32 files changed, 351 insertions, 235 deletions

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/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/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/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/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/midi.c b/sound/usb/midi.c
index 2c1558c327bb..9c23d89066e4 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..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;
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..06ebf24d3a4d 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -74,6 +74,7 @@ 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,