Merge branch 'master' into for-next

63 files changed, 990 insertions, 583 deletions

diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c
index 1cd9b301df03..3fd1a7e24928 100644
--- a/sound/aoa/fabrics/layout.c
+++ b/sound/aoa/fabrics/layout.c
@@ -992,7 +992,7 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
 		return -ENODEV;
 
 	/* by breaking out we keep a reference */
-	while ((sound = of_get_next_child(sdev->ofdev.node, sound))) {
+	while ((sound = of_get_next_child(sdev->ofdev.dev.of_node, sound))) {
 		if (sound->type && strcasecmp(sound->type, "soundchip") == 0)
 			break;
 	}
diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c
index fa8ab2815a98..99ca7120e269 100644
--- a/sound/aoa/soundbus/core.c
+++ b/sound/aoa/soundbus/core.c
@@ -74,11 +74,11 @@ static int soundbus_uevent(struct device *dev, struct kobj_uevent_env *env)
 	of = &soundbus_dev->ofdev;
 
 	/* stuff we want to pass to /sbin/hotplug */
-	retval = add_uevent_var(env, "OF_NAME=%s", of->node->name);
+	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->node->type);
+	retval = add_uevent_var(env, "OF_TYPE=%s", of->dev.of_node->type);
 	if (retval)
 		return retval;
 
@@ -86,7 +86,7 @@ static int soundbus_uevent(struct device *dev, struct kobj_uevent_env *env)
 	 * 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->node, "compatible", &cplen);
+	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);
@@ -169,7 +169,7 @@ int soundbus_add_one(struct soundbus_dev *dev)
 
 	/* sanity checks */
 	if (!dev->attach_codec ||
-	    !dev->ofdev.node ||
+	    !dev->ofdev.dev.of_node ||
 	    dev->pcmname ||
 	    dev->pcmid != -1) {
 		printk(KERN_ERR "soundbus: adding device failed sanity check!\n");
diff --git a/sound/aoa/soundbus/i2sbus/control.c b/sound/aoa/soundbus/i2sbus/control.c
index 47f854c2001f..4dc9b49c02cf 100644
--- a/sound/aoa/soundbus/i2sbus/control.c
+++ b/sound/aoa/soundbus/i2sbus/control.c
@@ -42,7 +42,7 @@ int i2sbus_control_add_dev(struct i2sbus_control *c,
 {
 	struct device_node *np;
 
-	np = i2sdev->sound.ofdev.node;
+	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");
diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c
index 9d6f3b176ed1..3ff8cc5f487a 100644
--- a/sound/aoa/soundbus/i2sbus/core.c
+++ b/sound/aoa/soundbus/i2sbus/core.c
@@ -221,9 +221,9 @@ static int i2sbus_add_dev(struct macio_dev *macio,
 
 	mutex_init(&dev->lock);
 	spin_lock_init(&dev->low_lock);
-	dev->sound.ofdev.node = np;
-	dev->sound.ofdev.dma_mask = macio->ofdev.dma_mask;
-	dev->sound.ofdev.dev.dma_mask = &dev->sound.ofdev.dma_mask;
+	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;
@@ -346,7 +346,7 @@ static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match)
 		return -ENODEV;
 	}
 
-	while ((np = of_get_next_child(dev->ofdev.node, np))) {
+	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);
@@ -437,9 +437,11 @@ static int i2sbus_shutdown(struct macio_dev* dev)
 }
 
 static struct macio_driver i2sbus_drv = {
-	.name = "soundbus-i2s",
-	.owner = THIS_MODULE,
-	.match_table = i2sbus_match,
+	.driver = {
+		.name = "soundbus-i2s",
+		.owner = THIS_MODULE,
+		.of_match_table = i2sbus_match,
+	},
 	.probe = i2sbus_probe,
 	.remove = i2sbus_remove,
 #ifdef CONFIG_PM
diff --git a/sound/aoa/soundbus/sysfs.c b/sound/aoa/soundbus/sysfs.c
index f580942b5c09..6496e754f00a 100644
--- a/sound/aoa/soundbus/sysfs.c
+++ b/sound/aoa/soundbus/sysfs.c
@@ -9,7 +9,7 @@ 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.node->field);	\
+	return sprintf (buf, format_string, mdev->ofdev.dev.of_node->field); \
 }
 
 static ssize_t modalias_show(struct device *dev, struct device_attribute *attr,
@@ -25,7 +25,7 @@ static ssize_t modalias_show(struct device *dev, struct device_attribute *attr,
 		length = strlen(buf);
 	} else {
 		length = sprintf(buf, "of:N%sT%s\n",
-				 of->node->name, of->node->type);
+				 of->dev.of_node->name, of->dev.of_node->type);
 	}
 
 	return length;
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index 428121a7e705..10c3a871a12d 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -657,7 +657,7 @@ static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev)
 	if (sr & AC97C_SR_CAEVT) {
 		struct snd_pcm_runtime *runtime;
 		int offset, next_period, block_size;
-		dev_info(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n",
+		dev_dbg(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n",
 				casr & AC97C_CSR_OVRUN   ? " OVRUN"   : "",
 				casr & AC97C_CSR_RXRDY   ? " RXRDY"   : "",
 				casr & AC97C_CSR_UNRUN   ? " UNRUN"   : "",
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index a2ff86189d2a..e9d98be190c5 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -345,7 +345,9 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 		new_hw_ptr = hw_base + pos;
 	}
       __delta:
-	delta = (new_hw_ptr - old_hw_ptr) % runtime->boundary;
+	delta = new_hw_ptr - old_hw_ptr;
+	if (delta < 0)
+		delta += runtime->boundary;
 	if (xrun_debug(substream, in_interrupt ?
 			XRUN_DEBUG_PERIODUPDATE : XRUN_DEBUG_HWPTRUPDATE)) {
 		char name[16];
@@ -439,8 +441,13 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 		snd_pcm_playback_silence(substream, new_hw_ptr);
 
 	if (in_interrupt) {
-		runtime->hw_ptr_interrupt = new_hw_ptr -
-				(new_hw_ptr % runtime->period_size);
+		delta = new_hw_ptr - runtime->hw_ptr_interrupt;
+		if (delta < 0)
+			delta += runtime->boundary;
+		delta -= (snd_pcm_uframes_t)delta % runtime->period_size;
+		runtime->hw_ptr_interrupt += delta;
+		if (runtime->hw_ptr_interrupt >= runtime->boundary)
+			runtime->hw_ptr_interrupt -= runtime->boundary;
 	}
 	runtime->hw_ptr_base = hw_base;
 	runtime->status->hw_ptr = new_hw_ptr;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 644c2bb17b86..303ac04ff6e4 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -27,7 +27,6 @@
 #include <linux/pm_qos_params.h>
 #include <linux/uio.h>
 #include <linux/dma-mapping.h>
-#include <linux/math64.h>
 #include <sound/core.h>
 #include <sound/control.h>
 #include <sound/info.h>
@@ -370,38 +369,6 @@ static int period_to_usecs(struct snd_pcm_runtime *runtime)
 	return usecs;
 }
 
-static int calc_boundary(struct snd_pcm_runtime *runtime)
-{
-	u_int64_t boundary;
-
-	boundary = (u_int64_t)runtime->buffer_size *
-		   (u_int64_t)runtime->period_size;
-#if BITS_PER_LONG < 64
-	/* try to find lowest common multiple for buffer and period */
-	if (boundary > LONG_MAX - runtime->buffer_size) {
-		u_int32_t remainder = -1;
-		u_int32_t divident = runtime->buffer_size;
-		u_int32_t divisor = runtime->period_size;
-		while (remainder) {
-			remainder = divident % divisor;
-			if (remainder) {
-				divident = divisor;
-				divisor = remainder;
-			}
-		}
-		boundary = div_u64(boundary, divisor);
-		if (boundary > LONG_MAX - runtime->buffer_size)
-			return -ERANGE;
-	}
-#endif
-	if (boundary == 0)
-		return -ERANGE;
-	runtime->boundary = boundary;
-	while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size)
-		runtime->boundary *= 2;
-	return 0;
-}
-
 static int snd_pcm_hw_params(struct snd_pcm_substream *substream,
 			     struct snd_pcm_hw_params *params)
 {
@@ -477,9 +444,9 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream,
 	runtime->stop_threshold = runtime->buffer_size;
 	runtime->silence_threshold = 0;
 	runtime->silence_size = 0;
-	err = calc_boundary(runtime);
-	if (err < 0)
-		goto _error;
+	runtime->boundary = runtime->buffer_size;
+	while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size)
+		runtime->boundary *= 2;
 
 	snd_pcm_timer_resolution_change(substream);
 	runtime->status->state = SNDRV_PCM_STATE_SETUP;
diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c
index 3e763d6a5d67..446cf9748664 100644
--- a/sound/mips/au1x00.c
+++ b/sound/mips/au1x00.c
@@ -516,6 +516,7 @@ get the interrupt driven case to work efficiently */
 			break;
 	if (i == 0x5000) {
 		printk(KERN_ERR "au1000 AC97: AC97 command read timeout\n");
+		spin_unlock(&au1000->ac97_lock);
 		return 0;
 	}
 
diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c
index 1f4774123064..13c214466d3b 100644
--- a/sound/oss/dmasound/dmasound_atari.c
+++ b/sound/oss/dmasound/dmasound_atari.c
@@ -1277,7 +1277,7 @@ static irqreturn_t AtaInterrupt(int irq, void *dummy)
 		 * (almost) like on the TT.
 		 */
 		write_sq_ignore_int = 0;
-		return IRQ_HANDLED;
+		goto out;
 	}
 
 	if (!write_sq.active) {
@@ -1285,7 +1285,7 @@ static irqreturn_t AtaInterrupt(int irq, void *dummy)
 		 * the sq variables, so better don't do anything here.
 		 */
 		WAKE_UP(write_sq.sync_queue);
-		return IRQ_HANDLED;
+		goto out;
 	}
 
 	/* Probably ;) one frame is finished. Well, in fact it may be that a
@@ -1322,6 +1322,7 @@ static irqreturn_t AtaInterrupt(int irq, void *dummy)
 	/* We are not playing after AtaPlay(), so there
 	   is nothing to play any more. Wake up a process
 	   waiting for audio output to drain. */
+out:
 	spin_unlock(&dmasound.lock);
 	return IRQ_HANDLED;
 }
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index f74c7372b3d1..1db586af4f9c 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -2578,6 +2578,9 @@ static int __devinit snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi)
 	if (err)
 		return -err;
 
+	memset(&prev_ctl, 0, sizeof(prev_ctl));
+	prev_ctl.control_type = -1;
+
 	for (idx = 0; idx < 2000; idx++) {
 		err = hpi_mixer_get_control_by_index(
 				ss, asihpi->h_mixer,
diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h
index 99400de6c075..0173bbe62b67 100644
--- a/sound/pci/asihpi/hpi.h
+++ b/sound/pci/asihpi/hpi.h
@@ -50,7 +50,7 @@ i.e 3.05.02 is a development version
 #define HPI_VER_RELEASE(v) ((int)(v & 0xFF))
 
 /* Use single digits for versions less that 10 to avoid octal. */
-#define HPI_VER HPI_VERSION_CONSTRUCTOR(4L, 3, 18)
+#define HPI_VER HPI_VERSION_CONSTRUCTOR(4L, 3, 25)
 
 /* Library version as documented in hpi-api-versions.txt */
 #define HPI_LIB_VER  HPI_VERSION_CONSTRUCTOR(9, 0, 0)
@@ -1632,6 +1632,12 @@ u16 hpi_tuner_get_hd_radio_sdk_version(const struct hpi_hsubsys *ph_subsys,
 u16 hpi_tuner_get_hd_radio_signal_quality(const struct hpi_hsubsys *ph_subsys,
 	u32 h_control, u32 *pquality);
 
+u16 hpi_tuner_get_hd_radio_signal_blend(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, u32 *pblend);
+
+u16 hpi_tuner_set_hd_radio_signal_blend(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, const u32 blend);
+
 /****************************/
 /* PADs control             */
 /****************************/
diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c
index 839ecb2e4b64..12dab5e4892c 100644
--- a/sound/pci/asihpi/hpi6000.c
+++ b/sound/pci/asihpi/hpi6000.c
@@ -691,9 +691,6 @@ static short hpi6000_adapter_boot_load_dsp(struct hpi_adapter_obj *pao,
 	case 0x6200:
 		boot_load_family = HPI_ADAPTER_FAMILY_ASI(0x6200);
 		break;
-	case 0x8800:
-		boot_load_family = HPI_ADAPTER_FAMILY_ASI(0x8800);
-		break;
 	default:
 		return HPI6000_ERROR_UNHANDLED_SUBSYS_ID;
 	}
@@ -1775,7 +1772,6 @@ static void hw_message(struct hpi_adapter_obj *pao, struct hpi_message *phm,
 	u16 error = 0;
 	u16 dsp_index = 0;
 	u16 num_dsp = ((struct hpi_hw_obj *)pao->priv)->num_dsp;
-	hpios_dsplock_lock(pao);
 
 	if (num_dsp < 2)
 		dsp_index = 0;
@@ -1796,6 +1792,8 @@ static void hw_message(struct hpi_adapter_obj *pao, struct hpi_message *phm,
 			}
 		}
 	}
+
+	hpios_dsplock_lock(pao);
 	error = hpi6000_message_response_sequence(pao, dsp_index, phm, phr);
 
 	/* maybe an error response */
diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c
index 5e88c1fc2b9e..e89991ea3543 100644
--- a/sound/pci/asihpi/hpi6205.c
+++ b/sound/pci/asihpi/hpi6205.c
@@ -966,23 +966,16 @@ static void outstream_write(struct hpi_adapter_obj *pao,
 	status = &interface->outstream_host_buffer_status[phm->obj_index];
 
 	if (phw->flag_outstream_just_reset[phm->obj_index]) {
-		/* Format can only change after reset. Must tell DSP. */
-		u16 function = phm->function;
-		phw->flag_outstream_just_reset[phm->obj_index] = 0;
-		phm->function = HPI_OSTREAM_SET_FORMAT;
-		hw_message(pao, phm, phr);	/* send the format to the DSP */
-		phm->function = function;
-		if (phr->error)
-			return;
-	}
-#if 1
-	if (phw->flag_outstream_just_reset[phm->obj_index]) {
 		/* First OutStremWrite() call following reset will write data to the
-		   adapter's buffers, reducing delay before stream can start
+		   adapter's buffers, reducing delay before stream can start. The DSP
+		   takes care of setting the stream data format using format information
+		   embedded in phm.
 		 */
 		int partial_write = 0;
 		unsigned int original_size = 0;
 
+		phw->flag_outstream_just_reset[phm->obj_index] = 0;
+
 		/* Send the first buffer to the DSP the old way. */
 		/* Limit size of first transfer - */
 		/* expect that this will not usually be triggered. */
@@ -1012,7 +1005,6 @@ static void outstream_write(struct hpi_adapter_obj *pao,
 			original_size - HPI6205_SIZEOF_DATA;
 		phm->u.d.u.data.pb_data += HPI6205_SIZEOF_DATA;
 	}
-#endif
 
 	space_available = outstream_get_space_available(status);
 	if (space_available < (long)phm->u.d.u.data.data_size) {
@@ -1369,6 +1361,9 @@ static u16 adapter_boot_load_dsp(struct hpi_adapter_obj *pao,
 	case HPI_ADAPTER_FAMILY_ASI(0x6500):
 		firmware_id = HPI_ADAPTER_FAMILY_ASI(0x6600);
 		break;
+	case HPI_ADAPTER_FAMILY_ASI(0x8800):
+		firmware_id = HPI_ADAPTER_FAMILY_ASI(0x8900);
+		break;
 	}
 	boot_code_id[1] = firmware_id;
 
diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h
index f1cd6f1a0d44..fdd0ce02aa68 100644
--- a/sound/pci/asihpi/hpi_internal.h
+++ b/sound/pci/asihpi/hpi_internal.h
@@ -232,6 +232,8 @@ enum HPI_BUSES {
 #define HPI_TUNER_HDRADIO_SDK_VERSION   HPI_CTL_ATTR(TUNER, 13)
 /** HD Radio DSP firmware version. */
 #define HPI_TUNER_HDRADIO_DSP_VERSION   HPI_CTL_ATTR(TUNER, 14)
+/** HD Radio signal blend (force analog, or automatic). */
+#define HPI_TUNER_HDRADIO_BLEND         HPI_CTL_ATTR(TUNER, 15)
 
 /** \} */
 
@@ -478,8 +480,10 @@ Threshold is a -ve number in units of dB/100,
 
 /** First 2 hex digits define the adapter family */
 #define HPI_ADAPTER_FAMILY_MASK         0xff00
+#define HPI_MODULE_FAMILY_MASK          0xfff0
 
 #define HPI_ADAPTER_FAMILY_ASI(f)   (f & HPI_ADAPTER_FAMILY_MASK)
+#define HPI_MODULE_FAMILY_ASI(f)   (f & HPI_MODULE_FAMILY_MASK)
 #define HPI_ADAPTER_ASI(f)   (f)
 
 /******************************************* message types */
@@ -970,6 +974,7 @@ struct hpi_control_union_msg {
 				u32 mode;
 				u32 value;
 			} mode;
+			u32 blend;
 		} tuner;
 	} u;
 };
diff --git a/sound/pci/asihpi/hpicmn.c b/sound/pci/asihpi/hpicmn.c
index 565102cae4f8..fcd64539d9ef 100644
--- a/sound/pci/asihpi/hpicmn.c
+++ b/sound/pci/asihpi/hpicmn.c
@@ -347,20 +347,15 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache,
 			found = 0;
 		break;
 	case HPI_CONTROL_TUNER:
-		{
-			struct hpi_control_cache_single *pCT =
-				(struct hpi_control_cache_single *)pI;
-			if (phm->u.c.attribute == HPI_TUNER_FREQ)
-				phr->u.c.param1 = pCT->u.t.freq_ink_hz;
-			else if (phm->u.c.attribute == HPI_TUNER_BAND)
-				phr->u.c.param1 = pCT->u.t.band;
-			else if ((phm->u.c.attribute == HPI_TUNER_LEVEL)
-				&& (phm->u.c.param1 ==
-					HPI_TUNER_LEVEL_AVERAGE))
-				phr->u.c.param1 = pCT->u.t.level;
-			else
-				found = 0;
-		}
+		if (phm->u.c.attribute == HPI_TUNER_FREQ)
+			phr->u.c.param1 = pC->u.t.freq_ink_hz;
+		else if (phm->u.c.attribute == HPI_TUNER_BAND)
+			phr->u.c.param1 = pC->u.t.band;
+		else if ((phm->u.c.attribute == HPI_TUNER_LEVEL)
+			&& (phm->u.c.param1 == HPI_TUNER_LEVEL_AVERAGE))
+			phr->u.c.param1 = pC->u.t.level;
+		else
+			found = 0;
 		break;
 	case HPI_CONTROL_AESEBU_RECEIVER:
 		if (phm->u.c.attribute == HPI_AESEBURX_ERRORSTATUS)
@@ -503,6 +498,9 @@ void hpi_sync_control_cache(struct hpi_control_cache *p_cache,
 	struct hpi_control_cache_single *pC;
 	struct hpi_control_cache_info *pI;
 
+	if (phr->error)
+		return;
+
 	if (!find_control(phm, p_cache, &pI, &control_index))
 		return;
 
@@ -520,8 +518,6 @@ void hpi_sync_control_cache(struct hpi_control_cache *p_cache,
 		break;
 	case HPI_CONTROL_MULTIPLEXER:
 		/* mux does not return its setting on Set command. */
-		if (phr->error)
-			return;
 		if (phm->u.c.attribute == HPI_MULTIPLEXER_SOURCE) {
 			pC->u.x.source_node_type = (u16)phm->u.c.param1;
 			pC->u.x.source_node_index = (u16)phm->u.c.param2;
@@ -529,8 +525,6 @@ void hpi_sync_control_cache(struct hpi_control_cache *p_cache,
 		break;
 	case HPI_CONTROL_CHANNEL_MODE:
 		/* mode does not return its setting on Set command. */
-		if (phr->error)
-			return;
 		if (phm->u.c.attribute == HPI_CHANNEL_MODE_MODE)
 			pC->u.m.mode = (u16)phm->u.c.param1;
 		break;
@@ -545,20 +539,14 @@ void hpi_sync_control_cache(struct hpi_control_cache *p_cache,
 			pC->u.phantom_power.state = (u16)phm->u.c.param1;
 		break;
 	case HPI_CONTROL_AESEBU_TRANSMITTER:
-		if (phr->error)
-			return;
 		if (phm->u.c.attribute == HPI_AESEBUTX_FORMAT)
 			pC->u.aes3tx.format = phm->u.c.param1;
 		break;
 	case HPI_CONTROL_AESEBU_RECEIVER:
-		if (phr->error)
-			return;
 		if (phm->u.c.attribute == HPI_AESEBURX_FORMAT)
 			pC->u.aes3rx.source = phm->u.c.param1;
 		break;
 	case HPI_CONTROL_SAMPLECLOCK:
-		if (phr->error)
-			return;
 		if (phm->u.c.attribute == HPI_SAMPLECLOCK_SOURCE)
 			pC->u.clk.source = (u16)phm->u.c.param1;
 		else if (phm->u.c.attribute == HPI_SAMPLECLOCK_SOURCE_INDEX)
@@ -590,7 +578,7 @@ struct hpi_control_cache *hpi_alloc_control_cache(const u32
 
 void hpi_free_control_cache(struct hpi_control_cache *p_cache)
 {
-	if ((p_cache->init) && (p_cache->p_info)) {
+	if (p_cache->init) {
 		kfree(p_cache->p_info);
 		p_cache->p_info = NULL;
 		p_cache->init = 0;
diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c
index eda26b312324..298eef3e20e9 100644
--- a/sound/pci/asihpi/hpifunc.c
+++ b/sound/pci/asihpi/hpifunc.c
@@ -2946,6 +2946,20 @@ u16 hpi_tuner_get_hd_radio_signal_quality(const struct hpi_hsubsys *ph_subsys,
 		HPI_TUNER_HDRADIO_SIGNAL_QUALITY, 0, 0, pquality, NULL);
 }
 
+u16 hpi_tuner_get_hd_radio_signal_blend(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, u32 *pblend)
+{
+	return hpi_control_param_get(ph_subsys, h_control,
+		HPI_TUNER_HDRADIO_BLEND, 0, 0, pblend, NULL);
+}
+
+u16 hpi_tuner_set_hd_radio_signal_blend(const struct hpi_hsubsys *ph_subsys,
+	u32 h_control, const u32 blend)
+{
+	return hpi_control_param_set(ph_subsys, h_control,
+		HPI_TUNER_HDRADIO_BLEND, blend, 0);
+}
+
 u16 hpi_tuner_getRDS(const struct hpi_hsubsys *ph_subsys, u32 h_control,
 	char *p_data)
 {
@@ -3266,8 +3280,7 @@ u16 hpi_entity_find_next(struct hpi_entity *container_entity,
 
 void hpi_entity_free(struct hpi_entity *entity)
 {
-	if (entity != NULL)
-		kfree(entity);
+	kfree(entity);
 }
 
 static u16 hpi_entity_alloc_and_copy(struct hpi_entity *src,
diff --git a/sound/pci/asihpi/hpios.c b/sound/pci/asihpi/hpios.c
index de615cfdb950..742ee12a9e17 100644
--- a/sound/pci/asihpi/hpios.c
+++ b/sound/pci/asihpi/hpios.c
@@ -89,26 +89,3 @@ u16 hpios_locked_mem_free(struct consistent_dma_area *p_mem_area)
 void hpios_locked_mem_free_all(void)
 {
 }
-
-void __iomem *hpios_map_io(struct pci_dev *pci_dev, int idx,
-	unsigned int length)
-{
-	HPI_DEBUG_LOG(DEBUG, "mapping %d %s %08llx-%08llx %04llx len 0x%x\n",
-		idx, pci_dev->resource[idx].name,
-		(unsigned long long)pci_resource_start(pci_dev, idx),
-		(unsigned long long)pci_resource_end(pci_dev, idx),
-		(unsigned long long)pci_resource_flags(pci_dev, idx), length);
-
-	if (!(pci_resource_flags(pci_dev, idx) & IORESOURCE_MEM)) {
-		HPI_DEBUG_LOG(ERROR, "not an io memory resource\n");
-		return NULL;
-	}
-
-	if (length > pci_resource_len(pci_dev, idx)) {
-		HPI_DEBUG_LOG(ERROR, "resource too small for requested %d \n",
-			length);
-		return NULL;
-	}
-
-	return ioremap(pci_resource_start(pci_dev, idx), length);
-}
diff --git a/sound/pci/asihpi/hpios.h b/sound/pci/asihpi/hpios.h
index a62c3f1e5f09..370f39b43f85 100644
--- a/sound/pci/asihpi/hpios.h
+++ b/sound/pci/asihpi/hpios.h
@@ -166,13 +166,4 @@ struct hpi_adapter {
 	void __iomem *ap_remapped_mem_base[HPI_MAX_ADAPTER_MEM_SPACES];
 };
 
-static inline void hpios_unmap_io(void __iomem *addr,
-	unsigned long size)
-{
-	iounmap(addr);
-}
-
-void __iomem *hpios_map_io(struct pci_dev *pci_dev, int idx,
-	unsigned int length);
-
 #endif
diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c
index 67921f93a41e..c15002242d98 100644
--- a/sound/pci/aw2/aw2-alsa.c
+++ b/sound/pci/aw2/aw2-alsa.c
@@ -26,7 +26,7 @@
 #include <linux/slab.h>
 #include <linux/interrupt.h>
 #include <linux/delay.h>
-#include <asm/io.h>
+#include <linux/io.h>
 #include <sound/core.h>
 #include <sound/initval.h>
 #include <sound/pcm.h>
@@ -44,9 +44,6 @@ MODULE_LICENSE("GPL");
 /*********************************
  * DEFINES
  ********************************/
-#define PCI_VENDOR_ID_SAA7146		  0x1131
-#define PCI_DEVICE_ID_SAA7146		  0x7146
-
 #define CTL_ROUTE_ANALOG 0
 #define CTL_ROUTE_DIGITAL 1
 
@@ -165,7 +162,7 @@ module_param_array(enable, bool, NULL, 0444);
 MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard.");
 
 static DEFINE_PCI_DEVICE_TABLE(snd_aw2_ids) = {
-	{PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0,
+	{PCI_VENDOR_ID_PHILIPS, PCI_DEVICE_ID_PHILIPS_SAA7146, 0, 0,
 	 0, 0, 0},
 	{0}
 };
@@ -419,7 +416,7 @@ static int snd_aw2_pcm_playback_open(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
-	snd_printdd(KERN_DEBUG "aw2: Playback_open \n");
+	snd_printdd(KERN_DEBUG "aw2: Playback_open\n");
 	runtime->hw = snd_aw2_playback_hw;
 	return 0;
 }
@@ -435,7 +432,7 @@ static int snd_aw2_pcm_capture_open(struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 
-	snd_printdd(KERN_DEBUG "aw2: Capture_open \n");
+	snd_printdd(KERN_DEBUG "aw2: Capture_open\n");
 	runtime->hw = snd_aw2_capture_hw;
 	return 0;
 }
diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c
index 4b302d86f5f2..7a9401462c1c 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -35,6 +35,7 @@
 #include <linux/vmalloc.h>
 #include <linux/init.h>
 #include <linux/mutex.h>
+#include <linux/moduleparam.h>
 
 #include <sound/core.h>
 #include <sound/tlv.h>
@@ -50,6 +51,10 @@
 #define EMU10K1_CENTER_LFE_FROM_FRONT
 #endif
 
+static bool high_res_gpr_volume;
+module_param(high_res_gpr_volume, bool, 0444);
+MODULE_PARM_DESC(high_res_gpr_volume, "GPR mixer controls use 31-bit range.");
+
 /*
  *  Tables
  */ 
@@ -296,6 +301,7 @@ static const u32 db_table[101] = {
 
 /* EMU10k1/EMU10k2 DSP control db gain */
 static const DECLARE_TLV_DB_SCALE(snd_emu10k1_db_scale1, -4000, 40, 1);
+static const DECLARE_TLV_DB_LINEAR(snd_emu10k1_db_linear, TLV_DB_GAIN_MUTE, 0);
 
 static const u32 onoff_table[2] = {
 	0x00000000, 0x00000001
@@ -1072,10 +1078,17 @@ snd_emu10k1_init_mono_control(struct snd_emu10k1_fx8010_control_gpr *ctl,
 	strcpy(ctl->id.name, name);
 	ctl->vcount = ctl->count = 1;
 	ctl->gpr[0] = gpr + 0; ctl->value[0] = defval;
-	ctl->min = 0;
-	ctl->max = 100;
-	ctl->tlv = snd_emu10k1_db_scale1;
-	ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100;	
+	if (high_res_gpr_volume) {
+		ctl->min = 0;
+		ctl->max = 0x7fffffff;
+		ctl->tlv = snd_emu10k1_db_linear;
+		ctl->translation = EMU10K1_GPR_TRANSLATION_NONE;
+	} else {
+		ctl->min = 0;
+		ctl->max = 100;
+		ctl->tlv = snd_emu10k1_db_scale1;
+		ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100;
+	}
 }
 
 static void __devinit
@@ -1087,10 +1100,17 @@ snd_emu10k1_init_stereo_control(struct snd_emu10k1_fx8010_control_gpr *ctl,
 	ctl->vcount = ctl->count = 2;
 	ctl->gpr[0] = gpr + 0; ctl->value[0] = defval;
 	ctl->gpr[1] = gpr + 1; ctl->value[1] = defval;
-	ctl->min = 0;
-	ctl->max = 100;
-	ctl->tlv = snd_emu10k1_db_scale1;
-	ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100;
+	if (high_res_gpr_volume) {
+		ctl->min = 0;
+		ctl->max = 0x7fffffff;
+		ctl->tlv = snd_emu10k1_db_linear;
+		ctl->translation = EMU10K1_GPR_TRANSLATION_NONE;
+	} else {
+		ctl->min = 0;
+		ctl->max = 100;
+		ctl->tlv = snd_emu10k1_db_scale1;
+		ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100;
+	}
 }
 
 static void __devinit
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 170610e1d7da..1df25cf5ce38 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1097,6 +1097,7 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
 	struct azx *chip = dev_id;
 	struct azx_dev *azx_dev;
 	u32 status;
+	u8 sd_status;
 	int i, ok;
 
 	spin_lock(&chip->reg_lock);
@@ -1110,8 +1111,10 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
 	for (i = 0; i < chip->num_streams; i++) {
 		azx_dev = &chip->azx_dev[i];
 		if (status & azx_dev->sd_int_sta_mask) {
+			sd_status = azx_sd_readb(azx_dev, SD_STS);
 			azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK);
-			if (!azx_dev->substream || !azx_dev->running)
+			if (!azx_dev->substream || !azx_dev->running ||
+			    !(sd_status & SD_INT_COMPLETE))
 				continue;
 			/* check whether this IRQ is really acceptable */
 			ok = azx_position_ok(chip, azx_dev);
@@ -1910,11 +1913,11 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
 	if (WARN_ONCE(!azx_dev->period_bytes,
 		      "hda-intel: zero azx_dev->period_bytes"))
 		return -1; /* this shouldn't happen! */
-	if (wallclk <= azx_dev->period_wallclk &&
+	if (wallclk < (azx_dev->period_wallclk * 5) / 4 &&
 	    pos % azx_dev->period_bytes > azx_dev->period_bytes / 2)
 		/* NG - it's below the first next period boundary */
 		return bdl_pos_adj[chip->dev_index] ? 0 : -1;
-	azx_dev->start_wallclk = wallclk;
+	azx_dev->start_wallclk += wallclk;
 	return 1; /* OK, it's fine */
 }
 
@@ -2279,16 +2282,23 @@ static int azx_dev_free(struct snd_device *device)
  * white/black-listing for position_fix
  */
 static struct snd_pci_quirk position_fix_list[] __devinitdata = {
+	SND_PCI_QUIRK(0x1025, 0x009f, "Acer Aspire 5110", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB),
-	SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB),
+	SND_PCI_QUIRK(0x1849, 0x0888, "775Dual-VSTA", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x8086, 0x2503, "DG965OT AAD63733-203", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB),
 	{}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index e863649d31f5..2bf2cb5da956 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -2975,6 +2975,8 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
 	SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD),
+	SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD),
+	SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD),
 	SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
 	{}
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 53538b0f9991..fc767b6b4785 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -7025,6 +7025,14 @@ static struct hda_input_mux alc889A_mb31_capture_source = {
 	},
 };
 
+static struct hda_input_mux alc889A_imac91_capture_source = {
+	.num_items = 2,
+	.items = {
+		{ "Mic", 0x01 },
+		{ "Line", 0x2 }, /* Not sure! */
+	},
+};
+
 /*
  * 2ch mode
  */
@@ -7486,15 +7494,8 @@ static struct snd_kcontrol_new alc885_macmini3_mixer[] = {
 };
 
 static struct snd_kcontrol_new alc885_imac91_mixer[] = {
-	HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
-	HDA_BIND_MUTE   ("Line-Out Playback Switch", 0x0c, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE  ("Speaker Playback Switch", 0x14, 0x00, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
-	HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_MUTE  ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT),
-	HDA_CODEC_MUTE  ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
-	HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT),
+	HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
+	HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT),
 	{ } /* end */
 };
 
@@ -7995,61 +7996,56 @@ static struct hda_verb alc885_mbp3_init_verbs[] = {
 
 /* iMac 9,1 */
 static struct hda_verb alc885_imac91_init_verbs[] = {
-	/* Line-Out mixer: unmute input/output amp left and right (volume = 0) */
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	/* Rear mixer */
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-	/* HP Pin: output 0 (0x0c) */
+	/* Internal Speaker Pin (0x0c) */
+	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
+	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
+	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
+	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+	/* HP Pin: Rear */
 	{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
 	{0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
 	{0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
-	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-	/* Internal Speakers: output 0 (0x0d) */
-	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+	{0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC880_HP_EVENT | AC_USRSP_EN)},
+	/* Line in Rear */
+	{0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50},
 	{0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
-	/* Mic (rear) pin: input vref at 80% */
-	{0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-	{0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 	/* Front Mic pin: input vref at 80% */
 	{0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
 	{0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	/* Line In pin: use output 1 when in LineOut mode */
-	{0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-	{0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-	{0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
-
-	/* FIXME: use matrix-type input source selection */
-	/* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
-	/* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+	/* Rear mixer */
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* Line-Out mixer: unmute input/output amp left and right (volume = 0) */
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+	{0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+	/* 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
 	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
 	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
 	{0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	/* Input mixer2 */
+	/* 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
 	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
 	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
 	{0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	/* Input mixer3 */
+	/* 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
 	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
 	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
 	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
 	{0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-	/* ADC1: mute amp left and right */
+	/* 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
 	{0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
-	/* ADC2: mute amp left and right */
+	/* 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
 	{0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
-	/* ADC3: mute amp left and right */
+	/* 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
 	{0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
 	{0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
-
 	{ }
 };
 
@@ -8118,7 +8114,7 @@ static void alc885_imac91_setup(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 
 	spec->autocfg.hp_pins[0] = 0x14;
-	spec->autocfg.speaker_pins[0] = 0x15;
+	spec->autocfg.speaker_pins[0] = 0x18;
 	spec->autocfg.speaker_pins[1] = 0x1a;
 }
 
@@ -9480,6 +9476,10 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
 	SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_IMAC24),
 	SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_IMAC24),
 	SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3),
+	SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889A_MB31),
+	SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_ASUS_A7M),
+	SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC885_MBP3),
+	SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC885_MBA21),
 	SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31),
 	SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3),
 	SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24),
@@ -9627,14 +9627,14 @@ static struct alc_config_preset alc882_presets[] = {
 		.init_hook = alc885_imac24_init_hook,
 	},
 	[ALC885_IMAC91] = {
-		.mixers = { alc885_imac91_mixer, alc882_chmode_mixer },
+		.mixers = {alc885_imac91_mixer},
 		.init_verbs = { alc885_imac91_init_verbs,
 				alc880_gpio1_init_verbs },
 		.num_dacs = ARRAY_SIZE(alc882_dac_nids),
 		.dac_nids = alc882_dac_nids,
-		.channel_mode = alc885_mbp_4ch_modes,
-		.num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes),
-		.input_mux = &alc882_capture_source,
+		.channel_mode = alc885_mba21_ch_modes,
+		.num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
+		.input_mux = &alc889A_imac91_capture_source,
 		.dig_out_nid = ALC882_DIGOUT_NID,
 		.dig_in_nid = ALC882_DIGIN_NID,
 		.unsol_event = alc_automute_amp_unsol_event,
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index a0e06d82da1f..f1e7babd6920 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2078,12 +2078,12 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = {
 	SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2000,
 			   "Intel D965", STAC_D965_3ST),
 	/* Dell 3 stack systems */
-	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01f7, "Dell XPS M1730", STAC_DELL_3ST),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01dd, "Dell Dimension E520", STAC_DELL_3ST),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01ed, "Dell     ", STAC_DELL_3ST),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01f4, "Dell     ", STAC_DELL_3ST),
 	/* Dell 3 stack systems with verb table in BIOS */
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01f3, "Dell Inspiron 1420", STAC_DELL_BIOS),
+	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01f7, "Dell XPS M1730", STAC_DELL_BIOS),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x0227, "Dell Vostro 1400  ", STAC_DELL_BIOS),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x022e, "Dell     ", STAC_DELL_BIOS),
 	SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x022f, "Dell Inspiron 1525", STAC_DELL_BIOS),
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 8ae20208e7be..0221ca79b3ae 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -426,8 +426,8 @@ static const struct soc_enum wm8350_enum[] = {
 	SOC_ENUM_SINGLE(WM8350_INPUT_MIXER_VOLUME, 15, 2, wm8350_lr),
 };
 
-static DECLARE_TLV_DB_LINEAR(pre_amp_tlv, -1200, 3525);
-static DECLARE_TLV_DB_LINEAR(out_pga_tlv, -5700, 600);
+static DECLARE_TLV_DB_SCALE(pre_amp_tlv, -1200, 3525, 0);
+static DECLARE_TLV_DB_SCALE(out_pga_tlv, -5700, 600, 0);
 static DECLARE_TLV_DB_SCALE(dac_pcm_tlv, -7163, 36, 1);
 static DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -12700, 50, 1);
 static DECLARE_TLV_DB_SCALE(out_mix_tlv, -1500, 300, 1);
diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c
index 7f5d080536a0..8f294066b0ed 100644
--- a/sound/soc/codecs/wm8400.c
+++ b/sound/soc/codecs/wm8400.c
@@ -107,21 +107,21 @@ static void wm8400_codec_reset(struct snd_soc_codec *codec)
 	wm8400_reset_codec_reg_cache(wm8400->wm8400);
 }
 
-static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600);
+static const DECLARE_TLV_DB_SCALE(rec_mix_tlv, -1500, 600, 0);
 
-static const DECLARE_TLV_DB_LINEAR(in_pga_tlv, -1650, 3000);
+static const DECLARE_TLV_DB_SCALE(in_pga_tlv, -1650, 3000, 0);
 
-static const DECLARE_TLV_DB_LINEAR(out_mix_tlv, -2100, 0);
+static const DECLARE_TLV_DB_SCALE(out_mix_tlv, -2100, 0, 0);
 
-static const DECLARE_TLV_DB_LINEAR(out_pga_tlv, -7300, 600);
+static const DECLARE_TLV_DB_SCALE(out_pga_tlv, -7300, 600, 0);
 
-static const DECLARE_TLV_DB_LINEAR(out_omix_tlv, -600, 0);
+static const DECLARE_TLV_DB_SCALE(out_omix_tlv, -600, 0, 0);
 
-static const DECLARE_TLV_DB_LINEAR(out_dac_tlv, -7163, 0);
+static const DECLARE_TLV_DB_SCALE(out_dac_tlv, -7163, 0, 0);
 
-static const DECLARE_TLV_DB_LINEAR(in_adc_tlv, -7163, 1763);
+static const DECLARE_TLV_DB_SCALE(in_adc_tlv, -7163, 1763, 0);
 
-static const DECLARE_TLV_DB_LINEAR(out_sidetone_tlv, -3600, 0);
+static const DECLARE_TLV_DB_SCALE(out_sidetone_tlv, -3600, 0, 0);
 
 static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
         struct snd_ctl_elem_value *ucontrol)
@@ -440,7 +440,7 @@ static int outmixer_event (struct snd_soc_dapm_widget *w,
 /* INMIX dB values */
 static const unsigned int in_mix_tlv[] = {
 	TLV_DB_RANGE_HEAD(1),
-	0,7, TLV_DB_LINEAR_ITEM(-1200, 600),
+	0,7, TLV_DB_SCALE_ITEM(-1200, 600, 0),
 };
 
 /* Left In PGA Connections */
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 7b536d923ea9..c018772cc430 100644
--- a/sound/soc/codecs/wm8990.c
+++ b/sound/soc/codecs/wm8990.c
@@ -111,21 +111,21 @@ static const u16 wm8990_reg[] = {
 
 #define wm8990_reset(c) snd_soc_write(c, WM8990_RESET, 0)
 
-static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600);
+static const DECLARE_TLV_DB_SCALE(rec_mix_tlv, -1500, 600, 0);
 
-static const DECLARE_TLV_DB_LINEAR(in_pga_tlv, -1650, 3000);
+static const DECLARE_TLV_DB_SCALE(in_pga_tlv, -1650, 3000, 0);
 
-static const DECLARE_TLV_DB_LINEAR(out_mix_tlv, 0, -2100);
+static const DECLARE_TLV_DB_SCALE(out_mix_tlv, 0, -2100, 0);
 
-static const DECLARE_TLV_DB_LINEAR(out_pga_tlv, -7300, 600);
+static const DECLARE_TLV_DB_SCALE(out_pga_tlv, -7300, 600, 0);
 
-static const DECLARE_TLV_DB_LINEAR(out_omix_tlv, -600, 0);
+static const DECLARE_TLV_DB_SCALE(out_omix_tlv, -600, 0, 0);
 
-static const DECLARE_TLV_DB_LINEAR(out_dac_tlv, -7163, 0);
+static const DECLARE_TLV_DB_SCALE(out_dac_tlv, -7163, 0, 0);
 
-static const DECLARE_TLV_DB_LINEAR(in_adc_tlv, -7163, 1763);
+static const DECLARE_TLV_DB_SCALE(in_adc_tlv, -7163, 1763, 0);
 
-static const DECLARE_TLV_DB_LINEAR(out_sidetone_tlv, -3600, 0);
+static const DECLARE_TLV_DB_SCALE(out_sidetone_tlv, -3600, 0, 0);
 
 static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
 	struct snd_ctl_elem_value *ucontrol)
@@ -451,7 +451,7 @@ static int outmixer_event(struct snd_soc_dapm_widget *w,
 /* INMIX dB values */
 static const unsigned int in_mix_tlv[] = {
 	TLV_DB_RANGE_HEAD(1),
-	0, 7, TLV_DB_LINEAR_ITEM(-1200, 600),
+	0, 7, TLV_DB_SCALE_ITEM(-1200, 600, 0),
 };
 
 /* Left In PGA Connections */
diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c
index d639e55c5124..1d4e7164e80a 100644
--- a/sound/soc/fsl/mpc5200_dma.c
+++ b/sound/soc/fsl/mpc5200_dma.c
@@ -380,8 +380,8 @@ int mpc5200_audio_dma_create(struct of_device *op)
 	int ret;
 
 	/* Fetch the registers and IRQ of the PSC */
-	irq = irq_of_parse_and_map(op->node, 0);
-	if (of_address_to_resource(op->node, 0, &res)) {
+	irq = irq_of_parse_and_map(op->dev.of_node, 0);
+	if (of_address_to_resource(op->dev.of_node, 0, &res)) {
 		dev_err(&op->dev, "Missing reg property\n");
 		return -ENODEV;
 	}
@@ -399,7 +399,7 @@ int mpc5200_audio_dma_create(struct of_device *op)
 	}
 
 	/* Get the PSC ID */
-	prop = of_get_property(op->node, "cell-index", &size);
+	prop = of_get_property(op->dev.of_node, "cell-index", &size);
 	if (!prop || size < sizeof *prop) {
 		ret = -ENODEV;
 		goto out_free;
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
index 3dbc7f7cd7b9..e2ee220bfb7e 100644
--- a/sound/soc/fsl/mpc5200_psc_ac97.c
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -317,12 +317,12 @@ static struct of_device_id psc_ac97_match[] __devinitdata = {
 MODULE_DEVICE_TABLE(of, psc_ac97_match);
 
 static struct of_platform_driver psc_ac97_driver = {
-	.match_table = psc_ac97_match,
 	.probe = psc_ac97_of_probe,
 	.remove = __devexit_p(psc_ac97_of_remove),
 	.driver = {
 		.name = "mpc5200-psc-ac97",
 		.owner = THIS_MODULE,
+		.of_match_table = psc_ac97_match,
 	},
 };
 
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index ce8de90fb94a..4f455bd6851f 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -181,7 +181,7 @@ static int __devinit psc_i2s_of_probe(struct of_device *op,
 
 	/* Check for the codec handle.  If it is not present then we
 	 * are done */
-	if (!of_get_property(op->node, "codec-handle", NULL))
+	if (!of_get_property(op->dev.of_node, "codec-handle", NULL))
 		return 0;
 
 	/* Due to errata in the dma mode; need to line up enabling
@@ -220,12 +220,12 @@ static struct of_device_id psc_i2s_match[] __devinitdata = {
 MODULE_DEVICE_TABLE(of, psc_i2s_match);
 
 static struct of_platform_driver psc_i2s_driver = {
-	.match_table = psc_i2s_match,
 	.probe = psc_i2s_of_probe,
 	.remove = __devexit_p(psc_i2s_of_remove),
 	.driver = {
 		.name = "mpc5200-psc-i2s",
 		.owner = THIS_MODULE,
+		.of_match_table = psc_i2s_match,
 	},
 };
 
diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c
index 604a91fa31bc..3a501062c244 100644
--- a/sound/soc/fsl/mpc8610_hpcd.c
+++ b/sound/soc/fsl/mpc8610_hpcd.c
@@ -203,7 +203,7 @@ static struct snd_soc_ops mpc8610_hpcd_ops = {
 static int mpc8610_hpcd_probe(struct of_device *ofdev,
 	const struct of_device_id *match)
 {
-	struct device_node *np = ofdev->node;
+	struct device_node *np = ofdev->dev.of_node;
 	struct device_node *codec_np = NULL;
 	struct device_node *guts_np = NULL;
 	struct device_node *dma_np = NULL;
@@ -580,9 +580,11 @@ static struct of_device_id mpc8610_hpcd_match[] = {
 MODULE_DEVICE_TABLE(of, mpc8610_hpcd_match);
 
 static struct of_platform_driver mpc8610_hpcd_of_driver = {
-	.owner  	= THIS_MODULE,
-	.name   	= "mpc8610_hpcd",
-	.match_table    = mpc8610_hpcd_match,
+	.driver = {
+		.name = "mpc8610_hpcd",
+		.owner = THIS_MODULE,
+		.of_match_table = mpc8610_hpcd_match,
+	},
 	.probe  	= mpc8610_hpcd_probe,
 	.remove 	= mpc8610_hpcd_remove,
 };
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
index eba9b9d257a1..252defea93b5 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -13,9 +13,18 @@ config SND_MXC_SOC_SSI
 
 config SND_MXC_SOC_WM1133_EV1
 	tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted"
-	depends on SND_IMX_SOC && EXPERIMENTAL
+	depends on SND_IMX_SOC && MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL
 	select SND_SOC_WM8350
 	select SND_MXC_SOC_SSI
 	help
 	  Enable support for audio on the i.MX31ADS with the WM1133-EV1
 	  PMIC board with WM8835x fitted.
+
+config SND_SOC_PHYCORE_AC97
+	tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards"
+	depends on MACH_PCM043 || MACH_PCA100
+	select SND_MXC_SOC_SSI
+	select SND_SOC_WM9712
+	help
+	  Say Y if you want to add support for SoC audio on Phytec phyCORE
+	  and phyCARD boards in AC97 mode
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index 2b31ac673ea4..05f19c9284f4 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -73,7 +73,8 @@ static void snd_imx_dma_err_callback(int channel, void *data, int err)
 {
 	struct snd_pcm_substream *substream = data;
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
-	struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data;
+	struct imx_pcm_dma_params *dma_params = 
+		snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
 	int ret;
@@ -102,7 +103,7 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream)
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
 	int ret;
 
-	dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream);
+	dma_params = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 
 	iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH);
 	if (iprtd->dma < 0) {
@@ -212,7 +213,7 @@ static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream)
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
 	int err;
 
-	dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream);
+	dma_params = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 
 	iprtd->substream = substream;
 	iprtd->buf = (unsigned int *)substream->dma_buffer.area;
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index 1941a357e8c4..d256f5f313b5 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -328,38 +328,6 @@ static struct snd_soc_device spitz_snd_devdata = {
 	.codec_dev = &soc_codec_dev_wm8750,
 };
 
-/*
- * FIXME: This is a temporary bodge to avoid cross-tree merge issues.
- * New drivers should register the wm8750 I2C device in the machine
- * setup code (under arch/arm for ARM systems).
- */
-static int wm8750_i2c_register(void)
-{
-	struct i2c_board_info info;
-	struct i2c_adapter *adapter;
-	struct i2c_client *client;
-
-	memset(&info, 0, sizeof(struct i2c_board_info));
-	info.addr = 0x1b;
-	strlcpy(info.type, "wm8750", I2C_NAME_SIZE);
-
-	adapter = i2c_get_adapter(0);
-	if (!adapter) {
-		printk(KERN_ERR "can't get i2c adapter 0\n");
-		return -ENODEV;
-	}
-
-	client = i2c_new_device(adapter, &info);
-	i2c_put_adapter(adapter);
-	if (!client) {
-		printk(KERN_ERR "can't add i2c device at 0x%x\n",
-		(unsigned int)info.addr);
-		return -ENODEV;
-	}
-
-	return 0;
-}
-
 static struct platform_device *spitz_snd_device;
 
 static int __init spitz_init(void)
@@ -369,10 +337,6 @@ static int __init spitz_init(void)
 	if (!(machine_is_spitz() || machine_is_borzoi() || machine_is_akita()))
 		return -ENODEV;
 
-	ret = wm8750_i2c_setup();
-	if (ret != 0)
-		return ret;
-
 	spitz_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!spitz_snd_device)
 		return -ENOMEM;
diff --git a/sound/soc/sh/siu.h b/sound/soc/sh/siu.h
index c0bfab8fed3d..492b1cae24cc 100644
--- a/sound/soc/sh/siu.h
+++ b/sound/soc/sh/siu.h
@@ -71,8 +71,7 @@ struct siu_firmware {
 #include <linux/dmaengine.h>
 #include <linux/interrupt.h>
 #include <linux/io.h>
-
-#include <asm/dmaengine.h>
+#include <linux/sh_dma.h>
 
 #include <sound/core.h>
 #include <sound/pcm.h>
diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c
index d86ee1bfc03a..eeed5edd722b 100644
--- a/sound/soc/sh/siu_dai.c
+++ b/sound/soc/sh/siu_dai.c
@@ -588,6 +588,8 @@ static int siu_dai_prepare(struct snd_pcm_substream *substream,
 		ret = siu_dai_spbstart(port_info);
 		if (ret < 0)
 			goto fail;
+	} else {
+		ret = 0;
 	}
 
 	port_info->play_cap |= self;
diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c
index 8f85719212f9..36170be15aa7 100644
--- a/sound/soc/sh/siu_pcm.c
+++ b/sound/soc/sh/siu_pcm.c
@@ -31,7 +31,6 @@
 #include <sound/pcm_params.h>
 #include <sound/soc-dai.h>
 
-#include <asm/dmaengine.h>
 #include <asm/siu.h>
 
 #include "siu.h"
@@ -358,13 +357,13 @@ static int siu_pcm_open(struct snd_pcm_substream *ss)
 	if (ss->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		siu_stream = &port_info->playback;
 		param = &siu_stream->param;
-		param->slave_id = port ? SHDMA_SLAVE_SIUB_TX :
-			SHDMA_SLAVE_SIUA_TX;
+		param->slave_id = port ? pdata->dma_slave_tx_b :
+			pdata->dma_slave_tx_a;
 	} else {
 		siu_stream = &port_info->capture;
 		param = &siu_stream->param;
-		param->slave_id = port ? SHDMA_SLAVE_SIUB_RX :
-			SHDMA_SLAVE_SIUA_RX;
+		param->slave_id = port ? pdata->dma_slave_rx_b :
+			pdata->dma_slave_rx_a;
 	}
 
 	param->dma_dev = pdata->dma_dev;
diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c
index 49cc7ea9a518..0e3452303ea6 100644
--- a/sound/soc/txx9/txx9aclc.c
+++ b/sound/soc/txx9/txx9aclc.c
@@ -160,7 +160,7 @@ static void txx9aclc_dma_tasklet(unsigned long data)
 		void __iomem *base = drvdata->base;
 
 		spin_unlock_irqrestore(&dmadata->dma_lock, flags);
-		chan->device->device_terminate_all(chan);
+		chan->device->device_control(chan, DMA_TERMINATE_ALL, 0);
 		/* first time */
 		for (i = 0; i < NR_DMA_CHAIN; i++) {
 			desc = txx9aclc_dma_submit(dmadata,
@@ -268,7 +268,7 @@ static int txx9aclc_pcm_close(struct snd_pcm_substream *substream)
 	struct dma_chan *chan = dmadata->dma_chan;
 
 	dmadata->frag_count = -1;
-	chan->device->device_terminate_all(chan);
+	chan->device->device_control(chan, DMA_TERMINATE_ALL, 0);
 	return 0;
 }
 
@@ -397,7 +397,8 @@ static int txx9aclc_pcm_remove(struct platform_device *pdev)
 		struct dma_chan *chan = dmadata->dma_chan;
 		if (chan) {
 			dmadata->frag_count = -1;
-			chan->device->device_terminate_all(chan);
+			chan->device->device_control(chan,
+						     DMA_TERMINATE_ALL, 0);
 			dma_release_channel(chan);
 		}
 		dev->dmadata[i].dma_chan = NULL;
diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c
index 574af56ba8a6..71221fd20944 100644
--- a/sound/sparc/amd7930.c
+++ b/sound/sparc/amd7930.c
@@ -1065,8 +1065,11 @@ static const struct of_device_id amd7930_match[] = {
 };
 
 static struct of_platform_driver amd7930_sbus_driver = {
-	.name		= "audio",
-	.match_table	= amd7930_match,
+	.driver = {
+		.name = "audio",
+		.owner = THIS_MODULE,
+		.of_match_table = amd7930_match,
+	},
 	.probe		= amd7930_sbus_probe,
 };
 
diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c
index 7dcc06512e86..fb4c6f2f29e5 100644
--- a/sound/sparc/cs4231.c
+++ b/sound/sparc/cs4231.c
@@ -2075,12 +2075,12 @@ static int __devinit cs4231_ebus_probe(struct of_device *op, const struct of_dev
 static int __devinit cs4231_probe(struct of_device *op, const struct of_device_id *match)
 {
 #ifdef EBUS_SUPPORT
-	if (!strcmp(op->node->parent->name, "ebus"))
+	if (!strcmp(op->dev.of_node->parent->name, "ebus"))
 		return cs4231_ebus_probe(op, match);
 #endif
 #ifdef SBUS_SUPPORT
-	if (!strcmp(op->node->parent->name, "sbus") ||
-	    !strcmp(op->node->parent->name, "sbi"))
+	if (!strcmp(op->dev.of_node->parent->name, "sbus") ||
+	    !strcmp(op->dev.of_node->parent->name, "sbi"))
 		return cs4231_sbus_probe(op, match);
 #endif
 	return -ENODEV;
@@ -2109,8 +2109,11 @@ static const struct of_device_id cs4231_match[] = {
 MODULE_DEVICE_TABLE(of, cs4231_match);
 
 static struct of_platform_driver cs4231_driver = {
-	.name		= "audio",
-	.match_table	= cs4231_match,
+	.driver = {
+		.name = "audio",
+		.owner = THIS_MODULE,
+		.of_match_table = cs4231_match,
+	},
 	.probe		= cs4231_probe,
 	.remove		= __devexit_p(cs4231_remove),
 };
diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c
index 2eab6ce48852..1557bf132e73 100644
--- a/sound/sparc/dbri.c
+++ b/sound/sparc/dbri.c
@@ -2651,7 +2651,7 @@ static int __devinit dbri_probe(struct of_device *op, const struct of_device_id
 
 	printk(KERN_INFO "audio%d at %p (irq %d) is DBRI(%c)+CS4215(%d)\n",
 	       dev, dbri->regs,
-	       dbri->irq, op->node->name[9], dbri->mm.version);
+	       dbri->irq, op->dev.of_node->name[9], dbri->mm.version);
 	dev++;
 
 	return 0;
@@ -2687,8 +2687,11 @@ static const struct of_device_id dbri_match[] = {
 MODULE_DEVICE_TABLE(of, dbri_match);
 
 static struct of_platform_driver dbri_sbus_driver = {
-	.name		= "dbri",
-	.match_table	= dbri_match,
+	.driver = {
+		.name = "dbri",
+		.owner = THIS_MODULE,
+		.of_match_table = dbri_match,
+	},
 	.probe		= dbri_probe,
 	.remove		= __devexit_p(dbri_remove),
 };
diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c
index 4c7b051f9d17..1bc56b2b94e2 100644
--- a/sound/spi/at73c213.c
+++ b/sound/spi/at73c213.c
@@ -69,7 +69,6 @@ struct snd_at73c213 {
 	int				irq;
 	int				period;
 	unsigned long			bitrate;
-	struct clk			*bitclk;
 	struct ssc_device		*ssc;
 	struct spi_device		*spi;
 	u8				spi_wbuffer[2];
diff --git a/sound/usb/Makefile b/sound/usb/Makefile
index e7ac7f493a8f..1e362bf8834f 100644
--- a/sound/usb/Makefile
+++ b/sound/usb/Makefile
@@ -11,7 +11,8 @@ snd-usb-audio-objs := 	card.o \
 			endpoint.o \
 			urb.o \
 			pcm.o \
-			helper.o
+			helper.o \
+			clock.o
 
 snd-usbmidi-lib-objs := midi.o
 
diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c
index 36ed703a7416..91c804cd2782 100644
--- a/sound/usb/caiaq/control.c
+++ b/sound/usb/caiaq/control.c
@@ -42,21 +42,12 @@ static int control_info(struct snd_kcontrol *kcontrol,
 
 	switch (dev->chip.usb_id) {
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ):
-		if (pos == 0) {
-			/* current input mode of A8DJ */
-			uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
-			uinfo->value.integer.min = 0;
-			uinfo->value.integer.max = 2;
-			return 0;
-		}
-		break;
-
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ):
 		if (pos == 0) {
-			/* current input mode of A4DJ */
+			/* current input mode of A8DJ and A4DJ */
 			uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
 			uinfo->value.integer.min = 0;
-			uinfo->value.integer.max = 1;
+			uinfo->value.integer.max = 2;
 			return 0;
 		}
 		break;
@@ -86,14 +77,6 @@ static int control_get(struct snd_kcontrol *kcontrol,
 	struct snd_usb_caiaqdev *dev = caiaqdev(chip->card);
 	int pos = kcontrol->private_value;
 
-	if (dev->chip.usb_id ==
-		USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ)) {
-		/* A4DJ has only one control */
-		/* do not expose hardware input mode 0 */
-		ucontrol->value.integer.value[0] = dev->control_state[0] - 1;
-		return 0;
-	}
-
 	if (pos & CNT_INTVAL)
 		ucontrol->value.integer.value[0]
 			= dev->control_state[pos & ~CNT_INTVAL];
@@ -112,20 +95,9 @@ static int control_put(struct snd_kcontrol *kcontrol,
 	int pos = kcontrol->private_value;
 	unsigned char cmd = EP1_CMD_WRITE_IO;
 
-	switch (dev->chip.usb_id) {
-	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): {
-		/* A4DJ has only one control */
-		/* do not expose hardware input mode 0 */
-		dev->control_state[0] = ucontrol->value.integer.value[0] + 1;
-		snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO,
-				dev->control_state, sizeof(dev->control_state));
-		return 1;
-	}
-
-	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1):
+	if (dev->chip.usb_id ==
+		USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1))
 		cmd = EP1_CMD_DIMM_LEDS;
-		break;
-	}
 
 	if (pos & CNT_INTVAL) {
 		dev->control_state[pos & ~CNT_INTVAL]
diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c
index 805271827675..cdfb856bddd2 100644
--- a/sound/usb/caiaq/device.c
+++ b/sound/usb/caiaq/device.c
@@ -36,7 +36,7 @@
 #include "input.h"
 
 MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>");
-MODULE_DESCRIPTION("caiaq USB audio, version 1.3.20");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.21");
 MODULE_LICENSE("GPL");
 MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
 			 "{Native Instruments, RigKontrol3},"
@@ -320,12 +320,6 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev)
 		}
 
 		break;
-	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ):
-		/* Audio 4 DJ - default input mode to phono */
-		dev->control_state[0] = 2;
-		snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO,
-			dev->control_state, 1);
-		break;
 	}
 
 	if (dev->spec.num_analog_audio_out +
diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c
index 8bbfbfd4c658..dcb620796d9e 100644
--- a/sound/usb/caiaq/input.c
+++ b/sound/usb/caiaq/input.c
@@ -171,7 +171,7 @@ static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *dev,
 		input_report_abs(input_dev, ABS_HAT0Y, (buf[4] << 8)  | buf[5]);
 		input_report_abs(input_dev, ABS_HAT1X, (buf[12] << 8) | buf[13]);
 		input_report_abs(input_dev, ABS_HAT1Y, (buf[2] << 8)  | buf[3]);
-		input_report_abs(input_dev, ABS_HAT2X, (buf[15] << 8) | buf[15]);
+		input_report_abs(input_dev, ABS_HAT2X, (buf[14] << 8) | buf[15]);
 		input_report_abs(input_dev, ABS_HAT2Y, (buf[0] << 8)  | buf[1]);
 		input_report_abs(input_dev, ABS_HAT3X, (buf[10] << 8) | buf[11]);
 		input_report_abs(input_dev, ABS_HAT3Y, (buf[6] << 8)  | buf[7]);
diff --git a/sound/usb/card.c b/sound/usb/card.c
index da1346bd4856..7a8ac1d81be7 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -236,7 +236,6 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
 	}
 
 	case UAC_VERSION_2: {
-		struct uac_clock_source_descriptor *cs;
 		struct usb_interface_assoc_descriptor *assoc =
 			usb_ifnum_to_if(dev, ctrlif)->intf_assoc;
 
@@ -245,21 +244,6 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
 			return -EINVAL;
 		}
 
-		/* FIXME: for now, we expect there is at least one clock source
-		 * descriptor and we always take the first one.
-		 * We should properly support devices with multiple clock sources,
-		 * clock selectors and sample rate conversion units. */
-
-		cs = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen,
-						NULL, UAC2_CLOCK_SOURCE);
-
-		if (!cs) {
-			snd_printk(KERN_ERR "CLOCK_SOURCE descriptor not found\n");
-			return -EINVAL;
-		}
-
-		chip->clock_id = cs->bClockID;
-
 		for (i = 0; i < assoc->bInterfaceCount; i++) {
 			int intf = assoc->bFirstInterface + i;
 
@@ -481,6 +465,8 @@ static void *snd_usb_audio_probe(struct usb_device *dev,
 			goto __error;
 	}
 
+	chip->ctrl_intf = alts;
+
 	if (err > 0) {
 		/* create normal USB audio interfaces */
 		if (snd_usb_create_streams(chip, ifnum) < 0 ||
diff --git a/sound/usb/card.h b/sound/usb/card.h
index ed92420c1095..1febf2f23754 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -25,6 +25,7 @@ struct audioformat {
 	unsigned int rate_min, rate_max;	/* min/max rates */
 	unsigned int nr_rates;		/* number of rate table entries */
 	unsigned int *rate_table;	/* rate table */
+	unsigned char clock;		/* associated clock */
 };
 
 struct snd_usb_substream;
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
new file mode 100644
index 000000000000..b7aadd614c70
--- /dev/null
+++ b/sound/usb/clock.c
@@ -0,0 +1,311 @@
+/*
+ *   Clock domain and sample rate management functions
+ *
+ *   This program is free software; you can redistribute it and/or modify
+ *   it under the terms of the GNU General Public License as published by
+ *   the Free Software Foundation; either version 2 of the License, or
+ *   (at your option) any later version.
+ *
+ *   This program is distributed in the hope that it will be useful,
+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *   GNU General Public License for more details.
+ *
+ *   You should have received a copy of the GNU General Public License
+ *   along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+#include <linux/bitops.h>
+#include <linux/init.h>
+#include <linux/list.h>
+#include <linux/slab.h>
+#include <linux/string.h>
+#include <linux/usb.h>
+#include <linux/moduleparam.h>
+#include <linux/mutex.h>
+#include <linux/usb/audio.h>
+#include <linux/usb/audio-v2.h>
+
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+
+#include "usbaudio.h"
+#include "card.h"
+#include "midi.h"
+#include "mixer.h"
+#include "proc.h"
+#include "quirks.h"
+#include "endpoint.h"
+#include "helper.h"
+#include "debug.h"
+#include "pcm.h"
+#include "urb.h"
+#include "format.h"
+
+static struct uac_clock_source_descriptor *
+	snd_usb_find_clock_source(struct usb_host_interface *ctrl_iface,
+				  int clock_id)
+{
+	struct uac_clock_source_descriptor *cs = NULL;
+
+	while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra,
+					     ctrl_iface->extralen,
+					     cs, UAC2_CLOCK_SOURCE))) {
+		if (cs->bClockID == clock_id)
+			return cs;
+	}
+
+	return NULL;
+}
+
+static struct uac_clock_selector_descriptor *
+	snd_usb_find_clock_selector(struct usb_host_interface *ctrl_iface,
+				    int clock_id)
+{
+	struct uac_clock_selector_descriptor *cs = NULL;
+
+	while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra,
+					     ctrl_iface->extralen,
+					     cs, UAC2_CLOCK_SELECTOR))) {
+		if (cs->bClockID == clock_id)
+			return cs;
+	}
+
+	return NULL;
+}
+
+static struct uac_clock_multiplier_descriptor *
+	snd_usb_find_clock_multiplier(struct usb_host_interface *ctrl_iface,
+				      int clock_id)
+{
+	struct uac_clock_multiplier_descriptor *cs = NULL;
+
+	while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra,
+					     ctrl_iface->extralen,
+					     cs, UAC2_CLOCK_MULTIPLIER))) {
+		if (cs->bClockID == clock_id)
+			return cs;
+	}
+
+	return NULL;
+}
+
+static int uac_clock_selector_get_val(struct snd_usb_audio *chip, int selector_id)
+{
+	unsigned char buf;
+	int ret;
+
+	ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0),
+			      UAC2_CS_CUR,
+			      USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
+			      UAC2_CX_CLOCK_SELECTOR << 8, selector_id << 8,
+			      &buf, sizeof(buf), 1000);
+
+	if (ret < 0)
+		return ret;
+
+	return buf;
+}
+
+static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id)
+{
+	int err;
+	unsigned char data;
+	struct usb_device *dev = chip->dev;
+
+	err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
+			      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+			      UAC2_CS_CONTROL_CLOCK_VALID << 8, source_id << 8,
+			      &data, sizeof(data), 1000);
+
+	if (err < 0) {
+		snd_printk(KERN_WARNING "%s(): cannot get clock validity for id %d\n",
+			   __func__, source_id);
+		return err;
+	}
+
+	return !!data;
+}
+
+/* Try to find the clock source ID of a given clock entity */
+
+static int __uac_clock_find_source(struct snd_usb_audio *chip,
+				   struct usb_host_interface *host_iface,
+				   int entity_id, unsigned long *visited)
+{
+	struct uac_clock_source_descriptor *source;
+	struct uac_clock_selector_descriptor *selector;
+	struct uac_clock_multiplier_descriptor *multiplier;
+
+	entity_id &= 0xff;
+
+	if (test_and_set_bit(entity_id, visited)) {
+		snd_printk(KERN_WARNING
+			"%s(): recursive clock topology detected, id %d.\n",
+			__func__, entity_id);
+		return -EINVAL;
+	}
+
+	/* first, see if the ID we're looking for is a clock source already */
+	source = snd_usb_find_clock_source(host_iface, entity_id);
+	if (source)
+		return source->bClockID;
+
+	selector = snd_usb_find_clock_selector(host_iface, entity_id);
+	if (selector) {
+		int ret;
+
+		/* the entity ID we are looking for is a selector.
+		 * find out what it currently selects */
+		ret = uac_clock_selector_get_val(chip, selector->bClockID);
+		if (ret < 0)
+			return ret;
+
+		if (ret > selector->bNrInPins || ret < 1) {
+			printk(KERN_ERR
+				"%s(): selector reported illegal value, id %d, ret %d\n",
+				__func__, selector->bClockID, ret);
+
+			return -EINVAL;
+		}
+
+		return __uac_clock_find_source(chip, host_iface,
+					       selector->baCSourceID[ret-1],
+					       visited);
+	}
+
+	/* FIXME: multipliers only act as pass-thru element for now */
+	multiplier = snd_usb_find_clock_multiplier(host_iface, entity_id);
+	if (multiplier)
+		return __uac_clock_find_source(chip, host_iface,
+					       multiplier->bCSourceID, visited);
+
+	return -EINVAL;
+}
+
+int snd_usb_clock_find_source(struct snd_usb_audio *chip,
+			      struct usb_host_interface *host_iface,
+			      int entity_id)
+{
+	DECLARE_BITMAP(visited, 256);
+	memset(visited, 0, sizeof(visited));
+	return __uac_clock_find_source(chip, host_iface, entity_id, visited);
+}
+
+static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface,
+			      struct usb_host_interface *alts,
+			      struct audioformat *fmt, int rate)
+{
+	struct usb_device *dev = chip->dev;
+	unsigned int ep;
+	unsigned char data[3];
+	int err, crate;
+
+	ep = get_endpoint(alts, 0)->bEndpointAddress;
+
+	/* if endpoint doesn't have sampling rate control, bail out */
+	if (!(fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE)) {
+		snd_printk(KERN_WARNING "%d:%d:%d: endpoint lacks sample rate attribute bit, cannot set.\n",
+				   dev->devnum, iface, fmt->altsetting);
+		return 0;
+	}
+
+	data[0] = rate;
+	data[1] = rate >> 8;
+	data[2] = rate >> 16;
+	if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR,
+				   USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT,
+				   UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep,
+				   data, sizeof(data), 1000)) < 0) {
+		snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n",
+			   dev->devnum, iface, fmt->altsetting, rate, ep);
+		return err;
+	}
+
+	if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR,
+				   USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN,
+				   UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep,
+				   data, sizeof(data), 1000)) < 0) {
+		snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n",
+			   dev->devnum, iface, fmt->altsetting, ep);
+		return 0; /* some devices don't support reading */
+	}
+
+	crate = data[0] | (data[1] << 8) | (data[2] << 16);
+	if (crate != rate) {
+		snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate);
+		// runtime->rate = crate;
+	}
+
+	return 0;
+}
+
+static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
+			      struct usb_host_interface *alts,
+			      struct audioformat *fmt, int rate)
+{
+	struct usb_device *dev = chip->dev;
+	unsigned char data[4];
+	int err, crate;
+	int clock = snd_usb_clock_find_source(chip, chip->ctrl_intf, fmt->clock);
+
+	if (clock < 0)
+		return clock;
+
+	if (!uac_clock_source_is_valid(chip, clock)) {
+		snd_printk(KERN_ERR "%d:%d:%d: clock source %d is not valid, cannot use\n",
+			   dev->devnum, iface, fmt->altsetting, clock);
+		return -ENXIO;
+	}
+
+	data[0] = rate;
+	data[1] = rate >> 8;
+	data[2] = rate >> 16;
+	data[3] = rate >> 24;
+	if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR,
+				   USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT,
+				   UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8,
+				   data, sizeof(data), 1000)) < 0) {
+		snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2)\n",
+			   dev->devnum, iface, fmt->altsetting, rate);
+		return err;
+	}
+
+	if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
+				   USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+				   UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8,
+				   data, sizeof(data), 1000)) < 0) {
+		snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
+			   dev->devnum, iface, fmt->altsetting);
+		return err;
+	}
+
+	crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
+	if (crate != rate)
+		snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate);
+
+	return 0;
+}
+
+int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface,
+			     struct usb_host_interface *alts,
+			     struct audioformat *fmt, int rate)
+{
+	struct usb_interface_descriptor *altsd = get_iface_desc(alts);
+
+	switch (altsd->bInterfaceProtocol) {
+	case UAC_VERSION_1:
+		return set_sample_rate_v1(chip, iface, alts, fmt, rate);
+
+	case UAC_VERSION_2:
+		return set_sample_rate_v2(chip, iface, alts, fmt, rate);
+	}
+
+	return -EINVAL;
+}
+
diff --git a/sound/usb/clock.h b/sound/usb/clock.h
new file mode 100644
index 000000000000..beb253684e2d
--- /dev/null
+++ b/sound/usb/clock.h
@@ -0,0 +1,12 @@
+#ifndef __USBAUDIO_CLOCK_H
+#define __USBAUDIO_CLOCK_H
+
+int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface,
+			     struct usb_host_interface *alts,
+			     struct audioformat *fmt, int rate);
+
+int snd_usb_clock_find_source(struct snd_usb_audio *chip,
+			      struct usb_host_interface *host_iface,
+			      int entity_id);
+
+#endif /* __USBAUDIO_CLOCK_H */
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index ef07a6d0dd5f..9593b91452b9 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -149,6 +149,79 @@ int snd_usb_add_audio_endpoint(struct snd_usb_audio *chip, int stream, struct au
 	return 0;
 }
 
+static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip,
+					 struct usb_host_interface *alts,
+					 int protocol, int iface_no)
+{
+	/* parsed with a v1 header here. that's ok as we only look at the
+	 * header first which is the same for both versions */
+	struct uac_iso_endpoint_descriptor *csep;
+	struct usb_interface_descriptor *altsd = get_iface_desc(alts);
+	int attributes = 0;
+
+	csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT);
+
+	/* Creamware Noah has this descriptor after the 2nd endpoint */
+	if (!csep && altsd->bNumEndpoints >= 2)
+		csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT);
+
+	if (!csep || csep->bLength < 7 ||
+	    csep->bDescriptorSubtype != UAC_EP_GENERAL) {
+		snd_printk(KERN_WARNING "%d:%u:%d : no or invalid"
+			   " class specific endpoint descriptor\n",
+			   chip->dev->devnum, iface_no,
+			   altsd->bAlternateSetting);
+		return 0;
+	}
+
+	if (protocol == UAC_VERSION_1) {
+		attributes = csep->bmAttributes;
+	} else {
+		struct uac2_iso_endpoint_descriptor *csep2 =
+			(struct uac2_iso_endpoint_descriptor *) csep;
+
+		attributes = csep->bmAttributes & UAC_EP_CS_ATTR_FILL_MAX;
+
+		/* emulate the endpoint attributes of a v1 device */
+		if (csep2->bmControls & UAC2_CONTROL_PITCH)
+			attributes |= UAC_EP_CS_ATTR_PITCH_CONTROL;
+	}
+
+	return attributes;
+}
+
+static struct uac2_input_terminal_descriptor *
+	snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface,
+					       int terminal_id)
+{
+	struct uac2_input_terminal_descriptor *term = NULL;
+
+	while ((term = snd_usb_find_csint_desc(ctrl_iface->extra,
+					       ctrl_iface->extralen,
+					       term, UAC_INPUT_TERMINAL))) {
+		if (term->bTerminalID == terminal_id)
+			return term;
+	}
+
+	return NULL;
+}
+
+static struct uac2_output_terminal_descriptor *
+	snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface,
+						int terminal_id)
+{
+	struct uac2_output_terminal_descriptor *term = NULL;
+
+	while ((term = snd_usb_find_csint_desc(ctrl_iface->extra,
+					       ctrl_iface->extralen,
+					       term, UAC_OUTPUT_TERMINAL))) {
+		if (term->bTerminalID == terminal_id)
+			return term;
+	}
+
+	return NULL;
+}
+
 int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 {
 	struct usb_device *dev;
@@ -158,8 +231,8 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 	int i, altno, err, stream;
 	int format = 0, num_channels = 0;
 	struct audioformat *fp = NULL;
-	unsigned char *fmt, *csep;
-	int num, protocol;
+	int num, protocol, clock = 0;
+	struct uac_format_type_i_continuous_descriptor *fmt;
 
 	dev = chip->dev;
 
@@ -222,6 +295,8 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 		}
 
 		case UAC_VERSION_2: {
+			struct uac2_input_terminal_descriptor *input_term;
+			struct uac2_output_terminal_descriptor *output_term;
 			struct uac_as_header_descriptor_v2 *as =
 				snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL);
 
@@ -240,7 +315,25 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 			num_channels = as->bNrChannels;
 			format = le32_to_cpu(as->bmFormats);
 
-			break;
+			/* lookup the terminal associated to this interface
+			 * to extract the clock */
+			input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
+									    as->bTerminalLink);
+			if (input_term) {
+				clock = input_term->bCSourceID;
+				break;
+			}
+
+			output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf,
+									      as->bTerminalLink);
+			if (output_term) {
+				clock = output_term->bCSourceID;
+				break;
+			}
+
+			snd_printk(KERN_ERR "%d:%u:%d : bogus bTerminalLink %d\n",
+				   dev->devnum, iface_no, altno, as->bTerminalLink);
+			continue;
 		}
 
 		default:
@@ -256,8 +349,8 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 				   dev->devnum, iface_no, altno);
 			continue;
 		}
-		if (((protocol == UAC_VERSION_1) && (fmt[0] < 8)) ||
-		    ((protocol == UAC_VERSION_2) && (fmt[0] != 6))) {
+		if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) ||
+		    ((protocol == UAC_VERSION_2) && (fmt->bLength != 6))) {
 			snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n",
 				   dev->devnum, iface_no, altno);
 			continue;
@@ -268,7 +361,9 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 		 * with the previous one, except for a larger packet size, but
 		 * is actually a mislabeled two-channel setting; ignore it.
 		 */
-		if (fmt[4] == 1 && fmt[5] == 2 && altno == 2 && num == 3 &&
+		if (fmt->bNrChannels == 1 &&
+		    fmt->bSubframeSize == 2 &&
+		    altno == 2 && num == 3 &&
 		    fp && fp->altsetting == 1 && fp->channels == 1 &&
 		    fp->formats == SNDRV_PCM_FMTBIT_S16_LE &&
 		    protocol == UAC_VERSION_1 &&
@@ -276,17 +371,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 							fp->maxpacksize * 2)
 			continue;
 
-		csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT);
-		/* Creamware Noah has this descriptor after the 2nd endpoint */
-		if (!csep && altsd->bNumEndpoints >= 2)
-			csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT);
-		if (!csep || csep[0] < 7 || csep[2] != UAC_EP_GENERAL) {
-			snd_printk(KERN_WARNING "%d:%u:%d : no or invalid"
-				   " class specific endpoint descriptor\n",
-				   dev->devnum, iface_no, altno);
-			csep = NULL;
-		}
-
 		fp = kzalloc(sizeof(*fp), GFP_KERNEL);
 		if (! fp) {
 			snd_printk(KERN_ERR "cannot malloc\n");
@@ -305,7 +389,8 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 		if (snd_usb_get_speed(dev) == USB_SPEED_HIGH)
 			fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1)
 					* (fp->maxpacksize & 0x7ff);
-		fp->attributes = csep ? csep[3] : 0;
+		fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no);
+		fp->clock = clock;
 
 		/* some quirks for attributes here */
 
diff --git a/sound/usb/format.c b/sound/usb/format.c
index b87cf87c4e7b..5367cd1e52d9 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -29,6 +29,7 @@
 #include "quirks.h"
 #include "helper.h"
 #include "debug.h"
+#include "clock.h"
 
 /*
  * parse the audio format type I descriptor
@@ -215,15 +216,17 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
 	struct usb_device *dev = chip->dev;
 	unsigned char tmp[2], *data;
 	int i, nr_rates, data_size, ret = 0;
+	int clock = snd_usb_clock_find_source(chip, chip->ctrl_intf, fp->clock);
 
 	/* get the number of sample rates first by only fetching 2 bytes */
 	ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE,
 			      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
-			      UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8,
+			      UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8,
 			      tmp, sizeof(tmp), 1000);
 
 	if (ret < 0) {
-		snd_printk(KERN_ERR "unable to retrieve number of sample rates\n");
+		snd_printk(KERN_ERR "%s(): unable to retrieve number of sample rates (clock %d)\n",
+				__func__, clock);
 		goto err;
 	}
 
@@ -237,12 +240,13 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
 
 	/* now get the full information */
 	ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE,
-			       USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
-			       UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8,
-			       data, data_size, 1000);
+			      USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+			      UAC2_CS_CONTROL_SAM_FREQ << 8, clock << 8,
+			      data, data_size, 1000);
 
 	if (ret < 0) {
-		snd_printk(KERN_ERR "unable to retrieve sample rate range\n");
+		snd_printk(KERN_ERR "%s(): unable to retrieve sample rate range (clock %d)\n",
+				__func__, clock);
 		ret = -EINVAL;
 		goto err_free;
 	}
@@ -278,12 +282,11 @@ err:
  * parse the format type I and III descriptors
  */
 static int parse_audio_format_i(struct snd_usb_audio *chip,
-				struct audioformat *fp,
-				int format, void *_fmt,
+				struct audioformat *fp, int format,
+				struct uac_format_type_i_continuous_descriptor *fmt,
 				struct usb_host_interface *iface)
 {
 	struct usb_interface_descriptor *altsd = get_iface_desc(iface);
-	struct uac_format_type_i_discrete_descriptor *fmt = _fmt;
 	int protocol = altsd->bInterfaceProtocol;
 	int pcm_format, ret;
 
@@ -320,7 +323,7 @@ static int parse_audio_format_i(struct snd_usb_audio *chip,
 	switch (protocol) {
 	case UAC_VERSION_1:
 		fp->channels = fmt->bNrChannels;
-		ret = parse_audio_format_rates_v1(chip, fp, _fmt, 7);
+		ret = parse_audio_format_rates_v1(chip, fp, (unsigned char *) fmt, 7);
 		break;
 	case UAC_VERSION_2:
 		/* fp->channels is already set in this case */
@@ -392,12 +395,12 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip,
 }
 
 int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp,
-		       int format, unsigned char *fmt, int stream,
-		       struct usb_host_interface *iface)
+			       int format, struct uac_format_type_i_continuous_descriptor *fmt,
+			       int stream, struct usb_host_interface *iface)
 {
 	int err;
 
-	switch (fmt[3]) {
+	switch (fmt->bFormatType) {
 	case UAC_FORMAT_TYPE_I:
 	case UAC_FORMAT_TYPE_III:
 		err = parse_audio_format_i(chip, fp, format, fmt, iface);
@@ -407,10 +410,11 @@ int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *f
 		break;
 	default:
 		snd_printd(KERN_INFO "%d:%u:%d : format type %d is not supported yet\n",
-			   chip->dev->devnum, fp->iface, fp->altsetting, fmt[3]);
-		return -1;
+			   chip->dev->devnum, fp->iface, fp->altsetting,
+			   fmt->bFormatType);
+		return -ENOTSUPP;
 	}
-	fp->fmt_type = fmt[3];
+	fp->fmt_type = fmt->bFormatType;
 	if (err < 0)
 		return err;
 #if 1
@@ -421,10 +425,10 @@ int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *f
 	if (chip->usb_id == USB_ID(0x041e, 0x3000) ||
 	    chip->usb_id == USB_ID(0x041e, 0x3020) ||
 	    chip->usb_id == USB_ID(0x041e, 0x3061)) {
-		if (fmt[3] == UAC_FORMAT_TYPE_I &&
+		if (fmt->bFormatType == UAC_FORMAT_TYPE_I &&
 		    fp->rates != SNDRV_PCM_RATE_48000 &&
 		    fp->rates != SNDRV_PCM_RATE_96000)
-			return -1;
+			return -ENOTSUPP;
 	}
 #endif
 	return 0;
diff --git a/sound/usb/format.h b/sound/usb/format.h
index 8298c4e8ddfa..387924f0af85 100644
--- a/sound/usb/format.h
+++ b/sound/usb/format.h
@@ -1,8 +1,9 @@
 #ifndef __USBAUDIO_FORMAT_H
 #define __USBAUDIO_FORMAT_H
 
-int snd_usb_parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp,
-			       int format, unsigned char *fmt, int stream,
-			       struct usb_host_interface *iface);
+int snd_usb_parse_audio_format(struct snd_usb_audio *chip,
+			       struct audioformat *fp, int format,
+			       struct uac_format_type_i_continuous_descriptor *fmt,
+			       int stream, struct usb_host_interface *iface);
 
 #endif /*  __USBAUDIO_FORMAT_H */
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 8b1e4b124a9f..46785643c66d 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -645,6 +645,105 @@ static struct usb_protocol_ops snd_usbmidi_cme_ops = {
 };
 
 /*
+ * AKAI MPD16 protocol:
+ *
+ * For control port (endpoint 1):
+ * ==============================
+ * One or more chunks consisting of first byte of (0x10 | msg_len) and then a
+ * SysEx message (msg_len=9 bytes long).
+ *
+ * For data port (endpoint 2):
+ * ===========================
+ * One or more chunks consisting of first byte of (0x20 | msg_len) and then a
+ * MIDI message (msg_len bytes long)
+ *
+ * Messages sent: Active Sense, Note On, Poly Pressure, Control Change.
+ */
+static void snd_usbmidi_akai_input(struct snd_usb_midi_in_endpoint *ep,
+				   uint8_t *buffer, int buffer_length)
+{
+	unsigned int pos = 0;
+	unsigned int len = (unsigned int)buffer_length;
+	while (pos < len) {
+		unsigned int port = (buffer[pos] >> 4) - 1;
+		unsigned int msg_len = buffer[pos] & 0x0f;
+		pos++;
+		if (pos + msg_len <= len && port < 2)
+			snd_usbmidi_input_data(ep, 0, &buffer[pos], msg_len);
+		pos += msg_len;
+	}
+}
+
+#define MAX_AKAI_SYSEX_LEN 9
+
+static void snd_usbmidi_akai_output(struct snd_usb_midi_out_endpoint *ep,
+				    struct urb *urb)
+{
+	uint8_t *msg;
+	int pos, end, count, buf_end;
+	uint8_t tmp[MAX_AKAI_SYSEX_LEN];
+	struct snd_rawmidi_substream *substream = ep->ports[0].substream;
+
+	if (!ep->ports[0].active)
+		return;
+
+	msg = urb->transfer_buffer + urb->transfer_buffer_length;
+	buf_end = ep->max_transfer - MAX_AKAI_SYSEX_LEN - 1;
+
+	/* only try adding more data when there's space for at least 1 SysEx */
+	while (urb->transfer_buffer_length < buf_end) {
+		count = snd_rawmidi_transmit_peek(substream,
+						  tmp, MAX_AKAI_SYSEX_LEN);
+		if (!count) {
+			ep->ports[0].active = 0;
+			return;
+		}
+		/* try to skip non-SysEx data */
+		for (pos = 0; pos < count && tmp[pos] != 0xF0; pos++)
+			;
+
+		if (pos > 0) {
+			snd_rawmidi_transmit_ack(substream, pos);
+			continue;
+		}
+
+		/* look for the start or end marker */
+		for (end = 1; end < count && tmp[end] < 0xF0; end++)
+			;
+
+		/* next SysEx started before the end of current one */
+		if (end < count && tmp[end] == 0xF0) {
+			/* it's incomplete - drop it */
+			snd_rawmidi_transmit_ack(substream, end);
+			continue;
+		}
+		/* SysEx complete */
+		if (end < count && tmp[end] == 0xF7) {
+			/* queue it, ack it, and get the next one */
+			count = end + 1;
+			msg[0] = 0x10 | count;
+			memcpy(&msg[1], tmp, count);
+			snd_rawmidi_transmit_ack(substream, count);
+			urb->transfer_buffer_length += count + 1;
+			msg += count + 1;
+			continue;
+		}
+		/* less than 9 bytes and no end byte - wait for more */
+		if (count < MAX_AKAI_SYSEX_LEN) {
+			ep->ports[0].active = 0;
+			return;
+		}
+		/* 9 bytes and no end marker in sight - malformed, skip it */
+		snd_rawmidi_transmit_ack(substream, count);
+	}
+}
+
+static struct usb_protocol_ops snd_usbmidi_akai_ops = {
+	.input = snd_usbmidi_akai_input,
+	.output = snd_usbmidi_akai_output,
+};
+
+/*
  * Novation USB MIDI protocol: number of data bytes is in the first byte
  * (when receiving) (+1!) or in the second byte (when sending); data begins
  * at the third byte.
@@ -1434,6 +1533,11 @@ static struct port_info {
 	EXTERNAL_PORT(0x086a, 0x0001, 8, "%s Broadcast"),
 	EXTERNAL_PORT(0x086a, 0x0002, 8, "%s Broadcast"),
 	EXTERNAL_PORT(0x086a, 0x0003, 4, "%s Broadcast"),
+	/* Akai MPD16 */
+	CONTROL_PORT(0x09e8, 0x0062, 0, "%s Control"),
+	PORT_INFO(0x09e8, 0x0062, 1, "%s MIDI", 0,
+		SNDRV_SEQ_PORT_TYPE_MIDI_GENERIC |
+		SNDRV_SEQ_PORT_TYPE_HARDWARE),
 	/* Access Music Virus TI */
 	EXTERNAL_PORT(0x133e, 0x0815, 0, "%s MIDI"),
 	PORT_INFO(0x133e, 0x0815, 1, "%s Synth", 0,
@@ -2035,6 +2139,12 @@ int snd_usbmidi_create(struct snd_card *card,
 		umidi->usb_protocol_ops = &snd_usbmidi_cme_ops;
 		err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
 		break;
+	case QUIRK_MIDI_AKAI:
+		umidi->usb_protocol_ops = &snd_usbmidi_akai_ops;
+		err = snd_usbmidi_detect_per_port_endpoints(umidi, endpoints);
+		/* endpoint 1 is input-only */
+		endpoints[1].out_cables = 0;
+		break;
 	default:
 		snd_printd(KERN_ERR "invalid quirk type %d\n", quirk->type);
 		err = -ENXIO;
diff --git a/sound/usb/midi.h b/sound/usb/midi.h
index 2089ec987c66..2fca80b744c0 100644
--- a/sound/usb/midi.h
+++ b/sound/usb/midi.h
@@ -37,6 +37,8 @@ struct snd_usb_midi_endpoint_info {
 
 /* for QUIRK_MIDI_CME, data is NULL */
 
+/* for QUIRK_MIDI_AKAI, data is NULL */
+
 int snd_usbmidi_create(struct snd_card *card,
 		       struct usb_interface *iface,
 		       struct list_head *midi_list,
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 97dd17655104..a060d005e209 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -78,39 +78,6 @@ enum {
 	USB_MIXER_U16,
 };
 
-enum {
-	USB_PROC_UPDOWN = 1,
-	USB_PROC_UPDOWN_SWITCH = 1,
-	USB_PROC_UPDOWN_MODE_SEL = 2,
-
-	USB_PROC_PROLOGIC = 2,
-	USB_PROC_PROLOGIC_SWITCH = 1,
-	USB_PROC_PROLOGIC_MODE_SEL = 2,
-
-	USB_PROC_3DENH = 3,
-	USB_PROC_3DENH_SWITCH = 1,
-	USB_PROC_3DENH_SPACE = 2,
-
-	USB_PROC_REVERB = 4,
-	USB_PROC_REVERB_SWITCH = 1,
-	USB_PROC_REVERB_LEVEL = 2,
-	USB_PROC_REVERB_TIME = 3,
-	USB_PROC_REVERB_DELAY = 4,
-
-	USB_PROC_CHORUS = 5,
-	USB_PROC_CHORUS_SWITCH = 1,
-	USB_PROC_CHORUS_LEVEL = 2,
-	USB_PROC_CHORUS_RATE = 3,
-	USB_PROC_CHORUS_DEPTH = 4,
-
-	USB_PROC_DCR = 6,
-	USB_PROC_DCR_SWITCH = 1,
-	USB_PROC_DCR_RATIO = 2,
-	USB_PROC_DCR_MAX_AMP = 3,
-	USB_PROC_DCR_THRESHOLD = 4,
-	USB_PROC_DCR_ATTACK = 5,
-	USB_PROC_DCR_RELEASE = 6,
-};
 
 /*E-mu 0202(0404) eXtension Unit(XU) control*/
 enum {
@@ -198,22 +165,24 @@ static int check_mapped_selector_name(struct mixer_build *state, int unitid,
 
 /*
  * find an audio control unit with the given unit id
- * this doesn't return any clock related units, so they need to be handled elsewhere
  */
 static void *find_audio_control_unit(struct mixer_build *state, unsigned char unit)
 {
-	unsigned char *p;
+	/* we just parse the header */
+	struct uac_feature_unit_descriptor *hdr = NULL;
 
-	p = NULL;
-	while ((p = snd_usb_find_desc(state->buffer, state->buflen, p,
-				      USB_DT_CS_INTERFACE)) != NULL) {
-		if (p[0] >= 4 && p[2] >= UAC_INPUT_TERMINAL && p[2] <= UAC2_EXTENSION_UNIT_V2 && p[3] == unit)
-			return p;
+	while ((hdr = snd_usb_find_desc(state->buffer, state->buflen, hdr,
+					USB_DT_CS_INTERFACE)) != NULL) {
+		if (hdr->bLength >= 4 &&
+		    hdr->bDescriptorSubtype >= UAC_INPUT_TERMINAL &&
+		    hdr->bDescriptorSubtype <= UAC2_SAMPLE_RATE_CONVERTER &&
+		    hdr->bUnitID == unit)
+			return hdr;
 	}
+
 	return NULL;
 }
 
-
 /*
  * copy a string with the given id
  */
@@ -344,8 +313,8 @@ static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int v
 			      buf, sizeof(buf), 1000);
 
 	if (ret < 0) {
-		snd_printdd(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n",
-			    request, validx, cval->mixer->ctrlif | (cval->id << 8), cval->val_type);
+		snd_printk(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n",
+			   request, validx, cval->mixer->ctrlif | (cval->id << 8), cval->val_type);
 		return ret;
 	}
 
@@ -462,6 +431,16 @@ static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel,
 			     int index, int value)
 {
 	int err;
+	unsigned int read_only = (channel == 0) ?
+		cval->master_readonly :
+		cval->ch_readonly & (1 << (channel - 1));
+
+	if (read_only) {
+		snd_printdd(KERN_INFO "%s(): channel %d of control %d is read_only\n",
+			    __func__, channel, cval->control);
+		return 0;
+	}
+
 	err = snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, (cval->control << 8) | channel,
 			    value);
 	if (err < 0)
@@ -631,6 +610,7 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm
  */
 static int check_input_term(struct mixer_build *state, int id, struct usb_audio_term *term)
 {
+	int err;
 	void *p1;
 
 	memset(term, 0, sizeof(*term));
@@ -651,6 +631,11 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_
 				term->channels = d->bNrChannels;
 				term->chconfig = le32_to_cpu(d->bmChannelConfig);
 				term->name = d->iTerminal;
+
+				/* call recursively to get the clock selectors */
+				err = check_input_term(state, d->bCSourceID, term);
+				if (err < 0)
+					return err;
 			}
 			return 0;
 		case UAC_FEATURE_UNIT: {
@@ -667,7 +652,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_
 			term->name = uac_mixer_unit_iMixer(d);
 			return 0;
 		}
-		case UAC_SELECTOR_UNIT: {
+		case UAC_SELECTOR_UNIT:
+		case UAC2_CLOCK_SELECTOR: {
 			struct uac_selector_unit_descriptor *d = p1;
 			/* call recursively to retrieve the channel info */
 			if (check_input_term(state, d->baSourceID[0], term) < 0)
@@ -690,6 +676,13 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_
 			term->name = uac_processing_unit_iProcessing(d, state->mixer->protocol);
 			return 0;
 		}
+		case UAC2_CLOCK_SOURCE: {
+			struct uac_clock_source_descriptor *d = p1;
+			term->type = d->bDescriptorSubtype << 16; /* virtual type */
+			term->id = id;
+			term->name = d->iClockSource;
+			return 0;
+		}
 		default:
 			return -ENODEV;
 		}
@@ -709,16 +702,20 @@ struct usb_feature_control_info {
 };
 
 static struct usb_feature_control_info audio_feature_info[] = {
-	{ "Mute",		USB_MIXER_INV_BOOLEAN },
-	{ "Volume",		USB_MIXER_S16 },
+	{ "Mute",			USB_MIXER_INV_BOOLEAN },
+	{ "Volume",			USB_MIXER_S16 },
 	{ "Tone Control - Bass",	USB_MIXER_S8 },
 	{ "Tone Control - Mid",		USB_MIXER_S8 },
 	{ "Tone Control - Treble",	USB_MIXER_S8 },
 	{ "Graphic Equalizer",		USB_MIXER_S8 }, /* FIXME: not implemeted yet */
-	{ "Auto Gain Control",	USB_MIXER_BOOLEAN },
-	{ "Delay Control",	USB_MIXER_U16 },
-	{ "Bass Boost",		USB_MIXER_BOOLEAN },
-	{ "Loudness",		USB_MIXER_BOOLEAN },
+	{ "Auto Gain Control",		USB_MIXER_BOOLEAN },
+	{ "Delay Control",		USB_MIXER_U16 },
+	{ "Bass Boost",			USB_MIXER_BOOLEAN },
+	{ "Loudness",			USB_MIXER_BOOLEAN },
+	/* UAC2 specific */
+	{ "Input Gain Control",		USB_MIXER_U16 },
+	{ "Input Gain Pad Control",	USB_MIXER_BOOLEAN },
+	{ "Phase Inverter Control",	USB_MIXER_BOOLEAN },
 };
 
 
@@ -958,7 +955,7 @@ static size_t append_ctl_name(struct snd_kcontrol *kctl, const char *str)
 static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 			      unsigned int ctl_mask, int control,
 			      struct usb_audio_term *iterm, int unitid,
-			      int read_only)
+			      int readonly_mask)
 {
 	struct uac_feature_unit_descriptor *desc = raw_desc;
 	unsigned int len = 0;
@@ -970,7 +967,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 
 	control++; /* change from zero-based to 1-based value */
 
-	if (control == UAC_GRAPHIC_EQUALIZER_CONTROL) {
+	if (control == UAC_FU_GRAPHIC_EQUALIZER) {
 		/* FIXME: not supported yet */
 		return;
 	}
@@ -989,20 +986,25 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 	cval->control = control;
 	cval->cmask = ctl_mask;
 	cval->val_type = audio_feature_info[control-1].type;
-	if (ctl_mask == 0)
+	if (ctl_mask == 0) {
 		cval->channels = 1;	/* master channel */
-	else {
+		cval->master_readonly = readonly_mask;
+	} else {
 		int i, c = 0;
 		for (i = 0; i < 16; i++)
 			if (ctl_mask & (1 << i))
 				c++;
 		cval->channels = c;
+		cval->ch_readonly = readonly_mask;
 	}
 
 	/* get min/max values */
 	get_min_max(cval, 0);
 
-	if (read_only)
+	/* if all channels in the mask are marked read-only, make the control
+	 * read-only. set_cur_mix_value() will check the mask again and won't
+	 * issue write commands to read-only channels. */
+	if (cval->channels == readonly_mask)
 		kctl = snd_ctl_new1(&usb_feature_unit_ctl_ro, cval);
 	else
 		kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval);
@@ -1021,8 +1023,8 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 				kctl->id.name, sizeof(kctl->id.name));
 
 	switch (control) {
-	case UAC_MUTE_CONTROL:
-	case UAC_VOLUME_CONTROL:
+	case UAC_FU_MUTE:
+	case UAC_FU_VOLUME:
 		/* determine the control name.  the rule is:
 		 * - if a name id is given in descriptor, use it.
 		 * - if the connected input can be determined, then use the name
@@ -1049,9 +1051,9 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 				len = append_ctl_name(kctl, " Playback");
 			}
 		}
-		append_ctl_name(kctl, control == UAC_MUTE_CONTROL ?
+		append_ctl_name(kctl, control == UAC_FU_MUTE ?
 				" Switch" : " Volume");
-		if (control == UAC_VOLUME_CONTROL) {
+		if (control == UAC_FU_VOLUME) {
 			kctl->tlv.c = mixer_vol_tlv;
 			kctl->vd[0].access |= 
 				SNDRV_CTL_ELEM_ACCESS_TLV_READ |
@@ -1126,7 +1128,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void
 	} else {
 		struct uac2_feature_unit_descriptor *ftr = _ftr;
 		csize = 4;
-		channels = (hdr->bLength - 6) / 4;
+		channels = (hdr->bLength - 6) / 4 - 1;
 		bmaControls = ftr->bmaControls;
 	}
 
@@ -1150,7 +1152,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void
 		snd_printk(KERN_INFO
 			   "usbmixer: master volume quirk for PCM2702 chip\n");
 		/* disable non-functional volume control */
-		master_bits &= ~UAC_FU_VOLUME;
+		master_bits &= ~UAC_CONTROL_BIT(UAC_FU_VOLUME);
 		break;
 	}
 	if (channels > 0)
@@ -1188,19 +1190,22 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void
 
 			for (j = 0; j < channels; j++) {
 				unsigned int mask = snd_usb_combine_bytes(bmaControls + csize * (j+1), csize);
-				if (mask & (1 << (i * 2))) {
+				if (uac2_control_is_readable(mask, i)) {
 					ch_bits |= (1 << j);
-					if (~mask & (1 << ((i * 2) + 1)))
+					if (!uac2_control_is_writeable(mask, i))
 						ch_read_only |= (1 << j);
 				}
 			}
 
-			/* FIXME: the whole unit is read-only if any of the channels is marked read-only */
+			/* NOTE: build_feature_ctl() will mark the control read-only if all channels
+			 * are marked read-only in the descriptors. Otherwise, the control will be
+			 * reported as writeable, but the driver will not actually issue a write
+			 * command for read-only channels */
 			if (ch_bits & 1) /* the first channel must be set (for ease of programming) */
-				build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, !!ch_read_only);
-			if (master_bits & (1 << i * 2))
+				build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, ch_read_only);
+			if (uac2_control_is_readable(master_bits, i))
 				build_feature_ctl(state, _ftr, 0, i, &iterm, unitid,
-						  ~master_bits & (1 << ((i * 2) + 1)));
+						  !uac2_control_is_writeable(master_bits, i));
 		}
 	}
 
@@ -1392,51 +1397,51 @@ struct procunit_info {
 };
 
 static struct procunit_value_info updown_proc_info[] = {
-	{ USB_PROC_UPDOWN_SWITCH, "Switch", USB_MIXER_BOOLEAN },
-	{ USB_PROC_UPDOWN_MODE_SEL, "Mode Select", USB_MIXER_U8, 1 },
+	{ UAC_UD_ENABLE, "Switch", USB_MIXER_BOOLEAN },
+	{ UAC_UD_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 },
 	{ 0 }
 };
 static struct procunit_value_info prologic_proc_info[] = {
-	{ USB_PROC_PROLOGIC_SWITCH, "Switch", USB_MIXER_BOOLEAN },
-	{ USB_PROC_PROLOGIC_MODE_SEL, "Mode Select", USB_MIXER_U8, 1 },
+	{ UAC_DP_ENABLE, "Switch", USB_MIXER_BOOLEAN },
+	{ UAC_DP_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 },
 	{ 0 }
 };
 static struct procunit_value_info threed_enh_proc_info[] = {
-	{ USB_PROC_3DENH_SWITCH, "Switch", USB_MIXER_BOOLEAN },
-	{ USB_PROC_3DENH_SPACE, "Spaciousness", USB_MIXER_U8 },
+	{ UAC_3D_ENABLE, "Switch", USB_MIXER_BOOLEAN },
+	{ UAC_3D_SPACE, "Spaciousness", USB_MIXER_U8 },
 	{ 0 }
 };
 static struct procunit_value_info reverb_proc_info[] = {
-	{ USB_PROC_REVERB_SWITCH, "Switch", USB_MIXER_BOOLEAN },
-	{ USB_PROC_REVERB_LEVEL, "Level", USB_MIXER_U8 },
-	{ USB_PROC_REVERB_TIME, "Time", USB_MIXER_U16 },
-	{ USB_PROC_REVERB_DELAY, "Delay", USB_MIXER_U8 },
+	{ UAC_REVERB_ENABLE, "Switch", USB_MIXER_BOOLEAN },
+	{ UAC_REVERB_LEVEL, "Level", USB_MIXER_U8 },
+	{ UAC_REVERB_TIME, "Time", USB_MIXER_U16 },
+	{ UAC_REVERB_FEEDBACK, "Feedback", USB_MIXER_U8 },
 	{ 0 }
 };
 static struct procunit_value_info chorus_proc_info[] = {
-	{ USB_PROC_CHORUS_SWITCH, "Switch", USB_MIXER_BOOLEAN },
-	{ USB_PROC_CHORUS_LEVEL, "Level", USB_MIXER_U8 },
-	{ USB_PROC_CHORUS_RATE, "Rate", USB_MIXER_U16 },
-	{ USB_PROC_CHORUS_DEPTH, "Depth", USB_MIXER_U16 },
+	{ UAC_CHORUS_ENABLE, "Switch", USB_MIXER_BOOLEAN },
+	{ UAC_CHORUS_LEVEL, "Level", USB_MIXER_U8 },
+	{ UAC_CHORUS_RATE, "Rate", USB_MIXER_U16 },
+	{ UAC_CHORUS_DEPTH, "Depth", USB_MIXER_U16 },
 	{ 0 }
 };
 static struct procunit_value_info dcr_proc_info[] = {
-	{ USB_PROC_DCR_SWITCH, "Switch", USB_MIXER_BOOLEAN },
-	{ USB_PROC_DCR_RATIO, "Ratio", USB_MIXER_U16 },
-	{ USB_PROC_DCR_MAX_AMP, "Max Amp", USB_MIXER_S16 },
-	{ USB_PROC_DCR_THRESHOLD, "Threshold", USB_MIXER_S16 },
-	{ USB_PROC_DCR_ATTACK, "Attack Time", USB_MIXER_U16 },
-	{ USB_PROC_DCR_RELEASE, "Release Time", USB_MIXER_U16 },
+	{ UAC_DCR_ENABLE, "Switch", USB_MIXER_BOOLEAN },
+	{ UAC_DCR_RATE, "Ratio", USB_MIXER_U16 },
+	{ UAC_DCR_MAXAMPL, "Max Amp", USB_MIXER_S16 },
+	{ UAC_DCR_THRESHOLD, "Threshold", USB_MIXER_S16 },
+	{ UAC_DCR_ATTACK_TIME, "Attack Time", USB_MIXER_U16 },
+	{ UAC_DCR_RELEASE_TIME, "Release Time", USB_MIXER_U16 },
 	{ 0 }
 };
 
 static struct procunit_info procunits[] = {
-	{ USB_PROC_UPDOWN, "Up Down", updown_proc_info },
-	{ USB_PROC_PROLOGIC, "Dolby Prologic", prologic_proc_info },
-	{ USB_PROC_3DENH, "3D Stereo Extender", threed_enh_proc_info },
-	{ USB_PROC_REVERB, "Reverb", reverb_proc_info },
-	{ USB_PROC_CHORUS, "Chorus", chorus_proc_info },
-	{ USB_PROC_DCR, "DCR", dcr_proc_info },
+	{ UAC_PROCESS_UP_DOWNMIX, "Up Down", updown_proc_info },
+	{ UAC_PROCESS_DOLBY_PROLOGIC, "Dolby Prologic", prologic_proc_info },
+	{ UAC_PROCESS_STEREO_EXTENDER, "3D Stereo Extender", threed_enh_proc_info },
+	{ UAC_PROCESS_REVERB, "Reverb", reverb_proc_info },
+	{ UAC_PROCESS_CHORUS, "Chorus", chorus_proc_info },
+	{ UAC_PROCESS_DYN_RANGE_COMP, "DCR", dcr_proc_info },
 	{ 0 },
 };
 /*
@@ -1524,7 +1529,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw
 		cval->channels = 1;
 
 		/* get min/max values */
-		if (type == USB_PROC_UPDOWN && cval->control == USB_PROC_UPDOWN_MODE_SEL) {
+		if (type == UAC_PROCESS_UP_DOWNMIX && cval->control == UAC_UD_MODE_SELECT) {
 			__u8 *control_spec = uac_processing_unit_specific(desc, state->mixer->protocol);
 			/* FIXME: hard-coded */
 			cval->min = 1;
@@ -1619,7 +1624,7 @@ static int mixer_ctl_selector_get(struct snd_kcontrol *kcontrol, struct snd_ctl_
 	struct usb_mixer_elem_info *cval = kcontrol->private_data;
 	int val, err;
 
-	err = get_cur_ctl_value(cval, 0, &val);
+	err = get_cur_ctl_value(cval, cval->control << 8, &val);
 	if (err < 0) {
 		if (cval->mixer->ignore_ctl_error) {
 			ucontrol->value.enumerated.item[0] = 0;
@@ -1638,7 +1643,7 @@ static int mixer_ctl_selector_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
 	struct usb_mixer_elem_info *cval = kcontrol->private_data;
 	int val, oval, err;
 
-	err = get_cur_ctl_value(cval, 0, &oval);
+	err = get_cur_ctl_value(cval, cval->control << 8, &oval);
 	if (err < 0) {
 		if (cval->mixer->ignore_ctl_error)
 			return 0;
@@ -1647,7 +1652,7 @@ static int mixer_ctl_selector_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
 	val = ucontrol->value.enumerated.item[0];
 	val = get_abs_value(cval, val);
 	if (val != oval) {
-		set_cur_ctl_value(cval, 0, val);
+		set_cur_ctl_value(cval, cval->control << 8, val);
 		return 1;
 	}
 	return 0;
@@ -1729,6 +1734,11 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void
 	cval->res = 1;
 	cval->initialized = 1;
 
+	if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR)
+		cval->control = UAC2_CX_CLOCK_SELECTOR;
+	else
+		cval->control = 0;
+
 	namelist = kmalloc(sizeof(char *) * desc->bNrInPins, GFP_KERNEL);
 	if (! namelist) {
 		snd_printk(KERN_ERR "cannot malloc\n");
@@ -1778,7 +1788,9 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void
 		if (! len)
 			strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name));
 
-		if ((state->oterm.type & 0xff00) == 0x0100)
+		if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR)
+			append_ctl_name(kctl, " Clock Source");
+		else if ((state->oterm.type & 0xff00) == 0x0100)
 			append_ctl_name(kctl, " Capture Source");
 		else
 			append_ctl_name(kctl, " Playback Source");
@@ -1812,10 +1824,12 @@ static int parse_audio_unit(struct mixer_build *state, int unitid)
 
 	switch (p1[2]) {
 	case UAC_INPUT_TERMINAL:
+	case UAC2_CLOCK_SOURCE:
 		return 0; /* NOP */
 	case UAC_MIXER_UNIT:
 		return parse_audio_mixer_unit(state, unitid, p1);
 	case UAC_SELECTOR_UNIT:
+	case UAC2_CLOCK_SELECTOR:
 		return parse_audio_selector_unit(state, unitid, p1);
 	case UAC_FEATURE_UNIT:
 		return parse_audio_feature_unit(state, unitid, p1);
@@ -1912,6 +1926,11 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
 			err = parse_audio_unit(&state, desc->bSourceID);
 			if (err < 0)
 				return err;
+
+			/* for UAC2, use the same approach to also add the clock selectors */
+			err = parse_audio_unit(&state, desc->bCSourceID);
+			if (err < 0)
+				return err;
 		}
 	}
 
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index 130123854a6c..a7cf1007fbb0 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -34,6 +34,8 @@ struct usb_mixer_elem_info {
 	unsigned int id;
 	unsigned int control;	/* CS or ICN (high byte) */
 	unsigned int cmask; /* channel mask bitmap: 0 = master */
+	unsigned int ch_readonly;
+	unsigned int master_readonly;
 	int channels;
 	int val_type;
 	int min, max, res;
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index d93fc89beba8..f1324c423835 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -85,8 +85,8 @@ static struct usbmix_name_map extigy_map[] = {
 	/* 16: MU (w/o controls) */
 	{ 17, NULL, 1 }, /* DISABLED: PU-switch (any effect?) */
 	{ 17, "Channel Routing", 2 },	/* PU: mode select */
-	{ 18, "Tone Control - Bass", UAC_BASS_CONTROL }, /* FU */
-	{ 18, "Tone Control - Treble", UAC_TREBLE_CONTROL }, /* FU */
+	{ 18, "Tone Control - Bass", UAC_FU_BASS }, /* FU */
+	{ 18, "Tone Control - Treble", UAC_FU_TREBLE }, /* FU */
 	{ 18, "Master Playback" }, /* FU; others */
 	/* 19: OT speaker */
 	/* 20: OT headphone */
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 2bf0d77d1768..456829882f40 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -31,6 +31,7 @@
 #include "urb.h"
 #include "helper.h"
 #include "pcm.h"
+#include "clock.h"
 
 /*
  * return the current pcm pointer.  just based on the hwptr_done value.
@@ -120,10 +121,6 @@ static int init_pitch_v1(struct snd_usb_audio *chip, int iface,
 
 	ep = get_endpoint(alts, 0)->bEndpointAddress;
 
-	/* if endpoint doesn't have pitch control, bail out */
-	if (!(fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL))
-		return 0;
-
 	data[0] = 1;
 	if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR,
 				   USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT,
@@ -137,119 +134,49 @@ static int init_pitch_v1(struct snd_usb_audio *chip, int iface,
 	return 0;
 }
 
-/*
- * initialize the picth control and sample rate
- */
-int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface,
-		       struct usb_host_interface *alts,
-		       struct audioformat *fmt)
-{
-	struct usb_interface_descriptor *altsd = get_iface_desc(alts);
-
-	switch (altsd->bInterfaceProtocol) {
-	case UAC_VERSION_1:
-		return init_pitch_v1(chip, iface, alts, fmt);
-
-	case UAC_VERSION_2:
-		/* not implemented yet */
-		return 0;
-	}
-
-	return -EINVAL;
-}
-
-static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface,
-			      struct usb_host_interface *alts,
-			      struct audioformat *fmt, int rate)
+static int init_pitch_v2(struct snd_usb_audio *chip, int iface,
+			 struct usb_host_interface *alts,
+			 struct audioformat *fmt)
 {
 	struct usb_device *dev = chip->dev;
+	unsigned char data[1];
 	unsigned int ep;
-	unsigned char data[3];
-	int err, crate;
+	int err;
 
 	ep = get_endpoint(alts, 0)->bEndpointAddress;
-	/* if endpoint doesn't have sampling rate control, bail out */
-	if (!(fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE)) {
-		snd_printk(KERN_WARNING "%d:%d:%d: endpoint lacks sample rate attribute bit, cannot set.\n",
-				   dev->devnum, iface, fmt->altsetting);
-		return 0;
-	}
 
-	data[0] = rate;
-	data[1] = rate >> 8;
-	data[2] = rate >> 16;
-	if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR,
-				   USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT,
-				   UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep,
-				   data, sizeof(data), 1000)) < 0) {
-		snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n",
-			   dev->devnum, iface, fmt->altsetting, rate, ep);
-		return err;
-	}
-	if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR,
-				   USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_IN,
-				   UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep,
-				   data, sizeof(data), 1000)) < 0) {
-		snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n",
-			   dev->devnum, iface, fmt->altsetting, ep);
-		return 0; /* some devices don't support reading */
-	}
-	crate = data[0] | (data[1] << 8) | (data[2] << 16);
-	if (crate != rate) {
-		snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate);
-		// runtime->rate = crate;
-	}
-
-	return 0;
-}
-
-static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
-			      struct usb_host_interface *alts,
-			      struct audioformat *fmt, int rate)
-{
-	struct usb_device *dev = chip->dev;
-	unsigned char data[4];
-	int err, crate;
-
-	data[0] = rate;
-	data[1] = rate >> 8;
-	data[2] = rate >> 16;
-	data[3] = rate >> 24;
+	data[0] = 1;
 	if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR,
-				   USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT,
-				   UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8,
-				   data, sizeof(data), 1000)) < 0) {
-		snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2)\n",
-			   dev->devnum, iface, fmt->altsetting, rate);
-		return err;
-	}
-	if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
-				   USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
-				   UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8,
+				   USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT,
+				   UAC2_EP_CS_PITCH << 8, 0,
 				   data, sizeof(data), 1000)) < 0) {
-		snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
+		snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH (v2)\n",
 			   dev->devnum, iface, fmt->altsetting);
 		return err;
 	}
-	crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
-	if (crate != rate)
-		snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate);
 
 	return 0;
 }
 
-int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface,
-			     struct usb_host_interface *alts,
-			     struct audioformat *fmt, int rate)
+/*
+ * initialize the pitch control and sample rate
+ */
+int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface,
+		       struct usb_host_interface *alts,
+		       struct audioformat *fmt)
 {
 	struct usb_interface_descriptor *altsd = get_iface_desc(alts);
 
+	/* if endpoint doesn't have pitch control, bail out */
+	if (!(fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL))
+		return 0;
+
 	switch (altsd->bInterfaceProtocol) {
 	case UAC_VERSION_1:
-		return set_sample_rate_v1(chip, iface, alts, fmt, rate);
+		return init_pitch_v1(chip, iface, alts, fmt);
 
 	case UAC_VERSION_2:
-		return set_sample_rate_v2(chip, iface, alts, fmt, rate);
+		return init_pitch_v2(chip, iface, alts, fmt);
 	}
 
 	return -EINVAL;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 91ddef31bcbd..f8797f61a24b 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -1973,6 +1973,17 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 
+/* AKAI devices */
+{
+	USB_DEVICE(0x09e8, 0x0062),
+	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
+		.vendor_name = "AKAI",
+		.product_name = "MPD16",
+		.ifnum = 0,
+		.type = QUIRK_MIDI_AKAI,
+	}
+},
+
 /* TerraTec devices */
 {
 	USB_DEVICE_VENDOR_SPEC(0x0ccd, 0x0012),
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 136e5b4cf6de..b45e54c09ba2 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -289,6 +289,7 @@ int snd_usb_create_quirk(struct snd_usb_audio *chip,
 		[QUIRK_MIDI_FASTLANE] = create_any_midi_quirk,
 		[QUIRK_MIDI_EMAGIC] = create_any_midi_quirk,
 		[QUIRK_MIDI_CME] = create_any_midi_quirk,
+		[QUIRK_MIDI_AKAI] = create_any_midi_quirk,
 		[QUIRK_AUDIO_STANDARD_INTERFACE] = create_standard_audio_quirk,
 		[QUIRK_AUDIO_FIXED_ENDPOINT] = create_fixed_stream_quirk,
 		[QUIRK_AUDIO_EDIROL_UAXX] = create_uaxx_quirk,
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index d679e72a3e5c..24d3319cc34d 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -40,9 +40,6 @@ struct snd_usb_audio {
 	int num_interfaces;
 	int num_suspended_intf;
 
-	/* for audio class v2 */
-	int clock_id;
-
 	struct list_head pcm_list;	/* list of pcm streams */
 	int pcm_devs;
 
@@ -53,6 +50,8 @@ struct snd_usb_audio {
 	int setup;			/* from the 'device_setup' module param */
 	int nrpacks;			/* from the 'nrpacks' module param */
 	int async_unlink;		/* from the 'async_unlink' module param */
+
+	struct usb_host_interface *ctrl_intf;	/* the audio control interface */
 };
 
 /*
@@ -74,6 +73,7 @@ enum quirk_type {
 	QUIRK_MIDI_FASTLANE,
 	QUIRK_MIDI_EMAGIC,
 	QUIRK_MIDI_CME,
+	QUIRK_MIDI_AKAI,
 	QUIRK_MIDI_US122L,
 	QUIRK_AUDIO_STANDARD_INTERFACE,
 	QUIRK_AUDIO_FIXED_ENDPOINT,