76 files changed, 4443 insertions, 400 deletions
diff --git a/sound/Kconfig b/sound/Kconfig
index 5a240e050ae6..ee2e69a9ecd1 100644
--- a/sound/Kconfig
+++ b/sound/Kconfig
@@ -108,6 +108,8 @@ source "sound/parisc/Kconfig"
 source "sound/soc/Kconfig"
+source "sound/x86/Kconfig"
 endif # SND
 menuconfig SOUND_PRIME
diff --git a/sound/Makefile b/sound/Makefile
index c41bdf5fdf24..6de45d2c32f7 100644
--- a/sound/Makefile
+++ b/sound/Makefile
@@ -5,7 +5,7 @@ obj-$(CONFIG_SOUND) += soundcore.o
 obj-$(CONFIG_SOUND_PRIME) += oss/
 obj-$(CONFIG_DMASOUND) += oss/
 obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ \
-        firewire/ sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/ hda/
+        firewire/ sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/ hda/ x86/
 obj-$(CONFIG_SND_AOA) += aoa/
 # This one must be compilable even if sound is configured out
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index 2096bb0835c8..8da9cb245d01 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -1749,7 +1749,7 @@ static int snd_rawmidi_dev_disconnect(struct snd_device *device)
 * Sets the rawmidi operators for the given stream direction.
 */
 void snd_rawmidi_set_ops(struct snd_rawmidi *rmidi, int stream,
-                         struct snd_rawmidi_ops *ops)
+                         const struct snd_rawmidi_ops *ops)
 {
        struct snd_rawmidi_substream *substream;
        
diff --git a/sound/core/seq/seq_memory.c b/sound/core/seq/seq_memory.c
index c850345c43b5..dfa5156f3585 100644
--- a/sound/core/seq/seq_memory.c
+++ b/sound/core/seq/seq_memory.c
@@ -419,7 +419,6 @@ int snd_seq_pool_done(struct snd_seq_pool *pool)
 {
        unsigned long flags;
        struct snd_seq_event_cell *ptr;
-        int max_count = 5 * HZ;
        if (snd_BUG_ON(!pool))
                return -EINVAL;
@@ -432,14 +431,8 @@ int snd_seq_pool_done(struct snd_seq_pool *pool)
        if (waitqueue_active(&pool->output_sleep))
                wake_up(&pool->output_sleep);
-        while (atomic_read(&pool->counter) > 0) {
+        while (atomic_read(&pool->counter) > 0)
-                if (max_count == 0) {
-                        pr_warn("ALSA: snd_seq_pool_done timeout: %d cells remain\n", atomic_read(&pool->counter));
-                        break;
-                }
                schedule_timeout_uninterruptible(1);
-                max_count--;
-        }
        
        /* release all resources */
        spin_lock_irqsave(&pool->lock, flags);
diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c
index 0bec02e89d51..450c5187eecb 100644
--- a/sound/core/seq/seq_queue.c
+++ b/sound/core/seq/seq_queue.c
@@ -181,6 +181,8 @@ void __exit snd_seq_queues_delete(void)
        }
 }
+static void queue_use(struct snd_seq_queue *queue, int client, int use);
 /* allocate a new queue -
 * return queue index value or negative value for error
 */
@@ -192,11 +194,11 @@ int snd_seq_queue_alloc(int client, int locked, unsigned int info_flags)
        if (q == NULL)
                return -ENOMEM;
        q->info_flags = info_flags;
+        queue_use(q, client, 1);
        if (queue_list_add(q) < 0) {
                queue_delete(q);
                return -ENOMEM;
        }
-        snd_seq_queue_use(q->queue, client, 1); /* use this queue */
        return q->queue;
 }
@@ -502,19 +504,9 @@ int snd_seq_queue_timer_set_tempo(int queueid, int client,
        return result;
 }
+/* use or unuse this queue */
-/* use or unuse this queue -
+static void queue_use(struct snd_seq_queue *queue, int client, int use)
- * if it is the first client, starts the timer.
- * if it is not longer used by any clients, stop the timer.
- */
-int snd_seq_queue_use(int queueid, int client, int use)
 {
-        struct snd_seq_queue *queue;
-        queue = queueptr(queueid);
-        if (queue == NULL)
-                return -EINVAL;
-        mutex_lock(&queue->timer_mutex);
        if (use) {
                if (!test_and_set_bit(client, queue->clients_bitmap))
                        queue->clients++;
@@ -529,6 +521,21 @@ int snd_seq_queue_use(int queueid, int client, int use)
        } else {
                snd_seq_timer_close(queue);
        }
+}
+/* use or unuse this queue -
+ * if it is the first client, starts the timer.
+ * if it is not longer used by any clients, stop the timer.
+ */
+int snd_seq_queue_use(int queueid, int client, int use)
+{
+        struct snd_seq_queue *queue;
+        queue = queueptr(queueid);
+        if (queue == NULL)
+                return -EINVAL;
+        mutex_lock(&queue->timer_mutex);
+        queue_use(queue, client, use);
        mutex_unlock(&queue->timer_mutex);
        queuefree(queue);
        return 0;
diff --git a/sound/core/seq/seq_virmidi.c b/sound/core/seq/seq_virmidi.c
index c82ed3e70506..52f31f1498f9 100644
--- a/sound/core/seq/seq_virmidi.c
+++ b/sound/core/seq/seq_virmidi.c
@@ -349,13 +349,13 @@ static int snd_virmidi_unuse(void *private_data,
 *  Register functions
 */
-static struct snd_rawmidi_ops snd_virmidi_input_ops = {
+static const struct snd_rawmidi_ops snd_virmidi_input_ops = {
        .open = snd_virmidi_input_open,
        .close = snd_virmidi_input_close,
        .trigger = snd_virmidi_input_trigger,
 };
-static struct snd_rawmidi_ops snd_virmidi_output_ops = {
+static const struct snd_rawmidi_ops snd_virmidi_output_ops = {
        .open = snd_virmidi_output_open,
        .close = snd_virmidi_output_close,
        .trigger = snd_virmidi_output_trigger,
diff --git a/sound/drivers/mpu401/mpu401_uart.c b/sound/drivers/mpu401/mpu401_uart.c
index 776596b5ee05..3a7c317ae012 100644
--- a/sound/drivers/mpu401/mpu401_uart.c
+++ b/sound/drivers/mpu401/mpu401_uart.c
@@ -481,14 +481,14 @@ snd_mpu401_uart_output_trigger(struct snd_rawmidi_substream *substream, int up)
 */
-static struct snd_rawmidi_ops snd_mpu401_uart_output =
+static const struct snd_rawmidi_ops snd_mpu401_uart_output =
 {
        .open =         snd_mpu401_uart_output_open,
        .close =        snd_mpu401_uart_output_close,
        .trigger =      snd_mpu401_uart_output_trigger,
 };
-static struct snd_rawmidi_ops snd_mpu401_uart_input =
+static const struct snd_rawmidi_ops snd_mpu401_uart_input =
 {
        .open =         snd_mpu401_uart_input_open,
        .close =        snd_mpu401_uart_input_close,
diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c
index 30e8a1d5bc87..00b31f92c504 100644
--- a/sound/drivers/mtpav.c
+++ b/sound/drivers/mtpav.c
@@ -600,13 +600,13 @@ static int snd_mtpav_get_ISA(struct mtpav *mcard)
 /*
 */
-static struct snd_rawmidi_ops snd_mtpav_output = {
+static const struct snd_rawmidi_ops snd_mtpav_output = {
        .open =         snd_mtpav_output_open,
        .close =        snd_mtpav_output_close,
        .trigger =      snd_mtpav_output_trigger,
 };
-static struct snd_rawmidi_ops snd_mtpav_input = {
+static const struct snd_rawmidi_ops snd_mtpav_input = {
        .open =         snd_mtpav_input_open,
        .close =        snd_mtpav_input_close,
        .trigger =      snd_mtpav_input_trigger,
diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c
index fd4d18df84d3..f32e81342247 100644
--- a/sound/drivers/mts64.c
+++ b/sound/drivers/mts64.c
@@ -749,13 +749,13 @@ static void snd_mts64_rawmidi_input_trigger(struct snd_rawmidi_substream *substr
        spin_unlock_irqrestore(&mts->lock, flags);
 }
-static struct snd_rawmidi_ops snd_mts64_rawmidi_output_ops = {
+static const struct snd_rawmidi_ops snd_mts64_rawmidi_output_ops = {
        .open    = snd_mts64_rawmidi_open,
        .close   = snd_mts64_rawmidi_close,
        .trigger = snd_mts64_rawmidi_output_trigger
 };
-static struct snd_rawmidi_ops snd_mts64_rawmidi_input_ops = {
+static const struct snd_rawmidi_ops snd_mts64_rawmidi_input_ops = {
        .open    = snd_mts64_rawmidi_open,
        .close   = snd_mts64_rawmidi_close,
        .trigger = snd_mts64_rawmidi_input_trigger
diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c
index 189e3e7028af..ec8a94325ef6 100644
--- a/sound/drivers/portman2x4.c
+++ b/sound/drivers/portman2x4.c
@@ -546,13 +546,13 @@ static void snd_portman_midi_output_trigger(struct snd_rawmidi_substream *substr
        spin_unlock_irqrestore(&pm->reg_lock, flags);
 }
-static struct snd_rawmidi_ops snd_portman_midi_output = {
+static const struct snd_rawmidi_ops snd_portman_midi_output = {
        .open =         snd_portman_midi_open,
        .close =        snd_portman_midi_close,
        .trigger =      snd_portman_midi_output_trigger,
 };
-static struct snd_rawmidi_ops snd_portman_midi_input = {
+static const struct snd_rawmidi_ops snd_portman_midi_input = {
        .open =         snd_portman_midi_open,
        .close =        snd_portman_midi_close,
        .trigger =      snd_portman_midi_input_trigger,
diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c
index 1927b89e1d1f..60d51ac4ccfe 100644
--- a/sound/drivers/serial-u16550.c
+++ b/sound/drivers/serial-u16550.c
@@ -752,14 +752,14 @@ static void snd_uart16550_output_trigger(struct snd_rawmidi_substream *substream
                snd_uart16550_output_write(substream);
 }
-static struct snd_rawmidi_ops snd_uart16550_output =
+static const struct snd_rawmidi_ops snd_uart16550_output =
 {
        .open =         snd_uart16550_output_open,
        .close =        snd_uart16550_output_close,
        .trigger =      snd_uart16550_output_trigger,
 };
-static struct snd_rawmidi_ops snd_uart16550_input =
+static const struct snd_rawmidi_ops snd_uart16550_input =
 {
        .open =         snd_uart16550_input_open,
        .close =        snd_uart16550_input_close,
diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c
index 11467272089e..ea7b377f0378 100644
--- a/sound/drivers/vx/vx_pcm.c
+++ b/sound/drivers/vx/vx_pcm.c
@@ -1015,7 +1015,7 @@ static void vx_pcm_capture_update(struct vx_core *chip, struct snd_pcm_substream
        int size, space, count;
        struct snd_pcm_runtime *runtime = subs->runtime;
-        if (! pipe->prepared || (chip->chip_status & VX_STAT_IS_STALE))
+        if (!pipe->running || (chip->chip_status & VX_STAT_IS_STALE))
                return;
        size = runtime->buffer_size - snd_pcm_capture_avail(runtime);
@@ -1048,8 +1048,10 @@ static void vx_pcm_capture_update(struct vx_core *chip, struct snd_pcm_substream
                /* ok, let's accelerate! */
                int align = pipe->align * 3;
                space = (count / align) * align;
-                vx_pseudo_dma_read(chip, runtime, pipe, space);
+                if (space > 0) {
-                count -= space;
+                        vx_pseudo_dma_read(chip, runtime, pipe, space);
+                        count -= space;
+                }
        }
        /* read the rest of bytes */
        while (count > 0) {
diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig
index ab894ed1ff67..9f00696c4e4a 100644
--- a/sound/firewire/Kconfig
+++ b/sound/firewire/Kconfig
@@ -34,6 +34,7 @@ config SND_OXFW
           * LaCie Firewire Speakers
           * Behringer F-Control Audio 202
           * Mackie(Loud) Onyx-i series (former models)
+           * Mackie(Loud) Onyx 1640i (former model)
           * Mackie(Loud) Onyx Satellite
           * Mackie(Loud) Tapco Link.Firewire
           * Mackie(Loud) d.2 pro/d.4 pro
diff --git a/sound/firewire/bebob/bebob_hwdep.c b/sound/firewire/bebob/bebob_hwdep.c
index ce731f4d8b4f..2b367c21b80c 100644
--- a/sound/firewire/bebob/bebob_hwdep.c
+++ b/sound/firewire/bebob/bebob_hwdep.c
@@ -172,16 +172,15 @@ hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file,
 #define hwdep_compat_ioctl NULL
 #endif
-static const struct snd_hwdep_ops hwdep_ops = {
-        .read           = hwdep_read,
-        .release        = hwdep_release,
-        .poll           = hwdep_poll,
-        .ioctl          = hwdep_ioctl,
-        .ioctl_compat   = hwdep_compat_ioctl,
-};
 int snd_bebob_create_hwdep_device(struct snd_bebob *bebob)
 {
+        static const struct snd_hwdep_ops ops = {
+                .read           = hwdep_read,
+                .release        = hwdep_release,
+                .poll           = hwdep_poll,
+                .ioctl          = hwdep_ioctl,
+                .ioctl_compat   = hwdep_compat_ioctl,
+        };
        struct snd_hwdep *hwdep;
        int err;
@@ -190,7 +189,7 @@ int snd_bebob_create_hwdep_device(struct snd_bebob *bebob)
                goto end;
        strcpy(hwdep->name, "BeBoB");
        hwdep->iface = SNDRV_HWDEP_IFACE_FW_BEBOB;
-        hwdep->ops = hwdep_ops;
+        hwdep->ops = ops;
        hwdep->private_data = bebob;
        hwdep->exclusive = true;
 end:
diff --git a/sound/firewire/bebob/bebob_midi.c b/sound/firewire/bebob/bebob_midi.c
index 868eb0decbec..3befa3eca6ef 100644
--- a/sound/firewire/bebob/bebob_midi.c
+++ b/sound/firewire/bebob/bebob_midi.c
@@ -106,18 +106,6 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
        spin_unlock_irqrestore(&bebob->lock, flags);
 }
-static struct snd_rawmidi_ops midi_capture_ops = {
-        .open           = midi_capture_open,
-        .close          = midi_capture_close,
-        .trigger        = midi_capture_trigger,
-};
-static struct snd_rawmidi_ops midi_playback_ops = {
-        .open           = midi_playback_open,
-        .close          = midi_playback_close,
-        .trigger        = midi_playback_trigger,
-};
 static void set_midi_substream_names(struct snd_bebob *bebob,
                                     struct snd_rawmidi_str *str)
 {
@@ -132,6 +120,16 @@ static void set_midi_substream_names(struct snd_bebob *bebob,
 int snd_bebob_create_midi_devices(struct snd_bebob *bebob)
 {
+        static const struct snd_rawmidi_ops capture_ops = {
+                .open           = midi_capture_open,
+                .close          = midi_capture_close,
+                .trigger        = midi_capture_trigger,
+        };
+        static const struct snd_rawmidi_ops playback_ops = {
+                .open           = midi_playback_open,
+                .close          = midi_playback_close,
+                .trigger        = midi_playback_trigger,
+        };
        struct snd_rawmidi *rmidi;
        struct snd_rawmidi_str *str;
        int err;
@@ -151,7 +149,7 @@ int snd_bebob_create_midi_devices(struct snd_bebob *bebob)
                rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT;
                snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
-                                    &midi_capture_ops);
+                                    &capture_ops);
                str = &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT];
@@ -162,7 +160,7 @@ int snd_bebob_create_midi_devices(struct snd_bebob *bebob)
                rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT;
                snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
-                                    &midi_playback_ops);
+                                    &playback_ops);
                str = &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT];
diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c
index 5d7b9343fa85..9e27eb8e1dd4 100644
--- a/sound/firewire/bebob/bebob_pcm.c
+++ b/sound/firewire/bebob/bebob_pcm.c
@@ -359,32 +359,31 @@ pcm_playback_pointer(struct snd_pcm_substream *sbstrm)
        return amdtp_stream_pcm_pointer(&bebob->rx_stream);
 }
-static const struct snd_pcm_ops pcm_capture_ops = {
-        .open           = pcm_open,
-        .close          = pcm_close,
-        .ioctl          = snd_pcm_lib_ioctl,
-        .hw_params      = pcm_capture_hw_params,
-        .hw_free        = pcm_capture_hw_free,
-        .prepare        = pcm_capture_prepare,
-        .trigger        = pcm_capture_trigger,
-        .pointer        = pcm_capture_pointer,
-        .page           = snd_pcm_lib_get_vmalloc_page,
-};
-static const struct snd_pcm_ops pcm_playback_ops = {
-        .open           = pcm_open,
-        .close          = pcm_close,
-        .ioctl          = snd_pcm_lib_ioctl,
-        .hw_params      = pcm_playback_hw_params,
-        .hw_free        = pcm_playback_hw_free,
-        .prepare        = pcm_playback_prepare,
-        .trigger        = pcm_playback_trigger,
-        .pointer        = pcm_playback_pointer,
-        .page           = snd_pcm_lib_get_vmalloc_page,
-        .mmap           = snd_pcm_lib_mmap_vmalloc,
-};
 int snd_bebob_create_pcm_devices(struct snd_bebob *bebob)
 {
+        static const struct snd_pcm_ops capture_ops = {
+                .open           = pcm_open,
+                .close          = pcm_close,
+                .ioctl          = snd_pcm_lib_ioctl,
+                .hw_params      = pcm_capture_hw_params,
+                .hw_free        = pcm_capture_hw_free,
+                .prepare        = pcm_capture_prepare,
+                .trigger        = pcm_capture_trigger,
+                .pointer        = pcm_capture_pointer,
+                .page           = snd_pcm_lib_get_vmalloc_page,
+        };
+        static const struct snd_pcm_ops playback_ops = {
+                .open           = pcm_open,
+                .close          = pcm_close,
+                .ioctl          = snd_pcm_lib_ioctl,
+                .hw_params      = pcm_playback_hw_params,
+                .hw_free        = pcm_playback_hw_free,
+                .prepare        = pcm_playback_prepare,
+                .trigger        = pcm_playback_trigger,
+                .pointer        = pcm_playback_pointer,
+                .page           = snd_pcm_lib_get_vmalloc_page,
+                .mmap           = snd_pcm_lib_mmap_vmalloc,
+        };
        struct snd_pcm *pcm;
        int err;
@@ -395,8 +394,8 @@ int snd_bebob_create_pcm_devices(struct snd_bebob *bebob)
        pcm->private_data = bebob;
        snprintf(pcm->name, sizeof(pcm->name),
                 "%s PCM", bebob->card->shortname);
-        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_playback_ops);
+        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_ops);
-        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_capture_ops);
+        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_ops);
 end:
        return err;
 }
diff --git a/sound/firewire/dice/dice-interface.h b/sound/firewire/dice/dice-interface.h
index 27b044f84c81..47f2c0a6f5d9 100644
--- a/sound/firewire/dice/dice-interface.h
+++ b/sound/firewire/dice/dice-interface.h
@@ -251,6 +251,7 @@
 /*
 * The speed at which the packets are sent, SCODE_100-_400; read/write.
+ * SCODE_800 is only available in Dice III.
 */
 #define TX_SPEED                        0x014
diff --git a/sound/firewire/dice/dice-midi.c b/sound/firewire/dice/dice-midi.c
index a040617505a7..8ff6da3c51f7 100644
--- a/sound/firewire/dice/dice-midi.c
+++ b/sound/firewire/dice/dice-midi.c
@@ -78,18 +78,6 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
        spin_unlock_irqrestore(&dice->lock, flags);
 }
-static struct snd_rawmidi_ops capture_ops = {
-        .open           = midi_open,
-        .close          = midi_close,
-        .trigger        = midi_capture_trigger,
-};
-static struct snd_rawmidi_ops playback_ops = {
-        .open           = midi_open,
-        .close          = midi_close,
-        .trigger        = midi_playback_trigger,
-};
 static void set_midi_substream_names(struct snd_dice *dice,
                                     struct snd_rawmidi_str *str)
 {
@@ -103,6 +91,16 @@ static void set_midi_substream_names(struct snd_dice *dice,
 int snd_dice_create_midi(struct snd_dice *dice)
 {
+        static const struct snd_rawmidi_ops capture_ops = {
+                .open           = midi_open,
+                .close          = midi_close,
+                .trigger        = midi_capture_trigger,
+        };
+        static const struct snd_rawmidi_ops playback_ops = {
+                .open           = midi_open,
+                .close          = midi_close,
+                .trigger        = midi_playback_trigger,
+        };
        __be32 reg;
        struct snd_rawmidi *rmidi;
        struct snd_rawmidi_str *str;
diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c
index ec4db3a514fc..8573289c381e 100644
--- a/sound/firewire/dice/dice-stream.c
+++ b/sound/firewire/dice/dice-stream.c
@@ -195,6 +195,7 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir,
        unsigned int i, pcm_chs, midi_ports;
        struct amdtp_stream *streams;
        struct fw_iso_resources *resources;
+        struct fw_device *fw_dev = fw_parent_device(dice->unit);
        int err = 0;
        if (dir == AMDTP_IN_STREAM) {
@@ -237,8 +238,17 @@ static int start_streams(struct snd_dice *dice, enum amdtp_stream_direction dir,
                if (err < 0)
                        return err;
+                if (dir == AMDTP_IN_STREAM) {
+                        reg[0] = cpu_to_be32(fw_dev->max_speed);
+                        err = snd_dice_transaction_write_tx(dice,
+                                        params->size * i + TX_SPEED,
+                                        reg, sizeof(reg[0]));
+                        if (err < 0)
+                                return err;
+                }
                err = amdtp_stream_start(&streams[i], resources[i].channel,
-                                fw_parent_device(dice->unit)->max_speed);
+                                         fw_dev->max_speed);
                if (err < 0)
                        return err;
        }
diff --git a/sound/firewire/digi00x/digi00x-hwdep.c b/sound/firewire/digi00x/digi00x-hwdep.c
index f188e4758fd2..463c6b8e864d 100644
--- a/sound/firewire/digi00x/digi00x-hwdep.c
+++ b/sound/firewire/digi00x/digi00x-hwdep.c
@@ -173,16 +173,15 @@ static int hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file,
 #define hwdep_compat_ioctl NULL
 #endif
-static const struct snd_hwdep_ops hwdep_ops = {
-        .read           = hwdep_read,
-        .release        = hwdep_release,
-        .poll           = hwdep_poll,
-        .ioctl          = hwdep_ioctl,
-        .ioctl_compat   = hwdep_compat_ioctl,
-};
 int snd_dg00x_create_hwdep_device(struct snd_dg00x *dg00x)
 {
+        static const struct snd_hwdep_ops ops = {
+                .read           = hwdep_read,
+                .release        = hwdep_release,
+                .poll           = hwdep_poll,
+                .ioctl          = hwdep_ioctl,
+                .ioctl_compat   = hwdep_compat_ioctl,
+        };
        struct snd_hwdep *hwdep;
        int err;
@@ -192,7 +191,7 @@ int snd_dg00x_create_hwdep_device(struct snd_dg00x *dg00x)
        strcpy(hwdep->name, "Digi00x");
        hwdep->iface = SNDRV_HWDEP_IFACE_FW_DIGI00X;
-        hwdep->ops = hwdep_ops;
+        hwdep->ops = ops;
        hwdep->private_data = dg00x;
        hwdep->exclusive = true;
diff --git a/sound/firewire/digi00x/digi00x-midi.c b/sound/firewire/digi00x/digi00x-midi.c
index 1a72a382b384..915d2a21223e 100644
--- a/sound/firewire/digi00x/digi00x-midi.c
+++ b/sound/firewire/digi00x/digi00x-midi.c
@@ -76,18 +76,6 @@ static void midi_phys_playback_trigger(struct snd_rawmidi_substream *substream,
        spin_unlock_irqrestore(&dg00x->lock, flags);
 }
-static struct snd_rawmidi_ops midi_phys_capture_ops = {
-        .open           = midi_phys_open,
-        .close          = midi_phys_close,
-        .trigger        = midi_phys_capture_trigger,
-};
-static struct snd_rawmidi_ops midi_phys_playback_ops = {
-        .open           = midi_phys_open,
-        .close          = midi_phys_close,
-        .trigger        = midi_phys_playback_trigger,
-};
 static int midi_ctl_open(struct snd_rawmidi_substream *substream)
 {
        /* Do nothing. */
@@ -139,18 +127,6 @@ static void midi_ctl_playback_trigger(struct snd_rawmidi_substream *substream,
        spin_unlock_irqrestore(&dg00x->lock, flags);
 }
-static struct snd_rawmidi_ops midi_ctl_capture_ops = {
-        .open           = midi_ctl_open,
-        .close          = midi_ctl_capture_close,
-        .trigger        = midi_ctl_capture_trigger,
-};
-static struct snd_rawmidi_ops midi_ctl_playback_ops = {
-        .open           = midi_ctl_open,
-        .close          = midi_ctl_playback_close,
-        .trigger        = midi_ctl_playback_trigger,
-};
 static void set_midi_substream_names(struct snd_dg00x *dg00x,
                                     struct snd_rawmidi_str *str,
                                     bool is_ctl)
@@ -172,6 +148,26 @@ static void set_midi_substream_names(struct snd_dg00x *dg00x,
 int snd_dg00x_create_midi_devices(struct snd_dg00x *dg00x)
 {
+        static const struct snd_rawmidi_ops phys_capture_ops = {
+                .open           = midi_phys_open,
+                .close          = midi_phys_close,
+                .trigger        = midi_phys_capture_trigger,
+        };
+        static const struct snd_rawmidi_ops phys_playback_ops = {
+                .open           = midi_phys_open,
+                .close          = midi_phys_close,
+                .trigger        = midi_phys_playback_trigger,
+        };
+        static const struct snd_rawmidi_ops ctl_capture_ops = {
+                .open           = midi_ctl_open,
+                .close          = midi_ctl_capture_close,
+                .trigger        = midi_ctl_capture_trigger,
+        };
+        static const struct snd_rawmidi_ops ctl_playback_ops = {
+                .open           = midi_ctl_open,
+                .close          = midi_ctl_playback_close,
+                .trigger        = midi_ctl_playback_trigger,
+        };
        struct snd_rawmidi *rmidi[2];
        struct snd_rawmidi_str *str;
        unsigned int i;
@@ -187,9 +183,9 @@ int snd_dg00x_create_midi_devices(struct snd_dg00x *dg00x)
                 "%s MIDI", dg00x->card->shortname);
        snd_rawmidi_set_ops(rmidi[0], SNDRV_RAWMIDI_STREAM_INPUT,
-                            &midi_phys_capture_ops);
+                            &phys_capture_ops);
        snd_rawmidi_set_ops(rmidi[0], SNDRV_RAWMIDI_STREAM_OUTPUT,
-                            &midi_phys_playback_ops);
+                            &phys_playback_ops);
        /* Add a pair of control midi ports. */
        err = snd_rawmidi_new(dg00x->card, dg00x->card->driver, 1,
@@ -201,9 +197,9 @@ int snd_dg00x_create_midi_devices(struct snd_dg00x *dg00x)
                 "%s control", dg00x->card->shortname);
        snd_rawmidi_set_ops(rmidi[1], SNDRV_RAWMIDI_STREAM_INPUT,
-                            &midi_ctl_capture_ops);
+                            &ctl_capture_ops);
        snd_rawmidi_set_ops(rmidi[1], SNDRV_RAWMIDI_STREAM_OUTPUT,
-                            &midi_ctl_playback_ops);
+                            &ctl_playback_ops);
        for (i = 0; i < ARRAY_SIZE(rmidi); i++) {
                rmidi[i]->private_data = dg00x;
diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c
index 613f05872770..68d1c52db051 100644
--- a/sound/firewire/digi00x/digi00x-pcm.c
+++ b/sound/firewire/digi00x/digi00x-pcm.c
@@ -329,33 +329,31 @@ static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm)
        return amdtp_stream_pcm_pointer(&dg00x->rx_stream);
 }
-static const struct snd_pcm_ops pcm_capture_ops = {
-        .open           = pcm_open,
-        .close          = pcm_close,
-        .ioctl          = snd_pcm_lib_ioctl,
-        .hw_params      = pcm_capture_hw_params,
-        .hw_free        = pcm_capture_hw_free,
-        .prepare        = pcm_capture_prepare,
-        .trigger        = pcm_capture_trigger,
-        .pointer        = pcm_capture_pointer,
-        .page           = snd_pcm_lib_get_vmalloc_page,
-};
-static const struct snd_pcm_ops pcm_playback_ops = {
-        .open           = pcm_open,
-        .close          = pcm_close,
-        .ioctl          = snd_pcm_lib_ioctl,
-        .hw_params      = pcm_playback_hw_params,
-        .hw_free        = pcm_playback_hw_free,
-        .prepare        = pcm_playback_prepare,
-        .trigger        = pcm_playback_trigger,
-        .pointer        = pcm_playback_pointer,
-        .page           = snd_pcm_lib_get_vmalloc_page,
-        .mmap           = snd_pcm_lib_mmap_vmalloc,
-};
 int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x)
 {
+        static const struct snd_pcm_ops capture_ops = {
+                .open           = pcm_open,
+                .close          = pcm_close,
+                .ioctl          = snd_pcm_lib_ioctl,
+                .hw_params      = pcm_capture_hw_params,
+                .hw_free        = pcm_capture_hw_free,
+                .prepare        = pcm_capture_prepare,
+                .trigger        = pcm_capture_trigger,
+                .pointer        = pcm_capture_pointer,
+                .page           = snd_pcm_lib_get_vmalloc_page,
+        };
+        static const struct snd_pcm_ops playback_ops = {
+                .open           = pcm_open,
+                .close          = pcm_close,
+                .ioctl          = snd_pcm_lib_ioctl,
+                .hw_params      = pcm_playback_hw_params,
+                .hw_free        = pcm_playback_hw_free,
+                .prepare        = pcm_playback_prepare,
+                .trigger        = pcm_playback_trigger,
+                .pointer        = pcm_playback_pointer,
+                .page           = snd_pcm_lib_get_vmalloc_page,
+                .mmap           = snd_pcm_lib_mmap_vmalloc,
+        };
        struct snd_pcm *pcm;
        int err;
@@ -366,8 +364,8 @@ int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x)
        pcm->private_data = dg00x;
        snprintf(pcm->name, sizeof(pcm->name),
                 "%s PCM", dg00x->card->shortname);
-        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_playback_ops);
+        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_ops);
-        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_capture_ops);
+        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_ops);
        return 0;
 }
diff --git a/sound/firewire/fireworks/fireworks_hwdep.c b/sound/firewire/fireworks/fireworks_hwdep.c
index 2e1d9a23920c..a3a3a16f5e08 100644
--- a/sound/firewire/fireworks/fireworks_hwdep.c
+++ b/sound/firewire/fireworks/fireworks_hwdep.c
@@ -303,17 +303,16 @@ hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file,
 #define hwdep_compat_ioctl NULL
 #endif
-static const struct snd_hwdep_ops hwdep_ops = {
-        .read           = hwdep_read,
-        .write          = hwdep_write,
-        .release        = hwdep_release,
-        .poll           = hwdep_poll,
-        .ioctl          = hwdep_ioctl,
-        .ioctl_compat   = hwdep_compat_ioctl,
-};
 int snd_efw_create_hwdep_device(struct snd_efw *efw)
 {
+        static const struct snd_hwdep_ops ops = {
+                .read           = hwdep_read,
+                .write          = hwdep_write,
+                .release        = hwdep_release,
+                .poll           = hwdep_poll,
+                .ioctl          = hwdep_ioctl,
+                .ioctl_compat   = hwdep_compat_ioctl,
+        };
        struct snd_hwdep *hwdep;
        int err;
@@ -322,7 +321,7 @@ int snd_efw_create_hwdep_device(struct snd_efw *efw)
                goto end;
        strcpy(hwdep->name, "Fireworks");
        hwdep->iface = SNDRV_HWDEP_IFACE_FW_FIREWORKS;
-        hwdep->ops = hwdep_ops;
+        hwdep->ops = ops;
        hwdep->private_data = efw;
        hwdep->exclusive = true;
 end:
diff --git a/sound/firewire/fireworks/fireworks_midi.c b/sound/firewire/fireworks/fireworks_midi.c
index 3e8c4cf9fe1e..f5da2cd4ce42 100644
--- a/sound/firewire/fireworks/fireworks_midi.c
+++ b/sound/firewire/fireworks/fireworks_midi.c
@@ -107,18 +107,6 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
        spin_unlock_irqrestore(&efw->lock, flags);
 }
-static struct snd_rawmidi_ops midi_capture_ops = {
-        .open           = midi_capture_open,
-        .close          = midi_capture_close,
-        .trigger        = midi_capture_trigger,
-};
-static struct snd_rawmidi_ops midi_playback_ops = {
-        .open           = midi_playback_open,
-        .close          = midi_playback_close,
-        .trigger        = midi_playback_trigger,
-};
 static void set_midi_substream_names(struct snd_efw *efw,
                                     struct snd_rawmidi_str *str)
 {
@@ -132,6 +120,16 @@ static void set_midi_substream_names(struct snd_efw *efw,
 int snd_efw_create_midi_devices(struct snd_efw *efw)
 {
+        static const struct snd_rawmidi_ops capture_ops = {
+                .open           = midi_capture_open,
+                .close          = midi_capture_close,
+                .trigger        = midi_capture_trigger,
+        };
+        static const struct snd_rawmidi_ops playback_ops = {
+                .open           = midi_playback_open,
+                .close          = midi_playback_close,
+                .trigger        = midi_playback_trigger,
+        };
        struct snd_rawmidi *rmidi;
        struct snd_rawmidi_str *str;
        int err;
@@ -151,7 +149,7 @@ int snd_efw_create_midi_devices(struct snd_efw *efw)
                rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT;
                snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
-                                    &midi_capture_ops);
+                                    &capture_ops);
                str = &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT];
@@ -162,7 +160,7 @@ int snd_efw_create_midi_devices(struct snd_efw *efw)
                rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT;
                snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
-                                    &midi_playback_ops);
+                                    &playback_ops);
                str = &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT];
diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c
index f4fbf75ed198..9171702f7d0b 100644
--- a/sound/firewire/fireworks/fireworks_pcm.c
+++ b/sound/firewire/fireworks/fireworks_pcm.c
@@ -383,33 +383,31 @@ static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm)
        return amdtp_stream_pcm_pointer(&efw->rx_stream);
 }
-static const struct snd_pcm_ops pcm_capture_ops = {
-        .open           = pcm_open,
-        .close          = pcm_close,
-        .ioctl          = snd_pcm_lib_ioctl,
-        .hw_params      = pcm_capture_hw_params,
-        .hw_free        = pcm_capture_hw_free,
-        .prepare        = pcm_capture_prepare,
-        .trigger        = pcm_capture_trigger,
-        .pointer        = pcm_capture_pointer,
-        .page           = snd_pcm_lib_get_vmalloc_page,
-};
-static const struct snd_pcm_ops pcm_playback_ops = {
-        .open           = pcm_open,
-        .close          = pcm_close,
-        .ioctl          = snd_pcm_lib_ioctl,
-        .hw_params      = pcm_playback_hw_params,
-        .hw_free        = pcm_playback_hw_free,
-        .prepare        = pcm_playback_prepare,
-        .trigger        = pcm_playback_trigger,
-        .pointer        = pcm_playback_pointer,
-        .page           = snd_pcm_lib_get_vmalloc_page,
-        .mmap           = snd_pcm_lib_mmap_vmalloc,
-};
 int snd_efw_create_pcm_devices(struct snd_efw *efw)
 {
+        static const struct snd_pcm_ops capture_ops = {
+                .open           = pcm_open,
+                .close          = pcm_close,
+                .ioctl          = snd_pcm_lib_ioctl,
+                .hw_params      = pcm_capture_hw_params,
+                .hw_free        = pcm_capture_hw_free,
+                .prepare        = pcm_capture_prepare,
+                .trigger        = pcm_capture_trigger,
+                .pointer        = pcm_capture_pointer,
+                .page           = snd_pcm_lib_get_vmalloc_page,
+        };
+        static const struct snd_pcm_ops playback_ops = {
+                .open           = pcm_open,
+                .close          = pcm_close,
+                .ioctl          = snd_pcm_lib_ioctl,
+                .hw_params      = pcm_playback_hw_params,
+                .hw_free        = pcm_playback_hw_free,
+                .prepare        = pcm_playback_prepare,
+                .trigger        = pcm_playback_trigger,
+                .pointer        = pcm_playback_pointer,
+                .page           = snd_pcm_lib_get_vmalloc_page,
+                .mmap           = snd_pcm_lib_mmap_vmalloc,
+        };
        struct snd_pcm *pcm;
        int err;
@@ -419,8 +417,8 @@ int snd_efw_create_pcm_devices(struct snd_efw *efw)
        pcm->private_data = efw;
        snprintf(pcm->name, sizeof(pcm->name), "%s PCM", efw->card->shortname);
-        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_playback_ops);
+        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_ops);
-        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_capture_ops);
+        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_ops);
 end:
        return err;
 }
diff --git a/sound/firewire/oxfw/oxfw-midi.c b/sound/firewire/oxfw/oxfw-midi.c
index 8665e1043d41..b7bbd77dfff1 100644
--- a/sound/firewire/oxfw/oxfw-midi.c
+++ b/sound/firewire/oxfw/oxfw-midi.c
@@ -116,18 +116,6 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
        spin_unlock_irqrestore(&oxfw->lock, flags);
 }
-static struct snd_rawmidi_ops midi_capture_ops = {
-        .open           = midi_capture_open,
-        .close          = midi_capture_close,
-        .trigger        = midi_capture_trigger,
-};
-static struct snd_rawmidi_ops midi_playback_ops = {
-        .open           = midi_playback_open,
-        .close          = midi_playback_close,
-        .trigger        = midi_playback_trigger,
-};
 static void set_midi_substream_names(struct snd_oxfw *oxfw,
                                     struct snd_rawmidi_str *str)
 {
@@ -142,6 +130,16 @@ static void set_midi_substream_names(struct snd_oxfw *oxfw,
 int snd_oxfw_create_midi(struct snd_oxfw *oxfw)
 {
+        static const struct snd_rawmidi_ops capture_ops = {
+                .open           = midi_capture_open,
+                .close          = midi_capture_close,
+                .trigger        = midi_capture_trigger,
+        };
+        static const struct snd_rawmidi_ops playback_ops = {
+                .open           = midi_playback_open,
+                .close          = midi_playback_close,
+                .trigger        = midi_playback_trigger,
+        };
        struct snd_rawmidi *rmidi;
        struct snd_rawmidi_str *str;
        int err;
@@ -164,7 +162,7 @@ int snd_oxfw_create_midi(struct snd_oxfw *oxfw)
                rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT;
                snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
-                                    &midi_capture_ops);
+                                    &capture_ops);
                str = &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT];
@@ -175,7 +173,7 @@ int snd_oxfw_create_midi(struct snd_oxfw *oxfw)
                rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT;
                snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
-                                    &midi_playback_ops);
+                                    &playback_ops);
                str = &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT];
diff --git a/sound/firewire/oxfw/oxfw-scs1x.c b/sound/firewire/oxfw/oxfw-scs1x.c
index f897c9831077..93209ebd9121 100644
--- a/sound/firewire/oxfw/oxfw-scs1x.c
+++ b/sound/firewire/oxfw/oxfw-scs1x.c
@@ -297,7 +297,7 @@ static void midi_capture_trigger(struct snd_rawmidi_substream *stream, int up)
        }
 }
-static struct snd_rawmidi_ops midi_capture_ops = {
+static const struct snd_rawmidi_ops midi_capture_ops = {
        .open    = midi_capture_open,
        .close   = midi_capture_close,
        .trigger = midi_capture_trigger,
@@ -338,12 +338,6 @@ static void midi_playback_drain(struct snd_rawmidi_substream *stream)
        wait_event(scs->idle_wait, scs->output_idle);
 }
-static struct snd_rawmidi_ops midi_playback_ops = {
-        .open    = midi_playback_open,
-        .close   = midi_playback_close,
-        .trigger = midi_playback_trigger,
-        .drain   = midi_playback_drain,
-};
 static int register_address(struct snd_oxfw *oxfw)
 {
        struct fw_scs1x *scs = oxfw->spec;
@@ -369,6 +363,12 @@ void snd_oxfw_scs1x_update(struct snd_oxfw *oxfw)
 int snd_oxfw_scs1x_add(struct snd_oxfw *oxfw)
 {
+        static const struct snd_rawmidi_ops midi_playback_ops = {
+                .open    = midi_playback_open,
+                .close   = midi_playback_close,
+                .trigger = midi_playback_trigger,
+                .drain   = midi_playback_drain,
+        };
        struct snd_rawmidi *rmidi;
        struct fw_scs1x *scs;
        int err;
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index e629b88f7d93..74d7fb6efce6 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -43,6 +43,7 @@ static bool detect_loud_models(struct fw_unit *unit)
        const char *const models[] = {
                "Onyxi",
                "Onyx-i",
+                "Onyx 1640i",
                "d.Pro",
                "Mackie Onyx Satellite",
                "Tapco LINK.firewire 4x6",
diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c
index 106406cbfaa3..8c4437d0051d 100644
--- a/sound/firewire/tascam/tascam-hwdep.c
+++ b/sound/firewire/tascam/tascam-hwdep.c
@@ -163,16 +163,15 @@ static int hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file,
 #define hwdep_compat_ioctl NULL
 #endif
-static const struct snd_hwdep_ops hwdep_ops = {
-        .read           = hwdep_read,
-        .release        = hwdep_release,
-        .poll           = hwdep_poll,
-        .ioctl          = hwdep_ioctl,
-        .ioctl_compat   = hwdep_compat_ioctl,
-};
 int snd_tscm_create_hwdep_device(struct snd_tscm *tscm)
 {
+        static const struct snd_hwdep_ops ops = {
+                .read           = hwdep_read,
+                .release        = hwdep_release,
+                .poll           = hwdep_poll,
+                .ioctl          = hwdep_ioctl,
+                .ioctl_compat   = hwdep_compat_ioctl,
+        };
        struct snd_hwdep *hwdep;
        int err;
@@ -182,7 +181,7 @@ int snd_tscm_create_hwdep_device(struct snd_tscm *tscm)
        strcpy(hwdep->name, "Tascam");
        hwdep->iface = SNDRV_HWDEP_IFACE_FW_TASCAM;
-        hwdep->ops = hwdep_ops;
+        hwdep->ops = ops;
        hwdep->private_data = tscm;
        hwdep->exclusive = true;
diff --git a/sound/firewire/tascam/tascam-midi.c b/sound/firewire/tascam/tascam-midi.c
index 41f842079d9d..df4f95d65925 100644
--- a/sound/firewire/tascam/tascam-midi.c
+++ b/sound/firewire/tascam/tascam-midi.c
@@ -68,20 +68,18 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
        spin_unlock_irqrestore(&tscm->lock, flags);
 }
-static struct snd_rawmidi_ops midi_capture_ops = {
-        .open           = midi_capture_open,
-        .close          = midi_capture_close,
-        .trigger        = midi_capture_trigger,
-};
-static struct snd_rawmidi_ops midi_playback_ops = {
-        .open           = midi_playback_open,
-        .close          = midi_playback_close,
-        .trigger        = midi_playback_trigger,
-};
 int snd_tscm_create_midi_devices(struct snd_tscm *tscm)
 {
+        static const struct snd_rawmidi_ops capture_ops = {
+                .open           = midi_capture_open,
+                .close          = midi_capture_close,
+                .trigger        = midi_capture_trigger,
+        };
+        static const struct snd_rawmidi_ops playback_ops = {
+                .open           = midi_playback_open,
+                .close          = midi_playback_close,
+                .trigger        = midi_playback_trigger,
+        };
        struct snd_rawmidi *rmidi;
        struct snd_rawmidi_str *stream;
        struct snd_rawmidi_substream *subs;
@@ -100,7 +98,7 @@ int snd_tscm_create_midi_devices(struct snd_tscm *tscm)
        rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT;
        snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
-                            &midi_capture_ops);
+                            &capture_ops);
        stream = &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT];
        /* Set port names for MIDI input. */
@@ -116,7 +114,7 @@ int snd_tscm_create_midi_devices(struct snd_tscm *tscm)
        rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT;
        snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
-                            &midi_playback_ops);
+                            &playback_ops);
        stream = &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT];
        /* Set port names for MIDI ourput. */
diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c
index 79db1b651f5c..f5dd6ce6b6f1 100644
--- a/sound/firewire/tascam/tascam-pcm.c
+++ b/sound/firewire/tascam/tascam-pcm.c
@@ -268,33 +268,31 @@ static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm)
        return amdtp_stream_pcm_pointer(&tscm->rx_stream);
 }
-static const struct snd_pcm_ops pcm_capture_ops = {
-        .open           = pcm_open,
-        .close          = pcm_close,
-        .ioctl          = snd_pcm_lib_ioctl,
-        .hw_params      = pcm_capture_hw_params,
-        .hw_free        = pcm_capture_hw_free,
-        .prepare        = pcm_capture_prepare,
-        .trigger        = pcm_capture_trigger,
-        .pointer        = pcm_capture_pointer,
-        .page           = snd_pcm_lib_get_vmalloc_page,
-};
-static const struct snd_pcm_ops pcm_playback_ops = {
-        .open           = pcm_open,
-        .close          = pcm_close,
-        .ioctl          = snd_pcm_lib_ioctl,
-        .hw_params      = pcm_playback_hw_params,
-        .hw_free        = pcm_playback_hw_free,
-        .prepare        = pcm_playback_prepare,
-        .trigger        = pcm_playback_trigger,
-        .pointer        = pcm_playback_pointer,
-        .page           = snd_pcm_lib_get_vmalloc_page,
-        .mmap           = snd_pcm_lib_mmap_vmalloc,
-};
 int snd_tscm_create_pcm_devices(struct snd_tscm *tscm)
 {
+        static const struct snd_pcm_ops capture_ops = {
+                .open           = pcm_open,
+                .close          = pcm_close,
+                .ioctl          = snd_pcm_lib_ioctl,
+                .hw_params      = pcm_capture_hw_params,
+                .hw_free        = pcm_capture_hw_free,
+                .prepare        = pcm_capture_prepare,
+                .trigger        = pcm_capture_trigger,
+                .pointer        = pcm_capture_pointer,
+                .page           = snd_pcm_lib_get_vmalloc_page,
+        };
+        static const struct snd_pcm_ops playback_ops = {
+                .open           = pcm_open,
+                .close          = pcm_close,
+                .ioctl          = snd_pcm_lib_ioctl,
+                .hw_params      = pcm_playback_hw_params,
+                .hw_free        = pcm_playback_hw_free,
+                .prepare        = pcm_playback_prepare,
+                .trigger        = pcm_playback_trigger,
+                .pointer        = pcm_playback_pointer,
+                .page           = snd_pcm_lib_get_vmalloc_page,
+                .mmap           = snd_pcm_lib_mmap_vmalloc,
+        };
        struct snd_pcm *pcm;
        int err;
@@ -305,8 +303,8 @@ int snd_tscm_create_pcm_devices(struct snd_tscm *tscm)
        pcm->private_data = tscm;
        snprintf(pcm->name, sizeof(pcm->name),
                 "%s PCM", tscm->card->shortname);
-        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_playback_ops);
+        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_ops);
-        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_capture_ops);
+        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_ops);
        return 0;
 }
diff --git a/sound/isa/gus/gus_uart.c b/sound/isa/gus/gus_uart.c
index 3992912743f5..ac5f5687d1a3 100644
--- a/sound/isa/gus/gus_uart.c
+++ b/sound/isa/gus/gus_uart.c
@@ -227,14 +227,14 @@ static void snd_gf1_uart_output_trigger(struct snd_rawmidi_substream *substream,
        spin_unlock_irqrestore(&gus->uart_cmd_lock, flags);
 }
-static struct snd_rawmidi_ops snd_gf1_uart_output =
+static const struct snd_rawmidi_ops snd_gf1_uart_output =
 {
        .open =         snd_gf1_uart_output_open,
        .close =        snd_gf1_uart_output_close,
        .trigger =      snd_gf1_uart_output_trigger,
 };
-static struct snd_rawmidi_ops snd_gf1_uart_input =
+static const struct snd_rawmidi_ops snd_gf1_uart_input =
 {
        .open =         snd_gf1_uart_input_open,
        .close =        snd_gf1_uart_input_close,
diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c
index ffc67fd80c23..912b5a9ccbab 100644
--- a/sound/isa/msnd/msnd_midi.c
+++ b/sound/isa/msnd/msnd_midi.c
@@ -142,7 +142,7 @@ void snd_msndmidi_input_read(void *mpuv)
 }
 EXPORT_SYMBOL(snd_msndmidi_input_read);
-static struct snd_rawmidi_ops snd_msndmidi_input = {
+static const struct snd_rawmidi_ops snd_msndmidi_input = {
        .open =         snd_msndmidi_input_open,
        .close =        snd_msndmidi_input_close,
        .trigger =      snd_msndmidi_input_trigger,
diff --git a/sound/isa/sb/sb8_midi.c b/sound/isa/sb/sb8_midi.c
index d551c50e549f..bd672abb4854 100644
--- a/sound/isa/sb/sb8_midi.c
+++ b/sound/isa/sb/sb8_midi.c
@@ -247,14 +247,14 @@ static void snd_sb8dsp_midi_output_trigger(struct snd_rawmidi_substream *substre
                snd_sb8dsp_midi_output_write(substream);
 }
-static struct snd_rawmidi_ops snd_sb8dsp_midi_output =
+static const struct snd_rawmidi_ops snd_sb8dsp_midi_output =
 {
        .open =         snd_sb8dsp_midi_output_open,
        .close =        snd_sb8dsp_midi_output_close,
        .trigger =      snd_sb8dsp_midi_output_trigger,
 };
-static struct snd_rawmidi_ops snd_sb8dsp_midi_input =
+static const struct snd_rawmidi_ops snd_sb8dsp_midi_input =
 {
        .open =         snd_sb8dsp_midi_input_open,
        .close =        snd_sb8dsp_midi_input_close,
diff --git a/sound/isa/wavefront/wavefront_midi.c b/sound/isa/wavefront/wavefront_midi.c
index 8a80fc6a616b..2aa05f3aaa38 100644
--- a/sound/isa/wavefront/wavefront_midi.c
+++ b/sound/isa/wavefront/wavefront_midi.c
@@ -559,14 +559,14 @@ snd_wavefront_midi_start (snd_wavefront_card_t *card)
        return 0;
 }
-struct snd_rawmidi_ops snd_wavefront_midi_output =
+const struct snd_rawmidi_ops snd_wavefront_midi_output =
 {
        .open =         snd_wavefront_midi_output_open,
        .close =        snd_wavefront_midi_output_close,
        .trigger =      snd_wavefront_midi_output_trigger,
 };
-struct snd_rawmidi_ops snd_wavefront_midi_input =
+const struct snd_rawmidi_ops snd_wavefront_midi_input =
 {
        .open =         snd_wavefront_midi_input_open,
        .close =        snd_wavefront_midi_input_close,
diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c
index ede449f0b50d..00fc9241d266 100644
--- a/sound/mips/hal2.c
+++ b/sound/mips/hal2.c
@@ -219,6 +219,8 @@ static int hal2_gain_get(struct snd_kcontrol *kcontrol,
                l = (tmp >> H2I_C2_L_GAIN_SHIFT) & 15;
                r = (tmp >> H2I_C2_R_GAIN_SHIFT) & 15;
                break;
+        default:
+                return -EINVAL;
        }
        ucontrol->value.integer.value[0] = l;
        ucontrol->value.integer.value[1] = r;
@@ -256,6 +258,8 @@ static int hal2_gain_put(struct snd_kcontrol *kcontrol,
                new |= (r << H2I_C2_R_GAIN_SHIFT);
                hal2_i_write32(hal2, H2I_ADC_C2, new);
                break;
+        default:
+                return -EINVAL;
        }
        return old != new;
 }
diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c
index 6368e5c7d0ba..f6156d8169d0 100644
--- a/sound/oss/ad1848.c
+++ b/sound/oss/ad1848.c
@@ -121,11 +121,6 @@ static bool deskpro_xl;
 static bool deskpro_m;
 static bool soundpro;
-static volatile signed char irq2dev[17] = {
-        -1, -1, -1, -1, -1, -1, -1, -1,
-        -1, -1, -1, -1, -1, -1, -1, -1, -1
-};
 #ifndef EXCLUDE_TIMERS
 static int timer_installed = -1;
 #endif
@@ -2060,7 +2055,7 @@ int ad1848_init (char *name, struct resource *ports, int irq, int dma_playback,
                else
                        devc->irq_ok = 1;       /* Couldn't test. assume it's OK */
        } else if (irq < 0)
-                irq2dev[-irq] = devc->dev_no = my_dev;
+                devc->dev_no = my_dev;
 #ifndef EXCLUDE_TIMERS
        if ((capabilities[devc->model].flags & CAP_F_TIMER) &&
diff --git a/sound/pci/ca0106/ca_midi.c b/sound/pci/ca0106/ca_midi.c
index b91c7f6d19f9..4d4d385205eb 100644
--- a/sound/pci/ca0106/ca_midi.c
+++ b/sound/pci/ca0106/ca_midi.c
@@ -255,14 +255,14 @@ static void ca_midi_output_trigger(struct snd_rawmidi_substream *substream, int
        }
 }
-static struct snd_rawmidi_ops ca_midi_output =
+static const struct snd_rawmidi_ops ca_midi_output =
 {
        .open =         ca_midi_output_open,
        .close =        ca_midi_output_close,
        .trigger =      ca_midi_output_trigger,
 };
-static struct snd_rawmidi_ops ca_midi_input =
+static const struct snd_rawmidi_ops ca_midi_input =
 {
        .open =         ca_midi_input_open,
        .close =        ca_midi_input_close,
diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c
index 8f0f5f24e40e..fa7c51684dd2 100644
--- a/sound/pci/cs4281.c
+++ b/sound/pci/cs4281.c
@@ -1767,14 +1767,14 @@ static void snd_cs4281_midi_output_trigger(struct snd_rawmidi_substream *substre
        spin_unlock_irqrestore(&chip->reg_lock, flags);
 }
-static struct snd_rawmidi_ops snd_cs4281_midi_output =
+static const struct snd_rawmidi_ops snd_cs4281_midi_output =
 {
        .open =         snd_cs4281_midi_output_open,
        .close =        snd_cs4281_midi_output_close,
        .trigger =      snd_cs4281_midi_output_trigger,
 };
-static struct snd_rawmidi_ops snd_cs4281_midi_input =
+static const struct snd_rawmidi_ops snd_cs4281_midi_input =
 {
        .open =         snd_cs4281_midi_input_open,
        .close =        snd_cs4281_midi_input_close,
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index fde3cd48258c..e4cf3187b4dd 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -72,18 +72,18 @@
 static void amp_voyetra(struct snd_cs46xx *chip, int change);
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
-static struct snd_pcm_ops snd_cs46xx_playback_rear_ops;
+static const struct snd_pcm_ops snd_cs46xx_playback_rear_ops;
-static struct snd_pcm_ops snd_cs46xx_playback_indirect_rear_ops;
+static const struct snd_pcm_ops snd_cs46xx_playback_indirect_rear_ops;
-static struct snd_pcm_ops snd_cs46xx_playback_clfe_ops;
+static const struct snd_pcm_ops snd_cs46xx_playback_clfe_ops;
-static struct snd_pcm_ops snd_cs46xx_playback_indirect_clfe_ops;
+static const struct snd_pcm_ops snd_cs46xx_playback_indirect_clfe_ops;
-static struct snd_pcm_ops snd_cs46xx_playback_iec958_ops;
+static const struct snd_pcm_ops snd_cs46xx_playback_iec958_ops;
-static struct snd_pcm_ops snd_cs46xx_playback_indirect_iec958_ops;
+static const struct snd_pcm_ops snd_cs46xx_playback_indirect_iec958_ops;
 #endif
-static struct snd_pcm_ops snd_cs46xx_playback_ops;
+static const struct snd_pcm_ops snd_cs46xx_playback_ops;
-static struct snd_pcm_ops snd_cs46xx_playback_indirect_ops;
+static const struct snd_pcm_ops snd_cs46xx_playback_indirect_ops;
-static struct snd_pcm_ops snd_cs46xx_capture_ops;
+static const struct snd_pcm_ops snd_cs46xx_capture_ops;
-static struct snd_pcm_ops snd_cs46xx_capture_indirect_ops;
+static const struct snd_pcm_ops snd_cs46xx_capture_indirect_ops;
 static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip,
                                            unsigned short reg,
@@ -1654,7 +1654,7 @@ static int snd_cs46xx_capture_close(struct snd_pcm_substream *substream)
 }
 #ifdef CONFIG_SND_CS46XX_NEW_DSP
-static struct snd_pcm_ops snd_cs46xx_playback_rear_ops = {
+static const struct snd_pcm_ops snd_cs46xx_playback_rear_ops = {
        .open =                 snd_cs46xx_playback_open_rear,
        .close =                snd_cs46xx_playback_close,
        .ioctl =                snd_pcm_lib_ioctl,
@@ -1665,7 +1665,7 @@ static struct snd_pcm_ops snd_cs46xx_playback_rear_ops = {
        .pointer =              snd_cs46xx_playback_direct_pointer,
 };
-static struct snd_pcm_ops snd_cs46xx_playback_indirect_rear_ops = {
+static const struct snd_pcm_ops snd_cs46xx_playback_indirect_rear_ops = {
        .open =                 snd_cs46xx_playback_open_rear,
        .close =                snd_cs46xx_playback_close,
        .ioctl =                snd_pcm_lib_ioctl,
@@ -1677,7 +1677,7 @@ static struct snd_pcm_ops snd_cs46xx_playback_indirect_rear_ops = {
        .ack =                  snd_cs46xx_playback_transfer,
 };
-static struct snd_pcm_ops snd_cs46xx_playback_clfe_ops = {
+static const struct snd_pcm_ops snd_cs46xx_playback_clfe_ops = {
        .open =                 snd_cs46xx_playback_open_clfe,
        .close =                snd_cs46xx_playback_close,
        .ioctl =                snd_pcm_lib_ioctl,
@@ -1688,7 +1688,7 @@ static struct snd_pcm_ops snd_cs46xx_playback_clfe_ops = {
        .pointer =              snd_cs46xx_playback_direct_pointer,
 };
-static struct snd_pcm_ops snd_cs46xx_playback_indirect_clfe_ops = {
+static const struct snd_pcm_ops snd_cs46xx_playback_indirect_clfe_ops = {
        .open =                 snd_cs46xx_playback_open_clfe,
        .close =                snd_cs46xx_playback_close,
        .ioctl =                snd_pcm_lib_ioctl,
@@ -1700,7 +1700,7 @@ static struct snd_pcm_ops snd_cs46xx_playback_indirect_clfe_ops = {
        .ack =                  snd_cs46xx_playback_transfer,
 };
-static struct snd_pcm_ops snd_cs46xx_playback_iec958_ops = {
+static const struct snd_pcm_ops snd_cs46xx_playback_iec958_ops = {
        .open =                 snd_cs46xx_playback_open_iec958,
        .close =                snd_cs46xx_playback_close_iec958,
        .ioctl =                snd_pcm_lib_ioctl,
@@ -1711,7 +1711,7 @@ static struct snd_pcm_ops snd_cs46xx_playback_iec958_ops = {
        .pointer =              snd_cs46xx_playback_direct_pointer,
 };
-static struct snd_pcm_ops snd_cs46xx_playback_indirect_iec958_ops = {
+static const struct snd_pcm_ops snd_cs46xx_playback_indirect_iec958_ops = {
        .open =                 snd_cs46xx_playback_open_iec958,
        .close =                snd_cs46xx_playback_close_iec958,
        .ioctl =                snd_pcm_lib_ioctl,
@@ -1725,7 +1725,7 @@ static struct snd_pcm_ops snd_cs46xx_playback_indirect_iec958_ops = {
 #endif
-static struct snd_pcm_ops snd_cs46xx_playback_ops = {
+static const struct snd_pcm_ops snd_cs46xx_playback_ops = {
        .open =                 snd_cs46xx_playback_open,
        .close =                snd_cs46xx_playback_close,
        .ioctl =                snd_pcm_lib_ioctl,
@@ -1736,7 +1736,7 @@ static struct snd_pcm_ops snd_cs46xx_playback_ops = {
        .pointer =              snd_cs46xx_playback_direct_pointer,
 };
-static struct snd_pcm_ops snd_cs46xx_playback_indirect_ops = {
+static const struct snd_pcm_ops snd_cs46xx_playback_indirect_ops = {
        .open =                 snd_cs46xx_playback_open,
        .close =                snd_cs46xx_playback_close,
        .ioctl =                snd_pcm_lib_ioctl,
@@ -1748,7 +1748,7 @@ static struct snd_pcm_ops snd_cs46xx_playback_indirect_ops = {
        .ack =                  snd_cs46xx_playback_transfer,
 };
-static struct snd_pcm_ops snd_cs46xx_capture_ops = {
+static const struct snd_pcm_ops snd_cs46xx_capture_ops = {
        .open =                 snd_cs46xx_capture_open,
        .close =                snd_cs46xx_capture_close,
        .ioctl =                snd_pcm_lib_ioctl,
@@ -1759,7 +1759,7 @@ static struct snd_pcm_ops snd_cs46xx_capture_ops = {
        .pointer =              snd_cs46xx_capture_direct_pointer,
 };
-static struct snd_pcm_ops snd_cs46xx_capture_indirect_ops = {
+static const struct snd_pcm_ops snd_cs46xx_capture_indirect_ops = {
        .open =                 snd_cs46xx_capture_open,
        .close =                snd_cs46xx_capture_close,
        .ioctl =                snd_pcm_lib_ioctl,
@@ -2683,14 +2683,14 @@ static void snd_cs46xx_midi_output_trigger(struct snd_rawmidi_substream *substre
        spin_unlock_irqrestore(&chip->reg_lock, flags);
 }
-static struct snd_rawmidi_ops snd_cs46xx_midi_output =
+static const struct snd_rawmidi_ops snd_cs46xx_midi_output =
 {
        .open =         snd_cs46xx_midi_output_open,
        .close =        snd_cs46xx_midi_output_close,
        .trigger =      snd_cs46xx_midi_output_trigger,
 };
-static struct snd_rawmidi_ops snd_cs46xx_midi_input =
+static const struct snd_rawmidi_ops snd_cs46xx_midi_input =
 {
        .open =         snd_cs46xx_midi_input_open,
        .close =        snd_cs46xx_midi_input_close,
diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c
index 06ac5d8da362..82bd10b68a77 100644
--- a/sound/pci/cs5535audio/cs5535audio_pm.c
+++ b/sound/pci/cs5535audio/cs5535audio_pm.c
@@ -55,7 +55,7 @@ static void snd_cs5535audio_stop_hardware(struct cs5535audio *cs5535au)
 }
-static int snd_cs5535audio_suspend(struct device *dev)
+static int __maybe_unused snd_cs5535audio_suspend(struct device *dev)
 {
        struct snd_card *card = dev_get_drvdata(dev);
        struct cs5535audio *cs5535au = card->private_data;
@@ -74,7 +74,7 @@ static int snd_cs5535audio_suspend(struct device *dev)
        return 0;
 }
-static int snd_cs5535audio_resume(struct device *dev)
+static int __maybe_unused snd_cs5535audio_resume(struct device *dev)
 {
        struct snd_card *card = dev_get_drvdata(dev);
        struct cs5535audio *cs5535au = card->private_data;
diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c
index a8fe58335ddc..8c685ddb1a41 100644
--- a/sound/pci/echoaudio/midi.c
+++ b/sound/pci/echoaudio/midi.c
@@ -288,13 +288,13 @@ static int snd_echo_midi_output_close(struct snd_rawmidi_substream *substream)
-static struct snd_rawmidi_ops snd_echo_midi_input = {
+static const struct snd_rawmidi_ops snd_echo_midi_input = {
        .open = snd_echo_midi_input_open,
        .close = snd_echo_midi_input_close,
        .trigger = snd_echo_midi_input_trigger,
 };
-static struct snd_rawmidi_ops snd_echo_midi_output = {
+static const struct snd_rawmidi_ops snd_echo_midi_output = {
        .open = snd_echo_midi_output_open,
        .close = snd_echo_midi_output_close,
        .trigger = snd_echo_midi_output_trigger,
diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c
index d2c7ea3a7610..aa2cc27b8491 100644
--- a/sound/pci/emu10k1/emu10k1_callback.c
+++ b/sound/pci/emu10k1/emu10k1_callback.c
@@ -61,7 +61,7 @@ static void set_filterQ(struct snd_emu10k1 *hw, struct snd_emux_voice *vp);
 /*
 * set up operators
 */
-static struct snd_emux_operators emu10k1_ops = {
+static const struct snd_emux_operators emu10k1_ops = {
        .owner =        THIS_MODULE,
        .get_voice =    get_voice,
        .prepare =      start_voice,
diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c
index 921037ed8468..32842734ada6 100644
--- a/sound/pci/emu10k1/emu10k1x.c
+++ b/sound/pci/emu10k1/emu10k1x.c
@@ -1486,14 +1486,14 @@ static void snd_emu10k1x_midi_output_trigger(struct snd_rawmidi_substream *subst
 */
-static struct snd_rawmidi_ops snd_emu10k1x_midi_output =
+static const struct snd_rawmidi_ops snd_emu10k1x_midi_output =
 {
        .open =         snd_emu10k1x_midi_output_open,
        .close =        snd_emu10k1x_midi_output_close,
        .trigger =      snd_emu10k1x_midi_output_trigger,
 };
-static struct snd_rawmidi_ops snd_emu10k1x_midi_input =
+static const struct snd_rawmidi_ops snd_emu10k1x_midi_input =
 {
        .open =         snd_emu10k1x_midi_input_open,
        .close =        snd_emu10k1x_midi_input_close,
diff --git a/sound/pci/emu10k1/emumpu401.c b/sound/pci/emu10k1/emumpu401.c
index fdf2b0ada489..b6650f5c1621 100644
--- a/sound/pci/emu10k1/emumpu401.c
+++ b/sound/pci/emu10k1/emumpu401.c
@@ -308,14 +308,14 @@ static void snd_emu10k1_midi_output_trigger(struct snd_rawmidi_substream *substr
 */
-static struct snd_rawmidi_ops snd_emu10k1_midi_output =
+static const struct snd_rawmidi_ops snd_emu10k1_midi_output =
 {
        .open =         snd_emu10k1_midi_output_open,
        .close =        snd_emu10k1_midi_output_close,
        .trigger =      snd_emu10k1_midi_output_trigger,
 };
-static struct snd_rawmidi_ops snd_emu10k1_midi_input =
+static const struct snd_rawmidi_ops snd_emu10k1_midi_input =
 {
        .open =         snd_emu10k1_midi_input_open,
        .close =        snd_emu10k1_midi_input_close,
diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c
index 51736c2b5a00..164adad91650 100644
--- a/sound/pci/ens1370.c
+++ b/sound/pci/ens1370.c
@@ -2317,14 +2317,14 @@ static void snd_ensoniq_midi_output_trigger(struct snd_rawmidi_substream *substr
        spin_unlock_irqrestore(&ensoniq->reg_lock, flags);
 }
-static struct snd_rawmidi_ops snd_ensoniq_midi_output =
+static const struct snd_rawmidi_ops snd_ensoniq_midi_output =
 {
        .open =         snd_ensoniq_midi_output_open,
        .close =        snd_ensoniq_midi_output_close,
        .trigger =      snd_ensoniq_midi_output_trigger,
 };
-static struct snd_rawmidi_ops snd_ensoniq_midi_input =
+static const struct snd_rawmidi_ops snd_ensoniq_midi_input =
 {
        .open =         snd_ensoniq_midi_input_open,
        .close =        snd_ensoniq_midi_input_close,
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 500878556578..3715a5725613 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -861,6 +861,10 @@ static int azx_rirb_get_response(struct hdac_bus *bus, unsigned int addr,
                return -EIO;
        }
+        /* no fallback mechanism? */
+        if (!chip->fallback_to_single_cmd)
+                return -EIO;
        /* a fatal communication error; need either to reset or to fallback
         * to the single_cmd mode
         */
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index a50e0532622a..35a9ab2cac46 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -150,6 +150,7 @@ struct azx {
        int bdl_pos_adj;
        int poll_count;
        unsigned int running:1;
+        unsigned int fallback_to_single_cmd:1;
        unsigned int single_cmd:1;
        unsigned int polling_mode:1;
        unsigned int msi:1;
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index c64d986009a9..16108f0eb688 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -128,7 +128,7 @@ static int bdl_pos_adj[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1};
 static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1};
 static int probe_only[SNDRV_CARDS];
 static int jackpoll_ms[SNDRV_CARDS];
-static bool single_cmd;
+static int single_cmd = -1;
 static int enable_msi = -1;
 #ifdef CONFIG_SND_HDA_PATCH_LOADER
 static char *patch[SNDRV_CARDS];
@@ -157,7 +157,7 @@ module_param_array(probe_only, int, NULL, 0444);
 MODULE_PARM_DESC(probe_only, "Only probing and no codec initialization.");
 module_param_array(jackpoll_ms, int, NULL, 0444);
 MODULE_PARM_DESC(jackpoll_ms, "Ms between polling for jack events (default = 0, using unsol events only)");
-module_param(single_cmd, bool, 0444);
+module_param(single_cmd, bint, 0444);
 MODULE_PARM_DESC(single_cmd, "Use single command to communicate with codecs "
                 "(for debugging only).");
 module_param(enable_msi, bint, 0444);
@@ -1596,7 +1596,11 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci,
        check_probe_mask(chip, dev);
-        chip->single_cmd = single_cmd;
+        if (single_cmd < 0) /* allow fallback to single_cmd at errors */
+                chip->fallback_to_single_cmd = 1;
+        else /* explicitly set to single_cmd or not */
+                chip->single_cmd = single_cmd;
        azx_check_snoop_available(chip);
        if (bdl_pos_adj[dev] < 0)
@@ -1774,6 +1778,14 @@ static int azx_first_init(struct azx *chip)
        chip->playback_index_offset = chip->capture_streams;
        chip->num_streams = chip->playback_streams + chip->capture_streams;
+        /* sanity check for the SDxCTL.STRM field overflow */
+        if (chip->num_streams > 15 &&
+            (chip->driver_caps & AZX_DCAPS_SEPARATE_STREAM_TAG) == 0) {
+                dev_warn(chip->card->dev, "number of I/O streams is %d, "
+                         "forcing separate stream tags", chip->num_streams);
+                chip->driver_caps |= AZX_DCAPS_SEPARATE_STREAM_TAG;
+        }
        /* initialize streams */
        err = azx_init_streams(chip);
        if (err < 0)
@@ -2155,7 +2167,20 @@ static void azx_remove(struct pci_dev *pci)
                /* cancel the pending probing work */
                chip = card->private_data;
                hda = container_of(chip, struct hda_intel, chip);
+                /* FIXME: below is an ugly workaround.
+                 * Both device_release_driver() and driver_probe_device()
+                 * take *both* the device's and its parent's lock before
+                 * calling the remove() and probe() callbacks.  The codec
+                 * probe takes the locks of both the codec itself and its
+                 * parent, i.e. the PCI controller dev.  Meanwhile, when
+                 * the PCI controller is unbound, it takes its lock, too
+                 * ==> ouch, a deadlock!
+                 * As a workaround, we unlock temporarily here the controller
+                 * device during cancel_work_sync() call.
+                 */
+                device_unlock(&pci->dev);
                cancel_work_sync(&hda->probe_work);
+                device_lock(&pci->dev);
                snd_card_free(card);
        }
@@ -2197,9 +2222,9 @@ static const struct pci_device_id azx_ids[] = {
          .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
        /* Lewisburg */
        { PCI_DEVICE(0x8086, 0xa1f0),
-          .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
+          .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE },
        { PCI_DEVICE(0x8086, 0xa270),
-          .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
+          .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_SKYLAKE },
        /* Lynx Point-LP */
        { PCI_DEVICE(0x8086, 0x9c20),
          .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH },
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 11b9b2f17a2e..9ec4dba8a793 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -1482,6 +1482,9 @@ static int dspio_scp(struct hda_codec *codec,
                } else if (ret_size != reply_data_size) {
                        codec_dbg(codec, "RetLen and HdrLen .NE.\n");
                        return -EINVAL;
+                } else if (!reply) {
+                        codec_dbg(codec, "NULL reply\n");
+                        return -EINVAL;
                } else {
                        *reply_len = ret_size*sizeof(unsigned int);
                        memcpy(reply, scp_reply.data, *reply_len);
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index cf9bc042fe96..3fc201c3b95a 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -3639,6 +3639,7 @@ HDA_CODEC_ENTRY(0x10de0070, "GPU 70 HDMI/DP",	patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de0071, "GPU 71 HDMI/DP",   patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de0072, "GPU 72 HDMI/DP",   patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de007d, "GPU 7d HDMI/DP",   patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0080, "GPU 80 HDMI/DP",   patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de0082, "GPU 82 HDMI/DP",   patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de0083, "GPU 83 HDMI/DP",   patch_nvhdmi),
 HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI",       patch_nvhdmi_2ch),
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 7d660ee1d5e8..73a00460b5c1 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -337,6 +337,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
        case 0x10ec0288:
        case 0x10ec0295:
        case 0x10ec0298:
+        case 0x10ec0299:
                alc_update_coef_idx(codec, 0x10, 1<<9, 0);
                break;
        case 0x10ec0285:
@@ -379,6 +380,7 @@ static void alc_fill_eapd_coef(struct hda_codec *codec)
                break;
        case 0x10ec0899:
        case 0x10ec0900:
+        case 0x10ec1220:
                alc_update_coef_idx(codec, 0x7, 1<<1, 0);
                break;
        }
@@ -912,6 +914,7 @@ static struct alc_codec_rename_pci_table rename_pci_tbl[] = {
        { 0x10ec0256, 0x1028, 0, "ALC3246" },
        { 0x10ec0225, 0x1028, 0, "ALC3253" },
        { 0x10ec0295, 0x1028, 0, "ALC3254" },
+        { 0x10ec0299, 0x1028, 0, "ALC3271" },
        { 0x10ec0670, 0x1025, 0, "ALC669X" },
        { 0x10ec0676, 0x1025, 0, "ALC679X" },
        { 0x10ec0282, 0x1043, 0, "ALC3229" },
@@ -2309,6 +2312,7 @@ static int patch_alc882(struct hda_codec *codec)
        case 0x10ec0882:
        case 0x10ec0885:
        case 0x10ec0900:
+        case 0x10ec1220:
                break;
        default:
                /* ALC883 and variants */
@@ -3717,6 +3721,7 @@ static void alc_headset_mode_unplugged(struct hda_codec *codec)
                break;
        case 0x10ec0225:
        case 0x10ec0295:
+        case 0x10ec0299:
                alc_process_coef_fw(codec, coef0225);
                break;
        case 0x10ec0867:
@@ -3812,6 +3817,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin,
        case 0x10ec0867:
                alc_update_coefex_idx(codec, 0x57, 0x5, 0, 1<<14);
                /* fallthru */
+        case 0x10ec0221:
        case 0x10ec0662:
                snd_hda_set_pin_ctl_cache(codec, hp_pin, 0);
                snd_hda_set_pin_ctl_cache(codec, mic_pin, PIN_VREF50);
@@ -3824,6 +3830,7 @@ static void alc_headset_mode_mic_in(struct hda_codec *codec, hda_nid_t hp_pin,
                break;
        case 0x10ec0225:
        case 0x10ec0295:
+        case 0x10ec0299:
                alc_update_coef_idx(codec, 0x45, 0x3f<<10, 0x31<<10);
                snd_hda_set_pin_ctl_cache(codec, hp_pin, 0);
                alc_process_coef_fw(codec, coef0225);
@@ -3882,6 +3889,7 @@ static void alc_headset_mode_default(struct hda_codec *codec)
        switch (codec->core.vendor_id) {
        case 0x10ec0225:
        case 0x10ec0295:
+        case 0x10ec0299:
                alc_process_coef_fw(codec, coef0225);
                break;
        case 0x10ec0255:
@@ -3997,6 +4005,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec)
                break;
        case 0x10ec0225:
        case 0x10ec0295:
+        case 0x10ec0299:
                alc_process_coef_fw(codec, coef0225);
                break;
        case 0x10ec0867:
@@ -4090,6 +4099,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec)
                break;
        case 0x10ec0225:
        case 0x10ec0295:
+        case 0x10ec0299:
                alc_process_coef_fw(codec, coef0225);
                break;
        }
@@ -4174,6 +4184,7 @@ static void alc_determine_headset_type(struct hda_codec *codec)
                break;
        case 0x10ec0225:
        case 0x10ec0295:
+        case 0x10ec0299:
                alc_process_coef_fw(codec, coef0225);
                msleep(800);
                val = alc_read_coef_idx(codec, 0x46);
@@ -4401,7 +4412,7 @@ static void alc_no_shutup(struct hda_codec *codec)
 static void alc_fixup_no_shutup(struct hda_codec *codec,
                                const struct hda_fixup *fix, int action)
 {
-        if (action == HDA_FIXUP_ACT_PRE_PROBE) {
+        if (action == HDA_FIXUP_ACT_PROBE) {
                struct alc_spec *spec = codec->spec;
                spec->shutup = alc_no_shutup;
        }
@@ -4857,6 +4868,8 @@ enum {
        ALC292_FIXUP_TPT460,
        ALC298_FIXUP_SPK_VOLUME,
        ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER,
+        ALC269_FIXUP_ATIV_BOOK_8,
+        ALC221_FIXUP_HP_MIC_NO_PRESENCE,
 };
 static const struct hda_fixup alc269_fixups[] = {
@@ -5529,6 +5542,22 @@ static const struct hda_fixup alc269_fixups[] = {
                .chained = true,
                .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE
        },
+        [ALC269_FIXUP_ATIV_BOOK_8] = {
+                .type = HDA_FIXUP_FUNC,
+                .v.func = alc_fixup_auto_mute_via_amp,
+                .chained = true,
+                .chain_id = ALC269_FIXUP_NO_SHUTUP
+        },
+        [ALC221_FIXUP_HP_MIC_NO_PRESENCE] = {
+                .type = HDA_FIXUP_PINS,
+                .v.pins = (const struct hda_pintbl[]) {
+                        { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */
+                        { 0x1a, 0x01a1913d }, /* use as headphone mic, without its own jack detect */
+                        { }
+                },
+                .chained = true,
+                .chain_id = ALC269_FIXUP_HEADSET_MODE
+        },
 };
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -5639,6 +5668,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
        SND_PCI_QUIRK(0x103c, 0x2337, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1),
        SND_PCI_QUIRK(0x103c, 0x221c, "HP EliteBook 755 G2", ALC280_FIXUP_HP_HEADSET_MIC),
        SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC),
+        SND_PCI_QUIRK(0x103c, 0x82bf, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
+        SND_PCI_QUIRK(0x103c, 0x82c0, "HP", ALC221_FIXUP_HP_MIC_NO_PRESENCE),
        SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300),
        SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
        SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST),
@@ -5665,6 +5696,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
        SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN),
        SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC),
        SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC),
+        SND_PCI_QUIRK(0x144d, 0xc740, "Samsung Ativ book 8 (NP870Z5G)", ALC269_FIXUP_ATIV_BOOK_8),
        SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC),
        SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC),
        SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE),
@@ -6065,6 +6097,12 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
        SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE,
                ALC298_STANDARD_PINS,
                {0x17, 0x90170150}),
+        SND_HDA_PIN_QUIRK(0x10ec0298, 0x1028, "Dell", ALC298_FIXUP_SPK_VOLUME,
+                {0x12, 0xb7a60140},
+                {0x13, 0xb7a60150},
+                {0x17, 0x90170110},
+                {0x1a, 0x03011020},
+                {0x21, 0x03211030}),
        {}
 };
@@ -6212,6 +6250,7 @@ static int patch_alc269(struct hda_codec *codec)
                break;
        case 0x10ec0225:
        case 0x10ec0295:
+        case 0x10ec0299:
                spec->codec_variant = ALC269_TYPE_ALC225;
                break;
        case 0x10ec0234:
@@ -7250,6 +7289,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = {
        HDA_CODEC_ENTRY(0x10ec0294, "ALC294", patch_alc269),
        HDA_CODEC_ENTRY(0x10ec0295, "ALC295", patch_alc269),
        HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269),
+        HDA_CODEC_ENTRY(0x10ec0299, "ALC299", patch_alc269),
        HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861),
        HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd),
        HDA_CODEC_ENTRY(0x10ec0861, "ALC861", patch_alc861),
@@ -7281,6 +7321,7 @@ static const struct hda_device_id snd_hda_id_realtek[] = {
        HDA_CODEC_ENTRY(0x10ec0892, "ALC892", patch_alc662),
        HDA_CODEC_ENTRY(0x10ec0899, "ALC898", patch_alc882),
        HDA_CODEC_ENTRY(0x10ec0900, "ALC1150", patch_alc882),
+        HDA_CODEC_ENTRY(0x10ec1220, "ALC1220", patch_alc882),
        {} /* terminator */
 };
 MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index 37b70f8e878f..faa3d38bac0b 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -166,6 +166,7 @@ enum {
        STAC_D965_VERBS,
        STAC_DELL_3ST,
        STAC_DELL_BIOS,
+        STAC_NEMO_DEFAULT,
        STAC_DELL_BIOS_AMIC,
        STAC_DELL_BIOS_SPDIF,
        STAC_927X_DELL_DMIC,
@@ -1360,6 +1361,27 @@ static const struct hda_pintbl oqo9200_pin_configs[] = {
        {}
 };
+/*
+ *  STAC 92HD700
+ *  18881000 Amigaone X1000
+ */
+static const struct hda_pintbl nemo_pin_configs[] = {
+        { 0x0a, 0x02214020 },   /* Front panel HP socket */
+        { 0x0b, 0x02a19080 },   /* Front Mic */
+        { 0x0c, 0x0181304e },   /* Line in */
+        { 0x0d, 0x01014010 },   /* Line out */
+        { 0x0e, 0x01a19040 },   /* Rear Mic */
+        { 0x0f, 0x01011012 },   /* Rear speakers */
+        { 0x10, 0x01016011 },   /* Center speaker */
+        { 0x11, 0x01012014 },   /* Side speakers (7.1) */
+        { 0x12, 0x103301f0 },   /* Motherboard CD line in connector */
+        { 0x13, 0x411111f0 },   /* Unused */
+        { 0x14, 0x411111f0 },   /* Unused */
+        { 0x21, 0x01442170 },   /* S/PDIF line out */
+        { 0x22, 0x411111f0 },   /* Unused */
+        { 0x23, 0x411111f0 },   /* Unused */
+        {}
+};
 static void stac9200_fixup_panasonic(struct hda_codec *codec,
                                     const struct hda_fixup *fix, int action)
@@ -3883,6 +3905,10 @@ static const struct hda_fixup stac927x_fixups[] = {
                .type = HDA_FIXUP_PINS,
                .v.pins = d965_5st_no_fp_pin_configs,
        },
+        [STAC_NEMO_DEFAULT] = {
+                .type = HDA_FIXUP_PINS,
+                .v.pins = nemo_pin_configs,
+        },
        [STAC_DELL_3ST] = {
                .type = HDA_FIXUP_PINS,
                .v.pins = dell_3st_pin_configs,
@@ -3939,6 +3965,7 @@ static const struct hda_model_fixup stac927x_models[] = {
        { .id = STAC_D965_5ST_NO_FP, .name = "5stack-no-fp" },
        { .id = STAC_DELL_3ST, .name = "dell-3stack" },
        { .id = STAC_DELL_BIOS, .name = "dell-bios" },
+        { .id = STAC_NEMO_DEFAULT, .name = "nemo-default" },
        { .id = STAC_DELL_BIOS_AMIC, .name = "dell-bios-amic" },
        { .id = STAC_927X_VOLKNOB, .name = "volknob" },
        {}
@@ -3977,6 +4004,8 @@ static const struct snd_pci_quirk stac927x_fixup_tbl[] = {
                           "Intel D965", STAC_D965_5ST),
        SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2500,
                           "Intel D965", STAC_D965_5ST),
+        /* Nemo */
+        SND_PCI_QUIRK(0x1888, 0x1000, "AmigaOne X1000", STAC_NEMO_DEFAULT),
        /* volume-knob fixes */
        SND_PCI_QUIRK_VENDOR(0x10cf, "FSC", STAC_927X_VOLKNOB),
        {} /* terminator */
@@ -5036,6 +5065,7 @@ static const struct hda_device_id snd_hda_id_sigmatel[] = {
        HDA_CODEC_ENTRY(0x83847683, "STAC9221D A2", patch_stac922x),
        HDA_CODEC_ENTRY(0x83847618, "STAC9227", patch_stac927x),
        HDA_CODEC_ENTRY(0x83847619, "STAC9227", patch_stac927x),
+        HDA_CODEC_ENTRY(0x83847638, "STAC92HD700", patch_stac927x),
        HDA_CODEC_ENTRY(0x83847616, "STAC9228", patch_stac927x),
        HDA_CODEC_ENTRY(0x83847617, "STAC9228", patch_stac927x),
        HDA_CODEC_ENTRY(0x83847614, "STAC9229", patch_stac927x),
diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c
index e5c52ed9b674..842744e7a139 100644
--- a/sound/pci/ice1712/ice1724.c
+++ b/sound/pci/ice1712/ice1724.c
@@ -367,7 +367,7 @@ static void vt1724_midi_output_drain(struct snd_rawmidi_substream *s)
        } while (time_after(timeout, jiffies));
 }
-static struct snd_rawmidi_ops vt1724_midi_output_ops = {
+static const struct snd_rawmidi_ops vt1724_midi_output_ops = {
        .open = vt1724_midi_output_open,
        .close = vt1724_midi_output_close,
        .trigger = vt1724_midi_output_trigger,
@@ -402,7 +402,7 @@ static void vt1724_midi_input_trigger(struct snd_rawmidi_substream *s, int up)
        spin_unlock_irqrestore(&ice->reg_lock, flags);
 }
-static struct snd_rawmidi_ops vt1724_midi_input_ops = {
+static const struct snd_rawmidi_ops vt1724_midi_input_ops = {
        .open = vt1724_midi_input_open,
        .close = vt1724_midi_input_close,
        .trigger = vt1724_midi_input_trigger,
diff --git a/sound/pci/mixart/mixart.h b/sound/pci/mixart/mixart.h
index 0cc17e0ea34a..426743871540 100644
--- a/sound/pci/mixart/mixart.h
+++ b/sound/pci/mixart/mixart.h
@@ -86,7 +86,7 @@ struct mixart_mgr {
        u32 msg_fifo[MSG_FIFO_SIZE];
        int msg_fifo_readptr;
        int msg_fifo_writeptr;
-        atomic_t msg_processed;       /* number of messages to be processed in takslet */
+        atomic_t msg_processed;       /* number of messages to be processed in tasklet */
        struct mutex lock;              /* interrupt lock */
        struct mutex msg_lock;          /* mailbox lock */
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index b94fc6357139..fc0face6cdc6 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -1510,14 +1510,14 @@ static int snd_hdsp_midi_output_close(struct snd_rawmidi_substream *substream)
        return 0;
 }
-static struct snd_rawmidi_ops snd_hdsp_midi_output =
+static const struct snd_rawmidi_ops snd_hdsp_midi_output =
 {
        .open =         snd_hdsp_midi_output_open,
        .close =        snd_hdsp_midi_output_close,
        .trigger =      snd_hdsp_midi_output_trigger,
 };
-static struct snd_rawmidi_ops snd_hdsp_midi_input =
+static const struct snd_rawmidi_ops snd_hdsp_midi_input =
 {
        .open =         snd_hdsp_midi_input_open,
        .close =        snd_hdsp_midi_input_close,
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 14bbf55c1ef9..c48acdb0e186 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -2043,14 +2043,14 @@ static int snd_hdspm_midi_output_close(struct snd_rawmidi_substream *substream)
        return 0;
 }
-static struct snd_rawmidi_ops snd_hdspm_midi_output =
+static const struct snd_rawmidi_ops snd_hdspm_midi_output =
 {
        .open =         snd_hdspm_midi_output_open,
        .close =        snd_hdspm_midi_output_close,
        .trigger =      snd_hdspm_midi_output_trigger,
 };
-static struct snd_rawmidi_ops snd_hdspm_midi_input =
+static const struct snd_rawmidi_ops snd_hdspm_midi_input =
 {
        .open =         snd_hdspm_midi_input_open,
        .close =        snd_hdspm_midi_input_close,
diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c
index af83b3b38052..8e457ea27f89 100644
--- a/sound/pci/vx222/vx222_ops.c
+++ b/sound/pci/vx222/vx222_ops.c
@@ -269,12 +269,12 @@ static void vx2_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime,
        /* Transfer using pseudo-dma.
         */
-        if (offset + count > pipe->buffer_bytes) {
+        if (offset + count >= pipe->buffer_bytes) {
                int length = pipe->buffer_bytes - offset;
                count -= length;
                length >>= 2; /* in 32bit words */
                /* Transfer using pseudo-dma. */
-                while (length-- > 0) {
+                for (; length > 0; length--) {
                        outl(cpu_to_le32(*addr), port);
                        addr++;
                }
@@ -284,7 +284,7 @@ static void vx2_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime,
        pipe->hw_ptr += count;
        count >>= 2; /* in 32bit words */
        /* Transfer using pseudo-dma. */
-        while (count-- > 0) {
+        for (; count > 0; count--) {
                outl(cpu_to_le32(*addr), port);
                addr++;
        }
@@ -307,12 +307,12 @@ static void vx2_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime,
        vx2_setup_pseudo_dma(chip, 0);
        /* Transfer using pseudo-dma.
         */
-        if (offset + count > pipe->buffer_bytes) {
+        if (offset + count >= pipe->buffer_bytes) {
                int length = pipe->buffer_bytes - offset;
                count -= length;
                length >>= 2; /* in 32bit words */
                /* Transfer using pseudo-dma. */
-                while (length-- > 0)
+                for (; length > 0; length--)
                        *addr++ = le32_to_cpu(inl(port));
                addr = (u32 *)runtime->dma_area;
                pipe->hw_ptr = 0;
@@ -320,7 +320,7 @@ static void vx2_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime,
        pipe->hw_ptr += count;
        count >>= 2; /* in 32bit words */
        /* Transfer using pseudo-dma. */
-        while (count-- > 0)
+        for (; count > 0; count--)
                *addr++ = le32_to_cpu(inl(port));
        vx2_release_pseudo_dma(chip);
diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c
index 281972913c32..56aa1ba73ccc 100644
--- a/sound/pcmcia/vx/vxp_ops.c
+++ b/sound/pcmcia/vx/vxp_ops.c
@@ -369,12 +369,12 @@ static void vxp_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime,
        unsigned short *addr = (unsigned short *)(runtime->dma_area + offset);
        vx_setup_pseudo_dma(chip, 1);
-        if (offset + count > pipe->buffer_bytes) {
+        if (offset + count >= pipe->buffer_bytes) {
                int length = pipe->buffer_bytes - offset;
                count -= length;
                length >>= 1; /* in 16bit words */
                /* Transfer using pseudo-dma. */
-                while (length-- > 0) {
+                for (; length > 0; length--) {
                        outw(cpu_to_le16(*addr), port);
                        addr++;
                }
@@ -384,7 +384,7 @@ static void vxp_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime,
        pipe->hw_ptr += count;
        count >>= 1; /* in 16bit words */
        /* Transfer using pseudo-dma. */
-        while (count-- > 0) {
+        for (; count > 0; count--) {
                outw(cpu_to_le16(*addr), port);
                addr++;
        }
@@ -411,12 +411,12 @@ static void vxp_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime,
        if (snd_BUG_ON(count % 2))
                return;
        vx_setup_pseudo_dma(chip, 0);
-        if (offset + count > pipe->buffer_bytes) {
+        if (offset + count >= pipe->buffer_bytes) {
                int length = pipe->buffer_bytes - offset;
                count -= length;
                length >>= 1; /* in 16bit words */
                /* Transfer using pseudo-dma. */
-                while (length-- > 0)
+                for (; length > 0; length--)
                        *addr++ = le16_to_cpu(inw(port));
                addr = (unsigned short *)runtime->dma_area;
                pipe->hw_ptr = 0;
@@ -424,7 +424,7 @@ static void vxp_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime,
        pipe->hw_ptr += count;
        count >>= 1; /* in 16bit words */
        /* Transfer using pseudo-dma. */
-        while (count-- > 1)
+        for (; count > 1; count--)
                *addr++ = le16_to_cpu(inw(port));
        /* Disable DMA */
        pchip->regDIALOG &= ~VXP_DLG_DMAREAD_SEL_MASK;
diff --git a/sound/synth/emux/emux_seq.c b/sound/synth/emux/emux_seq.c
index a0209204ae48..55579f6b8cb2 100644
--- a/sound/synth/emux/emux_seq.c
+++ b/sound/synth/emux/emux_seq.c
@@ -33,13 +33,13 @@ static int snd_emux_unuse(void *private_data, struct snd_seq_port_subscribe *inf
 * MIDI emulation operators
 */
 static struct snd_midi_op emux_ops = {
-        snd_emux_note_on,
+        .note_on = snd_emux_note_on,
-        snd_emux_note_off,
+        .note_off = snd_emux_note_off,
-        snd_emux_key_press,
+        .key_press = snd_emux_key_press,
-        snd_emux_terminate_note,
+        .note_terminate = snd_emux_terminate_note,
-        snd_emux_control,
+        .control = snd_emux_control,
-        snd_emux_nrpn,
+        .nrpn = snd_emux_nrpn,
-        snd_emux_sysex,
+        .sysex = snd_emux_sysex,
 };
diff --git a/sound/usb/6fire/midi.c b/sound/usb/6fire/midi.c
index 3d410969553e..aa5adbb6eb5d 100644
--- a/sound/usb/6fire/midi.c
+++ b/sound/usb/6fire/midi.c
@@ -139,14 +139,14 @@ static void usb6fire_midi_in_trigger(
        spin_unlock_irqrestore(&rt->in_lock, flags);
 }
-static struct snd_rawmidi_ops out_ops = {
+static const struct snd_rawmidi_ops out_ops = {
        .open = usb6fire_midi_out_open,
        .close = usb6fire_midi_out_close,
        .trigger = usb6fire_midi_out_trigger,
        .drain = usb6fire_midi_out_drain
 };
-static struct snd_rawmidi_ops in_ops = {
+static const struct snd_rawmidi_ops in_ops = {
        .open = usb6fire_midi_in_open,
        .close = usb6fire_midi_in_close,
        .trigger = usb6fire_midi_in_trigger
diff --git a/sound/usb/Makefile b/sound/usb/Makefile
index 2d2d122b069f..42cb33b94f6a 100644
--- a/sound/usb/Makefile
+++ b/sound/usb/Makefile
@@ -10,6 +10,7 @@ snd-usb-audio-objs := 	card.o \
                        mixer.o \
                        mixer_quirks.o \
                        mixer_scarlett.o \
+                        mixer_us16x08.o \
                        pcm.o \
                        proc.o \
                        quirks.o \
diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c
index d060dddcc52d..2ff9d578753a 100644
--- a/sound/usb/bcd2000/bcd2000.c
+++ b/sound/usb/bcd2000/bcd2000.c
@@ -252,13 +252,13 @@ static void bcd2000_input_complete(struct urb *urb)
                        __func__, ret);
 }
-static struct snd_rawmidi_ops bcd2000_midi_output = {
+static const struct snd_rawmidi_ops bcd2000_midi_output = {
        .open =    bcd2000_midi_output_open,
        .close =   bcd2000_midi_output_close,
        .trigger = bcd2000_midi_output_trigger,
 };
-static struct snd_rawmidi_ops bcd2000_midi_input = {
+static const struct snd_rawmidi_ops bcd2000_midi_input = {
        .open =    bcd2000_midi_input_open,
        .close =   bcd2000_midi_input_close,
        .trigger = bcd2000_midi_input_trigger,
diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c
index 2d7588461b33..f8e5b1b57c4f 100644
--- a/sound/usb/caiaq/midi.c
+++ b/sound/usb/caiaq/midi.c
@@ -102,14 +102,14 @@ static void snd_usb_caiaq_midi_output_trigger(struct snd_rawmidi_substream *subs
 }
-static struct snd_rawmidi_ops snd_usb_caiaq_midi_output =
+static const struct snd_rawmidi_ops snd_usb_caiaq_midi_output =
 {
        .open =         snd_usb_caiaq_midi_output_open,
        .close =        snd_usb_caiaq_midi_output_close,
        .trigger =      snd_usb_caiaq_midi_output_trigger,
 };
-static struct snd_rawmidi_ops snd_usb_caiaq_midi_input =
+static const struct snd_rawmidi_ops snd_usb_caiaq_midi_input =
 {
        .open =         snd_usb_caiaq_midi_input_open,
        .close =        snd_usb_caiaq_midi_input_close,
diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c
index 90009c0b3a92..0ff5a7d2e19f 100644
--- a/sound/usb/line6/driver.c
+++ b/sound/usb/line6/driver.c
@@ -492,42 +492,46 @@ static void line6_destruct(struct snd_card *card)
        usb_put_dev(usbdev);
 }
-/* get data from endpoint descriptor (see usb_maxpacket): */
+static void line6_get_usb_properties(struct usb_line6 *line6)
-static void line6_get_interval(struct usb_line6 *line6)
 {
        struct usb_device *usbdev = line6->usbdev;
        const struct line6_properties *properties = line6->properties;
        int pipe;
-        struct usb_host_endpoint *ep;
+        struct usb_host_endpoint *ep = NULL;
-        if (properties->capabilities & LINE6_CAP_CONTROL_MIDI) {
+        if (properties->capabilities & LINE6_CAP_CONTROL) {
-                pipe =
+                if (properties->capabilities & LINE6_CAP_CONTROL_MIDI) {
-                        usb_rcvintpipe(line6->usbdev, line6->properties->ep_ctrl_r);
+                        pipe = usb_rcvintpipe(line6->usbdev,
-        } else {
+                                line6->properties->ep_ctrl_r);
-                pipe =
+                } else {
-                        usb_rcvbulkpipe(line6->usbdev, line6->properties->ep_ctrl_r);
+                        pipe = usb_rcvbulkpipe(line6->usbdev,
+                                line6->properties->ep_ctrl_r);
+                }
+                ep = usbdev->ep_in[usb_pipeendpoint(pipe)];
        }
-        ep = usbdev->ep_in[usb_pipeendpoint(pipe)];
+        /* Control data transfer properties */
        if (ep) {
                line6->interval = ep->desc.bInterval;
-                if (usbdev->speed == USB_SPEED_LOW) {
-                        line6->intervals_per_second = USB_LOW_INTERVALS_PER_SECOND;
-                        line6->iso_buffers = USB_LOW_ISO_BUFFERS;
-                } else {
-                        line6->intervals_per_second = USB_HIGH_INTERVALS_PER_SECOND;
-                        line6->iso_buffers = USB_HIGH_ISO_BUFFERS;
-                }
                line6->max_packet_size = le16_to_cpu(ep->desc.wMaxPacketSize);
        } else {
-                dev_err(line6->ifcdev,
+                if (properties->capabilities & LINE6_CAP_CONTROL) {
-                        "endpoint not available, using fallback values");
+                        dev_err(line6->ifcdev,
+                                "endpoint not available, using fallback values");
+                }
                line6->interval = LINE6_FALLBACK_INTERVAL;
                line6->max_packet_size = LINE6_FALLBACK_MAXPACKETSIZE;
        }
-}
+        /* Isochronous transfer properties */
+        if (usbdev->speed == USB_SPEED_LOW) {
+                line6->intervals_per_second = USB_LOW_INTERVALS_PER_SECOND;
+                line6->iso_buffers = USB_LOW_ISO_BUFFERS;
+        } else {
+                line6->intervals_per_second = USB_HIGH_INTERVALS_PER_SECOND;
+                line6->iso_buffers = USB_HIGH_ISO_BUFFERS;
+        }
+}
 /* Enable buffering of incoming messages, flush the buffer */
 static int line6_hwdep_open(struct snd_hwdep *hw, struct file *file)
@@ -754,8 +758,9 @@ int line6_probe(struct usb_interface *interface,
                goto error;
        }
+        line6_get_usb_properties(line6);
        if (properties->capabilities & LINE6_CAP_CONTROL) {
-                line6_get_interval(line6);
                ret = line6_init_cap_control(line6);
                if (ret < 0)
                        goto error;
diff --git a/sound/usb/line6/midi.c b/sound/usb/line6/midi.c
index d0fb2f205bd9..1d3a23b02d68 100644
--- a/sound/usb/line6/midi.c
+++ b/sound/usb/line6/midi.c
@@ -200,14 +200,14 @@ static void line6_midi_input_trigger(struct snd_rawmidi_substream *substream,
                line6->line6midi->substream_receive = NULL;
 }
-static struct snd_rawmidi_ops line6_midi_output_ops = {
+static const struct snd_rawmidi_ops line6_midi_output_ops = {
        .open = line6_midi_output_open,
        .close = line6_midi_output_close,
        .trigger = line6_midi_output_trigger,
        .drain = line6_midi_output_drain,
 };
-static struct snd_rawmidi_ops line6_midi_input_ops = {
+static const struct snd_rawmidi_ops line6_midi_input_ops = {
        .open = line6_midi_input_open,
        .close = line6_midi_input_close,
        .trigger = line6_midi_input_trigger,
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index 7ba92921bf28..6e763bc8d7db 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -1234,14 +1234,14 @@ static void snd_usbmidi_input_trigger(struct snd_rawmidi_substream *substream,
                clear_bit(substream->number, &umidi->input_triggered);
 }
-static struct snd_rawmidi_ops snd_usbmidi_output_ops = {
+static const struct snd_rawmidi_ops snd_usbmidi_output_ops = {
        .open = snd_usbmidi_output_open,
        .close = snd_usbmidi_output_close,
        .trigger = snd_usbmidi_output_trigger,
        .drain = snd_usbmidi_output_drain,
 };
-static struct snd_rawmidi_ops snd_usbmidi_input_ops = {
+static const struct snd_rawmidi_ops snd_usbmidi_input_ops = {
        .open = snd_usbmidi_input_open,
        .close = snd_usbmidi_input_close,
        .trigger = snd_usbmidi_input_trigger
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 04991b009132..4fa0053a40af 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -43,6 +43,7 @@
 #include "mixer.h"
 #include "mixer_quirks.h"
 #include "mixer_scarlett.h"
+#include "mixer_us16x08.h"
 #include "helper.h"
 extern struct snd_kcontrol_new *snd_usb_feature_unit_ctl;
@@ -1729,6 +1730,10 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
                return err;
        switch (mixer->chip->usb_id) {
+        /* Tascam US-16x08 */
+        case USB_ID(0x0644, 0x8047):
+                err = snd_us16x08_controls_create(mixer);
+                break;
        case USB_ID(0x041e, 0x3020):
        case USB_ID(0x041e, 0x3040):
        case USB_ID(0x041e, 0x3042):
diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c
new file mode 100644
index 000000000000..301939b8f472
--- /dev/null
+++ b/sound/usb/mixer_us16x08.c
@@ -0,0 +1,1465 @@
+/*
+ *   Tascam US-16x08 ALSA driver
+ *
+ *   Copyright (c) 2016 by Detlef Urban (onkel@paraair.de)
+ *
+ *   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.
+ *
+ */
+#include <linux/slab.h>
+#include <linux/usb.h>
+#include <linux/usb/audio-v2.h>
+#include <sound/core.h>
+#include <sound/control.h>
+#include "usbaudio.h"
+#include "mixer.h"
+#include "helper.h"
+#include "mixer_us16x08.h"
+/* USB control message templates */
+static const char route_msg[] = {
+        0x61,
+        0x02,
+        0x03, /* input from master (0x02) or input from computer bus (0x03) */
+        0x62,
+        0x02,
+        0x01, /* input index (0x01/0x02 eq. left/right) or bus (0x01-0x08) */
+        0x41,
+        0x01,
+        0x61,
+        0x02,
+        0x01,
+        0x62,
+        0x02,
+        0x01, /* output index (0x01-0x08) */
+        0x42,
+        0x01,
+        0x43,
+        0x01,
+        0x00,
+        0x00
+};
+static const char mix_init_msg1[] = {
+        0x71, 0x01, 0x00, 0x00
+};
+static const char mix_init_msg2[] = {
+        0x62, 0x02, 0x00, 0x61, 0x02, 0x04, 0xb1, 0x01, 0x00, 0x00
+};
+static const char mix_msg_in[] = {
+        /* default message head, equal to all mixers */
+        0x61, 0x02, 0x04, 0x62, 0x02, 0x01,
+        0x81, /* 0x06: Controller ID */
+        0x02, /* 0x07:  */
+        0x00, /* 0x08: Value of common mixer */
+        0x00,
+        0x00
+};
+static const char mix_msg_out[] = {
+        /* default message head, equal to all mixers */
+        0x61, 0x02, 0x02, 0x62, 0x02, 0x01,
+        0x81, /* 0x06: Controller ID */
+        0x02, /*                    0x07:  */
+        0x00, /*                    0x08: Value of common mixer */
+        0x00,
+        0x00
+};
+static const char bypass_msg_out[] = {
+        0x45,
+        0x02,
+        0x01, /* on/off flag */
+        0x00,
+        0x00
+};
+static const char bus_msg_out[] = {
+        0x44,
+        0x02,
+        0x01, /* on/off flag */
+        0x00,
+        0x00
+};
+static const char comp_msg[] = {
+        /* default message head, equal to all mixers */
+        0x61, 0x02, 0x04, 0x62, 0x02, 0x01,
+        0x91,
+        0x02,
+        0xf0, /* 0x08: Threshold db (8) (e0 ... 00) (+-0dB -- -32dB) x-32 */
+        0x92,
+        0x02,
+        0x0a, /* 0x0b: Ratio (0a,0b,0d,0f,11,14,19,1e,23,28,32,3c,50,a0,ff)  */
+        0x93,
+        0x02,
+        0x02, /* 0x0e: Attack (0x02 ... 0xc0) (2ms ... 200ms) */
+        0x94,
+        0x02,
+        0x01, /* 0x11: Release (0x01 ... 0x64) (10ms ... 1000ms) x*10  */
+        0x95,
+        0x02,
+        0x03, /* 0x14: gain (0 ... 20) (0dB .. 20dB) */
+        0x96,
+        0x02,
+        0x01,
+        0x97,
+        0x02,
+        0x01, /* 0x1a: main Comp switch (0 ... 1) (off ... on)) */
+        0x00,
+        0x00
+};
+static const char eqs_msq[] = {
+        /* default message head, equal to all mixers */
+        0x61, 0x02, 0x04, 0x62, 0x02, 0x01,
+        0x51, /*                0x06: Controller ID  */
+        0x02,
+        0x04, /* 0x08: EQ set num (0x01..0x04) (LOW, LOWMID, HIGHMID, HIGH)) */
+        0x52,
+        0x02,
+        0x0c, /* 0x0b: value dB (0 ... 12) (-12db .. +12db)  x-6 */
+        0x53,
+        0x02,
+        0x0f, /* 0x0e: value freq (32-47) (1.7kHz..18kHz) */
+        0x54,
+        0x02,
+        0x02, /* 0x11: band width (0-6) (Q16-Q0.25)  2^x/4 (EQ xxMID only) */
+        0x55,
+        0x02,
+        0x01, /* 0x14: main EQ switch (0 ... 1) (off ... on)) */
+        0x00,
+        0x00
+};
+/* compressor ratio map */
+static const char ratio_map[] = {
+        0x0a, 0x0b, 0x0d, 0x0f, 0x11, 0x14, 0x19, 0x1e,
+        0x23, 0x28, 0x32, 0x3c, 0x50, 0xa0, 0xff
+};
+/* route enumeration names */
+const const char *route_names[] = {
+        "Master Left", "Master Right", "Output 1", "Output 2", "Output 3",
+        "Output 4", "Output 5", "Output 6", "Output 7", "Output 8",
+};
+static int snd_us16x08_recv_urb(struct snd_usb_audio *chip,
+        unsigned char *buf, int size)
+{
+        mutex_lock(&chip->mutex);
+        snd_usb_ctl_msg(chip->dev,
+                usb_rcvctrlpipe(chip->dev, 0),
+                SND_US16X08_URB_METER_REQUEST,
+                SND_US16X08_URB_METER_REQUESTTYPE, 0, 0, buf, size);
+        mutex_unlock(&chip->mutex);
+        return 0;
+}
+/* wrapper function to send prepared URB buffer to usb device. Return an error
+ * code if something went wrong
+ */
+static int snd_us16x08_send_urb(struct snd_usb_audio *chip, char *buf, int size)
+{
+        int err = 0;
+        if (chip) {
+                err = snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0),
+                        SND_US16X08_URB_REQUEST, SND_US16X08_URB_REQUESTTYPE,
+                        0, 0, buf, size);
+        }
+        return err;
+}
+static int snd_us16x08_route_info(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_info *uinfo)
+{
+        return snd_ctl_enum_info(uinfo, 1, 10, route_names);
+}
+static int snd_us16x08_route_get(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        struct usb_mixer_elem_info *elem = kcontrol->private_data;
+        int index = ucontrol->id.index;
+        /* route has no bias */
+        ucontrol->value.enumerated.item[0] = elem->cache_val[index];
+        return 0;
+}
+static int snd_us16x08_route_put(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        struct usb_mixer_elem_info *elem = kcontrol->private_data;
+        struct snd_usb_audio *chip = elem->head.mixer->chip;
+        int index = ucontrol->id.index;
+        char buf[sizeof(route_msg)];
+        int val, val_org, err = 0;
+        /* prepare the message buffer from template */
+        memcpy(buf, route_msg, sizeof(route_msg));
+        /*  get the new value (no bias for routes) */
+        val = ucontrol->value.enumerated.item[0];
+        /* sanity check */
+        if (val < 0 || val > 9)
+                return -EINVAL;
+        if (val < 2) {
+                /* input comes from a master channel */
+                val_org = val;
+                buf[2] = 0x02;
+        } else {
+                /* input comes from a computer channel */
+                buf[2] = 0x03;
+                val_org = val - 2;
+        }
+        /* place new route selection in URB message */
+        buf[5] = (unsigned char) (val_org & 0x0f) + 1;
+        /* place route selector in URB message */
+        buf[13] = index + 1;
+        err = snd_us16x08_send_urb(chip, buf, sizeof(route_msg));
+        if (err > 0) {
+                elem->cached |= 1 << index;
+                elem->cache_val[index] = val;
+        } else {
+                usb_audio_dbg(chip, "Failed to set routing, err:%d\n", err);
+        }
+        return err > 0 ? 1 : 0;
+}
+static int snd_us16x08_master_info(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_info *uinfo)
+{
+        uinfo->count = 1;
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+        uinfo->value.integer.max = SND_US16X08_KCMAX(kcontrol);
+        uinfo->value.integer.min = SND_US16X08_KCMIN(kcontrol);
+        uinfo->value.integer.step = SND_US16X08_KCSTEP(kcontrol);
+        return 0;
+}
+static int snd_us16x08_master_get(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        struct usb_mixer_elem_info *elem = kcontrol->private_data;
+        int index = ucontrol->id.index;
+        ucontrol->value.integer.value[0] = elem->cache_val[index];
+        return 0;
+}
+static int snd_us16x08_master_put(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        struct usb_mixer_elem_info *elem = kcontrol->private_data;
+        struct snd_usb_audio *chip = elem->head.mixer->chip;
+        char buf[sizeof(mix_msg_out)];
+        int val, err = 0;
+        int index = ucontrol->id.index;
+        /* prepare the message buffer from template */
+        memcpy(buf, mix_msg_out, sizeof(mix_msg_out));
+        /* new control value incl. bias*/
+        val = ucontrol->value.integer.value[0];
+        /* sanity check */
+        if (val < SND_US16X08_KCMIN(kcontrol)
+                || val > SND_US16X08_KCMAX(kcontrol))
+                return -EINVAL;
+        buf[8] = val - SND_US16X08_KCBIAS(kcontrol);
+        buf[6] = elem->head.id;
+        /* place channel selector in URB message */
+        buf[5] = index + 1;
+        err = snd_us16x08_send_urb(chip, buf, sizeof(mix_msg_out));
+        if (err > 0) {
+                elem->cached |= 1 << index;
+                elem->cache_val[index] = val;
+        } else {
+                usb_audio_dbg(chip, "Failed to set master, err:%d\n", err);
+        }
+        return err > 0 ? 1 : 0;
+}
+static int snd_us16x08_bus_put(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        struct usb_mixer_elem_info *elem = kcontrol->private_data;
+        struct snd_usb_audio *chip = elem->head.mixer->chip;
+        char buf[sizeof(mix_msg_out)];
+        int val, err = 0;
+        val = ucontrol->value.integer.value[0];
+        /* prepare the message buffer from template */
+        switch (elem->head.id) {
+        case SND_US16X08_ID_BYPASS:
+                memcpy(buf, bypass_msg_out, sizeof(bypass_msg_out));
+                buf[2] = val;
+                err = snd_us16x08_send_urb(chip, buf, sizeof(bypass_msg_out));
+                break;
+        case SND_US16X08_ID_BUSS_OUT:
+                memcpy(buf, bus_msg_out, sizeof(bus_msg_out));
+                buf[2] = val;
+                err = snd_us16x08_send_urb(chip, buf, sizeof(bus_msg_out));
+                break;
+        case SND_US16X08_ID_MUTE:
+                memcpy(buf, mix_msg_out, sizeof(mix_msg_out));
+                buf[8] = val;
+                buf[6] = elem->head.id;
+                buf[5] = 1;
+                err = snd_us16x08_send_urb(chip, buf, sizeof(mix_msg_out));
+                break;
+        }
+        if (err > 0) {
+                elem->cached |= 1;
+                elem->cache_val[0] = val;
+        } else {
+                usb_audio_dbg(chip, "Failed to set buss param, err:%d\n", err);
+        }
+        return err > 0 ? 1 : 0;
+}
+static int snd_us16x08_bus_get(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        struct usb_mixer_elem_info *elem = kcontrol->private_data;
+        switch (elem->head.id) {
+        case SND_US16X08_ID_BUSS_OUT:
+                ucontrol->value.integer.value[0] = elem->cache_val[0];
+                break;
+        case SND_US16X08_ID_BYPASS:
+                ucontrol->value.integer.value[0] = elem->cache_val[0];
+                break;
+        case SND_US16X08_ID_MUTE:
+                ucontrol->value.integer.value[0] = elem->cache_val[0];
+                break;
+        }
+        return 0;
+}
+/* gets a current mixer value from common store */
+static int snd_us16x08_channel_get(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        struct usb_mixer_elem_info *elem = kcontrol->private_data;
+        int index = ucontrol->id.index;
+        ucontrol->value.integer.value[0] = elem->cache_val[index];
+        return 0;
+}
+static int snd_us16x08_channel_put(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        struct usb_mixer_elem_info *elem = kcontrol->private_data;
+        struct snd_usb_audio *chip = elem->head.mixer->chip;
+        char buf[sizeof(mix_msg_in)];
+        int val, err;
+        int index = ucontrol->id.index;
+        /* prepare URB message from template */
+        memcpy(buf, mix_msg_in, sizeof(mix_msg_in));
+        val = ucontrol->value.integer.value[0];
+        /* sanity check */
+        if (val < SND_US16X08_KCMIN(kcontrol)
+                || val > SND_US16X08_KCMAX(kcontrol))
+                return -EINVAL;
+        /* add the bias to the new value */
+        buf[8] = val - SND_US16X08_KCBIAS(kcontrol);
+        buf[6] = elem->head.id;
+        buf[5] = index + 1;
+        err = snd_us16x08_send_urb(chip, buf, sizeof(mix_msg_in));
+        if (err > 0) {
+                elem->cached |= 1 << index;
+                elem->cache_val[index] = val;
+        } else {
+                usb_audio_dbg(chip, "Failed to set channel, err:%d\n", err);
+        }
+        return err > 0 ? 1 : 0;
+}
+static int snd_us16x08_mix_info(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_info *uinfo)
+{
+        uinfo->count = 1;
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+        uinfo->value.integer.max = SND_US16X08_KCMAX(kcontrol);
+        uinfo->value.integer.min = SND_US16X08_KCMIN(kcontrol);
+        uinfo->value.integer.step = SND_US16X08_KCSTEP(kcontrol);
+        return 0;
+}
+static int snd_us16x08_comp_get(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        struct usb_mixer_elem_info *elem = kcontrol->private_data;
+        struct snd_us16x08_comp_store *store =
+                ((struct snd_us16x08_comp_store *) elem->private_data);
+        int index = ucontrol->id.index;
+        int val_idx = COMP_STORE_IDX(elem->head.id);
+        ucontrol->value.integer.value[0] = store->val[val_idx][index];
+        return 0;
+}
+static int snd_us16x08_comp_put(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        struct usb_mixer_elem_info *elem = kcontrol->private_data;
+        struct snd_usb_audio *chip = elem->head.mixer->chip;
+        struct snd_us16x08_comp_store *store =
+                ((struct snd_us16x08_comp_store *) elem->private_data);
+        int index = ucontrol->id.index;
+        char buf[sizeof(comp_msg)];
+        int val_idx, val;
+        int err = 0;
+        /* prepare compressor URB message from template  */
+        memcpy(buf, comp_msg, sizeof(comp_msg));
+        /* new control value incl. bias*/
+        val_idx = elem->head.id - SND_US16X08_ID_COMP_BASE;
+        val = ucontrol->value.integer.value[0];
+        /* sanity check */
+        if (val < SND_US16X08_KCMIN(kcontrol)
+                || val > SND_US16X08_KCMAX(kcontrol))
+                return -EINVAL;
+        store->val[val_idx][index] = ucontrol->value.integer.value[0];
+        /* place comp values in message buffer watch bias! */
+        buf[8] = store->val[
+                COMP_STORE_IDX(SND_US16X08_ID_COMP_THRESHOLD)][index]
+                - SND_US16X08_COMP_THRESHOLD_BIAS;
+        buf[11] = ratio_map[store->val[
+                COMP_STORE_IDX(SND_US16X08_ID_COMP_RATIO)][index]];
+        buf[14] = store->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_ATTACK)][index]
+                + SND_US16X08_COMP_ATTACK_BIAS;
+        buf[17] = store->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_RELEASE)][index]
+                + SND_US16X08_COMP_RELEASE_BIAS;
+        buf[20] = store->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_GAIN)][index];
+        buf[26] = store->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_SWITCH)][index];
+        /* place channel selector in message buffer */
+        buf[5] = index + 1;
+        err = snd_us16x08_send_urb(chip, buf, sizeof(comp_msg));
+        if (err > 0) {
+                elem->cached |= 1 << index;
+                elem->cache_val[index] = val;
+        } else {
+                usb_audio_dbg(chip, "Failed to set compressor, err:%d\n", err);
+        }
+        return 1;
+}
+static int snd_us16x08_eqswitch_get(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        int val = 0;
+        struct usb_mixer_elem_info *elem = kcontrol->private_data;
+        struct snd_us16x08_eq_store *store =
+                ((struct snd_us16x08_eq_store *) elem->private_data);
+        int index = ucontrol->id.index;
+        /* get low switch from cache is enough, cause all bands are together */
+        val = store->val[EQ_STORE_BAND_IDX(elem->head.id)]
+                [EQ_STORE_PARAM_IDX(elem->head.id)][index];
+        ucontrol->value.integer.value[0] = val;
+        return 0;
+}
+static int snd_us16x08_eqswitch_put(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        struct usb_mixer_elem_info *elem = kcontrol->private_data;
+        struct snd_usb_audio *chip = elem->head.mixer->chip;
+        struct snd_us16x08_eq_store *store =
+                ((struct snd_us16x08_eq_store *) elem->private_data);
+        int index = ucontrol->id.index;
+        char buf[sizeof(eqs_msq)];
+        int val, err = 0;
+        int b_idx;
+        /* new control value incl. bias*/
+        val = ucontrol->value.integer.value[0] + SND_US16X08_KCBIAS(kcontrol);
+        /* prepare URB message from EQ template */
+        memcpy(buf, eqs_msq, sizeof(eqs_msq));
+        /* place channel index in URB message */
+        buf[5] = index + 1;
+        for (b_idx = 0; b_idx < SND_US16X08_ID_EQ_BAND_COUNT; b_idx++) {
+                /* all four EQ bands have to be enabled/disabled in once */
+                buf[20] = val;
+                buf[17] = store->val[b_idx][2][index];
+                buf[14] = store->val[b_idx][1][index];
+                buf[11] = store->val[b_idx][0][index];
+                buf[8] = b_idx + 1;
+                err = snd_us16x08_send_urb(chip, buf, sizeof(eqs_msq));
+                if (err < 0)
+                        break;
+                store->val[b_idx][3][index] = val;
+                msleep(15);
+        }
+        if (err > 0) {
+                elem->cached |= 1 << index;
+                elem->cache_val[index] = val;
+        } else {
+                usb_audio_dbg(chip, "Failed to set eq switch, err:%d\n", err);
+        }
+        return 1;
+}
+static int snd_us16x08_eq_get(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        int val = 0;
+        struct usb_mixer_elem_info *elem = kcontrol->private_data;
+        struct snd_us16x08_eq_store *store =
+                ((struct snd_us16x08_eq_store *) elem->private_data);
+        int index = ucontrol->id.index;
+        int b_idx = EQ_STORE_BAND_IDX(elem->head.id) - 1;
+        int p_idx = EQ_STORE_PARAM_IDX(elem->head.id);
+        val = store->val[b_idx][p_idx][index];
+        ucontrol->value.integer.value[0] = val;
+        return 0;
+}
+static int snd_us16x08_eq_put(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        struct usb_mixer_elem_info *elem = kcontrol->private_data;
+        struct snd_usb_audio *chip = elem->head.mixer->chip;
+        struct snd_us16x08_eq_store *store =
+                ((struct snd_us16x08_eq_store *) elem->private_data);
+        int index = ucontrol->id.index;
+        char buf[sizeof(eqs_msq)];
+        int val, err = 0;
+        int b_idx = EQ_STORE_BAND_IDX(elem->head.id) - 1;
+        int p_idx = EQ_STORE_PARAM_IDX(elem->head.id);
+        /* copy URB buffer from EQ template */
+        memcpy(buf, eqs_msq, sizeof(eqs_msq));
+        val = ucontrol->value.integer.value[0];
+        /* sanity check */
+        if (val < SND_US16X08_KCMIN(kcontrol)
+                || val > SND_US16X08_KCMAX(kcontrol))
+                return -EINVAL;
+        store->val[b_idx][p_idx][index] = val;
+        buf[20] = store->val[b_idx][3][index];
+        buf[17] = store->val[b_idx][2][index];
+        buf[14] = store->val[b_idx][1][index];
+        buf[11] = store->val[b_idx][0][index];
+        /* place channel index in URB buffer */
+        buf[5] = index + 1;
+        /* place EQ band in URB buffer */
+        buf[8] = b_idx + 1;
+        err = snd_us16x08_send_urb(chip, buf, sizeof(eqs_msq));
+        if (err > 0) {
+                /* store new value in EQ band cache */
+                elem->cached |= 1 << index;
+                elem->cache_val[index] = val;
+        } else {
+                usb_audio_dbg(chip, "Failed to set eq param, err:%d\n", err);
+        }
+        return 1;
+}
+static int snd_us16x08_meter_info(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_info *uinfo)
+{
+        uinfo->count = 1;
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+        uinfo->value.integer.max = 0x7FFF;
+        uinfo->value.integer.min = 0;
+        return 0;
+}
+/* calculate compressor index for reduction level request */
+static int snd_get_meter_comp_index(struct snd_us16x08_meter_store *store)
+{
+        int ret;
+        /* any channel active */
+        if (store->comp_active_index) {
+                /* check for stereo link */
+                if (store->comp_active_index & 0x20) {
+                        /* reset comp_index to left channel*/
+                        if (store->comp_index -
+                                store->comp_active_index > 1)
+                                store->comp_index =
+                                store->comp_active_index;
+                        ret = store->comp_index++ & 0x1F;
+                } else {
+                        /* no stereo link */
+                        ret = store->comp_active_index;
+                }
+        } else {
+                /* skip channels with no compressor active */
+                while (!store->comp_store->val[
+                        COMP_STORE_IDX(SND_US16X08_ID_COMP_SWITCH)]
+                        [store->comp_index - 1]
+                        && store->comp_index <= SND_US16X08_MAX_CHANNELS) {
+                        store->comp_index++;
+                }
+                ret = store->comp_index++;
+                if (store->comp_index > SND_US16X08_MAX_CHANNELS)
+                        store->comp_index = 1;
+        }
+        return ret;
+}
+/* retrieve the meter level values from URB message */
+static void get_meter_levels_from_urb(int s,
+        struct snd_us16x08_meter_store *store,
+        u8 *meter_urb)
+{
+        int val = MUC2(meter_urb, s) + (MUC3(meter_urb, s) << 8);
+        if (MUA0(meter_urb, s) == 0x61 && MUA1(meter_urb, s) == 0x02 &&
+                MUA2(meter_urb, s) == 0x04 && MUB0(meter_urb, s) == 0x62) {
+                if (MUC0(meter_urb, s) == 0x72)
+                        store->meter_level[MUB2(meter_urb, s) - 1] = val;
+                if (MUC0(meter_urb, s) == 0xb2)
+                        store->comp_level[MUB2(meter_urb, s) - 1] = val;
+        }
+        if (MUA0(meter_urb, s) == 0x61 && MUA1(meter_urb, s) == 0x02 &&
+                MUA2(meter_urb, s) == 0x02 && MUB0(meter_urb, s) == 0x62)
+                store->master_level[MUB2(meter_urb, s) - 1] = val;
+}
+/* Function to retrieve current meter values from the device.
+ *
+ * The device needs to be polled for meter values with an initial
+ * requests. It will return with a sequence of different meter value
+ * packages. The first request (case 0:) initiate this meter response sequence.
+ * After the third response, an additional request can be placed,
+ * to retrieve compressor reduction level value for given channel. This round
+ * trip channel selector will skip all inactive compressors.
+ * A mixer can interrupt this round-trip by selecting one ore two (stereo-link)
+ * specific channels.
+ */
+static int snd_us16x08_meter_get(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        int i, set;
+        struct usb_mixer_elem_info *elem = kcontrol->private_data;
+        struct snd_usb_audio *chip = elem->head.mixer->chip;
+        struct snd_us16x08_meter_store *store = elem->private_data;
+        u8 meter_urb[64];
+        char tmp[max(sizeof(mix_init_msg1), sizeof(mix_init_msg2))];
+        if (elem) {
+                store = (struct snd_us16x08_meter_store *) elem->private_data;
+                chip = elem->head.mixer->chip;
+        } else
+                return 0;
+        switch (kcontrol->private_value) {
+        case 0:
+                memcpy(tmp, mix_init_msg1, sizeof(mix_init_msg1));
+                snd_us16x08_send_urb(chip, tmp, 4);
+                snd_us16x08_recv_urb(chip, meter_urb,
+                        sizeof(meter_urb));
+                kcontrol->private_value++;
+                break;
+        case 1:
+                snd_us16x08_recv_urb(chip, meter_urb,
+                        sizeof(meter_urb));
+                kcontrol->private_value++;
+                break;
+        case 2:
+                snd_us16x08_recv_urb(chip, meter_urb,
+                        sizeof(meter_urb));
+                kcontrol->private_value++;
+                break;
+        case 3:
+                memcpy(tmp, mix_init_msg2, sizeof(mix_init_msg2));
+                tmp[2] = snd_get_meter_comp_index(store);
+                snd_us16x08_send_urb(chip, tmp, 10);
+                snd_us16x08_recv_urb(chip, meter_urb,
+                        sizeof(meter_urb));
+                kcontrol->private_value = 0;
+                break;
+        }
+        for (set = 0; set < 6; set++)
+                get_meter_levels_from_urb(set, store, meter_urb);
+        for (i = 0; i < SND_US16X08_MAX_CHANNELS; i++) {
+                ucontrol->value.integer.value[i] =
+                        store ? store->meter_level[i] : 0;
+        }
+        ucontrol->value.integer.value[i++] = store ? store->master_level[0] : 0;
+        ucontrol->value.integer.value[i++] = store ? store->master_level[1] : 0;
+        for (i = 2; i < SND_US16X08_MAX_CHANNELS + 2; i++)
+                ucontrol->value.integer.value[i + SND_US16X08_MAX_CHANNELS] =
+                store ? store->comp_level[i - 2] : 0;
+        return 1;
+}
+static int snd_us16x08_meter_put(struct snd_kcontrol *kcontrol,
+        struct snd_ctl_elem_value *ucontrol)
+{
+        struct usb_mixer_elem_info *elem = kcontrol->private_data;
+        struct snd_us16x08_meter_store *store = elem->private_data;
+        int val;
+        val = ucontrol->value.integer.value[0];
+        /* sanity check */
+        if (val < 0 || val >= SND_US16X08_MAX_CHANNELS)
+                return -EINVAL;
+        store->comp_active_index = val;
+        store->comp_index = val;
+        return 1;
+}
+static struct snd_kcontrol_new snd_us16x08_ch_boolean_ctl = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .count = 16,
+        .info = snd_us16x08_switch_info,
+        .get = snd_us16x08_channel_get,
+        .put = snd_us16x08_channel_put,
+        .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 1)
+};
+static struct snd_kcontrol_new snd_us16x08_ch_int_ctl = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .count = 16,
+        .info = snd_us16x08_mix_info,
+        .get = snd_us16x08_channel_get,
+        .put = snd_us16x08_channel_put,
+        .private_value = SND_US16X08_KCSET(SND_US16X08_FADER_BIAS, 1, 0, 133)
+};
+static struct snd_kcontrol_new snd_us16x08_pan_int_ctl = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .count = 16,
+        .info = snd_us16x08_mix_info,
+        .get = snd_us16x08_channel_get,
+        .put = snd_us16x08_channel_put,
+        .private_value = SND_US16X08_KCSET(SND_US16X08_FADER_BIAS, 1, 0, 255)
+};
+static struct snd_kcontrol_new snd_us16x08_master_ctl = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .count = 1,
+        .info = snd_us16x08_master_info,
+        .get = snd_us16x08_master_get,
+        .put = snd_us16x08_master_put,
+        .private_value = SND_US16X08_KCSET(SND_US16X08_FADER_BIAS, 1, 0, 133)
+};
+static struct snd_kcontrol_new snd_us16x08_route_ctl = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .count = 8,
+        .info = snd_us16x08_route_info,
+        .get = snd_us16x08_route_get,
+        .put = snd_us16x08_route_put,
+        .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 9)
+};
+static struct snd_kcontrol_new snd_us16x08_bus_ctl = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .count = 1,
+        .info = snd_us16x08_switch_info,
+        .get = snd_us16x08_bus_get,
+        .put = snd_us16x08_bus_put,
+        .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 1)
+};
+static struct snd_kcontrol_new snd_us16x08_compswitch_ctl = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .count = 16,
+        .info = snd_us16x08_switch_info,
+        .get = snd_us16x08_comp_get,
+        .put = snd_us16x08_comp_put,
+        .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 1)
+};
+static struct snd_kcontrol_new snd_us16x08_comp_threshold_ctl = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .count = 16,
+        .info = snd_us16x08_mix_info,
+        .get = snd_us16x08_comp_get,
+        .put = snd_us16x08_comp_put,
+        .private_value = SND_US16X08_KCSET(SND_US16X08_COMP_THRESHOLD_BIAS, 1,
+        0, 0x20)
+};
+static struct snd_kcontrol_new snd_us16x08_comp_ratio_ctl = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .count = 16,
+        .info = snd_us16x08_mix_info,
+        .get = snd_us16x08_comp_get,
+        .put = snd_us16x08_comp_put,
+        .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0,
+        sizeof(ratio_map) - 1), /*max*/
+};
+static struct snd_kcontrol_new snd_us16x08_comp_gain_ctl = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .count = 16,
+        .info = snd_us16x08_mix_info,
+        .get = snd_us16x08_comp_get,
+        .put = snd_us16x08_comp_put,
+        .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 0x14)
+};
+static struct snd_kcontrol_new snd_us16x08_comp_attack_ctl = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .count = 16,
+        .info = snd_us16x08_mix_info,
+        .get = snd_us16x08_comp_get,
+        .put = snd_us16x08_comp_put,
+        .private_value =
+        SND_US16X08_KCSET(SND_US16X08_COMP_ATTACK_BIAS, 1, 0, 0xc6),
+};
+static struct snd_kcontrol_new snd_us16x08_comp_release_ctl = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .count = 16,
+        .info = snd_us16x08_mix_info,
+        .get = snd_us16x08_comp_get,
+        .put = snd_us16x08_comp_put,
+        .private_value =
+        SND_US16X08_KCSET(SND_US16X08_COMP_RELEASE_BIAS, 1, 0, 0x63),
+};
+static struct snd_kcontrol_new snd_us16x08_eq_gain_ctl = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .count = 16,
+        .info = snd_us16x08_mix_info,
+        .get = snd_us16x08_eq_get,
+        .put = snd_us16x08_eq_put,
+        .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 24),
+};
+static struct snd_kcontrol_new snd_us16x08_eq_low_freq_ctl = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .count = 16,
+        .info = snd_us16x08_mix_info,
+        .get = snd_us16x08_eq_get,
+        .put = snd_us16x08_eq_put,
+        .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 0x1F),
+};
+static struct snd_kcontrol_new snd_us16x08_eq_mid_freq_ctl = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .count = 16,
+        .info = snd_us16x08_mix_info,
+        .get = snd_us16x08_eq_get,
+        .put = snd_us16x08_eq_put,
+        .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 0x3F)
+};
+static struct snd_kcontrol_new snd_us16x08_eq_mid_width_ctl = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .count = 16,
+        .info = snd_us16x08_mix_info,
+        .get = snd_us16x08_eq_get,
+        .put = snd_us16x08_eq_put,
+        .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 0x06)
+};
+static struct snd_kcontrol_new snd_us16x08_eq_high_freq_ctl = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .count = 16,
+        .info = snd_us16x08_mix_info,
+        .get = snd_us16x08_eq_get,
+        .put = snd_us16x08_eq_put,
+        .private_value =
+        SND_US16X08_KCSET(SND_US16X08_EQ_HIGHFREQ_BIAS, 1, 0, 0x1F)
+};
+static struct snd_kcontrol_new snd_us16x08_eq_switch_ctl = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .count = 16,
+        .info = snd_us16x08_switch_info,
+        .get = snd_us16x08_eqswitch_get,
+        .put = snd_us16x08_eqswitch_put,
+        .private_value = SND_US16X08_KCSET(SND_US16X08_NO_BIAS, 1, 0, 1)
+};
+static struct snd_kcontrol_new snd_us16x08_meter_ctl = {
+        .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+        .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+        .count = 1,
+        .info = snd_us16x08_meter_info,
+        .get = snd_us16x08_meter_get,
+        .put = snd_us16x08_meter_put
+};
+/* control store preparation */
+/* setup compressor store and assign default value */
+static struct snd_us16x08_comp_store *snd_us16x08_create_comp_store(void)
+{
+        int i = 0;
+        struct snd_us16x08_comp_store *tmp =
+                kmalloc(sizeof(struct snd_us16x08_comp_store), GFP_KERNEL);
+        if (tmp == NULL)
+                return NULL;
+        for (i = 0; i < SND_US16X08_MAX_CHANNELS; i++) {
+                tmp->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_THRESHOLD)][i]
+                        = 0x20;
+                tmp->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_RATIO)][i] = 0x00;
+                tmp->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_GAIN)][i] = 0x00;
+                tmp->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_SWITCH)][i] = 0x00;
+                tmp->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_ATTACK)][i] = 0x00;
+                tmp->val[COMP_STORE_IDX(SND_US16X08_ID_COMP_RELEASE)][i] = 0x00;
+        }
+        return tmp;
+}
+/* setup EQ store and assign default values */
+static struct snd_us16x08_eq_store *snd_us16x08_create_eq_store(void)
+{
+        int i, b_idx;
+        struct snd_us16x08_eq_store *tmp =
+                kmalloc(sizeof(struct snd_us16x08_eq_store), GFP_KERNEL);
+        if (tmp == NULL)
+                return NULL;
+        for (i = 0; i < SND_US16X08_MAX_CHANNELS; i++) {
+                for (b_idx = 0; b_idx < SND_US16X08_ID_EQ_BAND_COUNT; b_idx++) {
+                        tmp->val[b_idx][0][i] = 0x0c;
+                        tmp->val[b_idx][3][i] = 0x00;
+                        switch (b_idx) {
+                        case 0: /* EQ Low */
+                                tmp->val[b_idx][1][i] = 0x05;
+                                tmp->val[b_idx][2][i] = 0xff;
+                                break;
+                        case 1: /* EQ Mid low */
+                                tmp->val[b_idx][1][i] = 0x0e;
+                                tmp->val[b_idx][2][i] = 0x02;
+                                break;
+                        case 2: /* EQ Mid High */
+                                tmp->val[b_idx][1][i] = 0x1b;
+                                tmp->val[b_idx][2][i] = 0x02;
+                                break;
+                        case 3: /* EQ High */
+                                tmp->val[b_idx][1][i] = 0x2f
+                                        - SND_US16X08_EQ_HIGHFREQ_BIAS;
+                                tmp->val[b_idx][2][i] = 0xff;
+                                break;
+                        }
+                }
+        }
+        return tmp;
+}
+struct snd_us16x08_meter_store *snd_us16x08_create_meter_store(void)
+{
+        struct snd_us16x08_meter_store *tmp =
+                kzalloc(sizeof(struct snd_us16x08_meter_store), GFP_KERNEL);
+        if (!tmp)
+                return NULL;
+        tmp->comp_index = 1;
+        tmp->comp_active_index = 0;
+        return tmp;
+}
+static int add_new_ctl(struct usb_mixer_interface *mixer,
+        const struct snd_kcontrol_new *ncontrol,
+        int index, int val_type, int channels,
+        const char *name, const void *opt,
+        void (*freeer)(struct snd_kcontrol *kctl),
+        struct usb_mixer_elem_info **elem_ret)
+{
+        struct snd_kcontrol *kctl;
+        struct usb_mixer_elem_info *elem;
+        int err;
+        usb_audio_dbg(mixer->chip, "us16x08 add mixer %s\n", name);
+        elem = kzalloc(sizeof(*elem), GFP_KERNEL);
+        if (!elem)
+                return -ENOMEM;
+        elem->head.mixer = mixer;
+        elem->head.resume = NULL;
+        elem->control = 0;
+        elem->idx_off = 0;
+        elem->head.id = index;
+        elem->val_type = val_type;
+        elem->channels = channels;
+        elem->private_data = (void *) opt;
+        kctl = snd_ctl_new1(ncontrol, elem);
+        if (!kctl) {
+                kfree(elem);
+                return -ENOMEM;
+        }
+        kctl->private_free = freeer;
+        strlcpy(kctl->id.name, name, sizeof(kctl->id.name));
+        err = snd_usb_mixer_add_control(&elem->head, kctl);
+        if (err < 0)
+                return err;
+        if (elem_ret)
+                *elem_ret = elem;
+        return 0;
+}
+static struct snd_us16x08_control_params control_params;
+/* table of EQ controls */
+static struct snd_us16x08_control_params eq_controls[] = {
+        { /* EQ switch */
+                .kcontrol_new = &snd_us16x08_eq_switch_ctl,
+                .control_id = SND_US16X08_ID_EQENABLE,
+                .type = USB_MIXER_BOOLEAN,
+                .num_channels = 16,
+                .name = "EQ Switch",
+                .freeer = snd_usb_mixer_elem_free
+        },
+        { /* EQ low gain */
+                .kcontrol_new = &snd_us16x08_eq_gain_ctl,
+                .control_id = SND_US16X08_ID_EQLOWLEVEL,
+                .type = USB_MIXER_U8,
+                .num_channels = 16,
+                .name = "EQ Low Volume",
+                .freeer = snd_usb_mixer_elem_free
+        },
+        { /* EQ low freq */
+                .kcontrol_new = &snd_us16x08_eq_low_freq_ctl,
+                .control_id = SND_US16X08_ID_EQLOWFREQ,
+                .type = USB_MIXER_U8,
+                .num_channels = 16,
+                .name = "EQ Low Frequence",
+                .freeer = NULL
+        },
+        { /* EQ mid low gain */
+                .kcontrol_new = &snd_us16x08_eq_gain_ctl,
+                .control_id = SND_US16X08_ID_EQLOWMIDLEVEL,
+                .type = USB_MIXER_U8,
+                .num_channels = 16,
+                .name = "EQ MidLow Volume",
+                .freeer = snd_usb_mixer_elem_free
+        },
+        { /* EQ mid low freq */
+                .kcontrol_new = &snd_us16x08_eq_mid_freq_ctl,
+                .control_id = SND_US16X08_ID_EQLOWMIDFREQ,
+                .type = USB_MIXER_U8,
+                .num_channels = 16,
+                .name = "EQ MidLow Frequence",
+                .freeer = NULL
+        },
+        { /* EQ mid low Q */
+                .kcontrol_new = &snd_us16x08_eq_mid_width_ctl,
+                .control_id = SND_US16X08_ID_EQLOWMIDWIDTH,
+                .type = USB_MIXER_U8,
+                .num_channels = 16,
+                .name = "EQ MidQLow Q",
+                .freeer = NULL
+        },
+        { /* EQ mid high gain */
+                .kcontrol_new = &snd_us16x08_eq_gain_ctl,
+                .control_id = SND_US16X08_ID_EQHIGHMIDLEVEL,
+                .type = USB_MIXER_U8,
+                .num_channels = 16,
+                .name = "EQ MidHigh Volume",
+                .freeer = snd_usb_mixer_elem_free
+        },
+        { /* EQ mid high freq */
+                .kcontrol_new = &snd_us16x08_eq_mid_freq_ctl,
+                .control_id = SND_US16X08_ID_EQHIGHMIDFREQ,
+                .type = USB_MIXER_U8,
+                .num_channels = 16,
+                .name = "EQ MidHigh Frequence",
+                .freeer = NULL
+        },
+        { /* EQ mid high Q */
+                .kcontrol_new = &snd_us16x08_eq_mid_width_ctl,
+                .control_id = SND_US16X08_ID_EQHIGHMIDWIDTH,
+                .type = USB_MIXER_U8,
+                .num_channels = 16,
+                .name = "EQ MidHigh Q",
+                .freeer = NULL
+        },
+        { /* EQ high gain */
+                .kcontrol_new = &snd_us16x08_eq_gain_ctl,
+                .control_id = SND_US16X08_ID_EQHIGHLEVEL,
+                .type = USB_MIXER_U8,
+                .num_channels = 16,
+                .name = "EQ High Volume",
+                .freeer = snd_usb_mixer_elem_free
+        },
+        { /* EQ low freq */
+                .kcontrol_new = &snd_us16x08_eq_high_freq_ctl,
+                .control_id = SND_US16X08_ID_EQHIGHFREQ,
+                .type = USB_MIXER_U8,
+                .num_channels = 16,
+                .name = "EQ High Frequence",
+                .freeer = NULL
+        },
+};
+/* table of compressor controls */
+static struct snd_us16x08_control_params comp_controls[] = {
+        { /* Comp enable */
+                .kcontrol_new = &snd_us16x08_compswitch_ctl,
+                .control_id = SND_US16X08_ID_COMP_SWITCH,
+                .type = USB_MIXER_BOOLEAN,
+                .num_channels = 16,
+                .name = "Compressor Switch",
+                .freeer = snd_usb_mixer_elem_free
+        },
+        { /* Comp threshold */
+                .kcontrol_new = &snd_us16x08_comp_threshold_ctl,
+                .control_id = SND_US16X08_ID_COMP_THRESHOLD,
+                .type = USB_MIXER_U8,
+                .num_channels = 16,
+                .name = "Compressor Threshold Volume",
+                .freeer = NULL
+        },
+        { /* Comp ratio */
+                .kcontrol_new = &snd_us16x08_comp_ratio_ctl,
+                .control_id = SND_US16X08_ID_COMP_RATIO,
+                .type = USB_MIXER_U8,
+                .num_channels = 16,
+                .name = "Compressor Ratio",
+                .freeer = NULL
+        },
+        { /* Comp attack */
+                .kcontrol_new = &snd_us16x08_comp_attack_ctl,
+                .control_id = SND_US16X08_ID_COMP_ATTACK,
+                .type = USB_MIXER_U8,
+                .num_channels = 16,
+                .name = "Compressor Attack",
+                .freeer = NULL
+        },
+        { /* Comp release */
+                .kcontrol_new = &snd_us16x08_comp_release_ctl,
+                .control_id = SND_US16X08_ID_COMP_RELEASE,
+                .type = USB_MIXER_U8,
+                .num_channels = 16,
+                .name = "Compressor Release",
+                .freeer = NULL
+        },
+        { /* Comp gain */
+                .kcontrol_new = &snd_us16x08_comp_gain_ctl,
+                .control_id = SND_US16X08_ID_COMP_GAIN,
+                .type = USB_MIXER_U8,
+                .num_channels = 16,
+                .name = "Compressor Volume",
+                .freeer = NULL
+        },
+};
+/* table of channel controls */
+static struct snd_us16x08_control_params channel_controls[] = {
+        { /* Phase */
+                .kcontrol_new = &snd_us16x08_ch_boolean_ctl,
+                .control_id = SND_US16X08_ID_PHASE,
+                .type = USB_MIXER_BOOLEAN,
+                .num_channels = 16,
+                .name = "Phase Switch",
+                .freeer = snd_usb_mixer_elem_free,
+                .default_val = 0
+        },
+        { /* Fader */
+                .kcontrol_new = &snd_us16x08_ch_int_ctl,
+                .control_id = SND_US16X08_ID_FADER,
+                .type = USB_MIXER_U8,
+                .num_channels = 16,
+                .name = "Line Volume",
+                .freeer = NULL,
+                .default_val = 127
+        },
+        { /* Mute */
+                .kcontrol_new = &snd_us16x08_ch_boolean_ctl,
+                .control_id = SND_US16X08_ID_MUTE,
+                .type = USB_MIXER_BOOLEAN,
+                .num_channels = 16,
+                .name = "Mute Switch",
+                .freeer = NULL,
+                .default_val = 0
+        },
+        { /* Pan */
+                .kcontrol_new = &snd_us16x08_pan_int_ctl,
+                .control_id = SND_US16X08_ID_PAN,
+                .type = USB_MIXER_U16,
+                .num_channels = 16,
+                .name = "Pan Left-Right Volume",
+                .freeer = NULL,
+                .default_val = 127
+        },
+};
+/* table of master controls */
+static struct snd_us16x08_control_params master_controls[] = {
+        { /* Master */
+                .kcontrol_new = &snd_us16x08_master_ctl,
+                .control_id = SND_US16X08_ID_FADER,
+                .type = USB_MIXER_U8,
+                .num_channels = 16,
+                .name = "Master Volume",
+                .freeer = NULL,
+                .default_val = 127
+        },
+        { /* Bypass */
+                .kcontrol_new = &snd_us16x08_bus_ctl,
+                .control_id = SND_US16X08_ID_BYPASS,
+                .type = USB_MIXER_BOOLEAN,
+                .num_channels = 16,
+                .name = "DSP Bypass Switch",
+                .freeer = NULL,
+                .default_val = 0
+        },
+        { /* Buss out */
+                .kcontrol_new = &snd_us16x08_bus_ctl,
+                .control_id = SND_US16X08_ID_BUSS_OUT,
+                .type = USB_MIXER_BOOLEAN,
+                .num_channels = 16,
+                .name = "Buss Out Switch",
+                .freeer = NULL,
+                .default_val = 0
+        },
+        { /* Master mute */
+                .kcontrol_new = &snd_us16x08_bus_ctl,
+                .control_id = SND_US16X08_ID_MUTE,
+                .type = USB_MIXER_BOOLEAN,
+                .num_channels = 16,
+                .name = "Master Mute Switch",
+                .freeer = NULL,
+                .default_val = 0
+        },
+};
+int snd_us16x08_controls_create(struct usb_mixer_interface *mixer)
+{
+        int i, j;
+        int err;
+        struct usb_mixer_elem_info *elem;
+        struct snd_us16x08_comp_store *comp_store;
+        struct snd_us16x08_meter_store *meter_store;
+        struct snd_us16x08_eq_store *eq_store;
+        /* just check for non-MIDI interface */
+        if (mixer->hostif->desc.bInterfaceNumber == 3) {
+                /* create compressor mixer elements */
+                comp_store = snd_us16x08_create_comp_store();
+                if (comp_store == NULL)
+                        return -ENOMEM;
+                /* create eq store */
+                eq_store = snd_us16x08_create_eq_store();
+                if (eq_store == NULL) {
+                        kfree(comp_store);
+                        return -ENOMEM;
+                }
+                /* create meters store */
+                meter_store = snd_us16x08_create_meter_store();
+                if (meter_store == NULL) {
+                        kfree(comp_store);
+                        kfree(eq_store);
+                        return -ENOMEM;
+                }
+                /* add routing control */
+                err = add_new_ctl(mixer, &snd_us16x08_route_ctl,
+                        SND_US16X08_ID_ROUTE, USB_MIXER_U8, 8, "Line Out Route",
+                        NULL, NULL, &elem);
+                if (err < 0) {
+                        usb_audio_dbg(mixer->chip,
+                                "Failed to create route control, err:%d\n",
+                                err);
+                        return err;
+                }
+                for (i = 0; i < 8; i++)
+                        elem->cache_val[i] = i < 2 ? i : i + 2;
+                elem->cached = 0xff;
+                /* add master controls */
+                for (i = 0;
+                        i < sizeof(master_controls)
+                        / sizeof(control_params);
+                        i++) {
+                        err = add_new_ctl(mixer,
+                                master_controls[i].kcontrol_new,
+                                master_controls[i].control_id,
+                                master_controls[i].type,
+                                master_controls[i].num_channels,
+                                master_controls[i].name,
+                                comp_store,
+                                master_controls[i].freeer, &elem);
+                        if (err < 0)
+                                return err;
+                        elem->cache_val[0] = master_controls[i].default_val;
+                        elem->cached = 1;
+                }
+                /* add channel controls */
+                for (i = 0;
+                        i < sizeof(channel_controls)
+                        / sizeof(control_params);
+                        i++) {
+                        err = add_new_ctl(mixer,
+                                channel_controls[i].kcontrol_new,
+                                channel_controls[i].control_id,
+                                channel_controls[i].type,
+                                channel_controls[i].num_channels,
+                                channel_controls[i].name,
+                                comp_store,
+                                channel_controls[i].freeer, &elem);
+                        if (err < 0)
+                                return err;
+                        for (j = 0; j < SND_US16X08_MAX_CHANNELS; j++) {
+                                elem->cache_val[j] =
+                                        channel_controls[i].default_val;
+                        }
+                        elem->cached = 0xffff;
+                }
+                /* add EQ controls */
+                for (i = 0; i < sizeof(eq_controls) /
+                        sizeof(control_params); i++) {
+                        err = add_new_ctl(mixer,
+                                eq_controls[i].kcontrol_new,
+                                eq_controls[i].control_id,
+                                eq_controls[i].type,
+                                eq_controls[i].num_channels,
+                                eq_controls[i].name,
+                                eq_store,
+                                eq_controls[i].freeer, NULL);
+                        if (err < 0)
+                                return err;
+                }
+                /* add compressor controls */
+                for (i = 0;
+                        i < sizeof(comp_controls)
+                        / sizeof(control_params);
+                        i++) {
+                        err = add_new_ctl(mixer,
+                                comp_controls[i].kcontrol_new,
+                                comp_controls[i].control_id,
+                                comp_controls[i].type,
+                                comp_controls[i].num_channels,
+                                comp_controls[i].name,
+                                comp_store,
+                                comp_controls[i].freeer, NULL);
+                        if (err < 0)
+                                return err;
+                }
+                /* meter function 'get' must access to compressor store
+                 * so place a reference here
+                 */
+                meter_store->comp_store = comp_store;
+                err = add_new_ctl(mixer, &snd_us16x08_meter_ctl,
+                        SND_US16X08_ID_METER, USB_MIXER_U16, 0, "Level Meter",
+                        (void *) meter_store, snd_usb_mixer_elem_free, NULL);
+                if (err < 0)
+                        return err;
+        }
+        return 0;
+}
diff --git a/sound/usb/mixer_us16x08.h b/sound/usb/mixer_us16x08.h
new file mode 100644
index 000000000000..64f89b5eca2d
--- /dev/null
+++ b/sound/usb/mixer_us16x08.h
@@ -0,0 +1,122 @@
+#ifndef __USB_MIXER_US16X08_H
+#define __USB_MIXER_US16X08_H
+#define SND_US16X08_MAX_CHANNELS 16
+/* define some bias, cause some alsa-mixers wont work with
+ * negative ranges or if mixer-min != 0
+ */
+#define SND_US16X08_NO_BIAS 0
+#define SND_US16X08_FADER_BIAS 127
+#define SND_US16X08_EQ_HIGHFREQ_BIAS 0x20
+#define SND_US16X08_COMP_THRESHOLD_BIAS 0x20
+#define SND_US16X08_COMP_ATTACK_BIAS 2
+#define SND_US16X08_COMP_RELEASE_BIAS 1
+/* get macro for components of kcontrol private_value */
+#define SND_US16X08_KCBIAS(x) (((x)->private_value >> 24) & 0xff)
+#define SND_US16X08_KCSTEP(x) (((x)->private_value >> 16) & 0xff)
+#define SND_US16X08_KCMIN(x) (((x)->private_value >> 8) & 0xff)
+#define SND_US16X08_KCMAX(x) (((x)->private_value >> 0) & 0xff)
+/* set macro for kcontrol private_value */
+#define SND_US16X08_KCSET(bias, step, min, max)  \
+        (((bias) << 24) | ((step) << 16) | ((min) << 8) | (max))
+/* the URB request/type to control Tascam mixers */
+#define SND_US16X08_URB_REQUEST 0x1D
+#define SND_US16X08_URB_REQUESTTYPE 0x40
+/* the URB params to retrieve meter ranges */
+#define SND_US16X08_URB_METER_REQUEST       0x1e
+#define SND_US16X08_URB_METER_REQUESTTYPE   0xc0
+#define MUA0(x, y) ((x)[(y) * 10 + 4])
+#define MUA1(x, y) ((x)[(y) * 10 + 5])
+#define MUA2(x, y) ((x)[(y) * 10 + 6])
+#define MUB0(x, y) ((x)[(y) * 10 + 7])
+#define MUB1(x, y) ((x)[(y) * 10 + 8])
+#define MUB2(x, y) ((x)[(y) * 10 + 9])
+#define MUC0(x, y) ((x)[(y) * 10 + 10])
+#define MUC1(x, y) ((x)[(y) * 10 + 11])
+#define MUC2(x, y) ((x)[(y) * 10 + 12])
+#define MUC3(x, y) ((x)[(y) * 10 + 13])
+/* Common Channel control IDs */
+#define SND_US16X08_ID_BYPASS 0x45
+#define SND_US16X08_ID_BUSS_OUT 0x44
+#define SND_US16X08_ID_PHASE 0x85
+#define SND_US16X08_ID_MUTE 0x83
+#define SND_US16X08_ID_FADER 0x81
+#define SND_US16X08_ID_PAN 0x82
+#define SND_US16X08_ID_METER 0xB1
+#define SND_US16X08_ID_EQ_BAND_COUNT 4
+#define SND_US16X08_ID_EQ_PARAM_COUNT 4
+/* EQ level IDs */
+#define SND_US16X08_ID_EQLOWLEVEL 0x01
+#define SND_US16X08_ID_EQLOWMIDLEVEL 0x02
+#define SND_US16X08_ID_EQHIGHMIDLEVEL 0x03
+#define SND_US16X08_ID_EQHIGHLEVEL 0x04
+/* EQ frequence IDs */
+#define SND_US16X08_ID_EQLOWFREQ 0x11
+#define SND_US16X08_ID_EQLOWMIDFREQ 0x12
+#define SND_US16X08_ID_EQHIGHMIDFREQ 0x13
+#define SND_US16X08_ID_EQHIGHFREQ 0x14
+/* EQ width IDs */
+#define SND_US16X08_ID_EQLOWMIDWIDTH 0x22
+#define SND_US16X08_ID_EQHIGHMIDWIDTH 0x23
+#define SND_US16X08_ID_EQENABLE 0x30
+#define EQ_STORE_BAND_IDX(x) ((x) & 0xf)
+#define EQ_STORE_PARAM_IDX(x) (((x) & 0xf0) >> 4)
+#define SND_US16X08_ID_ROUTE 0x00
+/* Compressor Ids */
+#define SND_US16X08_ID_COMP_BASE        0x32
+#define SND_US16X08_ID_COMP_THRESHOLD   SND_US16X08_ID_COMP_BASE
+#define SND_US16X08_ID_COMP_RATIO       (SND_US16X08_ID_COMP_BASE + 1)
+#define SND_US16X08_ID_COMP_ATTACK      (SND_US16X08_ID_COMP_BASE + 2)
+#define SND_US16X08_ID_COMP_RELEASE     (SND_US16X08_ID_COMP_BASE + 3)
+#define SND_US16X08_ID_COMP_GAIN        (SND_US16X08_ID_COMP_BASE + 4)
+#define SND_US16X08_ID_COMP_SWITCH      (SND_US16X08_ID_COMP_BASE + 5)
+#define SND_US16X08_ID_COMP_COUNT       6
+#define COMP_STORE_IDX(x) ((x) - SND_US16X08_ID_COMP_BASE)
+struct snd_us16x08_eq_store {
+        u8 val[SND_US16X08_ID_EQ_BAND_COUNT][SND_US16X08_ID_EQ_PARAM_COUNT]
+                [SND_US16X08_MAX_CHANNELS];
+};
+struct snd_us16x08_comp_store {
+        u8 val[SND_US16X08_ID_COMP_COUNT][SND_US16X08_MAX_CHANNELS];
+};
+struct snd_us16x08_meter_store {
+        int meter_level[SND_US16X08_MAX_CHANNELS];
+        int master_level[2]; /* level of meter for master output */
+        int comp_index; /* round trip channel selector */
+        int comp_active_index; /* channel select from user space mixer */
+        int comp_level[16]; /* compressor reduction level */
+        struct snd_us16x08_comp_store *comp_store;
+};
+struct snd_us16x08_control_params {
+        struct snd_kcontrol_new *kcontrol_new;
+        int control_id;
+        int type;
+        int num_channels;
+        const char *name;
+        void (*freeer)(struct snd_kcontrol *kctl);
+        int default_val;
+};
+#define snd_us16x08_switch_info snd_ctl_boolean_mono_info
+int snd_us16x08_controls_create(struct usb_mixer_interface *mixer);
+#endif /* __USB_MIXER_US16X08_H */
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index eb4b9f7a571e..01eff6ce6401 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1360,6 +1360,21 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip,
                if (fp->altsetting == 3)
                        return SNDRV_PCM_FMTBIT_DSD_U32_BE;
                break;
+        /* Amanero Combo384 USB interface with native DSD support */
+        case USB_ID(0x16d0, 0x071a):
+                if (fp->altsetting == 2) {
+                        switch (chip->dev->descriptor.bcdDevice) {
+                        case 0x199:
+                                return SNDRV_PCM_FMTBIT_DSD_U32_LE;
+                        case 0x19b:
+                                return SNDRV_PCM_FMTBIT_DSD_U32_BE;
+                        default:
+                                break;
+                        }
+                }
+                break;
        default:
                break;
        }
diff --git a/sound/x86/Kconfig b/sound/x86/Kconfig
new file mode 100644
index 000000000000..30d066e4885e
--- /dev/null
+++ b/sound/x86/Kconfig
@@ -0,0 +1,15 @@
+menuconfig SND_X86
+        tristate "X86 sound devices"
+        depends on X86
+        ---help---
+          X86 sound devices that don't fall under SoC or PCI categories
+if SND_X86
+config HDMI_LPE_AUDIO
+        tristate "HDMI audio without HDaudio on Intel Atom platforms"
+        depends on DRM_I915
+        help
+         Choose this option to support HDMI LPE Audio mode
+endif   # SND_X86
diff --git a/sound/x86/Makefile b/sound/x86/Makefile
new file mode 100644
index 000000000000..7ff919808320
--- /dev/null
+++ b/sound/x86/Makefile
@@ -0,0 +1,4 @@
+snd-hdmi-lpe-audio-objs += \
+        intel_hdmi_audio.o
+obj-$(CONFIG_HDMI_LPE_AUDIO) += snd-hdmi-lpe-audio.o
diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c
new file mode 100644
index 000000000000..360cff35b239
--- /dev/null
+++ b/sound/x86/intel_hdmi_audio.c
@@ -0,0 +1,1851 @@
+/*
+ *   intel_hdmi_audio.c - Intel HDMI audio driver
+ *
+ *  Copyright (C) 2016 Intel Corp
+ *  Authors:    Sailaja Bandarupalli <sailaja.bandarupalli@intel.com>
+ *              Ramesh Babu K V <ramesh.babu@intel.com>
+ *              Vaibhav Agarwal <vaibhav.agarwal@intel.com>
+ *              Jerome Anand <jerome.anand@intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  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; version 2 of the License.
+ *
+ *  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.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ * ALSA driver for Intel HDMI audio
+ */
+#include <linux/types.h>
+#include <linux/platform_device.h>
+#include <linux/io.h>
+#include <linux/slab.h>
+#include <linux/module.h>
+#include <linux/interrupt.h>
+#include <linux/pm_runtime.h>
+#include <linux/dma-mapping.h>
+#include <linux/delay.h>
+#include <asm/cacheflush.h>
+#include <sound/core.h>
+#include <sound/asoundef.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <sound/jack.h>
+#include <drm/drm_edid.h>
+#include <drm/intel_lpe_audio.h>
+#include "intel_hdmi_audio.h"
+/*standard module options for ALSA. This module supports only one card*/
+static int hdmi_card_index = SNDRV_DEFAULT_IDX1;
+static char *hdmi_card_id = SNDRV_DEFAULT_STR1;
+module_param_named(index, hdmi_card_index, int, 0444);
+MODULE_PARM_DESC(index,
+                "Index value for INTEL Intel HDMI Audio controller.");
+module_param_named(id, hdmi_card_id, charp, 0444);
+MODULE_PARM_DESC(id,
+                "ID string for INTEL Intel HDMI Audio controller.");
+/*
+ * ELD SA bits in the CEA Speaker Allocation data block
+ */
+static const int eld_speaker_allocation_bits[] = {
+        [0] = FL | FR,
+        [1] = LFE,
+        [2] = FC,
+        [3] = RL | RR,
+        [4] = RC,
+        [5] = FLC | FRC,
+        [6] = RLC | RRC,
+        /* the following are not defined in ELD yet */
+        [7] = 0,
+};
+/*
+ * This is an ordered list!
+ *
+ * The preceding ones have better chances to be selected by
+ * hdmi_channel_allocation().
+ */
+static struct cea_channel_speaker_allocation channel_allocations[] = {
+/*                        channel:   7     6    5    4    3     2    1    0  */
+{ .ca_index = 0x00,  .speakers = {   0,    0,   0,   0,   0,    0,  FR,  FL } },
+                                /* 2.1 */
+{ .ca_index = 0x01,  .speakers = {   0,    0,   0,   0,   0,  LFE,  FR,  FL } },
+                                /* Dolby Surround */
+{ .ca_index = 0x02,  .speakers = {   0,    0,   0,   0,  FC,    0,  FR,  FL } },
+                                /* surround40 */
+{ .ca_index = 0x08,  .speakers = {   0,    0,  RR,  RL,   0,    0,  FR,  FL } },
+                                /* surround41 */
+{ .ca_index = 0x09,  .speakers = {   0,    0,  RR,  RL,   0,  LFE,  FR,  FL } },
+                                /* surround50 */
+{ .ca_index = 0x0a,  .speakers = {   0,    0,  RR,  RL,  FC,    0,  FR,  FL } },
+                                /* surround51 */
+{ .ca_index = 0x0b,  .speakers = {   0,    0,  RR,  RL,  FC,  LFE,  FR,  FL } },
+                                /* 6.1 */
+{ .ca_index = 0x0f,  .speakers = {   0,   RC,  RR,  RL,  FC,  LFE,  FR,  FL } },
+                                /* surround71 */
+{ .ca_index = 0x13,  .speakers = { RRC,  RLC,  RR,  RL,  FC,  LFE,  FR,  FL } },
+{ .ca_index = 0x03,  .speakers = {   0,    0,   0,   0,  FC,  LFE,  FR,  FL } },
+{ .ca_index = 0x04,  .speakers = {   0,    0,   0,  RC,   0,    0,  FR,  FL } },
+{ .ca_index = 0x05,  .speakers = {   0,    0,   0,  RC,   0,  LFE,  FR,  FL } },
+{ .ca_index = 0x06,  .speakers = {   0,    0,   0,  RC,  FC,    0,  FR,  FL } },
+{ .ca_index = 0x07,  .speakers = {   0,    0,   0,  RC,  FC,  LFE,  FR,  FL } },
+{ .ca_index = 0x0c,  .speakers = {   0,   RC,  RR,  RL,   0,    0,  FR,  FL } },
+{ .ca_index = 0x0d,  .speakers = {   0,   RC,  RR,  RL,   0,  LFE,  FR,  FL } },
+{ .ca_index = 0x0e,  .speakers = {   0,   RC,  RR,  RL,  FC,    0,  FR,  FL } },
+{ .ca_index = 0x10,  .speakers = { RRC,  RLC,  RR,  RL,   0,    0,  FR,  FL } },
+{ .ca_index = 0x11,  .speakers = { RRC,  RLC,  RR,  RL,   0,  LFE,  FR,  FL } },
+{ .ca_index = 0x12,  .speakers = { RRC,  RLC,  RR,  RL,  FC,    0,  FR,  FL } },
+{ .ca_index = 0x14,  .speakers = { FRC,  FLC,   0,   0,   0,    0,  FR,  FL } },
+{ .ca_index = 0x15,  .speakers = { FRC,  FLC,   0,   0,   0,  LFE,  FR,  FL } },
+{ .ca_index = 0x16,  .speakers = { FRC,  FLC,   0,   0,  FC,    0,  FR,  FL } },
+{ .ca_index = 0x17,  .speakers = { FRC,  FLC,   0,   0,  FC,  LFE,  FR,  FL } },
+{ .ca_index = 0x18,  .speakers = { FRC,  FLC,   0,  RC,   0,    0,  FR,  FL } },
+{ .ca_index = 0x19,  .speakers = { FRC,  FLC,   0,  RC,   0,  LFE,  FR,  FL } },
+{ .ca_index = 0x1a,  .speakers = { FRC,  FLC,   0,  RC,  FC,    0,  FR,  FL } },
+{ .ca_index = 0x1b,  .speakers = { FRC,  FLC,   0,  RC,  FC,  LFE,  FR,  FL } },
+{ .ca_index = 0x1c,  .speakers = { FRC,  FLC,  RR,  RL,   0,    0,  FR,  FL } },
+{ .ca_index = 0x1d,  .speakers = { FRC,  FLC,  RR,  RL,   0,  LFE,  FR,  FL } },
+{ .ca_index = 0x1e,  .speakers = { FRC,  FLC,  RR,  RL,  FC,    0,  FR,  FL } },
+{ .ca_index = 0x1f,  .speakers = { FRC,  FLC,  RR,  RL,  FC,  LFE,  FR,  FL } },
+};
+static const struct channel_map_table map_tables[] = {
+        { SNDRV_CHMAP_FL,       0x00,   FL },
+        { SNDRV_CHMAP_FR,       0x01,   FR },
+        { SNDRV_CHMAP_RL,       0x04,   RL },
+        { SNDRV_CHMAP_RR,       0x05,   RR },
+        { SNDRV_CHMAP_LFE,      0x02,   LFE },
+        { SNDRV_CHMAP_FC,       0x03,   FC },
+        { SNDRV_CHMAP_RLC,      0x06,   RLC },
+        { SNDRV_CHMAP_RRC,      0x07,   RRC },
+        {} /* terminator */
+};
+/* hardware capability structure */
+static const struct snd_pcm_hardware had_pcm_hardware = {
+        .info = (SNDRV_PCM_INFO_INTERLEAVED |
+                SNDRV_PCM_INFO_MMAP |
+                SNDRV_PCM_INFO_MMAP_VALID |
+                SNDRV_PCM_INFO_NO_PERIOD_WAKEUP),
+        .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+                    SNDRV_PCM_FMTBIT_S24_LE |
+                    SNDRV_PCM_FMTBIT_S32_LE),
+        .rates = SNDRV_PCM_RATE_32000 |
+                SNDRV_PCM_RATE_44100 |
+                SNDRV_PCM_RATE_48000 |
+                SNDRV_PCM_RATE_88200 |
+                SNDRV_PCM_RATE_96000 |
+                SNDRV_PCM_RATE_176400 |
+                SNDRV_PCM_RATE_192000,
+        .rate_min = HAD_MIN_RATE,
+        .rate_max = HAD_MAX_RATE,
+        .channels_min = HAD_MIN_CHANNEL,
+        .channels_max = HAD_MAX_CHANNEL,
+        .buffer_bytes_max = HAD_MAX_BUFFER,
+        .period_bytes_min = HAD_MIN_PERIOD_BYTES,
+        .period_bytes_max = HAD_MAX_PERIOD_BYTES,
+        .periods_min = HAD_MIN_PERIODS,
+        .periods_max = HAD_MAX_PERIODS,
+        .fifo_size = HAD_FIFO_SIZE,
+};
+/* Get the active PCM substream;
+ * Call had_substream_put() for unreferecing.
+ * Don't call this inside had_spinlock, as it takes by itself
+ */
+static struct snd_pcm_substream *
+had_substream_get(struct snd_intelhad *intelhaddata)
+{
+        struct snd_pcm_substream *substream;
+        unsigned long flags;
+        spin_lock_irqsave(&intelhaddata->had_spinlock, flags);
+        substream = intelhaddata->stream_info.substream;
+        if (substream)
+                intelhaddata->stream_info.substream_refcount++;
+        spin_unlock_irqrestore(&intelhaddata->had_spinlock, flags);
+        return substream;
+}
+/* Unref the active PCM substream;
+ * Don't call this inside had_spinlock, as it takes by itself
+ */
+static void had_substream_put(struct snd_intelhad *intelhaddata)
+{
+        unsigned long flags;
+        spin_lock_irqsave(&intelhaddata->had_spinlock, flags);
+        intelhaddata->stream_info.substream_refcount--;
+        spin_unlock_irqrestore(&intelhaddata->had_spinlock, flags);
+}
+/* Register access functions */
+static u32 had_read_register_raw(struct snd_intelhad *ctx, u32 reg)
+{
+        return ioread32(ctx->mmio_start + ctx->had_config_offset + reg);
+}
+static void had_write_register_raw(struct snd_intelhad *ctx, u32 reg, u32 val)
+{
+        iowrite32(val, ctx->mmio_start + ctx->had_config_offset + reg);
+}
+static void had_read_register(struct snd_intelhad *ctx, u32 reg, u32 *val)
+{
+        if (!ctx->connected)
+                *val = 0;
+        else
+                *val = had_read_register_raw(ctx, reg);
+}
+static void had_write_register(struct snd_intelhad *ctx, u32 reg, u32 val)
+{
+        if (ctx->connected)
+                had_write_register_raw(ctx, reg, val);
+}
+/*
+ * enable / disable audio configuration
+ *
+ * The normal read/modify should not directly be used on VLV2 for
+ * updating AUD_CONFIG register.
+ * This is because:
+ * Bit6 of AUD_CONFIG register is writeonly due to a silicon bug on VLV2
+ * HDMI IP. As a result a read-modify of AUD_CONFIG regiter will always
+ * clear bit6. AUD_CONFIG[6:4] represents the "channels" field of the
+ * register. This field should be 1xy binary for configuration with 6 or
+ * more channels. Read-modify of AUD_CONFIG (Eg. for enabling audio)
+ * causes the "channels" field to be updated as 0xy binary resulting in
+ * bad audio. The fix is to always write the AUD_CONFIG[6:4] with
+ * appropriate value when doing read-modify of AUD_CONFIG register.
+ */
+static void had_enable_audio(struct snd_intelhad *intelhaddata,
+                             bool enable)
+{
+        /* update the cached value */
+        intelhaddata->aud_config.regx.aud_en = enable;
+        had_write_register(intelhaddata, AUD_CONFIG,
+                           intelhaddata->aud_config.regval);
+}
+/* forcibly ACKs to both BUFFER_DONE and BUFFER_UNDERRUN interrupts */
+static void had_ack_irqs(struct snd_intelhad *ctx)
+{
+        u32 status_reg;
+        if (!ctx->connected)
+                return;
+        had_read_register(ctx, AUD_HDMI_STATUS, &status_reg);
+        status_reg |= HDMI_AUDIO_BUFFER_DONE | HDMI_AUDIO_UNDERRUN;
+        had_write_register(ctx, AUD_HDMI_STATUS, status_reg);
+        had_read_register(ctx, AUD_HDMI_STATUS, &status_reg);
+}
+/* Reset buffer pointers */
+static void had_reset_audio(struct snd_intelhad *intelhaddata)
+{
+        had_write_register(intelhaddata, AUD_HDMI_STATUS,
+                           AUD_HDMI_STATUSG_MASK_FUNCRST);
+        had_write_register(intelhaddata, AUD_HDMI_STATUS, 0);
+}
+/*
+ * initialize audio channel status registers
+ * This function is called in the prepare callback
+ */
+static int had_prog_status_reg(struct snd_pcm_substream *substream,
+                        struct snd_intelhad *intelhaddata)
+{
+        union aud_cfg cfg_val = {.regval = 0};
+        union aud_ch_status_0 ch_stat0 = {.regval = 0};
+        union aud_ch_status_1 ch_stat1 = {.regval = 0};
+        ch_stat0.regx.lpcm_id = (intelhaddata->aes_bits &
+                                          IEC958_AES0_NONAUDIO) >> 1;
+        ch_stat0.regx.clk_acc = (intelhaddata->aes_bits &
+                                          IEC958_AES3_CON_CLOCK) >> 4;
+        cfg_val.regx.val_bit = ch_stat0.regx.lpcm_id;
+        switch (substream->runtime->rate) {
+        case AUD_SAMPLE_RATE_32:
+                ch_stat0.regx.samp_freq = CH_STATUS_MAP_32KHZ;
+                break;
+        case AUD_SAMPLE_RATE_44_1:
+                ch_stat0.regx.samp_freq = CH_STATUS_MAP_44KHZ;
+                break;
+        case AUD_SAMPLE_RATE_48:
+                ch_stat0.regx.samp_freq = CH_STATUS_MAP_48KHZ;
+                break;
+        case AUD_SAMPLE_RATE_88_2:
+                ch_stat0.regx.samp_freq = CH_STATUS_MAP_88KHZ;
+                break;
+        case AUD_SAMPLE_RATE_96:
+                ch_stat0.regx.samp_freq = CH_STATUS_MAP_96KHZ;
+                break;
+        case AUD_SAMPLE_RATE_176_4:
+                ch_stat0.regx.samp_freq = CH_STATUS_MAP_176KHZ;
+                break;
+        case AUD_SAMPLE_RATE_192:
+                ch_stat0.regx.samp_freq = CH_STATUS_MAP_192KHZ;
+                break;
+        default:
+                /* control should never come here */
+                return -EINVAL;
+        }
+        had_write_register(intelhaddata,
+                           AUD_CH_STATUS_0, ch_stat0.regval);
+        switch (substream->runtime->format) {
+        case SNDRV_PCM_FORMAT_S16_LE:
+                ch_stat1.regx.max_wrd_len = MAX_SMPL_WIDTH_20;
+                ch_stat1.regx.wrd_len = SMPL_WIDTH_16BITS;
+                break;
+        case SNDRV_PCM_FORMAT_S24_LE:
+        case SNDRV_PCM_FORMAT_S32_LE:
+                ch_stat1.regx.max_wrd_len = MAX_SMPL_WIDTH_24;
+                ch_stat1.regx.wrd_len = SMPL_WIDTH_24BITS;
+                break;
+        default:
+                return -EINVAL;
+        }
+        had_write_register(intelhaddata,
+                           AUD_CH_STATUS_1, ch_stat1.regval);
+        return 0;
+}
+/*
+ * function to initialize audio
+ * registers and buffer confgiuration registers
+ * This function is called in the prepare callback
+ */
+static int had_init_audio_ctrl(struct snd_pcm_substream *substream,
+                               struct snd_intelhad *intelhaddata)
+{
+        union aud_cfg cfg_val = {.regval = 0};
+        union aud_buf_config buf_cfg = {.regval = 0};
+        u8 channels;
+        had_prog_status_reg(substream, intelhaddata);
+        buf_cfg.regx.audio_fifo_watermark = FIFO_THRESHOLD;
+        buf_cfg.regx.dma_fifo_watermark = DMA_FIFO_THRESHOLD;
+        buf_cfg.regx.aud_delay = 0;
+        had_write_register(intelhaddata, AUD_BUF_CONFIG, buf_cfg.regval);
+        channels = substream->runtime->channels;
+        cfg_val.regx.num_ch = channels - 2;
+        if (channels <= 2)
+                cfg_val.regx.layout = LAYOUT0;
+        else
+                cfg_val.regx.layout = LAYOUT1;
+        if (substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE)
+                cfg_val.regx.packet_mode = 1;
+        if (substream->runtime->format == SNDRV_PCM_FORMAT_S32_LE)
+                cfg_val.regx.left_align = 1;
+        cfg_val.regx.val_bit = 1;
+        /* fix up the DP bits */
+        if (intelhaddata->dp_output) {
+                cfg_val.regx.dp_modei = 1;
+                cfg_val.regx.set = 1;
+        }
+        had_write_register(intelhaddata, AUD_CONFIG, cfg_val.regval);
+        intelhaddata->aud_config = cfg_val;
+        return 0;
+}
+/*
+ * Compute derived values in channel_allocations[].
+ */
+static void init_channel_allocations(void)
+{
+        int i, j;
+        struct cea_channel_speaker_allocation *p;
+        for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) {
+                p = channel_allocations + i;
+                p->channels = 0;
+                p->spk_mask = 0;
+                for (j = 0; j < ARRAY_SIZE(p->speakers); j++)
+                        if (p->speakers[j]) {
+                                p->channels++;
+                                p->spk_mask |= p->speakers[j];
+                        }
+        }
+}
+/*
+ * The transformation takes two steps:
+ *
+ *      eld->spk_alloc => (eld_speaker_allocation_bits[]) => spk_mask
+ *            spk_mask => (channel_allocations[])         => ai->CA
+ *
+ * TODO: it could select the wrong CA from multiple candidates.
+ */
+static int had_channel_allocation(struct snd_intelhad *intelhaddata,
+                                  int channels)
+{
+        int i;
+        int ca = 0;
+        int spk_mask = 0;
+        /*
+         * CA defaults to 0 for basic stereo audio
+         */
+        if (channels <= 2)
+                return 0;
+        /*
+         * expand ELD's speaker allocation mask
+         *
+         * ELD tells the speaker mask in a compact(paired) form,
+         * expand ELD's notions to match the ones used by Audio InfoFrame.
+         */
+        for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) {
+                if (intelhaddata->eld[DRM_ELD_SPEAKER] & (1 << i))
+                        spk_mask |= eld_speaker_allocation_bits[i];
+        }
+        /* search for the first working match in the CA table */
+        for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) {
+                if (channels == channel_allocations[i].channels &&
+                (spk_mask & channel_allocations[i].spk_mask) ==
+                                channel_allocations[i].spk_mask) {
+                        ca = channel_allocations[i].ca_index;
+                        break;
+                }
+        }
+        dev_dbg(intelhaddata->dev, "select CA 0x%x for %d\n", ca, channels);
+        return ca;
+}
+/* from speaker bit mask to ALSA API channel position */
+static int spk_to_chmap(int spk)
+{
+        const struct channel_map_table *t = map_tables;
+        for (; t->map; t++) {
+                if (t->spk_mask == spk)
+                        return t->map;
+        }
+        return 0;
+}
+static void had_build_channel_allocation_map(struct snd_intelhad *intelhaddata)
+{
+        int i, c;
+        int spk_mask = 0;
+        struct snd_pcm_chmap_elem *chmap;
+        u8 eld_high, eld_high_mask = 0xF0;
+        u8 high_msb;
+        kfree(intelhaddata->chmap->chmap);
+        intelhaddata->chmap->chmap = NULL;
+        chmap = kzalloc(sizeof(*chmap), GFP_KERNEL);
+        if (!chmap)
+                return;
+        dev_dbg(intelhaddata->dev, "eld speaker = %x\n",
+                intelhaddata->eld[DRM_ELD_SPEAKER]);
+        /* WA: Fix the max channel supported to 8 */
+        /*
+         * Sink may support more than 8 channels, if eld_high has more than
+         * one bit set. SOC supports max 8 channels.
+         * Refer eld_speaker_allocation_bits, for sink speaker allocation
+         */
+        /* if 0x2F < eld < 0x4F fall back to 0x2f, else fall back to 0x4F */
+        eld_high = intelhaddata->eld[DRM_ELD_SPEAKER] & eld_high_mask;
+        if ((eld_high & (eld_high-1)) && (eld_high > 0x1F)) {
+                /* eld_high & (eld_high-1): if more than 1 bit set */
+                /* 0x1F: 7 channels */
+                for (i = 1; i < 4; i++) {
+                        high_msb = eld_high & (0x80 >> i);
+                        if (high_msb) {
+                                intelhaddata->eld[DRM_ELD_SPEAKER] &=
+                                        high_msb | 0xF;
+                                break;
+                        }
+                }
+        }
+        for (i = 0; i < ARRAY_SIZE(eld_speaker_allocation_bits); i++) {
+                if (intelhaddata->eld[DRM_ELD_SPEAKER] & (1 << i))
+                        spk_mask |= eld_speaker_allocation_bits[i];
+        }
+        for (i = 0; i < ARRAY_SIZE(channel_allocations); i++) {
+                if (spk_mask == channel_allocations[i].spk_mask) {
+                        for (c = 0; c < channel_allocations[i].channels; c++) {
+                                chmap->map[c] = spk_to_chmap(
+                                        channel_allocations[i].speakers[
+                                                (MAX_SPEAKERS - 1) - c]);
+                        }
+                        chmap->channels = channel_allocations[i].channels;
+                        intelhaddata->chmap->chmap = chmap;
+                        break;
+                }
+        }
+        if (i >= ARRAY_SIZE(channel_allocations))
+                kfree(chmap);
+}
+/*
+ * ALSA API channel-map control callbacks
+ */
+static int had_chmap_ctl_info(struct snd_kcontrol *kcontrol,
+                                struct snd_ctl_elem_info *uinfo)
+{
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+        uinfo->count = HAD_MAX_CHANNEL;
+        uinfo->value.integer.min = 0;
+        uinfo->value.integer.max = SNDRV_CHMAP_LAST;
+        return 0;
+}
+static int had_chmap_ctl_get(struct snd_kcontrol *kcontrol,
+                                struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol);
+        struct snd_intelhad *intelhaddata = info->private_data;
+        int i;
+        const struct snd_pcm_chmap_elem *chmap;
+        memset(ucontrol->value.integer.value, 0,
+               sizeof(long) * HAD_MAX_CHANNEL);
+        mutex_lock(&intelhaddata->mutex);
+        if (!intelhaddata->chmap->chmap) {
+                mutex_unlock(&intelhaddata->mutex);
+                return 0;
+        }
+        chmap = intelhaddata->chmap->chmap;
+        for (i = 0; i < chmap->channels; i++)
+                ucontrol->value.integer.value[i] = chmap->map[i];
+        mutex_unlock(&intelhaddata->mutex);
+        return 0;
+}
+static int had_register_chmap_ctls(struct snd_intelhad *intelhaddata,
+                                                struct snd_pcm *pcm)
+{
+        int err;
+        err = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+                        NULL, 0, (unsigned long)intelhaddata,
+                        &intelhaddata->chmap);
+        if (err < 0)
+                return err;
+        intelhaddata->chmap->private_data = intelhaddata;
+        intelhaddata->chmap->kctl->info = had_chmap_ctl_info;
+        intelhaddata->chmap->kctl->get = had_chmap_ctl_get;
+        intelhaddata->chmap->chmap = NULL;
+        return 0;
+}
+/*
+ * Initialize Data Island Packets registers
+ * This function is called in the prepare callback
+ */
+static void had_prog_dip(struct snd_pcm_substream *substream,
+                         struct snd_intelhad *intelhaddata)
+{
+        int i;
+        union aud_ctrl_st ctrl_state = {.regval = 0};
+        union aud_info_frame2 frame2 = {.regval = 0};
+        union aud_info_frame3 frame3 = {.regval = 0};
+        u8 checksum = 0;
+        u32 info_frame;
+        int channels;
+        int ca;
+        channels = substream->runtime->channels;
+        had_write_register(intelhaddata, AUD_CNTL_ST, ctrl_state.regval);
+        ca = had_channel_allocation(intelhaddata, channels);
+        if (intelhaddata->dp_output) {
+                info_frame = DP_INFO_FRAME_WORD1;
+                frame2.regval = (substream->runtime->channels - 1) | (ca << 24);
+        } else {
+                info_frame = HDMI_INFO_FRAME_WORD1;
+                frame2.regx.chnl_cnt = substream->runtime->channels - 1;
+                frame3.regx.chnl_alloc = ca;
+                /* Calculte the byte wide checksum for all valid DIP words */
+                for (i = 0; i < BYTES_PER_WORD; i++)
+                        checksum += (info_frame >> (i * 8)) & 0xff;
+                for (i = 0; i < BYTES_PER_WORD; i++)
+                        checksum += (frame2.regval >> (i * 8)) & 0xff;
+                for (i = 0; i < BYTES_PER_WORD; i++)
+                        checksum += (frame3.regval >> (i * 8)) & 0xff;
+                frame2.regx.chksum = -(checksum);
+        }
+        had_write_register(intelhaddata, AUD_HDMIW_INFOFR, info_frame);
+        had_write_register(intelhaddata, AUD_HDMIW_INFOFR, frame2.regval);
+        had_write_register(intelhaddata, AUD_HDMIW_INFOFR, frame3.regval);
+        /* program remaining DIP words with zero */
+        for (i = 0; i < HAD_MAX_DIP_WORDS-VALID_DIP_WORDS; i++)
+                had_write_register(intelhaddata, AUD_HDMIW_INFOFR, 0x0);
+        ctrl_state.regx.dip_freq = 1;
+        ctrl_state.regx.dip_en_sta = 1;
+        had_write_register(intelhaddata, AUD_CNTL_ST, ctrl_state.regval);
+}
+static int had_calculate_maud_value(u32 aud_samp_freq, u32 link_rate)
+{
+        u32 maud_val;
+        /* Select maud according to DP 1.2 spec */
+        if (link_rate == DP_2_7_GHZ) {
+                switch (aud_samp_freq) {
+                case AUD_SAMPLE_RATE_32:
+                        maud_val = AUD_SAMPLE_RATE_32_DP_2_7_MAUD_VAL;
+                        break;
+                case AUD_SAMPLE_RATE_44_1:
+                        maud_val = AUD_SAMPLE_RATE_44_1_DP_2_7_MAUD_VAL;
+                        break;
+                case AUD_SAMPLE_RATE_48:
+                        maud_val = AUD_SAMPLE_RATE_48_DP_2_7_MAUD_VAL;
+                        break;
+                case AUD_SAMPLE_RATE_88_2:
+                        maud_val = AUD_SAMPLE_RATE_88_2_DP_2_7_MAUD_VAL;
+                        break;
+                case AUD_SAMPLE_RATE_96:
+                        maud_val = AUD_SAMPLE_RATE_96_DP_2_7_MAUD_VAL;
+                        break;
+                case AUD_SAMPLE_RATE_176_4:
+                        maud_val = AUD_SAMPLE_RATE_176_4_DP_2_7_MAUD_VAL;
+                        break;
+                case HAD_MAX_RATE:
+                        maud_val = HAD_MAX_RATE_DP_2_7_MAUD_VAL;
+                        break;
+                default:
+                        maud_val = -EINVAL;
+                        break;
+                }
+        } else if (link_rate == DP_1_62_GHZ) {
+                switch (aud_samp_freq) {
+                case AUD_SAMPLE_RATE_32:
+                        maud_val = AUD_SAMPLE_RATE_32_DP_1_62_MAUD_VAL;
+                        break;
+                case AUD_SAMPLE_RATE_44_1:
+                        maud_val = AUD_SAMPLE_RATE_44_1_DP_1_62_MAUD_VAL;
+                        break;
+                case AUD_SAMPLE_RATE_48:
+                        maud_val = AUD_SAMPLE_RATE_48_DP_1_62_MAUD_VAL;
+                        break;
+                case AUD_SAMPLE_RATE_88_2:
+                        maud_val = AUD_SAMPLE_RATE_88_2_DP_1_62_MAUD_VAL;
+                        break;
+                case AUD_SAMPLE_RATE_96:
+                        maud_val = AUD_SAMPLE_RATE_96_DP_1_62_MAUD_VAL;
+                        break;
+                case AUD_SAMPLE_RATE_176_4:
+                        maud_val = AUD_SAMPLE_RATE_176_4_DP_1_62_MAUD_VAL;
+                        break;
+                case HAD_MAX_RATE:
+                        maud_val = HAD_MAX_RATE_DP_1_62_MAUD_VAL;
+                        break;
+                default:
+                        maud_val = -EINVAL;
+                        break;
+                }
+        } else
+                maud_val = -EINVAL;
+        return maud_val;
+}
+/*
+ * Program HDMI audio CTS value
+ *
+ * @aud_samp_freq: sampling frequency of audio data
+ * @tmds: sampling frequency of the display data
+ * @link_rate: DP link rate
+ * @n_param: N value, depends on aud_samp_freq
+ * @intelhaddata: substream private data
+ *
+ * Program CTS register based on the audio and display sampling frequency
+ */
+static void had_prog_cts(u32 aud_samp_freq, u32 tmds, u32 link_rate,
+                         u32 n_param, struct snd_intelhad *intelhaddata)
+{
+        u32 cts_val;
+        u64 dividend, divisor;
+        if (intelhaddata->dp_output) {
+                /* Substitute cts_val with Maud according to DP 1.2 spec*/
+                cts_val = had_calculate_maud_value(aud_samp_freq, link_rate);
+        } else {
+                /* Calculate CTS according to HDMI 1.3a spec*/
+                dividend = (u64)tmds * n_param*1000;
+                divisor = 128 * aud_samp_freq;
+                cts_val = div64_u64(dividend, divisor);
+        }
+        dev_dbg(intelhaddata->dev, "TMDS value=%d, N value=%d, CTS Value=%d\n",
+                 tmds, n_param, cts_val);
+        had_write_register(intelhaddata, AUD_HDMI_CTS, (BIT(24) | cts_val));
+}
+static int had_calculate_n_value(u32 aud_samp_freq)
+{
+        int n_val;
+        /* Select N according to HDMI 1.3a spec*/
+        switch (aud_samp_freq) {
+        case AUD_SAMPLE_RATE_32:
+                n_val = 4096;
+                break;
+        case AUD_SAMPLE_RATE_44_1:
+                n_val = 6272;
+                break;
+        case AUD_SAMPLE_RATE_48:
+                n_val = 6144;
+                break;
+        case AUD_SAMPLE_RATE_88_2:
+                n_val = 12544;
+                break;
+        case AUD_SAMPLE_RATE_96:
+                n_val = 12288;
+                break;
+        case AUD_SAMPLE_RATE_176_4:
+                n_val = 25088;
+                break;
+        case HAD_MAX_RATE:
+                n_val = 24576;
+                break;
+        default:
+                n_val = -EINVAL;
+                break;
+        }
+        return n_val;
+}
+/*
+ * Program HDMI audio N value
+ *
+ * @aud_samp_freq: sampling frequency of audio data
+ * @n_param: N value, depends on aud_samp_freq
+ * @intelhaddata: substream private data
+ *
+ * This function is called in the prepare callback.
+ * It programs based on the audio and display sampling frequency
+ */
+static int had_prog_n(u32 aud_samp_freq, u32 *n_param,
+                      struct snd_intelhad *intelhaddata)
+{
+        int n_val;
+        if (intelhaddata->dp_output) {
+                /*
+                 * According to DP specs, Maud and Naud values hold
+                 * a relationship, which is stated as:
+                 * Maud/Naud = 512 * fs / f_LS_Clk
+                 * where, fs is the sampling frequency of the audio stream
+                 * and Naud is 32768 for Async clock.
+                 */
+                n_val = DP_NAUD_VAL;
+        } else
+                n_val = had_calculate_n_value(aud_samp_freq);
+        if (n_val < 0)
+                return n_val;
+        had_write_register(intelhaddata, AUD_N_ENABLE, (BIT(24) | n_val));
+        *n_param = n_val;
+        return 0;
+}
+/*
+ * PCM ring buffer handling
+ *
+ * The hardware provides a ring buffer with the fixed 4 buffer descriptors
+ * (BDs).  The driver maps these 4 BDs onto the PCM ring buffer.  The mapping
+ * moves at each period elapsed.  The below illustrates how it works:
+ *
+ * At time=0
+ *  PCM | 0 | 1 | 2 | 3 | 4 | 5 | .... |n-1|
+ *  BD  | 0 | 1 | 2 | 3 |
+ *
+ * At time=1 (period elapsed)
+ *  PCM | 0 | 1 | 2 | 3 | 4 | 5 | .... |n-1|
+ *  BD      | 1 | 2 | 3 | 0 |
+ *
+ * At time=2 (second period elapsed)
+ *  PCM | 0 | 1 | 2 | 3 | 4 | 5 | .... |n-1|
+ *  BD          | 2 | 3 | 0 | 1 |
+ *
+ * The bd_head field points to the index of the BD to be read.  It's also the
+ * position to be filled at next.  The pcm_head and the pcm_filled fields
+ * point to the indices of the current position and of the next position to
+ * be filled, respectively.  For PCM buffer there are both _head and _filled
+ * because they may be difference when nperiods > 4.  For example, in the
+ * example above at t=1, bd_head=1 and pcm_head=1 while pcm_filled=5:
+ *
+ * pcm_head (=1) --v               v-- pcm_filled (=5)
+ *       PCM | 0 | 1 | 2 | 3 | 4 | 5 | .... |n-1|
+ *       BD      | 1 | 2 | 3 | 0 |
+ *  bd_head (=1) --^               ^-- next to fill (= bd_head)
+ *
+ * For nperiods < 4, the remaining BDs out of 4 are marked as invalid, so that
+ * the hardware skips those BDs in the loop.
+ *
+ * An exceptional setup is the case with nperiods=1.  Since we have to update
+ * BDs after finishing one BD processing, we'd need at least two BDs, where
+ * both BDs point to the same content, the same address, the same size of the
+ * whole PCM buffer.
+ */
+#define AUD_BUF_ADDR(x)         (AUD_BUF_A_ADDR + (x) * HAD_REG_WIDTH)
+#define AUD_BUF_LEN(x)          (AUD_BUF_A_LENGTH + (x) * HAD_REG_WIDTH)
+/* Set up a buffer descriptor at the "filled" position */
+static void had_prog_bd(struct snd_pcm_substream *substream,
+                        struct snd_intelhad *intelhaddata)
+{
+        int idx = intelhaddata->bd_head;
+        int ofs = intelhaddata->pcmbuf_filled * intelhaddata->period_bytes;
+        u32 addr = substream->runtime->dma_addr + ofs;
+        addr |= AUD_BUF_VALID;
+        if (!substream->runtime->no_period_wakeup)
+                addr |= AUD_BUF_INTR_EN;
+        had_write_register(intelhaddata, AUD_BUF_ADDR(idx), addr);
+        had_write_register(intelhaddata, AUD_BUF_LEN(idx),
+                           intelhaddata->period_bytes);
+        /* advance the indices to the next */
+        intelhaddata->bd_head++;
+        intelhaddata->bd_head %= intelhaddata->num_bds;
+        intelhaddata->pcmbuf_filled++;
+        intelhaddata->pcmbuf_filled %= substream->runtime->periods;
+}
+/* invalidate a buffer descriptor with the given index */
+static void had_invalidate_bd(struct snd_intelhad *intelhaddata,
+                              int idx)
+{
+        had_write_register(intelhaddata, AUD_BUF_ADDR(idx), 0);
+        had_write_register(intelhaddata, AUD_BUF_LEN(idx), 0);
+}
+/* Initial programming of ring buffer */
+static void had_init_ringbuf(struct snd_pcm_substream *substream,
+                             struct snd_intelhad *intelhaddata)
+{
+        struct snd_pcm_runtime *runtime = substream->runtime;
+        int i, num_periods;
+        num_periods = runtime->periods;
+        intelhaddata->num_bds = min(num_periods, HAD_NUM_OF_RING_BUFS);
+        /* set the minimum 2 BDs for num_periods=1 */
+        intelhaddata->num_bds = max(intelhaddata->num_bds, 2U);
+        intelhaddata->period_bytes =
+                frames_to_bytes(runtime, runtime->period_size);
+        WARN_ON(intelhaddata->period_bytes & 0x3f);
+        intelhaddata->bd_head = 0;
+        intelhaddata->pcmbuf_head = 0;
+        intelhaddata->pcmbuf_filled = 0;
+        for (i = 0; i < HAD_NUM_OF_RING_BUFS; i++) {
+                if (i < intelhaddata->num_bds)
+                        had_prog_bd(substream, intelhaddata);
+                else /* invalidate the rest */
+                        had_invalidate_bd(intelhaddata, i);
+        }
+        intelhaddata->bd_head = 0; /* reset at head again before starting */
+}
+/* process a bd, advance to the next */
+static void had_advance_ringbuf(struct snd_pcm_substream *substream,
+                                struct snd_intelhad *intelhaddata)
+{
+        int num_periods = substream->runtime->periods;
+        /* reprogram the next buffer */
+        had_prog_bd(substream, intelhaddata);
+        /* proceed to next */
+        intelhaddata->pcmbuf_head++;
+        intelhaddata->pcmbuf_head %= num_periods;
+}
+/* process the current BD(s);
+ * returns the current PCM buffer byte position, or -EPIPE for underrun.
+ */
+static int had_process_ringbuf(struct snd_pcm_substream *substream,
+                               struct snd_intelhad *intelhaddata)
+{
+        int len, processed;
+        unsigned long flags;
+        processed = 0;
+        spin_lock_irqsave(&intelhaddata->had_spinlock, flags);
+        for (;;) {
+                /* get the remaining bytes on the buffer */
+                had_read_register(intelhaddata,
+                                  AUD_BUF_LEN(intelhaddata->bd_head),
+                                  &len);
+                if (len < 0 || len > intelhaddata->period_bytes) {
+                        dev_dbg(intelhaddata->dev, "Invalid buf length %d\n",
+                                len);
+                        len = -EPIPE;
+                        goto out;
+                }
+                if (len > 0) /* OK, this is the current buffer */
+                        break;
+                /* len=0 => already empty, check the next buffer */
+                if (++processed >= intelhaddata->num_bds) {
+                        len = -EPIPE; /* all empty? - report underrun */
+                        goto out;
+                }
+                had_advance_ringbuf(substream, intelhaddata);
+        }
+        len = intelhaddata->period_bytes - len;
+        len += intelhaddata->period_bytes * intelhaddata->pcmbuf_head;
+ out:
+        spin_unlock_irqrestore(&intelhaddata->had_spinlock, flags);
+        return len;
+}
+/* called from irq handler */
+static void had_process_buffer_done(struct snd_intelhad *intelhaddata)
+{
+        struct snd_pcm_substream *substream;
+        substream = had_substream_get(intelhaddata);
+        if (!substream)
+                return; /* no stream? - bail out */
+        if (!intelhaddata->connected) {
+                snd_pcm_stop_xrun(substream);
+                goto out; /* disconnected? - bail out */
+        }
+        /* process or stop the stream */
+        if (had_process_ringbuf(substream, intelhaddata) < 0)
+                snd_pcm_stop_xrun(substream);
+        else
+                snd_pcm_period_elapsed(substream);
+ out:
+        had_substream_put(intelhaddata);
+}
+/*
+ * The interrupt status 'sticky' bits might not be cleared by
+ * setting '1' to that bit once...
+ */
+static void wait_clear_underrun_bit(struct snd_intelhad *intelhaddata)
+{
+        int i;
+        u32 val;
+        for (i = 0; i < 100; i++) {
+                /* clear bit30, 31 AUD_HDMI_STATUS */
+                had_read_register(intelhaddata, AUD_HDMI_STATUS, &val);
+                if (!(val & AUD_HDMI_STATUS_MASK_UNDERRUN))
+                        return;
+                udelay(100);
+                cond_resched();
+                had_write_register(intelhaddata, AUD_HDMI_STATUS, val);
+        }
+        dev_err(intelhaddata->dev, "Unable to clear UNDERRUN bits\n");
+}
+/* Perform some reset procedure but only when need_reset is set;
+ * this is called from prepare or hw_free callbacks once after trigger STOP
+ * or underrun has been processed in order to settle down the h/w state.
+ */
+static void had_do_reset(struct snd_intelhad *intelhaddata)
+{
+        if (!intelhaddata->need_reset || !intelhaddata->connected)
+                return;
+        /* Reset buffer pointers */
+        had_reset_audio(intelhaddata);
+        wait_clear_underrun_bit(intelhaddata);
+        intelhaddata->need_reset = false;
+}
+/* called from irq handler */
+static void had_process_buffer_underrun(struct snd_intelhad *intelhaddata)
+{
+        struct snd_pcm_substream *substream;
+        /* Report UNDERRUN error to above layers */
+        substream = had_substream_get(intelhaddata);
+        if (substream) {
+                snd_pcm_stop_xrun(substream);
+                had_substream_put(intelhaddata);
+        }
+        intelhaddata->need_reset = true;
+}
+/*
+ * ALSA PCM open callback
+ */
+static int had_pcm_open(struct snd_pcm_substream *substream)
+{
+        struct snd_intelhad *intelhaddata;
+        struct snd_pcm_runtime *runtime;
+        int retval;
+        intelhaddata = snd_pcm_substream_chip(substream);
+        runtime = substream->runtime;
+        pm_runtime_get_sync(intelhaddata->dev);
+        /* set the runtime hw parameter with local snd_pcm_hardware struct */
+        runtime->hw = had_pcm_hardware;
+        retval = snd_pcm_hw_constraint_integer(runtime,
+                         SNDRV_PCM_HW_PARAM_PERIODS);
+        if (retval < 0)
+                goto error;
+        /* Make sure, that the period size is always aligned
+         * 64byte boundary
+         */
+        retval = snd_pcm_hw_constraint_step(substream->runtime, 0,
+                        SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 64);
+        if (retval < 0)
+                goto error;
+        retval = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+        if (retval < 0)
+                goto error;
+        /* expose PCM substream */
+        spin_lock_irq(&intelhaddata->had_spinlock);
+        intelhaddata->stream_info.substream = substream;
+        intelhaddata->stream_info.substream_refcount++;
+        spin_unlock_irq(&intelhaddata->had_spinlock);
+        return retval;
+ error:
+        pm_runtime_put(intelhaddata->dev);
+        return retval;
+}
+/*
+ * ALSA PCM close callback
+ */
+static int had_pcm_close(struct snd_pcm_substream *substream)
+{
+        struct snd_intelhad *intelhaddata;
+        intelhaddata = snd_pcm_substream_chip(substream);
+        /* unreference and sync with the pending PCM accesses */
+        spin_lock_irq(&intelhaddata->had_spinlock);
+        intelhaddata->stream_info.substream = NULL;
+        intelhaddata->stream_info.substream_refcount--;
+        while (intelhaddata->stream_info.substream_refcount > 0) {
+                spin_unlock_irq(&intelhaddata->had_spinlock);
+                cpu_relax();
+                spin_lock_irq(&intelhaddata->had_spinlock);
+        }
+        spin_unlock_irq(&intelhaddata->had_spinlock);
+        pm_runtime_put(intelhaddata->dev);
+        return 0;
+}
+/*
+ * ALSA PCM hw_params callback
+ */
+static int had_pcm_hw_params(struct snd_pcm_substream *substream,
+                             struct snd_pcm_hw_params *hw_params)
+{
+        struct snd_intelhad *intelhaddata;
+        unsigned long addr;
+        int pages, buf_size, retval;
+        intelhaddata = snd_pcm_substream_chip(substream);
+        buf_size = params_buffer_bytes(hw_params);
+        retval = snd_pcm_lib_malloc_pages(substream, buf_size);
+        if (retval < 0)
+                return retval;
+        dev_dbg(intelhaddata->dev, "%s:allocated memory = %d\n",
+                __func__, buf_size);
+        /* mark the pages as uncached region */
+        addr = (unsigned long) substream->runtime->dma_area;
+        pages = (substream->runtime->dma_bytes + PAGE_SIZE - 1) / PAGE_SIZE;
+        retval = set_memory_uc(addr, pages);
+        if (retval) {
+                dev_err(intelhaddata->dev, "set_memory_uc failed.Error:%d\n",
+                        retval);
+                return retval;
+        }
+        memset(substream->runtime->dma_area, 0, buf_size);
+        return retval;
+}
+/*
+ * ALSA PCM hw_free callback
+ */
+static int had_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+        struct snd_intelhad *intelhaddata;
+        unsigned long addr;
+        u32 pages;
+        intelhaddata = snd_pcm_substream_chip(substream);
+        had_do_reset(intelhaddata);
+        /* mark back the pages as cached/writeback region before the free */
+        if (substream->runtime->dma_area != NULL) {
+                addr = (unsigned long) substream->runtime->dma_area;
+                pages = (substream->runtime->dma_bytes + PAGE_SIZE - 1) /
+                                                                PAGE_SIZE;
+                set_memory_wb(addr, pages);
+                return snd_pcm_lib_free_pages(substream);
+        }
+        return 0;
+}
+/*
+ * ALSA PCM trigger callback
+ */
+static int had_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+        int retval = 0;
+        struct snd_intelhad *intelhaddata;
+        intelhaddata = snd_pcm_substream_chip(substream);
+        spin_lock(&intelhaddata->had_spinlock);
+        switch (cmd) {
+        case SNDRV_PCM_TRIGGER_START:
+        case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+        case SNDRV_PCM_TRIGGER_RESUME:
+                /* Enable Audio */
+                had_ack_irqs(intelhaddata); /* FIXME: do we need this? */
+                had_enable_audio(intelhaddata, true);
+                break;
+        case SNDRV_PCM_TRIGGER_STOP:
+        case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+                /* Disable Audio */
+                had_enable_audio(intelhaddata, false);
+                intelhaddata->need_reset = true;
+                break;
+        default:
+                retval = -EINVAL;
+        }
+        spin_unlock(&intelhaddata->had_spinlock);
+        return retval;
+}
+/*
+ * ALSA PCM prepare callback
+ */
+static int had_pcm_prepare(struct snd_pcm_substream *substream)
+{
+        int retval;
+        u32 disp_samp_freq, n_param;
+        u32 link_rate = 0;
+        struct snd_intelhad *intelhaddata;
+        struct snd_pcm_runtime *runtime;
+        intelhaddata = snd_pcm_substream_chip(substream);
+        runtime = substream->runtime;
+        dev_dbg(intelhaddata->dev, "period_size=%d\n",
+                (int)frames_to_bytes(runtime, runtime->period_size));
+        dev_dbg(intelhaddata->dev, "periods=%d\n", runtime->periods);
+        dev_dbg(intelhaddata->dev, "buffer_size=%d\n",
+                (int)snd_pcm_lib_buffer_bytes(substream));
+        dev_dbg(intelhaddata->dev, "rate=%d\n", runtime->rate);
+        dev_dbg(intelhaddata->dev, "channels=%d\n", runtime->channels);
+        had_do_reset(intelhaddata);
+        /* Get N value in KHz */
+        disp_samp_freq = intelhaddata->tmds_clock_speed;
+        retval = had_prog_n(substream->runtime->rate, &n_param, intelhaddata);
+        if (retval) {
+                dev_err(intelhaddata->dev,
+                        "programming N value failed %#x\n", retval);
+                goto prep_end;
+        }
+        if (intelhaddata->dp_output)
+                link_rate = intelhaddata->link_rate;
+        had_prog_cts(substream->runtime->rate, disp_samp_freq, link_rate,
+                     n_param, intelhaddata);
+        had_prog_dip(substream, intelhaddata);
+        retval = had_init_audio_ctrl(substream, intelhaddata);
+        /* Prog buffer address */
+        had_init_ringbuf(substream, intelhaddata);
+        /*
+         * Program channel mapping in following order:
+         * FL, FR, C, LFE, RL, RR
+         */
+        had_write_register(intelhaddata, AUD_BUF_CH_SWAP, SWAP_LFE_CENTER);
+prep_end:
+        return retval;
+}
+/*
+ * ALSA PCM pointer callback
+ */
+static snd_pcm_uframes_t had_pcm_pointer(struct snd_pcm_substream *substream)
+{
+        struct snd_intelhad *intelhaddata;
+        int len;
+        intelhaddata = snd_pcm_substream_chip(substream);
+        if (!intelhaddata->connected)
+                return SNDRV_PCM_POS_XRUN;
+        len = had_process_ringbuf(substream, intelhaddata);
+        if (len < 0)
+                return SNDRV_PCM_POS_XRUN;
+        len = bytes_to_frames(substream->runtime, len);
+        /* wrapping may happen when periods=1 */
+        len %= substream->runtime->buffer_size;
+        return len;
+}
+/*
+ * ALSA PCM mmap callback
+ */
+static int had_pcm_mmap(struct snd_pcm_substream *substream,
+                        struct vm_area_struct *vma)
+{
+        vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+        return remap_pfn_range(vma, vma->vm_start,
+                        substream->dma_buffer.addr >> PAGE_SHIFT,
+                        vma->vm_end - vma->vm_start, vma->vm_page_prot);
+}
+/*
+ * ALSA PCM ops
+ */
+static const struct snd_pcm_ops had_pcm_ops = {
+        .open =         had_pcm_open,
+        .close =        had_pcm_close,
+        .ioctl =        snd_pcm_lib_ioctl,
+        .hw_params =    had_pcm_hw_params,
+        .hw_free =      had_pcm_hw_free,
+        .prepare =      had_pcm_prepare,
+        .trigger =      had_pcm_trigger,
+        .pointer =      had_pcm_pointer,
+        .mmap =         had_pcm_mmap,
+};
+/* process mode change of the running stream; called in mutex */
+static int had_process_mode_change(struct snd_intelhad *intelhaddata)
+{
+        struct snd_pcm_substream *substream;
+        int retval = 0;
+        u32 disp_samp_freq, n_param;
+        u32 link_rate = 0;
+        substream = had_substream_get(intelhaddata);
+        if (!substream)
+                return 0;
+        /* Disable Audio */
+        had_enable_audio(intelhaddata, false);
+        /* Update CTS value */
+        disp_samp_freq = intelhaddata->tmds_clock_speed;
+        retval = had_prog_n(substream->runtime->rate, &n_param, intelhaddata);
+        if (retval) {
+                dev_err(intelhaddata->dev,
+                        "programming N value failed %#x\n", retval);
+                goto out;
+        }
+        if (intelhaddata->dp_output)
+                link_rate = intelhaddata->link_rate;
+        had_prog_cts(substream->runtime->rate, disp_samp_freq, link_rate,
+                     n_param, intelhaddata);
+        /* Enable Audio */
+        had_enable_audio(intelhaddata, true);
+out:
+        had_substream_put(intelhaddata);
+        return retval;
+}
+/* process hot plug, called from wq with mutex locked */
+static void had_process_hot_plug(struct snd_intelhad *intelhaddata)
+{
+        struct snd_pcm_substream *substream;
+        spin_lock_irq(&intelhaddata->had_spinlock);
+        if (intelhaddata->connected) {
+                dev_dbg(intelhaddata->dev, "Device already connected\n");
+                spin_unlock_irq(&intelhaddata->had_spinlock);
+                return;
+        }
+        intelhaddata->connected = true;
+        dev_dbg(intelhaddata->dev,
+                "%s @ %d:DEBUG PLUG/UNPLUG : HAD_DRV_CONNECTED\n",
+                        __func__, __LINE__);
+        spin_unlock_irq(&intelhaddata->had_spinlock);
+        had_build_channel_allocation_map(intelhaddata);
+        /* Report to above ALSA layer */
+        substream = had_substream_get(intelhaddata);
+        if (substream) {
+                snd_pcm_stop_xrun(substream);
+                had_substream_put(intelhaddata);
+        }
+        snd_jack_report(intelhaddata->jack, SND_JACK_AVOUT);
+}
+/* process hot unplug, called from wq with mutex locked */
+static void had_process_hot_unplug(struct snd_intelhad *intelhaddata)
+{
+        struct snd_pcm_substream *substream;
+        spin_lock_irq(&intelhaddata->had_spinlock);
+        if (!intelhaddata->connected) {
+                dev_dbg(intelhaddata->dev, "Device already disconnected\n");
+                spin_unlock_irq(&intelhaddata->had_spinlock);
+                return;
+        }
+        /* Disable Audio */
+        had_enable_audio(intelhaddata, false);
+        intelhaddata->connected = false;
+        dev_dbg(intelhaddata->dev,
+                "%s @ %d:DEBUG PLUG/UNPLUG : HAD_DRV_DISCONNECTED\n",
+                        __func__, __LINE__);
+        spin_unlock_irq(&intelhaddata->had_spinlock);
+        kfree(intelhaddata->chmap->chmap);
+        intelhaddata->chmap->chmap = NULL;
+        /* Report to above ALSA layer */
+        substream = had_substream_get(intelhaddata);
+        if (substream) {
+                snd_pcm_stop_xrun(substream);
+                had_substream_put(intelhaddata);
+        }
+        snd_jack_report(intelhaddata->jack, 0);
+}
+/*
+ * ALSA iec958 and ELD controls
+ */
+static int had_iec958_info(struct snd_kcontrol *kcontrol,
+                                struct snd_ctl_elem_info *uinfo)
+{
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+        uinfo->count = 1;
+        return 0;
+}
+static int had_iec958_get(struct snd_kcontrol *kcontrol,
+                                struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_intelhad *intelhaddata = snd_kcontrol_chip(kcontrol);
+        mutex_lock(&intelhaddata->mutex);
+        ucontrol->value.iec958.status[0] = (intelhaddata->aes_bits >> 0) & 0xff;
+        ucontrol->value.iec958.status[1] = (intelhaddata->aes_bits >> 8) & 0xff;
+        ucontrol->value.iec958.status[2] =
+                                        (intelhaddata->aes_bits >> 16) & 0xff;
+        ucontrol->value.iec958.status[3] =
+                                        (intelhaddata->aes_bits >> 24) & 0xff;
+        mutex_unlock(&intelhaddata->mutex);
+        return 0;
+}
+static int had_iec958_mask_get(struct snd_kcontrol *kcontrol,
+                                struct snd_ctl_elem_value *ucontrol)
+{
+        ucontrol->value.iec958.status[0] = 0xff;
+        ucontrol->value.iec958.status[1] = 0xff;
+        ucontrol->value.iec958.status[2] = 0xff;
+        ucontrol->value.iec958.status[3] = 0xff;
+        return 0;
+}
+static int had_iec958_put(struct snd_kcontrol *kcontrol,
+                                struct snd_ctl_elem_value *ucontrol)
+{
+        unsigned int val;
+        struct snd_intelhad *intelhaddata = snd_kcontrol_chip(kcontrol);
+        int changed = 0;
+        val = (ucontrol->value.iec958.status[0] << 0) |
+                (ucontrol->value.iec958.status[1] << 8) |
+                (ucontrol->value.iec958.status[2] << 16) |
+                (ucontrol->value.iec958.status[3] << 24);
+        mutex_lock(&intelhaddata->mutex);
+        if (intelhaddata->aes_bits != val) {
+                intelhaddata->aes_bits = val;
+                changed = 1;
+        }
+        mutex_unlock(&intelhaddata->mutex);
+        return changed;
+}
+static int had_ctl_eld_info(struct snd_kcontrol *kcontrol,
+                            struct snd_ctl_elem_info *uinfo)
+{
+        uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES;
+        uinfo->count = HDMI_MAX_ELD_BYTES;
+        return 0;
+}
+static int had_ctl_eld_get(struct snd_kcontrol *kcontrol,
+                           struct snd_ctl_elem_value *ucontrol)
+{
+        struct snd_intelhad *intelhaddata = snd_kcontrol_chip(kcontrol);
+        mutex_lock(&intelhaddata->mutex);
+        memcpy(ucontrol->value.bytes.data, intelhaddata->eld,
+               HDMI_MAX_ELD_BYTES);
+        mutex_unlock(&intelhaddata->mutex);
+        return 0;
+}
+static const struct snd_kcontrol_new had_controls[] = {
+        {
+                .access = SNDRV_CTL_ELEM_ACCESS_READ,
+                .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+                .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, MASK),
+                .info = had_iec958_info, /* shared */
+                .get = had_iec958_mask_get,
+        },
+        {
+                .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+                .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT),
+                .info = had_iec958_info,
+                .get = had_iec958_get,
+                .put = had_iec958_put,
+        },
+        {
+                .access = (SNDRV_CTL_ELEM_ACCESS_READ |
+                           SNDRV_CTL_ELEM_ACCESS_VOLATILE),
+                .iface = SNDRV_CTL_ELEM_IFACE_PCM,
+                .name = "ELD",
+                .info = had_ctl_eld_info,
+                .get = had_ctl_eld_get,
+        },
+};
+/*
+ * audio interrupt handler
+ */
+static irqreturn_t display_pipe_interrupt_handler(int irq, void *dev_id)
+{
+        struct snd_intelhad *ctx = dev_id;
+        u32 audio_stat;
+        /* use raw register access to ack IRQs even while disconnected */
+        audio_stat = had_read_register_raw(ctx, AUD_HDMI_STATUS);
+        if (audio_stat & HDMI_AUDIO_UNDERRUN) {
+                had_write_register_raw(ctx, AUD_HDMI_STATUS,
+                                       HDMI_AUDIO_UNDERRUN);
+                had_process_buffer_underrun(ctx);
+        }
+        if (audio_stat & HDMI_AUDIO_BUFFER_DONE) {
+                had_write_register_raw(ctx, AUD_HDMI_STATUS,
+                                       HDMI_AUDIO_BUFFER_DONE);
+                had_process_buffer_done(ctx);
+        }
+        return IRQ_HANDLED;
+}
+/*
+ * monitor plug/unplug notification from i915; just kick off the work
+ */
+static void notify_audio_lpe(struct platform_device *pdev)
+{
+        struct snd_intelhad *ctx = platform_get_drvdata(pdev);
+        schedule_work(&ctx->hdmi_audio_wq);
+}
+/* the work to handle monitor hot plug/unplug */
+static void had_audio_wq(struct work_struct *work)
+{
+        struct snd_intelhad *ctx =
+                container_of(work, struct snd_intelhad, hdmi_audio_wq);
+        struct intel_hdmi_lpe_audio_pdata *pdata = ctx->dev->platform_data;
+        pm_runtime_get_sync(ctx->dev);
+        mutex_lock(&ctx->mutex);
+        if (!pdata->hdmi_connected) {
+                dev_dbg(ctx->dev, "%s: Event: HAD_NOTIFY_HOT_UNPLUG\n",
+                        __func__);
+                memset(ctx->eld, 0, sizeof(ctx->eld)); /* clear the old ELD */
+                had_process_hot_unplug(ctx);
+        } else {
+                struct intel_hdmi_lpe_audio_eld *eld = &pdata->eld;
+                dev_dbg(ctx->dev, "%s: HAD_NOTIFY_ELD : port = %d, tmds = %d\n",
+                        __func__, eld->port_id, pdata->tmds_clock_speed);
+                switch (eld->pipe_id) {
+                case 0:
+                        ctx->had_config_offset = AUDIO_HDMI_CONFIG_A;
+                        break;
+                case 1:
+                        ctx->had_config_offset = AUDIO_HDMI_CONFIG_B;
+                        break;
+                case 2:
+                        ctx->had_config_offset = AUDIO_HDMI_CONFIG_C;
+                        break;
+                default:
+                        dev_dbg(ctx->dev, "Invalid pipe %d\n",
+                                eld->pipe_id);
+                        break;
+                }
+                memcpy(ctx->eld, eld->eld_data, sizeof(ctx->eld));
+                ctx->dp_output = pdata->dp_output;
+                ctx->tmds_clock_speed = pdata->tmds_clock_speed;
+                ctx->link_rate = pdata->link_rate;
+                had_process_hot_plug(ctx);
+                /* Process mode change if stream is active */
+                had_process_mode_change(ctx);
+        }
+        mutex_unlock(&ctx->mutex);
+        pm_runtime_put(ctx->dev);
+}
+/*
+ * Jack interface
+ */
+static int had_create_jack(struct snd_intelhad *ctx)
+{
+        int err;
+        err = snd_jack_new(ctx->card, "HDMI/DP", SND_JACK_AVOUT, &ctx->jack,
+                           true, false);
+        if (err < 0)
+                return err;
+        ctx->jack->private_data = ctx;
+        return 0;
+}
+/*
+ * PM callbacks
+ */
+static int hdmi_lpe_audio_runtime_suspend(struct device *dev)
+{
+        struct snd_intelhad *ctx = dev_get_drvdata(dev);
+        struct snd_pcm_substream *substream;
+        substream = had_substream_get(ctx);
+        if (substream) {
+                snd_pcm_suspend(substream);
+                had_substream_put(ctx);
+        }
+        return 0;
+}
+static int __maybe_unused hdmi_lpe_audio_suspend(struct device *dev)
+{
+        struct snd_intelhad *ctx = dev_get_drvdata(dev);
+        int err;
+        err = hdmi_lpe_audio_runtime_suspend(dev);
+        if (!err)
+                snd_power_change_state(ctx->card, SNDRV_CTL_POWER_D3hot);
+        return err;
+}
+static int __maybe_unused hdmi_lpe_audio_resume(struct device *dev)
+{
+        struct snd_intelhad *ctx = dev_get_drvdata(dev);
+        snd_power_change_state(ctx->card, SNDRV_CTL_POWER_D0);
+        return 0;
+}
+/* release resources */
+static void hdmi_lpe_audio_free(struct snd_card *card)
+{
+        struct snd_intelhad *ctx = card->private_data;
+        cancel_work_sync(&ctx->hdmi_audio_wq);
+        if (ctx->mmio_start)
+                iounmap(ctx->mmio_start);
+        if (ctx->irq >= 0)
+                free_irq(ctx->irq, ctx);
+}
+/*
+ * hdmi_lpe_audio_probe - start bridge with i915
+ *
+ * This function is called when the i915 driver creates the
+ * hdmi-lpe-audio platform device.
+ */
+static int hdmi_lpe_audio_probe(struct platform_device *pdev)
+{
+        struct snd_card *card;
+        struct snd_intelhad *ctx;
+        struct snd_pcm *pcm;
+        struct intel_hdmi_lpe_audio_pdata *pdata;
+        int irq;
+        struct resource *res_mmio;
+        int i, ret;
+        pdata = pdev->dev.platform_data;
+        if (!pdata) {
+                dev_err(&pdev->dev, "%s: quit: pdata not allocated by i915!!\n", __func__);
+                return -EINVAL;
+        }
+        /* get resources */
+        irq = platform_get_irq(pdev, 0);
+        if (irq < 0) {
+                dev_err(&pdev->dev, "Could not get irq resource\n");
+                return -ENODEV;
+        }
+        res_mmio = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+        if (!res_mmio) {
+                dev_err(&pdev->dev, "Could not get IO_MEM resources\n");
+                return -ENXIO;
+        }
+        /* create a card instance with ALSA framework */
+        ret = snd_card_new(&pdev->dev, hdmi_card_index, hdmi_card_id,
+                           THIS_MODULE, sizeof(*ctx), &card);
+        if (ret)
+                return ret;
+        ctx = card->private_data;
+        spin_lock_init(&ctx->had_spinlock);
+        mutex_init(&ctx->mutex);
+        ctx->connected = false;
+        ctx->dev = &pdev->dev;
+        ctx->card = card;
+        ctx->aes_bits = SNDRV_PCM_DEFAULT_CON_SPDIF;
+        strcpy(card->driver, INTEL_HAD);
+        strcpy(card->shortname, "Intel HDMI/DP LPE Audio");
+        strcpy(card->longname, "Intel HDMI/DP LPE Audio");
+        ctx->irq = -1;
+        ctx->tmds_clock_speed = DIS_SAMPLE_RATE_148_5;
+        INIT_WORK(&ctx->hdmi_audio_wq, had_audio_wq);
+        card->private_free = hdmi_lpe_audio_free;
+        /* assume pipe A as default */
+        ctx->had_config_offset = AUDIO_HDMI_CONFIG_A;
+        platform_set_drvdata(pdev, ctx);
+        dev_dbg(&pdev->dev, "%s: mmio_start = 0x%x, mmio_end = 0x%x\n",
+                __func__, (unsigned int)res_mmio->start,
+                (unsigned int)res_mmio->end);
+        ctx->mmio_start = ioremap_nocache(res_mmio->start,
+                                          (size_t)(resource_size(res_mmio)));
+        if (!ctx->mmio_start) {
+                dev_err(&pdev->dev, "Could not get ioremap\n");
+                ret = -EACCES;
+                goto err;
+        }
+        /* setup interrupt handler */
+        ret = request_irq(irq, display_pipe_interrupt_handler, 0,
+                          pdev->name, ctx);
+        if (ret < 0) {
+                dev_err(&pdev->dev, "request_irq failed\n");
+                goto err;
+        }
+        ctx->irq = irq;
+        ret = snd_pcm_new(card, INTEL_HAD, PCM_INDEX, MAX_PB_STREAMS,
+                          MAX_CAP_STREAMS, &pcm);
+        if (ret)
+                goto err;
+        /* setup private data which can be retrieved when required */
+        pcm->private_data = ctx;
+        pcm->info_flags = 0;
+        strncpy(pcm->name, card->shortname, strlen(card->shortname));
+        /* setup the ops for playabck */
+        snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &had_pcm_ops);
+        /* only 32bit addressable */
+        dma_set_mask(&pdev->dev, DMA_BIT_MASK(32));
+        dma_set_coherent_mask(&pdev->dev, DMA_BIT_MASK(32));
+        /* allocate dma pages;
+         * try to allocate 600k buffer as default which is large enough
+         */
+        snd_pcm_lib_preallocate_pages_for_all(pcm,
+                        SNDRV_DMA_TYPE_DEV, NULL,
+                        HAD_DEFAULT_BUFFER, HAD_MAX_BUFFER);
+        /* create controls */
+        for (i = 0; i < ARRAY_SIZE(had_controls); i++) {
+                ret = snd_ctl_add(card, snd_ctl_new1(&had_controls[i], ctx));
+                if (ret < 0)
+                        goto err;
+        }
+        init_channel_allocations();
+        /* Register channel map controls */
+        ret = had_register_chmap_ctls(ctx, pcm);
+        if (ret < 0)
+                goto err;
+        ret = had_create_jack(ctx);
+        if (ret < 0)
+                goto err;
+        ret = snd_card_register(card);
+        if (ret)
+                goto err;
+        spin_lock_irq(&pdata->lpe_audio_slock);
+        pdata->notify_audio_lpe = notify_audio_lpe;
+        pdata->notify_pending = false;
+        spin_unlock_irq(&pdata->lpe_audio_slock);
+        pm_runtime_set_active(&pdev->dev);
+        pm_runtime_enable(&pdev->dev);
+        dev_dbg(&pdev->dev, "%s: handle pending notification\n", __func__);
+        schedule_work(&ctx->hdmi_audio_wq);
+        return 0;
+err:
+        snd_card_free(card);
+        return ret;
+}
+/*
+ * hdmi_lpe_audio_remove - stop bridge with i915
+ *
+ * This function is called when the platform device is destroyed.
+ */
+static int hdmi_lpe_audio_remove(struct platform_device *pdev)
+{
+        struct snd_intelhad *ctx = platform_get_drvdata(pdev);
+        snd_card_free(ctx->card);
+        return 0;
+}
+static const struct dev_pm_ops hdmi_lpe_audio_pm = {
+        SET_SYSTEM_SLEEP_PM_OPS(hdmi_lpe_audio_suspend, hdmi_lpe_audio_resume)
+        SET_RUNTIME_PM_OPS(hdmi_lpe_audio_runtime_suspend, NULL, NULL)
+};
+static struct platform_driver hdmi_lpe_audio_driver = {
+        .driver         = {
+                .name  = "hdmi-lpe-audio",
+                .pm = &hdmi_lpe_audio_pm,
+        },
+        .probe          = hdmi_lpe_audio_probe,
+        .remove         = hdmi_lpe_audio_remove,
+};
+module_platform_driver(hdmi_lpe_audio_driver);
+MODULE_ALIAS("platform:hdmi_lpe_audio");
+MODULE_AUTHOR("Sailaja Bandarupalli <sailaja.bandarupalli@intel.com>");
+MODULE_AUTHOR("Ramesh Babu K V <ramesh.babu@intel.com>");
+MODULE_AUTHOR("Vaibhav Agarwal <vaibhav.agarwal@intel.com>");
+MODULE_AUTHOR("Jerome Anand <jerome.anand@intel.com>");
+MODULE_DESCRIPTION("Intel HDMI Audio driver");
+MODULE_LICENSE("GPL v2");
+MODULE_SUPPORTED_DEVICE("{Intel,Intel_HAD}");
diff --git a/sound/x86/intel_hdmi_audio.h b/sound/x86/intel_hdmi_audio.h
new file mode 100644
index 000000000000..2d3e389f76b3
--- /dev/null
+++ b/sound/x86/intel_hdmi_audio.h
@@ -0,0 +1,136 @@
+/*
+ * Copyright (C) 2016 Intel Corporation
+ *  Authors:    Sailaja Bandarupalli <sailaja.bandarupalli@intel.com>
+ *              Ramesh Babu K V <ramesh.babu@intel.com>
+ *              Vaibhav Agarwal <vaibhav.agarwal@intel.com>
+ *              Jerome Anand <jerome.anand@intel.com>
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice (including the
+ * next paragraph) shall be included in all copies or substantial
+ * portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
+ * NONINFRINGEMENT.  IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS
+ * BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN
+ * ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
+ * CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+#ifndef _INTEL_HDMI_AUDIO_H_
+#define _INTEL_HDMI_AUDIO_H_
+#include "intel_hdmi_lpe_audio.h"
+#define PCM_INDEX               0
+#define MAX_PB_STREAMS          1
+#define MAX_CAP_STREAMS         0
+#define BYTES_PER_WORD          0x4
+#define INTEL_HAD               "HdmiLpeAudio"
+/*
+ *      CEA speaker placement:
+ *
+ *      FL  FLC   FC   FRC   FR
+ *
+ *                                              LFE
+ *
+ *      RL  RLC   RC   RRC   RR
+ *
+ *      The Left/Right Surround channel _notions_ LS/RS in SMPTE 320M
+ *      corresponds to CEA RL/RR; The SMPTE channel _assignment_ C/LFE is
+ *      swapped to CEA LFE/FC.
+ */
+enum cea_speaker_placement {
+        FL  = (1 <<  0),        /* Front Left           */
+        FC  = (1 <<  1),        /* Front Center         */
+        FR  = (1 <<  2),        /* Front Right          */
+        FLC = (1 <<  3),        /* Front Left Center    */
+        FRC = (1 <<  4),        /* Front Right Center   */
+        RL  = (1 <<  5),        /* Rear Left            */
+        RC  = (1 <<  6),        /* Rear Center          */
+        RR  = (1 <<  7),        /* Rear Right           */
+        RLC = (1 <<  8),        /* Rear Left Center     */
+        RRC = (1 <<  9),        /* Rear Right Center    */
+        LFE = (1 << 10),        /* Low Frequency Effect */
+};
+struct cea_channel_speaker_allocation {
+        int ca_index;
+        int speakers[8];
+        /* derived values, just for convenience */
+        int channels;
+        int spk_mask;
+};
+struct channel_map_table {
+        unsigned char map;              /* ALSA API channel map position */
+        unsigned char cea_slot;         /* CEA slot value */
+        int spk_mask;                   /* speaker position bit mask */
+};
+struct pcm_stream_info {
+        struct snd_pcm_substream *substream;
+        int substream_refcount;
+};
+/*
+ * struct snd_intelhad - intelhad driver structure
+ *
+ * @card: ptr to hold card details
+ * @connected: the monitor connection status
+ * @stream_info: stream information
+ * @eld: holds ELD info
+ * @curr_buf: pointer to hold current active ring buf
+ * @valid_buf_cnt: ring buffer count for stream
+ * @had_spinlock: driver lock
+ * @aes_bits: IEC958 status bits
+ * @buff_done: id of current buffer done intr
+ * @dev: platoform device handle
+ * @chmap: holds channel map info
+ */
+struct snd_intelhad {
+        struct snd_card *card;
+        bool            connected;
+        struct          pcm_stream_info stream_info;
+        unsigned char   eld[HDMI_MAX_ELD_BYTES];
+        bool dp_output;
+        unsigned int    aes_bits;
+        spinlock_t had_spinlock;
+        struct device *dev;
+        struct snd_pcm_chmap *chmap;
+        int tmds_clock_speed;
+        int link_rate;
+        /* ring buffer (BD) position index */
+        unsigned int bd_head;
+        /* PCM buffer position indices */
+        unsigned int pcmbuf_head;       /* being processed */
+        unsigned int pcmbuf_filled;     /* to be filled */
+        unsigned int num_bds;           /* number of BDs */
+        unsigned int period_bytes;      /* PCM period size in bytes */
+        /* internal stuff */
+        int irq;
+        void __iomem *mmio_start;
+        unsigned int had_config_offset;
+        union aud_cfg aud_config;       /* AUD_CONFIG reg value cache */
+        struct work_struct hdmi_audio_wq;
+        struct mutex mutex; /* for protecting chmap and eld */
+        bool need_reset;
+        struct snd_jack *jack;
+};
+#endif /* _INTEL_HDMI_AUDIO_ */
diff --git a/sound/x86/intel_hdmi_lpe_audio.h b/sound/x86/intel_hdmi_lpe_audio.h
new file mode 100644
index 000000000000..477e5153307c
--- /dev/null
+++ b/sound/x86/intel_hdmi_lpe_audio.h
@@ -0,0 +1,328 @@
+/*
+ *   intel_hdmi_lpe_audio.h - Intel HDMI LPE audio driver
+ *
+ *  Copyright (C) 2016 Intel Corp
+ *  Authors:    Sailaja Bandarupalli <sailaja.bandarupalli@intel.com>
+ *              Ramesh Babu K V <ramesh.babu@intel.com>
+ *              Vaibhav Agarwal <vaibhav.agarwal@intel.com>
+ *              Jerome Anand <jerome.anand@intel.com>
+ *              Aravind Siddappaji <aravindx.siddappaji@intel.com>
+ *  ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ *
+ *  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; version 2 of the License.
+ *
+ *  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.
+ *
+ * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+ */
+#ifndef __INTEL_HDMI_LPE_AUDIO_H
+#define __INTEL_HDMI_LPE_AUDIO_H
+#define HAD_MIN_CHANNEL         2
+#define HAD_MAX_CHANNEL         8
+#define HAD_NUM_OF_RING_BUFS    4
+/* max 20bit address, aligned to 64 */
+#define HAD_MAX_BUFFER          ((1024 * 1024 - 1) & ~0x3f)
+#define HAD_DEFAULT_BUFFER      (600 * 1024) /* default prealloc size */
+#define HAD_MAX_PERIODS         256     /* arbitrary, but should suffice */
+#define HAD_MIN_PERIODS         1
+#define HAD_MAX_PERIOD_BYTES    ((HAD_MAX_BUFFER / HAD_MIN_PERIODS) & ~0x3f)
+#define HAD_MIN_PERIOD_BYTES    1024    /* might be smaller */
+#define HAD_FIFO_SIZE           0 /* fifo not being used */
+#define MAX_SPEAKERS            8
+#define AUD_SAMPLE_RATE_32      32000
+#define AUD_SAMPLE_RATE_44_1    44100
+#define AUD_SAMPLE_RATE_48      48000
+#define AUD_SAMPLE_RATE_88_2    88200
+#define AUD_SAMPLE_RATE_96      96000
+#define AUD_SAMPLE_RATE_176_4   176400
+#define AUD_SAMPLE_RATE_192     192000
+#define HAD_MIN_RATE            AUD_SAMPLE_RATE_32
+#define HAD_MAX_RATE            AUD_SAMPLE_RATE_192
+#define DIS_SAMPLE_RATE_25_2    25200
+#define DIS_SAMPLE_RATE_27      27000
+#define DIS_SAMPLE_RATE_54      54000
+#define DIS_SAMPLE_RATE_74_25   74250
+#define DIS_SAMPLE_RATE_148_5   148500
+#define HAD_REG_WIDTH           0x08
+#define HAD_MAX_DIP_WORDS               16
+/* DP Link Rates */
+#define DP_2_7_GHZ                      270000
+#define DP_1_62_GHZ                     162000
+/* Maud Values */
+#define AUD_SAMPLE_RATE_32_DP_2_7_MAUD_VAL              1988
+#define AUD_SAMPLE_RATE_44_1_DP_2_7_MAUD_VAL            2740
+#define AUD_SAMPLE_RATE_48_DP_2_7_MAUD_VAL              2982
+#define AUD_SAMPLE_RATE_88_2_DP_2_7_MAUD_VAL            5480
+#define AUD_SAMPLE_RATE_96_DP_2_7_MAUD_VAL              5965
+#define AUD_SAMPLE_RATE_176_4_DP_2_7_MAUD_VAL           10961
+#define HAD_MAX_RATE_DP_2_7_MAUD_VAL                    11930
+#define AUD_SAMPLE_RATE_32_DP_1_62_MAUD_VAL             3314
+#define AUD_SAMPLE_RATE_44_1_DP_1_62_MAUD_VAL           4567
+#define AUD_SAMPLE_RATE_48_DP_1_62_MAUD_VAL             4971
+#define AUD_SAMPLE_RATE_88_2_DP_1_62_MAUD_VAL           9134
+#define AUD_SAMPLE_RATE_96_DP_1_62_MAUD_VAL             9942
+#define AUD_SAMPLE_RATE_176_4_DP_1_62_MAUD_VAL          18268
+#define HAD_MAX_RATE_DP_1_62_MAUD_VAL                   19884
+/* Naud Value */
+#define DP_NAUD_VAL                                     32768
+/* HDMI Controller register offsets - audio domain common */
+/* Base address for below regs = 0x65000 */
+enum hdmi_ctrl_reg_offset_common {
+        AUDIO_HDMI_CONFIG_A = 0x000,
+        AUDIO_HDMI_CONFIG_B = 0x800,
+        AUDIO_HDMI_CONFIG_C = 0x900,
+};
+/* HDMI controller register offsets */
+enum hdmi_ctrl_reg_offset {
+        AUD_CONFIG              = 0x0,
+        AUD_CH_STATUS_0         = 0x08,
+        AUD_CH_STATUS_1         = 0x0C,
+        AUD_HDMI_CTS            = 0x10,
+        AUD_N_ENABLE            = 0x14,
+        AUD_SAMPLE_RATE         = 0x18,
+        AUD_BUF_CONFIG          = 0x20,
+        AUD_BUF_CH_SWAP         = 0x24,
+        AUD_BUF_A_ADDR          = 0x40,
+        AUD_BUF_A_LENGTH        = 0x44,
+        AUD_BUF_B_ADDR          = 0x48,
+        AUD_BUF_B_LENGTH        = 0x4c,
+        AUD_BUF_C_ADDR          = 0x50,
+        AUD_BUF_C_LENGTH        = 0x54,
+        AUD_BUF_D_ADDR          = 0x58,
+        AUD_BUF_D_LENGTH        = 0x5c,
+        AUD_CNTL_ST             = 0x60,
+        AUD_HDMI_STATUS         = 0x64, /* v2 */
+        AUD_HDMIW_INFOFR        = 0x68, /* v2 */
+};
+/* Audio configuration */
+union aud_cfg {
+        struct {
+                u32 aud_en:1;
+                u32 layout:1;           /* LAYOUT[01], see below */
+                u32 fmt:2;
+                u32 num_ch:3;
+                u32 set:1;
+                u32 flat:1;
+                u32 val_bit:1;
+                u32 user_bit:1;
+                u32 underrun:1;         /* 0: send null packets,
+                                         * 1: send silence stream
+                                         */
+                u32 packet_mode:1;      /* 0: 32bit container, 1: 16bit */
+                u32 left_align:1;       /* 0: MSB bits 0-23, 1: bits 8-31 */
+                u32 bogus_sample:1;     /* bogus sample for odd channels */
+                u32 dp_modei:1;         /* 0: HDMI, 1: DP */
+                u32 rsvd:16;
+        } regx;
+        u32 regval;
+};
+#define AUD_CONFIG_VALID_BIT                    (1 << 9)
+#define AUD_CONFIG_DP_MODE                      (1 << 15)
+#define AUD_CONFIG_CH_MASK      0x70
+#define LAYOUT0                 0               /* interleaved stereo */
+#define LAYOUT1                 1               /* for channels > 2 */
+/* Audio Channel Status 0 Attributes */
+union aud_ch_status_0 {
+        struct {
+                u32 ch_status:1;
+                u32 lpcm_id:1;
+                u32 cp_info:1;
+                u32 format:3;
+                u32 mode:2;
+                u32 ctg_code:8;
+                u32 src_num:4;
+                u32 ch_num:4;
+                u32 samp_freq:4;        /* CH_STATUS_MAP_XXX */
+                u32 clk_acc:2;
+                u32 rsvd:2;
+        } regx;
+        u32 regval;
+};
+/* samp_freq values - Sampling rate as per IEC60958 Ver 3 */
+#define CH_STATUS_MAP_32KHZ     0x3
+#define CH_STATUS_MAP_44KHZ     0x0
+#define CH_STATUS_MAP_48KHZ     0x2
+#define CH_STATUS_MAP_88KHZ     0x8
+#define CH_STATUS_MAP_96KHZ     0xA
+#define CH_STATUS_MAP_176KHZ    0xC
+#define CH_STATUS_MAP_192KHZ    0xE
+/* Audio Channel Status 1 Attributes */
+union aud_ch_status_1 {
+        struct {
+                u32 max_wrd_len:1;
+                u32 wrd_len:3;
+                u32 rsvd:28;
+        } regx;
+        u32 regval;
+};
+#define MAX_SMPL_WIDTH_20       0x0
+#define MAX_SMPL_WIDTH_24       0x1
+#define SMPL_WIDTH_16BITS       0x1
+#define SMPL_WIDTH_24BITS       0x5
+/* CTS register */
+union aud_hdmi_cts {
+        struct {
+                u32 cts_val:24;
+                u32 en_cts_prog:1;
+                u32 rsvd:7;
+        } regx;
+        u32 regval;
+};
+/* N register */
+union aud_hdmi_n_enable {
+        struct {
+                u32 n_val:24;
+                u32 en_n_prog:1;
+                u32 rsvd:7;
+        } regx;
+        u32 regval;
+};
+/* Audio Buffer configurations */
+union aud_buf_config {
+        struct {
+                u32 audio_fifo_watermark:8;
+                u32 dma_fifo_watermark:3;
+                u32 rsvd0:5;
+                u32 aud_delay:8;
+                u32 rsvd1:8;
+        } regx;
+        u32 regval;
+};
+#define FIFO_THRESHOLD          0xFE
+#define DMA_FIFO_THRESHOLD      0x7
+/* Audio Sample Swapping offset */
+union aud_buf_ch_swap {
+        struct {
+                u32 first_0:3;
+                u32 second_0:3;
+                u32 first_1:3;
+                u32 second_1:3;
+                u32 first_2:3;
+                u32 second_2:3;
+                u32 first_3:3;
+                u32 second_3:3;
+                u32 rsvd:8;
+        } regx;
+        u32 regval;
+};
+#define SWAP_LFE_CENTER         0x00fac4c8      /* octal 76543210 */
+/* Address for Audio Buffer */
+union aud_buf_addr {
+        struct {
+                u32 valid:1;
+                u32 intr_en:1;
+                u32 rsvd:4;
+                u32 addr:26;
+        } regx;
+        u32 regval;
+};
+#define AUD_BUF_VALID           (1U << 0)
+#define AUD_BUF_INTR_EN         (1U << 1)
+/* Length of Audio Buffer */
+union aud_buf_len {
+        struct {
+                u32 buf_len:20;
+                u32 rsvd:12;
+        } regx;
+        u32 regval;
+};
+/* Audio Control State Register offset */
+union aud_ctrl_st {
+        struct {
+                u32 ram_addr:4;
+                u32 eld_ack:1;
+                u32 eld_addr:4;
+                u32 eld_buf_size:5;
+                u32 eld_valid:1;
+                u32 cp_ready:1;
+                u32 dip_freq:2;
+                u32 dip_idx:3;
+                u32 dip_en_sta:4;
+                u32 rsvd:7;
+        } regx;
+        u32 regval;
+};
+/* Audio HDMI Widget Data Island Packet offset */
+union aud_info_frame1 {
+        struct {
+                u32 pkt_type:8;
+                u32 ver_num:8;
+                u32 len:5;
+                u32 rsvd:11;
+        } regx;
+        u32 regval;
+};
+#define HDMI_INFO_FRAME_WORD1   0x000a0184
+#define DP_INFO_FRAME_WORD1     0x00441b84
+/* DIP frame 2 */
+union aud_info_frame2 {
+        struct {
+                u32 chksum:8;
+                u32 chnl_cnt:3;
+                u32 rsvd0:1;
+                u32 coding_type:4;
+                u32 smpl_size:2;
+                u32 smpl_freq:3;
+                u32 rsvd1:3;
+                u32 format:8;
+        } regx;
+        u32 regval;
+};
+/* DIP frame 3 */
+union aud_info_frame3 {
+        struct {
+                u32 chnl_alloc:8;
+                u32 rsvd0:3;
+                u32 lsv:4;
+                u32 dm_inh:1;
+                u32 rsvd1:16;
+        } regx;
+        u32 regval;
+};
+#define VALID_DIP_WORDS         3
+/* AUD_HDMI_STATUS bits */
+#define HDMI_AUDIO_UNDERRUN             (1U << 31)
+#define HDMI_AUDIO_BUFFER_DONE          (1U << 29)
+/* AUD_HDMI_STATUS register mask */
+#define AUD_HDMI_STATUS_MASK_UNDERRUN   0xC0000000
+#define AUD_HDMI_STATUS_MASK_SRDBG      0x00000002
+#define AUD_HDMI_STATUSG_MASK_FUNCRST   0x00000001
+#endif