Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (26 commits)
  ALSA: snd-usb-caiaq: Bump version number to 1.3.21
  ALSA: Revert "ALSA: snd-usb-caiaq: Set default input mode of A4DJ"
  ALSA: snd-usb-caiaq: Simplify single case to an 'if'
  ALSA: snd-usb-caiaq: Restore 'Control vinyl' input mode on A4DJ
  ALSA: hda: Use LPIB for a Shuttle device
  ALSA: hda: Add support for another Lenovo ThinkPad Edge in conexant codec
  ALSA: hda: Use LPIB for Sony VPCS11V9E
  ALSA: usb-audio: fix feature unit parser for UAC2
  ALSA: asihpi - Minor code cleanup
  ALSA: asihpi - Add support for new ASI8800 family
  ALSA: asihpi - Fix bug preventing outstream_write preload from happening
  ALSA: asihpi - Fix imbalanced lock path in hw_message
  ALSA: asihpi - Remove support for old ASI8800 family
  ALSA: asihpi - Add hd radio blend functions
  ALSA: asihpi - Remove unused io map functions
  ALSA: usb-audio: add support for UAC2 pitch control
  ALSA: usb-audio: parse UAC2 endpoint descriptors correctly
  ALSA: usb-audio: fix return values
  ALSA: usb-audio: parse more format descriptors with structs
  sound: Add missing spin_unlock
  ...

24 files changed, 184 insertions, 180 deletions

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/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/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 77e22c2a8caa..dc79564fea30 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -2288,8 +2288,10 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = {
 	SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB),
 	SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", 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),
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/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/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/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/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/endpoint.c b/sound/usb/endpoint.c
index ef07a6d0dd5f..28ee1ce3971a 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -149,6 +149,47 @@ 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;
@@ -158,8 +199,8 @@ 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;
 
@@ -256,8 +297,8 @@ 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 +309,9 @@ 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 +319,6 @@ 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 +337,7 @@ 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 = csep ? csep[3] : 0;
+		fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no);
 
 		/* some quirks for attributes here */
 
diff --git a/sound/usb/format.c b/sound/usb/format.c
index b87cf87c4e7b..fe29d61de19b 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -278,12 +278,11 @@ 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 +319,7 @@ 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 +391,12 @@ 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 +406,11 @@ 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 +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[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/mixer.c b/sound/usb/mixer.c
index 97dd17655104..03ce971e0027 100644
--- a/sound/usb/mixer.c
+++ b/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;
+		channels = (hdr->bLength - 6) / 4 - 1;
 		bmaControls = ftr->bmaControls;
 	}
 
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 2bf0d77d1768..056587de7be4 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -120,10 +120,6 @@ 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,
@@ -137,8 +133,32 @@ 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 picth control and sample 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,
@@ -146,13 +166,16 @@ 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:
-		/* not implemented yet */
-		return 0;
+		return init_pitch_v2(chip, iface, alts, fmt);
 	}
 
 	return -EINVAL;