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/codecs/onyx.c
@@ -0,0 +1,1135 @@
+/*
+ * Apple Onboard Audio driver for Onyx codec
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ *
+ *
+ * This is a driver for the pcm3052 codec chip (codenamed Onyx)
+ * that is present in newer Apple hardware (with digital output).
+ *
+ * The Onyx codec has the following connections (listed by the bit
+ * to be used in aoa_codec.connected):
+ *  0: analog output
+ *  1: digital output
+ *  2: line input
+ *  3: microphone input
+ * Note that even though I know of no machine that has for example
+ * the digital output connected but not the analog, I have handled
+ * all the different cases in the code so that this driver may serve
+ * as a good example of what to do.
+ *
+ * NOTE: This driver assumes that there's at most one chip to be
+ * 	 used with one alsa card, in form of creating all kinds
+ *	 of mixer elements without regard for their existence.
+ *	 But snd-aoa assumes that there's at most one card, so
+ *	 this means you can only have one onyx on a system. This
+ *	 should probably be fixed by changing the assumption of
+ *	 having just a single card on a system, and making the
+ *	 'card' pointer accessible to anyone who needs it instead
+ *	 of hiding it in the aoa_snd_* functions...
+ *
+ */
+#include <linux/delay.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("pcm3052 (onyx) codec driver for snd-aoa");
+
+#include "onyx.h"
+#include "../aoa.h"
+#include "../soundbus/soundbus.h"
+
+
+#define PFX "snd-aoa-codec-onyx: "
+
+struct onyx {
+	/* cache registers 65 to 80, they are write-only! */
+	u8			cache[16];
+	struct i2c_client	*i2c;
+	struct aoa_codec	codec;
+	u32			initialised:1,
+				spdif_locked:1,
+				analog_locked:1,
+				original_mute:2;
+	int			open_count;
+	struct codec_info	*codec_info;
+
+	/* mutex serializes concurrent access to the device
+	 * and this structure.
+	 */
+	struct mutex mutex;
+};
+#define codec_to_onyx(c) container_of(c, struct onyx, codec)
+
+/* both return 0 if all ok, else on error */
+static int onyx_read_register(struct onyx *onyx, u8 reg, u8 *value)
+{
+	s32 v;
+
+	if (reg != ONYX_REG_CONTROL) {
+		*value = onyx->cache[reg-FIRSTREGISTER];
+		return 0;
+	}
+	v = i2c_smbus_read_byte_data(onyx->i2c, reg);
+	if (v < 0)
+		return -1;
+	*value = (u8)v;
+	onyx->cache[ONYX_REG_CONTROL-FIRSTREGISTER] = *value;
+	return 0;
+}
+
+static int onyx_write_register(struct onyx *onyx, u8 reg, u8 value)
+{
+	int result;
+
+	result = i2c_smbus_write_byte_data(onyx->i2c, reg, value);
+	if (!result)
+		onyx->cache[reg-FIRSTREGISTER] = value;
+	return result;
+}
+
+/* alsa stuff */
+
+static int onyx_dev_register(struct snd_device *dev)
+{
+	return 0;
+}
+
+static struct snd_device_ops ops = {
+	.dev_register = onyx_dev_register,
+};
+
+/* this is necessary because most alsa mixer programs
+ * can't properly handle the negative range */
+#define VOLUME_RANGE_SHIFT	128
+
+static int onyx_snd_vol_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 = -128 + VOLUME_RANGE_SHIFT;
+	uinfo->value.integer.max = -1 + VOLUME_RANGE_SHIFT;
+	return 0;
+}
+
+static int onyx_snd_vol_get(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+	s8 l, r;
+
+	mutex_lock(&onyx->mutex);
+	onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, &l);
+	onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, &r);
+	mutex_unlock(&onyx->mutex);
+
+	ucontrol->value.integer.value[0] = l + VOLUME_RANGE_SHIFT;
+	ucontrol->value.integer.value[1] = r + VOLUME_RANGE_SHIFT;
+
+	return 0;
+}
+
+static int onyx_snd_vol_put(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+	s8 l, r;
+
+	if (ucontrol->value.integer.value[0] < -128 + VOLUME_RANGE_SHIFT ||
+	    ucontrol->value.integer.value[0] > -1 + VOLUME_RANGE_SHIFT)
+		return -EINVAL;
+	if (ucontrol->value.integer.value[1] < -128 + VOLUME_RANGE_SHIFT ||
+	    ucontrol->value.integer.value[1] > -1 + VOLUME_RANGE_SHIFT)
+		return -EINVAL;
+
+	mutex_lock(&onyx->mutex);
+	onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, &l);
+	onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, &r);
+
+	if (l + VOLUME_RANGE_SHIFT == ucontrol->value.integer.value[0] &&
+	    r + VOLUME_RANGE_SHIFT == ucontrol->value.integer.value[1]) {
+		mutex_unlock(&onyx->mutex);
+		return 0;
+	}
+
+	onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_LEFT,
+			    ucontrol->value.integer.value[0]
+			     - VOLUME_RANGE_SHIFT);
+	onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT,
+			    ucontrol->value.integer.value[1]
+			     - VOLUME_RANGE_SHIFT);
+	mutex_unlock(&onyx->mutex);
+
+	return 1;
+}
+
+static struct snd_kcontrol_new volume_control = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Master Playback Volume",
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.info = onyx_snd_vol_info,
+	.get = onyx_snd_vol_get,
+	.put = onyx_snd_vol_put,
+};
+
+/* like above, this is necessary because a lot
+ * of alsa mixer programs don't handle ranges
+ * that don't start at 0 properly.
+ * even alsamixer is one of them... */
+#define INPUTGAIN_RANGE_SHIFT	(-3)
+
+static int onyx_snd_inputgain_info(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 3 + INPUTGAIN_RANGE_SHIFT;
+	uinfo->value.integer.max = 28 + INPUTGAIN_RANGE_SHIFT;
+	return 0;
+}
+
+static int onyx_snd_inputgain_get(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+	u8 ig;
+
+	mutex_lock(&onyx->mutex);
+	onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &ig);
+	mutex_unlock(&onyx->mutex);
+
+	ucontrol->value.integer.value[0] =
+		(ig & ONYX_ADC_PGA_GAIN_MASK) + INPUTGAIN_RANGE_SHIFT;
+
+	return 0;
+}
+
+static int onyx_snd_inputgain_put(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+	u8 v, n;
+
+	if (ucontrol->value.integer.value[0] < 3 + INPUTGAIN_RANGE_SHIFT ||
+	    ucontrol->value.integer.value[0] > 28 + INPUTGAIN_RANGE_SHIFT)
+		return -EINVAL;
+	mutex_lock(&onyx->mutex);
+	onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &v);
+	n = v;
+	n &= ~ONYX_ADC_PGA_GAIN_MASK;
+	n |= (ucontrol->value.integer.value[0] - INPUTGAIN_RANGE_SHIFT)
+		& ONYX_ADC_PGA_GAIN_MASK;
+	onyx_write_register(onyx, ONYX_REG_ADC_CONTROL, n);
+	mutex_unlock(&onyx->mutex);
+
+	return n != v;
+}
+
+static struct snd_kcontrol_new inputgain_control = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Master Capture Volume",
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.info = onyx_snd_inputgain_info,
+	.get = onyx_snd_inputgain_get,
+	.put = onyx_snd_inputgain_put,
+};
+
+static int onyx_snd_capture_source_info(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo)
+{
+	static char *texts[] = { "Line-In", "Microphone" };
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = 2;
+	if (uinfo->value.enumerated.item > 1)
+		uinfo->value.enumerated.item = 1;
+	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+	return 0;
+}
+
+static int onyx_snd_capture_source_get(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+	s8 v;
+
+	mutex_lock(&onyx->mutex);
+	onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &v);
+	mutex_unlock(&onyx->mutex);
+
+	ucontrol->value.enumerated.item[0] = !!(v&ONYX_ADC_INPUT_MIC);
+
+	return 0;
+}
+
+static void onyx_set_capture_source(struct onyx *onyx, int mic)
+{
+	s8 v;
+
+	mutex_lock(&onyx->mutex);
+	onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &v);
+	v &= ~ONYX_ADC_INPUT_MIC;
+	if (mic)
+		v |= ONYX_ADC_INPUT_MIC;
+	onyx_write_register(onyx, ONYX_REG_ADC_CONTROL, v);
+	mutex_unlock(&onyx->mutex);
+}
+
+static int onyx_snd_capture_source_put(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	if (ucontrol->value.enumerated.item[0] > 1)
+		return -EINVAL;
+	onyx_set_capture_source(snd_kcontrol_chip(kcontrol),
+				ucontrol->value.enumerated.item[0]);
+	return 1;
+}
+
+static struct snd_kcontrol_new capture_source_control = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	/* If we name this 'Input Source', it properly shows up in
+	 * alsamixer as a selection, * but it's shown under the
+	 * 'Playback' category.
+	 * If I name it 'Capture Source', it shows up in strange
+	 * ways (two bools of which one can be selected at a
+	 * time) but at least it's shown in the 'Capture'
+	 * category.
+	 * I was told that this was due to backward compatibility,
+	 * but I don't understand then why the mangling is *not*
+	 * done when I name it "Input Source".....
+	 */
+	.name = "Capture Source",
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.info = onyx_snd_capture_source_info,
+	.get = onyx_snd_capture_source_get,
+	.put = onyx_snd_capture_source_put,
+};
+
+#define onyx_snd_mute_info	snd_ctl_boolean_stereo_info
+
+static int onyx_snd_mute_get(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+	u8 c;
+
+	mutex_lock(&onyx->mutex);
+	onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &c);
+	mutex_unlock(&onyx->mutex);
+
+	ucontrol->value.integer.value[0] = !(c & ONYX_MUTE_LEFT);
+	ucontrol->value.integer.value[1] = !(c & ONYX_MUTE_RIGHT);
+
+	return 0;
+}
+
+static int onyx_snd_mute_put(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+	u8 v = 0, c = 0;
+	int err = -EBUSY;
+
+	mutex_lock(&onyx->mutex);
+	if (onyx->analog_locked)
+		goto out_unlock;
+
+	onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v);
+	c = v;
+	c &= ~(ONYX_MUTE_RIGHT | ONYX_MUTE_LEFT);
+	if (!ucontrol->value.integer.value[0])
+		c |= ONYX_MUTE_LEFT;
+	if (!ucontrol->value.integer.value[1])
+		c |= ONYX_MUTE_RIGHT;
+	err = onyx_write_register(onyx, ONYX_REG_DAC_CONTROL, c);
+
+ out_unlock:
+	mutex_unlock(&onyx->mutex);
+
+	return !err ? (v != c) : err;
+}
+
+static struct snd_kcontrol_new mute_control = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Master Playback Switch",
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.info = onyx_snd_mute_info,
+	.get = onyx_snd_mute_get,
+	.put = onyx_snd_mute_put,
+};
+
+
+#define onyx_snd_single_bit_info	snd_ctl_boolean_mono_info
+
+#define FLAG_POLARITY_INVERT	1
+#define FLAG_SPDIFLOCK		2
+
+static int onyx_snd_single_bit_get(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+	u8 c;
+	long int pv = kcontrol->private_value;
+	u8 polarity = (pv >> 16) & FLAG_POLARITY_INVERT;
+	u8 address = (pv >> 8) & 0xff;
+	u8 mask = pv & 0xff;
+
+	mutex_lock(&onyx->mutex);
+	onyx_read_register(onyx, address, &c);
+	mutex_unlock(&onyx->mutex);
+
+	ucontrol->value.integer.value[0] = !!(c & mask) ^ polarity;
+
+	return 0;
+}
+
+static int onyx_snd_single_bit_put(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+	u8 v = 0, c = 0;
+	int err;
+	long int pv = kcontrol->private_value;
+	u8 polarity = (pv >> 16) & FLAG_POLARITY_INVERT;
+	u8 spdiflock = (pv >> 16) & FLAG_SPDIFLOCK;
+	u8 address = (pv >> 8) & 0xff;
+	u8 mask = pv & 0xff;
+
+	mutex_lock(&onyx->mutex);
+	if (spdiflock && onyx->spdif_locked) {
+		/* even if alsamixer doesn't care.. */
+		err = -EBUSY;
+		goto out_unlock;
+	}
+	onyx_read_register(onyx, address, &v);
+	c = v;
+	c &= ~(mask);
+	if (!!ucontrol->value.integer.value[0] ^ polarity)
+		c |= mask;
+	err = onyx_write_register(onyx, address, c);
+
+ out_unlock:
+	mutex_unlock(&onyx->mutex);
+
+	return !err ? (v != c) : err;
+}
+
+#define SINGLE_BIT(n, type, description, address, mask, flags)	 	\
+static struct snd_kcontrol_new n##_control = {				\
+	.iface = SNDRV_CTL_ELEM_IFACE_##type,				\
+	.name = description,						\
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,			\
+	.info = onyx_snd_single_bit_info,				\
+	.get = onyx_snd_single_bit_get,					\
+	.put = onyx_snd_single_bit_put,					\
+	.private_value = (flags << 16) | (address << 8) | mask		\
+}
+
+SINGLE_BIT(spdif,
+	   MIXER,
+	   SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH),
+	   ONYX_REG_DIG_INFO4,
+	   ONYX_SPDIF_ENABLE,
+	   FLAG_SPDIFLOCK);
+SINGLE_BIT(ovr1,
+	   MIXER,
+	   "Oversampling Rate",
+	   ONYX_REG_DAC_CONTROL,
+	   ONYX_OVR1,
+	   0);
+SINGLE_BIT(flt0,
+	   MIXER,
+	   "Fast Digital Filter Rolloff",
+	   ONYX_REG_DAC_FILTER,
+	   ONYX_ROLLOFF_FAST,
+	   FLAG_POLARITY_INVERT);
+SINGLE_BIT(hpf,
+	   MIXER,
+	   "Highpass Filter",
+	   ONYX_REG_ADC_HPF_BYPASS,
+	   ONYX_HPF_DISABLE,
+	   FLAG_POLARITY_INVERT);
+SINGLE_BIT(dm12,
+	   MIXER,
+	   "Digital De-Emphasis",
+	   ONYX_REG_DAC_DEEMPH,
+	   ONYX_DIGDEEMPH_CTRL,
+	   0);
+
+static int onyx_spdif_info(struct snd_kcontrol *kcontrol,
+			   struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+	uinfo->count = 1;
+	return 0;
+}
+
+static int onyx_spdif_mask_get(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	/* datasheet page 30, all others are 0 */
+	ucontrol->value.iec958.status[0] = 0x3e;
+	ucontrol->value.iec958.status[1] = 0xff;
+
+	ucontrol->value.iec958.status[3] = 0x3f;
+	ucontrol->value.iec958.status[4] = 0x0f;
+
+	return 0;
+}
+
+static struct snd_kcontrol_new onyx_spdif_mask = {
+	.access =	SNDRV_CTL_ELEM_ACCESS_READ,
+	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
+	.name =		SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK),
+	.info =		onyx_spdif_info,
+	.get =		onyx_spdif_mask_get,
+};
+
+static int onyx_spdif_get(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+	u8 v;
+
+	mutex_lock(&onyx->mutex);
+	onyx_read_register(onyx, ONYX_REG_DIG_INFO1, &v);
+	ucontrol->value.iec958.status[0] = v & 0x3e;
+
+	onyx_read_register(onyx, ONYX_REG_DIG_INFO2, &v);
+	ucontrol->value.iec958.status[1] = v;
+
+	onyx_read_register(onyx, ONYX_REG_DIG_INFO3, &v);
+	ucontrol->value.iec958.status[3] = v & 0x3f;
+
+	onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v);
+	ucontrol->value.iec958.status[4] = v & 0x0f;
+	mutex_unlock(&onyx->mutex);
+
+	return 0;
+}
+
+static int onyx_spdif_put(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	struct onyx *onyx = snd_kcontrol_chip(kcontrol);
+	u8 v;
+
+	mutex_lock(&onyx->mutex);
+	onyx_read_register(onyx, ONYX_REG_DIG_INFO1, &v);
+	v = (v & ~0x3e) | (ucontrol->value.iec958.status[0] & 0x3e);
+	onyx_write_register(onyx, ONYX_REG_DIG_INFO1, v);
+
+	v = ucontrol->value.iec958.status[1];
+	onyx_write_register(onyx, ONYX_REG_DIG_INFO2, v);
+
+	onyx_read_register(onyx, ONYX_REG_DIG_INFO3, &v);
+	v = (v & ~0x3f) | (ucontrol->value.iec958.status[3] & 0x3f);
+	onyx_write_register(onyx, ONYX_REG_DIG_INFO3, v);
+
+	onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v);
+	v = (v & ~0x0f) | (ucontrol->value.iec958.status[4] & 0x0f);
+	onyx_write_register(onyx, ONYX_REG_DIG_INFO4, v);
+	mutex_unlock(&onyx->mutex);
+
+	return 1;
+}
+
+static struct snd_kcontrol_new onyx_spdif_ctrl = {
+	.access =	SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.iface =	SNDRV_CTL_ELEM_IFACE_PCM,
+	.name =		SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
+	.info =		onyx_spdif_info,
+	.get =		onyx_spdif_get,
+	.put =		onyx_spdif_put,
+};
+
+/* our registers */
+
+static u8 register_map[] = {
+	ONYX_REG_DAC_ATTEN_LEFT,
+	ONYX_REG_DAC_ATTEN_RIGHT,
+	ONYX_REG_CONTROL,
+	ONYX_REG_DAC_CONTROL,
+	ONYX_REG_DAC_DEEMPH,
+	ONYX_REG_DAC_FILTER,
+	ONYX_REG_DAC_OUTPHASE,
+	ONYX_REG_ADC_CONTROL,
+	ONYX_REG_ADC_HPF_BYPASS,
+	ONYX_REG_DIG_INFO1,
+	ONYX_REG_DIG_INFO2,
+	ONYX_REG_DIG_INFO3,
+	ONYX_REG_DIG_INFO4
+};
+
+static u8 initial_values[ARRAY_SIZE(register_map)] = {
+	0x80, 0x80, /* muted */
+	ONYX_MRST | ONYX_SRST, /* but handled specially! */
+	ONYX_MUTE_LEFT | ONYX_MUTE_RIGHT,
+	0, /* no deemphasis */
+	ONYX_DAC_FILTER_ALWAYS,
+	ONYX_OUTPHASE_INVERTED,
+	(-1 /*dB*/ + 8) & 0xF, /* line in selected, -1 dB gain*/
+	ONYX_ADC_HPF_ALWAYS,
+	(1<<2),	/* pcm audio */
+	2,	/* category: pcm coder */
+	0,	/* sampling frequency 44.1 kHz, clock accuracy level II */
+	1	/* 24 bit depth */
+};
+
+/* reset registers of chip, either to initial or to previous values */
+static int onyx_register_init(struct onyx *onyx)
+{
+	int i;
+	u8 val;
+	u8 regs[sizeof(initial_values)];
+
+	if (!onyx->initialised) {
+		memcpy(regs, initial_values, sizeof(initial_values));
+		if (onyx_read_register(onyx, ONYX_REG_CONTROL, &val))
+			return -1;
+		val &= ~ONYX_SILICONVERSION;
+		val |= initial_values[3];
+		regs[3] = val;
+	} else {
+		for (i=0; i<sizeof(register_map); i++)
+			regs[i] = onyx->cache[register_map[i]-FIRSTREGISTER];
+	}
+
+	for (i=0; i<sizeof(register_map); i++) {
+		if (onyx_write_register(onyx, register_map[i], regs[i]))
+			return -1;
+	}
+	onyx->initialised = 1;
+	return 0;
+}
+
+static struct transfer_info onyx_transfers[] = {
+	/* this is first so we can skip it if no input is present...
+	 * No hardware exists with that, but it's here as an example
+	 * of what to do :) */
+	{
+		/* analog input */
+		.formats = SNDRV_PCM_FMTBIT_S8 |
+			   SNDRV_PCM_FMTBIT_S16_BE |
+			   SNDRV_PCM_FMTBIT_S24_BE,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.transfer_in = 1,
+		.must_be_clock_source = 0,
+		.tag = 0,
+	},
+	{
+		/* if analog and digital are currently off, anything should go,
+		 * so this entry describes everything we can do... */
+		.formats = SNDRV_PCM_FMTBIT_S8 |
+			   SNDRV_PCM_FMTBIT_S16_BE |
+			   SNDRV_PCM_FMTBIT_S24_BE
+#ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE
+			   | SNDRV_PCM_FMTBIT_COMPRESSED_16BE
+#endif
+		,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.tag = 0,
+	},
+	{
+		/* analog output */
+		.formats = SNDRV_PCM_FMTBIT_S8 |
+			   SNDRV_PCM_FMTBIT_S16_BE |
+			   SNDRV_PCM_FMTBIT_S24_BE,
+		.rates = SNDRV_PCM_RATE_8000_96000,
+		.transfer_in = 0,
+		.must_be_clock_source = 0,
+		.tag = 1,
+	},
+	{
+		/* digital pcm output, also possible for analog out */
+		.formats = SNDRV_PCM_FMTBIT_S8 |
+			   SNDRV_PCM_FMTBIT_S16_BE |
+			   SNDRV_PCM_FMTBIT_S24_BE,
+		.rates = SNDRV_PCM_RATE_32000 |
+			 SNDRV_PCM_RATE_44100 |
+			 SNDRV_PCM_RATE_48000,
+		.transfer_in = 0,
+		.must_be_clock_source = 0,
+		.tag = 2,
+	},
+#ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE
+	/* Once alsa gets supports for this kind of thing we can add it... */
+	{
+		/* digital compressed output */
+		.formats =  SNDRV_PCM_FMTBIT_COMPRESSED_16BE,
+		.rates = SNDRV_PCM_RATE_32000 |
+			 SNDRV_PCM_RATE_44100 |
+			 SNDRV_PCM_RATE_48000,
+		.tag = 2,
+	},
+#endif
+	{}
+};
+
+static int onyx_usable(struct codec_info_item *cii,
+		       struct transfer_info *ti,
+		       struct transfer_info *out)
+{
+	u8 v;
+	struct onyx *onyx = cii->codec_data;
+	int spdif_enabled, analog_enabled;
+
+	mutex_lock(&onyx->mutex);
+	onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v);
+	spdif_enabled = !!(v & ONYX_SPDIF_ENABLE);
+	onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v);
+	analog_enabled =
+		(v & (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT))
+		 != (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT);
+	mutex_unlock(&onyx->mutex);
+
+	switch (ti->tag) {
+	case 0: return 1;
+	case 1:	return analog_enabled;
+	case 2: return spdif_enabled;
+	}
+	return 1;
+}
+
+static int onyx_prepare(struct codec_info_item *cii,
+			struct bus_info *bi,
+			struct snd_pcm_substream *substream)
+{
+	u8 v;
+	struct onyx *onyx = cii->codec_data;
+	int err = -EBUSY;
+
+	mutex_lock(&onyx->mutex);
+
+#ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE
+	if (substream->runtime->format == SNDRV_PCM_FMTBIT_COMPRESSED_16BE) {
+		/* mute and lock analog output */
+		onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v);
+		if (onyx_write_register(onyx,
+					ONYX_REG_DAC_CONTROL,
+					v | ONYX_MUTE_RIGHT | ONYX_MUTE_LEFT))
+			goto out_unlock;
+		onyx->analog_locked = 1;
+		err = 0;
+		goto out_unlock;
+	}
+#endif
+	switch (substream->runtime->rate) {
+	case 32000:
+	case 44100:
+	case 48000:
+		/* these rates are ok for all outputs */
+		/* FIXME: program spdif channel control bits here so that
+		 *	  userspace doesn't have to if it only plays pcm! */
+		err = 0;
+		goto out_unlock;
+	default:
+		/* got some rate that the digital output can't do,
+		 * so disable and lock it */
+		onyx_read_register(cii->codec_data, ONYX_REG_DIG_INFO4, &v);
+		if (onyx_write_register(onyx,
+					ONYX_REG_DIG_INFO4,
+					v & ~ONYX_SPDIF_ENABLE))
+			goto out_unlock;
+		onyx->spdif_locked = 1;
+		err = 0;
+		goto out_unlock;
+	}
+
+ out_unlock:
+	mutex_unlock(&onyx->mutex);
+
+	return err;
+}
+
+static int onyx_open(struct codec_info_item *cii,
+		     struct snd_pcm_substream *substream)
+{
+	struct onyx *onyx = cii->codec_data;
+
+	mutex_lock(&onyx->mutex);
+	onyx->open_count++;
+	mutex_unlock(&onyx->mutex);
+
+	return 0;
+}
+
+static int onyx_close(struct codec_info_item *cii,
+		      struct snd_pcm_substream *substream)
+{
+	struct onyx *onyx = cii->codec_data;
+
+	mutex_lock(&onyx->mutex);
+	onyx->open_count--;
+	if (!onyx->open_count)
+		onyx->spdif_locked = onyx->analog_locked = 0;
+	mutex_unlock(&onyx->mutex);
+
+	return 0;
+}
+
+static int onyx_switch_clock(struct codec_info_item *cii,
+			     enum clock_switch what)
+{
+	struct onyx *onyx = cii->codec_data;
+
+	mutex_lock(&onyx->mutex);
+	/* this *MUST* be more elaborate later... */
+	switch (what) {
+	case CLOCK_SWITCH_PREPARE_SLAVE:
+		onyx->codec.gpio->methods->all_amps_off(onyx->codec.gpio);
+		break;
+	case CLOCK_SWITCH_SLAVE:
+		onyx->codec.gpio->methods->all_amps_restore(onyx->codec.gpio);
+		break;
+	default: /* silence warning */
+		break;
+	}
+	mutex_unlock(&onyx->mutex);
+
+	return 0;
+}
+
+#ifdef CONFIG_PM
+
+static int onyx_suspend(struct codec_info_item *cii, pm_message_t state)
+{
+	struct onyx *onyx = cii->codec_data;
+	u8 v;
+	int err = -ENXIO;
+
+	mutex_lock(&onyx->mutex);
+	if (onyx_read_register(onyx, ONYX_REG_CONTROL, &v))
+		goto out_unlock;
+	onyx_write_register(onyx, ONYX_REG_CONTROL, v | ONYX_ADPSV | ONYX_DAPSV);
+	/* Apple does a sleep here but the datasheet says to do it on resume */
+	err = 0;
+ out_unlock:
+	mutex_unlock(&onyx->mutex);
+
+	return err;
+}
+
+static int onyx_resume(struct codec_info_item *cii)
+{
+	struct onyx *onyx = cii->codec_data;
+	u8 v;
+	int err = -ENXIO;
+
+	mutex_lock(&onyx->mutex);
+
+	/* reset codec */
+	onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0);
+	msleep(1);
+	onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 1);
+	msleep(1);
+	onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0);
+	msleep(1);
+
+	/* take codec out of suspend (if it still is after reset) */
+	if (onyx_read_register(onyx, ONYX_REG_CONTROL, &v))
+		goto out_unlock;
+	onyx_write_register(onyx, ONYX_REG_CONTROL, v & ~(ONYX_ADPSV | ONYX_DAPSV));
+	/* FIXME: should divide by sample rate, but 8k is the lowest we go */
+	msleep(2205000/8000);
+	/* reset all values */
+	onyx_register_init(onyx);
+	err = 0;
+ out_unlock:
+	mutex_unlock(&onyx->mutex);
+
+	return err;
+}
+
+#endif /* CONFIG_PM */
+
+static struct codec_info onyx_codec_info = {
+	.transfers = onyx_transfers,
+	.sysclock_factor = 256,
+	.bus_factor = 64,
+	.owner = THIS_MODULE,
+	.usable = onyx_usable,
+	.prepare = onyx_prepare,
+	.open = onyx_open,
+	.close = onyx_close,
+	.switch_clock = onyx_switch_clock,
+#ifdef CONFIG_PM
+	.suspend = onyx_suspend,
+	.resume = onyx_resume,
+#endif
+};
+
+static int onyx_init_codec(struct aoa_codec *codec)
+{
+	struct onyx *onyx = codec_to_onyx(codec);
+	struct snd_kcontrol *ctl;
+	struct codec_info *ci = &onyx_codec_info;
+	u8 v;
+	int err;
+
+	if (!onyx->codec.gpio || !onyx->codec.gpio->methods) {
+		printk(KERN_ERR PFX "gpios not assigned!!\n");
+		return -EINVAL;
+	}
+
+	onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0);
+	msleep(1);
+	onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 1);
+	msleep(1);
+	onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0);
+	msleep(1);
+
+	if (onyx_register_init(onyx)) {
+		printk(KERN_ERR PFX "failed to initialise onyx registers\n");
+		return -ENODEV;
+	}
+
+	if (aoa_snd_device_new(SNDRV_DEV_LOWLEVEL, onyx, &ops)) {
+		printk(KERN_ERR PFX "failed to create onyx snd device!\n");
+		return -ENODEV;
+	}
+
+	/* nothing connected? what a joke! */
+	if ((onyx->codec.connected & 0xF) == 0)
+		return -ENOTCONN;
+
+	/* if no inputs are present... */
+	if ((onyx->codec.connected & 0xC) == 0) {
+		if (!onyx->codec_info)
+			onyx->codec_info = kmalloc(sizeof(struct codec_info), GFP_KERNEL);
+		if (!onyx->codec_info)
+			return -ENOMEM;
+		ci = onyx->codec_info;
+		*ci = onyx_codec_info;
+		ci->transfers++;
+	}
+
+	/* if no outputs are present... */
+	if ((onyx->codec.connected & 3) == 0) {
+		if (!onyx->codec_info)
+			onyx->codec_info = kmalloc(sizeof(struct codec_info), GFP_KERNEL);
+		if (!onyx->codec_info)
+			return -ENOMEM;
+		ci = onyx->codec_info;
+		/* this is fine as there have to be inputs
+		 * if we end up in this part of the code */
+		*ci = onyx_codec_info;
+		ci->transfers[1].formats = 0;
+	}
+
+	if (onyx->codec.soundbus_dev->attach_codec(onyx->codec.soundbus_dev,
+						   aoa_get_card(),
+						   ci, onyx)) {
+		printk(KERN_ERR PFX "error creating onyx pcm\n");
+		return -ENODEV;
+	}
+#define ADDCTL(n)							\
+	do {								\
+		ctl = snd_ctl_new1(&n, onyx);				\
+		if (ctl) {						\
+			ctl->id.device =				\
+				onyx->codec.soundbus_dev->pcm->device;	\
+			err = aoa_snd_ctl_add(ctl);			\
+			if (err)					\
+				goto error;				\
+		}							\
+	} while (0)
+
+	if (onyx->codec.soundbus_dev->pcm) {
+		/* give the user appropriate controls
+		 * depending on what inputs are connected */
+		if ((onyx->codec.connected & 0xC) == 0xC)
+			ADDCTL(capture_source_control);
+		else if (onyx->codec.connected & 4)
+			onyx_set_capture_source(onyx, 0);
+		else
+			onyx_set_capture_source(onyx, 1);
+		if (onyx->codec.connected & 0xC)
+			ADDCTL(inputgain_control);
+
+		/* depending on what output is connected,
+		 * give the user appropriate controls */
+		if (onyx->codec.connected & 1) {
+			ADDCTL(volume_control);
+			ADDCTL(mute_control);
+			ADDCTL(ovr1_control);
+			ADDCTL(flt0_control);
+			ADDCTL(hpf_control);
+			ADDCTL(dm12_control);
+			/* spdif control defaults to off */
+		}
+		if (onyx->codec.connected & 2) {
+			ADDCTL(onyx_spdif_mask);
+			ADDCTL(onyx_spdif_ctrl);
+		}
+		if ((onyx->codec.connected & 3) == 3)
+			ADDCTL(spdif_control);
+		/* if only S/PDIF is connected, enable it unconditionally */
+		if ((onyx->codec.connected & 3) == 2) {
+			onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v);
+			v |= ONYX_SPDIF_ENABLE;
+			onyx_write_register(onyx, ONYX_REG_DIG_INFO4, v);
+		}
+	}
+#undef ADDCTL
+	printk(KERN_INFO PFX "attached to onyx codec via i2c\n");
+
+	return 0;
+ error:
+	onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx);
+	snd_device_free(aoa_get_card(), onyx);
+	return err;
+}
+
+static void onyx_exit_codec(struct aoa_codec *codec)
+{
+	struct onyx *onyx = codec_to_onyx(codec);
+
+	if (!onyx->codec.soundbus_dev) {
+		printk(KERN_ERR PFX "onyx_exit_codec called without soundbus_dev!\n");
+		return;
+	}
+	onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx);
+}
+
+static int onyx_create(struct i2c_adapter *adapter,
+		       struct device_node *node,
+		       int addr)
+{
+	struct i2c_board_info info;
+	struct i2c_client *client;
+
+	memset(&info, 0, sizeof(struct i2c_board_info));
+	strlcpy(info.type, "aoa_codec_onyx", I2C_NAME_SIZE);
+	info.addr = addr;
+	info.platform_data = node;
+	client = i2c_new_device(adapter, &info);
+	if (!client)
+		return -ENODEV;
+
+	/*
+	 * We know the driver is already loaded, so the device should be
+	 * already bound. If not it means binding failed, which suggests
+	 * the device doesn't really exist and should be deleted.
+	 * Ideally this would be replaced by better checks _before_
+	 * instantiating the device.
+	 */
+	if (!client->driver) {
+		i2c_unregister_device(client);
+		return -ENODEV;
+	}
+
+	/*
+	 * Let i2c-core delete that device on driver removal.
+	 * This is safe because i2c-core holds the core_lock mutex for us.
+	 */
+	list_add_tail(&client->detected, &client->driver->clients);
+	return 0;
+}
+
+static int onyx_i2c_probe(struct i2c_client *client,
+			  const struct i2c_device_id *id)
+{
+	struct device_node *node = client->dev.platform_data;
+	struct onyx *onyx;
+	u8 dummy;
+
+	onyx = kzalloc(sizeof(struct onyx), GFP_KERNEL);
+
+	if (!onyx)
+		return -ENOMEM;
+
+	mutex_init(&onyx->mutex);
+	onyx->i2c = client;
+	i2c_set_clientdata(client, onyx);
+
+	/* we try to read from register ONYX_REG_CONTROL
+	 * to check if the codec is present */
+	if (onyx_read_register(onyx, ONYX_REG_CONTROL, &dummy) != 0) {
+		printk(KERN_ERR PFX "failed to read control register\n");
+		goto fail;
+	}
+
+	strlcpy(onyx->codec.name, "onyx", MAX_CODEC_NAME_LEN);
+	onyx->codec.owner = THIS_MODULE;
+	onyx->codec.init = onyx_init_codec;
+	onyx->codec.exit = onyx_exit_codec;
+	onyx->codec.node = of_node_get(node);
+
+	if (aoa_codec_register(&onyx->codec)) {
+		goto fail;
+	}
+	printk(KERN_DEBUG PFX "created and attached onyx instance\n");
+	return 0;
+ fail:
+	kfree(onyx);
+	return -ENODEV;
+}
+
+static int onyx_i2c_attach(struct i2c_adapter *adapter)
+{
+	struct device_node *busnode, *dev = NULL;
+	struct pmac_i2c_bus *bus;
+
+	bus = pmac_i2c_adapter_to_bus(adapter);
+	if (bus == NULL)
+		return -ENODEV;
+	busnode = pmac_i2c_get_bus_node(bus);
+
+	while ((dev = of_get_next_child(busnode, dev)) != NULL) {
+		if (of_device_is_compatible(dev, "pcm3052")) {
+			const u32 *addr;
+			printk(KERN_DEBUG PFX "found pcm3052\n");
+			addr = of_get_property(dev, "reg", NULL);
+			if (!addr)
+				return -ENODEV;
+			return onyx_create(adapter, dev, (*addr)>>1);
+		}
+	}
+
+	/* if that didn't work, try desperate mode for older
+	 * machines that have stuff missing from the device tree */
+
+	if (!of_device_is_compatible(busnode, "k2-i2c"))
+		return -ENODEV;
+
+	printk(KERN_DEBUG PFX "found k2-i2c, checking if onyx chip is on it\n");
+	/* probe both possible addresses for the onyx chip */
+	if (onyx_create(adapter, NULL, 0x46) == 0)
+		return 0;
+	return onyx_create(adapter, NULL, 0x47);
+}
+
+static int onyx_i2c_remove(struct i2c_client *client)
+{
+	struct onyx *onyx = i2c_get_clientdata(client);
+
+	aoa_codec_unregister(&onyx->codec);
+	of_node_put(onyx->codec.node);
+	kfree(onyx->codec_info);
+	kfree(onyx);
+	return 0;
+}
+
+static const struct i2c_device_id onyx_i2c_id[] = {
+	{ "aoa_codec_onyx", 0 },
+	{ }
+};
+
+static struct i2c_driver onyx_driver = {
+	.driver = {
+		.name = "aoa_codec_onyx",
+		.owner = THIS_MODULE,
+	},
+	.attach_adapter = onyx_i2c_attach,
+	.probe = onyx_i2c_probe,
+	.remove = onyx_i2c_remove,
+	.id_table = onyx_i2c_id,
+};
+
+module_i2c_driver(onyx_driver);
diff --git a/ANDROID_3.4.5/sound/aoa/codecs/onyx.h b/ANDROID_3.4.5/sound/aoa/codecs/onyx.h
new file mode 100644
index 00000000..ffd20254
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/codecs/onyx.h
@@ -0,0 +1,75 @@
+/*
+ * Apple Onboard Audio driver for Onyx codec (header)
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+#ifndef __SND_AOA_CODEC_ONYX_H
+#define __SND_AOA_CODEC_ONYX_H
+#include <stddef.h>
+#include <linux/i2c.h>
+#include <asm/pmac_low_i2c.h>
+#include <asm/prom.h>
+
+/* PCM3052 register definitions */
+
+/* the attenuation registers take values from
+ * -1 (0dB) to -127 (-63.0 dB) or others (muted) */
+#define ONYX_REG_DAC_ATTEN_LEFT		65
+#define FIRSTREGISTER			ONYX_REG_DAC_ATTEN_LEFT
+#define ONYX_REG_DAC_ATTEN_RIGHT	66
+
+#define ONYX_REG_CONTROL		67
+#	define ONYX_MRST		(1<<7)
+#	define ONYX_SRST		(1<<6)
+#	define ONYX_ADPSV		(1<<5)
+#	define ONYX_DAPSV		(1<<4)
+#	define ONYX_SILICONVERSION	(1<<0)
+/* all others reserved */
+
+#define ONYX_REG_DAC_CONTROL		68
+#	define ONYX_OVR1		(1<<6)
+#	define ONYX_MUTE_RIGHT		(1<<1)
+#	define ONYX_MUTE_LEFT		(1<<0)
+
+#define ONYX_REG_DAC_DEEMPH		69
+#	define ONYX_DIGDEEMPH_SHIFT	5
+#	define ONYX_DIGDEEMPH_MASK	(3<<ONYX_DIGDEEMPH_SHIFT)
+#	define ONYX_DIGDEEMPH_CTRL	(1<<4)
+
+#define ONYX_REG_DAC_FILTER		70
+#	define ONYX_ROLLOFF_FAST	(1<<5)
+#	define ONYX_DAC_FILTER_ALWAYS	(1<<2)
+
+#define	ONYX_REG_DAC_OUTPHASE		71
+#	define ONYX_OUTPHASE_INVERTED	(1<<0)
+
+#define ONYX_REG_ADC_CONTROL		72
+#	define ONYX_ADC_INPUT_MIC	(1<<5)
+/* 8 + input gain in dB, valid range for input gain is -4 .. 20 dB */
+#	define ONYX_ADC_PGA_GAIN_MASK	0x1f
+
+#define ONYX_REG_ADC_HPF_BYPASS		75
+#	define ONYX_HPF_DISABLE		(1<<3)
+#	define ONYX_ADC_HPF_ALWAYS	(1<<2)
+
+#define ONYX_REG_DIG_INFO1		77
+#	define ONYX_MASK_DIN_TO_BPZ	(1<<7)
+/* bits 1-5 control channel bits 1-5 */
+#	define ONYX_DIGOUT_DISABLE	(1<<0)
+
+#define ONYX_REG_DIG_INFO2		78
+/* controls channel bits 8-15 */
+
+#define ONYX_REG_DIG_INFO3		79
+/* control channel bits 24-29, high 2 bits reserved */
+
+#define ONYX_REG_DIG_INFO4		80
+#	define ONYX_VALIDL		(1<<7)
+#	define ONYX_VALIDR		(1<<6)
+#	define ONYX_SPDIF_ENABLE	(1<<5)
+/* lower 4 bits control bits 32-35 of channel control and word length */
+#	define ONYX_WORDLEN_MASK	(0xF)
+
+#endif /* __SND_AOA_CODEC_ONYX_H */
diff --git a/ANDROID_3.4.5/sound/aoa/codecs/tas-basstreble.h b/ANDROID_3.4.5/sound/aoa/codecs/tas-basstreble.h
new file mode 100644
index 00000000..69b61136
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/codecs/tas-basstreble.h
@@ -0,0 +1,134 @@
+/*
+ * This file is only included exactly once!
+ *
+ * The tables here are derived from the tas3004 datasheet,
+ * modulo typo corrections and some smoothing...
+ */
+
+#define TAS3004_TREBLE_MIN	0
+#define TAS3004_TREBLE_MAX	72
+#define TAS3004_BASS_MIN	0
+#define TAS3004_BASS_MAX	72
+#define TAS3004_TREBLE_ZERO	36
+#define TAS3004_BASS_ZERO	36
+
+static u8 tas3004_treble_table[] = {
+	150, /* -18 dB */
+	149,
+	148,
+	147,
+	146,
+	145,
+	144,
+	143,
+	142,
+	141,
+	140,
+	139,
+	138,
+	137,
+	136,
+	135,
+	134,
+	133,
+	132,
+	131,
+	130,
+	129,
+	128,
+	127,
+	126,
+	125,
+	124,
+	123,
+	122,
+	121,
+	120,
+	119,
+	118,
+	117,
+	116,
+	115,
+	114, /* 0 dB */
+	113,
+	112,
+	111,
+	109,
+	108,
+	107,
+	105,
+	104,
+	103,
+	101,
+	99,
+	98,
+	96,
+	93,
+	91,
+	89,
+	86,
+	83,
+	81,
+	77,
+	74,
+	71,
+	67,
+	63,
+	59,
+	54,
+	49,
+	44,
+	38,
+	32,
+	26,
+	19,
+	10,
+	4,
+	2,
+	1, /* +18 dB */
+};
+
+static inline u8 tas3004_treble(int idx)
+{
+	return tas3004_treble_table[idx];
+}
+
+/* I only save the difference here to the treble table
+ * so that the binary is smaller...
+ * I have also ignored completely differences of
+ * +/- 1
+ */
+static s8 tas3004_bass_diff_to_treble[] = {
+	2, /* 7 dB, offset 50 */
+	2,
+	2,
+	2,
+	2,
+	1,
+	2,
+	2,
+	2,
+	3,
+	4,
+	4,
+	5,
+	6,
+	7,
+	8,
+	9,
+	10,
+	11,
+	14,
+	13,
+	8,
+	1, /* 18 dB */
+};
+
+static inline u8 tas3004_bass(int idx)
+{
+	u8 result = tas3004_treble_table[idx];
+
+	if (idx >= 50)
+		result += tas3004_bass_diff_to_treble[idx-50];
+	return result;
+}
diff --git a/ANDROID_3.4.5/sound/aoa/codecs/tas-gain-table.h b/ANDROID_3.4.5/sound/aoa/codecs/tas-gain-table.h
new file mode 100644
index 00000000..4cfa6757
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/codecs/tas-gain-table.h
@@ -0,0 +1,209 @@
+/*
+ This is the program used to generate below table.
+
+#include <stdio.h>
+#include <math.h>
+int main() {
+  int dB2;
+  printf("/" "* This file is only included exactly once!\n");
+  printf(" *\n");
+  printf(" * If they'd only tell us that generating this table was\n");
+  printf(" * as easy as calculating\n");
+  printf(" *      hwvalue = 1048576.0*exp(0.057564628*dB*2)\n");
+  printf(" * :) *" "/\n");
+  printf("static int tas_gaintable[] = {\n");
+  printf("	0x000000, /" "* -infinity dB *" "/\n");
+  for (dB2=-140;dB2<=36;dB2++)
+    printf("	0x%.6x, /" "* %-02.1f dB *" "/\n", (int)(1048576.0*exp(0.057564628*dB2)), dB2/2.0);
+  printf("};\n\n");
+}
+
+*/
+
+/* This file is only included exactly once!
+ *
+ * If they'd only tell us that generating this table was
+ * as easy as calculating
+ *      hwvalue = 1048576.0*exp(0.057564628*dB*2)
+ * :) */
+static int tas_gaintable[] = {
+	0x000000, /* -infinity dB */
+	0x00014b, /* -70.0 dB */
+	0x00015f, /* -69.5 dB */
+	0x000174, /* -69.0 dB */
+	0x00018a, /* -68.5 dB */
+	0x0001a1, /* -68.0 dB */
+	0x0001ba, /* -67.5 dB */
+	0x0001d4, /* -67.0 dB */
+	0x0001f0, /* -66.5 dB */
+	0x00020d, /* -66.0 dB */
+	0x00022c, /* -65.5 dB */
+	0x00024d, /* -65.0 dB */
+	0x000270, /* -64.5 dB */
+	0x000295, /* -64.0 dB */
+	0x0002bc, /* -63.5 dB */
+	0x0002e6, /* -63.0 dB */
+	0x000312, /* -62.5 dB */
+	0x000340, /* -62.0 dB */
+	0x000372, /* -61.5 dB */
+	0x0003a6, /* -61.0 dB */
+	0x0003dd, /* -60.5 dB */
+	0x000418, /* -60.0 dB */
+	0x000456, /* -59.5 dB */
+	0x000498, /* -59.0 dB */
+	0x0004de, /* -58.5 dB */
+	0x000528, /* -58.0 dB */
+	0x000576, /* -57.5 dB */
+	0x0005c9, /* -57.0 dB */
+	0x000620, /* -56.5 dB */
+	0x00067d, /* -56.0 dB */
+	0x0006e0, /* -55.5 dB */
+	0x000748, /* -55.0 dB */
+	0x0007b7, /* -54.5 dB */
+	0x00082c, /* -54.0 dB */
+	0x0008a8, /* -53.5 dB */
+	0x00092b, /* -53.0 dB */
+	0x0009b6, /* -52.5 dB */
+	0x000a49, /* -52.0 dB */
+	0x000ae5, /* -51.5 dB */
+	0x000b8b, /* -51.0 dB */
+	0x000c3a, /* -50.5 dB */
+	0x000cf3, /* -50.0 dB */
+	0x000db8, /* -49.5 dB */
+	0x000e88, /* -49.0 dB */
+	0x000f64, /* -48.5 dB */
+	0x00104e, /* -48.0 dB */
+	0x001145, /* -47.5 dB */
+	0x00124b, /* -47.0 dB */
+	0x001361, /* -46.5 dB */
+	0x001487, /* -46.0 dB */
+	0x0015be, /* -45.5 dB */
+	0x001708, /* -45.0 dB */
+	0x001865, /* -44.5 dB */
+	0x0019d8, /* -44.0 dB */
+	0x001b60, /* -43.5 dB */
+	0x001cff, /* -43.0 dB */
+	0x001eb7, /* -42.5 dB */
+	0x002089, /* -42.0 dB */
+	0x002276, /* -41.5 dB */
+	0x002481, /* -41.0 dB */
+	0x0026ab, /* -40.5 dB */
+	0x0028f5, /* -40.0 dB */
+	0x002b63, /* -39.5 dB */
+	0x002df5, /* -39.0 dB */
+	0x0030ae, /* -38.5 dB */
+	0x003390, /* -38.0 dB */
+	0x00369e, /* -37.5 dB */
+	0x0039db, /* -37.0 dB */
+	0x003d49, /* -36.5 dB */
+	0x0040ea, /* -36.0 dB */
+	0x0044c3, /* -35.5 dB */
+	0x0048d6, /* -35.0 dB */
+	0x004d27, /* -34.5 dB */
+	0x0051b9, /* -34.0 dB */
+	0x005691, /* -33.5 dB */
+	0x005bb2, /* -33.0 dB */
+	0x006121, /* -32.5 dB */
+	0x0066e3, /* -32.0 dB */
+	0x006cfb, /* -31.5 dB */
+	0x007370, /* -31.0 dB */
+	0x007a48, /* -30.5 dB */
+	0x008186, /* -30.0 dB */
+	0x008933, /* -29.5 dB */
+	0x009154, /* -29.0 dB */
+	0x0099f1, /* -28.5 dB */
+	0x00a310, /* -28.0 dB */
+	0x00acba, /* -27.5 dB */
+	0x00b6f6, /* -27.0 dB */
+	0x00c1cd, /* -26.5 dB */
+	0x00cd49, /* -26.0 dB */
+	0x00d973, /* -25.5 dB */
+	0x00e655, /* -25.0 dB */
+	0x00f3fb, /* -24.5 dB */
+	0x010270, /* -24.0 dB */
+	0x0111c0, /* -23.5 dB */
+	0x0121f9, /* -23.0 dB */
+	0x013328, /* -22.5 dB */
+	0x01455b, /* -22.0 dB */
+	0x0158a2, /* -21.5 dB */
+	0x016d0e, /* -21.0 dB */
+	0x0182af, /* -20.5 dB */
+	0x019999, /* -20.0 dB */
+	0x01b1de, /* -19.5 dB */
+	0x01cb94, /* -19.0 dB */
+	0x01e6cf, /* -18.5 dB */
+	0x0203a7, /* -18.0 dB */
+	0x022235, /* -17.5 dB */
+	0x024293, /* -17.0 dB */
+	0x0264db, /* -16.5 dB */
+	0x02892c, /* -16.0 dB */
+	0x02afa3, /* -15.5 dB */
+	0x02d862, /* -15.0 dB */
+	0x03038a, /* -14.5 dB */
+	0x033142, /* -14.0 dB */
+	0x0361af, /* -13.5 dB */
+	0x0394fa, /* -13.0 dB */
+	0x03cb50, /* -12.5 dB */
+	0x0404de, /* -12.0 dB */
+	0x0441d5, /* -11.5 dB */
+	0x048268, /* -11.0 dB */
+	0x04c6d0, /* -10.5 dB */
+	0x050f44, /* -10.0 dB */
+	0x055c04, /* -9.5 dB */
+	0x05ad50, /* -9.0 dB */
+	0x06036e, /* -8.5 dB */
+	0x065ea5, /* -8.0 dB */
+	0x06bf44, /* -7.5 dB */
+	0x07259d, /* -7.0 dB */
+	0x079207, /* -6.5 dB */
+	0x0804dc, /* -6.0 dB */
+	0x087e80, /* -5.5 dB */
+	0x08ff59, /* -5.0 dB */
+	0x0987d5, /* -4.5 dB */
+	0x0a1866, /* -4.0 dB */
+	0x0ab189, /* -3.5 dB */
+	0x0b53be, /* -3.0 dB */
+	0x0bff91, /* -2.5 dB */
+	0x0cb591, /* -2.0 dB */
+	0x0d765a, /* -1.5 dB */
+	0x0e4290, /* -1.0 dB */
+	0x0f1adf, /* -0.5 dB */
+	0x100000, /* 0.0 dB */
+	0x10f2b4, /* 0.5 dB */
+	0x11f3c9, /* 1.0 dB */
+	0x13041a, /* 1.5 dB */
+	0x14248e, /* 2.0 dB */
+	0x15561a, /* 2.5 dB */
+	0x1699c0, /* 3.0 dB */
+	0x17f094, /* 3.5 dB */
+	0x195bb8, /* 4.0 dB */
+	0x1adc61, /* 4.5 dB */
+	0x1c73d5, /* 5.0 dB */
+	0x1e236d, /* 5.5 dB */
+	0x1fec98, /* 6.0 dB */
+	0x21d0d9, /* 6.5 dB */
+	0x23d1cd, /* 7.0 dB */
+	0x25f125, /* 7.5 dB */
+	0x2830af, /* 8.0 dB */
+	0x2a9254, /* 8.5 dB */
+	0x2d1818, /* 9.0 dB */
+	0x2fc420, /* 9.5 dB */
+	0x3298b0, /* 10.0 dB */
+	0x35982f, /* 10.5 dB */
+	0x38c528, /* 11.0 dB */
+	0x3c224c, /* 11.5 dB */
+	0x3fb278, /* 12.0 dB */
+	0x4378b0, /* 12.5 dB */
+	0x477829, /* 13.0 dB */
+	0x4bb446, /* 13.5 dB */
+	0x5030a1, /* 14.0 dB */
+	0x54f106, /* 14.5 dB */
+	0x59f980, /* 15.0 dB */
+	0x5f4e52, /* 15.5 dB */
+	0x64f403, /* 16.0 dB */
+	0x6aef5e, /* 16.5 dB */
+	0x714575, /* 17.0 dB */
+	0x77fbaa, /* 17.5 dB */
+	0x7f17af, /* 18.0 dB */
+};
+
diff --git a/ANDROID_3.4.5/sound/aoa/codecs/tas.c b/ANDROID_3.4.5/sound/aoa/codecs/tas.c
new file mode 100644
index 00000000..8e63d1f3
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/codecs/tas.c
@@ -0,0 +1,1029 @@
+/*
+ * Apple Onboard Audio driver for tas codec
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ *
+ * Open questions:
+ *  - How to distinguish between 3004 and versions?
+ *
+ * FIXMEs:
+ *  - This codec driver doesn't honour the 'connected'
+ *    property of the aoa_codec struct, hence if
+ *    it is used in machines where not everything is
+ *    connected it will display wrong mixer elements.
+ *  - Driver assumes that the microphone is always
+ *    monaureal and connected to the right channel of
+ *    the input. This should also be a codec-dependent
+ *    flag, maybe the codec should have 3 different
+ *    bits for the three different possibilities how
+ *    it can be hooked up...
+ *    But as long as I don't see any hardware hooked
+ *    up that way...
+ *  - As Apple notes in their code, the tas3004 seems
+ *    to delay the right channel by one sample. You can
+ *    see this when for example recording stereo in
+ *    audacity, or recording the tas output via cable
+ *    on another machine (use a sinus generator or so).
+ *    I tried programming the BiQuads but couldn't
+ *    make the delay work, maybe someone can read the
+ *    datasheet and fix it. The relevant Apple comment
+ *    is in AppleTAS3004Audio.cpp lines 1637 ff. Note
+ *    that their comment describing how they program
+ *    the filters sucks...
+ *
+ * Other things:
+ *  - this should actually register *two* aoa_codec
+ *    structs since it has two inputs. Then it must
+ *    use the prepare callback to forbid running the
+ *    secondary output on a different clock.
+ *    Also, whatever bus knows how to do this must
+ *    provide two soundbus_dev devices and the fabric
+ *    must be able to link them correctly.
+ *
+ *    I don't even know if Apple ever uses the second
+ *    port on the tas3004 though, I don't think their
+ *    i2s controllers can even do it. OTOH, they all
+ *    derive the clocks from common clocks, so it
+ *    might just be possible. The framework allows the
+ *    codec to refine the transfer_info items in the
+ *    usable callback, so we can simply remove the
+ *    rates the second instance is not using when it
+ *    actually is in use.
+ *    Maybe we'll need to make the sound busses have
+ *    a 'clock group id' value so the codec can
+ *    determine if the two outputs can be driven at
+ *    the same time. But that is likely overkill, up
+ *    to the fabric to not link them up incorrectly,
+ *    and up to the hardware designer to not wire
+ *    them up in some weird unusable way.
+ */
+#include <stddef.h>
+#include <linux/i2c.h>
+#include <asm/pmac_low_i2c.h>
+#include <asm/prom.h>
+#include <linux/delay.h>
+#include <linux/module.h>
+#include <linux/mutex.h>
+#include <linux/slab.h>
+
+MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("tas codec driver for snd-aoa");
+
+#include "tas.h"
+#include "tas-gain-table.h"
+#include "tas-basstreble.h"
+#include "../aoa.h"
+#include "../soundbus/soundbus.h"
+
+#define PFX "snd-aoa-codec-tas: "
+
+
+struct tas {
+	struct aoa_codec	codec;
+	struct i2c_client	*i2c;
+	u32			mute_l:1, mute_r:1 ,
+				controls_created:1 ,
+				drc_enabled:1,
+				hw_enabled:1;
+	u8			cached_volume_l, cached_volume_r;
+	u8			mixer_l[3], mixer_r[3];
+	u8			bass, treble;
+	u8			acr;
+	int			drc_range;
+	/* protects hardware access against concurrency from
+	 * userspace when hitting controls and during
+	 * codec init/suspend/resume */
+	struct mutex		mtx;
+};
+
+static int tas_reset_init(struct tas *tas);
+
+static struct tas *codec_to_tas(struct aoa_codec *codec)
+{
+	return container_of(codec, struct tas, codec);
+}
+
+static inline int tas_write_reg(struct tas *tas, u8 reg, u8 len, u8 *data)
+{
+	if (len == 1)
+		return i2c_smbus_write_byte_data(tas->i2c, reg, *data);
+	else
+		return i2c_smbus_write_i2c_block_data(tas->i2c, reg, len, data);
+}
+
+static void tas3004_set_drc(struct tas *tas)
+{
+	unsigned char val[6];
+
+	if (tas->drc_enabled)
+		val[0] = 0x50; /* 3:1 above threshold */
+	else
+		val[0] = 0x51; /* disabled */
+	val[1] = 0x02; /* 1:1 below threshold */
+	if (tas->drc_range > 0xef)
+		val[2] = 0xef;
+	else if (tas->drc_range < 0)
+		val[2] = 0x00;
+	else
+		val[2] = tas->drc_range;
+	val[3] = 0xb0;
+	val[4] = 0x60;
+	val[5] = 0xa0;
+
+	tas_write_reg(tas, TAS_REG_DRC, 6, val);
+}
+
+static void tas_set_treble(struct tas *tas)
+{
+	u8 tmp;
+
+	tmp = tas3004_treble(tas->treble);
+	tas_write_reg(tas, TAS_REG_TREBLE, 1, &tmp);
+}
+
+static void tas_set_bass(struct tas *tas)
+{
+	u8 tmp;
+
+	tmp = tas3004_bass(tas->bass);
+	tas_write_reg(tas, TAS_REG_BASS, 1, &tmp);
+}
+
+static void tas_set_volume(struct tas *tas)
+{
+	u8 block[6];
+	int tmp;
+	u8 left, right;
+
+	left = tas->cached_volume_l;
+	right = tas->cached_volume_r;
+
+	if (left > 177) left = 177;
+	if (right > 177) right = 177;
+
+	if (tas->mute_l) left = 0;
+	if (tas->mute_r) right = 0;
+
+	/* analysing the volume and mixer tables shows
+	 * that they are similar enough when we shift
+	 * the mixer table down by 4 bits. The error
+	 * is miniscule, in just one item the error
+	 * is 1, at a value of 0x07f17b (mixer table
+	 * value is 0x07f17a) */
+	tmp = tas_gaintable[left];
+	block[0] = tmp>>20;
+	block[1] = tmp>>12;
+	block[2] = tmp>>4;
+	tmp = tas_gaintable[right];
+	block[3] = tmp>>20;
+	block[4] = tmp>>12;
+	block[5] = tmp>>4;
+	tas_write_reg(tas, TAS_REG_VOL, 6, block);
+}
+
+static void tas_set_mixer(struct tas *tas)
+{
+	u8 block[9];
+	int tmp, i;
+	u8 val;
+
+	for (i=0;i<3;i++) {
+		val = tas->mixer_l[i];
+		if (val > 177) val = 177;
+		tmp = tas_gaintable[val];
+		block[3*i+0] = tmp>>16;
+		block[3*i+1] = tmp>>8;
+		block[3*i+2] = tmp;
+	}
+	tas_write_reg(tas, TAS_REG_LMIX, 9, block);
+
+	for (i=0;i<3;i++) {
+		val = tas->mixer_r[i];
+		if (val > 177) val = 177;
+		tmp = tas_gaintable[val];
+		block[3*i+0] = tmp>>16;
+		block[3*i+1] = tmp>>8;
+		block[3*i+2] = tmp;
+	}
+	tas_write_reg(tas, TAS_REG_RMIX, 9, block);
+}
+
+/* alsa stuff */
+
+static int tas_dev_register(struct snd_device *dev)
+{
+	return 0;
+}
+
+static struct snd_device_ops ops = {
+	.dev_register = tas_dev_register,
+};
+
+static int tas_snd_vol_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 = 177;
+	return 0;
+}
+
+static int tas_snd_vol_get(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+	mutex_lock(&tas->mtx);
+	ucontrol->value.integer.value[0] = tas->cached_volume_l;
+	ucontrol->value.integer.value[1] = tas->cached_volume_r;
+	mutex_unlock(&tas->mtx);
+	return 0;
+}
+
+static int tas_snd_vol_put(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+	if (ucontrol->value.integer.value[0] < 0 ||
+	    ucontrol->value.integer.value[0] > 177)
+		return -EINVAL;
+	if (ucontrol->value.integer.value[1] < 0 ||
+	    ucontrol->value.integer.value[1] > 177)
+		return -EINVAL;
+
+	mutex_lock(&tas->mtx);
+	if (tas->cached_volume_l == ucontrol->value.integer.value[0]
+	 && tas->cached_volume_r == ucontrol->value.integer.value[1]) {
+		mutex_unlock(&tas->mtx);
+		return 0;
+	}
+
+	tas->cached_volume_l = ucontrol->value.integer.value[0];
+	tas->cached_volume_r = ucontrol->value.integer.value[1];
+	if (tas->hw_enabled)
+		tas_set_volume(tas);
+	mutex_unlock(&tas->mtx);
+	return 1;
+}
+
+static struct snd_kcontrol_new volume_control = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Master Playback Volume",
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.info = tas_snd_vol_info,
+	.get = tas_snd_vol_get,
+	.put = tas_snd_vol_put,
+};
+
+#define tas_snd_mute_info	snd_ctl_boolean_stereo_info
+
+static int tas_snd_mute_get(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+	mutex_lock(&tas->mtx);
+	ucontrol->value.integer.value[0] = !tas->mute_l;
+	ucontrol->value.integer.value[1] = !tas->mute_r;
+	mutex_unlock(&tas->mtx);
+	return 0;
+}
+
+static int tas_snd_mute_put(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+	mutex_lock(&tas->mtx);
+	if (tas->mute_l == !ucontrol->value.integer.value[0]
+	 && tas->mute_r == !ucontrol->value.integer.value[1]) {
+		mutex_unlock(&tas->mtx);
+		return 0;
+	}
+
+	tas->mute_l = !ucontrol->value.integer.value[0];
+	tas->mute_r = !ucontrol->value.integer.value[1];
+	if (tas->hw_enabled)
+		tas_set_volume(tas);
+	mutex_unlock(&tas->mtx);
+	return 1;
+}
+
+static struct snd_kcontrol_new mute_control = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Master Playback Switch",
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.info = tas_snd_mute_info,
+	.get = tas_snd_mute_get,
+	.put = tas_snd_mute_put,
+};
+
+static int tas_snd_mixer_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 = 177;
+	return 0;
+}
+
+static int tas_snd_mixer_get(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct tas *tas = snd_kcontrol_chip(kcontrol);
+	int idx = kcontrol->private_value;
+
+	mutex_lock(&tas->mtx);
+	ucontrol->value.integer.value[0] = tas->mixer_l[idx];
+	ucontrol->value.integer.value[1] = tas->mixer_r[idx];
+	mutex_unlock(&tas->mtx);
+
+	return 0;
+}
+
+static int tas_snd_mixer_put(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct tas *tas = snd_kcontrol_chip(kcontrol);
+	int idx = kcontrol->private_value;
+
+	mutex_lock(&tas->mtx);
+	if (tas->mixer_l[idx] == ucontrol->value.integer.value[0]
+	 && tas->mixer_r[idx] == ucontrol->value.integer.value[1]) {
+		mutex_unlock(&tas->mtx);
+		return 0;
+	}
+
+	tas->mixer_l[idx] = ucontrol->value.integer.value[0];
+	tas->mixer_r[idx] = ucontrol->value.integer.value[1];
+
+	if (tas->hw_enabled)
+		tas_set_mixer(tas);
+	mutex_unlock(&tas->mtx);
+	return 1;
+}
+
+#define MIXER_CONTROL(n,descr,idx)			\
+static struct snd_kcontrol_new n##_control = {		\
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,		\
+	.name = descr " Playback Volume",		\
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,	\
+	.info = tas_snd_mixer_info,			\
+	.get = tas_snd_mixer_get,			\
+	.put = tas_snd_mixer_put,			\
+	.private_value = idx,				\
+}
+
+MIXER_CONTROL(pcm1, "PCM", 0);
+MIXER_CONTROL(monitor, "Monitor", 2);
+
+static int tas_snd_drc_range_info(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = TAS3004_DRC_MAX;
+	return 0;
+}
+
+static int tas_snd_drc_range_get(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+	mutex_lock(&tas->mtx);
+	ucontrol->value.integer.value[0] = tas->drc_range;
+	mutex_unlock(&tas->mtx);
+	return 0;
+}
+
+static int tas_snd_drc_range_put(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+	if (ucontrol->value.integer.value[0] < 0 ||
+	    ucontrol->value.integer.value[0] > TAS3004_DRC_MAX)
+		return -EINVAL;
+
+	mutex_lock(&tas->mtx);
+	if (tas->drc_range == ucontrol->value.integer.value[0]) {
+		mutex_unlock(&tas->mtx);
+		return 0;
+	}
+
+	tas->drc_range = ucontrol->value.integer.value[0];
+	if (tas->hw_enabled)
+		tas3004_set_drc(tas);
+	mutex_unlock(&tas->mtx);
+	return 1;
+}
+
+static struct snd_kcontrol_new drc_range_control = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "DRC Range",
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.info = tas_snd_drc_range_info,
+	.get = tas_snd_drc_range_get,
+	.put = tas_snd_drc_range_put,
+};
+
+#define tas_snd_drc_switch_info		snd_ctl_boolean_mono_info
+
+static int tas_snd_drc_switch_get(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+	mutex_lock(&tas->mtx);
+	ucontrol->value.integer.value[0] = tas->drc_enabled;
+	mutex_unlock(&tas->mtx);
+	return 0;
+}
+
+static int tas_snd_drc_switch_put(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+	mutex_lock(&tas->mtx);
+	if (tas->drc_enabled == ucontrol->value.integer.value[0]) {
+		mutex_unlock(&tas->mtx);
+		return 0;
+	}
+
+	tas->drc_enabled = !!ucontrol->value.integer.value[0];
+	if (tas->hw_enabled)
+		tas3004_set_drc(tas);
+	mutex_unlock(&tas->mtx);
+	return 1;
+}
+
+static struct snd_kcontrol_new drc_switch_control = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "DRC Range Switch",
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.info = tas_snd_drc_switch_info,
+	.get = tas_snd_drc_switch_get,
+	.put = tas_snd_drc_switch_put,
+};
+
+static int tas_snd_capture_source_info(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo)
+{
+	static char *texts[] = { "Line-In", "Microphone" };
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = 2;
+	if (uinfo->value.enumerated.item > 1)
+		uinfo->value.enumerated.item = 1;
+	strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
+	return 0;
+}
+
+static int tas_snd_capture_source_get(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+	mutex_lock(&tas->mtx);
+	ucontrol->value.enumerated.item[0] = !!(tas->acr & TAS_ACR_INPUT_B);
+	mutex_unlock(&tas->mtx);
+	return 0;
+}
+
+static int tas_snd_capture_source_put(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct tas *tas = snd_kcontrol_chip(kcontrol);
+	int oldacr;
+
+	if (ucontrol->value.enumerated.item[0] > 1)
+		return -EINVAL;
+	mutex_lock(&tas->mtx);
+	oldacr = tas->acr;
+
+	/*
+	 * Despite what the data sheet says in one place, the
+	 * TAS_ACR_B_MONAUREAL bit forces mono output even when
+	 * input A (line in) is selected.
+	 */
+	tas->acr &= ~(TAS_ACR_INPUT_B | TAS_ACR_B_MONAUREAL);
+	if (ucontrol->value.enumerated.item[0])
+		tas->acr |= TAS_ACR_INPUT_B | TAS_ACR_B_MONAUREAL |
+		      TAS_ACR_B_MON_SEL_RIGHT;
+	if (oldacr == tas->acr) {
+		mutex_unlock(&tas->mtx);
+		return 0;
+	}
+	if (tas->hw_enabled)
+		tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr);
+	mutex_unlock(&tas->mtx);
+	return 1;
+}
+
+static struct snd_kcontrol_new capture_source_control = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	/* If we name this 'Input Source', it properly shows up in
+	 * alsamixer as a selection, * but it's shown under the
+	 * 'Playback' category.
+	 * If I name it 'Capture Source', it shows up in strange
+	 * ways (two bools of which one can be selected at a
+	 * time) but at least it's shown in the 'Capture'
+	 * category.
+	 * I was told that this was due to backward compatibility,
+	 * but I don't understand then why the mangling is *not*
+	 * done when I name it "Input Source".....
+	 */
+	.name = "Capture Source",
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.info = tas_snd_capture_source_info,
+	.get = tas_snd_capture_source_get,
+	.put = tas_snd_capture_source_put,
+};
+
+static int tas_snd_treble_info(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	uinfo->value.integer.min = TAS3004_TREBLE_MIN;
+	uinfo->value.integer.max = TAS3004_TREBLE_MAX;
+	return 0;
+}
+
+static int tas_snd_treble_get(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+	mutex_lock(&tas->mtx);
+	ucontrol->value.integer.value[0] = tas->treble;
+	mutex_unlock(&tas->mtx);
+	return 0;
+}
+
+static int tas_snd_treble_put(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+	if (ucontrol->value.integer.value[0] < TAS3004_TREBLE_MIN ||
+	    ucontrol->value.integer.value[0] > TAS3004_TREBLE_MAX)
+		return -EINVAL;
+	mutex_lock(&tas->mtx);
+	if (tas->treble == ucontrol->value.integer.value[0]) {
+		mutex_unlock(&tas->mtx);
+		return 0;
+	}
+
+	tas->treble = ucontrol->value.integer.value[0];
+	if (tas->hw_enabled)
+		tas_set_treble(tas);
+	mutex_unlock(&tas->mtx);
+	return 1;
+}
+
+static struct snd_kcontrol_new treble_control = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Treble",
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.info = tas_snd_treble_info,
+	.get = tas_snd_treble_get,
+	.put = tas_snd_treble_put,
+};
+
+static int tas_snd_bass_info(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	uinfo->value.integer.min = TAS3004_BASS_MIN;
+	uinfo->value.integer.max = TAS3004_BASS_MAX;
+	return 0;
+}
+
+static int tas_snd_bass_get(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+	mutex_lock(&tas->mtx);
+	ucontrol->value.integer.value[0] = tas->bass;
+	mutex_unlock(&tas->mtx);
+	return 0;
+}
+
+static int tas_snd_bass_put(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct tas *tas = snd_kcontrol_chip(kcontrol);
+
+	if (ucontrol->value.integer.value[0] < TAS3004_BASS_MIN ||
+	    ucontrol->value.integer.value[0] > TAS3004_BASS_MAX)
+		return -EINVAL;
+	mutex_lock(&tas->mtx);
+	if (tas->bass == ucontrol->value.integer.value[0]) {
+		mutex_unlock(&tas->mtx);
+		return 0;
+	}
+
+	tas->bass = ucontrol->value.integer.value[0];
+	if (tas->hw_enabled)
+		tas_set_bass(tas);
+	mutex_unlock(&tas->mtx);
+	return 1;
+}
+
+static struct snd_kcontrol_new bass_control = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Bass",
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.info = tas_snd_bass_info,
+	.get = tas_snd_bass_get,
+	.put = tas_snd_bass_put,
+};
+
+static struct transfer_info tas_transfers[] = {
+	{
+		/* input */
+		.formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE,
+		.rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+		.transfer_in = 1,
+	},
+	{
+		/* output */
+		.formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE,
+		.rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+		.transfer_in = 0,
+	},
+	{}
+};
+
+static int tas_usable(struct codec_info_item *cii,
+		      struct transfer_info *ti,
+		      struct transfer_info *out)
+{
+	return 1;
+}
+
+static int tas_reset_init(struct tas *tas)
+{
+	u8 tmp;
+
+	tas->codec.gpio->methods->all_amps_off(tas->codec.gpio);
+	msleep(5);
+	tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 0);
+	msleep(5);
+	tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 1);
+	msleep(20);
+	tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 0);
+	msleep(10);
+	tas->codec.gpio->methods->all_amps_restore(tas->codec.gpio);
+
+	tmp = TAS_MCS_SCLK64 | TAS_MCS_SPORT_MODE_I2S | TAS_MCS_SPORT_WL_24BIT;
+	if (tas_write_reg(tas, TAS_REG_MCS, 1, &tmp))
+		goto outerr;
+
+	tas->acr |= TAS_ACR_ANALOG_PDOWN;
+	if (tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr))
+		goto outerr;
+
+	tmp = 0;
+	if (tas_write_reg(tas, TAS_REG_MCS2, 1, &tmp))
+		goto outerr;
+
+	tas3004_set_drc(tas);
+
+	/* Set treble & bass to 0dB */
+	tas->treble = TAS3004_TREBLE_ZERO;
+	tas->bass = TAS3004_BASS_ZERO;
+	tas_set_treble(tas);
+	tas_set_bass(tas);
+
+	tas->acr &= ~TAS_ACR_ANALOG_PDOWN;
+	if (tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr))
+		goto outerr;
+
+	return 0;
+ outerr:
+	return -ENODEV;
+}
+
+static int tas_switch_clock(struct codec_info_item *cii, enum clock_switch clock)
+{
+	struct tas *tas = cii->codec_data;
+
+	switch(clock) {
+	case CLOCK_SWITCH_PREPARE_SLAVE:
+		/* Clocks are going away, mute mute mute */
+		tas->codec.gpio->methods->all_amps_off(tas->codec.gpio);
+		tas->hw_enabled = 0;
+		break;
+	case CLOCK_SWITCH_SLAVE:
+		/* Clocks are back, re-init the codec */
+		mutex_lock(&tas->mtx);
+		tas_reset_init(tas);
+		tas_set_volume(tas);
+		tas_set_mixer(tas);
+		tas->hw_enabled = 1;
+		tas->codec.gpio->methods->all_amps_restore(tas->codec.gpio);
+		mutex_unlock(&tas->mtx);
+		break;
+	default:
+		/* doesn't happen as of now */
+		return -EINVAL;
+	}
+	return 0;
+}
+
+#ifdef CONFIG_PM
+/* we are controlled via i2c and assume that is always up
+ * If that wasn't the case, we'd have to suspend once
+ * our i2c device is suspended, and then take note of that! */
+static int tas_suspend(struct tas *tas)
+{
+	mutex_lock(&tas->mtx);
+	tas->hw_enabled = 0;
+	tas->acr |= TAS_ACR_ANALOG_PDOWN;
+	tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr);
+	mutex_unlock(&tas->mtx);
+	return 0;
+}
+
+static int tas_resume(struct tas *tas)
+{
+	/* reset codec */
+	mutex_lock(&tas->mtx);
+	tas_reset_init(tas);
+	tas_set_volume(tas);
+	tas_set_mixer(tas);
+	tas->hw_enabled = 1;
+	mutex_unlock(&tas->mtx);
+	return 0;
+}
+
+static int _tas_suspend(struct codec_info_item *cii, pm_message_t state)
+{
+	return tas_suspend(cii->codec_data);
+}
+
+static int _tas_resume(struct codec_info_item *cii)
+{
+	return tas_resume(cii->codec_data);
+}
+#else /* CONFIG_PM */
+#define _tas_suspend	NULL
+#define _tas_resume	NULL
+#endif /* CONFIG_PM */
+
+static struct codec_info tas_codec_info = {
+	.transfers = tas_transfers,
+	/* in theory, we can drive it at 512 too...
+	 * but so far the framework doesn't allow
+	 * for that and I don't see much point in it. */
+	.sysclock_factor = 256,
+	/* same here, could be 32 for just one 16 bit format */
+	.bus_factor = 64,
+	.owner = THIS_MODULE,
+	.usable = tas_usable,
+	.switch_clock = tas_switch_clock,
+	.suspend = _tas_suspend,
+	.resume = _tas_resume,
+};
+
+static int tas_init_codec(struct aoa_codec *codec)
+{
+	struct tas *tas = codec_to_tas(codec);
+	int err;
+
+	if (!tas->codec.gpio || !tas->codec.gpio->methods) {
+		printk(KERN_ERR PFX "gpios not assigned!!\n");
+		return -EINVAL;
+	}
+
+	mutex_lock(&tas->mtx);
+	if (tas_reset_init(tas)) {
+		printk(KERN_ERR PFX "tas failed to initialise\n");
+		mutex_unlock(&tas->mtx);
+		return -ENXIO;
+	}
+	tas->hw_enabled = 1;
+	mutex_unlock(&tas->mtx);
+
+	if (tas->codec.soundbus_dev->attach_codec(tas->codec.soundbus_dev,
+						   aoa_get_card(),
+						   &tas_codec_info, tas)) {
+		printk(KERN_ERR PFX "error attaching tas to soundbus\n");
+		return -ENODEV;
+	}
+
+	if (aoa_snd_device_new(SNDRV_DEV_LOWLEVEL, tas, &ops)) {
+		printk(KERN_ERR PFX "failed to create tas snd device!\n");
+		return -ENODEV;
+	}
+	err = aoa_snd_ctl_add(snd_ctl_new1(&volume_control, tas));
+	if (err)
+		goto error;
+
+	err = aoa_snd_ctl_add(snd_ctl_new1(&mute_control, tas));
+	if (err)
+		goto error;
+
+	err = aoa_snd_ctl_add(snd_ctl_new1(&pcm1_control, tas));
+	if (err)
+		goto error;
+
+	err = aoa_snd_ctl_add(snd_ctl_new1(&monitor_control, tas));
+	if (err)
+		goto error;
+
+	err = aoa_snd_ctl_add(snd_ctl_new1(&capture_source_control, tas));
+	if (err)
+		goto error;
+
+	err = aoa_snd_ctl_add(snd_ctl_new1(&drc_range_control, tas));
+	if (err)
+		goto error;
+
+	err = aoa_snd_ctl_add(snd_ctl_new1(&drc_switch_control, tas));
+	if (err)
+		goto error;
+
+	err = aoa_snd_ctl_add(snd_ctl_new1(&treble_control, tas));
+	if (err)
+		goto error;
+
+	err = aoa_snd_ctl_add(snd_ctl_new1(&bass_control, tas));
+	if (err)
+		goto error;
+
+	return 0;
+ error:
+	tas->codec.soundbus_dev->detach_codec(tas->codec.soundbus_dev, tas);
+	snd_device_free(aoa_get_card(), tas);
+	return err;
+}
+
+static void tas_exit_codec(struct aoa_codec *codec)
+{
+	struct tas *tas = codec_to_tas(codec);
+
+	if (!tas->codec.soundbus_dev)
+		return;
+	tas->codec.soundbus_dev->detach_codec(tas->codec.soundbus_dev, tas);
+}
+
+
+static int tas_create(struct i2c_adapter *adapter,
+		       struct device_node *node,
+		       int addr)
+{
+	struct i2c_board_info info;
+	struct i2c_client *client;
+
+	memset(&info, 0, sizeof(struct i2c_board_info));
+	strlcpy(info.type, "aoa_codec_tas", I2C_NAME_SIZE);
+	info.addr = addr;
+	info.platform_data = node;
+
+	client = i2c_new_device(adapter, &info);
+	if (!client)
+		return -ENODEV;
+	/*
+	 * We know the driver is already loaded, so the device should be
+	 * already bound. If not it means binding failed, and then there
+	 * is no point in keeping the device instantiated.
+	 */
+	if (!client->driver) {
+		i2c_unregister_device(client);
+		return -ENODEV;
+	}
+
+	/*
+	 * Let i2c-core delete that device on driver removal.
+	 * This is safe because i2c-core holds the core_lock mutex for us.
+	 */
+	list_add_tail(&client->detected, &client->driver->clients);
+	return 0;
+}
+
+static int tas_i2c_probe(struct i2c_client *client,
+			 const struct i2c_device_id *id)
+{
+	struct device_node *node = client->dev.platform_data;
+	struct tas *tas;
+
+	tas = kzalloc(sizeof(struct tas), GFP_KERNEL);
+
+	if (!tas)
+		return -ENOMEM;
+
+	mutex_init(&tas->mtx);
+	tas->i2c = client;
+	i2c_set_clientdata(client, tas);
+
+	/* seems that half is a saner default */
+	tas->drc_range = TAS3004_DRC_MAX / 2;
+
+	strlcpy(tas->codec.name, "tas", MAX_CODEC_NAME_LEN);
+	tas->codec.owner = THIS_MODULE;
+	tas->codec.init = tas_init_codec;
+	tas->codec.exit = tas_exit_codec;
+	tas->codec.node = of_node_get(node);
+
+	if (aoa_codec_register(&tas->codec)) {
+		goto fail;
+	}
+	printk(KERN_DEBUG
+	       "snd-aoa-codec-tas: tas found, addr 0x%02x on %s\n",
+	       (unsigned int)client->addr, node->full_name);
+	return 0;
+ fail:
+	mutex_destroy(&tas->mtx);
+	kfree(tas);
+	return -EINVAL;
+}
+
+static int tas_i2c_attach(struct i2c_adapter *adapter)
+{
+	struct device_node *busnode, *dev = NULL;
+	struct pmac_i2c_bus *bus;
+
+	bus = pmac_i2c_adapter_to_bus(adapter);
+	if (bus == NULL)
+		return -ENODEV;
+	busnode = pmac_i2c_get_bus_node(bus);
+
+	while ((dev = of_get_next_child(busnode, dev)) != NULL) {
+		if (of_device_is_compatible(dev, "tas3004")) {
+			const u32 *addr;
+			printk(KERN_DEBUG PFX "found tas3004\n");
+			addr = of_get_property(dev, "reg", NULL);
+			if (!addr)
+				continue;
+			return tas_create(adapter, dev, ((*addr) >> 1) & 0x7f);
+		}
+		/* older machines have no 'codec' node with a 'compatible'
+		 * property that says 'tas3004', they just have a 'deq'
+		 * node without any such property... */
+		if (strcmp(dev->name, "deq") == 0) {
+			const u32 *_addr;
+			u32 addr;
+			printk(KERN_DEBUG PFX "found 'deq' node\n");
+			_addr = of_get_property(dev, "i2c-address", NULL);
+			if (!_addr)
+				continue;
+			addr = ((*_addr) >> 1) & 0x7f;
+			/* now, if the address doesn't match any of the two
+			 * that a tas3004 can have, we cannot handle this.
+			 * I doubt it ever happens but hey. */
+			if (addr != 0x34 && addr != 0x35)
+				continue;
+			return tas_create(adapter, dev, addr);
+		}
+	}
+	return -ENODEV;
+}
+
+static int tas_i2c_remove(struct i2c_client *client)
+{
+	struct tas *tas = i2c_get_clientdata(client);
+	u8 tmp = TAS_ACR_ANALOG_PDOWN;
+
+	aoa_codec_unregister(&tas->codec);
+	of_node_put(tas->codec.node);
+
+	/* power down codec chip */
+	tas_write_reg(tas, TAS_REG_ACR, 1, &tmp);
+
+	mutex_destroy(&tas->mtx);
+	kfree(tas);
+	return 0;
+}
+
+static const struct i2c_device_id tas_i2c_id[] = {
+	{ "aoa_codec_tas", 0 },
+	{ }
+};
+
+static struct i2c_driver tas_driver = {
+	.driver = {
+		.name = "aoa_codec_tas",
+		.owner = THIS_MODULE,
+	},
+	.attach_adapter = tas_i2c_attach,
+	.probe = tas_i2c_probe,
+	.remove = tas_i2c_remove,
+	.id_table = tas_i2c_id,
+};
+
+module_i2c_driver(tas_driver);
diff --git a/ANDROID_3.4.5/sound/aoa/codecs/tas.h b/ANDROID_3.4.5/sound/aoa/codecs/tas.h
new file mode 100644
index 00000000..ae177e34
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/codecs/tas.h
@@ -0,0 +1,55 @@
+/*
+ * Apple Onboard Audio driver for tas codec (header)
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+#ifndef __SND_AOA_CODECTASH
+#define __SND_AOA_CODECTASH
+
+#define TAS_REG_MCS	0x01	/* main control */
+#	define TAS_MCS_FASTLOAD		(1<<7)
+#	define TAS_MCS_SCLK64		(1<<6)
+#	define TAS_MCS_SPORT_MODE_MASK	(3<<4)
+#	define TAS_MCS_SPORT_MODE_I2S	(2<<4)
+#	define TAS_MCS_SPORT_MODE_RJ	(1<<4)
+#	define TAS_MCS_SPORT_MODE_LJ	(0<<4)
+#	define TAS_MCS_SPORT_WL_MASK	(3<<0)
+#	define TAS_MCS_SPORT_WL_16BIT	(0<<0)
+#	define TAS_MCS_SPORT_WL_18BIT	(1<<0)
+#	define TAS_MCS_SPORT_WL_20BIT	(2<<0)
+#	define TAS_MCS_SPORT_WL_24BIT	(3<<0)
+
+#define TAS_REG_DRC	0x02
+#define TAS_REG_VOL	0x04
+#define TAS_REG_TREBLE	0x05
+#define TAS_REG_BASS	0x06
+#define TAS_REG_LMIX	0x07
+#define TAS_REG_RMIX	0x08
+
+#define TAS_REG_ACR	0x40	/* analog control */
+#	define TAS_ACR_B_MONAUREAL	(1<<7)
+#	define TAS_ACR_B_MON_SEL_RIGHT	(1<<6)
+#	define TAS_ACR_DEEMPH_MASK	(3<<2)
+#	define TAS_ACR_DEEMPH_OFF	(0<<2)
+#	define TAS_ACR_DEEMPH_48KHz	(1<<2)
+#	define TAS_ACR_DEEMPH_44KHz	(2<<2)
+#	define TAS_ACR_INPUT_B		(1<<1)
+#	define TAS_ACR_ANALOG_PDOWN	(1<<0)
+
+#define TAS_REG_MCS2	0x43	/* main control 2 */
+#	define TAS_MCS2_ALLPASS		(1<<1)
+
+#define TAS_REG_LEFT_BIQUAD6	0x10
+#define TAS_REG_RIGHT_BIQUAD6	0x19
+
+#define TAS_REG_LEFT_LOUDNESS		0x21
+#define TAS_REG_RIGHT_LOUDNESS		0x22
+#define TAS_REG_LEFT_LOUDNESS_GAIN	0x23
+#define TAS_REG_RIGHT_LOUDNESS_GAIN	0x24
+
+#define TAS3001_DRC_MAX		0x5f
+#define TAS3004_DRC_MAX		0xef
+
+#endif /* __SND_AOA_CODECTASH */
diff --git a/ANDROID_3.4.5/sound/aoa/codecs/toonie.c b/ANDROID_3.4.5/sound/aoa/codecs/toonie.c
new file mode 100644
index 00000000..69d2cb60
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/codecs/toonie.c
@@ -0,0 +1,151 @@
+/*
+ * Apple Onboard Audio driver for Toonie codec
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ *
+ *
+ * This is a driver for the toonie codec chip. This chip is present
+ * on the Mac Mini and is nothing but a DAC.
+ */
+#include <linux/delay.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("toonie codec driver for snd-aoa");
+
+#include "../aoa.h"
+#include "../soundbus/soundbus.h"
+
+
+#define PFX "snd-aoa-codec-toonie: "
+
+struct toonie {
+	struct aoa_codec	codec;
+};
+#define codec_to_toonie(c) container_of(c, struct toonie, codec)
+
+static int toonie_dev_register(struct snd_device *dev)
+{
+	return 0;
+}
+
+static struct snd_device_ops ops = {
+	.dev_register = toonie_dev_register,
+};
+
+static struct transfer_info toonie_transfers[] = {
+	/* This thing *only* has analog output,
+	 * the rates are taken from Info.plist
+	 * from Darwin. */
+	{
+		.formats = SNDRV_PCM_FMTBIT_S16_BE |
+			   SNDRV_PCM_FMTBIT_S24_BE,
+		.rates = SNDRV_PCM_RATE_32000 |
+			 SNDRV_PCM_RATE_44100 |
+			 SNDRV_PCM_RATE_48000 |
+			 SNDRV_PCM_RATE_88200 |
+			 SNDRV_PCM_RATE_96000,
+	},
+	{}
+};
+
+static int toonie_usable(struct codec_info_item *cii,
+			 struct transfer_info *ti,
+			 struct transfer_info *out)
+{
+	return 1;
+}
+
+#ifdef CONFIG_PM
+static int toonie_suspend(struct codec_info_item *cii, pm_message_t state)
+{
+	/* can we turn it off somehow? */
+	return 0;
+}
+
+static int toonie_resume(struct codec_info_item *cii)
+{
+	return 0;
+}
+#endif /* CONFIG_PM */
+
+static struct codec_info toonie_codec_info = {
+	.transfers = toonie_transfers,
+	.sysclock_factor = 256,
+	.bus_factor = 64,
+	.owner = THIS_MODULE,
+	.usable = toonie_usable,
+#ifdef CONFIG_PM
+	.suspend = toonie_suspend,
+	.resume = toonie_resume,
+#endif
+};
+
+static int toonie_init_codec(struct aoa_codec *codec)
+{
+	struct toonie *toonie = codec_to_toonie(codec);
+
+	/* nothing connected? what a joke! */
+	if (toonie->codec.connected != 1)
+		return -ENOTCONN;
+
+	if (aoa_snd_device_new(SNDRV_DEV_LOWLEVEL, toonie, &ops)) {
+		printk(KERN_ERR PFX "failed to create toonie snd device!\n");
+		return -ENODEV;
+	}
+
+	if (toonie->codec.soundbus_dev->attach_codec(toonie->codec.soundbus_dev,
+						     aoa_get_card(),
+						     &toonie_codec_info, toonie)) {
+		printk(KERN_ERR PFX "error creating toonie pcm\n");
+		snd_device_free(aoa_get_card(), toonie);
+		return -ENODEV;
+	}
+
+	return 0;
+}
+
+static void toonie_exit_codec(struct aoa_codec *codec)
+{
+	struct toonie *toonie = codec_to_toonie(codec);
+
+	if (!toonie->codec.soundbus_dev) {
+		printk(KERN_ERR PFX "toonie_exit_codec called without soundbus_dev!\n");
+		return;
+	}
+	toonie->codec.soundbus_dev->detach_codec(toonie->codec.soundbus_dev, toonie);
+}
+
+static struct toonie *toonie;
+
+static int __init toonie_init(void)
+{
+	toonie = kzalloc(sizeof(struct toonie), GFP_KERNEL);
+
+	if (!toonie)
+		return -ENOMEM;
+
+	strlcpy(toonie->codec.name, "toonie", sizeof(toonie->codec.name));
+	toonie->codec.owner = THIS_MODULE;
+	toonie->codec.init = toonie_init_codec;
+	toonie->codec.exit = toonie_exit_codec;
+
+	if (aoa_codec_register(&toonie->codec)) {
+		kfree(toonie);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static void __exit toonie_exit(void)
+{
+	aoa_codec_unregister(&toonie->codec);
+	kfree(toonie);
+}
+
+module_init(toonie_init);
+module_exit(toonie_exit);
diff --git a/ANDROID_3.4.5/sound/aoa/core/Makefile b/ANDROID_3.4.5/sound/aoa/core/Makefile
new file mode 100644
index 00000000..a1596e88
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/core/Makefile
@@ -0,0 +1,5 @@
+obj-$(CONFIG_SND_AOA) += snd-aoa.o
+snd-aoa-objs := core.o \
+		alsa.o \
+		gpio-pmf.o \
+		gpio-feature.o
diff --git a/ANDROID_3.4.5/sound/aoa/core/alsa.c b/ANDROID_3.4.5/sound/aoa/core/alsa.c
new file mode 100644
index 00000000..0fa3855b
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/core/alsa.c
@@ -0,0 +1,100 @@
+/*
+ * Apple Onboard Audio Alsa helpers
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+#include <linux/module.h>
+#include "alsa.h"
+
+static int index = -1;
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "index for AOA sound card.");
+
+static struct aoa_card *aoa_card;
+
+int aoa_alsa_init(char *name, struct module *mod, struct device *dev)
+{
+	struct snd_card *alsa_card;
+	int err;
+
+	if (aoa_card)
+		/* cannot be EEXIST due to usage in aoa_fabric_register */
+		return -EBUSY;
+
+	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;
+	strlcpy(alsa_card->driver, "AppleOnbdAudio", sizeof(alsa_card->driver));
+	strlcpy(alsa_card->shortname, name, sizeof(alsa_card->shortname));
+	strlcpy(alsa_card->longname, name, sizeof(alsa_card->longname));
+	strlcpy(alsa_card->mixername, name, sizeof(alsa_card->mixername));
+	err = snd_card_register(aoa_card->alsa_card);
+	if (err < 0) {
+		printk(KERN_ERR "snd-aoa: couldn't register alsa card\n");
+		snd_card_free(aoa_card->alsa_card);
+		aoa_card = NULL;
+		return err;
+	}
+	return 0;
+}
+
+struct snd_card *aoa_get_card(void)
+{
+	if (aoa_card)
+		return aoa_card->alsa_card;
+	return NULL;
+}
+EXPORT_SYMBOL_GPL(aoa_get_card);
+
+void aoa_alsa_cleanup(void)
+{
+	if (aoa_card) {
+		snd_card_free(aoa_card->alsa_card);
+		aoa_card = NULL;
+	}
+}
+
+int aoa_snd_device_new(snd_device_type_t type,
+		       void * device_data, struct snd_device_ops * ops)
+{
+	struct snd_card *card = aoa_get_card();
+	int err;
+
+	if (!card) return -ENOMEM;
+
+	err = snd_device_new(card, type, device_data, ops);
+	if (err) {
+		printk(KERN_ERR "snd-aoa: failed to create snd device (%d)\n", err);
+		return err;
+	}
+	err = snd_device_register(card, device_data);
+	if (err) {
+		printk(KERN_ERR "snd-aoa: failed to register "
+				"snd device (%d)\n", err);
+		printk(KERN_ERR "snd-aoa: have you forgotten the "
+				"dev_register callback?\n");
+		snd_device_free(card, device_data);
+	}
+	return err;
+}
+EXPORT_SYMBOL_GPL(aoa_snd_device_new);
+
+int aoa_snd_ctl_add(struct snd_kcontrol* control)
+{
+	int err;
+
+	if (!aoa_card) return -ENODEV;
+
+	err = snd_ctl_add(aoa_card->alsa_card, control);
+	if (err)
+		printk(KERN_ERR "snd-aoa: failed to add alsa control (%d)\n",
+		       err);
+	return err;
+}
+EXPORT_SYMBOL_GPL(aoa_snd_ctl_add);
diff --git a/ANDROID_3.4.5/sound/aoa/core/alsa.h b/ANDROID_3.4.5/sound/aoa/core/alsa.h
new file mode 100644
index 00000000..9669e448
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/core/alsa.h
@@ -0,0 +1,16 @@
+/*
+ * Apple Onboard Audio Alsa private helpers
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+
+#ifndef __SND_AOA_ALSA_H
+#define __SND_AOA_ALSA_H
+#include "../aoa.h"
+
+extern int aoa_alsa_init(char *name, struct module *mod, struct device *dev);
+extern void aoa_alsa_cleanup(void);
+
+#endif /* __SND_AOA_ALSA_H */
diff --git a/ANDROID_3.4.5/sound/aoa/core/core.c b/ANDROID_3.4.5/sound/aoa/core/core.c
new file mode 100644
index 00000000..10bec6c6
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/core/core.c
@@ -0,0 +1,162 @@
+/*
+ * Apple Onboard Audio driver core
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/list.h>
+#include "../aoa.h"
+#include "alsa.h"
+
+MODULE_DESCRIPTION("Apple Onboard Audio Sound Driver");
+MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
+MODULE_LICENSE("GPL");
+
+/* We allow only one fabric. This simplifies things,
+ * and more don't really make that much sense */
+static struct aoa_fabric *fabric;
+static LIST_HEAD(codec_list);
+
+static int attach_codec_to_fabric(struct aoa_codec *c)
+{
+	int err;
+
+	if (!try_module_get(c->owner))
+		return -EBUSY;
+	/* found_codec has to be assigned */
+	err = -ENOENT;
+	if (fabric->found_codec)
+		err = fabric->found_codec(c);
+	if (err) {
+		module_put(c->owner);
+		printk(KERN_ERR "snd-aoa: fabric didn't like codec %s\n",
+				c->name);
+		return err;
+	}
+	c->fabric = fabric;
+
+	err = 0;
+	if (c->init)
+		err = c->init(c);
+	if (err) {
+		printk(KERN_ERR "snd-aoa: codec %s didn't init\n", c->name);
+		c->fabric = NULL;
+		if (fabric->remove_codec)
+			fabric->remove_codec(c);
+		module_put(c->owner);
+		return err;
+	}
+	if (fabric->attached_codec)
+		fabric->attached_codec(c);
+	return 0;
+}
+
+int aoa_codec_register(struct aoa_codec *codec)
+{
+	int err = 0;
+
+	/* if there's a fabric already, we can tell if we
+	 * will want to have this codec, so propagate error
+	 * through. Otherwise, this will happen later... */
+	if (fabric)
+		err = attach_codec_to_fabric(codec);
+	if (!err)
+		list_add(&codec->list, &codec_list);
+	return err;
+}
+EXPORT_SYMBOL_GPL(aoa_codec_register);
+
+void aoa_codec_unregister(struct aoa_codec *codec)
+{
+	list_del(&codec->list);
+	if (codec->fabric && codec->exit)
+		codec->exit(codec);
+	if (fabric && fabric->remove_codec)
+		fabric->remove_codec(codec);
+	codec->fabric = NULL;
+	module_put(codec->owner);
+}
+EXPORT_SYMBOL_GPL(aoa_codec_unregister);
+
+int aoa_fabric_register(struct aoa_fabric *new_fabric, struct device *dev)
+{
+	struct aoa_codec *c;
+	int err;
+
+	/* allow querying for presence of fabric
+	 * (i.e. do this test first!) */
+	if (new_fabric == fabric) {
+		err = -EALREADY;
+		goto attach;
+	}
+	if (fabric)
+		return -EEXIST;
+	if (!new_fabric)
+		return -EINVAL;
+
+	err = aoa_alsa_init(new_fabric->name, new_fabric->owner, dev);
+	if (err)
+		return err;
+
+	fabric = new_fabric;
+
+ attach:
+	list_for_each_entry(c, &codec_list, list) {
+		if (c->fabric != fabric)
+			attach_codec_to_fabric(c);
+	}
+	return err;
+}
+EXPORT_SYMBOL_GPL(aoa_fabric_register);
+
+void aoa_fabric_unregister(struct aoa_fabric *old_fabric)
+{
+	struct aoa_codec *c;
+
+	if (fabric != old_fabric)
+		return;
+
+	list_for_each_entry(c, &codec_list, list) {
+		if (c->fabric)
+			aoa_fabric_unlink_codec(c);
+	}
+
+	aoa_alsa_cleanup();
+
+	fabric = NULL;
+}
+EXPORT_SYMBOL_GPL(aoa_fabric_unregister);
+
+void aoa_fabric_unlink_codec(struct aoa_codec *codec)
+{
+	if (!codec->fabric) {
+		printk(KERN_ERR "snd-aoa: fabric unassigned "
+				"in aoa_fabric_unlink_codec\n");
+		dump_stack();
+		return;
+	}
+	if (codec->exit)
+		codec->exit(codec);
+	if (codec->fabric->remove_codec)
+		codec->fabric->remove_codec(codec);
+	codec->fabric = NULL;
+	module_put(codec->owner);
+}
+EXPORT_SYMBOL_GPL(aoa_fabric_unlink_codec);
+
+static int __init aoa_init(void)
+{
+	return 0;
+}
+
+static void __exit aoa_exit(void)
+{
+	aoa_alsa_cleanup();
+}
+
+module_init(aoa_init);
+module_exit(aoa_exit);
diff --git a/ANDROID_3.4.5/sound/aoa/core/gpio-feature.c b/ANDROID_3.4.5/sound/aoa/core/gpio-feature.c
new file mode 100644
index 00000000..faa31749
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/core/gpio-feature.c
@@ -0,0 +1,422 @@
+/*
+ * Apple Onboard Audio feature call GPIO control
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ *
+ * This file contains the GPIO control routines for
+ * direct (through feature calls) access to the GPIO
+ * registers.
+ */
+
+#include <asm/pmac_feature.h>
+#include <linux/interrupt.h>
+#include "../aoa.h"
+
+/* TODO: these are lots of global variables
+ * that aren't used on most machines...
+ * Move them into a dynamically allocated
+ * structure and use that.
+ */
+
+/* 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;
+static int lineout_detect_gpio;
+static int headphone_detect_gpio;
+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;
+static int lineout_detect_gpio_activestate;
+static int headphone_detect_gpio_activestate;
+static int linein_detect_gpio_activestate;
+
+/* node pointers that we save when getting the GPIO number
+ * to get the interrupt later */
+static struct device_node *lineout_detect_node;
+static struct device_node *linein_detect_node;
+static struct device_node *headphone_detect_node;
+
+static int lineout_detect_irq;
+static int linein_detect_irq;
+static int headphone_detect_irq;
+
+static struct device_node *get_gpio(char *name,
+				    char *altname,
+				    int *gpioptr,
+				    int *gpioactiveptr)
+{
+	struct device_node *np, *gpio;
+	const u32 *reg;
+	const char *audio_gpio;
+
+	*gpioptr = -1;
+
+	/* check if we can get it the easy way ... */
+	np = of_find_node_by_name(NULL, name);
+	if (!np) {
+		/* some machines have only gpioX/extint-gpioX nodes,
+		 * and an audio-gpio property saying what it is ...
+		 * So what we have to do is enumerate all children
+		 * of the gpio node and check them all. */
+		gpio = of_find_node_by_name(NULL, "gpio");
+		if (!gpio)
+			return NULL;
+		while ((np = of_get_next_child(gpio, np))) {
+			audio_gpio = of_get_property(np, "audio-gpio", NULL);
+			if (!audio_gpio)
+				continue;
+			if (strcmp(audio_gpio, name) == 0)
+				break;
+			if (altname && (strcmp(audio_gpio, altname) == 0))
+				break;
+		}
+		/* still not found, assume not there */
+		if (!np)
+			return NULL;
+	}
+
+	reg = of_get_property(np, "reg", NULL);
+	if (!reg)
+		return NULL;
+
+	*gpioptr = *reg;
+
+	/* this is a hack, usually the GPIOs 'reg' property
+	 * should have the offset based from the GPIO space
+	 * which is at 0x50, but apparently not always... */
+	if (*gpioptr < 0x50)
+		*gpioptr += 0x50;
+
+	reg = of_get_property(np, "audio-gpio-active-state", NULL);
+	if (!reg)
+		/* Apple seems to default to 1, but
+		 * that doesn't seem right at least on most
+		 * machines. So until proven that the opposite
+		 * is necessary, we default to 0
+		 * (which, incidentally, snd-powermac also does...) */
+		*gpioactiveptr = 0;
+	else
+		*gpioactiveptr = *reg;
+
+	return np;
+}
+
+static void get_irq(struct device_node * np, int *irqptr)
+{
+	if (np)
+		*irqptr = irq_of_parse_and_map(np, 0);
+	else
+		*irqptr = NO_IRQ;
+}
+
+/* 0x4 is outenable, 0x1 is out, thus 4 or 5 */
+#define SWITCH_GPIO(name, v, on)				\
+	(((v)&~1) | ((on)?					\
+			(name##_gpio_activestate==0?4:5):	\
+			(name##_gpio_activestate==0?5:4)))
+
+#define FTR_GPIO(name, bit)					\
+static void ftr_gpio_set_##name(struct gpio_runtime *rt, int on)\
+{								\
+	int v;							\
+								\
+	if (unlikely(!rt)) return;				\
+								\
+	if (name##_mute_gpio < 0)				\
+		return;						\
+								\
+	v = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL,		\
+			      name##_mute_gpio,			\
+			      0);				\
+								\
+	/* muted = !on... */					\
+	v = SWITCH_GPIO(name##_mute, v, !on);			\
+								\
+	pmac_call_feature(PMAC_FTR_WRITE_GPIO, NULL,		\
+			  name##_mute_gpio, v);			\
+								\
+	rt->implementation_private &= ~(1<<bit);		\
+	rt->implementation_private |= (!!on << bit);		\
+}								\
+static int ftr_gpio_get_##name(struct gpio_runtime *rt)		\
+{								\
+	if (unlikely(!rt)) return 0;				\
+	return (rt->implementation_private>>bit)&1;		\
+}
+
+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)
+{
+	int v;
+
+	if (unlikely(!rt)) return;
+	if (hw_reset_gpio < 0)
+		return;
+
+	v = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL,
+			      hw_reset_gpio, 0);
+	v = SWITCH_GPIO(hw_reset, v, on);
+	pmac_call_feature(PMAC_FTR_WRITE_GPIO, NULL,
+			  hw_reset_gpio, v);
+}
+
+static struct gpio_methods methods;
+
+static void ftr_gpio_all_amps_off(struct gpio_runtime *rt)
+{
+	int saved;
+
+	if (unlikely(!rt)) return;
+	saved = rt->implementation_private;
+	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;
+}
+
+static void ftr_gpio_all_amps_restore(struct gpio_runtime *rt)
+{
+	int s;
+
+	if (unlikely(!rt)) return;
+	s = rt->implementation_private;
+	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)
+{
+	struct gpio_notification *notif =
+		container_of(work, struct gpio_notification, work.work);
+
+	mutex_lock(&notif->mutex);
+	if (notif->notify)
+		notif->notify(notif->data);
+	mutex_unlock(&notif->mutex);
+}
+
+static void gpio_enable_dual_edge(int gpio)
+{
+	int v;
+
+	if (gpio == -1)
+		return;
+	v = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, gpio, 0);
+	v |= 0x80; /* enable dual edge */
+	pmac_call_feature(PMAC_FTR_WRITE_GPIO, NULL, gpio, v);
+}
+
+static void ftr_gpio_init(struct gpio_runtime *rt)
+{
+	get_gpio("headphone-mute", NULL,
+		 &headphone_mute_gpio,
+		 &headphone_mute_gpio_activestate);
+	get_gpio("amp-mute", NULL,
+		 &amp_mute_gpio,
+		 &amp_mute_gpio_activestate);
+	get_gpio("lineout-mute", NULL,
+		 &lineout_mute_gpio,
+		 &lineout_mute_gpio_activestate);
+	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,
+					 &headphone_detect_gpio_activestate);
+	/* go Apple, and thanks for giving these different names
+	 * across the board... */
+	lineout_detect_node = get_gpio("lineout-detect", "line-output-detect",
+				       &lineout_detect_gpio,
+				       &lineout_detect_gpio_activestate);
+	linein_detect_node = get_gpio("linein-detect", "line-input-detect",
+				      &linein_detect_gpio,
+				      &linein_detect_gpio_activestate);
+
+	gpio_enable_dual_edge(headphone_detect_gpio);
+	gpio_enable_dual_edge(lineout_detect_gpio);
+	gpio_enable_dual_edge(linein_detect_gpio);
+
+	get_irq(headphone_detect_node, &headphone_detect_irq);
+	get_irq(lineout_detect_node, &lineout_detect_irq);
+	get_irq(linein_detect_node, &linein_detect_irq);
+
+	ftr_gpio_all_amps_off(rt);
+	rt->implementation_private = 0;
+	INIT_DELAYED_WORK(&rt->headphone_notify.work, ftr_handle_notify);
+	INIT_DELAYED_WORK(&rt->line_in_notify.work, ftr_handle_notify);
+	INIT_DELAYED_WORK(&rt->line_out_notify.work, ftr_handle_notify);
+	mutex_init(&rt->headphone_notify.mutex);
+	mutex_init(&rt->line_in_notify.mutex);
+	mutex_init(&rt->line_out_notify.mutex);
+}
+
+static void ftr_gpio_exit(struct gpio_runtime *rt)
+{
+	ftr_gpio_all_amps_off(rt);
+	rt->implementation_private = 0;
+	if (rt->headphone_notify.notify)
+		free_irq(headphone_detect_irq, &rt->headphone_notify);
+	if (rt->line_in_notify.gpio_private)
+		free_irq(linein_detect_irq, &rt->line_in_notify);
+	if (rt->line_out_notify.gpio_private)
+		free_irq(lineout_detect_irq, &rt->line_out_notify);
+	cancel_delayed_work_sync(&rt->headphone_notify.work);
+	cancel_delayed_work_sync(&rt->line_in_notify.work);
+	cancel_delayed_work_sync(&rt->line_out_notify.work);
+	mutex_destroy(&rt->headphone_notify.mutex);
+	mutex_destroy(&rt->line_in_notify.mutex);
+	mutex_destroy(&rt->line_out_notify.mutex);
+}
+
+static irqreturn_t ftr_handle_notify_irq(int xx, void *data)
+{
+	struct gpio_notification *notif = data;
+
+	schedule_delayed_work(&notif->work, 0);
+
+	return IRQ_HANDLED;
+}
+
+static int ftr_set_notify(struct gpio_runtime *rt,
+			  enum notify_type type,
+			  notify_func_t notify,
+			  void *data)
+{
+	struct gpio_notification *notif;
+	notify_func_t old;
+	int irq;
+	char *name;
+	int err = -EBUSY;
+
+	switch (type) {
+	case AOA_NOTIFY_HEADPHONE:
+		notif = &rt->headphone_notify;
+		name = "headphone-detect";
+		irq = headphone_detect_irq;
+		break;
+	case AOA_NOTIFY_LINE_IN:
+		notif = &rt->line_in_notify;
+		name = "linein-detect";
+		irq = linein_detect_irq;
+		break;
+	case AOA_NOTIFY_LINE_OUT:
+		notif = &rt->line_out_notify;
+		name = "lineout-detect";
+		irq = lineout_detect_irq;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (irq == NO_IRQ)
+		return -ENODEV;
+
+	mutex_lock(&notif->mutex);
+
+	old = notif->notify;
+
+	if (!old && !notify) {
+		err = 0;
+		goto out_unlock;
+	}
+
+	if (old && notify) {
+		if (old == notify && notif->data == data)
+			err = 0;
+		goto out_unlock;
+	}
+
+	if (old && !notify)
+		free_irq(irq, notif);
+
+	if (!old && notify) {
+		err = request_irq(irq, ftr_handle_notify_irq, 0, name, notif);
+		if (err)
+			goto out_unlock;
+	}
+
+	notif->notify = notify;
+	notif->data = data;
+
+	err = 0;
+ out_unlock:
+	mutex_unlock(&notif->mutex);
+	return err;
+}
+
+static int ftr_get_detect(struct gpio_runtime *rt,
+			  enum notify_type type)
+{
+	int gpio, ret, active;
+
+	switch (type) {
+	case AOA_NOTIFY_HEADPHONE:
+		gpio = headphone_detect_gpio;
+		active = headphone_detect_gpio_activestate;
+		break;
+	case AOA_NOTIFY_LINE_IN:
+		gpio = linein_detect_gpio;
+		active = linein_detect_gpio_activestate;
+		break;
+	case AOA_NOTIFY_LINE_OUT:
+		gpio = lineout_detect_gpio;
+		active = lineout_detect_gpio_activestate;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (gpio == -1)
+		return -ENODEV;
+
+	ret = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, gpio, 0);
+	if (ret < 0)
+		return ret;
+	return ((ret >> 1) & 1) == active;
+}
+
+static struct gpio_methods methods = {
+	.init			= ftr_gpio_init,
+	.exit			= ftr_gpio_exit,
+	.all_amps_off		= ftr_gpio_all_amps_off,
+	.all_amps_restore	= ftr_gpio_all_amps_restore,
+	.set_headphone		= ftr_gpio_set_headphone,
+	.set_speakers		= ftr_gpio_set_amp,
+	.set_lineout		= ftr_gpio_set_lineout,
+	.set_hw_reset		= ftr_gpio_set_hw_reset,
+	.get_headphone		= ftr_gpio_get_headphone,
+	.get_speakers		= ftr_gpio_get_amp,
+	.get_lineout		= ftr_gpio_get_lineout,
+	.set_notify		= ftr_set_notify,
+	.get_detect		= ftr_get_detect,
+};
+
+struct gpio_methods *ftr_gpio_methods = &methods;
+EXPORT_SYMBOL_GPL(ftr_gpio_methods);
diff --git a/ANDROID_3.4.5/sound/aoa/core/gpio-pmf.c b/ANDROID_3.4.5/sound/aoa/core/gpio-pmf.c
new file mode 100644
index 00000000..c8d8a1a6
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/core/gpio-pmf.c
@@ -0,0 +1,253 @@
+/*
+ * Apple Onboard Audio pmf GPIOs
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+
+#include <linux/slab.h>
+#include <asm/pmac_feature.h>
+#include <asm/pmac_pfunc.h>
+#include "../aoa.h"
+
+#define PMF_GPIO(name, bit)					\
+static void pmf_gpio_set_##name(struct gpio_runtime *rt, int on)\
+{								\
+	struct pmf_args args = { .count = 1, .u[0].v = !on };	\
+	int rc;							\
+							\
+	if (unlikely(!rt)) return;				\
+	rc = pmf_call_function(rt->node, #name "-mute", &args);	\
+	if (rc && rc != -ENODEV)				\
+		printk(KERN_WARNING "pmf_gpio_set_" #name	\
+		" failed, rc: %d\n", rc);			\
+	rt->implementation_private &= ~(1<<bit);		\
+	rt->implementation_private |= (!!on << bit);		\
+}								\
+static int pmf_gpio_get_##name(struct gpio_runtime *rt)		\
+{								\
+	if (unlikely(!rt)) return 0;				\
+	return (rt->implementation_private>>bit)&1;		\
+}
+
+PMF_GPIO(headphone, 0);
+PMF_GPIO(amp, 1);
+PMF_GPIO(lineout, 2);
+
+static void pmf_gpio_set_hw_reset(struct gpio_runtime *rt, int on)
+{
+	struct pmf_args args = { .count = 1, .u[0].v = !!on };
+	int rc;
+
+	if (unlikely(!rt)) return;
+	rc = pmf_call_function(rt->node, "hw-reset", &args);
+	if (rc)
+		printk(KERN_WARNING "pmf_gpio_set_hw_reset"
+		       " failed, rc: %d\n", rc);
+}
+
+static void pmf_gpio_all_amps_off(struct gpio_runtime *rt)
+{
+	int saved;
+
+	if (unlikely(!rt)) return;
+	saved = rt->implementation_private;
+	pmf_gpio_set_headphone(rt, 0);
+	pmf_gpio_set_amp(rt, 0);
+	pmf_gpio_set_lineout(rt, 0);
+	rt->implementation_private = saved;
+}
+
+static void pmf_gpio_all_amps_restore(struct gpio_runtime *rt)
+{
+	int s;
+
+	if (unlikely(!rt)) return;
+	s = rt->implementation_private;
+	pmf_gpio_set_headphone(rt, (s>>0)&1);
+	pmf_gpio_set_amp(rt, (s>>1)&1);
+	pmf_gpio_set_lineout(rt, (s>>2)&1);
+}
+
+static void pmf_handle_notify(struct work_struct *work)
+{
+	struct gpio_notification *notif =
+		container_of(work, struct gpio_notification, work.work);
+
+	mutex_lock(&notif->mutex);
+	if (notif->notify)
+		notif->notify(notif->data);
+	mutex_unlock(&notif->mutex);
+}
+
+static void pmf_gpio_init(struct gpio_runtime *rt)
+{
+	pmf_gpio_all_amps_off(rt);
+	rt->implementation_private = 0;
+	INIT_DELAYED_WORK(&rt->headphone_notify.work, pmf_handle_notify);
+	INIT_DELAYED_WORK(&rt->line_in_notify.work, pmf_handle_notify);
+	INIT_DELAYED_WORK(&rt->line_out_notify.work, pmf_handle_notify);
+	mutex_init(&rt->headphone_notify.mutex);
+	mutex_init(&rt->line_in_notify.mutex);
+	mutex_init(&rt->line_out_notify.mutex);
+}
+
+static void pmf_gpio_exit(struct gpio_runtime *rt)
+{
+	pmf_gpio_all_amps_off(rt);
+	rt->implementation_private = 0;
+
+	if (rt->headphone_notify.gpio_private)
+		pmf_unregister_irq_client(rt->headphone_notify.gpio_private);
+	if (rt->line_in_notify.gpio_private)
+		pmf_unregister_irq_client(rt->line_in_notify.gpio_private);
+	if (rt->line_out_notify.gpio_private)
+		pmf_unregister_irq_client(rt->line_out_notify.gpio_private);
+
+	/* make sure no work is pending before freeing
+	 * all things */
+	cancel_delayed_work_sync(&rt->headphone_notify.work);
+	cancel_delayed_work_sync(&rt->line_in_notify.work);
+	cancel_delayed_work_sync(&rt->line_out_notify.work);
+
+	mutex_destroy(&rt->headphone_notify.mutex);
+	mutex_destroy(&rt->line_in_notify.mutex);
+	mutex_destroy(&rt->line_out_notify.mutex);
+
+	kfree(rt->headphone_notify.gpio_private);
+	kfree(rt->line_in_notify.gpio_private);
+	kfree(rt->line_out_notify.gpio_private);
+}
+
+static void pmf_handle_notify_irq(void *data)
+{
+	struct gpio_notification *notif = data;
+
+	schedule_delayed_work(&notif->work, 0);
+}
+
+static int pmf_set_notify(struct gpio_runtime *rt,
+			  enum notify_type type,
+			  notify_func_t notify,
+			  void *data)
+{
+	struct gpio_notification *notif;
+	notify_func_t old;
+	struct pmf_irq_client *irq_client;
+	char *name;
+	int err = -EBUSY;
+
+	switch (type) {
+	case AOA_NOTIFY_HEADPHONE:
+		notif = &rt->headphone_notify;
+		name = "headphone-detect";
+		break;
+	case AOA_NOTIFY_LINE_IN:
+		notif = &rt->line_in_notify;
+		name = "linein-detect";
+		break;
+	case AOA_NOTIFY_LINE_OUT:
+		notif = &rt->line_out_notify;
+		name = "lineout-detect";
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	mutex_lock(&notif->mutex);
+
+	old = notif->notify;
+
+	if (!old && !notify) {
+		err = 0;
+		goto out_unlock;
+	}
+
+	if (old && notify) {
+		if (old == notify && notif->data == data)
+			err = 0;
+		goto out_unlock;
+	}
+
+	if (old && !notify) {
+		irq_client = notif->gpio_private;
+		pmf_unregister_irq_client(irq_client);
+		kfree(irq_client);
+		notif->gpio_private = NULL;
+	}
+	if (!old && notify) {
+		irq_client = kzalloc(sizeof(struct pmf_irq_client),
+				     GFP_KERNEL);
+		if (!irq_client) {
+			err = -ENOMEM;
+			goto out_unlock;
+		}
+		irq_client->data = notif;
+		irq_client->handler = pmf_handle_notify_irq;
+		irq_client->owner = THIS_MODULE;
+		err = pmf_register_irq_client(rt->node,
+					      name,
+					      irq_client);
+		if (err) {
+			printk(KERN_ERR "snd-aoa: gpio layer failed to"
+					" register %s irq (%d)\n", name, err);
+			kfree(irq_client);
+			goto out_unlock;
+		}
+		notif->gpio_private = irq_client;
+	}
+	notif->notify = notify;
+	notif->data = data;
+
+	err = 0;
+ out_unlock:
+	mutex_unlock(&notif->mutex);
+	return err;
+}
+
+static int pmf_get_detect(struct gpio_runtime *rt,
+			  enum notify_type type)
+{
+	char *name;
+	int err = -EBUSY, ret;
+	struct pmf_args args = { .count = 1, .u[0].p = &ret };
+
+	switch (type) {
+	case AOA_NOTIFY_HEADPHONE:
+		name = "headphone-detect";
+		break;
+	case AOA_NOTIFY_LINE_IN:
+		name = "linein-detect";
+		break;
+	case AOA_NOTIFY_LINE_OUT:
+		name = "lineout-detect";
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	err = pmf_call_function(rt->node, name, &args);
+	if (err)
+		return err;
+	return ret;
+}
+
+static struct gpio_methods methods = {
+	.init			= pmf_gpio_init,
+	.exit			= pmf_gpio_exit,
+	.all_amps_off		= pmf_gpio_all_amps_off,
+	.all_amps_restore	= pmf_gpio_all_amps_restore,
+	.set_headphone		= pmf_gpio_set_headphone,
+	.set_speakers		= pmf_gpio_set_amp,
+	.set_lineout		= pmf_gpio_set_lineout,
+	.set_hw_reset		= pmf_gpio_set_hw_reset,
+	.get_headphone		= pmf_gpio_get_headphone,
+	.get_speakers		= pmf_gpio_get_amp,
+	.get_lineout		= pmf_gpio_get_lineout,
+	.set_notify		= pmf_set_notify,
+	.get_detect		= pmf_get_detect,
+};
+
+struct gpio_methods *pmf_gpio_methods = &methods;
+EXPORT_SYMBOL_GPL(pmf_gpio_methods);
diff --git a/ANDROID_3.4.5/sound/aoa/fabrics/Kconfig b/ANDROID_3.4.5/sound/aoa/fabrics/Kconfig
new file mode 100644
index 00000000..3ca475a8
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/fabrics/Kconfig
@@ -0,0 +1,11 @@
+config SND_AOA_FABRIC_LAYOUT
+	tristate "layout-id fabric"
+	select SND_AOA_SOUNDBUS
+	select SND_AOA_SOUNDBUS_I2S
+	---help---
+	This enables the layout-id fabric for the Apple Onboard
+	Audio driver, the module holding it all together
+	based on the device-tree's layout-id property.
+	
+	If you are unsure and have a later Apple machine,
+	compile it as a module.
diff --git a/ANDROID_3.4.5/sound/aoa/fabrics/Makefile b/ANDROID_3.4.5/sound/aoa/fabrics/Makefile
new file mode 100644
index 00000000..da37c10e
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/fabrics/Makefile
@@ -0,0 +1,3 @@
+snd-aoa-fabric-layout-objs += layout.o
+
+obj-$(CONFIG_SND_AOA_FABRIC_LAYOUT) += snd-aoa-fabric-layout.o
diff --git a/ANDROID_3.4.5/sound/aoa/fabrics/layout.c b/ANDROID_3.4.5/sound/aoa/fabrics/layout.c
new file mode 100644
index 00000000..552b97af
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/fabrics/layout.c
@@ -0,0 +1,1168 @@
+/*
+ * Apple Onboard Audio driver -- layout/machine id fabric
+ *
+ * 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 or device-id property in the device tree.
+ */
+#include <asm/prom.h>
+#include <linux/list.h>
+#include <linux/module.h>
+#include <linux/slab.h>
+#include "../aoa.h"
+#include "../soundbus/soundbus.h"
+
+MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Layout-ID fabric for snd-aoa");
+
+#define MAX_CODECS_PER_BUS	2
+
+/* These are the connections the layout fabric
+ * knows about. It doesn't really care about the
+ * input ones, but I thought I'd separate them
+ * to give them proper names. The thing is that
+ * Apple usually will distinguish the active output
+ * by GPIOs, while the active input is set directly
+ * on the codec. Hence we here tell the codec what
+ * we think is connected. This information is hard-
+ * coded below ... */
+#define CC_SPEAKERS	(1<<0)
+#define CC_HEADPHONE	(1<<1)
+#define CC_LINEOUT	(1<<2)
+#define CC_DIGITALOUT	(1<<3)
+#define CC_LINEIN	(1<<4)
+#define CC_MICROPHONE	(1<<5)
+#define CC_DIGITALIN	(1<<6)
+/* pretty bogus but users complain...
+ * This is a flag saying that the LINEOUT
+ * should be renamed to HEADPHONE.
+ * be careful with input detection! */
+#define CC_LINEOUT_LABELLED_HEADPHONE	(1<<7)
+
+struct codec_connection {
+	/* CC_ flags from above */
+	int connected;
+	/* codec dependent bit to be set in the aoa_codec.connected field.
+	 * This intentionally doesn't have any generic flags because the
+	 * fabric has to know the codec anyway and all codecs might have
+	 * different connectors */
+	int codec_bit;
+};
+
+struct codec_connect_info {
+	char *name;
+	struct codec_connection *connections;
+};
+
+#define LAYOUT_FLAG_COMBO_LINEOUT_SPDIF	(1<<0)
+
+struct layout {
+	unsigned int layout_id, device_id;
+	struct codec_connect_info codecs[MAX_CODECS_PER_BUS];
+	int flags;
+
+	/* if busname is not assigned, we use 'Master' below,
+	 * so that our layout table doesn't need to be filled
+	 * too much.
+	 * We only assign these two if we expect to find more
+	 * than one soundbus, i.e. on those machines with
+	 * multiple layout-ids */
+	char *busname;
+	int pcmid;
+};
+
+MODULE_ALIAS("sound-layout-36");
+MODULE_ALIAS("sound-layout-41");
+MODULE_ALIAS("sound-layout-45");
+MODULE_ALIAS("sound-layout-47");
+MODULE_ALIAS("sound-layout-48");
+MODULE_ALIAS("sound-layout-49");
+MODULE_ALIAS("sound-layout-50");
+MODULE_ALIAS("sound-layout-51");
+MODULE_ALIAS("sound-layout-56");
+MODULE_ALIAS("sound-layout-57");
+MODULE_ALIAS("sound-layout-58");
+MODULE_ALIAS("sound-layout-60");
+MODULE_ALIAS("sound-layout-61");
+MODULE_ALIAS("sound-layout-62");
+MODULE_ALIAS("sound-layout-64");
+MODULE_ALIAS("sound-layout-65");
+MODULE_ALIAS("sound-layout-66");
+MODULE_ALIAS("sound-layout-67");
+MODULE_ALIAS("sound-layout-68");
+MODULE_ALIAS("sound-layout-69");
+MODULE_ALIAS("sound-layout-70");
+MODULE_ALIAS("sound-layout-72");
+MODULE_ALIAS("sound-layout-76");
+MODULE_ALIAS("sound-layout-80");
+MODULE_ALIAS("sound-layout-82");
+MODULE_ALIAS("sound-layout-84");
+MODULE_ALIAS("sound-layout-86");
+MODULE_ALIAS("sound-layout-90");
+MODULE_ALIAS("sound-layout-92");
+MODULE_ALIAS("sound-layout-94");
+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[] = {
+	{
+		.connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT,
+		.codec_bit = 0,
+	},
+	{
+		.connected = CC_DIGITALOUT,
+		.codec_bit = 1,
+	},
+	{
+		.connected = CC_LINEIN,
+		.codec_bit = 2,
+	},
+	{} /* terminate array by .connected == 0 */
+};
+
+/* onyx on machines without headphone */
+static struct codec_connection onyx_connections_noheadphones[] = {
+	{
+		.connected = CC_SPEAKERS | CC_LINEOUT |
+			     CC_LINEOUT_LABELLED_HEADPHONE,
+		.codec_bit = 0,
+	},
+	{
+		.connected = CC_DIGITALOUT,
+		.codec_bit = 1,
+	},
+	/* FIXME: are these correct? probably not for all the machines
+	 * below ... If not this will need separating. */
+	{
+		.connected = CC_LINEIN,
+		.codec_bit = 2,
+	},
+	{
+		.connected = CC_MICROPHONE,
+		.codec_bit = 3,
+	},
+	{} /* terminate array by .connected == 0 */
+};
+
+/* onyx on machines with real line-out */
+static struct codec_connection onyx_connections_reallineout[] = {
+	{
+		.connected = CC_SPEAKERS | CC_LINEOUT | CC_HEADPHONE,
+		.codec_bit = 0,
+	},
+	{
+		.connected = CC_DIGITALOUT,
+		.codec_bit = 1,
+	},
+	{
+		.connected = CC_LINEIN,
+		.codec_bit = 2,
+	},
+	{} /* terminate array by .connected == 0 */
+};
+
+/* tas on machines without line out */
+static struct codec_connection tas_connections_nolineout[] = {
+	{
+		.connected = CC_SPEAKERS | CC_HEADPHONE,
+		.codec_bit = 0,
+	},
+	{
+		.connected = CC_LINEIN,
+		.codec_bit = 2,
+	},
+	{
+		.connected = CC_MICROPHONE,
+		.codec_bit = 3,
+	},
+	{} /* terminate array by .connected == 0 */
+};
+
+/* tas on machines with neither line out nor line in */
+static struct codec_connection tas_connections_noline[] = {
+	{
+		.connected = CC_SPEAKERS | CC_HEADPHONE,
+		.codec_bit = 0,
+	},
+	{
+		.connected = CC_MICROPHONE,
+		.codec_bit = 3,
+	},
+	{} /* terminate array by .connected == 0 */
+};
+
+/* tas on machines without microphone */
+static struct codec_connection tas_connections_nomic[] = {
+	{
+		.connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT,
+		.codec_bit = 0,
+	},
+	{
+		.connected = CC_LINEIN,
+		.codec_bit = 2,
+	},
+	{} /* terminate array by .connected == 0 */
+};
+
+/* tas on machines with everything connected */
+static struct codec_connection tas_connections_all[] = {
+	{
+		.connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT,
+		.codec_bit = 0,
+	},
+	{
+		.connected = CC_LINEIN,
+		.codec_bit = 2,
+	},
+	{
+		.connected = CC_MICROPHONE,
+		.codec_bit = 3,
+	},
+	{} /* terminate array by .connected == 0 */
+};
+
+static struct codec_connection toonie_connections[] = {
+	{
+		.connected = CC_SPEAKERS | CC_HEADPHONE,
+		.codec_bit = 0,
+	},
+	{} /* terminate array by .connected == 0 */
+};
+
+static struct codec_connection topaz_input[] = {
+	{
+		.connected = CC_DIGITALIN,
+		.codec_bit = 0,
+	},
+	{} /* terminate array by .connected == 0 */
+};
+
+static struct codec_connection topaz_output[] = {
+	{
+		.connected = CC_DIGITALOUT,
+		.codec_bit = 1,
+	},
+	{} /* terminate array by .connected == 0 */
+};
+
+static struct codec_connection topaz_inout[] = {
+	{
+		.connected = CC_DIGITALIN,
+		.codec_bit = 0,
+	},
+	{
+		.connected = CC_DIGITALOUT,
+		.codec_bit = 1,
+	},
+	{} /* terminate array by .connected == 0 */
+};
+
+static struct layout layouts[] = {
+	/* last PowerBooks (15" Oct 2005) */
+	{ .layout_id = 82,
+	  .flags = LAYOUT_FLAG_COMBO_LINEOUT_SPDIF,
+	  .codecs[0] = {
+		.name = "onyx",
+		.connections = onyx_connections_noheadphones,
+	  },
+	  .codecs[1] = {
+		.name = "topaz",
+		.connections = topaz_input,
+	  },
+	},
+	/* PowerMac9,1 */
+	{ .layout_id = 60,
+	  .codecs[0] = {
+		.name = "onyx",
+		.connections = onyx_connections_reallineout,
+	  },
+	},
+	/* PowerMac9,1 */
+	{ .layout_id = 61,
+	  .codecs[0] = {
+		.name = "topaz",
+		.connections = topaz_input,
+	  },
+	},
+	/* PowerBook5,7 */
+	{ .layout_id = 64,
+	  .flags = LAYOUT_FLAG_COMBO_LINEOUT_SPDIF,
+	  .codecs[0] = {
+		.name = "onyx",
+		.connections = onyx_connections_noheadphones,
+	  },
+	},
+	/* PowerBook5,7 */
+	{ .layout_id = 65,
+	  .codecs[0] = {
+		.name = "topaz",
+		.connections = topaz_input,
+	  },
+	},
+	/* PowerBook5,9 [17" Oct 2005] */
+	{ .layout_id = 84,
+	  .flags = LAYOUT_FLAG_COMBO_LINEOUT_SPDIF,
+	  .codecs[0] = {
+		.name = "onyx",
+		.connections = onyx_connections_noheadphones,
+	  },
+	  .codecs[1] = {
+		.name = "topaz",
+		.connections = topaz_input,
+	  },
+	},
+	/* PowerMac8,1 */
+	{ .layout_id = 45,
+	  .codecs[0] = {
+		.name = "onyx",
+		.connections = onyx_connections_noheadphones,
+	  },
+	  .codecs[1] = {
+		.name = "topaz",
+		.connections = topaz_input,
+	  },
+	},
+	/* Quad PowerMac (analog in, analog/digital out) */
+	{ .layout_id = 68,
+	  .codecs[0] = {
+		.name = "onyx",
+		.connections = onyx_connections_nomic,
+	  },
+	},
+	/* Quad PowerMac (digital in) */
+	{ .layout_id = 69,
+	  .codecs[0] = {
+		.name = "topaz",
+		.connections = topaz_input,
+	  },
+	  .busname = "digital in", .pcmid = 1 },
+	/* Early 2005 PowerBook (PowerBook 5,6) */
+	{ .layout_id = 70,
+	  .codecs[0] = {
+		.name = "tas",
+		.connections = tas_connections_nolineout,
+	  },
+	},
+	/* PowerBook 5,4 */
+	{ .layout_id = 51,
+	  .codecs[0] = {
+		.name = "tas",
+		.connections = tas_connections_nolineout,
+	  },
+	},
+	/* PowerBook6,7 */
+	{ .layout_id = 80,
+	  .codecs[0] = {
+		.name = "tas",
+		.connections = tas_connections_noline,
+	  },
+	},
+	/* PowerBook6,8 */
+	{ .layout_id = 72,
+	  .codecs[0] = {
+		.name = "tas",
+		.connections = tas_connections_nolineout,
+	  },
+	},
+	/* PowerMac8,2 */
+	{ .layout_id = 86,
+	  .codecs[0] = {
+		.name = "onyx",
+		.connections = onyx_connections_nomic,
+	  },
+	  .codecs[1] = {
+		.name = "topaz",
+		.connections = topaz_input,
+	  },
+	},
+	/* PowerBook6,7 */
+	{ .layout_id = 92,
+	  .codecs[0] = {
+		.name = "tas",
+		.connections = tas_connections_nolineout,
+	  },
+	},
+	/* PowerMac10,1 (Mac Mini) */
+	{ .layout_id = 58,
+	  .codecs[0] = {
+		.name = "toonie",
+		.connections = toonie_connections,
+	  },
+	},
+	{
+	  .layout_id = 96,
+	  .codecs[0] = {
+	  	.name = "onyx",
+	  	.connections = onyx_connections_noheadphones,
+	  },
+	},
+	/* unknown, untested, but this comes from Apple */
+	{ .layout_id = 41,
+	  .codecs[0] = {
+		.name = "tas",
+		.connections = tas_connections_all,
+	  },
+	},
+	{ .layout_id = 36,
+	  .codecs[0] = {
+		.name = "tas",
+		.connections = tas_connections_nomic,
+	  },
+	  .codecs[1] = {
+		.name = "topaz",
+		.connections = topaz_inout,
+	  },
+	},
+	{ .layout_id = 47,
+	  .codecs[0] = {
+		.name = "onyx",
+		.connections = onyx_connections_noheadphones,
+	  },
+	},
+	{ .layout_id = 48,
+	  .codecs[0] = {
+		.name = "topaz",
+		.connections = topaz_input,
+	  },
+	},
+	{ .layout_id = 49,
+	  .codecs[0] = {
+		.name = "onyx",
+		.connections = onyx_connections_nomic,
+	  },
+	},
+	{ .layout_id = 50,
+	  .codecs[0] = {
+		.name = "topaz",
+		.connections = topaz_input,
+	  },
+	},
+	{ .layout_id = 56,
+	  .codecs[0] = {
+		.name = "onyx",
+		.connections = onyx_connections_noheadphones,
+	  },
+	},
+	{ .layout_id = 57,
+	  .codecs[0] = {
+		.name = "topaz",
+		.connections = topaz_input,
+	  },
+	},
+	{ .layout_id = 62,
+	  .codecs[0] = {
+		.name = "onyx",
+		.connections = onyx_connections_noheadphones,
+	  },
+	  .codecs[1] = {
+		.name = "topaz",
+		.connections = topaz_output,
+	  },
+	},
+	{ .layout_id = 66,
+	  .codecs[0] = {
+		.name = "onyx",
+		.connections = onyx_connections_noheadphones,
+	  },
+	},
+	{ .layout_id = 67,
+	  .codecs[0] = {
+		.name = "topaz",
+		.connections = topaz_input,
+	  },
+	},
+	{ .layout_id = 76,
+	  .codecs[0] = {
+		.name = "tas",
+		.connections = tas_connections_nomic,
+	  },
+	  .codecs[1] = {
+		.name = "topaz",
+		.connections = topaz_inout,
+	  },
+	},
+	{ .layout_id = 90,
+	  .codecs[0] = {
+		.name = "tas",
+		.connections = tas_connections_noline,
+	  },
+	},
+	{ .layout_id = 94,
+	  .codecs[0] = {
+		.name = "onyx",
+		/* but it has an external mic?? how to select? */
+		.connections = onyx_connections_noheadphones,
+	  },
+	},
+	{ .layout_id = 98,
+	  .codecs[0] = {
+		.name = "toonie",
+		.connections = toonie_connections,
+	  },
+	},
+	{ .layout_id = 100,
+	  .codecs[0] = {
+		.name = "topaz",
+		.connections = topaz_input,
+	  },
+	  .codecs[1] = {
+		.name = "onyx",
+		.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,
+	  },
+	},
+	{}
+};
+
+static struct layout *find_layout_by_id(unsigned int id)
+{
+	struct layout *l;
+
+	l = layouts;
+	while (l->codecs[0].name) {
+		if (l->layout_id == id)
+			return l;
+		l++;
+	}
+	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;
+
+	for (i=0; i<MAX_CODECS_PER_BUS; i++) {
+		if (l->codecs[i].name) {
+			request_module("snd-aoa-codec-%s", l->codecs[i].name);
+		}
+	}
+	/* now we wait for the codecs to call us back */
+}
+
+struct layout_dev;
+
+struct layout_dev_ptr {
+	struct layout_dev *ptr;
+};
+
+struct layout_dev {
+	struct list_head list;
+	struct soundbus_dev *sdev;
+	struct device_node *sound;
+	struct aoa_codec *codecs[MAX_CODECS_PER_BUS];
+	struct layout *layout;
+	struct gpio_runtime gpio;
+
+	/* we need these for headphone/lineout detection */
+	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;
+
+	struct layout_dev_ptr selfptr_headphone;
+	struct layout_dev_ptr selfptr_lineout;
+
+	u32 have_lineout_detect:1,
+	    have_headphone_detect:1,
+	    switch_on_headphone:1,
+	    switch_on_lineout:1;
+};
+
+static LIST_HEAD(layouts_list);
+static int layouts_list_items;
+/* this can go away but only if we allow multiple cards,
+ * make the fabric handle all the card stuff, etc... */
+static struct layout_dev *layout_device;
+
+#define control_info	snd_ctl_boolean_mono_info
+
+#define AMP_CONTROL(n, description)					\
+static int n##_control_get(struct snd_kcontrol *kcontrol,		\
+			   struct snd_ctl_elem_value *ucontrol)		\
+{									\
+	struct gpio_runtime *gpio = snd_kcontrol_chip(kcontrol);	\
+	if (gpio->methods && gpio->methods->get_##n)			\
+		ucontrol->value.integer.value[0] =			\
+			gpio->methods->get_##n(gpio);			\
+	return 0;							\
+}									\
+static int n##_control_put(struct snd_kcontrol *kcontrol,		\
+			   struct snd_ctl_elem_value *ucontrol)		\
+{									\
+	struct gpio_runtime *gpio = snd_kcontrol_chip(kcontrol);	\
+	if (gpio->methods && gpio->methods->get_##n)			\
+		gpio->methods->set_##n(gpio,				\
+			!!ucontrol->value.integer.value[0]);		\
+	return 1;							\
+}									\
+static struct snd_kcontrol_new n##_ctl = {				\
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,				\
+	.name = description,						\
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,                      \
+	.info = control_info,						\
+	.get = n##_control_get,						\
+	.put = n##_control_put,						\
+}
+
+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)
+{
+	struct layout_dev *ldev = snd_kcontrol_chip(kcontrol);
+
+	switch (kcontrol->private_value) {
+	case 0:
+		ucontrol->value.integer.value[0] = ldev->switch_on_headphone;
+		break;
+	case 1:
+		ucontrol->value.integer.value[0] = ldev->switch_on_lineout;
+		break;
+	default:
+		return -ENODEV;
+	}
+	return 0;
+}
+
+static int detect_choice_put(struct snd_kcontrol *kcontrol,
+			     struct snd_ctl_elem_value *ucontrol)
+{
+	struct layout_dev *ldev = snd_kcontrol_chip(kcontrol);
+
+	switch (kcontrol->private_value) {
+	case 0:
+		ldev->switch_on_headphone = !!ucontrol->value.integer.value[0];
+		break;
+	case 1:
+		ldev->switch_on_lineout = !!ucontrol->value.integer.value[0];
+		break;
+	default:
+		return -ENODEV;
+	}
+	return 1;
+}
+
+static struct snd_kcontrol_new headphone_detect_choice = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Headphone Detect Autoswitch",
+	.info = control_info,
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.get = detect_choice_get,
+	.put = detect_choice_put,
+	.private_value = 0,
+};
+
+static struct snd_kcontrol_new lineout_detect_choice = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Line-Out Detect Autoswitch",
+	.info = control_info,
+	.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.get = detect_choice_get,
+	.put = detect_choice_put,
+	.private_value = 1,
+};
+
+static int detected_get(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct layout_dev *ldev = snd_kcontrol_chip(kcontrol);
+	int v;
+
+	switch (kcontrol->private_value) {
+	case 0:
+		v = ldev->gpio.methods->get_detect(&ldev->gpio,
+						   AOA_NOTIFY_HEADPHONE);
+		break;
+	case 1:
+		v = ldev->gpio.methods->get_detect(&ldev->gpio,
+						   AOA_NOTIFY_LINE_OUT);
+		break;
+	default:
+		return -ENODEV;
+	}
+	ucontrol->value.integer.value[0] = v;
+	return 0;
+}
+
+static struct snd_kcontrol_new headphone_detected = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Headphone Detected",
+	.info = control_info,
+	.access = SNDRV_CTL_ELEM_ACCESS_READ,
+	.get = detected_get,
+	.private_value = 0,
+};
+
+static struct snd_kcontrol_new lineout_detected = {
+	.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name = "Line-Out Detected",
+	.info = control_info,
+	.access = SNDRV_CTL_ELEM_ACCESS_READ,
+	.get = detected_get,
+	.private_value = 1,
+};
+
+static int check_codec(struct aoa_codec *codec,
+		       struct layout_dev *ldev,
+		       struct codec_connect_info *cci)
+{
+	const u32 *ref;
+	char propname[32];
+	struct codec_connection *cc;
+
+	/* if the codec has a 'codec' node, we require a reference */
+	if (codec->node && (strcmp(codec->node->name, "codec") == 0)) {
+		snprintf(propname, sizeof(propname),
+			 "platform-%s-codec-ref", codec->name);
+		ref = of_get_property(ldev->sound, propname, NULL);
+		if (!ref) {
+			printk(KERN_INFO "snd-aoa-fabric-layout: "
+				"required property %s not present\n", propname);
+			return -ENODEV;
+		}
+		if (*ref != codec->node->phandle) {
+			printk(KERN_INFO "snd-aoa-fabric-layout: "
+				"%s doesn't match!\n", propname);
+			return -ENODEV;
+		}
+	} else {
+		if (layouts_list_items != 1) {
+			printk(KERN_INFO "snd-aoa-fabric-layout: "
+				"more than one soundbus, but no references.\n");
+			return -ENODEV;
+		}
+	}
+	codec->soundbus_dev = ldev->sdev;
+	codec->gpio = &ldev->gpio;
+
+	cc = cci->connections;
+	if (!cc)
+		return -EINVAL;
+
+	printk(KERN_INFO "snd-aoa-fabric-layout: can use this codec\n");
+
+	codec->connected = 0;
+	codec->fabric_data = cc;
+
+	while (cc->connected) {
+		codec->connected |= 1<<cc->codec_bit;
+		cc++;
+	}
+
+	return 0;
+}
+
+static int layout_found_codec(struct aoa_codec *codec)
+{
+	struct layout_dev *ldev;
+	int i;
+
+	list_for_each_entry(ldev, &layouts_list, list) {
+		for (i=0; i<MAX_CODECS_PER_BUS; i++) {
+			if (!ldev->layout->codecs[i].name)
+				continue;
+			if (strcmp(ldev->layout->codecs[i].name, codec->name) == 0) {
+				if (check_codec(codec,
+						ldev,
+						&ldev->layout->codecs[i]) == 0)
+					return 0;
+			}
+		}
+	}
+	return -ENODEV;
+}
+
+static void layout_remove_codec(struct aoa_codec *codec)
+{
+	int i;
+	/* here remove the codec from the layout dev's
+	 * codec reference */
+
+	codec->soundbus_dev = NULL;
+	codec->gpio = NULL;
+	for (i=0; i<MAX_CODECS_PER_BUS; i++) {
+	}
+}
+
+static void layout_notify(void *data)
+{
+	struct layout_dev_ptr *dptr = data;
+	struct layout_dev *ldev;
+	int v, update;
+	struct snd_kcontrol *detected, *c;
+	struct snd_card *card = aoa_get_card();
+
+	ldev = dptr->ptr;
+	if (data == &ldev->selfptr_headphone) {
+		v = ldev->gpio.methods->get_detect(&ldev->gpio, AOA_NOTIFY_HEADPHONE);
+		detected = ldev->headphone_detected_ctrl;
+		update = ldev->switch_on_headphone;
+		if (update) {
+			ldev->gpio.methods->set_speakers(&ldev->gpio, !v);
+			ldev->gpio.methods->set_headphone(&ldev->gpio, v);
+			ldev->gpio.methods->set_lineout(&ldev->gpio, 0);
+		}
+	} else if (data == &ldev->selfptr_lineout) {
+		v = ldev->gpio.methods->get_detect(&ldev->gpio, AOA_NOTIFY_LINE_OUT);
+		detected = ldev->lineout_detected_ctrl;
+		update = ldev->switch_on_lineout;
+		if (update) {
+			ldev->gpio.methods->set_speakers(&ldev->gpio, !v);
+			ldev->gpio.methods->set_headphone(&ldev->gpio, 0);
+			ldev->gpio.methods->set_lineout(&ldev->gpio, v);
+		}
+	} else
+		return;
+
+	if (detected)
+		snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &detected->id);
+	if (update) {
+		c = ldev->headphone_ctrl;
+		if (c)
+			snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &c->id);
+		c = ldev->speaker_ctrl;
+		if (c)
+			snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &c->id);
+		c = ldev->lineout_ctrl;
+		if (c)
+			snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &c->id);
+	}
+}
+
+static void layout_attached_codec(struct aoa_codec *codec)
+{
+	struct codec_connection *cc;
+	struct snd_kcontrol *ctl;
+	int headphones, lineout;
+	struct layout_dev *ldev = layout_device;
+
+	/* need to add this codec to our codec array! */
+
+	cc = codec->fabric_data;
+
+	headphones = codec->gpio->methods->get_detect(codec->gpio,
+						      AOA_NOTIFY_HEADPHONE);
+ 	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)
+				ldev->gpio.methods->set_speakers(codec->gpio, 1);
+			ctl = snd_ctl_new1(&speakers_ctl, codec->gpio);
+			ldev->speaker_ctrl = ctl;
+			aoa_snd_ctl_add(ctl);
+		}
+		if (cc->connected & CC_HEADPHONE) {
+			if (headphones == 1)
+				ldev->gpio.methods->set_headphone(codec->gpio, 1);
+			ctl = snd_ctl_new1(&headphone_ctl, codec->gpio);
+			ldev->headphone_ctrl = ctl;
+			aoa_snd_ctl_add(ctl);
+			ldev->have_headphone_detect =
+				!ldev->gpio.methods
+					->set_notify(&ldev->gpio,
+						     AOA_NOTIFY_HEADPHONE,
+						     layout_notify,
+						     &ldev->selfptr_headphone);
+			if (ldev->have_headphone_detect) {
+				ctl = snd_ctl_new1(&headphone_detect_choice,
+						   ldev);
+				aoa_snd_ctl_add(ctl);
+				ctl = snd_ctl_new1(&headphone_detected,
+						   ldev);
+				ldev->headphone_detected_ctrl = ctl;
+				aoa_snd_ctl_add(ctl);
+			}
+		}
+		if (cc->connected & CC_LINEOUT) {
+			if (lineout == 1)
+				ldev->gpio.methods->set_lineout(codec->gpio, 1);
+			ctl = snd_ctl_new1(&lineout_ctl, codec->gpio);
+			if (cc->connected & CC_LINEOUT_LABELLED_HEADPHONE)
+				strlcpy(ctl->id.name,
+					"Headphone Switch", sizeof(ctl->id.name));
+			ldev->lineout_ctrl = ctl;
+			aoa_snd_ctl_add(ctl);
+			ldev->have_lineout_detect =
+				!ldev->gpio.methods
+					->set_notify(&ldev->gpio,
+						     AOA_NOTIFY_LINE_OUT,
+						     layout_notify,
+						     &ldev->selfptr_lineout);
+			if (ldev->have_lineout_detect) {
+				ctl = snd_ctl_new1(&lineout_detect_choice,
+						   ldev);
+				if (cc->connected & CC_LINEOUT_LABELLED_HEADPHONE)
+					strlcpy(ctl->id.name,
+						"Headphone Detect Autoswitch",
+						sizeof(ctl->id.name));
+				aoa_snd_ctl_add(ctl);
+				ctl = snd_ctl_new1(&lineout_detected,
+						   ldev);
+				if (cc->connected & CC_LINEOUT_LABELLED_HEADPHONE)
+					strlcpy(ctl->id.name,
+						"Headphone Detected",
+						sizeof(ctl->id.name));
+				ldev->lineout_detected_ctrl = ctl;
+				aoa_snd_ctl_add(ctl);
+			}
+		}
+		cc++;
+	}
+	/* now update initial state */
+	if (ldev->have_headphone_detect)
+		layout_notify(&ldev->selfptr_headphone);
+	if (ldev->have_lineout_detect)
+		layout_notify(&ldev->selfptr_lineout);
+}
+
+static struct aoa_fabric layout_fabric = {
+	.name = "SoundByLayout",
+	.owner = THIS_MODULE,
+	.found_codec = layout_found_codec,
+	.remove_codec = layout_remove_codec,
+	.attached_codec = layout_attached_codec,
+};
+
+static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
+{
+	struct device_node *sound = NULL;
+	const unsigned int *id;
+	struct layout *layout = NULL;
+	struct layout_dev *ldev = NULL;
+	int err;
+
+	/* hm, currently we can only have one ... */
+	if (layout_device)
+		return -ENODEV;
+
+	/* by breaking out we keep a reference */
+	while ((sound = of_get_next_child(sdev->ofdev.dev.of_node, sound))) {
+		if (sound->type && strcasecmp(sound->type, "soundchip") == 0)
+			break;
+	}
+	if (!sound)
+		return -ENODEV;
+
+	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);
+	}
+
+	if (!layout) {
+		printk(KERN_ERR "snd-aoa-fabric-layout: unknown layout\n");
+		goto outnodev;
+	}
+
+	ldev = kzalloc(sizeof(struct layout_dev), GFP_KERNEL);
+	if (!ldev)
+		goto outnodev;
+
+	layout_device = ldev;
+	ldev->sdev = sdev;
+	ldev->sound = sound;
+	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 */
+		ldev->gpio.methods = ftr_gpio_methods;
+		printk(KERN_DEBUG
+		       "snd-aoa-fabric-layout: Using direct GPIOs\n");
+		break;
+	default:
+		ldev->gpio.methods = pmf_gpio_methods;
+		printk(KERN_DEBUG
+		       "snd-aoa-fabric-layout: Using PMF GPIOs\n");
+	}
+	ldev->selfptr_headphone.ptr = ldev;
+	ldev->selfptr_lineout.ptr = ldev;
+	dev_set_drvdata(&sdev->ofdev.dev, ldev);
+	list_add(&ldev->list, &layouts_list);
+	layouts_list_items++;
+
+	/* assign these before registering ourselves, so
+	 * callbacks that are done during registration
+	 * already have the values */
+	sdev->pcmid = ldev->layout->pcmid;
+	if (ldev->layout->busname) {
+		sdev->pcmname = ldev->layout->busname;
+	} else {
+		sdev->pcmname = "Master";
+	}
+
+	ldev->gpio.methods->init(&ldev->gpio);
+
+	err = aoa_fabric_register(&layout_fabric, &sdev->ofdev.dev);
+	if (err && err != -EALREADY) {
+		printk(KERN_INFO "snd-aoa-fabric-layout: can't use,"
+				 " another fabric is active!\n");
+		goto outlistdel;
+	}
+
+	use_layout(layout);
+	ldev->switch_on_headphone = 1;
+	ldev->switch_on_lineout = 1;
+	return 0;
+ outlistdel:
+	/* we won't be using these then... */
+	ldev->gpio.methods->exit(&ldev->gpio);
+	/* reset if we didn't use it */
+	sdev->pcmname = NULL;
+	sdev->pcmid = -1;
+	list_del(&ldev->list);
+	layouts_list_items--;
+	kfree(ldev);
+ outnodev:
+ 	of_node_put(sound);
+ 	layout_device = NULL;
+	return -ENODEV;
+}
+
+static int aoa_fabric_layout_remove(struct soundbus_dev *sdev)
+{
+	struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev);
+	int i;
+
+	for (i=0; i<MAX_CODECS_PER_BUS; i++) {
+		if (ldev->codecs[i]) {
+			aoa_fabric_unlink_codec(ldev->codecs[i]);
+		}
+		ldev->codecs[i] = NULL;
+	}
+	list_del(&ldev->list);
+	layouts_list_items--;
+	of_node_put(ldev->sound);
+
+	ldev->gpio.methods->set_notify(&ldev->gpio,
+				       AOA_NOTIFY_HEADPHONE,
+				       NULL,
+				       NULL);
+	ldev->gpio.methods->set_notify(&ldev->gpio,
+				       AOA_NOTIFY_LINE_OUT,
+				       NULL,
+				       NULL);
+
+	ldev->gpio.methods->exit(&ldev->gpio);
+	layout_device = NULL;
+	kfree(ldev);
+	sdev->pcmid = -1;
+	sdev->pcmname = NULL;
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int aoa_fabric_layout_suspend(struct soundbus_dev *sdev, pm_message_t state)
+{
+	struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev);
+
+	if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off)
+		ldev->gpio.methods->all_amps_off(&ldev->gpio);
+
+	return 0;
+}
+
+static int aoa_fabric_layout_resume(struct soundbus_dev *sdev)
+{
+	struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev);
+
+	if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off)
+		ldev->gpio.methods->all_amps_restore(&ldev->gpio);
+
+	return 0;
+}
+#endif
+
+static struct soundbus_driver aoa_soundbus_driver = {
+	.name = "snd_aoa_soundbus_drv",
+	.owner = THIS_MODULE,
+	.probe = aoa_fabric_layout_probe,
+	.remove = aoa_fabric_layout_remove,
+#ifdef CONFIG_PM
+	.suspend = aoa_fabric_layout_suspend,
+	.resume = aoa_fabric_layout_resume,
+#endif
+	.driver = {
+		.owner = THIS_MODULE,
+	}
+};
+
+static int __init aoa_fabric_layout_init(void)
+{
+	int err;
+
+	err = soundbus_register_driver(&aoa_soundbus_driver);
+	if (err)
+		return err;
+	return 0;
+}
+
+static void __exit aoa_fabric_layout_exit(void)
+{
+	soundbus_unregister_driver(&aoa_soundbus_driver);
+	aoa_fabric_unregister(&layout_fabric);
+}
+
+module_init(aoa_fabric_layout_init);
+module_exit(aoa_fabric_layout_exit);
diff --git a/ANDROID_3.4.5/sound/aoa/soundbus/Kconfig b/ANDROID_3.4.5/sound/aoa/soundbus/Kconfig
new file mode 100644
index 00000000..839d1137
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/soundbus/Kconfig
@@ -0,0 +1,14 @@
+config SND_AOA_SOUNDBUS
+	tristate "Apple Soundbus support"
+	select SND_PCM
+	---help---
+	This option enables the generic driver for the soundbus
+	support on Apple machines.
+	
+	It is required for the sound bus implementations.
+
+config SND_AOA_SOUNDBUS_I2S
+	tristate "I2S bus support"
+	depends on SND_AOA_SOUNDBUS && PCI
+	---help---
+	This option enables support for Apple I2S busses.
diff --git a/ANDROID_3.4.5/sound/aoa/soundbus/Makefile b/ANDROID_3.4.5/sound/aoa/soundbus/Makefile
new file mode 100644
index 00000000..0e61f5aa
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/soundbus/Makefile
@@ -0,0 +1,3 @@
+obj-$(CONFIG_SND_AOA_SOUNDBUS) += snd-aoa-soundbus.o
+snd-aoa-soundbus-objs := core.o sysfs.o
+obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += i2sbus/
diff --git a/ANDROID_3.4.5/sound/aoa/soundbus/core.c b/ANDROID_3.4.5/sound/aoa/soundbus/core.c
new file mode 100644
index 00000000..7487eb76
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/soundbus/core.c
@@ -0,0 +1,219 @@
+/*
+ * soundbus
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+
+#include <linux/module.h>
+#include "soundbus.h"
+
+MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
+MODULE_LICENSE("GPL");
+MODULE_DESCRIPTION("Apple Soundbus");
+
+struct soundbus_dev *soundbus_dev_get(struct soundbus_dev *dev)
+{
+	struct device *tmp;
+
+	if (!dev)
+		return NULL;
+	tmp = get_device(&dev->ofdev.dev);
+	if (tmp)
+		return to_soundbus_device(tmp);
+	else
+		return NULL;
+}
+EXPORT_SYMBOL_GPL(soundbus_dev_get);
+
+void soundbus_dev_put(struct soundbus_dev *dev)
+{
+	if (dev)
+		put_device(&dev->ofdev.dev);
+}
+EXPORT_SYMBOL_GPL(soundbus_dev_put);
+
+static int soundbus_probe(struct device *dev)
+{
+	int error = -ENODEV;
+	struct soundbus_driver *drv;
+	struct soundbus_dev *soundbus_dev;
+
+	drv = to_soundbus_driver(dev->driver);
+	soundbus_dev = to_soundbus_device(dev);
+
+	if (!drv->probe)
+		return error;
+
+	soundbus_dev_get(soundbus_dev);
+
+	error = drv->probe(soundbus_dev);
+	if (error)
+		soundbus_dev_put(soundbus_dev);
+
+	return error;
+}
+
+
+static int soundbus_uevent(struct device *dev, struct kobj_uevent_env *env)
+{
+	struct soundbus_dev * soundbus_dev;
+	struct platform_device * of;
+	const char *compat;
+	int retval = 0;
+	int cplen, seen = 0;
+
+	if (!dev)
+		return -ENODEV;
+
+	soundbus_dev = to_soundbus_device(dev);
+	if (!soundbus_dev)
+		return -ENODEV;
+
+	of = &soundbus_dev->ofdev;
+
+	/* stuff we want to pass to /sbin/hotplug */
+	retval = add_uevent_var(env, "OF_NAME=%s", of->dev.of_node->name);
+	if (retval)
+		return retval;
+
+	retval = add_uevent_var(env, "OF_TYPE=%s", of->dev.of_node->type);
+	if (retval)
+		return retval;
+
+	/* Since the compatible field can contain pretty much anything
+	 * it's not really legal to split it out with commas. We split it
+	 * up using a number of environment variables instead. */
+
+	compat = of_get_property(of->dev.of_node, "compatible", &cplen);
+	while (compat && cplen > 0) {
+		int tmp = env->buflen;
+		retval = add_uevent_var(env, "OF_COMPATIBLE_%d=%s", seen, compat);
+		if (retval)
+			return retval;
+		compat += env->buflen - tmp;
+		cplen -= env->buflen - tmp;
+		seen += 1;
+	}
+
+	retval = add_uevent_var(env, "OF_COMPATIBLE_N=%d", seen);
+	if (retval)
+		return retval;
+	retval = add_uevent_var(env, "MODALIAS=%s", soundbus_dev->modalias);
+
+	return retval;
+}
+
+static int soundbus_device_remove(struct device *dev)
+{
+	struct soundbus_dev * soundbus_dev = to_soundbus_device(dev);
+	struct soundbus_driver * drv = to_soundbus_driver(dev->driver);
+
+	if (dev->driver && drv->remove)
+		drv->remove(soundbus_dev);
+	soundbus_dev_put(soundbus_dev);
+
+	return 0;
+}
+
+static void soundbus_device_shutdown(struct device *dev)
+{
+	struct soundbus_dev * soundbus_dev = to_soundbus_device(dev);
+	struct soundbus_driver * drv = to_soundbus_driver(dev->driver);
+
+	if (dev->driver && drv->shutdown)
+		drv->shutdown(soundbus_dev);
+}
+
+#ifdef CONFIG_PM
+
+static int soundbus_device_suspend(struct device *dev, pm_message_t state)
+{
+	struct soundbus_dev * soundbus_dev = to_soundbus_device(dev);
+	struct soundbus_driver * drv = to_soundbus_driver(dev->driver);
+
+	if (dev->driver && drv->suspend)
+		return drv->suspend(soundbus_dev, state);
+	return 0;
+}
+
+static int soundbus_device_resume(struct device * dev)
+{
+	struct soundbus_dev * soundbus_dev = to_soundbus_device(dev);
+	struct soundbus_driver * drv = to_soundbus_driver(dev->driver);
+
+	if (dev->driver && drv->resume)
+		return drv->resume(soundbus_dev);
+	return 0;
+}
+
+#endif /* CONFIG_PM */
+
+static struct bus_type soundbus_bus_type = {
+	.name		= "aoa-soundbus",
+	.probe		= soundbus_probe,
+	.uevent		= soundbus_uevent,
+	.remove		= soundbus_device_remove,
+	.shutdown	= soundbus_device_shutdown,
+#ifdef CONFIG_PM
+	.suspend	= soundbus_device_suspend,
+	.resume		= soundbus_device_resume,
+#endif
+	.dev_attrs	= soundbus_dev_attrs,
+};
+
+int soundbus_add_one(struct soundbus_dev *dev)
+{
+	static int devcount;
+
+	/* sanity checks */
+	if (!dev->attach_codec ||
+	    !dev->ofdev.dev.of_node ||
+	    dev->pcmname ||
+	    dev->pcmid != -1) {
+		printk(KERN_ERR "soundbus: adding device failed sanity check!\n");
+		return -EINVAL;
+	}
+
+	dev_set_name(&dev->ofdev.dev, "soundbus:%x", ++devcount);
+	dev->ofdev.dev.bus = &soundbus_bus_type;
+	return of_device_register(&dev->ofdev);
+}
+EXPORT_SYMBOL_GPL(soundbus_add_one);
+
+void soundbus_remove_one(struct soundbus_dev *dev)
+{
+	of_device_unregister(&dev->ofdev);
+}
+EXPORT_SYMBOL_GPL(soundbus_remove_one);
+
+int soundbus_register_driver(struct soundbus_driver *drv)
+{
+	/* initialize common driver fields */
+	drv->driver.name = drv->name;
+	drv->driver.bus = &soundbus_bus_type;
+
+	/* register with core */
+	return driver_register(&drv->driver);
+}
+EXPORT_SYMBOL_GPL(soundbus_register_driver);
+
+void soundbus_unregister_driver(struct soundbus_driver *drv)
+{
+	driver_unregister(&drv->driver);
+}
+EXPORT_SYMBOL_GPL(soundbus_unregister_driver);
+
+static int __init soundbus_init(void)
+{
+	return bus_register(&soundbus_bus_type);
+}
+
+static void __exit soundbus_exit(void)
+{
+	bus_unregister(&soundbus_bus_type);
+}
+
+subsys_initcall(soundbus_init);
+module_exit(soundbus_exit);
diff --git a/ANDROID_3.4.5/sound/aoa/soundbus/i2sbus/Makefile b/ANDROID_3.4.5/sound/aoa/soundbus/i2sbus/Makefile
new file mode 100644
index 00000000..1b949b2a
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/soundbus/i2sbus/Makefile
@@ -0,0 +1,2 @@
+obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += snd-aoa-i2sbus.o
+snd-aoa-i2sbus-objs := core.o pcm.o control.o
diff --git a/ANDROID_3.4.5/sound/aoa/soundbus/i2sbus/control.c b/ANDROID_3.4.5/sound/aoa/soundbus/i2sbus/control.c
new file mode 100644
index 00000000..4dc9b49c
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/soundbus/i2sbus/control.c
@@ -0,0 +1,194 @@
+/*
+ * i2sbus driver -- bus control routines
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+
+#include <linux/kernel.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+
+#include <asm/io.h>
+#include <asm/prom.h>
+#include <asm/macio.h>
+#include <asm/pmac_feature.h>
+#include <asm/pmac_pfunc.h>
+#include <asm/keylargo.h>
+
+#include "i2sbus.h"
+
+int i2sbus_control_init(struct macio_dev* dev, struct i2sbus_control **c)
+{
+	*c = kzalloc(sizeof(struct i2sbus_control), GFP_KERNEL);
+	if (!*c)
+		return -ENOMEM;
+
+	INIT_LIST_HEAD(&(*c)->list);
+
+	(*c)->macio = dev->bus->chip;
+	return 0;
+}
+
+void i2sbus_control_destroy(struct i2sbus_control *c)
+{
+	kfree(c);
+}
+
+/* this is serialised externally */
+int i2sbus_control_add_dev(struct i2sbus_control *c,
+			   struct i2sbus_dev *i2sdev)
+{
+	struct device_node *np;
+
+	np = i2sdev->sound.ofdev.dev.of_node;
+	i2sdev->enable = pmf_find_function(np, "enable");
+	i2sdev->cell_enable = pmf_find_function(np, "cell-enable");
+	i2sdev->clock_enable = pmf_find_function(np, "clock-enable");
+	i2sdev->cell_disable = pmf_find_function(np, "cell-disable");
+	i2sdev->clock_disable = pmf_find_function(np, "clock-disable");
+
+	/* if the bus number is not 0 or 1 we absolutely need to use
+	 * the platform functions -- there's nothing in Darwin that
+	 * would allow seeing a system behind what the FCRs are then,
+	 * and I don't want to go parsing a bunch of platform functions
+	 * by hand to try finding a system... */
+	if (i2sdev->bus_number != 0 && i2sdev->bus_number != 1 &&
+	    (!i2sdev->enable ||
+	     !i2sdev->cell_enable || !i2sdev->clock_enable ||
+	     !i2sdev->cell_disable || !i2sdev->clock_disable)) {
+		pmf_put_function(i2sdev->enable);
+		pmf_put_function(i2sdev->cell_enable);
+		pmf_put_function(i2sdev->clock_enable);
+		pmf_put_function(i2sdev->cell_disable);
+		pmf_put_function(i2sdev->clock_disable);
+		return -ENODEV;
+	}
+
+	list_add(&i2sdev->item, &c->list);
+
+	return 0;
+}
+
+void i2sbus_control_remove_dev(struct i2sbus_control *c,
+			       struct i2sbus_dev *i2sdev)
+{
+	/* this is serialised externally */
+	list_del(&i2sdev->item);
+	if (list_empty(&c->list))
+		i2sbus_control_destroy(c);
+}
+
+int i2sbus_control_enable(struct i2sbus_control *c,
+			  struct i2sbus_dev *i2sdev)
+{
+	struct pmf_args args = { .count = 0 };
+	struct macio_chip *macio = c->macio;
+
+	if (i2sdev->enable)
+		return pmf_call_one(i2sdev->enable, &args);
+
+	if (macio == NULL || macio->base == NULL)
+		return -ENODEV;
+
+	switch (i2sdev->bus_number) {
+	case 0:
+		/* these need to be locked or done through
+		 * newly created feature calls! */
+		MACIO_BIS(KEYLARGO_FCR1, KL1_I2S0_ENABLE);
+		break;
+	case 1:
+		MACIO_BIS(KEYLARGO_FCR1, KL1_I2S1_ENABLE);
+		break;
+	default:
+		return -ENODEV;
+	}
+	return 0;
+}
+
+int i2sbus_control_cell(struct i2sbus_control *c,
+			struct i2sbus_dev *i2sdev,
+			int enable)
+{
+	struct pmf_args args = { .count = 0 };
+	struct macio_chip *macio = c->macio;
+
+	switch (enable) {
+	case 0:
+		if (i2sdev->cell_disable)
+			return pmf_call_one(i2sdev->cell_disable, &args);
+		break;
+	case 1:
+		if (i2sdev->cell_enable)
+			return pmf_call_one(i2sdev->cell_enable, &args);
+		break;
+	default:
+		printk(KERN_ERR "i2sbus: INVALID CELL ENABLE VALUE\n");
+		return -ENODEV;
+	}
+
+	if (macio == NULL || macio->base == NULL)
+		return -ENODEV;
+
+	switch (i2sdev->bus_number) {
+	case 0:
+		if (enable)
+			MACIO_BIS(KEYLARGO_FCR1, KL1_I2S0_CELL_ENABLE);
+		else
+			MACIO_BIC(KEYLARGO_FCR1, KL1_I2S0_CELL_ENABLE);
+		break;
+	case 1:
+		if (enable)
+			MACIO_BIS(KEYLARGO_FCR1, KL1_I2S1_CELL_ENABLE);
+		else
+			MACIO_BIC(KEYLARGO_FCR1, KL1_I2S1_CELL_ENABLE);
+		break;
+	default:
+		return -ENODEV;
+	}
+	return 0;
+}
+
+int i2sbus_control_clock(struct i2sbus_control *c,
+			 struct i2sbus_dev *i2sdev,
+			 int enable)
+{
+	struct pmf_args args = { .count = 0 };
+	struct macio_chip *macio = c->macio;
+
+	switch (enable) {
+	case 0:
+		if (i2sdev->clock_disable)
+			return pmf_call_one(i2sdev->clock_disable, &args);
+		break;
+	case 1:
+		if (i2sdev->clock_enable)
+			return pmf_call_one(i2sdev->clock_enable, &args);
+		break;
+	default:
+		printk(KERN_ERR "i2sbus: INVALID CLOCK ENABLE VALUE\n");
+		return -ENODEV;
+	}
+
+	if (macio == NULL || macio->base == NULL)
+		return -ENODEV;
+
+	switch (i2sdev->bus_number) {
+	case 0:
+		if (enable)
+			MACIO_BIS(KEYLARGO_FCR1, KL1_I2S0_CLK_ENABLE_BIT);
+		else
+			MACIO_BIC(KEYLARGO_FCR1, KL1_I2S0_CLK_ENABLE_BIT);
+		break;
+	case 1:
+		if (enable)
+			MACIO_BIS(KEYLARGO_FCR1, KL1_I2S1_CLK_ENABLE_BIT);
+		else
+			MACIO_BIC(KEYLARGO_FCR1, KL1_I2S1_CLK_ENABLE_BIT);
+		break;
+	default:
+		return -ENODEV;
+	}
+	return 0;
+}
diff --git a/ANDROID_3.4.5/sound/aoa/soundbus/i2sbus/core.c b/ANDROID_3.4.5/sound/aoa/soundbus/i2sbus/core.c
new file mode 100644
index 00000000..01065833
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/soundbus/i2sbus/core.c
@@ -0,0 +1,464 @@
+/*
+ * i2sbus driver
+ *
+ * Copyright 2006-2008 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+
+#include <linux/module.h>
+#include <linux/slab.h>
+#include <linux/pci.h>
+#include <linux/interrupt.h>
+#include <linux/dma-mapping.h>
+
+#include <sound/core.h>
+
+#include <asm/macio.h>
+#include <asm/dbdma.h>
+
+#include "../soundbus.h"
+#include "i2sbus.h"
+
+MODULE_LICENSE("GPL");
+MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
+MODULE_DESCRIPTION("Apple Soundbus: I2S support");
+
+static int force;
+module_param(force, int, 0444);
+MODULE_PARM_DESC(force, "Force loading i2sbus even when"
+			" no layout-id property is present");
+
+static struct of_device_id i2sbus_match[] = {
+	{ .name = "i2s" },
+	{ }
+};
+
+MODULE_DEVICE_TABLE(of, i2sbus_match);
+
+static int alloc_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev,
+				       struct dbdma_command_mem *r,
+				       int numcmds)
+{
+	/* one more for rounding, one for branch back, one for stop command */
+	r->size = (numcmds + 3) * sizeof(struct dbdma_cmd);
+	/* We use the PCI APIs for now until the generic one gets fixed
+	 * enough or until we get some macio-specific versions
+	 */
+	r->space = dma_alloc_coherent(
+			&macio_get_pci_dev(i2sdev->macio)->dev,
+			r->size,
+			&r->bus_addr,
+			GFP_KERNEL);
+
+	if (!r->space) return -ENOMEM;
+
+	memset(r->space, 0, r->size);
+	r->cmds = (void*)DBDMA_ALIGN(r->space);
+	r->bus_cmd_start = r->bus_addr +
+			   (dma_addr_t)((char*)r->cmds - (char*)r->space);
+
+	return 0;
+}
+
+static void free_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev,
+				       struct dbdma_command_mem *r)
+{
+	if (!r->space) return;
+
+	dma_free_coherent(&macio_get_pci_dev(i2sdev->macio)->dev,
+			    r->size, r->space, r->bus_addr);
+}
+
+static void i2sbus_release_dev(struct device *dev)
+{
+	struct i2sbus_dev *i2sdev;
+	int i;
+
+	i2sdev = container_of(dev, struct i2sbus_dev, sound.ofdev.dev);
+
+ 	if (i2sdev->intfregs) iounmap(i2sdev->intfregs);
+ 	if (i2sdev->out.dbdma) iounmap(i2sdev->out.dbdma);
+ 	if (i2sdev->in.dbdma) iounmap(i2sdev->in.dbdma);
+	for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++)
+		if (i2sdev->allocated_resource[i])
+			release_and_free_resource(i2sdev->allocated_resource[i]);
+	free_dbdma_descriptor_ring(i2sdev, &i2sdev->out.dbdma_ring);
+	free_dbdma_descriptor_ring(i2sdev, &i2sdev->in.dbdma_ring);
+	for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++)
+		free_irq(i2sdev->interrupts[i], i2sdev);
+	i2sbus_control_remove_dev(i2sdev->control, i2sdev);
+	mutex_destroy(&i2sdev->lock);
+	kfree(i2sdev);
+}
+
+static irqreturn_t i2sbus_bus_intr(int irq, void *devid)
+{
+	struct i2sbus_dev *dev = devid;
+	u32 intreg;
+
+	spin_lock(&dev->low_lock);
+	intreg = in_le32(&dev->intfregs->intr_ctl);
+
+	/* acknowledge interrupt reasons */
+	out_le32(&dev->intfregs->intr_ctl, intreg);
+
+	spin_unlock(&dev->low_lock);
+
+	return IRQ_HANDLED;
+}
+
+
+/*
+ * XXX FIXME: We test the layout_id's here to get the proper way of
+ * mapping in various registers, thanks to bugs in Apple device-trees.
+ * We could instead key off the machine model and the name of the i2s
+ * node (i2s-a). This we'll do when we move it all to macio_asic.c
+ * and have that export items for each sub-node too.
+ */
+static int i2sbus_get_and_fixup_rsrc(struct device_node *np, int index,
+				     int layout, struct resource *res)
+{
+	struct device_node *parent;
+	int pindex, rc = -ENXIO;
+	const u32 *reg;
+
+	/* Machines with layout 76 and 36 (K2 based) have a weird device
+	 * tree what we need to special case.
+	 * Normal machines just fetch the resource from the i2s-X node.
+	 * Darwin further divides normal machines into old and new layouts
+	 * with a subtely different code path but that doesn't seem necessary
+	 * in practice, they just bloated it. In addition, even on our K2
+	 * case the i2s-modem node, if we ever want to handle it, uses the
+	 * normal layout
+	 */
+	if (layout != 76 && layout != 36)
+		return of_address_to_resource(np, index, res);
+
+	parent = of_get_parent(np);
+	pindex = (index == aoa_resource_i2smmio) ? 0 : 1;
+	rc = of_address_to_resource(parent, pindex, res);
+	if (rc)
+		goto bail;
+	reg = of_get_property(np, "reg", NULL);
+	if (reg == NULL) {
+		rc = -ENXIO;
+		goto bail;
+	}
+	res->start += reg[index * 2];
+	res->end = res->start + reg[index * 2 + 1] - 1;
+ bail:
+	of_node_put(parent);
+	return rc;
+}
+
+/* FIXME: look at device node refcounting */
+static int i2sbus_add_dev(struct macio_dev *macio,
+			  struct i2sbus_control *control,
+			  struct device_node *np)
+{
+	struct i2sbus_dev *dev;
+	struct device_node *child = NULL, *sound = NULL;
+	struct resource *r;
+	int i, layout = 0, rlen, ok = force;
+	static const char *rnames[] = { "i2sbus: %s (control)",
+					"i2sbus: %s (tx)",
+					"i2sbus: %s (rx)" };
+	static irq_handler_t ints[] = {
+		i2sbus_bus_intr,
+		i2sbus_tx_intr,
+		i2sbus_rx_intr
+	};
+
+	if (strlen(np->name) != 5)
+		return 0;
+	if (strncmp(np->name, "i2s-", 4))
+		return 0;
+
+	dev = kzalloc(sizeof(struct i2sbus_dev), GFP_KERNEL);
+	if (!dev)
+		return 0;
+
+	i = 0;
+	while ((child = of_get_next_child(np, child))) {
+		if (strcmp(child->name, "sound") == 0) {
+			i++;
+			sound = child;
+		}
+	}
+	if (i == 1) {
+		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
+	 * things in some fabric, refuse to attach if there is no
+	 * layout-id property or we haven't been forced to attach.
+	 * When there are two i2s busses and only one has a layout-id,
+	 * then this depends on the order, but that isn't important
+	 * either as the second one in that case is just a modem. */
+	if (!ok) {
+		kfree(dev);
+		return -ENODEV;
+	}
+
+	mutex_init(&dev->lock);
+	spin_lock_init(&dev->low_lock);
+	dev->sound.ofdev.archdata.dma_mask = macio->ofdev.archdata.dma_mask;
+	dev->sound.ofdev.dev.of_node = np;
+	dev->sound.ofdev.dev.dma_mask = &dev->sound.ofdev.archdata.dma_mask;
+	dev->sound.ofdev.dev.parent = &macio->ofdev.dev;
+	dev->sound.ofdev.dev.release = i2sbus_release_dev;
+	dev->sound.attach_codec = i2sbus_attach_codec;
+	dev->sound.detach_codec = i2sbus_detach_codec;
+	dev->sound.pcmid = -1;
+	dev->macio = macio;
+	dev->control = control;
+	dev->bus_number = np->name[4] - 'a';
+	INIT_LIST_HEAD(&dev->sound.codec_list);
+
+	for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) {
+		dev->interrupts[i] = -1;
+		snprintf(dev->rnames[i], sizeof(dev->rnames[i]),
+			 rnames[i], np->name);
+	}
+	for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) {
+		int irq = irq_of_parse_and_map(np, i);
+		if (request_irq(irq, ints[i], 0, dev->rnames[i], dev))
+			goto err;
+		dev->interrupts[i] = irq;
+	}
+
+
+	/* Resource handling is problematic as some device-trees contain
+	 * useless crap (ugh ugh ugh). We work around that here by calling
+	 * specific functions for calculating the appropriate resources.
+	 *
+	 * This will all be moved to macio_asic.c at one point
+	 */
+	for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) {
+		if (i2sbus_get_and_fixup_rsrc(np,i,layout,&dev->resources[i]))
+			goto err;
+		/* If only we could use our resource dev->resources[i]...
+		 * but request_resource doesn't know about parents and
+		 * contained resources...
+		 */
+		dev->allocated_resource[i] =
+			request_mem_region(dev->resources[i].start,
+					   resource_size(&dev->resources[i]),
+					   dev->rnames[i]);
+		if (!dev->allocated_resource[i]) {
+			printk(KERN_ERR "i2sbus: failed to claim resource %d!\n", i);
+			goto err;
+		}
+	}
+
+	r = &dev->resources[aoa_resource_i2smmio];
+	rlen = resource_size(r);
+	if (rlen < sizeof(struct i2s_interface_regs))
+		goto err;
+	dev->intfregs = ioremap(r->start, rlen);
+
+	r = &dev->resources[aoa_resource_txdbdma];
+	rlen = resource_size(r);
+	if (rlen < sizeof(struct dbdma_regs))
+		goto err;
+	dev->out.dbdma = ioremap(r->start, rlen);
+
+	r = &dev->resources[aoa_resource_rxdbdma];
+	rlen = resource_size(r);
+	if (rlen < sizeof(struct dbdma_regs))
+		goto err;
+	dev->in.dbdma = ioremap(r->start, rlen);
+
+	if (!dev->intfregs || !dev->out.dbdma || !dev->in.dbdma)
+		goto err;
+
+	if (alloc_dbdma_descriptor_ring(dev, &dev->out.dbdma_ring,
+					MAX_DBDMA_COMMANDS))
+		goto err;
+	if (alloc_dbdma_descriptor_ring(dev, &dev->in.dbdma_ring,
+					MAX_DBDMA_COMMANDS))
+		goto err;
+
+	if (i2sbus_control_add_dev(dev->control, dev)) {
+		printk(KERN_ERR "i2sbus: control layer didn't like bus\n");
+		goto err;
+	}
+
+	if (soundbus_add_one(&dev->sound)) {
+		printk(KERN_DEBUG "i2sbus: device registration error!\n");
+		goto err;
+	}
+
+	/* enable this cell */
+	i2sbus_control_cell(dev->control, dev, 1);
+	i2sbus_control_enable(dev->control, dev);
+	i2sbus_control_clock(dev->control, dev, 1);
+
+	return 1;
+ err:
+	for (i=0;i<3;i++)
+		if (dev->interrupts[i] != -1)
+			free_irq(dev->interrupts[i], dev);
+	free_dbdma_descriptor_ring(dev, &dev->out.dbdma_ring);
+	free_dbdma_descriptor_ring(dev, &dev->in.dbdma_ring);
+	if (dev->intfregs) iounmap(dev->intfregs);
+	if (dev->out.dbdma) iounmap(dev->out.dbdma);
+	if (dev->in.dbdma) iounmap(dev->in.dbdma);
+	for (i=0;i<3;i++)
+		if (dev->allocated_resource[i])
+			release_and_free_resource(dev->allocated_resource[i]);
+	mutex_destroy(&dev->lock);
+	kfree(dev);
+	return 0;
+}
+
+static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match)
+{
+	struct device_node *np = NULL;
+	int got = 0, err;
+	struct i2sbus_control *control = NULL;
+
+	err = i2sbus_control_init(dev, &control);
+	if (err)
+		return err;
+	if (!control) {
+		printk(KERN_ERR "i2sbus_control_init API breakage\n");
+		return -ENODEV;
+	}
+
+	while ((np = of_get_next_child(dev->ofdev.dev.of_node, np))) {
+		if (of_device_is_compatible(np, "i2sbus") ||
+		    of_device_is_compatible(np, "i2s-modem")) {
+			got += i2sbus_add_dev(dev, control, np);
+		}
+	}
+
+	if (!got) {
+		/* found none, clean up */
+		i2sbus_control_destroy(control);
+		return -ENODEV;
+	}
+
+	dev_set_drvdata(&dev->ofdev.dev, control);
+
+	return 0;
+}
+
+static int i2sbus_remove(struct macio_dev* dev)
+{
+	struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev);
+	struct i2sbus_dev *i2sdev, *tmp;
+
+	list_for_each_entry_safe(i2sdev, tmp, &control->list, item)
+		soundbus_remove_one(&i2sdev->sound);
+
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int i2sbus_suspend(struct macio_dev* dev, pm_message_t state)
+{
+	struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev);
+	struct codec_info_item *cii;
+	struct i2sbus_dev* i2sdev;
+	int err, ret = 0;
+
+	list_for_each_entry(i2sdev, &control->list, item) {
+		/* Notify Alsa */
+		if (i2sdev->sound.pcm) {
+			/* Suspend PCM streams */
+			snd_pcm_suspend_all(i2sdev->sound.pcm);
+		}
+
+		/* Notify codecs */
+		list_for_each_entry(cii, &i2sdev->sound.codec_list, list) {
+			err = 0;
+			if (cii->codec->suspend)
+				err = cii->codec->suspend(cii, state);
+			if (err)
+				ret = err;
+		}
+
+		/* wait until streams are stopped */
+		i2sbus_wait_for_stop_both(i2sdev);
+	}
+
+	return ret;
+}
+
+static int i2sbus_resume(struct macio_dev* dev)
+{
+	struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev);
+	struct codec_info_item *cii;
+	struct i2sbus_dev* i2sdev;
+	int err, ret = 0;
+
+	list_for_each_entry(i2sdev, &control->list, item) {
+		/* reset i2s bus format etc. */
+		i2sbus_pcm_prepare_both(i2sdev);
+
+		/* Notify codecs so they can re-initialize */
+		list_for_each_entry(cii, &i2sdev->sound.codec_list, list) {
+			err = 0;
+			if (cii->codec->resume)
+				err = cii->codec->resume(cii);
+			if (err)
+				ret = err;
+		}
+	}
+
+	return ret;
+}
+#endif /* CONFIG_PM */
+
+static int i2sbus_shutdown(struct macio_dev* dev)
+{
+	return 0;
+}
+
+static struct macio_driver i2sbus_drv = {
+	.driver = {
+		.name = "soundbus-i2s",
+		.owner = THIS_MODULE,
+		.of_match_table = i2sbus_match,
+	},
+	.probe = i2sbus_probe,
+	.remove = i2sbus_remove,
+#ifdef CONFIG_PM
+	.suspend = i2sbus_suspend,
+	.resume = i2sbus_resume,
+#endif
+	.shutdown = i2sbus_shutdown,
+};
+
+static int __init soundbus_i2sbus_init(void)
+{
+	return macio_register_driver(&i2sbus_drv);
+}
+
+static void __exit soundbus_i2sbus_exit(void)
+{
+	macio_unregister_driver(&i2sbus_drv);
+}
+
+module_init(soundbus_i2sbus_init);
+module_exit(soundbus_i2sbus_exit);
diff --git a/ANDROID_3.4.5/sound/aoa/soundbus/i2sbus/i2sbus.h b/ANDROID_3.4.5/sound/aoa/soundbus/i2sbus/i2sbus.h
new file mode 100644
index 00000000..befefd99
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/soundbus/i2sbus/i2sbus.h
@@ -0,0 +1,126 @@
+/*
+ * i2sbus driver -- private definitions
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+#ifndef __I2SBUS_H
+#define __I2SBUS_H
+#include <linux/interrupt.h>
+#include <linux/spinlock.h>
+#include <linux/mutex.h>
+#include <linux/completion.h>
+
+#include <sound/pcm.h>
+
+#include <asm/prom.h>
+#include <asm/pmac_feature.h>
+#include <asm/dbdma.h>
+
+#include "interface.h"
+#include "../soundbus.h"
+
+struct i2sbus_control {
+	struct list_head list;
+	struct macio_chip *macio;
+};
+
+#define MAX_DBDMA_COMMANDS	32
+
+struct dbdma_command_mem {
+	dma_addr_t bus_addr;
+	dma_addr_t bus_cmd_start;
+	struct dbdma_cmd *cmds;
+	void *space;
+	int size;
+	u32 running:1;
+	u32 stopping:1;
+};
+
+struct pcm_info {
+	u32 created:1, /* has this direction been created with alsa? */
+	    active:1;  /* is this stream active? */
+	/* runtime information */
+	struct snd_pcm_substream *substream;
+	int current_period;
+	u32 frame_count;
+	struct dbdma_command_mem dbdma_ring;
+	volatile struct dbdma_regs __iomem *dbdma;
+	struct completion *stop_completion;
+};
+
+enum {
+	aoa_resource_i2smmio = 0,
+	aoa_resource_txdbdma,
+	aoa_resource_rxdbdma,
+};
+
+struct i2sbus_dev {
+	struct soundbus_dev sound;
+	struct macio_dev *macio;
+	struct i2sbus_control *control;
+	volatile struct i2s_interface_regs __iomem *intfregs;
+
+	struct resource resources[3];
+	struct resource *allocated_resource[3];
+	int interrupts[3];
+	char rnames[3][32];
+
+	/* info about currently active substreams */
+	struct pcm_info out, in;
+	snd_pcm_format_t format;
+	unsigned int rate;
+
+	/* list for a single controller */
+	struct list_head item;
+	/* number of bus on controller */
+	int bus_number;
+	/* for use by control layer */
+	struct pmf_function *enable,
+			    *cell_enable,
+			    *cell_disable,
+			    *clock_enable,
+			    *clock_disable;
+
+	/* locks */
+	/* spinlock for low-level interrupt locking */
+	spinlock_t low_lock;
+	/* mutex for high-level consistency */
+	struct mutex lock;
+};
+
+#define soundbus_dev_to_i2sbus_dev(sdev) \
+		container_of(sdev, struct i2sbus_dev, sound)
+
+/* pcm specific functions */
+extern int
+i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card,
+		    struct codec_info *ci, void *data);
+extern void
+i2sbus_detach_codec(struct soundbus_dev *dev, void *data);
+extern irqreturn_t
+i2sbus_tx_intr(int irq, void *devid);
+extern irqreturn_t
+i2sbus_rx_intr(int irq, void *devid);
+
+extern void i2sbus_wait_for_stop_both(struct i2sbus_dev *i2sdev);
+extern void i2sbus_pcm_prepare_both(struct i2sbus_dev *i2sdev);
+
+/* control specific functions */
+extern int i2sbus_control_init(struct macio_dev* dev,
+			       struct i2sbus_control **c);
+extern void i2sbus_control_destroy(struct i2sbus_control *c);
+extern int i2sbus_control_add_dev(struct i2sbus_control *c,
+				  struct i2sbus_dev *i2sdev);
+extern void i2sbus_control_remove_dev(struct i2sbus_control *c,
+				      struct i2sbus_dev *i2sdev);
+extern int i2sbus_control_enable(struct i2sbus_control *c,
+				 struct i2sbus_dev *i2sdev);
+extern int i2sbus_control_cell(struct i2sbus_control *c,
+			       struct i2sbus_dev *i2sdev,
+			       int enable);
+extern int i2sbus_control_clock(struct i2sbus_control *c,
+				struct i2sbus_dev *i2sdev,
+				int enable);
+#endif /* __I2SBUS_H */
diff --git a/ANDROID_3.4.5/sound/aoa/soundbus/i2sbus/interface.h b/ANDROID_3.4.5/sound/aoa/soundbus/i2sbus/interface.h
new file mode 100644
index 00000000..c6b5f545
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/soundbus/i2sbus/interface.h
@@ -0,0 +1,187 @@
+/*
+ * i2sbus driver -- interface register definitions
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+#ifndef __I2SBUS_INTERFACE_H
+#define __I2SBUS_INTERFACE_H
+
+/* i2s bus control registers, at least what we know about them */
+
+#define __PAD(m,n) u8 __pad##m[n]
+#define _PAD(line, n) __PAD(line, n)
+#define PAD(n) _PAD(__LINE__, (n))
+struct i2s_interface_regs {
+	__le32 intr_ctl;	/* 0x00 */
+	PAD(12);
+	__le32 serial_format;	/* 0x10 */
+	PAD(12);
+	__le32 codec_msg_out;	/* 0x20 */
+	PAD(12);
+	__le32 codec_msg_in;	/* 0x30 */
+	PAD(12);
+	__le32 frame_count;	/* 0x40 */
+	PAD(12);
+	__le32 frame_match;	/* 0x50 */
+	PAD(12);
+	__le32 data_word_sizes;	/* 0x60 */
+	PAD(12);
+	__le32 peak_level_sel;	/* 0x70 */
+	PAD(12);
+	__le32 peak_level_in0;	/* 0x80 */
+	PAD(12);
+	__le32 peak_level_in1;	/* 0x90 */
+	PAD(12);
+	/* total size: 0x100 bytes */
+}  __attribute__((__packed__));
+
+/* interrupt register is just a bitfield with
+ * interrupt enable and pending bits */
+#define I2S_REG_INTR_CTL		0x00
+#	define I2S_INT_FRAME_COUNT		(1<<31)
+#	define I2S_PENDING_FRAME_COUNT		(1<<30)
+#	define I2S_INT_MESSAGE_FLAG		(1<<29)
+#	define I2S_PENDING_MESSAGE_FLAG		(1<<28)
+#	define I2S_INT_NEW_PEAK			(1<<27)
+#	define I2S_PENDING_NEW_PEAK		(1<<26)
+#	define I2S_INT_CLOCKS_STOPPED		(1<<25)
+#	define I2S_PENDING_CLOCKS_STOPPED	(1<<24)
+#	define I2S_INT_EXTERNAL_SYNC_ERROR	(1<<23)
+#	define I2S_PENDING_EXTERNAL_SYNC_ERROR	(1<<22)
+#	define I2S_INT_EXTERNAL_SYNC_OK		(1<<21)
+#	define I2S_PENDING_EXTERNAL_SYNC_OK	(1<<20)
+#	define I2S_INT_NEW_SAMPLE_RATE		(1<<19)
+#	define I2S_PENDING_NEW_SAMPLE_RATE	(1<<18)
+#	define I2S_INT_STATUS_FLAG		(1<<17)
+#	define I2S_PENDING_STATUS_FLAG		(1<<16)
+
+/* serial format register is more interesting :)
+ * It contains:
+ *  - clock source
+ *  - MClk divisor
+ *  - SClk divisor
+ *  - SClk master flag
+ *  - serial format (sony, i2s 64x, i2s 32x, dav, silabs)
+ *  - external sample frequency interrupt (don't understand)
+ *  - external sample frequency
+ */
+#define I2S_REG_SERIAL_FORMAT		0x10
+/* clock source. You get either 18.432, 45.1584 or 49.1520 MHz */
+#	define I2S_SF_CLOCK_SOURCE_SHIFT	30
+#	define I2S_SF_CLOCK_SOURCE_MASK		(3<<I2S_SF_CLOCK_SOURCE_SHIFT)
+#	define I2S_SF_CLOCK_SOURCE_18MHz	(0<<I2S_SF_CLOCK_SOURCE_SHIFT)
+#	define I2S_SF_CLOCK_SOURCE_45MHz	(1<<I2S_SF_CLOCK_SOURCE_SHIFT)
+#	define I2S_SF_CLOCK_SOURCE_49MHz	(2<<I2S_SF_CLOCK_SOURCE_SHIFT)
+/* also, let's define the exact clock speeds here, in Hz */
+#define I2S_CLOCK_SPEED_18MHz	18432000
+#define I2S_CLOCK_SPEED_45MHz	45158400
+#define I2S_CLOCK_SPEED_49MHz	49152000
+/* MClk is the clock that drives the codec, usually called its 'system clock'.
+ * It is derived by taking only every 'divisor' tick of the clock.
+ */
+#	define I2S_SF_MCLKDIV_SHIFT		24
+#	define I2S_SF_MCLKDIV_MASK		(0x1F<<I2S_SF_MCLKDIV_SHIFT)
+#	define I2S_SF_MCLKDIV_1			(0x14<<I2S_SF_MCLKDIV_SHIFT)
+#	define I2S_SF_MCLKDIV_3			(0x13<<I2S_SF_MCLKDIV_SHIFT)
+#	define I2S_SF_MCLKDIV_5			(0x12<<I2S_SF_MCLKDIV_SHIFT)
+#	define I2S_SF_MCLKDIV_14		(0x0E<<I2S_SF_MCLKDIV_SHIFT)
+#	define I2S_SF_MCLKDIV_OTHER(div)	(((div/2-1)<<I2S_SF_MCLKDIV_SHIFT)&I2S_SF_MCLKDIV_MASK)
+static inline int i2s_sf_mclkdiv(int div, int *out)
+{
+	int d;
+
+	switch(div) {
+	case 1: *out |= I2S_SF_MCLKDIV_1; return 0;
+	case 3: *out |= I2S_SF_MCLKDIV_3; return 0;
+	case 5: *out |= I2S_SF_MCLKDIV_5; return 0;
+	case 14: *out |= I2S_SF_MCLKDIV_14; return 0;
+	default:
+		if (div%2) return -1;
+		d = div/2-1;
+		if (d == 0x14 || d == 0x13 || d == 0x12 || d == 0x0E)
+			return -1;
+		*out |= I2S_SF_MCLKDIV_OTHER(div);
+		return 0;
+	}
+}
+/* SClk is the clock that drives the i2s wire bus. Note that it is
+ * derived from the MClk above by taking only every 'divisor' tick
+ * of MClk.
+ */
+#	define I2S_SF_SCLKDIV_SHIFT		20
+#	define I2S_SF_SCLKDIV_MASK		(0xF<<I2S_SF_SCLKDIV_SHIFT)
+#	define I2S_SF_SCLKDIV_1			(8<<I2S_SF_SCLKDIV_SHIFT)
+#	define I2S_SF_SCLKDIV_3			(9<<I2S_SF_SCLKDIV_SHIFT)
+#	define I2S_SF_SCLKDIV_OTHER(div)	(((div/2-1)<<I2S_SF_SCLKDIV_SHIFT)&I2S_SF_SCLKDIV_MASK)
+static inline int i2s_sf_sclkdiv(int div, int *out)
+{
+	int d;
+
+	switch(div) {
+	case 1: *out |= I2S_SF_SCLKDIV_1; return 0;
+	case 3: *out |= I2S_SF_SCLKDIV_3; return 0;
+	default:
+		if (div%2) return -1;
+		d = div/2-1;
+		if (d == 8 || d == 9) return -1;
+		*out |= I2S_SF_SCLKDIV_OTHER(div);
+		return 0;
+	}
+}
+#	define I2S_SF_SCLK_MASTER		(1<<19)
+/* serial format is the way the data is put to the i2s wire bus */
+#	define I2S_SF_SERIAL_FORMAT_SHIFT	16
+#	define I2S_SF_SERIAL_FORMAT_MASK	(7<<I2S_SF_SERIAL_FORMAT_SHIFT)
+#	define I2S_SF_SERIAL_FORMAT_SONY	(0<<I2S_SF_SERIAL_FORMAT_SHIFT)
+#	define I2S_SF_SERIAL_FORMAT_I2S_64X	(1<<I2S_SF_SERIAL_FORMAT_SHIFT)
+#	define I2S_SF_SERIAL_FORMAT_I2S_32X	(2<<I2S_SF_SERIAL_FORMAT_SHIFT)
+#	define I2S_SF_SERIAL_FORMAT_I2S_DAV	(4<<I2S_SF_SERIAL_FORMAT_SHIFT)
+#	define I2S_SF_SERIAL_FORMAT_I2S_SILABS	(5<<I2S_SF_SERIAL_FORMAT_SHIFT)
+/* unknown */
+#	define I2S_SF_EXT_SAMPLE_FREQ_INT_SHIFT	12
+#	define I2S_SF_EXT_SAMPLE_FREQ_INT_MASK	(0xF<<I2S_SF_SAMPLE_FREQ_INT_SHIFT)
+/* probably gives external frequency? */
+#	define I2S_SF_EXT_SAMPLE_FREQ_MASK	0xFFF
+
+/* used to send codec messages, but how isn't clear */
+#define I2S_REG_CODEC_MSG_OUT		0x20
+
+/* used to receive codec messages, but how isn't clear */
+#define I2S_REG_CODEC_MSG_IN		0x30
+
+/* frame count reg isn't clear to me yet, but probably useful */
+#define I2S_REG_FRAME_COUNT		0x40
+
+/* program to some value, and get interrupt if frame count reaches it */
+#define I2S_REG_FRAME_MATCH		0x50
+
+/* this register describes how the bus transfers data */
+#define I2S_REG_DATA_WORD_SIZES		0x60
+/* number of interleaved input channels */
+#	define I2S_DWS_NUM_CHANNELS_IN_SHIFT	24
+#	define I2S_DWS_NUM_CHANNELS_IN_MASK	(0x1F<<I2S_DWS_NUM_CHANNELS_IN_SHIFT)
+/* word size of input data */
+#	define I2S_DWS_DATA_IN_SIZE_SHIFT	16
+#	define I2S_DWS_DATA_IN_16BIT		(0<<I2S_DWS_DATA_IN_SIZE_SHIFT)
+#	define I2S_DWS_DATA_IN_24BIT		(3<<I2S_DWS_DATA_IN_SIZE_SHIFT)
+/* number of interleaved output channels */
+#	define I2S_DWS_NUM_CHANNELS_OUT_SHIFT	8
+#	define I2S_DWS_NUM_CHANNELS_OUT_MASK	(0x1F<<I2S_DWS_NUM_CHANNELS_OUT_SHIFT)
+/* word size of output data */
+#	define I2S_DWS_DATA_OUT_SIZE_SHIFT	0
+#	define I2S_DWS_DATA_OUT_16BIT		(0<<I2S_DWS_DATA_OUT_SIZE_SHIFT)
+#	define I2S_DWS_DATA_OUT_24BIT		(3<<I2S_DWS_DATA_OUT_SIZE_SHIFT)
+
+
+/* unknown */
+#define I2S_REG_PEAK_LEVEL_SEL		0x70
+
+/* unknown */
+#define I2S_REG_PEAK_LEVEL_IN0		0x80
+
+/* unknown */
+#define I2S_REG_PEAK_LEVEL_IN1		0x90
+
+#endif /* __I2SBUS_INTERFACE_H */
diff --git a/ANDROID_3.4.5/sound/aoa/soundbus/i2sbus/pcm.c b/ANDROID_3.4.5/sound/aoa/soundbus/i2sbus/pcm.c
new file mode 100644
index 00000000..19491ed9
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/soundbus/i2sbus/pcm.c
@@ -0,0 +1,1064 @@
+/*
+ * i2sbus driver -- pcm routines
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+
+#include <asm/io.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <asm/macio.h>
+#include <linux/pci.h>
+#include <linux/module.h>
+#include "../soundbus.h"
+#include "i2sbus.h"
+
+static inline void get_pcm_info(struct i2sbus_dev *i2sdev, int in,
+				struct pcm_info **pi, struct pcm_info **other)
+{
+	if (in) {
+		if (pi)
+			*pi = &i2sdev->in;
+		if (other)
+			*other = &i2sdev->out;
+	} else {
+		if (pi)
+			*pi = &i2sdev->out;
+		if (other)
+			*other = &i2sdev->in;
+	}
+}
+
+static int clock_and_divisors(int mclk, int sclk, int rate, int *out)
+{
+	/* sclk must be derived from mclk! */
+	if (mclk % sclk)
+		return -1;
+	/* derive sclk register value */
+	if (i2s_sf_sclkdiv(mclk / sclk, out))
+		return -1;
+
+	if (I2S_CLOCK_SPEED_18MHz % (rate * mclk) == 0) {
+		if (!i2s_sf_mclkdiv(I2S_CLOCK_SPEED_18MHz / (rate * mclk), out)) {
+			*out |= I2S_SF_CLOCK_SOURCE_18MHz;
+			return 0;
+		}
+	}
+	if (I2S_CLOCK_SPEED_45MHz % (rate * mclk) == 0) {
+		if (!i2s_sf_mclkdiv(I2S_CLOCK_SPEED_45MHz / (rate * mclk), out)) {
+			*out |= I2S_SF_CLOCK_SOURCE_45MHz;
+			return 0;
+		}
+	}
+	if (I2S_CLOCK_SPEED_49MHz % (rate * mclk) == 0) {
+		if (!i2s_sf_mclkdiv(I2S_CLOCK_SPEED_49MHz / (rate * mclk), out)) {
+			*out |= I2S_SF_CLOCK_SOURCE_49MHz;
+			return 0;
+		}
+	}
+	return -1;
+}
+
+#define CHECK_RATE(rate)						\
+	do { if (rates & SNDRV_PCM_RATE_ ##rate) {			\
+		int dummy;						\
+		if (clock_and_divisors(sysclock_factor,			\
+				       bus_factor, rate, &dummy))	\
+			rates &= ~SNDRV_PCM_RATE_ ##rate;		\
+	} } while (0)
+
+static int i2sbus_pcm_open(struct i2sbus_dev *i2sdev, int in)
+{
+	struct pcm_info *pi, *other;
+	struct soundbus_dev *sdev;
+	int masks_inited = 0, err;
+	struct codec_info_item *cii, *rev;
+	struct snd_pcm_hardware *hw;
+	u64 formats = 0;
+	unsigned int rates = 0;
+	struct transfer_info v;
+	int result = 0;
+	int bus_factor = 0, sysclock_factor = 0;
+	int found_this;
+
+	mutex_lock(&i2sdev->lock);
+
+	get_pcm_info(i2sdev, in, &pi, &other);
+
+	hw = &pi->substream->runtime->hw;
+	sdev = &i2sdev->sound;
+
+	if (pi->active) {
+		/* alsa messed up */
+		result = -EBUSY;
+		goto out_unlock;
+	}
+
+	/* we now need to assign the hw */
+	list_for_each_entry(cii, &sdev->codec_list, list) {
+		struct transfer_info *ti = cii->codec->transfers;
+		bus_factor = cii->codec->bus_factor;
+		sysclock_factor = cii->codec->sysclock_factor;
+		while (ti->formats && ti->rates) {
+			v = *ti;
+			if (ti->transfer_in == in
+			    && cii->codec->usable(cii, ti, &v)) {
+				if (masks_inited) {
+					formats &= v.formats;
+					rates &= v.rates;
+				} else {
+					formats = v.formats;
+					rates = v.rates;
+					masks_inited = 1;
+				}
+			}
+			ti++;
+		}
+	}
+	if (!masks_inited || !bus_factor || !sysclock_factor) {
+		result = -ENODEV;
+		goto out_unlock;
+	}
+	/* bus dependent stuff */
+	hw->info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
+		   SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_RESUME |
+		   SNDRV_PCM_INFO_JOINT_DUPLEX;
+
+	CHECK_RATE(5512);
+	CHECK_RATE(8000);
+	CHECK_RATE(11025);
+	CHECK_RATE(16000);
+	CHECK_RATE(22050);
+	CHECK_RATE(32000);
+	CHECK_RATE(44100);
+	CHECK_RATE(48000);
+	CHECK_RATE(64000);
+	CHECK_RATE(88200);
+	CHECK_RATE(96000);
+	CHECK_RATE(176400);
+	CHECK_RATE(192000);
+	hw->rates = rates;
+
+	/* well. the codec might want 24 bits only, and we'll
+	 * ever only transfer 24 bits, but they are top-aligned!
+	 * So for alsa, we claim that we're doing full 32 bit
+	 * while in reality we'll ignore the lower 8 bits of
+	 * that when doing playback (they're transferred as 0
+	 * as far as I know, no codecs we have are 32-bit capable
+	 * so I can't really test) and when doing recording we'll
+	 * always have those lower 8 bits recorded as 0 */
+	if (formats & SNDRV_PCM_FMTBIT_S24_BE)
+		formats |= SNDRV_PCM_FMTBIT_S32_BE;
+	if (formats & SNDRV_PCM_FMTBIT_U24_BE)
+		formats |= SNDRV_PCM_FMTBIT_U32_BE;
+	/* now mask off what we can support. I suppose we could
+	 * also support S24_3LE and some similar formats, but I
+	 * doubt there's a codec that would be able to use that,
+	 * so we don't support it here. */
+	hw->formats = formats & (SNDRV_PCM_FMTBIT_S16_BE |
+				 SNDRV_PCM_FMTBIT_U16_BE |
+				 SNDRV_PCM_FMTBIT_S32_BE |
+				 SNDRV_PCM_FMTBIT_U32_BE);
+
+	/* we need to set the highest and lowest rate possible.
+	 * These are the highest and lowest rates alsa can
+	 * support properly in its bitfield.
+	 * Below, we'll use that to restrict to the rate
+	 * currently in use (if any). */
+	hw->rate_min = 5512;
+	hw->rate_max = 192000;
+	/* if the other stream is active, then we can only
+	 * support what it is currently using.
+	 * FIXME: I lied. This comment is wrong. We can support
+	 * anything that works with the same serial format, ie.
+	 * when recording 24 bit sound we can well play 16 bit
+	 * sound at the same time iff using the same transfer mode.
+	 */
+	if (other->active) {
+		/* FIXME: is this guaranteed by the alsa api? */
+		hw->formats &= (1ULL << i2sdev->format);
+		/* see above, restrict rates to the one we already have */
+		hw->rate_min = i2sdev->rate;
+		hw->rate_max = i2sdev->rate;
+	}
+
+	hw->channels_min = 2;
+	hw->channels_max = 2;
+	/* these are somewhat arbitrary */
+	hw->buffer_bytes_max = 131072;
+	hw->period_bytes_min = 256;
+	hw->period_bytes_max = 16384;
+	hw->periods_min = 3;
+	hw->periods_max = MAX_DBDMA_COMMANDS;
+	err = snd_pcm_hw_constraint_integer(pi->substream->runtime,
+					    SNDRV_PCM_HW_PARAM_PERIODS);
+	if (err < 0) {
+		result = err;
+		goto out_unlock;
+	}
+	list_for_each_entry(cii, &sdev->codec_list, list) {
+		if (cii->codec->open) {
+			err = cii->codec->open(cii, pi->substream);
+			if (err) {
+				result = err;
+				/* unwind */
+				found_this = 0;
+				list_for_each_entry_reverse(rev,
+				    &sdev->codec_list, list) {
+					if (found_this && rev->codec->close) {
+						rev->codec->close(rev,
+								pi->substream);
+					}
+					if (rev == cii)
+						found_this = 1;
+				}
+				goto out_unlock;
+			}
+		}
+	}
+
+ out_unlock:
+	mutex_unlock(&i2sdev->lock);
+	return result;
+}
+
+#undef CHECK_RATE
+
+static int i2sbus_pcm_close(struct i2sbus_dev *i2sdev, int in)
+{
+	struct codec_info_item *cii;
+	struct pcm_info *pi;
+	int err = 0, tmp;
+
+	mutex_lock(&i2sdev->lock);
+
+	get_pcm_info(i2sdev, in, &pi, NULL);
+
+	list_for_each_entry(cii, &i2sdev->sound.codec_list, list) {
+		if (cii->codec->close) {
+			tmp = cii->codec->close(cii, pi->substream);
+			if (tmp)
+				err = tmp;
+		}
+	}
+
+	pi->substream = NULL;
+	pi->active = 0;
+	mutex_unlock(&i2sdev->lock);
+	return err;
+}
+
+static void i2sbus_wait_for_stop(struct i2sbus_dev *i2sdev,
+				 struct pcm_info *pi)
+{
+	unsigned long flags;
+	struct completion done;
+	long timeout;
+
+	spin_lock_irqsave(&i2sdev->low_lock, flags);
+	if (pi->dbdma_ring.stopping) {
+		init_completion(&done);
+		pi->stop_completion = &done;
+		spin_unlock_irqrestore(&i2sdev->low_lock, flags);
+		timeout = wait_for_completion_timeout(&done, HZ);
+		spin_lock_irqsave(&i2sdev->low_lock, flags);
+		pi->stop_completion = NULL;
+		if (timeout == 0) {
+			/* timeout expired, stop dbdma forcefully */
+			printk(KERN_ERR "i2sbus_wait_for_stop: timed out\n");
+			/* make sure RUN, PAUSE and S0 bits are cleared */
+			out_le32(&pi->dbdma->control, (RUN | PAUSE | 1) << 16);
+			pi->dbdma_ring.stopping = 0;
+			timeout = 10;
+			while (in_le32(&pi->dbdma->status) & ACTIVE) {
+				if (--timeout <= 0)
+					break;
+				udelay(1);
+			}
+		}
+	}
+	spin_unlock_irqrestore(&i2sdev->low_lock, flags);
+}
+
+#ifdef CONFIG_PM
+void i2sbus_wait_for_stop_both(struct i2sbus_dev *i2sdev)
+{
+	struct pcm_info *pi;
+
+	get_pcm_info(i2sdev, 0, &pi, NULL);
+	i2sbus_wait_for_stop(i2sdev, pi);
+	get_pcm_info(i2sdev, 1, &pi, NULL);
+	i2sbus_wait_for_stop(i2sdev, pi);
+}
+#endif
+
+static int i2sbus_hw_params(struct snd_pcm_substream *substream,
+			    struct snd_pcm_hw_params *params)
+{
+	return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+}
+
+static inline int i2sbus_hw_free(struct snd_pcm_substream *substream, int in)
+{
+	struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+	struct pcm_info *pi;
+
+	get_pcm_info(i2sdev, in, &pi, NULL);
+	if (pi->dbdma_ring.stopping)
+		i2sbus_wait_for_stop(i2sdev, pi);
+	snd_pcm_lib_free_pages(substream);
+	return 0;
+}
+
+static int i2sbus_playback_hw_free(struct snd_pcm_substream *substream)
+{
+	return i2sbus_hw_free(substream, 0);
+}
+
+static int i2sbus_record_hw_free(struct snd_pcm_substream *substream)
+{
+	return i2sbus_hw_free(substream, 1);
+}
+
+static int i2sbus_pcm_prepare(struct i2sbus_dev *i2sdev, int in)
+{
+	/* whee. Hard work now. The user has selected a bitrate
+	 * and bit format, so now we have to program our
+	 * I2S controller appropriately. */
+	struct snd_pcm_runtime *runtime;
+	struct dbdma_cmd *command;
+	int i, periodsize, nperiods;
+	dma_addr_t offset;
+	struct bus_info bi;
+	struct codec_info_item *cii;
+	int sfr = 0;		/* serial format register */
+	int dws = 0;		/* data word sizes reg */
+	int input_16bit;
+	struct pcm_info *pi, *other;
+	int cnt;
+	int result = 0;
+	unsigned int cmd, stopaddr;
+
+	mutex_lock(&i2sdev->lock);
+
+	get_pcm_info(i2sdev, in, &pi, &other);
+
+	if (pi->dbdma_ring.running) {
+		result = -EBUSY;
+		goto out_unlock;
+	}
+	if (pi->dbdma_ring.stopping)
+		i2sbus_wait_for_stop(i2sdev, pi);
+
+	if (!pi->substream || !pi->substream->runtime) {
+		result = -EINVAL;
+		goto out_unlock;
+	}
+
+	runtime = pi->substream->runtime;
+	pi->active = 1;
+	if (other->active &&
+	    ((i2sdev->format != runtime->format)
+	     || (i2sdev->rate != runtime->rate))) {
+		result = -EINVAL;
+		goto out_unlock;
+	}
+
+	i2sdev->format = runtime->format;
+	i2sdev->rate = runtime->rate;
+
+	periodsize = snd_pcm_lib_period_bytes(pi->substream);
+	nperiods = pi->substream->runtime->periods;
+	pi->current_period = 0;
+
+	/* generate dbdma command ring first */
+	command = pi->dbdma_ring.cmds;
+	memset(command, 0, (nperiods + 2) * sizeof(struct dbdma_cmd));
+
+	/* commands to DMA to/from the ring */
+	/*
+	 * For input, we need to do a graceful stop; if we abort
+	 * the DMA, we end up with leftover bytes that corrupt
+	 * the next recording.  To do this we set the S0 status
+	 * bit and wait for the DMA controller to stop.  Each
+	 * command has a branch condition to
+	 * make it branch to a stop command if S0 is set.
+	 * On input we also need to wait for the S7 bit to be
+	 * set before turning off the DMA controller.
+	 * In fact we do the graceful stop for output as well.
+	 */
+	offset = runtime->dma_addr;
+	cmd = (in? INPUT_MORE: OUTPUT_MORE) | BR_IFSET | INTR_ALWAYS;
+	stopaddr = pi->dbdma_ring.bus_cmd_start +
+		(nperiods + 1) * sizeof(struct dbdma_cmd);
+	for (i = 0; i < nperiods; i++, command++, offset += periodsize) {
+		command->command = cpu_to_le16(cmd);
+		command->cmd_dep = cpu_to_le32(stopaddr);
+		command->phy_addr = cpu_to_le32(offset);
+		command->req_count = cpu_to_le16(periodsize);
+	}
+
+	/* branch back to beginning of ring */
+	command->command = cpu_to_le16(DBDMA_NOP | BR_ALWAYS);
+	command->cmd_dep = cpu_to_le32(pi->dbdma_ring.bus_cmd_start);
+	command++;
+
+	/* set stop command */
+	command->command = cpu_to_le16(DBDMA_STOP);
+
+	/* ok, let's set the serial format and stuff */
+	switch (runtime->format) {
+	/* 16 bit formats */
+	case SNDRV_PCM_FORMAT_S16_BE:
+	case SNDRV_PCM_FORMAT_U16_BE:
+		/* FIXME: if we add different bus factors we need to
+		 * do more here!! */
+		bi.bus_factor = 0;
+		list_for_each_entry(cii, &i2sdev->sound.codec_list, list) {
+			bi.bus_factor = cii->codec->bus_factor;
+			break;
+		}
+		if (!bi.bus_factor) {
+			result = -ENODEV;
+			goto out_unlock;
+		}
+		input_16bit = 1;
+		break;
+	case SNDRV_PCM_FORMAT_S32_BE:
+	case SNDRV_PCM_FORMAT_U32_BE:
+		/* force 64x bus speed, otherwise the data cannot be
+		 * transferred quickly enough! */
+		bi.bus_factor = 64;
+		input_16bit = 0;
+		break;
+	default:
+		result = -EINVAL;
+		goto out_unlock;
+	}
+	/* we assume all sysclocks are the same! */
+	list_for_each_entry(cii, &i2sdev->sound.codec_list, list) {
+		bi.sysclock_factor = cii->codec->sysclock_factor;
+		break;
+	}
+
+	if (clock_and_divisors(bi.sysclock_factor,
+			       bi.bus_factor,
+			       runtime->rate,
+			       &sfr) < 0) {
+		result = -EINVAL;
+		goto out_unlock;
+	}
+	switch (bi.bus_factor) {
+	case 32:
+		sfr |= I2S_SF_SERIAL_FORMAT_I2S_32X;
+		break;
+	case 64:
+		sfr |= I2S_SF_SERIAL_FORMAT_I2S_64X;
+		break;
+	}
+	/* FIXME: THIS ASSUMES MASTER ALL THE TIME */
+	sfr |= I2S_SF_SCLK_MASTER;
+
+	list_for_each_entry(cii, &i2sdev->sound.codec_list, list) {
+		int err = 0;
+		if (cii->codec->prepare)
+			err = cii->codec->prepare(cii, &bi, pi->substream);
+		if (err) {
+			result = err;
+			goto out_unlock;
+		}
+	}
+	/* codecs are fine with it, so set our clocks */
+	if (input_16bit)
+		dws =	(2 << I2S_DWS_NUM_CHANNELS_IN_SHIFT) |
+			(2 << I2S_DWS_NUM_CHANNELS_OUT_SHIFT) |
+			I2S_DWS_DATA_IN_16BIT | I2S_DWS_DATA_OUT_16BIT;
+	else
+		dws =	(2 << I2S_DWS_NUM_CHANNELS_IN_SHIFT) |
+			(2 << I2S_DWS_NUM_CHANNELS_OUT_SHIFT) |
+			I2S_DWS_DATA_IN_24BIT | I2S_DWS_DATA_OUT_24BIT;
+
+	/* early exit if already programmed correctly */
+	/* not locking these is fine since we touch them only in this function */
+	if (in_le32(&i2sdev->intfregs->serial_format) == sfr
+	 && in_le32(&i2sdev->intfregs->data_word_sizes) == dws)
+		goto out_unlock;
+
+	/* let's notify the codecs about clocks going away.
+	 * For now we only do mastering on the i2s cell... */
+	list_for_each_entry(cii, &i2sdev->sound.codec_list, list)
+		if (cii->codec->switch_clock)
+			cii->codec->switch_clock(cii, CLOCK_SWITCH_PREPARE_SLAVE);
+
+	i2sbus_control_enable(i2sdev->control, i2sdev);
+	i2sbus_control_cell(i2sdev->control, i2sdev, 1);
+
+	out_le32(&i2sdev->intfregs->intr_ctl, I2S_PENDING_CLOCKS_STOPPED);
+
+	i2sbus_control_clock(i2sdev->control, i2sdev, 0);
+
+	msleep(1);
+
+	/* wait for clock stopped. This can apparently take a while... */
+	cnt = 100;
+	while (cnt-- &&
+	    !(in_le32(&i2sdev->intfregs->intr_ctl) & I2S_PENDING_CLOCKS_STOPPED)) {
+		msleep(5);
+	}
+	out_le32(&i2sdev->intfregs->intr_ctl, I2S_PENDING_CLOCKS_STOPPED);
+
+	/* not locking these is fine since we touch them only in this function */
+	out_le32(&i2sdev->intfregs->serial_format, sfr);
+	out_le32(&i2sdev->intfregs->data_word_sizes, dws);
+
+        i2sbus_control_enable(i2sdev->control, i2sdev);
+        i2sbus_control_cell(i2sdev->control, i2sdev, 1);
+        i2sbus_control_clock(i2sdev->control, i2sdev, 1);
+	msleep(1);
+
+	list_for_each_entry(cii, &i2sdev->sound.codec_list, list)
+		if (cii->codec->switch_clock)
+			cii->codec->switch_clock(cii, CLOCK_SWITCH_SLAVE);
+
+ out_unlock:
+	mutex_unlock(&i2sdev->lock);
+	return result;
+}
+
+#ifdef CONFIG_PM
+void i2sbus_pcm_prepare_both(struct i2sbus_dev *i2sdev)
+{
+	i2sbus_pcm_prepare(i2sdev, 0);
+	i2sbus_pcm_prepare(i2sdev, 1);
+}
+#endif
+
+static int i2sbus_pcm_trigger(struct i2sbus_dev *i2sdev, int in, int cmd)
+{
+	struct codec_info_item *cii;
+	struct pcm_info *pi;
+	int result = 0;
+	unsigned long flags;
+
+	spin_lock_irqsave(&i2sdev->low_lock, flags);
+
+	get_pcm_info(i2sdev, in, &pi, NULL);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+		if (pi->dbdma_ring.running) {
+			result = -EALREADY;
+			goto out_unlock;
+		}
+		list_for_each_entry(cii, &i2sdev->sound.codec_list, list)
+			if (cii->codec->start)
+				cii->codec->start(cii, pi->substream);
+		pi->dbdma_ring.running = 1;
+
+		if (pi->dbdma_ring.stopping) {
+			/* Clear the S0 bit, then see if we stopped yet */
+			out_le32(&pi->dbdma->control, 1 << 16);
+			if (in_le32(&pi->dbdma->status) & ACTIVE) {
+				/* possible race here? */
+				udelay(10);
+				if (in_le32(&pi->dbdma->status) & ACTIVE) {
+					pi->dbdma_ring.stopping = 0;
+					goto out_unlock; /* keep running */
+				}
+			}
+		}
+
+		/* make sure RUN, PAUSE and S0 bits are cleared */
+		out_le32(&pi->dbdma->control, (RUN | PAUSE | 1) << 16);
+
+		/* set branch condition select register */
+		out_le32(&pi->dbdma->br_sel, (1 << 16) | 1);
+
+		/* write dma command buffer address to the dbdma chip */
+		out_le32(&pi->dbdma->cmdptr, pi->dbdma_ring.bus_cmd_start);
+
+		/* initialize the frame count and current period */
+		pi->current_period = 0;
+		pi->frame_count = in_le32(&i2sdev->intfregs->frame_count);
+
+		/* set the DMA controller running */
+		out_le32(&pi->dbdma->control, (RUN << 16) | RUN);
+
+		/* off you go! */
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+		if (!pi->dbdma_ring.running) {
+			result = -EALREADY;
+			goto out_unlock;
+		}
+		pi->dbdma_ring.running = 0;
+
+		/* Set the S0 bit to make the DMA branch to the stop cmd */
+		out_le32(&pi->dbdma->control, (1 << 16) | 1);
+		pi->dbdma_ring.stopping = 1;
+
+		list_for_each_entry(cii, &i2sdev->sound.codec_list, list)
+			if (cii->codec->stop)
+				cii->codec->stop(cii, pi->substream);
+		break;
+	default:
+		result = -EINVAL;
+		goto out_unlock;
+	}
+
+ out_unlock:
+	spin_unlock_irqrestore(&i2sdev->low_lock, flags);
+	return result;
+}
+
+static snd_pcm_uframes_t i2sbus_pcm_pointer(struct i2sbus_dev *i2sdev, int in)
+{
+	struct pcm_info *pi;
+	u32 fc;
+
+	get_pcm_info(i2sdev, in, &pi, NULL);
+
+	fc = in_le32(&i2sdev->intfregs->frame_count);
+	fc = fc - pi->frame_count;
+
+	if (fc >= pi->substream->runtime->buffer_size)
+		fc %= pi->substream->runtime->buffer_size;
+	return fc;
+}
+
+static inline void handle_interrupt(struct i2sbus_dev *i2sdev, int in)
+{
+	struct pcm_info *pi;
+	u32 fc, nframes;
+	u32 status;
+	int timeout, i;
+	int dma_stopped = 0;
+	struct snd_pcm_runtime *runtime;
+
+	spin_lock(&i2sdev->low_lock);
+	get_pcm_info(i2sdev, in, &pi, NULL);
+	if (!pi->dbdma_ring.running && !pi->dbdma_ring.stopping)
+		goto out_unlock;
+
+	i = pi->current_period;
+	runtime = pi->substream->runtime;
+	while (pi->dbdma_ring.cmds[i].xfer_status) {
+		if (le16_to_cpu(pi->dbdma_ring.cmds[i].xfer_status) & BT)
+			/*
+			 * BT is the branch taken bit.  If it took a branch
+			 * it is because we set the S0 bit to make it
+			 * branch to the stop command.
+			 */
+			dma_stopped = 1;
+		pi->dbdma_ring.cmds[i].xfer_status = 0;
+
+		if (++i >= runtime->periods) {
+			i = 0;
+			pi->frame_count += runtime->buffer_size;
+		}
+		pi->current_period = i;
+
+		/*
+		 * Check the frame count.  The DMA tends to get a bit
+		 * ahead of the frame counter, which confuses the core.
+		 */
+		fc = in_le32(&i2sdev->intfregs->frame_count);
+		nframes = i * runtime->period_size;
+		if (fc < pi->frame_count + nframes)
+			pi->frame_count = fc - nframes;
+	}
+
+	if (dma_stopped) {
+		timeout = 1000;
+		for (;;) {
+			status = in_le32(&pi->dbdma->status);
+			if (!(status & ACTIVE) && (!in || (status & 0x80)))
+				break;
+			if (--timeout <= 0) {
+				printk(KERN_ERR "i2sbus: timed out "
+				       "waiting for DMA to stop!\n");
+				break;
+			}
+			udelay(1);
+		}
+
+		/* Turn off DMA controller, clear S0 bit */
+		out_le32(&pi->dbdma->control, (RUN | PAUSE | 1) << 16);
+
+		pi->dbdma_ring.stopping = 0;
+		if (pi->stop_completion)
+			complete(pi->stop_completion);
+	}
+
+	if (!pi->dbdma_ring.running)
+		goto out_unlock;
+	spin_unlock(&i2sdev->low_lock);
+	/* may call _trigger again, hence needs to be unlocked */
+	snd_pcm_period_elapsed(pi->substream);
+	return;
+
+ out_unlock:
+	spin_unlock(&i2sdev->low_lock);
+}
+
+irqreturn_t i2sbus_tx_intr(int irq, void *devid)
+{
+	handle_interrupt((struct i2sbus_dev *)devid, 0);
+	return IRQ_HANDLED;
+}
+
+irqreturn_t i2sbus_rx_intr(int irq, void *devid)
+{
+	handle_interrupt((struct i2sbus_dev *)devid, 1);
+	return IRQ_HANDLED;
+}
+
+static int i2sbus_playback_open(struct snd_pcm_substream *substream)
+{
+	struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+
+	if (!i2sdev)
+		return -EINVAL;
+	i2sdev->out.substream = substream;
+	return i2sbus_pcm_open(i2sdev, 0);
+}
+
+static int i2sbus_playback_close(struct snd_pcm_substream *substream)
+{
+	struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+	int err;
+
+	if (!i2sdev)
+		return -EINVAL;
+	if (i2sdev->out.substream != substream)
+		return -EINVAL;
+	err = i2sbus_pcm_close(i2sdev, 0);
+	if (!err)
+		i2sdev->out.substream = NULL;
+	return err;
+}
+
+static int i2sbus_playback_prepare(struct snd_pcm_substream *substream)
+{
+	struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+
+	if (!i2sdev)
+		return -EINVAL;
+	if (i2sdev->out.substream != substream)
+		return -EINVAL;
+	return i2sbus_pcm_prepare(i2sdev, 0);
+}
+
+static int i2sbus_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+
+	if (!i2sdev)
+		return -EINVAL;
+	if (i2sdev->out.substream != substream)
+		return -EINVAL;
+	return i2sbus_pcm_trigger(i2sdev, 0, cmd);
+}
+
+static snd_pcm_uframes_t i2sbus_playback_pointer(struct snd_pcm_substream
+						 *substream)
+{
+	struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+
+	if (!i2sdev)
+		return -EINVAL;
+	if (i2sdev->out.substream != substream)
+		return 0;
+	return i2sbus_pcm_pointer(i2sdev, 0);
+}
+
+static struct snd_pcm_ops i2sbus_playback_ops = {
+	.open =		i2sbus_playback_open,
+	.close =	i2sbus_playback_close,
+	.ioctl =	snd_pcm_lib_ioctl,
+	.hw_params =	i2sbus_hw_params,
+	.hw_free =	i2sbus_playback_hw_free,
+	.prepare =	i2sbus_playback_prepare,
+	.trigger =	i2sbus_playback_trigger,
+	.pointer =	i2sbus_playback_pointer,
+};
+
+static int i2sbus_record_open(struct snd_pcm_substream *substream)
+{
+	struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+
+	if (!i2sdev)
+		return -EINVAL;
+	i2sdev->in.substream = substream;
+	return i2sbus_pcm_open(i2sdev, 1);
+}
+
+static int i2sbus_record_close(struct snd_pcm_substream *substream)
+{
+	struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+	int err;
+
+	if (!i2sdev)
+		return -EINVAL;
+	if (i2sdev->in.substream != substream)
+		return -EINVAL;
+	err = i2sbus_pcm_close(i2sdev, 1);
+	if (!err)
+		i2sdev->in.substream = NULL;
+	return err;
+}
+
+static int i2sbus_record_prepare(struct snd_pcm_substream *substream)
+{
+	struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+
+	if (!i2sdev)
+		return -EINVAL;
+	if (i2sdev->in.substream != substream)
+		return -EINVAL;
+	return i2sbus_pcm_prepare(i2sdev, 1);
+}
+
+static int i2sbus_record_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+
+	if (!i2sdev)
+		return -EINVAL;
+	if (i2sdev->in.substream != substream)
+		return -EINVAL;
+	return i2sbus_pcm_trigger(i2sdev, 1, cmd);
+}
+
+static snd_pcm_uframes_t i2sbus_record_pointer(struct snd_pcm_substream
+					       *substream)
+{
+	struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
+
+	if (!i2sdev)
+		return -EINVAL;
+	if (i2sdev->in.substream != substream)
+		return 0;
+	return i2sbus_pcm_pointer(i2sdev, 1);
+}
+
+static struct snd_pcm_ops i2sbus_record_ops = {
+	.open =		i2sbus_record_open,
+	.close =	i2sbus_record_close,
+	.ioctl =	snd_pcm_lib_ioctl,
+	.hw_params =	i2sbus_hw_params,
+	.hw_free =	i2sbus_record_hw_free,
+	.prepare =	i2sbus_record_prepare,
+	.trigger =	i2sbus_record_trigger,
+	.pointer =	i2sbus_record_pointer,
+};
+
+static void i2sbus_private_free(struct snd_pcm *pcm)
+{
+	struct i2sbus_dev *i2sdev = snd_pcm_chip(pcm);
+	struct codec_info_item *p, *tmp;
+
+	i2sdev->sound.pcm = NULL;
+	i2sdev->out.created = 0;
+	i2sdev->in.created = 0;
+	list_for_each_entry_safe(p, tmp, &i2sdev->sound.codec_list, list) {
+		printk(KERN_ERR "i2sbus: a codec didn't unregister!\n");
+		list_del(&p->list);
+		module_put(p->codec->owner);
+		kfree(p);
+	}
+	soundbus_dev_put(&i2sdev->sound);
+	module_put(THIS_MODULE);
+}
+
+int
+i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card,
+		    struct codec_info *ci, void *data)
+{
+	int err, in = 0, out = 0;
+	struct transfer_info *tmp;
+	struct i2sbus_dev *i2sdev = soundbus_dev_to_i2sbus_dev(dev);
+	struct codec_info_item *cii;
+
+	if (!dev->pcmname || dev->pcmid == -1) {
+		printk(KERN_ERR "i2sbus: pcm name and id must be set!\n");
+		return -EINVAL;
+	}
+
+	list_for_each_entry(cii, &dev->codec_list, list) {
+		if (cii->codec_data == data)
+			return -EALREADY;
+	}
+
+	if (!ci->transfers || !ci->transfers->formats
+	    || !ci->transfers->rates || !ci->usable)
+		return -EINVAL;
+
+	/* we currently code the i2s transfer on the clock, and support only
+	 * 32 and 64 */
+	if (ci->bus_factor != 32 && ci->bus_factor != 64)
+		return -EINVAL;
+
+	/* If you want to fix this, you need to keep track of what transport infos
+	 * are to be used, which codecs they belong to, and then fix all the
+	 * sysclock/busclock stuff above to depend on which is usable */
+	list_for_each_entry(cii, &dev->codec_list, list) {
+		if (cii->codec->sysclock_factor != ci->sysclock_factor) {
+			printk(KERN_DEBUG
+			       "cannot yet handle multiple different sysclocks!\n");
+			return -EINVAL;
+		}
+		if (cii->codec->bus_factor != ci->bus_factor) {
+			printk(KERN_DEBUG
+			       "cannot yet handle multiple different bus clocks!\n");
+			return -EINVAL;
+		}
+	}
+
+	tmp = ci->transfers;
+	while (tmp->formats && tmp->rates) {
+		if (tmp->transfer_in)
+			in = 1;
+		else
+			out = 1;
+		tmp++;
+	}
+
+	cii = kzalloc(sizeof(struct codec_info_item), GFP_KERNEL);
+	if (!cii) {
+		printk(KERN_DEBUG "i2sbus: failed to allocate cii\n");
+		return -ENOMEM;
+	}
+
+	/* use the private data to point to the codec info */
+	cii->sdev = soundbus_dev_get(dev);
+	cii->codec = ci;
+	cii->codec_data = data;
+
+	if (!cii->sdev) {
+		printk(KERN_DEBUG
+		       "i2sbus: failed to get soundbus dev reference\n");
+		err = -ENODEV;
+		goto out_free_cii;
+	}
+
+	if (!try_module_get(THIS_MODULE)) {
+		printk(KERN_DEBUG "i2sbus: failed to get module reference!\n");
+		err = -EBUSY;
+		goto out_put_sdev;
+	}
+
+	if (!try_module_get(ci->owner)) {
+		printk(KERN_DEBUG
+		       "i2sbus: failed to get module reference to codec owner!\n");
+		err = -EBUSY;
+		goto out_put_this_module;
+	}
+
+	if (!dev->pcm) {
+		err = snd_pcm_new(card, dev->pcmname, dev->pcmid, 0, 0,
+				  &dev->pcm);
+		if (err) {
+			printk(KERN_DEBUG "i2sbus: failed to create pcm\n");
+			goto out_put_ci_module;
+		}
+		dev->pcm->dev = &dev->ofdev.dev;
+	}
+
+	/* ALSA yet again sucks.
+	 * If it is ever fixed, remove this line. See below. */
+	out = in = 1;
+
+	if (!i2sdev->out.created && out) {
+		if (dev->pcm->card != card) {
+			/* eh? */
+			printk(KERN_ERR
+			       "Can't attach same bus to different cards!\n");
+			err = -EINVAL;
+			goto out_put_ci_module;
+		}
+		err = snd_pcm_new_stream(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, 1);
+		if (err)
+			goto out_put_ci_module;
+		snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK,
+				&i2sbus_playback_ops);
+		i2sdev->out.created = 1;
+	}
+
+	if (!i2sdev->in.created && in) {
+		if (dev->pcm->card != card) {
+			printk(KERN_ERR
+			       "Can't attach same bus to different cards!\n");
+			err = -EINVAL;
+			goto out_put_ci_module;
+		}
+		err = snd_pcm_new_stream(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, 1);
+		if (err)
+			goto out_put_ci_module;
+		snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE,
+				&i2sbus_record_ops);
+		i2sdev->in.created = 1;
+	}
+
+	/* so we have to register the pcm after adding any substream
+	 * to it because alsa doesn't create the devices for the
+	 * substreams when we add them later.
+	 * Therefore, force in and out on both busses (above) and
+	 * register the pcm now instead of just after creating it.
+	 */
+	err = snd_device_register(card, dev->pcm);
+	if (err) {
+		printk(KERN_ERR "i2sbus: error registering new pcm\n");
+		goto out_put_ci_module;
+	}
+	/* no errors any more, so let's add this to our list */
+	list_add(&cii->list, &dev->codec_list);
+
+	dev->pcm->private_data = i2sdev;
+	dev->pcm->private_free = i2sbus_private_free;
+
+	/* well, we really should support scatter/gather DMA */
+	snd_pcm_lib_preallocate_pages_for_all(
+		dev->pcm, SNDRV_DMA_TYPE_DEV,
+		snd_dma_pci_data(macio_get_pci_dev(i2sdev->macio)),
+		64 * 1024, 64 * 1024);
+
+	return 0;
+ out_put_ci_module:
+	module_put(ci->owner);
+ out_put_this_module:
+	module_put(THIS_MODULE);
+ out_put_sdev:
+	soundbus_dev_put(dev);
+ out_free_cii:
+	kfree(cii);
+	return err;
+}
+
+void i2sbus_detach_codec(struct soundbus_dev *dev, void *data)
+{
+	struct codec_info_item *cii = NULL, *i;
+
+	list_for_each_entry(i, &dev->codec_list, list) {
+		if (i->codec_data == data) {
+			cii = i;
+			break;
+		}
+	}
+	if (cii) {
+		list_del(&cii->list);
+		module_put(cii->codec->owner);
+		kfree(cii);
+	}
+	/* no more codecs, but still a pcm? */
+	if (list_empty(&dev->codec_list) && dev->pcm) {
+		/* the actual cleanup is done by the callback above! */
+		snd_device_free(dev->pcm->card, dev->pcm);
+	}
+}
diff --git a/ANDROID_3.4.5/sound/aoa/soundbus/soundbus.h b/ANDROID_3.4.5/sound/aoa/soundbus/soundbus.h
new file mode 100644
index 00000000..adecbf36
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/soundbus/soundbus.h
@@ -0,0 +1,204 @@
+/*
+ * soundbus generic definitions
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+#ifndef __SOUNDBUS_H
+#define __SOUNDBUS_H
+
+#include <linux/of_device.h>
+#include <sound/pcm.h>
+#include <linux/list.h>
+
+
+/* When switching from master to slave or the other way around,
+ * you don't want to have the codec chip acting as clock source
+ * while the bus still is.
+ * More importantly, while switch from slave to master, you need
+ * to turn off the chip's master function first, but then there's
+ * no clock for a while and other chips might reset, so we notify
+ * their drivers after having switched.
+ * The constants here are codec-point of view, so when we switch
+ * the soundbus to master we tell the codec we're going to switch
+ * and give it CLOCK_SWITCH_PREPARE_SLAVE!
+ */
+enum clock_switch {
+	CLOCK_SWITCH_PREPARE_SLAVE,
+	CLOCK_SWITCH_PREPARE_MASTER,
+	CLOCK_SWITCH_SLAVE,
+	CLOCK_SWITCH_MASTER,
+	CLOCK_SWITCH_NOTIFY,
+};
+
+/* information on a transfer the codec can take */
+struct transfer_info {
+	u64 formats;		/* SNDRV_PCM_FMTBIT_* */
+	unsigned int rates;	/* SNDRV_PCM_RATE_* */
+	/* flags */
+	u32 transfer_in:1, /* input = 1, output = 0 */
+	    must_be_clock_source:1;
+	/* for codecs to distinguish among their TIs */
+	int tag;
+};
+
+struct codec_info_item {
+	struct codec_info *codec;
+	void *codec_data;
+	struct soundbus_dev *sdev;
+	/* internal, to be used by the soundbus provider */
+	struct list_head list;
+};
+
+/* for prepare, where the codecs need to know
+ * what we're going to drive the bus with */
+struct bus_info {
+	/* see below */
+	int sysclock_factor;
+	int bus_factor;
+};
+
+/* information on the codec itself, plus function pointers */
+struct codec_info {
+	/* the module this lives in */
+	struct module *owner;
+
+	/* supported transfer possibilities, array terminated by
+	 * formats or rates being 0. */
+	struct transfer_info *transfers;
+
+	/* Master clock speed factor
+	 * to be used (master clock speed = sysclock_factor * sampling freq)
+	 * Unused if the soundbus provider has no such notion.
+	 */
+	int sysclock_factor;
+
+	/* Bus factor, bus clock speed = bus_factor * sampling freq)
+	 * Unused if the soundbus provider has no such notion.
+	 */
+	int bus_factor;
+
+	/* operations */
+	/* clock switching, see above */
+	int (*switch_clock)(struct codec_info_item *cii,
+			    enum clock_switch clock);
+
+	/* called for each transfer_info when the user
+	 * opens the pcm device to determine what the
+	 * hardware can support at this point in time.
+	 * That can depend on other user-switchable controls.
+	 * Return 1 if usable, 0 if not.
+	 * out points to another instance of a transfer_info
+	 * which is initialised to the values in *ti, and
+	 * it's format and rate values can be modified by
+	 * the callback if it is necessary to further restrict
+	 * the formats that can be used at the moment, for
+	 * example when one codec has multiple logical codec
+	 * info structs for multiple inputs.
+	 */
+	int (*usable)(struct codec_info_item *cii,
+		      struct transfer_info *ti,
+		      struct transfer_info *out);
+
+	/* called when pcm stream is opened, probably not implemented
+	 * most of the time since it isn't too useful */
+	int (*open)(struct codec_info_item *cii,
+		    struct snd_pcm_substream *substream);
+
+	/* called when the pcm stream is closed, at this point
+	 * the user choices can all be unlocked (see below) */
+	int (*close)(struct codec_info_item *cii,
+		     struct snd_pcm_substream *substream);
+
+	/* if the codec must forbid some user choices because
+	 * they are not valid with the substream/transfer info,
+	 * it must do so here. Example: no digital output for
+	 * incompatible framerate, say 8KHz, on Onyx.
+	 * If the selected stuff in the substream is NOT
+	 * compatible, you have to reject this call! */
+	int (*prepare)(struct codec_info_item *cii,
+		       struct bus_info *bi,
+		       struct snd_pcm_substream *substream);
+
+	/* start() is called before data is pushed to the codec.
+	 * Note that start() must be atomic! */
+	int (*start)(struct codec_info_item *cii,
+		     struct snd_pcm_substream *substream);
+
+	/* stop() is called after data is no longer pushed to the codec.
+	 * Note that stop() must be atomic! */
+	int (*stop)(struct codec_info_item *cii,
+		    struct snd_pcm_substream *substream);
+
+	int (*suspend)(struct codec_info_item *cii, pm_message_t state);
+	int (*resume)(struct codec_info_item *cii);
+};
+
+/* information on a soundbus device */
+struct soundbus_dev {
+	/* the bus it belongs to */
+	struct list_head onbuslist;
+
+	/* the of device it represents */
+	struct platform_device ofdev;
+
+	/* what modules go by */
+	char modalias[32];
+
+	/* These fields must be before attach_codec can be called.
+	 * They should be set by the owner of the alsa card object
+	 * that is needed, and whoever sets them must make sure
+	 * that they are unique within that alsa card object. */
+	char *pcmname;
+	int pcmid;
+
+	/* this is assigned by the soundbus provider in attach_codec */
+	struct snd_pcm *pcm;
+
+	/* operations */
+	/* attach a codec to this soundbus, give the alsa
+	 * card object the PCMs for this soundbus should be in.
+	 * The 'data' pointer must be unique, it is used as the
+	 * key for detach_codec(). */
+	int (*attach_codec)(struct soundbus_dev *dev, struct snd_card *card,
+			    struct codec_info *ci, void *data);
+	void (*detach_codec)(struct soundbus_dev *dev, void *data);
+	/* TODO: suspend/resume */
+
+	/* private for the soundbus provider */
+	struct list_head codec_list;
+	u32 have_out:1, have_in:1;
+};
+#define to_soundbus_device(d) container_of(d, struct soundbus_dev, ofdev.dev)
+#define of_to_soundbus_device(d) container_of(d, struct soundbus_dev, ofdev)
+
+extern int soundbus_add_one(struct soundbus_dev *dev);
+extern void soundbus_remove_one(struct soundbus_dev *dev);
+
+extern struct soundbus_dev *soundbus_dev_get(struct soundbus_dev *dev);
+extern void soundbus_dev_put(struct soundbus_dev *dev);
+
+struct soundbus_driver {
+	char *name;
+	struct module *owner;
+
+	/* we don't implement any matching at all */
+
+	int	(*probe)(struct soundbus_dev* dev);
+	int	(*remove)(struct soundbus_dev* dev);
+
+	int	(*suspend)(struct soundbus_dev* dev, pm_message_t state);
+	int	(*resume)(struct soundbus_dev* dev);
+	int	(*shutdown)(struct soundbus_dev* dev);
+
+	struct device_driver driver;
+};
+#define to_soundbus_driver(drv) container_of(drv,struct soundbus_driver, driver)
+
+extern int soundbus_register_driver(struct soundbus_driver *drv);
+extern void soundbus_unregister_driver(struct soundbus_driver *drv);
+
+extern struct device_attribute soundbus_dev_attrs[];
+
+#endif /* __SOUNDBUS_H */
diff --git a/ANDROID_3.4.5/sound/aoa/soundbus/sysfs.c b/ANDROID_3.4.5/sound/aoa/soundbus/sysfs.c
new file mode 100644
index 00000000..e0980b5c
--- /dev/null
+++ b/ANDROID_3.4.5/sound/aoa/soundbus/sysfs.c
@@ -0,0 +1,42 @@
+#include <linux/kernel.h>
+#include <linux/stat.h>
+/* FIX UP */
+#include "soundbus.h"
+
+#define soundbus_config_of_attr(field, format_string)			\
+static ssize_t								\
+field##_show (struct device *dev, struct device_attribute *attr,	\
+              char *buf)						\
+{									\
+	struct soundbus_dev *mdev = to_soundbus_device (dev);		\
+	return sprintf (buf, format_string, mdev->ofdev.dev.of_node->field); \
+}
+
+static ssize_t modalias_show(struct device *dev, struct device_attribute *attr,
+			     char *buf)
+{
+	struct soundbus_dev *sdev = to_soundbus_device(dev);
+	struct platform_device *of = &sdev->ofdev;
+	int length;
+
+	if (*sdev->modalias) {
+		strlcpy(buf, sdev->modalias, sizeof(sdev->modalias) + 1);
+		strcat(buf, "\n");
+		length = strlen(buf);
+	} else {
+		length = sprintf(buf, "of:N%sT%s\n",
+				 of->dev.of_node->name, of->dev.of_node->type);
+	}
+
+	return length;
+}
+
+soundbus_config_of_attr (name, "%s\n");
+soundbus_config_of_attr (type, "%s\n");
+
+struct device_attribute soundbus_dev_attrs[] = {
+	__ATTR_RO(name),
+	__ATTR_RO(type),
+	__ATTR_RO(modalias),
+	__ATTR_NULL
+};