diff --git a/[refs] b/[refs]
index 0cc895eea7aa..b262e21bd025 100644
--- a/[refs]
+++ b/[refs]
@@ -1,2 +1,2 @@
 ---
-refs/heads/master: a98d3984c85222aa9efc63c4f9dd3d805ce469f2
+refs/heads/master: a39e33eb2a9d6afa79021ba1de2aa7d1039f4e53
diff --git a/trunk/include/linux/usb/audio-v2.h b/trunk/include/linux/usb/audio-v2.h
index 92f1d99f0f17..2389f93a28b5 100644
--- a/trunk/include/linux/usb/audio-v2.h
+++ b/trunk/include/linux/usb/audio-v2.h
@@ -105,22 +105,6 @@ struct uac_as_header_descriptor_v2 {
 	__u8 iChannelNames;
 } __attribute__((packed));
 
-/* 4.10.1.2 Class-Specific AS Isochronous Audio Data Endpoint Descriptor */
-
-struct uac2_iso_endpoint_descriptor {
-	__u8  bLength;			/* in bytes: 8 */
-	__u8  bDescriptorType;		/* USB_DT_CS_ENDPOINT */
-	__u8  bDescriptorSubtype;	/* EP_GENERAL */
-	__u8  bmAttributes;
-	__u8  bmControls;
-	__u8  bLockDelayUnits;
-	__le16 wLockDelay;
-} __attribute__((packed));
-
-#define UAC2_CONTROL_PITCH		(3 << 0)
-#define UAC2_CONTROL_DATA_OVERRUN	(3 << 2)
-#define UAC2_CONTROL_DATA_UNDERRUN	(3 << 4)
-
 /* 6.1 Interrupt Data Message */
 
 #define UAC2_INTERRUPT_DATA_MSG_VENDOR	(1 << 0)
diff --git a/trunk/sound/core/pcm_lib.c b/trunk/sound/core/pcm_lib.c
index e9d98be190c5..a2ff86189d2a 100644
--- a/trunk/sound/core/pcm_lib.c
+++ b/trunk/sound/core/pcm_lib.c
@@ -345,9 +345,7 @@ 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;
-	if (delta < 0)
-		delta += runtime->boundary;
+	delta = (new_hw_ptr - old_hw_ptr) % runtime->boundary;
 	if (xrun_debug(substream, in_interrupt ?
 			XRUN_DEBUG_PERIODUPDATE : XRUN_DEBUG_HWPTRUPDATE)) {
 		char name[16];
@@ -441,13 +439,8 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream,
 		snd_pcm_playback_silence(substream, new_hw_ptr);
 
 	if (in_interrupt) {
-		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_interrupt = new_hw_ptr -
+				(new_hw_ptr % runtime->period_size);
 	}
 	runtime->hw_ptr_base = hw_base;
 	runtime->status->hw_ptr = new_hw_ptr;
diff --git a/trunk/sound/core/pcm_native.c b/trunk/sound/core/pcm_native.c
index 303ac04ff6e4..644c2bb17b86 100644
--- a/trunk/sound/core/pcm_native.c
+++ b/trunk/sound/core/pcm_native.c
@@ -27,6 +27,7 @@
 #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>
@@ -369,6 +370,38 @@ 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)
 {
@@ -444,9 +477,9 @@ 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;
-	runtime->boundary = runtime->buffer_size;
-	while (runtime->boundary * 2 <= LONG_MAX - runtime->buffer_size)
-		runtime->boundary *= 2;
+	err = calc_boundary(runtime);
+	if (err < 0)
+		goto _error;
 
 	snd_pcm_timer_resolution_change(substream);
 	runtime->status->state = SNDRV_PCM_STATE_SETUP;
diff --git a/trunk/sound/mips/au1x00.c b/trunk/sound/mips/au1x00.c
index 446cf9748664..3e763d6a5d67 100644
--- a/trunk/sound/mips/au1x00.c
+++ b/trunk/sound/mips/au1x00.c
@@ -516,7 +516,6 @@ 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/trunk/sound/oss/dmasound/dmasound_atari.c b/trunk/sound/oss/dmasound/dmasound_atari.c
index 13c214466d3b..1f4774123064 100644
--- a/trunk/sound/oss/dmasound/dmasound_atari.c
+++ b/trunk/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;
-		goto out;
+		return IRQ_HANDLED;
 	}
 
 	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);
-		goto out;
+		return IRQ_HANDLED;
 	}
 
 	/* Probably ;) one frame is finished. Well, in fact it may be that a
@@ -1322,7 +1322,6 @@ 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/trunk/sound/pci/asihpi/hpi.h b/trunk/sound/pci/asihpi/hpi.h
index 0173bbe62b67..99400de6c075 100644
--- a/trunk/sound/pci/asihpi/hpi.h
+++ b/trunk/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, 25)
+#define HPI_VER HPI_VERSION_CONSTRUCTOR(4L, 3, 18)
 
 /* Library version as documented in hpi-api-versions.txt */
 #define HPI_LIB_VER  HPI_VERSION_CONSTRUCTOR(9, 0, 0)
@@ -1632,12 +1632,6 @@ 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/trunk/sound/pci/asihpi/hpi6000.c b/trunk/sound/pci/asihpi/hpi6000.c
index 12dab5e4892c..839ecb2e4b64 100644
--- a/trunk/sound/pci/asihpi/hpi6000.c
+++ b/trunk/sound/pci/asihpi/hpi6000.c
@@ -691,6 +691,9 @@ 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;
 	}
@@ -1772,6 +1775,7 @@ 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;
@@ -1792,8 +1796,6 @@ 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/trunk/sound/pci/asihpi/hpi6205.c b/trunk/sound/pci/asihpi/hpi6205.c
index e89991ea3543..5e88c1fc2b9e 100644
--- a/trunk/sound/pci/asihpi/hpi6205.c
+++ b/trunk/sound/pci/asihpi/hpi6205.c
@@ -965,17 +965,24 @@ static void outstream_write(struct hpi_adapter_obj *pao,
 	hpi_init_response(phr, phm->object, phm->function, 0);
 	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. The DSP
-		   takes care of setting the stream data format using format information
-		   embedded in phm.
+		   adapter's buffers, reducing delay before stream can start
 		 */
 		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. */
@@ -1005,6 +1012,7 @@ 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) {
@@ -1361,9 +1369,6 @@ 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/trunk/sound/pci/asihpi/hpi_internal.h b/trunk/sound/pci/asihpi/hpi_internal.h
index fdd0ce02aa68..f1cd6f1a0d44 100644
--- a/trunk/sound/pci/asihpi/hpi_internal.h
+++ b/trunk/sound/pci/asihpi/hpi_internal.h
@@ -232,8 +232,6 @@ 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)
 
 /** \} */
 
@@ -480,10 +478,8 @@ 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 */
@@ -974,7 +970,6 @@ struct hpi_control_union_msg {
 				u32 mode;
 				u32 value;
 			} mode;
-			u32 blend;
 		} tuner;
 	} u;
 };
diff --git a/trunk/sound/pci/asihpi/hpicmn.c b/trunk/sound/pci/asihpi/hpicmn.c
index fcd64539d9ef..565102cae4f8 100644
--- a/trunk/sound/pci/asihpi/hpicmn.c
+++ b/trunk/sound/pci/asihpi/hpicmn.c
@@ -347,15 +347,20 @@ short hpi_check_control_cache(struct hpi_control_cache *p_cache,
 			found = 0;
 		break;
 	case HPI_CONTROL_TUNER:
-		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;
+		{
+			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;
+		}
 		break;
 	case HPI_CONTROL_AESEBU_RECEIVER:
 		if (phm->u.c.attribute == HPI_AESEBURX_ERRORSTATUS)
@@ -498,9 +503,6 @@ 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;
 
@@ -518,6 +520,8 @@ 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;
@@ -525,6 +529,8 @@ 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;
@@ -539,14 +545,20 @@ 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)
@@ -578,7 +590,7 @@ 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) {
+	if ((p_cache->init) && (p_cache->p_info)) {
 		kfree(p_cache->p_info);
 		p_cache->p_info = NULL;
 		p_cache->init = 0;
diff --git a/trunk/sound/pci/asihpi/hpifunc.c b/trunk/sound/pci/asihpi/hpifunc.c
index 298eef3e20e9..eda26b312324 100644
--- a/trunk/sound/pci/asihpi/hpifunc.c
+++ b/trunk/sound/pci/asihpi/hpifunc.c
@@ -2946,20 +2946,6 @@ 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)
 {
@@ -3280,7 +3266,8 @@ u16 hpi_entity_find_next(struct hpi_entity *container_entity,
 
 void hpi_entity_free(struct hpi_entity *entity)
 {
-	kfree(entity);
+	if (entity != NULL)
+		kfree(entity);
 }
 
 static u16 hpi_entity_alloc_and_copy(struct hpi_entity *src,
diff --git a/trunk/sound/pci/asihpi/hpios.c b/trunk/sound/pci/asihpi/hpios.c
index 742ee12a9e17..de615cfdb950 100644
--- a/trunk/sound/pci/asihpi/hpios.c
+++ b/trunk/sound/pci/asihpi/hpios.c
@@ -89,3 +89,26 @@ 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/trunk/sound/pci/asihpi/hpios.h b/trunk/sound/pci/asihpi/hpios.h
index 370f39b43f85..a62c3f1e5f09 100644
--- a/trunk/sound/pci/asihpi/hpios.h
+++ b/trunk/sound/pci/asihpi/hpios.h
@@ -166,4 +166,13 @@ 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/trunk/sound/pci/aw2/aw2-alsa.c b/trunk/sound/pci/aw2/aw2-alsa.c
index c15002242d98..67921f93a41e 100644
--- a/trunk/sound/pci/aw2/aw2-alsa.c
+++ b/trunk/sound/pci/aw2/aw2-alsa.c
@@ -26,7 +26,7 @@
 #include <linux/slab.h>
 #include <linux/interrupt.h>
 #include <linux/delay.h>
-#include <linux/io.h>
+#include <asm/io.h>
 #include <sound/core.h>
 #include <sound/initval.h>
 #include <sound/pcm.h>
@@ -44,6 +44,9 @@ 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
 
@@ -162,7 +165,7 @@ 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_PHILIPS, PCI_DEVICE_ID_PHILIPS_SAA7146, 0, 0,
+	{PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0,
 	 0, 0, 0},
 	{0}
 };
@@ -416,7 +419,7 @@ 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;
 }
@@ -432,7 +435,7 @@ 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/trunk/sound/pci/emu10k1/emufx.c b/trunk/sound/pci/emu10k1/emufx.c
index 7a9401462c1c..4b302d86f5f2 100644
--- a/trunk/sound/pci/emu10k1/emufx.c
+++ b/trunk/sound/pci/emu10k1/emufx.c
@@ -35,7 +35,6 @@
 #include <linux/vmalloc.h>
 #include <linux/init.h>
 #include <linux/mutex.h>
-#include <linux/moduleparam.h>
 
 #include <sound/core.h>
 #include <sound/tlv.h>
@@ -51,10 +50,6 @@
 #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
  */ 
@@ -301,7 +296,6 @@ 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
@@ -1078,17 +1072,10 @@ 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;
-	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;
-	}
+	ctl->min = 0;
+	ctl->max = 100;
+	ctl->tlv = snd_emu10k1_db_scale1;
+	ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100;	
 }
 
 static void __devinit
@@ -1100,17 +1087,10 @@ 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;
-	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;
-	}
+	ctl->min = 0;
+	ctl->max = 100;
+	ctl->tlv = snd_emu10k1_db_scale1;
+	ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100;
 }
 
 static void __devinit
diff --git a/trunk/sound/pci/hda/patch_conexant.c b/trunk/sound/pci/hda/patch_conexant.c
index e863649d31f5..f4a2bd621663 100644
--- a/trunk/sound/pci/hda/patch_conexant.c
+++ b/trunk/sound/pci/hda/patch_conexant.c
@@ -2975,6 +2975,7 @@ 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, 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/trunk/sound/soc/codecs/wm8350.c b/trunk/sound/soc/codecs/wm8350.c
index 0221ca79b3ae..8ae20208e7be 100644
--- a/trunk/sound/soc/codecs/wm8350.c
+++ b/trunk/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_SCALE(pre_amp_tlv, -1200, 3525, 0);
-static DECLARE_TLV_DB_SCALE(out_pga_tlv, -5700, 600, 0);
+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(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/trunk/sound/soc/codecs/wm8400.c b/trunk/sound/soc/codecs/wm8400.c
index 8f294066b0ed..7f5d080536a0 100644
--- a/trunk/sound/soc/codecs/wm8400.c
+++ b/trunk/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_SCALE(rec_mix_tlv, -1500, 600, 0);
+static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600);
 
-static const DECLARE_TLV_DB_SCALE(in_pga_tlv, -1650, 3000, 0);
+static const DECLARE_TLV_DB_LINEAR(in_pga_tlv, -1650, 3000);
 
-static const DECLARE_TLV_DB_SCALE(out_mix_tlv, -2100, 0, 0);
+static const DECLARE_TLV_DB_LINEAR(out_mix_tlv, -2100, 0);
 
-static const DECLARE_TLV_DB_SCALE(out_pga_tlv, -7300, 600, 0);
+static const DECLARE_TLV_DB_LINEAR(out_pga_tlv, -7300, 600);
 
-static const DECLARE_TLV_DB_SCALE(out_omix_tlv, -600, 0, 0);
+static const DECLARE_TLV_DB_LINEAR(out_omix_tlv, -600, 0);
 
-static const DECLARE_TLV_DB_SCALE(out_dac_tlv, -7163, 0, 0);
+static const DECLARE_TLV_DB_LINEAR(out_dac_tlv, -7163, 0);
 
-static const DECLARE_TLV_DB_SCALE(in_adc_tlv, -7163, 1763, 0);
+static const DECLARE_TLV_DB_LINEAR(in_adc_tlv, -7163, 1763);
 
-static const DECLARE_TLV_DB_SCALE(out_sidetone_tlv, -3600, 0, 0);
+static const DECLARE_TLV_DB_LINEAR(out_sidetone_tlv, -3600, 0);
 
 static int wm8400_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol,
         struct snd_ctl_elem_value *ucontrol)
@@ -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_SCALE_ITEM(-1200, 600, 0),
+	0,7, TLV_DB_LINEAR_ITEM(-1200, 600),
 };
 
 /* Left In PGA Connections */
diff --git a/trunk/sound/soc/codecs/wm8990.c b/trunk/sound/soc/codecs/wm8990.c
index c018772cc430..7b536d923ea9 100644
--- a/trunk/sound/soc/codecs/wm8990.c
+++ b/trunk/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_SCALE(rec_mix_tlv, -1500, 600, 0);
+static const DECLARE_TLV_DB_LINEAR(rec_mix_tlv, -1500, 600);
 
-static const DECLARE_TLV_DB_SCALE(in_pga_tlv, -1650, 3000, 0);
+static const DECLARE_TLV_DB_LINEAR(in_pga_tlv, -1650, 3000);
 
-static const DECLARE_TLV_DB_SCALE(out_mix_tlv, 0, -2100, 0);
+static const DECLARE_TLV_DB_LINEAR(out_mix_tlv, 0, -2100);
 
-static const DECLARE_TLV_DB_SCALE(out_pga_tlv, -7300, 600, 0);
+static const DECLARE_TLV_DB_LINEAR(out_pga_tlv, -7300, 600);
 
-static const DECLARE_TLV_DB_SCALE(out_omix_tlv, -600, 0, 0);
+static const DECLARE_TLV_DB_LINEAR(out_omix_tlv, -600, 0);
 
-static const DECLARE_TLV_DB_SCALE(out_dac_tlv, -7163, 0, 0);
+static const DECLARE_TLV_DB_LINEAR(out_dac_tlv, -7163, 0);
 
-static const DECLARE_TLV_DB_SCALE(in_adc_tlv, -7163, 1763, 0);
+static const DECLARE_TLV_DB_LINEAR(in_adc_tlv, -7163, 1763);
 
-static const DECLARE_TLV_DB_SCALE(out_sidetone_tlv, -3600, 0, 0);
+static const DECLARE_TLV_DB_LINEAR(out_sidetone_tlv, -3600, 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_SCALE_ITEM(-1200, 600, 0),
+	0, 7, TLV_DB_LINEAR_ITEM(-1200, 600),
 };
 
 /* Left In PGA Connections */
diff --git a/trunk/sound/soc/imx/imx-pcm-dma-mx2.c b/trunk/sound/soc/imx/imx-pcm-dma-mx2.c
index 05f19c9284f4..2b31ac673ea4 100644
--- a/trunk/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/trunk/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -73,8 +73,7 @@ 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 = 
-		snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
+	struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data;
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct imx_pcm_runtime_data *iprtd = runtime->private_data;
 	int ret;
@@ -103,7 +102,7 @@ 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_dai_get_dma_data(rtd->dai->cpu_dai, substream);
+	dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream);
 
 	iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH);
 	if (iprtd->dma < 0) {
@@ -213,7 +212,7 @@ 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_dai_get_dma_data(rtd->dai->cpu_dai, substream);
+	dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream);
 
 	iprtd->substream = substream;
 	iprtd->buf = (unsigned int *)substream->dma_buffer.area;
diff --git a/trunk/sound/soc/sh/siu_dai.c b/trunk/sound/soc/sh/siu_dai.c
index eeed5edd722b..d86ee1bfc03a 100644
--- a/trunk/sound/soc/sh/siu_dai.c
+++ b/trunk/sound/soc/sh/siu_dai.c
@@ -588,8 +588,6 @@ 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/trunk/sound/usb/caiaq/control.c b/trunk/sound/usb/caiaq/control.c
index 91c804cd2782..36ed703a7416 100644
--- a/trunk/sound/usb/caiaq/control.c
+++ b/trunk/sound/usb/caiaq/control.c
@@ -42,9 +42,8 @@ static int control_info(struct snd_kcontrol *kcontrol,
 
 	switch (dev->chip.usb_id) {
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ):
-	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ):
 		if (pos == 0) {
-			/* current input mode of A8DJ and A4DJ */
+			/* current input mode of A8DJ */
 			uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
 			uinfo->value.integer.min = 0;
 			uinfo->value.integer.max = 2;
@@ -52,6 +51,16 @@ static int control_info(struct snd_kcontrol *kcontrol,
 		}
 		break;
 
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ):
+		if (pos == 0) {
+			/* current input mode of A4DJ */
+			uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+			uinfo->value.integer.min = 0;
+			uinfo->value.integer.max = 1;
+			return 0;
+		}
+		break;
+
 	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1):
 		maxval = 127;
 		break;
@@ -77,6 +86,14 @@ static int control_get(struct snd_kcontrol *kcontrol,
 	struct snd_usb_caiaqdev *dev = caiaqdev(chip->card);
 	int pos = kcontrol->private_value;
 
+	if (dev->chip.usb_id ==
+		USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ)) {
+		/* A4DJ has only one control */
+		/* do not expose hardware input mode 0 */
+		ucontrol->value.integer.value[0] = dev->control_state[0] - 1;
+		return 0;
+	}
+
 	if (pos & CNT_INTVAL)
 		ucontrol->value.integer.value[0]
 			= dev->control_state[pos & ~CNT_INTVAL];
@@ -95,9 +112,20 @@ static int control_put(struct snd_kcontrol *kcontrol,
 	int pos = kcontrol->private_value;
 	unsigned char cmd = EP1_CMD_WRITE_IO;
 
-	if (dev->chip.usb_id ==
-		USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_TRAKTORKONTROLX1))
+	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):
 		cmd = EP1_CMD_DIMM_LEDS;
+		break;
+	}
 
 	if (pos & CNT_INTVAL) {
 		dev->control_state[pos & ~CNT_INTVAL]
diff --git a/trunk/sound/usb/caiaq/device.c b/trunk/sound/usb/caiaq/device.c
index cdfb856bddd2..805271827675 100644
--- a/trunk/sound/usb/caiaq/device.c
+++ b/trunk/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 1.3.21");
+MODULE_DESCRIPTION("caiaq USB audio, version 1.3.20");
 MODULE_LICENSE("GPL");
 MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2},"
 			 "{Native Instruments, RigKontrol3},"
@@ -320,6 +320,12 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev)
 		}
 
 		break;
+	case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ):
+		/* Audio 4 DJ - default input mode to phono */
+		dev->control_state[0] = 2;
+		snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO,
+			dev->control_state, 1);
+		break;
 	}
 
 	if (dev->spec.num_analog_audio_out +
diff --git a/trunk/sound/usb/caiaq/input.c b/trunk/sound/usb/caiaq/input.c
index dcb620796d9e..8bbfbfd4c658 100644
--- a/trunk/sound/usb/caiaq/input.c
+++ b/trunk/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[14] << 8) | buf[15]);
+		input_report_abs(input_dev, ABS_HAT2X, (buf[15] << 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/trunk/sound/usb/endpoint.c b/trunk/sound/usb/endpoint.c
index 28ee1ce3971a..ef07a6d0dd5f 100644
--- a/trunk/sound/usb/endpoint.c
+++ b/trunk/sound/usb/endpoint.c
@@ -149,47 +149,6 @@ 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;
-}
-
 int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 {
 	struct usb_device *dev;
@@ -199,8 +158,8 @@ 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;
-	struct uac_format_type_i_continuous_descriptor *fmt;
 
 	dev = chip->dev;
 
@@ -297,8 +256,8 @@ 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->bLength < 8)) ||
-		    ((protocol == UAC_VERSION_2) && (fmt->bLength != 6))) {
+		if (((protocol == UAC_VERSION_1) && (fmt[0] < 8)) ||
+		    ((protocol == UAC_VERSION_2) && (fmt[0] != 6))) {
 			snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n",
 				   dev->devnum, iface_no, altno);
 			continue;
@@ -309,9 +268,7 @@ 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->bNrChannels == 1 &&
-		    fmt->bSubframeSize == 2 &&
-		    altno == 2 && num == 3 &&
+		if (fmt[4] == 1 && fmt[5] == 2 && altno == 2 && num == 3 &&
 		    fp && fp->altsetting == 1 && fp->channels == 1 &&
 		    fp->formats == SNDRV_PCM_FMTBIT_S16_LE &&
 		    protocol == UAC_VERSION_1 &&
@@ -319,6 +276,17 @@ 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");
@@ -337,7 +305,7 @@ 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 = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no);
+		fp->attributes = csep ? csep[3] : 0;
 
 		/* some quirks for attributes here */
 
diff --git a/trunk/sound/usb/format.c b/trunk/sound/usb/format.c
index fe29d61de19b..b87cf87c4e7b 100644
--- a/trunk/sound/usb/format.c
+++ b/trunk/sound/usb/format.c
@@ -278,11 +278,12 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
  * parse the format type I and III descriptors
  */
 static int parse_audio_format_i(struct snd_usb_audio *chip,
-				struct audioformat *fp, int format,
-				struct uac_format_type_i_continuous_descriptor *fmt,
+				struct audioformat *fp,
+				int format, void *_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;
 
@@ -319,7 +320,7 @@ 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, (unsigned char *) fmt, 7);
+		ret = parse_audio_format_rates_v1(chip, fp, _fmt, 7);
 		break;
 	case UAC_VERSION_2:
 		/* fp->channels is already set in this case */
@@ -391,12 +392,12 @@ 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, struct uac_format_type_i_continuous_descriptor *fmt,
-			       int stream, struct usb_host_interface *iface)
+		       int format, unsigned char *fmt, int stream,
+		       struct usb_host_interface *iface)
 {
 	int err;
 
-	switch (fmt->bFormatType) {
+	switch (fmt[3]) {
 	case UAC_FORMAT_TYPE_I:
 	case UAC_FORMAT_TYPE_III:
 		err = parse_audio_format_i(chip, fp, format, fmt, iface);
@@ -406,11 +407,10 @@ 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->bFormatType);
-		return -ENOTSUPP;
+			   chip->dev->devnum, fp->iface, fp->altsetting, fmt[3]);
+		return -1;
 	}
-	fp->fmt_type = fmt->bFormatType;
+	fp->fmt_type = fmt[3];
 	if (err < 0)
 		return err;
 #if 1
@@ -421,10 +421,10 @@ 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->bFormatType == UAC_FORMAT_TYPE_I &&
+		if (fmt[3] == UAC_FORMAT_TYPE_I &&
 		    fp->rates != SNDRV_PCM_RATE_48000 &&
 		    fp->rates != SNDRV_PCM_RATE_96000)
-			return -ENOTSUPP;
+			return -1;
 	}
 #endif
 	return 0;
diff --git a/trunk/sound/usb/format.h b/trunk/sound/usb/format.h
index 387924f0af85..8298c4e8ddfa 100644
--- a/trunk/sound/usb/format.h
+++ b/trunk/sound/usb/format.h
@@ -1,9 +1,8 @@
 #ifndef __USBAUDIO_FORMAT_H
 #define __USBAUDIO_FORMAT_H
 
-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);
+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);
 
 #endif /*  __USBAUDIO_FORMAT_H */
diff --git a/trunk/sound/usb/midi.c b/trunk/sound/usb/midi.c
index 9c23d89066e4..2c1558c327bb 100644
--- a/trunk/sound/usb/midi.c
+++ b/trunk/sound/usb/midi.c
@@ -644,105 +644,6 @@ static struct usb_protocol_ops snd_usbmidi_cme_ops = {
 	.output_packet = snd_usbmidi_output_standard_packet,
 };
 
-/*
- * 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
@@ -1533,11 +1434,6 @@ 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,
@@ -2139,12 +2035,6 @@ 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/trunk/sound/usb/midi.h b/trunk/sound/usb/midi.h
index 2fca80b744c0..2089ec987c66 100644
--- a/trunk/sound/usb/midi.h
+++ b/trunk/sound/usb/midi.h
@@ -37,8 +37,6 @@ 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/trunk/sound/usb/mixer.c b/trunk/sound/usb/mixer.c
index 03ce971e0027..97dd17655104 100644
--- a/trunk/sound/usb/mixer.c
+++ b/trunk/sound/usb/mixer.c
@@ -1126,7 +1126,7 @@ 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 - 1;
+		channels = (hdr->bLength - 6) / 4;
 		bmaControls = ftr->bmaControls;
 	}
 
diff --git a/trunk/sound/usb/pcm.c b/trunk/sound/usb/pcm.c
index 056587de7be4..2bf0d77d1768 100644
--- a/trunk/sound/usb/pcm.c
+++ b/trunk/sound/usb/pcm.c
@@ -120,6 +120,10 @@ 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,
@@ -133,32 +137,8 @@ static int init_pitch_v1(struct snd_usb_audio *chip, int iface,
 	return 0;
 }
 
-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;
-	int err;
-
-	ep = get_endpoint(alts, 0)->bEndpointAddress;
-
-	data[0] = 1;
-	if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR,
-				   USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT,
-				   UAC2_EP_CS_PITCH << 8, 0,
-				   data, sizeof(data), 1000)) < 0) {
-		snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH (v2)\n",
-			   dev->devnum, iface, fmt->altsetting);
-		return err;
-	}
-
-	return 0;
-}
-
 /*
- * initialize the pitch control and sample rate
+ * initialize the picth control and sample rate
  */
 int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface,
 		       struct usb_host_interface *alts,
@@ -166,16 +146,13 @@ int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface,
 {
 	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 init_pitch_v1(chip, iface, alts, fmt);
 
 	case UAC_VERSION_2:
-		return init_pitch_v2(chip, iface, alts, fmt);
+		/* not implemented yet */
+		return 0;
 	}
 
 	return -EINVAL;
diff --git a/trunk/sound/usb/quirks-table.h b/trunk/sound/usb/quirks-table.h
index f8797f61a24b..91ddef31bcbd 100644
--- a/trunk/sound/usb/quirks-table.h
+++ b/trunk/sound/usb/quirks-table.h
@@ -1973,17 +1973,6 @@ 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/trunk/sound/usb/quirks.c b/trunk/sound/usb/quirks.c
index b45e54c09ba2..136e5b4cf6de 100644
--- a/trunk/sound/usb/quirks.c
+++ b/trunk/sound/usb/quirks.c
@@ -289,7 +289,6 @@ 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/trunk/sound/usb/usbaudio.h b/trunk/sound/usb/usbaudio.h
index 06ebf24d3a4d..d679e72a3e5c 100644
--- a/trunk/sound/usb/usbaudio.h
+++ b/trunk/sound/usb/usbaudio.h
@@ -74,7 +74,6 @@ 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,