Merge branch 'for-next' into for-linus

47 files changed, 2078 insertions, 867 deletions

diff --git a/sound/ac97/ac97_core.h b/sound/ac97/ac97_core.h
index 0c5956e4b2f3..5a9677c3d4c3 100644
--- a/sound/ac97/ac97_core.h
+++ b/sound/ac97/ac97_core.h
@@ -3,7 +3,7 @@
  * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik@free.fr>
  */
 
-unsigned int snd_ac97_bus_scan_one(struct ac97_controller *ac97,
+unsigned int snd_ac97_bus_scan_one(struct ac97_controller *adrv,
 				   unsigned int codec_num);
 
 static inline bool ac97_ids_match(unsigned int id1, unsigned int id2,
diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c
index d350dbd24305..1c8e8131a716 100644
--- a/sound/aoa/soundbus/i2sbus/pcm.c
+++ b/sound/aoa/soundbus/i2sbus/pcm.c
@@ -254,12 +254,11 @@ static void i2sbus_wait_for_stop(struct i2sbus_dev *i2sdev,
 				 struct pcm_info *pi)
 {
 	unsigned long flags;
-	struct completion done;
+	DECLARE_COMPLETION_ONSTACK(done);
 	long timeout;
 
 	spin_lock_irqsave(&i2sdev->low_lock, flags);
 	if (pi->dbdma_ring.stopping) {
-		init_completion(&done);
 		pi->stop_completion = &done;
 		spin_unlock_irqrestore(&i2sdev->low_lock, flags);
 		timeout = wait_for_completion_timeout(&done, HZ);
diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c
index 1006458f7f85..66ecbd4d034e 100644
--- a/sound/atmel/ac97c.c
+++ b/sound/atmel/ac97c.c
@@ -475,12 +475,12 @@ static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev)
 		struct snd_pcm_runtime *runtime;
 		int offset, next_period, block_size;
 		dev_dbg(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n",
-				casr & AC97C_CSR_OVRUN   ? " OVRUN"   : "",
-				casr & AC97C_CSR_RXRDY   ? " RXRDY"   : "",
-				casr & AC97C_CSR_UNRUN   ? " UNRUN"   : "",
-				casr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "",
-				casr & AC97C_CSR_TXRDY   ? " TXRDY"   : "",
-				!casr                    ? " NONE"    : "");
+			(casr & AC97C_CSR_OVRUN)   ? " OVRUN"   : "",
+			(casr & AC97C_CSR_RXRDY)   ? " RXRDY"   : "",
+			(casr & AC97C_CSR_UNRUN)   ? " UNRUN"   : "",
+			(casr & AC97C_CSR_TXEMPTY) ? " TXEMPTY" : "",
+			(casr & AC97C_CSR_TXRDY)   ? " TXRDY"   : "",
+			!casr                      ? " NONE"    : "");
 		if ((casr & camr) & AC97C_CSR_ENDTX) {
 			runtime = chip->playback_substream->runtime;
 			block_size = frames_to_bytes(runtime, runtime->period_size);
@@ -521,11 +521,11 @@ static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev)
 
 	if (sr & AC97C_SR_COEVT) {
 		dev_info(&chip->pdev->dev, "codec channel event%s%s%s%s%s\n",
-				cosr & AC97C_CSR_OVRUN   ? " OVRUN"   : "",
-				cosr & AC97C_CSR_RXRDY   ? " RXRDY"   : "",
-				cosr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "",
-				cosr & AC97C_CSR_TXRDY   ? " TXRDY"   : "",
-				!cosr                    ? " NONE"    : "");
+			 (cosr & AC97C_CSR_OVRUN)   ? " OVRUN"   : "",
+			 (cosr & AC97C_CSR_RXRDY)   ? " RXRDY"   : "",
+			 (cosr & AC97C_CSR_TXEMPTY) ? " TXEMPTY" : "",
+			 (cosr & AC97C_CSR_TXRDY)   ? " TXRDY"   : "",
+			 !cosr                      ? " NONE"    : "");
 		retval = IRQ_HANDLED;
 	}
 
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c
index 0e53f6f31916..c1fec932c49d 100644
--- a/sound/core/compress_offload.c
+++ b/sound/core/compress_offload.c
@@ -513,10 +513,11 @@ EXPORT_SYMBOL(snd_compr_malloc_pages);
 
 int snd_compr_free_pages(struct snd_compr_stream *stream)
 {
-	struct snd_compr_runtime *runtime = stream->runtime;
+	struct snd_compr_runtime *runtime;
 
 	if (snd_BUG_ON(!(stream) || !(stream)->runtime))
 		return -EINVAL;
+	runtime = stream->runtime;
 	if (runtime->dma_area == NULL)
 		return 0;
 	if (runtime->dma_buffer_p != &stream->dma_buffer) {
@@ -1031,7 +1032,7 @@ static const struct file_operations snd_compr_file_ops = {
 
 static int snd_compress_dev_register(struct snd_device *device)
 {
-	int ret = -EINVAL;
+	int ret;
 	struct snd_compr *compr;
 
 	if (snd_BUG_ON(!device || !device->device_data))
diff --git a/sound/core/control.c b/sound/core/control.c
index aa0c0cf182af..421ddc76f264 100644
--- a/sound/core/control.c
+++ b/sound/core/control.c
@@ -150,14 +150,14 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask,
 		return;
 	if (card->shutdown)
 		return;
-	read_lock(&card->ctl_files_rwlock);
+	read_lock_irqsave(&card->ctl_files_rwlock, flags);
 #if IS_ENABLED(CONFIG_SND_MIXER_OSS)
 	card->mixer_oss_change_count++;
 #endif
 	list_for_each_entry(ctl, &card->ctl_files, list) {
 		if (!ctl->subscribed)
 			continue;
-		spin_lock_irqsave(&ctl->read_lock, flags);
+		spin_lock(&ctl->read_lock);
 		list_for_each_entry(ev, &ctl->events, list) {
 			if (ev->id.numid == id->numid) {
 				ev->mask |= mask;
@@ -174,10 +174,10 @@ void snd_ctl_notify(struct snd_card *card, unsigned int mask,
 		}
 	_found:
 		wake_up(&ctl->change_sleep);
-		spin_unlock_irqrestore(&ctl->read_lock, flags);
+		spin_unlock(&ctl->read_lock);
 		kill_fasync(&ctl->fasync, SIGIO, POLL_IN);
 	}
-	read_unlock(&card->ctl_files_rwlock);
+	read_unlock_irqrestore(&card->ctl_files_rwlock, flags);
 }
 EXPORT_SYMBOL(snd_ctl_notify);
 
@@ -717,22 +717,19 @@ static int snd_ctl_card_info(struct snd_card *card, struct snd_ctl_file * ctl,
 }
 
 static int snd_ctl_elem_list(struct snd_card *card,
-			     struct snd_ctl_elem_list __user *_list)
+			     struct snd_ctl_elem_list *list)
 {
-	struct snd_ctl_elem_list list;
 	struct snd_kcontrol *kctl;
 	struct snd_ctl_elem_id id;
 	unsigned int offset, space, jidx;
 	int err = 0;
 
-	if (copy_from_user(&list, _list, sizeof(list)))
-		return -EFAULT;
-	offset = list.offset;
-	space = list.space;
+	offset = list->offset;
+	space = list->space;
 
 	down_read(&card->controls_rwsem);
-	list.count = card->controls_count;
-	list.used = 0;
+	list->count = card->controls_count;
+	list->used = 0;
 	if (space > 0) {
 		list_for_each_entry(kctl, &card->controls, list) {
 			if (offset >= kctl->count) {
@@ -741,12 +738,12 @@ static int snd_ctl_elem_list(struct snd_card *card,
 			}
 			for (jidx = offset; jidx < kctl->count; jidx++) {
 				snd_ctl_build_ioff(&id, kctl, jidx);
-				if (copy_to_user(list.pids + list.used, &id,
+				if (copy_to_user(list->pids + list->used, &id,
 						 sizeof(id))) {
 					err = -EFAULT;
 					goto out;
 				}
-				list.used++;
+				list->used++;
 				if (!--space)
 					goto out;
 			}
@@ -755,11 +752,26 @@ static int snd_ctl_elem_list(struct snd_card *card,
 	}
  out:
 	up_read(&card->controls_rwsem);
-	if (!err && copy_to_user(_list, &list, sizeof(list)))
-		err = -EFAULT;
 	return err;
 }
 
+static int snd_ctl_elem_list_user(struct snd_card *card,
+				  struct snd_ctl_elem_list __user *_list)
+{
+	struct snd_ctl_elem_list list;
+	int err;
+
+	if (copy_from_user(&list, _list, sizeof(list)))
+		return -EFAULT;
+	err = snd_ctl_elem_list(card, &list);
+	if (err)
+		return err;
+	if (copy_to_user(_list, &list, sizeof(list)))
+		return -EFAULT;
+
+	return 0;
+}
+
 /* Check whether the given kctl info is valid */
 static int snd_ctl_check_elem_info(struct snd_card *card,
 				   const struct snd_ctl_elem_info *info)
@@ -1703,7 +1715,7 @@ static long snd_ctl_ioctl(struct file *file, unsigned int cmd, unsigned long arg
 	case SNDRV_CTL_IOCTL_CARD_INFO:
 		return snd_ctl_card_info(card, ctl, cmd, argp);
 	case SNDRV_CTL_IOCTL_ELEM_LIST:
-		return snd_ctl_elem_list(card, argp);
+		return snd_ctl_elem_list_user(card, argp);
 	case SNDRV_CTL_IOCTL_ELEM_INFO:
 		return snd_ctl_elem_info_user(ctl, argp);
 	case SNDRV_CTL_IOCTL_ELEM_READ:
@@ -1939,8 +1951,9 @@ int snd_ctl_get_preferred_subdevice(struct snd_card *card, int type)
 {
 	struct snd_ctl_file *kctl;
 	int subdevice = -1;
+	unsigned long flags;
 
-	read_lock(&card->ctl_files_rwlock);
+	read_lock_irqsave(&card->ctl_files_rwlock, flags);
 	list_for_each_entry(kctl, &card->ctl_files, list) {
 		if (kctl->pid == task_pid(current)) {
 			subdevice = kctl->preferred_subdevice[type];
@@ -1948,7 +1961,7 @@ int snd_ctl_get_preferred_subdevice(struct snd_card *card, int type)
 				break;
 		}
 	}
-	read_unlock(&card->ctl_files_rwlock);
+	read_unlock_irqrestore(&card->ctl_files_rwlock, flags);
 	return subdevice;
 }
 EXPORT_SYMBOL_GPL(snd_ctl_get_preferred_subdevice);
@@ -1997,13 +2010,14 @@ static int snd_ctl_dev_disconnect(struct snd_device *device)
 {
 	struct snd_card *card = device->device_data;
 	struct snd_ctl_file *ctl;
+	unsigned long flags;
 
-	read_lock(&card->ctl_files_rwlock);
+	read_lock_irqsave(&card->ctl_files_rwlock, flags);
 	list_for_each_entry(ctl, &card->ctl_files, list) {
 		wake_up(&ctl->change_sleep);
 		kill_fasync(&ctl->fasync, SIGIO, POLL_ERR);
 	}
-	read_unlock(&card->ctl_files_rwlock);
+	read_unlock_irqrestore(&card->ctl_files_rwlock, flags);
 
 	return snd_unregister_device(&card->ctl_dev);
 }
diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c
index 02df1d7db9a1..1d708aab9c98 100644
--- a/sound/core/control_compat.c
+++ b/sound/core/control_compat.c
@@ -22,24 +22,22 @@ struct snd_ctl_elem_list32 {
 static int snd_ctl_elem_list_compat(struct snd_card *card,
 				    struct snd_ctl_elem_list32 __user *data32)
 {
-	struct snd_ctl_elem_list __user *data;
+	struct snd_ctl_elem_list data = {};
 	compat_caddr_t ptr;
 	int err;
 
-	data = compat_alloc_user_space(sizeof(*data));
-
 	/* offset, space, used, count */
-	if (copy_in_user(data, data32, 4 * sizeof(u32)))
+	if (copy_from_user(&data, data32, 4 * sizeof(u32)))
 		return -EFAULT;
 	/* pids */
-	if (get_user(ptr, &data32->pids) ||
-	    put_user(compat_ptr(ptr), &data->pids))
+	if (get_user(ptr, &data32->pids))
 		return -EFAULT;
-	err = snd_ctl_elem_list(card, data);
+	data.pids = compat_ptr(ptr);
+	err = snd_ctl_elem_list(card, &data);
 	if (err < 0)
 		return err;
 	/* copy the result */
-	if (copy_in_user(data32, data, 4 * sizeof(u32)))
+	if (copy_to_user(data32, &data, 4 * sizeof(u32)))
 		return -EFAULT;
 	return 0;
 }
diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c
index c61ba52a530a..e97ff8cccb64 100644
--- a/sound/core/hrtimer.c
+++ b/sound/core/hrtimer.c
@@ -114,7 +114,7 @@ static int snd_hrtimer_stop(struct snd_timer *t)
 }
 
 static const struct snd_timer_hardware hrtimer_hw __initconst = {
-	.flags =	SNDRV_TIMER_HW_AUTO | SNDRV_TIMER_HW_TASKLET,
+	.flags =	SNDRV_TIMER_HW_AUTO | SNDRV_TIMER_HW_WORK,
 	.open =		snd_hrtimer_open,
 	.close =	snd_hrtimer_close,
 	.start =	snd_hrtimer_start,
diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c
index 21edb8ac95eb..0c029892880a 100644
--- a/sound/core/hwdep.c
+++ b/sound/core/hwdep.c
@@ -203,28 +203,35 @@ static int snd_hwdep_dsp_status(struct snd_hwdep *hw,
 }
 
 static int snd_hwdep_dsp_load(struct snd_hwdep *hw,
-			      struct snd_hwdep_dsp_image __user *_info)
+			      struct snd_hwdep_dsp_image *info)
 {
-	struct snd_hwdep_dsp_image info;
 	int err;
 	
 	if (! hw->ops.dsp_load)
 		return -ENXIO;
-	memset(&info, 0, sizeof(info));
-	if (copy_from_user(&info, _info, sizeof(info)))
-		return -EFAULT;
-	if (info.index >= 32)
+	if (info->index >= 32)
 		return -EINVAL;
 	/* check whether the dsp was already loaded */
-	if (hw->dsp_loaded & (1u << info.index))
+	if (hw->dsp_loaded & (1u << info->index))
 		return -EBUSY;
-	err = hw->ops.dsp_load(hw, &info);
+	err = hw->ops.dsp_load(hw, info);
 	if (err < 0)
 		return err;
-	hw->dsp_loaded |= (1u << info.index);
+	hw->dsp_loaded |= (1u << info->index);
 	return 0;
 }
 
+static int snd_hwdep_dsp_load_user(struct snd_hwdep *hw,
+				   struct snd_hwdep_dsp_image __user *_info)
+{
+	struct snd_hwdep_dsp_image info = {};
+
+	if (copy_from_user(&info, _info, sizeof(info)))
+		return -EFAULT;
+	return snd_hwdep_dsp_load(hw, &info);
+}
+
+
 static long snd_hwdep_ioctl(struct file * file, unsigned int cmd,
 			    unsigned long arg)
 {
@@ -238,7 +245,7 @@ static long snd_hwdep_ioctl(struct file * file, unsigned int cmd,
 	case SNDRV_HWDEP_IOCTL_DSP_STATUS:
 		return snd_hwdep_dsp_status(hw, argp);
 	case SNDRV_HWDEP_IOCTL_DSP_LOAD:
-		return snd_hwdep_dsp_load(hw, argp);
+		return snd_hwdep_dsp_load_user(hw, argp);
 	}
 	if (hw->ops.ioctl)
 		return hw->ops.ioctl(hw, file, cmd, arg);
diff --git a/sound/core/hwdep_compat.c b/sound/core/hwdep_compat.c
index bc81db9cb3d4..a0b76706c083 100644
--- a/sound/core/hwdep_compat.c
+++ b/sound/core/hwdep_compat.c
@@ -19,26 +19,17 @@ struct snd_hwdep_dsp_image32 {
 static int snd_hwdep_dsp_load_compat(struct snd_hwdep *hw,
 				     struct snd_hwdep_dsp_image32 __user *src)
 {
-	struct snd_hwdep_dsp_image __user *dst;
+	struct snd_hwdep_dsp_image info = {};
 	compat_caddr_t ptr;
-	u32 val;
 
-	dst = compat_alloc_user_space(sizeof(*dst));
-
-	/* index and name */
-	if (copy_in_user(dst, src, 4 + 64))
-		return -EFAULT;
-	if (get_user(ptr, &src->image) ||
-	    put_user(compat_ptr(ptr), &dst->image))
-		return -EFAULT;
-	if (get_user(val, &src->length) ||
-	    put_user(val, &dst->length))
-		return -EFAULT;
-	if (get_user(val, &src->driver_data) ||
-	    put_user(val, &dst->driver_data))
+	if (copy_from_user(&info, src, 4 + 64) ||
+	    get_user(ptr, &src->image) ||
+	    get_user(info.length, &src->length) ||
+	    get_user(info.driver_data, &src->driver_data))
 		return -EFAULT;
+	info.image = compat_ptr(ptr);
 
-	return snd_hwdep_dsp_load(hw, dst);
+	return snd_hwdep_dsp_load(hw, &info);
 }
 
 enum {
diff --git a/sound/core/init.c b/sound/core/init.c
index 0478847ba2b8..764dbe673d48 100644
--- a/sound/core/init.c
+++ b/sound/core/init.c
@@ -519,10 +519,9 @@ EXPORT_SYMBOL(snd_card_free_when_closed);
  */
 int snd_card_free(struct snd_card *card)
 {
-	struct completion released;
+	DECLARE_COMPLETION_ONSTACK(released);
 	int ret;
 
-	init_completion(&released);
 	card->release_completion = &released;
 	ret = snd_card_free_when_closed(card);
 	if (ret)
diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index ad74ea9cbff5..0aeeb6244ff6 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -157,8 +157,8 @@ int snd_dma_alloc_pages(int type, struct device *device, size_t size,
 		 * so if we fail to malloc, try to fetch memory traditionally.
 		 */
 		dmab->dev.type = SNDRV_DMA_TYPE_DEV;
-#endif /* CONFIG_GENERIC_ALLOCATOR */
 		fallthrough;
+#endif /* CONFIG_GENERIC_ALLOCATOR */
 	case SNDRV_DMA_TYPE_DEV:
 	case SNDRV_DMA_TYPE_DEV_UC:
 		snd_malloc_dev_pages(dmab, size);
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index b6d2331a82f7..be5714f1bb58 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -991,11 +991,13 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream)
 		       PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control)));
 	kfree(runtime->hw_constraints.rules);
 	/* Avoid concurrent access to runtime via PCM timer interface */
-	if (substream->timer)
+	if (substream->timer) {
 		spin_lock_irq(&substream->timer->lock);
-	substream->runtime = NULL;
-	if (substream->timer)
+		substream->runtime = NULL;
 		spin_unlock_irq(&substream->timer->lock);
+	} else {
+		substream->runtime = NULL;
+	}
 	kfree(runtime);
 	put_pid(substream->pid);
 	substream->pid = NULL;
diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c
index 1bf6a3d9e0c2..4f03ba8ed0ae 100644
--- a/sound/core/pcm_memory.c
+++ b/sound/core/pcm_memory.c
@@ -377,7 +377,7 @@ struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream, unsigne
  */
 int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size)
 {
-	struct snd_card *card = substream->pcm->card;
+	struct snd_card *card;
 	struct snd_pcm_runtime *runtime;
 	struct snd_dma_buffer *dmab = NULL;
 
@@ -387,6 +387,7 @@ int snd_pcm_lib_malloc_pages(struct snd_pcm_substream *substream, size_t size)
 		       SNDRV_DMA_TYPE_UNKNOWN))
 		return -EINVAL;
 	runtime = substream->runtime;
+	card = substream->pcm->card;
 
 	if (runtime->dma_buffer_p) {
 		/* perphaps, we might free the large DMA memory region
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index 2a688b711a9a..c78720a3299c 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -35,7 +35,7 @@ module_param_array(amidi_map, int, NULL, 0444);
 MODULE_PARM_DESC(amidi_map, "Raw MIDI device number assigned to 2nd OSS device.");
 #endif /* CONFIG_SND_OSSEMUL */
 
-static int snd_rawmidi_free(struct snd_rawmidi *rawmidi);
+static int snd_rawmidi_free(struct snd_rawmidi *rmidi);
 static int snd_rawmidi_dev_free(struct snd_device *device);
 static int snd_rawmidi_dev_register(struct snd_device *device);
 static int snd_rawmidi_dev_disconnect(struct snd_device *device);
diff --git a/sound/core/seq/oss/seq_oss.c b/sound/core/seq/oss/seq_oss.c
index c8b9c0b315d8..250a92b18726 100644
--- a/sound/core/seq/oss/seq_oss.c
+++ b/sound/core/seq/oss/seq_oss.c
@@ -174,9 +174,12 @@ odev_ioctl(struct file *file, unsigned int cmd, unsigned long arg)
 	if (snd_BUG_ON(!dp))
 		return -ENXIO;
 
-	mutex_lock(&register_mutex);
+	if (cmd != SNDCTL_SEQ_SYNC &&
+	    mutex_lock_interruptible(&register_mutex))
+		return -ERESTARTSYS;
 	rc = snd_seq_oss_ioctl(dp, cmd, arg);
-	mutex_unlock(&register_mutex);
+	if (cmd != SNDCTL_SEQ_SYNC)
+		mutex_unlock(&register_mutex);
 	return rc;
 }
 
diff --git a/sound/core/timer.c b/sound/core/timer.c
index 6e27d87b18ed..765ea66665a8 100644
--- a/sound/core/timer.c
+++ b/sound/core/timer.c
@@ -173,7 +173,7 @@ EXPORT_SYMBOL(snd_timer_instance_free);
  */
 static struct snd_timer *snd_timer_find(struct snd_timer_id *tid)
 {
-	struct snd_timer *timer = NULL;
+	struct snd_timer *timer;
 
 	list_for_each_entry(timer, &snd_timer_list, device_list) {
 		if (timer->tmr_class != tid->dev_class)
@@ -813,12 +813,12 @@ static void snd_timer_clear_callbacks(struct snd_timer *timer,
 }
 
 /*
- * timer tasklet
+ * timer work
  *
  */
-static void snd_timer_tasklet(struct tasklet_struct *t)
+static void snd_timer_work(struct work_struct *work)
 {
-	struct snd_timer *timer = from_tasklet(timer, t, task_queue);
+	struct snd_timer *timer = container_of(work, struct snd_timer, task_work);
 	unsigned long flags;
 
 	if (timer->card && timer->card->shutdown) {
@@ -843,7 +843,7 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left)
 	unsigned long resolution;
 	struct list_head *ack_list_head;
 	unsigned long flags;
-	int use_tasklet = 0;
+	bool use_work = false;
 
 	if (timer == NULL)
 		return;
@@ -884,7 +884,7 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left)
 			--timer->running;
 			list_del_init(&ti->active_list);
 		}
-		if ((timer->hw.flags & SNDRV_TIMER_HW_TASKLET) ||
+		if ((timer->hw.flags & SNDRV_TIMER_HW_WORK) ||
 		    (ti->flags & SNDRV_TIMER_IFLG_FAST))
 			ack_list_head = &timer->ack_list_head;
 		else
@@ -919,11 +919,11 @@ void snd_timer_interrupt(struct snd_timer * timer, unsigned long ticks_left)
 	snd_timer_process_callbacks(timer, &timer->ack_list_head);
 
 	/* do we have any slow callbacks? */
-	use_tasklet = !list_empty(&timer->sack_list_head);
+	use_work = !list_empty(&timer->sack_list_head);
 	spin_unlock_irqrestore(&timer->lock, flags);
 
-	if (use_tasklet)
-		tasklet_schedule(&timer->task_queue);
+	if (use_work)
+		queue_work(system_highpri_wq, &timer->task_work);
 }
 EXPORT_SYMBOL(snd_timer_interrupt);
 
@@ -967,7 +967,7 @@ int snd_timer_new(struct snd_card *card, char *id, struct snd_timer_id *tid,
 	INIT_LIST_HEAD(&timer->ack_list_head);
 	INIT_LIST_HEAD(&timer->sack_list_head);
 	spin_lock_init(&timer->lock);
-	tasklet_setup(&timer->task_queue, snd_timer_tasklet);
+	INIT_WORK(&timer->task_work, snd_timer_work);
 	timer->max_instances = 1000; /* default limit per timer */
 	if (card != NULL) {
 		timer->module = card->module;
@@ -1200,7 +1200,7 @@ static int snd_timer_s_close(struct snd_timer *timer)
 
 static const struct snd_timer_hardware snd_timer_system =
 {
-	.flags =	SNDRV_TIMER_HW_FIRST | SNDRV_TIMER_HW_TASKLET,
+	.flags =	SNDRV_TIMER_HW_FIRST | SNDRV_TIMER_HW_WORK,
 	.resolution =	1000000000L / HZ,
 	.ticks =	10000000L,
 	.close =	snd_timer_s_close,
@@ -1280,8 +1280,8 @@ static void snd_timer_proc_read(struct snd_info_entry *entry,
 		list_for_each_entry(ti, &timer->open_list_head, open_list)
 			snd_iprintf(buffer, "  Client %s : %s\n",
 				    ti->owner ? ti->owner : "unknown",
-				    ti->flags & (SNDRV_TIMER_IFLG_START |
-						 SNDRV_TIMER_IFLG_RUNNING)
+				    (ti->flags & (SNDRV_TIMER_IFLG_START |
+						  SNDRV_TIMER_IFLG_RUNNING))
 				    ? "running" : "stopped");
 	}
 	mutex_unlock(&register_mutex);
diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c
index 251eaf1152e2..c91356326699 100644
--- a/sound/drivers/aloop.c
+++ b/sound/drivers/aloop.c
@@ -110,7 +110,7 @@ struct loopback_cable {
 	struct {
 		int stream;
 		struct snd_timer_id id;
-		struct tasklet_struct event_tasklet;
+		struct work_struct event_work;
 		struct snd_timer_instance *instance;
 	} snd_timer;
 };
@@ -309,8 +309,8 @@ static int loopback_snd_timer_close_cable(struct loopback_pcm *dpcm)
 	 */
 	snd_timer_close(cable->snd_timer.instance);
 
-	/* wait till drain tasklet has finished if requested */
-	tasklet_kill(&cable->snd_timer.event_tasklet);
+	/* wait till drain work has finished if requested */
+	cancel_work_sync(&cable->snd_timer.event_work);
 
 	snd_timer_instance_free(cable->snd_timer.instance);
 	memset(&cable->snd_timer, 0, sizeof(cable->snd_timer));
@@ -794,11 +794,11 @@ static void loopback_snd_timer_function(struct snd_timer_instance *timeri,
 					  resolution);
 }
 
-static void loopback_snd_timer_tasklet(unsigned long arg)
+static void loopback_snd_timer_work(struct work_struct *work)
 {
-	struct snd_timer_instance *timeri = (struct snd_timer_instance *)arg;
-	struct loopback_cable *cable = timeri->callback_data;
+	struct loopback_cable *cable;
 
+	cable = container_of(work, struct loopback_cable, snd_timer.event_work);
 	loopback_snd_timer_period_elapsed(cable, SNDRV_TIMER_EVENT_MSTOP, 0);
 }
 
@@ -828,9 +828,9 @@ static void loopback_snd_timer_event(struct snd_timer_instance *timeri,
 		 * state the streaming will be aborted by the usual timeout. It
 		 * should not be aborted here because may be the timer sound
 		 * card does only a recovery and the timer is back soon.
-		 * This tasklet triggers loopback_snd_timer_tasklet()
+		 * This work triggers loopback_snd_timer_work()
 		 */
-		tasklet_schedule(&cable->snd_timer.event_tasklet);
+		schedule_work(&cable->snd_timer.event_work);
 	}
 }
 
@@ -1124,7 +1124,7 @@ static int loopback_snd_timer_open(struct loopback_pcm *dpcm)
 		err = -ENOMEM;
 		goto exit;
 	}
-	/* The callback has to be called from another tasklet. If
+	/* The callback has to be called from another work. If
 	 * SNDRV_TIMER_IFLG_FAST is specified it will be called from the
 	 * snd_pcm_period_elapsed() call of the selected sound card.
 	 * snd_pcm_period_elapsed() helds snd_pcm_stream_lock_irqsave().
@@ -1137,9 +1137,8 @@ static int loopback_snd_timer_open(struct loopback_pcm *dpcm)
 	timeri->callback_data = (void *)cable;
 	timeri->ccallback = loopback_snd_timer_event;
 
-	/* initialise a tasklet used for draining */
-	tasklet_init(&cable->snd_timer.event_tasklet,
-		     loopback_snd_timer_tasklet, (unsigned long)timeri);
+	/* initialise a work used for draining */
+	INIT_WORK(&cable->snd_timer.event_work, loopback_snd_timer_work);
 
 	/* The mutex loopback->cable_lock is kept locked.
 	 * Therefore snd_timer_open() cannot be called a second time
diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c
index 4e79293d7f11..ed40d0f7432c 100644
--- a/sound/drivers/pcsp/pcsp_lib.c
+++ b/sound/drivers/pcsp/pcsp_lib.c
@@ -23,10 +23,10 @@ MODULE_PARM_DESC(nforce_wa, "Apply NForce chipset workaround "
 #define DMIX_WANTS_S16	1
 
 /*
- * Call snd_pcm_period_elapsed in a tasklet
+ * Call snd_pcm_period_elapsed in a work
  * This avoids spinlock messes and long-running irq contexts
  */
-static void pcsp_call_pcm_elapsed(unsigned long priv)
+static void pcsp_call_pcm_elapsed(struct work_struct *work)
 {
 	if (atomic_read(&pcsp_chip.timer_active)) {
 		struct snd_pcm_substream *substream;
@@ -36,7 +36,7 @@ static void pcsp_call_pcm_elapsed(unsigned long priv)
 	}
 }
 
-static DECLARE_TASKLET_OLD(pcsp_pcm_tasklet, pcsp_call_pcm_elapsed);
+static DECLARE_WORK(pcsp_pcm_work, pcsp_call_pcm_elapsed);
 
 /* write the port and returns the next expire time in ns;
  * called at the trigger-start and in hrtimer callback
@@ -119,11 +119,9 @@ static void pcsp_pointer_update(struct snd_pcsp *chip)
 	if (periods_elapsed) {
 		chip->period_ptr += periods_elapsed * period_bytes;
 		chip->period_ptr %= buffer_bytes;
+		queue_work(system_highpri_wq, &pcsp_pcm_work);
 	}
 	spin_unlock_irqrestore(&chip->substream_lock, flags);
-
-	if (periods_elapsed)
-		tasklet_schedule(&pcsp_pcm_tasklet);
 }
 
 enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
@@ -196,7 +194,7 @@ void pcsp_sync_stop(struct snd_pcsp *chip)
 	pcsp_stop_playing(chip);
 	local_irq_enable();
 	hrtimer_cancel(&chip->timer);
-	tasklet_kill(&pcsp_pcm_tasklet);
+	cancel_work_sync(&pcsp_pcm_work);
 }
 
 static int snd_pcsp_playback_close(struct snd_pcm_substream *substream)
diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c
index 38603cb2bd5b..c876cf9b5005 100644
--- a/sound/drivers/portman2x4.c
+++ b/sound/drivers/portman2x4.c
@@ -467,7 +467,7 @@ static int portman_probe(struct parport *p)
 	parport_write_control(p, 0);	/* Reset Strobe=0. */
 
 	/* Check if Tx circuitry is functioning properly.  If initialized 
-	 * unit TxEmpty is false, send out char and see if if goes true.
+	 * unit TxEmpty is false, send out char and see if it goes true.
 	 */
 	/* 8 */
 	parport_write_control(p, TXDATA0);	/* Tx channel 0, strobe off. */
diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c
index 26d591fe6a6b..d5c65cab195b 100644
--- a/sound/drivers/vx/vx_core.c
+++ b/sound/drivers/vx/vx_core.c
@@ -597,9 +597,9 @@ static void vx_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *b
 	
 	snd_iprintf(buffer, "%s\n", chip->card->longname);
 	snd_iprintf(buffer, "Xilinx Firmware: %s\n",
-		    chip->chip_status & VX_STAT_XILINX_LOADED ? "Loaded" : "No");
+		    (chip->chip_status & VX_STAT_XILINX_LOADED) ? "Loaded" : "No");
 	snd_iprintf(buffer, "Device Initialized: %s\n",
-		    chip->chip_status & VX_STAT_DEVICE_INIT ? "Yes" : "No");
+		    (chip->chip_status & VX_STAT_DEVICE_INIT) ? "Yes" : "No");
 	snd_iprintf(buffer, "DSP audio info:");
 	if (chip->audio_info & VX_AUDIO_INFO_REAL_TIME)
 		snd_iprintf(buffer, " realtime");
diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c
index 664b9efa9a50..3d2e3bcafca8 100644
--- a/sound/drivers/vx/vx_pcm.c
+++ b/sound/drivers/vx/vx_pcm.c
@@ -60,7 +60,6 @@ static void vx_pcm_read_per_bytes(struct vx_core *chip, struct snd_pcm_runtime *
 	*buf++ = vx_inb(chip, RXL);
 	if (++offset >= pipe->buffer_bytes) {
 		offset = 0;
-		buf = (unsigned char *)runtime->dma_area;
 	}
 	pipe->hw_ptr = offset;
 }
@@ -530,7 +529,6 @@ static int vx_pcm_playback_open(struct snd_pcm_substream *subs)
 		err = vx_alloc_pipe(chip, 0, audio, 2, &pipe); /* stereo playback */
 		if (err < 0)
 			return err;
-		chip->playback_pipes[audio] = pipe;
 	}
 	/* open for playback */
 	pipe->references++;
diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c
index ee1c428b1fd3..4e2f2bb7879f 100644
--- a/sound/firewire/amdtp-stream.c
+++ b/sound/firewire/amdtp-stream.c
@@ -64,7 +64,7 @@
 #define IT_PKT_HEADER_SIZE_CIP		8 // For 2 CIP header.
 #define IT_PKT_HEADER_SIZE_NO_CIP	0 // Nothing.
 
-static void pcm_period_tasklet(struct tasklet_struct *t);
+static void pcm_period_work(struct work_struct *work);
 
 /**
  * amdtp_stream_init - initialize an AMDTP stream structure
@@ -94,7 +94,7 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
 	s->flags = flags;
 	s->context = ERR_PTR(-1);
 	mutex_init(&s->mutex);
-	tasklet_setup(&s->period_tasklet, pcm_period_tasklet);
+	INIT_WORK(&s->period_work, pcm_period_work);
 	s->packet_index = 0;
 
 	init_waitqueue_head(&s->callback_wait);
@@ -203,7 +203,7 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s,
 
 	// Linux driver for 1394 OHCI controller voluntarily flushes isoc
 	// context when total size of accumulated context header reaches
-	// PAGE_SIZE. This kicks tasklet for the isoc context and brings
+	// PAGE_SIZE. This kicks work for the isoc context and brings
 	// callback in the middle of scheduled interrupts.
 	// Although AMDTP streams in the same domain use the same events per
 	// IRQ, use the largest size of context header between IT/IR contexts.
@@ -333,7 +333,7 @@ EXPORT_SYMBOL(amdtp_stream_get_max_payload);
  */
 void amdtp_stream_pcm_prepare(struct amdtp_stream *s)
 {
-	tasklet_kill(&s->period_tasklet);
+	cancel_work_sync(&s->period_work);
 	s->pcm_buffer_pointer = 0;
 	s->pcm_period_pointer = 0;
 }
@@ -437,13 +437,14 @@ static void update_pcm_pointers(struct amdtp_stream *s,
 	s->pcm_period_pointer += frames;
 	if (s->pcm_period_pointer >= pcm->runtime->period_size) {
 		s->pcm_period_pointer -= pcm->runtime->period_size;
-		tasklet_hi_schedule(&s->period_tasklet);
+		queue_work(system_highpri_wq, &s->period_work);
 	}
 }
 
-static void pcm_period_tasklet(struct tasklet_struct *t)
+static void pcm_period_work(struct work_struct *work)
 {
-	struct amdtp_stream *s = from_tasklet(s, t, period_tasklet);
+	struct amdtp_stream *s = container_of(work, struct amdtp_stream,
+					      period_work);
 	struct snd_pcm_substream *pcm = READ_ONCE(s->pcm);
 
 	if (pcm)
@@ -794,7 +795,7 @@ static void generate_pkt_descs(struct amdtp_stream *s, struct pkt_desc *descs,
 static inline void cancel_stream(struct amdtp_stream *s)
 {
 	s->packet_index = -1;
-	if (in_interrupt())
+	if (current_work() == &s->period_work)
 		amdtp_stream_pcm_abort(s);
 	WRITE_ONCE(s->pcm_buffer_pointer, SNDRV_PCM_POS_XRUN);
 }
@@ -1184,7 +1185,7 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d,
 
 	if (irq_target && amdtp_stream_running(irq_target)) {
 		// This function is called in software IRQ context of
-		// period_tasklet or process context.
+		// period_work or process context.
 		//
 		// When the software IRQ context was scheduled by software IRQ
 		// context of IT contexts, queued packets were already handled.
@@ -1195,9 +1196,9 @@ unsigned long amdtp_domain_stream_pcm_pointer(struct amdtp_domain *d,
 		// immediately to keep better granularity of PCM pointer.
 		//
 		// Later, the process context will sometimes schedules software
-		// IRQ context of the period_tasklet. Then, no need to flush the
+		// IRQ context of the period_work. Then, no need to flush the
 		// queue by the same reason as described in the above
-		if (!in_interrupt()) {
+		if (current_work() != &s->period_work) {
 			// Queued packet should be processed without any kernel
 			// preemption to keep latency against bus cycle.
 			preempt_disable();
@@ -1263,7 +1264,7 @@ static void amdtp_stream_stop(struct amdtp_stream *s)
 		return;
 	}
 
-	tasklet_kill(&s->period_tasklet);
+	cancel_work_sync(&s->period_work);
 	fw_iso_context_stop(s->context);
 	fw_iso_context_destroy(s->context);
 	s->context = ERR_PTR(-1);
diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h
index 703b710aaf7f..2ceb57d1d58e 100644
--- a/sound/firewire/amdtp-stream.h
+++ b/sound/firewire/amdtp-stream.h
@@ -163,7 +163,7 @@ struct amdtp_stream {
 
 	/* For a PCM substream processing. */
 	struct snd_pcm_substream *pcm;
-	struct tasklet_struct period_tasklet;
+	struct work_struct period_work;
 	snd_pcm_uframes_t pcm_buffer_pointer;
 	unsigned int pcm_period_pointer;
 
diff --git a/sound/hda/hdac_component.c b/sound/hda/hdac_component.c
index 89126c6fd216..bb37e7e0bd79 100644
--- a/sound/hda/hdac_component.c
+++ b/sound/hda/hdac_component.c
@@ -210,12 +210,14 @@ static int hdac_component_master_bind(struct device *dev)
 			goto module_put;
 	}
 
+	complete_all(&acomp->master_bind_complete);
 	return 0;
 
  module_put:
 	module_put(acomp->ops->owner);
 out_unbind:
 	component_unbind_all(dev, acomp);
+	complete_all(&acomp->master_bind_complete);
 
 	return ret;
 }
@@ -296,6 +298,7 @@ int snd_hdac_acomp_init(struct hdac_bus *bus,
 	if (!acomp)
 		return -ENOMEM;
 	acomp->audio_ops = aops;
+	init_completion(&acomp->master_bind_complete);
 	bus->audio_component = acomp;
 	devres_add(dev, acomp);
 
diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c
index 3c2db3816029..454474ac5716 100644
--- a/sound/hda/hdac_i915.c
+++ b/sound/hda/hdac_i915.c
@@ -11,9 +11,7 @@
 #include <sound/hda_i915.h>
 #include <sound/hda_register.h>
 
-static struct completion bind_complete;
-
-#define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \
+#define IS_HSW_CONTROLLER(pci) (((pci)->device == 0x0a0c) || \
 				((pci)->device == 0x0c0c) || \
 				((pci)->device == 0x0d0c) || \
 				((pci)->device == 0x160c))
@@ -41,7 +39,7 @@ void snd_hdac_i915_set_bclk(struct hdac_bus *bus)
 
 	if (!acomp || !acomp->ops || !acomp->ops->get_cdclk_freq)
 		return; /* only for i915 binding */
-	if (!CONTROLLER_IN_GPU(pci))
+	if (!IS_HSW_CONTROLLER(pci))
 		return; /* only HSW/BDW */
 
 	cdclk_freq = acomp->ops->get_cdclk_freq(acomp->dev);
@@ -73,11 +71,49 @@ void snd_hdac_i915_set_bclk(struct hdac_bus *bus)
 }
 EXPORT_SYMBOL_GPL(snd_hdac_i915_set_bclk);
 
+/* returns true if the devices can be connected for audio */
+static bool connectivity_check(struct pci_dev *i915, struct pci_dev *hdac)
+{
+	struct pci_bus *bus_a = i915->bus, *bus_b = hdac->bus;
+
+	/* directly connected on the same bus */
+	if (bus_a == bus_b)
+		return true;
+
+	/*
+	 * on i915 discrete GPUs with embedded HDA audio, the two
+	 * devices are connected via 2nd level PCI bridge
+	 */
+	bus_a = bus_a->parent;
+	bus_b = bus_b->parent;
+	if (!bus_a || !bus_b)
+		return false;
+	bus_a = bus_a->parent;
+	bus_b = bus_b->parent;
+	if (bus_a && bus_a == bus_b)
+		return true;
+
+	return false;
+}
+
 static int i915_component_master_match(struct device *dev, int subcomponent,
 				       void *data)
 {
-	return !strcmp(dev->driver->name, "i915") &&
-	       subcomponent == I915_COMPONENT_AUDIO;
+	struct pci_dev *hdac_pci, *i915_pci;
+	struct hdac_bus *bus = data;
+
+	if (!dev_is_pci(dev))
+		return 0;
+
+	hdac_pci = to_pci_dev(bus->dev);
+	i915_pci = to_pci_dev(dev);
+
+	if (!strcmp(dev->driver->name, "i915") &&
+	    subcomponent == I915_COMPONENT_AUDIO &&
+	    connectivity_check(i915_pci, hdac_pci))
+		return 1;
+
+	return 0;
 }
 
 /* check whether intel graphics is present */
@@ -92,19 +128,6 @@ static bool i915_gfx_present(void)
 	return pci_dev_present(ids);
 }
 
-static int i915_master_bind(struct device *dev,
-			    struct drm_audio_component *acomp)
-{
-	complete_all(&bind_complete);
-	/* clear audio_ops here as it was needed only for completion call */
-	acomp->audio_ops = NULL;
-	return 0;
-}
-
-static const struct drm_audio_component_audio_ops i915_init_ops = {
-	.master_bind = i915_master_bind
-};
-
 /**
  * snd_hdac_i915_init - Initialize i915 audio component
  * @bus: HDA core bus
@@ -125,9 +148,7 @@ int snd_hdac_i915_init(struct hdac_bus *bus)
 	if (!i915_gfx_present())
 		return -ENODEV;
 
-	init_completion(&bind_complete);
-
-	err = snd_hdac_acomp_init(bus, &i915_init_ops,
+	err = snd_hdac_acomp_init(bus, NULL,
 				  i915_component_master_match,
 				  sizeof(struct i915_audio_component) - sizeof(*acomp));
 	if (err < 0)
@@ -139,8 +160,8 @@ int snd_hdac_i915_init(struct hdac_bus *bus)
 		if (!IS_ENABLED(CONFIG_MODULES) ||
 		    !request_module("i915")) {
 			/* 60s timeout */
-			wait_for_completion_timeout(&bind_complete,
-						   msecs_to_jiffies(60 * 1000));
+			wait_for_completion_timeout(&acomp->master_bind_complete,
+						    msecs_to_jiffies(60 * 1000));
 		}
 	}
 	if (!acomp->ops) {
diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c
index 35e76480306e..5e1f9f10051b 100644
--- a/sound/pci/asihpi/asihpi.c
+++ b/sound/pci/asihpi/asihpi.c
@@ -117,7 +117,6 @@ struct snd_card_asihpi {
 	 * snd_card_asihpi_timer_function().
 	 */
 	struct snd_card_asihpi_pcm *llmode_streampriv;
-	struct tasklet_struct t;
 	void (*pcm_start)(struct snd_pcm_substream *substream);
 	void (*pcm_stop)(struct snd_pcm_substream *substream);
 
@@ -258,15 +257,6 @@ static inline u16 hpi_stream_group_reset(u32 h_stream)
 		return hpi_instream_group_reset(h_stream);
 }
 
-static inline u16 hpi_stream_group_get_map(
-				u32 h_stream, u32 *mo, u32 *mi)
-{
-	if (hpi_handle_object(h_stream) ==  HPI_OBJ_OSTREAM)
-		return hpi_outstream_group_get_map(h_stream, mo, mi);
-	else
-		return hpi_instream_group_get_map(h_stream, mo, mi);
-}
-
 static u16 handle_error(u16 err, int line, char *filename)
 {
 	if (err)
@@ -547,9 +537,7 @@ static void snd_card_asihpi_pcm_int_start(struct snd_pcm_substream *substream)
 	card = snd_pcm_substream_chip(substream);
 
 	WARN_ON(in_interrupt());
-	tasklet_disable(&card->t);
 	card->llmode_streampriv = dpcm;
-	tasklet_enable(&card->t);
 
 	hpi_handle_error(hpi_adapter_set_property(card->hpi->adapter->index,
 		HPI_ADAPTER_PROPERTY_IRQ_RATE,
@@ -565,13 +553,7 @@ static void snd_card_asihpi_pcm_int_stop(struct snd_pcm_substream *substream)
 	hpi_handle_error(hpi_adapter_set_property(card->hpi->adapter->index,
 		HPI_ADAPTER_PROPERTY_IRQ_RATE, 0, 0));
 
-	if (in_interrupt())
-		card->llmode_streampriv = NULL;
-	else {
-		tasklet_disable(&card->t);
-		card->llmode_streampriv = NULL;
-		tasklet_enable(&card->t);
-	}
+	card->llmode_streampriv = NULL;
 }
 
 static int snd_card_asihpi_trigger(struct snd_pcm_substream *substream,
@@ -921,10 +903,9 @@ static void snd_card_asihpi_timer_function(struct timer_list *t)
 		add_timer(&dpcm->timer);
 }
 
-static void snd_card_asihpi_int_task(struct tasklet_struct *t)
+static void snd_card_asihpi_isr(struct hpi_adapter *a)
 {
-	struct snd_card_asihpi *asihpi = from_tasklet(asihpi, t, t);
-	struct hpi_adapter *a = asihpi->hpi;
+	struct snd_card_asihpi *asihpi;
 
 	WARN_ON(!a || !a->snd_card || !a->snd_card->private_data);
 	asihpi = (struct snd_card_asihpi *)a->snd_card->private_data;
@@ -933,15 +914,6 @@ static void snd_card_asihpi_int_task(struct tasklet_struct *t)
 			&asihpi->llmode_streampriv->timer);
 }
 
-static void snd_card_asihpi_isr(struct hpi_adapter *a)
-{
-	struct snd_card_asihpi *asihpi;
-
-	WARN_ON(!a || !a->snd_card || !a->snd_card->private_data);
-	asihpi = (struct snd_card_asihpi *)a->snd_card->private_data;
-	tasklet_schedule(&asihpi->t);
-}
-
 /***************************** PLAYBACK OPS ****************/
 static int snd_card_asihpi_playback_prepare(struct snd_pcm_substream *
 					    substream)
@@ -2871,7 +2843,6 @@ static int snd_asihpi_probe(struct pci_dev *pci_dev,
 	if (hpi->interrupt_mode) {
 		asihpi->pcm_start = snd_card_asihpi_pcm_int_start;
 		asihpi->pcm_stop = snd_card_asihpi_pcm_int_stop;
-		tasklet_setup(&asihpi->t, snd_card_asihpi_int_task);
 		hpi->interrupt_callback = snd_card_asihpi_isr;
 	} else {
 		asihpi->pcm_start = snd_card_asihpi_pcm_timer_start;
@@ -2960,14 +2931,12 @@ __nodev:
 static void snd_asihpi_remove(struct pci_dev *pci_dev)
 {
 	struct hpi_adapter *hpi = pci_get_drvdata(pci_dev);
-	struct snd_card_asihpi *asihpi = hpi->snd_card->private_data;
 
 	/* Stop interrupts */
 	if (hpi->interrupt_mode) {
 		hpi->interrupt_callback = NULL;
 		hpi_handle_error(hpi_adapter_set_property(hpi->adapter->index,
 			HPI_ADAPTER_PROPERTY_IRQ_RATE, 0, 0));
-		tasklet_kill(&asihpi->t);
 	}
 
 	snd_card_free(hpi->snd_card);
diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c
index 9790f5108a16..bb31b7fe867d 100644
--- a/sound/pci/asihpi/hpioctl.c
+++ b/sound/pci/asihpi/hpioctl.c
@@ -329,11 +329,20 @@ static irqreturn_t asihpi_isr(int irq, void *dev_id)
 	   asihpi_irq_count, a->adapter->type, a->adapter->index); */
 
 	if (a->interrupt_callback)
-		a->interrupt_callback(a);
+		return IRQ_WAKE_THREAD;
 
 	return IRQ_HANDLED;
 }
 
+static irqreturn_t asihpi_isr_thread(int irq, void *dev_id)
+{
+	struct hpi_adapter *a = dev_id;
+
+	if (a->interrupt_callback)
+		a->interrupt_callback(a);
+	return IRQ_HANDLED;
+}
+
 int asihpi_adapter_probe(struct pci_dev *pci_dev,
 			 const struct pci_device_id *pci_id)
 {
@@ -478,8 +487,9 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev,
 		}
 
 		/* Note: request_irq calls asihpi_isr here */
-		if (request_irq(pci_dev->irq, asihpi_isr, IRQF_SHARED,
-				"asihpi", &adapters[adapter_index])) {
+		if (request_threaded_irq(pci_dev->irq, asihpi_isr,
+					 asihpi_isr_thread, IRQF_SHARED,
+					 "asihpi", &adapters[adapter_index])) {
 			dev_err(&pci_dev->dev, "request_irq(%d) failed\n",
 				pci_dev->irq);
 			goto err;
diff --git a/sound/pci/asihpi/hpios.h b/sound/pci/asihpi/hpios.h
index 26f7cf455a1e..9e551bc46264 100644
--- a/sound/pci/asihpi/hpios.h
+++ b/sound/pci/asihpi/hpios.h
@@ -67,7 +67,7 @@ struct hpi_ioctl_linux {
 };
 
 /* Conflict?: H is already used by a number of drivers hid, bluetooth hci,
-   and some sound drivers sb16, hdsp, emu10k. AFAIK 0xFC is ununsed command
+   and some sound drivers sb16, hdsp, emu10k. AFAIK 0xFC is unused command
 */
 #define HPI_IOCTL_LINUX _IOWR('H', 0xFC, struct hpi_ioctl_linux)
 
diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c
index 824f4ac1a8ce..4dc01647753c 100644
--- a/sound/pci/hda/hda_auto_parser.c
+++ b/sound/pci/hda/hda_auto_parser.c
@@ -350,7 +350,7 @@ int snd_hda_parse_pin_defcfg(struct hda_codec *codec,
 	 */
 	if (!cfg->line_outs && cfg->hp_outs > 1 &&
 	    !(cond_flags & HDA_PINCFG_NO_HP_FIXUP)) {
-		int i = 0;
+		i = 0;
 		while (i < cfg->hp_outs) {
 			/* The real HPs should have the sequence 0x0f */
 			if ((hp_out[i].seq & 0x0f) == 0x0f) {
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 36a9dbc33aa0..61e495187b1a 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -368,7 +368,8 @@ enum {
 #define CONTROLLER_IN_GPU(pci) (((pci)->device == 0x0a0c) || \
 					((pci)->device == 0x0c0c) || \
 					((pci)->device == 0x0d0c) || \
-					((pci)->device == 0x160c))
+					((pci)->device == 0x160c) || \
+					((pci)->device == 0x490d))
 
 #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98)
 
@@ -2493,6 +2494,9 @@ static const struct pci_device_id azx_ids[] = {
 	/* Tigerlake-H */
 	{ PCI_DEVICE(0x8086, 0x43c8),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
+	/* DG1 */
+	{ PCI_DEVICE(0x8086, 0x490d),
+	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
 	/* Elkhart Lake */
 	{ PCI_DEVICE(0x8086, 0x4b55),
 	  .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE},
diff --git a/sound/pci/hda/hda_jack.h b/sound/pci/hda/hda_jack.h
index 727b6d3ba454..8ceaf0ef5df1 100644
--- a/sound/pci/hda/hda_jack.h
+++ b/sound/pci/hda/hda_jack.h
@@ -77,7 +77,7 @@ int snd_hda_jack_detect_enable(struct hda_codec *codec, hda_nid_t nid,
 
 struct hda_jack_callback *
 snd_hda_jack_detect_enable_callback_mst(struct hda_codec *codec, hda_nid_t nid,
-					int dev_id, hda_jack_callback_fn cb);
+					int dev_id, hda_jack_callback_fn func);
 
 /**
  * snd_hda_jack_detect_enable - enable the jack-detection
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 8c28b1022f49..5beb8aa44ecd 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -100,7 +100,7 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol,
 int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol,
 				 struct snd_ctl_elem_value *ucontrol);
 int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag,
-			  unsigned int size, unsigned int __user *tlv);
+			  unsigned int size, unsigned int __user *_tlv);
 int snd_hda_mixer_amp_switch_info(struct snd_kcontrol *kcontrol,
 				  struct snd_ctl_elem_info *uinfo);
 int snd_hda_mixer_amp_switch_get(struct snd_kcontrol *kcontrol,
@@ -119,7 +119,7 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol,
 int snd_hda_codec_amp_update(struct hda_codec *codec, hda_nid_t nid,
 			     int ch, int dir, int idx, int mask, int val);
 int snd_hda_codec_amp_stereo(struct hda_codec *codec, hda_nid_t nid,
-			     int dir, int idx, int mask, int val);
+			     int direction, int idx, int mask, int val);
 int snd_hda_codec_amp_init(struct hda_codec *codec, hda_nid_t nid, int ch,
 			   int direction, int idx, int mask, int val);
 int snd_hda_codec_amp_init_stereo(struct hda_codec *codec, hda_nid_t nid,
@@ -198,7 +198,7 @@ int snd_hda_input_mux_put(struct hda_codec *codec,
 			  unsigned int *cur_val);
 int snd_hda_add_imux_item(struct hda_codec *codec,
 			  struct hda_input_mux *imux, const char *label,
-			  int index, int *type_index_ret);
+			  int index, int *type_idx);
 
 /*
  * Multi-channel / digital-out PCM helper
@@ -642,7 +642,7 @@ unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec,
  */
 int snd_hda_enum_helper_info(struct snd_kcontrol *kcontrol,
 			     struct snd_ctl_elem_info *uinfo,
-			     int num_entries, const char * const *texts);
+			     int num_items, const char * const *texts);
 #define snd_hda_enum_bool_helper_info(kcontrol, uinfo) \
 	snd_hda_enum_helper_info(kcontrol, uinfo, 0, NULL)
 
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index b7dbf2e7f77a..9779978e4bc7 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -38,6 +38,8 @@
 #define FLOAT_ONE	0x3f800000
 #define FLOAT_TWO	0x40000000
 #define FLOAT_THREE     0x40400000
+#define FLOAT_FIVE	0x40a00000
+#define FLOAT_SIX       0x40c00000
 #define FLOAT_EIGHT     0x41000000
 #define FLOAT_MINUS_5	0xc0a00000
 
@@ -80,11 +82,11 @@ MODULE_FIRMWARE(R3DI_EFX_FILE);
 
 static const char *const dirstr[2] = { "Playback", "Capture" };
 
-#define NUM_OF_OUTPUTS 3
+#define NUM_OF_OUTPUTS 2
+static const char *const out_type_str[2] = { "Speakers", "Headphone" };
 enum {
 	SPEAKER_OUT,
 	HEADPHONE_OUT,
-	SURROUND_OUT
 };
 
 enum {
@@ -143,7 +145,12 @@ enum {
 	MIC_BOOST_ENUM,
 	AE5_HEADPHONE_GAIN_ENUM,
 	AE5_SOUND_FILTER_ENUM,
-	ZXR_HEADPHONE_GAIN
+	ZXR_HEADPHONE_GAIN,
+	SPEAKER_CHANNEL_CFG_ENUM,
+	SPEAKER_FULL_RANGE_FRONT,
+	SPEAKER_FULL_RANGE_REAR,
+	BASS_REDIRECTION,
+	BASS_REDIRECTION_XOVER,
 #define EFFECTS_COUNT  (EFFECT_END_NID - EFFECT_START_NID)
 };
 
@@ -589,46 +596,108 @@ static const struct ct_eq_preset ca0132_alt_eq_presets[] = {
 	}
 };
 
-/* DSP command sequences for ca0132_alt_select_out */
-#define ALT_OUT_SET_MAX_COMMANDS 9 /* Max number of commands in sequence */
-struct ca0132_alt_out_set {
-	char *name; /*preset name*/
-	unsigned char commands;
-	unsigned int mids[ALT_OUT_SET_MAX_COMMANDS];
-	unsigned int reqs[ALT_OUT_SET_MAX_COMMANDS];
-	unsigned int vals[ALT_OUT_SET_MAX_COMMANDS];
+/*
+ * DSP reqs for handling full-range speakers/bass redirection. If a speaker is
+ * set as not being full range, and bass redirection is enabled, all
+ * frequencies below the crossover frequency are redirected to the LFE
+ * channel. If the surround configuration has no LFE channel, this can't be
+ * enabled. X-Bass must be disabled when using these.
+ */
+enum speaker_range_reqs {
+	SPEAKER_BASS_REDIRECT            = 0x15,
+	SPEAKER_BASS_REDIRECT_XOVER_FREQ = 0x16,
+	/* Between 0x16-0x1a are the X-Bass reqs. */
+	SPEAKER_FULL_RANGE_FRONT_L_R     = 0x1a,
+	SPEAKER_FULL_RANGE_CENTER_LFE    = 0x1b,
+	SPEAKER_FULL_RANGE_REAR_L_R      = 0x1c,
+	SPEAKER_FULL_RANGE_SURROUND_L_R  = 0x1d,
+	SPEAKER_BASS_REDIRECT_SUB_GAIN   = 0x1e,
+};
+
+/*
+ * Definitions for the DSP req's to handle speaker tuning. These all belong to
+ * module ID 0x96, the output effects module.
+ */
+enum speaker_tuning_reqs {
+	/*
+	 * Currently, this value is always set to 0.0f. However, on Windows,
+	 * when selecting certain headphone profiles on the new Sound Blaster
+	 * connect software, the QUERY_SPEAKER_EQ_ADDRESS req on mid 0x80 is
+	 * sent. This gets the speaker EQ address area, which is then used to
+	 * send over (presumably) an equalizer profile for the specific
+	 * headphone setup. It is sent using the same method the DSP
+	 * firmware is uploaded with, which I believe is why the 'ctspeq.bin'
+	 * file exists in linux firmware tree but goes unused. It would also
+	 * explain why the QUERY_SPEAKER_EQ_ADDRESS req is defined but unused.
+	 * Once this profile is sent over, SPEAKER_TUNING_USE_SPEAKER_EQ is
+	 * set to 1.0f.
+	 */
+	SPEAKER_TUNING_USE_SPEAKER_EQ           = 0x1f,
+	SPEAKER_TUNING_ENABLE_CENTER_EQ         = 0x20,
+	SPEAKER_TUNING_FRONT_LEFT_VOL_LEVEL     = 0x21,
+	SPEAKER_TUNING_FRONT_RIGHT_VOL_LEVEL    = 0x22,
+	SPEAKER_TUNING_CENTER_VOL_LEVEL         = 0x23,
+	SPEAKER_TUNING_LFE_VOL_LEVEL            = 0x24,
+	SPEAKER_TUNING_REAR_LEFT_VOL_LEVEL      = 0x25,
+	SPEAKER_TUNING_REAR_RIGHT_VOL_LEVEL     = 0x26,
+	SPEAKER_TUNING_SURROUND_LEFT_VOL_LEVEL  = 0x27,
+	SPEAKER_TUNING_SURROUND_RIGHT_VOL_LEVEL = 0x28,
+	/*
+	 * Inversion is used when setting headphone virtualization to line
+	 * out. Not sure why this is, but it's the only place it's ever used.
+	 */
+	SPEAKER_TUNING_FRONT_LEFT_INVERT        = 0x29,
+	SPEAKER_TUNING_FRONT_RIGHT_INVERT       = 0x2a,
+	SPEAKER_TUNING_CENTER_INVERT            = 0x2b,
+	SPEAKER_TUNING_LFE_INVERT               = 0x2c,
+	SPEAKER_TUNING_REAR_LEFT_INVERT         = 0x2d,
+	SPEAKER_TUNING_REAR_RIGHT_INVERT        = 0x2e,
+	SPEAKER_TUNING_SURROUND_LEFT_INVERT     = 0x2f,
+	SPEAKER_TUNING_SURROUND_RIGHT_INVERT    = 0x30,
+	/* Delay is used when setting surround speaker distance in Windows. */
+	SPEAKER_TUNING_FRONT_LEFT_DELAY         = 0x31,
+	SPEAKER_TUNING_FRONT_RIGHT_DELAY        = 0x32,
+	SPEAKER_TUNING_CENTER_DELAY             = 0x33,
+	SPEAKER_TUNING_LFE_DELAY                = 0x34,
+	SPEAKER_TUNING_REAR_LEFT_DELAY          = 0x35,
+	SPEAKER_TUNING_REAR_RIGHT_DELAY         = 0x36,
+	SPEAKER_TUNING_SURROUND_LEFT_DELAY      = 0x37,
+	SPEAKER_TUNING_SURROUND_RIGHT_DELAY     = 0x38,
+	/* Of these two, only mute seems to ever be used. */
+	SPEAKER_TUNING_MAIN_VOLUME              = 0x39,
+	SPEAKER_TUNING_MUTE                     = 0x3a,
+};
+
+/* Surround output channel count configuration structures. */
+#define SPEAKER_CHANNEL_CFG_COUNT 5
+enum {
+	SPEAKER_CHANNELS_2_0,
+	SPEAKER_CHANNELS_2_1,
+	SPEAKER_CHANNELS_4_0,
+	SPEAKER_CHANNELS_4_1,
+	SPEAKER_CHANNELS_5_1,
+};
+
+struct ca0132_alt_speaker_channel_cfg {
+	char *name;
+	unsigned int val;
 };
 
-static const struct ca0132_alt_out_set alt_out_presets[] = {
-	{ .name = "Line Out",
-	  .commands = 7,
-	  .mids = { 0x96, 0x96, 0x96, 0x8F,
-		    0x96, 0x96, 0x96 },
-	  .reqs = { 0x19, 0x17, 0x18, 0x01,
-		    0x1F, 0x15, 0x3A },
-	  .vals = { 0x3F000000, 0x42A00000, 0x00000000,
-		    0x00000000, 0x00000000, 0x00000000,
-		    0x00000000 }
+static const struct ca0132_alt_speaker_channel_cfg speaker_channel_cfgs[] = {
+	{ .name = "2.0",
+	  .val = FLOAT_ONE
 	},
-	{ .name = "Headphone",
-	  .commands = 7,
-	  .mids = { 0x96, 0x96, 0x96, 0x8F,
-		    0x96, 0x96, 0x96 },
-	  .reqs = { 0x19, 0x17, 0x18, 0x01,
-		    0x1F, 0x15, 0x3A },
-	  .vals = { 0x3F000000, 0x42A00000, 0x00000000,
-		    0x00000000, 0x00000000, 0x00000000,
-		    0x00000000 }
+	{ .name = "2.1",
+	  .val = FLOAT_TWO
 	},
-	{ .name = "Surround",
-	  .commands = 8,
-	  .mids = { 0x96, 0x8F, 0x96, 0x96,
-		    0x96, 0x96, 0x96, 0x96 },
-	  .reqs = { 0x18, 0x01, 0x1F, 0x15,
-		    0x3A, 0x1A, 0x1B, 0x1C },
-	  .vals = { 0x00000000, 0x00000000, 0x00000000,
-		    0x00000000, 0x00000000, 0x00000000,
-		    0x00000000, 0x00000000 }
+	{ .name = "4.0",
+	  .val = FLOAT_FIVE
+	},
+	{ .name = "4.1",
+	  .val = FLOAT_SIX
+	},
+	{ .name = "5.1",
+	  .val = FLOAT_EIGHT
 	}
 };
 
@@ -658,26 +727,29 @@ static const struct ct_dsp_volume_ctl ca0132_alt_vol_ctls[] = {
 };
 
 /* Values for ca0113_mmio_command_set for selecting output. */
-#define AE5_CA0113_OUT_SET_COMMANDS 6
-struct ae5_ca0113_output_set {
-	unsigned int group[AE5_CA0113_OUT_SET_COMMANDS];
-	unsigned int target[AE5_CA0113_OUT_SET_COMMANDS];
-	unsigned int vals[AE5_CA0113_OUT_SET_COMMANDS];
+#define AE_CA0113_OUT_SET_COMMANDS 6
+struct ae_ca0113_output_set {
+	unsigned int group[AE_CA0113_OUT_SET_COMMANDS];
+	unsigned int target[AE_CA0113_OUT_SET_COMMANDS];
+	unsigned int vals[NUM_OF_OUTPUTS][AE_CA0113_OUT_SET_COMMANDS];
 };
 
-static const struct ae5_ca0113_output_set ae5_ca0113_output_presets[] = {
-	{ .group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
-	  .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
-	  .vals =   { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f }
-	},
-	{ .group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
-	  .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
-	  .vals =   { 0x3f, 0x3f, 0x00, 0x00, 0x00, 0x00 }
-	},
-	{ .group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
-	  .target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
-	  .vals =   { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f }
-	}
+static const struct ae_ca0113_output_set ae5_ca0113_output_presets = {
+	.group =  { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
+	.target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
+		    /* Speakers. */
+	.vals =   { { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f },
+		    /* Headphones. */
+		    { 0x3f, 0x3f, 0x00, 0x00, 0x00, 0x00 } },
+};
+
+static const struct ae_ca0113_output_set ae7_ca0113_output_presets = {
+	.group  = { 0x30, 0x30, 0x48, 0x48, 0x48, 0x30 },
+	.target = { 0x2e, 0x30, 0x0d, 0x17, 0x19, 0x32 },
+		    /* Speakers. */
+	.vals   = { { 0x00, 0x00, 0x40, 0x00, 0x00, 0x3f },
+		    /* Headphones. */
+		    { 0x3f, 0x3f, 0x00, 0x00, 0x02, 0x00 } },
 };
 
 /* ae5 ca0113 command sequences to set headphone gain levels. */
@@ -1009,8 +1081,12 @@ struct ca0132_spec {
 	/* ca0132_alt control related values */
 	unsigned char in_enum_val;
 	unsigned char out_enum_val;
+	unsigned char channel_cfg_val;
+	unsigned char speaker_range_val[2];
 	unsigned char mic_boost_enum_val;
 	unsigned char smart_volume_setting;
+	unsigned char bass_redirection_val;
+	long bass_redirect_xover_freq;
 	long fx_ctl_val[EFFECT_LEVEL_SLIDERS];
 	long xbass_xover_freq;
 	long eq_preset_val;
@@ -1065,6 +1141,7 @@ enum {
 	QUIRK_R3DI,
 	QUIRK_R3D,
 	QUIRK_AE5,
+	QUIRK_AE7,
 };
 
 #ifdef CONFIG_PCI
@@ -1168,6 +1245,20 @@ static const struct hda_pintbl r3di_pincfgs[] = {
 	{}
 };
 
+static const struct hda_pintbl ae7_pincfgs[] = {
+	{ 0x0b, 0x01017010 },
+	{ 0x0c, 0x014510f0 },
+	{ 0x0d, 0x414510f0 },
+	{ 0x0e, 0x01c520f0 },
+	{ 0x0f, 0x01017114 },
+	{ 0x10, 0x01017011 },
+	{ 0x11, 0x018170ff },
+	{ 0x12, 0x01a170f0 },
+	{ 0x13, 0x908700f0 },
+	{ 0x18, 0x500000f0 },
+	{}
+};
+
 static const struct snd_pci_quirk ca0132_quirks[] = {
 	SND_PCI_QUIRK(0x1028, 0x057b, "Alienware M17x R4", QUIRK_ALIENWARE_M17XR4),
 	SND_PCI_QUIRK(0x1028, 0x0685, "Alienware 15 2015", QUIRK_ALIENWARE),
@@ -1184,9 +1275,203 @@ static const struct snd_pci_quirk ca0132_quirks[] = {
 	SND_PCI_QUIRK(0x1102, 0x0013, "Recon3D", QUIRK_R3D),
 	SND_PCI_QUIRK(0x1102, 0x0018, "Recon3D", QUIRK_R3D),
 	SND_PCI_QUIRK(0x1102, 0x0051, "Sound Blaster AE-5", QUIRK_AE5),
+	SND_PCI_QUIRK(0x1102, 0x0081, "Sound Blaster AE-7", QUIRK_AE7),
 	{}
 };
 
+/* Output selection quirk info structures. */
+#define MAX_QUIRK_MMIO_GPIO_SET_VALS 3
+#define MAX_QUIRK_SCP_SET_VALS 2
+struct ca0132_alt_out_set_info {
+	unsigned int dac2port; /* ParamID 0x0d value. */
+
+	bool has_hda_gpio;
+	char hda_gpio_pin;
+	char hda_gpio_set;
+
+	unsigned int mmio_gpio_count;
+	char mmio_gpio_pin[MAX_QUIRK_MMIO_GPIO_SET_VALS];
+	char mmio_gpio_set[MAX_QUIRK_MMIO_GPIO_SET_VALS];
+
+	unsigned int scp_cmds_count;
+	unsigned int scp_cmd_mid[MAX_QUIRK_SCP_SET_VALS];
+	unsigned int scp_cmd_req[MAX_QUIRK_SCP_SET_VALS];
+	unsigned int scp_cmd_val[MAX_QUIRK_SCP_SET_VALS];
+
+	bool has_chipio_write;
+	unsigned int chipio_write_addr;
+	unsigned int chipio_write_data;
+};
+
+struct ca0132_alt_out_set_quirk_data {
+	int quirk_id;
+
+	bool has_headphone_gain;
+	bool is_ae_series;
+
+	struct ca0132_alt_out_set_info out_set_info[NUM_OF_OUTPUTS];
+};
+
+static const struct ca0132_alt_out_set_quirk_data quirk_out_set_data[] = {
+	{ .quirk_id = QUIRK_R3DI,
+	  .has_headphone_gain = false,
+	  .is_ae_series       = false,
+	  .out_set_info = {
+		/* Speakers. */
+		{ .dac2port         = 0x24,
+		  .has_hda_gpio     = true,
+		  .hda_gpio_pin     = 2,
+		  .hda_gpio_set     = 1,
+		  .mmio_gpio_count  = 0,
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		},
+		/* Headphones. */
+		{ .dac2port         = 0x21,
+		  .has_hda_gpio     = true,
+		  .hda_gpio_pin     = 2,
+		  .hda_gpio_set     = 0,
+		  .mmio_gpio_count  = 0,
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		} },
+	},
+	{ .quirk_id = QUIRK_R3D,
+	  .has_headphone_gain = false,
+	  .is_ae_series       = false,
+	  .out_set_info = {
+		/* Speakers. */
+		{ .dac2port         = 0x24,
+		  .has_hda_gpio     = false,
+		  .mmio_gpio_count  = 1,
+		  .mmio_gpio_pin    = { 1 },
+		  .mmio_gpio_set    = { 1 },
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		},
+		/* Headphones. */
+		{ .dac2port         = 0x21,
+		  .has_hda_gpio     = false,
+		  .mmio_gpio_count  = 1,
+		  .mmio_gpio_pin    = { 1 },
+		  .mmio_gpio_set    = { 0 },
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		} },
+	},
+	{ .quirk_id = QUIRK_SBZ,
+	  .has_headphone_gain = false,
+	  .is_ae_series       = false,
+	  .out_set_info = {
+		/* Speakers. */
+		{ .dac2port         = 0x18,
+		  .has_hda_gpio     = false,
+		  .mmio_gpio_count  = 3,
+		  .mmio_gpio_pin    = { 7, 4, 1 },
+		  .mmio_gpio_set    = { 0, 1, 1 },
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false, },
+		/* Headphones. */
+		{ .dac2port         = 0x12,
+		  .has_hda_gpio     = false,
+		  .mmio_gpio_count  = 3,
+		  .mmio_gpio_pin    = { 7, 4, 1 },
+		  .mmio_gpio_set    = { 1, 1, 0 },
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		} },
+	},
+	{ .quirk_id = QUIRK_ZXR,
+	  .has_headphone_gain = true,
+	  .is_ae_series       = false,
+	  .out_set_info = {
+		/* Speakers. */
+		{ .dac2port         = 0x24,
+		  .has_hda_gpio     = false,
+		  .mmio_gpio_count  = 3,
+		  .mmio_gpio_pin    = { 2, 3, 5 },
+		  .mmio_gpio_set    = { 1, 1, 0 },
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		},
+		/* Headphones. */
+		{ .dac2port         = 0x21,
+		  .has_hda_gpio     = false,
+		  .mmio_gpio_count  = 3,
+		  .mmio_gpio_pin    = { 2, 3, 5 },
+		  .mmio_gpio_set    = { 0, 1, 1 },
+		  .scp_cmds_count   = 0,
+		  .has_chipio_write = false,
+		} },
+	},
+	{ .quirk_id = QUIRK_AE5,
+	  .has_headphone_gain = true,
+	  .is_ae_series       = true,
+	  .out_set_info = {
+		/* Speakers. */
+		{ .dac2port          = 0xa4,
+		  .has_hda_gpio      = false,
+		  .mmio_gpio_count   = 0,
+		  .scp_cmds_count    = 2,
+		  .scp_cmd_mid       = { 0x96, 0x96 },
+		  .scp_cmd_req       = { SPEAKER_TUNING_FRONT_LEFT_INVERT,
+					 SPEAKER_TUNING_FRONT_RIGHT_INVERT },
+		  .scp_cmd_val       = { FLOAT_ZERO, FLOAT_ZERO },
+		  .has_chipio_write  = true,
+		  .chipio_write_addr = 0x0018b03c,
+		  .chipio_write_data = 0x00000012
+		},
+		/* Headphones. */
+		{ .dac2port          = 0xa1,
+		  .has_hda_gpio      = false,
+		  .mmio_gpio_count   = 0,
+		  .scp_cmds_count    = 2,
+		  .scp_cmd_mid       = { 0x96, 0x96 },
+		  .scp_cmd_req       = { SPEAKER_TUNING_FRONT_LEFT_INVERT,
+					 SPEAKER_TUNING_FRONT_RIGHT_INVERT },
+		  .scp_cmd_val       = { FLOAT_ONE, FLOAT_ONE },
+		  .has_chipio_write  = true,
+		  .chipio_write_addr = 0x0018b03c,
+		  .chipio_write_data = 0x00000012
+		} },
+	},
+	{ .quirk_id = QUIRK_AE7,
+	  .has_headphone_gain = true,
+	  .is_ae_series       = true,
+	  .out_set_info = {
+		/* Speakers. */
+		{ .dac2port          = 0x58,
+		  .has_hda_gpio      = false,
+		  .mmio_gpio_count   = 1,
+		  .mmio_gpio_pin     = { 0 },
+		  .mmio_gpio_set     = { 1 },
+		  .scp_cmds_count    = 2,
+		  .scp_cmd_mid       = { 0x96, 0x96 },
+		  .scp_cmd_req       = { SPEAKER_TUNING_FRONT_LEFT_INVERT,
+					 SPEAKER_TUNING_FRONT_RIGHT_INVERT },
+		  .scp_cmd_val       = { FLOAT_ZERO, FLOAT_ZERO },
+		  .has_chipio_write  = true,
+		  .chipio_write_addr = 0x0018b03c,
+		  .chipio_write_data = 0x00000000
+		},
+		/* Headphones. */
+		{ .dac2port          = 0x58,
+		  .has_hda_gpio      = false,
+		  .mmio_gpio_count   = 1,
+		  .mmio_gpio_pin     = { 0 },
+		  .mmio_gpio_set     = { 1 },
+		  .scp_cmds_count    = 2,
+		  .scp_cmd_mid       = { 0x96, 0x96 },
+		  .scp_cmd_req       = { SPEAKER_TUNING_FRONT_LEFT_INVERT,
+					 SPEAKER_TUNING_FRONT_RIGHT_INVERT },
+		  .scp_cmd_val       = { FLOAT_ONE, FLOAT_ONE },
+		  .has_chipio_write  = true,
+		  .chipio_write_addr = 0x0018b03c,
+		  .chipio_write_data = 0x00000010
+		} },
+	}
+};
+
 /*
  * CA0132 codec access
  */
@@ -3339,6 +3624,7 @@ static void ca0132_gpio_init(struct hda_codec *codec)
 	switch (ca0132_quirk(spec)) {
 	case QUIRK_SBZ:
 	case QUIRK_AE5:
+	case QUIRK_AE7:
 		snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
 		snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53);
 		snd_hda_codec_write(codec, 0x01, 0, 0x790, 0x23);
@@ -3444,26 +3730,6 @@ static void r3di_gpio_mic_set(struct hda_codec *codec,
 			    AC_VERB_SET_GPIO_DATA, cur_gpio);
 }
 
-static void r3di_gpio_out_set(struct hda_codec *codec,
-		enum r3di_out_select cur_out)
-{
-	unsigned int cur_gpio;
-
-	/* Get the current GPIO Data setup */
-	cur_gpio = snd_hda_codec_read(codec, 0x01, 0, AC_VERB_GET_GPIO_DATA, 0);
-
-	switch (cur_out) {
-	case R3DI_HEADPHONE_OUT:
-		cur_gpio &= ~(1 << R3DI_OUT_SELECT_BIT);
-		break;
-	case R3DI_LINE_OUT:
-		cur_gpio |= (1 << R3DI_OUT_SELECT_BIT);
-		break;
-	}
-	snd_hda_codec_write(codec, codec->core.afg, 0,
-			    AC_VERB_SET_GPIO_DATA, cur_gpio);
-}
-
 static void r3di_gpio_dsp_status_set(struct hda_codec *codec,
 		enum r3di_dsp_status dsp_status)
 {
@@ -4159,135 +4425,198 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val);
 static void ae5_mmio_select_out(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
+	const struct ae_ca0113_output_set *out_cmds;
 	unsigned int i;
 
-	for (i = 0; i < AE5_CA0113_OUT_SET_COMMANDS; i++)
-		ca0113_mmio_command_set(codec,
-			ae5_ca0113_output_presets[spec->cur_out_type].group[i],
-			ae5_ca0113_output_presets[spec->cur_out_type].target[i],
-			ae5_ca0113_output_presets[spec->cur_out_type].vals[i]);
+	if (ca0132_quirk(spec) == QUIRK_AE5)
+		out_cmds = &ae5_ca0113_output_presets;
+	else
+		out_cmds = &ae7_ca0113_output_presets;
+
+	for (i = 0; i < AE_CA0113_OUT_SET_COMMANDS; i++)
+		ca0113_mmio_command_set(codec, out_cmds->group[i],
+				out_cmds->target[i],
+				out_cmds->vals[spec->cur_out_type][i]);
+}
+
+static int ca0132_alt_set_full_range_speaker(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	int quirk = ca0132_quirk(spec);
+	unsigned int tmp;
+	int err;
+
+	/* 2.0/4.0 setup has no LFE channel, so setting full-range does nothing. */
+	if (spec->channel_cfg_val == SPEAKER_CHANNELS_4_0
+			|| spec->channel_cfg_val == SPEAKER_CHANNELS_2_0)
+		return 0;
+
+	/* Set front L/R full range. Zero for full-range, one for redirection. */
+	tmp = spec->speaker_range_val[0] ? FLOAT_ZERO : FLOAT_ONE;
+	err = dspio_set_uint_param(codec, 0x96,
+			SPEAKER_FULL_RANGE_FRONT_L_R, tmp);
+	if (err < 0)
+		return err;
+
+	/* When setting full-range rear, both rear and center/lfe are set. */
+	tmp = spec->speaker_range_val[1] ? FLOAT_ZERO : FLOAT_ONE;
+	err = dspio_set_uint_param(codec, 0x96,
+			SPEAKER_FULL_RANGE_CENTER_LFE, tmp);
+	if (err < 0)
+		return err;
+
+	err = dspio_set_uint_param(codec, 0x96,
+			SPEAKER_FULL_RANGE_REAR_L_R, tmp);
+	if (err < 0)
+		return err;
+
+	/*
+	 * Only the AE series cards set this value when setting full-range,
+	 * and it's always 1.0f.
+	 */
+	if (quirk == QUIRK_AE5 || quirk == QUIRK_AE7) {
+		err = dspio_set_uint_param(codec, 0x96,
+				SPEAKER_FULL_RANGE_SURROUND_L_R, FLOAT_ONE);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
+}
+
+static int ca0132_alt_surround_set_bass_redirection(struct hda_codec *codec,
+		bool val)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int tmp;
+	int err;
+
+	if (val && spec->channel_cfg_val != SPEAKER_CHANNELS_4_0 &&
+			spec->channel_cfg_val != SPEAKER_CHANNELS_2_0)
+		tmp = FLOAT_ONE;
+	else
+		tmp = FLOAT_ZERO;
+
+	err = dspio_set_uint_param(codec, 0x96, SPEAKER_BASS_REDIRECT, tmp);
+	if (err < 0)
+		return err;
+
+	/* If it is enabled, make sure to set the crossover frequency. */
+	if (tmp) {
+		tmp = float_xbass_xover_lookup[spec->xbass_xover_freq];
+		err = dspio_set_uint_param(codec, 0x96,
+				SPEAKER_BASS_REDIRECT_XOVER_FREQ, tmp);
+		if (err < 0)
+			return err;
+	}
+
+	return 0;
 }
 
 /*
  * These are the commands needed to setup output on each of the different card
  * types.
  */
-static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec)
+static void ca0132_alt_select_out_get_quirk_data(struct hda_codec *codec,
+		const struct ca0132_alt_out_set_quirk_data **quirk_data)
 {
 	struct ca0132_spec *spec = codec->spec;
-	unsigned int tmp;
+	int quirk = ca0132_quirk(spec);
+	unsigned int i;
 
-	switch (spec->cur_out_type) {
-	case SPEAKER_OUT:
-		switch (ca0132_quirk(spec)) {
-		case QUIRK_SBZ:
-			ca0113_mmio_gpio_set(codec, 7, false);
-			ca0113_mmio_gpio_set(codec, 4, true);
-			ca0113_mmio_gpio_set(codec, 1, true);
-			chipio_set_control_param(codec, 0x0d, 0x18);
-			break;
-		case QUIRK_ZXR:
-			ca0113_mmio_gpio_set(codec, 2, true);
-			ca0113_mmio_gpio_set(codec, 3, true);
-			ca0113_mmio_gpio_set(codec, 5, false);
-			zxr_headphone_gain_set(codec, 0);
-			chipio_set_control_param(codec, 0x0d, 0x24);
-			break;
-		case QUIRK_R3DI:
-			chipio_set_control_param(codec, 0x0d, 0x24);
-			r3di_gpio_out_set(codec, R3DI_LINE_OUT);
-			break;
-		case QUIRK_R3D:
-			chipio_set_control_param(codec, 0x0d, 0x24);
-			ca0113_mmio_gpio_set(codec, 1, true);
-			break;
-		case QUIRK_AE5:
-			ae5_mmio_select_out(codec);
-			ae5_headphone_gain_set(codec, 2);
-			tmp = FLOAT_ZERO;
-			dspio_set_uint_param(codec, 0x96, 0x29, tmp);
-			dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
-			chipio_set_control_param(codec, 0x0d, 0xa4);
-			chipio_write(codec, 0x18b03c, 0x00000012);
-			break;
-		default:
-			break;
+	*quirk_data = NULL;
+	for (i = 0; i < ARRAY_SIZE(quirk_out_set_data); i++) {
+		if (quirk_out_set_data[i].quirk_id == quirk) {
+			*quirk_data = &quirk_out_set_data[i];
+			return;
 		}
-		break;
-	case HEADPHONE_OUT:
-		switch (ca0132_quirk(spec)) {
-		case QUIRK_SBZ:
-			ca0113_mmio_gpio_set(codec, 7, true);
-			ca0113_mmio_gpio_set(codec, 4, true);
-			ca0113_mmio_gpio_set(codec, 1, false);
-			chipio_set_control_param(codec, 0x0d, 0x12);
-			break;
-		case QUIRK_ZXR:
-			ca0113_mmio_gpio_set(codec, 2, false);
-			ca0113_mmio_gpio_set(codec, 3, false);
-			ca0113_mmio_gpio_set(codec, 5, true);
-			zxr_headphone_gain_set(codec, spec->zxr_gain_set);
-			chipio_set_control_param(codec, 0x0d, 0x21);
-			break;
-		case QUIRK_R3DI:
-			chipio_set_control_param(codec, 0x0d, 0x21);
-			r3di_gpio_out_set(codec, R3DI_HEADPHONE_OUT);
-			break;
-		case QUIRK_R3D:
-			chipio_set_control_param(codec, 0x0d, 0x21);
-			ca0113_mmio_gpio_set(codec, 0x1, false);
-			break;
-		case QUIRK_AE5:
-			ae5_mmio_select_out(codec);
-			ae5_headphone_gain_set(codec,
-					spec->ae5_headphone_gain_val);
-			tmp = FLOAT_ONE;
-			dspio_set_uint_param(codec, 0x96, 0x29, tmp);
-			dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
-			chipio_set_control_param(codec, 0x0d, 0xa1);
-			chipio_write(codec, 0x18b03c, 0x00000012);
-			break;
-		default:
-			break;
+	}
+}
+
+static int ca0132_alt_select_out_quirk_set(struct hda_codec *codec)
+{
+	const struct ca0132_alt_out_set_quirk_data *quirk_data;
+	const struct ca0132_alt_out_set_info *out_info;
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int i, gpio_data;
+	int err;
+
+	ca0132_alt_select_out_get_quirk_data(codec, &quirk_data);
+	if (!quirk_data)
+		return 0;
+
+	out_info = &quirk_data->out_set_info[spec->cur_out_type];
+	if (quirk_data->is_ae_series)
+		ae5_mmio_select_out(codec);
+
+	if (out_info->has_hda_gpio) {
+		gpio_data = snd_hda_codec_read(codec, codec->core.afg, 0,
+				AC_VERB_GET_GPIO_DATA, 0);
+
+		if (out_info->hda_gpio_set)
+			gpio_data |= (1 << out_info->hda_gpio_pin);
+		else
+			gpio_data &= ~(1 << out_info->hda_gpio_pin);
+
+		snd_hda_codec_write(codec, codec->core.afg, 0,
+				    AC_VERB_SET_GPIO_DATA, gpio_data);
+	}
+
+	if (out_info->mmio_gpio_count) {
+		for (i = 0; i < out_info->mmio_gpio_count; i++) {
+			ca0113_mmio_gpio_set(codec, out_info->mmio_gpio_pin[i],
+					out_info->mmio_gpio_set[i]);
 		}
-		break;
-	case SURROUND_OUT:
-		switch (ca0132_quirk(spec)) {
-		case QUIRK_SBZ:
-			ca0113_mmio_gpio_set(codec, 7, false);
-			ca0113_mmio_gpio_set(codec, 4, true);
-			ca0113_mmio_gpio_set(codec, 1, true);
-			chipio_set_control_param(codec, 0x0d, 0x18);
-			break;
-		case QUIRK_ZXR:
-			ca0113_mmio_gpio_set(codec, 2, true);
-			ca0113_mmio_gpio_set(codec, 3, true);
-			ca0113_mmio_gpio_set(codec, 5, false);
-			zxr_headphone_gain_set(codec, 0);
-			chipio_set_control_param(codec, 0x0d, 0x24);
-			break;
-		case QUIRK_R3DI:
-			chipio_set_control_param(codec, 0x0d, 0x24);
-			r3di_gpio_out_set(codec, R3DI_LINE_OUT);
-			break;
-		case QUIRK_R3D:
-			ca0113_mmio_gpio_set(codec, 1, true);
-			chipio_set_control_param(codec, 0x0d, 0x24);
-			break;
-		case QUIRK_AE5:
-			ae5_mmio_select_out(codec);
-			ae5_headphone_gain_set(codec, 2);
-			tmp = FLOAT_ZERO;
-			dspio_set_uint_param(codec, 0x96, 0x29, tmp);
-			dspio_set_uint_param(codec, 0x96, 0x2a, tmp);
-			chipio_set_control_param(codec, 0x0d, 0xa4);
-			chipio_write(codec, 0x18b03c, 0x00000012);
-			break;
-		default:
-			break;
+	}
+
+	if (out_info->scp_cmds_count) {
+		for (i = 0; i < out_info->scp_cmds_count; i++) {
+			err = dspio_set_uint_param(codec,
+					out_info->scp_cmd_mid[i],
+					out_info->scp_cmd_req[i],
+					out_info->scp_cmd_val[i]);
+			if (err < 0)
+				return err;
 		}
-		break;
 	}
+
+	chipio_set_control_param(codec, 0x0d, out_info->dac2port);
+
+	if (out_info->has_chipio_write) {
+		chipio_write(codec, out_info->chipio_write_addr,
+				out_info->chipio_write_data);
+	}
+
+	if (quirk_data->has_headphone_gain) {
+		if (spec->cur_out_type != HEADPHONE_OUT) {
+			if (quirk_data->is_ae_series)
+				ae5_headphone_gain_set(codec, 2);
+			else
+				zxr_headphone_gain_set(codec, 0);
+		} else {
+			if (quirk_data->is_ae_series)
+				ae5_headphone_gain_set(codec,
+						spec->ae5_headphone_gain_val);
+			else
+				zxr_headphone_gain_set(codec,
+						spec->zxr_gain_set);
+		}
+	}
+
+	return 0;
+}
+
+static void ca0132_set_out_node_pincfg(struct hda_codec *codec, hda_nid_t nid,
+		bool out_enable, bool hp_enable)
+{
+	unsigned int pin_ctl;
+
+	pin_ctl = snd_hda_codec_read(codec, nid, 0,
+			AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
+
+	pin_ctl = hp_enable ? pin_ctl | PIN_HP_AMP : pin_ctl & ~PIN_HP_AMP;
+	pin_ctl = out_enable ? pin_ctl | PIN_OUT : pin_ctl & ~PIN_OUT;
+	snd_hda_set_pin_ctl(codec, nid, pin_ctl);
 }
 
 /*
@@ -4296,18 +4625,14 @@ static void ca0132_alt_select_out_quirk_handler(struct hda_codec *codec)
  * output with an enumerated control "output source" if the auto detect
  * mute switch is set to off. If the auto detect mute switch is enabled, it
  * will detect either headphone or lineout(SPEAKER_OUT) from jack detection.
- * It also adds the ability to auto-detect the front headphone port. The only
- * way to select surround is to disable auto detect, and set Surround with the
- * enumerated control.
+ * It also adds the ability to auto-detect the front headphone port.
  */
 static int ca0132_alt_select_out(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
-	unsigned int pin_ctl;
+	unsigned int tmp, outfx_set;
 	int jack_present;
 	int auto_jack;
-	unsigned int i;
-	unsigned int tmp;
 	int err;
 	/* Default Headphone is rear headphone */
 	hda_nid_t headphone_nid = spec->out_pins[1];
@@ -4334,115 +4659,112 @@ static int ca0132_alt_select_out(struct hda_codec *codec)
 	} else
 		spec->cur_out_type = spec->out_enum_val;
 
-	/* Begin DSP output switch */
-	tmp = FLOAT_ONE;
-	err = dspio_set_uint_param(codec, 0x96, 0x3A, tmp);
+	outfx_set = spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID];
+
+	/* Begin DSP output switch, mute DSP volume. */
+	err = dspio_set_uint_param(codec, 0x96, SPEAKER_TUNING_MUTE, FLOAT_ONE);
 	if (err < 0)
 		goto exit;
 
-	ca0132_alt_select_out_quirk_handler(codec);
+	if (ca0132_alt_select_out_quirk_set(codec) < 0)
+		goto exit;
 
 	switch (spec->cur_out_type) {
 	case SPEAKER_OUT:
 		codec_dbg(codec, "%s speaker\n", __func__);
 
-		/* disable headphone node */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0,
-					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[1],
-				    pin_ctl & ~PIN_HP);
-		/* enable line-out node */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0,
-				AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[0],
-				    pin_ctl | PIN_OUT);
 		/* Enable EAPD */
 		snd_hda_codec_write(codec, spec->out_pins[0], 0,
 			AC_VERB_SET_EAPD_BTLENABLE, 0x01);
 
-		/* If PlayEnhancement is enabled, set different source */
-		if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
-			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE);
+		/* Disable headphone node. */
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[1], 0, 0);
+		/* Set front L-R to output. */
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[0], 1, 0);
+		/* Set Center/LFE to output. */
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[2], 1, 0);
+		/* Set rear surround to output. */
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[3], 1, 0);
+
+		/*
+		 * Without PlayEnhancement being enabled, if we've got a 2.0
+		 * setup, set it to floating point eight to disable any DSP
+		 * processing effects.
+		 */
+		if (!outfx_set && spec->channel_cfg_val == SPEAKER_CHANNELS_2_0)
+			tmp = FLOAT_EIGHT;
 		else
-			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT);
+			tmp = speaker_channel_cfgs[spec->channel_cfg_val].val;
+
+		err = dspio_set_uint_param(codec, 0x80, 0x04, tmp);
+		if (err < 0)
+			goto exit;
+
 		break;
 	case HEADPHONE_OUT:
 		codec_dbg(codec, "%s hp\n", __func__);
-
 		snd_hda_codec_write(codec, spec->out_pins[0], 0,
 			AC_VERB_SET_EAPD_BTLENABLE, 0x00);
 
-		/* disable speaker*/
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0,
-					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[0],
-				pin_ctl & ~PIN_HP);
+		/* Disable all speaker nodes. */
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[0], 0, 0);
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[2], 0, 0);
+		ca0132_set_out_node_pincfg(codec, spec->out_pins[3], 0, 0);
 
 		/* enable headphone, either front or rear */
-
 		if (snd_hda_jack_detect(codec, spec->unsol_tag_front_hp))
 			headphone_nid = spec->out_pins[2];
 		else if (snd_hda_jack_detect(codec, spec->unsol_tag_hp))
 			headphone_nid = spec->out_pins[1];
 
-		pin_ctl = snd_hda_codec_read(codec, headphone_nid, 0,
-					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, headphone_nid,
-				    pin_ctl | PIN_HP);
+		ca0132_set_out_node_pincfg(codec, headphone_nid, 1, 1);
 
-		if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
-			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE);
+		if (outfx_set)
+			err = dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ONE);
 		else
-			dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ZERO);
-		break;
-	case SURROUND_OUT:
-		codec_dbg(codec, "%s surround\n", __func__);
+			err = dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_ZERO);
 
-		/* enable line out node */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[0], 0,
-				AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[0],
-						pin_ctl | PIN_OUT);
-		/* Disable headphone out */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[1], 0,
-					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[1],
-				    pin_ctl & ~PIN_HP);
-		/* Enable EAPD on line out */
-		snd_hda_codec_write(codec, spec->out_pins[0], 0,
-			AC_VERB_SET_EAPD_BTLENABLE, 0x01);
-		/* enable center/lfe out node */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[2], 0,
-					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[2],
-				    pin_ctl | PIN_OUT);
-		/* Now set rear surround node as out. */
-		pin_ctl = snd_hda_codec_read(codec, spec->out_pins[3], 0,
-					AC_VERB_GET_PIN_WIDGET_CONTROL, 0);
-		snd_hda_set_pin_ctl(codec, spec->out_pins[3],
-				    pin_ctl | PIN_OUT);
-
-		dspio_set_uint_param(codec, 0x80, 0x04, FLOAT_EIGHT);
+		if (err < 0)
+			goto exit;
 		break;
 	}
 	/*
-	 * Surround always sets it's scp command to req 0x04 to FLOAT_EIGHT.
-	 * With this set though, X_BASS cannot be enabled. So, if we have OutFX
-	 * enabled, we need to make sure X_BASS is off, otherwise everything
-	 * sounds all muffled. Running ca0132_effects_set with X_BASS as the
-	 * effect should sort this out.
+	 * If output effects are enabled, set the X-Bass effect value again to
+	 * make sure that it's properly enabled/disabled for speaker
+	 * configurations with an LFE channel.
 	 */
-	if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
+	if (outfx_set)
 		ca0132_effects_set(codec, X_BASS,
 			spec->effects_switch[X_BASS - EFFECT_START_NID]);
 
-	/* run through the output dsp commands for the selected output. */
-	for (i = 0; i < alt_out_presets[spec->cur_out_type].commands; i++) {
-		err = dspio_set_uint_param(codec,
-		alt_out_presets[spec->cur_out_type].mids[i],
-		alt_out_presets[spec->cur_out_type].reqs[i],
-		alt_out_presets[spec->cur_out_type].vals[i]);
+	/* Set speaker EQ bypass attenuation to 0. */
+	err = dspio_set_uint_param(codec, 0x8f, 0x01, FLOAT_ZERO);
+	if (err < 0)
+		goto exit;
+
+	/*
+	 * Although unused on all cards but the AE series, this is always set
+	 * to zero when setting the output.
+	 */
+	err = dspio_set_uint_param(codec, 0x96,
+			SPEAKER_TUNING_USE_SPEAKER_EQ, FLOAT_ZERO);
+	if (err < 0)
+		goto exit;
+
+	if (spec->cur_out_type == SPEAKER_OUT)
+		err = ca0132_alt_surround_set_bass_redirection(codec,
+				spec->bass_redirection_val);
+	else
+		err = ca0132_alt_surround_set_bass_redirection(codec, 0);
+
+	/* Unmute DSP now that we're done with output selection. */
+	err = dspio_set_uint_param(codec, 0x96,
+			SPEAKER_TUNING_MUTE, FLOAT_ZERO);
+	if (err < 0)
+		goto exit;
 
+	if (spec->cur_out_type == SPEAKER_OUT) {
+		err = ca0132_alt_set_full_range_speaker(codec);
 		if (err < 0)
 			goto exit;
 	}
@@ -4675,6 +4997,15 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
 			tmp = FLOAT_THREE;
 			break;
+		case QUIRK_AE7:
+			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
+			tmp = FLOAT_THREE;
+			chipio_set_conn_rate(codec, MEM_CONNID_MICIN2,
+					SR_96_000);
+			chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2,
+					SR_96_000);
+			dspio_set_uint_param(codec, 0x80, 0x01, FLOAT_ZERO);
+			break;
 		default:
 			tmp = FLOAT_ONE;
 			break;
@@ -4720,6 +5051,14 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 		case QUIRK_AE5:
 			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
 			break;
+		case QUIRK_AE7:
+			ca0113_mmio_command_set(codec, 0x30, 0x28, 0x3f);
+			chipio_set_conn_rate(codec, MEM_CONNID_MICIN2,
+					SR_96_000);
+			chipio_set_conn_rate(codec, MEM_CONNID_MICOUT2,
+					SR_96_000);
+			dspio_set_uint_param(codec, 0x80, 0x01, FLOAT_ZERO);
+			break;
 		default:
 			break;
 		}
@@ -4729,7 +5068,10 @@ static int ca0132_alt_select_in(struct hda_codec *codec)
 		if (ca0132_quirk(spec) == QUIRK_R3DI)
 			chipio_set_conn_rate(codec, 0x0F, SR_96_000);
 
-		tmp = FLOAT_ZERO;
+		if (ca0132_quirk(spec) == QUIRK_AE7)
+			tmp = FLOAT_THREE;
+		else
+			tmp = FLOAT_ZERO;
 		dspio_set_uint_param(codec, 0x80, 0x00, tmp);
 
 		switch (ca0132_quirk(spec)) {
@@ -4852,7 +5194,7 @@ static int ca0132_voicefx_set(struct hda_codec *codec, int enable)
 static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val)
 {
 	struct ca0132_spec *spec = codec->spec;
-	unsigned int on, tmp;
+	unsigned int on, tmp, channel_cfg;
 	int num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT;
 	int err = 0;
 	int idx = nid - EFFECT_START_NID;
@@ -4865,8 +5207,12 @@ static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val)
 		/* if PE if off, turn off out effects. */
 		if (!spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID])
 			val = 0;
-		if (spec->cur_out_type == SURROUND_OUT && nid == X_BASS)
-			val = 0;
+		if (spec->cur_out_type == SPEAKER_OUT && nid == X_BASS) {
+			channel_cfg = spec->channel_cfg_val;
+			if (channel_cfg != SPEAKER_CHANNELS_2_0 &&
+					channel_cfg != SPEAKER_CHANNELS_4_0)
+				val = 0;
+		}
 	}
 
 	/* for in effect, qualify with CrystalVoice */
@@ -5122,6 +5468,18 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol,
 	return ret;
 }
 /* End of control change helpers. */
+
+static void ca0132_alt_bass_redirection_xover_set(struct hda_codec *codec,
+		long idx)
+{
+	snd_hda_power_up(codec);
+
+	dspio_set_param(codec, 0x96, 0x20, SPEAKER_BASS_REDIRECT_XOVER_FREQ,
+			&(float_xbass_xover_lookup[idx]), sizeof(unsigned int));
+
+	snd_hda_power_down(codec);
+}
+
 /*
  * Below I've added controls to mess with the effect levels, I've only enabled
  * them on the Sound Blaster Z, but they would probably also work on the
@@ -5130,6 +5488,7 @@ static int ca0132_vnode_switch_set(struct snd_kcontrol *kcontrol,
  */
 
 /* Sets DSP effect level from the sliders above the controls */
+
 static int ca0132_alt_slider_ctl_set(struct hda_codec *codec, hda_nid_t nid,
 			  const unsigned int *lookup, int idx)
 {
@@ -5175,8 +5534,13 @@ static int ca0132_alt_xbass_xover_slider_ctl_get(struct snd_kcontrol *kcontrol,
 	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
 	struct ca0132_spec *spec = codec->spec;
 	long *valp = ucontrol->value.integer.value;
+	hda_nid_t nid = get_amp_nid(kcontrol);
+
+	if (nid == BASS_REDIRECTION_XOVER)
+		*valp = spec->bass_redirect_xover_freq;
+	else
+		*valp = spec->xbass_xover_freq;
 
-	*valp = spec->xbass_xover_freq;
 	return 0;
 }
 
@@ -5230,16 +5594,25 @@ static int ca0132_alt_xbass_xover_slider_put(struct snd_kcontrol *kcontrol,
 	struct ca0132_spec *spec = codec->spec;
 	hda_nid_t nid = get_amp_nid(kcontrol);
 	long *valp = ucontrol->value.integer.value;
+	long *cur_val;
 	int idx;
 
+	if (nid == BASS_REDIRECTION_XOVER)
+		cur_val = &spec->bass_redirect_xover_freq;
+	else
+		cur_val = &spec->xbass_xover_freq;
+
 	/* any change? */
-	if (spec->xbass_xover_freq == *valp)
+	if (*cur_val == *valp)
 		return 0;
 
-	spec->xbass_xover_freq = *valp;
+	*cur_val = *valp;
 
 	idx = *valp;
-	ca0132_alt_slider_ctl_set(codec, nid, float_xbass_xover_lookup, idx);
+	if (nid == BASS_REDIRECTION_XOVER)
+		ca0132_alt_bass_redirection_xover_set(codec, *cur_val);
+	else
+		ca0132_alt_slider_ctl_set(codec, nid, float_xbass_xover_lookup, idx);
 
 	return 0;
 }
@@ -5466,6 +5839,13 @@ static int ca0132_alt_input_source_put(struct snd_kcontrol *kcontrol,
 	int sel = ucontrol->value.enumerated.item[0];
 	unsigned int items = IN_SRC_NUM_OF_INPUTS;
 
+	/*
+	 * The AE-7 has no front microphone, so limit items to 2: rear mic and
+	 * line-in.
+	 */
+	if (ca0132_quirk(spec) == QUIRK_AE7)
+		items = 2;
+
 	if (sel >= items)
 		return 0;
 
@@ -5489,7 +5869,7 @@ static int ca0132_alt_output_select_get_info(struct snd_kcontrol *kcontrol,
 	if (uinfo->value.enumerated.item >= NUM_OF_OUTPUTS)
 		uinfo->value.enumerated.item = NUM_OF_OUTPUTS - 1;
 	strcpy(uinfo->value.enumerated.name,
-			alt_out_presets[uinfo->value.enumerated.item].name);
+			out_type_str[uinfo->value.enumerated.item]);
 	return 0;
 }
 
@@ -5516,7 +5896,7 @@ static int ca0132_alt_output_select_put(struct snd_kcontrol *kcontrol,
 		return 0;
 
 	codec_dbg(codec, "ca0132_alt_output_select: sel=%d, preset=%s\n",
-		    sel, alt_out_presets[sel].name);
+		    sel, out_type_str[sel]);
 
 	spec->out_enum_val = sel;
 
@@ -5528,6 +5908,54 @@ static int ca0132_alt_output_select_put(struct snd_kcontrol *kcontrol,
 	return 1;
 }
 
+/* Select surround output type: 2.1, 4.0, 4.1, or 5.1. */
+static int ca0132_alt_speaker_channel_cfg_get_info(struct snd_kcontrol *kcontrol,
+				 struct snd_ctl_elem_info *uinfo)
+{
+	unsigned int items = SPEAKER_CHANNEL_CFG_COUNT;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = items;
+	if (uinfo->value.enumerated.item >= items)
+		uinfo->value.enumerated.item = items - 1;
+	strcpy(uinfo->value.enumerated.name,
+			speaker_channel_cfgs[uinfo->value.enumerated.item].name);
+	return 0;
+}
+
+static int ca0132_alt_speaker_channel_cfg_get(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ca0132_spec *spec = codec->spec;
+
+	ucontrol->value.enumerated.item[0] = spec->channel_cfg_val;
+	return 0;
+}
+
+static int ca0132_alt_speaker_channel_cfg_put(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct hda_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct ca0132_spec *spec = codec->spec;
+	int sel = ucontrol->value.enumerated.item[0];
+	unsigned int items = SPEAKER_CHANNEL_CFG_COUNT;
+
+	if (sel >= items)
+		return 0;
+
+	codec_dbg(codec, "ca0132_alt_speaker_channels: sel=%d, channels=%s\n",
+		    sel, speaker_channel_cfgs[sel].name);
+
+	spec->channel_cfg_val = sel;
+
+	if (spec->out_enum_val == SPEAKER_OUT)
+		ca0132_alt_select_out(codec);
+
+	return 1;
+}
+
 /*
  * Smart Volume output setting control. Three different settings, Normal,
  * which takes the value from the smart volume slider. The two others, loud
@@ -5754,6 +6182,16 @@ static int ca0132_switch_get(struct snd_kcontrol *kcontrol,
 		return 0;
 	}
 
+	if (nid == SPEAKER_FULL_RANGE_FRONT || nid == SPEAKER_FULL_RANGE_REAR) {
+		*valp = spec->speaker_range_val[nid - SPEAKER_FULL_RANGE_FRONT];
+		return 0;
+	}
+
+	if (nid == BASS_REDIRECTION) {
+		*valp = spec->bass_redirection_val;
+		return 0;
+	}
+
 	return 0;
 }
 
@@ -5832,6 +6270,22 @@ static int ca0132_switch_put(struct snd_kcontrol *kcontrol,
 		goto exit;
 	}
 
+	if (nid == SPEAKER_FULL_RANGE_FRONT || nid == SPEAKER_FULL_RANGE_REAR) {
+		spec->speaker_range_val[nid - SPEAKER_FULL_RANGE_FRONT] = *valp;
+		if (spec->cur_out_type == SPEAKER_OUT)
+			ca0132_alt_set_full_range_speaker(codec);
+
+		changed = 0;
+	}
+
+	if (nid == BASS_REDIRECTION) {
+		spec->bass_redirection_val = *valp;
+		if (spec->cur_out_type == SPEAKER_OUT)
+			ca0132_alt_surround_set_bass_redirection(codec, *valp);
+
+		changed = 0;
+	}
+
 exit:
 	snd_hda_power_down(codec);
 	return changed;
@@ -6173,6 +6627,81 @@ static int ca0132_alt_add_output_enum(struct hda_codec *codec)
 }
 
 /*
+ * Add a control for selecting channel count on speaker output. Setting this
+ * allows the DSP to do bass redirection and channel upmixing on surround
+ * configurations.
+ */
+static int ca0132_alt_add_speaker_channel_cfg_enum(struct hda_codec *codec)
+{
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_MUTE_MONO("Surround Channel Config",
+				    SPEAKER_CHANNEL_CFG_ENUM, 1, 0, HDA_OUTPUT);
+	knew.info = ca0132_alt_speaker_channel_cfg_get_info;
+	knew.get = ca0132_alt_speaker_channel_cfg_get;
+	knew.put = ca0132_alt_speaker_channel_cfg_put;
+	return snd_hda_ctl_add(codec, SPEAKER_CHANNEL_CFG_ENUM,
+				snd_ctl_new1(&knew, codec));
+}
+
+/*
+ * Full range front stereo and rear surround switches. When these are set to
+ * full range, the lower frequencies from these channels are no longer
+ * redirected to the LFE channel.
+ */
+static int ca0132_alt_add_front_full_range_switch(struct hda_codec *codec)
+{
+	struct snd_kcontrol_new knew =
+		CA0132_CODEC_MUTE_MONO("Full-Range Front Speakers",
+				    SPEAKER_FULL_RANGE_FRONT, 1, HDA_OUTPUT);
+
+	return snd_hda_ctl_add(codec, SPEAKER_FULL_RANGE_FRONT,
+				snd_ctl_new1(&knew, codec));
+}
+
+static int ca0132_alt_add_rear_full_range_switch(struct hda_codec *codec)
+{
+	struct snd_kcontrol_new knew =
+		CA0132_CODEC_MUTE_MONO("Full-Range Rear Speakers",
+				    SPEAKER_FULL_RANGE_REAR, 1, HDA_OUTPUT);
+
+	return snd_hda_ctl_add(codec, SPEAKER_FULL_RANGE_REAR,
+				snd_ctl_new1(&knew, codec));
+}
+
+/*
+ * Bass redirection redirects audio below the crossover frequency to the LFE
+ * channel on speakers that are set as not being full-range. On configurations
+ * without an LFE channel, it does nothing. Bass redirection seems to be the
+ * replacement for X-Bass on configurations with an LFE channel.
+ */
+static int ca0132_alt_add_bass_redirection_crossover(struct hda_codec *codec)
+{
+	const char *namestr = "Bass Redirection Crossover";
+	struct snd_kcontrol_new knew =
+		HDA_CODEC_VOLUME_MONO(namestr, BASS_REDIRECTION_XOVER, 1, 0,
+				HDA_OUTPUT);
+
+	knew.tlv.c = NULL;
+	knew.info = ca0132_alt_xbass_xover_slider_info;
+	knew.get = ca0132_alt_xbass_xover_slider_ctl_get;
+	knew.put = ca0132_alt_xbass_xover_slider_put;
+
+	return snd_hda_ctl_add(codec, BASS_REDIRECTION_XOVER,
+			snd_ctl_new1(&knew, codec));
+}
+
+static int ca0132_alt_add_bass_redirection_switch(struct hda_codec *codec)
+{
+	const char *namestr = "Bass Redirection";
+	struct snd_kcontrol_new knew =
+		CA0132_CODEC_MUTE_MONO(namestr, BASS_REDIRECTION, 1,
+				HDA_OUTPUT);
+
+	return snd_hda_ctl_add(codec, BASS_REDIRECTION,
+			snd_ctl_new1(&knew, codec));
+}
+
+/*
  * Create an Input Source enumerated control for the alternate ca0132 codecs
  * because the front microphone has no auto-detect, and Line-in has to be set
  * somehow.
@@ -6478,6 +7007,21 @@ static int ca0132_build_controls(struct hda_codec *codec)
 		err = ca0132_alt_add_output_enum(codec);
 		if (err < 0)
 			return err;
+		err = ca0132_alt_add_speaker_channel_cfg_enum(codec);
+		if (err < 0)
+			return err;
+		err = ca0132_alt_add_front_full_range_switch(codec);
+		if (err < 0)
+			return err;
+		err = ca0132_alt_add_rear_full_range_switch(codec);
+		if (err < 0)
+			return err;
+		err = ca0132_alt_add_bass_redirection_crossover(codec);
+		if (err < 0)
+			return err;
+		err = ca0132_alt_add_bass_redirection_switch(codec);
+		if (err < 0)
+			return err;
 		err = ca0132_alt_add_mic_boost_enum(codec);
 		if (err < 0)
 			return err;
@@ -6492,20 +7036,25 @@ static int ca0132_build_controls(struct hda_codec *codec)
 		}
 	}
 
-	if (ca0132_quirk(spec) == QUIRK_AE5) {
+	switch (ca0132_quirk(spec)) {
+	case QUIRK_AE5:
+	case QUIRK_AE7:
 		err = ae5_add_headphone_gain_enum(codec);
 		if (err < 0)
 			return err;
 		err = ae5_add_sound_filter_enum(codec);
 		if (err < 0)
 			return err;
-	}
-
-	if (ca0132_quirk(spec) == QUIRK_ZXR) {
+		break;
+	case QUIRK_ZXR:
 		err = zxr_add_headphone_gain_switch(codec);
 		if (err < 0)
 			return err;
+		break;
+	default:
+		break;
 	}
+
 #ifdef ENABLE_TUNING_CONTROLS
 	add_tuning_ctls(codec);
 #endif
@@ -6875,6 +7424,68 @@ static void ca0132_refresh_widget_caps(struct hda_codec *codec)
 }
 
 /*
+ * Default speaker tuning values setup for alternative codecs.
+ */
+static const unsigned int sbz_default_delay_values[] = {
+	/* Non-zero values are floating point 0.000198. */
+	0x394f9e38, 0x394f9e38, 0x00000000, 0x00000000, 0x00000000, 0x00000000
+};
+
+static const unsigned int zxr_default_delay_values[] = {
+	/* Non-zero values are floating point 0.000220. */
+	0x00000000, 0x00000000, 0x3966afcd, 0x3966afcd, 0x3966afcd, 0x3966afcd
+};
+
+static const unsigned int ae5_default_delay_values[] = {
+	/* Non-zero values are floating point 0.000100. */
+	0x00000000, 0x00000000, 0x38d1b717, 0x38d1b717, 0x38d1b717, 0x38d1b717
+};
+
+/*
+ * If we never change these, probably only need them on initialization.
+ */
+static void ca0132_alt_init_speaker_tuning(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int i, tmp, start_req, end_req;
+	const unsigned int *values;
+
+	switch (ca0132_quirk(spec)) {
+	case QUIRK_SBZ:
+		values = sbz_default_delay_values;
+		break;
+	case QUIRK_ZXR:
+		values = zxr_default_delay_values;
+		break;
+	case QUIRK_AE5:
+	case QUIRK_AE7:
+		values = ae5_default_delay_values;
+		break;
+	default:
+		values = sbz_default_delay_values;
+		break;
+	}
+
+	tmp = FLOAT_ZERO;
+	dspio_set_uint_param(codec, 0x96, SPEAKER_TUNING_ENABLE_CENTER_EQ, tmp);
+
+	start_req = SPEAKER_TUNING_FRONT_LEFT_VOL_LEVEL;
+	end_req = SPEAKER_TUNING_REAR_RIGHT_VOL_LEVEL;
+	for (i = start_req; i < end_req + 1; i++)
+		dspio_set_uint_param(codec, 0x96, i, tmp);
+
+	start_req = SPEAKER_TUNING_FRONT_LEFT_INVERT;
+	end_req = SPEAKER_TUNING_REAR_RIGHT_INVERT;
+	for (i = start_req; i < end_req + 1; i++)
+		dspio_set_uint_param(codec, 0x96, i, tmp);
+
+
+	for (i = 0; i < 6; i++)
+		dspio_set_uint_param(codec, 0x96,
+				SPEAKER_TUNING_FRONT_LEFT_DELAY + i, values[i]);
+}
+
+/*
  * Creates a dummy stream to bind the output to. This seems to have to be done
  * after changing the main outputs source and destination streams.
  */
@@ -7021,6 +7632,7 @@ static void ca0132_alt_dsp_scp_startup(struct hda_codec *codec)
 		switch (ca0132_quirk(spec)) {
 		case QUIRK_SBZ:
 		case QUIRK_AE5:
+		case QUIRK_AE7:
 			tmp = 0x00000003;
 			dspio_set_uint_param_no_source(codec, 0x80, 0x0C, tmp);
 			tmp = 0x00000000;
@@ -7230,6 +7842,206 @@ static void ae5_post_dsp_startup_data(struct hda_codec *codec)
 	mutex_unlock(&spec->chipio_mutex);
 }
 
+static const unsigned int ae7_port_set_data[] = {
+	0x0001e0c0, 0x0001e1c1, 0x0001e4c2, 0x0001e5c3, 0x0001e2c4, 0x0001e3c5,
+	0x0001e8c6, 0x0001e9c7, 0x0001ecc8, 0x0001edc9, 0x0001eaca, 0x0001ebcb
+};
+
+static void ae7_post_dsp_setup_ports(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int i, count, addr;
+
+	mutex_lock(&spec->chipio_mutex);
+
+	chipio_set_stream_channels(codec, 0x0c, 6);
+	chipio_set_stream_control(codec, 0x0c, 1);
+
+	count = ARRAY_SIZE(ae7_port_set_data);
+	addr = 0x190030;
+	for (i = 0; i < count; i++) {
+		chipio_write_no_mutex(codec, addr, ae7_port_set_data[i]);
+
+		/* Addresses are incremented by 4-bytes. */
+		addr += 0x04;
+	}
+
+	/*
+	 * Port setting always ends with a write of 0x1 to address 0x19042c.
+	 */
+	chipio_write_no_mutex(codec, 0x19042c, 0x00000001);
+
+	ca0113_mmio_command_set(codec, 0x30, 0x30, 0x00);
+	ca0113_mmio_command_set(codec, 0x48, 0x0d, 0x40);
+	ca0113_mmio_command_set(codec, 0x48, 0x17, 0x00);
+	ca0113_mmio_command_set(codec, 0x48, 0x19, 0x00);
+	ca0113_mmio_command_set(codec, 0x48, 0x11, 0xff);
+	ca0113_mmio_command_set(codec, 0x48, 0x12, 0xff);
+	ca0113_mmio_command_set(codec, 0x48, 0x13, 0xff);
+	ca0113_mmio_command_set(codec, 0x48, 0x14, 0x7f);
+
+	mutex_unlock(&spec->chipio_mutex);
+}
+
+static void ae7_post_dsp_asi_stream_setup(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	mutex_lock(&spec->chipio_mutex);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x725, 0x81);
+	ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x00);
+
+	chipio_set_conn_rate_no_mutex(codec, 0x70, SR_96_000);
+	chipio_set_stream_channels(codec, 0x0c, 6);
+	chipio_set_stream_control(codec, 0x0c, 1);
+
+	chipio_set_stream_source_dest(codec, 0x05, 0x43, 0x00);
+	chipio_set_stream_source_dest(codec, 0x18, 0x09, 0xd0);
+
+	chipio_set_conn_rate_no_mutex(codec, 0xd0, SR_96_000);
+	chipio_set_stream_channels(codec, 0x18, 6);
+	chipio_set_stream_control(codec, 0x18, 1);
+
+	chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 4);
+
+	mutex_unlock(&spec->chipio_mutex);
+}
+
+static void ae7_post_dsp_pll_setup(struct hda_codec *codec)
+{
+	const unsigned int addr[] = { 0x41, 0x45, 0x40, 0x43, 0x51 };
+	const unsigned int data[] = { 0xc8, 0xcc, 0xcb, 0xc7, 0x8d };
+	unsigned int i;
+
+	for (i = 0; i < ARRAY_SIZE(addr); i++) {
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				    VENDOR_CHIPIO_8051_ADDRESS_LOW, addr[i]);
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				    VENDOR_CHIPIO_PLL_PMU_WRITE, data[i]);
+	}
+}
+
+static void ae7_post_dsp_asi_setup_ports(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	const unsigned int target[] = { 0x0b, 0x04, 0x06, 0x0a, 0x0c, 0x11,
+					0x12, 0x13, 0x14 };
+	const unsigned int data[]   = { 0x12, 0x00, 0x48, 0x05, 0x5f, 0xff,
+					0xff, 0xff, 0x7f };
+	unsigned int i;
+
+	mutex_lock(&spec->chipio_mutex);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7);
+
+	chipio_write_no_mutex(codec, 0x189000, 0x0001f101);
+	chipio_write_no_mutex(codec, 0x189004, 0x0001f101);
+	chipio_write_no_mutex(codec, 0x189024, 0x00014004);
+	chipio_write_no_mutex(codec, 0x189028, 0x0002000f);
+
+	ae7_post_dsp_pll_setup(codec);
+	chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 7);
+
+	for (i = 0; i < ARRAY_SIZE(target); i++)
+		ca0113_mmio_command_set(codec, 0x48, target[i], data[i]);
+
+	ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83);
+	ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x00);
+	ca0113_mmio_command_set(codec, 0x48, 0x10, 0x00);
+
+	chipio_set_stream_source_dest(codec, 0x21, 0x64, 0x56);
+	chipio_set_stream_channels(codec, 0x21, 2);
+	chipio_set_conn_rate_no_mutex(codec, 0x56, SR_8_000);
+
+	chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_NODE_ID, 0x09);
+	/*
+	 * In the 8051's memory, this param is referred to as 'n2sid', which I
+	 * believe is 'node to streamID'. It seems to be a way to assign a
+	 * stream to a given HDA node.
+	 */
+	chipio_set_control_param_no_mutex(codec, 0x20, 0x21);
+
+	chipio_write_no_mutex(codec, 0x18b038, 0x00000088);
+
+	/*
+	 * Now, at this point on Windows, an actual stream is setup and
+	 * seemingly sends data to the HDA node 0x09, which is the digital
+	 * audio input node. This is left out here, because obviously I don't
+	 * know what data is being sent. Interestingly, the AE-5 seems to go
+	 * through the motions of getting here and never actually takes this
+	 * step, but the AE-7 does.
+	 */
+
+	ca0113_mmio_gpio_set(codec, 0, 1);
+	ca0113_mmio_gpio_set(codec, 1, 1);
+
+	ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83);
+	chipio_write_no_mutex(codec, 0x18b03c, 0x00000000);
+	ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x00);
+	ca0113_mmio_command_set(codec, 0x48, 0x10, 0x00);
+
+	chipio_set_stream_source_dest(codec, 0x05, 0x43, 0x00);
+	chipio_set_stream_source_dest(codec, 0x18, 0x09, 0xd0);
+
+	chipio_set_conn_rate_no_mutex(codec, 0xd0, SR_96_000);
+	chipio_set_stream_channels(codec, 0x18, 6);
+
+	/*
+	 * Runs again, this has been repeated a few times, but I'm just
+	 * following what the Windows driver does.
+	 */
+	ae7_post_dsp_pll_setup(codec);
+	chipio_set_control_param_no_mutex(codec, CONTROL_PARAM_ASI, 7);
+
+	mutex_unlock(&spec->chipio_mutex);
+}
+
+/*
+ * The Windows driver has commands that seem to setup ASI, which I believe to
+ * be some sort of audio serial interface. My current speculation is that it's
+ * related to communicating with the new DAC.
+ */
+static void ae7_post_dsp_asi_setup(struct hda_codec *codec)
+{
+	chipio_8051_write_direct(codec, 0x93, 0x10);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x44);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2);
+
+	ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83);
+	ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f);
+
+	chipio_set_control_param(codec, 3, 3);
+	chipio_set_control_flag(codec, CONTROL_FLAG_ASI_96KHZ, 1);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0, 0x724, 0x83);
+	chipio_set_control_param(codec, CONTROL_PARAM_ASI, 0);
+	snd_hda_codec_write(codec, 0x17, 0, 0x794, 0x00);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x92);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_HIGH, 0xfa);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_DATA_WRITE, 0x22);
+
+	ae7_post_dsp_pll_setup(codec);
+	ae7_post_dsp_asi_stream_setup(codec);
+
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x43);
+	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc7);
+
+	ae7_post_dsp_asi_setup_ports(codec);
+}
+
 /*
  * Setup default parameters for DSP
  */
@@ -7306,6 +8118,12 @@ static void r3d_setup_defaults(struct hda_codec *codec)
 	if (ca0132_quirk(spec) == QUIRK_R3DI)
 		r3di_gpio_dsp_status_set(codec, R3DI_DSP_DOWNLOADED);
 
+	/* Disable mute on Center/LFE. */
+	if (ca0132_quirk(spec) == QUIRK_R3D) {
+		ca0113_mmio_gpio_set(codec, 2, false);
+		ca0113_mmio_gpio_set(codec, 4, true);
+	}
+
 	/* Setup effect defaults */
 	num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1;
 	for (idx = 0; idx < num_fx; idx++) {
@@ -7373,6 +8191,8 @@ static void sbz_setup_defaults(struct hda_codec *codec)
 		}
 	}
 
+	ca0132_alt_init_speaker_tuning(codec);
+
 	ca0132_alt_create_dummy_stream(codec);
 }
 
@@ -7440,6 +8260,93 @@ static void ae5_setup_defaults(struct hda_codec *codec)
 		}
 	}
 
+	ca0132_alt_init_speaker_tuning(codec);
+
+	ca0132_alt_create_dummy_stream(codec);
+}
+
+/*
+ * Setup default parameters for the Sound Blaster AE-7 DSP.
+ */
+static void ae7_setup_defaults(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int tmp;
+	int num_fx;
+	int idx, i;
+
+	if (spec->dsp_state != DSP_DOWNLOADED)
+		return;
+
+	ca0132_alt_dsp_scp_startup(codec);
+	ca0132_alt_init_analog_mics(codec);
+	ae7_post_dsp_setup_ports(codec);
+
+	tmp = FLOAT_ZERO;
+	dspio_set_uint_param(codec, 0x96,
+			SPEAKER_TUNING_FRONT_LEFT_INVERT, tmp);
+	dspio_set_uint_param(codec, 0x96,
+			SPEAKER_TUNING_FRONT_RIGHT_INVERT, tmp);
+
+	ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f);
+
+	/* New, unknown SCP req's */
+	dspio_set_uint_param(codec, 0x80, 0x0d, tmp);
+	dspio_set_uint_param(codec, 0x80, 0x0e, tmp);
+
+	ca0113_mmio_gpio_set(codec, 0, false);
+
+	/* Internal loopback off */
+	tmp = FLOAT_ONE;
+	dspio_set_uint_param(codec, 0x37, 0x08, tmp);
+	dspio_set_uint_param(codec, 0x37, 0x10, tmp);
+
+	/*remove DSP headroom*/
+	tmp = FLOAT_ZERO;
+	dspio_set_uint_param(codec, 0x96, 0x3C, tmp);
+
+	/* set WUH source */
+	tmp = FLOAT_TWO;
+	dspio_set_uint_param(codec, 0x31, 0x00, tmp);
+	chipio_set_conn_rate(codec, MEM_CONNID_WUH, SR_48_000);
+
+	/* Set speaker source? */
+	dspio_set_uint_param(codec, 0x32, 0x00, tmp);
+	ca0113_mmio_command_set(codec, 0x30, 0x28, 0x00);
+
+	/*
+	 * This is the second time we've called this, but this is seemingly
+	 * what Windows does.
+	 */
+	ca0132_alt_init_analog_mics(codec);
+
+	ae7_post_dsp_asi_setup(codec);
+
+	/*
+	 * Not sure why, but these are both set to 1. They're only set to 0
+	 * upon shutdown.
+	 */
+	ca0113_mmio_gpio_set(codec, 0, true);
+	ca0113_mmio_gpio_set(codec, 1, true);
+
+	/* Volume control related. */
+	ca0113_mmio_command_set(codec, 0x48, 0x0f, 0x04);
+	ca0113_mmio_command_set(codec, 0x48, 0x10, 0x04);
+	ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x80);
+
+	/* out, in effects + voicefx */
+	num_fx = OUT_EFFECTS_COUNT + IN_EFFECTS_COUNT + 1;
+	for (idx = 0; idx < num_fx; idx++) {
+		for (i = 0; i <= ca0132_effects[idx].params; i++) {
+			dspio_set_uint_param(codec,
+					ca0132_effects[idx].mid,
+					ca0132_effects[idx].reqs[i],
+					ca0132_effects[idx].def_vals[i]);
+		}
+	}
+
+	ca0132_alt_init_speaker_tuning(codec);
+
 	ca0132_alt_create_dummy_stream(codec);
 }
 
@@ -7757,9 +8664,15 @@ static void ca0132_init_chip(struct hda_codec *codec)
 	 * ca0132 codecs. Also sets x-bass crossover frequency to 80hz.
 	 */
 	if (ca0132_use_alt_controls(spec)) {
+		/* Set speakers to default to full range. */
+		spec->speaker_range_val[0] = 1;
+		spec->speaker_range_val[1] = 1;
+
 		spec->xbass_xover_freq = 8;
 		for (i = 0; i < EFFECT_LEVEL_SLIDERS; i++)
 			spec->fx_ctl_val[i] = effect_slider_defaults[i];
+
+		spec->bass_redirect_xover_freq = 8;
 	}
 
 	spec->voicefx_val = 0;
@@ -7925,6 +8838,32 @@ static void ae5_exit_chip(struct hda_codec *codec)
 	snd_hda_codec_write(codec, 0x01, 0, 0x724, 0x83);
 }
 
+static void ae7_exit_chip(struct hda_codec *codec)
+{
+	chipio_set_stream_control(codec, 0x18, 0);
+	chipio_set_stream_source_dest(codec, 0x21, 0xc8, 0xc8);
+	chipio_set_stream_channels(codec, 0x21, 0);
+	chipio_set_control_param(codec, CONTROL_PARAM_NODE_ID, 0x09);
+	chipio_set_control_param(codec, 0x20, 0x01);
+
+	chipio_set_control_param(codec, CONTROL_PARAM_ASI, 0);
+
+	chipio_set_stream_control(codec, 0x18, 0);
+	chipio_set_stream_control(codec, 0x0c, 0);
+
+	ca0113_mmio_command_set(codec, 0x30, 0x2b, 0x00);
+	snd_hda_codec_write(codec, 0x15, 0, 0x724, 0x83);
+	ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83);
+	ca0113_mmio_command_set(codec, 0x30, 0x30, 0x00);
+	ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x00);
+	ca0113_mmio_gpio_set(codec, 0, false);
+	ca0113_mmio_gpio_set(codec, 1, false);
+	ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f);
+
+	snd_hda_codec_write(codec, 0x01, 0, 0x793, 0x00);
+	snd_hda_codec_write(codec, 0x01, 0, 0x794, 0x53);
+}
+
 static void zxr_exit_chip(struct hda_codec *codec)
 {
 	chipio_set_stream_control(codec, 0x03, 0);
@@ -8108,81 +9047,149 @@ static void r3di_pre_dsp_setup(struct hda_codec *codec)
  * what they do, or if they're necessary. Could possibly
  * be removed. Figure they're better to leave in.
  */
-static void ca0132_mmio_init(struct hda_codec *codec)
+static const unsigned int ca0113_mmio_init_address_sbz[] = {
+	0x400, 0x408, 0x40c, 0x01c, 0xc0c, 0xc00, 0xc04, 0xc0c, 0xc0c, 0xc0c,
+	0xc0c, 0xc08, 0xc08, 0xc08, 0xc08, 0xc08, 0xc04
+};
+
+static const unsigned int ca0113_mmio_init_data_sbz[] = {
+	0x00000030, 0x00000000, 0x00000003, 0x00000003, 0x00000003,
+	0x00000003, 0x000000c1, 0x000000f1, 0x00000001, 0x000000c7,
+	0x000000c1, 0x00000080
+};
+
+static const unsigned int ca0113_mmio_init_data_zxr[] = {
+	0x00000030, 0x00000000, 0x00000000, 0x00000003, 0x00000003,
+	0x00000003, 0x00000001, 0x000000f1, 0x00000001, 0x000000c7,
+	0x000000c1, 0x00000080
+};
+
+static const unsigned int ca0113_mmio_init_address_ae5[] = {
+	0x400, 0x42c, 0x46c, 0x4ac, 0x4ec, 0x43c, 0x47c, 0x4bc, 0x4fc, 0x408,
+	0x100, 0x410, 0x40c, 0x100, 0x100, 0x830, 0x86c, 0x800, 0x86c, 0x800,
+	0x804, 0x20c, 0x01c, 0xc0c, 0xc00, 0xc04, 0xc0c, 0xc0c, 0xc0c, 0xc0c,
+	0xc08, 0xc08, 0xc08, 0xc08, 0xc08, 0xc04, 0x01c
+};
+
+static const unsigned int ca0113_mmio_init_data_ae5[] = {
+	0x00000001, 0x00000000, 0x00000000, 0x00000000, 0x00000000,
+	0x00000000, 0x00000000, 0x00000000, 0x00000000, 0x00000001,
+	0x00000600, 0x00000014, 0x00000001, 0x0000060f, 0x0000070f,
+	0x00000aff, 0x00000000, 0x0000006b, 0x00000001, 0x0000006b,
+	0x00000057, 0x00800000, 0x00880680, 0x00000080, 0x00000030,
+	0x00000000, 0x00000000, 0x00000003, 0x00000003, 0x00000003,
+	0x00000001, 0x000000f1, 0x00000001, 0x000000c7, 0x000000c1,
+	0x00000080, 0x00880680
+};
+
+static void ca0132_mmio_init_sbz(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
+	unsigned int tmp[2], i, count, cur_addr;
+	const unsigned int *addr, *data;
 
-	if (ca0132_quirk(spec) == QUIRK_AE5)
-		writel(0x00000001, spec->mem_base + 0x400);
-	else
-		writel(0x00000000, spec->mem_base + 0x400);
+	addr = ca0113_mmio_init_address_sbz;
+	for (i = 0; i < 3; i++)
+		writel(0x00000000, spec->mem_base + addr[i]);
 
-	if (ca0132_quirk(spec) == QUIRK_AE5)
-		writel(0x00000001, spec->mem_base + 0x408);
-	else
-		writel(0x00000000, spec->mem_base + 0x408);
+	cur_addr = i;
+	switch (ca0132_quirk(spec)) {
+	case QUIRK_ZXR:
+		tmp[0] = 0x00880480;
+		tmp[1] = 0x00000080;
+		break;
+	case QUIRK_SBZ:
+		tmp[0] = 0x00820680;
+		tmp[1] = 0x00000083;
+		break;
+	case QUIRK_R3D:
+		tmp[0] = 0x00880680;
+		tmp[1] = 0x00000083;
+		break;
+	default:
+		tmp[0] = 0x00000000;
+		tmp[1] = 0x00000000;
+		break;
+	}
 
-	if (ca0132_quirk(spec) == QUIRK_AE5)
-		writel(0x00000001, spec->mem_base + 0x40c);
-	else
-		writel(0x00000000, spec->mem_base + 0x40C);
+	for (i = 0; i < 2; i++)
+		writel(tmp[i], spec->mem_base + addr[cur_addr + i]);
 
-	if (ca0132_quirk(spec) == QUIRK_ZXR)
-		writel(0x00880640, spec->mem_base + 0x01C);
-	else
-		writel(0x00880680, spec->mem_base + 0x01C);
+	cur_addr += i;
 
-	if (ca0132_quirk(spec) == QUIRK_AE5)
-		writel(0x00000080, spec->mem_base + 0xC0C);
-	else
-		writel(0x00000083, spec->mem_base + 0xC0C);
+	switch (ca0132_quirk(spec)) {
+	case QUIRK_ZXR:
+		count = ARRAY_SIZE(ca0113_mmio_init_data_zxr);
+		data = ca0113_mmio_init_data_zxr;
+		break;
+	default:
+		count = ARRAY_SIZE(ca0113_mmio_init_data_sbz);
+		data = ca0113_mmio_init_data_sbz;
+		break;
+	}
 
-	writel(0x00000030, spec->mem_base + 0xC00);
-	writel(0x00000000, spec->mem_base + 0xC04);
+	for (i = 0; i < count; i++)
+		writel(data[i], spec->mem_base + addr[cur_addr + i]);
+}
 
-	if (ca0132_quirk(spec) == QUIRK_AE5)
-		writel(0x00000000, spec->mem_base + 0xC0C);
-	else
-		writel(0x00000003, spec->mem_base + 0xC0C);
+static void ca0132_mmio_init_ae5(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+	const unsigned int *addr, *data;
+	unsigned int i, count;
+
+	addr = ca0113_mmio_init_address_ae5;
+	data = ca0113_mmio_init_data_ae5;
+	count = ARRAY_SIZE(ca0113_mmio_init_data_ae5);
 
-	writel(0x00000003, spec->mem_base + 0xC0C);
-	writel(0x00000003, spec->mem_base + 0xC0C);
-	writel(0x00000003, spec->mem_base + 0xC0C);
+	if (ca0132_quirk(spec) == QUIRK_AE7) {
+		writel(0x00000680, spec->mem_base + 0x1c);
+		writel(0x00880680, spec->mem_base + 0x1c);
+	}
+
+	for (i = 0; i < count; i++) {
+		/*
+		 * AE-7 shares all writes with the AE-5, except that it writes
+		 * a different value to 0x20c.
+		 */
+		if (i == 21 && ca0132_quirk(spec) == QUIRK_AE7) {
+			writel(0x00800001, spec->mem_base + addr[i]);
+			continue;
+		}
+
+		writel(data[i], spec->mem_base + addr[i]);
+	}
 
 	if (ca0132_quirk(spec) == QUIRK_AE5)
-		writel(0x00000001, spec->mem_base + 0xC08);
-	else
-		writel(0x000000C1, spec->mem_base + 0xC08);
-
-	writel(0x000000F1, spec->mem_base + 0xC08);
-	writel(0x00000001, spec->mem_base + 0xC08);
-	writel(0x000000C7, spec->mem_base + 0xC08);
-	writel(0x000000C1, spec->mem_base + 0xC08);
-	writel(0x00000080, spec->mem_base + 0xC04);
-
-	if (ca0132_quirk(spec) == QUIRK_AE5) {
-		writel(0x00000000, spec->mem_base + 0x42c);
-		writel(0x00000000, spec->mem_base + 0x46c);
-		writel(0x00000000, spec->mem_base + 0x4ac);
-		writel(0x00000000, spec->mem_base + 0x4ec);
-		writel(0x00000000, spec->mem_base + 0x43c);
-		writel(0x00000000, spec->mem_base + 0x47c);
-		writel(0x00000000, spec->mem_base + 0x4bc);
-		writel(0x00000000, spec->mem_base + 0x4fc);
-		writel(0x00000600, spec->mem_base + 0x100);
-		writel(0x00000014, spec->mem_base + 0x410);
-		writel(0x0000060f, spec->mem_base + 0x100);
-		writel(0x0000070f, spec->mem_base + 0x100);
-		writel(0x00000aff, spec->mem_base + 0x830);
-		writel(0x00000000, spec->mem_base + 0x86c);
-		writel(0x0000006b, spec->mem_base + 0x800);
-		writel(0x00000001, spec->mem_base + 0x86c);
-		writel(0x0000006b, spec->mem_base + 0x800);
-		writel(0x00000057, spec->mem_base + 0x804);
-		writel(0x00800000, spec->mem_base + 0x20c);
+		writel(0x00880680, spec->mem_base + 0x1c);
+}
+
+static void ca0132_mmio_init(struct hda_codec *codec)
+{
+	struct ca0132_spec *spec = codec->spec;
+
+	switch (ca0132_quirk(spec)) {
+	case QUIRK_R3D:
+	case QUIRK_SBZ:
+	case QUIRK_ZXR:
+		ca0132_mmio_init_sbz(codec);
+		break;
+	case QUIRK_AE5:
+		ca0132_mmio_init_ae5(codec);
+		break;
 	}
 }
 
+static const unsigned int ca0132_ae5_register_set_addresses[] = {
+	0x304, 0x304, 0x304, 0x304, 0x100, 0x304, 0x100, 0x304, 0x100, 0x304,
+	0x100, 0x304, 0x86c, 0x800, 0x86c, 0x800, 0x804
+};
+
+static const unsigned char ca0132_ae5_register_set_data[] = {
+	0x0f, 0x0e, 0x1f, 0x0c, 0x3f, 0x08, 0x7f, 0x00, 0xff, 0x00, 0x6b,
+	0x01, 0x6b, 0x57
+};
+
 /*
  * This function writes to some SFR's, does some region2 writes, and then
  * eventually resets the codec with the 0x7ff verb. Not quite sure why it does
@@ -8191,6 +9198,18 @@ static void ca0132_mmio_init(struct hda_codec *codec)
 static void ae5_register_set(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
+	unsigned int count = ARRAY_SIZE(ca0132_ae5_register_set_addresses);
+	const unsigned int *addr = ca0132_ae5_register_set_addresses;
+	const unsigned char *data = ca0132_ae5_register_set_data;
+	unsigned int i, cur_addr;
+	unsigned char tmp[3];
+
+	if (ca0132_quirk(spec) == QUIRK_AE7) {
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				    VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x41);
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc8);
+	}
 
 	chipio_8051_write_direct(codec, 0x93, 0x10);
 	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
@@ -8198,25 +9217,43 @@ static void ae5_register_set(struct hda_codec *codec)
 	snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
 			    VENDOR_CHIPIO_PLL_PMU_WRITE, 0xc2);
 
-	writeb(0x0f, spec->mem_base + 0x304);
-	writeb(0x0f, spec->mem_base + 0x304);
-	writeb(0x0f, spec->mem_base + 0x304);
-	writeb(0x0f, spec->mem_base + 0x304);
-	writeb(0x0e, spec->mem_base + 0x100);
-	writeb(0x1f, spec->mem_base + 0x304);
-	writeb(0x0c, spec->mem_base + 0x100);
-	writeb(0x3f, spec->mem_base + 0x304);
-	writeb(0x08, spec->mem_base + 0x100);
-	writeb(0x7f, spec->mem_base + 0x304);
-	writeb(0x00, spec->mem_base + 0x100);
-	writeb(0xff, spec->mem_base + 0x304);
+	if (ca0132_quirk(spec) == QUIRK_AE7) {
+		tmp[0] = 0x03;
+		tmp[1] = 0x03;
+		tmp[2] = 0x07;
+	} else {
+		tmp[0] = 0x0f;
+		tmp[1] = 0x0f;
+		tmp[2] = 0x0f;
+	}
 
-	ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x3f);
+	for (i = cur_addr = 0; i < 3; i++, cur_addr++)
+		writeb(tmp[i], spec->mem_base + addr[cur_addr]);
+
+	/*
+	 * First writes are in single bytes, final are in 4 bytes. So, we use
+	 * writeb, then writel.
+	 */
+	for (i = 0; cur_addr < 12; i++, cur_addr++)
+		writeb(data[i], spec->mem_base + addr[cur_addr]);
+
+	for (; cur_addr < count; i++, cur_addr++)
+		writel(data[i], spec->mem_base + addr[cur_addr]);
+
+	writel(0x00800001, spec->mem_base + 0x20c);
+
+	if (ca0132_quirk(spec) == QUIRK_AE7) {
+		ca0113_mmio_command_set_type2(codec, 0x48, 0x07, 0x83);
+		ca0113_mmio_command_set(codec, 0x30, 0x2e, 0x3f);
+	} else {
+		ca0113_mmio_command_set(codec, 0x30, 0x2d, 0x3f);
+	}
 
 	chipio_8051_write_direct(codec, 0x90, 0x00);
 	chipio_8051_write_direct(codec, 0x90, 0x10);
 
-	ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
+	if (ca0132_quirk(spec) == QUIRK_AE5)
+		ca0113_mmio_command_set(codec, 0x48, 0x07, 0x83);
 
 	chipio_write(codec, 0x18b0a4, 0x000000c2);
 
@@ -8268,6 +9305,19 @@ static void ca0132_alt_init(struct hda_codec *codec)
 		snd_hda_sequence_write(codec, spec->desktop_init_verbs);
 		ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f);
 		break;
+	case QUIRK_AE7:
+		ca0132_gpio_init(codec);
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				VENDOR_CHIPIO_8051_ADDRESS_LOW, 0x49);
+		snd_hda_codec_write(codec, WIDGET_CHIP_CTRL, 0,
+				VENDOR_CHIPIO_PLL_PMU_WRITE, 0x88);
+		snd_hda_sequence_write(codec, spec->chip_init_verbs);
+		snd_hda_sequence_write(codec, spec->desktop_init_verbs);
+		chipio_write(codec, 0x18b008, 0x000000f8);
+		chipio_write(codec, 0x18b008, 0x000000f0);
+		chipio_write(codec, 0x18b030, 0x00000020);
+		ca0113_mmio_command_set(codec, 0x30, 0x32, 0x3f);
+		break;
 	case QUIRK_ZXR:
 		snd_hda_sequence_write(codec, spec->chip_init_verbs);
 		snd_hda_sequence_write(codec, spec->desktop_init_verbs);
@@ -8315,7 +9365,7 @@ static int ca0132_init(struct hda_codec *codec)
 
 	snd_hda_power_up_pm(codec);
 
-	if (ca0132_quirk(spec) == QUIRK_AE5)
+	if (ca0132_quirk(spec) == QUIRK_AE5 || ca0132_quirk(spec) == QUIRK_AE7)
 		ae5_register_set(codec);
 
 	ca0132_init_unsol(codec);
@@ -8343,6 +9393,9 @@ static int ca0132_init(struct hda_codec *codec)
 	case QUIRK_AE5:
 		ae5_setup_defaults(codec);
 		break;
+	case QUIRK_AE7:
+		ae7_setup_defaults(codec);
+		break;
 	default:
 		ca0132_setup_defaults(codec);
 		ca0132_init_analog_mic2(codec);
@@ -8430,6 +9483,9 @@ static void ca0132_free(struct hda_codec *codec)
 	case QUIRK_AE5:
 		ae5_exit_chip(codec);
 		break;
+	case QUIRK_AE7:
+		ae7_exit_chip(codec);
+		break;
 	case QUIRK_R3DI:
 		r3di_gpio_shutdown(codec);
 		break;
@@ -8534,6 +9590,10 @@ static void ca0132_config(struct hda_codec *codec)
 		codec_dbg(codec, "%s: QUIRK_AE5 applied.\n", __func__);
 		snd_hda_apply_pincfgs(codec, ae5_pincfgs);
 		break;
+	case QUIRK_AE7:
+		codec_dbg(codec, "%s: QUIRK_AE7 applied.\n", __func__);
+		snd_hda_apply_pincfgs(codec, ae7_pincfgs);
+		break;
 	default:
 		break;
 	}
@@ -8615,6 +9675,7 @@ static void ca0132_config(struct hda_codec *codec)
 		spec->dig_in = 0x09;
 		break;
 	case QUIRK_AE5:
+	case QUIRK_AE7:
 		spec->num_outputs = 2;
 		spec->out_pins[0] = 0x0B; /* Line out */
 		spec->out_pins[1] = 0x11; /* Rear headphone out */
@@ -8813,6 +9874,10 @@ static int patch_ca0132(struct hda_codec *codec)
 		spec->mixers[0] = desktop_mixer;
 		snd_hda_codec_set_name(codec, "Sound BlasterX AE-5");
 		break;
+	case QUIRK_AE7:
+		spec->mixers[0] = desktop_mixer;
+		snd_hda_codec_set_name(codec, "Sound Blaster AE-7");
+		break;
 	default:
 		spec->mixers[0] = ca0132_mixer;
 		break;
@@ -8823,6 +9888,7 @@ static int patch_ca0132(struct hda_codec *codec)
 	case QUIRK_SBZ:
 	case QUIRK_R3D:
 	case QUIRK_AE5:
+	case QUIRK_AE7:
 	case QUIRK_ZXR:
 		spec->use_alt_controls = true;
 		spec->use_alt_functions = true;
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index 402050088090..055440740184 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -4269,6 +4269,7 @@ HDA_CODEC_ENTRY(0x8086280c, "Cannonlake HDMI",	patch_i915_glk_hdmi),
 HDA_CODEC_ENTRY(0x8086280d, "Geminilake HDMI",	patch_i915_glk_hdmi),
 HDA_CODEC_ENTRY(0x8086280f, "Icelake HDMI",	patch_i915_icl_hdmi),
 HDA_CODEC_ENTRY(0x80862812, "Tigerlake HDMI",	patch_i915_tgl_hdmi),
+HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI",	patch_i915_tgl_hdmi),
 HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI",	patch_i915_tgl_hdmi),
 HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI",	patch_i915_icl_hdmi),
 HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI",	patch_i915_icl_hdmi),
diff --git a/sound/pci/mixart/mixart.h b/sound/pci/mixart/mixart.h
index 42111562e9bc..cbed6d9a9f2e 100644
--- a/sound/pci/mixart/mixart.h
+++ b/sound/pci/mixart/mixart.h
@@ -69,7 +69,7 @@ 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 tasklet */
+	atomic_t msg_processed;       /* number of messages to be processed in irq thread */
 
 	struct mutex lock;              /* interrupt lock */
 	struct mutex msg_lock;		/* mailbox lock */
diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c
index 098c69b3b7aa..fcc2073c5025 100644
--- a/sound/pci/riptide/riptide.c
+++ b/sound/pci/riptide/riptide.c
@@ -445,7 +445,6 @@ struct snd_riptide {
 	union firmware_version firmware;
 
 	spinlock_t lock;
-	struct tasklet_struct riptide_tq;
 	struct snd_info_entry *proc_entry;
 
 	unsigned long received_irqs;
@@ -1070,9 +1069,9 @@ getmixer(struct cmdif *cif, short num, unsigned short *rval,
 	return 0;
 }
 
-static void riptide_handleirq(struct tasklet_struct *t)
+static irqreturn_t riptide_handleirq(int irq, void *dev_id)
 {
-	struct snd_riptide *chip = from_tasklet(chip, t, riptide_tq);
+	struct snd_riptide *chip = dev_id;
 	struct cmdif *cif = chip->cif;
 	struct snd_pcm_substream *substream[PLAYBACK_SUBSTREAMS + 1];
 	struct snd_pcm_runtime *runtime;
@@ -1083,7 +1082,7 @@ static void riptide_handleirq(struct tasklet_struct *t)
 	unsigned int flag;
 
 	if (!cif)
-		return;
+		return IRQ_HANDLED;
 
 	for (i = 0; i < PLAYBACK_SUBSTREAMS; i++)
 		substream[i] = chip->playback_substream[i];
@@ -1134,6 +1133,8 @@ static void riptide_handleirq(struct tasklet_struct *t)
 			}
 		}
 	}
+
+	return IRQ_HANDLED;
 }
 
 #ifdef CONFIG_PM_SLEEP
@@ -1699,13 +1700,14 @@ snd_riptide_interrupt(int irq, void *dev_id)
 {
 	struct snd_riptide *chip = dev_id;
 	struct cmdif *cif = chip->cif;
+	irqreturn_t ret = IRQ_HANDLED;
 
 	if (cif) {
 		chip->received_irqs++;
 		if (IS_EOBIRQ(cif->hwport) || IS_EOSIRQ(cif->hwport) ||
 		    IS_EOCIRQ(cif->hwport)) {
 			chip->handled_irqs++;
-			tasklet_schedule(&chip->riptide_tq);
+			ret = IRQ_WAKE_THREAD;
 		}
 		if (chip->rmidi && IS_MPUIRQ(cif->hwport)) {
 			chip->handled_irqs++;
@@ -1714,7 +1716,7 @@ snd_riptide_interrupt(int irq, void *dev_id)
 		}
 		SET_AIACK(cif->hwport);
 	}
-	return IRQ_HANDLED;
+	return ret;
 }
 
 static void
@@ -1843,7 +1845,6 @@ snd_riptide_create(struct snd_card *card, struct pci_dev *pci,
 	chip->received_irqs = 0;
 	chip->handled_irqs = 0;
 	chip->cif = NULL;
-	tasklet_setup(&chip->riptide_tq, riptide_handleirq);
 
 	if ((chip->res_port =
 	     request_region(chip->port, 64, "RIPTIDE")) == NULL) {
@@ -1856,8 +1857,9 @@ snd_riptide_create(struct snd_card *card, struct pci_dev *pci,
 	hwport = (struct riptideport *)chip->port;
 	UNSET_AIE(hwport);
 
-	if (request_irq(pci->irq, snd_riptide_interrupt, IRQF_SHARED,
-			KBUILD_MODNAME, chip)) {
+	if (request_threaded_irq(pci->irq, snd_riptide_interrupt,
+				 riptide_handleirq, IRQF_SHARED,
+				 KBUILD_MODNAME, chip)) {
 		snd_printk(KERN_ERR "Riptide: unable to grab IRQ %d\n",
 			   pci->irq);
 		snd_riptide_free(chip);
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index dda56ecfd33b..cea53a878c36 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -447,8 +447,8 @@ struct hdsp {
 	struct snd_pcm_substream *capture_substream;
 	struct snd_pcm_substream *playback_substream;
         struct hdsp_midi      midi[2];
-	struct tasklet_struct midi_tasklet;
-	int		      use_midi_tasklet;
+	struct work_struct    midi_work;
+	int		      use_midi_work;
 	int                   precise_ptr;
 	u32                   control_register;	     /* cached value */
 	u32                   control2_register;     /* cached value */
@@ -1385,7 +1385,6 @@ static void snd_hdsp_midi_input_trigger(struct snd_rawmidi_substream *substream,
 		}
 	} else {
 		hdsp->control_register &= ~ie;
-		tasklet_kill(&hdsp->midi_tasklet);
 	}
 
 	hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
@@ -2542,37 +2541,37 @@ static int snd_hdsp_put_precise_pointer(struct snd_kcontrol *kcontrol, struct sn
 	return change;
 }
 
-#define HDSP_USE_MIDI_TASKLET(xname, xindex) \
+#define HDSP_USE_MIDI_WORK(xname, xindex) \
 { .iface = SNDRV_CTL_ELEM_IFACE_CARD, \
   .name = xname, \
   .index = xindex, \
-  .info = snd_hdsp_info_use_midi_tasklet, \
-  .get = snd_hdsp_get_use_midi_tasklet, \
-  .put = snd_hdsp_put_use_midi_tasklet \
+  .info = snd_hdsp_info_use_midi_work, \
+  .get = snd_hdsp_get_use_midi_work, \
+  .put = snd_hdsp_put_use_midi_work \
 }
 
-static int hdsp_set_use_midi_tasklet(struct hdsp *hdsp, int use_tasklet)
+static int hdsp_set_use_midi_work(struct hdsp *hdsp, int use_work)
 {
-	if (use_tasklet)
-		hdsp->use_midi_tasklet = 1;
+	if (use_work)
+		hdsp->use_midi_work = 1;
 	else
-		hdsp->use_midi_tasklet = 0;
+		hdsp->use_midi_work = 0;
 	return 0;
 }
 
-#define snd_hdsp_info_use_midi_tasklet		snd_ctl_boolean_mono_info
+#define snd_hdsp_info_use_midi_work		snd_ctl_boolean_mono_info
 
-static int snd_hdsp_get_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int snd_hdsp_get_use_midi_work(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 
 	spin_lock_irq(&hdsp->lock);
-	ucontrol->value.integer.value[0] = hdsp->use_midi_tasklet;
+	ucontrol->value.integer.value[0] = hdsp->use_midi_work;
 	spin_unlock_irq(&hdsp->lock);
 	return 0;
 }
 
-static int snd_hdsp_put_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
+static int snd_hdsp_put_use_midi_work(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol)
 {
 	struct hdsp *hdsp = snd_kcontrol_chip(kcontrol);
 	int change;
@@ -2582,8 +2581,8 @@ static int snd_hdsp_put_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct s
 		return -EBUSY;
 	val = ucontrol->value.integer.value[0] & 1;
 	spin_lock_irq(&hdsp->lock);
-	change = (int)val != hdsp->use_midi_tasklet;
-	hdsp_set_use_midi_tasklet(hdsp, val);
+	change = (int)val != hdsp->use_midi_work;
+	hdsp_set_use_midi_work(hdsp, val);
 	spin_unlock_irq(&hdsp->lock);
 	return change;
 }
@@ -2950,7 +2949,7 @@ HDSP_SPDIF_SYNC_CHECK("SPDIF Lock Status", 0),
 HDSP_ADATSYNC_SYNC_CHECK("ADAT Sync Lock Status", 0),
 HDSP_TOGGLE_SETTING("Line Out", HDSP_LineOut),
 HDSP_PRECISE_POINTER("Precise Pointer", 0),
-HDSP_USE_MIDI_TASKLET("Use Midi Tasklet", 0),
+HDSP_USE_MIDI_WORK("Use Midi Tasklet", 0),
 };
 
 
@@ -3370,7 +3369,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer)
 	snd_iprintf(buffer, "MIDI1 Input status: 0x%x\n", hdsp_read(hdsp, HDSP_midiStatusIn0));
 	snd_iprintf(buffer, "MIDI2 Output status: 0x%x\n", hdsp_read(hdsp, HDSP_midiStatusOut1));
 	snd_iprintf(buffer, "MIDI2 Input status: 0x%x\n", hdsp_read(hdsp, HDSP_midiStatusIn1));
-	snd_iprintf(buffer, "Use Midi Tasklet: %s\n", hdsp->use_midi_tasklet ? "on" : "off");
+	snd_iprintf(buffer, "Use Midi Tasklet: %s\n", hdsp->use_midi_work ? "on" : "off");
 
 	snd_iprintf(buffer, "\n");
 
@@ -3791,9 +3790,9 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp)
 	return 0;
 }
 
-static void hdsp_midi_tasklet(struct tasklet_struct *t)
+static void hdsp_midi_work(struct work_struct *work)
 {
-	struct hdsp *hdsp = from_tasklet(hdsp, t, midi_tasklet);
+	struct hdsp *hdsp = container_of(work, struct hdsp, midi_work);
 
 	if (hdsp->midi[0].pending)
 		snd_hdsp_midi_input_read (&hdsp->midi[0]);
@@ -3838,7 +3837,7 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id)
 	}
 
 	if (midi0 && midi0status) {
-		if (hdsp->use_midi_tasklet) {
+		if (hdsp->use_midi_work) {
 			/* we disable interrupts for this input until processing is done */
 			hdsp->control_register &= ~HDSP_Midi0InterruptEnable;
 			hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
@@ -3849,7 +3848,7 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id)
 		}
 	}
 	if (hdsp->io_type != Multiface && hdsp->io_type != RPM && hdsp->io_type != H9632 && midi1 && midi1status) {
-		if (hdsp->use_midi_tasklet) {
+		if (hdsp->use_midi_work) {
 			/* we disable interrupts for this input until processing is done */
 			hdsp->control_register &= ~HDSP_Midi1InterruptEnable;
 			hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register);
@@ -3859,8 +3858,8 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id)
 			snd_hdsp_midi_input_read (&hdsp->midi[1]);
 		}
 	}
-	if (hdsp->use_midi_tasklet && schedule)
-		tasklet_schedule(&hdsp->midi_tasklet);
+	if (hdsp->use_midi_work && schedule)
+		queue_work(system_highpri_wq, &hdsp->midi_work);
 	return IRQ_HANDLED;
 }
 
@@ -5182,7 +5181,7 @@ static int snd_hdsp_create(struct snd_card *card,
 
 	spin_lock_init(&hdsp->lock);
 
-	tasklet_setup(&hdsp->midi_tasklet, hdsp_midi_tasklet);
+	INIT_WORK(&hdsp->midi_work, hdsp_midi_work);
 
 	pci_read_config_word(hdsp->pci, PCI_CLASS_REVISION, &hdsp->firmware_rev);
 	hdsp->firmware_rev &= 0xff;
@@ -5235,7 +5234,7 @@ static int snd_hdsp_create(struct snd_card *card,
 	hdsp->irq = pci->irq;
 	card->sync_irq = hdsp->irq;
 	hdsp->precise_ptr = 0;
-	hdsp->use_midi_tasklet = 1;
+	hdsp->use_midi_work = 1;
 	hdsp->dds_value = 0;
 
 	if ((err = snd_hdsp_initialize_memory(hdsp)) < 0)
@@ -5305,7 +5304,7 @@ static int snd_hdsp_free(struct hdsp *hdsp)
 {
 	if (hdsp->port) {
 		/* stop the audio, and cancel all interrupts */
-		tasklet_kill(&hdsp->midi_tasklet);
+		cancel_work_sync(&hdsp->midi_work);
 		hdsp->control_register &= ~(HDSP_Start|HDSP_AudioInterruptEnable|HDSP_Midi0InterruptEnable|HDSP_Midi1InterruptEnable);
 		hdsp_write (hdsp, HDSP_controlRegister, hdsp->control_register);
 	}
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index 572350aaf18d..4a1f576dd9cf 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -997,7 +997,7 @@ struct hdspm {
 	u32 settings_register;  /* cached value for AIO / RayDat (sync reference, master/slave) */
 
 	struct hdspm_midi midi[4];
-	struct tasklet_struct midi_tasklet;
+	struct work_struct midi_work;
 
 	size_t period_bytes;
 	unsigned char ss_in_channels;
@@ -1217,7 +1217,7 @@ static int snd_hdspm_use_is_exclusive(struct hdspm *hdspm)
 	return ret;
 }
 
-/* round arbitary sample rates to commonly known rates */
+/* round arbitrary sample rates to commonly known rates */
 static int hdspm_round_frequency(int rate)
 {
 	if (rate < 38050)
@@ -2169,9 +2169,9 @@ static int snd_hdspm_create_midi(struct snd_card *card,
 }
 
 
-static void hdspm_midi_tasklet(struct tasklet_struct *t)
+static void hdspm_midi_work(struct work_struct *work)
 {
-	struct hdspm *hdspm = from_tasklet(hdspm, t, midi_tasklet);
+	struct hdspm *hdspm = container_of(work, struct hdspm, midi_work);
 	int i = 0;
 
 	while (i < hdspm->midiPorts) {
@@ -5449,7 +5449,7 @@ static irqreturn_t snd_hdspm_interrupt(int irq, void *dev_id)
 		}
 
 		if (schedule)
-			tasklet_hi_schedule(&hdspm->midi_tasklet);
+			queue_work(system_highpri_wq, &hdspm->midi_work);
 	}
 
 	return IRQ_HANDLED;
@@ -6538,6 +6538,7 @@ static int snd_hdspm_create(struct snd_card *card,
 	hdspm->card = card;
 
 	spin_lock_init(&hdspm->lock);
+	INIT_WORK(&hdspm->midi_work, hdspm_midi_work);
 
 	pci_read_config_word(hdspm->pci,
 			PCI_CLASS_REVISION, &hdspm->firmware_rev);
@@ -6836,9 +6837,6 @@ static int snd_hdspm_create(struct snd_card *card,
 
 	}
 
-	tasklet_setup(&hdspm->midi_tasklet, hdspm_midi_tasklet);
-
-
 	if (hdspm->io_type != MADIface) {
 		hdspm->serial = (hdspm_read(hdspm,
 				HDSPM_midiStatusIn0)>>8) & 0xFFFFFF;
@@ -6873,6 +6871,7 @@ static int snd_hdspm_free(struct hdspm * hdspm)
 {
 
 	if (hdspm->port) {
+		cancel_work_sync(&hdspm->midi_work);
 
 		/* stop th audio, and cancel all interrupts */
 		hdspm->control_register &=
diff --git a/sound/usb/card.c b/sound/usb/card.c
index 696e788c5d31..fa764b61fe9c 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -333,6 +333,106 @@ static int snd_usb_create_streams(struct snd_usb_audio *chip, int ctrlif)
 }
 
 /*
+ * Profile name preset table
+ */
+struct usb_audio_device_name {
+	u32 id;
+	const char *vendor_name;
+	const char *product_name;
+	const char *profile_name;	/* override card->longname */
+};
+
+#define PROFILE_NAME(vid, pid, vendor, product, profile)	 \
+	{ .id = USB_ID(vid, pid), .vendor_name = (vendor),	 \
+	  .product_name = (product), .profile_name = (profile) }
+#define DEVICE_NAME(vid, pid, vendor, product) \
+	PROFILE_NAME(vid, pid, vendor, product, NULL)
+
+/* vendor/product and profile name presets, sorted in device id order */
+static const struct usb_audio_device_name usb_audio_names[] = {
+	/* HP Thunderbolt Dock Audio Headset */
+	PROFILE_NAME(0x03f0, 0x0269, "HP", "Thunderbolt Dock Audio Headset",
+		     "HP-Thunderbolt-Dock-Audio-Headset"),
+	/* HP Thunderbolt Dock Audio Module */
+	PROFILE_NAME(0x03f0, 0x0567, "HP", "Thunderbolt Dock Audio Module",
+		     "HP-Thunderbolt-Dock-Audio-Module"),
+
+	/* Two entries for Gigabyte TRX40 Aorus Master:
+	 * TRX40 Aorus Master has two USB-audio devices, one for the front
+	 * headphone with ESS SABRE9218 DAC chip, while another for the rest
+	 * I/O (the rear panel and the front mic) with Realtek ALC1220-VB.
+	 * Here we provide two distinct names for making UCM profiles easier.
+	 */
+	PROFILE_NAME(0x0414, 0xa000, "Gigabyte", "Aorus Master Front Headphone",
+		     "Gigabyte-Aorus-Master-Front-Headphone"),
+	PROFILE_NAME(0x0414, 0xa001, "Gigabyte", "Aorus Master Main Audio",
+		     "Gigabyte-Aorus-Master-Main-Audio"),
+
+	/* Gigabyte TRX40 Aorus Pro WiFi */
+	PROFILE_NAME(0x0414, 0xa002,
+		     "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"),
+
+	/* Creative/E-Mu devices */
+	DEVICE_NAME(0x041e, 0x3010, "Creative Labs", "Sound Blaster MP3+"),
+	/* Creative/Toshiba Multimedia Center SB-0500 */
+	DEVICE_NAME(0x041e, 0x3048, "Toshiba", "SB-0500"),
+
+	DEVICE_NAME(0x046d, 0x0990, "Logitech, Inc.", "QuickCam Pro 9000"),
+
+	/* Dell WD15 Dock */
+	PROFILE_NAME(0x0bda, 0x4014, "Dell", "WD15 Dock", "Dell-WD15-Dock"),
+	/* Dell WD19 Dock */
+	PROFILE_NAME(0x0bda, 0x402e, "Dell", "WD19 Dock", "Dell-WD15-Dock"),
+
+	DEVICE_NAME(0x0ccd, 0x0028, "TerraTec", "Aureon5.1MkII"),
+
+	/*
+	 * The original product_name is "USB Sound Device", however this name
+	 * is also used by the CM106 based cards, so make it unique.
+	 */
+	DEVICE_NAME(0x0d8c, 0x0102, NULL, "ICUSBAUDIO7D"),
+	DEVICE_NAME(0x0d8c, 0x0103, NULL, "Audio Advantage MicroII"),
+
+	/* MSI TRX40 Creator */
+	PROFILE_NAME(0x0db0, 0x0d64,
+		     "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"),
+	/* MSI TRX40 */
+	PROFILE_NAME(0x0db0, 0x543d,
+		     "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"),
+
+	/* Stanton/N2IT Final Scratch v1 device ('Scratchamp') */
+	DEVICE_NAME(0x103d, 0x0100, "Stanton", "ScratchAmp"),
+	DEVICE_NAME(0x103d, 0x0101, "Stanton", "ScratchAmp"),
+
+	/* aka. Serato Scratch Live DJ Box */
+	DEVICE_NAME(0x13e5, 0x0001, "Rane", "SL-1"),
+
+	/* Lenovo ThinkStation P620 Rear Line-in, Line-out and Microphone */
+	PROFILE_NAME(0x17aa, 0x1046, "Lenovo", "ThinkStation P620 Rear",
+		     "Lenovo-ThinkStation-P620-Rear"),
+	/* Lenovo ThinkStation P620 Internal Speaker + Front Headset */
+	PROFILE_NAME(0x17aa, 0x104d, "Lenovo", "ThinkStation P620 Main",
+		     "Lenovo-ThinkStation-P620-Main"),
+
+	/* Asrock TRX40 Creator */
+	PROFILE_NAME(0x26ce, 0x0a01,
+		     "Realtek", "ALC1220-VB-DT", "Realtek-ALC1220-VB-Desktop"),
+
+	{ } /* terminator */
+};
+
+static const struct usb_audio_device_name *
+lookup_device_name(u32 id)
+{
+	static const struct usb_audio_device_name *p;
+
+	for (p = usb_audio_names; p->id; p++)
+		if (p->id == id)
+			return p;
+	return NULL;
+}
+
+/*
  * free the chip instance
  *
  * here we have to do not much, since pcm and controls are already freed
@@ -357,10 +457,16 @@ static void usb_audio_make_shortname(struct usb_device *dev,
 				     const struct snd_usb_audio_quirk *quirk)
 {
 	struct snd_card *card = chip->card;
-
-	if (quirk && quirk->product_name && *quirk->product_name) {
-		strlcpy(card->shortname, quirk->product_name,
-			sizeof(card->shortname));
+	const struct usb_audio_device_name *preset;
+	const char *s = NULL;
+
+	preset = lookup_device_name(chip->usb_id);
+	if (preset && preset->product_name)
+		s = preset->product_name;
+	else if (quirk && quirk->product_name)
+		s = quirk->product_name;
+	if (s && *s) {
+		strlcpy(card->shortname, s, sizeof(card->shortname));
 		return;
 	}
 
@@ -382,17 +488,26 @@ static void usb_audio_make_longname(struct usb_device *dev,
 				    const struct snd_usb_audio_quirk *quirk)
 {
 	struct snd_card *card = chip->card;
+	const struct usb_audio_device_name *preset;
+	const char *s = NULL;
 	int len;
 
+	preset = lookup_device_name(chip->usb_id);
+
 	/* shortcut - if any pre-defined string is given, use it */
-	if (quirk && quirk->profile_name && *quirk->profile_name) {
-		strlcpy(card->longname, quirk->profile_name,
-			sizeof(card->longname));
+	if (preset && preset->profile_name)
+		s = preset->profile_name;
+	if (s && *s) {
+		strlcpy(card->longname, s, sizeof(card->longname));
 		return;
 	}
 
-	if (quirk && quirk->vendor_name && *quirk->vendor_name) {
-		len = strlcpy(card->longname, quirk->vendor_name, sizeof(card->longname));
+	if (preset && preset->vendor_name)
+		s = preset->vendor_name;
+	else if (quirk && quirk->vendor_name)
+		s = quirk->vendor_name;
+	if (s && *s) {
+		len = strlcpy(card->longname, s, sizeof(card->longname));
 	} else {
 		/* retrieve the vendor and device strings as longname */
 		if (dev->descriptor.iManufacturer)
diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c
index 5fbc8dd2f409..e2f9ce2f5b8b 100644
--- a/sound/usb/endpoint.c
+++ b/sound/usb/endpoint.c
@@ -318,7 +318,7 @@ static inline void prepare_inbound_urb(struct snd_usb_endpoint *ep,
 
 /*
  * Send output urbs that have been prepared previously. URBs are dequeued
- * from ep->ready_playback_urbs and in case there there aren't any available
+ * from ep->ready_playback_urbs and in case there aren't any available
  * or there are no packets that have been prepared, this function does
  * nothing.
  *
diff --git a/sound/usb/midi.c b/sound/usb/midi.c
index e8287a05e36b..c8213652470c 100644
--- a/sound/usb/midi.c
+++ b/sound/usb/midi.c
@@ -142,7 +142,7 @@ struct snd_usb_midi_out_endpoint {
 	unsigned int active_urbs;
 	unsigned int drain_urbs;
 	int max_transfer;		/* size of urb buffer */
-	struct tasklet_struct tasklet;
+	struct work_struct work;
 	unsigned int next_urb;
 	spinlock_t buffer_lock;
 
@@ -344,9 +344,10 @@ static void snd_usbmidi_do_output(struct snd_usb_midi_out_endpoint *ep)
 	spin_unlock_irqrestore(&ep->buffer_lock, flags);
 }
 
-static void snd_usbmidi_out_tasklet(struct tasklet_struct *t)
+static void snd_usbmidi_out_work(struct work_struct *work)
 {
-	struct snd_usb_midi_out_endpoint *ep = from_tasklet(ep, t, tasklet);
+	struct snd_usb_midi_out_endpoint *ep =
+		container_of(work, struct snd_usb_midi_out_endpoint, work);
 
 	snd_usbmidi_do_output(ep);
 }
@@ -1177,7 +1178,7 @@ static void snd_usbmidi_output_trigger(struct snd_rawmidi_substream *substream,
 			snd_rawmidi_proceed(substream);
 			return;
 		}
-		tasklet_schedule(&port->ep->tasklet);
+		queue_work(system_highpri_wq, &port->ep->work);
 	}
 }
 
@@ -1440,7 +1441,7 @@ static int snd_usbmidi_out_endpoint_create(struct snd_usb_midi *umidi,
 	}
 
 	spin_lock_init(&ep->buffer_lock);
-	tasklet_setup(&ep->tasklet, snd_usbmidi_out_tasklet);
+	INIT_WORK(&ep->work, snd_usbmidi_out_work);
 	init_waitqueue_head(&ep->drain_wait);
 
 	for (i = 0; i < 0x10; ++i)
@@ -1503,7 +1504,7 @@ void snd_usbmidi_disconnect(struct list_head *p)
 	for (i = 0; i < MIDI_MAX_ENDPOINTS; ++i) {
 		struct snd_usb_midi_endpoint *ep = &umidi->endpoints[i];
 		if (ep->out)
-			tasklet_kill(&ep->out->tasklet);
+			cancel_work_sync(&ep->out->work);
 		if (ep->out) {
 			for (j = 0; j < OUTPUT_URBS; ++j)
 				usb_kill_urb(ep->out->urbs[j].urb);
diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c
index 3b2dce1043f5..6b30155964ec 100644
--- a/sound/usb/misc/ua101.c
+++ b/sound/usb/misc/ua101.c
@@ -96,7 +96,7 @@ struct ua101 {
 	u8 rate_feedback[MAX_QUEUE_LENGTH];
 
 	struct list_head ready_playback_urbs;
-	struct tasklet_struct playback_tasklet;
+	struct work_struct playback_work;
 	wait_queue_head_t alsa_capture_wait;
 	wait_queue_head_t rate_feedback_wait;
 	wait_queue_head_t alsa_playback_wait;
@@ -188,7 +188,7 @@ static void playback_urb_complete(struct urb *usb_urb)
 		spin_lock_irqsave(&ua->lock, flags);
 		list_add_tail(&urb->ready_list, &ua->ready_playback_urbs);
 		if (ua->rate_feedback_count > 0)
-			tasklet_schedule(&ua->playback_tasklet);
+			queue_work(system_highpri_wq, &ua->playback_work);
 		ua->playback.substream->runtime->delay -=
 				urb->urb.iso_frame_desc[0].length /
 						ua->playback.frame_bytes;
@@ -247,9 +247,9 @@ static inline void add_with_wraparound(struct ua101 *ua,
 		*value -= ua->playback.queue_length;
 }
 
-static void playback_tasklet(struct tasklet_struct *t)
+static void playback_work(struct work_struct *work)
 {
-	struct ua101 *ua = from_tasklet(ua, t, playback_tasklet);
+	struct ua101 *ua = container_of(work, struct ua101, playback_work);
 	unsigned long flags;
 	unsigned int frames;
 	struct ua101_urb *urb;
@@ -401,7 +401,7 @@ static void capture_urb_complete(struct urb *urb)
 		}
 		if (test_bit(USB_PLAYBACK_RUNNING, &ua->states) &&
 		    !list_empty(&ua->ready_playback_urbs))
-			tasklet_schedule(&ua->playback_tasklet);
+			queue_work(system_highpri_wq, &ua->playback_work);
 	}
 
 	spin_unlock_irqrestore(&ua->lock, flags);
@@ -532,7 +532,7 @@ static void stop_usb_playback(struct ua101 *ua)
 
 	kill_stream_urbs(&ua->playback);
 
-	tasklet_kill(&ua->playback_tasklet);
+	cancel_work_sync(&ua->playback_work);
 
 	disable_iso_interface(ua, INTF_PLAYBACK);
 }
@@ -550,7 +550,7 @@ static int start_usb_playback(struct ua101 *ua)
 		return 0;
 
 	kill_stream_urbs(&ua->playback);
-	tasklet_kill(&ua->playback_tasklet);
+	cancel_work_sync(&ua->playback_work);
 
 	err = enable_iso_interface(ua, INTF_PLAYBACK);
 	if (err < 0)
@@ -1218,7 +1218,7 @@ static int ua101_probe(struct usb_interface *interface,
 	spin_lock_init(&ua->lock);
 	mutex_init(&ua->mutex);
 	INIT_LIST_HEAD(&ua->ready_playback_urbs);
-	tasklet_setup(&ua->playback_tasklet, playback_tasklet);
+	INIT_WORK(&ua->playback_work, playback_work);
 	init_waitqueue_head(&ua->alsa_capture_wait);
 	init_waitqueue_head(&ua->rate_feedback_wait);
 	init_waitqueue_head(&ua->alsa_playback_wait);
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index 199cdbfdc761..df036a359f2f 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -2602,6 +2602,216 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer)
 	return 0;
 }
 
+/*
+ * Pioneer DJ DJM-250MK2 and maybe other DJM models
+ *
+ * For playback, no duplicate mapping should be set.
+ * There are three mixer stereo channels (CH1, CH2, AUX)
+ * and three stereo sources (Playback 1-2, Playback 3-4, Playback 5-6).
+ * Each channel should be mapped just once to one source.
+ * If mapped multiple times, only one source will play on given channel
+ * (sources are not mixed together).
+ *
+ * For recording, duplicate mapping is OK. We will get the same signal multiple times.
+ *
+ * Channels 7-8 are in both directions fixed to FX SEND / FX RETURN.
+ *
+ * See also notes in the quirks-table.h file.
+ */
+
+struct snd_pioneer_djm_option {
+	const u16 wIndex;
+	const u16 wValue;
+	const char *name;
+};
+
+static const struct snd_pioneer_djm_option snd_pioneer_djm_options_capture_level[] = {
+	{ .name =  "-5 dB",                  .wValue = 0x0300, .wIndex = 0x8003 },
+	{ .name = "-10 dB",                  .wValue = 0x0200, .wIndex = 0x8003 },
+	{ .name = "-15 dB",                  .wValue = 0x0100, .wIndex = 0x8003 },
+	{ .name = "-19 dB",                  .wValue = 0x0000, .wIndex = 0x8003 }
+};
+
+static const struct snd_pioneer_djm_option snd_pioneer_djm_options_capture_ch12[] = {
+	{ .name =  "CH1 Control Tone PHONO", .wValue = 0x0103, .wIndex = 0x8002 },
+	{ .name =  "CH1 Control Tone LINE",  .wValue = 0x0100, .wIndex = 0x8002 },
+	{ .name =  "Post CH1 Fader",         .wValue = 0x0106, .wIndex = 0x8002 },
+	{ .name =  "Cross Fader A",          .wValue = 0x0107, .wIndex = 0x8002 },
+	{ .name =  "Cross Fader B",          .wValue = 0x0108, .wIndex = 0x8002 },
+	{ .name =  "MIC",                    .wValue = 0x0109, .wIndex = 0x8002 },
+	{ .name =  "AUX",                    .wValue = 0x010d, .wIndex = 0x8002 },
+	{ .name =  "REC OUT",                .wValue = 0x010a, .wIndex = 0x8002 }
+};
+
+static const struct snd_pioneer_djm_option snd_pioneer_djm_options_capture_ch34[] = {
+	{ .name =  "CH2 Control Tone PHONO", .wValue = 0x0203, .wIndex = 0x8002 },
+	{ .name =  "CH2 Control Tone LINE",  .wValue = 0x0200, .wIndex = 0x8002 },
+	{ .name =  "Post CH2 Fader",         .wValue = 0x0206, .wIndex = 0x8002 },
+	{ .name =  "Cross Fader A",          .wValue = 0x0207, .wIndex = 0x8002 },
+	{ .name =  "Cross Fader B",          .wValue = 0x0208, .wIndex = 0x8002 },
+	{ .name =  "MIC",                    .wValue = 0x0209, .wIndex = 0x8002 },
+	{ .name =  "AUX",                    .wValue = 0x020d, .wIndex = 0x8002 },
+	{ .name =  "REC OUT",                .wValue = 0x020a, .wIndex = 0x8002 }
+};
+
+static const struct snd_pioneer_djm_option snd_pioneer_djm_options_capture_ch56[] = {
+	{ .name =  "REC OUT",                .wValue = 0x030a, .wIndex = 0x8002 },
+	{ .name =  "Post CH1 Fader",         .wValue = 0x0311, .wIndex = 0x8002 },
+	{ .name =  "Post CH2 Fader",         .wValue = 0x0312, .wIndex = 0x8002 },
+	{ .name =  "Cross Fader A",          .wValue = 0x0307, .wIndex = 0x8002 },
+	{ .name =  "Cross Fader B",          .wValue = 0x0308, .wIndex = 0x8002 },
+	{ .name =  "MIC",                    .wValue = 0x0309, .wIndex = 0x8002 },
+	{ .name =  "AUX",                    .wValue = 0x030d, .wIndex = 0x8002 }
+};
+
+static const struct snd_pioneer_djm_option snd_pioneer_djm_options_playback_12[] = {
+	{ .name =  "CH1",                    .wValue = 0x0100, .wIndex = 0x8016 },
+	{ .name =  "CH2",                    .wValue = 0x0101, .wIndex = 0x8016 },
+	{ .name =  "AUX",                    .wValue = 0x0104, .wIndex = 0x8016 }
+};
+
+static const struct snd_pioneer_djm_option snd_pioneer_djm_options_playback_34[] = {
+	{ .name =  "CH1",                    .wValue = 0x0200, .wIndex = 0x8016 },
+	{ .name =  "CH2",                    .wValue = 0x0201, .wIndex = 0x8016 },
+	{ .name =  "AUX",                    .wValue = 0x0204, .wIndex = 0x8016 }
+};
+
+static const struct snd_pioneer_djm_option snd_pioneer_djm_options_playback_56[] = {
+	{ .name =  "CH1",                    .wValue = 0x0300, .wIndex = 0x8016 },
+	{ .name =  "CH2",                    .wValue = 0x0301, .wIndex = 0x8016 },
+	{ .name =  "AUX",                    .wValue = 0x0304, .wIndex = 0x8016 }
+};
+
+struct snd_pioneer_djm_option_group {
+	const char *name;
+	const struct snd_pioneer_djm_option *options;
+	const size_t count;
+	const u16 default_value;
+};
+
+#define snd_pioneer_djm_option_group_item(_name, suffix, _default_value) { \
+	.name = _name, \
+	.options = snd_pioneer_djm_options_##suffix, \
+	.count = ARRAY_SIZE(snd_pioneer_djm_options_##suffix), \
+	.default_value = _default_value }
+
+static const struct snd_pioneer_djm_option_group snd_pioneer_djm_option_groups[] = {
+	snd_pioneer_djm_option_group_item("Master Capture Level Capture Switch", capture_level, 0),
+	snd_pioneer_djm_option_group_item("Capture 1-2 Capture Switch",          capture_ch12,  2),
+	snd_pioneer_djm_option_group_item("Capture 3-4 Capture Switch",          capture_ch34,  2),
+	snd_pioneer_djm_option_group_item("Capture 5-6 Capture Switch",          capture_ch56,  0),
+	snd_pioneer_djm_option_group_item("Playback 1-2 Playback Switch",        playback_12,   0),
+	snd_pioneer_djm_option_group_item("Playback 3-4 Playback Switch",        playback_34,   1),
+	snd_pioneer_djm_option_group_item("Playback 5-6 Playback Switch",        playback_56,   2)
+};
+
+// layout of the kcontrol->private_value:
+#define SND_PIONEER_DJM_VALUE_MASK 0x0000ffff
+#define SND_PIONEER_DJM_GROUP_MASK 0xffff0000
+#define SND_PIONEER_DJM_GROUP_SHIFT 16
+
+static int snd_pioneer_djm_controls_info(struct snd_kcontrol *kctl, struct snd_ctl_elem_info *info)
+{
+	u16 group_index = kctl->private_value >> SND_PIONEER_DJM_GROUP_SHIFT;
+	size_t count;
+	const char *name;
+	const struct snd_pioneer_djm_option_group *group;
+
+	if (group_index >= ARRAY_SIZE(snd_pioneer_djm_option_groups))
+		return -EINVAL;
+
+	group = &snd_pioneer_djm_option_groups[group_index];
+	count = group->count;
+	if (info->value.enumerated.item >= count)
+		info->value.enumerated.item = count - 1;
+	name = group->options[info->value.enumerated.item].name;
+	strlcpy(info->value.enumerated.name, name, sizeof(info->value.enumerated.name));
+	info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	info->count = 1;
+	info->value.enumerated.items = count;
+	return 0;
+}
+
+static int snd_pioneer_djm_controls_update(struct usb_mixer_interface *mixer, u16 group, u16 value)
+{
+	int err;
+
+	if (group >= ARRAY_SIZE(snd_pioneer_djm_option_groups)
+			|| value >= snd_pioneer_djm_option_groups[group].count)
+		return -EINVAL;
+
+	err = snd_usb_lock_shutdown(mixer->chip);
+	if (err)
+		return err;
+
+	err = snd_usb_ctl_msg(
+		mixer->chip->dev, usb_sndctrlpipe(mixer->chip->dev, 0),
+		USB_REQ_SET_FEATURE,
+		USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE,
+		snd_pioneer_djm_option_groups[group].options[value].wValue,
+		snd_pioneer_djm_option_groups[group].options[value].wIndex,
+		NULL, 0);
+
+	snd_usb_unlock_shutdown(mixer->chip);
+	return err;
+}
+
+static int snd_pioneer_djm_controls_get(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *elem)
+{
+	elem->value.enumerated.item[0] = kctl->private_value & SND_PIONEER_DJM_VALUE_MASK;
+	return 0;
+}
+
+static int snd_pioneer_djm_controls_put(struct snd_kcontrol *kctl, struct snd_ctl_elem_value *elem)
+{
+	struct usb_mixer_elem_list *list = snd_kcontrol_chip(kctl);
+	struct usb_mixer_interface *mixer = list->mixer;
+	unsigned long private_value = kctl->private_value;
+	u16 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT;
+	u16 value = elem->value.enumerated.item[0];
+
+	kctl->private_value = (group << SND_PIONEER_DJM_GROUP_SHIFT) | value;
+
+	return snd_pioneer_djm_controls_update(mixer, group, value);
+}
+
+static int snd_pioneer_djm_controls_resume(struct usb_mixer_elem_list *list)
+{
+	unsigned long private_value = list->kctl->private_value;
+	u16 group = (private_value & SND_PIONEER_DJM_GROUP_MASK) >> SND_PIONEER_DJM_GROUP_SHIFT;
+	u16 value = (private_value & SND_PIONEER_DJM_VALUE_MASK);
+
+	return snd_pioneer_djm_controls_update(list->mixer, group, value);
+}
+
+static int snd_pioneer_djm_controls_create(struct usb_mixer_interface *mixer)
+{
+	int err, i;
+	const struct snd_pioneer_djm_option_group *group;
+	struct snd_kcontrol_new knew = {
+		.iface  = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+		.index = 0,
+		.info = snd_pioneer_djm_controls_info,
+		.get  = snd_pioneer_djm_controls_get,
+		.put  = snd_pioneer_djm_controls_put
+	};
+
+	for (i = 0; i < ARRAY_SIZE(snd_pioneer_djm_option_groups); i++) {
+		group = &snd_pioneer_djm_option_groups[i];
+		knew.name = group->name;
+		knew.private_value = (i << SND_PIONEER_DJM_GROUP_SHIFT) | group->default_value;
+		err = snd_pioneer_djm_controls_update(mixer, i, group->default_value);
+		if (err)
+			return err;
+		err = add_single_ctl_with_resume(mixer, 0, snd_pioneer_djm_controls_resume,
+						 &knew, NULL);
+		if (err)
+			return err;
+	}
+	return 0;
+}
+
 int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 {
 	int err = 0;
@@ -2706,6 +2916,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer)
 	case USB_ID(0x2a39, 0x3fb0): /* RME Babyface Pro FS */
 		err = snd_bbfpro_controls_create(mixer);
 		break;
+	case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */
+		err = snd_pioneer_djm_controls_create(mixer);
+		break;
 	}
 
 	return err;
diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c
index 0ffff7640892..d33df146d6ce 100644
--- a/sound/usb/mixer_scarlett_gen2.c
+++ b/sound/usb/mixer_scarlett_gen2.c
@@ -1946,7 +1946,7 @@ static void scarlett2_mixer_interrupt(struct urb *urb)
 		goto requeue;
 
 	if (len == 8) {
-		data = le32_to_cpu(*(u32 *)urb->transfer_buffer);
+		data = le32_to_cpu(*(__le32 *)urb->transfer_buffer);
 		if (data & SCARLETT2_USB_INTERRUPT_VOL_CHANGE)
 			scarlett2_mixer_interrupt_vol_change(mixer);
 		if (data & SCARLETT2_USB_INTERRUPT_BUTTON_CHANGE)
diff --git a/sound/usb/mixer_us16x08.c b/sound/usb/mixer_us16x08.c
index a4d4d71db55b..92b1a6d9c931 100644
--- a/sound/usb/mixer_us16x08.c
+++ b/sound/usb/mixer_us16x08.c
@@ -1109,7 +1109,7 @@ static const struct snd_us16x08_control_params eq_controls[] = {
 		.control_id = SND_US16X08_ID_EQLOWFREQ,
 		.type = USB_MIXER_U8,
 		.num_channels = 16,
-		.name = "EQ Low Frequence",
+		.name = "EQ Low Frequency",
 	},
 	{ /* EQ mid low gain */
 		.kcontrol_new = &snd_us16x08_eq_gain_ctl,
@@ -1123,7 +1123,7 @@ static const struct snd_us16x08_control_params eq_controls[] = {
 		.control_id = SND_US16X08_ID_EQLOWMIDFREQ,
 		.type = USB_MIXER_U8,
 		.num_channels = 16,
-		.name = "EQ MidLow Frequence",
+		.name = "EQ MidLow Frequency",
 	},
 	{ /* EQ mid low Q */
 		.kcontrol_new = &snd_us16x08_eq_mid_width_ctl,
@@ -1144,7 +1144,7 @@ static const struct snd_us16x08_control_params eq_controls[] = {
 		.control_id = SND_US16X08_ID_EQHIGHMIDFREQ,
 		.type = USB_MIXER_U8,
 		.num_channels = 16,
-		.name = "EQ MidHigh Frequence",
+		.name = "EQ MidHigh Frequency",
 	},
 	{ /* EQ mid high Q */
 		.kcontrol_new = &snd_us16x08_eq_mid_width_ctl,
@@ -1165,7 +1165,7 @@ static const struct snd_us16x08_control_params eq_controls[] = {
 		.control_id = SND_US16X08_ID_EQHIGHFREQ,
 		.type = USB_MIXER_U8,
 		.num_channels = 16,
-		.name = "EQ High Frequence",
+		.name = "EQ High Frequency",
 	},
 };
 
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index 23eafd50126f..3c1697f6b60c 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -25,33 +25,16 @@
 	.idProduct = prod, \
 	.bInterfaceClass = USB_CLASS_VENDOR_SPEC
 
-#define QUIRK_RENAME_DEVICE(_vendor, _device)				\
-	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \
-		.vendor_name = _vendor,					\
-		.product_name = _device,				\
-		.ifnum = QUIRK_NO_INTERFACE				\
-	}
-
-#define QUIRK_DEVICE_PROFILE(_vendor, _device, _profile)		\
-	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { \
-		.vendor_name = _vendor,					\
-		.product_name = _device,				\
-		.profile_name = _profile,				\
-		.ifnum = QUIRK_NO_INTERFACE				\
-	}
+/* A standard entry matching with vid/pid and the audio class/subclass */
+#define USB_AUDIO_DEVICE(vend, prod) \
+	.match_flags = USB_DEVICE_ID_MATCH_DEVICE | \
+		       USB_DEVICE_ID_MATCH_INT_CLASS | \
+		       USB_DEVICE_ID_MATCH_INT_SUBCLASS, \
+	.idVendor = vend, \
+	.idProduct = prod, \
+	.bInterfaceClass = USB_CLASS_AUDIO, \
+	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
 
-/* HP Thunderbolt Dock Audio Headset */
-{
-	USB_DEVICE(0x03f0, 0x0269),
-	QUIRK_DEVICE_PROFILE("HP", "Thunderbolt Dock Audio Headset",
-			     "HP-Thunderbolt-Dock-Audio-Headset"),
-},
-/* HP Thunderbolt Dock Audio Module */
-{
-	USB_DEVICE(0x03f0, 0x0567),
-	QUIRK_DEVICE_PROFILE("HP", "Thunderbolt Dock Audio Module",
-			     "HP-Thunderbolt-Dock-Audio-Module"),
-},
 /* FTDI devices */
 {
 	USB_DEVICE(0x0403, 0xb8d8),
@@ -85,44 +68,14 @@
 	}
 },
 
-/* Creative/E-Mu devices */
-{
-	USB_DEVICE(0x041e, 0x3010),
-	QUIRK_RENAME_DEVICE("Creative Labs", "Sound Blaster MP3+")
-},
-/* Creative/Toshiba Multimedia Center SB-0500 */
-{
-	USB_DEVICE(0x041e, 0x3048),
-	QUIRK_RENAME_DEVICE("Toshiba", "SB-0500")
-},
-{
-	/* E-Mu 0202 USB */
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE,
-	.idVendor = 0x041e,
-	.idProduct = 0x3f02,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-},
-{
-	/* E-Mu 0404 USB */
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE,
-	.idVendor = 0x041e,
-	.idProduct = 0x3f04,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-},
-{
-	/* E-Mu Tracker Pre */
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE,
-	.idVendor = 0x041e,
-	.idProduct = 0x3f0a,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-},
-{
-	/* E-Mu 0204 USB */
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE,
-	.idVendor = 0x041e,
-	.idProduct = 0x3f19,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-},
+/* E-Mu 0202 USB */
+{ USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f02) },
+/* E-Mu 0404 USB */
+{ USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f04) },
+/* E-Mu Tracker Pre */
+{ USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f0a) },
+/* E-Mu 0204 USB */
+{ USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f19) },
 
 /*
  * HP Wireless Audio
@@ -164,70 +117,13 @@
  * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface
  * class matches do not take effect without an explicit ID match.
  */
-{
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
-		       USB_DEVICE_ID_MATCH_INT_CLASS |
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
-	.idVendor = 0x046d,
-	.idProduct = 0x0850,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
-},
-{
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
-		       USB_DEVICE_ID_MATCH_INT_CLASS |
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
-	.idVendor = 0x046d,
-	.idProduct = 0x08ae,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
-},
-{
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
-		       USB_DEVICE_ID_MATCH_INT_CLASS |
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
-	.idVendor = 0x046d,
-	.idProduct = 0x08c6,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
-},
-{
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
-		       USB_DEVICE_ID_MATCH_INT_CLASS |
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
-	.idVendor = 0x046d,
-	.idProduct = 0x08f0,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
-},
-{
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
-		       USB_DEVICE_ID_MATCH_INT_CLASS |
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
-	.idVendor = 0x046d,
-	.idProduct = 0x08f5,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
-},
-{
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
-		       USB_DEVICE_ID_MATCH_INT_CLASS |
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
-	.idVendor = 0x046d,
-	.idProduct = 0x08f6,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL
-},
-{
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
-		       USB_DEVICE_ID_MATCH_INT_CLASS |
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
-	.idVendor = 0x046d,
-	.idProduct = 0x0990,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
-	QUIRK_RENAME_DEVICE("Logitech, Inc.", "QuickCam Pro 9000")
-},
+{ USB_AUDIO_DEVICE(0x046d, 0x0850) },
+{ USB_AUDIO_DEVICE(0x046d, 0x08ae) },
+{ USB_AUDIO_DEVICE(0x046d, 0x08c6) },
+{ USB_AUDIO_DEVICE(0x046d, 0x08f0) },
+{ USB_AUDIO_DEVICE(0x046d, 0x08f5) },
+{ USB_AUDIO_DEVICE(0x046d, 0x08f6) },
+{ USB_AUDIO_DEVICE(0x046d, 0x0990) },
 
 /*
  * Yamaha devices
@@ -2610,10 +2506,6 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 {
-	USB_DEVICE(0x0ccd, 0x0028),
-	QUIRK_RENAME_DEVICE("TerraTec", "Aureon5.1MkII")
-},
-{
 	USB_DEVICE(0x0ccd, 0x0035),
 	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
 		.vendor_name = "Miditech",
@@ -2623,16 +2515,6 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 
-/* Stanton/N2IT Final Scratch v1 device ('Scratchamp') */
-{
-	USB_DEVICE(0x103d, 0x0100),
-	QUIRK_RENAME_DEVICE("Stanton", "ScratchAmp")
-},
-{
-	USB_DEVICE(0x103d, 0x0101),
-	QUIRK_RENAME_DEVICE("Stanton", "ScratchAmp")
-},
-
 /* Novation EMS devices */
 {
 	USB_DEVICE_VENDOR_SPEC(0x1235, 0x0001),
@@ -2817,20 +2699,10 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 	}
 },
 
-/* */
-{
-	/* aka. Serato Scratch Live DJ Box */
-	USB_DEVICE(0x13e5, 0x0001),
-	QUIRK_RENAME_DEVICE("Rane", "SL-1")
-},
-
 /* Lenovo ThinkStation P620 Rear Line-in, Line-out and Microphone */
 {
 	USB_DEVICE(0x17aa, 0x1046),
 	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
-		.vendor_name = "Lenovo",
-		.product_name = "ThinkStation P620 Rear",
-		.profile_name = "Lenovo-ThinkStation-P620-Rear",
 		.ifnum = QUIRK_ANY_INTERFACE,
 		.type = QUIRK_SETUP_DISABLE_AUTOSUSPEND
 	}
@@ -2839,9 +2711,6 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 {
 	USB_DEVICE(0x17aa, 0x104d),
 	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
-		.vendor_name = "Lenovo",
-		.product_name = "ThinkStation P620 Main",
-		.profile_name = "Lenovo-ThinkStation-P620-Main",
 		.ifnum = QUIRK_ANY_INTERFACE,
 		.type = QUIRK_SETUP_DISABLE_AUTOSUSPEND
 	}
@@ -2879,10 +2748,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
 },
 
 /* KeithMcMillen Stringport */
-{
-	USB_DEVICE(0x1f38, 0x0001),
-	.bInterfaceClass = USB_CLASS_AUDIO,
-},
+{ USB_DEVICE(0x1f38, 0x0001) }, /* FIXME: should be more restrictive matching */
 
 /* Miditech devices */
 {
@@ -2913,13 +2779,7 @@ YAMAHA_DEVICE(0x7010, "UB99"),
  */
 
 #define AU0828_DEVICE(vid, pid, vname, pname) { \
-	.idVendor = vid, \
-	.idProduct = pid, \
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE | \
-		       USB_DEVICE_ID_MATCH_INT_CLASS | \
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS, \
-	.bInterfaceClass = USB_CLASS_AUDIO, \
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL, \
+	USB_AUDIO_DEVICE(vid, pid), \
 	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { \
 		.vendor_name = vname, \
 		.product_name = pname, \
@@ -2949,13 +2809,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 
 /* Syntek STK1160 */
 {
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
-		       USB_DEVICE_ID_MATCH_INT_CLASS |
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
-	.idVendor = 0x05e1,
-	.idProduct = 0x0408,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
+	USB_AUDIO_DEVICE(0x05e1, 0x0408),
 	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
 		.vendor_name = "Syntek",
 		.product_name = "STK1160",
@@ -3117,10 +2971,7 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 },
 {
 	/* Tascam US122 MKII - playback-only support */
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE,
-	.idVendor = 0x0644,
-	.idProduct = 0x8021,
-	.bInterfaceClass = USB_CLASS_AUDIO,
+	USB_DEVICE_VENDOR_SPEC(0x0644, 0x8021),
 	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
 		.vendor_name = "TASCAM",
 		.product_name = "US122 MKII",
@@ -3305,19 +3156,6 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 	}
 },
 
-/*
- * The original product_name is "USB Sound Device", however this name
- * is also used by the CM106 based cards, so make it unique.
- */
-{
-	USB_DEVICE(0x0d8c, 0x0102),
-	QUIRK_RENAME_DEVICE(NULL, "ICUSBAUDIO7D")
-},
-{
-	USB_DEVICE(0x0d8c, 0x0103),
-	QUIRK_RENAME_DEVICE(NULL, "Audio Advantage MicroII")
-},
-
 /* disabled due to regression for other devices;
  * see https://bugzilla.kernel.org/show_bug.cgi?id=199905
  */
@@ -3418,18 +3256,10 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 		}
 	}
 },
-/* Dell WD15 Dock */
-{
-	USB_DEVICE(0x0bda, 0x4014),
-	QUIRK_DEVICE_PROFILE("Dell", "WD15 Dock", "Dell-WD15-Dock")
-},
 /* Dell WD19 Dock */
 {
 	USB_DEVICE(0x0bda, 0x402e),
 	.driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) {
-		.vendor_name = "Dell",
-		.product_name = "WD19 Dock",
-		.profile_name = "Dell-WD15-Dock",
 		.ifnum = QUIRK_ANY_INTERFACE,
 		.type = QUIRK_SETUP_FMT_AFTER_RESUME
 	}
@@ -3701,33 +3531,6 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"),
 	}
 },
 
-#define ALC1220_VB_DESKTOP(vend, prod) { \
-	USB_DEVICE(vend, prod),	\
-	QUIRK_DEVICE_PROFILE("Realtek", "ALC1220-VB-DT", \
-			     "Realtek-ALC1220-VB-Desktop") \
-}
-ALC1220_VB_DESKTOP(0x0414, 0xa002), /* Gigabyte TRX40 Aorus Pro WiFi */
-ALC1220_VB_DESKTOP(0x0db0, 0x0d64), /* MSI TRX40 Creator */
-ALC1220_VB_DESKTOP(0x0db0, 0x543d), /* MSI TRX40 */
-ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */
-#undef ALC1220_VB_DESKTOP
-
-/* Two entries for Gigabyte TRX40 Aorus Master:
- * TRX40 Aorus Master has two USB-audio devices, one for the front headphone
- * with ESS SABRE9218 DAC chip, while another for the rest I/O (the rear
- * panel and the front mic) with Realtek ALC1220-VB.
- * Here we provide two distinct names for making UCM profiles easier.
- */
-{
-	USB_DEVICE(0x0414, 0xa000),
-	QUIRK_DEVICE_PROFILE("Gigabyte", "Aorus Master Front Headphone",
-			     "Gigabyte-Aorus-Master-Front-Headphone")
-},
-{
-	USB_DEVICE(0x0414, 0xa001),
-	QUIRK_DEVICE_PROFILE("Gigabyte", "Aorus Master Main Audio",
-			     "Gigabyte-Aorus-Master-Main-Audio")
-},
 {
 	/*
 	 * Pioneer DJ DJM-900NXS2
@@ -3804,13 +3607,7 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */
  * channels to be swapped and out of phase, which is dealt with in quirks.c.
  */
 {
-	.match_flags = USB_DEVICE_ID_MATCH_DEVICE |
-		       USB_DEVICE_ID_MATCH_INT_CLASS |
-		       USB_DEVICE_ID_MATCH_INT_SUBCLASS,
-	.idVendor = 0x534d,
-	.idProduct = 0x2109,
-	.bInterfaceClass = USB_CLASS_AUDIO,
-	.bInterfaceSubClass = USB_SUBCLASS_AUDIOCONTROL,
+	USB_AUDIO_DEVICE(0x534d, 0x2109),
 	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
 		.vendor_name = "MacroSilicon",
 		.product_name = "MS2109",
@@ -3851,3 +3648,4 @@ ALC1220_VB_DESKTOP(0x26ce, 0x0a01), /* Asrock TRX40 Creator */
 },
 
 #undef USB_DEVICE_VENDOR_SPEC
+#undef USB_AUDIO_DEVICE
diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h
index 6839915a0128..0805b7f21272 100644
--- a/sound/usb/usbaudio.h
+++ b/sound/usb/usbaudio.h
@@ -110,7 +110,6 @@ enum quirk_type {
 struct snd_usb_audio_quirk {
 	const char *vendor_name;
 	const char *product_name;
-	const char *profile_name;	/* override the card->longname */
 	int16_t ifnum;
 	uint16_t type;
 	bool shares_media_device;