62 files changed, 1197 insertions, 643 deletions

diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c
index 67893372173..3ff8cc5f487 100644
--- a/sound/aoa/soundbus/i2sbus/core.c
+++ b/sound/aoa/soundbus/i2sbus/core.c
@@ -437,9 +437,11 @@ static int i2sbus_shutdown(struct macio_dev* dev)
 }
 
 static struct macio_driver i2sbus_drv = {
-        .name = "soundbus-i2s",
-        .owner = THIS_MODULE,
-        .match_table = i2sbus_match,
+        .driver = {
+                .name = "soundbus-i2s",
+                .owner = THIS_MODULE,
+                .of_match_table = i2sbus_match,
+        },
         .probe = i2sbus_probe,
         .remove = i2sbus_remove,
 #ifdef CONFIG_PM
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index 428121a7e70..10c3a871a12 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -657,7 +657,7 @@ static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev)
         if (sr & AC97C_SR_CAEVT) {
                 struct snd_pcm_runtime *runtime;
                 int offset, next_period, block_size;
-                dev_info(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n",
+                dev_dbg(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n",
                                 casr & AC97C_CSR_OVRUN   ? " OVRUN"   : "",
                                 casr & AC97C_CSR_RXRDY   ? " RXRDY"   : "",
                                 casr & AC97C_CSR_UNRUN   ? " UNRUN"   : "",
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index a2ff86189d2..e9d98be190c 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 644c2bb17b8..a3b2a647924 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)
 {
@@ -477,20 +444,18 @@ 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;
 
-        if (substream->latency_pm_qos_req) {
-                pm_qos_remove_request(substream->latency_pm_qos_req);
-                substream->latency_pm_qos_req = NULL;
-        }
+        if (pm_qos_request_active(&substream->latency_pm_qos_req))
+                pm_qos_remove_request(&substream->latency_pm_qos_req);
         if ((usecs = period_to_usecs(runtime)) >= 0)
-                substream->latency_pm_qos_req = pm_qos_add_request(
-                                        PM_QOS_CPU_DMA_LATENCY, usecs);
+                pm_qos_add_request(&substream->latency_pm_qos_req,
+                                   PM_QOS_CPU_DMA_LATENCY, usecs);
         return 0;
  _error:
         /* hardware might be unuseable from this time,
@@ -545,8 +510,7 @@ static int snd_pcm_hw_free(struct snd_pcm_substream *substream)
         if (substream->ops->hw_free)
                 result = substream->ops->hw_free(substream);
         runtime->status->state = SNDRV_PCM_STATE_OPEN;
-        pm_qos_remove_request(substream->latency_pm_qos_req);
-        substream->latency_pm_qos_req = NULL;
+        pm_qos_remove_request(&substream->latency_pm_qos_req);
         return result;
 }
 
diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c
index 3e763d6a5d6..446cf974866 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 1f477412306..13c214466d3 100644
--- a/sound/oss/dmasound/dmasound_atari.c
+++ b/sound/oss/dmasound/dmasound_atari.c
@@ -1277,7 +1277,7 @@ static irqreturn_t AtaInterrupt(int irq, void *dummy)
                  * (almost) like on the TT.
                  */
                 write_sq_ignore_int = 0;
-                return IRQ_HANDLED;
+                goto out;
         }
 
         if (!write_sq.active) {
@@ -1285,7 +1285,7 @@ static irqreturn_t AtaInterrupt(int irq, void *dummy)
                  * the sq variables, so better don't do anything here.
                  */
                 WAKE_UP(write_sq.sync_queue);
-                return IRQ_HANDLED;
+                goto out;
         }
 
         /* Probably ;) one frame is finished. Well, in fact it may be that a
@@ -1322,6 +1322,7 @@ static irqreturn_t AtaInterrupt(int irq, void *dummy)
         /* We are not playing after AtaPlay(), so there
            is nothing to play any more. Wake up a process
            waiting for audio output to drain. */
+out:
         spin_unlock(&dmasound.lock);
         return IRQ_HANDLED;
 }
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index f74c7372b3d..1db586af4f9 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -2578,6 +2578,9 @@ static int __devinit snd_card_asihpi_mixer_new(struct snd_card_asihpi *asihpi)
         if (err)
                 return -err;
 
+        memset(&prev_ctl, 0, sizeof(prev_ctl));
+        prev_ctl.control_type = -1;
+
         for (idx = 0; idx < 2000; idx++) {
                 err = hpi_mixer_get_control_by_index(
                                 ss, asihpi->h_mixer,
diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h
index 99400de6c07..0173bbe62b6 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 839ecb2e4b6..12dab5e4892 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 5e88c1fc2b9..3b441344822 100644
--- a/sound/pci/asihpi/hpi6205.c
+++ b/sound/pci/asihpi/hpi6205.c
@@ -941,11 +941,11 @@ static void outstream_host_buffer_free(struct hpi_adapter_obj *pao,
 
 }
 
-static long outstream_get_space_available(struct hpi_hostbuffer_status
+static u32 outstream_get_space_available(struct hpi_hostbuffer_status
         *status)
 {
-        return status->size_in_bytes - ((long)(status->host_index) -
-                (long)(status->dSP_index));
+        return status->size_in_bytes - (status->host_index -
+                status->dSP_index);
 }
 
 static void outstream_write(struct hpi_adapter_obj *pao,
@@ -954,7 +954,7 @@ static void outstream_write(struct hpi_adapter_obj *pao,
         struct hpi_hw_obj *phw = pao->priv;
         struct bus_master_interface *interface = phw->p_interface_buffer;
         struct hpi_hostbuffer_status *status;
-        long space_available;
+        u32 space_available;
 
         if (!phw->outstream_host_buffer_size[phm->obj_index]) {
                 /* there  is no BBM buffer, write via message */
@@ -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. */
@@ -1012,10 +1005,9 @@ 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) {
+        if (space_available < phm->u.d.u.data.data_size) {
                 phr->error = HPI_ERROR_INVALID_DATASIZE;
                 return;
         }
@@ -1026,7 +1018,7 @@ static void outstream_write(struct hpi_adapter_obj *pao,
                 && hpios_locked_mem_valid(&phw->outstream_host_buffers[phm->
                                 obj_index])) {
                 u8 *p_bbm_data;
-                long l_first_write;
+                u32 l_first_write;
                 u8 *p_app_data = (u8 *)phm->u.d.u.data.pb_data;
 
                 if (hpios_locked_mem_get_virt_addr(&phw->
@@ -1256,9 +1248,9 @@ static void instream_start(struct hpi_adapter_obj *pao,
         hw_message(pao, phm, phr);
 }
 
-static long instream_get_bytes_available(struct hpi_hostbuffer_status *status)
+static u32 instream_get_bytes_available(struct hpi_hostbuffer_status *status)
 {
-        return (long)(status->dSP_index) - (long)(status->host_index);
+        return status->dSP_index - status->host_index;
 }
 
 static void instream_read(struct hpi_adapter_obj *pao,
@@ -1267,9 +1259,9 @@ static void instream_read(struct hpi_adapter_obj *pao,
         struct hpi_hw_obj *phw = pao->priv;
         struct bus_master_interface *interface = phw->p_interface_buffer;
         struct hpi_hostbuffer_status *status;
-        long data_available;
+        u32 data_available;
         u8 *p_bbm_data;
-        long l_first_read;
+        u32 l_first_read;
         u8 *p_app_data = (u8 *)phm->u.d.u.data.pb_data;
 
         if (!phw->instream_host_buffer_size[phm->obj_index]) {
@@ -1280,7 +1272,7 @@ static void instream_read(struct hpi_adapter_obj *pao,
 
         status = &interface->instream_host_buffer_status[phm->obj_index];
         data_available = instream_get_bytes_available(status);
-        if (data_available < (long)phm->u.d.u.data.data_size) {
+        if (data_available < phm->u.d.u.data.data_size) {
                 phr->error = HPI_ERROR_INVALID_DATASIZE;
                 return;
         }
@@ -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 f1cd6f1a0d4..fdd0ce02aa6 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 565102cae4f..fcd64539d9e 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 eda26b31232..298eef3e20e 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 de615cfdb95..742ee12a9e1 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 a62c3f1e5f0..370f39b43f8 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 67921f93a41..c15002242d9 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 4b302d86f5f..7a9401462c1 100644
--- a/sound/pci/emu10k1/emufx.c
+++ b/sound/pci/emu10k1/emufx.c
@@ -35,6 +35,7 @@
 #include <linux/vmalloc.h>
 #include <linux/init.h>
 #include <linux/mutex.h>
+#include <linux/moduleparam.h>
 
 #include <sound/core.h>
 #include <sound/tlv.h>
@@ -50,6 +51,10 @@
 #define EMU10K1_CENTER_LFE_FROM_FRONT
 #endif
 
+static bool high_res_gpr_volume;
+module_param(high_res_gpr_volume, bool, 0444);
+MODULE_PARM_DESC(high_res_gpr_volume, "GPR mixer controls use 31-bit range.");
+
 /*
  *  Tables
  */ 
@@ -296,6 +301,7 @@ static const u32 db_table[101] = {
 
 /* EMU10k1/EMU10k2 DSP control db gain */
 static const DECLARE_TLV_DB_SCALE(snd_emu10k1_db_scale1, -4000, 40, 1);
+static const DECLARE_TLV_DB_LINEAR(snd_emu10k1_db_linear, TLV_DB_GAIN_MUTE, 0);
 
 static const u32 onoff_table[2] = {
         0x00000000, 0x00000001
@@ -1072,10 +1078,17 @@ snd_emu10k1_init_mono_control(struct snd_emu10k1_fx8010_control_gpr *ctl,
         strcpy(ctl->id.name, name);
         ctl->vcount = ctl->count = 1;
         ctl->gpr[0] = gpr + 0; ctl->value[0] = defval;
-        ctl->min = 0;
-        ctl->max = 100;
-        ctl->tlv = snd_emu10k1_db_scale1;
-        ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100;    
+        if (high_res_gpr_volume) {
+                ctl->min = 0;
+                ctl->max = 0x7fffffff;
+                ctl->tlv = snd_emu10k1_db_linear;
+                ctl->translation = EMU10K1_GPR_TRANSLATION_NONE;
+        } else {
+                ctl->min = 0;
+                ctl->max = 100;
+                ctl->tlv = snd_emu10k1_db_scale1;
+                ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100;
+        }
 }
 
 static void __devinit
@@ -1087,10 +1100,17 @@ snd_emu10k1_init_stereo_control(struct snd_emu10k1_fx8010_control_gpr *ctl,
         ctl->vcount = ctl->count = 2;
         ctl->gpr[0] = gpr + 0; ctl->value[0] = defval;
         ctl->gpr[1] = gpr + 1; ctl->value[1] = defval;
-        ctl->min = 0;
-        ctl->max = 100;
-        ctl->tlv = snd_emu10k1_db_scale1;
-        ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100;
+        if (high_res_gpr_volume) {
+                ctl->min = 0;
+                ctl->max = 0x7fffffff;
+                ctl->tlv = snd_emu10k1_db_linear;
+                ctl->translation = EMU10K1_GPR_TRANSLATION_NONE;
+        } else {
+                ctl->min = 0;
+                ctl->max = 100;
+                ctl->tlv = snd_emu10k1_db_scale1;
+                ctl->translation = EMU10K1_GPR_TRANSLATION_TABLE100;
+        }
 }
 
 static void __devinit
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index a3d638c8c1f..ba2098d20cc 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -784,6 +784,9 @@ static int read_pin_defaults(struct hda_codec *codec)
                 pin->nid = nid;
                 pin->cfg = snd_hda_codec_read(codec, nid, 0,
                                               AC_VERB_GET_CONFIG_DEFAULT, 0);
+                pin->ctrl = snd_hda_codec_read(codec, nid, 0,
+                                               AC_VERB_GET_PIN_WIDGET_CONTROL,
+                                               0);
         }
         return 0;
 }
@@ -912,15 +915,38 @@ static void restore_pincfgs(struct hda_codec *codec)
 void snd_hda_shutup_pins(struct hda_codec *codec)
 {
         int i;
+        /* don't shut up pins when unloading the driver; otherwise it breaks
+         * the default pin setup at the next load of the driver
+         */
+        if (codec->bus->shutdown)
+                return;
         for (i = 0; i < codec->init_pins.used; i++) {
                 struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
                 /* use read here for syncing after issuing each verb */
                 snd_hda_codec_read(codec, pin->nid, 0,
                                    AC_VERB_SET_PIN_WIDGET_CONTROL, 0);
         }
+        codec->pins_shutup = 1;
 }
 EXPORT_SYMBOL_HDA(snd_hda_shutup_pins);
 
+/* Restore the pin controls cleared previously via snd_hda_shutup_pins() */
+static void restore_shutup_pins(struct hda_codec *codec)
+{
+        int i;
+        if (!codec->pins_shutup)
+                return;
+        if (codec->bus->shutdown)
+                return;
+        for (i = 0; i < codec->init_pins.used; i++) {
+                struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i);
+                snd_hda_codec_write(codec, pin->nid, 0,
+                                    AC_VERB_SET_PIN_WIDGET_CONTROL,
+                                    pin->ctrl);
+        }
+        codec->pins_shutup = 0;
+}
+
 static void init_hda_cache(struct hda_cache_rec *cache,
                            unsigned int record_size);
 static void free_hda_cache(struct hda_cache_rec *cache);
@@ -2907,6 +2933,7 @@ static void hda_call_codec_resume(struct hda_codec *codec)
                             codec->afg ? codec->afg : codec->mfg,
                             AC_PWRST_D0);
         restore_pincfgs(codec); /* restore all current pin configs */
+        restore_shutup_pins(codec);
         hda_exec_init_verbs(codec);
         if (codec->patch_ops.resume)
                 codec->patch_ops.resume(codec);
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 49e939e7e5c..5991d14e1ec 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -821,6 +821,7 @@ struct hda_codec {
         unsigned int pin_amp_workaround:1; /* pin out-amp takes index
                                             * (e.g. Conexant codecs)
                                             */
+        unsigned int pins_shutup:1;     /* pins are shut up */
         unsigned int no_trigger_sense:1; /* don't trigger at pin-sensing */
 #ifdef CONFIG_SND_HDA_POWER_SAVE
         unsigned int power_on :1;       /* current (global) power-state */
@@ -897,7 +898,9 @@ void snd_hda_codec_resume_cache(struct hda_codec *codec);
 /* the struct for codec->pin_configs */
 struct hda_pincfg {
         hda_nid_t nid;
-        unsigned int cfg;
+        unsigned char ctrl;     /* current pin control value */
+        unsigned char pad;      /* reserved */
+        unsigned int cfg;       /* default configuration */
 };
 
 unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid);
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 170610e1d7d..1df25cf5ce3 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -1097,6 +1097,7 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
         struct azx *chip = dev_id;
         struct azx_dev *azx_dev;
         u32 status;
+        u8 sd_status;
         int i, ok;
 
         spin_lock(&chip->reg_lock);
@@ -1110,8 +1111,10 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id)
         for (i = 0; i < chip->num_streams; i++) {
                 azx_dev = &chip->azx_dev[i];
                 if (status & azx_dev->sd_int_sta_mask) {
+                        sd_status = azx_sd_readb(azx_dev, SD_STS);
                         azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK);
-                        if (!azx_dev->substream || !azx_dev->running)
+                        if (!azx_dev->substream || !azx_dev->running ||
+                            !(sd_status & SD_INT_COMPLETE))
                                 continue;
                         /* check whether this IRQ is really acceptable */
                         ok = azx_position_ok(chip, azx_dev);
@@ -1910,11 +1913,11 @@ static int azx_position_ok(struct azx *chip, struct azx_dev *azx_dev)
         if (WARN_ONCE(!azx_dev->period_bytes,
                       "hda-intel: zero azx_dev->period_bytes"))
                 return -1; /* this shouldn't happen! */
-        if (wallclk <= azx_dev->period_wallclk &&
+        if (wallclk < (azx_dev->period_wallclk * 5) / 4 &&
             pos % azx_dev->period_bytes > azx_dev->period_bytes / 2)
                 /* NG - it's below the first next period boundary */
                 return bdl_pos_adj[chip->dev_index] ? 0 : -1;
-        azx_dev->start_wallclk = wallclk;
+        azx_dev->start_wallclk += wallclk;
         return 1; /* OK, it's fine */
 }
 
@@ -2279,16 +2282,23 @@ static int azx_dev_free(struct snd_device *device)
  * white/black-listing for position_fix
  */
 static struct snd_pci_quirk position_fix_list[] __devinitdata = {
+        SND_PCI_QUIRK(0x1025, 0x009f, "Acer Aspire 5110", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB),
-        SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB),
+        SND_PCI_QUIRK(0x1043, 0x81b3, "ASUS", POS_FIX_LPIB),
+        SND_PCI_QUIRK(0x1043, 0x81e7, "ASUS M2V", POS_FIX_LPIB),
+        SND_PCI_QUIRK(0x104d, 0x9069, "Sony VPCS11V9E", POS_FIX_LPIB),
+        SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB),
+        SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba A100-259", POS_FIX_LPIB),
+        SND_PCI_QUIRK(0x1297, 0x3166, "Shuttle", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x1565, 0x8218, "Biostar Microtech", POS_FIX_LPIB),
+        SND_PCI_QUIRK(0x1849, 0x0888, "775Dual-VSTA", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x8086, 0x2503, "DG965OT AAD63733-203", POS_FIX_LPIB),
         SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB),
         {}
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index e863649d31f..2bf2cb5da95 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -2975,6 +2975,8 @@ static struct snd_pci_quirk cxt5066_cfg_tbl[] = {
         SND_PCI_QUIRK(0x1179, 0xff50, "Toshiba Satellite P500-PSPGSC-01800T", CXT5066_OLPC_XO_1_5),
         SND_PCI_QUIRK(0x1179, 0xffe0, "Toshiba Satellite Pro T130-15F", CXT5066_OLPC_XO_1_5),
         SND_PCI_QUIRK(0x17aa, 0x21b2, "Thinkpad X100e", CXT5066_IDEAPAD),
+        SND_PCI_QUIRK(0x17aa, 0x21b3, "Thinkpad Edge 13 (197)", CXT5066_IDEAPAD),
+        SND_PCI_QUIRK(0x17aa, 0x21b4, "Thinkpad Edge", CXT5066_IDEAPAD),
         SND_PCI_QUIRK(0x17aa, 0x3a0d, "ideapad", CXT5066_IDEAPAD),
         SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo Thinkpad", CXT5066_THINKPAD),
         {}
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 86067ee7863..2fc53961054 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -52,6 +52,10 @@ struct hdmi_spec {
          */
         struct hda_multi_out multiout;
         unsigned int codec_type;
+
+        /* misc flags */
+        /* PD bit indicates only the update, not the current state */
+        unsigned int old_pin_detect:1;
 };
 
 
@@ -616,6 +620,9 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hda_nid_t nid,
  * Unsolicited events
  */
 
+static void hdmi_present_sense(struct hda_codec *codec, hda_nid_t pin_nid,
+                               struct hdmi_eld *eld);
+
 static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
 {
         struct hdmi_spec *spec = codec->spec;
@@ -632,6 +639,12 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res)
         if (index < 0)
                 return;
 
+        if (spec->old_pin_detect) {
+                if (pind)
+                        hdmi_present_sense(codec, tag, &spec->sink_eld[index]);
+                pind = spec->sink_eld[index].monitor_present;
+        }
+
         spec->sink_eld[index].monitor_present = pind;
         spec->sink_eld[index].eld_valid = eldv;
 
diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c
index 3c10c0b149f..b0652acee9b 100644
--- a/sound/pci/hda/patch_nvhdmi.c
+++ b/sound/pci/hda/patch_nvhdmi.c
@@ -478,6 +478,7 @@ static int patch_nvhdmi_8ch_89(struct hda_codec *codec)
 
         codec->spec = spec;
         spec->codec_type = HDA_CODEC_NVIDIA_MCP89;
+        spec->old_pin_detect = 1;
 
         if (hdmi_parse_codec(codec) < 0) {
                 codec->spec = NULL;
@@ -508,6 +509,7 @@ static int patch_nvhdmi_8ch_7x(struct hda_codec *codec)
         spec->multiout.max_channels = 8;
         spec->multiout.dig_out_nid = nvhdmi_master_con_nid_7x;
         spec->codec_type = HDA_CODEC_NVIDIA_MCP7X;
+        spec->old_pin_detect = 1;
 
         codec->patch_ops = nvhdmi_patch_ops_8ch_7x;
 
@@ -528,6 +530,7 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec)
         spec->multiout.max_channels = 2;
         spec->multiout.dig_out_nid = nvhdmi_master_con_nid_7x;
         spec->codec_type = HDA_CODEC_NVIDIA_MCP7X;
+        spec->old_pin_detect = 1;
 
         codec->patch_ops = nvhdmi_patch_ops_2ch;
 
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 53538b0f999..596ea2f12cf 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -1267,6 +1267,8 @@ static int alc_auto_parse_customize_define(struct hda_codec *codec)
         unsigned nid = 0;
         struct alc_spec *spec = codec->spec;
 
+        spec->cdefine.enable_pcbeep = 1; /* assume always enabled */
+
         ass = codec->subsystem_id & 0xffff;
         if (ass != codec->bus->pci->subsystem_device && (ass & 1))
                 goto do_sku;
@@ -2547,7 +2549,7 @@ static struct snd_kcontrol_new alc_beep_mixer[] = {
 static int alc_build_controls(struct hda_codec *codec)
 {
         struct alc_spec *spec = codec->spec;
-        struct snd_kcontrol *kctl;
+        struct snd_kcontrol *kctl = NULL;
         struct snd_kcontrol_new *knew;
         int i, j, err;
         unsigned int u;
@@ -2619,16 +2621,18 @@ static int alc_build_controls(struct hda_codec *codec)
         }
 
         /* assign Capture Source enums to NID */
-        kctl = snd_hda_find_mixer_ctl(codec, "Capture Source");
-        if (!kctl)
-                kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
-        for (i = 0; kctl && i < kctl->count; i++) {
-                hda_nid_t *nids = spec->capsrc_nids;
-                if (!nids)
-                        nids = spec->adc_nids;
-                err = snd_hda_add_nid(codec, kctl, i, nids[i]);
-                if (err < 0)
-                        return err;
+        if (spec->capsrc_nids || spec->adc_nids) {
+                kctl = snd_hda_find_mixer_ctl(codec, "Capture Source");
+                if (!kctl)
+                        kctl = snd_hda_find_mixer_ctl(codec, "Input Source");
+                for (i = 0; kctl && i < kctl->count; i++) {
+                        hda_nid_t *nids = spec->capsrc_nids;
+                        if (!nids)
+                                nids = spec->adc_nids;
+                        err = snd_hda_add_nid(codec, kctl, i, nids[i]);
+                        if (err < 0)
+                                return err;
+                }
         }
         if (spec->cap_mixer) {
                 const char *kname = kctl ? kctl->id.name : NULL;
@@ -5176,8 +5180,24 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids,
 #ifdef CONFIG_SND_HDA_INPUT_BEEP
 #define set_beep_amp(spec, nid, idx, dir) \
         ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir))
+
+static struct snd_pci_quirk beep_white_list[] = {
+        SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1),
+        {}
+};
+
+static inline int has_cdefine_beep(struct hda_codec *codec)
+{
+        struct alc_spec *spec = codec->spec;
+        const struct snd_pci_quirk *q;
+        q = snd_pci_quirk_lookup(codec->bus->pci, beep_white_list);
+        if (q)
+                return q->value;
+        return spec->cdefine.enable_pcbeep;
+}
 #else
 #define set_beep_amp(spec, nid, idx, dir) /* NOP */
+#define has_cdefine_beep(codec)         0
 #endif
 
 /*
@@ -6948,7 +6968,7 @@ static struct hda_input_mux mb5_capture_source = {
         .num_items = 3,
         .items = {
                 { "Mic", 0x1 },
-                { "Line", 0x2 },
+                { "Line", 0x7 },
                 { "CD", 0x4 },
         },
 };
@@ -7025,6 +7045,14 @@ static struct hda_input_mux alc889A_mb31_capture_source = {
         },
 };
 
+static struct hda_input_mux alc889A_imac91_capture_source = {
+        .num_items = 2,
+        .items = {
+                { "Mic", 0x01 },
+                { "Line", 0x2 }, /* Not sure! */
+        },
+};
+
 /*
  * 2ch mode
  */
@@ -7461,8 +7489,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = {
         HDA_BIND_MUTE   ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT),
         HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT),
         HDA_BIND_MUTE   ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT),
-        HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
-        HDA_CODEC_MUTE  ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
+        HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x07, HDA_INPUT),
+        HDA_CODEC_MUTE  ("Line Playback Switch", 0x0b, 0x07, HDA_INPUT),
         HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT),
         HDA_CODEC_MUTE  ("Mic Playback Switch", 0x0b, 0x01, HDA_INPUT),
         HDA_CODEC_VOLUME("Line Boost", 0x15, 0x00, HDA_INPUT),
@@ -7486,15 +7514,8 @@ static struct snd_kcontrol_new alc885_macmini3_mixer[] = {
 };
 
 static struct snd_kcontrol_new alc885_imac91_mixer[] = {
-        HDA_CODEC_VOLUME("Line-Out Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
-        HDA_BIND_MUTE   ("Line-Out Playback Switch", 0x0c, 0x02, HDA_INPUT),
-        HDA_CODEC_MUTE  ("Speaker Playback Switch", 0x14, 0x00, HDA_OUTPUT),
-        HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x00, HDA_OUTPUT),
-        HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT),
-        HDA_CODEC_MUTE  ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT),
-        HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x00, HDA_INPUT),
-        HDA_CODEC_MUTE  ("Mic Playback Switch", 0x0b, 0x00, HDA_INPUT),
-        HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x00, HDA_INPUT),
+        HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0c, 0x00, HDA_OUTPUT),
+        HDA_BIND_MUTE("Speaker Playback Switch", 0x0c, 0x02, HDA_INPUT),
         { } /* end */
 };
 
@@ -7852,10 +7873,9 @@ static struct hda_verb alc885_mb5_init_verbs[] = {
         {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
         {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
 
-        {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)},
-        {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
-        {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
-        {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
+        {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0x1)},
+        {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x7)},
+        {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0x4)},
         { }
 };
 
@@ -7995,61 +8015,56 @@ static struct hda_verb alc885_mbp3_init_verbs[] = {
 
 /* iMac 9,1 */
 static struct hda_verb alc885_imac91_init_verbs[] = {
-        /* Line-Out mixer: unmute input/output amp left and right (volume = 0) */
-        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-        /* Rear mixer */
-        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
-        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
-        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
-        /* HP Pin: output 0 (0x0c) */
+        /* Internal Speaker Pin (0x0c) */
+        {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
+        {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+        {0x18, AC_VERB_SET_CONNECT_SEL, 0x00},
+        {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, (PIN_OUT | AC_PINCTL_VREF_50) },
+        {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
+        {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00},
+        /* HP Pin: Rear */
         {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},
         {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE},
         {0x14, AC_VERB_SET_CONNECT_SEL, 0x00},
-        {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN},
-        /* Internal Speakers: output 0 (0x0d) */
-        {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT},
+        {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, (ALC880_HP_EVENT | AC_USRSP_EN)},
+        /* Line in Rear */
+        {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_VREF_50},
         {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-        {0x15, AC_VERB_SET_CONNECT_SEL, 0x00},
-        /* Mic (rear) pin: input vref at 80% */
-        {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
-        {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
         /* Front Mic pin: input vref at 80% */
         {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80},
         {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-        /* Line In pin: use output 1 when in LineOut mode */
-        {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN},
-        {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},
-        {0x1a, AC_VERB_SET_CONNECT_SEL, 0x01},
-
-        /* FIXME: use matrix-type input source selection */
-        /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */
-        /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */
+        /* Rear mixer */
+        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+        {0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+        /* Line-Out mixer: unmute input/output amp left and right (volume = 0) */
+        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO},
+        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
+        {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)},
+        /* 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
         {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
         {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
         {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
         {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-        /* Input mixer2 */
+        /* 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
         {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
         {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
         {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
         {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-        /* Input mixer3 */
+        /* 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In */
         {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
         {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)},
         {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)},
         {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)},
-        /* ADC1: mute amp left and right */
+        /* 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
         {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
         {0x07, AC_VERB_SET_CONNECT_SEL, 0x00},
-        /* ADC2: mute amp left and right */
+        /* 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
         {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
         {0x08, AC_VERB_SET_CONNECT_SEL, 0x00},
-        /* ADC3: mute amp left and right */
+        /* 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In */
         {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)},
         {0x09, AC_VERB_SET_CONNECT_SEL, 0x00},
-
         { }
 };
 
@@ -8118,7 +8133,7 @@ static void alc885_imac91_setup(struct hda_codec *codec)
         struct alc_spec *spec = codec->spec;
 
         spec->autocfg.hp_pins[0] = 0x14;
-        spec->autocfg.speaker_pins[0] = 0x15;
+        spec->autocfg.speaker_pins[0] = 0x18;
         spec->autocfg.speaker_pins[1] = 0x1a;
 }
 
@@ -9480,11 +9495,16 @@ static struct snd_pci_quirk alc882_ssid_cfg_tbl[] = {
         SND_PCI_QUIRK(0x106b, 0x1000, "iMac 24", ALC885_IMAC24),
         SND_PCI_QUIRK(0x106b, 0x2800, "AppleTV", ALC885_IMAC24),
         SND_PCI_QUIRK(0x106b, 0x2c00, "MacbookPro rev3", ALC885_MBP3),
+        SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889A_MB31),
+        SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_ASUS_A7M),
+        SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC885_MBP3),
+        SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC885_MBA21),
         SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889A_MB31),
         SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC885_MBP3),
         SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_IMAC24),
         SND_PCI_QUIRK(0x106b, 0x4900, "iMac 9,1 Aluminum", ALC885_IMAC91),
         SND_PCI_QUIRK(0x106b, 0x3f00, "Macbook 5,1", ALC885_MB5),
+        SND_PCI_QUIRK(0x106b, 0x4a00, "Macbook 5,2", ALC885_MB5),
         /* FIXME: HP jack sense seems not working for MBP 5,1 or 5,2,
          * so apparently no perfect solution yet
          */
@@ -9627,14 +9647,14 @@ static struct alc_config_preset alc882_presets[] = {
                 .init_hook = alc885_imac24_init_hook,
         },
         [ALC885_IMAC91] = {
-                .mixers = { alc885_imac91_mixer, alc882_chmode_mixer },
+                .mixers = {alc885_imac91_mixer},
                 .init_verbs = { alc885_imac91_init_verbs,
                                 alc880_gpio1_init_verbs },
                 .num_dacs = ARRAY_SIZE(alc882_dac_nids),
                 .dac_nids = alc882_dac_nids,
-                .channel_mode = alc885_mbp_4ch_modes,
-                .num_channel_mode = ARRAY_SIZE(alc885_mbp_4ch_modes),
-                .input_mux = &alc882_capture_source,
+                .channel_mode = alc885_mba21_ch_modes,
+                .num_channel_mode = ARRAY_SIZE(alc885_mba21_ch_modes),
+                .input_mux = &alc889A_imac91_capture_source,
                 .dig_out_nid = ALC882_DIGOUT_NID,
                 .dig_in_nid = ALC882_DIGIN_NID,
                 .unsol_event = alc_automute_amp_unsol_event,
@@ -10562,10 +10582,12 @@ static int patch_alc882(struct hda_codec *codec)
                 }
         }
 
-        err = snd_hda_attach_beep_device(codec, 0x1);
-        if (err < 0) {
-                alc_free(codec);
-                return err;
+        if (has_cdefine_beep(codec)) {
+                err = snd_hda_attach_beep_device(codec, 0x1);
+                if (err < 0) {
+                        alc_free(codec);
+                        return err;
+                }
         }
 
         if (board_config != ALC882_AUTO)
@@ -10615,7 +10637,7 @@ static int patch_alc882(struct hda_codec *codec)
 
         set_capture_mixer(codec);
 
-        if (spec->cdefine.enable_pcbeep)
+        if (has_cdefine_beep(codec))
                 set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
         if (board_config == ALC882_AUTO)
@@ -12431,7 +12453,7 @@ static int patch_alc262(struct hda_codec *codec)
                 }
         }
 
-        if (!spec->no_analog) {
+        if (!spec->no_analog && has_cdefine_beep(codec)) {
                 err = snd_hda_attach_beep_device(codec, 0x1);
                 if (err < 0) {
                         alc_free(codec);
@@ -12482,7 +12504,7 @@ static int patch_alc262(struct hda_codec *codec)
         }
         if (!spec->cap_mixer && !spec->no_analog)
                 set_capture_mixer(codec);
-        if (!spec->no_analog && spec->cdefine.enable_pcbeep)
+        if (!spec->no_analog && has_cdefine_beep(codec))
                 set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
 
         spec->vmaster_nid = 0x0c;
@@ -14454,10 +14476,12 @@ static int patch_alc269(struct hda_codec *codec)
                 }
         }
 
-        err = snd_hda_attach_beep_device(codec, 0x1);
-        if (err < 0) {
-                alc_free(codec);
-                return err;
+        if (has_cdefine_beep(codec)) {
+                err = snd_hda_attach_beep_device(codec, 0x1);
+                if (err < 0) {
+                        alc_free(codec);
+                        return err;
+                }
         }
 
         if (board_config != ALC269_AUTO)
@@ -14490,7 +14514,7 @@ static int patch_alc269(struct hda_codec *codec)
 
         if (!spec->cap_mixer)
                 set_capture_mixer(codec);
-        if (spec->cdefine.enable_pcbeep)
+        if (has_cdefine_beep(codec))
                 set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
 
         if (board_config == ALC269_AUTO)
@@ -18687,10 +18711,12 @@ static int patch_alc662(struct hda_codec *codec)
                 }
         }
 
-        err = snd_hda_attach_beep_device(codec, 0x1);
-        if (err < 0) {
-                alc_free(codec);
-                return err;
+        if (has_cdefine_beep(codec)) {
+                err = snd_hda_attach_beep_device(codec, 0x1);
+                if (err < 0) {
+                        alc_free(codec);
+                        return err;
+                }
         }
 
         if (board_config != ALC662_AUTO)
@@ -18712,7 +18738,7 @@ static int patch_alc662(struct hda_codec *codec)
         if (!spec->cap_mixer)
                 set_capture_mixer(codec);
 
-        if (spec->cdefine.enable_pcbeep) {
+        if (has_cdefine_beep(codec)) {
                 switch (codec->vendor_id) {
                 case 0x10ec0662:
                         set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index a0e06d82da1..f1e7babd692 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -2078,12 +2078,12 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = {
         SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2000,
                            "Intel D965", STAC_D965_3ST),
         /* Dell 3 stack systems */
-        SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01f7, "Dell XPS M1730", STAC_DELL_3ST),
         SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01dd, "Dell Dimension E520", STAC_DELL_3ST),
         SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01ed, "Dell     ", STAC_DELL_3ST),
         SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01f4, "Dell     ", STAC_DELL_3ST),
         /* Dell 3 stack systems with verb table in BIOS */
         SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01f3, "Dell Inspiron 1420", STAC_DELL_BIOS),
+        SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x01f7, "Dell XPS M1730", STAC_DELL_BIOS),
         SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x0227, "Dell Vostro 1400  ", STAC_DELL_BIOS),
         SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x022e, "Dell     ", STAC_DELL_BIOS),
         SND_PCI_QUIRK(PCI_VENDOR_ID_DELL,  0x022f, "Dell Inspiron 1525", STAC_DELL_BIOS),
diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c
index 495be6e7193..24454c98d0e 100644
--- a/sound/soc/au1x/psc-i2s.c
+++ b/sound/soc/au1x/psc-i2s.c
@@ -300,7 +300,7 @@ struct snd_soc_dai au1xpsc_i2s_dai = {
 };
 EXPORT_SYMBOL(au1xpsc_i2s_dai);
 
-static int __init au1xpsc_i2s_drvprobe(struct platform_device *pdev)
+static int __devinit au1xpsc_i2s_drvprobe(struct platform_device *pdev)
 {
         struct resource *r;
         unsigned long sel;
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 31ac5538fe7..5da30eb6ad0 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -83,8 +83,8 @@ config SND_SOC_ALL_CODECS
 
 config SND_SOC_WM_HUBS
         tristate
-        default y if SND_SOC_WM8993=y
-        default m if SND_SOC_WM8993=m
+        default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y
+        default m if SND_SOC_WM8993=m || SND_SOC_WM8994=m
 
 config SND_SOC_AC97_CODEC
         tristate
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 8ae20208e7b..0221ca79b3a 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 7f5d080536a..8f294066b0e 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/wm8727.c b/sound/soc/codecs/wm8727.c
index 1072621e93f..9d1df262813 100644
--- a/sound/soc/codecs/wm8727.c
+++ b/sound/soc/codecs/wm8727.c
@@ -127,6 +127,8 @@ static __devinit int wm8727_platform_probe(struct platform_device *pdev)
                 goto err_codec;
         }
 
+        return 0;
+
 err_codec:
         snd_soc_unregister_codec(codec);
 err:
diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c
index 7e4a627b4c7..4e212ed62ea 100644
--- a/sound/soc/codecs/wm8776.c
+++ b/sound/soc/codecs/wm8776.c
@@ -94,7 +94,6 @@ SOC_DAPM_SINGLE("Bypass Switch", WM8776_OUTMUX, 2, 1, 0),
 
 static const struct snd_soc_dapm_widget wm8776_dapm_widgets[] = {
 SND_SOC_DAPM_INPUT("AUX"),
-SND_SOC_DAPM_INPUT("AUX"),
 
 SND_SOC_DAPM_INPUT("AIN1"),
 SND_SOC_DAPM_INPUT("AIN2"),
diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c
index 0417dae32e6..19ad590ca0b 100644
--- a/sound/soc/codecs/wm8988.c
+++ b/sound/soc/codecs/wm8988.c
@@ -885,7 +885,6 @@ static int wm8988_register(struct wm8988_priv *wm8988,
         ret = snd_soc_register_dai(&wm8988_dai);
         if (ret != 0) {
                 dev_err(codec->dev, "Failed to register DAI: %d\n", ret);
-                snd_soc_unregister_codec(codec);
                 goto err_codec;
         }
 
diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c
index 7b536d923ea..c018772cc43 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/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c
index 79f0f4ad242..d3955096d87 100644
--- a/sound/soc/davinci/davinci-mcasp.c
+++ b/sound/soc/davinci/davinci-mcasp.c
@@ -612,7 +612,6 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream)
                                                                 NUMDMA_MASK);
                 mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL,
                                 ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK);
-                mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE);
         }
 
         if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) {
@@ -623,7 +622,6 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream)
                                                                 NUMDMA_MASK);
                 mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL,
                                 ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK);
-                mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE);
         }
 }
 
diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h
index 22208b373fb..e1ec6d91ea3 100644
--- a/sound/soc/fsl/mpc5200_dma.h
+++ b/sound/soc/fsl/mpc5200_dma.h
@@ -73,7 +73,7 @@ struct psc_dma {
 };
 
 /* Utility for retrieving psc_dma_stream structure from a substream */
-inline struct psc_dma_stream *
+static inline struct psc_dma_stream *
 to_psc_dma_stream(struct snd_pcm_substream *substream, struct psc_dma *psc_dma)
 {
         if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig
index eba9b9d257a..252defea93b 100644
--- a/sound/soc/imx/Kconfig
+++ b/sound/soc/imx/Kconfig
@@ -13,9 +13,18 @@ config SND_MXC_SOC_SSI
 
 config SND_MXC_SOC_WM1133_EV1
         tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted"
-        depends on SND_IMX_SOC && EXPERIMENTAL
+        depends on SND_IMX_SOC && MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL
         select SND_SOC_WM8350
         select SND_MXC_SOC_SSI
         help
           Enable support for audio on the i.MX31ADS with the WM1133-EV1
           PMIC board with WM8835x fitted.
+
+config SND_SOC_PHYCORE_AC97
+        tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards"
+        depends on MACH_PCM043 || MACH_PCA100
+        select SND_MXC_SOC_SSI
+        select SND_SOC_WM9712
+        help
+          Say Y if you want to add support for SoC audio on Phytec phyCORE
+          and phyCARD boards in AC97 mode
diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c
index 2b31ac673ea..05f19c9284f 100644
--- a/sound/soc/imx/imx-pcm-dma-mx2.c
+++ b/sound/soc/imx/imx-pcm-dma-mx2.c
@@ -73,7 +73,8 @@ static void snd_imx_dma_err_callback(int channel, void *data, int err)
 {
         struct snd_pcm_substream *substream = data;
         struct snd_soc_pcm_runtime *rtd = substream->private_data;
-        struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data;
+        struct imx_pcm_dma_params *dma_params = 
+                snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
         struct snd_pcm_runtime *runtime = substream->runtime;
         struct imx_pcm_runtime_data *iprtd = runtime->private_data;
         int ret;
@@ -102,7 +103,7 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream)
         struct imx_pcm_runtime_data *iprtd = runtime->private_data;
         int ret;
 
-        dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream);
+        dma_params = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 
         iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH);
         if (iprtd->dma < 0) {
@@ -212,7 +213,7 @@ static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream)
         struct imx_pcm_runtime_data *iprtd = runtime->private_data;
         int err;
 
-        dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream);
+        dma_params = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream);
 
         iprtd->substream = substream;
         iprtd->buf = (unsigned int *)substream->dma_buffer.area;
diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c
index 1941a357e8c..d256f5f313b 100644
--- a/sound/soc/pxa/spitz.c
+++ b/sound/soc/pxa/spitz.c
@@ -328,38 +328,6 @@ static struct snd_soc_device spitz_snd_devdata = {
         .codec_dev = &soc_codec_dev_wm8750,
 };
 
-/*
- * FIXME: This is a temporary bodge to avoid cross-tree merge issues.
- * New drivers should register the wm8750 I2C device in the machine
- * setup code (under arch/arm for ARM systems).
- */
-static int wm8750_i2c_register(void)
-{
-        struct i2c_board_info info;
-        struct i2c_adapter *adapter;
-        struct i2c_client *client;
-
-        memset(&info, 0, sizeof(struct i2c_board_info));
-        info.addr = 0x1b;
-        strlcpy(info.type, "wm8750", I2C_NAME_SIZE);
-
-        adapter = i2c_get_adapter(0);
-        if (!adapter) {
-                printk(KERN_ERR "can't get i2c adapter 0\n");
-                return -ENODEV;
-        }
-
-        client = i2c_new_device(adapter, &info);
-        i2c_put_adapter(adapter);
-        if (!client) {
-                printk(KERN_ERR "can't add i2c device at 0x%x\n",
-                (unsigned int)info.addr);
-                return -ENODEV;
-        }
-
-        return 0;
-}
-
 static struct platform_device *spitz_snd_device;
 
 static int __init spitz_init(void)
@@ -369,10 +337,6 @@ static int __init spitz_init(void)
         if (!(machine_is_spitz() || machine_is_borzoi() || machine_is_akita()))
                 return -ENODEV;
 
-        ret = wm8750_i2c_setup();
-        if (ret != 0)
-                return ret;
-
         spitz_snd_device = platform_device_alloc("soc-audio", -1);
         if (!spitz_snd_device)
                 return -ENOMEM;
diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c
index 3396a0db06b..ec4acac49eb 100644
--- a/sound/soc/sh/fsi.c
+++ b/sound/soc/sh/fsi.c
@@ -683,20 +683,15 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
 
         /* clock inversion (CKG2) */
         data = 0;
-        switch (SH_FSI_INVERSION_MASK & flags) {
-        case SH_FSI_LRM_INV:
-                data = 1 << 12;
-                break;
-        case SH_FSI_BRM_INV:
-                data = 1 << 8;
-                break;
-        case SH_FSI_LRS_INV:
-                data = 1 << 4;
-                break;
-        case SH_FSI_BRS_INV:
-                data = 1 << 0;
-                break;
-        }
+        if (SH_FSI_LRM_INV & flags)
+                data |= 1 << 12;
+        if (SH_FSI_BRM_INV & flags)
+                data |= 1 << 8;
+        if (SH_FSI_LRS_INV & flags)
+                data |= 1 << 4;
+        if (SH_FSI_BRS_INV & flags)
+                data |= 1 << 0;
+
         fsi_reg_write(fsi, CKG2, data);
 
         /* do fmt, di fmt */
@@ -726,15 +721,15 @@ static int fsi_dai_startup(struct snd_pcm_substream *substream,
                 break;
         case SH_FSI_FMT_TDM:
                 msg = "TDM";
-                data = CR_FMT(CR_TDM) | (fsi->chan - 1);
                 fsi->chan = is_play ?
                         SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags);
+                data = CR_FMT(CR_TDM) | (fsi->chan - 1);
                 break;
         case SH_FSI_FMT_TDM_DELAY:
                 msg = "TDM Delay";
-                data = CR_FMT(CR_TDM_D) | (fsi->chan - 1);
                 fsi->chan = is_play ?
                         SH_FSI_GET_CH_O(flags) : SH_FSI_GET_CH_I(flags);
+                data = CR_FMT(CR_TDM_D) | (fsi->chan - 1);
                 break;
         default:
                 dev_err(dai->dev, "unknown format.\n");
diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c
index d86ee1bfc03..eeed5edd722 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/spi/at73c213.c b/sound/spi/at73c213.c
index 4c7b051f9d1..1bc56b2b94e 100644
--- a/sound/spi/at73c213.c
+++ b/sound/spi/at73c213.c
@@ -69,7 +69,6 @@ struct snd_at73c213 {
         int                             irq;
         int                             period;
         unsigned long                   bitrate;
-        struct clk                      *bitclk;
         struct ssc_device               *ssc;
         struct spi_device               *spi;
         u8                              spi_wbuffer[2];
diff --git a/sound/usb/Makefile b/sound/usb/Makefile
index e7ac7f493a8..1e362bf8834 100644
--- a/sound/usb/Makefile
+++ b/sound/usb/Makefile
@@ -11,7 +11,8 @@ snd-usb-audio-objs := 	card.o \
                         endpoint.o \
                         urb.o \
                         pcm.o \
-                        helper.o
+                        helper.o \
+                        clock.o
 
 snd-usbmidi-lib-objs := midi.o
 
diff --git a/sound/usb/caiaq/control.c b/sound/usb/caiaq/control.c
index 36ed703a741..91c804cd278 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 80527182767..cdfb856bddd 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 8bbfbfd4c65..dcb620796d9 100644
--- a/sound/usb/caiaq/input.c
+++ b/sound/usb/caiaq/input.c
@@ -171,7 +171,7 @@ static void snd_caiaq_input_read_analog(struct snd_usb_caiaqdev *dev,
                 input_report_abs(input_dev, ABS_HAT0Y, (buf[4] << 8)  | buf[5]);
                 input_report_abs(input_dev, ABS_HAT1X, (buf[12] << 8) | buf[13]);
                 input_report_abs(input_dev, ABS_HAT1Y, (buf[2] << 8)  | buf[3]);
-                input_report_abs(input_dev, ABS_HAT2X, (buf[15] << 8) | buf[15]);
+                input_report_abs(input_dev, ABS_HAT2X, (buf[14] << 8) | buf[15]);
                 input_report_abs(input_dev, ABS_HAT2Y, (buf[0] << 8)  | buf[1]);
                 input_report_abs(input_dev, ABS_HAT3X, (buf[10] << 8) | buf[11]);
                 input_report_abs(input_dev, ABS_HAT3Y, (buf[6] << 8)  | buf[7]);
diff --git a/sound/usb/card.c b/sound/usb/card.c
index da1346bd485..7a8ac1d81be 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -236,7 +236,6 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
         }
 
         case UAC_VERSION_2: {
-                struct uac_clock_source_descriptor *cs;
                 struct usb_interface_assoc_descriptor *assoc =
                         usb_ifnum_to_if(dev, ctrlif)->intf_assoc;
 
@@ -245,21 +244,6 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
                         return -EINVAL;
                 }
 
-                /* FIXME: for now, we expect there is at least one clock source
-                 * descriptor and we always take the first one.
-                 * We should properly support devices with multiple clock sources,
-                 * clock selectors and sample rate conversion units. */
-
-                cs = snd_usb_find_csint_desc(host_iface->extra, host_iface->extralen,
-                                                NULL, UAC2_CLOCK_SOURCE);
-
-                if (!cs) {
-                        snd_printk(KERN_ERR "CLOCK_SOURCE descriptor not found\n");
-                        return -EINVAL;
-                }
-
-                chip->clock_id = cs->bClockID;
-
                 for (i = 0; i < assoc->bInterfaceCount; i++) {
                         int intf = assoc->bFirstInterface + i;
 
@@ -481,6 +465,8 @@ static void *snd_usb_audio_probe(struct usb_device *dev,
                         goto __error;
         }
 
+        chip->ctrl_intf = alts;
+
         if (err > 0) {
                 /* create normal USB audio interfaces */
                 if (snd_usb_create_streams(chip, ifnum) < 0 ||
diff --git a/sound/usb/card.h b/sound/usb/card.h
index ed92420c109..1febf2f2375 100644
--- a/sound/usb/card.h
+++ b/sound/usb/card.h
@@ -25,6 +25,7 @@ struct audioformat {
         unsigned int rate_min, rate_max;        /* min/max rates */
         unsigned int nr_rates;          /* number of rate table entries */
         unsigned int *rate_table;       /* rate table */
+        unsigned char clock;            /* associated clock */
 };
 
 struct snd_usb_substream;
diff --git a/sound/usb/clock.c b/sound/usb/clock.c
new file mode 100644
index 00000000000..b5855114667
--- /dev/null
+++ b/sound/usb/clock.c
@@ -0,0 +1,315 @@
+/*
+ *   Clock domain and sample rate management functions
+ *
+ *   This program is free software; you can redistribute it and/or modify
+ *   it under the terms of the GNU General Public License as published by
+ *   the Free Software Foundation; either version 2 of the License, or
+ *   (at your option) any later version.
+ *
+ *   This program is distributed in the hope that it will be useful,
+ *   but WITHOUT ANY WARRANTY; without even the implied warranty of
+ *   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ *   GNU General Public License for more details.
+ *
+ *   You should have received a copy of the GNU General Public License
+ *   along with this program; if not, write to the Free Software
+ *   Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+#include <linux/bitops.h>
+#include <linux/init.h>
+#include <linux/list.h>
+#include <linux/slab.h>
+#include <linux/string.h>
+#include <linux/usb.h>
+#include <linux/moduleparam.h>
+#include <linux/mutex.h>
+#include <linux/usb/audio.h>
+#include <linux/usb/audio-v2.h>
+
+#include <sound/core.h>
+#include <sound/info.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+
+#include "usbaudio.h"
+#include "card.h"
+#include "midi.h"
+#include "mixer.h"
+#include "proc.h"
+#include "quirks.h"
+#include "endpoint.h"
+#include "helper.h"
+#include "debug.h"
+#include "pcm.h"
+#include "urb.h"
+#include "format.h"
+
+static struct uac_clock_source_descriptor *
+        snd_usb_find_clock_source(struct usb_host_interface *ctrl_iface,
+                                  int clock_id)
+{
+        struct uac_clock_source_descriptor *cs = NULL;
+
+        while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra,
+                                             ctrl_iface->extralen,
+                                             cs, UAC2_CLOCK_SOURCE))) {
+                if (cs->bClockID == clock_id)
+                        return cs;
+        }
+
+        return NULL;
+}
+
+static struct uac_clock_selector_descriptor *
+        snd_usb_find_clock_selector(struct usb_host_interface *ctrl_iface,
+                                    int clock_id)
+{
+        struct uac_clock_selector_descriptor *cs = NULL;
+
+        while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra,
+                                             ctrl_iface->extralen,
+                                             cs, UAC2_CLOCK_SELECTOR))) {
+                if (cs->bClockID == clock_id)
+                        return cs;
+        }
+
+        return NULL;
+}
+
+static struct uac_clock_multiplier_descriptor *
+        snd_usb_find_clock_multiplier(struct usb_host_interface *ctrl_iface,
+                                      int clock_id)
+{
+        struct uac_clock_multiplier_descriptor *cs = NULL;
+
+        while ((cs = snd_usb_find_csint_desc(ctrl_iface->extra,
+                                             ctrl_iface->extralen,
+                                             cs, UAC2_CLOCK_MULTIPLIER))) {
+                if (cs->bClockID == clock_id)
+                        return cs;
+        }
+
+        return NULL;
+}
+
+static int uac_clock_selector_get_val(struct snd_usb_audio *chip, int selector_id)
+{
+        unsigned char buf;
+        int ret;
+
+        ret = snd_usb_ctl_msg(chip->dev, usb_rcvctrlpipe(chip->dev, 0),
+                              UAC2_CS_CUR,
+                              USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
+                              UAC2_CX_CLOCK_SELECTOR << 8,
+                              snd_usb_ctrl_intf(chip) | (selector_id << 8),
+                              &buf, sizeof(buf), 1000);
+
+        if (ret < 0)
+                return ret;
+
+        return buf;
+}
+
+static bool uac_clock_source_is_valid(struct snd_usb_audio *chip, int source_id)
+{
+        int err;
+        unsigned char data;
+        struct usb_device *dev = chip->dev;
+
+        err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
+                              USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+                              UAC2_CS_CONTROL_CLOCK_VALID << 8,
+                              snd_usb_ctrl_intf(chip) | (source_id << 8),
+                              &data, sizeof(data), 1000);
+
+        if (err < 0) {
+                snd_printk(KERN_WARNING "%s(): cannot get clock validity for id %d\n",
+                           __func__, source_id);
+                return err;
+        }
+
+        return !!data;
+}
+
+/* Try to find the clock source ID of a given clock entity */
+
+static int __uac_clock_find_source(struct snd_usb_audio *chip,
+                                   struct usb_host_interface *host_iface,
+                                   int entity_id, unsigned long *visited)
+{
+        struct uac_clock_source_descriptor *source;
+        struct uac_clock_selector_descriptor *selector;
+        struct uac_clock_multiplier_descriptor *multiplier;
+
+        entity_id &= 0xff;
+
+        if (test_and_set_bit(entity_id, visited)) {
+                snd_printk(KERN_WARNING
+                        "%s(): recursive clock topology detected, id %d.\n",
+                        __func__, entity_id);
+                return -EINVAL;
+        }
+
+        /* first, see if the ID we're looking for is a clock source already */
+        source = snd_usb_find_clock_source(host_iface, entity_id);
+        if (source)
+                return source->bClockID;
+
+        selector = snd_usb_find_clock_selector(host_iface, entity_id);
+        if (selector) {
+                int ret;
+
+                /* the entity ID we are looking for is a selector.
+                 * find out what it currently selects */
+                ret = uac_clock_selector_get_val(chip, selector->bClockID);
+                if (ret < 0)
+                        return ret;
+
+                if (ret > selector->bNrInPins || ret < 1) {
+                        printk(KERN_ERR
+                                "%s(): selector reported illegal value, id %d, ret %d\n",
+                                __func__, selector->bClockID, ret);
+
+                        return -EINVAL;
+                }
+
+                return __uac_clock_find_source(chip, host_iface,
+                                               selector->baCSourceID[ret-1],
+                                               visited);
+        }
+
+        /* FIXME: multipliers only act as pass-thru element for now */
+        multiplier = snd_usb_find_clock_multiplier(host_iface, entity_id);
+        if (multiplier)
+                return __uac_clock_find_source(chip, host_iface,
+                                               multiplier->bCSourceID, visited);
+
+        return -EINVAL;
+}
+
+int snd_usb_clock_find_source(struct snd_usb_audio *chip,
+                              struct usb_host_interface *host_iface,
+                              int entity_id)
+{
+        DECLARE_BITMAP(visited, 256);
+        memset(visited, 0, sizeof(visited));
+        return __uac_clock_find_source(chip, host_iface, entity_id, visited);
+}
+
+static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface,
+                              struct usb_host_interface *alts,
+                              struct audioformat *fmt, int rate)
+{
+        struct usb_device *dev = chip->dev;
+        unsigned int ep;
+        unsigned char data[3];
+        int err, crate;
+
+        ep = get_endpoint(alts, 0)->bEndpointAddress;
+
+        /* if endpoint doesn't have sampling rate control, bail out */
+        if (!(fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE)) {
+                snd_printk(KERN_WARNING "%d:%d:%d: endpoint lacks sample rate attribute bit, cannot set.\n",
+                                   dev->devnum, iface, fmt->altsetting);
+                return 0;
+        }
+
+        data[0] = rate;
+        data[1] = rate >> 8;
+        data[2] = rate >> 16;
+        if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR,
+                                   USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT,
+                                   UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep,
+                                   data, sizeof(data), 1000)) < 0) {
+                snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n",
+                           dev->devnum, iface, fmt->altsetting, rate, ep);
+                return err;
+        }
+
+        if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR,
+                                   USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_IN,
+                                   UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep,
+                                   data, sizeof(data), 1000)) < 0) {
+                snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n",
+                           dev->devnum, iface, fmt->altsetting, ep);
+                return 0; /* some devices don't support reading */
+        }
+
+        crate = data[0] | (data[1] << 8) | (data[2] << 16);
+        if (crate != rate) {
+                snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate);
+                // runtime->rate = crate;
+        }
+
+        return 0;
+}
+
+static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
+                              struct usb_host_interface *alts,
+                              struct audioformat *fmt, int rate)
+{
+        struct usb_device *dev = chip->dev;
+        unsigned char data[4];
+        int err, crate;
+        int clock = snd_usb_clock_find_source(chip, chip->ctrl_intf, fmt->clock);
+
+        if (clock < 0)
+                return clock;
+
+        if (!uac_clock_source_is_valid(chip, clock)) {
+                snd_printk(KERN_ERR "%d:%d:%d: clock source %d is not valid, cannot use\n",
+                           dev->devnum, iface, fmt->altsetting, clock);
+                return -ENXIO;
+        }
+
+        data[0] = rate;
+        data[1] = rate >> 8;
+        data[2] = rate >> 16;
+        data[3] = rate >> 24;
+        if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR,
+                                   USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT,
+                                   UAC2_CS_CONTROL_SAM_FREQ << 8,
+                                   snd_usb_ctrl_intf(chip) | (clock << 8),
+                                   data, sizeof(data), 1000)) < 0) {
+                snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2)\n",
+                           dev->devnum, iface, fmt->altsetting, rate);
+                return err;
+        }
+
+        if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
+                                   USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+                                   UAC2_CS_CONTROL_SAM_FREQ << 8,
+                                   snd_usb_ctrl_intf(chip) | (clock << 8),
+                                   data, sizeof(data), 1000)) < 0) {
+                snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
+                           dev->devnum, iface, fmt->altsetting);
+                return err;
+        }
+
+        crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
+        if (crate != rate)
+                snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate);
+
+        return 0;
+}
+
+int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface,
+                             struct usb_host_interface *alts,
+                             struct audioformat *fmt, int rate)
+{
+        struct usb_interface_descriptor *altsd = get_iface_desc(alts);
+
+        switch (altsd->bInterfaceProtocol) {
+        case UAC_VERSION_1:
+                return set_sample_rate_v1(chip, iface, alts, fmt, rate);
+
+        case UAC_VERSION_2:
+                return set_sample_rate_v2(chip, iface, alts, fmt, rate);
+        }
+
+        return -EINVAL;
+}
+
diff --git a/sound/usb/clock.h b/sound/usb/clock.h
new file mode 100644
index 00000000000..beb253684e2
--- /dev/null
+++ b/sound/usb/clock.h
@@ -0,0 +1,12 @@
+#ifndef __USBAUDIO_CLOCK_H
+#define __USBAUDIO_CLOCK_H
+
+int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface,
+                             struct usb_host_interface *alts,
+                             struct audioformat *fmt, int rate);
+
+int snd_usb_clock_find_source(struct snd_usb_audio *chip,
+                              struct usb_host_interface *host_iface,
+                              int entity_id);
+
+#endif /* __USBAUDIO_CLOCK_H */
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index ef07a6d0dd5..6f6596cf2b1 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -149,6 +149,79 @@ int snd_usb_add_audio_endpoint(struct snd_usb_audio *chip, int stream, struct au
         return 0;
 }
 
+static int parse_uac_endpoint_attributes(struct snd_usb_audio *chip,
+                                         struct usb_host_interface *alts,
+                                         int protocol, int iface_no)
+{
+        /* parsed with a v1 header here. that's ok as we only look at the
+         * header first which is the same for both versions */
+        struct uac_iso_endpoint_descriptor *csep;
+        struct usb_interface_descriptor *altsd = get_iface_desc(alts);
+        int attributes = 0;
+
+        csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT);
+
+        /* Creamware Noah has this descriptor after the 2nd endpoint */
+        if (!csep && altsd->bNumEndpoints >= 2)
+                csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT);
+
+        if (!csep || csep->bLength < 7 ||
+            csep->bDescriptorSubtype != UAC_EP_GENERAL) {
+                snd_printk(KERN_WARNING "%d:%u:%d : no or invalid"
+                           " class specific endpoint descriptor\n",
+                           chip->dev->devnum, iface_no,
+                           altsd->bAlternateSetting);
+                return 0;
+        }
+
+        if (protocol == UAC_VERSION_1) {
+                attributes = csep->bmAttributes;
+        } else {
+                struct uac2_iso_endpoint_descriptor *csep2 =
+                        (struct uac2_iso_endpoint_descriptor *) csep;
+
+                attributes = csep->bmAttributes & UAC_EP_CS_ATTR_FILL_MAX;
+
+                /* emulate the endpoint attributes of a v1 device */
+                if (csep2->bmControls & UAC2_CONTROL_PITCH)
+                        attributes |= UAC_EP_CS_ATTR_PITCH_CONTROL;
+        }
+
+        return attributes;
+}
+
+static struct uac2_input_terminal_descriptor *
+        snd_usb_find_input_terminal_descriptor(struct usb_host_interface *ctrl_iface,
+                                               int terminal_id)
+{
+        struct uac2_input_terminal_descriptor *term = NULL;
+
+        while ((term = snd_usb_find_csint_desc(ctrl_iface->extra,
+                                               ctrl_iface->extralen,
+                                               term, UAC_INPUT_TERMINAL))) {
+                if (term->bTerminalID == terminal_id)
+                        return term;
+        }
+
+        return NULL;
+}
+
+static struct uac2_output_terminal_descriptor *
+        snd_usb_find_output_terminal_descriptor(struct usb_host_interface *ctrl_iface,
+                                                int terminal_id)
+{
+        struct uac2_output_terminal_descriptor *term = NULL;
+
+        while ((term = snd_usb_find_csint_desc(ctrl_iface->extra,
+                                               ctrl_iface->extralen,
+                                               term, UAC_OUTPUT_TERMINAL))) {
+                if (term->bTerminalID == terminal_id)
+                        return term;
+        }
+
+        return NULL;
+}
+
 int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
 {
         struct usb_device *dev;
@@ -158,8 +231,8 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
         int i, altno, err, stream;
         int format = 0, num_channels = 0;
         struct audioformat *fp = NULL;
-        unsigned char *fmt, *csep;
-        int num, protocol;
+        int num, protocol, clock = 0;
+        struct uac_format_type_i_continuous_descriptor *fmt;
 
         dev = chip->dev;
 
@@ -222,6 +295,8 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
                 }
 
                 case UAC_VERSION_2: {
+                        struct uac2_input_terminal_descriptor *input_term;
+                        struct uac2_output_terminal_descriptor *output_term;
                         struct uac_as_header_descriptor_v2 *as =
                                 snd_usb_find_csint_desc(alts->extra, alts->extralen, NULL, UAC_AS_GENERAL);
 
@@ -240,7 +315,25 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
                         num_channels = as->bNrChannels;
                         format = le32_to_cpu(as->bmFormats);
 
-                        break;
+                        /* lookup the terminal associated to this interface
+                         * to extract the clock */
+                        input_term = snd_usb_find_input_terminal_descriptor(chip->ctrl_intf,
+                                                                            as->bTerminalLink);
+                        if (input_term) {
+                                clock = input_term->bCSourceID;
+                                break;
+                        }
+
+                        output_term = snd_usb_find_output_terminal_descriptor(chip->ctrl_intf,
+                                                                              as->bTerminalLink);
+                        if (output_term) {
+                                clock = output_term->bCSourceID;
+                                break;
+                        }
+
+                        snd_printk(KERN_ERR "%d:%u:%d : bogus bTerminalLink %d\n",
+                                   dev->devnum, iface_no, altno, as->bTerminalLink);
+                        continue;
                 }
 
                 default:
@@ -256,8 +349,8 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
                                    dev->devnum, iface_no, altno);
                         continue;
                 }
-                if (((protocol == UAC_VERSION_1) && (fmt[0] < 8)) ||
-                    ((protocol == UAC_VERSION_2) && (fmt[0] != 6))) {
+                if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) ||
+                    ((protocol == UAC_VERSION_2) && (fmt->bLength != 6))) {
                         snd_printk(KERN_ERR "%d:%u:%d : invalid UAC_FORMAT_TYPE desc\n",
                                    dev->devnum, iface_no, altno);
                         continue;
@@ -268,7 +361,9 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
                  * with the previous one, except for a larger packet size, but
                  * is actually a mislabeled two-channel setting; ignore it.
                  */
-                if (fmt[4] == 1 && fmt[5] == 2 && altno == 2 && num == 3 &&
+                if (fmt->bNrChannels == 1 &&
+                    fmt->bSubframeSize == 2 &&
+                    altno == 2 && num == 3 &&
                     fp && fp->altsetting == 1 && fp->channels == 1 &&
                     fp->formats == SNDRV_PCM_FMTBIT_S16_LE &&
                     protocol == UAC_VERSION_1 &&
@@ -276,17 +371,6 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
                                                         fp->maxpacksize * 2)
                         continue;
 
-                csep = snd_usb_find_desc(alts->endpoint[0].extra, alts->endpoint[0].extralen, NULL, USB_DT_CS_ENDPOINT);
-                /* Creamware Noah has this descriptor after the 2nd endpoint */
-                if (!csep && altsd->bNumEndpoints >= 2)
-                        csep = snd_usb_find_desc(alts->endpoint[1].extra, alts->endpoint[1].extralen, NULL, USB_DT_CS_ENDPOINT);
-                if (!csep || csep[0] < 7 || csep[2] != UAC_EP_GENERAL) {
-                        snd_printk(KERN_WARNING "%d:%u:%d : no or invalid"
-                                   " class specific endpoint descriptor\n",
-                                   dev->devnum, iface_no, altno);
-                        csep = NULL;
-                }
-
                 fp = kzalloc(sizeof(*fp), GFP_KERNEL);
                 if (! fp) {
                         snd_printk(KERN_ERR "cannot malloc\n");
@@ -305,7 +389,8 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
                 if (snd_usb_get_speed(dev) == USB_SPEED_HIGH)
                         fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1)
                                         * (fp->maxpacksize & 0x7ff);
-                fp->attributes = csep ? csep[3] : 0;
+                fp->attributes = parse_uac_endpoint_attributes(chip, alts, protocol, iface_no);
+                fp->clock = clock;
 
                 /* some quirks for attributes here */
 
@@ -342,6 +427,7 @@ int snd_usb_parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no)
                 if (snd_usb_parse_audio_format(chip, fp, format, fmt, stream, alts) < 0) {
                         kfree(fp->rate_table);
                         kfree(fp);
+                        fp = NULL;
                         continue;
                 }
 
diff --git a/sound/usb/format.c b/sound/usb/format.c
index b87cf87c4e7..30364aba79c 100644
--- a/sound/usb/format.c
+++ b/sound/usb/format.c
@@ -29,6 +29,7 @@
 #include "quirks.h"
 #include "helper.h"
 #include "debug.h"
+#include "clock.h"
 
 /*
  * parse the audio format type I descriptor
@@ -205,6 +206,60 @@ static int parse_audio_format_rates_v1(struct snd_usb_audio *chip, struct audiof
 }
 
 /*
+ * Helper function to walk the array of sample rate triplets reported by
+ * the device. The problem is that we need to parse whole array first to
+ * get to know how many sample rates we have to expect.
+ * Then fp->rate_table can be allocated and filled.
+ */
+static int parse_uac2_sample_rate_range(struct audioformat *fp, int nr_triplets,
+                                        const unsigned char *data)
+{
+        int i, nr_rates = 0;
+
+        fp->rates = fp->rate_min = fp->rate_max = 0;
+
+        for (i = 0; i < nr_triplets; i++) {
+                int min = combine_quad(&data[2 + 12 * i]);
+                int max = combine_quad(&data[6 + 12 * i]);
+                int res = combine_quad(&data[10 + 12 * i]);
+                int rate;
+
+                if ((max < 0) || (min < 0) || (res < 0) || (max < min))
+                        continue;
+
+                /*
+                 * for ranges with res == 1, we announce a continuous sample
+                 * rate range, and this function should return 0 for no further
+                 * parsing.
+                 */
+                if (res == 1) {
+                        fp->rate_min = min;
+                        fp->rate_max = max;
+                        fp->rates = SNDRV_PCM_RATE_CONTINUOUS;
+                        return 0;
+                }
+
+                for (rate = min; rate <= max; rate += res) {
+                        if (fp->rate_table)
+                                fp->rate_table[nr_rates] = rate;
+                        if (!fp->rate_min || rate < fp->rate_min)
+                                fp->rate_min = rate;
+                        if (!fp->rate_max || rate > fp->rate_max)
+                                fp->rate_max = rate;
+                        fp->rates |= snd_pcm_rate_to_rate_bit(rate);
+
+                        nr_rates++;
+
+                        /* avoid endless loop */
+                        if (res == 0)
+                                break;
+                }
+        }
+
+        return nr_rates;
+}
+
+/*
  * parse the format descriptor and stores the possible sample rates
  * on the audioformat table (audio class v2).
  */
@@ -214,21 +269,30 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
 {
         struct usb_device *dev = chip->dev;
         unsigned char tmp[2], *data;
-        int i, nr_rates, data_size, ret = 0;
+        int nr_triplets, data_size, ret = 0;
+        int clock = snd_usb_clock_find_source(chip, chip->ctrl_intf, fp->clock);
+
+        if (clock < 0) {
+                snd_printk(KERN_ERR "%s(): unable to find clock source (clock %d)\n",
+                                __func__, clock);
+                goto err;
+        }
 
         /* get the number of sample rates first by only fetching 2 bytes */
         ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE,
                               USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
-                              UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8,
+                              UAC2_CS_CONTROL_SAM_FREQ << 8,
+                              snd_usb_ctrl_intf(chip) | (clock << 8),
                               tmp, sizeof(tmp), 1000);
 
         if (ret < 0) {
-                snd_printk(KERN_ERR "unable to retrieve number of sample rates\n");
+                snd_printk(KERN_ERR "%s(): unable to retrieve number of sample rates (clock %d)\n",
+                                __func__, clock);
                 goto err;
         }
 
-        nr_rates = (tmp[1] << 8) | tmp[0];
-        data_size = 2 + 12 * nr_rates;
+        nr_triplets = (tmp[1] << 8) | tmp[0];
+        data_size = 2 + 12 * nr_triplets;
         data = kzalloc(data_size, GFP_KERNEL);
         if (!data) {
                 ret = -ENOMEM;
@@ -237,36 +301,40 @@ static int parse_audio_format_rates_v2(struct snd_usb_audio *chip,
 
         /* now get the full information */
         ret = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_RANGE,
-                               USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
-                               UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8,
-                               data, data_size, 1000);
+                              USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
+                              UAC2_CS_CONTROL_SAM_FREQ << 8,
+                              snd_usb_ctrl_intf(chip) | (clock << 8),
+                              data, data_size, 1000);
 
         if (ret < 0) {
-                snd_printk(KERN_ERR "unable to retrieve sample rate range\n");
+                snd_printk(KERN_ERR "%s(): unable to retrieve sample rate range (clock %d)\n",
+                                __func__, clock);
                 ret = -EINVAL;
                 goto err_free;
         }
 
-        fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL);
+        /* Call the triplet parser, and make sure fp->rate_table is NULL.
+         * We just use the return value to know how many sample rates we
+         * will have to deal with. */
+        kfree(fp->rate_table);
+        fp->rate_table = NULL;
+        fp->nr_rates = parse_uac2_sample_rate_range(fp, nr_triplets, data);
+
+        if (fp->nr_rates == 0) {
+                /* SNDRV_PCM_RATE_CONTINUOUS */
+                ret = 0;
+                goto err_free;
+        }
+
+        fp->rate_table = kmalloc(sizeof(int) * fp->nr_rates, GFP_KERNEL);
         if (!fp->rate_table) {
                 ret = -ENOMEM;
                 goto err_free;
         }
 
-        fp->nr_rates = 0;
-        fp->rate_min = fp->rate_max = 0;
-
-        for (i = 0; i < nr_rates; i++) {
-                int rate = combine_quad(&data[2 + 12 * i]);
-
-                fp->rate_table[fp->nr_rates] = rate;
-                if (!fp->rate_min || rate < fp->rate_min)
-                        fp->rate_min = rate;
-                if (!fp->rate_max || rate > fp->rate_max)
-                        fp->rate_max = rate;
-                fp->rates |= snd_pcm_rate_to_rate_bit(rate);
-                fp->nr_rates++;
-        }
+        /* Call the triplet parser again, but this time, fp->rate_table is
+         * allocated, so the rates will be stored */
+        parse_uac2_sample_rate_range(fp, nr_triplets, data);
 
 err_free:
         kfree(data);
@@ -278,12 +346,11 @@ 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 +387,7 @@ 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 +459,12 @@ 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 +474,11 @@ 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 +489,10 @@ 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 8298c4e8ddf..387924f0af8 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/helper.h b/sound/usb/helper.h
index a6b0e51b3a9..09bd943c43b 100644
--- a/sound/usb/helper.h
+++ b/sound/usb/helper.h
@@ -28,5 +28,9 @@ unsigned char snd_usb_parse_datainterval(struct snd_usb_audio *chip,
 #define snd_usb_get_speed(dev) ((dev)->speed)
 #endif
 
+static inline int snd_usb_ctrl_intf(struct snd_usb_audio *chip)
+{
+        return get_iface_desc(chip->ctrl_intf)->bInterfaceNumber;
+}
 
 #endif /* __USBAUDIO_HELPER_H */
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 8b1e4b124a9..46785643c66 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 2089ec987c6..2fca80b744c 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 97dd1765510..736d134cc03 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -78,39 +78,6 @@ enum {
         USB_MIXER_U16,
 };
 
-enum {
-        USB_PROC_UPDOWN = 1,
-        USB_PROC_UPDOWN_SWITCH = 1,
-        USB_PROC_UPDOWN_MODE_SEL = 2,
-
-        USB_PROC_PROLOGIC = 2,
-        USB_PROC_PROLOGIC_SWITCH = 1,
-        USB_PROC_PROLOGIC_MODE_SEL = 2,
-
-        USB_PROC_3DENH = 3,
-        USB_PROC_3DENH_SWITCH = 1,
-        USB_PROC_3DENH_SPACE = 2,
-
-        USB_PROC_REVERB = 4,
-        USB_PROC_REVERB_SWITCH = 1,
-        USB_PROC_REVERB_LEVEL = 2,
-        USB_PROC_REVERB_TIME = 3,
-        USB_PROC_REVERB_DELAY = 4,
-
-        USB_PROC_CHORUS = 5,
-        USB_PROC_CHORUS_SWITCH = 1,
-        USB_PROC_CHORUS_LEVEL = 2,
-        USB_PROC_CHORUS_RATE = 3,
-        USB_PROC_CHORUS_DEPTH = 4,
-
-        USB_PROC_DCR = 6,
-        USB_PROC_DCR_SWITCH = 1,
-        USB_PROC_DCR_RATIO = 2,
-        USB_PROC_DCR_MAX_AMP = 3,
-        USB_PROC_DCR_THRESHOLD = 4,
-        USB_PROC_DCR_ATTACK = 5,
-        USB_PROC_DCR_RELEASE = 6,
-};
 
 /*E-mu 0202(0404) eXtension Unit(XU) control*/
 enum {
@@ -198,22 +165,24 @@ static int check_mapped_selector_name(struct mixer_build *state, int unitid,
 
 /*
  * find an audio control unit with the given unit id
- * this doesn't return any clock related units, so they need to be handled elsewhere
  */
 static void *find_audio_control_unit(struct mixer_build *state, unsigned char unit)
 {
-        unsigned char *p;
+        /* we just parse the header */
+        struct uac_feature_unit_descriptor *hdr = NULL;
 
-        p = NULL;
-        while ((p = snd_usb_find_desc(state->buffer, state->buflen, p,
-                                      USB_DT_CS_INTERFACE)) != NULL) {
-                if (p[0] >= 4 && p[2] >= UAC_INPUT_TERMINAL && p[2] <= UAC2_EXTENSION_UNIT_V2 && p[3] == unit)
-                        return p;
+        while ((hdr = snd_usb_find_desc(state->buffer, state->buflen, hdr,
+                                        USB_DT_CS_INTERFACE)) != NULL) {
+                if (hdr->bLength >= 4 &&
+                    hdr->bDescriptorSubtype >= UAC_INPUT_TERMINAL &&
+                    hdr->bDescriptorSubtype <= UAC2_SAMPLE_RATE_CONVERTER &&
+                    hdr->bUnitID == unit)
+                        return hdr;
         }
+
         return NULL;
 }
 
-
 /*
  * copy a string with the given id
  */
@@ -328,27 +297,36 @@ static int get_ctl_value_v1(struct usb_mixer_elem_info *cval, int request, int v
 
 static int get_ctl_value_v2(struct usb_mixer_elem_info *cval, int request, int validx, int *value_ret)
 {
-        unsigned char buf[14]; /* enough space for one range of 4 bytes */
+        unsigned char buf[2 + 3*sizeof(__u16)]; /* enough space for one range */
         unsigned char *val;
-        int ret;
+        int ret, size;
         __u8 bRequest;
 
-        bRequest = (request == UAC_GET_CUR) ?
-                UAC2_CS_CUR : UAC2_CS_RANGE;
+        if (request == UAC_GET_CUR) {
+                bRequest = UAC2_CS_CUR;
+                size = sizeof(__u16);
+        } else {
+                bRequest = UAC2_CS_RANGE;
+                size = sizeof(buf);
+        }
+
+        memset(buf, 0, sizeof(buf));
 
         ret = snd_usb_ctl_msg(cval->mixer->chip->dev,
                               usb_rcvctrlpipe(cval->mixer->chip->dev, 0),
                               bRequest,
                               USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN,
                               validx, cval->mixer->ctrlif | (cval->id << 8),
-                              buf, sizeof(buf), 1000);
+                              buf, size, 1000);
 
         if (ret < 0) {
-                snd_printdd(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n",
-                            request, validx, cval->mixer->ctrlif | (cval->id << 8), cval->val_type);
+                snd_printk(KERN_ERR "cannot get ctl value: req = %#x, wValue = %#x, wIndex = %#x, type = %d\n",
+                           request, validx, cval->mixer->ctrlif | (cval->id << 8), cval->val_type);
                 return ret;
         }
 
+        /* FIXME: how should we handle multiple triplets here? */
+
         switch (request) {
         case UAC_GET_CUR:
                 val = buf;
@@ -462,6 +440,16 @@ static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel,
                              int index, int value)
 {
         int err;
+        unsigned int read_only = (channel == 0) ?
+                cval->master_readonly :
+                cval->ch_readonly & (1 << (channel - 1));
+
+        if (read_only) {
+                snd_printdd(KERN_INFO "%s(): channel %d of control %d is read_only\n",
+                            __func__, channel, cval->control);
+                return 0;
+        }
+
         err = snd_usb_mixer_set_ctl_value(cval, UAC_SET_CUR, (cval->control << 8) | channel,
                             value);
         if (err < 0)
@@ -631,6 +619,7 @@ static int get_term_name(struct mixer_build *state, struct usb_audio_term *iterm
  */
 static int check_input_term(struct mixer_build *state, int id, struct usb_audio_term *term)
 {
+        int err;
         void *p1;
 
         memset(term, 0, sizeof(*term));
@@ -651,6 +640,11 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_
                                 term->channels = d->bNrChannels;
                                 term->chconfig = le32_to_cpu(d->bmChannelConfig);
                                 term->name = d->iTerminal;
+
+                                /* call recursively to get the clock selectors */
+                                err = check_input_term(state, d->bCSourceID, term);
+                                if (err < 0)
+                                        return err;
                         }
                         return 0;
                 case UAC_FEATURE_UNIT: {
@@ -667,7 +661,8 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_
                         term->name = uac_mixer_unit_iMixer(d);
                         return 0;
                 }
-                case UAC_SELECTOR_UNIT: {
+                case UAC_SELECTOR_UNIT:
+                case UAC2_CLOCK_SELECTOR: {
                         struct uac_selector_unit_descriptor *d = p1;
                         /* call recursively to retrieve the channel info */
                         if (check_input_term(state, d->baSourceID[0], term) < 0)
@@ -690,6 +685,13 @@ static int check_input_term(struct mixer_build *state, int id, struct usb_audio_
                         term->name = uac_processing_unit_iProcessing(d, state->mixer->protocol);
                         return 0;
                 }
+                case UAC2_CLOCK_SOURCE: {
+                        struct uac_clock_source_descriptor *d = p1;
+                        term->type = d->bDescriptorSubtype << 16; /* virtual type */
+                        term->id = id;
+                        term->name = d->iClockSource;
+                        return 0;
+                }
                 default:
                         return -ENODEV;
                 }
@@ -709,16 +711,20 @@ struct usb_feature_control_info {
 };
 
 static struct usb_feature_control_info audio_feature_info[] = {
-        { "Mute",               USB_MIXER_INV_BOOLEAN },
-        { "Volume",             USB_MIXER_S16 },
+        { "Mute",                       USB_MIXER_INV_BOOLEAN },
+        { "Volume",                     USB_MIXER_S16 },
         { "Tone Control - Bass",        USB_MIXER_S8 },
         { "Tone Control - Mid",         USB_MIXER_S8 },
         { "Tone Control - Treble",      USB_MIXER_S8 },
         { "Graphic Equalizer",          USB_MIXER_S8 }, /* FIXME: not implemeted yet */
-        { "Auto Gain Control",  USB_MIXER_BOOLEAN },
-        { "Delay Control",      USB_MIXER_U16 },
-        { "Bass Boost",         USB_MIXER_BOOLEAN },
-        { "Loudness",           USB_MIXER_BOOLEAN },
+        { "Auto Gain Control",          USB_MIXER_BOOLEAN },
+        { "Delay Control",              USB_MIXER_U16 },
+        { "Bass Boost",                 USB_MIXER_BOOLEAN },
+        { "Loudness",                   USB_MIXER_BOOLEAN },
+        /* UAC2 specific */
+        { "Input Gain Control",         USB_MIXER_U16 },
+        { "Input Gain Pad Control",     USB_MIXER_BOOLEAN },
+        { "Phase Inverter Control",     USB_MIXER_BOOLEAN },
 };
 
 
@@ -958,7 +964,7 @@ static size_t append_ctl_name(struct snd_kcontrol *kctl, const char *str)
 static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
                               unsigned int ctl_mask, int control,
                               struct usb_audio_term *iterm, int unitid,
-                              int read_only)
+                              int readonly_mask)
 {
         struct uac_feature_unit_descriptor *desc = raw_desc;
         unsigned int len = 0;
@@ -970,7 +976,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
 
         control++; /* change from zero-based to 1-based value */
 
-        if (control == UAC_GRAPHIC_EQUALIZER_CONTROL) {
+        if (control == UAC_FU_GRAPHIC_EQUALIZER) {
                 /* FIXME: not supported yet */
                 return;
         }
@@ -989,20 +995,25 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
         cval->control = control;
         cval->cmask = ctl_mask;
         cval->val_type = audio_feature_info[control-1].type;
-        if (ctl_mask == 0)
+        if (ctl_mask == 0) {
                 cval->channels = 1;     /* master channel */
-        else {
+                cval->master_readonly = readonly_mask;
+        } else {
                 int i, c = 0;
                 for (i = 0; i < 16; i++)
                         if (ctl_mask & (1 << i))
                                 c++;
                 cval->channels = c;
+                cval->ch_readonly = readonly_mask;
         }
 
         /* get min/max values */
         get_min_max(cval, 0);
 
-        if (read_only)
+        /* if all channels in the mask are marked read-only, make the control
+         * read-only. set_cur_mix_value() will check the mask again and won't
+         * issue write commands to read-only channels. */
+        if (cval->channels == readonly_mask)
                 kctl = snd_ctl_new1(&usb_feature_unit_ctl_ro, cval);
         else
                 kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval);
@@ -1021,8 +1032,8 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
                                 kctl->id.name, sizeof(kctl->id.name));
 
         switch (control) {
-        case UAC_MUTE_CONTROL:
-        case UAC_VOLUME_CONTROL:
+        case UAC_FU_MUTE:
+        case UAC_FU_VOLUME:
                 /* determine the control name.  the rule is:
                  * - if a name id is given in descriptor, use it.
                  * - if the connected input can be determined, then use the name
@@ -1049,9 +1060,9 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
                                 len = append_ctl_name(kctl, " Playback");
                         }
                 }
-                append_ctl_name(kctl, control == UAC_MUTE_CONTROL ?
+                append_ctl_name(kctl, control == UAC_FU_MUTE ?
                                 " Switch" : " Volume");
-                if (control == UAC_VOLUME_CONTROL) {
+                if (control == UAC_FU_VOLUME) {
                         kctl->tlv.c = mixer_vol_tlv;
                         kctl->vd[0].access |= 
                                 SNDRV_CTL_ELEM_ACCESS_TLV_READ |
@@ -1096,6 +1107,19 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc,
                 }
                 break;
 
+        case USB_ID(0x046d, 0x0809):
+        case USB_ID(0x046d, 0x0991):
+        /* Most audio usb devices lie about volume resolution.
+         * Most Logitech webcams have res = 384.
+         * Proboly there is some logitech magic behind this number --fishor
+         */
+                if (!strcmp(kctl->id.name, "Mic Capture Volume")) {
+                        snd_printk(KERN_INFO
+                                "set resolution quirk: cval->res = 384\n");
+                        cval->res = 384;
+                }
+                break;
+
         }
 
         snd_printdd(KERN_INFO "[%d] FU [%s] ch = %d, val = %d/%d/%d\n",
@@ -1126,7 +1150,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void
         } else {
                 struct uac2_feature_unit_descriptor *ftr = _ftr;
                 csize = 4;
-                channels = (hdr->bLength - 6) / 4;
+                channels = (hdr->bLength - 6) / 4 - 1;
                 bmaControls = ftr->bmaControls;
         }
 
@@ -1150,7 +1174,7 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void
                 snd_printk(KERN_INFO
                            "usbmixer: master volume quirk for PCM2702 chip\n");
                 /* disable non-functional volume control */
-                master_bits &= ~UAC_FU_VOLUME;
+                master_bits &= ~UAC_CONTROL_BIT(UAC_FU_VOLUME);
                 break;
         }
         if (channels > 0)
@@ -1188,19 +1212,22 @@ static int parse_audio_feature_unit(struct mixer_build *state, int unitid, void
 
                         for (j = 0; j < channels; j++) {
                                 unsigned int mask = snd_usb_combine_bytes(bmaControls + csize * (j+1), csize);
-                                if (mask & (1 << (i * 2))) {
+                                if (uac2_control_is_readable(mask, i)) {
                                         ch_bits |= (1 << j);
-                                        if (~mask & (1 << ((i * 2) + 1)))
+                                        if (!uac2_control_is_writeable(mask, i))
                                                 ch_read_only |= (1 << j);
                                 }
                         }
 
-                        /* FIXME: the whole unit is read-only if any of the channels is marked read-only */
+                        /* NOTE: build_feature_ctl() will mark the control read-only if all channels
+                         * are marked read-only in the descriptors. Otherwise, the control will be
+                         * reported as writeable, but the driver will not actually issue a write
+                         * command for read-only channels */
                         if (ch_bits & 1) /* the first channel must be set (for ease of programming) */
-                                build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, !!ch_read_only);
-                        if (master_bits & (1 << i * 2))
+                                build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, ch_read_only);
+                        if (uac2_control_is_readable(master_bits, i))
                                 build_feature_ctl(state, _ftr, 0, i, &iterm, unitid,
-                                                  ~master_bits & (1 << ((i * 2) + 1)));
+                                                  !uac2_control_is_writeable(master_bits, i));
                 }
         }
 
@@ -1392,51 +1419,51 @@ struct procunit_info {
 };
 
 static struct procunit_value_info updown_proc_info[] = {
-        { USB_PROC_UPDOWN_SWITCH, "Switch", USB_MIXER_BOOLEAN },
-        { USB_PROC_UPDOWN_MODE_SEL, "Mode Select", USB_MIXER_U8, 1 },
+        { UAC_UD_ENABLE, "Switch", USB_MIXER_BOOLEAN },
+        { UAC_UD_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 },
         { 0 }
 };
 static struct procunit_value_info prologic_proc_info[] = {
-        { USB_PROC_PROLOGIC_SWITCH, "Switch", USB_MIXER_BOOLEAN },
-        { USB_PROC_PROLOGIC_MODE_SEL, "Mode Select", USB_MIXER_U8, 1 },
+        { UAC_DP_ENABLE, "Switch", USB_MIXER_BOOLEAN },
+        { UAC_DP_MODE_SELECT, "Mode Select", USB_MIXER_U8, 1 },
         { 0 }
 };
 static struct procunit_value_info threed_enh_proc_info[] = {
-        { USB_PROC_3DENH_SWITCH, "Switch", USB_MIXER_BOOLEAN },
-        { USB_PROC_3DENH_SPACE, "Spaciousness", USB_MIXER_U8 },
+        { UAC_3D_ENABLE, "Switch", USB_MIXER_BOOLEAN },
+        { UAC_3D_SPACE, "Spaciousness", USB_MIXER_U8 },
         { 0 }
 };
 static struct procunit_value_info reverb_proc_info[] = {
-        { USB_PROC_REVERB_SWITCH, "Switch", USB_MIXER_BOOLEAN },
-        { USB_PROC_REVERB_LEVEL, "Level", USB_MIXER_U8 },
-        { USB_PROC_REVERB_TIME, "Time", USB_MIXER_U16 },
-        { USB_PROC_REVERB_DELAY, "Delay", USB_MIXER_U8 },
+        { UAC_REVERB_ENABLE, "Switch", USB_MIXER_BOOLEAN },
+        { UAC_REVERB_LEVEL, "Level", USB_MIXER_U8 },
+        { UAC_REVERB_TIME, "Time", USB_MIXER_U16 },
+        { UAC_REVERB_FEEDBACK, "Feedback", USB_MIXER_U8 },
         { 0 }
 };
 static struct procunit_value_info chorus_proc_info[] = {
-        { USB_PROC_CHORUS_SWITCH, "Switch", USB_MIXER_BOOLEAN },
-        { USB_PROC_CHORUS_LEVEL, "Level", USB_MIXER_U8 },
-        { USB_PROC_CHORUS_RATE, "Rate", USB_MIXER_U16 },
-        { USB_PROC_CHORUS_DEPTH, "Depth", USB_MIXER_U16 },
+        { UAC_CHORUS_ENABLE, "Switch", USB_MIXER_BOOLEAN },
+        { UAC_CHORUS_LEVEL, "Level", USB_MIXER_U8 },
+        { UAC_CHORUS_RATE, "Rate", USB_MIXER_U16 },
+        { UAC_CHORUS_DEPTH, "Depth", USB_MIXER_U16 },
         { 0 }
 };
 static struct procunit_value_info dcr_proc_info[] = {
-        { USB_PROC_DCR_SWITCH, "Switch", USB_MIXER_BOOLEAN },
-        { USB_PROC_DCR_RATIO, "Ratio", USB_MIXER_U16 },
-        { USB_PROC_DCR_MAX_AMP, "Max Amp", USB_MIXER_S16 },
-        { USB_PROC_DCR_THRESHOLD, "Threshold", USB_MIXER_S16 },
-        { USB_PROC_DCR_ATTACK, "Attack Time", USB_MIXER_U16 },
-        { USB_PROC_DCR_RELEASE, "Release Time", USB_MIXER_U16 },
+        { UAC_DCR_ENABLE, "Switch", USB_MIXER_BOOLEAN },
+        { UAC_DCR_RATE, "Ratio", USB_MIXER_U16 },
+        { UAC_DCR_MAXAMPL, "Max Amp", USB_MIXER_S16 },
+        { UAC_DCR_THRESHOLD, "Threshold", USB_MIXER_S16 },
+        { UAC_DCR_ATTACK_TIME, "Attack Time", USB_MIXER_U16 },
+        { UAC_DCR_RELEASE_TIME, "Release Time", USB_MIXER_U16 },
         { 0 }
 };
 
 static struct procunit_info procunits[] = {
-        { USB_PROC_UPDOWN, "Up Down", updown_proc_info },
-        { USB_PROC_PROLOGIC, "Dolby Prologic", prologic_proc_info },
-        { USB_PROC_3DENH, "3D Stereo Extender", threed_enh_proc_info },
-        { USB_PROC_REVERB, "Reverb", reverb_proc_info },
-        { USB_PROC_CHORUS, "Chorus", chorus_proc_info },
-        { USB_PROC_DCR, "DCR", dcr_proc_info },
+        { UAC_PROCESS_UP_DOWNMIX, "Up Down", updown_proc_info },
+        { UAC_PROCESS_DOLBY_PROLOGIC, "Dolby Prologic", prologic_proc_info },
+        { UAC_PROCESS_STEREO_EXTENDER, "3D Stereo Extender", threed_enh_proc_info },
+        { UAC_PROCESS_REVERB, "Reverb", reverb_proc_info },
+        { UAC_PROCESS_CHORUS, "Chorus", chorus_proc_info },
+        { UAC_PROCESS_DYN_RANGE_COMP, "DCR", dcr_proc_info },
         { 0 },
 };
 /*
@@ -1524,7 +1551,7 @@ static int build_audio_procunit(struct mixer_build *state, int unitid, void *raw
                 cval->channels = 1;
 
                 /* get min/max values */
-                if (type == USB_PROC_UPDOWN && cval->control == USB_PROC_UPDOWN_MODE_SEL) {
+                if (type == UAC_PROCESS_UP_DOWNMIX && cval->control == UAC_UD_MODE_SELECT) {
                         __u8 *control_spec = uac_processing_unit_specific(desc, state->mixer->protocol);
                         /* FIXME: hard-coded */
                         cval->min = 1;
@@ -1619,7 +1646,7 @@ static int mixer_ctl_selector_get(struct snd_kcontrol *kcontrol, struct snd_ctl_
         struct usb_mixer_elem_info *cval = kcontrol->private_data;
         int val, err;
 
-        err = get_cur_ctl_value(cval, 0, &val);
+        err = get_cur_ctl_value(cval, cval->control << 8, &val);
         if (err < 0) {
                 if (cval->mixer->ignore_ctl_error) {
                         ucontrol->value.enumerated.item[0] = 0;
@@ -1638,7 +1665,7 @@ static int mixer_ctl_selector_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
         struct usb_mixer_elem_info *cval = kcontrol->private_data;
         int val, oval, err;
 
-        err = get_cur_ctl_value(cval, 0, &oval);
+        err = get_cur_ctl_value(cval, cval->control << 8, &oval);
         if (err < 0) {
                 if (cval->mixer->ignore_ctl_error)
                         return 0;
@@ -1647,7 +1674,7 @@ static int mixer_ctl_selector_put(struct snd_kcontrol *kcontrol, struct snd_ctl_
         val = ucontrol->value.enumerated.item[0];
         val = get_abs_value(cval, val);
         if (val != oval) {
-                set_cur_ctl_value(cval, 0, val);
+                set_cur_ctl_value(cval, cval->control << 8, val);
                 return 1;
         }
         return 0;
@@ -1729,6 +1756,11 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void
         cval->res = 1;
         cval->initialized = 1;
 
+        if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR)
+                cval->control = UAC2_CX_CLOCK_SELECTOR;
+        else
+                cval->control = 0;
+
         namelist = kmalloc(sizeof(char *) * desc->bNrInPins, GFP_KERNEL);
         if (! namelist) {
                 snd_printk(KERN_ERR "cannot malloc\n");
@@ -1778,7 +1810,9 @@ static int parse_audio_selector_unit(struct mixer_build *state, int unitid, void
                 if (! len)
                         strlcpy(kctl->id.name, "USB", sizeof(kctl->id.name));
 
-                if ((state->oterm.type & 0xff00) == 0x0100)
+                if (desc->bDescriptorSubtype == UAC2_CLOCK_SELECTOR)
+                        append_ctl_name(kctl, " Clock Source");
+                else if ((state->oterm.type & 0xff00) == 0x0100)
                         append_ctl_name(kctl, " Capture Source");
                 else
                         append_ctl_name(kctl, " Playback Source");
@@ -1812,10 +1846,12 @@ static int parse_audio_unit(struct mixer_build *state, int unitid)
 
         switch (p1[2]) {
         case UAC_INPUT_TERMINAL:
+        case UAC2_CLOCK_SOURCE:
                 return 0; /* NOP */
         case UAC_MIXER_UNIT:
                 return parse_audio_mixer_unit(state, unitid, p1);
         case UAC_SELECTOR_UNIT:
+        case UAC2_CLOCK_SELECTOR:
                 return parse_audio_selector_unit(state, unitid, p1);
         case UAC_FEATURE_UNIT:
                 return parse_audio_feature_unit(state, unitid, p1);
@@ -1912,6 +1948,11 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer)
                         err = parse_audio_unit(&state, desc->bSourceID);
                         if (err < 0)
                                 return err;
+
+                        /* for UAC2, use the same approach to also add the clock selectors */
+                        err = parse_audio_unit(&state, desc->bCSourceID);
+                        if (err < 0)
+                                return err;
                 }
         }
 
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index 130123854a6..a7cf1007fbb 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -34,6 +34,8 @@ struct usb_mixer_elem_info {
         unsigned int id;
         unsigned int control;   /* CS or ICN (high byte) */
         unsigned int cmask; /* channel mask bitmap: 0 = master */
+        unsigned int ch_readonly;
+        unsigned int master_readonly;
         int channels;
         int val_type;
         int min, max, res;
diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c
index d93fc89beba..f1324c42383 100644
--- a/sound/usb/mixer_maps.c
+++ b/sound/usb/mixer_maps.c
@@ -85,8 +85,8 @@ static struct usbmix_name_map extigy_map[] = {
         /* 16: MU (w/o controls) */
         { 17, NULL, 1 }, /* DISABLED: PU-switch (any effect?) */
         { 17, "Channel Routing", 2 },   /* PU: mode select */
-        { 18, "Tone Control - Bass", UAC_BASS_CONTROL }, /* FU */
-        { 18, "Tone Control - Treble", UAC_TREBLE_CONTROL }, /* FU */
+        { 18, "Tone Control - Bass", UAC_FU_BASS }, /* FU */
+        { 18, "Tone Control - Treble", UAC_FU_TREBLE }, /* FU */
         { 18, "Master Playback" }, /* FU; others */
         /* 19: OT speaker */
         /* 20: OT headphone */
diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c
index 2bf0d77d176..456829882f4 100644
--- a/sound/usb/pcm.c
+++ b/sound/usb/pcm.c
@@ -31,6 +31,7 @@
 #include "urb.h"
 #include "helper.h"
 #include "pcm.h"
+#include "clock.h"
 
 /*
  * return the current pcm pointer.  just based on the hwptr_done value.
@@ -120,10 +121,6 @@ static int init_pitch_v1(struct snd_usb_audio *chip, int iface,
 
         ep = get_endpoint(alts, 0)->bEndpointAddress;
 
-        /* if endpoint doesn't have pitch control, bail out */
-        if (!(fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL))
-                return 0;
-
         data[0] = 1;
         if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR,
                                    USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT,
@@ -137,119 +134,49 @@ static int init_pitch_v1(struct snd_usb_audio *chip, int iface,
         return 0;
 }
 
-/*
- * initialize the picth control and sample rate
- */
-int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface,
-                       struct usb_host_interface *alts,
-                       struct audioformat *fmt)
-{
-        struct usb_interface_descriptor *altsd = get_iface_desc(alts);
-
-        switch (altsd->bInterfaceProtocol) {
-        case UAC_VERSION_1:
-                return init_pitch_v1(chip, iface, alts, fmt);
-
-        case UAC_VERSION_2:
-                /* not implemented yet */
-                return 0;
-        }
-
-        return -EINVAL;
-}
-
-static int set_sample_rate_v1(struct snd_usb_audio *chip, int iface,
-                              struct usb_host_interface *alts,
-                              struct audioformat *fmt, int rate)
+static int init_pitch_v2(struct snd_usb_audio *chip, int iface,
+                         struct usb_host_interface *alts,
+                         struct audioformat *fmt)
 {
         struct usb_device *dev = chip->dev;
+        unsigned char data[1];
         unsigned int ep;
-        unsigned char data[3];
-        int err, crate;
+        int err;
 
         ep = get_endpoint(alts, 0)->bEndpointAddress;
-        /* if endpoint doesn't have sampling rate control, bail out */
-        if (!(fmt->attributes & UAC_EP_CS_ATTR_SAMPLE_RATE)) {
-                snd_printk(KERN_WARNING "%d:%d:%d: endpoint lacks sample rate attribute bit, cannot set.\n",
-                                   dev->devnum, iface, fmt->altsetting);
-                return 0;
-        }
 
-        data[0] = rate;
-        data[1] = rate >> 8;
-        data[2] = rate >> 16;
-        if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC_SET_CUR,
-                                   USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT,
-                                   UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep,
-                                   data, sizeof(data), 1000)) < 0) {
-                snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n",
-                           dev->devnum, iface, fmt->altsetting, rate, ep);
-                return err;
-        }
-        if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC_GET_CUR,
-                                   USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_IN,
-                                   UAC_EP_CS_ATTR_SAMPLE_RATE << 8, ep,
-                                   data, sizeof(data), 1000)) < 0) {
-                snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n",
-                           dev->devnum, iface, fmt->altsetting, ep);
-                return 0; /* some devices don't support reading */
-        }
-        crate = data[0] | (data[1] << 8) | (data[2] << 16);
-        if (crate != rate) {
-                snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate);
-                // runtime->rate = crate;
-        }
-
-        return 0;
-}
-
-static int set_sample_rate_v2(struct snd_usb_audio *chip, int iface,
-                              struct usb_host_interface *alts,
-                              struct audioformat *fmt, int rate)
-{
-        struct usb_device *dev = chip->dev;
-        unsigned char data[4];
-        int err, crate;
-
-        data[0] = rate;
-        data[1] = rate >> 8;
-        data[2] = rate >> 16;
-        data[3] = rate >> 24;
+        data[0] = 1;
         if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), UAC2_CS_CUR,
-                                   USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_OUT,
-                                   UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8,
-                                   data, sizeof(data), 1000)) < 0) {
-                snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d (v2)\n",
-                           dev->devnum, iface, fmt->altsetting, rate);
-                return err;
-        }
-        if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), UAC2_CS_CUR,
-                                   USB_TYPE_CLASS | USB_RECIP_INTERFACE | USB_DIR_IN,
-                                   UAC2_CS_CONTROL_SAM_FREQ << 8, chip->clock_id << 8,
+                                   USB_TYPE_CLASS | USB_RECIP_ENDPOINT | USB_DIR_OUT,
+                                   UAC2_EP_CS_PITCH << 8, 0,
                                    data, sizeof(data), 1000)) < 0) {
-                snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq (v2)\n",
+                snd_printk(KERN_ERR "%d:%d:%d: cannot set enable PITCH (v2)\n",
                            dev->devnum, iface, fmt->altsetting);
                 return err;
         }
-        crate = data[0] | (data[1] << 8) | (data[2] << 16) | (data[3] << 24);
-        if (crate != rate)
-                snd_printd(KERN_WARNING "current rate %d is different from the runtime rate %d\n", crate, rate);
 
         return 0;
 }
 
-int snd_usb_init_sample_rate(struct snd_usb_audio *chip, int iface,
-                             struct usb_host_interface *alts,
-                             struct audioformat *fmt, int rate)
+/*
+ * initialize the pitch control and sample rate
+ */
+int snd_usb_init_pitch(struct snd_usb_audio *chip, int iface,
+                       struct usb_host_interface *alts,
+                       struct audioformat *fmt)
 {
         struct usb_interface_descriptor *altsd = get_iface_desc(alts);
 
+        /* if endpoint doesn't have pitch control, bail out */
+        if (!(fmt->attributes & UAC_EP_CS_ATTR_PITCH_CONTROL))
+                return 0;
+
         switch (altsd->bInterfaceProtocol) {
         case UAC_VERSION_1:
-                return set_sample_rate_v1(chip, iface, alts, fmt, rate);
+                return init_pitch_v1(chip, iface, alts, fmt);
 
         case UAC_VERSION_2:
-                return set_sample_rate_v2(chip, iface, alts, fmt, rate);
+                return init_pitch_v2(chip, iface, alts, fmt);
         }
 
         return -EINVAL;
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 91ddef31bcb..f8797f61a24 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 136e5b4cf6d..b45e54c09ba 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 d679e72a3e5..24d3319cc34 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -40,9 +40,6 @@ struct snd_usb_audio {
         int num_interfaces;
         int num_suspended_intf;
 
-        /* for audio class v2 */
-        int clock_id;
-
         struct list_head pcm_list;      /* list of pcm streams */
         int pcm_devs;
 
@@ -53,6 +50,8 @@ struct snd_usb_audio {
         int setup;                      /* from the 'device_setup' module param */
         int nrpacks;                    /* from the 'nrpacks' module param */
         int async_unlink;               /* from the 'async_unlink' module param */
+
+        struct usb_host_interface *ctrl_intf;   /* the audio control interface */
 };
 
 /*
@@ -74,6 +73,7 @@ enum quirk_type {
         QUIRK_MIDI_FASTLANE,
         QUIRK_MIDI_EMAGIC,
         QUIRK_MIDI_CME,
+        QUIRK_MIDI_AKAI,
         QUIRK_MIDI_US122L,
         QUIRK_AUDIO_STANDARD_INTERFACE,
         QUIRK_AUDIO_FIXED_ENDPOINT,